diff options
Diffstat (limited to 'pd/portaudio_v18')
134 files changed, 0 insertions, 38202 deletions
diff --git a/pd/portaudio_v18/LICENSE.txt b/pd/portaudio_v18/LICENSE.txt deleted file mode 100644 index 105da3f7..00000000 --- a/pd/portaudio_v18/LICENSE.txt +++ /dev/null @@ -1,65 +0,0 @@ -Portable header file to contain: -/* - * PortAudio Portable Real-Time Audio Library - * PortAudio API Header File - * Latest version available at: http://www.audiomulch.com/portaudio/ - * - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ - - -Implementation files to contain: -/* - * PortAudio Portable Real-Time Audio Library - * Latest version at: http://www.audiomulch.com/portaudio/ - * <platform> Implementation - * Copyright (c) 1999-2000 <author(s)> - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */
\ No newline at end of file diff --git a/pd/portaudio_v18/Makefile.in b/pd/portaudio_v18/Makefile.in deleted file mode 100644 index a8e0434f..00000000 --- a/pd/portaudio_v18/Makefile.in +++ /dev/null @@ -1,96 +0,0 @@ -# -# PortAudio Makefile.in -# -# Dominic Mazzoni -# - -PREFIX = @prefix@ -CC = @CC@ -CFLAGS = @CFLAGS@ -Ipa_common -LIBS = @LIBS@ -AR = @AR@ -RANLIB = @RANLIB@ -INSTALL = @INSTALL@ -SHARED_FLAGS = @SHARED_FLAGS@ -DLL_LIBS = @DLL_LIBS@ - -OTHER_OBJS = @OTHER_OBJS@ - -PALIB = libportaudio.a -PADLL = @PADLL@ -PADLLV = $(PADLL).0.0.18 -PAINC = pa_common/portaudio.h - -COMMON_OBJS = \ - pa_common/pa_convert.o \ - pa_common/pa_lib.o - -TESTS = \ - bin/patest_buffer \ - bin/patest_clip \ - bin/patest_dither \ - bin/patest_hang \ - bin/patest_latency \ - bin/patest_leftright \ - bin/patest_longsine \ - bin/patest_many \ - bin/patest_maxsines \ - bin/patest_multi_sine \ - bin/patest_pink \ - bin/patest_record \ - bin/patest_ringmix \ - bin/patest_saw \ - bin/patest_sine8 \ - bin/patest_sine \ - bin/patest_sine_formats \ - bin/patest_sine_time \ - bin/patest_stop \ - bin/patest_sync \ - bin/patest_toomanysines \ - bin/patest_underflow \ - bin/patest_wire - -OBJS = $(COMMON_OBJS) $(OTHER_OBJS) - -all: lib/$(PALIB) lib/$(PADLLV) tests - -tests: bin/ $(TESTS) - -lib/$(PALIB): lib/ $(OBJS) Makefile $(PAINC) - $(AR) ruv lib/$(PALIB) $(OBJS) - $(RANLIB) lib/$(PALIB) - -lib/$(PADLLV): lib/ $(OBJS) Makefile $(PAINC) - $(CC) $(SHARED_FLAGS) -o lib/$(PADLLV) $(OBJS) $(DLL_LIBS) - -$(TESTS): bin/%: lib/$(PALIB) Makefile $(PAINC) pa_tests/%.c - $(CC) -o $@ $(CFLAGS) pa_tests/$*.c lib/$(PALIB) $(LIBS) - -install: lib/$(PALIB) lib/$(PADLLV) - $(INSTALL) -m 644 lib/$(PADLLV) $(PREFIX)/lib/$(PADLLV) - $(INSTALL) -m 644 lib/$(PALIB) $(PREFIX)/lib/$(PALIB) - cd $(PREFIX)/lib && rm -f $(PADLL) && ln -s $(PADLLV) $(PADLL) - $(INSTALL) -m 644 pa_common/portaudio.h $(PREFIX)/include/portaudio.h - @echo "" - @echo "------------------------------------------------------------" - @echo "PortAudio was successfully installed." - @echo "" - @echo "On some systems (e.g. Linux) you should run 'ldconfig' now" - @echo "to make the shared object available. You may also need to" - @echo "modify your LD_LIBRARY_PATH environment variable to include" - @echo "the directory $(PREFIX)/lib" - @echo "------------------------------------------------------------" - @echo "" - -clean: - rm -f $(OBJS) $(TESTS) lib/$(PALIB) - rm -f config.log config.status - -%.o: %.c Makefile $(PAINC) - $(CC) -c $(CFLAGS) $< -o $@ - -bin: - mkdir bin - -lib: - mkdir lib diff --git a/pd/portaudio_v18/Makefile.linux b/pd/portaudio_v18/Makefile.linux deleted file mode 100644 index a58d57b3..00000000 --- a/pd/portaudio_v18/Makefile.linux +++ /dev/null @@ -1,59 +0,0 @@ -# Make PortAudio for Linux -# Updated 2001/08/25 Bill Eldridge bill@rfa.org -# Updated 2001/10/16, philburk@softsynth.com, s/unix_oss/unix_oss/ -# Updated 2002/04/30 Bill Eldridge bill@rfa.org -# Made the libinstall and tests compile a bit cleaner - -# A pretty bare makefile, that figures out all the test files -# and compiles them against the library in the pa_unix_oss directory. - -# Do "make all" and then when happy, "make libinstall" -# (if not happy, "make clean") - -# The ldconfig stuff in libinstall is the wrong way to do it - -# someone tell me the right way, please - - -LIBS = -lm -lpthread - -CDEFINES = -I../pa_common -CFLAGS = -g -LIBINST = /usr/local/lib - -TESTS:= $(wildcard pa_tests/pa*.c pa_tests/debug*.c) -TESTO:= $(wildcard pa_tests/pa*.o pa_tests/debug*.o) - -LIBFILES:= ./pa_common/pa_lib.c ./pa_unix_oss/pa_unix_oss.c ./pa_unix_oss/pa_unix.c - -#all: sharedlib libinstall tests -all: sharedlib libinstall testo testq - -.c.o: - -gcc $(CFLAGS) -c -I./pa_common $< -o $*.o - -.o: - -gcc $*.o -o $* -Lpa_unix_oss -lportaudio $(LIBS) - -#.c.o: -# -gcc -c -I./pa_common $< -o $*.o -# -gcc $*.o -o $* -Lpa_unix_oss $(LIBS) -lportaudio - - -sharedlib: $(LIBFILES:.c=.o) - gcc -shared -o ./pa_unix_oss/libportaudio.so ./pa_common/pa_lib.o ./pa_unix_oss/pa_unix_oss.o ./pa_unix_oss/pa_unix.o - -libinstall: ./pa_unix_oss/libportaudio.so - @cp -f ./pa_unix_oss/libportaudio.so $(LIBINST) - @/sbin/ldconfig - -testo: $(TESTS:.c=.o) - -testq: $(TESTO:.o=) - -clean: - -@rm -f $(TESTS:.c=.o) - -@rm -f $(TESTS:.c=) - -@rm -f $(LIBFILES:.c=.o) - -@rm -f ./pa_unix_oss/libportaudio.so - - diff --git a/pd/portaudio_v18/Makefile.mingw b/pd/portaudio_v18/Makefile.mingw deleted file mode 100644 index 2043a161..00000000 --- a/pd/portaudio_v18/Makefile.mingw +++ /dev/null @@ -1,57 +0,0 @@ - -# Makefile for PortAudio on mingw (http://mingw.sourceforge.net) - -# Contributed by Bill Eldridge, bill@rfa.org, Radio Free Asia -# Copyright 2002/02/20, GPL - -# Uses a common mingw32 cross-compiler that defaults -# to everything in /usr/local/cross-tools - -# First edit your path with -# export PATH=/usr/local/cross-tools/bin:$PATH - -# Usage: make -f Makefile.mingw all -# or make -f Makefile.mingw sharedlib -# make -f Makefile.mingw tests -# -# Then copy executables & portaudio.dll to your Windows machine -# -# To make work with pa_win_ds, you'll have to substitue -# all the pa_win_wmme files with pa_win_ds files, no biggie. - -CC= i586-mingw32msvc-gcc -DLLTOOL= i586-mingw32msvc-dlltool -DLLWRAP= i586-mingw32msvc-dllwrap - -ARCH= pa_win_wmme - -TESTS:= $(wildcard pa_tests/pa*.c pa_tests/debug*.c) - -.c.o: - -$(CC) -c -I./pa_common $< -o $*.o - -$(CC) $*.o -o $*.exe -L/usr/local/lib -L$(ARCH) -lportaudio.dll -lwinmm - -all: sharedlib tests - -sharedlib: ./pa_common/pa_lib.c - $(CC) -c -I./pa_common pa_common/pa_lib.c -o pa_common/pa_lib.o - $(CC) -c -I./pa_common pa_win_wmme/pa_win_wmme.c -o pa_win_wmme/pa_win_wmme.o - $(CC) -shared -mthreads -o portaudio.dll pa_common/pa_lib.o pa_win_wmme/pa_win_wmme.o -L/usr/local/cross-tools/i586-win32msvc/lib -lwinmm -lm - $(DLLWRAP) --export-all --output-def=libportaudio.def --output-lib=libportaudio.a --dllname=portaudio.dll --drivername=i586-mingw32msvc-gcc pa_common/pa_lib.o pa_win_wmme/pa_win_wmme.o -L/usr/local/cross-tools/i586-win32msvc/lib -lwinmm -lm - $(CC) -shared -Wl,--enable-auto-image-base -o portaudio.dll -Wl,--out-implib=pa_win_wmme/libportaudio.dll.a pa_common/pa_lib.o pa_win_wmme/pa_win_wmme.o -L/usr/local/cross-tools/i586-win32msvc/lib -lwinmm - - -tests: $(TESTS:.c=.o) - -sine: - $(CC) -c -I./pa_common pa_tests/patest_sine.c -o pa_tests/patest_sine.o - $(CC) pa_tests/patest_sine.o -o pa_tests/patest_sine.exe -L/usr/local/lib -lportaudio.dll -lwinmm - -clean: - -rm ./pa_tests/*.exe - -rm ./pa_tests/*.o - -nothing: - $(CC) pa_tests/patest_sine.o -L/usr/lib/w32api -L./pa_win_wmme -lportaudio.dll -lwinmm - - diff --git a/pd/portaudio_v18/Makefile.solaris b/pd/portaudio_v18/Makefile.solaris deleted file mode 100644 index ef3a4185..00000000 --- a/pd/portaudio_v18/Makefile.solaris +++ /dev/null @@ -1,59 +0,0 @@ -# Make PortAudio for Linux -# Updated 2001/08/25 Bill Eldridge bill@rfa.org -# Updated 2001/10/16, philburk@softsynth.com, s/unix_oss/unix_oss/ -# Updated 2002/04/30 Bill Eldridge bill@rfa.org -# Made the libinstall and tests compile a bit cleaner - -# A pretty bare makefile, that figures out all the test files -# and compiles them against the library in the pa_unix_oss directory. - -# Do "make all" and then when happy, "make libinstall" -# (if not happy, "make clean") - -# The ldconfig stuff in libinstall is the wrong way to do it - -# someone tell me the right way, please - - -LIBS = -lm -lpthread -lrt - -CDEFINES = -I../pa_common -CFLAGS = -g -LIBINST = /usr/local/lib - -TESTS:= $(wildcard pa_tests/pa*.c pa_tests/debug*.c) -TESTO:= $(wildcard pa_tests/pa*.o pa_tests/debug*.o) - -LIBFILES:= ./pa_common/pa_lib.c ./pa_unix_oss/pa_unix_solaris.c ./pa_unix_oss/pa_unix.c - -#all: sharedlib libinstall tests -all: sharedlib libinstall testo testq - -.c.o: - -gcc $(CFLAGS) -c -I./pa_common $< -o $*.o - -.o: - -gcc $*.o -o $* -Lpa_unix_oss -lportaudio $(LIBS) - -#.c.o: -# -gcc -c -I./pa_common $< -o $*.o -# -gcc $*.o -o $* -Lpa_unix_oss $(LIBS) -lportaudio - - -sharedlib: $(LIBFILES:.c=.o) - gcc -shared -o ./pa_unix_oss/libportaudio.so ./pa_common/pa_lib.o ./pa_unix_oss/pa_unix_solaris.o ./pa_unix_oss/pa_unix.o - -libinstall: ./pa_unix_oss/libportaudio.so - @cp -f ./pa_unix_oss/libportaudio.so $(LIBINST) - @/sbin/ldconfig - -testo: $(TESTS:.c=.o) - -testq: $(TESTO:.o=) - -clean: - -@rm -f $(TESTS:.c=.o) - -@rm -f $(TESTS:.c=) - -@rm -f $(LIBFILES:.c=.o) - -@rm -f ./pa_unix_oss/libportaudio.so - - diff --git a/pd/portaudio_v18/README.txt b/pd/portaudio_v18/README.txt deleted file mode 100644 index d1e5d7d6..00000000 --- a/pd/portaudio_v18/README.txt +++ /dev/null @@ -1,81 +0,0 @@ -README for PortAudio -Implementations for PC DirectSound and Mac SoundManager - -/* - * PortAudio Portable Real-Time Audio Library - * Latest Version at: http://www.portaudio.com// - * - * Copyright (c) 1999-2000 Phil Burk and Ross Bencina - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ - -PortAudio is a portable audio I/O library designed for cross-platform -support of audio. It uses a callback mechanism to request audio processing. -Audio can be generated in various formats, including 32 bit floating point, -and will be converted to the native format internally. - -Documentation: - See "pa_common/portaudio.h" for API spec. - See docs folder for a tutorial. - Also see http://www.portaudio.com/docs/ - And see "pa_tests/patest_saw.c" for an example. - -For information on compiling programs with PortAudio, please see the -tutorial at: - - http://www.portaudio.com/docs/pa_tutorial.html - -Important Files and Folders: - pa_common/ = platform independant code - pa_common/portaudio.h = header file for PortAudio API. Specifies API. - pa_common/pa_lib.c = host independant code for all implementations. - - pablio = simple blocking read/write interface - -Platform Implementations - pa_asio = ASIO for Windows and Macintosh - pa_beos = BeOS - pa_mac = Macintosh Sound Manager for OS 8,9 and Carbon - pa_mac_core = Macintosh Core Audio for OS X - pa_sgi = Silicon Graphics AL - pa_unix_oss = OSS implementation for various Unixes - pa_win_ds = Windows Direct Sound - pa_win_wmme = Windows MME (most widely supported) - -Test Programs - pa_tests/pa_fuzz.c = guitar fuzz box - pa_tests/pa_devs.c = print a list of available devices - pa_tests/pa_minlat.c = determine minimum latency for your machine - pa_tests/paqa_devs.c = self test that opens all devices - pa_tests/paqa_errs.c = test error detection and reporting - pa_tests/patest_clip.c = hear a sine wave clipped and unclipped - pa_tests/patest_dither.c = hear effects of dithering (extremely subtle) - pa_tests/patest_pink.c = fun with pink noise - pa_tests/patest_record.c = record and playback some audio - pa_tests/patest_maxsines.c = how many sine waves can we play? Tests Pa_GetCPULoad(). - pa_tests/patest_sine.c = output a sine wave in a simple PA app - pa_tests/patest_sync.c = test syncronization of audio and video - pa_tests/patest_wire.c = pass input to output, wire simulator diff --git a/pd/portaudio_v18/config.guess b/pd/portaudio_v18/config.guess deleted file mode 100644 index 297e5c30..00000000 --- a/pd/portaudio_v18/config.guess +++ /dev/null @@ -1,1308 +0,0 @@ -#! /bin/sh -# Attempt to guess a canonical system name. -# Copyright (C) 1992, 1993, 1994, 1995, 1996, 1997, 1998, 1999, 2000, 2001 -# Free Software Foundation, Inc. - -timestamp='2001-10-05' - -# This file is free software; you can redistribute it and/or modify it -# under the terms of the GNU General Public License as published by -# the Free Software Foundation; either version 2 of the License, or -# (at your option) any later version. -# -# This program is distributed in the hope that it will be useful, but -# WITHOUT ANY WARRANTY; without even the implied warranty of -# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -# General Public License for more details. -# -# You should have received a copy of the GNU General Public License -# along with this program; if not, write to the Free Software -# Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. -# -# As a special exception to the GNU General Public License, if you -# distribute this file as part of a program that contains a -# configuration script generated by Autoconf, you may include it under -# the same distribution terms that you use for the rest of that program. - -# Originally written by Per Bothner <bothner@cygnus.com>. -# Please send patches to <config-patches@gnu.org>. Submit a context -# diff and a properly formatted ChangeLog entry. -# -# This script attempts to guess a canonical system name similar to -# config.sub. If it succeeds, it prints the system name on stdout, and -# exits with 0. Otherwise, it exits with 1. -# -# The plan is that this can be called by configure scripts if you -# don't specify an explicit build system type. - -me=`echo "$0" | sed -e 's,.*/,,'` - -usage="\ -Usage: $0 [OPTION] - -Output the configuration name of the system \`$me' is run on. - -Operation modes: - -h, --help print this help, then exit - -t, --time-stamp print date of last modification, then exit - -v, --version print version number, then exit - -Report bugs and patches to <config-patches@gnu.org>." - -version="\ -GNU config.guess ($timestamp) - -Originally written by Per Bothner. -Copyright (C) 1992, 1993, 1994, 1995, 1996, 1997, 1998, 1999, 2000, 2001 -Free Software Foundation, Inc. - -This is free software; see the source for copying conditions. There is NO -warranty; not even for MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE." - -help=" -Try \`$me --help' for more information." - -# Parse command line -while test $# -gt 0 ; do - case $1 in - --time-stamp | --time* | -t ) - echo "$timestamp" ; exit 0 ;; - --version | -v ) - echo "$version" ; exit 0 ;; - --help | --h* | -h ) - echo "$usage"; exit 0 ;; - -- ) # Stop option processing - shift; break ;; - - ) # Use stdin as input. - break ;; - -* ) - echo "$me: invalid option $1$help" >&2 - exit 1 ;; - * ) - break ;; - esac -done - -if test $# != 0; then - echo "$me: too many arguments$help" >&2 - exit 1 -fi - - -dummy=dummy-$$ -trap 'rm -f $dummy.c $dummy.o $dummy.rel $dummy; exit 1' 1 2 15 - -# CC_FOR_BUILD -- compiler used by this script. -# Historically, `CC_FOR_BUILD' used to be named `HOST_CC'. We still -# use `HOST_CC' if defined, but it is deprecated. - -set_cc_for_build='case $CC_FOR_BUILD,$HOST_CC,$CC in - ,,) echo "int dummy(){}" > $dummy.c ; - for c in cc gcc c89 ; do - ($c $dummy.c -c -o $dummy.o) >/dev/null 2>&1 ; - if test $? = 0 ; then - CC_FOR_BUILD="$c"; break ; - fi ; - done ; - rm -f $dummy.c $dummy.o $dummy.rel ; - if test x"$CC_FOR_BUILD" = x ; then - CC_FOR_BUILD=no_compiler_found ; - fi - ;; - ,,*) CC_FOR_BUILD=$CC ;; - ,*,*) CC_FOR_BUILD=$HOST_CC ;; -esac' - -# This is needed to find uname on a Pyramid OSx when run in the BSD universe. -# (ghazi@noc.rutgers.edu 1994-08-24) -if (test -f /.attbin/uname) >/dev/null 2>&1 ; then - PATH=$PATH:/.attbin ; export PATH -fi - -UNAME_MACHINE=`(uname -m) 2>/dev/null` || UNAME_MACHINE=unknown -UNAME_RELEASE=`(uname -r) 2>/dev/null` || UNAME_RELEASE=unknown -UNAME_SYSTEM=`(uname -s) 2>/dev/null` || UNAME_SYSTEM=unknown -UNAME_VERSION=`(uname -v) 2>/dev/null` || UNAME_VERSION=unknown - -# Note: order is significant - the case branches are not exclusive. - -case "${UNAME_MACHINE}:${UNAME_SYSTEM}:${UNAME_RELEASE}:${UNAME_VERSION}" in - *:NetBSD:*:*) - # NetBSD (nbsd) targets should (where applicable) match one or - # more of the tupples: *-*-netbsdelf*, *-*-netbsdaout*, - # *-*-netbsdecoff* and *-*-netbsd*. For targets that recently - # switched to ELF, *-*-netbsd* would select the old - # object file format. This provides both forward - # compatibility and a consistent mechanism for selecting the - # object file format. - # Determine the machine/vendor (is the vendor relevant). - case "${UNAME_MACHINE}" in - amiga) machine=m68k-unknown ;; - arm32) machine=arm-unknown ;; - atari*) machine=m68k-atari ;; - sun3*) machine=m68k-sun ;; - mac68k) machine=m68k-apple ;; - macppc) machine=powerpc-apple ;; - hp3[0-9][05]) machine=m68k-hp ;; - ibmrt|romp-ibm) machine=romp-ibm ;; - sparc*) machine=`uname -p`-unknown ;; - *) machine=${UNAME_MACHINE}-unknown ;; - esac - # The Operating System including object format, if it has switched - # to ELF recently, or will in the future. - case "${UNAME_MACHINE}" in - i386|sparc|amiga|arm*|hp300|mvme68k|vax|atari|luna68k|mac68k|news68k|next68k|pc532|sun3*|x68k) - eval $set_cc_for_build - if echo __ELF__ | $CC_FOR_BUILD -E - 2>/dev/null \ - | grep __ELF__ >/dev/null - then - # Once all utilities can be ECOFF (netbsdecoff) or a.out (netbsdaout). - # Return netbsd for either. 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Ake) contributed MIS and NILE. - if test "`(/bin/universe) 2>/dev/null`" = att ; then - echo pyramid-pyramid-sysv3 - else - echo pyramid-pyramid-bsd - fi - exit 0 ;; - NILE*:*:*:dcosx) - echo pyramid-pyramid-svr4 - exit 0 ;; - sun4H:SunOS:5.*:*) - echo sparc-hal-solaris2`echo ${UNAME_RELEASE}|sed -e 's/[^.]*//'` - exit 0 ;; - sun4*:SunOS:5.*:* | tadpole*:SunOS:5.*:*) - echo sparc-sun-solaris2`echo ${UNAME_RELEASE}|sed -e 's/[^.]*//'` - exit 0 ;; - i86pc:SunOS:5.*:*) - echo i386-pc-solaris2`echo ${UNAME_RELEASE}|sed -e 's/[^.]*//'` - exit 0 ;; - sun4*:SunOS:6*:*) - # According to config.sub, this is the proper way to canonicalize - # SunOS6. Hard to guess exactly what SunOS6 will be like, but - # it's likely to be more like Solaris than SunOS4. - echo sparc-sun-solaris3`echo ${UNAME_RELEASE}|sed -e 's/[^.]*//'` - exit 0 ;; - sun4*:SunOS:*:*) - case "`/usr/bin/arch -k`" in - Series*|S4*) - UNAME_RELEASE=`uname -v` - ;; - esac - # Japanese Language versions have a version number like `4.1.3-JL'. - echo sparc-sun-sunos`echo ${UNAME_RELEASE}|sed -e 's/-/_/'` - exit 0 ;; - sun3*:SunOS:*:*) - echo m68k-sun-sunos${UNAME_RELEASE} - exit 0 ;; - sun*:*:4.2BSD:*) - UNAME_RELEASE=`(head -1 /etc/motd | awk '{print substr($5,1,3)}') 2>/dev/null` - test "x${UNAME_RELEASE}" = "x" && UNAME_RELEASE=3 - case "`/bin/arch`" in - sun3) - echo m68k-sun-sunos${UNAME_RELEASE} - ;; - sun4) - echo sparc-sun-sunos${UNAME_RELEASE} - ;; - esac - exit 0 ;; - aushp:SunOS:*:*) - echo sparc-auspex-sunos${UNAME_RELEASE} - exit 0 ;; - # The situation for MiNT is a little confusing. The machine name - # can be virtually everything (everything which is not - # "atarist" or "atariste" at least should have a processor - # > m68000). The system name ranges from "MiNT" over "FreeMiNT" - # to the lowercase version "mint" (or "freemint"). Finally - # the system name "TOS" denotes a system which is actually not - # MiNT. But MiNT is downward compatible to TOS, so this should - # be no problem. - atarist[e]:*MiNT:*:* | atarist[e]:*mint:*:* | atarist[e]:*TOS:*:*) - echo m68k-atari-mint${UNAME_RELEASE} - exit 0 ;; - atari*:*MiNT:*:* | atari*:*mint:*:* | atarist[e]:*TOS:*:*) - echo m68k-atari-mint${UNAME_RELEASE} - exit 0 ;; - *falcon*:*MiNT:*:* | *falcon*:*mint:*:* | *falcon*:*TOS:*:*) - echo m68k-atari-mint${UNAME_RELEASE} - exit 0 ;; - milan*:*MiNT:*:* | milan*:*mint:*:* | *milan*:*TOS:*:*) - echo m68k-milan-mint${UNAME_RELEASE} - exit 0 ;; - hades*:*MiNT:*:* | hades*:*mint:*:* | *hades*:*TOS:*:*) - echo m68k-hades-mint${UNAME_RELEASE} - exit 0 ;; - *:*MiNT:*:* | *:*mint:*:* | *:*TOS:*:*) - echo m68k-unknown-mint${UNAME_RELEASE} - exit 0 ;; - powerpc:machten:*:*) - echo powerpc-apple-machten${UNAME_RELEASE} - exit 0 ;; - RISC*:Mach:*:*) - echo mips-dec-mach_bsd4.3 - exit 0 ;; - RISC*:ULTRIX:*:*) - echo mips-dec-ultrix${UNAME_RELEASE} - exit 0 ;; - VAX*:ULTRIX*:*:*) - echo vax-dec-ultrix${UNAME_RELEASE} - exit 0 ;; - 2020:CLIX:*:* | 2430:CLIX:*:*) - echo clipper-intergraph-clix${UNAME_RELEASE} - exit 0 ;; - mips:*:*:UMIPS | mips:*:*:RISCos) - eval $set_cc_for_build - sed 's/^ //' << EOF >$dummy.c -#ifdef __cplusplus -#include <stdio.h> /* for printf() prototype */ - int main (int argc, char *argv[]) { -#else - int main (argc, argv) int argc; char *argv[]; { -#endif - #if defined (host_mips) && defined (MIPSEB) - #if defined (SYSTYPE_SYSV) - printf ("mips-mips-riscos%ssysv\n", argv[1]); exit (0); - #endif - #if defined (SYSTYPE_SVR4) - printf ("mips-mips-riscos%ssvr4\n", argv[1]); exit (0); - #endif - #if defined (SYSTYPE_BSD43) || defined(SYSTYPE_BSD) - printf ("mips-mips-riscos%sbsd\n", argv[1]); exit (0); - #endif - #endif - exit (-1); - } -EOF - $CC_FOR_BUILD $dummy.c -o $dummy \ - && ./$dummy `echo "${UNAME_RELEASE}" | sed -n 's/\([0-9]*\).*/\1/p'` \ - && rm -f $dummy.c $dummy && exit 0 - rm -f $dummy.c $dummy - echo mips-mips-riscos${UNAME_RELEASE} - exit 0 ;; - Motorola:PowerMAX_OS:*:*) - echo powerpc-motorola-powermax - exit 0 ;; - Night_Hawk:Power_UNIX:*:*) - echo powerpc-harris-powerunix - exit 0 ;; - m88k:CX/UX:7*:*) - echo m88k-harris-cxux7 - exit 0 ;; - m88k:*:4*:R4*) - echo m88k-motorola-sysv4 - exit 0 ;; - m88k:*:3*:R3*) - echo m88k-motorola-sysv3 - exit 0 ;; - AViiON:dgux:*:*) - # DG/UX returns AViiON for all architectures - UNAME_PROCESSOR=`/usr/bin/uname -p` - if [ $UNAME_PROCESSOR = mc88100 ] || [ $UNAME_PROCESSOR = mc88110 ] - then - if [ ${TARGET_BINARY_INTERFACE}x = m88kdguxelfx ] || \ - [ ${TARGET_BINARY_INTERFACE}x = x ] - then - echo m88k-dg-dgux${UNAME_RELEASE} - else - echo m88k-dg-dguxbcs${UNAME_RELEASE} - fi - else - echo i586-dg-dgux${UNAME_RELEASE} - fi - exit 0 ;; - M88*:DolphinOS:*:*) # DolphinOS (SVR3) - echo m88k-dolphin-sysv3 - exit 0 ;; - M88*:*:R3*:*) - # Delta 88k system running SVR3 - echo m88k-motorola-sysv3 - exit 0 ;; - XD88*:*:*:*) # Tektronix XD88 system running UTekV (SVR3) - echo m88k-tektronix-sysv3 - exit 0 ;; - Tek43[0-9][0-9]:UTek:*:*) # Tektronix 4300 system running UTek (BSD) - echo m68k-tektronix-bsd - exit 0 ;; - *:IRIX*:*:*) - echo mips-sgi-irix`echo ${UNAME_RELEASE}|sed -e 's/-/_/g'` - exit 0 ;; - ????????:AIX?:[12].1:2) # AIX 2.2.1 or AIX 2.1.1 is RT/PC AIX. - echo romp-ibm-aix # uname -m gives an 8 hex-code CPU id - exit 0 ;; # Note that: echo "'`uname -s`'" gives 'AIX ' - i*86:AIX:*:*) - echo i386-ibm-aix - exit 0 ;; - ia64:AIX:*:*) - if [ -x /usr/bin/oslevel ] ; then - IBM_REV=`/usr/bin/oslevel` - else - IBM_REV=${UNAME_VERSION}.${UNAME_RELEASE} - fi - echo ${UNAME_MACHINE}-ibm-aix${IBM_REV} - exit 0 ;; - *:AIX:2:3) - if grep bos325 /usr/include/stdio.h >/dev/null 2>&1; then - eval $set_cc_for_build - sed 's/^ //' << EOF >$dummy.c - #include <sys/systemcfg.h> - - main() - { - if (!__power_pc()) - exit(1); - puts("powerpc-ibm-aix3.2.5"); - exit(0); - } -EOF - $CC_FOR_BUILD $dummy.c -o $dummy && ./$dummy && rm -f $dummy.c $dummy && exit 0 - rm -f $dummy.c $dummy - echo rs6000-ibm-aix3.2.5 - elif grep bos324 /usr/include/stdio.h >/dev/null 2>&1; then - echo rs6000-ibm-aix3.2.4 - else - echo rs6000-ibm-aix3.2 - fi - exit 0 ;; - *:AIX:*:[45]) - IBM_CPU_ID=`/usr/sbin/lsdev -C -c processor -S available | head -1 | awk '{ print $1 }'` - if /usr/sbin/lsattr -El ${IBM_CPU_ID} | grep ' POWER' >/dev/null 2>&1; then - IBM_ARCH=rs6000 - else - IBM_ARCH=powerpc - fi - if [ -x /usr/bin/oslevel ] ; then - IBM_REV=`/usr/bin/oslevel` - else - IBM_REV=${UNAME_VERSION}.${UNAME_RELEASE} - fi - echo ${IBM_ARCH}-ibm-aix${IBM_REV} - exit 0 ;; - *:AIX:*:*) - echo rs6000-ibm-aix - exit 0 ;; - ibmrt:4.4BSD:*|romp-ibm:BSD:*) - echo romp-ibm-bsd4.4 - exit 0 ;; - ibmrt:*BSD:*|romp-ibm:BSD:*) # covers RT/PC BSD and - echo romp-ibm-bsd${UNAME_RELEASE} # 4.3 with uname added to - exit 0 ;; # report: romp-ibm BSD 4.3 - *:BOSX:*:*) - echo rs6000-bull-bosx - exit 0 ;; - DPX/2?00:B.O.S.:*:*) - echo m68k-bull-sysv3 - exit 0 ;; - 9000/[34]??:4.3bsd:1.*:*) - echo m68k-hp-bsd - exit 0 ;; - hp300:4.4BSD:*:* | 9000/[34]??:4.3bsd:2.*:*) - echo m68k-hp-bsd4.4 - exit 0 ;; - 9000/[34678]??:HP-UX:*:*) - HPUX_REV=`echo ${UNAME_RELEASE}|sed -e 's/[^.]*.[0B]*//'` - case "${UNAME_MACHINE}" in - 9000/31? ) HP_ARCH=m68000 ;; - 9000/[34]?? ) HP_ARCH=m68k ;; - 9000/[678][0-9][0-9]) - if [ -x /usr/bin/getconf ]; then - sc_cpu_version=`/usr/bin/getconf SC_CPU_VERSION 2>/dev/null` - sc_kernel_bits=`/usr/bin/getconf SC_KERNEL_BITS 2>/dev/null` - case "${sc_cpu_version}" in - 523) HP_ARCH="hppa1.0" ;; # CPU_PA_RISC1_0 - 528) HP_ARCH="hppa1.1" ;; # CPU_PA_RISC1_1 - 532) # CPU_PA_RISC2_0 - case "${sc_kernel_bits}" in - 32) HP_ARCH="hppa2.0n" ;; - 64) HP_ARCH="hppa2.0w" ;; - '') HP_ARCH="hppa2.0" ;; # HP-UX 10.20 - esac ;; - esac - fi - if [ "${HP_ARCH}" = "" ]; then - eval $set_cc_for_build - sed 's/^ //' << EOF >$dummy.c - - #define _HPUX_SOURCE - #include <stdlib.h> - #include <unistd.h> - - int main () - { - #if defined(_SC_KERNEL_BITS) - long bits = sysconf(_SC_KERNEL_BITS); - #endif - long cpu = sysconf (_SC_CPU_VERSION); - - switch (cpu) - { - case CPU_PA_RISC1_0: puts ("hppa1.0"); break; - case CPU_PA_RISC1_1: puts ("hppa1.1"); break; - case CPU_PA_RISC2_0: - #if defined(_SC_KERNEL_BITS) - switch (bits) - { - case 64: puts ("hppa2.0w"); break; - case 32: puts ("hppa2.0n"); break; - default: puts ("hppa2.0"); break; - } break; - #else /* !defined(_SC_KERNEL_BITS) */ - puts ("hppa2.0"); break; - #endif - default: puts ("hppa1.0"); break; - } - exit (0); - } -EOF - (CCOPTS= $CC_FOR_BUILD $dummy.c -o $dummy 2>/dev/null) && HP_ARCH=`./$dummy` - if test -z "$HP_ARCH"; then HP_ARCH=hppa; fi - rm -f $dummy.c $dummy - fi ;; - esac - echo ${HP_ARCH}-hp-hpux${HPUX_REV} - exit 0 ;; - ia64:HP-UX:*:*) - HPUX_REV=`echo ${UNAME_RELEASE}|sed -e 's/[^.]*.[0B]*//'` - echo ia64-hp-hpux${HPUX_REV} - exit 0 ;; - 3050*:HI-UX:*:*) - eval $set_cc_for_build - sed 's/^ //' << EOF >$dummy.c - #include <unistd.h> - int - main () - { - long cpu = sysconf (_SC_CPU_VERSION); - /* The order matters, because CPU_IS_HP_MC68K erroneously returns - true for CPU_PA_RISC1_0. CPU_IS_PA_RISC returns correct - results, however. */ - if (CPU_IS_PA_RISC (cpu)) - { - switch (cpu) - { - case CPU_PA_RISC1_0: puts ("hppa1.0-hitachi-hiuxwe2"); break; - case CPU_PA_RISC1_1: puts ("hppa1.1-hitachi-hiuxwe2"); break; - case CPU_PA_RISC2_0: puts ("hppa2.0-hitachi-hiuxwe2"); break; - default: puts ("hppa-hitachi-hiuxwe2"); break; - } - } - else if (CPU_IS_HP_MC68K (cpu)) - puts ("m68k-hitachi-hiuxwe2"); - else puts ("unknown-hitachi-hiuxwe2"); - exit (0); - } -EOF - $CC_FOR_BUILD $dummy.c -o $dummy && ./$dummy && rm -f $dummy.c $dummy && exit 0 - rm -f $dummy.c $dummy - echo unknown-hitachi-hiuxwe2 - exit 0 ;; - 9000/7??:4.3bsd:*:* | 9000/8?[79]:4.3bsd:*:* ) - echo hppa1.1-hp-bsd - exit 0 ;; - 9000/8??:4.3bsd:*:*) - echo hppa1.0-hp-bsd - exit 0 ;; - *9??*:MPE/iX:*:* | *3000*:MPE/iX:*:*) - echo hppa1.0-hp-mpeix - exit 0 ;; - hp7??:OSF1:*:* | hp8?[79]:OSF1:*:* ) - echo hppa1.1-hp-osf - exit 0 ;; - hp8??:OSF1:*:*) - echo hppa1.0-hp-osf - exit 0 ;; - i*86:OSF1:*:*) - if [ -x /usr/sbin/sysversion ] ; then - echo ${UNAME_MACHINE}-unknown-osf1mk - else - echo ${UNAME_MACHINE}-unknown-osf1 - fi - exit 0 ;; - parisc*:Lites*:*:*) - echo hppa1.1-hp-lites - exit 0 ;; - C1*:ConvexOS:*:* | convex:ConvexOS:C1*:*) - echo c1-convex-bsd - exit 0 ;; - C2*:ConvexOS:*:* | convex:ConvexOS:C2*:*) - if getsysinfo -f scalar_acc - then echo c32-convex-bsd - else echo c2-convex-bsd - fi - exit 0 ;; - C34*:ConvexOS:*:* | convex:ConvexOS:C34*:*) - echo c34-convex-bsd - exit 0 ;; - C38*:ConvexOS:*:* | convex:ConvexOS:C38*:*) - echo c38-convex-bsd - exit 0 ;; - C4*:ConvexOS:*:* | convex:ConvexOS:C4*:*) - echo c4-convex-bsd - exit 0 ;; - CRAY*X-MP:*:*:*) - echo xmp-cray-unicos - exit 0 ;; - CRAY*Y-MP:*:*:*) - echo ymp-cray-unicos${UNAME_RELEASE} | sed -e 's/\.[^.]*$/.X/' - exit 0 ;; - CRAY*[A-Z]90:*:*:*) - echo ${UNAME_MACHINE}-cray-unicos${UNAME_RELEASE} \ - | sed -e 's/CRAY.*\([A-Z]90\)/\1/' \ - -e y/ABCDEFGHIJKLMNOPQRSTUVWXYZ/abcdefghijklmnopqrstuvwxyz/ \ - -e 's/\.[^.]*$/.X/' - exit 0 ;; - CRAY*TS:*:*:*) - echo t90-cray-unicos${UNAME_RELEASE} | sed -e 's/\.[^.]*$/.X/' - exit 0 ;; - CRAY*T3D:*:*:*) - echo alpha-cray-unicosmk${UNAME_RELEASE} | sed -e 's/\.[^.]*$/.X/' - exit 0 ;; - CRAY*T3E:*:*:*) - echo alphaev5-cray-unicosmk${UNAME_RELEASE} | sed -e 's/\.[^.]*$/.X/' - exit 0 ;; - CRAY*SV1:*:*:*) - echo sv1-cray-unicos${UNAME_RELEASE} | sed -e 's/\.[^.]*$/.X/' - exit 0 ;; - CRAY-2:*:*:*) - echo cray2-cray-unicos - exit 0 ;; - F30[01]:UNIX_System_V:*:* | F700:UNIX_System_V:*:*) - FUJITSU_PROC=`uname -m | tr 'ABCDEFGHIJKLMNOPQRSTUVWXYZ' 'abcdefghijklmnopqrstuvwxyz'` - FUJITSU_SYS=`uname -p | tr 'ABCDEFGHIJKLMNOPQRSTUVWXYZ' 'abcdefghijklmnopqrstuvwxyz' | sed -e 's/\///'` - FUJITSU_REL=`echo ${UNAME_RELEASE} | sed -e 's/ /_/'` - echo "${FUJITSU_PROC}-fujitsu-${FUJITSU_SYS}${FUJITSU_REL}" - exit 0 ;; - i*86:BSD/386:*:* | i*86:BSD/OS:*:* | *:Ascend\ Embedded/OS:*:*) - echo ${UNAME_MACHINE}-pc-bsdi${UNAME_RELEASE} - exit 0 ;; - sparc*:BSD/OS:*:*) - echo sparc-unknown-bsdi${UNAME_RELEASE} - exit 0 ;; - *:BSD/OS:*:*) - echo ${UNAME_MACHINE}-unknown-bsdi${UNAME_RELEASE} - exit 0 ;; - *:FreeBSD:*:*) - echo ${UNAME_MACHINE}-unknown-freebsd`echo ${UNAME_RELEASE}|sed -e 's/[-(].*//'` - exit 0 ;; - i*:CYGWIN*:*) - echo ${UNAME_MACHINE}-pc-cygwin - exit 0 ;; - i*:MINGW*:*) - echo ${UNAME_MACHINE}-pc-mingw32 - exit 0 ;; - i*:PW*:*) - echo ${UNAME_MACHINE}-pc-pw32 - exit 0 ;; - i*:Windows_NT*:* | Pentium*:Windows_NT*:*) - # How do we know it's Interix rather than the generic POSIX subsystem? - # It also conflicts with pre-2.0 versions of AT&T UWIN. Should we - # UNAME_MACHINE based on the output of uname instead of i386? - echo i386-pc-interix - exit 0 ;; - i*:UWIN*:*) - echo ${UNAME_MACHINE}-pc-uwin - exit 0 ;; - p*:CYGWIN*:*) - echo powerpcle-unknown-cygwin - exit 0 ;; - prep*:SunOS:5.*:*) - echo powerpcle-unknown-solaris2`echo ${UNAME_RELEASE}|sed -e 's/[^.]*//'` - exit 0 ;; - *:GNU:*:*) - echo `echo ${UNAME_MACHINE}|sed -e 's,[-/].*$,,'`-unknown-gnu`echo ${UNAME_RELEASE}|sed -e 's,/.*$,,'` - exit 0 ;; - i*86:Minix:*:*) - echo ${UNAME_MACHINE}-pc-minix - exit 0 ;; - arm*:Linux:*:*) - echo ${UNAME_MACHINE}-unknown-linux-gnu - exit 0 ;; - ia64:Linux:*:*) - echo ${UNAME_MACHINE}-unknown-linux - exit 0 ;; - m68*:Linux:*:*) - echo ${UNAME_MACHINE}-unknown-linux-gnu - exit 0 ;; - mips:Linux:*:*) - case `sed -n '/^byte/s/^.*: \(.*\) endian/\1/p' < /proc/cpuinfo` in - big) echo mips-unknown-linux-gnu && exit 0 ;; - little) echo mipsel-unknown-linux-gnu && exit 0 ;; - esac - ;; - ppc:Linux:*:*) - echo powerpc-unknown-linux-gnu - exit 0 ;; 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- sh*:Linux:*:*) - echo ${UNAME_MACHINE}-unknown-linux-gnu - exit 0 ;; - sparc:Linux:*:* | sparc64:Linux:*:*) - echo ${UNAME_MACHINE}-unknown-linux-gnu - exit 0 ;; - x86_64:Linux:*:*) - echo x86_64-unknown-linux-gnu - exit 0 ;; - i*86:Linux:*:*) - # The BFD linker knows what the default object file format is, so - # first see if it will tell us. cd to the root directory to prevent - # problems with other programs or directories called `ld' in the path. - ld_supported_targets=`cd /; ld --help 2>&1 \ - | sed -ne '/supported targets:/!d - s/[ ][ ]*/ /g - s/.*supported targets: *// - s/ .*// - p'` - case "$ld_supported_targets" in - elf32-i386) - TENTATIVE="${UNAME_MACHINE}-pc-linux-gnu" - ;; - a.out-i386-linux) - echo "${UNAME_MACHINE}-pc-linux-gnuaout" - exit 0 ;; - coff-i386) - echo "${UNAME_MACHINE}-pc-linux-gnucoff" - exit 0 ;; - "") - # Either a pre-BFD a.out linker (linux-gnuoldld) or - # one that does not give us useful --help. - echo "${UNAME_MACHINE}-pc-linux-gnuoldld" - exit 0 ;; - esac - # Determine whether the default compiler is a.out or elf - eval $set_cc_for_build - cat >$dummy.c <<EOF -#include <features.h> -#ifdef __cplusplus -#include <stdio.h> /* for printf() prototype */ - int main (int argc, char *argv[]) { -#else - int main (argc, argv) int argc; char *argv[]; { -#endif -#ifdef __ELF__ -# ifdef __GLIBC__ -# if __GLIBC__ >= 2 - printf ("%s-pc-linux-gnu\n", argv[1]); -# else - printf ("%s-pc-linux-gnulibc1\n", argv[1]); -# endif -# else - printf ("%s-pc-linux-gnulibc1\n", argv[1]); -# endif -#else - printf ("%s-pc-linux-gnuaout\n", argv[1]); -#endif - return 0; -} -EOF - $CC_FOR_BUILD $dummy.c -o $dummy 2>/dev/null && ./$dummy "${UNAME_MACHINE}" && rm -f $dummy.c $dummy && exit 0 - rm -f $dummy.c $dummy - test x"${TENTATIVE}" != x && echo "${TENTATIVE}" && exit 0 - ;; - i*86:DYNIX/ptx:4*:*) - # ptx 4.0 does uname -s correctly, with DYNIX/ptx in there. - # earlier versions are messed up and put the nodename in both - # sysname and nodename. - echo i386-sequent-sysv4 - exit 0 ;; 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- rs6000:LynxOS:2.*:*) - echo rs6000-unknown-lynxos${UNAME_RELEASE} - exit 0 ;; - PowerPC:LynxOS:2.*:* | PowerPC:LynxOS:3.[01]*:* | PowerPC:LynxOS:4.0*:*) - echo powerpc-unknown-lynxos${UNAME_RELEASE} - exit 0 ;; - SM[BE]S:UNIX_SV:*:*) - echo mips-dde-sysv${UNAME_RELEASE} - exit 0 ;; - RM*:ReliantUNIX-*:*:*) - echo mips-sni-sysv4 - exit 0 ;; - RM*:SINIX-*:*:*) - echo mips-sni-sysv4 - exit 0 ;; - *:SINIX-*:*:*) - if uname -p 2>/dev/null >/dev/null ; then - UNAME_MACHINE=`(uname -p) 2>/dev/null` - echo ${UNAME_MACHINE}-sni-sysv4 - else - echo ns32k-sni-sysv - fi - exit 0 ;; - PENTIUM:*:4.0*:*) # Unisys `ClearPath HMP IX 4000' SVR4/MP effort - # says <Richard.M.Bartel@ccMail.Census.GOV> - echo i586-unisys-sysv4 - exit 0 ;; - *:UNIX_System_V:4*:FTX*) - # From Gerald Hewes <hewes@openmarket.com>. - # How about differentiating between stratus architectures? -djm - echo hppa1.1-stratus-sysv4 - exit 0 ;; - *:*:*:FTX*) - # From seanf@swdc.stratus.com. - echo i860-stratus-sysv4 - exit 0 ;; - *:VOS:*:*) - # From Paul.Green@stratus.com. - echo hppa1.1-stratus-vos - exit 0 ;; - mc68*:A/UX:*:*) - echo m68k-apple-aux${UNAME_RELEASE} - exit 0 ;; - news*:NEWS-OS:6*:*) - echo mips-sony-newsos6 - exit 0 ;; - R[34]000:*System_V*:*:* | R4000:UNIX_SYSV:*:* | R*000:UNIX_SV:*:*) - if [ -d /usr/nec ]; then - echo mips-nec-sysv${UNAME_RELEASE} - else - echo mips-unknown-sysv${UNAME_RELEASE} - fi - exit 0 ;; - BeBox:BeOS:*:*) # BeOS running on hardware made by Be, PPC only. - echo powerpc-be-beos - exit 0 ;; - BeMac:BeOS:*:*) # BeOS running on Mac or Mac clone, PPC only. - echo powerpc-apple-beos - exit 0 ;; - BePC:BeOS:*:*) # BeOS running on Intel PC compatible. - echo i586-pc-beos - exit 0 ;; - SX-4:SUPER-UX:*:*) - echo sx4-nec-superux${UNAME_RELEASE} - exit 0 ;; - SX-5:SUPER-UX:*:*) - echo sx5-nec-superux${UNAME_RELEASE} - exit 0 ;; - Power*:Rhapsody:*:*) - echo powerpc-apple-rhapsody${UNAME_RELEASE} - exit 0 ;; - *:Rhapsody:*:*) - echo ${UNAME_MACHINE}-apple-rhapsody${UNAME_RELEASE} - exit 0 ;; - *:Darwin:*:*) - echo `uname -p`-apple-darwin${UNAME_RELEASE} - exit 0 ;; - *:procnto*:*:* | *:QNX:[0123456789]*:*) - if test "${UNAME_MACHINE}" = "x86pc"; then - UNAME_MACHINE=pc - fi - echo `uname -p`-${UNAME_MACHINE}-nto-qnx - exit 0 ;; - *:QNX:*:4*) - echo i386-pc-qnx - exit 0 ;; - NSR-[KW]:NONSTOP_KERNEL:*:*) - echo nsr-tandem-nsk${UNAME_RELEASE} - exit 0 ;; - *:NonStop-UX:*:*) - echo mips-compaq-nonstopux - exit 0 ;; - BS2000:POSIX*:*:*) - echo bs2000-siemens-sysv - exit 0 ;; - DS/*:UNIX_System_V:*:*) - echo ${UNAME_MACHINE}-${UNAME_SYSTEM}-${UNAME_RELEASE} - exit 0 ;; - *:Plan9:*:*) - # "uname -m" is not consistent, so use $cputype instead. 386 - # is converted to i386 for consistency with other x86 - # operating systems. - if test "$cputype" = "386"; then - UNAME_MACHINE=i386 - else - UNAME_MACHINE="$cputype" - fi - echo ${UNAME_MACHINE}-unknown-plan9 - exit 0 ;; - i*86:OS/2:*:*) - # If we were able to find `uname', then EMX Unix compatibility - # is probably installed. - echo ${UNAME_MACHINE}-pc-os2-emx - exit 0 ;; - *:TOPS-10:*:*) - echo pdp10-unknown-tops10 - exit 0 ;; - *:TENEX:*:*) - echo pdp10-unknown-tenex - exit 0 ;; - KS10:TOPS-20:*:* | KL10:TOPS-20:*:* | TYPE4:TOPS-20:*:*) - echo pdp10-dec-tops20 - exit 0 ;; - XKL-1:TOPS-20:*:* | TYPE5:TOPS-20:*:*) - echo pdp10-xkl-tops20 - exit 0 ;; - *:TOPS-20:*:*) - echo pdp10-unknown-tops20 - exit 0 ;; - *:ITS:*:*) - echo pdp10-unknown-its - exit 0 ;; - i*86:XTS-300:*:STOP) - echo ${UNAME_MACHINE}-unknown-stop - exit 0 ;; - i*86:atheos:*:*) - echo ${UNAME_MACHINE}-unknown-atheos - exit 0 ;; -esac - -#echo '(No uname command or uname output not recognized.)' 1>&2 -#echo "${UNAME_MACHINE}:${UNAME_SYSTEM}:${UNAME_RELEASE}:${UNAME_VERSION}" 1>&2 - -eval $set_cc_for_build -cat >$dummy.c <<EOF -#ifdef _SEQUENT_ -# include <sys/types.h> -# include <sys/utsname.h> -#endif -main () -{ -#if defined (sony) -#if defined (MIPSEB) - /* BFD wants "bsd" instead of "newsos". 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It is advised that you -download the most up to date version of the config scripts from - - ftp://ftp.gnu.org/pub/gnu/config/ - -If the version you run ($0) is already up to date, please -send the following data and any information you think might be -pertinent to <config-patches@gnu.org> in order to provide the needed -information to handle your system. - -config.guess timestamp = $timestamp - -uname -m = `(uname -m) 2>/dev/null || echo unknown` -uname -r = `(uname -r) 2>/dev/null || echo unknown` -uname -s = `(uname -s) 2>/dev/null || echo unknown` -uname -v = `(uname -v) 2>/dev/null || echo unknown` - -/usr/bin/uname -p = `(/usr/bin/uname -p) 2>/dev/null` -/bin/uname -X = `(/bin/uname -X) 2>/dev/null` - -hostinfo = `(hostinfo) 2>/dev/null` -/bin/universe = `(/bin/universe) 2>/dev/null` -/usr/bin/arch -k = `(/usr/bin/arch -k) 2>/dev/null` -/bin/arch = `(/bin/arch) 2>/dev/null` -/usr/bin/oslevel = `(/usr/bin/oslevel) 2>/dev/null` -/usr/convex/getsysinfo = `(/usr/convex/getsysinfo) 2>/dev/null` - -UNAME_MACHINE = ${UNAME_MACHINE} -UNAME_RELEASE = ${UNAME_RELEASE} -UNAME_SYSTEM = ${UNAME_SYSTEM} -UNAME_VERSION = ${UNAME_VERSION} -EOF - -exit 1 - -# Local variables: -# eval: (add-hook 'write-file-hooks 'time-stamp) -# time-stamp-start: "timestamp='" -# time-stamp-format: "%:y-%02m-%02d" -# time-stamp-end: "'" -# End: diff --git a/pd/portaudio_v18/config.sub b/pd/portaudio_v18/config.sub deleted file mode 100644 index 791bcded..00000000 --- a/pd/portaudio_v18/config.sub +++ /dev/null @@ -1,1413 +0,0 @@ -#! /bin/sh -# Configuration validation subroutine script. -# Copyright (C) 1992, 1993, 1994, 1995, 1996, 1997, 1998, 1999, 2000, 2001 -# Free Software Foundation, Inc. - -timestamp='2001-10-05' - -# This file is (in principle) common to ALL GNU software. -# The presence of a machine in this file suggests that SOME GNU software -# can handle that machine. 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Should this be sysv3.2? - i*86v32) - basic_machine=`echo $1 | sed -e 's/86.*/86-pc/'` - os=-sysv32 - ;; - i*86v4*) - basic_machine=`echo $1 | sed -e 's/86.*/86-pc/'` - os=-sysv4 - ;; - i*86v) - basic_machine=`echo $1 | sed -e 's/86.*/86-pc/'` - os=-sysv - ;; - i*86sol2) - basic_machine=`echo $1 | sed -e 's/86.*/86-pc/'` - os=-solaris2 - ;; - i386mach) - basic_machine=i386-mach - os=-mach - ;; - i386-vsta | vsta) - basic_machine=i386-unknown - os=-vsta - ;; - iris | iris4d) - basic_machine=mips-sgi - case $os in - -irix*) - ;; - *) - os=-irix4 - ;; - esac - ;; - isi68 | isi) - basic_machine=m68k-isi - os=-sysv - ;; - m88k-omron*) - basic_machine=m88k-omron - ;; - magnum | m3230) - basic_machine=mips-mips - os=-sysv - ;; - merlin) - basic_machine=ns32k-utek - os=-sysv - ;; - mingw32) - basic_machine=i386-pc - os=-mingw32 - ;; - miniframe) - basic_machine=m68000-convergent - ;; - *mint | -mint[0-9]* | *MiNT | *MiNT[0-9]*) - basic_machine=m68k-atari - os=-mint - ;; - mipsel*-linux*) - basic_machine=mipsel-unknown - os=-linux-gnu - ;; 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- ps2) - basic_machine=i386-ibm - ;; - pw32) - basic_machine=i586-unknown - os=-pw32 - ;; - rom68k) - basic_machine=m68k-rom68k - os=-coff - ;; - rm[46]00) - basic_machine=mips-siemens - ;; - rtpc | rtpc-*) - basic_machine=romp-ibm - ;; - sa29200) - basic_machine=a29k-amd - os=-udi - ;; - sequent) - basic_machine=i386-sequent - ;; - sh) - basic_machine=sh-hitachi - os=-hms - ;; - sparclite-wrs | simso-wrs) - basic_machine=sparclite-wrs - os=-vxworks - ;; - sps7) - basic_machine=m68k-bull - os=-sysv2 - ;; - spur) - basic_machine=spur-unknown - ;; - st2000) - basic_machine=m68k-tandem - ;; - stratus) - basic_machine=i860-stratus - os=-sysv4 - ;; - sun2) - basic_machine=m68000-sun - ;; - sun2os3) - basic_machine=m68000-sun - os=-sunos3 - ;; - sun2os4) - basic_machine=m68000-sun - os=-sunos4 - ;; - sun3os3) - basic_machine=m68k-sun - os=-sunos3 - ;; - sun3os4) - basic_machine=m68k-sun - os=-sunos4 - ;; - sun4os3) - basic_machine=sparc-sun - os=-sunos3 - ;; - sun4os4) - basic_machine=sparc-sun - os=-sunos4 - ;; 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When coming back to configure, we -# need to make the FD available again. -if test "$no_create" != yes; then - ac_cs_success=: - exec 5>/dev/null - $SHELL $CONFIG_STATUS || ac_cs_success=false - exec 5>>config.log - # Use ||, not &&, to avoid exiting from the if with $? = 1, which - # would make configure fail if this is the last instruction. - $ac_cs_success || { (exit 1); exit 1; } -fi - -echo "" -echo "Finished configure." - -echo "" -echo "Type 'make' to build PortAudio and examples." diff --git a/pd/portaudio_v18/configure.in b/pd/portaudio_v18/configure.in deleted file mode 100644 index d3a9946c..00000000 --- a/pd/portaudio_v18/configure.in +++ /dev/null @@ -1,81 +0,0 @@ -dnl -dnl PortAudio configure.in script -dnl -dnl Dominic Mazzoni -dnl - -dnl Require autoconf >= 2.13 -AC_PREREQ(2.13) - -dnl Init autoconf and make sure configure is being called -dnl from the right directory -AC_INIT([pa_common/portaudio.h]) - -dnl Checks for programs -AC_PROG_CC -AC_PROG_RANLIB -AC_PROG_INSTALL -AC_PATH_PROG(AR, ar, no) -if [[ $AR = "no" ]] ; then - AC_MSG_ERROR("Could not find ar - needed to create a library"); -fi - -dnl Extra variables we want to substitute -AC_SUBST(OTHER_OBJS) -AC_SUBST(PADLL) -AC_SUBST(SHARED_FLAGS) -AC_SUBST(DLL_LIBS) - -dnl Determine the host operating system / platform -AC_CANONICAL_HOST - -case "${host_os}" in - darwin* ) - dnl Mac OS X configuration - - OTHER_OBJS="pa_mac_core/pa_mac_core.o"; - LIBS="-framework CoreAudio -lm"; - PADLL="libportaudio.dylib"; - SHARED_FLAGS="-framework CoreAudio -dynamiclib"; - ;; - - mingw* ) - dnl MingW configuration - - OTHER_OBJS="pa_win_wmme/pa_win_wmme.o"; - LIBS="-lwinmm -lm"; - PADLL="portaudio.dll"; - SHARED_FLAGS="-shared -mthreads"; - DLL_LIBS="-lwinmm"; - ;; - - cygwin* ) - dnl Cygwin configuration - - OTHER_OBJS="pa_win_wmme/pa_win_wmme.o"; - LIBS="-lwinmm -lm"; - PADLL="portaudio.dll"; - SHARED_FLAGS="-shared -mthreads"; - DLL_LIBS="-lwinmm"; - ;; - - *) - dnl Unix OSS configuration - - AC_CHECK_LIB(pthread, pthread_create, - , - AC_MSG_ERROR([libpthread not found!])) - - OTHER_OBJS="pa_unix_oss/pa_unix_oss.o pa_unix_oss/pa_unix.o"; - LIBS="-lm -lpthread"; - PADLL="libportaudio.so"; - SHARED_FLAGS="-shared"; -esac - -AC_OUTPUT([Makefile]) - -echo "" -echo "Finished configure." - -echo "" -echo "Type 'make' to build PortAudio and examples." diff --git a/pd/portaudio_v18/docs/index.html b/pd/portaudio_v18/docs/index.html deleted file mode 100644 index f92cd110..00000000 --- a/pd/portaudio_v18/docs/index.html +++ /dev/null @@ -1,94 +0,0 @@ -<!doctype html public "-//w3c//dtd html 4.0 transitional//en"> -<html> -<head> - <meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1"> - <meta name="GENERATOR" content="Mozilla/4.79 [en] (Windows NT 5.0; U) [Netscape]"> - <meta name="Author" content="Phil Burk"> - <meta name="Description" content="PortAudio Docs, a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function."> - <meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,"> - <title>PortAudio Docs</title> -</head> -<body> - -<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" > -<tr> -<td> -<center> -<h1> -PortAudio Documentation</h1></center> -</td> -</tr> -</table></center> - -<p>Copyright 2000 Phil Burk and Ross Bencina -<br> -<hr WIDTH="100%"> -<h2> -V18</h2> - -<h3> -<a href="portaudio_h.txt">API Reference for V18</a></h3> - -<blockquote>The Application Programmer Interface is documented in "portaudio.h".</blockquote> - -<h3> -<a href="pa_tutorial.html">Tutorial</a></h3> - -<blockquote>Describes how to write audio programs using the PortAudio API.</blockquote> - -<h3> -<a href="pa_impl_guide.html">Implementation Guide</a></h3> - -<blockquote>Describes how to write an implementation of PortAudio for a -new computer platform.</blockquote> - -<h3> -<a href="portaudio_icmc2001.pdf">Paper Presented at ICMC2001</a> (PDF)</h3> - -<blockquote>Describes the PortAudio API and discusses implementation issues. -Written July 2001.</blockquote> - -<hr WIDTH="100%"> -<h2> -V19 - improved API</h2> - -<h3> -<a href="proposals/index.html">Proposed V19 Changes</a></h3> - -<blockquote>Describes API changes being considered by the developer community. -Feedback welcome.</blockquote> - -<h3> -<a href="v19-doxydocs/">API Reference for V19</a></h3> - -<blockquote> - <p>Reference documents for the Application Programmer Interface - for V19 generated by doxygen.</p> -</blockquote> - <h3> Paper presented at ACMC2003</h3> - <blockquote> - <p>"PortAudio and Media Synchronisation" - Discusses how the V19 API relates to - various synchronisation tasks including MIDI controlled playback, GUI/Audio - sync and MIDI sync. <a href="http://www.portaudio.com/docs/portaudio_sync_acmc2003.pdf">http://www.portaudio.com/docs/portaudio_sync_acmc2003.pdf </a> (PDF)</p> -</blockquote> - <h3>Programmer's Guides</h3> - -<ul> -<li> -<a href="http://www.numerix-dsp.com/files/portaudio.html" target="_blank"> -Compilation Information For Borland C/C++ And wxWindows by John Edwards -</li> -</ul> - -<hr WIDTH="100%"> -<h2> -Miscellaneous</h2> - -<h3> -<a href="latency.html">Improving Latency</a></h3> - -<blockquote>How to tune your computer to achieve the lowest possible audio -delay.</blockquote> -<a href="http://www.portaudio.com/">Return to PortAudio Home Page</a> -</body> -</html> diff --git a/pd/portaudio_v18/docs/latency.html b/pd/portaudio_v18/docs/latency.html deleted file mode 100644 index 87f1d122..00000000 --- a/pd/portaudio_v18/docs/latency.html +++ /dev/null @@ -1,192 +0,0 @@ -<!doctype html public "-//w3c//dtd html 4.0 transitional//en"> -<html> -<head> - <meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1"> - <meta name="GENERATOR" content="Mozilla/4.79 [en] (Windows NT 5.0; U) [Netscape]"> - <meta name="Author" content="Phil Burk"> - <meta name="Description" content="Internal docs. How a stream is started or stopped."> - <meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,"> - <title>PortAudio Implementation - Start/Stop</title> -</head> -<body> - -<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" > -<tr> -<td> -<center> -<h1> -<a href="http://www.portaudio.com">PortAudio</a> Latency</h1></center> -</td> -</tr> -</table></center> - -<p>This page discusses the issues of audio latency for <a href="http://www.portaudio.com">PortAudio</a> -. It offers suggestions on how to lower latency to improve the responsiveness -of applications. -<blockquote><b><a href="#what">What is Latency?</a></b> -<br><b><a href="#portaudio">PortAudio and Latency</a></b> -<br><b><a href="#macintosh">Macintosh</a></b> -<br><b><a href="#unix">Unix</a></b> -<br><b><a href="#windows">WIndows</a></b></blockquote> -By Phil Burk, Copyright 2002 Phil Burk and Ross Bencina -<h2> -<a NAME="what"></a>What is Latency?</h2> -Latency is basically longest time that you have to wait before you obtain -a desired result. For digital audio output it is the time between making -a sound in software and finally hearing it. -<p>Consider the example of pressing a key on the ASCII keyboard to play -a note. There are several stages in this process which each contribute -their own latency. First the operating system must respond to the keypress. -Then the audio signal generated must work its way through the PortAudio -buffers. Then it must work its way through the audio card hardware. Then -it must go through the audio amplifier which is very quick and then travel -through the air. Sound travels at abous one foot per millisecond through -air so placing speakers across the room can add 5-20 msec of delay. -<p>The reverse process occurs when recording or responding to audio input. -If you are processing audio, for example if you implement a software guitar -fuzz box, then you have both the audio input and audio output latencies -added together. -<p>The audio buffers are used to prevent glitches in the audio stream. -The user software writes audio into the output buffers. That audio is read -by the low level audio driver or by DMA and sent to the DAC. If the computer -gets busy doing something like reading the disk or redrawing the screen, -then it may not have time to fill the audio buffer. The audio hardware -then runs out of audio data, which causes a glitch. By using a large enough -buffer we can ensure that there is always enough audio data for the audio -hardware to play. But if the buffer is too large then the latency is high -and the system feels sluggish. If you play notes on the keyboard then the -"instrument" will feel unresponsive. So you want the buffers to be as small -as possible without glitching. -<h2> -<a NAME="portaudio"></a>PortAudio and Latency</h2> -The only delay that PortAudio can control is the total length of its buffers. -The Pa_OpenStream() call takes two parameters: numBuffers and framesPerBuffer. -The latency is also affected by the sample rate which we will call framesPerSecond. -A frame is a set of samples that occur simultaneously. For a stereo stream, -a frame is two samples. -<p>The latency in milliseconds due to this buffering is: -<blockquote><tt>latency_msec = 1000 * numBuffers * framesPerBuffer / framesPerSecond</tt></blockquote> -This is not the total latency, as we have seen, but it is the part we can -control. -<p>If you call Pa_OpenStream() with numBuffers equal to zero, then PortAudio -will select a conservative number that will prevent audio glitches. If -you still get glitches, then you can pass a larger value for numBuffers -until the glitching stops. if you try to pass a numBuffers value that is -too small, then PortAudio will use its own idea of the minimum value. -<p>PortAudio decides on the minimum number of buffers in a conservative -way based on the frameRate, operating system and other variables. You can -query the value that PortAudio will use by calling: -<blockquote><tt>int Pa_GetMinNumBuffers( int framesPerBuffer, double sampleRate -);</tt></blockquote> -On some systems you can override the PortAudio minimum if you know your -system can handle a lower value. You do this by setting an environment -variable called PA_MIN_LATENCY_MSEC which is read by PortAudio when it -starts up. This is supported on the PortAudio implementations for Windows -MME, Windows DirectSound, and Unix OSS. -<h2> -<a NAME="macintosh"></a>Macintosh</h2> -The best thing you can do to improve latency on Mac OS 8 and 9 is to turn -off Virtual Memory. PortAudio V18 will detect that Virtual Memory is turned -off and use a very low latency. -<p>For Mac OS X the latency is very low because Apple Core Audio is so -well written. You can set the PA_MIN_LATENCY_MSEC variable using: -<blockquote><tt>setenv PA_MIN_LATENCY_MSEC 4</tt></blockquote> - -<h2> -<a NAME="unix"></a>Unix</h2> -PortAudio under Unix currently uses a backgroud thread that reads and writes -to OSS. This gives you decent but not great latency. But if you raise the -priority of the background thread to a very priority then you can get under -10 milliseconds latency. In order to raise your priority you must run the -PortAudio program as root! You must also set PA_MIN_LATENCY_MSEC using -the appropriate command for your shell. -<h2> -<a NAME="windows"></a>Windows</h2> -Latency under Windows is a complex issue because of all the alternative -operating system versions and device drivers. I have seen latency range -from 8 milliseconds to 400 milliseconds. The worst case is when using Windows -NT. Windows 98 is a little better, and Windows XP can be quite good if -properly tuned. -<p>The underlying audio API also makes a lot of difference. If the audio -device has its own DirectSound driver then DirectSound can often provide -better latency than WMME. But if a real DirectSound driver is not available -for your device then it is emulated using WMME and the latency can be very -high. That's where I saw the 400 millisecond latency. The ASIO implementation -is generally very good and will give the lowest latency if available. -<p>You can set the PA_MIN_LATENCY_MSEC variable to 50, for example, by -entering in MS-DOS: -<blockquote><tt>set PA_MIN_LATENCY_MSEC=50</tt></blockquote> -If you enter this in a DOS window then you must run the PortAudio program -from that same window for the variable to have an effect. You can add that -line to your C:\AUTOEXEC.BAT file and reboot if you want it to affect any -PortAudio based program. -<p>For Windows XP, you can set environment variables as follows: -<ol> -<li> -Select "Control Panel" from the "Start Menu".</li> - -<li> -Launch the "System" Control Panel</li> - -<li> -Click on the "Advanced" tab.</li> - -<li> -Click on the "Environment Variables" button.</li> - -<li> -Click "New" button under User Variables.</li> - -<li> -Enter PA_MIN_LATENCY_MSEC for the name and some optimistic number for the -value.</li> - -<li> -Click OK, OK, OK.</li> -</ol> - -<h3> -Improving Latency on Windows</h3> -There are several steps you can take to improve latency under windows. -<ol> -<li> -Avoid reading or writng to disk when doing audio.</li> - -<li> -Turn off all automated background tasks such as email clients, virus scanners, -backup programs, FTP servers, web servers, etc. when doing audio.</li> - -<li> -Disconnect from the network to prevent network traffic from interrupting -your CPU.</li> -</ol> -<b>Important: </b>Windows XP users can also tune the OS to favor background -tasks, such as audio, over foreground tasks, such as word processing. I -lowered my latency from 40 to 10 milliseconds using this simple technique. -<ol> -<li> -Select "Control Panel" from the "Start Menu".</li> - -<li> -Launch the "System" Control Panel</li> - -<li> -Click on the "Advanced" tab.</li> - -<li> -Click on the "Settings" button in the Performance area.</li> - -<li> -Click on the "Advanced" tab.</li> - -<li> -Select "Background services" in the Processor Scheduling area.</li> - -<li> -Click OK, OK.</li> -</ol> -Please let us know if you have others sugestions for lowering latency. -<br> -<br> -</body> -</html> diff --git a/pd/portaudio_v18/docs/pa_drivermodel.c.txt b/pd/portaudio_v18/docs/pa_drivermodel.c.txt deleted file mode 100644 index 956f664c..00000000 --- a/pd/portaudio_v18/docs/pa_drivermodel.c.txt +++ /dev/null @@ -1,488 +0,0 @@ -/* - This file contains the host-neutral code for implementing multiple driver model - support in PortAudio. - - It has not been compiled, but it is supplied only for example purposes at this stage. - - TODO: use of CHECK_DRIVER_MODEL is bogus in some instances since some - of those functions don't return a PaError - - -*/ - -#include "pa_drivermodel.h.txt" - - -#ifndef PA_MULTIDRIVER -/* single driver support, most functions will stay in the implementation files */ - -PaDriverModelID Pa_CountDriverModels() -{ - return 1; -} - -/* -Perhaps all drivers should define this with this signature -const PaDriverModelInfo* Pa_GetDriverModelInfo( PaDriverModelID driverModelID ) -{ -} -*/ - -PaDeviceID Pa_DriverModelDefaultInputDeviceID( PaDriverModelID driverModelID ) -{ - return Pa_GetDefaultInputDeviceID(); -} - - -PaDeviceID Pa_DriverModelDefaultOutputDeviceID( PaDriverModelID driverModelID ) -{ - return Pa_GetDefaultInputDeviceID(); -} - -/* -Perhaps all drivers should define with this signature -int Pa_DriverModelMinNumBuffers( PaDriverModelID driverModelID, int framesPerBuffer, double sampleRate ) -{ - -} -*/ - -int Pa_DriverModelCountDevices( PaDriverModelID driverModelID ) -{ - return Pa_CountDevices(); -} - -PaDeviceID Pa_DriverModelGetDeviceID(PaDriverModelID driverModelID, int perDriverModelIndex ) -{ - return perDriverModelIndex; -} - - -#else -/* multidriver support */ - - -typedef PaError (*PaInitializeFunPtr)( PaDriverModelImplementation** ); - -/* - the initializers array is a static table of function pointers - to all the available driverModels on the current platform. - - the order of pointers in the array is important. the first - pointer is always considered to be the "default" driver model. -*/ - -static PaInitializeFunPtr driverModelInitializers[] = { -#ifdef WINDOWS - PaWin32WMME_MultiDriverInitialize, - PaWin32DS_MultiDriverInitialize, - PaASIO_MultiDriverInitialize -#endif -#ifdef MAC - PaMacSM_MultiDriverInitialize, - PaMacCA_MultiDriverInitialize, - PaASIO_MultiDriverInitialize -#endif -/* other platforms here */ - (PaInitializeFunPtr*)0 /* NULL terminate the list */ -}; - - -/* - the driverModels array is a dynamically created table of - currently available driverModels. -*/ -static PaDriverModelImplementation* driverModels = 0; -static int numDriverModels = 0; - - -#define PA_CHECK_INITIALIZED\ - if( driverModels == 0 ) - return paLibraryNotInitialised - -#define PA_CHECK_DRIVER_MODEL_ID( id ) - if( id < 0 || id >= numDriverModels ) - return paBadDriverModelID; - - -/* - ConvertPublicDeviceIdToImplementationDeviceId converts deviceId - from a public device id, to a device id for a particular - PortAudio implementation. On return impl will either point - to a valid implementation or will be NULL. -*/ -static void ConvertPublicDeviceIDToImplementationDeviceID( - PaDriverModelImplementation *impl, PaDeviceID deviceID ) -{ - int i, count; - - impl = NULL; - - for( i=0; i < numDriverModels; ++i ){ - count = driverModels[i]->countDevices(); - if( deviceID < count ){ - impl = driverModels[i]; - return NULL; - }else{ - deviceID -= count; - } - } -} - -static PaDeviceID ConvertImplementationDeviceIDToPublicDeviceID( - PaDriverModelID driverModelID, PaDeviceID deviceID ) -{ - int i; - - for( i=0; i < driverModelID; ++i ) - deviceID += driverModels[i]->countDevices(); -} - - -PaError Pa_Initialize( void ) -{ - PaError result = paNoError; - int i, initializerCount; - PaDriverModelImplementation *impl; - - if( driverModels != 0 ) - return paAlreadyInitialized; - - /* count the number of driverModels */ - initializerCount=0; - while( driverModelInitializers[initializerCount] != 0 ){ - ++initializerCount; - } - - driverModels = malloc( sizeof(PaDriverModelImplementation*) * initializerCount ); - if( driverModels == NULL ) - return paInsufficientMemory; - - numDriverModels = 0; - for( i=0; i<initializerCount; ++i ){ - result = (*driverModelInitializers[i])( &impl ); - if( result == paNoError ){ - driverModels[numDriverModels] = impl; - ++numDriverModels; - }else{ - // TODO: terminate the drivers which have already been initialized. - } - } - - return result; -} - - - -PaError Pa_Terminate( void ) -{ - int i; - - PA_CHECK_INITIALIZED; - - /* - rather than require each implementation to do it separately we could - keep track of all open streams and close them here - */ - - for( i=0; i<numDriverModels; ++i ) - driverModels[i]->terminate( driverModels[i] ); -} - - -long Pa_GetHostError( void ) -{ - PA_CHECK_INITIALIZED; - - under construction. depends on error text proposal. -} - - -const char *Pa_GetErrorText( PaError errnum ) -{ - PA_CHECK_INITIALIZED; - - under construction. may need to call driver model specific code - depending on how the error text proposal pans out. -} - - - -int Pa_CountDevices() -{ - int i, result; - - PA_CHECK_INITIALIZED; - - result = 0; - for( i=0; i < numDriverModels; ++i ) - result += driverModels[i]->countDevices(); - - return result; -} - - -PaDeviceID Pa_GetDefaultInputDeviceID( void ) -{ - PA_CHECK_INITIALIZED; - - return driverModels[0]->getDefaultInputDeviceID(); -} - -PaDeviceID Pa_GetDefaultOutputDeviceID( void ) -{ - PA_CHECK_INITIALIZED; - - return driverModels[0]->getDefaultInputDeviceID(); -} - - -const PaDeviceInfo* Pa_GetDeviceInfo( PaDeviceID deviceID ) -{ - PaDriverModelImplementation *impl; - - PA_CHECK_INITIALIZED; - - ConvertPublicDeviceIDToImplementationDeviceID( impl, deviceID ); - if( impl == NULL ) - return paInvalidDeviceID; - - return impl->getDeviceInfo( deviceID ); -} - -/* NEW MULTIPLE DRIVER MODEL FUNCTIONS ---------------------------------- */ - -PaDriverModelID Pa_CountDriverModels() -{ - PA_CHECK_INITIALIZED; - - return numDriverModels; -} - - -const PaDriverModelInfo* Pa_GetDriverModelInfo( PaDriverModelID driverModelID ) -{ - PA_CHECK_INITIALIZED; - PA_CHECK_DRIVER_MODEL_ID( driverModelID ); - - return driverModels[ driverModelID ]->getDriverModelInfo(); -} - - -PaDeviceID Pa_DriverModelDefaultInputDeviceID( PaDriverModelID driverModelID ) -{ - PA_CHECK_INITIALIZED; - PA_CHECK_DRIVER_MODEL_ID( driverModelID ); - - return ConvertImplementationDeviceIDToPublicDeviceID( driverModelID, - driverModels[ driverModelID ]->getDefaultInputDeviceID(); -} - - -PaDeviceID Pa_DriverModelDefaultOutputDeviceID( PaDriverModelID driverModelID ) -{ - PA_CHECK_INITIALIZED; - PA_CHECK_DRIVER_MODEL_ID( driverModelID ); - - return ConvertImplementationDeviceIDToPublicDeviceID( driverModelID, - driverModels[ driverModelID ]->getDefaultOutputDeviceID(); -} - - -int Pa_DriverModelMinNumBuffers( PaDriverModelID driverModelID, int framesPerBuffer, double sampleRate ) -{ - PA_CHECK_INITIALIZED; - PA_CHECK_DRIVER_MODEL_ID( driverModelID ); - - return driverModels[ driverModelID ]->getMinNumBuffers( int framesPerBuffer, double sampleRate ); -} - - -int Pa_DriverModelCountDevices( PaDriverModelID driverModelID ) -{ - PA_CHECK_INITIALIZED; - PA_CHECK_DRIVER_MODEL_ID( driverModelID ); - - return driverModels[ driverModelID ]->coundDevices(); -} - -PaDeviceID Pa_DriverModelGetDeviceID(PaDriverModelID driverModelID, int perDriverModelIndex ) -{ - PA_CHECK_INITIALIZED; - PA_CHECK_DRIVER_MODEL_ID( driverModelID ); - - return ConvertImplementationDeviceIDToPublicDeviceID( driverModelID, perDriverModelIndex ); -} - -/* END NEW MULTIPLE DRIVER MODEL FUNCTIONS ------------------------------ */ - - -PaError Pa_OpenStream( PortAudioStream** stream, - PaDeviceID inputDevice, - int numInputChannels, - PaSampleFormat inputSampleFormat, - void *inputDriverInfo, - PaDeviceID outputDevice, - int numOutputChannels, - PaSampleFormat outputSampleFormat, - void *outputDriverInfo, - double sampleRate, - unsigned long framesPerBuffer, - unsigned long numberOfBuffers, - PaStreamFlags streamFlags, - PortAudioCallback *callback, - void *userData ) -{ - PaError result; - PaDriverModelImplementation *inputImpl, *outputImpl, impl; - - PA_CHECK_INITIALIZED; - - if( inputDevice != paNoDevice ){ - ConvertPublicDeviceIDToImplementationDeviceID( inputImpl, inputDevice ); - if( inputImpl == NULL ) - return paInvalidDeviceID; - else - impl = inputImpl; - } - - if( outputDevice != paNoDevice ){ - ConvertPublicDeviceIDToImplementationDeviceID( outputImpl, outputDevice ); - if( outputImpl == NULL ) - return paInvalidDeviceID; - else - impl = outputImpl; - } - - if( inputDevice != paNoDevice && outputDevice != paNoDevice ){ - if( inputImpl != outputImpl ) - return paDevicesMustBelongToTheSameDriverModel; - } - - - result = impl->openStream( stream, inputDevice, numInputChannels, inputSampleFormat, inputDriverInfo, - outputDevice, numOutputChannels, outputSampleFormat, outputDriverInfo, - sampleRate, framesPerBuffer, numberOfBuffers, streamFlags, callback, userData ); - - - if( result == paNoError ) - ((PaStreamImplementation*)stream)->magic = PA_STREAM_MAGIC; - - return result; -} - - -PaError Pa_OpenDefaultStream( PortAudioStream** stream, - int numInputChannels, - int numOutputChannels, - PaSampleFormat sampleFormat, - double sampleRate, - unsigned long framesPerBuffer, - unsigned long numberOfBuffers, - PortAudioCallback *callback, - void *userData ) -{ - PaError result; - int inputDevice = driverModels[0]->getDefaultInputDeviceID; - int outputDevice = driverModels[0]->getDefaultOutputDeviceID; - - result = driverModels[0]->openStream( stream, inputDevice, numInputChannels, sampleFormat, 0, - outputDevice, numOutputChannels, sampleFormat, 0, - sampleRate, framesPerBuffer, numberOfBuffers, - streamFlags, callback, userData ); - - if( result == paNoError ) - ((PaStreamImplementation*)stream)->magic = PA_STREAM_MAGIC; - - return result; -} - - -PaError Pa_CloseStream( PortAudioStream* stream ) -{ - PA_CHECK_INITIALIZED; - - PaError result = ((PaStreamImplementation*)stream)->close(); - - if( result == PaNoError ) - ((PaStreamImplementation*)stream)->magic = 0; /* clear magic number */ - - return result; -} - - -PaError Pa_StartStream( PortAudioStream *stream ); -{ - PA_CHECK_INITIALIZED; - - return ((PaStreamImplementation*)stream)->start(); -} - - -PaError Pa_StopStream( PortAudioStream *stream ); -{ - PA_CHECK_INITIALIZED; - - return ((PaStreamImplementation*)stream)->stop(); -} - - -PaError Pa_AbortStream( PortAudioStream *stream ); -{ - PA_CHECK_INITIALIZED; - - return ((PaStreamImplementation*)stream)->abort(); -} - - -PaError Pa_StreamActive( PortAudioStream *stream ) -{ - PA_CHECK_INITIALIZED; - - return ((PaStreamImplementation*)stream)->active(); -} - -PaTimestamp Pa_StreamTime( PortAudioStream *stream ) -{ - PA_CHECK_INITIALIZED; - - return ((PaStreamImplementation*)stream)->time(); -} - - -double Pa_StreamCPULoad( PortAudioStream* stream ) -{ - PA_CHECK_INITIALIZED; - - return ((PaStreamImplementation*)stream)->cpuLoad(); -} - - - -int Pa_GetMinNumBuffers( PaDeviceID deviceID, int framesPerBuffer, double sampleRate ) -{ - PaDriverModelImplementation *impl; - PA_CHECK_INITIALIZED; - - ConvertPublicDeviceIDToImplementationDeviceID( impl, deviceID ); - if( impl == NULL ) - return paInvalidDeviceID; - - return impl->getMinNumBuffers( framesPerBuffer, sampleRate ); -} - - -void Pa_Sleep( long msec ) -{ - same as existing implementaion -} - - -PaError Pa_GetSampleSize( PaSampleFormat format ) -{ - same as existing implementation -} - -#endif /* PA_MULTIDRIVER */ - - diff --git a/pd/portaudio_v18/docs/pa_drivermodel.h.txt b/pd/portaudio_v18/docs/pa_drivermodel.h.txt deleted file mode 100644 index f3f2ca2a..00000000 --- a/pd/portaudio_v18/docs/pa_drivermodel.h.txt +++ /dev/null @@ -1,143 +0,0 @@ -#ifndef PA_MULTIDRIVERMODEL_H -#define PA_MULTIDRIVERMODEL_H - -#ifdef __cplusplus -extern "C" { -#endif /* __cplusplus */ - - -/* - This file contains the host-neutral code for implementing multiple driver model - support in PortAudio. - - It has not been compiled, but it is supplied only for example purposes at this stage. -*/ - - -#include "portaudio.h" - - -#define PA_MULTIDRIVER // for multidriver support - - - -TODO: declare function pointer types for the following function pointers - -/* - Each driver model implementation needs to implement an initialize function - which is added to the driverModelInitializers array in pa_multidrivermodel.c - - the initializer function needs to return a pointer to a - PaDriverModelImplementation structure, or NULL if initiliazation failed. TODO: need error code instead - - the function pointer members of this structure point to funtions - which operate in exactly the same way as the corresponding functions - in the PortAudio API. -*/ - -struct{ - fptr terminate; /* takes the PaDriverModelImplementation* returned by initialize */ - fptr getDriverModelInfo; - fptr getHostError; - fptr getHostErrorText; - fptr countDevices; - fptr getDefaultInputDeviceID; - fptr getDefaultOutputDeviceID; - fptr getDeviceInfo; - fptr openStream; - fptr getMinNumBuffers; -} PaDriverModelImplementation; - -/* - whenever an implementaion's openstream method is called it should return a - PortAudioStream* whose first segment is actually the following structure. - - the functions pointer members of this structure point to funcitons - which operate in exactly the same way as the corresponding functions - in the PortAudio API. -*/ -struct{ - unsigned long magic; - fptr close; - fptr start; - fptr stop; - fptr abort; - fptr active; - fptr time; - fptr cpuLoad; -} PaStreamImplementation; - -/* - Implementations should set magic to PA_STREAM_MAGIC when opening - a stream _and_ clear it to zero when closing a stream. - All functions which operate on streams should check the validity - of magic. -*/ - -#define PA_STREAM_MAGIC 0x12345678 - -#define PA_CHECK_STREAM( stream )\ - if( ((PaStreamImplementation*)stream)->magic != PA_STREAM_MAGIC )\ - return paBadStreamPtr; - - -/* - PA_API allows the same implementation to be used for single - driver model and multi-driver model operation. If - PA_MULTIDRIVER not defined, PA_API will declare api - functions as global, otherwise they will be static, and include - the drivermodel code. - - Usage would be something like: - - int PA_API(CountDevices)(); - - The PA_MULTIDRIVER_SUPPORT macro declares the initialization and - termination functions required by the multidriver support. it also - allocates and deallocates the PaDriverModelImplementation structure. - - TODO: add macros for initializing PaStreamImplementation PortAudioStream - these would be PA_INITIALIZE_STREAM and PA_TERMINATE_STREAM - they would assign and clear the magic number and assign the - interface functions if neceassary. -*/ - -#ifdef PA_MULTIDRIVER - -#define PA_API( model, name ) static Pa ## model ## _ ## name - -#define PA_MULTIDRIVER_SUPPORT( model )\ -PaError Pa_ ## model ## _MultiDriverTerminate( PaStreamImplementation *impl )\ -{\ - free( impl );\ - return Pa ## model ## _Terminate();\ -}\ -PaError Pa ## model ## _MultiDriverInitialize( PaStreamImplementation** impl )\ -{\ - PaError result = Pa ## model ## _Initialize();\ -\ - if( result == paNoError ){\ - *impl = malloc( sizeof( PaDriverModelImplementation ) );\ - if( impl == NULL ){\ - // TODO: call terminate, return an error - }else{\ - (*impl)->terminate = Pa ## model ## _MultiDriverTerminate();\ - (*impl)->getDriverModelInfo = Pa ## model ## _GetDriverModelInfo();\ - (*impl)->getHostError = Pa ## model ## _GetHostError();\ - // TODO: assign the rest of the interface functions - }\ - }\ - return result;\ -} - -#else /* !PA_MULTIDRIVER */ - -#define PA_API( model, name ) Pa_ ## name - -#define PA_MULTIDRIVER_SUPPORT - -#endif /* PA_MULTIDRIVER */ - - - -#endif /* PA_MULTIDRIVERMODEL_H */ diff --git a/pd/portaudio_v18/docs/pa_impl_guide.html b/pd/portaudio_v18/docs/pa_impl_guide.html deleted file mode 100644 index 50abc304..00000000 --- a/pd/portaudio_v18/docs/pa_impl_guide.html +++ /dev/null @@ -1,197 +0,0 @@ -<!doctype html public "-//w3c//dtd html 4.0 transitional//en"> -<html> -<head> - <meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1"> - <meta name="GENERATOR" content="Mozilla/4.79 [en] (Windows NT 5.0; U) [Netscape]"> - <meta name="Author" content="Phil Burk"> - <meta name="Description" content="Internal docs. How a stream is started or stopped."> - <meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,"> - <title>PortAudio Implementation - Start/Stop</title> -</head> -<body> - -<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" > -<tr> -<td> -<center> -<h1> -<a href="http://www.portaudio.com">PortAudio</a> Implementation Guide</h1></center> -</td> -</tr> -</table></center> - -<p>This document describes how to implement the PortAudio API on a new -computer platform. Implementing PortAudio on a new platform, makes it possible -to port many existing audio applications to that platform. -<p>By Phil Burk -<br>Copyright 2000 Phil Burk and Ross Bencina -<p>Note that the license says: <b>"Any person wishing to distribute modifications -to the Software is requested to send the modifications to the original -developer so that they can be incorporated into the canonical version."</b>. -So when you have finished a new implementation, please send it back to -us at "<a href="http://www.portaudio.com">http://www.portaudio.com</a>" -so that we can make it available for other users. Thank you! -<h2> -Download the Latest PortAudio Implementation</h2> -Always start with the latest implementation available at "<a href="http://www.portaudio.com">http://www.portaudio.com</a>". -Look for the nightly snapshot under the CVS section. -<h2> -Select an Existing Implementation as a Basis</h2> -The fastest way to get started is to take an existing implementation and -translate it for your new platform. Choose an implementation whose architecture -is as close as possible to your target. -<ul> -<li> -DirectSound Implementation - pa_win_ds - Uses a timer callback for the -background "thread". Polls a circular buffer and writes blocks of data -to keep it full.</li> - -<li> -Windows MME - pa_win_wmme - Spawns an actual Win32 thread. Writes blocks -of data to the HW device and waits for events that signal buffer completion.</li> - -<li> -Linux OSS - pa_linux - Spawns a real thread that writes to the "/dev/dsp" -stream using blocking I/O calls.</li> -</ul> -When you write a new implementation, you will be using some code that is -in common with all implementations. This code is in the folder "pa_common". -It provides various functions such as parameter checking, error code to -text conversion, sample format conversion, clipping and dithering, etc. -<p>The code that you write will go into a separate folder called "pa_{os}_{api}". -For example, code specific to the DirectSound interface for Windows goes -in "pa_win_ds". -<h2> -Read Docs and Code</h2> -Famialiarize yourself with the system by reading the documentation provided. -here is a suggested order: -<ol> -<li> -User Programming <a href="pa_tutorial.html">Tutorial</a></li> - -<li> -Header file "pa_common/portaudio.h" which defines API.</li> - -<li> -Header file "pa_common/pa_host.h" for host dependant code. This definces -the routine you will need to provide.</li> - -<li> -Shared code in "pa_common/pa_lib.c".</li> - -<li> -Docs on Implementation of <a href="pa_impl_startstop.html">Start/Stop</a> -code.</li> -</ol> - -<h2> -Implement Output to Default Device</h2> -Now we are ready to crank some code. For instant gratification, let's try -to play a sine wave. -<ol> -<li> -Link the test program "pa_tests/patest_sine.c" with the file "pa_lib.c" -and the implementation specific file you are creating.</li> - -<li> -For now, just stub out the device query code and the audio input code.</li> - -<li> -Modify PaHost_OpenStream() to open your default target device and get everything -setup.</li> - -<li> -Modify PaHost_StartOutput() to start playing audio.</li> - -<li> -Modify PaHost_StopOutput() to stop audio.</li> - -<li> -Modify PaHost_CloseStream() to clean up. Free all memory that you allocated -in PaHost_OpenStream().</li> - -<li> -Keep cranking until you can play a sine wave using "patest_sine.c".</li> - -<li> -Once that works, try "patest_pink.c", "patest_clip.c", "patest_sine8.c".</li> - -<li> -To test your Open and Close code, try "patest_many.c".</li> - -<li> -Now test to make sure that the three modes of stopping are properly supported -by running "patest_stop.c".</li> - -<li> -Test your implementation of time stamping with "patest_sync.c".</li> -</ol> - -<h2> -Implement Device Queries</h2> -Now that output is working, lets implement the code for querying what devices -are available to the user. Run "pa_tests/pa_devs.c". It should print all -of the devices available and their characteristics. -<h2> -Implement Input</h2> -Implement audio input and test it with: -<ol> -<li> -patest_record.c - record in half duplex, play back as recorded.</li> - -<li> -patest_wire.c - full duplex, copies input to output. Note that some HW -may not support full duplex.</li> - -<li> -patest_fuzz.c - plug in your guitar and get a feel for why latency is an -important issue in computer music.</li> - -<li> -paqa_devs.c - try to open every device and use it with every possible format</li> -</ol> - -<h2> -Debugging Tools</h2> -You generally cannot use printf() calls to debug real-time processes because -they disturb the timing. Also calling printf() from your background thread -or interrupt could crash the machine. So PA includes a tool for capturing -events and storing the information while it is running. It then prints -the events when Pa_Terminate() is called. -<ol> -<li> -To enable trace mode, change TRACE_REALTIME_EVENTS in "pa_common/pa_trace.h" -from a (0) to a (1).</li> - -<li> -Link with "pa_common/pa_trace.c".</li> - -<li> -Add trace messages to your code by calling:</li> - -<br><tt> void AddTraceMessage( char *msg, int data );</tt> -<br><tt>for example</tt> -<br><tt> AddTraceMessage("Pa_TimeSlice: past_NumCallbacks ", -past->past_NumCallbacks );</tt> -<li> -Run your program. You will get a dump of events at the end.</li> - -<li> -You can leave the trace messages in your code. They will turn to NOOPs -when you change TRACE_REALTIME_EVENTS back to (0).</li> -</ol> - -<h2> -Delivery</h2> -Please send your new code along with notes on the implementation back to -us at "<a href="http://www.portaudio.com">http://www.portaudio.com</a>". -We will review the implementation and post it with your name. If you had -to make any modifications to the code in "pa_common" or "pa_tests" <b>please</b> -send us those modifications and your notes. We will try to merge your changes -so that the "pa_common" code works with <b>all</b> implementations. -<p>If you have suggestions for how to make future implementations easier, -please let us know. -<br>THANKS! -<br> -</body> -</html> diff --git a/pd/portaudio_v18/docs/pa_impl_startstop.html b/pd/portaudio_v18/docs/pa_impl_startstop.html deleted file mode 100644 index 0f2d0ce5..00000000 --- a/pd/portaudio_v18/docs/pa_impl_startstop.html +++ /dev/null @@ -1,190 +0,0 @@ -<!doctype html public "-//w3c//dtd html 4.0 transitional//en"> -<html> -<head> - <meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1"> - <meta name="GENERATOR" content="Mozilla/4.75 [en]C-gatewaynet (Win98; U) [Netscape]"> - <meta name="Author" content="Phil Burk"> - <meta name="Description" content="Internal docs. How a stream is started or stopped."> - <meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,"> - <title>PortAudio Implementation - Start/Stop</title> -</head> -<body> - -<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" > -<tr> -<td> -<center> -<h1> -PortAudio Implementation</h1></center> -</td> -</tr> -</table></center> - -<h2> -Starting and Stopping Streams</h2> -PortAudio is generally executed in two "threads". The foreground thread -is the application thread. The background "thread" may be implemented as -an actual thread, an interrupt handler, or a callback from a timer thread. -<p>There are three ways that PortAudio can stop a stream. In each case -we look at the sequence of events and the messages sent between the two -threads. The following variables are contained in the internalPortAudioStream. -<blockquote><tt>int past_IsActive; -/* Background is still playing. */</tt> -<br><tt>int past_StopSoon; /* Stop -when last buffer done. */</tt> -<br><tt>int past_StopNow; /* -Stop IMMEDIATELY. */</tt></blockquote> - -<h3> -Pa_AbortStream()</h3> -This function causes the background thread to terminate as soon as possible -and audio I/O to stop abruptly. -<br> -<table BORDER COLS=2 WIDTH="60%" > -<tr> -<td><b>Foreground Thread</b></td> - -<td><b>Background Thread</b></td> -</tr> - -<tr> -<td>sets <tt>StopNow</tt></td> - -<td></td> -</tr> - -<tr> -<td></td> - -<td>sees <tt>StopNow</tt>, </td> -</tr> - -<tr> -<td></td> - -<td>clears IsActive, stops thread</td> -</tr> - -<tr> -<td>waits for thread to exit</td> - -<td></td> -</tr> - -<tr> -<td>turns off audio I/O</td> - -<td></td> -</tr> -</table> - -<h3> -Pa_StopStream()</h3> -This function stops the user callback function from being called and then -waits for all audio data written to the output buffer to be played. In -a system with very low latency, you may not hear any difference between -<br> -<table BORDER COLS=2 WIDTH="60%" > -<tr> -<td><b>Foreground Thread</b></td> - -<td><b>Background Thread</b></td> -</tr> - -<tr> -<td>sets StopSoon</td> - -<td></td> -</tr> - -<tr> -<td></td> - -<td>stops calling user callback</td> -</tr> - -<tr> -<td></td> - -<td>continues until output buffer empty</td> -</tr> - -<tr> -<td></td> - -<td>clears IsActive, stops thread</td> -</tr> - -<tr> -<td>waits for thread to exit</td> - -<td></td> -</tr> - -<tr> -<td>turns off audio I/O</td> - -<td></td> -</tr> -</table> - -<h3> -User callback returns one.</h3> -If the user callback returns one then the user callback function will no -longer be called. Audio output will continue until all audio data written -to the output buffer has been played. Then the audio I/O is stopped, the -background thread terminates, and the stream becomes inactive. -<br> -<table BORDER COLS=2 WIDTH="60%" > -<tr> -<td><b>Foreground Thread</b></td> - -<td><b>Background Thread</b></td> -</tr> - -<tr> -<td></td> - -<td>callback returns 1</td> -</tr> - -<tr> -<td></td> - -<td>sets StopSoon</td> -</tr> - -<tr> -<td></td> - -<td>stops calling user callback</td> -</tr> - -<tr> -<td></td> - -<td>continues until output buffer empty</td> -</tr> - -<tr> -<td></td> - -<td>clears IsActive, stops thread</td> -</tr> - -<tr> -<td>waits for thread to exit</td> - -<td></td> -</tr> - -<tr> -<td>turns off audio I/O</td> - -<td></td> -</tr> -</table> - -<br> -</body> -</html> diff --git a/pd/portaudio_v18/docs/pa_tut_asio.html b/pd/portaudio_v18/docs/pa_tut_asio.html deleted file mode 100644 index a7fa7d3a..00000000 --- a/pd/portaudio_v18/docs/pa_tut_asio.html +++ /dev/null @@ -1,108 +0,0 @@ -<!doctype html public "-//w3c//dtd html 4.0 transitional//en"> -<html> -<head> - <meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1"> - <meta name="GENERATOR" content="Mozilla/4.79 [en] (Windows NT 5.0; U) [Netscape]"> - <meta name="Author" content="Phil Burk"> - <meta name="Description" content="Tutorial for PortAudio, a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function."> - <meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,"> - <title>PortAudio Tutorial</title> -</head> -<body> - -<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" > -<tr> -<td> -<center> -<h1> -PortAudio Tutorial</h1></center> -</td> -</tr> -</table></center> - -<h2> -<font size=+2><a href="http://www.portaudio.com/">home</a> | -<a href="pa_tutorial.html">contents</a> -| <a href="pa_tut_over.html">previous</a> | <a href="pa_tut_callback.html">next</a></font></h2> - -<h2> -Compiling for ASIO (Windows or Macintosh)</h2> - -<blockquote>ASIO is a low latency audio API from Steinberg. To compile -an ASIO application, you must first <a href="http://www.steinberg.net/en/ps/support/3rdparty/">download -the ASIO SDK</a> from Steinberg. You also need to obtain ASIO drivers from -the manufacturer of your audio hardware. -<p>Note: I am using '/' as a file separator below. On Macintosh replace -'/' with ':'. On Windows, replace '/' with '\'. -<p>You may try compiling the "pa_tests/patest_saw.c" file first because -it is the simplest. -<p>Several files are common to all PortAudio implementations. Add the following -source files to your project: -<blockquote>pa_common/pa_lib.c -<br>pa_common/portaudio.h -<br>pa_common/pa_host.h</blockquote> -To use ASIO with the PortAudio library add the following: -<blockquote> -<pre>pa_asio/pa_asio.cpp</pre> -</blockquote> -</blockquote> - -<h3> -Macintosh Specific</h3> - -<blockquote>Note: there is a bug in the <b>Macintosh</b> ASIO code. Mac -users should read the file "pa_asio:readme_asio_sdk_patch.txt" for information -on how to fix the bug. -<p>Add these files from the ASIO SDK downloaded from Steinberg:</blockquote> - -<blockquote> -<blockquote><tt>host/asiodrivers.cpp</tt> -<br><tt>host/mac/asioshlib.cpp</tt> -<br><tt>host/mac/codefragements.cpp</tt></blockquote> -The ASIO drivers should be in a folder called "ASIO Drivers" beneath your -application.</blockquote> - -<h3> -Windows Specific</h3> - -<blockquote>Add these files from the ASIO SDK downloaded from Steinberg:</blockquote> - -<blockquote> -<blockquote><tt>host/asiodrivers.cpp</tt> -<br><tt>host/asiolist.cpp</tt> -<br><tt>common/asio.cpp</tt></blockquote> -</blockquote> - -<blockquote>Add these directories to the path for include files:</blockquote> - -<blockquote> -<blockquote><tt>host</tt> -<br><tt>host/pc</tt> -<br><tt>common</tt></blockquote> -</blockquote> - -<blockquote>and link with the system library "<b>winmm.lib</b>". For MS -Visual C++: -<ul> -<li> -select "Settings..." from the "Project" menu,</li> - -<li> -select the project name in the tree on the left,</li> - -<li> -choose "All Configurations" in the popup menu above the tree,</li> - -<li> -select the "Link" tab,</li> - -<li> -enter "winmm.lib", without quotes, as the first item in the "Object/library -modules:" field.</li> -</ul> -</blockquote> -<font size=+2><a href="http://www.portaudio.com/">home</a> | -<a href="pa_tutorial.html">contents</a> -| <a href="pa_tut_over.html">previous</a> | <a href="pa_tut_callback.html">next</a></font> -</body> -</html> diff --git a/pd/portaudio_v18/docs/pa_tut_callback.html b/pd/portaudio_v18/docs/pa_tut_callback.html deleted file mode 100644 index f5ccaf0f..00000000 --- a/pd/portaudio_v18/docs/pa_tut_callback.html +++ /dev/null @@ -1,91 +0,0 @@ -<!doctype html public "-//w3c//dtd html 4.0 transitional//en"> -<html> -<head> - <meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1"> - <meta name="GENERATOR" content="Mozilla/4.77 [en]C-gatewaynet (Win98; U) [Netscape]"> - <meta name="Author" content="Phil Burk"> - <meta name="Description" content="Tutorial for PortAudio, a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function."> - <meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,"> - <title>PortAudio Tutorial</title> -</head> -<body> - -<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" > -<tr> -<td> -<center> -<h1> -PortAudio Tutorial</h1></center> -</td> -</tr> -</table></center> - -<h2> -Writing a Callback Function</h2> - -<blockquote>To write a program using PortAudio, you must include the "portaudio.h" -include file. You may wish to read "<a href="portaudio_h.txt">portaudio.h</a>" -because it contains a complete description of the PortAudio functions and -constants. -<blockquote> -<pre>#include "portaudio.h"</pre> -</blockquote> -The next task is to write your custom callback function. It is a function -that is called by the PortAudio engine whenever it has captured audio data, -or when it needs more audio data for output. -<p>Your callback function is often called by an interrupt, or low level -process so you should not do any complex system activities like allocating -memory, or reading or writing files, or printf(). Just crunch numbers and -generate audio signals. What is safe or not safe will vary from platform -to platform. On the Macintosh, for example, you can only call "interrupt -safe" routines. Also do not call any PortAudio functions in the callback -except for Pa_StreamTime() and Pa_GetCPULoad(). -<p>Your callback function must return an int and accept the exact parameters -specified in this typedef: -<blockquote> -<pre>typedef int (PortAudioCallback)( - void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData );</pre> -</blockquote> -Here is an example callback function from the test file "patests/patest_saw.c". -It calculates a simple left and right sawtooth signal and writes it to -the output buffer. Notice that in this example, the signals are of <tt>float</tt> -data type. The signals must be between -1.0 and +1.0. You can also use -16 bit integers or other formats which are specified during setup. You -can pass a pointer to your data structure through PortAudio which will -appear as <tt>userData</tt>. -<blockquote> -<pre>int patestCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - unsigned int i; -/* Cast data passed through stream to our structure type. */ - paTestData *data = (paTestData*)userData; - float *out = (float*)outputBuffer; - - for( i=0; i<framesPerBuffer; i++ ) - { - /* Stereo channels are interleaved. */ - *out++ = data->left_phase; /* left */ - *out++ = data->right_phase; /* right */ - - /* Generate simple sawtooth phaser that ranges between -1.0 and 1.0. */ - data->left_phase += 0.01f; - /* When signal reaches top, drop back down. */ - if( data->left_phase >= 1.0f ) data->left_phase -= 2.0f; - - /* higher pitch so we can distinguish left and right. */ - data->right_phase += 0.03f; - if( data->right_phase >= 1.0f ) data->right_phase -= 2.0f; - } - return 0; -}</pre> -</blockquote> -</blockquote> -<font size=+2><a href="http://www.portaudio.com/">home</a> | -<a href="pa_tutorial.html">contents</a> -| <a href="pa_tut_over.html">previous</a> | <a href="pa_tut_init.html">next</a></font> -</body> -</html> diff --git a/pd/portaudio_v18/docs/pa_tut_devs.html b/pd/portaudio_v18/docs/pa_tut_devs.html deleted file mode 100644 index 1756992c..00000000 --- a/pd/portaudio_v18/docs/pa_tut_devs.html +++ /dev/null @@ -1,65 +0,0 @@ -<!doctype html public "-//w3c//dtd html 4.0 transitional//en"> -<html> -<head> - <meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1"> - <meta name="GENERATOR" content="Mozilla/4.75 [en]C-gatewaynet (Win98; U) [Netscape]"> - <meta name="Author" content="Phil Burk"> - <meta name="Description" content="Tutorial for PortAudio, a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function."> - <meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,"> - <title>PortAudio Tutorial</title> -</head> -<body> - -<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" > -<tr> -<td> -<center> -<h1> -PortAudio Tutorial</h1></center> -</td> -</tr> -</table></center> - -<h2> -Querying for Available Devices</h2> - -<blockquote>There are often several different audio devices available in -a computer with different capabilities. They can differ in the sample rates -supported, bit widths, etc. PortAudio provides a simple way to query for -the available devices, and then pass the selected device to Pa_OpenStream(). -For an example, see the file "pa_tests/pa_devs.c". -<p>To determine the number of devices: -<blockquote> -<pre>numDevices = Pa_CountDevices();</pre> -</blockquote> -You can then query each device in turn by calling Pa_GetDeviceInfo() with -an index. -<blockquote> -<pre>for( i=0; i<numDevices; i++ ) { - pdi = Pa_GetDeviceInfo( i );</pre> -</blockquote> -It will return a pointer to a <tt>PaDeviceInfo</tt> structure which is -defined as: -<blockquote> -<pre>typedef struct{ - int structVersion; - const char *name; - int maxInputChannels; - int maxOutputChannels; -/* Number of discrete rates, or -1 if range supported. */ - int numSampleRates; -/* Array of supported sample rates, or {min,max} if range supported. */ - const double *sampleRates; - PaSampleFormat nativeSampleFormat; -}PaDeviceInfo;</pre> -</blockquote> -If the device supports a continuous range of sample rates, then numSampleRates -will equal -1, and the sampleRates array will have two values, the minimum -and maximum rate. -<p>The device information is allocated by Pa_Initialize() and freed by -Pa_Terminate() so you do not have to free() the structure returned by Pa_GetDeviceInfo().</blockquote> -<font size=+2><a href="http://www.portaudio.com/">home</a> | -<a href="pa_tutorial.html">contents</a> -| <a href="pa_tut_util.html">previous</a> | <a href="pa_tut_rw.html">next</a></font> -</body> -</html> diff --git a/pd/portaudio_v18/docs/pa_tut_explore.html b/pd/portaudio_v18/docs/pa_tut_explore.html deleted file mode 100644 index 91c08a5b..00000000 --- a/pd/portaudio_v18/docs/pa_tut_explore.html +++ /dev/null @@ -1,42 +0,0 @@ -<!doctype html public "-//w3c//dtd html 4.0 transitional//en"> -<html> -<head> - <meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1"> - <meta name="GENERATOR" content="Mozilla/4.73 [en]C-gatewaynet (Win98; U) [Netscape]"> - <meta name="Author" content="Phil Burk"> - <meta name="Description" content="Tutorial for PortAudio, a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function."> - <meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,"> - <title>PortAudio Tutorial</title> -</head> -<body> - -<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" > -<tr> -<td> -<center> -<h1> -PortAudio Tutorial</h1></center> -</td> -</tr> -</table></center> - -<h2> -Exploring PortAudio</h2> - -<blockquote>Now that you have a good idea of how PortAudio works, you can -try out the test programs. -<ul> -<li> -For an example of playing a sine wave, see "pa_tests/patest_sine.c".</li> - -<li> -For an example of recording and playing back a sound, see "pa_tests/patest_record.c".</li> -</ul> -I also encourage you to examine the source for the PortAudio libraries. -If you have suggestions on ways to improve them, please let us know. if -you want to implement PortAudio on a new platform, please let us know as -well so we can coordinate people's efforts.</blockquote> -<font size=+2><a href="http://www.portaudio.com/">home</a> | <a href="pa_tutorial.html">contents</a> -| <a href="pa_tut_rw.html">previous</a> | next</font> -</body> -</html> diff --git a/pd/portaudio_v18/docs/pa_tut_init.html b/pd/portaudio_v18/docs/pa_tut_init.html deleted file mode 100644 index 91bfa8d9..00000000 --- a/pd/portaudio_v18/docs/pa_tut_init.html +++ /dev/null @@ -1,43 +0,0 @@ -<!doctype html public "-//w3c//dtd html 4.0 transitional//en"> -<html> -<head> - <meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1"> - <meta name="GENERATOR" content="Mozilla/4.73 [en]C-gatewaynet (Win98; U) [Netscape]"> - <meta name="Author" content="Phil Burk"> - <meta name="Description" content="Tutorial for PortAudio, a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function."> - <meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,"> - <title>PortAudio Tutorial</title> -</head> -<body> - -<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" > -<tr> -<td> -<center> -<h1> -PortAudio Tutorial</h1></center> -</td> -</tr> -</table></center> - -<h2> -Initializing PortAudio</h2> - -<blockquote>Before making any other calls to PortAudio, you must call <tt>Pa_Initialize</tt>(). -This will trigger a scan of available devices which can be queried later. -Like most PA functions, it will return a result of type <tt>paError</tt>. -If the result is not <tt>paNoError</tt>, then an error has occurred. -<blockquote> -<pre>err = Pa_Initialize(); -if( err != paNoError ) goto error;</pre> -</blockquote> -You can get a text message that explains the error message by passing it -to -<blockquote> -<pre>printf( "PortAudio error: %s\n", Pa_GetErrorText( err ) );</pre> -</blockquote> -</blockquote> -<font size=+2><a href="http://www.portaudio.com/">home</a> | <a href="pa_tutorial.html">contents</a> -| <a href="pa_tut_callback.html">previous</a> | <a href="pa_tut_open.html">next</a></font> -</body> -</html> diff --git a/pd/portaudio_v18/docs/pa_tut_mac.html b/pd/portaudio_v18/docs/pa_tut_mac.html deleted file mode 100644 index bf3dafd1..00000000 --- a/pd/portaudio_v18/docs/pa_tut_mac.html +++ /dev/null @@ -1,41 +0,0 @@ -<!doctype html public "-//w3c//dtd html 4.0 transitional//en"> -<html> -<head> - <meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1"> - <meta name="GENERATOR" content="Mozilla/4.77 [en]C-gatewaynet (Win98; U) [Netscape]"> - <meta name="Author" content="Phil Burk"> - <meta name="Description" content="Tutorial for PortAudio, a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function."> - <meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,"> - <title>PortAudio Tutorial</title> -</head> -<body> - -<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" > -<tr> -<td> -<center> -<h1> -PortAudio Tutorial</h1></center> -</td> -</tr> -</table></center> - -<h2> -Compiling for Macintosh</h2> - -<blockquote>To compile a Macintosh application with the PortAudio library, -add the following source files to your project: -<blockquote> -<pre>pa_mac:pa_mac.c -pa_common:pa_lib.c -pa_common:portaudio.h -pa_common:pa_host.h</pre> -</blockquote> -Also add the Apple <b>SoundLib</b> to your project. -<p>You may try compiling the "pa_tests:patest_saw.c" file first because -it is the simplest.</blockquote> -<font size=+2><a href="http://www.portaudio.com/">home</a> | -<a href="pa_tutorial.html">contents</a> -| <a href="pa_tut_over.html">previous</a> | <a href="pa_tut_callback.html">next</a></font> -</body> -</html> diff --git a/pd/portaudio_v18/docs/pa_tut_mac_osx.html b/pd/portaudio_v18/docs/pa_tut_mac_osx.html deleted file mode 100644 index bc33f44a..00000000 --- a/pd/portaudio_v18/docs/pa_tut_mac_osx.html +++ /dev/null @@ -1,86 +0,0 @@ -<!doctype html public "-//w3c//dtd html 4.0 transitional//en"> -<html> -<head> - <meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1"> - <meta name="GENERATOR" content="Mozilla/4.79 [en] (Windows NT 5.0; U) [Netscape]"> - <meta name="Author" content="Phil Burk"> - <meta name="Description" content="Tutorial for PortAudio, a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function."> - <meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,"> - <title>PortAudio Tutorial</title> -</head> -<body> - -<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" > -<tr> -<td> -<center> -<h1> -PortAudio Tutorial</h1></center> -</td> -</tr> -</table></center> - -<h2> -<font size=+2><a href="http://www.portaudio.com/">home</a> | -<a href="pa_tutorial.html">contents</a> -| <a href="pa_tut_over.html">previous</a> | <a href="pa_tut_callback.html">next</a></font></h2> - -<h2> -Compiling for Macintosh OS X</h2> - -<blockquote>To compile a Macintosh OS X CoreAudio application with the -PortAudio library you will use the following source files: -<blockquote>pa_mac_core/pa_mac_core.c -<br>pa_common/pa_lib.c -<br>pa_common/portaudio.h -<br>pa_common/pa_host.h -<br>pa_common/pa_convert.c -<br>pa_common/pa_trace.c -<br>pa_common/pa_trace.h -<br>pablio/ringbuffer.c -<br>pablio/ringbuffer.h</blockquote> -</blockquote> - -<h3> -Using Apple Project Builder</h3> - -<blockquote>Create a new ProjectBuilder project. You can use a "Tool" project -to run the PortAudio examples. -<p>Add the source files from above to your Project. -<p>Add both the Apple CoreAudio.framework and the AudioToolbox.framework -to your project by selecting "Add FrameWorks..." from the Project menu. -<p>Compile and run the "pa_tests:patest_saw.c" file first because it is -the simplest.</blockquote> - -<h3> -Or Using Metrowerks CodeWarrior 8</h3> - -<blockquote>by James Van Lommel</blockquote> - -<blockquote>Create a new CodeWarrior project using Mac OS C++ Stationery. -<br>Then choose Mac OS X Mach-O > Standard Console > C++ Console Mach-O. -<p>In the project window, Clear the HelloWorld.cpp file and add the source -files from above to your Project. -<p>Add a test file of your choosing, like -<br> patests /patest_sine8.c -<p>Add the frameworks to the Frameworks tab using Project > Add Files... -<br> CoreAudio -<br> AudioToolbox -<p>(The System framework should already be a part of the project.) -<p>Open the current target's settings, and in Language Settings > C/C++ -Language, uncheck (disable) the "ANSI Strict" setting. (Do this for both -Debug and Release projects, if necessary.) You may also choose to disable -RTTI and exceptions; the most important settings are 'Force C++ compilation' -(enabled) and 'ANSI Strict' (disabled)." -<p>Edit pa_mac_core.c: -<br> On line 1546, cast the PaHost_AllocateFastMemory() result -to a (char *) or you will get a compile error. -<p>Compile and run. (may need to run from a terminal window) -<p>I've successfully built patest_sine8.c this way using the CVS .tar version -of portaudio (date: 2003-04-27). I get 17 warnings during compilation, -all of which deal with unused variables or arguments.</blockquote> -<font size=+2><a href="http://www.portaudio.com/">home</a> | -<a href="pa_tutorial.html">contents</a> -| <a href="pa_tut_over.html">previous</a> | <a href="pa_tut_callback.html">next</a></font> -</body> -</html> diff --git a/pd/portaudio_v18/docs/pa_tut_open.html b/pd/portaudio_v18/docs/pa_tut_open.html deleted file mode 100644 index 12772811..00000000 --- a/pd/portaudio_v18/docs/pa_tut_open.html +++ /dev/null @@ -1,56 +0,0 @@ -<!doctype html public "-//w3c//dtd html 4.0 transitional//en"> -<html> -<head> - <meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1"> - <meta name="GENERATOR" content="Mozilla/4.79 [en] (Windows NT 5.0; U) [Netscape]"> - <meta name="Author" content="Phil Burk"> - <meta name="Description" content="Tutorial for PortAudio, a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function."> - <meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,"> - <title>PortAudio Tutorial</title> -</head> -<body> - -<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" > -<tr> -<td> -<center> -<h1> -PortAudio Tutorial</h1></center> -</td> -</tr> -</table></center> - -<h2> -Opening a Stream using Defaults</h2> - -<blockquote>The next step is to open a stream which is similar to opening -a file. You can specify whether you want audio input and/or output, how -many channels, the data format, sample rate, etc. -<p>First declare a variable to receive the stream pointer: -<blockquote> -<pre>PortAudioStream *stream;</pre> -</blockquote> -There are two calls for opening streams, <tt>Pa_OpenStream</tt>() and <tt>Pa_OpenDefaultStream</tt>(). -P<tt>a_OpenStream()</tt> takes extra parameters which give you more -control. You can normally just use <tt>Pa_OpenDefaultStream</tt>() which -just calls <tt>Pa_OpenStream()</tt> <tt>with</tt> some reasonable default -values. Let's open a stream for stereo output, using floating point -data, at 44100 Hz. -<blockquote> -<pre>err = Pa_OpenDefaultStream( - &stream, /* passes back stream pointer */ - 0, /* no input channels */ - 2, /* stereo output */ - paFloat32, /* 32 bit floating point output */ - 44100, /* sample rate */ - 256, /* frames per buffer */ - 0, /* number of buffers, if zero then use default minimum */ - patestCallback, /* specify our custom callback */ - &data ); /* pass our data through to callback */</pre> -</blockquote> -If you want to use 16 bit integer data, pass <tt>paInt16</tt> instead of -<tt>paFloat32</tt>.</blockquote> -<font size=+2><a href="http://www.portaudio.com/">home</a> | <a href="pa_tutorial.html">contents</a> -| <a href="pa_tut_init.html">previous</a> | <a href="pa_tut_run.html">next</a></font> -</body> -</html> diff --git a/pd/portaudio_v18/docs/pa_tut_oss.html b/pd/portaudio_v18/docs/pa_tut_oss.html deleted file mode 100644 index 1bb76f25..00000000 --- a/pd/portaudio_v18/docs/pa_tut_oss.html +++ /dev/null @@ -1,46 +0,0 @@ -<!doctype html public "-//w3c//dtd html 4.0 transitional//en"> -<html> -<head> - <meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1"> - <meta name="GENERATOR" content="Mozilla/4.77 [en]C-gatewaynet (Win98; U) [Netscape]"> - <meta name="Author" content="Phil Burk"> - <meta name="Description" content="Tutorial for PortAudio, a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function."> - <meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,"> - <title>PortAudio Tutorial</title> -</head> -<body> - -<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" > -<tr> -<td> -<center> -<h1> -PortAudio Tutorial</h1></center> -</td> -</tr> -</table></center> - -<h2> -Compiling for Unix OSS</h2> - -<blockquote>[Skip this page if you are not using Unix and OSS] -<p>We currently support the <a href="http://www.opensound.com/">OSS</a> -audio drivers for Linux, Solaris, and FreeBSD. We hope to someday support -the newer ALSA drivers. -<ol> -<li> -cd to pa_unix_oss directory</li> - -<li> -Edit the Makefile and uncomment one of the tests. You may try compiling -the "patest_sine.c" file first because it is very simple.</li> - -<li> -gmake run</li> -</ol> -</blockquote> -<font size=+2><a href="http://www.portaudio.com/">home</a> | -<a href="pa_tutorial.html">contents</a> -| <a href="pa_tut_pc.html">previous</a> | <a href="pa_tut_callback.html">next</a></font> -</body> -</html> diff --git a/pd/portaudio_v18/docs/pa_tut_over.html b/pd/portaudio_v18/docs/pa_tut_over.html deleted file mode 100644 index baa99205..00000000 --- a/pd/portaudio_v18/docs/pa_tut_over.html +++ /dev/null @@ -1,92 +0,0 @@ -<!doctype html public "-//w3c//dtd html 4.0 transitional//en"> -<html> -<head> - <meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1"> - <meta name="GENERATOR" content="Mozilla/4.79 [en] (Windows NT 5.0; U) [Netscape]"> - <meta name="Author" content="Phil Burk"> - <meta name="Description" content="Tutorial for PortAudio, a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function."> - <meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,"> - <title>PortAudio Tutorial</title> -</head> -<body> - -<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" > -<tr> -<td> -<center> -<h1> -PortAudio Tutorial</h1></center> -</td> -</tr> -</table></center> - -<h2> -Overview of PortAudio</h2> - -<blockquote>PortAudio is a library that provides streaming audio input -and output. It is a cross-platform API (Application Programming Interface) -that works on Windows, Macintosh, Unix running OSS, SGI, BeOS, and perhaps -other platforms by the time you read this. This means that you can write -a simple 'C' program to process or generate an audio signal, and that program -can run on several different types of computer just by recompiling the -source code. -<p>Here are the steps to writing a PortAudio application: -<ol> -<li> -Write a callback function that will be called by PortAudio when audio processing -is needed.</li> - -<li> -Initialize the PA library and open a stream for audio I/O.</li> - -<li> -Start the stream. Your callback function will be now be called repeatedly -by PA in the background.</li> - -<li> -In your callback you can read audio data from the inputBuffer and/or write -data to the outputBuffer.</li> - -<li> -Stop the stream by returning 1 from your callback, or by calling a stop -function.</li> - -<li> -Close the stream and terminate the library.</li> -</ol> -</blockquote> - -<blockquote>There is also <a href="pa_tut_rw.html">another interface</a> -provided that allows you to generate audio in the foreground. You then -simply write data to the stream and the tool will not return until it is -ready to accept more data. This interface is simpler to use but is usually -not preferred for large applications because it requires that you launch -a thread to perform the synthesis. Launching a thread may be difficult -on non-multi-tasking systems such as the Macintosh prior to MacOS X. -<p>Let's continue by building a simple application that will play a sawtooth -wave. -<p>Please select the page for the specific implementation you would like -to use: -<ul> -<li> -<a href="pa_tut_pc.html">Windows (WMME or DirectSound)</a></li> - -<li> -<a href="pa_tut_mac.html">Macintosh SoundManager for OS 7,8,9</a></li> - -<li> -<a href="pa_tut_mac_osx.html">Macintosh CoreAudio for OS X</a></li> - -<li> -<a href="pa_tut_asio.html">ASIO on Windows or Macintosh</a></li> - -<li> -<a href="pa_tut_oss.html">Unix OSS</a></li> -</ul> -or continue with the <a href="pa_tut_callback.html">next page of the programming -tutorial</a>.</blockquote> -<font size=+2><a href="http://www.portaudio.com/">home</a> | -<a href="pa_tutorial.html">contents</a> -| <a href="pa_tutorial.html">previous</a></font> -</body> -</html> diff --git a/pd/portaudio_v18/docs/pa_tut_pc.html b/pd/portaudio_v18/docs/pa_tut_pc.html deleted file mode 100644 index f7a70101..00000000 --- a/pd/portaudio_v18/docs/pa_tut_pc.html +++ /dev/null @@ -1,114 +0,0 @@ -<!doctype html public "-//w3c//dtd html 4.0 transitional//en"> -<html> -<head> - <meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1"> - <meta name="GENERATOR" content="Mozilla/4.79 [en] (Windows NT 5.0; U) [Netscape]"> - <meta name="Author" content="Phil Burk"> - <meta name="Description" content="Tutorial for PortAudio, a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function."> - <meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,"> - <title>PortAudio Tutorial</title> -</head> -<body> - -<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" > -<tr> -<td> -<center> -<h1> -PortAudio Tutorial</h1></center> -</td> -</tr> -</table></center> - -<h2> -Compiling for Windows (WMME or DirectSound)</h2> - -<blockquote>To compile PortAudio for Windows, you can choose between three -options: -<ul> -<li> -DirectSound API.</li> - -<li> -Windows MultiMedia Extensions API (aka WMME or WAVE).</li> - -<li> -<a href="pa_tut_asio.html">Steinberg's ASIO API</a></li> -</ul> -Some advantages of using DirectSound are that DirectSound may have lower -latency than WMME, and supports effects processing plugins. But one disadvantage -is that DirectSound is not installed on all PCs, and is not well supported -under Windows NT. <b>So WMME is the best choice for most projects.</b><b></b> -<p><b>Note: </b>If you are compiling one of the PortAudio test programs -with Visual C++, then create a new Project of type "Win32 Console Application". -<h3> -All</h3> -For any Windows implementation, add the following source files to your -project: -<blockquote> -<pre><b>pa_common\pa_lib.c -pa_common\portaudio.h -pa_common\pa_host.h</b></pre> -</blockquote> -Link with the system library "<b>winmm.lib</b>". For Visual C++: -<ol> -<li> -select "Settings..." from the "Project" menu,</li> - -<li> -select the project name in the tree on the left,</li> - -<li> -choose "All Configurations" in the popup menu above the tree,</li> - -<li> -select the "Link" tab,</li> - -<li> -enter "winmm.lib", without quotes, as the first item in the "Object/library -modules:" field.</li> -</ol> - -<h3> -WMME</h3> -To use the WMME implementation, add the following source files to your -project: -<blockquote><b><tt>pa_win_wmme/pa_win_wmme.c</tt></b></blockquote> - -<h3> -DirectSound</h3> -If you want to use the DirectSound implementation of PortAudio then you -must have a recent copy of the free -<a href="http://www.microsoft.com/directx/download.asp">DirectX</a> -SDK for Developers from Microsoft installed on your computer. To compile -an application add the following source files to your project: -<blockquote> -<pre><b>pa_win_ds\dsound_wrapper.c -pa_win_ds\pa_dsound.c</b></pre> -</blockquote> -Link with both system libraries "<b>dsound.lib</b>" and "<b>winmm.lib</b>" -using the procedure described above for "winmm.lib". -<br> -<table BORDER > -<tr> -<td><b>Borland</b> users cannot link with the "dsound.lib" from Microsoft -directly. Emmanuel offered this advice: -<p>One can use implib from Borland to generate a new .lib file which is -compatible with Borland C++. -<p>Use: "implib dsound.dll dsound.lib" and include dsound.lib into your -project. -<p>I still had a problem executing the patest_record example. The thread -ended with an error like 'Floating point overflow at...'. This problem -was caused due to a fault in the compiler. Now I'm using Borland 5.02 (instead -of 5.01). Everything seems to be working fine at the moment.</td> -</tr> -</table> -</blockquote> - -<blockquote>You might try compiling the "pa_tests\patest_saw.c" file first -because it is the simplest.</blockquote> -<font size=+2><a href="http://www.portaudio.com/">home</a> | -<a href="pa_tutorial.html">contents</a> -| <a href="pa_tut_over.html">previous</a> | <a href="pa_tut_callback.html">next</a></font> -</body> -</html> diff --git a/pd/portaudio_v18/docs/pa_tut_run.html b/pd/portaudio_v18/docs/pa_tut_run.html deleted file mode 100644 index 5c70d089..00000000 --- a/pd/portaudio_v18/docs/pa_tut_run.html +++ /dev/null @@ -1,56 +0,0 @@ -<!doctype html public "-//w3c//dtd html 4.0 transitional//en"> -<html> -<head> - <meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1"> - <meta name="GENERATOR" content="Mozilla/4.73 [en]C-gatewaynet (Win98; U) [Netscape]"> - <meta name="Author" content="Phil Burk"> - <meta name="Description" content="Tutorial for PortAudio, a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function."> - <meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,"> - <title>PortAudio Tutorial</title> -</head> -<body> - -<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" > -<tr> -<td> -<center> -<h1> -PortAudio Tutorial</h1></center> -</td> -</tr> -</table></center> - -<h2> -Starting and Stopping a Stream</h2> - -<blockquote>The stream will not start running until you call Pa_StartStream(). -Then it will start calling your callback function to perform the audio -processing. -<blockquote> -<pre>err = Pa_StartStream( stream ); -if( err != paNoError ) goto error;</pre> -</blockquote> -At this point, audio is being generated. You can communicate to your callback -routine through the data structure you passed in on the open call, or through -global variables, or using other interprocess communication techniques. -Please be aware that your callback function may be called at interrupt -time when your foreground process is least expecting it. So avoid sharing -complex data structures that are easily corrupted like double linked lists. -<p>In many of the tests we simply sleep for a few seconds so we can hear -the sound. This is easy to do with Pa_Sleep() which will sleep for some -number of milliseconds. Do not rely on this function for accurate scheduling. -it is mostly for writing examples. -<blockquote> -<pre>/* Sleep for several seconds. */ -Pa_Sleep(NUM_SECONDS*1000);</pre> -</blockquote> -When you are through, you can stop the stream from the foreground. -<blockquote> -<pre>err = Pa_StopStream( stream ); -if( err != paNoError ) goto error;</pre> -</blockquote> -You can also stop the stream by returning 1 from your custom callback function.</blockquote> -<font size=+2><a href="http://www.portaudio.com/">home</a> | <a href="pa_tutorial.html">contents</a> -| <a href="pa_tut_open.html">previous</a> | <a href="pa_tut_term.html">next</a></font> -</body> -</html> diff --git a/pd/portaudio_v18/docs/pa_tut_rw.html b/pd/portaudio_v18/docs/pa_tut_rw.html deleted file mode 100644 index 3e8889fb..00000000 --- a/pd/portaudio_v18/docs/pa_tut_rw.html +++ /dev/null @@ -1,79 +0,0 @@ -<!doctype html public "-//w3c//dtd html 4.0 transitional//en"> -<html> -<head> - <meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1"> - <meta name="GENERATOR" content="Mozilla/4.77 [en]C-gatewaynet (Win98; U) [Netscape]"> - <meta name="Author" content="Phil Burk"> - <meta name="Description" content="Tutorial for PortAudio, a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function."> - <meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,"> - <title>PortAudio Tutorial</title> -</head> -<body> - -<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" > -<tr> -<td> -<center> -<h1> -PortAudio Tutorial</h1></center> -</td> -</tr> -</table></center> - -<h2> -Blocking Read/Write Functions</h2> - -<blockquote>[Note: These functions are not part of the official PortAudio -API. They are simply built on top of PortAudio as an extra utility. Also -note that they are under evaluation and their definition may change.] -<p>There are two fundamentally different ways to design an audio API. One -is to use callback functions the way we have already shown. The callback -function operates under an interrupt or background thread This leaves the -foreground application free to do other things while the audio just runs -in the background. But this can sometimes be awkward. -<p>So we have provided an alternative technique that lets a program generate -audio in the foreground and then just write it to the audio stream as if -it was a file. If there is not enough room in the audio buffer for more -data, then the write function will just block until more room is available. -This can make it very easy to write an audio example. To use this tool, -you must add the files "pablio/pablio.c" and "pablio/ringbuffer.c" to your -project. You must also: -<blockquote> -<pre>#include "pablio.h"</pre> -</blockquote> -Here is a short excerpt of a program that opens a stream for input and -output. It then reads a block of samples from input, and writes them to -output, in a loop. The complete example can be found in "pablio/test_rw.c". -<blockquote> -<pre> #define SAMPLES_PER_FRAME (2) - #define FRAMES_PER_BLOCK (1024) - SAMPLE samples[SAMPLES_PER_FRAME * FRAMES_PER_BLOCK]; - PaError err; - PABLIO_Stream *aStream; - -/* Open simplified blocking I/O layer on top of PortAudio. */ - err = OpenAudioStream( &aStream, SAMPLE_RATE, paFloat32, - (PABLIO_READ_WRITE | PABLIO_STEREO) ); - if( err != paNoError ) goto error; - -/* Process samples in the foreground. */ - for( i=0; i<(NUM_SECONDS * SAMPLE_RATE); i++ ) - { - /* Read one block of data into sample array from audio input. */ - ReadAudioStream( aStream, samples, FRAMES_PER_BLOCK ); - /* - ** At this point you could process the data in samples array, - ** and write the result back to the same samples array. - */ - /* Write that same frame of data to output. */ - WriteAudioStream( aStream, samples, FRAMES_PER_BLOCK ); - } - - CloseAudioStream( aStream );</pre> -</blockquote> -</blockquote> -<font size=+2><a href="http://www.portaudio.com/">home</a> | -<a href="pa_tutorial.html">contents</a> -| <a href="pa_tut_devs.html">previous</a> | <a href="pa_tut_explore.html">next</a></font> -</body> -</html> diff --git a/pd/portaudio_v18/docs/pa_tut_term.html b/pd/portaudio_v18/docs/pa_tut_term.html deleted file mode 100644 index 1c72209f..00000000 --- a/pd/portaudio_v18/docs/pa_tut_term.html +++ /dev/null @@ -1,47 +0,0 @@ -<!doctype html public "-//w3c//dtd html 4.0 transitional//en"> -<html> -<head> - <meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1"> - <meta name="GENERATOR" content="Mozilla/4.73 [en]C-gatewaynet (Win98; U) [Netscape]"> - <meta name="Author" content="Phil Burk"> - <meta name="Description" content="Tutorial for PortAudio, a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function."> - <meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,"> - <title>PortAudio Tutorial</title> -</head> -<body> - -<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" > -<tr> -<td> -<center> -<h1> -PortAudio Tutorial</h1></center> -</td> -</tr> -</table></center> - -<h2> -Terminating PortAudio</h2> - -<blockquote>You can start and stop a stream as many times as you like. -But when you are done using it, you should close it by calling:</blockquote> - -<blockquote> -<blockquote> -<pre>err = Pa_CloseStream( stream ); -if( err != paNoError ) goto error;</pre> -</blockquote> -Then when you are done using PortAudio, you should terminate the whole -system by calling: -<blockquote> -<pre>Pa_Terminate();</pre> -</blockquote> -That's basically it. You can now write an audio program in 'C' that will -run on multiple platforms, for example PCs and Macintosh. -<p>In the rest of the tutorial we will look at some additional utility -functions, and a different way of using PortAudio that does not require -the use of a callback function.</blockquote> -<font size=+2><a href="http://www.portaudio.com/">home</a> | <a href="pa_tutorial.html">contents</a> -| <a href="pa_tut_run.html">previous</a> | <a href="pa_tut_util.html">next</a></font> -</body> -</html> diff --git a/pd/portaudio_v18/docs/pa_tut_util.html b/pd/portaudio_v18/docs/pa_tut_util.html deleted file mode 100644 index f4b54750..00000000 --- a/pd/portaudio_v18/docs/pa_tut_util.html +++ /dev/null @@ -1,55 +0,0 @@ -<!doctype html public "-//w3c//dtd html 4.0 transitional//en"> -<html> -<head> - <meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1"> - <meta name="GENERATOR" content="Mozilla/4.75 [en]C-gatewaynet (Win98; U) [Netscape]"> - <meta name="Author" content="Phil Burk"> - <meta name="Description" content="Tutorial for PortAudio, a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function."> - <meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,"> - <title>PortAudio Tutorial</title> -</head> -<body> - -<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" > -<tr> -<td> -<center> -<h1> -PortAudio Tutorial</h1></center> -</td> -</tr> -</table></center> - -<h2> -Utility Functions</h2> - -<blockquote>Here are several more functions that are not critical, but -may be handy when using PortAudio. -<p>Pa_StreamActive() returns one when the stream in playing audio, zero -when not playing, or a negative error number if the stream is invalid. -The stream is active between calls to Pa_StartStream() and Pa_StopStream(), -but may also become inactive if the callback returns a non-zero value. -In the latter case, the stream is considered inactive after the last buffer -has finished playing. -<blockquote> -<pre>PaError Pa_StreamActive( PortAudioStream *stream );</pre> -</blockquote> -Pa_StreamTime() returns the number of samples that have been generated. -PaTimeStamp is a double precision number which is a convenient way to pass -big numbers around even though we only need integers. -<blockquote> -<pre>PaTimestamp Pa_StreamTime( PortAudioStream *stream );</pre> -</blockquote> -The "CPU Load" is a fraction of total CPU time consumed by the stream's -audio processing. A value of 0.5 would imply that PortAudio and the sound -generating callback was consuming roughly 50% of the available CPU time. -This function may be called from the callback function or the application. -<blockquote> -<pre>double Pa_GetCPULoad( PortAudioStream* stream );</pre> -</blockquote> -</blockquote> -<font size=+2><a href="http://www.portaudio.com/">home</a> | -<a href="pa_tutorial.html">contents</a> | <a href="pa_tut_term.html">previous</a> -| <a href="pa_tut_devs.html">next</a></font> -</body> -</html> diff --git a/pd/portaudio_v18/docs/pa_tutorial.html b/pd/portaudio_v18/docs/pa_tutorial.html deleted file mode 100644 index 1371c44f..00000000 --- a/pd/portaudio_v18/docs/pa_tutorial.html +++ /dev/null @@ -1,46 +0,0 @@ -<!doctype html public "-//w3c//dtd html 4.0 transitional//en"> -<html> -<head> - <meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1"> - <meta name="GENERATOR" content="Mozilla/4.79 [en] (Windows NT 5.0; U) [Netscape]"> - <meta name="Author" content="Phil Burk"> - <meta name="Description" content="Tutorial for PortAudio, a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function."> - <meta name="KeyWords" content="audio, tutorial, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,"> - <title>PortAudio Tutorial</title> -</head> -<body> - -<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" > -<tr> -<td> -<center> -<h1> -PortAudio Tutorial</h1></center> -</td> -</tr> -</table></center> - -<p>Copyright 2000 Phil Burk and Ross Bencina -<h2> -Table of Contents</h2> - -<blockquote><a href="pa_tut_over.html">Overview of PortAudio</a> -<br><a href="pa_tut_mac.html">Compiling for Macintosh OS 7,8,9</a> -<br><a href="pa_tut_mac_osx.html">Compiling for Macintosh OS X</a> -<br><a href="pa_tut_pc.html">Compiling for Windows (DirectSound and WMME)</a> -<br><a href="pa_tut_asio.html">Compiling for ASIO on Windows or Mac OS -8,9</a> -<br><a href="pa_tut_oss.html">Compiling for Unix OSS</a> -<br><a href="pa_tut_callback.html">Writing a Callback Function</a> -<br><a href="pa_tut_init.html">Initializing PortAudio</a> -<br><a href="pa_tut_open.html">Opening a Stream using Defaults</a> -<br><a href="pa_tut_run.html">Starting and Stopping a Stream</a> -<br><a href="pa_tut_term.html">Cleaning Up</a> -<br><a href="pa_tut_util.html">Utilities</a> -<br><a href="pa_tut_devs.html">Querying for Devices</a> -<br><a href="pa_tut_rw.html">Blocking Read/Write Functions</a> -<br><a href="pa_tut_explore.html">Exploring the PortAudio Package</a></blockquote> -<font size=+2><a href="http://www.portaudio.com/">home</a> | contents | -previous | <a href="pa_tut_over.html">next</a></font> -</body> -</html> diff --git a/pd/portaudio_v18/docs/portaudio_h.txt b/pd/portaudio_v18/docs/portaudio_h.txt deleted file mode 100644 index 6d60086f..00000000 --- a/pd/portaudio_v18/docs/portaudio_h.txt +++ /dev/null @@ -1,425 +0,0 @@ -#ifndef PORT_AUDIO_H -#define PORT_AUDIO_H - -#ifdef __cplusplus -extern "C" { -#endif /* __cplusplus */ - -/* - * PortAudio Portable Real-Time Audio Library - * PortAudio API Header File - * Latest version available at: http://www.audiomulch.com/portaudio/ - * - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ - -typedef int PaError; -typedef enum { - paNoError = 0, - - paHostError = -10000, - paInvalidChannelCount, - paInvalidSampleRate, - paInvalidDeviceId, - paInvalidFlag, - paSampleFormatNotSupported, - paBadIODeviceCombination, - paInsufficientMemory, - paBufferTooBig, - paBufferTooSmall, - paNullCallback, - paBadStreamPtr, - paTimedOut, - paInternalError -} PaErrorNum; - -/* - Pa_Initialize() is the library initialisation function - call this before - using the library. -*/ - -PaError Pa_Initialize( void ); - -/* - Pa_Terminate() is the library termination function - call this after - using the library. -*/ - -PaError Pa_Terminate( void ); - -/* - Return host specific error. - This can be called after receiving a paHostError. -*/ -long Pa_GetHostError( void ); - -/* - Translate the error number into a human readable message. -*/ -const char *Pa_GetErrorText( PaError errnum ); - -/* - Sample formats - - These are formats used to pass sound data between the callback and the - stream. Each device has a "native" format which may be used when optimum - efficiency or control over conversion is required. - - Formats marked "always available" are supported (emulated) by all devices. - - The floating point representation uses +1.0 and -1.0 as the respective - maximum and minimum. - -*/ - -typedef unsigned long PaSampleFormat; -#define paFloat32 ((PaSampleFormat) (1<<0)) /*always available*/ -#define paInt16 ((PaSampleFormat) (1<<1)) /*always available*/ -#define paInt32 ((PaSampleFormat) (1<<2)) /*always available*/ -#define paInt24 ((PaSampleFormat) (1<<3)) -#define paPackedInt24 ((PaSampleFormat) (1<<4)) -#define paInt8 ((PaSampleFormat) (1<<5)) -#define paUInt8 ((PaSampleFormat) (1<<6)) /* unsigned 8 bit, 128 is "ground" */ -#define paCustomFormat ((PaSampleFormat) (1<<16)) - -/* - Device enumeration mechanism. - - Device ids range from 0 to Pa_CountDevices()-1. - - Devices may support input, output or both. Device 0 is always the "default" - device and should support at least stereo in and out if that is available - on the taget platform _even_ if this involves kludging an input/output - device on platforms that usually separate input from output. Other platform - specific devices are specified by positive device ids. -*/ - -typedef int PaDeviceID; -#define paNoDevice -1 - -typedef struct{ - int structVersion; - const char *name; - int maxInputChannels; - int maxOutputChannels; -/* Number of discrete rates, or -1 if range supported. */ - int numSampleRates; -/* Array of supported sample rates, or {min,max} if range supported. */ - const double *sampleRates; - PaSampleFormat nativeSampleFormats; -} PaDeviceInfo; - - -int Pa_CountDevices(); -/* - Pa_GetDefaultInputDeviceID(), Pa_GetDefaultOutputDeviceID() - - Return the default device ID or paNoDevice if there is no devices. - The result can be passed to Pa_OpenStream(). - - On the PC, the user can specify a default device by - setting an environment variable. For example, to use device #1. - - set PA_RECOMMENDED_OUTPUT_DEVICE=1 - - The user should first determine the available device ID by using - the supplied application "pa_devs". -*/ -PaDeviceID Pa_GetDefaultInputDeviceID( void ); -PaDeviceID Pa_GetDefaultOutputDeviceID( void ); - -/* - PaTimestamp is used to represent a continuous sample clock with arbitrary - start time useful for syncronisation. The type is used in the outTime - argument to the callback function and the result of Pa_StreamTime() -*/ - -typedef double PaTimestamp; - -/* - Pa_GetDeviceInfo() returns a pointer to an immutable PaDeviceInfo structure - referring to the device specified by id. - If id is out of range the function returns NULL. - - The returned structure is owned by the PortAudio implementation and must - not be manipulated or freed. The pointer is guaranteed to be valid until - between calls to Pa_Initialize() and Pa_Terminate(). -*/ - -const PaDeviceInfo* Pa_GetDeviceInfo( PaDeviceID devID ); - -/* - PortAudioCallback is implemented by clients of the portable audio api. - - inputBuffer and outputBuffer are arrays of interleaved samples, - the format, packing and number of channels used by the buffers are - determined by parameters to Pa_OpenStream() (see below). - - framesPerBuffer is the number of sample frames to be processed by the callback. - - outTime is the time in samples when the buffer(s) processed by - this callback will begin being played at the audio output. - See also Pa_StreamTime() - - userData is the value of a user supplied pointer passed to Pa_OpenStream() - intended for storing synthesis data etc. - - return value: - The callback can return a nonzero value to stop the stream. This may be - useful in applications such as soundfile players where a specific duration - of output is required. However, it is not necessary to utilise this mechanism - as StopStream() will also terminate the stream. A callback returning a - nonzero value must fill the entire outputBuffer. - - NOTE: None of the other stream functions may be called from within the - callback function except for Pa_GetCPULoad(). - -*/ - -typedef int (PortAudioCallback)( - void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ); - - -/* - Stream flags - - These flags may be supplied (ored together) in the streamFlags argument to - the Pa_OpenStream() function. - - [ suggestions? ] -*/ - -#define paNoFlag (0) -#define paClipOff (1<<0) /* disable defult clipping of out of range samples */ -#define paDitherOff (1<<1) /* disable default dithering */ -#define paPlatformSpecificFlags (0x00010000) -typedef unsigned long PaStreamFlags; - -/* - A single PortAudioStream provides multiple channels of real-time - input and output audio streaming to a client application. - Pointers to PortAudioStream objects are passed between PortAudio functions. -*/ - -typedef void PortAudioStream; -#define PaStream PortAudioStream - -/* - Pa_OpenStream() opens a stream for either input, output or both. - - stream is the address of a PortAudioStream pointer which will receive - a pointer to the newly opened stream. - - inputDevice is the id of the device used for input (see PaDeviceID above.) - inputDevice may be paNoDevice to indicate that an input device is not required. - - numInputChannels is the number of channels of sound to be delivered to the - callback. It can range from 1 to the value of maxInputChannels in the - device input record for the device specified in the inputDevice parameter. - If inputDevice is paNoDevice numInputChannels is ignored. - - inputSampleFormat is the format of inputBuffer provided to the callback - function. inputSampleFormat may be any of the formats described by the - PaSampleFormat enumeration (see above). PortAudio guarantees support for - the sound devices native formats (nativeSampleFormats in the device info - record) and additionally 16 and 32 bit integer and 32 bit floating point - formats. Support for other formats is implementation defined. - - inputDriverInfo is a pointer to an optional driver specific data structure - containing additional information for device setup or stream processing. - inputDriverInfo is never required for correct operation. If not used - inputDriverInfo should be NULL. - - outputDevice is the id of the device used for output (see PaDeviceID above.) - outputDevice may be paNoDevice to indicate that an output device is not required. - - numOutputChannels is the number of channels of sound to be supplied by the - callback. See the definition of numInputChannels above for more details. - - outputSampleFormat is the sample format of the outputBuffer filled by the - callback function. See the definition of inputSampleFormat above for more - details. - - outputDriverInfo is a pointer to an optional driver specific data structure - containing additional information for device setup or stream processing. - outputDriverInfo is never required for correct operation. If not used - outputDriverInfo should be NULL. - - sampleRate is the desired sampleRate for input and output - - framesPerBuffer is the length in sample frames of all internal sample buffers - used for communication with platform specific audio routines. Wherever - possible this corresponds to the framesPerBuffer parameter passed to the - callback function. - - numberOfBuffers is the number of buffers used for multibuffered - communication with the platform specific audio routines. This parameter is - provided only as a guide - and does not imply that an implementation must - use multibuffered i/o when reliable double buffering is available (such as - SndPlayDoubleBuffer() on the Macintosh.) - - streamFlags may contain a combination of flags ORed together. - These flags modify the behavior of the - streaming process. Some flags may only be relevant to certain buffer formats. - - callback is a pointer to a client supplied function that is responsible - for processing and filling input and output buffers (see above for details.) - - userData is a client supplied pointer which is passed to the callback - function. It could for example, contain a pointer to instance data necessary - for processing the audio buffers. - - return value: - Apon success Pa_OpenStream() returns PaNoError and places a pointer to a - valid PortAudioStream in the stream argument. The stream is inactive (stopped). - If a call to Pa_OpenStream() fails a nonzero error code is returned (see - PAError above) and the value of stream is invalid. - -*/ - -PaError Pa_OpenStream( PortAudioStream** stream, - PaDeviceID inputDevice, - int numInputChannels, - PaSampleFormat inputSampleFormat, - void *inputDriverInfo, - PaDeviceID outputDevice, - int numOutputChannels, - PaSampleFormat outputSampleFormat, - void *outputDriverInfo, - double sampleRate, - unsigned long framesPerBuffer, - unsigned long numberOfBuffers, - PaStreamFlags streamFlags, - PortAudioCallback *callback, - void *userData ); - - -/* - Pa_OpenDefaultStream() is a simplified version of Pa_OpenStream() that - opens the default input and/or ouput devices. Most parameters have - identical meaning to their Pa_OpenStream() counterparts, with the following - exceptions: - - If either numInputChannels or numOutputChannels is 0 the respective device - is not opened (same as passing paNoDevice in the device arguments to Pa_OpenStream() ) - - sampleFormat applies to both the input and output buffers. -*/ - -PaError Pa_OpenDefaultStream( PortAudioStream** stream, - int numInputChannels, - int numOutputChannels, - PaSampleFormat sampleFormat, - double sampleRate, - unsigned long framesPerBuffer, - unsigned long numberOfBuffers, - PortAudioCallback *callback, - void *userData ); - -/* - Pa_CloseStream() closes an audio stream, flushing any pending buffers. -*/ - -PaError Pa_CloseStream( PortAudioStream* ); - -/* - Pa_StartStream() and Pa_StopStream() begin and terminate audio processing. - Pa_StopStream() waits until all pending audio buffers have been played. - Pa_AbortStream() stops playing immediately without waiting for pending - buffers to complete. -*/ - -PaError Pa_StartStream( PortAudioStream *stream ); - -PaError Pa_StopStream( PortAudioStream *stream ); - -PaError Pa_AbortStream( PortAudioStream *stream ); - -/* - Pa_StreamActive() returns one when the stream is playing audio, - zero when not playing, or a negative error number if the - stream is invalid. - The stream is active between calls to Pa_StartStream() and Pa_StopStream(), - but may also become inactive if the callback returns a non-zero value. - In the latter case, the stream is considered inactive after the last - buffer has finished playing. -*/ - -PaError Pa_StreamActive( PortAudioStream *stream ); - -/* - Pa_StreamTime() returns the current output time for the stream in samples. - This time may be used as a time reference (for example syncronising audio to - MIDI). -*/ - -PaTimestamp Pa_StreamTime( PortAudioStream *stream ); - -/* - The "CPU Load" is a fraction of total CPU time consumed by the - stream's audio processing. - A value of 0.5 would imply that PortAudio and the sound generating - callback was consuming roughly 50% of the available CPU time. - This function may be called from the callback function or the application. -*/ -double Pa_GetCPULoad( PortAudioStream* stream ); - -/* - Use Pa_GetMinNumBuffers() to determine minimum number of buffers required for - the current host based on minimum latency. - On the PC, for the DirectSound implementation, latency can be optionally set - by user by setting an environment variable. - For example, to set latency to 200 msec, put: - - set PA_MIN_LATENCY_MSEC=200 - - in the AUTOEXEC.BAT file and reboot. - If the environment variable is not set, then the latency will be determined - based on the OS. Windows NT has higher latency than Win95. -*/ - -int Pa_GetMinNumBuffers( int framesPerBuffer, double sampleRate ); - -/* - Sleep for at least 'msec' milliseconds. - You may sleep longer than the requested time so don't rely - on this for accurate musical timing. -*/ -void Pa_Sleep( long msec ); - -/* - Return size in bytes of a single sample in a given PaSampleFormat - or paSampleFormatNotSupported. -*/ -PaError Pa_GetSampleSize( PaSampleFormat format ); - -#ifdef __cplusplus -} -#endif /* __cplusplus */ -#endif /* PORT_AUDIO_H */ diff --git a/pd/portaudio_v18/docs/portaudio_icmc2001.pdf b/pd/portaudio_v18/docs/portaudio_icmc2001.pdf Binary files differdeleted file mode 100644 index dd074b7d..00000000 --- a/pd/portaudio_v18/docs/portaudio_icmc2001.pdf +++ /dev/null diff --git a/pd/portaudio_v18/docs/proposals.html b/pd/portaudio_v18/docs/proposals.html Binary files differdeleted file mode 100644 index 88dd2d07..00000000 --- a/pd/portaudio_v18/docs/proposals.html +++ /dev/null diff --git a/pd/portaudio_v18/docs/releases.html b/pd/portaudio_v18/docs/releases.html deleted file mode 100644 index aec80a1c..00000000 --- a/pd/portaudio_v18/docs/releases.html +++ /dev/null @@ -1,339 +0,0 @@ -<!doctype html public "-//w3c//dtd html 4.0 transitional//en"> -<html> -<head> - <meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1"> - <meta name="GENERATOR" content="Mozilla/4.79 [en] (Windows NT 5.0; U) [Netscape]"> - <meta name="Author" content="Phil Burk"> - <meta name="Description" content="PortAudio is a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function."> - <meta name="KeyWords" content="audio, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,"> - <title>PortAudio Release Notes</title> -</head> -<body> - -<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" > -<tr> -<td> -<center> -<h1> -PortAudio - Release Notes</h1></center> -</td> -</tr> -</table></center> - -<p>Link to <a href="http://www.portaudio.com">PortAudio Home Page</a> -<h2> -<b>V18 - 5/6/02</b></h2> - -<blockquote>All source code and documentation now under <a href="http://www.portaudio.com/usingcvs.html">CVS</a>. -<p>Ran most of the code through <a href="http://astyle.sourceforge.net/">AStyle</a> -to cleanup ragged indentation caused by using different editors. Used this -command: -<br><tt> astyle --style=ansi -c -o --convert-tabs --indent-preprocessor -*.c</tt></blockquote> - -<blockquote>Added "pa_common/pa_convert.c" for Mac OS X. Start of new conversion -utilities. -<p><b>ASIO</b> -<ul> -<li> -New Pa_ASIO_Adaptor_Init function to init Callback adpatation variables,</li> - -<li> -Cleanup of Pa_ASIO_Callback_Input</li> - -<li> -Break apart device loading to debug random failure in Pa_ASIO_QueryDeviceInfo</li> - -<li> -Deallocate all resources in PaHost_Term for cases where Pa_CloseStream -is not called properly</li> - -<li> -New Pa_ASIO_loadDriver that calls CoInitialize on each thread on Windows. -Allows use by multiple threads.</li> - -<li> -Correct error code management in PaHost_Term, removed various compiler -warning</li> - -<li> -Add Mac includes for <Devices.h> and <Timer.h></li> - -<li> -Pa_ASIO_QueryDeviceInfo bug correction, memory allocation checking, better -error handling</li> -</ul> -<b>Mac OS X</b> -<ul> -<li> -Major cleanup and improvements.</li> - -<li> -Fixed device queries for numChannels and sampleRates,</li> - -<li> -Audio input works if using same CoreAudio device (some HW devices make -separate CoreAudio devices).</li> - -<li> -Added paInt16, paInt8, format using new "pa_common/pa_convert.c" file.</li> - -<li> -Return error if opened in mono mode cuz not supported.</li> - -<li> -Check for getenv("PA_MIN_LATEWNCY_MSEC") to set latency externally.</li> - -<li> -Use getrusage() instead of gettimeofday() for CPU Load calculation.</li> -</ul> -<b>Windows MME</b> -<ul> -<li> -Fixed bug that caused TIMEOUT in Pa_StopStream(). Added check for past_StopSoon() -in Pa_TimeSlice(). Thanks Julien Maillard.</li> - -<li> -Detect Win XP versus NT, use lower latency.</li> - -<li> -Fix DBUG typo;</li> - -<li> -removed init of CurrentCount which was not compiling on Borland</li> - -<li> -general cleanup, factored streamData alloc and cpu usage initialization</li> - -<li> -stopped counting WAVE_MAPPER when there were no audio cards plugged in</li> -</ul> -<b>Windows DirectSound</b> -<ul> -<li> -Detect Win XP and Win 2K properly when determining latency.</li> -</ul> -<b>Unix OSS</b> -<ul> -<li> -Use high real-time priority if app is running with root priveledges. Lowers -latency.</li> - -<li> -Added watch dog thread that prevents real-time thread from hogging CPU -and hanging the computer.</li> - -<li> -Check error return from read() and write().</li> - -<li> -Check CPU endianness instead of assuming Little Endian.</li> -</ul> -</blockquote> - -<h2> -<b>V17 - 10/15/01</b></h2> - -<blockquote><b>Unix OSS</b> -<ul> -<li> -Set num channels back to two after device query for ALSA. This fixed a -bug in V16 that sometimes caused a failure when querying for the sample -rates. Thanks Stweart Greenhill.</li> -</ul> -</blockquote> - -<blockquote> -<h4> -<b>Macintosh Sound Manager</b></h4> - -<ul> -<li> -Use NewSndCallBackUPP() for CARBON compatibility.</li> -</ul> -</blockquote> - -<h2> -<b>V16 - 9/27/01</b></h2> - -<blockquote><b>Added Alpha implementations for ASIO, SGI, and BeOS!</b> -<br> -<li> -CPULoad is now calculated based on the time spent to generate a known number -of frames. This is more accurate than a simple percentage of real-time. -Implemented in pa_unix_oss, pa_win_wmme and pa_win_ds.</li> - -<li> -Fix dither and shift for recording PaUInt8 format data.</li> - -<li> -Added "patest_maxsines.c" which tests <tt>Pa_GetCPULoad().</tt></li> -</blockquote> - -<blockquote> -<h4> -Windows WMME</h4> - -<ul> -<li> -sDevicePtrs now allocated using <tt>GlobalAlloc()</tt>. This prevents a -crash in Pa_Terminate() on Win2000. Thanks Mike Berry for finding this. -Thanks Mike Berry.</li> - -<li> -Pass process instead of thread to <tt>SetPriorityClass</tt>(). This fixes -a bug that caused the priority to not be increased. Thanks to Alberto di -Bene for spotting this.</li> -</ul> - -<h4> -Windows DirectSound</h4> - -<ul> -<li> -Casts for compiling with __MWERKS__ CodeWarrior.</li> -</ul> - -<h4> -UNIX OSS</h4> - -<ul> -<li> -Derived from Linux OSS implementation.</li> - -<li> -Numerous patches from Heiko Purnhagen, Stephen Brandon, etc.</li> - -<li> -Improved query mechanism which often bailed out unnecessarily.</li> - -<li> -Removed sNumDevices and potential related bugs,</li> - -<li> -Use <tt>getenv("PA_MIN_LATENCY_MSEC")</tt> in code to set desired latency. -User can set by entering:</li> - -<br> <tt>export PA_MIN_LATENCY_MSEC=40</tt></ul> - -<h4> -Macintosh Sound Manager</h4> - -<ul> -<li> -Pass unused event to WaitNextEvent instead of NULL to prevent Mac OSX crash. -Thanks Dominic Mazzoni.</li> - -<li> -Use requested number of input channels.</li> - -<br> </ul> -</blockquote> - -<h2> -<b>V15 - 5/29/01</b></h2> - -<blockquote> -<ul> -<li> -<b>New Linux OSS Beta</b></li> -</ul> - -<h4> -Windows WMME</h4> - -<ul> -<li> - sDevicePtrs now allocated based on sizeof(pointer). Was allocating -too much space.</li> - -<li> - Check for excessive numbers of channels. Some drivers reported bogus -numbers.</li> - -<li> -Apply Mike Berry's changes for CodeWarrior on PC including condition including -of memory.h, and explicit typecasting on memory allocation.</li> -</ul> - -<h4> -Macintosh Sound Manager</h4> - -<ul> -<li> -ScanInputDevices was setting sDefaultOutputDeviceID instead of sDefaultInputDeviceID.</li> - -<li> -Device Scan was crashing for anything other than siBadSoundInDevice, but -some Macs may return other errors! Caused failure to init on some G4s under -OS9.</li> - -<li> -Fix TIMEOUT in record mode.</li> - -<li> -Change CARBON_COMPATIBLE to TARGET_API_MAC_CARBON</li> -</ul> -</blockquote> - -<h2> -<b>V14 - 2/6/01</b></h2> - -<blockquote> -<ul> -<li> -Added implementation for Windows MultiMedia Extensions (WMME) by Ross and -Phil</li> - -<li> -Changed Pa_StopStream() so that it waits for the buffers to drain.</li> - -<li> -Added Pa_AbortStream() that stops immediately without waiting.</li> - -<li> -Added new test: patest_stop.c to test above two mods.</li> - -<li> -Fixed Pa_StreamTime() so that it returns current play position instead -of the write position. Added "patest_sync.c" to demo audio/video sync.</li> - -<li> -Improved stability of Macintosh implementation. Added timeouts to prevent -hangs.</li> - -<li> -Added Pa_GetSampleSize( PaSampleFormat format );</li> - -<li> -Changes some "int"s to "long"s so that PA works properly on Macintosh which -often compiles using 16 bit ints.</li> - -<li> -Added Implementation Guide</li> -</ul> -</blockquote> - -<h2> -<b>V12 - 1/9/01</b></h2> - -<blockquote> -<ul> -<li> -Mac now scans for and queries all devices. But it does not yet support -selecting any other than the default device.</li> - -<li> -Blocking I/O calls renamed to separate them from the PortAudio API.</li> - -<li> -Cleaned up indentation problems with tabs versus spaces.</li> - -<li> -Now attempts to correct bogus sample rate info returned from DirectSound -device queries.</li> -</ul> -</blockquote> - -</body> -</html> diff --git a/pd/portaudio_v18/fixdir.bat b/pd/portaudio_v18/fixdir.bat deleted file mode 100644 index 92d6c747..00000000 --- a/pd/portaudio_v18/fixdir.bat +++ /dev/null @@ -1,19 +0,0 @@ -rem Use Astyle to fix style in 'C' files -cd %1% - -fixlines -p *.c -fixlines -p *.cpp -fixlines -p *.cc - -astyle --style=ansi -c -o --convert-tabs --indent-preprocessor *.c -astyle --style=ansi -c -o --convert-tabs --indent-preprocessor *.cpp -astyle --style=ansi -c -o --convert-tabs --indent-preprocessor *.cc -del *.orig -@rem convert line terminators to Unix style LFs -fixlines -u *.c -fixlines -u *.cpp -fixlines -u *.cc -fixlines -u *.h -del *.bak - -cd ..\ diff --git a/pd/portaudio_v18/fixfile.bat b/pd/portaudio_v18/fixfile.bat deleted file mode 100644 index 48f6fbc2..00000000 --- a/pd/portaudio_v18/fixfile.bat +++ /dev/null @@ -1,7 +0,0 @@ -rem Use Astyle to fix style in a file -fixlines -p %1% -astyle --style=ansi -c -o --convert-tabs --indent-preprocessor %1% -del %1%.orig -@rem convert line terminators to Unix style LFs -fixlines -u %1% -del %1%.bak diff --git a/pd/portaudio_v18/index.html b/pd/portaudio_v18/index.html deleted file mode 100644 index ff6cfeec..00000000 --- a/pd/portaudio_v18/index.html +++ /dev/null @@ -1,89 +0,0 @@ -<!doctype html public "-//w3c//dtd html 4.0 transitional//en"> -<html> -<head> - <meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1"> - <meta name="GENERATOR" content="Mozilla/4.79 [en] (Windows NT 5.0; U) [Netscape]"> - <meta name="Author" content="Phil Burk"> - <meta name="Description" content="PortAudio is a cross platform, open-source, audio I/O library.It provides a very simple API for recording and/or playing sound using a simple callback function."> - <meta name="KeyWords" content="audio, library, portable, open-source, DirectSound,sound, music, JSyn, synthesis,"> - <title>PortAudio - Cross-Platform Audio API</title> -</head> -<body> - -<center><table COLS=1 WIDTH="100%" BGCOLOR="#FADA7A" > -<tr> -<td> -<center> -<h1> -PortAudio - Portable Audio Library</h1></center> -</td> -</tr> -</table></center> - -<p>Last updated 5/6/02. -<p>PortAudio is a cross platform, <a href="#License">open-source</a>, audio -I/O library proposed by <b>Ross Bencina</b> to the <a href="http://shoko.calarts.edu/~glmrboy/musicdsp/music-dsp.html">music-dsp</a> -mailing list. It lets you write simple audio programs in 'C' that will -compile and run on <b>Windows, Macintosh, Unix, BeOS</b>. PortAudio is -intended to promote the exchange of audio synthesis software between developers -on different platforms. -<p>For complete information on PortAudio and to download the latest releases, -please visit "<b><font size=+2><a href="http://www.portaudio.com">http://www.portaudio.com</a></font></b>". -<br> -<br> -<center> -<h2> -<b><a href="docs/index.html">Click here for Documentation</a></b></h2></center> - -<h2> -<b><font size=+2></font></b></h2> - -<h2> -<b><font size=+2>Contacts and E-Mail List</font></b></h2> - -<ul> -<li> -If you are using or implementing PortAudio then please join the <b><font size=+1><a href="http://techweb.rfa.org/mailman/listinfo/portaudio">PortAudio -mail list</a></font><font size=+2> </font></b>generously administered by -<b>Bill -Eldridge</b>.</li> - -<li> -If you find bugs in one of these implementations, or have suggestions, -please e-mail them to <a href="mailto:philburk@softsynth.com">Phil Burk</a>.</li> - -<li> -If you make improvements to the library, please send them to us so we can -incorporate the improvements.</li> -</ul> - -<h2> -<a NAME="License"></a>License</h2> -PortAudio Portable Real-Time Audio Library -<br>Copyright (c) 1999-2000 Ross Bencina and Phil Burk -<p>Permission is hereby granted, free of charge, to any person obtaining -a copy of this software and associated documentation files (the "Software"), -to deal in the Software without restriction, including without limitation -the rights to use, copy, modify, merge, publish, distribute, sublicense, -and/or sell copies of the Software, and to permit persons to whom the Software -is furnished to do so, subject to the following conditions: -<ul> -<li> -The above copyright notice and this permission notice shall be included -in all copies or substantial portions of the Software.</li> - -<li> -Any person wishing to distribute modifications to the Software is requested -to send the modifications to the original developer so that they can be -incorporated into the canonical version.</li> -</ul> -THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS -OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, -FITNESS FOR A PARTICULAR PURPOSE AND ON INFRINGEMENT. -<br>IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY -CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT -OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR -THE USE OR OTHER DEALINGS IN THE SOFTWARE. -<br> -</body> -</html> diff --git a/pd/portaudio_v18/install-sh b/pd/portaudio_v18/install-sh deleted file mode 100644 index e9de2384..00000000 --- a/pd/portaudio_v18/install-sh +++ /dev/null @@ -1,251 +0,0 @@ -#!/bin/sh -# -# install - install a program, script, or datafile -# This comes from X11R5 (mit/util/scripts/install.sh). -# -# Copyright 1991 by the Massachusetts Institute of Technology -# -# Permission to use, copy, modify, distribute, and sell this software and its -# documentation for any purpose is hereby granted without fee, provided that -# the above copyright notice appear in all copies and that both that -# copyright notice and this permission notice appear in supporting -# documentation, and that the name of M.I.T. not be used in advertising or -# publicity pertaining to distribution of the software without specific, -# written prior permission. M.I.T. makes no representations about the -# suitability of this software for any purpose. It is provided "as is" -# without express or implied warranty. -# -# Calling this script install-sh is preferred over install.sh, to prevent -# `make' implicit rules from creating a file called install from it -# when there is no Makefile. -# -# This script is compatible with the BSD install script, but was written -# from scratch. It can only install one file at a time, a restriction -# shared with many OS's install programs. - - -# set DOITPROG to echo to test this script - -# Don't use :- since 4.3BSD and earlier shells don't like it. -doit="${DOITPROG-}" - - -# put in absolute paths if you don't have them in your path; or use env. vars. - -mvprog="${MVPROG-mv}" -cpprog="${CPPROG-cp}" -chmodprog="${CHMODPROG-chmod}" -chownprog="${CHOWNPROG-chown}" -chgrpprog="${CHGRPPROG-chgrp}" -stripprog="${STRIPPROG-strip}" -rmprog="${RMPROG-rm}" -mkdirprog="${MKDIRPROG-mkdir}" - -transformbasename="" -transform_arg="" -instcmd="$mvprog" -chmodcmd="$chmodprog 0755" -chowncmd="" -chgrpcmd="" -stripcmd="" -rmcmd="$rmprog -f" -mvcmd="$mvprog" -src="" -dst="" -dir_arg="" - -while [ x"$1" != x ]; do - case $1 in - -c) instcmd="$cpprog" - shift - continue;; - - -d) dir_arg=true - shift - continue;; - - -m) chmodcmd="$chmodprog $2" - shift - shift - continue;; - - -o) chowncmd="$chownprog $2" - shift - shift - continue;; - - -g) chgrpcmd="$chgrpprog $2" - shift - shift - continue;; - - -s) stripcmd="$stripprog" - shift - continue;; - - -t=*) transformarg=`echo $1 | sed 's/-t=//'` - shift - continue;; - - -b=*) transformbasename=`echo $1 | sed 's/-b=//'` - shift - continue;; - - *) if [ x"$src" = x ] - then - src=$1 - else - # this colon is to work around a 386BSD /bin/sh bug - : - dst=$1 - fi - shift - continue;; - esac -done - -if [ x"$src" = x ] -then - echo "install: no input file specified" - exit 1 -else - true -fi - -if [ x"$dir_arg" != x ]; then - dst=$src - src="" - - if [ -d $dst ]; then - instcmd=: - chmodcmd="" - else - instcmd=mkdir - fi -else - -# Waiting for this to be detected by the "$instcmd $src $dsttmp" command -# might cause directories to be created, which would be especially bad -# if $src (and thus $dsttmp) contains '*'. - - if [ -f $src -o -d $src ] - then - true - else - echo "install: $src does not exist" - exit 1 - fi - - if [ x"$dst" = x ] - then - echo "install: no destination specified" - exit 1 - else - true - fi - -# If destination is a directory, append the input filename; if your system -# does not like double slashes in filenames, you may need to add some logic - - if [ -d $dst ] - then - dst="$dst"/`basename $src` - else - true - fi -fi - -## this sed command emulates the dirname command -dstdir=`echo $dst | sed -e 's,[^/]*$,,;s,/$,,;s,^$,.,'` - -# Make sure that the destination directory exists. -# this part is taken from Noah Friedman's mkinstalldirs script - -# Skip lots of stat calls in the usual case. -if [ ! -d "$dstdir" ]; then -defaultIFS=' -' -IFS="${IFS-${defaultIFS}}" - -oIFS="${IFS}" -# Some sh's can't handle IFS=/ for some reason. -IFS='%' -set - `echo ${dstdir} | sed -e 's@/@%@g' -e 's@^%@/@'` -IFS="${oIFS}" - -pathcomp='' - -while [ $# -ne 0 ] ; do - pathcomp="${pathcomp}${1}" - shift - - if [ ! -d "${pathcomp}" ] ; - then - $mkdirprog "${pathcomp}" - else - true - fi - - pathcomp="${pathcomp}/" -done -fi - -if [ x"$dir_arg" != x ] -then - $doit $instcmd $dst && - - if [ x"$chowncmd" != x ]; then $doit $chowncmd $dst; else true ; fi && - if [ x"$chgrpcmd" != x ]; then $doit $chgrpcmd $dst; else true ; fi && - if [ x"$stripcmd" != x ]; then $doit $stripcmd $dst; else true ; fi && - if [ x"$chmodcmd" != x ]; then $doit $chmodcmd $dst; else true ; fi -else - -# If we're going to rename the final executable, determine the name now. - - if [ x"$transformarg" = x ] - then - dstfile=`basename $dst` - else - dstfile=`basename $dst $transformbasename | - sed $transformarg`$transformbasename - fi - -# don't allow the sed command to completely eliminate the filename - - if [ x"$dstfile" = x ] - then - dstfile=`basename $dst` - else - true - fi - -# Make a temp file name in the proper directory. - - dsttmp=$dstdir/#inst.$$# - -# Move or copy the file name to the temp name - - $doit $instcmd $src $dsttmp && - - trap "rm -f ${dsttmp}" 0 && - -# and set any options; do chmod last to preserve setuid bits - -# If any of these fail, we abort the whole thing. If we want to -# ignore errors from any of these, just make sure not to ignore -# errors from the above "$doit $instcmd $src $dsttmp" command. - - if [ x"$chowncmd" != x ]; then $doit $chowncmd $dsttmp; else true;fi && - if [ x"$chgrpcmd" != x ]; then $doit $chgrpcmd $dsttmp; else true;fi && - if [ x"$stripcmd" != x ]; then $doit $stripcmd $dsttmp; else true;fi && - if [ x"$chmodcmd" != x ]; then $doit $chmodcmd $dsttmp; else true;fi && - -# Now rename the file to the real destination. - - $doit $rmcmd -f $dstdir/$dstfile && - $doit $mvcmd $dsttmp $dstdir/$dstfile - -fi && - - -exit 0 diff --git a/pd/portaudio_v18/pa_asio/Callback_adaptation_.pdf b/pd/portaudio_v18/pa_asio/Callback_adaptation_.pdf Binary files differdeleted file mode 100644 index 76bf6786..00000000 --- a/pd/portaudio_v18/pa_asio/Callback_adaptation_.pdf +++ /dev/null diff --git a/pd/portaudio_v18/pa_asio/Pa_ASIO.pdf b/pd/portaudio_v18/pa_asio/Pa_ASIO.pdf Binary files differdeleted file mode 100644 index ac5ecadb..00000000 --- a/pd/portaudio_v18/pa_asio/Pa_ASIO.pdf +++ /dev/null diff --git a/pd/portaudio_v18/pa_asio/pa_asio.cpp b/pd/portaudio_v18/pa_asio/pa_asio.cpp deleted file mode 100644 index 717ee558..00000000 --- a/pd/portaudio_v18/pa_asio/pa_asio.cpp +++ /dev/null @@ -1,3096 +0,0 @@ -/* - * $Id: pa_asio.cpp,v 1.7.4.8 2003/08/14 06:44:25 stephane Exp $ - * Portable Audio I/O Library for ASIO Drivers - * - * Author: Stephane Letz - * Based on the Open Source API proposed by Ross Bencina - * Copyright (c) 2000-2001 Stephane Letz, Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - */ - -/* Modification History - - 08-03-01 First version : Stephane Letz - 08-06-01 Tweaks for PC, use C++, buffer allocation, Float32 to Int32 conversion : Phil Burk - 08-20-01 More conversion, PA_StreamTime, Pa_GetHostError : Stephane Letz - 08-21-01 PaUInt8 bug correction, implementation of ASIOSTFloat32LSB and ASIOSTFloat32MSB native formats : Stephane Letz - 08-24-01 MAX_INT32_FP hack, another Uint8 fix : Stephane and Phil - 08-27-01 Implementation of hostBufferSize < userBufferSize case, better management of the ouput buffer when - the stream is stopped : Stephane Letz - 08-28-01 Check the stream pointer for null in bufferSwitchTimeInfo, correct bug in bufferSwitchTimeInfo when - the stream is stopped : Stephane Letz - 10-12-01 Correct the PaHost_CalcNumHostBuffers function: computes FramesPerHostBuffer to be the lowest that - respect requested FramesPerUserBuffer and userBuffersPerHostBuffer : Stephane Letz - 10-26-01 Management of hostBufferSize and userBufferSize of any size : Stephane Letz - 10-27-01 Improve calculus of hostBufferSize to be multiple or divisor of userBufferSize if possible : Stephane and Phil - 10-29-01 Change MAX_INT32_FP to (2147483520.0f) to prevent roundup to 0x80000000 : Phil Burk - 10-31-01 Clear the ouput buffer and user buffers in PaHost_StartOutput, correct bug in GetFirstMultiple : Stephane Letz - 11-06-01 Rename functions : Stephane Letz - 11-08-01 New Pa_ASIO_Adaptor_Init function to init Callback adpatation variables, cleanup of Pa_ASIO_Callback_Input: Stephane Letz - 11-29-01 Break apart device loading to debug random failure in Pa_ASIO_QueryDeviceInfo ; Phil Burk - 01-03-02 Desallocate all resources in PaHost_Term for cases where Pa_CloseStream is not called properly : Stephane Letz - 02-01-02 Cleanup, test of multiple-stream opening : Stephane Letz - 19-02-02 New Pa_ASIO_loadDriver that calls CoInitialize on each thread on Windows : Stephane Letz - 09-04-02 Correct error code management in PaHost_Term, removes various compiler warning : Stephane Letz - 12-04-02 Add Mac includes for <Devices.h> and <Timer.h> : Phil Burk - 13-04-02 Removes another compiler warning : Stephane Letz - 30-04-02 Pa_ASIO_QueryDeviceInfo bug correction, memory allocation checking, better error handling : D Viens, P Burk, S Letz - 01-12-02 Fix Pa_GetDefaultInputDeviceID and Pa_GetDefaultOuputDeviceID result when no driver are available : S Letz - 05-12-02 More debug messages : S Letz - 01-23-03 Increased max channels to 128. Fixed comparison of (OutputChannels > kMaxInputChannels) : P Burk - 02-17-03 Better termination handling : PaHost_CloseStream is called in PaHost_term is the the stream was not explicitely closed by the application : S Letz - 04-02-03 More robust ASIO driver buffer size initialization : some buggy drivers (like the Hoontech DSP24) give incorrect [min, preferred, max] values - They should work with the preferred size value, thus if Pa_ASIO_CreateBuffers fails with the hostBufferSize computed in PaHost_CalcNumHostBuffers, - we try again with the preferred size. Fix an old (never detected?) bug in the buffer adapdation code : S Letz - 30-06-03 The audio callback was not protected against reentrancy : some drivers (like the Hoontech DSP24) seems to cause this behaviour - that corrupted the buffer adapdation state and finally caused crashes. The reentrancy state is now checked in bufferSwitchTimeInfo : S Letz - 17-07-03 Correct bug in Pa_ASIO_Convert_Inter_Output : parameter past_InputSampleFormat was used instead of past_OutputSampleFormat : J Maillard, S Letz - 25-07-03 Use of atomic operations for reenterCounter management on Windows, need to be implemented on MacOS9 : S Letz - 14-08-03 OutTime value in the audio callback was not updated correctly : S Letz - - TO DO : - - - Check Pa_StopSteam and Pa_AbortStream - - Optimization for Input only or Ouput only (really necessary ??) -*/ - - -#include <stdio.h> -#include <assert.h> -#include <string.h> - -#include "portaudio.h" -#include "pa_host.h" -#include "pa_trace.h" - -#include "asiosys.h" -#include "asio.h" -#include "asiodrivers.h" - - -#if MAC -#include <Devices.h> -#include <Timer.h> -#include <Math64.h> -#else -#include <math.h> -#include <windows.h> -#include <mmsystem.h> -#endif - -enum { - // number of input and outputs supported by the host application - // you can change these to higher or lower values - kMaxInputChannels = 128, - kMaxOutputChannels = 128 -}; - -/* ASIO specific device information. */ -typedef struct internalPortAudioDevice -{ - PaDeviceInfo pad_Info; -} internalPortAudioDevice; - - -/* ASIO driver internal data storage */ -typedef struct PaHostSoundControl -{ - // ASIOInit() - ASIODriverInfo pahsc_driverInfo; - - // ASIOGetChannels() - int32 pahsc_NumInputChannels; - int32 pahsc_NumOutputChannels; - - // ASIOGetBufferSize() - sizes in frames per buffer - int32 pahsc_minSize; - int32 pahsc_maxSize; - int32 pahsc_preferredSize; - int32 pahsc_granularity; - - // ASIOGetSampleRate() - ASIOSampleRate pahsc_sampleRate; - - // ASIOOutputReady() - bool pahsc_postOutput; - - // ASIOGetLatencies () - int32 pahsc_inputLatency; - int32 pahsc_outputLatency; - - // ASIOCreateBuffers () - ASIOBufferInfo bufferInfos[kMaxInputChannels + kMaxOutputChannels]; // buffer info's - - // ASIOGetChannelInfo() - ASIOChannelInfo pahsc_channelInfos[kMaxInputChannels + kMaxOutputChannels]; // channel info's - // The above two arrays share the same indexing, as the data in them are linked together - - // Information from ASIOGetSamplePosition() - // data is converted to double floats for easier use, however 64 bit integer can be used, too - double nanoSeconds; - double samples; - double tcSamples; // time code samples - - // bufferSwitchTimeInfo() - ASIOTime tInfo; // time info state - unsigned long sysRefTime; // system reference time, when bufferSwitch() was called - - // Signal the end of processing in this example - bool stopped; - - ASIOCallbacks pahsc_asioCallbacks; - - int32 pahsc_userInputBufferFrameOffset; // Position in Input user buffer - int32 pahsc_userOutputBufferFrameOffset; // Position in Output user buffer - int32 pahsc_hostOutputBufferFrameOffset; // Position in Output ASIO buffer - - int32 past_FramesPerHostBuffer; // Number of frames in ASIO buffer - - int32 pahsc_InputBufferOffset; // Number of null frames for input buffer alignement - int32 pahsc_OutputBufferOffset; // Number of null frames for ouput buffer alignement - -#if MAC - UInt64 pahsc_EntryCount; - UInt64 pahsc_LastExitCount; -#elif WINDOWS - LARGE_INTEGER pahsc_EntryCount; - LARGE_INTEGER pahsc_LastExitCount; -#endif - - PaTimestamp pahsc_NumFramesDone; - - internalPortAudioStream *past; - - int32 reenterCount; // Counter of audio callback reentrancy - int32 reenterError; // Counter of audio callback reentrancy detection - -} PaHostSoundControl; - - -//---------------------------------------------------------- -#define PRINT(x) { printf x; fflush(stdout); } -#define ERR_RPT(x) PRINT(x) - -#define DBUG(x) /* PRINT(x) */ -#define DBUGX(x) /* PRINT(x) */ - -/* We are trying to be compatible with CARBON but this has not been thoroughly tested. */ -#define CARBON_COMPATIBLE (0) -#define PA_MAX_DEVICE_INFO (32) - -#define MIN_INT8 (-0x80) -#define MAX_INT8 (0x7F) - -#define MIN_INT8_FP ((float)-0x80) -#define MAX_INT8_FP ((float)0x7F) - -#define MIN_INT16_FP ((float)-0x8000) -#define MAX_INT16_FP ((float)0x7FFF) - -#define MIN_INT16 (-0x8000) -#define MAX_INT16 (0x7FFF) - -#define MAX_INT32_FP (2147483520.0f) /* 0x0x7FFFFF80 - seems safe */ - -/************************************************************************************/ -/****************** Data ************************************************************/ -/************************************************************************************/ -static int sNumDevices = 0; -static internalPortAudioDevice sDevices[PA_MAX_DEVICE_INFO] = { 0 }; -static int32 sPaHostError = 0; -static int sDefaultOutputDeviceID = 0; -static int sDefaultInputDeviceID = 0; - -PaHostSoundControl asioDriverInfo = {0}; - -#ifdef MAC -static bool swap = true; -#elif WINDOWS -static bool swap = false; -#endif - -// Prototypes -static void bufferSwitch(long index, ASIOBool processNow); -static ASIOTime *bufferSwitchTimeInfo(ASIOTime *timeInfo, long index, ASIOBool processNow); -static void sampleRateChanged(ASIOSampleRate sRate); -static long asioMessages(long selector, long value, void* message, double* opt); -static void Pa_StartUsageCalculation( internalPortAudioStream *past ); -static void Pa_EndUsageCalculation( internalPortAudioStream *past ); - -static void Pa_ASIO_Convert_Inter_Input( - ASIOBufferInfo* nativeBuffer, - void* inputBuffer, - long NumInputChannels, - long NumOuputChannels, - long framePerBuffer, - long hostFrameOffset, - long userFrameOffset, - ASIOSampleType nativeFormat, - PaSampleFormat paFormat, - PaStreamFlags flags, - long index); - -static void Pa_ASIO_Convert_Inter_Output( - ASIOBufferInfo* nativeBuffer, - void* outputBuffer, - long NumInputChannels, - long NumOuputChannels, - long framePerBuffer, - long hostFrameOffset, - long userFrameOffset, - ASIOSampleType nativeFormat, - PaSampleFormat paFormat, - PaStreamFlags flags, - long index); - -static void Pa_ASIO_Clear_Output(ASIOBufferInfo* nativeBuffer, - ASIOSampleType nativeFormat, - long NumInputChannels, - long NumOuputChannels, - long index, - long hostFrameOffset, - long frames); - -static void Pa_ASIO_Callback_Input(long index); -static void Pa_ASIO_Callback_Output(long index, long framePerBuffer); -static void Pa_ASIO_Callback_End(); -static void Pa_ASIO_Clear_User_Buffers(); - -// Some external references -extern AsioDrivers* asioDrivers ; -bool loadAsioDriver(char *name); -unsigned long get_sys_reference_time(); - - -/************************************************************************************/ -/****************** Macro ************************************************************/ -/************************************************************************************/ - -#define SwapLong(v) ((((v)>>24)&0xFF)|(((v)>>8)&0xFF00)|(((v)&0xFF00)<<8)|(((v)&0xFF)<<24)) ; -#define SwapShort(v) ((((v)>>8)&0xFF)|(((v)&0xFF)<<8)) ; - -#define ClipShort(v) (((v)<MIN_INT16)?MIN_INT16:(((v)>MAX_INT16)?MAX_INT16:(v))) -#define ClipChar(v) (((v)<MIN_INT8)?MIN_INT8:(((v)>MAX_INT8)?MAX_INT8:(v))) -#define ClipFloat(v) (((v)<-1.0f)?-1.0f:(((v)>1.0f)?1.0f:(v))) - -#ifndef min -#define min(a,b) ((a)<(b)?(a):(b)) -#endif - -#ifndef max -#define max(a,b) ((a)>=(b)?(a):(b)) -#endif - -static bool Pa_ASIO_loadAsioDriver(char *name) -{ - #ifdef WINDOWS - CoInitialize(0); - #endif - return loadAsioDriver(name); -} - - -// Utilities for alignement buffer size computation -static int PGCD (int a, int b) {return (b == 0) ? a : PGCD (b,a%b);} -static int PPCM (int a, int b) {return (a*b) / PGCD (a,b);} - -// Takes the size of host buffer and user buffer : returns the number of frames needed for buffer adaptation -static int Pa_ASIO_CalcFrameShift (int M, int N) -{ - int res = 0; - for (int i = M; i < PPCM (M,N) ; i+=M) { res = max (res, i%N); } - return res; -} - -// We have the following relation : -// Pa_ASIO_CalcFrameShift (M,N) + M = Pa_ASIO_CalcFrameShift (N,M) + N - -/* ASIO sample type to PortAudio sample type conversion */ -static PaSampleFormat Pa_ASIO_Convert_SampleFormat(ASIOSampleType type) -{ - switch (type) { - - case ASIOSTInt16MSB: - case ASIOSTInt16LSB: - case ASIOSTInt32MSB16: - case ASIOSTInt32LSB16: - return paInt16; - - case ASIOSTFloat32MSB: - case ASIOSTFloat32LSB: - case ASIOSTFloat64MSB: - case ASIOSTFloat64LSB: - return paFloat32; - - case ASIOSTInt32MSB: - case ASIOSTInt32LSB: - case ASIOSTInt32MSB18: - case ASIOSTInt32MSB20: - case ASIOSTInt32MSB24: - case ASIOSTInt32LSB18: - case ASIOSTInt32LSB20: - case ASIOSTInt32LSB24: - return paInt32; - - case ASIOSTInt24MSB: - case ASIOSTInt24LSB: - return paInt24; - - default: - return paCustomFormat; - } -} - - - -//-------------------------------------------------------------------------------------------------------------------- -static void PaHost_CalcBufferOffset(internalPortAudioStream *past) -{ - if (asioDriverInfo.past_FramesPerHostBuffer > past->past_FramesPerUserBuffer){ - // Computes the MINIMUM value of null frames shift for the output buffer alignement - asioDriverInfo.pahsc_OutputBufferOffset = Pa_ASIO_CalcFrameShift (asioDriverInfo.past_FramesPerHostBuffer,past->past_FramesPerUserBuffer); - asioDriverInfo.pahsc_InputBufferOffset = 0; - DBUG(("PaHost_CalcBufferOffset : Minimum BufferOffset for Output = %d\n", asioDriverInfo.pahsc_OutputBufferOffset)); - }else{ - - //Computes the MINIMUM value of null frames shift for the input buffer alignement - asioDriverInfo.pahsc_InputBufferOffset = Pa_ASIO_CalcFrameShift (asioDriverInfo.past_FramesPerHostBuffer,past->past_FramesPerUserBuffer); - asioDriverInfo.pahsc_OutputBufferOffset = 0; - DBUG(("PaHost_CalcBufferOffset : Minimum BufferOffset for Input = %d\n", asioDriverInfo.pahsc_InputBufferOffset)); - } -} - -//-------------------------------------------------------------------------------------------------------------------- -/* Allocate ASIO buffers, initialise channels */ -static ASIOError Pa_ASIO_CreateBuffers (PaHostSoundControl *asioDriverInfo, long InputChannels, - long OutputChannels, long framesPerBuffer) -{ - ASIOError err; - int i; - - ASIOBufferInfo *info = asioDriverInfo->bufferInfos; - - // Check parameters - if ((InputChannels > kMaxInputChannels) || (OutputChannels > kMaxOutputChannels)) return ASE_InvalidParameter; - - for(i = 0; i < InputChannels; i++, info++){ - info->isInput = ASIOTrue; - info->channelNum = i; - info->buffers[0] = info->buffers[1] = 0; - } - - for(i = 0; i < OutputChannels; i++, info++){ - info->isInput = ASIOFalse; - info->channelNum = i; - info->buffers[0] = info->buffers[1] = 0; - } - - // Set up the asioCallback structure and create the ASIO data buffer - asioDriverInfo->pahsc_asioCallbacks.bufferSwitch = &bufferSwitch; - asioDriverInfo->pahsc_asioCallbacks.sampleRateDidChange = &sampleRateChanged; - asioDriverInfo->pahsc_asioCallbacks.asioMessage = &asioMessages; - asioDriverInfo->pahsc_asioCallbacks.bufferSwitchTimeInfo = &bufferSwitchTimeInfo; - - DBUG(("Pa_ASIO_CreateBuffers : ASIOCreateBuffers with inputChannels = %ld \n", InputChannels)); - DBUG(("Pa_ASIO_CreateBuffers : ASIOCreateBuffers with OutputChannels = %ld \n", OutputChannels)); - DBUG(("Pa_ASIO_CreateBuffers : ASIOCreateBuffers with size = %ld \n", framesPerBuffer)); - - err = ASIOCreateBuffers( asioDriverInfo->bufferInfos, InputChannels+OutputChannels, - framesPerBuffer, &asioDriverInfo->pahsc_asioCallbacks); - if (err != ASE_OK) return err; - - // Initialise buffers - for (i = 0; i < InputChannels + OutputChannels; i++) - { - asioDriverInfo->pahsc_channelInfos[i].channel = asioDriverInfo->bufferInfos[i].channelNum; - asioDriverInfo->pahsc_channelInfos[i].isInput = asioDriverInfo->bufferInfos[i].isInput; - err = ASIOGetChannelInfo(&asioDriverInfo->pahsc_channelInfos[i]); - if (err != ASE_OK) break; - } - - err = ASIOGetLatencies(&asioDriverInfo->pahsc_inputLatency, &asioDriverInfo->pahsc_outputLatency); - - DBUG(("Pa_ASIO_CreateBuffers : InputLatency = %ld latency = %ld msec \n", - asioDriverInfo->pahsc_inputLatency, - (long)((asioDriverInfo->pahsc_inputLatency*1000)/ asioDriverInfo->past->past_SampleRate))); - DBUG(("Pa_ASIO_CreateBuffers : OuputLatency = %ld latency = %ld msec \n", - asioDriverInfo->pahsc_outputLatency, - (long)((asioDriverInfo->pahsc_outputLatency*1000)/ asioDriverInfo->past->past_SampleRate))); - - return err; -} - - -/* - Query ASIO driver info : - - First we get all available ASIO drivers located in the ASIO folder, - then try to load each one. For each loaded driver, get all needed informations. -*/ -static PaError Pa_ASIO_QueryDeviceInfo( internalPortAudioDevice * ipad ) -{ - -#define NUM_STANDARDSAMPLINGRATES 3 /* 11.025, 22.05, 44.1 */ -#define NUM_CUSTOMSAMPLINGRATES 9 /* must be the same number of elements as in the array below */ -#define MAX_NUMSAMPLINGRATES (NUM_STANDARDSAMPLINGRATES+NUM_CUSTOMSAMPLINGRATES) - - ASIOSampleRate possibleSampleRates[] - = {8000.0, 9600.0, 11025.0, 12000.0, 16000.0, 22050.0, 24000.0, 32000.0, 44100.0, 48000.0, 88200.0, 96000.0}; - - ASIOChannelInfo channelInfos; - long InputChannels,OutputChannels; - double *sampleRates; - char* names[PA_MAX_DEVICE_INFO] ; - PaDeviceInfo *dev; - int i; - int numDrivers; - ASIOError asioError; - - /* Allocate names */ - for (i = 0 ; i < PA_MAX_DEVICE_INFO ; i++) { - names[i] = (char*)PaHost_AllocateFastMemory(32); - /* check memory */ - if(!names[i]) return paInsufficientMemory; - } - - /* MUST BE CHECKED : to force fragments loading on Mac */ - Pa_ASIO_loadAsioDriver("dummy"); - - /* Get names of all available ASIO drivers */ - asioDrivers->getDriverNames(names,PA_MAX_DEVICE_INFO); - - /* Check all available ASIO drivers */ -#if MAC - numDrivers = asioDrivers->getNumFragments(); -#elif WINDOWS - numDrivers = asioDrivers->asioGetNumDev(); -#endif - - DBUG(("PaASIO_QueryDeviceInfo: number of installed drivers = %d\n", numDrivers )); - - for (int driver = 0 ; driver < numDrivers ; driver++) - { - - #if WINDOWS - asioDriverInfo.pahsc_driverInfo.asioVersion = 2; // FIXME - is this right? PLB - asioDriverInfo.pahsc_driverInfo.sysRef = GetDesktopWindow(); // FIXME - is this right? PLB - #endif - - DBUG(("---------------------------------------\n")); - - DBUG(("PaASIO_QueryDeviceInfo: Driver name = %s\n", names[driver])); - - /* If the driver can be loaded : */ - if ( !Pa_ASIO_loadAsioDriver(names[driver]) ){ - DBUG(("PaASIO_QueryDeviceInfo could not loadAsioDriver %s\n", names[driver])); - } else { - - DBUG(("PaASIO_QueryDeviceInfo: loadAsioDriver OK\n")); - - if((asioError = ASIOInit(&asioDriverInfo.pahsc_driverInfo)) != ASE_OK){ - - DBUG(("PaASIO_QueryDeviceInfo: ASIOInit returned %d for %s\n", asioError, names[driver])); - - }else { - - DBUG(("PaASIO_QueryDeviceInfo: ASIOInit OK \n")); - - if(ASIOGetChannels(&InputChannels, &OutputChannels) != ASE_OK){ - - DBUG(("PaASIO_QueryDeviceInfo could not ASIOGetChannels for %s\n", names[driver])); - - }else { - - DBUG(("PaASIO_QueryDeviceInfo: ASIOGetChannels OK \n")); - - /* Gets the name */ - dev = &(ipad[sNumDevices].pad_Info); - dev->name = names[driver]; - names[driver] = 0; - - /* Gets Input and Output channels number */ - dev->maxInputChannels = InputChannels; - dev->maxOutputChannels = OutputChannels; - - DBUG(("PaASIO_QueryDeviceInfo: InputChannels = %d\n", InputChannels )); - DBUG(("PaASIO_QueryDeviceInfo: OutputChannels = %d\n", OutputChannels )); - - /* Make room in case device supports all rates. */ - sampleRates = (double*)PaHost_AllocateFastMemory(MAX_NUMSAMPLINGRATES * sizeof(double)); - /* check memory */ - if (!sampleRates) { - ASIOExit(); - return paInsufficientMemory; - } - dev->sampleRates = sampleRates; - dev->numSampleRates = 0; - - /* Loop through the possible sampling rates and check each to see if the device supports it. */ - for (int index = 0; index < MAX_NUMSAMPLINGRATES; index++) { - if (ASIOCanSampleRate(possibleSampleRates[index]) != ASE_NoClock) { - DBUG(("PaASIO_QueryDeviceInfo: possible sample rate = %d\n", (long)possibleSampleRates[index])); - dev->numSampleRates += 1; - *sampleRates = possibleSampleRates[index]; - sampleRates++; - } - } - - /* We assume that all channels have the same SampleType, so check the first */ - channelInfos.channel = 0; - channelInfos.isInput = 1; - - if ((asioError = ASIOGetChannelInfo(&channelInfos)) == ASE_NotPresent) { - DBUG(("PaASIO_QueryDeviceInfo: ASIOGetChannelInfo returned %d \n",asioError)); - } - - dev->nativeSampleFormats = Pa_ASIO_Convert_SampleFormat(channelInfos.type); - - /* unload the driver */ - if ((asioError = ASIOExit()) != ASE_OK) { - DBUG(("PaASIO_QueryDeviceInfo: ASIOExit returned %d \n",asioError)); - } - - sNumDevices++; - } - } - } - } - - /* free only unused names */ - for (i = 0 ; i < PA_MAX_DEVICE_INFO ; i++) if (names[i]) PaHost_FreeFastMemory(names[i],32); - - return paNoError; -} - -//---------------------------------------------------------------------------------- -// TAKEN FROM THE ASIO SDK: -void sampleRateChanged(ASIOSampleRate sRate) -{ - // do whatever you need to do if the sample rate changed - // usually this only happens during external sync. - // Audio processing is not stopped by the driver, actual sample rate - // might not have even changed, maybe only the sample rate status of an - // AES/EBU or S/PDIF digital input at the audio device. - // You might have to update time/sample related conversion routines, etc. -} - -//---------------------------------------------------------------------------------- -// TAKEN FROM THE ASIO SDK: -long asioMessages(long selector, long value, void* message, double* opt) -{ - // currently the parameters "value", "message" and "opt" are not used. - long ret = 0; - switch(selector) - { - case kAsioSelectorSupported: - if(value == kAsioResetRequest - || value == kAsioEngineVersion - || value == kAsioResyncRequest - || value == kAsioLatenciesChanged - // the following three were added for ASIO 2.0, you don't necessarily have to support them - || value == kAsioSupportsTimeInfo - || value == kAsioSupportsTimeCode - || value == kAsioSupportsInputMonitor) - ret = 1L; - break; - - case kAsioBufferSizeChange: - //printf("kAsioBufferSizeChange \n"); - break; - - case kAsioResetRequest: - // defer the task and perform the reset of the driver during the next "safe" situation - // You cannot reset the driver right now, as this code is called from the driver. - // Reset the driver is done by completely destruct is. I.e. ASIOStop(), ASIODisposeBuffers(), Destruction - // Afterwards you initialize the driver again. - asioDriverInfo.stopped; // In this sample the processing will just stop - ret = 1L; - break; - case kAsioResyncRequest: - // This informs the application, that the driver encountered some non fatal data loss. - // It is used for synchronization purposes of different media. - // Added mainly to work around the Win16Mutex problems in Windows 95/98 with the - // Windows Multimedia system, which could loose data because the Mutex was hold too long - // by another thread. - // However a driver can issue it in other situations, too. - ret = 1L; - break; - case kAsioLatenciesChanged: - // This will inform the host application that the drivers were latencies changed. - // Beware, it this does not mean that the buffer sizes have changed! - // You might need to update internal delay data. - ret = 1L; - //printf("kAsioLatenciesChanged \n"); - break; - case kAsioEngineVersion: - // return the supported ASIO version of the host application - // If a host applications does not implement this selector, ASIO 1.0 is assumed - // by the driver - ret = 2L; - break; - case kAsioSupportsTimeInfo: - // informs the driver wether the asioCallbacks.bufferSwitchTimeInfo() callback - // is supported. - // For compatibility with ASIO 1.0 drivers the host application should always support - // the "old" bufferSwitch method, too. - ret = 1; - break; - case kAsioSupportsTimeCode: - // informs the driver wether application is interested in time code info. - // If an application does not need to know about time code, the driver has less work - // to do. - ret = 0; - break; - } - return ret; -} - -//---------------------------------------------------------------------------------- -// Atomic increment and decrement operations -#if MAC - /* need to be implemented on Mac */ - inline long Pa_AtomicIncrement(long* v) {return ++(*v);} - inline long Pa_AtomicDecrement(long* v) {return --(*v);} -#elif WINDOWS - inline long Pa_AtomicIncrement(long* v) {return InterlockedIncrement(v);} - inline long Pa_AtomicDecrement(long* v) {return InterlockedDecrement(v);} -#endif - - -//---------------------------------------------------------------------------------- -// conversion from 64 bit ASIOSample/ASIOTimeStamp to double float -#if NATIVE_INT64 - #define ASIO64toDouble(a) (a) -#else - const double twoRaisedTo32 = 4294967296.; - #define ASIO64toDouble(a) ((a).lo + (a).hi * twoRaisedTo32) -#endif - - -static ASIOTime *bufferSwitchTimeInfo(ASIOTime *timeInfo, long index, ASIOBool processNow) -{ - // the actual processing callback. - // Beware that this is normally in a seperate thread, hence be sure that you take care - // about thread synchronization. This is omitted here for simplicity. - - // store the timeInfo for later use - asioDriverInfo.tInfo = *timeInfo; - - // get the time stamp of the buffer, not necessary if no - // synchronization to other media is required - - if (timeInfo->timeInfo.flags & kSystemTimeValid) - asioDriverInfo.nanoSeconds = ASIO64toDouble(timeInfo->timeInfo.systemTime); - else - asioDriverInfo.nanoSeconds = 0; - - if (timeInfo->timeInfo.flags & kSamplePositionValid) - asioDriverInfo.samples = ASIO64toDouble(timeInfo->timeInfo.samplePosition); - else - asioDriverInfo.samples = 0; - - if (timeInfo->timeCode.flags & kTcValid) - asioDriverInfo.tcSamples = ASIO64toDouble(timeInfo->timeCode.timeCodeSamples); - else - asioDriverInfo.tcSamples = 0; - - // get the system reference time - asioDriverInfo.sysRefTime = get_sys_reference_time(); - -#if 0 - // a few debug messages for the Windows device driver developer - // tells you the time when driver got its interrupt and the delay until the app receives - // the event notification. - static double last_samples = 0; - char tmp[128]; - sprintf (tmp, "diff: %d / %d ms / %d ms / %d samples \n", asioDriverInfo.sysRefTime - (long)(asioDriverInfo.nanoSeconds / 1000000.0), asioDriverInfo.sysRefTime, (long)(asioDriverInfo.nanoSeconds / 1000000.0), (long)(asioDriverInfo.samples - last_samples)); - OutputDebugString (tmp); - last_samples = asioDriverInfo.samples; -#endif - - // To avoid the callback accessing a desallocated stream - if (asioDriverInfo.past == NULL) return 0L; - - // Keep sample position - asioDriverInfo.pahsc_NumFramesDone = timeInfo->timeInfo.samplePosition.lo; - - // Reentrancy control - if(Pa_AtomicIncrement(&asioDriverInfo.reenterCount)) { - asioDriverInfo.reenterError++; - DBUG(("bufferSwitchTimeInfo : reentrancy detection = %d\n", asioDriverInfo.reenterError)); - return 0L; - } - - do { - - /* Has a user callback returned '1' to indicate finished at the last ASIO callback? */ - if( asioDriverInfo.past->past_StopSoon ) { - - Pa_ASIO_Clear_Output(asioDriverInfo.bufferInfos, - asioDriverInfo.pahsc_channelInfos[0].type, - asioDriverInfo.pahsc_NumInputChannels , - asioDriverInfo.pahsc_NumOutputChannels, - index, - 0, - asioDriverInfo.past_FramesPerHostBuffer); - - asioDriverInfo.past->past_IsActive = 0; - - // Finally if the driver supports the ASIOOutputReady() optimization, do it here, all data are in place - if (asioDriverInfo.pahsc_postOutput) ASIOOutputReady(); - - }else{ - - /* CPU usage */ - Pa_StartUsageCalculation(asioDriverInfo.past); - - Pa_ASIO_Callback_Input(index); - - // Finally if the driver supports the ASIOOutputReady() optimization, do it here, all data are in place - if (asioDriverInfo.pahsc_postOutput) ASIOOutputReady(); - - Pa_ASIO_Callback_End(); - - /* CPU usage */ - Pa_EndUsageCalculation(asioDriverInfo.past); - } - - } while(Pa_AtomicDecrement(&asioDriverInfo.reenterCount) >= 0); - - return 0L; -} - - -//---------------------------------------------------------------------------------- -void bufferSwitch(long index, ASIOBool processNow) -{ - // the actual processing callback. - // Beware that this is normally in a seperate thread, hence be sure that you take care - // about thread synchronization. This is omitted here for simplicity. - - // as this is a "back door" into the bufferSwitchTimeInfo a timeInfo needs to be created - // though it will only set the timeInfo.samplePosition and timeInfo.systemTime fields and the according flags - - ASIOTime timeInfo; - memset (&timeInfo, 0, sizeof (timeInfo)); - - // get the time stamp of the buffer, not necessary if no - // synchronization to other media is required - if(ASIOGetSamplePosition(&timeInfo.timeInfo.samplePosition, &timeInfo.timeInfo.systemTime) == ASE_OK) - timeInfo.timeInfo.flags = kSystemTimeValid | kSamplePositionValid; - - // Call the real callback - bufferSwitchTimeInfo (&timeInfo, index, processNow); -} - -//---------------------------------------------------------------------------------- -unsigned long get_sys_reference_time() -{ - // get the system reference time - #if WINDOWS - return timeGetTime(); - #elif MAC - static const double twoRaisedTo32 = 4294967296.; - UnsignedWide ys; - Microseconds(&ys); - double r = ((double)ys.hi * twoRaisedTo32 + (double)ys.lo); - return (unsigned long)(r / 1000.); - #endif -} - - -/************************************************************* -** Calculate 2 LSB dither signal with a triangular distribution. -** Ranged properly for adding to a 32 bit integer prior to >>15. -*/ -#define DITHER_BITS (15) -#define DITHER_SCALE (1.0f / ((1<<DITHER_BITS)-1)) -inline static long Pa_TriangularDither( void ) -{ - static unsigned long previous = 0; - static unsigned long randSeed1 = 22222; - static unsigned long randSeed2 = 5555555; - long current, highPass; -/* Generate two random numbers. */ - randSeed1 = (randSeed1 * 196314165) + 907633515; - randSeed2 = (randSeed2 * 196314165) + 907633515; -/* Generate triangular distribution about 0. */ - current = (((long)randSeed1)>>(32-DITHER_BITS)) + (((long)randSeed2)>>(32-DITHER_BITS)); - /* High pass filter to reduce audibility. */ - highPass = current - previous; - previous = current; - return highPass; -} - -// TO BE COMPLETED WITH ALL SUPPORTED PA SAMPLE TYPES - -//------------------------------------------------------------------------------------------------------------------------------------------------------- -static void Input_Int16_Float32 (ASIOBufferInfo* nativeBuffer, float *inBufPtr, int framePerBuffer, int NumInputChannels, int index, int hostFrameOffset,int userFrameOffset, bool swap) -{ - long temp; - int i,j; - - for( j=0; j<NumInputChannels; j++ ) { - short *asioBufPtr = &((short*)nativeBuffer[j].buffers[index])[hostFrameOffset]; - float *userBufPtr = &inBufPtr[j+(userFrameOffset*NumInputChannels)]; - for (i= 0; i < framePerBuffer; i++) - { - temp = asioBufPtr[i]; - if (swap) temp = SwapShort(temp); - *userBufPtr = (1.0f / MAX_INT16_FP) * temp; - userBufPtr += NumInputChannels; - } - } - -} - -//------------------------------------------------------------------------------------------------------------------------------------------------------- -static void Input_Int32_Float32 (ASIOBufferInfo* nativeBuffer, float *inBufPtr, int framePerBuffer, int NumInputChannels, int index, int hostFrameOffset,int userFrameOffset,bool swap) -{ - long temp; - int i,j; - - for( j=0; j<NumInputChannels; j++ ) { - long *asioBufPtr = &((long*)nativeBuffer[j].buffers[index])[hostFrameOffset]; - float *userBufPtr = &inBufPtr[j+(userFrameOffset*NumInputChannels)]; - for (i= 0; i < framePerBuffer; i++) - { - temp = asioBufPtr[i]; - if (swap) temp = SwapLong(temp); - *userBufPtr = (1.0f / MAX_INT32_FP) * temp; - userBufPtr += NumInputChannels; - } - } -} - -//------------------------------------------------------------------------------------------------------------------------------------------------------- -// MUST BE TESTED -static void Input_Float32_Float32 (ASIOBufferInfo* nativeBuffer, float *inBufPtr, int framePerBuffer, int NumInputChannels, int index, int hostFrameOffset,int userFrameOffset,bool swap) -{ - unsigned long temp; - int i,j; - - for( j=0; j<NumInputChannels; j++ ) { - unsigned long *asioBufPtr = &((unsigned long*)nativeBuffer[j].buffers[index])[hostFrameOffset]; - float *userBufPtr = &inBufPtr[j+(userFrameOffset*NumInputChannels)]; - for (i= 0; i < framePerBuffer; i++) - { - temp = asioBufPtr[i]; - if (swap) temp = SwapLong(temp); - *userBufPtr = (float)temp; - userBufPtr += NumInputChannels; - } - } -} - -//------------------------------------------------------------------------------------------------------------------------------------------------------- -static void Input_Int16_Int32 (ASIOBufferInfo* nativeBuffer, long *inBufPtr, int framePerBuffer, int NumInputChannels, int index, int hostFrameOffset,int userFrameOffset,bool swap) -{ - long temp; - int i,j; - - for( j=0; j<NumInputChannels; j++ ) { - short *asioBufPtr = &((short*)nativeBuffer[j].buffers[index])[hostFrameOffset]; - long *userBufPtr = &inBufPtr[j+(userFrameOffset*NumInputChannels)]; - for (i= 0; i < framePerBuffer; i++) - { - temp = asioBufPtr[i]; - if (swap) temp = SwapShort(temp); - *userBufPtr = temp<<16; - userBufPtr += NumInputChannels; - } - } -} - -//------------------------------------------------------------------------------------------------------------------------------------------------------- -static void Input_Int32_Int32 (ASIOBufferInfo* nativeBuffer, long *inBufPtr, int framePerBuffer, int NumInputChannels, int index, int hostFrameOffset,int userFrameOffset,bool swap) -{ - long temp; - int i,j; - - for( j=0; j<NumInputChannels; j++ ) { - long *asioBufPtr = &((long*)nativeBuffer[j].buffers[index])[hostFrameOffset]; - long *userBufPtr = &inBufPtr[j+(userFrameOffset*NumInputChannels)]; - for (i= 0; i < framePerBuffer; i++) - { - temp = asioBufPtr[i]; - if (swap) temp = SwapLong(temp); - *userBufPtr = temp; - userBufPtr += NumInputChannels; - } - } -} - -//------------------------------------------------------------------------------------------------------------------------------------------------------- -// MUST BE TESTED -static void Input_Float32_Int32 (ASIOBufferInfo* nativeBuffer, long *inBufPtr, int framePerBuffer, int NumInputChannels, int index, int hostFrameOffset,int userFrameOffset,bool swap) -{ - unsigned long temp; - int i,j; - - for( j=0; j<NumInputChannels; j++ ) { - unsigned long *asioBufPtr = &((unsigned long*)nativeBuffer[j].buffers[index])[hostFrameOffset]; - long *userBufPtr = &inBufPtr[j+(userFrameOffset*NumInputChannels)]; - for (i= 0; i < framePerBuffer; i++) - { - temp = asioBufPtr[i]; - if (swap) temp = SwapLong(temp); - *userBufPtr = (long)((float)temp * MAX_INT32_FP); // Is temp a value between -1.0 and 1.0 ?? - userBufPtr += NumInputChannels; - } - } -} - - -//------------------------------------------------------------------------------------------------------------------------------------------------------- -static void Input_Int16_Int16 (ASIOBufferInfo* nativeBuffer, short *inBufPtr, int framePerBuffer, int NumInputChannels, int index, int hostFrameOffset,int userFrameOffset,bool swap) -{ - long temp; - int i,j; - - for( j=0; j<NumInputChannels; j++ ) { - short *asioBufPtr = &((short*)nativeBuffer[j].buffers[index])[hostFrameOffset]; - short *userBufPtr = &inBufPtr[j+(userFrameOffset*NumInputChannels)]; - for (i= 0; i < framePerBuffer; i++) - { - temp = asioBufPtr[i]; - if (swap) temp = SwapShort(temp); - *userBufPtr = (short)temp; - userBufPtr += NumInputChannels; - } - } -} - - //------------------------------------------------------------------------------------------------------------------------------------------------------- -static void Input_Int32_Int16 (ASIOBufferInfo* nativeBuffer, short *inBufPtr, int framePerBuffer, int NumInputChannels, int index, int hostFrameOffset, int userFrameOffset,uint32 flags,bool swap) -{ - long temp; - int i,j; - - if( flags & paDitherOff ) - { - for( j=0; j<NumInputChannels; j++ ) { - long *asioBufPtr = &((long*)nativeBuffer[j].buffers[index])[hostFrameOffset]; - short *userBufPtr = &inBufPtr[j+(userFrameOffset*NumInputChannels)]; - for (i= 0; i < framePerBuffer; i++) - { - temp = asioBufPtr[i]; - if (swap) temp = SwapLong(temp); - *userBufPtr = (short)(temp>>16); - userBufPtr += NumInputChannels; - } - } - } - else - { - for( j=0; j<NumInputChannels; j++ ) { - long *asioBufPtr = &((long*)nativeBuffer[j].buffers[index])[hostFrameOffset]; - short *userBufPtr = &inBufPtr[j+(userFrameOffset*NumInputChannels)]; - for (i= 0; i < framePerBuffer; i++) - { - temp = asioBufPtr[i]; - if (swap) temp = SwapLong(temp); - temp = (temp >> 1) + Pa_TriangularDither(); - temp = temp >> 15; - temp = (short) ClipShort(temp); - *userBufPtr = (short)temp; - userBufPtr += NumInputChannels; - } - } - - } -} - -//------------------------------------------------------------------------------------------------------------------------------------------------------- -// MUST BE TESTED -static void Input_Float32_Int16 (ASIOBufferInfo* nativeBuffer, short *inBufPtr, int framePerBuffer, int NumInputChannels, int index, int hostFrameOffset,int userFrameOffset,uint32 flags,bool swap) -{ - unsigned long temp; - int i,j; - - if( flags & paDitherOff ) - { - for( j=0; j<NumInputChannels; j++ ) { - unsigned long *asioBufPtr = &((unsigned long*)nativeBuffer[j].buffers[index])[hostFrameOffset]; - short *userBufPtr = &inBufPtr[j+(userFrameOffset*NumInputChannels)]; - for (i= 0; i < framePerBuffer; i++) - { - temp = asioBufPtr[i]; - if (swap) temp = SwapLong(temp); - *userBufPtr = (short)((float)temp * MAX_INT16_FP); // Is temp a value between -1.0 and 1.0 ?? - userBufPtr += NumInputChannels; - } - } - } - else - { - for( j=0; j<NumInputChannels; j++ ) { - unsigned long *asioBufPtr = &((unsigned long*)nativeBuffer[j].buffers[index])[hostFrameOffset]; - short *userBufPtr = &inBufPtr[j+(userFrameOffset*NumInputChannels)]; - for (i= 0; i < framePerBuffer; i++) - { - float dither = Pa_TriangularDither()*DITHER_SCALE; - temp = asioBufPtr[i]; - if (swap) temp = SwapLong(temp); - temp = (short)(((float)temp * MAX_INT16_FP) + dither); - temp = ClipShort(temp); - *userBufPtr = (short)temp; - userBufPtr += NumInputChannels; - } - } - } -} - -//------------------------------------------------------------------------------------------------------------------------------------------------------- -static void Input_Int16_Int8 (ASIOBufferInfo* nativeBuffer, char *inBufPtr, int framePerBuffer, int NumInputChannels, int index, int hostFrameOffset,int userFrameOffset, uint32 flags,bool swap) -{ - long temp; - int i,j; - - if( flags & paDitherOff ) - { - for( j=0; j<NumInputChannels; j++ ) { - short *asioBufPtr = &((short*)nativeBuffer[j].buffers[index])[hostFrameOffset]; - char *userBufPtr = &inBufPtr[j+(userFrameOffset*NumInputChannels)]; - for (i= 0; i < framePerBuffer; i++) - { - temp = asioBufPtr[i]; - if (swap) temp = SwapShort(temp); - *userBufPtr = (char)(temp>>8); - userBufPtr += NumInputChannels; - } - } - } - else - { - for( j=0; j<NumInputChannels; j++ ) { - short *asioBufPtr = &((short*)nativeBuffer[j].buffers[index])[hostFrameOffset]; - char *userBufPtr = &inBufPtr[j+(userFrameOffset*NumInputChannels)]; - for (i= 0; i < framePerBuffer; i++) - { - temp = asioBufPtr[i]; - if (swap) temp = SwapShort(temp); - temp += Pa_TriangularDither() >> 8; - temp = ClipShort(temp); - *userBufPtr = (char)(temp>>8); - userBufPtr += NumInputChannels; - } - } - } -} - -//------------------------------------------------------------------------------------------------------------------------------------------------------- -static void Input_Int32_Int8 (ASIOBufferInfo* nativeBuffer, char *inBufPtr, int framePerBuffer, int NumInputChannels, int index, int hostFrameOffset, int userFrameOffset, uint32 flags,bool swap) -{ - long temp; - int i,j; - - if( flags & paDitherOff ) - { - for( j=0; j<NumInputChannels; j++ ) { - long *asioBufPtr = &((long*)nativeBuffer[j].buffers[index])[hostFrameOffset]; - char *userBufPtr = &inBufPtr[j+(userFrameOffset*NumInputChannels)]; - for (i= 0; i < framePerBuffer; i++) - { - temp = asioBufPtr[i]; - if (swap) temp = SwapLong(temp); - *userBufPtr = (char)(temp>>24); - userBufPtr += NumInputChannels; - } - } - } - else - { - for( j=0; j<NumInputChannels; j++ ) { - long *asioBufPtr = &((long*)nativeBuffer[j].buffers[index])[hostFrameOffset]; - char *userBufPtr = &inBufPtr[j+(userFrameOffset*NumInputChannels)]; - for (i= 0; i < framePerBuffer; i++) - { - temp = asioBufPtr[i]; - if (swap) temp = SwapLong(temp); - temp = temp>>16; // Shift to get a 16 bit value, then use the 16 bits to 8 bits code (MUST BE CHECHED) - temp += Pa_TriangularDither() >> 8; - temp = ClipShort(temp); - *userBufPtr = (char)(temp >> 8); - userBufPtr += NumInputChannels; - } - } - } -} - -//------------------------------------------------------------------------------------------------------------------------------------------------------- -// MUST BE TESTED - -static void Input_Float32_Int8 (ASIOBufferInfo* nativeBuffer, char *inBufPtr, int framePerBuffer, int NumInputChannels, int index, int hostFrameOffset,int userFrameOffset, uint32 flags,bool swap) -{ - unsigned long temp; - int i,j; - - if( flags & paDitherOff ) - { - for( j=0; j<NumInputChannels; j++ ) { - unsigned long *asioBufPtr = &((unsigned long*)nativeBuffer[j].buffers[index])[hostFrameOffset]; - char *userBufPtr = &inBufPtr[j+(userFrameOffset*NumInputChannels)]; - for (i= 0; i < framePerBuffer; i++) - { - temp = asioBufPtr[i]; - if (swap) temp = SwapLong(temp); - *userBufPtr = (char)((float)temp*MAX_INT8_FP); // Is temp a value between -1.0 and 1.0 ?? - userBufPtr += NumInputChannels; - } - } - } - else - { - for( j=0; j<NumInputChannels; j++ ) { - unsigned long *asioBufPtr = &((unsigned long*)nativeBuffer[j].buffers[index])[hostFrameOffset]; - char *userBufPtr = &inBufPtr[j+(userFrameOffset*NumInputChannels)]; - for (i= 0; i < framePerBuffer; i++) - { - float dither = Pa_TriangularDither()*DITHER_SCALE; - temp = asioBufPtr[i]; - if (swap) temp = SwapLong(temp); - temp = (char)(((float)temp * MAX_INT8_FP) + dither); - temp = ClipChar(temp); - *userBufPtr = (char)temp; - userBufPtr += NumInputChannels; - } - } - } -} - -//------------------------------------------------------------------------------------------------------------------------------------------------------- -static void Input_Int16_IntU8 (ASIOBufferInfo* nativeBuffer, unsigned char *inBufPtr, int framePerBuffer, int NumInputChannels, int index, int hostFrameOffset,int userFrameOffset, uint32 flags,bool swap) -{ - long temp; - int i,j; - - if( flags & paDitherOff ) - { - for( j=0; j<NumInputChannels; j++ ) { - short *asioBufPtr = &((short*)nativeBuffer[j].buffers[index])[hostFrameOffset]; - unsigned char *userBufPtr = &inBufPtr[j+(userFrameOffset*NumInputChannels)]; - for (i= 0; i < framePerBuffer; i++) - { - temp = asioBufPtr[i]; - if (swap) temp = SwapShort(temp); - *userBufPtr = (unsigned char)((temp>>8) + 0x80); - userBufPtr += NumInputChannels; - } - } - } - else - { - for( j=0; j<NumInputChannels; j++ ) { - short *asioBufPtr = &((short*)nativeBuffer[j].buffers[index])[hostFrameOffset]; - unsigned char *userBufPtr = &inBufPtr[j+(userFrameOffset*NumInputChannels)]; - for (i= 0; i < framePerBuffer; i++) - { - temp = asioBufPtr[i]; - if (swap) temp = SwapShort(temp); - temp += Pa_TriangularDither() >> 8; - temp = ClipShort(temp); - *userBufPtr = (unsigned char)((temp>>8) + 0x80); - userBufPtr += NumInputChannels; - } - } - } -} - -//------------------------------------------------------------------------------------------------------------------------------------------------------- -static void Input_Int32_IntU8 (ASIOBufferInfo* nativeBuffer, unsigned char *inBufPtr, int framePerBuffer, int NumInputChannels, int index, int hostFrameOffset, int userFrameOffset,uint32 flags,bool swap) -{ - long temp; - int i,j; - - if( flags & paDitherOff ) - { - for( j=0; j<NumInputChannels; j++ ) { - long *asioBufPtr = &((long*)nativeBuffer[j].buffers[index])[hostFrameOffset]; - unsigned char *userBufPtr = &inBufPtr[j+(userFrameOffset*NumInputChannels)]; - for (i= 0; i < framePerBuffer; i++) - { - temp = asioBufPtr[i]; - if (swap) temp = SwapLong(temp); - *userBufPtr = (unsigned char)((temp>>24) + 0x80); - userBufPtr += NumInputChannels; - } - } - } - else - { - for( j=0; j<NumInputChannels; j++ ) { - long *asioBufPtr = &((long*)nativeBuffer[j].buffers[index])[hostFrameOffset]; - unsigned char *userBufPtr = &inBufPtr[j+(userFrameOffset*NumInputChannels)]; - for (i= 0; i < framePerBuffer; i++) - { - temp = asioBufPtr[i]; - if (swap) temp = SwapLong(temp); - temp = temp>>16; // Shift to get a 16 bit value, then use the 16 bits to 8 bits code (MUST BE CHECHED) - temp += Pa_TriangularDither() >> 8; - temp = ClipShort(temp); - *userBufPtr = (unsigned char)((temp>>8) + 0x80); - userBufPtr += NumInputChannels; - } - } - } -} - -//------------------------------------------------------------------------------------------------------------------------------------------------------- -// MUST BE TESTED - -static void Input_Float32_IntU8 (ASIOBufferInfo* nativeBuffer, unsigned char *inBufPtr, int framePerBuffer, int NumInputChannels, int index, int hostFrameOffset,int userFrameOffset, uint32 flags,bool swap) -{ - unsigned long temp; - int i,j; - - if( flags & paDitherOff ) - { - for( j=0; j<NumInputChannels; j++ ) { - unsigned long *asioBufPtr = &((unsigned long*)nativeBuffer[j].buffers[index])[hostFrameOffset]; - unsigned char *userBufPtr = &inBufPtr[j+(userFrameOffset*NumInputChannels)]; - for (i= 0; i < framePerBuffer; i++) - { - temp = asioBufPtr[i]; - if (swap) temp = SwapLong(temp); - *userBufPtr = (unsigned char)(((float)temp*MAX_INT8_FP) + 0x80); - userBufPtr += NumInputChannels; - } - } - } - else - { - for( j=0; j<NumInputChannels; j++ ) { - unsigned long *asioBufPtr = &((unsigned long*)nativeBuffer[j].buffers[index])[hostFrameOffset]; - unsigned char *userBufPtr = &inBufPtr[j+(userFrameOffset*NumInputChannels)]; - for (i= 0; i < framePerBuffer; i++) - { - float dither = Pa_TriangularDither()*DITHER_SCALE; - temp = asioBufPtr[i]; - if (swap) temp = SwapLong(temp); - temp = (char)(((float)temp * MAX_INT8_FP) + dither); - temp = ClipChar(temp); - *userBufPtr = (unsigned char)(temp + 0x80); - userBufPtr += NumInputChannels; - } - } - } -} - - -// OUPUT -//------------------------------------------------------------------------------------------------------------------------------------------------------- -static void Output_Float32_Int16 (ASIOBufferInfo* nativeBuffer, float *outBufPtr, int framePerBuffer, int NumInputChannels, int NumOuputChannels, int index, int hostFrameOffset, int userFrameOffset,uint32 flags, bool swap) -{ - long temp; - int i,j; - - if( flags & paDitherOff ) - { - if( flags & paClipOff ) /* NOTHING */ - { - for( j=0; j<NumOuputChannels; j++ ) { - short *asioBufPtr = &((short*)nativeBuffer[j+NumInputChannels].buffers[index])[hostFrameOffset]; - float *userBufPtr = &outBufPtr[j+(userFrameOffset*NumOuputChannels)]; - - for (i= 0; i < framePerBuffer; i++) - { - temp = (short) (*userBufPtr * MAX_INT16_FP); - if (swap) temp = SwapShort(temp); - asioBufPtr[i] = (short)temp; - userBufPtr += NumOuputChannels; - } - } - } - else /* CLIP */ - { - for( j=0; j<NumOuputChannels; j++ ) { - short *asioBufPtr = &((short*)nativeBuffer[j+NumInputChannels].buffers[index])[hostFrameOffset]; - float *userBufPtr = &outBufPtr[j+(userFrameOffset*NumOuputChannels)]; - - for (i= 0; i < framePerBuffer; i++) - { - temp = (long) (*userBufPtr * MAX_INT16_FP); - temp = ClipShort(temp); - if (swap) temp = SwapShort(temp); - asioBufPtr[i] = (short)temp; - userBufPtr += NumOuputChannels; - } - } - } - } - else - { - /* If you dither then you have to clip because dithering could push the signal out of range! */ - for( j=0; j<NumOuputChannels; j++ ) { - short *asioBufPtr = &((short*)nativeBuffer[j+NumInputChannels].buffers[index])[hostFrameOffset]; - float *userBufPtr = &outBufPtr[j+(userFrameOffset*NumOuputChannels)]; - - for (i= 0; i < framePerBuffer; i++) - { - float dither = Pa_TriangularDither()*DITHER_SCALE; - temp = (long) ((*userBufPtr * MAX_INT16_FP) + dither); - temp = ClipShort(temp); - if (swap) temp = SwapShort(temp); - asioBufPtr[i] = (short)temp; - userBufPtr += NumOuputChannels; - } - } - } -} - -//------------------------------------------------------------------------------------------------------------------------------------------------------- -static void Output_Float32_Int32 (ASIOBufferInfo* nativeBuffer, float *outBufPtr, int framePerBuffer, int NumInputChannels, int NumOuputChannels, int index, int hostFrameOffset, int userFrameOffset,uint32 flags,bool swap) -{ - long temp; - int i,j; - - if( flags & paClipOff ) - { - for (j= 0; j < NumOuputChannels; j++) - { - long *asioBufPtr = &((long*)nativeBuffer[j+NumInputChannels].buffers[index])[hostFrameOffset]; - float *userBufPtr = &outBufPtr[j+(userFrameOffset*NumOuputChannels)]; - for( i=0; i<framePerBuffer; i++ ) - { - temp = (long) (*userBufPtr * MAX_INT32_FP); - if (swap) temp = SwapLong(temp); - asioBufPtr[i] = temp; - userBufPtr += NumOuputChannels; - } - } - } - else // CLIP * - { - for (j= 0; j < NumOuputChannels; j++) - { - long *asioBufPtr = &((long*)nativeBuffer[j+NumInputChannels].buffers[index])[hostFrameOffset]; - float *userBufPtr = &outBufPtr[j+(userFrameOffset*NumOuputChannels)]; - for( i=0; i<framePerBuffer; i++ ) - { - float temp1 = *userBufPtr; - temp1 = ClipFloat(temp1); - temp = (long) (temp1*MAX_INT32_FP); - if (swap) temp = SwapLong(temp); - asioBufPtr[i] = temp; - userBufPtr += NumOuputChannels; - } - } - } - -} - - -//------------------------------------------------------------------------------------------------------------------------------------------------------- -// MUST BE TESTED - - static void Output_Float32_Float32 (ASIOBufferInfo* nativeBuffer, float *outBufPtr, int framePerBuffer, int NumInputChannels, int NumOuputChannels, int index, int hostFrameOffset, int userFrameOffset,uint32 flags,bool swap) -{ - long temp; - int i,j; - - if( flags & paClipOff ) - { - for (j= 0; j < NumOuputChannels; j++) - { - float *asioBufPtr = &((float*)nativeBuffer[j+NumInputChannels].buffers[index])[hostFrameOffset]; - float *userBufPtr = &outBufPtr[j+(userFrameOffset*NumOuputChannels)]; - for( i=0; i<framePerBuffer; i++ ) - { - temp = (long) *userBufPtr; - if (swap) temp = SwapLong(temp); - asioBufPtr[i] = (float)temp; - userBufPtr += NumOuputChannels; - } - } - - } - else /* CLIP */ - { - for (j= 0; j < NumOuputChannels; j++) - { - float *asioBufPtr = &((float*)nativeBuffer[j+NumInputChannels].buffers[index])[hostFrameOffset]; - float *userBufPtr = &outBufPtr[j+(userFrameOffset*NumOuputChannels)]; - for( i=0; i<framePerBuffer; i++ ) - { - float temp1 = *userBufPtr; - temp1 = ClipFloat(temp1); // Is is necessary?? - temp = (long) temp1; - if (swap) temp = SwapLong(temp); - asioBufPtr[i] = (float)temp; - userBufPtr += NumOuputChannels; - } - } - } - -} - -//------------------------------------------------------------------------------------------------------------------------------------------------------- -static void Output_Int32_Int16(ASIOBufferInfo* nativeBuffer, long *outBufPtr, int framePerBuffer, int NumInputChannels, int NumOuputChannels, int index, int hostFrameOffset,int userFrameOffset,uint32 flags,bool swap) -{ - long temp; - int i,j; - - if( flags & paDitherOff ) - { - for (j= 0; j < NumOuputChannels; j++) - { - short *asioBufPtr = &((short*)nativeBuffer[j+NumInputChannels].buffers[index])[hostFrameOffset]; - long *userBufPtr = &outBufPtr[j+(userFrameOffset*NumOuputChannels)]; - for( i=0; i<framePerBuffer; i++ ) - { - temp = (short) ((*userBufPtr) >> 16); - if (swap) temp = SwapShort(temp); - asioBufPtr[i] = (short)temp; - userBufPtr += NumOuputChannels; - } - } - } - else - { - for (j= 0; j < NumOuputChannels; j++) - { - short *asioBufPtr = &((short*)nativeBuffer[j+NumInputChannels].buffers[index])[hostFrameOffset]; - long *userBufPtr = &outBufPtr[j+(userFrameOffset*NumOuputChannels)]; - for( i=0; i<framePerBuffer; i++ ) - { - temp = (*userBufPtr >> 1) + Pa_TriangularDither(); - temp = temp >> 15; - temp = (short) ClipShort(temp); - if (swap) temp = SwapShort(temp); - asioBufPtr[i] = (short)temp; - userBufPtr += NumOuputChannels; - } - } - } -} - -//------------------------------------------------------------------------------------------------------------------------------------------------------- -static void Output_Int32_Int32(ASIOBufferInfo* nativeBuffer, long *outBufPtr, int framePerBuffer, int NumInputChannels, int NumOuputChannels, int index, int hostFrameOffset,int userFrameOffset,uint32 flags,bool swap) -{ - long temp; - int i,j; - - for (j= 0; j < NumOuputChannels; j++) - { - long *asioBufPtr = &((long*)nativeBuffer[j+NumInputChannels].buffers[index])[hostFrameOffset]; - long *userBufPtr = &outBufPtr[j+(userFrameOffset*NumOuputChannels)]; - for( i=0; i<framePerBuffer; i++ ) - { - temp = *userBufPtr; - if (swap) temp = SwapLong(temp); - asioBufPtr[i] = temp; - userBufPtr += NumOuputChannels; - } - } -} - -//------------------------------------------------------------------------------------------------------------------------------------------------------- -// MUST BE CHECKED - -static void Output_Int32_Float32(ASIOBufferInfo* nativeBuffer, long *outBufPtr, int framePerBuffer, int NumInputChannels, int NumOuputChannels, int index, int hostFrameOffset,int userFrameOffset,uint32 flags,bool swap) -{ - long temp; - int i,j; - - for (j= 0; j < NumOuputChannels; j++) - { - float *asioBufPtr = &((float*)nativeBuffer[j+NumInputChannels].buffers[index])[hostFrameOffset]; - long *userBufPtr = &outBufPtr[j+(userFrameOffset*NumOuputChannels)]; - for( i=0; i<framePerBuffer; i++ ) - { - temp = *userBufPtr; - if (swap) temp = SwapLong(temp); - asioBufPtr[i] = ((float)temp) * (1.0f / MAX_INT32_FP); - userBufPtr += NumOuputChannels; - } - } -} - -//------------------------------------------------------------------------------------------------------------------------------------------------------- -static void Output_Int16_Int16(ASIOBufferInfo* nativeBuffer, short *outBufPtr, int framePerBuffer, int NumInputChannels, int NumOuputChannels, int index, int hostFrameOffset, int userFrameOffset,bool swap) -{ - long temp; - int i,j; - - for (j= 0; j < NumOuputChannels; j++) - { - short *asioBufPtr = &((short*)nativeBuffer[j+NumInputChannels].buffers[index])[hostFrameOffset]; - short *userBufPtr = &outBufPtr[j+(userFrameOffset*NumOuputChannels)]; - for( i=0; i<framePerBuffer; i++ ) - { - temp = *userBufPtr; - if (swap) temp = SwapShort(temp); - asioBufPtr[i] = (short)temp; - userBufPtr += NumOuputChannels; - } - } -} - -//------------------------------------------------------------------------------------------------------------------------------------------------------- -static void Output_Int16_Int32(ASIOBufferInfo* nativeBuffer, short *outBufPtr, int framePerBuffer, int NumInputChannels, int NumOuputChannels, int index, int hostFrameOffset,int userFrameOffset, bool swap) -{ - long temp; - int i,j; - - for (j= 0; j < NumOuputChannels; j++) - { - long *asioBufPtr = &((long*)nativeBuffer[j+NumInputChannels].buffers[index])[hostFrameOffset]; - short *userBufPtr = &outBufPtr[j+(userFrameOffset*NumOuputChannels)]; - for( i=0; i<framePerBuffer; i++ ) - { - temp = (*userBufPtr)<<16; - if (swap) temp = SwapLong(temp); - asioBufPtr[i] = temp; - userBufPtr += NumOuputChannels; - } - } -} - -//------------------------------------------------------------------------------------------------------------------------------------------------------- -// MUST BE CHECKED -static void Output_Int16_Float32(ASIOBufferInfo* nativeBuffer, short *outBufPtr, int framePerBuffer, int NumInputChannels, int NumOuputChannels, int index, int hostFrameOffset,int userFrameOffset, bool swap) -{ - long temp; - int i,j; - - for (j= 0; j < NumOuputChannels; j++) - { - float *asioBufPtr = &((float*)nativeBuffer[j+NumInputChannels].buffers[index])[hostFrameOffset]; - short *userBufPtr = &outBufPtr[j+(userFrameOffset*NumOuputChannels)]; - for( i=0; i<framePerBuffer; i++ ) - { - temp = *userBufPtr; - asioBufPtr[i] = ((float)temp) * (1.0f / MAX_INT16_FP); - userBufPtr += NumOuputChannels; - } - } -} -//------------------------------------------------------------------------------------------------------------------------------------------------------- -static void Output_Int8_Int16(ASIOBufferInfo* nativeBuffer, char *outBufPtr, int framePerBuffer, int NumInputChannels, int NumOuputChannels, int index, int hostFrameOffset,int userFrameOffset, bool swap) -{ - long temp; - int i,j; - - for (j= 0; j < NumOuputChannels; j++) - { - short *asioBufPtr = &((short*)nativeBuffer[j+NumInputChannels].buffers[index])[hostFrameOffset]; - char *userBufPtr = &outBufPtr[j+(userFrameOffset*NumOuputChannels)]; - for( i=0; i<framePerBuffer; i++ ) - { - temp = (short)(*userBufPtr)<<8; - if (swap) temp = SwapShort(temp); - asioBufPtr[i] = (short)temp; - userBufPtr += NumOuputChannels; - } - } -} - -//------------------------------------------------------------------------------------------------------------------------------------------------------- -static void Output_Int8_Int32(ASIOBufferInfo* nativeBuffer, char *outBufPtr, int framePerBuffer, int NumInputChannels, int NumOuputChannels, int index, int hostFrameOffset,int userFrameOffset, bool swap) -{ - long temp; - int i,j; - - for (j= 0; j < NumOuputChannels; j++) - { - long *asioBufPtr = &((long*)nativeBuffer[j+NumInputChannels].buffers[index])[hostFrameOffset]; - char *userBufPtr = &outBufPtr[j+(userFrameOffset*NumOuputChannels)]; - for( i=0; i<framePerBuffer; i++ ) - { - temp = (short)(*userBufPtr)<<24; - if (swap) temp = SwapLong(temp); - asioBufPtr[i] = temp; - userBufPtr += NumOuputChannels; - } - } -} - - -//------------------------------------------------------------------------------------------------------------------------------------------------------- -// MUST BE CHECKED -static void Output_Int8_Float32(ASIOBufferInfo* nativeBuffer, char *outBufPtr, int framePerBuffer, int NumInputChannels, int NumOuputChannels, int index, int hostFrameOffset,int userFrameOffset, bool swap) -{ - long temp; - int i,j; - - for (j= 0; j < NumOuputChannels; j++) - { - long *asioBufPtr = &((long*)nativeBuffer[j+NumInputChannels].buffers[index])[hostFrameOffset]; - char *userBufPtr = &outBufPtr[j+(userFrameOffset*NumOuputChannels)]; - for( i=0; i<framePerBuffer; i++ ) - { - temp = *userBufPtr; - asioBufPtr[i] = (long)(((float)temp) * (1.0f / MAX_INT8_FP)); - userBufPtr += NumOuputChannels; - } - } -} - -//------------------------------------------------------------------------------------------------------------------------------------------------------- -static void Output_IntU8_Int16(ASIOBufferInfo* nativeBuffer, unsigned char *outBufPtr, int framePerBuffer, int NumInputChannels, int NumOuputChannels, int index, int hostFrameOffset,int userFrameOffset, bool swap) -{ - long temp; - int i,j; - - for (j= 0; j < NumOuputChannels; j++) - { - short *asioBufPtr = &((short*)nativeBuffer[j+NumInputChannels].buffers[index])[hostFrameOffset]; - unsigned char *userBufPtr = &outBufPtr[j+(userFrameOffset*NumOuputChannels)]; - for( i=0; i<framePerBuffer; i++ ) - { - temp = ((short)((*userBufPtr) - 0x80)) << 8; - if (swap) temp = SwapShort(temp); - asioBufPtr[i] = (short)temp; - userBufPtr += NumOuputChannels; - } - } -} - -//------------------------------------------------------------------------------------------------------------------------------------------------------- -static void Output_IntU8_Int32(ASIOBufferInfo* nativeBuffer, unsigned char *outBufPtr, int framePerBuffer, int NumInputChannels, int NumOuputChannels, int index, int hostFrameOffset,int userFrameOffset, bool swap) -{ - long temp; - int i,j; - - for (j= 0; j < NumOuputChannels; j++) - { - long *asioBufPtr = &((long*)nativeBuffer[j+NumInputChannels].buffers[index])[hostFrameOffset]; - unsigned char *userBufPtr = &outBufPtr[j+(userFrameOffset*NumOuputChannels)]; - for( i=0; i<framePerBuffer; i++ ) - { - temp = ((short)((*userBufPtr) - 0x80)) << 24; - if (swap) temp = SwapLong(temp); - asioBufPtr[i] = temp; - userBufPtr += NumOuputChannels; - } - } -} - -//------------------------------------------------------------------------------------------------------------------------------------------------------- -// MUST BE CHECKED - -static void Output_IntU8_Float32(ASIOBufferInfo* nativeBuffer, unsigned char *outBufPtr, int framePerBuffer, int NumInputChannels, int NumOuputChannels, int index, int hostFrameOffset,int userFrameOffset, bool swap) -{ - long temp; - int i,j; - - for (j= 0; j < NumOuputChannels; j++) - { - float *asioBufPtr = &((float*)nativeBuffer[j+NumInputChannels].buffers[index])[hostFrameOffset]; - unsigned char *userBufPtr = &outBufPtr[j+(userFrameOffset*NumOuputChannels)]; - for( i=0; i<framePerBuffer; i++ ) - { - temp = ((short)((*userBufPtr) - 0x80)) << 24; - asioBufPtr[i] = ((float)temp) * (1.0f / MAX_INT32_FP); - userBufPtr += NumOuputChannels; - } - } -} - -//------------------------------------------------------------------------------------------------------------------------------------------------------- -static void Pa_ASIO_Clear_Output_16 (ASIOBufferInfo* nativeBuffer, long frames, long NumInputChannels, long NumOuputChannels, long index, long hostFrameOffset) -{ - int i,j; - - for( j=0; j<NumOuputChannels; j++ ) { - short *asioBufPtr = &((short*)nativeBuffer[j+NumInputChannels].buffers[index])[hostFrameOffset]; - for (i= 0; i < frames; i++) {asioBufPtr[i] = 0; } - } -} - -//------------------------------------------------------------------------------------------------------------------------------------------------------- -static void Pa_ASIO_Clear_Output_32 (ASIOBufferInfo* nativeBuffer, long frames, long NumInputChannels, long NumOuputChannels, long index, long hostFrameOffset) -{ - int i,j; - - for( j=0; j<NumOuputChannels; j++ ) { - long *asioBufPtr = &((long*)nativeBuffer[j+NumInputChannels].buffers[index])[hostFrameOffset]; - for (i= 0; i < frames; i++) {asioBufPtr[i] = 0; } - } -} - - -//------------------------------------------------------------------------------------------------------------------------------------------------------- -static void Pa_ASIO_Adaptor_Init() -{ - if (asioDriverInfo.past->past_FramesPerUserBuffer <= asioDriverInfo.past_FramesPerHostBuffer) { - asioDriverInfo.pahsc_hostOutputBufferFrameOffset = asioDriverInfo.pahsc_OutputBufferOffset; - asioDriverInfo.pahsc_userInputBufferFrameOffset = 0; // empty - asioDriverInfo.pahsc_userOutputBufferFrameOffset = asioDriverInfo.past->past_FramesPerUserBuffer; // empty - DBUG(("Pa_ASIO_Adaptor_Init : shift output\n")); - DBUG(("Pa_ASIO_Adaptor_Init : userInputBufferFrameOffset %d\n",asioDriverInfo.pahsc_userInputBufferFrameOffset)); - DBUG(("Pa_ASIO_Adaptor_Init : userOutputBufferFrameOffset %d\n",asioDriverInfo.pahsc_userOutputBufferFrameOffset)); - DBUG(("Pa_ASIO_Adaptor_Init : hostOutputBufferFrameOffset %d\n",asioDriverInfo.pahsc_hostOutputBufferFrameOffset)); - - }else { - asioDriverInfo.pahsc_hostOutputBufferFrameOffset = 0; // empty - asioDriverInfo.pahsc_userInputBufferFrameOffset = asioDriverInfo.pahsc_InputBufferOffset; - asioDriverInfo.pahsc_userOutputBufferFrameOffset = asioDriverInfo.past->past_FramesPerUserBuffer; // empty - DBUG(("Pa_ASIO_Adaptor_Init : shift input\n")); - DBUG(("Pa_ASIO_Adaptor_Init : userInputBufferFrameOffset %d\n",asioDriverInfo.pahsc_userInputBufferFrameOffset)); - DBUG(("Pa_ASIO_Adaptor_Init : userOutputBufferFrameOffset %d\n",asioDriverInfo.pahsc_userOutputBufferFrameOffset)); - DBUG(("Pa_ASIO_Adaptor_Init : hostOutputBufferFrameOffset %d\n",asioDriverInfo.pahsc_hostOutputBufferFrameOffset)); - } -} - -//------------------------------------------------------------------------------------------------------------------------------------------------------- -// FIXME : optimization for Input only or output only modes (really necessary ??) -static void Pa_ASIO_Callback_Input(long index) -{ - internalPortAudioStream *past = asioDriverInfo.past; - long framesInputHostBuffer = asioDriverInfo.past_FramesPerHostBuffer; // number of frames available into the host input buffer - long framesInputUserBuffer; // number of frames needed to complete the user input buffer - long framesOutputHostBuffer; // number of frames needed to complete the host output buffer - long framesOuputUserBuffer; // number of frames available into the user output buffer - long userResult; - long tmp; - - /* Fill host ASIO output with remaining frames in user output */ - framesOutputHostBuffer = asioDriverInfo.past_FramesPerHostBuffer - asioDriverInfo.pahsc_hostOutputBufferFrameOffset; - framesOuputUserBuffer = asioDriverInfo.past->past_FramesPerUserBuffer - asioDriverInfo.pahsc_userOutputBufferFrameOffset; - tmp = min(framesOutputHostBuffer, framesOuputUserBuffer); - framesOutputHostBuffer -= tmp; - Pa_ASIO_Callback_Output(index,tmp); - - /* Available frames in hostInputBuffer */ - while (framesInputHostBuffer > 0) { - - /* Number of frames needed to complete an user input buffer */ - framesInputUserBuffer = asioDriverInfo.past->past_FramesPerUserBuffer - asioDriverInfo.pahsc_userInputBufferFrameOffset; - - if (framesInputHostBuffer >= framesInputUserBuffer) { - - /* Convert ASIO input to user input */ - Pa_ASIO_Convert_Inter_Input (asioDriverInfo.bufferInfos, - past->past_InputBuffer, - asioDriverInfo.pahsc_NumInputChannels , - asioDriverInfo.pahsc_NumOutputChannels, - framesInputUserBuffer, - asioDriverInfo.past_FramesPerHostBuffer - framesInputHostBuffer, - asioDriverInfo.pahsc_userInputBufferFrameOffset, - asioDriverInfo.pahsc_channelInfos[0].type, - past->past_InputSampleFormat, - past->past_Flags, - index); - - /* Call PortAudio callback */ - userResult = asioDriverInfo.past->past_Callback(past->past_InputBuffer, past->past_OutputBuffer, - past->past_FramesPerUserBuffer,past->past_FrameCount,past->past_UserData ); - - past->past_FrameCount += (PaTimestamp) past->past_FramesPerUserBuffer; - - /* User callback has asked us to stop in the middle of the host buffer */ - if( userResult != 0) { - - /* Put 0 in the end of the output buffer */ - Pa_ASIO_Clear_Output(asioDriverInfo.bufferInfos, - asioDriverInfo.pahsc_channelInfos[0].type, - asioDriverInfo.pahsc_NumInputChannels , - asioDriverInfo.pahsc_NumOutputChannels, - index, - asioDriverInfo.pahsc_hostOutputBufferFrameOffset, - asioDriverInfo.past_FramesPerHostBuffer - asioDriverInfo.pahsc_hostOutputBufferFrameOffset); - - past->past_StopSoon = 1; - return; - } - - - /* Full user ouput buffer : write offset */ - asioDriverInfo.pahsc_userOutputBufferFrameOffset = 0; - - /* Empty user input buffer : read offset */ - asioDriverInfo.pahsc_userInputBufferFrameOffset = 0; - - /* Fill host ASIO output */ - tmp = min (past->past_FramesPerUserBuffer,framesOutputHostBuffer); - Pa_ASIO_Callback_Output(index,tmp); - - framesOutputHostBuffer -= tmp; - framesInputHostBuffer -= framesInputUserBuffer; - - }else { - - /* Convert ASIO input to user input */ - Pa_ASIO_Convert_Inter_Input (asioDriverInfo.bufferInfos, - past->past_InputBuffer, - asioDriverInfo.pahsc_NumInputChannels , - asioDriverInfo.pahsc_NumOutputChannels, - framesInputHostBuffer, - asioDriverInfo.past_FramesPerHostBuffer - framesInputHostBuffer, - asioDriverInfo.pahsc_userInputBufferFrameOffset, - asioDriverInfo.pahsc_channelInfos[0].type, - past->past_InputSampleFormat, - past->past_Flags, - index); - - /* Update pahsc_userInputBufferFrameOffset */ - asioDriverInfo.pahsc_userInputBufferFrameOffset += framesInputHostBuffer; - - /* Update framesInputHostBuffer */ - framesInputHostBuffer = 0; - } - } - -} - -//------------------------------------------------------------------------------------------------------------------------------------------------------- -static void Pa_ASIO_Callback_Output(long index, long framePerBuffer) -{ - internalPortAudioStream *past = asioDriverInfo.past; - - if (framePerBuffer > 0) { - - /* Convert user output to ASIO ouput */ - Pa_ASIO_Convert_Inter_Output (asioDriverInfo.bufferInfos, - past->past_OutputBuffer, - asioDriverInfo.pahsc_NumInputChannels, - asioDriverInfo.pahsc_NumOutputChannels, - framePerBuffer, - asioDriverInfo.pahsc_hostOutputBufferFrameOffset, - asioDriverInfo.pahsc_userOutputBufferFrameOffset, - asioDriverInfo.pahsc_channelInfos[0].type, - past->past_OutputSampleFormat, - past->past_Flags, - index); - - /* Update hostOuputFrameOffset */ - asioDriverInfo.pahsc_hostOutputBufferFrameOffset += framePerBuffer; - - /* Update userOutputFrameOffset */ - asioDriverInfo.pahsc_userOutputBufferFrameOffset += framePerBuffer; - } -} -//------------------------------------------------------------------------------------------------------------------------------------------------------- -static void Pa_ASIO_Callback_End() - { - /* Empty ASIO ouput : write offset */ - asioDriverInfo.pahsc_hostOutputBufferFrameOffset = 0; - } - -//------------------------------------------------------------------------------------------------------------------------------------------------------- -static void Pa_ASIO_Clear_User_Buffers() -{ - if( asioDriverInfo.past->past_InputBuffer != NULL ) - { - memset( asioDriverInfo.past->past_InputBuffer, 0, asioDriverInfo.past->past_InputBufferSize ); - } - if( asioDriverInfo.past->past_OutputBuffer != NULL ) - { - memset( asioDriverInfo.past->past_OutputBuffer, 0, asioDriverInfo.past->past_OutputBufferSize ); - } -} - -//------------------------------------------------------------------------------------------------------------------------------------------------------- - static void Pa_ASIO_Clear_Output(ASIOBufferInfo* nativeBuffer, - ASIOSampleType nativeFormat, - long NumInputChannels, - long NumOuputChannels, - long index, - long hostFrameOffset, - long frames) -{ - - switch (nativeFormat) { - - case ASIOSTInt16MSB: - case ASIOSTInt16LSB: - case ASIOSTInt32MSB16: - case ASIOSTInt32LSB16: - Pa_ASIO_Clear_Output_16(nativeBuffer, frames, NumInputChannels, NumOuputChannels, index, hostFrameOffset); - break; - - case ASIOSTFloat64MSB: - case ASIOSTFloat64LSB: - break; - - case ASIOSTFloat32MSB: - case ASIOSTFloat32LSB: - case ASIOSTInt32MSB: - case ASIOSTInt32LSB: - case ASIOSTInt32MSB18: - case ASIOSTInt32MSB20: - case ASIOSTInt32MSB24: - case ASIOSTInt32LSB18: - case ASIOSTInt32LSB20: - case ASIOSTInt32LSB24: - Pa_ASIO_Clear_Output_32(nativeBuffer, frames, NumInputChannels, NumOuputChannels, index, hostFrameOffset); - break; - - case ASIOSTInt24MSB: - case ASIOSTInt24LSB: - break; - - default: - break; - } -} - - -//--------------------------------------------------------------------------------------- -static void Pa_ASIO_Convert_Inter_Input( - ASIOBufferInfo* nativeBuffer, - void* inputBuffer, - long NumInputChannels, - long NumOuputChannels, - long framePerBuffer, - long hostFrameOffset, - long userFrameOffset, - ASIOSampleType nativeFormat, - PaSampleFormat paFormat, - PaStreamFlags flags, - long index) -{ - - if((NumInputChannels > 0) && (nativeBuffer != NULL)) - { - /* Convert from native format to PA format. */ - switch(paFormat) - { - case paFloat32: - { - float *inBufPtr = (float *) inputBuffer; - - switch (nativeFormat) { - case ASIOSTInt16LSB: - Input_Int16_Float32(nativeBuffer, inBufPtr, framePerBuffer, NumInputChannels, index, hostFrameOffset, userFrameOffset, swap); - break; - case ASIOSTInt16MSB: - Input_Int16_Float32(nativeBuffer, inBufPtr, framePerBuffer, NumInputChannels, index, hostFrameOffset, userFrameOffset,!swap); - break; - case ASIOSTInt32LSB: - Input_Int32_Float32(nativeBuffer, inBufPtr, framePerBuffer, NumInputChannels, index, hostFrameOffset, userFrameOffset,swap); - break; - case ASIOSTInt32MSB: - Input_Int32_Float32(nativeBuffer, inBufPtr, framePerBuffer, NumInputChannels, index, hostFrameOffset, userFrameOffset,!swap); - break; - case ASIOSTFloat32LSB: // IEEE 754 32 bit float, as found on Intel x86 architecture - Input_Float32_Float32(nativeBuffer, inBufPtr, framePerBuffer, NumInputChannels, index, hostFrameOffset, userFrameOffset,swap); - break; - case ASIOSTFloat32MSB: // IEEE 754 32 bit float, as found on Intel x86 architecture - Input_Float32_Float32(nativeBuffer, inBufPtr, framePerBuffer, NumInputChannels, index, hostFrameOffset, userFrameOffset,!swap); - break; - - case ASIOSTInt24LSB: // used for 20 bits as well - case ASIOSTInt24MSB: // used for 20 bits as well - - case ASIOSTFloat64LSB: // IEEE 754 64 bit double float, as found on Intel x86 architecture - case ASIOSTFloat64MSB: // IEEE 754 64 bit double float, as found on Intel x86 architecture - - // these are used for 32 bit data buffer, with different alignment of the data inside - // 32 bit PCI bus systems can more easily used with these - - case ASIOSTInt32LSB16: // 32 bit data with 16 bit alignment - case ASIOSTInt32LSB18: // 32 bit data with 18 bit alignment - case ASIOSTInt32LSB20: // 32 bit data with 20 bit alignment - case ASIOSTInt32LSB24: // 32 bit data with 24 bit alignment - - - case ASIOSTInt32MSB16: // 32 bit data with 16 bit alignment - case ASIOSTInt32MSB18: // 32 bit data with 18 bit alignment - case ASIOSTInt32MSB20: // 32 bit data with 20 bit alignment - case ASIOSTInt32MSB24: // 32 bit data with 24 bit alignment - DBUG(("Not yet implemented : please report the problem\n")); - break; - } - - break; - } - - case paInt32: - { - long *inBufPtr = (long *)inputBuffer; - - switch (nativeFormat) { - case ASIOSTInt16LSB: - Input_Int16_Int32(nativeBuffer, inBufPtr, framePerBuffer, NumInputChannels, index, hostFrameOffset,userFrameOffset, swap); - break; - case ASIOSTInt16MSB: - Input_Int16_Int32(nativeBuffer, inBufPtr, framePerBuffer, NumInputChannels, index, hostFrameOffset,userFrameOffset, !swap); - break; - case ASIOSTInt32LSB: - Input_Int32_Int32(nativeBuffer, inBufPtr, framePerBuffer, NumInputChannels, index, hostFrameOffset,userFrameOffset, swap); - break; - case ASIOSTInt32MSB: - Input_Int32_Int32(nativeBuffer, inBufPtr, framePerBuffer, NumInputChannels, index, hostFrameOffset,userFrameOffset, !swap); - break; - case ASIOSTFloat32LSB: // IEEE 754 32 bit float, as found on Intel x86 architecture - Input_Float32_Int32(nativeBuffer, inBufPtr, framePerBuffer, NumInputChannels, index, hostFrameOffset,userFrameOffset, swap); - break; - case ASIOSTFloat32MSB: // IEEE 754 32 bit float, as found on Intel x86 architecture - Input_Float32_Int32(nativeBuffer, inBufPtr, framePerBuffer, NumInputChannels, index, hostFrameOffset,userFrameOffset, !swap); - break; - - case ASIOSTInt24LSB: // used for 20 bits as well - case ASIOSTInt24MSB: // used for 20 bits as well - - case ASIOSTFloat64LSB: // IEEE 754 64 bit double float, as found on Intel x86 architecture - case ASIOSTFloat64MSB: // IEEE 754 64 bit double float, as found on Intel x86 architecture - - // these are used for 32 bit data buffer, with different alignment of the data inside - // 32 bit PCI bus systems can more easily used with these - - case ASIOSTInt32LSB16: // 32 bit data with 16 bit alignment - case ASIOSTInt32LSB18: // 32 bit data with 18 bit alignment - case ASIOSTInt32LSB20: // 32 bit data with 20 bit alignment - case ASIOSTInt32LSB24: // 32 bit data with 24 bit alignment - - - case ASIOSTInt32MSB16: // 32 bit data with 16 bit alignment - case ASIOSTInt32MSB18: // 32 bit data with 18 bit alignment - case ASIOSTInt32MSB20: // 32 bit data with 20 bit alignment - case ASIOSTInt32MSB24: // 32 bit data with 24 bit alignment - DBUG(("Not yet implemented : please report the problem\n")); - break; - - } - break; - } - - case paInt16: - { - short *inBufPtr = (short *) inputBuffer; - - switch (nativeFormat) { - case ASIOSTInt16LSB: - Input_Int16_Int16(nativeBuffer, inBufPtr, framePerBuffer , NumInputChannels, index , hostFrameOffset,userFrameOffset, swap); - break; - case ASIOSTInt16MSB: - Input_Int16_Int16(nativeBuffer, inBufPtr, framePerBuffer , NumInputChannels, index , hostFrameOffset,userFrameOffset, !swap); - break; - case ASIOSTInt32LSB: - Input_Int32_Int16(nativeBuffer, inBufPtr, framePerBuffer, NumInputChannels, index, hostFrameOffset,userFrameOffset, flags,swap); - break; - case ASIOSTInt32MSB: - Input_Int32_Int16(nativeBuffer, inBufPtr, framePerBuffer, NumInputChannels, index, hostFrameOffset,userFrameOffset, flags,!swap); - break; - case ASIOSTFloat32LSB: // IEEE 754 32 bit float, as found on Intel x86 architecture - Input_Float32_Int16(nativeBuffer, inBufPtr, framePerBuffer, NumInputChannels, index, hostFrameOffset,userFrameOffset, flags,swap); - break; - case ASIOSTFloat32MSB: // IEEE 754 32 bit float, as found on Intel x86 architecture - Input_Float32_Int16(nativeBuffer, inBufPtr, framePerBuffer, NumInputChannels, index, hostFrameOffset,userFrameOffset, flags,!swap); - break; - - case ASIOSTInt24LSB: // used for 20 bits as well - case ASIOSTInt24MSB: // used for 20 bits as well - - case ASIOSTFloat64LSB: // IEEE 754 64 bit double float, as found on Intel x86 architecture - case ASIOSTFloat64MSB: // IEEE 754 64 bit double float, as found on Intel x86 architecture - - // these are used for 32 bit data buffer, with different alignment of the data inside - // 32 bit PCI bus systems can more easily used with these - - case ASIOSTInt32LSB16: // 32 bit data with 16 bit alignment - case ASIOSTInt32LSB18: // 32 bit data with 18 bit alignment - case ASIOSTInt32LSB20: // 32 bit data with 20 bit alignment - case ASIOSTInt32LSB24: // 32 bit data with 24 bit alignment - - - case ASIOSTInt32MSB16: // 32 bit data with 16 bit alignment - case ASIOSTInt32MSB18: // 32 bit data with 18 bit alignment - case ASIOSTInt32MSB20: // 32 bit data with 20 bit alignment - case ASIOSTInt32MSB24: // 32 bit data with 24 bit alignment - DBUG(("Not yet implemented : please report the problem\n")); - break; - - } - break; - } - - case paInt8: - { - /* Convert 16 bit data to 8 bit chars */ - - char *inBufPtr = (char *) inputBuffer; - - switch (nativeFormat) { - case ASIOSTInt16LSB: - Input_Int16_Int8(nativeBuffer, inBufPtr, framePerBuffer, NumInputChannels, index, hostFrameOffset,userFrameOffset,flags,swap); - break; - case ASIOSTInt16MSB: - Input_Int16_Int8(nativeBuffer, inBufPtr, framePerBuffer, NumInputChannels, index, hostFrameOffset,userFrameOffset, flags,!swap); - break; - case ASIOSTInt32LSB: - Input_Int32_Int8(nativeBuffer, inBufPtr, framePerBuffer, NumInputChannels, index, hostFrameOffset,userFrameOffset, flags,swap); - break; - case ASIOSTInt32MSB: - Input_Int32_Int8(nativeBuffer, inBufPtr, framePerBuffer, NumInputChannels, index, hostFrameOffset,userFrameOffset, flags,!swap); - break; - case ASIOSTFloat32LSB: // IEEE 754 32 bit float, as found on Intel x86 architecture - Input_Float32_Int8(nativeBuffer, inBufPtr, framePerBuffer, NumInputChannels, index, hostFrameOffset,userFrameOffset, flags,swap); - break; - case ASIOSTFloat32MSB: // IEEE 754 32 bit float, as found on Intel x86 architecture - Input_Float32_Int8(nativeBuffer, inBufPtr, framePerBuffer, NumInputChannels, index, hostFrameOffset,userFrameOffset, flags,!swap); - break; - - case ASIOSTInt24LSB: // used for 20 bits as well - case ASIOSTInt24MSB: // used for 20 bits as well - - case ASIOSTFloat64LSB: // IEEE 754 64 bit double float, as found on Intel x86 architecture - case ASIOSTFloat64MSB: // IEEE 754 64 bit double float, as found on Intel x86 architecture - - // these are used for 32 bit data buffer, with different alignment of the data inside - // 32 bit PCI bus systems can more easily used with these - - case ASIOSTInt32LSB16: // 32 bit data with 16 bit alignment - case ASIOSTInt32LSB18: // 32 bit data with 18 bit alignment - case ASIOSTInt32LSB20: // 32 bit data with 20 bit alignment - case ASIOSTInt32LSB24: // 32 bit data with 24 bit alignment - - - case ASIOSTInt32MSB16: // 32 bit data with 16 bit alignment - case ASIOSTInt32MSB18: // 32 bit data with 18 bit alignment - case ASIOSTInt32MSB20: // 32 bit data with 20 bit alignment - case ASIOSTInt32MSB24: // 32 bit data with 24 bit alignment - DBUG(("Not yet implemented : please report the problem\n")); - break; - } - break; - } - - case paUInt8: - { - /* Convert 16 bit data to 8 bit unsigned chars */ - - unsigned char *inBufPtr = (unsigned char *)inputBuffer; - - switch (nativeFormat) { - case ASIOSTInt16LSB: - Input_Int16_IntU8(nativeBuffer, inBufPtr, framePerBuffer, NumInputChannels, index, hostFrameOffset,userFrameOffset, flags,swap); - break; - case ASIOSTInt16MSB: - Input_Int16_IntU8(nativeBuffer, inBufPtr, framePerBuffer, NumInputChannels, index, hostFrameOffset,userFrameOffset, flags,!swap); - break; - case ASIOSTInt32LSB: - Input_Int32_IntU8(nativeBuffer, inBufPtr, framePerBuffer, NumInputChannels, index, hostFrameOffset,userFrameOffset,flags,swap); - break; - case ASIOSTInt32MSB: - Input_Int32_IntU8(nativeBuffer, inBufPtr, framePerBuffer, NumInputChannels, index, hostFrameOffset,userFrameOffset, flags,!swap); - break; - case ASIOSTFloat32LSB: // IEEE 754 32 bit float, as found on Intel x86 architecture - Input_Float32_IntU8(nativeBuffer, inBufPtr, framePerBuffer, NumInputChannels, index, hostFrameOffset,userFrameOffset,flags,swap); - break; - case ASIOSTFloat32MSB: // IEEE 754 32 bit float, as found on Intel x86 architecture - Input_Float32_IntU8(nativeBuffer, inBufPtr, framePerBuffer, NumInputChannels, index, hostFrameOffset,userFrameOffset,flags,!swap); - break; - - case ASIOSTInt24LSB: // used for 20 bits as well - case ASIOSTInt24MSB: // used for 20 bits as well - - case ASIOSTFloat64LSB: // IEEE 754 64 bit double float, as found on Intel x86 architecture - case ASIOSTFloat64MSB: // IEEE 754 64 bit double float, as found on Intel x86 architecture - - // these are used for 32 bit data buffer, with different alignment of the data inside - // 32 bit PCI bus systems can more easily used with these - - case ASIOSTInt32LSB16: // 32 bit data with 16 bit alignment - case ASIOSTInt32LSB18: // 32 bit data with 18 bit alignment - case ASIOSTInt32LSB20: // 32 bit data with 20 bit alignment - case ASIOSTInt32LSB24: // 32 bit data with 24 bit alignment - - - case ASIOSTInt32MSB16: // 32 bit data with 16 bit alignment - case ASIOSTInt32MSB18: // 32 bit data with 18 bit alignment - case ASIOSTInt32MSB20: // 32 bit data with 20 bit alignment - case ASIOSTInt32MSB24: // 32 bit data with 24 bit alignment - DBUG(("Not yet implemented : please report the problem\n")); - break; - - } - break; - } - - default: - break; - } - } -} - - -//--------------------------------------------------------------------------------------- -static void Pa_ASIO_Convert_Inter_Output(ASIOBufferInfo* nativeBuffer, - void* outputBuffer, - long NumInputChannels, - long NumOuputChannels, - long framePerBuffer, - long hostFrameOffset, - long userFrameOffset, - ASIOSampleType nativeFormat, - PaSampleFormat paFormat, - PaStreamFlags flags, - long index) -{ - - if((NumOuputChannels > 0) && (nativeBuffer != NULL)) - { - /* Convert from PA format to native format */ - - switch(paFormat) - { - case paFloat32: - { - float *outBufPtr = (float *) outputBuffer; - - switch (nativeFormat) { - case ASIOSTInt16LSB: - Output_Float32_Int16(nativeBuffer, outBufPtr, framePerBuffer, NumInputChannels, NumOuputChannels, index, hostFrameOffset, userFrameOffset, flags, swap); - break; - case ASIOSTInt16MSB: - Output_Float32_Int16(nativeBuffer, outBufPtr, framePerBuffer, NumInputChannels, NumOuputChannels, index, hostFrameOffset, userFrameOffset, flags,!swap); - break; - case ASIOSTInt32LSB: - Output_Float32_Int32(nativeBuffer, outBufPtr, framePerBuffer, NumInputChannels, NumOuputChannels, index, hostFrameOffset, userFrameOffset, flags,swap); - break; - case ASIOSTInt32MSB: - Output_Float32_Int32(nativeBuffer, outBufPtr, framePerBuffer, NumInputChannels, NumOuputChannels, index, hostFrameOffset,userFrameOffset, flags,!swap); - break; - case ASIOSTFloat32LSB: - Output_Float32_Float32(nativeBuffer, outBufPtr, framePerBuffer, NumInputChannels, NumOuputChannels, index, hostFrameOffset,userFrameOffset,flags,swap); - break; - case ASIOSTFloat32MSB: - Output_Float32_Float32(nativeBuffer, outBufPtr, framePerBuffer, NumInputChannels, NumOuputChannels, index, hostFrameOffset,userFrameOffset, flags,!swap); - break; - - case ASIOSTInt24LSB: // used for 20 bits as well - case ASIOSTInt24MSB: // used for 20 bits as well - - case ASIOSTFloat64LSB: // IEEE 754 64 bit double float, as found on Intel x86 architecture - case ASIOSTFloat64MSB: // IEEE 754 64 bit double float, as found on Intel x86 architecture - - // these are used for 32 bit data buffer, with different alignment of the data inside - // 32 bit PCI bus systems can more easily used with these - - case ASIOSTInt32LSB16: // 32 bit data with 16 bit alignment - case ASIOSTInt32LSB18: // 32 bit data with 18 bit alignment - case ASIOSTInt32LSB20: // 32 bit data with 20 bit alignment - case ASIOSTInt32LSB24: // 32 bit data with 24 bit alignment - - - case ASIOSTInt32MSB16: // 32 bit data with 16 bit alignment - case ASIOSTInt32MSB18: // 32 bit data with 18 bit alignment - case ASIOSTInt32MSB20: // 32 bit data with 20 bit alignment - case ASIOSTInt32MSB24: // 32 bit data with 24 bit alignment - DBUG(("Not yet implemented : please report the problem\n")); - break; - } - break; - } - - case paInt32: - { - long *outBufPtr = (long *) outputBuffer; - - switch (nativeFormat) { - case ASIOSTInt16LSB: - Output_Int32_Int16(nativeBuffer, outBufPtr, framePerBuffer, NumInputChannels, NumOuputChannels, index, hostFrameOffset,userFrameOffset, flags,swap); - break; - case ASIOSTInt16MSB: - Output_Int32_Int16(nativeBuffer, outBufPtr, framePerBuffer, NumInputChannels, NumOuputChannels, index, hostFrameOffset,userFrameOffset, flags,!swap); - break; - case ASIOSTInt32LSB: - Output_Int32_Int32(nativeBuffer, outBufPtr, framePerBuffer, NumInputChannels, NumOuputChannels, index, hostFrameOffset,userFrameOffset, flags,swap); - break; - case ASIOSTInt32MSB: - Output_Int32_Int32(nativeBuffer, outBufPtr, framePerBuffer, NumInputChannels, NumOuputChannels, index, hostFrameOffset,userFrameOffset, flags,!swap); - break; - case ASIOSTFloat32LSB: - Output_Int32_Float32(nativeBuffer, outBufPtr, framePerBuffer, NumInputChannels, NumOuputChannels, index, hostFrameOffset,userFrameOffset, flags,swap); - break; - case ASIOSTFloat32MSB: - Output_Int32_Float32(nativeBuffer, outBufPtr, framePerBuffer, NumInputChannels, NumOuputChannels, index, hostFrameOffset,userFrameOffset, flags,!swap); - break; - - case ASIOSTInt24LSB: // used for 20 bits as well - case ASIOSTInt24MSB: // used for 20 bits as well - - case ASIOSTFloat64LSB: // IEEE 754 64 bit double float, as found on Intel x86 architecture - case ASIOSTFloat64MSB: // IEEE 754 64 bit double float, as found on Intel x86 architecture - - // these are used for 32 bit data buffer, with different alignment of the data inside - // 32 bit PCI bus systems can more easily used with these - - case ASIOSTInt32LSB16: // 32 bit data with 16 bit alignment - case ASIOSTInt32LSB18: // 32 bit data with 18 bit alignment - case ASIOSTInt32LSB20: // 32 bit data with 20 bit alignment - case ASIOSTInt32LSB24: // 32 bit data with 24 bit alignment - - - case ASIOSTInt32MSB16: // 32 bit data with 16 bit alignment - case ASIOSTInt32MSB18: // 32 bit data with 18 bit alignment - case ASIOSTInt32MSB20: // 32 bit data with 20 bit alignment - case ASIOSTInt32MSB24: // 32 bit data with 24 bit alignment - DBUG(("Not yet implemented : please report the problem\n")); - break; - } - break; - } - - case paInt16: - { - short *outBufPtr = (short *) outputBuffer; - - switch (nativeFormat) { - case ASIOSTInt16LSB: - Output_Int16_Int16(nativeBuffer, outBufPtr, framePerBuffer, NumInputChannels, NumOuputChannels, index, hostFrameOffset,userFrameOffset, swap); - break; - case ASIOSTInt16MSB: - Output_Int16_Int16(nativeBuffer, outBufPtr, framePerBuffer, NumInputChannels, NumOuputChannels, index, hostFrameOffset,userFrameOffset, !swap); - break; - case ASIOSTInt32LSB: - Output_Int16_Int32(nativeBuffer, outBufPtr, framePerBuffer, NumInputChannels, NumOuputChannels, index, hostFrameOffset,userFrameOffset, swap); - break; - case ASIOSTInt32MSB: - Output_Int16_Int32(nativeBuffer, outBufPtr, framePerBuffer, NumInputChannels, NumOuputChannels, index, hostFrameOffset,userFrameOffset, !swap); - break; - case ASIOSTFloat32LSB: - Output_Int16_Float32(nativeBuffer, outBufPtr, framePerBuffer, NumInputChannels, NumOuputChannels, index, hostFrameOffset,userFrameOffset, swap); - break; - case ASIOSTFloat32MSB: - Output_Int16_Float32(nativeBuffer, outBufPtr, framePerBuffer, NumInputChannels, NumOuputChannels, index, hostFrameOffset,userFrameOffset, !swap); - break; - - case ASIOSTInt24LSB: // used for 20 bits as well - case ASIOSTInt24MSB: // used for 20 bits as well - - case ASIOSTFloat64LSB: // IEEE 754 64 bit double float, as found on Intel x86 architecture - case ASIOSTFloat64MSB: // IEEE 754 64 bit double float, as found on Intel x86 architecture - - // these are used for 32 bit data buffer, with different alignment of the data inside - // 32 bit PCI bus systems can more easily used with these - - case ASIOSTInt32LSB16: // 32 bit data with 16 bit alignment - case ASIOSTInt32LSB18: // 32 bit data with 18 bit alignment - case ASIOSTInt32LSB20: // 32 bit data with 20 bit alignment - case ASIOSTInt32LSB24: // 32 bit data with 24 bit alignment - - - case ASIOSTInt32MSB16: // 32 bit data with 16 bit alignment - case ASIOSTInt32MSB18: // 32 bit data with 18 bit alignment - case ASIOSTInt32MSB20: // 32 bit data with 20 bit alignment - case ASIOSTInt32MSB24: // 32 bit data with 24 bit alignment - DBUG(("Not yet implemented : please report the problem\n")); - break; - - } - break; - } - - - case paInt8: - { - char *outBufPtr = (char *) outputBuffer; - - switch (nativeFormat) { - case ASIOSTInt16LSB: - Output_Int8_Int16(nativeBuffer, outBufPtr, framePerBuffer, NumInputChannels, NumOuputChannels, index, hostFrameOffset,userFrameOffset, swap); - break; - case ASIOSTInt16MSB: - Output_Int8_Int16(nativeBuffer, outBufPtr, framePerBuffer, NumInputChannels, NumOuputChannels, index, hostFrameOffset,userFrameOffset, !swap); - break; - case ASIOSTInt32LSB: - Output_Int8_Int32(nativeBuffer, outBufPtr, framePerBuffer, NumInputChannels, NumOuputChannels, index, hostFrameOffset,userFrameOffset, swap); - break; - case ASIOSTInt32MSB: - Output_Int8_Int32(nativeBuffer, outBufPtr, framePerBuffer, NumInputChannels, NumOuputChannels, index, hostFrameOffset,userFrameOffset, !swap); - break; - case ASIOSTFloat32LSB: - Output_Int8_Float32(nativeBuffer, outBufPtr, framePerBuffer, NumInputChannels, NumOuputChannels, index, hostFrameOffset,userFrameOffset, swap); - break; - case ASIOSTFloat32MSB: - Output_Int8_Float32(nativeBuffer, outBufPtr, framePerBuffer, NumInputChannels, NumOuputChannels, index, hostFrameOffset,userFrameOffset, !swap); - break; - - case ASIOSTInt24LSB: // used for 20 bits as well - case ASIOSTInt24MSB: // used for 20 bits as well - - case ASIOSTFloat64LSB: // IEEE 754 64 bit double float, as found on Intel x86 architecture - case ASIOSTFloat64MSB: // IEEE 754 64 bit double float, as found on Intel x86 architecture - - // these are used for 32 bit data buffer, with different alignment of the data inside - // 32 bit PCI bus systems can more easily used with these - - case ASIOSTInt32LSB16: // 32 bit data with 16 bit alignment - case ASIOSTInt32LSB18: // 32 bit data with 18 bit alignment - case ASIOSTInt32LSB20: // 32 bit data with 20 bit alignment - case ASIOSTInt32LSB24: // 32 bit data with 24 bit alignment - - - case ASIOSTInt32MSB16: // 32 bit data with 16 bit alignment - case ASIOSTInt32MSB18: // 32 bit data with 18 bit alignment - case ASIOSTInt32MSB20: // 32 bit data with 20 bit alignment - case ASIOSTInt32MSB24: // 32 bit data with 24 bit alignment - DBUG(("Not yet implemented : please report the problem\n")); - break; - } - break; - } - - case paUInt8: - { - unsigned char *outBufPtr = (unsigned char *) outputBuffer; - - switch (nativeFormat) { - case ASIOSTInt16LSB: - Output_IntU8_Int16(nativeBuffer, outBufPtr, framePerBuffer, NumInputChannels, NumOuputChannels, index, hostFrameOffset,userFrameOffset, swap); - break; - case ASIOSTInt16MSB: - Output_IntU8_Int16(nativeBuffer, outBufPtr, framePerBuffer, NumInputChannels, NumOuputChannels, index, hostFrameOffset,userFrameOffset, !swap); - break; - case ASIOSTInt32LSB: - Output_IntU8_Int32(nativeBuffer, outBufPtr, framePerBuffer, NumInputChannels, NumOuputChannels, index, hostFrameOffset,userFrameOffset, swap); - break; - case ASIOSTInt32MSB: - Output_IntU8_Int32(nativeBuffer, outBufPtr, framePerBuffer, NumInputChannels, NumOuputChannels, index, hostFrameOffset,userFrameOffset, !swap); - break; - case ASIOSTFloat32LSB: - Output_IntU8_Float32(nativeBuffer, outBufPtr, framePerBuffer, NumInputChannels, NumOuputChannels, index, hostFrameOffset,userFrameOffset, swap); - break; - case ASIOSTFloat32MSB: - Output_IntU8_Float32(nativeBuffer, outBufPtr, framePerBuffer, NumInputChannels, NumOuputChannels, index, hostFrameOffset,userFrameOffset, !swap); - break; - - case ASIOSTInt24LSB: // used for 20 bits as well - case ASIOSTInt24MSB: // used for 20 bits as well - - case ASIOSTFloat64LSB: // IEEE 754 64 bit double float, as found on Intel x86 architecture - case ASIOSTFloat64MSB: // IEEE 754 64 bit double float, as found on Intel x86 architecture - - // these are used for 32 bit data buffer, with different alignment of the data inside - // 32 bit PCI bus systems can more easily used with these - - case ASIOSTInt32LSB16: // 32 bit data with 16 bit alignment - case ASIOSTInt32LSB18: // 32 bit data with 18 bit alignment - case ASIOSTInt32LSB20: // 32 bit data with 20 bit alignment - case ASIOSTInt32LSB24: // 32 bit data with 24 bit alignment - - - case ASIOSTInt32MSB16: // 32 bit data with 16 bit alignment - case ASIOSTInt32MSB18: // 32 bit data with 18 bit alignment - case ASIOSTInt32MSB20: // 32 bit data with 20 bit alignment - case ASIOSTInt32MSB24: // 32 bit data with 24 bit alignment - DBUG(("Not yet implemented : please report the problem\n")); - break; - } - break; - } - - default: - break; - } - } - -} - - - -/* Load a ASIO driver corresponding to the required device */ -static PaError Pa_ASIO_loadDevice (long device) -{ - PaDeviceInfo * dev = &(sDevices[device].pad_Info); - - if (!Pa_ASIO_loadAsioDriver((char *) dev->name)) return paHostError; - if (ASIOInit(&asioDriverInfo.pahsc_driverInfo) != ASE_OK) return paHostError; - if (ASIOGetChannels(&asioDriverInfo.pahsc_NumInputChannels, &asioDriverInfo.pahsc_NumOutputChannels) != ASE_OK) return paHostError; - if (ASIOGetBufferSize(&asioDriverInfo.pahsc_minSize, &asioDriverInfo.pahsc_maxSize, &asioDriverInfo.pahsc_preferredSize, &asioDriverInfo.pahsc_granularity) != ASE_OK) return paHostError; - - if(ASIOOutputReady() == ASE_OK) - asioDriverInfo.pahsc_postOutput = true; - else - asioDriverInfo.pahsc_postOutput = false; - - return paNoError; -} - -//--------------------------------------------------- -static int GetHighestBitPosition (unsigned long n) -{ - int pos = -1; - while( n != 0 ) - { - pos++; - n = n >> 1; - } - return pos; -} - -//------------------------------------------------------------------------------------------ -static int GetFirstMultiple(long min, long val ){ return ((min + val - 1) / val) * val; } - -//------------------------------------------------------------------------------------------ -static int GetFirstPossibleDivisor(long max, long val ) -{ - for (int i = 2; i < 20; i++) {if (((val%i) == 0) && ((val/i) <= max)) return (val/i); } - return val; -} - -//------------------------------------------------------------------------ -static int IsPowerOfTwo( unsigned long n ) { return ((n & (n-1)) == 0); } - -/******************************************************************* -* Determine size of native ASIO audio buffer size -* Input parameters : FramesPerUserBuffer, NumUserBuffers -* Output values : FramesPerHostBuffer, OutputBufferOffset or InputtBufferOffset -*/ - -static PaError PaHost_CalcNumHostBuffers( internalPortAudioStream *past ) -{ - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - long requestedBufferSize; - long firstMultiple, firstDivisor; - - // Compute requestedBufferSize - if( past->past_NumUserBuffers < 1 ){ - requestedBufferSize = past->past_FramesPerUserBuffer; - }else{ - requestedBufferSize = past->past_NumUserBuffers * past->past_FramesPerUserBuffer; - } - - // Adjust FramesPerHostBuffer using requestedBufferSize, ASIO minSize and maxSize, - if (requestedBufferSize < asioDriverInfo.pahsc_minSize){ - - firstMultiple = GetFirstMultiple(asioDriverInfo.pahsc_minSize, requestedBufferSize); - - if (firstMultiple <= asioDriverInfo.pahsc_maxSize) - asioDriverInfo.past_FramesPerHostBuffer = firstMultiple; - else - asioDriverInfo.past_FramesPerHostBuffer = asioDriverInfo.pahsc_minSize; - - }else if (requestedBufferSize > asioDriverInfo.pahsc_maxSize){ - - firstDivisor = GetFirstPossibleDivisor(asioDriverInfo.pahsc_maxSize, requestedBufferSize); - - if ((firstDivisor >= asioDriverInfo.pahsc_minSize) && (firstDivisor <= asioDriverInfo.pahsc_maxSize)) - asioDriverInfo.past_FramesPerHostBuffer = firstDivisor; - else - asioDriverInfo.past_FramesPerHostBuffer = asioDriverInfo.pahsc_maxSize; - }else{ - asioDriverInfo.past_FramesPerHostBuffer = requestedBufferSize; - } - - // If ASIO buffer size needs to be a power of two - if( asioDriverInfo.pahsc_granularity < 0 ){ - // Needs to be a power of two. - - if( !IsPowerOfTwo( asioDriverInfo.past_FramesPerHostBuffer ) ) - { - int highestBit = GetHighestBitPosition(asioDriverInfo.past_FramesPerHostBuffer); - asioDriverInfo.past_FramesPerHostBuffer = 1 << (highestBit + 1); - } - } - - DBUG(("----------------------------------\n")); - DBUG(("PaHost_CalcNumHostBuffers : minSize = %ld \n",asioDriverInfo.pahsc_minSize)); - DBUG(("PaHost_CalcNumHostBuffers : preferredSize = %ld \n",asioDriverInfo.pahsc_preferredSize)); - DBUG(("PaHost_CalcNumHostBuffers : maxSize = %ld \n",asioDriverInfo.pahsc_maxSize)); - DBUG(("PaHost_CalcNumHostBuffers : granularity = %ld \n",asioDriverInfo.pahsc_granularity)); - DBUG(("PaHost_CalcNumHostBuffers : User buffer size = %d\n", asioDriverInfo.past->past_FramesPerUserBuffer )); - DBUG(("PaHost_CalcNumHostBuffers : ASIO buffer size = %d\n", asioDriverInfo.past_FramesPerHostBuffer )); - - return paNoError; -} - - -/***********************************************************************/ -int Pa_CountDevices() -{ - PaError err ; - - if( sNumDevices <= 0 ) - { - /* Force loading of ASIO drivers */ - err = Pa_ASIO_QueryDeviceInfo(sDevices); - if( err != paNoError ) goto error; - } - - return sNumDevices; - -error: - PaHost_Term(); - DBUG(("Pa_CountDevices: returns %d\n", err )); - return err; -} - -/***********************************************************************/ -PaError PaHost_Init( void ) -{ - /* Have we already initialized the device info? */ - PaError err = (PaError) Pa_CountDevices(); - return ( err < 0 ) ? err : paNoError; -} - -/***********************************************************************/ -PaError PaHost_Term( void ) -{ - int i; - PaDeviceInfo *dev; - double *rates; - PaError result = paNoError; - - if (sNumDevices > 0) { - - /* Free allocated sample rate arrays and names*/ - for( i=0; i<sNumDevices; i++ ){ - dev = &sDevices[i].pad_Info; - rates = (double *) dev->sampleRates; - if ((rates != NULL)) PaHost_FreeFastMemory(rates, MAX_NUMSAMPLINGRATES * sizeof(double)); - dev->sampleRates = NULL; - if(dev->name != NULL) PaHost_FreeFastMemory((void *) dev->name, 32); - dev->name = NULL; - } - - sNumDevices = 0; - - /* If the stream has been closed with PaHost_CloseStream, asioDriverInfo.past == null, otherwise close it now */ - if(asioDriverInfo.past != NULL) Pa_CloseStream(asioDriverInfo.past); - - /* remove the loaded ASIO driver */ - asioDrivers->removeCurrentDriver(); - } - - return result; -} - -/***********************************************************************/ -PaError PaHost_OpenStream( internalPortAudioStream *past ) -{ - PaError result = paNoError; - ASIOError err; - int32 device; - - /* Check if a stream already runs */ - if (asioDriverInfo.past != NULL) return paHostError; - - /* Check the device number */ - if ((past->past_InputDeviceID != paNoDevice) - &&(past->past_OutputDeviceID != paNoDevice) - &&(past->past_InputDeviceID != past->past_OutputDeviceID)) - { - return paInvalidDeviceId; - } - - /* Allocation */ - memset(&asioDriverInfo, 0, sizeof(PaHostSoundControl)); - past->past_DeviceData = (void*) &asioDriverInfo; - - /* Reentrancy counter initialisation */ - asioDriverInfo.reenterCount = -1; - asioDriverInfo.reenterError = 0; - - /* FIXME */ - asioDriverInfo.past = past; - - /* load the ASIO device */ - device = (past->past_InputDeviceID < 0) ? past->past_OutputDeviceID : past->past_InputDeviceID; - result = Pa_ASIO_loadDevice(device); - if (result != paNoError) goto error; - - /* Check ASIO parameters and input parameters */ - if ((past->past_NumInputChannels > asioDriverInfo.pahsc_NumInputChannels) - || (past->past_NumOutputChannels > asioDriverInfo.pahsc_NumOutputChannels)) { - result = paInvalidChannelCount; - goto error; - } - - /* Set sample rate */ - if (ASIOSetSampleRate(past->past_SampleRate) != ASE_OK) { - result = paInvalidSampleRate; - goto error; - } - - /* if OK calc buffer size */ - result = PaHost_CalcNumHostBuffers( past ); - if (result != paNoError) goto error; - - - /* - Allocating input and output buffers number for the real past_NumInputChannels and past_NumOutputChannels - optimize the data transfer. - */ - - asioDriverInfo.pahsc_NumInputChannels = past->past_NumInputChannels; - asioDriverInfo.pahsc_NumOutputChannels = past->past_NumOutputChannels; - - /* Allocate ASIO buffers and callback*/ - err = Pa_ASIO_CreateBuffers(&asioDriverInfo, - asioDriverInfo.pahsc_NumInputChannels, - asioDriverInfo.pahsc_NumOutputChannels, - asioDriverInfo.past_FramesPerHostBuffer); - - - /* - Some buggy drivers (like the Hoontech DSP24) give incorrect [min, preferred, max] values - They should work with the preferred size value, thus if Pa_ASIO_CreateBuffers fails with - the hostBufferSize computed in PaHost_CalcNumHostBuffers, we try again with the preferred size. - */ - - if (err != ASE_OK) { - - DBUG(("PaHost_OpenStream : Pa_ASIO_CreateBuffers failed with the requested framesPerBuffer = %ld \n", asioDriverInfo.past_FramesPerHostBuffer)); - - err = Pa_ASIO_CreateBuffers(&asioDriverInfo, - asioDriverInfo.pahsc_NumInputChannels, - asioDriverInfo.pahsc_NumOutputChannels, - asioDriverInfo.pahsc_preferredSize); - - if (err == ASE_OK) { - // Adjust FramesPerHostBuffer to take the preferredSize instead of the value computed in PaHost_CalcNumHostBuffers - asioDriverInfo.past_FramesPerHostBuffer = asioDriverInfo.pahsc_preferredSize; - DBUG(("PaHost_OpenStream : Adjust FramesPerHostBuffer to take the preferredSize instead of the value computed in PaHost_CalcNumHostBuffers\n")); - } else { - DBUG(("PaHost_OpenStream : Pa_ASIO_CreateBuffers failed with the preferred framesPerBuffer = %ld \n", asioDriverInfo.pahsc_preferredSize)); - } - } - - /* Compute buffer adapdation offset */ - PaHost_CalcBufferOffset(past); - - if (err == ASE_OK) - return paNoError; - else if (err == ASE_NoMemory) - result = paInsufficientMemory; - else if (err == ASE_InvalidParameter) - result = paInvalidChannelCount; - else if (err == ASE_InvalidMode) - result = paBufferTooBig; - else - result = paHostError; - -error: - ASIOExit(); - return result; - -} - -/***********************************************************************/ -PaError PaHost_CloseStream( internalPortAudioStream *past ) -{ - PaHostSoundControl *pahsc; - PaError result = paNoError; - - if( past == NULL ) return paBadStreamPtr; - pahsc = (PaHostSoundControl *) past->past_DeviceData; - if( pahsc == NULL ) return paNoError; - - #if PA_TRACE_START_STOP - AddTraceMessage( "PaHost_CloseStream: pahsc_HWaveOut ", (int) pahsc->pahsc_HWaveOut ); - #endif - - /* Free data and device for output. */ - past->past_DeviceData = NULL; - asioDriverInfo.past = NULL; - - /* Dispose */ - if(ASIODisposeBuffers() != ASE_OK) result = paHostError; - if(ASIOExit() != ASE_OK) result = paHostError; - - return result; -} - -/***********************************************************************/ -PaError PaHost_StartOutput( internalPortAudioStream *past ) -{ - /* Clear the index 0 host output buffer */ - Pa_ASIO_Clear_Output(asioDriverInfo.bufferInfos, - asioDriverInfo.pahsc_channelInfos[0].type, - asioDriverInfo.pahsc_NumInputChannels, - asioDriverInfo.pahsc_NumOutputChannels, - 0, - 0, - asioDriverInfo.past_FramesPerHostBuffer); - - /* Clear the index 1 host output buffer */ - Pa_ASIO_Clear_Output(asioDriverInfo.bufferInfos, - asioDriverInfo.pahsc_channelInfos[0].type, - asioDriverInfo.pahsc_NumInputChannels, - asioDriverInfo.pahsc_NumOutputChannels, - 1, - 0, - asioDriverInfo.past_FramesPerHostBuffer); - - Pa_ASIO_Clear_User_Buffers(); - - Pa_ASIO_Adaptor_Init(); - - return paNoError; -} - -/***********************************************************************/ -PaError PaHost_StopOutput( internalPortAudioStream *past, int abort ) -{ - /* Nothing to do ?? */ - return paNoError; -} - -/***********************************************************************/ -PaError PaHost_StartInput( internalPortAudioStream *past ) -{ - /* Nothing to do ?? */ - return paNoError; -} - -/***********************************************************************/ -PaError PaHost_StopInput( internalPortAudioStream *past, int abort ) -{ - /* Nothing to do */ - return paNoError; -} - -/***********************************************************************/ -PaError PaHost_StartEngine( internalPortAudioStream *past ) -{ - // TO DO : count of samples - past->past_IsActive = 1; - return (ASIOStart() == ASE_OK) ? paNoError : paHostError; -} - -/***********************************************************************/ -PaError PaHost_StopEngine( internalPortAudioStream *past, int abort ) -{ - // TO DO : count of samples - past->past_IsActive = 0; - return (ASIOStop() == ASE_OK) ? paNoError : paHostError; -} - -/***********************************************************************/ -// TO BE CHECKED -PaError PaHost_StreamActive( internalPortAudioStream *past ) -{ - PaHostSoundControl *pahsc; - if( past == NULL ) return paBadStreamPtr; - pahsc = (PaHostSoundControl *) past->past_DeviceData; - if( pahsc == NULL ) return paInternalError; - return (PaError) past->past_IsActive; -} - -/*************************************************************************/ -PaTimestamp Pa_StreamTime( PortAudioStream *stream ) -{ - PaHostSoundControl *pahsc; - internalPortAudioStream *past = (internalPortAudioStream *) stream; - if( past == NULL ) return paBadStreamPtr; - pahsc = (PaHostSoundControl *) past->past_DeviceData; - return pahsc->pahsc_NumFramesDone; -} - -/************************************************************************* - * Allocate memory that can be accessed in real-time. - * This may need to be held in physical memory so that it is not - * paged to virtual memory. - * This call MUST be balanced with a call to PaHost_FreeFastMemory(). - */ -void *PaHost_AllocateFastMemory( long numBytes ) -{ - #if MAC - void *addr = NewPtrClear( numBytes ); - if( (addr == NULL) || (MemError () != 0) ) return NULL; - - #if (CARBON_COMPATIBLE == 0) - if( HoldMemory( addr, numBytes ) != noErr ) - { - DisposePtr( (Ptr) addr ); - return NULL; - } - #endif - return addr; - #elif WINDOWS - void *addr = malloc( numBytes ); /* FIXME - do we need physical memory? */ - if( addr != NULL ) memset( addr, 0, numBytes ); - return addr; - #endif -} - -/************************************************************************* - * Free memory that could be accessed in real-time. - * This call MUST be balanced with a call to PaHost_AllocateFastMemory(). - */ -void PaHost_FreeFastMemory( void *addr, long numBytes ) -{ - #if MAC - if( addr == NULL ) return; - #if CARBON_COMPATIBLE - (void) numBytes; - #else - UnholdMemory( addr, numBytes ); - #endif - DisposePtr( (Ptr) addr ); - #elif WINDOWS - if( addr != NULL ) free( addr ); - #endif -} - - -/*************************************************************************/ -void Pa_Sleep( long msec ) -{ - #if MAC - int32 sleepTime, endTime; - /* Convert to ticks. Round up so we sleep a MINIMUM of msec time. */ - sleepTime = ((msec * 60) + 999) / 1000; - if( sleepTime < 1 ) sleepTime = 1; - endTime = TickCount() + sleepTime; - do{ - DBUGX(("Sleep for %d ticks.\n", sleepTime )); - WaitNextEvent( 0, NULL, sleepTime, NULL ); /* Use this just to sleep without getting events. */ - sleepTime = endTime - TickCount(); - } while( sleepTime > 0 ); - #elif WINDOWS - Sleep( msec ); - #endif -} - -/*************************************************************************/ -const PaDeviceInfo* Pa_GetDeviceInfo( PaDeviceID id ) -{ - if( (id < 0) || ( id >= Pa_CountDevices()) ) return NULL; - return &sDevices[id].pad_Info; -} - -/*************************************************************************/ -PaDeviceID Pa_GetDefaultInputDeviceID( void ) -{ - return (sNumDevices > 0) ? sDefaultInputDeviceID : paNoDevice; -} - -/*************************************************************************/ -PaDeviceID Pa_GetDefaultOutputDeviceID( void ) -{ - return (sNumDevices > 0) ? sDefaultOutputDeviceID : paNoDevice; -} - -/*************************************************************************/ -int Pa_GetMinNumBuffers( int framesPerUserBuffer, double sampleRate ) -{ - // TO BE IMPLEMENTED : using the ASIOGetLatency call?? - return 2; -} - -/*************************************************************************/ -int32 Pa_GetHostError( void ) -{ - int32 err = sPaHostError; - sPaHostError = 0; - return err; -} - - -#ifdef MAC - -/**************************************************************************/ -static void Pa_StartUsageCalculation( internalPortAudioStream *past ) -{ - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - UnsignedWide widePad; - if( pahsc == NULL ) return; -/* Query system timer for usage analysis and to prevent overuse of CPU. */ - Microseconds( &widePad ); - pahsc->pahsc_EntryCount = UnsignedWideToUInt64( widePad ); -} -/**************************************************************************/ -static void Pa_EndUsageCalculation( internalPortAudioStream *past ) -{ - UnsignedWide widePad; - UInt64 CurrentCount; - long InsideCount; - long TotalCount; - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - if( pahsc == NULL ) return; -/* Measure CPU utilization during this callback. Note that this calculation -** assumes that we had the processor the whole time. -*/ -#define LOWPASS_COEFFICIENT_0 (0.9) -#define LOWPASS_COEFFICIENT_1 (0.99999 - LOWPASS_COEFFICIENT_0) - Microseconds( &widePad ); - CurrentCount = UnsignedWideToUInt64( widePad ); - if( past->past_IfLastExitValid ) - { - InsideCount = (long) U64Subtract(CurrentCount, pahsc->pahsc_EntryCount); - TotalCount = (long) U64Subtract(CurrentCount, pahsc->pahsc_LastExitCount); -/* Low pass filter the result because sometimes we get called several times in a row. -* That can cause the TotalCount to be very low which can cause the usage to appear -* unnaturally high. So we must filter numerator and denominator separately!!! -*/ - past->past_AverageInsideCount = (( LOWPASS_COEFFICIENT_0 * past->past_AverageInsideCount) + - (LOWPASS_COEFFICIENT_1 * InsideCount)); - past->past_AverageTotalCount = (( LOWPASS_COEFFICIENT_0 * past->past_AverageTotalCount) + - (LOWPASS_COEFFICIENT_1 * TotalCount)); - past->past_Usage = past->past_AverageInsideCount / past->past_AverageTotalCount; - } - pahsc->pahsc_LastExitCount = CurrentCount; - past->past_IfLastExitValid = 1; -} - -#elif WINDOWS - -/********************************* BEGIN CPU UTILIZATION MEASUREMENT ****/ -static void Pa_StartUsageCalculation( internalPortAudioStream *past ) -{ - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - if( pahsc == NULL ) return; -/* Query system timer for usage analysis and to prevent overuse of CPU. */ - QueryPerformanceCounter( &pahsc->pahsc_EntryCount ); -} - -static void Pa_EndUsageCalculation( internalPortAudioStream *past ) -{ - LARGE_INTEGER CurrentCount = { 0, 0 }; - LONGLONG InsideCount; - LONGLONG TotalCount; -/* -** Measure CPU utilization during this callback. Note that this calculation -** assumes that we had the processor the whole time. -*/ -#define LOWPASS_COEFFICIENT_0 (0.9) -#define LOWPASS_COEFFICIENT_1 (0.99999 - LOWPASS_COEFFICIENT_0) - - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - if( pahsc == NULL ) return; - - if( QueryPerformanceCounter( &CurrentCount ) ) - { - if( past->past_IfLastExitValid ) - { - InsideCount = CurrentCount.QuadPart - pahsc->pahsc_EntryCount.QuadPart; - TotalCount = CurrentCount.QuadPart - pahsc->pahsc_LastExitCount.QuadPart; -/* Low pass filter the result because sometimes we get called several times in a row. - * That can cause the TotalCount to be very low which can cause the usage to appear - * unnaturally high. So we must filter numerator and denominator separately!!! - */ - past->past_AverageInsideCount = (( LOWPASS_COEFFICIENT_0 * past->past_AverageInsideCount) + - (LOWPASS_COEFFICIENT_1 * InsideCount)); - past->past_AverageTotalCount = (( LOWPASS_COEFFICIENT_0 * past->past_AverageTotalCount) + - (LOWPASS_COEFFICIENT_1 * TotalCount)); - past->past_Usage = past->past_AverageInsideCount / past->past_AverageTotalCount; - } - pahsc->pahsc_LastExitCount = CurrentCount; - past->past_IfLastExitValid = 1; - } -} - -#endif - - - - diff --git a/pd/portaudio_v18/pa_asio/readme_asio_sdk_patch.txt b/pd/portaudio_v18/pa_asio/readme_asio_sdk_patch.txt deleted file mode 100644 index c0fdca7f..00000000 --- a/pd/portaudio_v18/pa_asio/readme_asio_sdk_patch.txt +++ /dev/null @@ -1,25 +0,0 @@ -There is a bug in the ASIO SDK that causes the Macintosh version to often fail during initialization. Here is a patch that you can apply. - -In codefragments.cpp replace getFrontProcessDirectory function with -the following one (GetFrontProcess replaced by GetCurrentProcess) - - -bool CodeFragments::getFrontProcessDirectory(void *specs) -{ - FSSpec *fss = (FSSpec *)specs; - ProcessInfoRec pif; - ProcessSerialNumber psn; - - memset(&psn,0,(long)sizeof(ProcessSerialNumber)); - // if(GetFrontProcess(&psn) == noErr) // wrong !!! - if(GetCurrentProcess(&psn) == noErr) // correct !!! - { - pif.processName = 0; - pif.processAppSpec = fss; - pif.processInfoLength = sizeof(ProcessInfoRec); - if(GetProcessInformation(&psn, &pif) == noErr) - return true; - } - return false; -} - diff --git a/pd/portaudio_v18/pa_beos/PlaybackNode.cc b/pd/portaudio_v18/pa_beos/PlaybackNode.cc deleted file mode 100644 index a93b8412..00000000 --- a/pd/portaudio_v18/pa_beos/PlaybackNode.cc +++ /dev/null @@ -1,538 +0,0 @@ -/* - * $Id: PlaybackNode.cc,v 1.1.1.1 2002/01/22 00:52:07 phil Exp $ - * PortAudio Portable Real-Time Audio Library - * Latest Version at: http://www.portaudio.com - * BeOS Media Kit Implementation by Joshua Haberman - * - * Copyright (c) 2001 Joshua Haberman <joshua@haberman.com> - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - * --- - * - * Significant portions of this file are based on sample code from Be. The - * Be Sample Code Licence follows: - * - * Copyright 1991-1999, Be Incorporated. - * All rights reserved. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions - * are met: - * - * 1. Redistributions of source code must retain the above copyright - * notice, this list of conditions, and the following disclaimer. - * - * 2. Redistributions in binary form must reproduce the above copyright - * notice, this list of conditions, and the following disclaimer in the - * documentation and/or other materials provided with the distribution. - * - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR "AS IS" AND ANY EXPRESS OR - * IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES - * OF TITLE, NON-INFRINGEMENT, MERCHANTABILITY AND FITNESS FOR A PARTICULAR - * PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY - * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES - * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; - * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED - * AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR - * TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE - * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#include <stdio.h> - -#include <be/media/BufferGroup.h> -#include <be/media/Buffer.h> -#include <be/media/TimeSource.h> - -#include "PlaybackNode.h" - -#define PRINT(x) { printf x; fflush(stdout); } - -#ifdef DEBUG -#define DBUG(x) PRINT(x) -#else -#define DBUG(x) -#endif - - -PaPlaybackNode::PaPlaybackNode(uint32 channels, float frame_rate, uint32 frames_per_buffer, - PortAudioCallback* callback, void *user_data) : - BMediaNode("PortAudio input node"), - BBufferProducer(B_MEDIA_RAW_AUDIO), - BMediaEventLooper(), - mAborted(false), - mRunning(false), - mBufferGroup(NULL), - mDownstreamLatency(0), - mStartTime(0), - mCallback(callback), - mUserData(user_data), - mFramesPerBuffer(frames_per_buffer) -{ - DBUG(("Constructor called.\n")); - - mPreferredFormat.type = B_MEDIA_RAW_AUDIO; - mPreferredFormat.u.raw_audio.channel_count = channels; - mPreferredFormat.u.raw_audio.frame_rate = frame_rate; - mPreferredFormat.u.raw_audio.byte_order = - (B_HOST_IS_BENDIAN) ? B_MEDIA_BIG_ENDIAN : B_MEDIA_LITTLE_ENDIAN; - mPreferredFormat.u.raw_audio.buffer_size = - media_raw_audio_format::wildcard.buffer_size; - - mOutput.destination = media_destination::null; - mOutput.format = mPreferredFormat; - - /* The amount of time it takes for this node to produce a buffer when - * asked. Essentially, it is how long the user's callback takes to run. - * We set this to be the length of the sound data each buffer of the - * requested size can hold. */ - //mInternalLatency = (bigtime_t)(1000000 * frames_per_buffer / frame_rate); - - /* ACK! it seems that the mixer (at least on my machine) demands that IT - * specify the buffer size, so for now I'll just make a generic guess here */ - mInternalLatency = 1000000 / 20; -} - - - -PaPlaybackNode::~PaPlaybackNode() -{ - DBUG(("Destructor called.\n")); - Quit(); /* Stop the BMediaEventLooper thread */ -} - - -/************************* - * - * Local methods - * - */ - -bool PaPlaybackNode::IsRunning() -{ - return mRunning; -} - - -PaTimestamp PaPlaybackNode::GetStreamTime() -{ - BTimeSource *timeSource = TimeSource(); - PaTimestamp time = (timeSource->Now() - mStartTime) * - mPreferredFormat.u.raw_audio.frame_rate / 1000000; - return time; -} - - -void PaPlaybackNode::SetSampleFormat(PaSampleFormat inFormat, - PaSampleFormat outFormat) -{ - uint32 beOutFormat; - - switch(outFormat) - { - case paFloat32: - beOutFormat = media_raw_audio_format::B_AUDIO_FLOAT; - mOutputSampleWidth = 4; - break; - - case paInt16: - beOutFormat = media_raw_audio_format::B_AUDIO_SHORT; - mOutputSampleWidth = 2; - break; - - case paInt32: - beOutFormat = media_raw_audio_format::B_AUDIO_INT; - mOutputSampleWidth = 4; - break; - - case paInt8: - beOutFormat = media_raw_audio_format::B_AUDIO_CHAR; - mOutputSampleWidth = 1; - break; - - case paUInt8: - beOutFormat = media_raw_audio_format::B_AUDIO_UCHAR; - mOutputSampleWidth = 1; - break; - - case paInt24: - case paPackedInt24: - case paCustomFormat: - DBUG(("Unsupported output format: %x\n", outFormat)); - break; - - default: - DBUG(("Unknown output format: %x\n", outFormat)); - } - - mPreferredFormat.u.raw_audio.format = beOutFormat; - mFramesPerBuffer * mPreferredFormat.u.raw_audio.channel_count * mOutputSampleWidth; -} - -BBuffer *PaPlaybackNode::FillNextBuffer(bigtime_t time) -{ - /* Get a buffer from the buffer group */ - BBuffer *buf = mBufferGroup->RequestBuffer( - mOutput.format.u.raw_audio.buffer_size, BufferDuration()); - unsigned long frames = mOutput.format.u.raw_audio.buffer_size / - mOutputSampleWidth / mOutput.format.u.raw_audio.channel_count; - bigtime_t start_time; - int ret; - - if( !buf ) - { - DBUG(("Unable to allocate a buffer\n")); - return NULL; - } - - start_time = mStartTime + - (bigtime_t)((double)mSamplesSent / - (double)mOutput.format.u.raw_audio.frame_rate / - (double)mOutput.format.u.raw_audio.channel_count * - 1000000.0); - - /* Now call the user callback to get the data */ - ret = mCallback(NULL, /* Input buffer */ - buf->Data(), /* Output buffer */ - frames, /* Frames per buffer */ - mSamplesSent / mOutput.format.u.raw_audio.channel_count, /* timestamp */ - mUserData); - - if( ret ) - mAborted = true; - - media_header *hdr = buf->Header(); - - hdr->type = B_MEDIA_RAW_AUDIO; - hdr->size_used = mOutput.format.u.raw_audio.buffer_size; - hdr->time_source = TimeSource()->ID(); - hdr->start_time = start_time; - - return buf; -} - - - - -/************************* - * - * BMediaNode methods - * - */ - -BMediaAddOn *PaPlaybackNode::AddOn( int32 * ) const -{ - DBUG(("AddOn() called.\n")); - return NULL; /* we don't provide service to outside applications */ -} - - -status_t PaPlaybackNode::HandleMessage( int32 message, const void *data, - size_t size ) -{ - DBUG(("HandleMessage() called.\n")); - return B_ERROR; /* we don't define any custom messages */ -} - - - - -/************************* - * - * BMediaEventLooper methods - * - */ - -void PaPlaybackNode::NodeRegistered() -{ - DBUG(("NodeRegistered() called.\n")); - - /* Start the BMediaEventLooper thread */ - SetPriority(B_REAL_TIME_PRIORITY); - Run(); - - /* set up as much information about our output as we can */ - mOutput.source.port = ControlPort(); - mOutput.source.id = 0; - mOutput.node = Node(); - ::strcpy(mOutput.name, "PortAudio Playback"); -} - - -void PaPlaybackNode::HandleEvent( const media_timed_event *event, - bigtime_t lateness, bool realTimeEvent ) -{ - // DBUG(("HandleEvent() called.\n")); - status_t err; - - switch(event->type) - { - case BTimedEventQueue::B_START: - DBUG((" Handling a B_START event\n")); - if( RunState() != B_STARTED ) - { - mStartTime = event->event_time + EventLatency(); - mSamplesSent = 0; - mAborted = false; - mRunning = true; - media_timed_event firstEvent( mStartTime, - BTimedEventQueue::B_HANDLE_BUFFER ); - EventQueue()->AddEvent( firstEvent ); - } - break; - - case BTimedEventQueue::B_STOP: - DBUG((" Handling a B_STOP event\n")); - mRunning = false; - EventQueue()->FlushEvents( 0, BTimedEventQueue::B_ALWAYS, true, - BTimedEventQueue::B_HANDLE_BUFFER ); - break; - - case BTimedEventQueue::B_HANDLE_BUFFER: - //DBUG((" Handling a B_HANDLE_BUFFER event\n")); - - /* make sure we're started and connected */ - if( RunState() != BMediaEventLooper::B_STARTED || - mOutput.destination == media_destination::null ) - break; - - BBuffer *buffer = FillNextBuffer(event->event_time); - - /* make sure we weren't aborted while this routine was running. - * this can happen in one of two ways: either the callback returned - * nonzero (in which case mAborted is set in FillNextBuffer() ) or - * the client called AbortStream */ - if( mAborted ) - { - if( buffer ) - buffer->Recycle(); - Stop(0, true); - break; - } - - if( buffer ) - { - err = SendBuffer(buffer, mOutput.destination); - if( err != B_OK ) - buffer->Recycle(); - } - - mSamplesSent += mOutput.format.u.raw_audio.buffer_size / mOutputSampleWidth; - - /* Now schedule the next buffer event, so we can send another - * buffer when this one runs out. We calculate when it should - * happen by calculating when the data we just sent will finish - * playing. - * - * NOTE, however, that the event will actually get generated - * earlier than we specify, to account for the latency it will - * take to produce the buffer. It uses the latency value we - * specified in SetEventLatency() to determine just how early - * to generate it. */ - - /* totalPerformanceTime includes the time represented by the buffer - * we just sent */ - bigtime_t totalPerformanceTime = (bigtime_t)((double)mSamplesSent / - (double)mOutput.format.u.raw_audio.channel_count / - (double)mOutput.format.u.raw_audio.frame_rate * 1000000.0); - - bigtime_t nextEventTime = mStartTime + totalPerformanceTime; - - media_timed_event nextBufferEvent(nextEventTime, - BTimedEventQueue::B_HANDLE_BUFFER); - EventQueue()->AddEvent(nextBufferEvent); - - break; - - } -} - - - - -/************************* - * - * BBufferProducer methods - * - */ - -status_t PaPlaybackNode::FormatSuggestionRequested( media_type type, - int32 /*quality*/, media_format* format ) -{ - /* the caller wants to know this node's preferred format and provides - * a suggestion, asking if we support it */ - DBUG(("FormatSuggestionRequested() called.\n")); - - if(!format) - return B_BAD_VALUE; - - *format = mPreferredFormat; - - /* we only support raw audio (a wildcard is okay too) */ - if ( type == B_MEDIA_UNKNOWN_TYPE || type == B_MEDIA_RAW_AUDIO ) - return B_OK; - else - return B_MEDIA_BAD_FORMAT; -} - - -status_t PaPlaybackNode::FormatProposal( const media_source& output, - media_format* format ) -{ - /* This is similar to FormatSuggestionRequested(), but it is actually part - * of the negotiation process. We're given the opportunity to specify any - * properties that are wildcards (ie. properties that the other node doesn't - * care one way or another about) */ - DBUG(("FormatProposal() called.\n")); - - /* Make sure this proposal really applies to our output */ - if( output != mOutput.source ) - return B_MEDIA_BAD_SOURCE; - - /* We return two things: whether we support the proposed format, and our own - * preferred format */ - *format = mPreferredFormat; - - if( format->type == B_MEDIA_UNKNOWN_TYPE || format->type == B_MEDIA_RAW_AUDIO ) - return B_OK; - else - return B_MEDIA_BAD_FORMAT; -} - - -status_t PaPlaybackNode::FormatChangeRequested( const media_source& source, - const media_destination& destination, media_format* io_format, int32* ) -{ - /* we refuse to change formats, supporting only 1 */ - DBUG(("FormatChangeRequested() called.\n")); - - return B_ERROR; -} - - -status_t PaPlaybackNode::GetNextOutput( int32* cookie, media_output* out_output ) -{ - /* this is where we allow other to enumerate our outputs -- the cookie is - * an integer we can use to keep track of where we are in enumeration. */ - DBUG(("GetNextOutput() called.\n")); - - if( *cookie == 0 ) - { - *out_output = mOutput; - *cookie = 1; - return B_OK; - } - - return B_BAD_INDEX; -} - - -status_t PaPlaybackNode::DisposeOutputCookie( int32 cookie ) -{ - DBUG(("DisposeOutputCookie() called.\n")); - return B_OK; -} - - -void PaPlaybackNode::LateNoticeReceived( const media_source& what, - bigtime_t how_much, bigtime_t performance_time ) -{ - /* This function is called as notification that a buffer we sent wasn't - * received by the time we stamped it with -- it got there late. Basically, - * it means we underestimated our own latency, so we should increase it */ - DBUG(("LateNoticeReceived() called.\n")); - - if( what != mOutput.source ) - return; - - if( RunMode() == B_INCREASE_LATENCY ) - { - mInternalLatency += how_much; - SetEventLatency( mDownstreamLatency + mInternalLatency ); - DBUG(("Increasing latency to %Ld\n", mDownstreamLatency + mInternalLatency)); - } - else - DBUG(("I don't know what to do with this notice!")); -} - - -void PaPlaybackNode::EnableOutput( const media_source& what, bool enabled, - int32* ) -{ - DBUG(("EnableOutput() called.\n")); - /* stub -- we don't support this yet */ -} - - -status_t PaPlaybackNode::PrepareToConnect( const media_source& what, - const media_destination& where, media_format* format, - media_source* out_source, char* out_name ) -{ - /* the final stage of format negotiations. here we _must_ make specific any - * remaining wildcards */ - DBUG(("PrepareToConnect() called.\n")); - - /* make sure this really refers to our source */ - if( what != mOutput.source ) - return B_MEDIA_BAD_SOURCE; - - /* make sure we're not already connected */ - if( mOutput.destination != media_destination::null ) - return B_MEDIA_ALREADY_CONNECTED; - - if( format->type != B_MEDIA_RAW_AUDIO ) - return B_MEDIA_BAD_FORMAT; - - if( format->u.raw_audio.format != mPreferredFormat.u.raw_audio.format ) - return B_MEDIA_BAD_FORMAT; - - if( format->u.raw_audio.buffer_size == - media_raw_audio_format::wildcard.buffer_size ) - { - DBUG(("We were left to decide buffer size: choosing 2048")); - format->u.raw_audio.buffer_size = 2048; - } - else - DBUG(("Using consumer specified buffer size of %lu.\n", - format->u.raw_audio.buffer_size)); - - /* Reserve the connection, return the information */ - mOutput.destination = where; - mOutput.format = *format; - *out_source = mOutput.source; - strncpy( out_name, mOutput.name, B_MEDIA_NAME_LENGTH ); - - return B_OK; -} - - -void PaPlaybackNode::Connect(status_t error, const media_source& source, - const media_destination& destination, const media_format& format, char* io_name) -{ - DBUG(("Connect() called.\n")); - diff --git a/pd/portaudio_v18/pa_beos/PlaybackNode.h b/pd/portaudio_v18/pa_beos/PlaybackNode.h deleted file mode 100644 index c9582056..00000000 --- a/pd/portaudio_v18/pa_beos/PlaybackNode.h +++ /dev/null @@ -1,108 +0,0 @@ -/* - * $Id: PlaybackNode.h,v 1.1.1.1 2002/01/22 00:52:08 phil Exp $ - * PortAudio Portable Real-Time Audio Library - * Latest Version at: http://www.portaudio.com - * BeOS Media Kit Implementation by Joshua Haberman - * - * Copyright (c) 2001 Joshua Haberman <joshua@haberman.com> - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ - -#include <be/media/MediaRoster.h> -#include <be/media/MediaEventLooper.h> -#include <be/media/BufferProducer.h> - -#include "portaudio.h" - -class PaPlaybackNode : - public BBufferProducer, - public BMediaEventLooper -{ - -public: - PaPlaybackNode( uint32 channels, float frame_rate, uint32 frames_per_buffer, - PortAudioCallback *callback, void *user_data ); - ~PaPlaybackNode(); - - - /* Local methods ******************************************/ - - BBuffer *FillNextBuffer(bigtime_t time); - void SetSampleFormat(PaSampleFormat inFormat, PaSampleFormat outFormat); - bool IsRunning(); - PaTimestamp GetStreamTime(); - - /* BMediaNode methods *************************************/ - - BMediaAddOn* AddOn( int32 * ) const; - status_t HandleMessage( int32 message, const void *data, size_t size ); - - /* BMediaEventLooper methods ******************************/ - - void HandleEvent( const media_timed_event *event, bigtime_t lateness, - bool realTimeEvent ); - void NodeRegistered(); - - /* BBufferProducer methods ********************************/ - - status_t FormatSuggestionRequested( media_type type, int32 quality, - media_format* format ); - status_t FormatProposal( const media_source& output, media_format* format ); - status_t FormatChangeRequested( const media_source& source, - const media_destination& destination, media_format* io_format, int32* ); - - status_t GetNextOutput( int32* cookie, media_output* out_output ); - status_t DisposeOutputCookie( int32 cookie ); - - void LateNoticeReceived( const media_source& what, bigtime_t how_much, - bigtime_t performance_time ); - void EnableOutput( const media_source& what, bool enabled, int32* _deprecated_ ); - - status_t PrepareToConnect( const media_source& what, - const media_destination& where, media_format* format, - media_source* out_source, char* out_name ); - void Connect(status_t error, const media_source& source, - const media_destination& destination, const media_format& format, - char* io_name); - void Disconnect(const media_source& what, const media_destination& where); - - status_t SetBufferGroup(const media_source& for_source, BBufferGroup* newGroup); - - bool mAborted; - -private: - media_output mOutput; - media_format mPreferredFormat; - uint32 mOutputSampleWidth, mFramesPerBuffer; - BBufferGroup *mBufferGroup; - bigtime_t mDownstreamLatency, mInternalLatency, mStartTime; - uint64 mSamplesSent; - PortAudioCallback *mCallback; - void *mUserData; - bool mRunning; - -}; - diff --git a/pd/portaudio_v18/pa_beos/pa_beos_mk.cc b/pd/portaudio_v18/pa_beos/pa_beos_mk.cc deleted file mode 100644 index ff35cdc3..00000000 --- a/pd/portaudio_v18/pa_beos/pa_beos_mk.cc +++ /dev/null @@ -1,441 +0,0 @@ -/* - * $Id: pa_beos_mk.cc,v 1.1.1.1 2002/01/22 00:52:09 phil Exp $ - * PortAudio Portable Real-Time Audio Library - * Latest Version at: http://www.portaudio.com - * BeOS Media Kit Implementation by Joshua Haberman - * - * Copyright (c) 2001 Joshua Haberman <joshua@haberman.com> - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ - -#include <be/app/Application.h> -#include <be/kernel/OS.h> -#include <be/media/RealtimeAlloc.h> -#include <be/media/MediaRoster.h> -#include <be/media/TimeSource.h> - -#include <stdio.h> -#include <string.h> - -#include "portaudio.h" -#include "pa_host.h" - -#include "PlaybackNode.h" - -#define PRINT(x) { printf x; fflush(stdout); } - -#ifdef DEBUG -#define DBUG(x) PRINT(x) -#else -#define DBUG(x) -#endif - -typedef struct PaHostSoundControl -{ - /* These members are common to all modes of operation */ - media_node pahsc_TimeSource; /* the sound card's DAC. */ - media_format pahsc_Format; - - /* These methods are specific to playing mode */ - media_node pahsc_OutputNode; /* output to the mixer */ - media_node pahsc_InputNode; /* reads data from user callback -- PA specific */ - - media_input pahsc_MixerInput; /* input jack on the soundcard's mixer. */ - media_output pahsc_PaOutput; /* output jack from the PA node */ - - PaPlaybackNode *pahsc_InputNodeInstance; - -} -PaHostSoundControl; - -/*************************************************************************/ -PaDeviceID Pa_GetDefaultOutputDeviceID( void ) -{ - /* stub */ - return 0; -} - -/*************************************************************************/ -PaDeviceID Pa_GetDefaultInputDeviceID( void ) -{ - /* stub */ - return 0; -} - -/*************************************************************************/ -const PaDeviceInfo* Pa_GetDeviceInfo( PaDeviceID id ) -{ - /* stub */ - return NULL; -} - -/*************************************************************************/ -int Pa_CountDevices() -{ - /* stub */ - return 1; -} - -/*************************************************************************/ -PaError PaHost_Init( void ) -{ - /* we have to create this in order to use BMediaRoster. I hope it doesn't - * cause problems */ - be_app = new BApplication("application/x-vnd.portaudio-app"); - - return paNoError; -} - -PaError PaHost_Term( void ) -{ - delete be_app; - return paNoError; -} - -/*************************************************************************/ -PaError PaHost_StreamActive( internalPortAudioStream *past ) -{ - PaHostSoundControl *pahsc = (PaHostSoundControl *)past->past_DeviceData; - DBUG(("IsRunning returning: %s\n", - pahsc->pahsc_InputNodeInstance->IsRunning() ? "true" : "false")); - - return (PaError)pahsc->pahsc_InputNodeInstance->IsRunning(); -} - -PaError PaHost_StartOutput( internalPortAudioStream *past ) -{ - return paNoError; -} - -/*************************************************************************/ -PaError PaHost_StartInput( internalPortAudioStream *past ) -{ - return paNoError; -} - -/*************************************************************************/ -PaError PaHost_StopInput( internalPortAudioStream *past, int abort ) -{ - return paNoError; -} - -/*************************************************************************/ -PaError PaHost_StopOutput( internalPortAudioStream *past, int abort ) -{ - return paNoError; -} - - -/*************************************************************************/ -PaError PaHost_StartEngine( internalPortAudioStream *past ) -{ - bigtime_t very_soon, start_latency; - status_t err; - BMediaRoster *roster = BMediaRoster::Roster(&err); - PaHostSoundControl *pahsc = (PaHostSoundControl *)past->past_DeviceData; - - /* for some reason, err indicates an error (though nothing it wrong) - * when the DBUG macro in pa_lib.c is enabled. It's reproducably - * linked. Weird. */ - if( !roster /* || err != B_OK */ ) - { - DBUG(("No media server! err=%d, roster=%x\n", err, roster)); - return paHostError; - } - - /* tell the node when to start -- since there aren't any other nodes - * starting that we have to wait for, just tell it to start now - */ - - BTimeSource *timeSource = roster->MakeTimeSourceFor(pahsc->pahsc_TimeSource); - very_soon = timeSource->PerformanceTimeFor( BTimeSource::RealTime() ); - timeSource->Release(); - - /* Add the latency of starting the network of nodes */ - err = roster->GetStartLatencyFor( pahsc->pahsc_TimeSource, &start_latency ); - very_soon += start_latency; - - err = roster->StartNode( pahsc->pahsc_InputNode, very_soon ); - /* No need to start the mixer -- it's always running */ - - return paNoError; -} - - -/*************************************************************************/ -PaError PaHost_StopEngine( internalPortAudioStream *past, int abort ) -{ - PaHostSoundControl *pahsc = (PaHostSoundControl *)past->past_DeviceData; - BMediaRoster *roster = BMediaRoster::Roster(); - - if( !roster ) - { - DBUG(("No media roster!\n")); - return paHostError; - } - - if( !pahsc ) - return paHostError; - - /* this crashes, and I don't know why yet */ - // if( abort ) - // pahsc->pahsc_InputNodeInstance->mAborted = true; - - roster->StopNode(pahsc->pahsc_InputNode, 0, /* immediate = */ true); - - return paNoError; -} - - -/*************************************************************************/ -PaError PaHost_OpenStream( internalPortAudioStream *past ) -{ - status_t err; - BMediaRoster *roster = BMediaRoster::Roster(&err); - PaHostSoundControl *pahsc; - - /* Allocate and initialize host data. */ - pahsc = (PaHostSoundControl *) PaHost_AllocateFastMemory(sizeof(PaHostSoundControl)); - if( pahsc == NULL ) - { - goto error; - } - memset( pahsc, 0, sizeof(PaHostSoundControl) ); - past->past_DeviceData = (void *) pahsc; - - if( !roster /* || err != B_OK */ ) - { - /* no media server! */ - DBUG(("No media server.\n")); - goto error; - } - - if ( past->past_NumInputChannels > 0 && past->past_NumOutputChannels > 0 ) - { - /* filter -- not implemented yet */ - goto error; - } - else if ( past->past_NumInputChannels > 0 ) - { - /* recorder -- not implemented yet */ - goto error; - } - else - { - /* player ****************************************************************/ - - status_t err; - int32 num; - - /* First we need to create the three components (like components in a stereo - * system). The mixer component is our interface to the sound card, data - * we write there will get played. The BePA_InputNode component is the node - * which represents communication with the PA client (it is what calls the - * client's callbacks). The time source component is the sound card's DAC, - * which allows us to slave the other components to it instead of the system - * clock. */ - err = roster->GetAudioMixer( &pahsc->pahsc_OutputNode ); - if( err != B_OK ) - { - DBUG(("Couldn't get default mixer.\n")); - goto error; - } - - err = roster->GetTimeSource( &pahsc->pahsc_TimeSource ); - if( err != B_OK ) - { - DBUG(("Couldn't get time source.\n")); - goto error; - } - - pahsc->pahsc_InputNodeInstance = new PaPlaybackNode(2, 44100, - past->past_FramesPerUserBuffer, past->past_Callback, past->past_UserData ); - pahsc->pahsc_InputNodeInstance->SetSampleFormat(0, - past->past_OutputSampleFormat); - err = roster->RegisterNode( pahsc->pahsc_InputNodeInstance ); - if( err != B_OK ) - { - DBUG(("Unable to register node.\n")); - goto error; - } - - roster->GetNodeFor( pahsc->pahsc_InputNodeInstance->Node().node, - &pahsc->pahsc_InputNode ); - if( err != B_OK ) - { - DBUG(("Unable to get input node.\n")); - goto error; - } - - /* Now we have three components (nodes) sitting next to each other. The - * next step is to look at them and find their inputs and outputs so we can - * wire them together. */ - err = roster->GetFreeInputsFor( pahsc->pahsc_OutputNode, - &pahsc->pahsc_MixerInput, 1, &num, B_MEDIA_RAW_AUDIO ); - if( err != B_OK || num < 1 ) - { - DBUG(("Couldn't get the mixer input.\n")); - goto error; - } - - err = roster->GetFreeOutputsFor( pahsc->pahsc_InputNode, - &pahsc->pahsc_PaOutput, 1, &num, B_MEDIA_RAW_AUDIO ); - if( err != B_OK || num < 1 ) - { - DBUG(("Couldn't get PortAudio output.\n")); - goto error; - } - - - /* We've found the input and output -- the final step is to run a wire - * between them so they are connected. */ - - /* try to make the mixer input adapt to what PA sends it */ - pahsc->pahsc_Format = pahsc->pahsc_PaOutput.format; - roster->Connect( pahsc->pahsc_PaOutput.source, - pahsc->pahsc_MixerInput.destination, &pahsc->pahsc_Format, - &pahsc->pahsc_PaOutput, &pahsc->pahsc_MixerInput ); - - - /* Actually, there's one final step -- tell them all to sync to the - * sound card's DAC */ - roster->SetTimeSourceFor( pahsc->pahsc_InputNode.node, - pahsc->pahsc_TimeSource.node ); - roster->SetTimeSourceFor( pahsc->pahsc_OutputNode.node, - pahsc->pahsc_TimeSource.node ); - - } - - return paNoError; - -error: - PaHost_CloseStream( past ); - return paHostError; -} - -/*************************************************************************/ -PaError PaHost_CloseStream( internalPortAudioStream *past ) -{ - PaHostSoundControl *pahsc = (PaHostSoundControl *)past->past_DeviceData; - status_t err; - BMediaRoster *roster = BMediaRoster::Roster(&err); - - if( !roster ) - { - DBUG(("Couldn't get media roster\n")); - return paHostError; - } - - if( !pahsc ) - return paHostError; - - /* Disconnect all the connections we made when opening the stream */ - - roster->Disconnect(pahsc->pahsc_InputNode.node, pahsc->pahsc_PaOutput.source, - pahsc->pahsc_OutputNode.node, pahsc->pahsc_MixerInput.destination); - - DBUG(("Calling ReleaseNode()")); - roster->ReleaseNode(pahsc->pahsc_InputNode); - - /* deleting the node shouldn't be necessary -- it is reference counted, and will - * delete itself when its references drop to zero. the call to ReleaseNode() - * above should decrease its reference count */ - pahsc->pahsc_InputNodeInstance = NULL; - - return paNoError; -} - -/*************************************************************************/ -PaTimestamp Pa_StreamTime( PortAudioStream *stream ) -{ - internalPortAudioStream *past = (internalPortAudioStream *) stream; - PaHostSoundControl *pahsc = (PaHostSoundControl *)past->past_DeviceData; - - return pahsc->pahsc_InputNodeInstance->GetStreamTime(); -} - -/*************************************************************************/ -void Pa_Sleep( long msec ) -{ - /* snooze() takes microseconds */ - snooze( msec * 1000 ); -} - -/************************************************************************* - * Allocate memory that can be accessed in real-time. - * This may need to be held in physical memory so that it is not - * paged to virtual memory. - * This call MUST be balanced with a call to PaHost_FreeFastMemory(). - * Memory will be set to zero. - */ -void *PaHost_AllocateFastMemory( long numBytes ) -{ - /* BeOS supports non-pagable memory through pools -- a pool is an area - * of physical memory that is locked. It would be best to pre-allocate - * that pool and then hand out memory from it, but we don't know in - * advance how much we'll need. So for now, we'll allocate a pool - * for every request we get, storing a pointer to the pool at the - * beginning of the allocated memory */ - rtm_pool *pool; - void *addr; - long size = numBytes + sizeof(rtm_pool *); - static int counter = 0; - char pool_name[100]; - - /* Every pool needs a unique name. */ - sprintf(pool_name, "PaPoolNumber%d", counter++); - - if( rtm_create_pool( &pool, size, pool_name ) != B_OK ) - return 0; - - addr = rtm_alloc( pool, size ); - if( addr == NULL ) - return 0; - - memset( addr, 0, numBytes ); - *((rtm_pool **)addr) = pool; // store the pointer to the pool - addr = (rtm_pool **)addr + 1; // and return the next location in memory - - return addr; -} - -/************************************************************************* - * Free memory that could be accessed in real-time. - * This call MUST be balanced with a call to PaHost_AllocateFastMemory(). - */ -void PaHost_FreeFastMemory( void *addr, long numBytes ) -{ - rtm_pool *pool; - - if( addr == NULL ) - return; - - addr = (rtm_pool **)addr - 1; - pool = *((rtm_pool **)addr); - - rtm_free( addr ); - rtm_delete_pool( pool ); -} diff --git a/pd/portaudio_v18/pa_common/pa_convert.c b/pd/portaudio_v18/pa_common/pa_convert.c deleted file mode 100644 index 72e021eb..00000000 --- a/pd/portaudio_v18/pa_common/pa_convert.c +++ /dev/null @@ -1,470 +0,0 @@ -/* - * pa_conversions.c - * portaudio - * - * Created by Phil Burk on Mon Mar 18 2002. - * - */ -#include <stdio.h> - -#include "portaudio.h" -#include "pa_host.h" - -#define CLIP( val, min, max ) { val = ((val) < (min)) ? min : (((val) < (max)) ? (max) : (val)); } - -/*************************************************************************/ -static void PaConvert_Float32_Int16( - float *sourceBuffer, int sourceStride, - short *targetBuffer, int targetStride, - int numSamples ) -{ - int i; - for( i=0; i<numSamples; i++ ) - { - short samp = (short) (*sourceBuffer * (32767.0f)); - *targetBuffer = samp; - sourceBuffer += sourceStride; - targetBuffer += targetStride; - } -} - -/*************************************************************************/ -static void PaConvert_Float32_Int16_Clip( - float *sourceBuffer, int sourceStride, - short *targetBuffer, int targetStride, - int numSamples ) -{ - int i; - for( i=0; i<numSamples; i++ ) - { - long samp = (long) (*sourceBuffer * (32767.0f)); - CLIP( samp, -0x8000, 0x7FFF ); - *targetBuffer = (short) samp; - sourceBuffer += sourceStride; - targetBuffer += targetStride; - } -} - -/*************************************************************************/ -static void PaConvert_Float32_Int16_ClipDither( - float *sourceBuffer, int sourceStride, - short *targetBuffer, int targetStride, - int numSamples ) -{ - int i; - for( i=0; i<numSamples; i++ ) - { - // use smaller scaler to prevent overflow when we add the dither - float dither = PaConvert_TriangularDither() * PA_DITHER_SCALE; - float dithered = (*sourceBuffer * (32766.0f)) + dither; - long samp = (long) dithered; - CLIP( samp, -0x8000, 0x7FFF ); - *targetBuffer = (short) samp; - sourceBuffer += sourceStride; - targetBuffer += targetStride; - } -} - -/*************************************************************************/ -static void PaConvert_Float32_Int16_Dither( - float *sourceBuffer, int sourceStride, - short *targetBuffer, int targetStride, - int numSamples ) -{ - int i; - for( i=0; i<numSamples; i++ ) - { - // use smaller scaler to prevent overflow when we add the dither - float dither = PaConvert_TriangularDither() * PA_DITHER_SCALE; - float dithered = (*sourceBuffer * (32766.0f)) + dither; - *targetBuffer = (short) dithered; - sourceBuffer += sourceStride; - targetBuffer += targetStride; - } -} - - -/*************************************************************************/ -static void PaConvert_Int16_Float32( - short *sourceBuffer, int sourceStride, - float *targetBuffer, int targetStride, - int numSamples ) -{ - int i; - for( i=0; i<numSamples; i++ ) - { - float samp = *sourceBuffer * (1.0f / 32768.0f); - *targetBuffer = samp; - sourceBuffer += sourceStride; - targetBuffer += targetStride; - } -} - -/*************************************************************************/ -static void PaConvert_Float32_Int8( - float *sourceBuffer, int sourceStride, - char *targetBuffer, int targetStride, - int numSamples ) -{ - int i; - for( i=0; i<numSamples; i++ ) - { - char samp = (char) (*sourceBuffer * (127.0)); - *targetBuffer = samp; - sourceBuffer += sourceStride; - targetBuffer += targetStride; - } -} - - -/*************************************************************************/ -static void PaConvert_Float32_Int8_Clip( - float *sourceBuffer, int sourceStride, - char *targetBuffer, int targetStride, - int numSamples ) -{ - int i; - for( i=0; i<numSamples; i++ ) - { - long samp = (long) (*sourceBuffer * 127.0f); - CLIP( samp, -0x80, 0x7F ); - *targetBuffer = (char) samp; - sourceBuffer += sourceStride; - targetBuffer += targetStride; - } -} - -/*************************************************************************/ -static void PaConvert_Float32_Int8_ClipDither( - float *sourceBuffer, int sourceStride, - char *targetBuffer, int targetStride, - int numSamples ) -{ - int i; - for( i=0; i<numSamples; i++ ) - { - // use smaller scaler to prevent overflow when we add the dither - float dither = PaConvert_TriangularDither() * PA_DITHER_SCALE; - float dithered = (*sourceBuffer * (126.0f)) + dither; - long samp = (long) dithered; - CLIP( samp, -0x80, 0x7F ); - *targetBuffer = (char) samp; - sourceBuffer += sourceStride; - targetBuffer += targetStride; - } -} - -/*************************************************************************/ -static void PaConvert_Float32_Int8_Dither( - float *sourceBuffer, int sourceStride, - char *targetBuffer, int targetStride, - int numSamples ) -{ - int i; - for( i=0; i<numSamples; i++ ) - { - // use smaller scaler to prevent overflow when we add the dither - float dither = PaConvert_TriangularDither() * PA_DITHER_SCALE; //FIXME - float dithered = (*sourceBuffer * (126.0f)) + dither; - long samp = (long) dithered; - *targetBuffer = (char) samp; - sourceBuffer += sourceStride; - targetBuffer += targetStride; - } -} - -/*************************************************************************/ -static void PaConvert_Int8_Float32( - char *sourceBuffer, int sourceStride, - float *targetBuffer, int targetStride, - int numSamples ) -{ - int i; - for( i=0; i<numSamples; i++ ) - { - float samp = *sourceBuffer * (1.0f / 128.0f); - *targetBuffer = samp; - sourceBuffer += sourceStride; - targetBuffer += targetStride; - } -} - -/*************************************************************************/ -static void PaConvert_Float32_UInt8( - float *sourceBuffer, int sourceStride, - unsigned char *targetBuffer, int targetStride, - int numSamples ) -{ - int i; - for( i=0; i<numSamples; i++ ) - { - unsigned char samp = (unsigned char)(128 + (*sourceBuffer * (127.0))); - *targetBuffer = samp; - sourceBuffer += sourceStride; - targetBuffer += targetStride; - } -} - -/*************************************************************************/ -static void PaConvert_UInt8_Float32( - unsigned char *sourceBuffer, int sourceStride, - float *targetBuffer, int targetStride, - int numSamples ) -{ - int i; - for( i=0; i<numSamples; i++ ) - { - float samp = (*sourceBuffer - 128) * (1.0f / 128.0f); - *targetBuffer = samp; - sourceBuffer += sourceStride; - targetBuffer += targetStride; - } -} - -/*************************************************************************/ -static void PaConvert_Float32_Int32( - float *sourceBuffer, int sourceStride, - long *targetBuffer, int targetStride, - int numSamples ) -{ - int i; - for( i=0; i<numSamples; i++ ) - { - int samp = (int) (*sourceBuffer * 0x7FFFFFFF); - *targetBuffer = samp; - sourceBuffer += sourceStride; - targetBuffer += targetStride; - } -} - -/*************************************************************************/ -static void PaConvert_Float32_Int32_Clip( - float *sourceBuffer, int sourceStride, - long *targetBuffer, int targetStride, - int numSamples ) -{ - int i; - for( i=0; i<numSamples; i++ ) - { - int samp; - float fs = *sourceBuffer; - CLIP( fs, -1.0f, 0.999999f ); - samp = (int) (*sourceBuffer * 0x7FFFFFFF); - *targetBuffer = samp; - sourceBuffer += sourceStride; - targetBuffer += targetStride; - } -} - -/*************************************************************************/ -static void PaConvert_Int32_Float32( - long *sourceBuffer, int sourceStride, - float *targetBuffer, int targetStride, - int numSamples ) -{ - int i; - for( i=0; i<numSamples; i++ ) - { - float samp = *sourceBuffer * (1.0f / 0x7FFFFFFF); - *targetBuffer = samp; - sourceBuffer += sourceStride; - targetBuffer += targetStride; - } -} - -/*************************************************************************/ -static PortAudioConverter *PaConvert_SelectProc( PaSampleFormat sourceFormat, - PaSampleFormat targetFormat, int ifClip, int ifDither ) -{ - PortAudioConverter *proc = NULL; - switch( sourceFormat ) - { - case paUInt8: - switch( targetFormat ) - { - case paFloat32: - proc = (PortAudioConverter *) PaConvert_UInt8_Float32; - break; - default: - break; - } - break; - case paInt8: - switch( targetFormat ) - { - case paFloat32: - proc = (PortAudioConverter *) PaConvert_Int8_Float32; - break; - default: - break; - } - break; - case paInt16: - switch( targetFormat ) - { - case paFloat32: - proc = (PortAudioConverter *) PaConvert_Int16_Float32; - break; - default: - break; - } - break; - - case paInt32: - switch( targetFormat ) - { - case paFloat32: - proc = (PortAudioConverter *) PaConvert_Int32_Float32; - break; - default: - break; - } - break; - - case paFloat32: - switch( targetFormat ) - { - case paUInt8: - proc = (PortAudioConverter *) PaConvert_Float32_UInt8; - break; - case paInt8: - if( ifClip && ifDither ) proc = (PortAudioConverter *) PaConvert_Float32_Int8_ClipDither; - else if( ifClip ) proc = (PortAudioConverter *) PaConvert_Float32_Int8_Clip; - else if( ifDither ) proc = (PortAudioConverter *) PaConvert_Float32_Int8_Dither; - else proc = (PortAudioConverter *) PaConvert_Float32_Int8; - break; - case paInt16: - if( ifClip && ifDither ) proc = (PortAudioConverter *) PaConvert_Float32_Int16_ClipDither; - else if( ifClip ) proc = (PortAudioConverter *) PaConvert_Float32_Int16_Clip; - else if( ifDither ) proc = (PortAudioConverter *) PaConvert_Float32_Int16_Dither; - else proc = (PortAudioConverter *) PaConvert_Float32_Int16; - break; - case paInt32: - /* Don't bother dithering a 32 bit integer! */ - if( ifClip ) proc = (PortAudioConverter *) PaConvert_Float32_Int32_Clip; - else proc = (PortAudioConverter *) PaConvert_Float32_Int32; - break; - default: - break; - } - break; - default: - break; - } - return proc; - -} - -/*************************************************************************/ -PaError PaConvert_SetupInput( internalPortAudioStream *past, - PaSampleFormat nativeInputSampleFormat ) -{ - past->past_NativeInputSampleFormat = nativeInputSampleFormat; - past->past_InputConversionSourceStride = 1; - past->past_InputConversionTargetStride = 1; - - if( nativeInputSampleFormat != past->past_InputSampleFormat ) - { - int ifDither = (past->past_Flags & paDitherOff) == 0; - past->past_InputConversionProc = PaConvert_SelectProc( nativeInputSampleFormat, - past->past_InputSampleFormat, 0, ifDither ); - if( past->past_InputConversionProc == NULL ) return paSampleFormatNotSupported; - } - else - { - past->past_InputConversionProc = NULL; /* no conversion necessary */ - } - - return paNoError; -} - -/*************************************************************************/ -PaError PaConvert_SetupOutput( internalPortAudioStream *past, - PaSampleFormat nativeOutputSampleFormat ) -{ - - past->past_NativeOutputSampleFormat = nativeOutputSampleFormat; - past->past_OutputConversionSourceStride = 1; - past->past_OutputConversionTargetStride = 1; - - if( nativeOutputSampleFormat != past->past_OutputSampleFormat ) - { - int ifDither = (past->past_Flags & paDitherOff) == 0; - int ifClip = (past->past_Flags & paClipOff) == 0; - - past->past_OutputConversionProc = PaConvert_SelectProc( past->past_OutputSampleFormat, - nativeOutputSampleFormat, ifClip, ifDither ); - if( past->past_OutputConversionProc == NULL ) return paSampleFormatNotSupported; - } - else - { - past->past_OutputConversionProc = NULL; /* no conversion necessary */ - } - - return paNoError; -} - -/************************************************************************* -** Called by host code. -** Convert input from native format to user format, -** call user code, -** then convert output to native format. -** Returns result from user callback. -*/ -long PaConvert_Process( internalPortAudioStream *past, - void *nativeInputBuffer, - void *nativeOutputBuffer ) -{ - int userResult; - void *inputBuffer = NULL; - void *outputBuffer = NULL; - - /* Get native input data. */ - if( (past->past_NumInputChannels > 0) && (nativeInputBuffer != NULL) ) - { - if( past->past_InputSampleFormat == past->past_NativeInputSampleFormat ) - { - /* Already in native format so just read directly from native buffer. */ - inputBuffer = nativeInputBuffer; - } - else - { - inputBuffer = past->past_InputBuffer; - /* Convert input data to user format. */ - (*past->past_InputConversionProc)(nativeInputBuffer, past->past_InputConversionSourceStride, - inputBuffer, past->past_InputConversionTargetStride, - past->past_FramesPerUserBuffer * past->past_NumInputChannels ); - } - } - - /* Are we doing output? */ - if( (past->past_NumOutputChannels > 0) && (nativeOutputBuffer != NULL) ) - { - outputBuffer = (past->past_OutputConversionProc == NULL) ? - nativeOutputBuffer : past->past_OutputBuffer; - } - /* - AddTraceMessage("Pa_CallConvertInt16: inputBuffer = ", (int) inputBuffer ); - AddTraceMessage("Pa_CallConvertInt16: outputBuffer = ", (int) outputBuffer ); - */ - /* Call user callback routine. */ - userResult = past->past_Callback( - inputBuffer, - outputBuffer, - past->past_FramesPerUserBuffer, - past->past_FrameCount, - past->past_UserData ); - - /* Advance frame counter for timestamp. */ - past->past_FrameCount += past->past_FramesPerUserBuffer; // FIXME - should this be in here? - - /* Convert to native format if necessary. */ - if( (past->past_OutputConversionProc != NULL ) && (outputBuffer != NULL) ) - { - (*past->past_OutputConversionProc)( outputBuffer, past->past_OutputConversionSourceStride, - nativeOutputBuffer, past->past_OutputConversionTargetStride, - past->past_FramesPerUserBuffer * past->past_NumOutputChannels ); - } - - return userResult; -} diff --git a/pd/portaudio_v18/pa_common/pa_host.h b/pd/portaudio_v18/pa_common/pa_host.h deleted file mode 100644 index db898fe0..00000000 --- a/pd/portaudio_v18/pa_common/pa_host.h +++ /dev/null @@ -1,189 +0,0 @@ -#ifndef PA_HOST_H -#define PA_HOST_H - -/* - * $Id: pa_host.h,v 1.3.4.1 2003/02/11 21:33:58 philburk Exp $ - * Host dependant internal API for PortAudio - * - * Author: Phil Burk <philburk@softsynth.com> - * - * PortAudio Portable Real-Time Audio Library - * Latest Version at: http://www.softsynth.com/portaudio/ - * DirectSound and Macintosh Implementation - * Copyright (c) 1999-2000 Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ - -#include "portaudio.h" - -#ifdef __cplusplus -extern "C" -{ -#endif /* __cplusplus */ - -#ifndef SUPPORT_AUDIO_CAPTURE -#define SUPPORT_AUDIO_CAPTURE (1) -#endif - -#ifndef int32 - typedef long int32; -#endif -#ifndef uint32 - typedef unsigned long uint32; -#endif -#ifndef int16 - typedef short int16; -#endif -#ifndef uint16 - typedef unsigned short uint16; -#endif - -/* Used to convert between various sample formats. */ -typedef void (PortAudioConverter)( - void *inputBuffer, int inputStride, - void *outputBuffer, int outputStride, - int numSamples ); - -#define PA_MAGIC (0x18273645) - -/************************************************************************************/ -/****************** Structures ******************************************************/ -/************************************************************************************/ - -typedef struct internalPortAudioStream -{ - uint32 past_Magic; /* ID for struct to catch bugs. */ - - /* Begin user specified information. */ - uint32 past_FramesPerUserBuffer; - uint32 past_NumUserBuffers; - double past_SampleRate; /* Closest supported sample rate. */ - int past_NumInputChannels; - int past_NumOutputChannels; - PaDeviceID past_InputDeviceID; - PaDeviceID past_OutputDeviceID; - PaSampleFormat past_InputSampleFormat; - PaSampleFormat past_OutputSampleFormat; - PortAudioCallback *past_Callback; - void *past_UserData; - uint32 past_Flags; - /* End user specified information. */ - - void *past_DeviceData; - PaSampleFormat past_NativeOutputSampleFormat; - PaSampleFormat past_NativeInputSampleFormat; - - /* Flags for communicating between foreground and background. */ - volatile int past_IsActive; /* Background is still playing. */ - volatile int past_StopSoon; /* Background should keep playing when buffers empty. */ - volatile int past_StopNow; /* Background should stop playing now. */ - /* These buffers are used when the native format does not match the user format. */ - void *past_InputBuffer; - uint32 past_InputBufferSize; /* Size in bytes of the input buffer. */ - void *past_OutputBuffer; - uint32 past_OutputBufferSize; - /* Measurements */ - uint32 past_NumCallbacks; - PaTimestamp past_FrameCount; /* Frames output to buffer. */ - /* For measuring CPU utilization. */ - double past_AverageInsideCount; - double past_AverageTotalCount; - double past_Usage; - int past_IfLastExitValid; - /* Format Conversion */ - /* These are setup by PaConversion_Setup() */ - PortAudioConverter *past_InputConversionProc; - int past_InputConversionSourceStride; - int past_InputConversionTargetStride; - PortAudioConverter *past_OutputConversionProc; - int past_OutputConversionSourceStride; - int past_OutputConversionTargetStride; -} -internalPortAudioStream; - -/************************************************************************************/ -/******** These functions must be provided by a platform implementation. ************/ -/************************************************************************************/ - -PaError PaHost_Init( void ); -PaError PaHost_Term( void ); - -PaError PaHost_OpenStream( internalPortAudioStream *past ); -PaError PaHost_CloseStream( internalPortAudioStream *past ); - -PaError PaHost_StartOutput( internalPortAudioStream *past ); -PaError PaHost_StopOutput( internalPortAudioStream *past, int abort ); -PaError PaHost_StartInput( internalPortAudioStream *past ); -PaError PaHost_StopInput( internalPortAudioStream *past, int abort ); -PaError PaHost_StartEngine( internalPortAudioStream *past ); -PaError PaHost_StopEngine( internalPortAudioStream *past, int abort ); -PaError PaHost_StreamActive( internalPortAudioStream *past ); - -void *PaHost_AllocateFastMemory( long numBytes ); -void PaHost_FreeFastMemory( void *addr, long numBytes ); - -/* This only called if PA_VALIDATE_RATE IS CALLED. */ -PaError PaHost_ValidateSampleRate( PaDeviceID id, double requestedFrameRate, - double *closestFrameRatePtr ); - -/**********************************************************************/ -/************ Common Utility Routines provided by PA ******************/ -/**********************************************************************/ - -/* PaHost_IsInitialized() returns non-zero if PA is initialized, 0 otherwise */ -int PaHost_IsInitialized( void ); - -internalPortAudioStream* PaHost_GetStreamRepresentation( PortAudioStream *stream ); - -int PaHost_FindClosestTableEntry( double allowableError, const double *rateTable, - int numRates, double frameRate ); - -long Pa_CallConvertInt16( internalPortAudioStream *past, - short *nativeInputBuffer, - short *nativeOutputBuffer ); - -/* Calculate 2 LSB dither signal with a triangular distribution. -** Ranged properly for adding to a 32 bit 1.31 fixed point value prior to >>15. -** Range of output is +/- 65535 -** Multiply by PA_DITHER_SCALE to get a float between -2.0 and 2.0. */ -#define PA_DITHER_BITS (15) -#define PA_DITHER_SCALE (1.0f / ((1<<PA_DITHER_BITS)-1)) -long PaConvert_TriangularDither( void ); - -PaError PaConvert_SetupInput( internalPortAudioStream *past, - PaSampleFormat nativeInputSampleFormat ); - -PaError PaConvert_SetupOutput( internalPortAudioStream *past, - PaSampleFormat nativeOutputSampleFormat ); - -long PaConvert_Process( internalPortAudioStream *past, - void *nativeInputBuffer, - void *nativeOutputBuffer ); - -#ifdef __cplusplus -} -#endif /* __cplusplus */ -#endif /* PA_HOST_H */ diff --git a/pd/portaudio_v18/pa_common/pa_lib.c b/pd/portaudio_v18/pa_common/pa_lib.c deleted file mode 100644 index bf97de22..00000000 --- a/pd/portaudio_v18/pa_common/pa_lib.c +++ /dev/null @@ -1,806 +0,0 @@ -/* - * $Id: pa_lib.c,v 1.3.4.2 2003/03/15 02:50:14 pieter Exp $ - * Portable Audio I/O Library - * Host Independant Layer - * - * Based on the Open Source API proposed by Ross Bencina - * Copyright (c) 1999-2000 Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ - -/* Modification History: - PLB20010422 - apply Mike Berry's changes for CodeWarrior on PC - PLB20010820 - fix dither and shift for recording PaUInt8 format -*/ - -#include <stdio.h> -#include <stdlib.h> -#include <string.h> -#include <math.h> - -/* PLB20010422 - "memory.h" doesn't work on CodeWarrior for PC. Thanks Mike Berry for the mod. */ -#ifdef _WIN32 -#ifndef __MWERKS__ -#include <memory.h> -#endif /* __MWERKS__ */ -#else /* !_WIN32 */ -#include <memory.h> -#endif /* _WIN32 */ - -#include "portaudio.h" -#include "pa_host.h" -#include "pa_trace.h" - -/* The reason we might NOT want to validate the rate before opening the stream - * is because many DirectSound drivers lie about the rates they actually support. - */ -#define PA_VALIDATE_RATE (0) /* If true validate sample rate against driver info. */ - -/* -O- maybe not allocate past_InputBuffer and past_OutputBuffer if not needed for conversion -*/ - -#ifndef FALSE - #define FALSE (0) - #define TRUE (!FALSE) -#endif - -#define PRINT(x) { printf x; fflush(stdout); } -#define ERR_RPT(x) PRINT(x) -#define DBUG(x) /* PRINT(x) */ -#define DBUGX(x) /* PRINT(x) */ - -static int gInitCount = 0; /* Count number of times Pa_Initialize() called to allow nesting and overlapping. */ - -static PaError Pa_KillStream( PortAudioStream *stream, int abort ); - -/***********************************************************************/ -int PaHost_FindClosestTableEntry( double allowableError, const double *rateTable, int numRates, double frameRate ) -{ - double err, minErr = allowableError; - int i, bestFit = -1; - - for( i=0; i<numRates; i++ ) - { - err = fabs( frameRate - rateTable[i] ); - if( err < minErr ) - { - minErr = err; - bestFit = i; - } - } - return bestFit; -} - -/************************************************************************** -** Make sure sample rate is legal and also convert to enumeration for driver. -*/ -PaError PaHost_ValidateSampleRate( PaDeviceID id, double requestedFrameRate, - double *closestFrameRatePtr ) -{ - long bestRateIndex; - const PaDeviceInfo *pdi; - pdi = Pa_GetDeviceInfo( id ); - if( pdi == NULL ) - { - return paInvalidDeviceId; - } - - if( pdi->numSampleRates == -1 ) - { - /* Is it out of range? */ - if( (requestedFrameRate < pdi->sampleRates[0]) || - (requestedFrameRate > pdi->sampleRates[1]) ) - { - return paInvalidSampleRate; - } - - *closestFrameRatePtr = requestedFrameRate; - } - else - { - bestRateIndex = PaHost_FindClosestTableEntry( 1.0, pdi->sampleRates, pdi->numSampleRates, requestedFrameRate ); - if( bestRateIndex < 0 ) return paInvalidSampleRate; - *closestFrameRatePtr = pdi->sampleRates[bestRateIndex]; - } - return paNoError; -} - -/*************************************************************************/ -PaError Pa_OpenStream( - PortAudioStream** streamPtrPtr, - PaDeviceID inputDeviceID, - int numInputChannels, - PaSampleFormat inputSampleFormat, - void *inputDriverInfo, - PaDeviceID outputDeviceID, - int numOutputChannels, - PaSampleFormat outputSampleFormat, - void *outputDriverInfo, - double sampleRate, - unsigned long framesPerBuffer, - unsigned long numberOfBuffers, - unsigned long streamFlags, - PortAudioCallback *callback, - void *userData ) -{ - internalPortAudioStream *past = NULL; - PaError result = paNoError; - int bitsPerInputSample; - int bitsPerOutputSample; - /* Print passed parameters. */ - DBUG(("Pa_OpenStream( %p, %d, %d, %d, %p, /* input */ \n", - streamPtrPtr, inputDeviceID, numInputChannels, - inputSampleFormat, inputDriverInfo )); - DBUG((" %d, %d, %d, %p, /* output */\n", - outputDeviceID, numOutputChannels, - outputSampleFormat, outputDriverInfo )); - DBUG((" %g, %d, %d, 0x%x, , %p )\n", - sampleRate, framesPerBuffer, numberOfBuffers, - streamFlags, userData )); - - /* Check for parameter errors. */ - if( (streamFlags & ~(paClipOff | paDitherOff)) != 0 ) return paInvalidFlag; - if( streamPtrPtr == NULL ) return paBadStreamPtr; - if( inputDriverInfo != NULL ) return paHostError; /* REVIEW */ - if( outputDriverInfo != NULL ) return paHostError; /* REVIEW */ - if( (inputDeviceID < 0) && ( outputDeviceID < 0) ) return paInvalidDeviceId; - if( (outputDeviceID >= Pa_CountDevices()) || (inputDeviceID >= Pa_CountDevices()) ) - { - return paInvalidDeviceId; - } - if( (numInputChannels <= 0) && ( numOutputChannels <= 0) ) return paInvalidChannelCount; - -#if SUPPORT_AUDIO_CAPTURE - if( inputDeviceID >= 0 ) - { - PaError size = Pa_GetSampleSize( inputSampleFormat ); - if( size < 0 ) return size; - bitsPerInputSample = 8 * size; - if( (numInputChannels <= 0) ) return paInvalidChannelCount; - } -#else - if( inputDeviceID >= 0 ) - { - return paInvalidChannelCount; - } -#endif /* SUPPORT_AUDIO_CAPTURE */ - else - { - if( numInputChannels > 0 ) return paInvalidChannelCount; - bitsPerInputSample = 0; - } - - if( outputDeviceID >= 0 ) - { - PaError size = Pa_GetSampleSize( outputSampleFormat ); - if( size < 0 ) return size; - bitsPerOutputSample = 8 * size; - if( (numOutputChannels <= 0) ) return paInvalidChannelCount; - } - else - { - if( numOutputChannels > 0 ) return paInvalidChannelCount; - bitsPerOutputSample = 0; - } - - if( callback == NULL ) return paNullCallback; - - /* Allocate and clear stream structure. */ - past = (internalPortAudioStream *) PaHost_AllocateFastMemory( sizeof(internalPortAudioStream) ); - if( past == NULL ) return paInsufficientMemory; - memset( past, 0, sizeof(internalPortAudioStream) ); - AddTraceMessage("Pa_OpenStream: past", (long) past ); - - past->past_Magic = PA_MAGIC; /* Set ID to catch bugs. */ - past->past_FramesPerUserBuffer = framesPerBuffer; - past->past_NumUserBuffers = numberOfBuffers; /* NOTE - PaHost_OpenStream() MUST CHECK FOR ZERO! */ - past->past_Callback = callback; - past->past_UserData = userData; - past->past_OutputSampleFormat = outputSampleFormat; - past->past_InputSampleFormat = inputSampleFormat; - past->past_OutputDeviceID = outputDeviceID; - past->past_InputDeviceID = inputDeviceID; - past->past_NumInputChannels = numInputChannels; - past->past_NumOutputChannels = numOutputChannels; - past->past_Flags = streamFlags; - - /* Check for absurd sample rates. */ - if( (sampleRate < 1000.0) || (sampleRate > 200000.0) ) - { - result = paInvalidSampleRate; - goto cleanup; - } - - /* Allocate buffers that may be used for format conversion from user to native buffers. */ - if( numInputChannels > 0 ) - { - -#if PA_VALIDATE_RATE - result = PaHost_ValidateSampleRate( inputDeviceID, sampleRate, &past->past_SampleRate ); - if( result < 0 ) - { - goto cleanup; - } -#else - past->past_SampleRate = sampleRate; -#endif - /* Allocate single Input buffer for passing formatted samples to user callback. */ - past->past_InputBufferSize = framesPerBuffer * numInputChannels * ((bitsPerInputSample+7) / 8); - past->past_InputBuffer = PaHost_AllocateFastMemory(past->past_InputBufferSize); - if( past->past_InputBuffer == NULL ) - { - result = paInsufficientMemory; - goto cleanup; - } - } - else - { - past->past_InputBuffer = NULL; - } - - /* Allocate single Output buffer. */ - if( numOutputChannels > 0 ) - { -#if PA_VALIDATE_RATE - result = PaHost_ValidateSampleRate( outputDeviceID, sampleRate, &past->past_SampleRate ); - if( result < 0 ) - { - goto cleanup; - } -#else - past->past_SampleRate = sampleRate; -#endif - past->past_OutputBufferSize = framesPerBuffer * numOutputChannels * ((bitsPerOutputSample+7) / 8); - past->past_OutputBuffer = PaHost_AllocateFastMemory(past->past_OutputBufferSize); - if( past->past_OutputBuffer == NULL ) - { - result = paInsufficientMemory; - goto cleanup; - } - } - else - { - past->past_OutputBuffer = NULL; - } - - result = PaHost_OpenStream( past ); - if( result < 0 ) goto cleanup; - - *streamPtrPtr = (void *) past; - - return result; - -cleanup: - if( past != NULL ) Pa_CloseStream( past ); - *streamPtrPtr = NULL; - return result; -} - - -/*************************************************************************/ -PaError Pa_OpenDefaultStream( PortAudioStream** stream, - int numInputChannels, - int numOutputChannels, - PaSampleFormat sampleFormat, - double sampleRate, - unsigned long framesPerBuffer, - unsigned long numberOfBuffers, - PortAudioCallback *callback, - void *userData ) -{ - return Pa_OpenStream( - stream, - ((numInputChannels > 0) ? Pa_GetDefaultInputDeviceID() : paNoDevice), - numInputChannels, sampleFormat, NULL, - ((numOutputChannels > 0) ? Pa_GetDefaultOutputDeviceID() : paNoDevice), - numOutputChannels, sampleFormat, NULL, - sampleRate, framesPerBuffer, numberOfBuffers, paNoFlag, callback, userData ); -} - -/*************************************************************************/ -PaError Pa_CloseStream( PortAudioStream* stream) -{ - PaError result; - internalPortAudioStream *past; - - DBUG(("Pa_CloseStream()\n")); - if( stream == NULL ) return paBadStreamPtr; - past = (internalPortAudioStream *) stream; - - Pa_AbortStream( past ); - result = PaHost_CloseStream( past ); - - if( past->past_InputBuffer ) PaHost_FreeFastMemory( past->past_InputBuffer, past->past_InputBufferSize ); - if( past->past_OutputBuffer ) PaHost_FreeFastMemory( past->past_OutputBuffer, past->past_OutputBufferSize ); - PaHost_FreeFastMemory( past, sizeof(internalPortAudioStream) ); - - return result; -} - -/*************************************************************************/ -PaError Pa_StartStream( PortAudioStream *stream ) -{ - PaError result = paHostError; - internalPortAudioStream *past; - - if( stream == NULL ) return paBadStreamPtr; - past = (internalPortAudioStream *) stream; - - past->past_FrameCount = 0.0; - - if( past->past_NumInputChannels > 0 ) - { - result = PaHost_StartInput( past ); - DBUG(("Pa_StartStream: PaHost_StartInput returned = 0x%X.\n", result)); - if( result < 0 ) goto error; - } - - if( past->past_NumOutputChannels > 0 ) - { - result = PaHost_StartOutput( past ); - DBUG(("Pa_StartStream: PaHost_StartOutput returned = 0x%X.\n", result)); - if( result < 0 ) goto error; - } - - result = PaHost_StartEngine( past ); - DBUG(("Pa_StartStream: PaHost_StartEngine returned = 0x%X.\n", result)); - if( result < 0 ) goto error; - - return paNoError; - -error: - return result; -} - -/*************************************************************************/ -PaError Pa_StopStream( PortAudioStream *stream ) -{ - return Pa_KillStream( stream, 0 ); -} - -/*************************************************************************/ -PaError Pa_AbortStream( PortAudioStream *stream ) -{ - return Pa_KillStream( stream, 1 ); -} - -/*************************************************************************/ -static PaError Pa_KillStream( PortAudioStream *stream, int abort ) -{ - PaError result = paNoError; - internalPortAudioStream *past; - - DBUG(("Pa_StopStream().\n")); - if( stream == NULL ) return paBadStreamPtr; - past = (internalPortAudioStream *) stream; - - if( (past->past_NumInputChannels > 0) || (past->past_NumOutputChannels > 0) ) - { - result = PaHost_StopEngine( past, abort ); - DBUG(("Pa_StopStream: PaHost_StopEngine returned = 0x%X.\n", result)); - if( result < 0 ) goto error; - } - - if( past->past_NumInputChannels > 0 ) - { - result = PaHost_StopInput( past, abort ); - DBUG(("Pa_StopStream: PaHost_StopInput returned = 0x%X.\n", result)); - if( result != paNoError ) goto error; - } - - if( past->past_NumOutputChannels > 0 ) - { - result = PaHost_StopOutput( past, abort ); - DBUG(("Pa_StopStream: PaHost_StopOutput returned = 0x%X.\n", result)); - if( result != paNoError ) goto error; - } - -error: - past->past_Usage = 0; - past->past_IfLastExitValid = 0; - - return result; -} - -/*************************************************************************/ -PaError Pa_StreamActive( PortAudioStream *stream ) -{ - internalPortAudioStream *past; - if( stream == NULL ) return paBadStreamPtr; - past = (internalPortAudioStream *) stream; - return PaHost_StreamActive( past ); -} - -/*************************************************************************/ -const char *Pa_GetErrorText( PaError errnum ) -{ - const char *msg; - - switch(errnum) - { - case paNoError: msg = "Success"; break; - case paHostError: msg = "Host error."; break; - case paInvalidChannelCount: msg = "Invalid number of channels."; break; - case paInvalidSampleRate: msg = "Invalid sample rate."; break; - case paInvalidDeviceId: msg = "Invalid device ID."; break; - case paInvalidFlag: msg = "Invalid flag."; break; - case paSampleFormatNotSupported: msg = "Sample format not supported"; break; - case paBadIODeviceCombination: msg = "Illegal combination of I/O devices."; break; - case paInsufficientMemory: msg = "Insufficient memory."; break; - case paBufferTooBig: msg = "Buffer too big."; break; - case paBufferTooSmall: msg = "Buffer too small."; break; - case paNullCallback: msg = "No callback routine specified."; break; - case paBadStreamPtr: msg = "Invalid stream pointer."; break; - case paTimedOut : msg = "Wait Timed Out."; break; - case paInternalError: msg = "Internal PortAudio Error."; break; - case paDeviceUnavailable: msg = "Device Unavailable."; break; - default: msg = "Illegal error number."; break; - } - return msg; -} - -/* - Get CPU Load as a fraction of total CPU time. - A value of 0.5 would imply that PortAudio and the sound generating - callback was consuming roughly 50% of the available CPU time. - The amount may vary depending on CPU load. - This function may be called from the callback function. -*/ -double Pa_GetCPULoad( PortAudioStream* stream) -{ - internalPortAudioStream *past; - if( stream == NULL ) return (double) paBadStreamPtr; - past = (internalPortAudioStream *) stream; - return past->past_Usage; -} - -/*************************************************************************/ -internalPortAudioStream* PaHost_GetStreamRepresentation( PortAudioStream *stream ) -{ - internalPortAudioStream* result = (internalPortAudioStream*) stream; - - if( result == NULL || result->past_Magic != PA_MAGIC ) - return NULL; - else - return result; -} - -/************************************************************* -** Calculate 2 LSB dither signal with a triangular distribution. -** Ranged properly for adding to a 32 bit integer prior to >>15. -** Range of output is +/- 32767 -*/ -#define PA_DITHER_BITS (15) -#define PA_DITHER_SCALE (1.0f / ((1<<PA_DITHER_BITS)-1)) -long PaConvert_TriangularDither( void ) -{ - static unsigned long previous = 0; - static unsigned long randSeed1 = 22222; - static unsigned long randSeed2 = 5555555; - long current, highPass; - /* Generate two random numbers. */ - randSeed1 = (randSeed1 * 196314165) + 907633515; - randSeed2 = (randSeed2 * 196314165) + 907633515; - /* Generate triangular distribution about 0. - * Shift before adding to prevent overflow which would skew the distribution. - * Also shift an extra bit for the high pass filter. - */ -#define DITHER_SHIFT ((32 - PA_DITHER_BITS) + 1) - current = (((long)randSeed1)>>DITHER_SHIFT) + (((long)randSeed2)>>DITHER_SHIFT); - /* High pass filter to reduce audibility. */ - highPass = current - previous; - previous = current; - return highPass; -} - -/************************************************************************* -** Called by host code. -** Convert input from Int16, call user code, then convert output -** to Int16 format for native use. -** Assumes host native format is paInt16. -** Returns result from user callback. -*/ -long Pa_CallConvertInt16( internalPortAudioStream *past, - short *nativeInputBuffer, - short *nativeOutputBuffer ) -{ - long temp; - int userResult; - unsigned int i; - void *inputBuffer = NULL; - void *outputBuffer = NULL; - -#if SUPPORT_AUDIO_CAPTURE - /* Get native data from DirectSound. */ - if( (past->past_NumInputChannels > 0) && (nativeInputBuffer != NULL) ) - { - /* Convert from native format to PA format. */ - unsigned int samplesPerBuffer = past->past_FramesPerUserBuffer * past->past_NumInputChannels; - switch(past->past_InputSampleFormat) - { - - case paFloat32: - { - float *inBufPtr = (float *) past->past_InputBuffer; - inputBuffer = past->past_InputBuffer; - for( i=0; i<samplesPerBuffer; i++ ) - { - inBufPtr[i] = nativeInputBuffer[i] * (1.0f / 32767.0f); - } - break; - } - - case paInt32: - { - /* Convert 16 bit data to 32 bit integers */ - int *inBufPtr = (int *) past->past_InputBuffer; - inputBuffer = past->past_InputBuffer; - for( i=0; i<samplesPerBuffer; i++ ) - { - inBufPtr[i] = nativeInputBuffer[i] << 16; - } - break; - } - - case paInt16: - { - /* Already in correct format so don't copy. */ - inputBuffer = nativeInputBuffer; - break; - } - - case paInt8: - { - /* Convert 16 bit data to 8 bit chars */ - char *inBufPtr = (char *) past->past_InputBuffer; - inputBuffer = past->past_InputBuffer; - if( past->past_Flags & paDitherOff ) - { - for( i=0; i<samplesPerBuffer; i++ ) - { - inBufPtr[i] = (char)(nativeInputBuffer[i] >> 8); - } - } - else - { - for( i=0; i<samplesPerBuffer; i++ ) - { - temp = nativeInputBuffer[i]; - temp += PaConvert_TriangularDither() >> 8; /* PLB20010820 */ - temp = ((temp < -0x8000) ? -0x8000 : ((temp > 0x7FFF) ? 0x7FFF : temp)); - inBufPtr[i] = (char)(temp >> 8); - } - } - break; - } - - case paUInt8: - { - /* Convert 16 bit data to 8 bit unsigned chars */ - unsigned char *inBufPtr = (unsigned char *) past->past_InputBuffer; - inputBuffer = past->past_InputBuffer; - if( past->past_Flags & paDitherOff ) - { - for( i=0; i<samplesPerBuffer; i++ ) - { - inBufPtr[i] = (unsigned char)((nativeInputBuffer[i] >> 8) + 0x80); - } - } - else - { - /* If you dither then you have to clip because dithering could push the signal out of range! */ - for( i=0; i<samplesPerBuffer; i++ ) - { - temp = nativeInputBuffer[i]; - temp += PaConvert_TriangularDither() >> 8; /* PLB20010820 */ - temp = ((temp < -0x8000) ? -0x8000 : ((temp > 0x7FFF) ? 0x7FFF : temp)); - inBufPtr[i] = (unsigned char)((temp>>8) + 0x80); /* PLB20010820 */ - } - } - break; - } - - default: - break; - } - } -#endif /* SUPPORT_AUDIO_CAPTURE */ - - /* Are we doing output time? */ - if( (past->past_NumOutputChannels > 0) && (nativeOutputBuffer != NULL) ) - { - /* May already be in native format so just write directly to native buffer. */ - outputBuffer = (past->past_OutputSampleFormat == paInt16) ? - (void*)nativeOutputBuffer : past->past_OutputBuffer; - } - /* - AddTraceMessage("Pa_CallConvertInt16: inputBuffer = ", (int) inputBuffer ); - AddTraceMessage("Pa_CallConvertInt16: outputBuffer = ", (int) outputBuffer ); - */ - /* Call user callback routine. */ - userResult = past->past_Callback( - inputBuffer, - outputBuffer, - past->past_FramesPerUserBuffer, - past->past_FrameCount, - past->past_UserData ); - - past->past_FrameCount += (PaTimestamp) past->past_FramesPerUserBuffer; - - /* Convert to native format if necessary. */ - if( outputBuffer != NULL ) - { - unsigned int samplesPerBuffer = past->past_FramesPerUserBuffer * past->past_NumOutputChannels; - switch(past->past_OutputSampleFormat) - { - case paFloat32: - { - float *outBufPtr = (float *) past->past_OutputBuffer; - if( past->past_Flags & paDitherOff ) - { - if( past->past_Flags & paClipOff ) /* NOTHING */ - { - for( i=0; i<samplesPerBuffer; i++ ) - { - *nativeOutputBuffer++ = (short) (outBufPtr[i] * (32767.0f)); - } - } - else /* CLIP */ - { - for( i=0; i<samplesPerBuffer; i++ ) - { - temp = (long)(outBufPtr[i] * 32767.0f); - *nativeOutputBuffer++ = (short)((temp < -0x8000) ? -0x8000 : ((temp > 0x7FFF) ? 0x7FFF : temp)); - } - } - } - else - { - /* If you dither then you have to clip because dithering could push the signal out of range! */ - for( i=0; i<samplesPerBuffer; i++ ) - { - float dither = PaConvert_TriangularDither()*PA_DITHER_SCALE; - float dithered = (outBufPtr[i] * (32767.0f)) + dither; - temp = (long) (dithered); - *nativeOutputBuffer++ = (short)((temp < -0x8000) ? -0x8000 : ((temp > 0x7FFF) ? 0x7FFF : temp)); - } - } - break; - } - - case paInt32: - { - int *outBufPtr = (int *) past->past_OutputBuffer; - if( past->past_Flags & paDitherOff ) - { - for( i=0; i<samplesPerBuffer; i++ ) - { - *nativeOutputBuffer++ = (short) (outBufPtr[i] >> 16 ); - } - } - else - { - for( i=0; i<samplesPerBuffer; i++ ) - { - /* Shift one bit down before dithering so that we have room for overflow from add. */ - temp = (outBufPtr[i] >> 1) + PaConvert_TriangularDither(); - temp = temp >> 15; - *nativeOutputBuffer++ = (short)((temp < -0x8000) ? -0x8000 : ((temp > 0x7FFF) ? 0x7FFF : temp)); - } - } - break; - } - - case paInt8: - { - char *outBufPtr = (char *) past->past_OutputBuffer; - for( i=0; i<samplesPerBuffer; i++ ) - { - *nativeOutputBuffer++ = (short) (((int)outBufPtr[i]) << 8); - } - break; - } - - case paUInt8: - { - unsigned char *outBufPtr = (unsigned char *) past->past_OutputBuffer; - for( i=0; i<samplesPerBuffer; i++ ) - { - *nativeOutputBuffer++ = (short) (((int)(outBufPtr[i] - 0x80)) << 8); - } - break; - } - - default: - break; - } - - } - - return userResult; -} - -/*************************************************************************/ -PaError Pa_Initialize( void ) -{ - if( gInitCount++ > 0 ) return paNoError; - ResetTraceMessages(); - return PaHost_Init(); -} - -PaError Pa_Terminate( void ) -{ - PaError result = paNoError; - - if( gInitCount == 0 ) return paNoError; - else if( --gInitCount == 0 ) - { - result = PaHost_Term(); - DumpTraceMessages(); - } - return result; -} - -int PaHost_IsInitialized() -{ - return gInitCount; -} - -/*************************************************************************/ -PaError Pa_GetSampleSize( PaSampleFormat format ) -{ - int size; - switch(format ) - { - - case paUInt8: - case paInt8: - size = 1; - break; - - case paInt16: - size = 2; - break; - - case paPackedInt24: - size = 3; - break; - - case paFloat32: - case paInt32: - case paInt24: - size = 4; - break; - - default: - size = paSampleFormatNotSupported; - break; - } - return (PaError) size; -} - - diff --git a/pd/portaudio_v18/pa_common/pa_trace.c b/pd/portaudio_v18/pa_common/pa_trace.c deleted file mode 100644 index d55a6d37..00000000 --- a/pd/portaudio_v18/pa_common/pa_trace.c +++ /dev/null @@ -1,83 +0,0 @@ -/* - * $Id: pa_trace.c,v 1.1.1.1 2002/01/22 00:52:11 phil Exp $ - * Portable Audio I/O Library Trace Facility - * Store trace information in real-time for later printing. - * - * Based on the Open Source API proposed by Ross Bencina - * Copyright (c) 1999-2000 Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - */ - -#include <stdio.h> -#include <stdlib.h> -#include <string.h> -#include "pa_trace.h" - -#if TRACE_REALTIME_EVENTS - -static char *traceTextArray[MAX_TRACE_RECORDS]; -static int traceIntArray[MAX_TRACE_RECORDS]; -static int traceIndex = 0; -static int traceBlock = 0; - -/*********************************************************************/ -void ResetTraceMessages() -{ - traceIndex = 0; -} - -/*********************************************************************/ -void DumpTraceMessages() -{ - int i; - int numDump = (traceIndex < MAX_TRACE_RECORDS) ? traceIndex : MAX_TRACE_RECORDS; - - printf("DumpTraceMessages: traceIndex = %d\n", traceIndex ); - for( i=0; i<numDump; i++ ) - { - printf("%3d: %s = 0x%08X\n", - i, traceTextArray[i], traceIntArray[i] ); - } - ResetTraceMessages(); - fflush(stdout); -} - -/*********************************************************************/ -void AddTraceMessage( char *msg, int data ) -{ - if( (traceIndex == MAX_TRACE_RECORDS) && (traceBlock == 0) ) - { - traceBlock = 1; - /* DumpTraceMessages(); */ - } - else if( traceIndex < MAX_TRACE_RECORDS ) - { - traceTextArray[traceIndex] = msg; - traceIntArray[traceIndex] = data; - traceIndex++; - } -} - -#endif diff --git a/pd/portaudio_v18/pa_common/pa_trace.h b/pd/portaudio_v18/pa_common/pa_trace.h deleted file mode 100644 index d0fc904c..00000000 --- a/pd/portaudio_v18/pa_common/pa_trace.h +++ /dev/null @@ -1,67 +0,0 @@ -#ifndef PA_TRACE_H -#define PA_TRACE_H -/* - * $Id: pa_trace.h,v 1.1.1.1 2002/01/22 00:52:11 phil Exp $ - * Portable Audio I/O Library Trace Facility - * Store trace information in real-time for later printing. - * - * Based on the Open Source API proposed by Ross Bencina - * Copyright (c) 1999-2000 Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - */ - - -#define TRACE_REALTIME_EVENTS (0) /* Keep log of various real-time events. */ -#define MAX_TRACE_RECORDS (2048) - -#ifdef __cplusplus -extern "C" -{ -#endif /* __cplusplus */ - - - /************************************************************************************/ - /****************** Prototypes ******************************************************/ - /************************************************************************************/ - -#if TRACE_REALTIME_EVENTS - - void DumpTraceMessages(); - void ResetTraceMessages(); - void AddTraceMessage( char *msg, int data ); - -#else - -#define AddTraceMessage(msg,data) /* noop */ -#define ResetTraceMessages() /* noop */ -#define DumpTraceMessages() /* noop */ - -#endif - -#ifdef __cplusplus -} -#endif /* __cplusplus */ - -#endif /* PA_TRACE_H */ diff --git a/pd/portaudio_v18/pa_common/portaudio.h b/pd/portaudio_v18/pa_common/portaudio.h deleted file mode 100644 index 06f7079b..00000000 --- a/pd/portaudio_v18/pa_common/portaudio.h +++ /dev/null @@ -1,463 +0,0 @@ -#ifndef PORT_AUDIO_H -#define PORT_AUDIO_H - -#ifdef __cplusplus -extern "C" -{ -#endif /* __cplusplus */ - -/* - * $Id: portaudio.h,v 1.5 2002/03/26 18:04:22 philburk Exp $ - * PortAudio Portable Real-Time Audio Library - * PortAudio API Header File - * Latest version available at: http://www.audiomulch.com/portaudio/ - * - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ - -typedef int PaError; -typedef enum { - paNoError = 0, - - paHostError = -10000, - paInvalidChannelCount, - paInvalidSampleRate, - paInvalidDeviceId, - paInvalidFlag, - paSampleFormatNotSupported, - paBadIODeviceCombination, - paInsufficientMemory, - paBufferTooBig, - paBufferTooSmall, - paNullCallback, - paBadStreamPtr, - paTimedOut, - paInternalError, - paDeviceUnavailable -} PaErrorNum; - -/* - Pa_Initialize() is the library initialisation function - call this before - using the library. - -*/ - -PaError Pa_Initialize( void ); - -/* - Pa_Terminate() is the library termination function - call this after - using the library. - -*/ - -PaError Pa_Terminate( void ); - -/* - Pa_GetHostError() returns a host specific error code. - This can be called after receiving a PortAudio error code of paHostError. - -*/ - -long Pa_GetHostError( void ); - -/* - Pa_GetErrorText() translates the supplied PortAudio error number - into a human readable message. - -*/ - -const char *Pa_GetErrorText( PaError errnum ); - -/* - Sample formats - - These are formats used to pass sound data between the callback and the - stream. Each device has a "native" format which may be used when optimum - efficiency or control over conversion is required. - - Formats marked "always available" are supported (emulated) by all - PortAudio implementations. - - The floating point representation (paFloat32) uses +1.0 and -1.0 as the - maximum and minimum respectively. - - paUInt8 is an unsigned 8 bit format where 128 is considered "ground" - -*/ - -typedef unsigned long PaSampleFormat; -#define paFloat32 ((PaSampleFormat) (1<<0)) /*always available*/ -#define paInt16 ((PaSampleFormat) (1<<1)) /*always available*/ -#define paInt32 ((PaSampleFormat) (1<<2)) /*always available*/ -#define paInt24 ((PaSampleFormat) (1<<3)) -#define paPackedInt24 ((PaSampleFormat) (1<<4)) -#define paInt8 ((PaSampleFormat) (1<<5)) -#define paUInt8 ((PaSampleFormat) (1<<6)) -#define paCustomFormat ((PaSampleFormat) (1<<16)) - -/* - Device enumeration mechanism. - - Device ids range from 0 to Pa_CountDevices()-1. - - Devices may support input, output or both. - -*/ - -typedef int PaDeviceID; -#define paNoDevice -1 - -int Pa_CountDevices( void ); - -typedef struct -{ - int structVersion; - const char *name; - int maxInputChannels; - int maxOutputChannels; - /* Number of discrete rates, or -1 if range supported. */ - int numSampleRates; - /* Array of supported sample rates, or {min,max} if range supported. */ - const double *sampleRates; - PaSampleFormat nativeSampleFormats; -} -PaDeviceInfo; - -/* - Pa_GetDefaultInputDeviceID(), Pa_GetDefaultOutputDeviceID() return the - default device ids for input and output respectively, or paNoDevice if - no device is available. - The result can be passed to Pa_OpenStream(). - - On the PC, the user can specify a default device by - setting an environment variable. For example, to use device #1. - - set PA_RECOMMENDED_OUTPUT_DEVICE=1 - - The user should first determine the available device ids by using - the supplied application "pa_devs". - -*/ - -PaDeviceID Pa_GetDefaultInputDeviceID( void ); -PaDeviceID Pa_GetDefaultOutputDeviceID( void ); - - - -/* - Pa_GetDeviceInfo() returns a pointer to an immutable PaDeviceInfo structure - for the device specified. - If the device parameter is out of range the function returns NULL. - - PortAudio manages the memory referenced by the returned pointer, the client - must not manipulate or free the memory. The pointer is only guaranteed to be - valid between calls to Pa_Initialize() and Pa_Terminate(). - -*/ - -const PaDeviceInfo* Pa_GetDeviceInfo( PaDeviceID device ); - -/* - PaTimestamp is used to represent a continuous sample clock with arbitrary - start time that can be used for syncronization. The type is used for the - outTime argument to the PortAudioCallback and as the result of Pa_StreamTime() - -*/ - -typedef double PaTimestamp; - -/* - PortAudioCallback is implemented by PortAudio clients. - - inputBuffer and outputBuffer are arrays of interleaved samples, - the format, packing and number of channels used by the buffers are - determined by parameters to Pa_OpenStream() (see below). - - framesPerBuffer is the number of sample frames to be processed by the callback. - - outTime is the time in samples when the buffer(s) processed by - this callback will begin being played at the audio output. - See also Pa_StreamTime() - - userData is the value of a user supplied pointer passed to Pa_OpenStream() - intended for storing synthesis data etc. - - return value: - The callback can return a non-zero value to stop the stream. This may be - useful in applications such as soundfile players where a specific duration - of output is required. However, it is not necessary to utilise this mechanism - as StopStream() will also terminate the stream. A callback returning a - non-zero value must fill the entire outputBuffer. - - NOTE: None of the other stream functions may be called from within the - callback function except for Pa_GetCPULoad(). - -*/ - -typedef int (PortAudioCallback)( - void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ); - - -/* - Stream flags - - These flags may be supplied (ored together) in the streamFlags argument to - the Pa_OpenStream() function. - -*/ - -#define paNoFlag (0) -#define paClipOff (1<<0) /* disable default clipping of out of range samples */ -#define paDitherOff (1<<1) /* disable default dithering */ -#define paPlatformSpecificFlags (0x00010000) -typedef unsigned long PaStreamFlags; - -/* - A single PortAudioStream provides multiple channels of real-time - input and output audio streaming to a client application. - Pointers to PortAudioStream objects are passed between PortAudio functions. -*/ - -typedef void PortAudioStream; -#define PaStream PortAudioStream - -/* - Pa_OpenStream() opens a stream for either input, output or both. - - stream is the address of a PortAudioStream pointer which will receive - a pointer to the newly opened stream. - - inputDevice is the id of the device used for input (see PaDeviceID above.) - inputDevice may be paNoDevice to indicate that an input device is not required. - - numInputChannels is the number of channels of sound to be delivered to the - callback. It can range from 1 to the value of maxInputChannels in the - PaDeviceInfo record for the device specified by the inputDevice parameter. - If inputDevice is paNoDevice numInputChannels is ignored. - - inputSampleFormat is the sample format of inputBuffer provided to the callback - function. inputSampleFormat may be any of the formats described by the - PaSampleFormat enumeration (see above). PortAudio guarantees support for - the device's native formats (nativeSampleFormats in the device info record) - and additionally 16 and 32 bit integer and 32 bit floating point formats. - Support for other formats is implementation defined. - - inputDriverInfo is a pointer to an optional driver specific data structure - containing additional information for device setup or stream processing. - inputDriverInfo is never required for correct operation. If not used - inputDriverInfo should be NULL. - - outputDevice is the id of the device used for output (see PaDeviceID above.) - outputDevice may be paNoDevice to indicate that an output device is not required. - - numOutputChannels is the number of channels of sound to be supplied by the - callback. See the definition of numInputChannels above for more details. - - outputSampleFormat is the sample format of the outputBuffer filled by the - callback function. See the definition of inputSampleFormat above for more - details. - - outputDriverInfo is a pointer to an optional driver specific data structure - containing additional information for device setup or stream processing. - outputDriverInfo is never required for correct operation. If not used - outputDriverInfo should be NULL. - - sampleRate is the desired sampleRate. For full-duplex streams it is the - sample rate for both input and output - - framesPerBuffer is the length in sample frames of all internal sample buffers - used for communication with platform specific audio routines. Wherever - possible this corresponds to the framesPerBuffer parameter passed to the - callback function. - - numberOfBuffers is the number of buffers used for multibuffered communication - with the platform specific audio routines. If you pass zero, then an optimum - value will be chosen for you internally. This parameter is provided only - as a guide - and does not imply that an implementation must use multibuffered - i/o when reliable double buffering is available (such as SndPlayDoubleBuffer() - on the Macintosh.) - - streamFlags may contain a combination of flags ORed together. - These flags modify the behaviour of the streaming process. Some flags may only - be relevant to certain buffer formats. - - callback is a pointer to a client supplied function that is responsible - for processing and filling input and output buffers (see above for details.) - - userData is a client supplied pointer which is passed to the callback - function. It could for example, contain a pointer to instance data necessary - for processing the audio buffers. - - return value: - Upon success Pa_OpenStream() returns PaNoError and places a pointer to a - valid PortAudioStream in the stream argument. The stream is inactive (stopped). - If a call to Pa_OpenStream() fails a non-zero error code is returned (see - PaError above) and the value of stream is invalid. - -*/ - -PaError Pa_OpenStream( PortAudioStream** stream, - PaDeviceID inputDevice, - int numInputChannels, - PaSampleFormat inputSampleFormat, - void *inputDriverInfo, - PaDeviceID outputDevice, - int numOutputChannels, - PaSampleFormat outputSampleFormat, - void *outputDriverInfo, - double sampleRate, - unsigned long framesPerBuffer, - unsigned long numberOfBuffers, - PaStreamFlags streamFlags, - PortAudioCallback *callback, - void *userData ); - - -/* - Pa_OpenDefaultStream() is a simplified version of Pa_OpenStream() that opens - the default input and/or output devices. Most parameters have identical meaning - to their Pa_OpenStream() counterparts, with the following exceptions: - - If either numInputChannels or numOutputChannels is 0 the respective device - is not opened. This has the same effect as passing paNoDevice in the device - arguments to Pa_OpenStream(). - - sampleFormat applies to both the input and output buffers. - -*/ - -PaError Pa_OpenDefaultStream( PortAudioStream** stream, - int numInputChannels, - int numOutputChannels, - PaSampleFormat sampleFormat, - double sampleRate, - unsigned long framesPerBuffer, - unsigned long numberOfBuffers, - PortAudioCallback *callback, - void *userData ); - -/* - Pa_CloseStream() closes an audio stream, flushing any pending buffers. - -*/ - -PaError Pa_CloseStream( PortAudioStream* ); - -/* - Pa_StartStream() and Pa_StopStream() begin and terminate audio processing. - Pa_StopStream() waits until all pending audio buffers have been played. - Pa_AbortStream() stops playing immediately without waiting for pending - buffers to complete. - -*/ - -PaError Pa_StartStream( PortAudioStream *stream ); - -PaError Pa_StopStream( PortAudioStream *stream ); - -PaError Pa_AbortStream( PortAudioStream *stream ); - -/* - Pa_StreamActive() returns one (1) when the stream is active (ie playing - or recording audio), zero (0) when not playing, or a negative error number - if the stream is invalid. - The stream is active between calls to Pa_StartStream() and Pa_StopStream(), - but may also become inactive if the callback returns a non-zero value. - In the latter case, the stream is considered inactive after the last - buffer has finished playing. - -*/ - -PaError Pa_StreamActive( PortAudioStream *stream ); - -/* - Pa_StreamTime() returns the current output time in samples for the stream. - This time may be used as a time reference (for example synchronizing audio to - MIDI). - -*/ - -PaTimestamp Pa_StreamTime( PortAudioStream *stream ); - -/* - Pa_GetCPULoad() returns the CPU Load for the stream. - The "CPU Load" is a fraction of total CPU time consumed by the stream's - audio processing routines including, but not limited to the client supplied - callback. - A value of 0.5 would imply that PortAudio and the sound generating - callback was consuming roughly 50% of the available CPU time. - This function may be called from the callback function or the application. - -*/ - -double Pa_GetCPULoad( PortAudioStream* stream ); - -/* - Pa_GetMinNumBuffers() returns the minimum number of buffers required by - the current host based on minimum latency. - On the PC, for the DirectSound implementation, latency can be optionally set - by user by setting an environment variable. - For example, to set latency to 200 msec, put: - - set PA_MIN_LATENCY_MSEC=200 - - in the AUTOEXEC.BAT file and reboot. - If the environment variable is not set, then the latency will be determined - based on the OS. Windows NT has higher latency than Win95. - -*/ - -int Pa_GetMinNumBuffers( int framesPerBuffer, double sampleRate ); - -/* - Pa_Sleep() puts the caller to sleep for at least 'msec' milliseconds. - You may sleep longer than the requested time so don't rely on this for - accurate musical timing. - - Pa_Sleep() is provided as a convenience for authors of portable code (such as - the tests and examples in the PortAudio distribution.) - -*/ - -void Pa_Sleep( long msec ); - -/* - Pa_GetSampleSize() returns the size in bytes of a single sample in the - supplied PaSampleFormat, or paSampleFormatNotSupported if the format is - no supported. - -*/ - -PaError Pa_GetSampleSize( PaSampleFormat format ); - - -#ifdef __cplusplus -} -#endif /* __cplusplus */ -#endif /* PORT_AUDIO_H */ diff --git a/pd/portaudio_v18/pa_dll_switch/PaDllEntry.h b/pd/portaudio_v18/pa_dll_switch/PaDllEntry.h deleted file mode 100644 index e070054b..00000000 --- a/pd/portaudio_v18/pa_dll_switch/PaDllEntry.h +++ /dev/null @@ -1,184 +0,0 @@ - -/* - * PortAudio Portable Real-Time Audio Library - * PortAudio DLL Header File - * Latest version available at: http://www.audiomulch.com/portaudio/ - * - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ - -// changed by zplane.developement in order to generate a DLL - -#ifndef __PADLLENTRY_HEADER_INCLUDED__ - -#define __PADLLENTRY_HEADER_INCLUDED__ - -typedef int PaError; -typedef enum { - paNoError = 0, - - paHostError = -10000, - paInvalidChannelCount, - paInvalidSampleRate, - paInvalidDeviceId, - paInvalidFlag, - paSampleFormatNotSupported, - paBadIODeviceCombination, - paInsufficientMemory, - paBufferTooBig, - paBufferTooSmall, - paNullCallback, - paBadStreamPtr, - paTimedOut, - paInternalError -} PaErrorNum; - -typedef unsigned long PaSampleFormat; -#define paFloat32 ((PaSampleFormat) (1<<0)) /*always available*/ -#define paInt16 ((PaSampleFormat) (1<<1)) /*always available*/ -#define paInt32 ((PaSampleFormat) (1<<2)) /*always available*/ -#define paInt24 ((PaSampleFormat) (1<<3)) -#define paPackedInt24 ((PaSampleFormat) (1<<4)) -#define paInt8 ((PaSampleFormat) (1<<5)) -#define paUInt8 ((PaSampleFormat) (1<<6)) /* unsigned 8 bit, 128 is "ground" */ -#define paCustomFormat ((PaSampleFormat) (1<<16)) - - -typedef int PaDeviceID; -#define paNoDevice -1 - -typedef struct -{ - int structVersion; - const char *name; - int maxInputChannels; - int maxOutputChannels; - /* Number of discrete rates, or -1 if range supported. */ - int numSampleRates; - /* Array of supported sample rates, or {min,max} if range supported. */ - const double *sampleRates; - PaSampleFormat nativeSampleFormats; -} -PaDeviceInfo; - - -typedef double PaTimestamp; - - -typedef int (PortAudioCallback)( - void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ); - - -#define paNoFlag (0) -#define paClipOff (1<<0) /* disable default clipping of out of range samples */ -#define paDitherOff (1<<1) /* disable default dithering */ -#define paPlatformSpecificFlags (0x00010000) -typedef unsigned long PaStreamFlags; - -typedef void PortAudioStream; -#define PaStream PortAudioStream - -extern PaError (__cdecl* Pa_Initialize)( void ); - - - -extern PaError (__cdecl* Pa_Terminate)( void ); - - -extern long (__cdecl* Pa_GetHostError)( void ); - - -extern const char* (__cdecl* Pa_GetErrorText)( PaError ); - - - -extern int (__cdecl* Pa_CountDevices)(void); - -extern PaDeviceID (__cdecl* Pa_GetDefaultInputDeviceID)( void ); - -extern PaDeviceID (__cdecl* Pa_GetDefaultOutputDeviceID)( void ); - - -extern const PaDeviceInfo* (__cdecl* Pa_GetDeviceInfo)( PaDeviceID); - - - -extern PaError (__cdecl* Pa_OpenStream)( - PortAudioStream ** , - PaDeviceID , - int , - PaSampleFormat , - void *, - PaDeviceID , - int , - PaSampleFormat , - void *, - double , - unsigned long , - unsigned long , - unsigned long , - PortAudioCallback *, - void * ); - - - -extern PaError (__cdecl* Pa_OpenDefaultStream)( PortAudioStream** stream, - int numInputChannels, - int numOutputChannels, - PaSampleFormat sampleFormat, - double sampleRate, - unsigned long framesPerBuffer, - unsigned long numberOfBuffers, - PortAudioCallback *callback, - void *userData ); - - -extern PaError (__cdecl* Pa_CloseStream)( PortAudioStream* ); - - -extern PaError (__cdecl* Pa_StartStream)( PortAudioStream *stream ); - -extern PaError (__cdecl* Pa_StopStream)( PortAudioStream *stream ); - -extern PaError (__cdecl* Pa_AbortStream)( PortAudioStream *stream ); - -extern PaError (__cdecl* Pa_StreamActive)( PortAudioStream *stream ); - -extern PaTimestamp (__cdecl* Pa_StreamTime)( PortAudioStream *stream ); - -extern double (__cdecl* Pa_GetCPULoad)( PortAudioStream* stream ); - -extern int (__cdecl* Pa_GetMinNumBuffers)( int framesPerBuffer, double sampleRate ); - -extern void (__cdecl* Pa_Sleep)( long msec ); - -extern PaError (__cdecl* Pa_GetSampleSize)( PaSampleFormat format ); - -#endif // __PADLLENTRY_HEADER_INCLUDED__ - diff --git a/pd/portaudio_v18/pa_dll_switch/letter_from_tim_010817.txt b/pd/portaudio_v18/pa_dll_switch/letter_from_tim_010817.txt Binary files differdeleted file mode 100644 index a535cd1d..00000000 --- a/pd/portaudio_v18/pa_dll_switch/letter_from_tim_010817.txt +++ /dev/null diff --git a/pd/portaudio_v18/pa_dll_switch/loadPA_DLL.cpp b/pd/portaudio_v18/pa_dll_switch/loadPA_DLL.cpp deleted file mode 100644 index 043eda87..00000000 --- a/pd/portaudio_v18/pa_dll_switch/loadPA_DLL.cpp +++ /dev/null @@ -1,203 +0,0 @@ -////////////////////////////////////////////////////////////////////////// - - -HINSTANCE pPaDll; - -/* - the function pointers to the PortAudio DLLs -*/ - -PaError (__cdecl* Pa_Initialize)( void ); - - - -PaError (__cdecl* Pa_Terminate)( void ); - - -long (__cdecl* Pa_GetHostError)( void ); - - -const char* (__cdecl* Pa_GetErrorText)( PaError ); - - -int (__cdecl* Pa_CountDevices)(void); - -PaDeviceID (__cdecl* Pa_GetDefaultInputDeviceID)( void ); - -PaDeviceID (__cdecl* Pa_GetDefaultOutputDeviceID)( void ); - - -const PaDeviceInfo* (__cdecl* Pa_GetDeviceInfo)( PaDeviceID); - - - -PaError (__cdecl* Pa_OpenStream)( - PortAudioStream ** , - PaDeviceID , - int , - PaSampleFormat , - void *, - PaDeviceID , - int , - PaSampleFormat , - void *, - double , - unsigned long , - unsigned long , - unsigned long , - PortAudioCallback *, - void * ); - - - -PaError (__cdecl* Pa_OpenDefaultStream)( PortAudioStream** stream, - int numInputChannels, - int numOutputChannels, - PaSampleFormat sampleFormat, - double sampleRate, - unsigned long framesPerBuffer, - unsigned long numberOfBuffers, - PortAudioCallback *callback, - void *userData ); - - -PaError (__cdecl* Pa_CloseStream)( PortAudioStream* ); - - -PaError (__cdecl* Pa_StartStream)( PortAudioStream *stream ); - -PaError (__cdecl* Pa_StopStream)( PortAudioStream *stream ); - -PaError (__cdecl* Pa_AbortStream)( PortAudioStream *stream ); - -PaError (__cdecl* Pa_StreamActive)( PortAudioStream *stream ); - -PaTimestamp (__cdecl* Pa_StreamTime)( PortAudioStream *stream ); - -double (__cdecl* Pa_GetCPULoad)( PortAudioStream* stream ); - -int (__cdecl* Pa_GetMinNumBuffers)( int framesPerBuffer, double sampleRate ); - -void (__cdecl* Pa_Sleep)( long msec ); - -PaError (__cdecl* Pa_GetSampleSize)( PaSampleFormat format ); - - -////////////////////////////////////////////////////////////////////////// - -... - -ZERROR AudioEngine::DirectXSupport(ZBOOL bSupDX) -{ - if (bSupDX) - if (CheckForDirectXSupport()) - bSupportDirectX = _TRUE; - else - return _NO_SOUND; - else - bSupportDirectX = _FALSE; - return _NO_ERROR; -} - - - -ZBOOL AudioEngine::CheckForDirectXSupport() -{ - HMODULE pTestDXLib; - FARPROC pFunctionality; - - pTestDXLib=LoadLibrary("DSOUND"); - if (pTestDXLib!=NULL) // check if there is a DirectSound - { - pFunctionality = GetProcAddress(pTestDXLib, (char*) 7); - if (pFunctionality!=NULL) - { - FreeLibrary(pTestDXLib); - return _TRUE; - } - else - { - FreeLibrary(pTestDXLib); - return _FALSE; - } - } - else - return _FALSE; -} - - -ZERROR AudioEngine::LoadPALib() -{ -#ifdef _DEBUG - if (bSupportDirectX) - pPaDll = LoadLibrary("PA_DXD"); - else - pPaDll = LoadLibrary("PA_MMED"); -#else - if (bSupportDirectX) - pPaDll = LoadLibrary("PA_DX"); - else - pPaDll = LoadLibrary("PA_MME"); -#endif - if (pPaDll!=NULL) - { - - Pa_Initialize = (int (__cdecl*)(void))GetProcAddress(pPaDll,"Pa_Initialize"); - Pa_Terminate = (int (__cdecl*)(void))GetProcAddress(pPaDll,"Pa_Terminate"); - Pa_GetHostError = (long (__cdecl* )( void )) GetProcAddress(pPaDll,"Pa_GetHostError"); - Pa_GetErrorText = (const char* (__cdecl* )( PaError )) GetProcAddress(pPaDll,"Pa_GetErrorText"); - Pa_CountDevices = (int (__cdecl*)(void))GetProcAddress(pPaDll,"Pa_CountDevices"); - Pa_GetDefaultInputDeviceID = (int (__cdecl*)(void))GetProcAddress(pPaDll,"Pa_GetDefaultInputDeviceID"); - Pa_GetDefaultOutputDeviceID = (int (__cdecl*)(void))GetProcAddress(pPaDll,"Pa_GetDefaultOutputDeviceID"); - Pa_GetDeviceInfo = (const PaDeviceInfo* (__cdecl* )( PaDeviceID)) GetProcAddress(pPaDll,"Pa_GetDeviceInfo"); - Pa_OpenStream = ( PaError (__cdecl* )( - PortAudioStream ** , - PaDeviceID , - int , - PaSampleFormat , - void *, - PaDeviceID , - int , - PaSampleFormat , - void *, - double , - unsigned long , - unsigned long , - unsigned long , - PortAudioCallback *, - void * )) GetProcAddress(pPaDll,"Pa_OpenStream"); - - Pa_OpenDefaultStream = (PaError (__cdecl* )( PortAudioStream** , - int , - int , - PaSampleFormat , - double , - unsigned long , - unsigned long , - PortAudioCallback *, - void * )) GetProcAddress(pPaDll,"Pa_OpenDefaultStream"); - Pa_CloseStream = (PaError (__cdecl* )( PortAudioStream* )) GetProcAddress(pPaDll,"Pa_CloseStream"); - Pa_StartStream = (PaError (__cdecl* )( PortAudioStream* )) GetProcAddress(pPaDll,"Pa_StartStream"); - Pa_StopStream = (PaError (__cdecl* )( PortAudioStream* ))GetProcAddress(pPaDll,"Pa_StopStream"); - Pa_AbortStream = (PaError (__cdecl* )( PortAudioStream* )) GetProcAddress(pPaDll,"Pa_AbortStream"); - Pa_StreamActive = (PaError (__cdecl* )( PortAudioStream* )) GetProcAddress(pPaDll,"Pa_StreamActive"); - Pa_StreamTime = (PaTimestamp (__cdecl* )( PortAudioStream *))GetProcAddress(pPaDll,"Pa_StreamTime"); - Pa_GetCPULoad = (double (__cdecl* )( PortAudioStream* ))GetProcAddress(pPaDll,"Pa_GetCPULoad"); - Pa_GetMinNumBuffers = (int (__cdecl* )( int , double )) GetProcAddress(pPaDll,"Pa_GetMinNumBuffers"); - Pa_Sleep = (void (__cdecl* )( long )) GetProcAddress(pPaDll,"Pa_Sleep"); - Pa_GetSampleSize = (PaError (__cdecl* )( PaSampleFormat )) GetProcAddress(pPaDll,"Pa_GetSampleSize"); - - return _NO_ERROR; - } - else - return _DLL_NOT_FOUND; -} - -ZERROR AudioEngine::UnLoadPALib() -{ - if (pPaDll!=NULL) - FreeLibrary(pPaDll); - return _NO_ERROR; -} - -... diff --git a/pd/portaudio_v18/pa_dll_switch/pa_lib.c b/pd/portaudio_v18/pa_dll_switch/pa_lib.c deleted file mode 100644 index 86601592..00000000 --- a/pd/portaudio_v18/pa_dll_switch/pa_lib.c +++ /dev/null @@ -1,827 +0,0 @@ -/* - * Portable Audio I/O Library - * Host Independant Layer - * - * Based on the Open Source API proposed by Ross Bencina - * Copyright (c) 1999-2000 Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ - -/* Modification History: - PLB20010422 - apply Mike Berry's changes for CodeWarrior on PC -*/ - -#include <stdio.h> -#include <stdlib.h> -#include <string.h> -#include <math.h> - -/* PLB20010422 - "memory.h" doesn't work on CodeWarrior for PC. Thanks Mike Berry for the mod. */ -#ifdef _WIN32 -#ifndef __MWERKS__ -#include <memory.h> -#endif /* __MWERKS__ */ -#else /* !_WIN32 */ -#include <memory.h> -#endif /* _WIN32 */ - -#include "portaudio.h" -#include "pa_host.h" -#include "pa_trace.h" - -/* The reason we might NOT want to validate the rate before opening the stream - * is because many DirectSound drivers lie about the rates they actually support. - */ -#define PA_VALIDATE_RATE (0) /* If true validate sample rate against driver info. */ - -/* -O- maybe not allocate past_InputBuffer and past_OutputBuffer if not needed for conversion -*/ - -#ifndef FALSE - #define FALSE (0) - #define TRUE (!FALSE) -#endif - -#define PRINT(x) { printf x; fflush(stdout); } -#define ERR_RPT(x) PRINT(x) -#define DBUG(x) /* PRINT(x) */ -#define DBUGX(x) /* PRINT(x) */ - -static int gInitCount = 0; /* Count number of times Pa_Initialize() called to allow nesting and overlapping. */ - -static PaError Pa_KillStream( PortAudioStream *stream, int abort ); - -/***********************************************************************/ -int PaHost_FindClosestTableEntry( double allowableError, const double *rateTable, int numRates, double frameRate ) -{ - double err, minErr = allowableError; - int i, bestFit = -1; - - for( i=0; i<numRates; i++ ) - { - err = fabs( frameRate - rateTable[i] ); - if( err < minErr ) - { - minErr = err; - bestFit = i; - } - } - return bestFit; -} - -/************************************************************************** -** Make sure sample rate is legal and also convert to enumeration for driver. -*/ -PaError PaHost_ValidateSampleRate( PaDeviceID id, double requestedFrameRate, - double *closestFrameRatePtr ) -{ - long bestRateIndex; - const PaDeviceInfo *pdi; - pdi = Pa_GetDeviceInfo( id ); - if( pdi == NULL ) return paInvalidDeviceId; - - if( pdi->numSampleRates == -1 ) - { - /* Is it out of range? */ - if( (requestedFrameRate < pdi->sampleRates[0]) || - (requestedFrameRate > pdi->sampleRates[1]) ) - { - return paInvalidSampleRate; - } - - *closestFrameRatePtr = requestedFrameRate; - } - else - { - bestRateIndex = PaHost_FindClosestTableEntry( 1.0, pdi->sampleRates, pdi->numSampleRates, requestedFrameRate ); - if( bestRateIndex < 0 ) return paInvalidSampleRate; - *closestFrameRatePtr = pdi->sampleRates[bestRateIndex]; - } - return paNoError; -} - -/*************************************************************************/ -DLL_API PaError Pa_OpenStream( - PortAudioStream** streamPtrPtr, - PaDeviceID inputDeviceID, - int numInputChannels, - PaSampleFormat inputSampleFormat, - void *inputDriverInfo, - PaDeviceID outputDeviceID, - int numOutputChannels, - PaSampleFormat outputSampleFormat, - void *outputDriverInfo, - double sampleRate, - unsigned long framesPerBuffer, - unsigned long numberOfBuffers, - unsigned long streamFlags, - PortAudioCallback *callback, - void *userData ) -{ - internalPortAudioStream *past = NULL; - PaError result = paNoError; - int bitsPerInputSample; - int bitsPerOutputSample; - /* Print passed parameters. */ - DBUG(("Pa_OpenStream( %p, %d, %d, %d, %p, /* input */ \n", - streamPtrPtr, inputDeviceID, numInputChannels, - inputSampleFormat, inputDriverInfo )); - DBUG((" %d, %d, %d, %p, /* output */\n", - outputDeviceID, numOutputChannels, - outputSampleFormat, outputDriverInfo )); - DBUG((" %g, %d, %d, 0x%x, , %p )\n", - sampleRate, framesPerBuffer, numberOfBuffers, - streamFlags, userData )); - - /* Check for parameter errors. */ - if( (streamFlags & ~(paClipOff | paDitherOff)) != 0 ) return paInvalidFlag; - if( streamPtrPtr == NULL ) return paBadStreamPtr; - if( inputDriverInfo != NULL ) return paHostError; /* REVIEW */ - if( outputDriverInfo != NULL ) return paHostError; /* REVIEW */ - if( (inputDeviceID < 0) && ( outputDeviceID < 0) ) return paInvalidDeviceId; - if( (outputDeviceID >= Pa_CountDevices()) || (inputDeviceID >= Pa_CountDevices()) ) return paInvalidDeviceId; - if( (numInputChannels <= 0) && ( numOutputChannels <= 0) ) return paInvalidChannelCount; - -#if SUPPORT_AUDIO_CAPTURE - if( inputDeviceID >= 0 ) - { - PaError size = Pa_GetSampleSize( inputSampleFormat ); - if( size < 0 ) return size; - bitsPerInputSample = 8 * size; - if( (numInputChannels <= 0) ) return paInvalidChannelCount; - } -#else - if( inputDeviceID >= 0 ) - { - return paInvalidChannelCount; - } -#endif /* SUPPORT_AUDIO_CAPTURE */ - else - { - if( numInputChannels > 0 ) return paInvalidChannelCount; - bitsPerInputSample = 0; - } - - if( outputDeviceID >= 0 ) - { - PaError size = Pa_GetSampleSize( outputSampleFormat ); - if( size < 0 ) return size; - bitsPerOutputSample = 8 * size; - if( (numOutputChannels <= 0) ) return paInvalidChannelCount; - } - else - { - if( numOutputChannels > 0 ) return paInvalidChannelCount; - bitsPerOutputSample = 0; - } - - if( callback == NULL ) return paNullCallback; - - /* Allocate and clear stream structure. */ - past = (internalPortAudioStream *) PaHost_AllocateFastMemory( sizeof(internalPortAudioStream) ); - if( past == NULL ) return paInsufficientMemory; - memset( past, 0, sizeof(internalPortAudioStream) ); - AddTraceMessage("Pa_OpenStream: past", (long) past ); - - past->past_Magic = PA_MAGIC; /* Set ID to catch bugs. */ - past->past_FramesPerUserBuffer = framesPerBuffer; - past->past_NumUserBuffers = numberOfBuffers; /* NOTE - PaHost_OpenStream() NMUST CHECK FOR ZERO! */ - past->past_Callback = callback; - past->past_UserData = userData; - past->past_OutputSampleFormat = outputSampleFormat; - past->past_InputSampleFormat = inputSampleFormat; - past->past_OutputDeviceID = outputDeviceID; - past->past_InputDeviceID = inputDeviceID; - past->past_NumInputChannels = numInputChannels; - past->past_NumOutputChannels = numOutputChannels; - past->past_Flags = streamFlags; - - /* Check for absurd sample rates. */ - if( (sampleRate < 1000.0) || (sampleRate > 200000.0) ) - { - result = paInvalidSampleRate; - goto cleanup; - } - - /* Allocate buffers that may be used for format conversion from user to native buffers. */ - if( numInputChannels > 0 ) - { - -#if PA_VALIDATE_RATE - result = PaHost_ValidateSampleRate( inputDeviceID, sampleRate, &past->past_SampleRate ); - if( result < 0 ) - { - goto cleanup; - } -#else - past->past_SampleRate = sampleRate; -#endif - /* Allocate single Input buffer. */ - past->past_InputBufferSize = framesPerBuffer * numInputChannels * ((bitsPerInputSample+7) / 8); - past->past_InputBuffer = PaHost_AllocateFastMemory(past->past_InputBufferSize); - if( past->past_InputBuffer == NULL ) - { - result = paInsufficientMemory; - goto cleanup; - } - } - else - { - past->past_InputBuffer = NULL; - } - - /* Allocate single Output buffer. */ - if( numOutputChannels > 0 ) - { -#if PA_VALIDATE_RATE - result = PaHost_ValidateSampleRate( outputDeviceID, sampleRate, &past->past_SampleRate ); - if( result < 0 ) - { - goto cleanup; - } -#else - past->past_SampleRate = sampleRate; -#endif - past->past_OutputBufferSize = framesPerBuffer * numOutputChannels * ((bitsPerOutputSample+7) / 8); - past->past_OutputBuffer = PaHost_AllocateFastMemory(past->past_OutputBufferSize); - if( past->past_OutputBuffer == NULL ) - { - result = paInsufficientMemory; - goto cleanup; - } - } - else - { - past->past_OutputBuffer = NULL; - } - - result = PaHost_OpenStream( past ); - if( result < 0 ) goto cleanup; - - *streamPtrPtr = (void *) past; - - return result; - -cleanup: - if( past != NULL ) Pa_CloseStream( past ); - *streamPtrPtr = NULL; - return result; -} - - -/*************************************************************************/ -DLL_API PaError Pa_OpenDefaultStream( PortAudioStream** stream, - int numInputChannels, - int numOutputChannels, - PaSampleFormat sampleFormat, - double sampleRate, - unsigned long framesPerBuffer, - unsigned long numberOfBuffers, - PortAudioCallback *callback, - void *userData ) -{ - return Pa_OpenStream( - stream, - ((numInputChannels > 0) ? Pa_GetDefaultInputDeviceID() : paNoDevice), - numInputChannels, sampleFormat, NULL, - ((numOutputChannels > 0) ? Pa_GetDefaultOutputDeviceID() : paNoDevice), - numOutputChannels, sampleFormat, NULL, - sampleRate, framesPerBuffer, numberOfBuffers, paNoFlag, callback, userData ); -} - -/*************************************************************************/ -DLL_API PaError Pa_CloseStream( PortAudioStream* stream) -{ - PaError result; - internalPortAudioStream *past; - - DBUG(("Pa_CloseStream()\n")); - if( stream == NULL ) return paBadStreamPtr; - past = (internalPortAudioStream *) stream; - - Pa_AbortStream( past ); - result = PaHost_CloseStream( past ); - - if( past->past_InputBuffer ) PaHost_FreeFastMemory( past->past_InputBuffer, past->past_InputBufferSize ); - if( past->past_OutputBuffer ) PaHost_FreeFastMemory( past->past_OutputBuffer, past->past_OutputBufferSize ); - PaHost_FreeFastMemory( past, sizeof(internalPortAudioStream) ); - - return result; -} - -/*************************************************************************/ -DLL_API PaError Pa_StartStream( PortAudioStream *stream ) -{ - PaError result = paHostError; - internalPortAudioStream *past; - - if( stream == NULL ) return paBadStreamPtr; - past = (internalPortAudioStream *) stream; - - past->past_FrameCount = 0.0; - - if( past->past_NumInputChannels > 0 ) - { - result = PaHost_StartInput( past ); - DBUG(("Pa_StartStream: PaHost_StartInput returned = 0x%X.\n", result)); - if( result < 0 ) goto error; - } - - if( past->past_NumOutputChannels > 0 ) - { - result = PaHost_StartOutput( past ); - DBUG(("Pa_StartStream: PaHost_StartOutput returned = 0x%X.\n", result)); - if( result < 0 ) goto error; - } - - result = PaHost_StartEngine( past ); - DBUG(("Pa_StartStream: PaHost_StartEngine returned = 0x%X.\n", result)); - if( result < 0 ) goto error; - - return paNoError; - -error: - return result; -} - -/*************************************************************************/ -DLL_API PaError Pa_StopStream( PortAudioStream *stream ) -{ - return Pa_KillStream( stream, 0 ); -} - -/*************************************************************************/ -DLL_API PaError Pa_AbortStream( PortAudioStream *stream ) -{ - return Pa_KillStream( stream, 1 ); -} - -/*************************************************************************/ -static PaError Pa_KillStream( PortAudioStream *stream, int abort ) -{ - PaError result = paNoError; - internalPortAudioStream *past; - - DBUG(("Pa_StopStream().\n")); - if( stream == NULL ) return paBadStreamPtr; - past = (internalPortAudioStream *) stream; - - if( (past->past_NumInputChannels > 0) || (past->past_NumOutputChannels > 0) ) - { - result = PaHost_StopEngine( past, abort ); - DBUG(("Pa_StopStream: PaHost_StopEngine returned = 0x%X.\n", result)); - if( result < 0 ) goto error; - } - - if( past->past_NumInputChannels > 0 ) - { - result = PaHost_StopInput( past, abort ); - DBUG(("Pa_StopStream: PaHost_StopInput returned = 0x%X.\n", result)); - if( result != paNoError ) goto error; - } - - if( past->past_NumOutputChannels > 0 ) - { - result = PaHost_StopOutput( past, abort ); - DBUG(("Pa_StopStream: PaHost_StopOutput returned = 0x%X.\n", result)); - if( result != paNoError ) goto error; - } - -error: - past->past_Usage = 0; - past->past_IfLastExitValid = 0; - - return result; -} - -/*************************************************************************/ -DLL_API PaError Pa_StreamActive( PortAudioStream *stream ) -{ - internalPortAudioStream *past; - if( stream == NULL ) return paBadStreamPtr; - past = (internalPortAudioStream *) stream; - return PaHost_StreamActive( past ); -} - -/*************************************************************************/ -DLL_API const char *Pa_GetErrorText( PaError errnum ) -{ - const char *msg; - - switch(errnum) - { - case paNoError: msg = "Success"; break; - case paHostError: msg = "Host error."; break; - case paInvalidChannelCount: msg = "Invalid number of channels."; break; - case paInvalidSampleRate: msg = "Invalid sample rate."; break; - case paInvalidDeviceId: msg = "Invalid device ID."; break; - case paInvalidFlag: msg = "Invalid flag."; break; - case paSampleFormatNotSupported: msg = "Sample format not supported"; break; - case paBadIODeviceCombination: msg = "Illegal combination of I/O devices."; break; - case paInsufficientMemory: msg = "Insufficient memory."; break; - case paBufferTooBig: msg = "Buffer too big."; break; - case paBufferTooSmall: msg = "Buffer too small."; break; - case paNullCallback: msg = "No callback routine specified."; break; - case paBadStreamPtr: msg = "Invalid stream pointer."; break; - case paTimedOut : msg = "Wait Timed Out."; break; - case paInternalError: msg = "Internal PortAudio Error."; break; - default: msg = "Illegal error number."; break; - } - return msg; -} - -/* - Get CPU Load as a fraction of total CPU time. - A value of 0.5 would imply that PortAudio and the sound generating - callback was consuming roughly 50% of the available CPU time. - The amount may vary depending on CPU load. - This function may be called from the callback function. -*/ -DLL_API double Pa_GetCPULoad( PortAudioStream* stream) -{ - internalPortAudioStream *past; - if( stream == NULL ) return (double) paBadStreamPtr; - past = (internalPortAudioStream *) stream; - return past->past_Usage; -} - -/************************************************************* -** Calculate 2 LSB dither signal with a triangular distribution. -** Ranged properly for adding to a 32 bit integer prior to >>15. -*/ -#define DITHER_BITS (15) -#define DITHER_SCALE (1.0f / ((1<<DITHER_BITS)-1)) -static long Pa_TriangularDither( void ) -{ - static unsigned long previous = 0; - static unsigned long randSeed1 = 22222; - static unsigned long randSeed2 = 5555555; - long current, highPass; - /* Generate two random numbers. */ - randSeed1 = (randSeed1 * 196314165) + 907633515; - randSeed2 = (randSeed2 * 196314165) + 907633515; - /* Generate triangular distribution about 0. */ - current = (((long)randSeed1)>>(32-DITHER_BITS)) + (((long)randSeed2)>>(32-DITHER_BITS)); - /* High pass filter to reduce audibility. */ - highPass = current - previous; - previous = current; - return highPass; -} - -/************************************************************************* -** Called by host code. -** Convert input from Int16, call user code, then convert output -** to Int16 format for native use. -** Assumes host native format is paInt16. -** Returns result from user callback. -*/ -long Pa_CallConvertInt16( internalPortAudioStream *past, - short *nativeInputBuffer, - short *nativeOutputBuffer ) -{ - long temp; - long bytesEmpty = 0; - long bytesFilled = 0; - int userResult; - unsigned int i; - void *inputBuffer = NULL; - void *outputBuffer = NULL; - -#if SUPPORT_AUDIO_CAPTURE - /* Get native data from DirectSound. */ - if( (past->past_NumInputChannels > 0) && (nativeInputBuffer != NULL) ) - { - /* Convert from native format to PA format. */ - unsigned int samplesPerBuffer = past->past_FramesPerUserBuffer * past->past_NumInputChannels; - switch(past->past_InputSampleFormat) - { - - case paFloat32: - { - float *inBufPtr = (float *) past->past_InputBuffer; - inputBuffer = past->past_InputBuffer; - for( i=0; i<samplesPerBuffer; i++ ) - { - inBufPtr[i] = nativeInputBuffer[i] * (1.0f / 32767.0f); - } - break; - } - - case paInt32: - { - /* Convert 16 bit data to 32 bit integers */ - int *inBufPtr = (int *) past->past_InputBuffer; - inputBuffer = past->past_InputBuffer; - for( i=0; i<samplesPerBuffer; i++ ) - { - inBufPtr[i] = nativeInputBuffer[i] << 16; - } - break; - } - - case paInt16: - { - /* Already in correct format so don't copy. */ - inputBuffer = nativeInputBuffer; - break; - } - - case paInt8: - { - /* Convert 16 bit data to 8 bit chars */ - char *inBufPtr = (char *) past->past_InputBuffer; - inputBuffer = past->past_InputBuffer; - if( past->past_Flags & paDitherOff ) - { - for( i=0; i<samplesPerBuffer; i++ ) - { - inBufPtr[i] = (char)(nativeInputBuffer[i] >> 8); - } - } - else - { - for( i=0; i<samplesPerBuffer; i++ ) - { - temp = nativeInputBuffer[i]; - temp += Pa_TriangularDither() >> 7; - temp = ((temp < -0x8000) ? -0x8000 : ((temp > 0x7FFF) ? 0x7FFF : temp)); - inBufPtr[i] = (char)(temp >> 8); - } - } - break; - } - - case paUInt8: - { - /* Convert 16 bit data to 8 bit unsigned chars */ - unsigned char *inBufPtr = (unsigned char *) past->past_InputBuffer; - inputBuffer = past->past_InputBuffer; - if( past->past_Flags & paDitherOff ) - { - for( i=0; i<samplesPerBuffer; i++ ) - { - inBufPtr[i] = ((unsigned char)(nativeInputBuffer[i] >> 8)) + 0x80; - } - } - else - { - /* If you dither then you have to clip because dithering could push the signal out of range! */ - for( i=0; i<samplesPerBuffer; i++ ) - { - temp = nativeInputBuffer[i]; - temp += Pa_TriangularDither() >> 7; - temp = ((temp < -0x8000) ? -0x8000 : ((temp > 0x7FFF) ? 0x7FFF : temp)); - inBufPtr[i] = (unsigned char)(temp + 0x80); - } - } - break; - } - - default: - break; - } - } -#endif /* SUPPORT_AUDIO_CAPTURE */ - - /* Are we doing output time? */ - if( (past->past_NumOutputChannels > 0) && (nativeOutputBuffer != NULL) ) - { - /* May already be in native format so just write directly to native buffer. */ - outputBuffer = (past->past_OutputSampleFormat == paInt16) ? - nativeOutputBuffer : past->past_OutputBuffer; - } - /* - AddTraceMessage("Pa_CallConvertInt16: inputBuffer = ", (int) inputBuffer ); - AddTraceMessage("Pa_CallConvertInt16: outputBuffer = ", (int) outputBuffer ); - */ - /* Call user callback routine. */ - userResult = past->past_Callback( - inputBuffer, - outputBuffer, - past->past_FramesPerUserBuffer, - past->past_FrameCount, - past->past_UserData ); - - past->past_FrameCount += (PaTimestamp) past->past_FramesPerUserBuffer; - - /* Convert to native format if necessary. */ - if( outputBuffer != NULL ) - { - unsigned int samplesPerBuffer = past->past_FramesPerUserBuffer * past->past_NumOutputChannels; - switch(past->past_OutputSampleFormat) - { - case paFloat32: - { - float *outBufPtr = (float *) past->past_OutputBuffer; - if( past->past_Flags & paDitherOff ) - { - if( past->past_Flags & paClipOff ) /* NOTHING */ - { - for( i=0; i<samplesPerBuffer; i++ ) - { - *nativeOutputBuffer++ = (short) (outBufPtr[i] * (32767.0f)); - } - } - else /* CLIP */ - { - for( i=0; i<samplesPerBuffer; i++ ) - { - temp = (long)(outBufPtr[i] * 32767.0f); - *nativeOutputBuffer++ = (short)((temp < -0x8000) ? -0x8000 : ((temp > 0x7FFF) ? 0x7FFF : temp)); - } - } - } - else - { - /* If you dither then you have to clip because dithering could push the signal out of range! */ - for( i=0; i<samplesPerBuffer; i++ ) - { - float dither = Pa_TriangularDither()*DITHER_SCALE; - float dithered = (outBufPtr[i] * (32767.0f)) + dither; - temp = (long) (dithered); - *nativeOutputBuffer++ = (short)((temp < -0x8000) ? -0x8000 : ((temp > 0x7FFF) ? 0x7FFF : temp)); - } - } - break; - } - - case paInt32: - { - int *outBufPtr = (int *) past->past_OutputBuffer; - if( past->past_Flags & paDitherOff ) - { - for( i=0; i<samplesPerBuffer; i++ ) - { - *nativeOutputBuffer++ = (short) (outBufPtr[i] >> 16 ); - } - } - else - { - for( i=0; i<samplesPerBuffer; i++ ) - { - /* Shift one bit down before dithering so that we have room for overflow from add. */ - temp = (outBufPtr[i] >> 1) + Pa_TriangularDither(); - temp = temp >> 15; - *nativeOutputBuffer++ = (short)((temp < -0x8000) ? -0x8000 : ((temp > 0x7FFF) ? 0x7FFF : temp)); - } - } - break; - } - - case paInt8: - { - char *outBufPtr = (char *) past->past_OutputBuffer; - for( i=0; i<samplesPerBuffer; i++ ) - { - *nativeOutputBuffer++ = ((short)outBufPtr[i]) << 8; - } - break; - } - - case paUInt8: - { - unsigned char *outBufPtr = (unsigned char *) past->past_OutputBuffer; - for( i=0; i<samplesPerBuffer; i++ ) - { - *nativeOutputBuffer++ = ((short)(outBufPtr[i] - 0x80)) << 8; - } - break; - } - - default: - break; - } - - } - - return userResult; -} - -/************************************************************************* -** Called by host code. -** Convert input from Float32, call user code, then convert output -** to Float32 format for native use. -** Assumes host native format is Float32. -** Returns result from user callback. -** FIXME - Unimplemented for formats other than paFloat32!!!! -*/ -long Pa_CallConvertFloat32( internalPortAudioStream *past, - float *nativeInputBuffer, - float *nativeOutputBuffer ) -{ - long bytesEmpty = 0; - long bytesFilled = 0; - int userResult; - void *inputBuffer = NULL; - void *outputBuffer = NULL; - - /* Get native data from DirectSound. */ - if( (past->past_NumInputChannels > 0) && (nativeInputBuffer != NULL) ) - { - inputBuffer = nativeInputBuffer; // FIXME - } - - /* Are we doing output time? */ - if( (past->past_NumOutputChannels > 0) && (nativeOutputBuffer != NULL) ) - { - /* May already be in native format so just write directly to native buffer. */ - outputBuffer = (past->past_OutputSampleFormat == paFloat32) ? - nativeOutputBuffer : past->past_OutputBuffer; - } - /* - AddTraceMessage("Pa_CallConvertInt16: inputBuffer = ", (int) inputBuffer ); - AddTraceMessage("Pa_CallConvertInt16: outputBuffer = ", (int) outputBuffer ); - */ - /* Call user callback routine. */ - userResult = past->past_Callback( - inputBuffer, - outputBuffer, - past->past_FramesPerUserBuffer, - past->past_FrameCount, - past->past_UserData ); - - past->past_FrameCount += (PaTimestamp) past->past_FramesPerUserBuffer; - - /* Convert to native format if necessary. */ // FIXME - return userResult; -} - -/*************************************************************************/ -DLL_API PaError Pa_Initialize( void ) -{ - if( gInitCount++ > 0 ) return paNoError; - ResetTraceMessages(); - return PaHost_Init(); -} - -DLL_API PaError Pa_Terminate( void ) -{ - PaError result = paNoError; - - if( gInitCount == 0 ) return paNoError; - else if( --gInitCount == 0 ) - { - result = PaHost_Term(); - DumpTraceMessages(); - } - return result; -} - -/*************************************************************************/ -DLL_API PaError Pa_GetSampleSize( PaSampleFormat format ) -{ - int size; - switch(format ) - { - - case paUInt8: - case paInt8: - size = 1; - break; - - case paInt16: - size = 2; - break; - - case paPackedInt24: - size = 3; - break; - - case paFloat32: - case paInt32: - case paInt24: - size = 4; - break; - - default: - size = paSampleFormatNotSupported; - break; - } - return (PaError) size; -} - - diff --git a/pd/portaudio_v18/pa_dll_switch/portaudio.h b/pd/portaudio_v18/pa_dll_switch/portaudio.h deleted file mode 100644 index 9632521e..00000000 --- a/pd/portaudio_v18/pa_dll_switch/portaudio.h +++ /dev/null @@ -1,439 +0,0 @@ -#ifndef PORT_AUDIO_H -#define PORT_AUDIO_H - -#ifdef __cplusplus -extern "C" -{ -#endif /* __cplusplus */ - -/* - * PortAudio Portable Real-Time Audio Library - * PortAudio API Header File - * Latest version available at: http://www.audiomulch.com/portaudio/ - * - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ - -// added by zplane.developement in order to generate a DLL - -#if defined(PA_MME_EXPORTS) || defined(PA_DX_EXPORTS) -#define DLL_API __declspec( dllexport ) -#elif defined(_LIB) || defined(_STATIC_LINK) || defined(_STATIC_APP) -#define DLL_API -#else -#define DLL_API __declspec(dllexport) -#endif - - -typedef int PaError; -typedef enum { - paNoError = 0, - - paHostError = -10000, - paInvalidChannelCount, - paInvalidSampleRate, - paInvalidDeviceId, - paInvalidFlag, - paSampleFormatNotSupported, - paBadIODeviceCombination, - paInsufficientMemory, - paBufferTooBig, - paBufferTooSmall, - paNullCallback, - paBadStreamPtr, - paTimedOut, - paInternalError -} PaErrorNum; - -/* - Pa_Initialize() is the library initialisation function - call this before - using the library. -*/ - -DLL_API PaError Pa_Initialize( void ); - -/* - Pa_Terminate() is the library termination function - call this after - using the library. -*/ - -DLL_API PaError Pa_Terminate( void ); - -/* - Return host specific error. - This can be called after receiving a paHostError. -*/ -DLL_API long Pa_GetHostError( void ); - -/* - Translate the error number into a human readable message. -*/ -DLL_API const char *Pa_GetErrorText( PaError errnum ); - -/* - Sample formats - - These are formats used to pass sound data between the callback and the - stream. Each device has a "native" format which may be used when optimum - efficiency or control over conversion is required. - - Formats marked "always available" are supported (emulated) by all devices. - - The floating point representation uses +1.0 and -1.0 as the respective - maximum and minimum. - -*/ - -typedef unsigned long PaSampleFormat; -#define paFloat32 ((PaSampleFormat) (1<<0)) /*always available*/ -#define paInt16 ((PaSampleFormat) (1<<1)) /*always available*/ -#define paInt32 ((PaSampleFormat) (1<<2)) /*always available*/ -#define paInt24 ((PaSampleFormat) (1<<3)) -#define paPackedInt24 ((PaSampleFormat) (1<<4)) -#define paInt8 ((PaSampleFormat) (1<<5)) -#define paUInt8 ((PaSampleFormat) (1<<6)) /* unsigned 8 bit, 128 is "ground" */ -#define paCustomFormat ((PaSampleFormat) (1<<16)) - -/* - Device enumeration mechanism. - - Device ids range from 0 to Pa_CountDevices()-1. - - Devices may support input, output or both. Device 0 is always the "default" - device and should support at least stereo in and out if that is available - on the taget platform _even_ if this involves kludging an input/output - device on platforms that usually separate input from output. Other platform - specific devices are specified by positive device ids. -*/ - -typedef int PaDeviceID; -#define paNoDevice -1 - -typedef struct -{ - int structVersion; - const char *name; - int maxInputChannels; - int maxOutputChannels; - /* Number of discrete rates, or -1 if range supported. */ - int numSampleRates; - /* Array of supported sample rates, or {min,max} if range supported. */ - const double *sampleRates; - PaSampleFormat nativeSampleFormats; -} -PaDeviceInfo; - - -DLL_API int Pa_CountDevices(); -/* - Pa_GetDefaultInputDeviceID(), Pa_GetDefaultOutputDeviceID() - - Return the default device ID or paNoDevice if there is no devices. - The result can be passed to Pa_OpenStream(). - - On the PC, the user can specify a default device by - setting an environment variable. For example, to use device #1. - - set PA_RECOMMENDED_OUTPUT_DEVICE=1 - - The user should first determine the available device ID by using - the supplied application "pa_devs". -*/ -DLL_API PaDeviceID Pa_GetDefaultInputDeviceID( void ); -DLL_API PaDeviceID Pa_GetDefaultOutputDeviceID( void ); - -/* - PaTimestamp is used to represent a continuous sample clock with arbitrary - start time useful for syncronisation. The type is used in the outTime - argument to the callback function and the result of Pa_StreamTime() -*/ - -typedef double PaTimestamp; - -/* - Pa_GetDeviceInfo() returns a pointer to an immutable PaDeviceInfo structure - referring to the device specified by id. - If id is out of range the function returns NULL. - - The returned structure is owned by the PortAudio implementation and must - not be manipulated or freed. The pointer is guaranteed to be valid until - between calls to Pa_Initialize() and Pa_Terminate(). -*/ - -DLL_API const PaDeviceInfo* Pa_GetDeviceInfo( PaDeviceID id ); - -/* - PortAudioCallback is implemented by clients of the portable audio api. - - inputBuffer and outputBuffer are arrays of interleaved samples, - the format, packing and number of channels used by the buffers are - determined by parameters to Pa_OpenStream() (see below). - - framesPerBuffer is the number of sample frames to be processed by the callback. - - outTime is the time in samples when the buffer(s) processed by - this callback will begin being played at the audio output. - See also Pa_StreamTime() - - userData is the value of a user supplied pointer passed to Pa_OpenStream() - intended for storing synthesis data etc. - - return value: - The callback can return a nonzero value to stop the stream. This may be - useful in applications such as soundfile players where a specific duration - of output is required. However, it is not necessary to utilise this mechanism - as StopStream() will also terminate the stream. A callback returning a - nonzero value must fill the entire outputBuffer. - - NOTE: None of the other stream functions may be called from within the - callback function except for Pa_GetCPULoad(). - -*/ - -typedef int (PortAudioCallback)( - void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ); - - -/* - Stream flags - - These flags may be supplied (ored together) in the streamFlags argument to - the Pa_OpenStream() function. - - [ suggestions? ] -*/ - -#define paNoFlag (0) -#define paClipOff (1<<0) /* disable defult clipping of out of range samples */ -#define paDitherOff (1<<1) /* disable default dithering */ -#define paPlatformSpecificFlags (0x00010000) -typedef unsigned long PaStreamFlags; - -/* - A single PortAudioStream provides multiple channels of real-time - input and output audio streaming to a client application. - Pointers to PortAudioStream objects are passed between PortAudio functions. -*/ - -typedef void PortAudioStream; -#define PaStream PortAudioStream - -/* - Pa_OpenStream() opens a stream for either input, output or both. - - stream is the address of a PortAudioStream pointer which will receive - a pointer to the newly opened stream. - - inputDevice is the id of the device used for input (see PaDeviceID above.) - inputDevice may be paNoDevice to indicate that an input device is not required. - - numInputChannels is the number of channels of sound to be delivered to the - callback. It can range from 1 to the value of maxInputChannels in the - device input record for the device specified in the inputDevice parameter. - If inputDevice is paNoDevice numInputChannels is ignored. - - inputSampleFormat is the format of inputBuffer provided to the callback - function. inputSampleFormat may be any of the formats described by the - PaSampleFormat enumeration (see above). PortAudio guarantees support for - the sound devices native formats (nativeSampleFormats in the device info - record) and additionally 16 and 32 bit integer and 32 bit floating point - formats. Support for other formats is implementation defined. - - inputDriverInfo is a pointer to an optional driver specific data structure - containing additional information for device setup or stream processing. - inputDriverInfo is never required for correct operation. If not used - inputDriverInfo should be NULL. - - outputDevice is the id of the device used for output (see PaDeviceID above.) - outputDevice may be paNoDevice to indicate that an output device is not required. - - numOutputChannels is the number of channels of sound to be supplied by the - callback. See the definition of numInputChannels above for more details. - - outputSampleFormat is the sample format of the outputBuffer filled by the - callback function. See the definition of inputSampleFormat above for more - details. - - outputDriverInfo is a pointer to an optional driver specific data structure - containing additional information for device setup or stream processing. - outputDriverInfo is never required for correct operation. If not used - outputDriverInfo should be NULL. - - sampleRate is the desired sampleRate for input and output - - framesPerBuffer is the length in sample frames of all internal sample buffers - used for communication with platform specific audio routines. Wherever - possible this corresponds to the framesPerBuffer parameter passed to the - callback function. - - numberOfBuffers is the number of buffers used for multibuffered - communication with the platform specific audio routines. This parameter is - provided only as a guide - and does not imply that an implementation must - use multibuffered i/o when reliable double buffering is available (such as - SndPlayDoubleBuffer() on the Macintosh.) - - streamFlags may contain a combination of flags ORed together. - These flags modify the behavior of the - streaming process. Some flags may only be relevant to certain buffer formats. - - callback is a pointer to a client supplied function that is responsible - for processing and filling input and output buffers (see above for details.) - - userData is a client supplied pointer which is passed to the callback - function. It could for example, contain a pointer to instance data necessary - for processing the audio buffers. - - return value: - Apon success Pa_OpenStream() returns PaNoError and places a pointer to a - valid PortAudioStream in the stream argument. The stream is inactive (stopped). - If a call to Pa_OpenStream() fails a nonzero error code is returned (see - PAError above) and the value of stream is invalid. - -*/ - -DLL_API PaError Pa_OpenStream( PortAudioStream** stream, - PaDeviceID inputDevice, - int numInputChannels, - PaSampleFormat inputSampleFormat, - void *inputDriverInfo, - PaDeviceID outputDevice, - int numOutputChannels, - PaSampleFormat outputSampleFormat, - void *outputDriverInfo, - double sampleRate, - unsigned long framesPerBuffer, - unsigned long numberOfBuffers, - PaStreamFlags streamFlags, - PortAudioCallback *callback, - void *userData ); - - -/* - Pa_OpenDefaultStream() is a simplified version of Pa_OpenStream() that - opens the default input and/or ouput devices. Most parameters have - identical meaning to their Pa_OpenStream() counterparts, with the following - exceptions: - - If either numInputChannels or numOutputChannels is 0 the respective device - is not opened (same as passing paNoDevice in the device arguments to Pa_OpenStream() ) - - sampleFormat applies to both the input and output buffers. -*/ - -DLL_API PaError Pa_OpenDefaultStream( PortAudioStream** stream, - int numInputChannels, - int numOutputChannels, - PaSampleFormat sampleFormat, - double sampleRate, - unsigned long framesPerBuffer, - unsigned long numberOfBuffers, - PortAudioCallback *callback, - void *userData ); - -/* - Pa_CloseStream() closes an audio stream, flushing any pending buffers. -*/ - -DLL_API PaError Pa_CloseStream( PortAudioStream* ); - -/* - Pa_StartStream() and Pa_StopStream() begin and terminate audio processing. - When Pa_StopStream() returns, all pending audio buffers have been played. - Pa_AbortStream() stops playing immediately without waiting for pending - buffers to complete. -*/ - -DLL_API PaError Pa_StartStream( PortAudioStream *stream ); - -DLL_API PaError Pa_StopStream( PortAudioStream *stream ); - -DLL_API PaError Pa_AbortStream( PortAudioStream *stream ); - -/* - Pa_StreamActive() returns one when the stream is playing audio, - zero when not playing, or a negative error number if the - stream is invalid. - The stream is active between calls to Pa_StartStream() and Pa_StopStream(), - but may also become inactive if the callback returns a non-zero value. - In the latter case, the stream is considered inactive after the last - buffer has finished playing. -*/ - -DLL_API PaError Pa_StreamActive( PortAudioStream *stream ); - -/* - Pa_StreamTime() returns the current output time for the stream in samples. - This time may be used as a time reference (for example syncronising audio to - MIDI). -*/ - -DLL_API PaTimestamp Pa_StreamTime( PortAudioStream *stream ); - -/* - The "CPU Load" is a fraction of total CPU time consumed by the - stream's audio processing. - A value of 0.5 would imply that PortAudio and the sound generating - callback was consuming roughly 50% of the available CPU time. - This function may be called from the callback function or the application. -*/ -DLL_API double Pa_GetCPULoad( PortAudioStream* stream ); - -/* - Use Pa_GetMinNumBuffers() to determine minimum number of buffers required for - the current host based on minimum latency. - On the PC, for the DirectSound implementation, latency can be optionally set - by user by setting an environment variable. - For example, to set latency to 200 msec, put: - - set PA_MIN_LATENCY_MSEC=200 - - in the AUTOEXEC.BAT file and reboot. - If the environment variable is not set, then the latency will be determined - based on the OS. Windows NT has higher latency than Win95. -*/ - -DLL_API int Pa_GetMinNumBuffers( int framesPerBuffer, double sampleRate ); - -/* - Sleep for at least 'msec' milliseconds. - You may sleep longer than the requested time so don't rely - on this for accurate musical timing. -*/ -DLL_API void Pa_Sleep( long msec ); - -/* - Return size in bytes of a single sample in a given PaSampleFormat - or paSampleFormatNotSupported. -*/ -DLL_API PaError Pa_GetSampleSize( PaSampleFormat format ); - -#ifdef __cplusplus -} -#endif /* __cplusplus */ -#endif /* PORT_AUDIO_H */ diff --git a/pd/portaudio_v18/pa_mac/pa_mac.c b/pd/portaudio_v18/pa_mac/pa_mac.c deleted file mode 100644 index 118ea510..00000000 --- a/pd/portaudio_v18/pa_mac/pa_mac.c +++ /dev/null @@ -1,1687 +0,0 @@ -/* - * $Id: pa_mac.c,v 1.4.4.2 2002/10/15 03:14:08 dmazzoni Exp $ - * Portable Audio I/O Library for Macintosh - * - * Based on the Open Source API proposed by Ross Bencina - * Copyright (c) 1999-2000 Phil Burk - * - * Special thanks to Chris Rolfe for his many helpful suggestions, bug fixes, - * and code contributions. - * Thanks also to Tue Haste Andersen, Alberto Ricci, Nico Wald, - * Roelf Toxopeus and Tom Erbe for testing the code and making - * numerous suggestions. - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - */ -/* Modification History - PLB20010415 - ScanInputDevices was setting sDefaultOutputDeviceID instead of sDefaultInputDeviceID - PLB20010415 - Device Scan was crashing for anything other than siBadSoundInDevice, but some Macs may return other errors! - PLB20010420 - Fix TIMEOUT in record mode. - PLB20010420 - Change CARBON_COMPATIBLE to TARGET_API_MAC_CARBON - PLB20010907 - Pass unused event to WaitNextEvent to prevent Mac OSX crash. Thanks Dominic Mazzoni. - PLB20010908 - Use requested number of input channels. Thanks Dominic Mazzoni. - PLB20011009 - Use NewSndCallBackUPP() for CARBON - PLB20020417 - I used to call Pa_GetMinNumBuffers() which doesn't take into account the - variable minFramesPerHostBuffer. Now I call PaMac_GetMinNumBuffers() which will - give lower latency when virtual memory is turned off. - Thanks Kristoffer Jensen and Georgios Marentakis for spotting this bug. - PLB20020423 - Use new method to calculate CPU load similar to other ports. Based on num frames calculated. - Fixed Pa_StreamTime(). Now estimates how many frames have played based on MicroSecond timer. - Added PA_MAX_USAGE_ALLOWED to prevent Mac from hanging when CPU load approaches 100%. - PLB20020424 - Fixed return value in Pa_StreamTime - PLB20020612 - Fix allocation error on Mac 8600 by casting *nameH as uchar* so that we get a proper Str255 length. -*/ - -/* -COMPATIBILITY -This Macintosh implementation is designed for use with Mac OS 7, 8 and -9 on PowerMacs, and OS X if compiled with CARBON - -OUTPUT -A circular array of CmpSoundHeaders is used as a queue. For low latency situations -there will only be two small buffers used. For higher latency, more and larger buffers -may be used. -To play the sound we use SndDoCommand() with bufferCmd. Each buffer is followed -by a callbackCmd which informs us when the buffer has been processsed. - -INPUT -The SndInput Manager SPBRecord call is used for sound input. If only -input is used, then the PA user callback is called from the Input completion proc. -For full-duplex, or output only operation, the PA callback is called from the -HostBuffer output completion proc. In that case, input sound is passed to the -callback by a simple FIFO. - -TODO: -O- Add support for native sample data formats other than int16. -O- Review buffer sizing. Should it be based on result of siDeviceBufferInfo query? -O- Determine default devices somehow. -*/ -#include <stdio.h> -#include <stdlib.h> -#include <string.h> -#include <memory.h> -#include <math.h> - -/* Mac specific includes */ -#include "OSUtils.h" -#include <MacTypes.h> -#include <Math64.h> -#include <Errors.h> -#include <Sound.h> -#include <SoundInput.h> -#include <SoundComponents.h> -#include <Devices.h> -#include <DateTimeUtils.h> -#include <Timer.h> -#include <Gestalt.h> - -#include "portaudio.h" -#include "pa_host.h" -#include "pa_trace.h" - -#ifndef FALSE - #define FALSE (0) - #define TRUE (!FALSE) -#endif - -/* #define TARGET_API_MAC_CARBON (1) */ - -/* - * Define maximum CPU load that will be allowed. User callback will - * be skipped if load exceeds this limit. This is to prevent the Mac - * from hanging when the CPU is hogged by the sound thread. - * On my PowerBook G3, the mac hung when I used 94% of CPU ( usage = 0.94 ). - */ -#define PA_MAX_USAGE_ALLOWED (0.92) - -/* Debugging output macros. */ -#define PRINT(x) { printf x; fflush(stdout); } -#define ERR_RPT(x) PRINT(x) -#define DBUG(x) /* PRINT(x) */ -#define DBUGX(x) /* PRINT(x) */ - -#define MAC_PHYSICAL_FRAMES_PER_BUFFER (512) /* Minimum number of stereo frames per SoundManager double buffer. */ -#define MAC_VIRTUAL_FRAMES_PER_BUFFER (4096) /* Need this many when using Virtual Memory for recording. */ -#define PA_MIN_NUM_HOST_BUFFERS (2) -#define PA_MAX_NUM_HOST_BUFFERS (16) /* Do not exceed!! */ -#define PA_MAX_DEVICE_INFO (32) - -/* Conversions for 16.16 fixed point code. */ -#define DoubleToUnsignedFixed(x) ((UnsignedFixed) ((x) * 65536.0)) -#define UnsignedFixedToDouble(fx) (((double)(fx)) * (1.0/(1<<16))) - -/************************************************************************************/ -/****************** Structures ******************************************************/ -/************************************************************************************/ -/* Use for passing buffers from input callback to output callback for processing. */ -typedef struct MultiBuffer -{ - char *buffers[PA_MAX_NUM_HOST_BUFFERS]; - int numBuffers; - int nextWrite; - int nextRead; -} -MultiBuffer; - -/* Define structure to contain all Macintosh specific data. */ -typedef struct PaHostSoundControl -{ - UInt64 pahsc_EntryCount; - double pahsc_InverseMicrosPerHostBuffer; /* 1/Microseconds of real-time audio per user buffer. */ - - /* Use char instead of Boolean for atomic operation. */ - volatile char pahsc_IsRecording; /* Recording in progress. Set by foreground. Cleared by background. */ - volatile char pahsc_StopRecording; /* Signal sent to background. */ - volatile char pahsc_IfInsideCallback; - /* Input */ - SPB pahsc_InputParams; - SICompletionUPP pahsc_InputCompletionProc; - MultiBuffer pahsc_InputMultiBuffer; - int32 pahsc_BytesPerInputHostBuffer; - int32 pahsc_InputRefNum; - /* Output */ - CmpSoundHeader pahsc_SoundHeaders[PA_MAX_NUM_HOST_BUFFERS]; - int32 pahsc_BytesPerOutputHostBuffer; - SndChannelPtr pahsc_Channel; - SndCallBackUPP pahsc_OutputCompletionProc; - int32 pahsc_NumOutsQueued; - int32 pahsc_NumOutsPlayed; - PaTimestamp pahsc_NumFramesDone; - UInt64 pahsc_WhenFramesDoneIncremented; - /* Init Time -------------- */ - int32 pahsc_NumHostBuffers; - int32 pahsc_FramesPerHostBuffer; - int32 pahsc_UserBuffersPerHostBuffer; - int32 pahsc_MinFramesPerHostBuffer; /* Can vary depending on virtual memory usage. */ -} -PaHostSoundControl; - -/* Mac specific device information. */ -typedef struct internalPortAudioDevice -{ - long pad_DeviceRefNum; - long pad_DeviceBufferSize; - Component pad_Identifier; - PaDeviceInfo pad_Info; -} -internalPortAudioDevice; - -/************************************************************************************/ -/****************** Data ************************************************************/ -/************************************************************************************/ -static int sNumDevices = 0; -static internalPortAudioDevice sDevices[PA_MAX_DEVICE_INFO] = { 0 }; -static int32 sPaHostError = 0; -static int sDefaultOutputDeviceID; -static int sDefaultInputDeviceID; - -/************************************************************************************/ -/****************** Prototypes ******************************************************/ -/************************************************************************************/ -static PaError PaMac_TimeSlice( internalPortAudioStream *past, int16 *macOutputBufPtr ); -static PaError PaMac_CallUserLoop( internalPortAudioStream *past, int16 *outPtr ); -static PaError PaMac_RecordNext( internalPortAudioStream *past ); -static void PaMac_StartLoadCalculation( internalPortAudioStream *past ); -static int PaMac_GetMinNumBuffers( int minFramesPerHostBuffer, int framesPerBuffer, double sampleRate ); -static double *PaMac_GetSampleRatesFromHandle ( int numRates, Handle h ); -static PaError PaMac_ScanInputDevices( void ); -static PaError PaMac_ScanOutputDevices( void ); -static PaError PaMac_QueryOutputDeviceInfo( Component identifier, internalPortAudioDevice *ipad ); -static PaError PaMac_QueryInputDeviceInfo( Str255 deviceName, internalPortAudioDevice *ipad ); -static void PaMac_InitSoundHeader( internalPortAudioStream *past, CmpSoundHeader *sndHeader ); -static void PaMac_EndLoadCalculation( internalPortAudioStream *past ); -static void PaMac_PlayNext ( internalPortAudioStream *past, int index ); -static long PaMac_FillNextOutputBuffer( internalPortAudioStream *past, int index ); -static pascal void PaMac_InputCompletionProc(SPBPtr recParams); -static pascal void PaMac_OutputCompletionProc (SndChannelPtr theChannel, SndCommand * theCmd); -static PaError PaMac_BackgroundManager( internalPortAudioStream *past, int index ); -long PaHost_GetTotalBufferFrames( internalPortAudioStream *past ); -static int Mac_IsVirtualMemoryOn( void ); -static void PToCString(unsigned char* inString, char* outString); -static void CToPString(char *inString, unsigned char* outString); -char *MultiBuffer_GetNextWriteBuffer( MultiBuffer *mbuf ); -char *MultiBuffer_GetNextReadBuffer( MultiBuffer *mbuf ); -int MultiBuffer_GetNextReadIndex( MultiBuffer *mbuf ); -int MultiBuffer_GetNextWriteIndex( MultiBuffer *mbuf ); -int MultiBuffer_IsWriteable( MultiBuffer *mbuf ); -int MultiBuffer_IsReadable( MultiBuffer *mbuf ); -void MultiBuffer_AdvanceReadIndex( MultiBuffer *mbuf ); -void MultiBuffer_AdvanceWriteIndex( MultiBuffer *mbuf ); - -/************************************************************************* -** Simple FIFO index control for multiple buffers. -** Read and Write indices range from 0 to 2N-1. -** This allows us to distinguish between full and empty. -*/ -char *MultiBuffer_GetNextWriteBuffer( MultiBuffer *mbuf ) -{ - return mbuf->buffers[mbuf->nextWrite % mbuf->numBuffers]; -} -char *MultiBuffer_GetNextReadBuffer( MultiBuffer *mbuf ) -{ - return mbuf->buffers[mbuf->nextRead % mbuf->numBuffers]; -} -int MultiBuffer_GetNextReadIndex( MultiBuffer *mbuf ) -{ - return mbuf->nextRead % mbuf->numBuffers; -} -int MultiBuffer_GetNextWriteIndex( MultiBuffer *mbuf ) -{ - return mbuf->nextWrite % mbuf->numBuffers; -} - -int MultiBuffer_IsWriteable( MultiBuffer *mbuf ) -{ - int bufsFull = mbuf->nextWrite - mbuf->nextRead; - if( bufsFull < 0 ) bufsFull += (2 * mbuf->numBuffers); - return (bufsFull < mbuf->numBuffers); -} -int MultiBuffer_IsReadable( MultiBuffer *mbuf ) -{ - int bufsFull = mbuf->nextWrite - mbuf->nextRead; - if( bufsFull < 0 ) bufsFull += (2 * mbuf->numBuffers); - return (bufsFull > 0); -} -void MultiBuffer_AdvanceReadIndex( MultiBuffer *mbuf ) -{ - int temp = mbuf->nextRead + 1; - mbuf->nextRead = (temp >= (2 * mbuf->numBuffers)) ? 0 : temp; -} -void MultiBuffer_AdvanceWriteIndex( MultiBuffer *mbuf ) -{ - int temp = mbuf->nextWrite + 1; - mbuf->nextWrite = (temp >= (2 * mbuf->numBuffers)) ? 0 : temp; -} - -/************************************************************************* -** String Utility by Chris Rolfe -*/ -static void PToCString(unsigned char* inString, char* outString) -{ - long i; - for(i=0; i<inString[0]; i++) /* convert Pascal to C string */ - outString[i] = inString[i+1]; - outString[i]=0; -} - -/************************************************************************* -** String Utility by Dominic Mazzoni -*/ -static void CToPString(char* inString, unsigned char* outString) -{ - long len = strlen(inString); - long i; - - if (len > 255) - len = 255; - - /* Length is stored in first char of Pascal string */ - outString[0] = (unsigned char)len; - for(i=0; i<len; i++) - outString[i+1] = inString[i]; -} - -/*************************************************************************/ -PaError PaHost_Term( void ) -{ - int i; - PaDeviceInfo *dev; - double *rates; - /* Free any allocated sample rate arrays. */ - for( i=0; i<sNumDevices; i++ ) - { - dev = &sDevices[i].pad_Info; - rates = (double *) dev->sampleRates; - if( (rates != NULL) ) free( rates ); /* MEM_011 */ - dev->sampleRates = NULL; - if( dev->name != NULL ) free( (void *) dev->name ); /* MEM_010 */ - dev->name = NULL; - } - sNumDevices = 0; - return paNoError; -} - -/************************************************************************* - PaHost_Init() is the library initialization function - call this before - using the library. -*/ -PaError PaHost_Init( void ) -{ - PaError err; - NumVersionVariant version; - - version.parts = SndSoundManagerVersion(); - DBUG(("SndSoundManagerVersion = 0x%x\n", version.whole)); - - /* Have we already initialized the device info? */ - err = (PaError) Pa_CountDevices(); - if( err < 0 ) return err; - else return paNoError; -} - -/************************************************************************* - PaMac_ScanOutputDevices() queries the properties of all output devices. -*/ -static PaError PaMac_ScanOutputDevices( void ) -{ - PaError err; - Component identifier=0; - ComponentDescription criteria = { kSoundOutputDeviceType, 0, 0, 0, 0 }; - long numComponents, i; - - /* Search the system linked list for output components */ - numComponents = CountComponents (&criteria); - identifier = 0; - sDefaultOutputDeviceID = sNumDevices; /* FIXME - query somehow */ - for (i = 0; i < numComponents; i++) - { - /* passing nil returns first matching component. */ - identifier = FindNextComponent( identifier, &criteria); - sDevices[sNumDevices].pad_Identifier = identifier; - - /* Set up for default OUTPUT devices. */ - err = PaMac_QueryOutputDeviceInfo( identifier, &sDevices[sNumDevices] ); - if( err < 0 ) return err; - else sNumDevices++; - - } - - return paNoError; -} - -/************************************************************************* - PaMac_ScanInputDevices() queries the properties of all input devices. -*/ -static PaError PaMac_ScanInputDevices( void ) -{ - Str255 deviceName; - int count; - Handle iconHandle; - PaError err; - OSErr oserr; - count = 1; - sDefaultInputDeviceID = sNumDevices; /* FIXME - query somehow */ /* PLB20010415 - was setting sDefaultOutputDeviceID */ - while(true) - { - /* Thanks Chris Rolfe and Alberto Ricci for this trick. */ - oserr = SPBGetIndexedDevice(count++, deviceName, &iconHandle); - DBUG(("PaMac_ScanInputDevices: SPBGetIndexedDevice returned %d\n", oserr )); -#if 1 - /* PLB20010415 - was giving error for anything other than siBadSoundInDevice, but some Macs may return other errors! */ - if(oserr != noErr) break; /* Some type of error is expected when count > devices */ -#else - if(oserr == siBadSoundInDevice) - { /* it's expected when count > devices */ - oserr = noErr; - break; - } - if(oserr != noErr) - { - ERR_RPT(("ERROR: SPBGetIndexedDevice(%d,,) returned %d\n", count-1, oserr )); - sPaHostError = oserr; - return paHostError; - } -#endif - DisposeHandle(iconHandle); /* Don't need the icon */ - - err = PaMac_QueryInputDeviceInfo( deviceName, &sDevices[sNumDevices] ); - DBUG(("PaMac_ScanInputDevices: PaMac_QueryInputDeviceInfo returned %d\n", err )); - if( err < 0 ) return err; - else if( err == 1 ) sNumDevices++; - } - - return paNoError; -} - -/* Sample rate info returned by using siSampleRateAvailable selector in SPBGetDeviceInfo() */ -/* Thanks to Chris Rolfe for help with this query. */ -#pragma options align=mac68k -typedef struct -{ - int16 numRates; - UnsignedFixed (**rates)[]; /* Handle created by SPBGetDeviceInfo */ -} -SRateInfo; -#pragma options align=reset - -/************************************************************************* -** PaMac_QueryOutputDeviceInfo() -** Query information about a named output device. -** Clears contents of ipad and writes info based on queries. -** Return one if OK, -** zero if device cannot be used, -** or negative error. -*/ -static PaError PaMac_QueryOutputDeviceInfo( Component identifier, internalPortAudioDevice *ipad ) -{ - int len; - OSErr err; - PaDeviceInfo *dev = &ipad->pad_Info; - SRateInfo srinfo = {0}; - int numRates; - ComponentDescription tempD; - Handle nameH=nil, infoH=nil, iconH=nil; - - memset( ipad, 0, sizeof(internalPortAudioDevice) ); - - dev->structVersion = 1; - dev->maxInputChannels = 0; - dev->maxOutputChannels = 2; - dev->nativeSampleFormats = paInt16; /* FIXME - query to see if 24 or 32 bit data can be handled. */ - - /* Get sample rates supported. */ - err = GetSoundOutputInfo(identifier, siSampleRateAvailable, (Ptr) &srinfo); - if(err != noErr) - { - ERR_RPT(("Error in PaMac_QueryOutputDeviceInfo: GetSoundOutputInfo siSampleRateAvailable returned %d\n", err )); - goto error; - } - numRates = srinfo.numRates; - DBUG(("PaMac_QueryOutputDeviceInfo: srinfo.numRates = %d\n", srinfo.numRates )); - if( numRates == 0 ) - { - dev->numSampleRates = -1; - numRates = 2; - } - else - { - dev->numSampleRates = numRates; - } - dev->sampleRates = PaMac_GetSampleRatesFromHandle( numRates, (Handle) srinfo.rates ); - if(dev->sampleRates == NULL) - { - DBUG(("PaMac_QueryOutputDeviceInfo: PaMac_GetSampleRatesFromHandle alloc failed.\n")); - return paInsufficientMemory; - } - - /* SPBGetDeviceInfo created the handle, but it's OUR job to release it. */ - DisposeHandle((Handle) srinfo.rates); - - /* Device name */ - /* we pass an existing handle for the component name; - we don't care about the info (type, subtype, etc.) or icon, so set them to nil */ - DBUG(("PaMac_QueryOutputDeviceInfo: get component name.\n")); - infoH = nil; - iconH = nil; - nameH = NewHandle(0); - if(nameH == nil) return paInsufficientMemory; - err = GetComponentInfo(identifier, &tempD, nameH, infoH, iconH); - if (err) - { - ERR_RPT(("Error in PaMac_QueryOutputDeviceInfo: GetComponentInfo returned %d\n", err )); - goto error; - } - /* Cast as uchar* so that we get a proper pascal string length. */ - len = ((unsigned char *)(*nameH))[0] + 1; /* PLB20020612 - fix allocation error on Mac 8600 */ - DBUG(("PaMac_QueryOutputDeviceInfo: new len = %d\n", len )); - - dev->name = (char *) malloc(len); /* MEM_010 */ - if( dev->name == NULL ) - { - DisposeHandle(nameH); - return paInsufficientMemory; - } - else - { - PToCString((unsigned char *)(*nameH), (char *) dev->name); - DisposeHandle(nameH); - } - - DBUG(("PaMac_QueryOutputDeviceInfo: dev->name = %s\n", dev->name )); - return paNoError; - -error: - sPaHostError = err; - return paHostError; - -} - -/************************************************************************* -** PaMac_QueryInputDeviceInfo() -** Query information about a named input device. -** Clears contents of ipad and writes info based on queries. -** Return one if OK, -** zero if device cannot be used, -** or negative error. -*/ -static PaError PaMac_QueryInputDeviceInfo( Str255 deviceName, internalPortAudioDevice *ipad ) -{ - PaError result = paNoError; - int len; - OSErr err; - long mRefNum = 0; - long tempL; - int16 tempS; - Fixed tempF; - PaDeviceInfo *dev = &ipad->pad_Info; - SRateInfo srinfo = {0}; - int numRates; - - memset( ipad, 0, sizeof(internalPortAudioDevice) ); - dev->maxOutputChannels = 0; - - /* Open device based on name. If device is in use, it may not be able to open in write mode. */ - err = SPBOpenDevice( deviceName, siWritePermission, &mRefNum); - if (err) - { - /* If device is in use, it may not be able to open in write mode so try read mode. */ - err = SPBOpenDevice( deviceName, siReadPermission, &mRefNum); - if (err) - { - ERR_RPT(("Error in PaMac_QueryInputDeviceInfo: SPBOpenDevice returned %d\n", err )); - sPaHostError = err; - return paHostError; - } - } - - /* Define macros for printing out device info. */ -#define PrintDeviceInfo(selector,var) \ - err = SPBGetDeviceInfo(mRefNum, selector, (Ptr) &var); \ - if (err) { \ - DBUG(("query %s failed\n", #selector )); \ - }\ - else { \ - DBUG(("query %s = 0x%x\n", #selector, var )); \ - } - - PrintDeviceInfo( siContinuous, tempS ); - PrintDeviceInfo( siAsync, tempS ); - PrintDeviceInfo( siNumberChannels, tempS ); - PrintDeviceInfo( siSampleSize, tempS ); - PrintDeviceInfo( siSampleRate, tempF ); - PrintDeviceInfo( siChannelAvailable, tempS ); - PrintDeviceInfo( siActiveChannels, tempL ); - PrintDeviceInfo( siDeviceBufferInfo, tempL ); - - err = SPBGetDeviceInfo(mRefNum, siActiveChannels, (Ptr) &tempL); - if (err == 0) DBUG(("%s = 0x%x\n", "siActiveChannels", tempL )); - /* Can we use this device? */ - err = SPBGetDeviceInfo(mRefNum, siAsync, (Ptr) &tempS); - if (err) - { - ERR_RPT(("Error in PaMac_QueryInputDeviceInfo: SPBGetDeviceInfo siAsync returned %d\n", err )); - goto error; - } - if( tempS == 0 ) goto useless; /* Does not support async recording so forget about it. */ - - err = SPBGetDeviceInfo(mRefNum, siChannelAvailable, (Ptr) &tempS); - if (err) - { - ERR_RPT(("Error in PaMac_QueryInputDeviceInfo: SPBGetDeviceInfo siChannelAvailable returned %d\n", err )); - goto error; - } - dev->maxInputChannels = tempS; - - /* Get sample rates supported. */ - err = SPBGetDeviceInfo(mRefNum, siSampleRateAvailable, (Ptr) &srinfo); - if (err) - { - ERR_RPT(("Error in PaMac_QueryInputDeviceInfo: SPBGetDeviceInfo siSampleRateAvailable returned %d\n", err )); - goto error; - } - - numRates = srinfo.numRates; - DBUG(("numRates = 0x%x\n", numRates )); - if( numRates == 0 ) - { - dev->numSampleRates = -1; - numRates = 2; - } - else - { - dev->numSampleRates = numRates; - } - dev->sampleRates = PaMac_GetSampleRatesFromHandle( numRates, (Handle) srinfo.rates ); - /* SPBGetDeviceInfo created the handle, but it's OUR job to release it. */ - DisposeHandle((Handle) srinfo.rates); - - /* Get size of device buffer. */ - err = SPBGetDeviceInfo(mRefNum, siDeviceBufferInfo, (Ptr) &tempL); - if (err) - { - ERR_RPT(("Error in PaMac_QueryInputDeviceInfo: SPBGetDeviceInfo siDeviceBufferInfo returned %d\n", err )); - goto error; - } - ipad->pad_DeviceBufferSize = tempL; - DBUG(("siDeviceBufferInfo = %d\n", tempL )); - - /* Set format based on sample size. */ - err = SPBGetDeviceInfo(mRefNum, siSampleSize, (Ptr) &tempS); - if (err) - { - ERR_RPT(("Error in PaMac_QueryInputDeviceInfo: SPBGetDeviceInfo siSampleSize returned %d\n", err )); - goto error; - } - switch( tempS ) - { - case 0x0020: - dev->nativeSampleFormats = paInt32; /* FIXME - warning, code probably won't support this! */ - break; - case 0x0010: - default: /* FIXME - What about other formats? */ - dev->nativeSampleFormats = paInt16; - break; - } - DBUG(("nativeSampleFormats = %d\n", dev->nativeSampleFormats )); - - /* Device name */ - len = deviceName[0] + 1; /* Get length of Pascal string */ - dev->name = (char *) malloc(len); /* MEM_010 */ - if( dev->name == NULL ) - { - result = paInsufficientMemory; - goto cleanup; - } - PToCString(deviceName, (char *) dev->name); - DBUG(("deviceName = %s\n", dev->name )); - result = (PaError) 1; - /* All done so close up device. */ -cleanup: - if( mRefNum ) SPBCloseDevice(mRefNum); - return result; - -error: - if( mRefNum ) SPBCloseDevice(mRefNum); - sPaHostError = err; - return paHostError; - -useless: - if( mRefNum ) SPBCloseDevice(mRefNum); - return (PaError) 0; -} - -/************************************************************************* -** Allocate a double array and fill it with listed sample rates. -*/ -static double * PaMac_GetSampleRatesFromHandle ( int numRates, Handle h ) -{ - OSErr err = noErr; - SInt8 hState; - int i; - UnsignedFixed *fixedRates; - double *rates = (double *) malloc( numRates * sizeof(double) ); /* MEM_011 */ - if( rates == NULL ) return NULL; - /* Save and restore handle state as suggested by TechNote at: - http://developer.apple.com/technotes/tn/tn1122.html - */ - hState = HGetState (h); - if (!(err = MemError ())) - { - HLock (h); - if (!(err = MemError ( ))) - { - fixedRates = (UInt32 *) *h; - for( i=0; i<numRates; i++ ) - { - rates[i] = UnsignedFixedToDouble(fixedRates[i]); - } - - HSetState (h,hState); - err = MemError ( ); - } - } - if( err ) - { - free( rates ); - ERR_RPT(("Error in PaMac_GetSampleRatesFromHandle = %d\n", err )); - } - return rates; -} - -/*************************************************************************/ -int Pa_CountDevices() -{ - PaError err; - DBUG(("Pa_CountDevices()\n")); - /* If no devices, go find some. */ - if( sNumDevices <= 0 ) - { - err = PaMac_ScanOutputDevices(); - if( err != paNoError ) goto error; - err = PaMac_ScanInputDevices(); - if( err != paNoError ) goto error; - } - return sNumDevices; - -error: - PaHost_Term(); - DBUG(("Pa_CountDevices: returns %d\n", err )); - return err; - -} - -/*************************************************************************/ -const PaDeviceInfo* Pa_GetDeviceInfo( PaDeviceID id ) -{ - if( (id < 0) || ( id >= Pa_CountDevices()) ) return NULL; - return &sDevices[id].pad_Info; -} -/*************************************************************************/ -PaDeviceID Pa_GetDefaultInputDeviceID( void ) -{ - return sDefaultInputDeviceID; -} - -/*************************************************************************/ -PaDeviceID Pa_GetDefaultOutputDeviceID( void ) -{ - return sDefaultOutputDeviceID; -} - -/********************************* BEGIN CPU UTILIZATION MEASUREMENT ****/ -static void PaMac_StartLoadCalculation( internalPortAudioStream *past ) -{ - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - UnsignedWide widePad; - if( pahsc == NULL ) return; - /* Query system timer for usage analysis and to prevent overuse of CPU. */ - Microseconds( &widePad ); - pahsc->pahsc_EntryCount = UnsignedWideToUInt64( widePad ); -} - -/****************************************************************************** -** Measure fractional CPU load based on real-time it took to calculate -** buffers worth of output. -*/ -/**************************************************************************/ -static void PaMac_EndLoadCalculation( internalPortAudioStream *past ) -{ - UnsignedWide widePad; - UInt64 currentCount; - long usecsElapsed; - double newUsage; - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - if( pahsc == NULL ) return; - - /* Measure CPU utilization during this callback. Note that this calculation - ** assumes that we had the processor the whole time. - */ -#define LOWPASS_COEFFICIENT_0 (0.95) -#define LOWPASS_COEFFICIENT_1 (0.99999 - LOWPASS_COEFFICIENT_0) - Microseconds( &widePad ); - currentCount = UnsignedWideToUInt64( widePad ); - - usecsElapsed = (long) U64Subtract(currentCount, pahsc->pahsc_EntryCount); - - /* Use inverse because it is faster than the divide. */ - newUsage = usecsElapsed * pahsc->pahsc_InverseMicrosPerHostBuffer; - - past->past_Usage = (LOWPASS_COEFFICIENT_0 * past->past_Usage) + - (LOWPASS_COEFFICIENT_1 * newUsage); - -} - -/*********************************************************************** -** Called by Pa_StartStream() -*/ -PaError PaHost_StartInput( internalPortAudioStream *past ) -{ - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - pahsc->pahsc_IsRecording = 0; - pahsc->pahsc_StopRecording = 0; - pahsc->pahsc_InputMultiBuffer.nextWrite = 0; - pahsc->pahsc_InputMultiBuffer.nextRead = 0; - return PaMac_RecordNext( past ); -} - -/*********************************************************************** -** Called by Pa_StopStream(). -** May be called during error recovery or cleanup code -** so protect against NULL pointers. -*/ -PaError PaHost_StopInput( internalPortAudioStream *past, int abort ) -{ - int32 timeOutMsec; - PaError result = paNoError; - OSErr err = 0; - long mRefNum; - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - if( pahsc == NULL ) return paNoError; - - (void) abort; - - mRefNum = pahsc->pahsc_InputRefNum; - - DBUG(("PaHost_StopInput: mRefNum = %d\n", mRefNum )); - if( mRefNum ) - { - DBUG(("PaHost_StopInput: pahsc_IsRecording = %d\n", pahsc->pahsc_IsRecording )); - if( pahsc->pahsc_IsRecording ) - { - /* PLB20010420 - Fix TIMEOUT in record mode. */ - pahsc->pahsc_StopRecording = 1; /* Request that we stop recording. */ - err = SPBStopRecording(mRefNum); - DBUG(("PaHost_StopInput: then pahsc_IsRecording = %d\n", pahsc->pahsc_IsRecording )); - - /* Calculate timeOut longer than longest time it could take to play one buffer. */ - timeOutMsec = (int32) ((1500.0 * pahsc->pahsc_FramesPerHostBuffer) / past->past_SampleRate); - /* Keep querying sound channel until it is no longer busy playing. */ - while( !err && pahsc->pahsc_IsRecording && (timeOutMsec > 0)) - { - Pa_Sleep(20); - timeOutMsec -= 20; - } - if( timeOutMsec <= 0 ) - { - ERR_RPT(("PaHost_StopInput: timed out!\n")); - return paTimedOut; - } - } - } - if( err ) - { - sPaHostError = err; - result = paHostError; - } - - DBUG(("PaHost_StopInput: finished.\n", mRefNum )); - return result; -} - -/***********************************************************************/ -static void PaMac_InitSoundHeader( internalPortAudioStream *past, CmpSoundHeader *sndHeader ) -{ - sndHeader->numChannels = past->past_NumOutputChannels; - sndHeader->sampleRate = DoubleToUnsignedFixed(past->past_SampleRate); - sndHeader->loopStart = 0; - sndHeader->loopEnd = 0; - sndHeader->encode = cmpSH; - sndHeader->baseFrequency = kMiddleC; - sndHeader->markerChunk = nil; - sndHeader->futureUse2 = nil; - sndHeader->stateVars = nil; - sndHeader->leftOverSamples = nil; - sndHeader->compressionID = 0; - sndHeader->packetSize = 0; - sndHeader->snthID = 0; - sndHeader->sampleSize = 8 * sizeof(int16); // FIXME - might be 24 or 32 bits some day; - sndHeader->sampleArea[0] = 0; - sndHeader->format = kSoundNotCompressed; -} - -static void SetFramesDone( PaHostSoundControl *pahsc, PaTimestamp framesDone ) -{ - UnsignedWide now; - Microseconds( &now ); - pahsc->pahsc_NumFramesDone = framesDone; - pahsc->pahsc_WhenFramesDoneIncremented = UnsignedWideToUInt64( now ); -} - -/***********************************************************************/ -PaError PaHost_StartOutput( internalPortAudioStream *past ) -{ - SndCommand pauseCommand; - SndCommand resumeCommand; - int i; - OSErr error; - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - if( pahsc == NULL ) return paInternalError; - if( pahsc->pahsc_Channel == NULL ) return paInternalError; - - past->past_StopSoon = 0; - past->past_IsActive = 1; - pahsc->pahsc_NumOutsQueued = 0; - pahsc->pahsc_NumOutsPlayed = 0; - - SetFramesDone( pahsc, 0.0 ); - - /* Pause channel so it does not do back ground processing while we are still filling the queue. */ - pauseCommand.cmd = pauseCmd; - pauseCommand.param1 = pauseCommand.param2 = 0; - error = SndDoCommand (pahsc->pahsc_Channel, &pauseCommand, true); - if (noErr != error) goto exit; - - /* Queue all of the buffers so we start off full. */ - for (i = 0; i<pahsc->pahsc_NumHostBuffers; i++) - { - PaMac_PlayNext( past, i ); - } - - /* Resume channel now that the queue is full. */ - resumeCommand.cmd = resumeCmd; - resumeCommand.param1 = resumeCommand.param2 = 0; - error = SndDoImmediate( pahsc->pahsc_Channel, &resumeCommand ); - if (noErr != error) goto exit; - - return paNoError; -exit: - past->past_IsActive = 0; - sPaHostError = error; - ERR_RPT(("Error in PaHost_StartOutput: SndDoCommand returned %d\n", error )); - return paHostError; -} - -/*******************************************************************/ -long PaHost_GetTotalBufferFrames( internalPortAudioStream *past ) -{ - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - return (long) (pahsc->pahsc_NumHostBuffers * pahsc->pahsc_FramesPerHostBuffer); -} - -/*********************************************************************** -** Called by Pa_StopStream(). -** May be called during error recovery or cleanup code -** so protect against NULL pointers. -*/ -PaError PaHost_StopOutput( internalPortAudioStream *past, int abort ) -{ - int32 timeOutMsec; - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - if( pahsc == NULL ) return paNoError; - if( pahsc->pahsc_Channel == NULL ) return paNoError; - - DBUG(("PaHost_StopOutput()\n")); - if( past->past_IsActive == 0 ) return paNoError; - - /* Set flags for callback function to see. */ - if( abort ) past->past_StopNow = 1; - past->past_StopSoon = 1; - /* Calculate timeOut longer than longest time it could take to play all buffers. */ - timeOutMsec = (int32) ((1500.0 * PaHost_GetTotalBufferFrames( past )) / past->past_SampleRate); - /* Keep querying sound channel until it is no longer busy playing. */ - while( past->past_IsActive && (timeOutMsec > 0)) - { - Pa_Sleep(20); - timeOutMsec -= 20; - } - if( timeOutMsec <= 0 ) - { - ERR_RPT(("PaHost_StopOutput: timed out!\n")); - return paTimedOut; - } - else return paNoError; -} - -/***********************************************************************/ -PaError PaHost_StartEngine( internalPortAudioStream *past ) -{ - (void) past; /* Prevent unused variable warnings. */ - return paNoError; -} - -/***********************************************************************/ -PaError PaHost_StopEngine( internalPortAudioStream *past, int abort ) -{ - (void) past; /* Prevent unused variable warnings. */ - (void) abort; /* Prevent unused variable warnings. */ - return paNoError; -} -/***********************************************************************/ -PaError PaHost_StreamActive( internalPortAudioStream *past ) -{ - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - return (PaError) ( past->past_IsActive + pahsc->pahsc_IsRecording ); -} -int Mac_IsVirtualMemoryOn( void ) -{ - long attr; - OSErr result = Gestalt( gestaltVMAttr, &attr ); - DBUG(("gestaltVMAttr : 0x%x\n", attr )); - return ((attr >> gestaltVMHasPagingControl ) & 1); -} - -/******************************************************************* -* Determine number of host Buffers -* and how many User Buffers we can put into each host buffer. -*/ -static void PaHost_CalcNumHostBuffers( internalPortAudioStream *past ) -{ - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - int32 minNumBuffers; - int32 minFramesPerHostBuffer; - int32 minTotalFrames; - int32 userBuffersPerHostBuffer; - int32 framesPerHostBuffer; - int32 numHostBuffers; - - minFramesPerHostBuffer = pahsc->pahsc_MinFramesPerHostBuffer; - minFramesPerHostBuffer = (minFramesPerHostBuffer + 7) & ~7; - DBUG(("PaHost_CalcNumHostBuffers: minFramesPerHostBuffer = %d\n", minFramesPerHostBuffer )); - - /* Determine number of user buffers based on minimum latency. */ - /* PLB20020417 I used to call Pa_GetMinNumBuffers() which doesn't take into account the - ** variable minFramesPerHostBuffer. Now I call PaMac_GetMinNumBuffers() which will - ** gove lower latency when virtual memory is turned off. */ - /* minNumBuffers = Pa_GetMinNumBuffers( past->past_FramesPerUserBuffer, past->past_SampleRate ); WRONG */ - minNumBuffers = PaMac_GetMinNumBuffers( minFramesPerHostBuffer, past->past_FramesPerUserBuffer, past->past_SampleRate ); - - past->past_NumUserBuffers = ( minNumBuffers > past->past_NumUserBuffers ) ? minNumBuffers : past->past_NumUserBuffers; - DBUG(("PaHost_CalcNumHostBuffers: min past_NumUserBuffers = %d\n", past->past_NumUserBuffers )); - minTotalFrames = past->past_NumUserBuffers * past->past_FramesPerUserBuffer; - - /* We cannot make the buffers too small because they may not get serviced quickly enough. */ - if( (int32) past->past_FramesPerUserBuffer < minFramesPerHostBuffer ) - { - userBuffersPerHostBuffer = - (minFramesPerHostBuffer + past->past_FramesPerUserBuffer - 1) / - past->past_FramesPerUserBuffer; - } - else - { - userBuffersPerHostBuffer = 1; - } - framesPerHostBuffer = past->past_FramesPerUserBuffer * userBuffersPerHostBuffer; - - /* Calculate number of host buffers needed. Round up to cover minTotalFrames. */ - numHostBuffers = (minTotalFrames + framesPerHostBuffer - 1) / framesPerHostBuffer; - /* Make sure we have enough host buffers. */ - if( numHostBuffers < PA_MIN_NUM_HOST_BUFFERS) - { - numHostBuffers = PA_MIN_NUM_HOST_BUFFERS; - } - else - { - /* If we have too many host buffers, try to put more user buffers in a host buffer. */ - while(numHostBuffers > PA_MAX_NUM_HOST_BUFFERS) - { - userBuffersPerHostBuffer += 1; - framesPerHostBuffer = past->past_FramesPerUserBuffer * userBuffersPerHostBuffer; - numHostBuffers = (minTotalFrames + framesPerHostBuffer - 1) / framesPerHostBuffer; - } - } - - pahsc->pahsc_UserBuffersPerHostBuffer = userBuffersPerHostBuffer; - pahsc->pahsc_FramesPerHostBuffer = framesPerHostBuffer; - pahsc->pahsc_NumHostBuffers = numHostBuffers; - DBUG(("PaHost_CalcNumHostBuffers: pahsc_UserBuffersPerHostBuffer = %d\n", pahsc->pahsc_UserBuffersPerHostBuffer )); - DBUG(("PaHost_CalcNumHostBuffers: pahsc_NumHostBuffers = %d\n", pahsc->pahsc_NumHostBuffers )); - DBUG(("PaHost_CalcNumHostBuffers: pahsc_FramesPerHostBuffer = %d\n", pahsc->pahsc_FramesPerHostBuffer )); - DBUG(("PaHost_CalcNumHostBuffers: past_NumUserBuffers = %d\n", past->past_NumUserBuffers )); -} - -/*******************************************************************/ -PaError PaHost_OpenStream( internalPortAudioStream *past ) -{ - OSErr err; - PaError result = paHostError; - PaHostSoundControl *pahsc; - int i; - /* Allocate and initialize host data. */ - pahsc = (PaHostSoundControl *) PaHost_AllocateFastMemory(sizeof(PaHostSoundControl)); - if( pahsc == NULL ) - { - return paInsufficientMemory; - } - past->past_DeviceData = (void *) pahsc; - - /* If recording, and virtual memory is turned on, then use bigger buffers to prevent glitches. */ - if( (past->past_NumInputChannels > 0) && Mac_IsVirtualMemoryOn() ) - { - pahsc->pahsc_MinFramesPerHostBuffer = MAC_VIRTUAL_FRAMES_PER_BUFFER; - } - else - { - pahsc->pahsc_MinFramesPerHostBuffer = MAC_PHYSICAL_FRAMES_PER_BUFFER; - } - - PaHost_CalcNumHostBuffers( past ); - - /* Setup constants for CPU load measurement. */ - pahsc->pahsc_InverseMicrosPerHostBuffer = past->past_SampleRate / (1000000.0 * pahsc->pahsc_FramesPerHostBuffer); - - /* ------------------ OUTPUT */ - if( past->past_NumOutputChannels > 0 ) - { - /* Create sound channel to which we can send commands. */ - pahsc->pahsc_Channel = 0L; - err = SndNewChannel(&pahsc->pahsc_Channel, sampledSynth, 0, nil); /* FIXME - use kUseOptionalOutputDevice if not default. */ - if(err != 0) - { - ERR_RPT(("Error in PaHost_OpenStream: SndNewChannel returned 0x%x\n", err )); - goto error; - } - - /* Install our callback function pointer straight into the sound channel structure */ - /* Use new CARBON name for callback procedure. */ -#if TARGET_API_MAC_CARBON - pahsc->pahsc_OutputCompletionProc = NewSndCallBackUPP(PaMac_OutputCompletionProc); -#else - pahsc->pahsc_OutputCompletionProc = NewSndCallBackProc(PaMac_OutputCompletionProc); -#endif - - pahsc->pahsc_Channel->callBack = pahsc->pahsc_OutputCompletionProc; - - pahsc->pahsc_BytesPerOutputHostBuffer = pahsc->pahsc_FramesPerHostBuffer * past->past_NumOutputChannels * sizeof(int16); - for (i = 0; i<pahsc->pahsc_NumHostBuffers; i++) - { - char *buf = (char *)PaHost_AllocateFastMemory(pahsc->pahsc_BytesPerOutputHostBuffer); - if (buf == NULL) - { - ERR_RPT(("Error in PaHost_OpenStream: could not allocate output buffer. Size = \n", pahsc->pahsc_BytesPerOutputHostBuffer )); - goto memerror; - } - - PaMac_InitSoundHeader( past, &pahsc->pahsc_SoundHeaders[i] ); - pahsc->pahsc_SoundHeaders[i].samplePtr = buf; - pahsc->pahsc_SoundHeaders[i].numFrames = (unsigned long) pahsc->pahsc_FramesPerHostBuffer; - - } - } -#ifdef SUPPORT_AUDIO_CAPTURE - /* ------------------ INPUT */ - /* Use double buffer scheme that matches output. */ - if( past->past_NumInputChannels > 0 ) - { - int16 tempS; - long tempL; - Fixed tempF; - long mRefNum; - Str255 namePString; -#if TARGET_API_MAC_CARBON - pahsc->pahsc_InputCompletionProc = NewSICompletionUPP((SICompletionProcPtr)PaMac_InputCompletionProc); -#else - pahsc->pahsc_InputCompletionProc = NewSICompletionProc((ProcPtr)PaMac_InputCompletionProc); -#endif - pahsc->pahsc_BytesPerInputHostBuffer = pahsc->pahsc_FramesPerHostBuffer * past->past_NumInputChannels * sizeof(int16); - for (i = 0; i<pahsc->pahsc_NumHostBuffers; i++) - { - char *buf = (char *) PaHost_AllocateFastMemory(pahsc->pahsc_BytesPerInputHostBuffer); - if ( buf == NULL ) - { - ERR_RPT(("PaHost_OpenStream: could not allocate input buffer. Size = \n", pahsc->pahsc_BytesPerInputHostBuffer )); - goto memerror; - } - pahsc->pahsc_InputMultiBuffer.buffers[i] = buf; - } - pahsc->pahsc_InputMultiBuffer.numBuffers = pahsc->pahsc_NumHostBuffers; - - // err = SPBOpenDevice( (const unsigned char *) &noname, siWritePermission, &mRefNum); - CToPString((char *)sDevices[past->past_InputDeviceID].pad_Info.name, namePString); - err = SPBOpenDevice(namePString, siWritePermission, &mRefNum); - - if (err) goto error; - pahsc->pahsc_InputRefNum = mRefNum; - DBUG(("PaHost_OpenStream: mRefNum = %d\n", mRefNum )); - - /* Set input device characteristics. */ - tempS = 1; - err = SPBSetDeviceInfo(mRefNum, siContinuous, (Ptr) &tempS); - if (err) - { - ERR_RPT(("Error in PaHost_OpenStream: SPBSetDeviceInfo siContinuous returned %d\n", err )); - goto error; - } - - tempL = 0x03; - err = SPBSetDeviceInfo(mRefNum, siActiveChannels, (Ptr) &tempL); - if (err) - { - DBUG(("PaHost_OpenStream: setting siActiveChannels returned 0x%x. Error ignored.\n", err )); - } - - /* PLB20010908 - Use requested number of input channels. Thanks Dominic Mazzoni. */ - tempS = past->past_NumInputChannels; - err = SPBSetDeviceInfo(mRefNum, siNumberChannels, (Ptr) &tempS); - if (err) - { - ERR_RPT(("Error in PaHost_OpenStream: SPBSetDeviceInfo siNumberChannels returned %d\n", err )); - goto error; - } - - tempF = ((unsigned long)past->past_SampleRate) << 16; - err = SPBSetDeviceInfo(mRefNum, siSampleRate, (Ptr) &tempF); - if (err) - { - ERR_RPT(("Error in PaHost_OpenStream: SPBSetDeviceInfo siSampleRate returned %d\n", err )); - goto error; - } - - /* Setup record-parameter block */ - pahsc->pahsc_InputParams.inRefNum = mRefNum; - pahsc->pahsc_InputParams.milliseconds = 0; // not used - pahsc->pahsc_InputParams.completionRoutine = pahsc->pahsc_InputCompletionProc; - pahsc->pahsc_InputParams.interruptRoutine = 0; - pahsc->pahsc_InputParams.userLong = (long) past; - pahsc->pahsc_InputParams.unused1 = 0; - } -#endif /* SUPPORT_AUDIO_CAPTURE */ - DBUG(("PaHost_OpenStream: complete.\n")); - return paNoError; - -error: - PaHost_CloseStream( past ); - ERR_RPT(("PaHost_OpenStream: sPaHostError = 0x%x.\n", err )); - sPaHostError = err; - return paHostError; - -memerror: - PaHost_CloseStream( past ); - return paInsufficientMemory; -} - -/*********************************************************************** -** Called by Pa_CloseStream(). -** May be called during error recovery or cleanup code -** so protect against NULL pointers. -*/ -PaError PaHost_CloseStream( internalPortAudioStream *past ) -{ - PaError result = paNoError; - OSErr err = 0; - int i; - PaHostSoundControl *pahsc; - - DBUG(("PaHost_CloseStream( 0x%x )\n", past )); - - if( past == NULL ) return paBadStreamPtr; - - pahsc = (PaHostSoundControl *) past->past_DeviceData; - if( pahsc == NULL ) return paNoError; - - if( past->past_NumOutputChannels > 0 ) - { - /* TRUE means flush now instead of waiting for quietCmd to be processed. */ - if( pahsc->pahsc_Channel != NULL ) SndDisposeChannel(pahsc->pahsc_Channel, TRUE); - { - for (i = 0; i<pahsc->pahsc_NumHostBuffers; i++) - { - Ptr p = (Ptr) pahsc->pahsc_SoundHeaders[i].samplePtr; - if( p != NULL ) PaHost_FreeFastMemory( p, pahsc->pahsc_BytesPerOutputHostBuffer ); - } - } - } - - if( past->past_NumInputChannels > 0 ) - { - if( pahsc->pahsc_InputRefNum ) - { - err = SPBCloseDevice(pahsc->pahsc_InputRefNum); - pahsc->pahsc_InputRefNum = 0; - if( err ) - { - sPaHostError = err; - result = paHostError; - } - } - { - for (i = 0; i<pahsc->pahsc_InputMultiBuffer.numBuffers; i++) - { - Ptr p = (Ptr) pahsc->pahsc_InputMultiBuffer.buffers[i]; - if( p != NULL ) PaHost_FreeFastMemory( p, pahsc->pahsc_BytesPerInputHostBuffer ); - } - } - } - - past->past_DeviceData = NULL; - PaHost_FreeFastMemory( pahsc, sizeof(PaHostSoundControl) ); - - DBUG(("PaHost_CloseStream: complete.\n", past )); - return result; -} -/*************************************************************************/ -int Pa_GetMinNumBuffers( int framesPerUserBuffer, double sampleRate ) -{ -/* We use the MAC_VIRTUAL_FRAMES_PER_BUFFER because we might be recording. -** This routine doesn't have enough information to determine the best value -** and is being depracated. */ - return PaMac_GetMinNumBuffers( MAC_VIRTUAL_FRAMES_PER_BUFFER, framesPerUserBuffer, sampleRate ); -} -/*************************************************************************/ -static int PaMac_GetMinNumBuffers( int minFramesPerHostBuffer, int framesPerUserBuffer, double sampleRate ) -{ - int minUserPerHost = ( minFramesPerHostBuffer + framesPerUserBuffer - 1) / framesPerUserBuffer; - int numBufs = PA_MIN_NUM_HOST_BUFFERS * minUserPerHost; - if( numBufs < PA_MIN_NUM_HOST_BUFFERS ) numBufs = PA_MIN_NUM_HOST_BUFFERS; - (void) sampleRate; - return numBufs; -} - -/*************************************************************************/ -void Pa_Sleep( int32 msec ) -{ - EventRecord event; - int32 sleepTime, endTime; - /* Convert to ticks. Round up so we sleep a MINIMUM of msec time. */ - sleepTime = ((msec * 60) + 999) / 1000; - if( sleepTime < 1 ) sleepTime = 1; - endTime = TickCount() + sleepTime; - do - { - DBUGX(("Sleep for %d ticks.\n", sleepTime )); - /* Use WaitNextEvent() to sleep without getting events. */ - /* PLB20010907 - Pass unused event to WaitNextEvent instead of NULL to prevent - * Mac OSX crash. Thanks Dominic Mazzoni. */ - WaitNextEvent( 0, &event, sleepTime, NULL ); - sleepTime = endTime - TickCount(); - } - while( sleepTime > 0 ); -} -/*************************************************************************/ -int32 Pa_GetHostError( void ) -{ - int32 err = sPaHostError; - sPaHostError = 0; - return err; -} - -/************************************************************************* - * Allocate memory that can be accessed in real-time. - * This may need to be held in physical memory so that it is not - * paged to virtual memory. - * This call MUST be balanced with a call to PaHost_FreeFastMemory(). - */ -void *PaHost_AllocateFastMemory( long numBytes ) -{ - void *addr = NewPtrClear( numBytes ); - if( (addr == NULL) || (MemError () != 0) ) return NULL; - -#if (TARGET_API_MAC_CARBON == 0) - if( HoldMemory( addr, numBytes ) != noErr ) - { - DisposePtr( (Ptr) addr ); - return NULL; - } -#endif - return addr; -} - -/************************************************************************* - * Free memory that could be accessed in real-time. - * This call MUST be balanced with a call to PaHost_AllocateFastMemory(). - */ -void PaHost_FreeFastMemory( void *addr, long numBytes ) -{ - if( addr == NULL ) return; -#if TARGET_API_MAC_CARBON - (void) numBytes; -#else - UnholdMemory( addr, numBytes ); -#endif - DisposePtr( (Ptr) addr ); -} - -/*************************************************************************/ -PaTimestamp Pa_StreamTime( PortAudioStream *stream ) -{ - PaTimestamp framesDone1; - PaTimestamp framesDone2; - UInt64 whenIncremented; - UnsignedWide now; - UInt64 now64; - long microsElapsed; - long framesElapsed; - - PaHostSoundControl *pahsc; - internalPortAudioStream *past = (internalPortAudioStream *) stream; - if( past == NULL ) return paBadStreamPtr; - pahsc = (PaHostSoundControl *) past->past_DeviceData; - -/* Capture information from audio thread. - * We have to be careful that we don't get interrupted in the middle. - * So we grab the pahsc_NumFramesDone twice and make sure it didn't change. - */ - do - { - framesDone1 = pahsc->pahsc_NumFramesDone; - whenIncremented = pahsc->pahsc_WhenFramesDoneIncremented; - framesDone2 = pahsc->pahsc_NumFramesDone; - } while( framesDone1 != framesDone2 ); - - /* Calculate how many microseconds have elapsed and convert to frames. */ - Microseconds( &now ); - now64 = UnsignedWideToUInt64( now ); - microsElapsed = U64Subtract( now64, whenIncremented ); - framesElapsed = microsElapsed * past->past_SampleRate * 0.000001; - - return framesDone1 + framesElapsed; -} - -/************************************************************************** -** Callback for Input, SPBRecord() -*/ -int gRecordCounter = 0; -int gPlayCounter = 0; -pascal void PaMac_InputCompletionProc(SPBPtr recParams) -{ - PaError result = paNoError; - int finished = 1; - internalPortAudioStream *past; - PaHostSoundControl *pahsc; - - gRecordCounter += 1; /* debug hack to see if engine running */ - - /* Get our PA data from Mac structure. */ - past = (internalPortAudioStream *) recParams->userLong; - if( past == NULL ) return; - - if( past->past_Magic != PA_MAGIC ) - { - AddTraceMessage("PaMac_InputCompletionProc: bad MAGIC, past", (long) past ); - AddTraceMessage("PaMac_InputCompletionProc: bad MAGIC, magic", (long) past->past_Magic ); - goto error; - } - pahsc = (PaHostSoundControl *) past->past_DeviceData; - past->past_NumCallbacks += 1; - - /* Have we been asked to stop recording? */ - if( (recParams->error == abortErr) || pahsc->pahsc_StopRecording ) goto error; - - /* If there are no output channels, then we need to call the user callback function from here. - * Otherwise we will call the user code during the output completion routine. - */ - if(past->past_NumOutputChannels == 0) - { - SetFramesDone( pahsc, - pahsc->pahsc_NumFramesDone + pahsc->pahsc_FramesPerHostBuffer ); - result = PaMac_CallUserLoop( past, NULL ); - } - - /* Did user code ask us to stop? If not, issue another recording request. */ - if( (result == paNoError) && (pahsc->pahsc_StopRecording == 0) ) - { - result = PaMac_RecordNext( past ); - if( result != paNoError ) pahsc->pahsc_IsRecording = 0; - } - else goto error; - - return; - -error: - pahsc->pahsc_IsRecording = 0; - pahsc->pahsc_StopRecording = 0; - return; -} - -/*********************************************************************** -** Called by either input or output completion proc. -** Grabs input data if any present, and calls PA conversion code, -** that in turn calls user code. -*/ -static PaError PaMac_CallUserLoop( internalPortAudioStream *past, int16 *outPtr ) -{ - PaError result = paNoError; - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - int16 *inPtr = NULL; - int i; - - - /* Advance read index for sound input FIFO here, independantly of record/write process. */ - if(past->past_NumInputChannels > 0) - { - if( MultiBuffer_IsReadable( &pahsc->pahsc_InputMultiBuffer ) ) - { - inPtr = (int16 *) MultiBuffer_GetNextReadBuffer( &pahsc->pahsc_InputMultiBuffer ); - MultiBuffer_AdvanceReadIndex( &pahsc->pahsc_InputMultiBuffer ); - } - } - - /* Call user code enough times to fill buffer. */ - if( (inPtr != NULL) || (outPtr != NULL) ) - { - PaMac_StartLoadCalculation( past ); /* CPU usage */ - -#ifdef PA_MAX_USAGE_ALLOWED - /* If CPU usage exceeds limit, skip user callback to prevent hanging CPU. */ - if( past->past_Usage > PA_MAX_USAGE_ALLOWED ) - { - past->past_FrameCount += (PaTimestamp) pahsc->pahsc_FramesPerHostBuffer; - } - else -#endif - { - - for( i=0; i<pahsc->pahsc_UserBuffersPerHostBuffer; i++ ) - { - result = (PaError) Pa_CallConvertInt16( past, inPtr, outPtr ); - if( result != 0) - { - /* Recording might be in another process, so tell it to stop with a flag. */ - pahsc->pahsc_StopRecording = pahsc->pahsc_IsRecording; - break; - } - /* Advance sample pointers. */ - if(inPtr != NULL) inPtr += past->past_FramesPerUserBuffer * past->past_NumInputChannels; - if(outPtr != NULL) outPtr += past->past_FramesPerUserBuffer * past->past_NumOutputChannels; - } - } - - PaMac_EndLoadCalculation( past ); - } - return result; -} - -/*********************************************************************** -** Setup next recording buffer in FIFO and issue recording request to Snd Input Manager. -*/ -static PaError PaMac_RecordNext( internalPortAudioStream *past ) -{ - PaError result = paNoError; - OSErr err; - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - /* Get pointer to next buffer to record into. */ - pahsc->pahsc_InputParams.bufferPtr = MultiBuffer_GetNextWriteBuffer( &pahsc->pahsc_InputMultiBuffer ); - - /* Advance write index if there is room. Otherwise keep writing same buffer. */ - if( MultiBuffer_IsWriteable( &pahsc->pahsc_InputMultiBuffer ) ) - { - MultiBuffer_AdvanceWriteIndex( &pahsc->pahsc_InputMultiBuffer ); - } - - AddTraceMessage("PaMac_RecordNext: bufferPtr", (long) pahsc->pahsc_InputParams.bufferPtr ); - AddTraceMessage("PaMac_RecordNext: nextWrite", pahsc->pahsc_InputMultiBuffer.nextWrite ); - - /* Setup parameters and issue an asynchronous recording request. */ - pahsc->pahsc_InputParams.bufferLength = pahsc->pahsc_BytesPerInputHostBuffer; - pahsc->pahsc_InputParams.count = pahsc->pahsc_BytesPerInputHostBuffer; - err = SPBRecord(&pahsc->pahsc_InputParams, true); - if( err ) - { - AddTraceMessage("PaMac_RecordNext: SPBRecord error ", err ); - sPaHostError = err; - result = paHostError; - } - else - { - pahsc->pahsc_IsRecording = 1; - } - return result; -} - -/************************************************************************** -** Callback for Output Playback() -** Return negative error, 0 to continue, 1 to stop. -*/ -long PaMac_FillNextOutputBuffer( internalPortAudioStream *past, int index ) -{ - PaHostSoundControl *pahsc; - long result = 0; - int finished = 1; - char *outPtr; - - gPlayCounter += 1; /* debug hack */ - - past->past_NumCallbacks += 1; - pahsc = (PaHostSoundControl *) past->past_DeviceData; - if( pahsc == NULL ) return -1; - /* Are we nested?! */ - if( pahsc->pahsc_IfInsideCallback ) return 0; - pahsc->pahsc_IfInsideCallback = 1; - /* Get pointer to buffer to fill. */ - outPtr = pahsc->pahsc_SoundHeaders[index].samplePtr; - /* Combine with any sound input, and call user callback. */ - result = PaMac_CallUserLoop( past, (int16 *) outPtr ); - - pahsc->pahsc_IfInsideCallback = 0; - return result; -} - -/************************************************************************************* -** Called by SoundManager when ready for another buffer. -*/ -static pascal void PaMac_OutputCompletionProc (SndChannelPtr theChannel, SndCommand * theCallBackCmd) -{ - internalPortAudioStream *past; - PaHostSoundControl *pahsc; - (void) theChannel; - (void) theCallBackCmd; - - /* Get our data from Mac structure. */ - past = (internalPortAudioStream *) theCallBackCmd->param2; - if( past == NULL ) return; - - pahsc = (PaHostSoundControl *) past->past_DeviceData; - pahsc->pahsc_NumOutsPlayed += 1; - - SetFramesDone( pahsc, - pahsc->pahsc_NumFramesDone + pahsc->pahsc_FramesPerHostBuffer ); - - PaMac_BackgroundManager( past, theCallBackCmd->param1 ); -} - -/*******************************************************************/ -static PaError PaMac_BackgroundManager( internalPortAudioStream *past, int index ) -{ - PaError result = paNoError; - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - /* Has someone asked us to abort by calling Pa_AbortStream()? */ - if( past->past_StopNow ) - { - SndCommand command; - /* Clear the queue of any pending commands. */ - command.cmd = flushCmd; - command.param1 = command.param2 = 0; - SndDoImmediate( pahsc->pahsc_Channel, &command ); - /* Then stop currently playing buffer, if any. */ - command.cmd = quietCmd; - SndDoImmediate( pahsc->pahsc_Channel, &command ); - past->past_IsActive = 0; - } - /* Has someone asked us to stop by calling Pa_StopStream() - * OR has a user callback returned '1' to indicate finished. - */ - else if( past->past_StopSoon ) - { - if( (pahsc->pahsc_NumOutsQueued - pahsc->pahsc_NumOutsPlayed) <= 0 ) - { - past->past_IsActive = 0; /* We're finally done. */ - } - } - else - { - PaMac_PlayNext( past, index ); - } - return result; -} - -/************************************************************************************* -** Fill next buffer with sound and queue it for playback. -*/ -static void PaMac_PlayNext ( internalPortAudioStream *past, int index ) -{ - OSErr error; - long result; - SndCommand playCmd; - SndCommand callbackCmd; - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - - /* If this was the last buffer, or abort requested, then just be done. */ - if ( past->past_StopSoon ) goto done; - - /* Load buffer with sound. */ - result = PaMac_FillNextOutputBuffer ( past, index ); - if( result > 0 ) past->past_StopSoon = 1; /* Stop generating audio but wait until buffers play. */ - else if( result < 0 ) goto done; - - /* Play the next buffer. */ - playCmd.cmd = bufferCmd; - playCmd.param1 = 0; - playCmd.param2 = (long) &pahsc->pahsc_SoundHeaders[ index ]; - error = SndDoCommand (pahsc->pahsc_Channel, &playCmd, true ); - if( error != noErr ) goto gotError; - - /* Ask for a callback when it is done. */ - callbackCmd.cmd = callBackCmd; - callbackCmd.param1 = index; - callbackCmd.param2 = (long)past; - error = SndDoCommand (pahsc->pahsc_Channel, &callbackCmd, true ); - if( error != noErr ) goto gotError; - pahsc->pahsc_NumOutsQueued += 1; - - return; - -gotError: - sPaHostError = error; -done: - return; -} diff --git a/pd/portaudio_v18/pa_mac_core/notes.txt b/pd/portaudio_v18/pa_mac_core/notes.txt deleted file mode 100644 index 0abdd3db..00000000 --- a/pd/portaudio_v18/pa_mac_core/notes.txt +++ /dev/null @@ -1,36 +0,0 @@ -Notes on Core Audio Implementation of PortAudio - -by Phil Burk and Darren Gibbs - -Document last updated October 18, 2002 - -WHAT WORKS - -Output with very low latency, <10 msec. -Half duplex input or output. -Full duplex -The paFLoat32, paInt16, paInt8, paUInt8 sample formats. -Pa_GetCPULoad() -Pa_StreamTime() - -KNOWN BUGS OR LIMITATIONS - -The iMic supports multiple sample rates. -But there is a bug when changing sample rates: - Run patest_record.c at rate A - it works. - Then run patest_record.c at rate B - it FAIL! - Then run patest_record.c again at rate B - it works! - - -DEVICE MAPPING - -CoreAudio devices can support both input and output. But the sample -rates supported may be different. So we have map one or two PortAudio -device to each CoreAudio device depending on whether it supports -input, output or both. - -When we query devices, we first get a list of CoreAudio devices. Then -we scan the list and add a PortAudio device for each CoreAudio device -that supports input. Then we make a scan for output devices. - -(minor change to test CVS) diff --git a/pd/portaudio_v18/pa_mac_core/pa_mac_core.c b/pd/portaudio_v18/pa_mac_core/pa_mac_core.c deleted file mode 100644 index 983d0e4f..00000000 --- a/pd/portaudio_v18/pa_mac_core/pa_mac_core.c +++ /dev/null @@ -1,2136 +0,0 @@ -/* - * $Id: pa_mac_core.c,v 1.8.4.13 2003/12/09 19:04:54 philburk Exp $ - * pa_mac_core.c - * Implementation of PortAudio for Mac OS X Core Audio - * - * PortAudio Portable Real-Time Audio Library - * Latest Version at: http://www.portaudio.com - * - * Authors: Ross Bencina and Phil Burk - * Copyright (c) 1999-2002 Ross Bencina and Phil Burk - * - * Theory of Operation - * - * This code uses the HAL (Hardware Access Layer) of the Apple CoreAudio library. - * This is the layer closes to the hardware. - * The HAL layer only supports the native HW supported sample rates. - * So if the chip only supports 44100 Hz, then the HAL only supports 44100. - * To provide other rates we use the handy Apple AudioConverter which provides - * sample rate conversion, mono-to-stereo conversion, and buffer size adaptation. - * - * There are four modes of operation: - * PA_MODE_OUTPUT_ONLY, - * PA_MODE_INPUT_ONLY, - * PA_MODE_IO_ONE_DEVICE, - * PA_MODE_IO_TWO_DEVICES - * - * The processing pipeline for PA_MODE_IO_ONE_DEVICE is in one thread: - * - * PaOSX_CoreAudioIOCallback() input buffers -> RingBuffer -> input.AudioConverter -> - * PortAudio callback -> output.AudioConverter -> PaOSX_CoreAudioIOCallback() output buffers - * - * For two separate devices, we have to use two separate callbacks. - * We pass data between them using a RingBuffer FIFO. - * The processing pipeline for PA_MODE_IO_TWO_DEVICES is split into two threads: - * - * PaOSX_CoreAudioInputCallback() input buffers -> RingBuffer - * - * RingBuffer -> input.AudioConverter -> - * PortAudio callback -> output.AudioConverter -> PaOSX_CoreAudioIOCallback() output buffers - * - * License - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - * CHANGE HISTORY: - - 3.29.2001 - Phil Burk - First pass... converted from Window MME code with help from Darren. - 3.30.2001 - Darren Gibbs - Added more support for dynamically querying device info. - 12.7.2001 - Gord Peters - Tweaks to compile on PA V17 and OS X 10.1 - 2.7.2002 - Darren and Phil - fixed isInput so GetProperty works better, - fixed device queries for numChannels and sampleRates, - one CoreAudio device now maps to separate input and output PaDevices, - audio input works if using same CoreAudio device (some HW devices make separate CoreAudio devices). - 2.22.2002 - Stephane Letz - Explicit cast needed for compilation with Code Warrior 7 - 3.19.2002 - Phil Burk - Added paInt16, paInt8, format using new "pa_common/pa_convert.c" file. - Return error if opened in mono mode cuz not supported. [Supported 10.12.2002] - Add support for Pa_GetCPULoad(); - Fixed timestamp in callback and Pa_StreamTime() (Thanks n++k for the advice!) - Check for invalid sample rates and return an error. - Check for getenv("PA_MIN_LATENCY_MSEC") to set latency externally. - Better error checking for invalid channel counts and invalid devices. - 3.29.2002 - Phil Burk - Fixed Pa_GetCPULoad() for small buffers. - 3.31.2002 - Phil Burk - Use getrusage() instead of gettimeofday() for CPU Load calculation. - 10.12.2002 - Phil Burk - Use AudioConverter to allow wide range of sample rates, and mono. - Use FIFO (from pablio/rinbuffer.h) so that we can pull data through converter. - Added PaOSX_FixVolumeScalar() to make iMic audible. - 10.17.2002 - Phil Burk - Support full duplex between two different devices. - Name internal functions PaOSX_* - Dumped useless PA_MIN_LATENCY_MSEC environment variable. - Use kAudioDevicePropertyStreamFormatMatch to determine max channels. - 02.03.2003 - Phil Burk - always use AudioConverters so that we can adapt when format changes. - Synchronize with device when format changes. - 02.13.2003 - Phil Burk - scan for maxChannels because FormatMatch won't tell us. - 03.05.2003 - Phil Burk and Dominic Mazzoni - interleave and deinterleave multiple - CoreAudio buffers. Needed for MOTU828 and some other N>2 channel devices. - See code related to "streamInterleavingBuffer". - 03.06.2003 - Phil Burk and Ryan Francesconi - fixed numChannels query for MOTU828. - Handle fact that MOTU828 gives you 8 channels even when you ask for 2! - 04.06.2003 - Phil Burk - Combine Dominic Mazzoni's technique of using Configuration to query maxChannels - with old technique of scanning for mormat. - Increase channel scan by 1 to handle mono USB microphones. - Do not merge or split channels in AudioConverter to handle 2+2 channels - of Quattro which has a format of 2 channels. - 04.07.2003 - Phil Burk - use AudioGetCurrentHostTime instead of getrusage() which can lock threads. - 04.10.2003 - Phil Burk - fixed pointer bug with input deinterleaving loop. - Detect and ignore NULL inputData and outputData in CodeAudio callback. - Overlap creation and deletion of AudioConverters to prevent thread death when device rate changes. - 04.16.2003 - Phil Burk - Fixed input channel scrambling when numChannels != 2^N. Caused by alignment - error when filling RingBuffer with 2^N zero bytes. - 04.26.2003 - Phil Burk - Removed code to turn up volume and unmute to prevent blown eardrums. - 12.08.2003 - Phil Burk - Move declaration of oldConverter to top of PAOSX_DevicePropertyListener() - 12.08.2003 - Phil Burk - Removed need for #include "pa_trace.h", just for debug - 12.09.2003 - Phil Burk - Only change sampleRate or numChannels if we need to improve over - current setting. -*/ - -#include <CoreServices/CoreServices.h> -#include <CoreAudio/CoreAudio.h> -#include <sys/time.h> -#include <sys/resource.h> -#include <AudioUnit/AudioUnit.h> -#include <AudioToolbox/DefaultAudioOutput.h> -#include <AudioToolbox/AudioConverter.h> -#include <CoreAudio/HostTime.h> - -#include "portaudio.h" -#include "pa_host.h" -#include "ringbuffer.h" - -/************************************************* Configuration ********/ -#define PA_ENABLE_LOAD_MEASUREMENT (1) - -/************************************************* Constants ********/ -#define SET_DEVICE_BUFFER_SIZE (1) - -/* To trace program, enable TRACE_REALTIME_EVENTS in pa_trace.h */ -#define PA_TRACE_RUN (0) -#define PA_TRACE_START_STOP (0) - -#define PA_MIN_LATENCY_MSEC (20) /* FIXME */ -#define MIN_TIMEOUT_MSEC (3000) - -#define PRINT(x) { printf x; fflush(stdout); } -#define PRINT_ERR( msg, err ) PRINT(( msg ": error = 0x%0lX = '%s'\n", (err), ErrorToString(err)) ) -#define DBUG(x) /* PRINT(x) */ -#define DBUGBACK(x) /* if( sMaxBackgroundErrorMessages-- > 0 ) PRINT(x) */ -#define DBUGX(x) - -// define value of isInput passed to CoreAudio routines -#define IS_INPUT (true) -#define IS_OUTPUT (false) - -typedef enum PaDeviceMode -{ - PA_MODE_OUTPUT_ONLY, - PA_MODE_INPUT_ONLY, - PA_MODE_IO_ONE_DEVICE, - PA_MODE_IO_TWO_DEVICES -} PaDeviceMode; - -#define PA_USING_OUTPUT (pahsc->mode != PA_MODE_INPUT_ONLY) -#define PA_USING_INPUT (pahsc->mode != PA_MODE_OUTPUT_ONLY) - -/************************************************************** - * Information needed by PortAudio specific to a CoreAudio device. - */ -typedef struct PaHostInOut_s -{ - AudioDeviceID audioDeviceID; /* CoreAudio specific ID */ - int bytesPerUserNativeBuffer; /* User buffer size in native host format. Depends on numChannels. */ - AudioConverterRef converter; - void *converterBuffer; - int numChannels; - /** Used for interleaving or de-interleaving multiple streams for devices like MOTU828. */ - int streamInterleavingBufferLen; /**< size in bytes */ - Float32 *streamInterleavingBuffer; -} PaHostInOut; - -/************************************************************** - * Structure for internal host specific stream data. - * This is allocated on a per stream basis. - */ -typedef struct PaHostSoundControl -{ - PaHostInOut input; - PaHostInOut output; - AudioDeviceID primaryDeviceID; - PaDeviceMode mode; - RingBuffer ringBuffer; - char *ringBufferData; - Boolean formatListenerCalled; - /* For measuring CPU utilization. */ - UInt64 entryTime; - double inverseHostTicksPerBuffer; /* 1/Ticks of real-time audio per user buffer. */ -} PaHostSoundControl; - -/************************************************************** - * Structure for internal extended device info query. - * There will be one or two PortAudio devices for each Core Audio device: - * one input and or one output. - */ -typedef struct PaHostDeviceInfo -{ - PaDeviceInfo paInfo; - AudioDeviceID audioDeviceID; -} -PaHostDeviceInfo; - -/************************************************* Shared Data ********/ -/* FIXME - put Mutex around this shared data. */ -static int sNumPaDevices = 0; /* Total number of PaDeviceInfos */ -static int sNumInputDevices = 0; /* Total number of input PaDeviceInfos */ -static int sNumOutputDevices = 0; -static int sNumCoreDevices = 0; -static AudioDeviceID *sCoreDeviceIDs; // Array of Core AudioDeviceIDs -static PaHostDeviceInfo *sDeviceInfos = NULL; -static int sDefaultInputDeviceID = paNoDevice; -static int sDefaultOutputDeviceID = paNoDevice; -static int sSavedHostError = 0; - -static const double supportedSampleRateRange[] = { 8000.0, 96000.0 }; /* FIXME - go to double HW rate. */ -static const char sMapperSuffixInput[] = " - Input"; -static const char sMapperSuffixOutput[] = " - Output"; - -/* Debug support. */ -//static int sMaxBackgroundErrorMessages = 100; -//static int sCoverageCounter = 1; // used to check code coverage during validation - -/* We index the input devices first, then the output devices. */ -#define LOWEST_INPUT_DEVID (0) -#define HIGHEST_INPUT_DEVID (sNumInputDevices - 1) -#define LOWEST_OUTPUT_DEVID (sNumInputDevices) -#define HIGHEST_OUTPUT_DEVID (sNumPaDevices - 1) - -/************************************************* Macros ********/ - -/************************************************* Prototypes **********/ - -static PaError PaOSX_QueryDevices( void ); -static int PaOSX_ScanDevices( Boolean isInput ); -static int PaOSX_QueryDeviceInfo( PaHostDeviceInfo *hostDeviceInfo, int coreDeviceIndex, Boolean isInput ); -static PaDeviceID PaOSX_QueryDefaultInputDevice( void ); -static PaDeviceID PaOSX_QueryDefaultOutputDevice( void ); -static void PaOSX_CalcHostBufferSize( internalPortAudioStream *past ); - -static OSStatus PAOSX_DevicePropertyListener (AudioDeviceID inDevice, - UInt32 inChannel, - Boolean isInput, - AudioDevicePropertyID inPropertyID, - void* inClientData); - -/**********************************************************************/ -/* OS X errors are 4 character ID that can be printed. - * Note that uses a static pad so result must be printed immediately. - */ -static OSStatus statusText[2] = { 0, 0 }; -static const char *ErrorToString( OSStatus err ) -{ - const char *str; - - switch (err) - { - case kAudioHardwareUnspecifiedError: - str = "kAudioHardwareUnspecifiedError"; - break; - case kAudioHardwareNotRunningError: - str = "kAudioHardwareNotRunningError"; - break; - case kAudioHardwareUnknownPropertyError: - str = "kAudioHardwareUnknownPropertyError"; - break; - case kAudioDeviceUnsupportedFormatError: - str = "kAudioDeviceUnsupportedFormatError"; - break; - case kAudioHardwareBadPropertySizeError: - str = "kAudioHardwareBadPropertySizeError"; - break; - case kAudioHardwareIllegalOperationError: - str = "kAudioHardwareIllegalOperationError"; - break; - default: - statusText[0] = err; - str = (const char *)statusText; - break; - } - - return str; -} - -/**********************************************************************/ -static unsigned long RoundUpToNextPowerOf2( unsigned long n ) -{ - long numBits = 0; - if( ((n-1) & n) == 0) return n; /* Already Power of two. */ - while( n > 0 ) - { - n= n>>1; - numBits++; - } - return (1<<numBits); -} - -/********************************* BEGIN CPU UTILIZATION MEASUREMENT ****/ -static void Pa_StartUsageCalculation( internalPortAudioStream *past ) -{ - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - if( pahsc == NULL ) return; - /* Query user CPU timer for usage analysis and to prevent overuse of CPU. */ - pahsc->entryTime = AudioGetCurrentHostTime(); -} - -/****************************************************************************** -** Measure fractional CPU load based on real-time it took to calculate -** buffers worth of output. -*/ -static void Pa_EndUsageCalculation( internalPortAudioStream *past ) -{ - UInt64 exitTime; - UInt64 ticksElapsed; - double newUsage; - -#define LOWPASS_COEFFICIENT_0 (0.95) -#define LOWPASS_COEFFICIENT_1 (0.99999 - LOWPASS_COEFFICIENT_0) - - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - if( pahsc == NULL ) return; - - exitTime = AudioGetCurrentHostTime(); - - ticksElapsed = exitTime - pahsc->entryTime; - - /* Use inverse because it is faster than the divide. */ - newUsage = ticksElapsed * pahsc->inverseHostTicksPerBuffer; - /* Low pass filter result. */ - past->past_Usage = (LOWPASS_COEFFICIENT_0 * past->past_Usage) + - (LOWPASS_COEFFICIENT_1 * newUsage); -} -/****************************************** END CPU UTILIZATION *******/ - -/************************************************************************/ -static PaDeviceID PaOSX_QueryDefaultInputDevice( void ) -{ - OSStatus err = noErr; - UInt32 count; - int i; - AudioDeviceID tempDeviceID = kAudioDeviceUnknown; - PaDeviceID defaultDeviceID = paNoDevice; - - // get the default output device for the HAL - // it is required to pass the size of the data to be returned - count = sizeof(tempDeviceID); - err = AudioHardwareGetProperty( kAudioHardwarePropertyDefaultInputDevice, &count, (void *) &tempDeviceID); - if (err != noErr) goto error; - - // scan input devices to see which one matches this device - defaultDeviceID = paNoDevice; - for( i=LOWEST_INPUT_DEVID; i<=HIGHEST_INPUT_DEVID; i++ ) - { - DBUG(("PaOSX_QueryDefaultInputDevice: i = %d, aDevId = %ld\n", i, sDeviceInfos[i].audioDeviceID )); - if( sDeviceInfos[i].audioDeviceID == tempDeviceID ) - { - defaultDeviceID = i; - break; - } - } -error: - return defaultDeviceID; -} - -/************************************************************************/ -static PaDeviceID PaOSX_QueryDefaultOutputDevice( void ) -{ - OSStatus err = noErr; - UInt32 count; - int i; - AudioDeviceID tempDeviceID = kAudioDeviceUnknown; - PaDeviceID defaultDeviceID = paNoDevice; - - // get the default output device for the HAL - // it is required to pass the size of the data to be returned - count = sizeof(tempDeviceID); - err = AudioHardwareGetProperty( kAudioHardwarePropertyDefaultOutputDevice, &count, (void *) &tempDeviceID); - if (err != noErr) goto error; - - // scan output devices to see which one matches this device - defaultDeviceID = paNoDevice; - for( i=LOWEST_OUTPUT_DEVID; i<=HIGHEST_OUTPUT_DEVID; i++ ) - { - DBUG(("PaOSX_QueryDefaultOutputDevice: i = %d, aDevId = %ld\n", i, sDeviceInfos[i].audioDeviceID )); - if( sDeviceInfos[i].audioDeviceID == tempDeviceID ) - { - defaultDeviceID = i; - break; - } - } -error: - return defaultDeviceID; -} - -/******************************************************************/ -static PaError PaOSX_QueryDevices( void ) -{ - OSStatus err = noErr; - UInt32 outSize; - Boolean outWritable; - int numBytes; - - // find out how many Core Audio devices there are, if any - outSize = sizeof(outWritable); - err = AudioHardwareGetPropertyInfo(kAudioHardwarePropertyDevices, &outSize, &outWritable); - if (err != noErr) - { - PRINT_ERR("Couldn't get info about list of audio devices", err); - sSavedHostError = err; - return paHostError; - } - - // calculate the number of device available - sNumCoreDevices = outSize / sizeof(AudioDeviceID); - - // Bail if there aren't any devices - if (sNumCoreDevices < 1) - { - PRINT(("No Devices Available")); - return paHostError; - } - - // make space for the devices we are about to get - sCoreDeviceIDs = (AudioDeviceID *)malloc(outSize); - - // get an array of AudioDeviceIDs - err = AudioHardwareGetProperty(kAudioHardwarePropertyDevices, &outSize, (void *)sCoreDeviceIDs); - if (err != noErr) - { - PRINT_ERR("Couldn't get list of audio device IDs", err); - sSavedHostError = err; - return paHostError; - } - - // Allocate structures to hold device info pointers. - // There will be a maximum of two Pa devices per Core Audio device, input and/or output. - numBytes = sNumCoreDevices * 2 * sizeof(PaHostDeviceInfo); - sDeviceInfos = (PaHostDeviceInfo *) PaHost_AllocateFastMemory( numBytes ); - if( sDeviceInfos == NULL ) return paInsufficientMemory; - - // Scan all the Core Audio devices to see which support input and allocate a - // PaHostDeviceInfo structure for each one. - DBUG(("PaOSX_QueryDevices: scan for input ======================\n")); - PaOSX_ScanDevices( IS_INPUT ); - sNumInputDevices = sNumPaDevices; - // Now scan all the output devices. - DBUG(("PaOSX_QueryDevices: scan for output ======================\n")); - PaOSX_ScanDevices( IS_OUTPUT ); - sNumOutputDevices = sNumPaDevices - sNumInputDevices; - - // Figure out which of the devices that we scanned is the default device. - sDefaultInputDeviceID = PaOSX_QueryDefaultInputDevice(); - sDefaultOutputDeviceID = PaOSX_QueryDefaultOutputDevice(); - - return paNoError; -} - - -/*************************************************************************/ -/* Query a device for its sample rate. - * @return positive rate or 0.0 on error. - */ -static Float64 PaOSX_GetDeviceSampleRate( AudioDeviceID deviceID, Boolean isInput ) -{ - OSStatus err = noErr; - AudioStreamBasicDescription formatDesc; - UInt32 dataSize; - dataSize = sizeof(formatDesc); - err = AudioDeviceGetProperty( deviceID, 0, isInput, - kAudioDevicePropertyStreamFormat, &dataSize, &formatDesc); - if( err != noErr ) return 0.0; - else return formatDesc.mSampleRate; -} - -/*************************************************************************/ -/* Allocate a string containing the device name. */ -static char *PaOSX_DeviceNameFromID(AudioDeviceID deviceID, Boolean isInput ) -{ - OSStatus err = noErr; - UInt32 outSize; - Boolean outWritable; - char *deviceName = nil; - - // query size of name - err = AudioDeviceGetPropertyInfo(deviceID, 0, isInput, kAudioDevicePropertyDeviceName, &outSize, &outWritable); - if (err == noErr) - { - deviceName = (char*)malloc( outSize + 1); - if( deviceName ) - { - err = AudioDeviceGetProperty(deviceID, 0, isInput, kAudioDevicePropertyDeviceName, &outSize, deviceName); - if (err != noErr) - PRINT_ERR("Couldn't get audio device name", err); - } - } - - return deviceName; -} - -/************************************************************************* -** Scan all of the Core Audio devices to see which support selected -** input or output mode. -** Changes sNumDevices, and fills in sDeviceInfos. -*/ -static int PaOSX_ScanDevices( Boolean isInput ) -{ - int coreDeviceIndex; - int result; - PaHostDeviceInfo *hostDeviceInfo; - int numAdded = 0; - - for( coreDeviceIndex=0; coreDeviceIndex<sNumCoreDevices; coreDeviceIndex++ ) - { - // try to fill in next PaHostDeviceInfo - hostDeviceInfo = &sDeviceInfos[sNumPaDevices]; - result = PaOSX_QueryDeviceInfo( hostDeviceInfo, coreDeviceIndex, isInput ); - DBUG(("PaOSX_ScanDevices: paDevId = %d, coreDevId = %d, result = %d\n", sNumPaDevices, coreDeviceIndex, result )); - if( result > 0 ) - { - sNumPaDevices += 1; // bump global counter if we got one - numAdded += 1; - } - else if( result < 0 ) return result; - } - return numAdded; -} - -/************************************************************************* -** Determine the maximum number of channels based on the configuration. -** @return maxChannels or negative error. -*/ -static int PaOSX_GetMaxChannels_Config( AudioDeviceID devID, Boolean isInput ) -{ - OSStatus err; - UInt32 outSize; - Boolean outWritable; - AudioBufferList *list; - int numChannels; - int i; - - // Determine maximum number of channels supported. - // dmazzoni: new method - - outSize = 0; - err = AudioDeviceGetPropertyInfo(devID, 0, isInput, - kAudioDevicePropertyStreamConfiguration, - &outSize, &outWritable); - if ( err != noErr ) - { - PRINT_ERR("PaOSX_GetMaxChannels_Config: Could not get stream configuration info", err); - sSavedHostError = err; - return paHostError; - } - - list = (AudioBufferList *)PaHost_AllocateFastMemory( outSize ); - err = AudioDeviceGetProperty(devID, 0, isInput, - kAudioDevicePropertyStreamConfiguration, - &outSize, list); - if ( err != noErr ) - { - PRINT_ERR("PaOSX_GetMaxChannels_Config: Could not get stream configuration", err); - sSavedHostError = err; - return paHostError; - } - - numChannels = 0; - for( i=0; i<list->mNumberBuffers; i++ ) - { - int bufChannels = list->mBuffers[i].mNumberChannels; - DBUG(("PaOSX_GetMaxChannels_Config: buffer %d has %d channels.\n", i, bufChannels )); - numChannels += bufChannels; - } - - PaHost_FreeFastMemory( list, outSize ); - - return numChannels; -} - -/************************************************************************* -** Determine the maximum number of channels a device will support based on scanning the format. -** @return maxChannels or negative error. -*/ -static int PaOSX_GetMaxChannels_Format( AudioDeviceID devID, Boolean isInput ) -{ - OSStatus err; - UInt32 outSize; - AudioStreamBasicDescription formatDesc; - int maxChannels; - int numChannels; - Boolean gotMax; - - // Scan to find highest matching format. - // Unfortunately some devices won't just return maxChannels for the match. - // For example, some 8 channel devices return 2 when given 256 as input. - gotMax = false; - maxChannels = 0; - numChannels = 0; - while( !gotMax ) - { - - memset( &formatDesc, 0, sizeof(formatDesc)); - numChannels = numChannels + 1; - DBUG(("PaOSX_GetMaxChannels: try numChannels = %d = %d + 1\n", - numChannels, numChannels )); - formatDesc.mChannelsPerFrame = numChannels; - outSize = sizeof(formatDesc); - - err = AudioDeviceGetProperty( devID, 0, - isInput, kAudioDevicePropertyStreamFormatMatch, &outSize, &formatDesc); - - DBUG(("PaOSX_GetMaxChannels: err 0x%0x, formatDesc.mChannelsPerFrame= %d\n", - err, formatDesc.mChannelsPerFrame )); - if( err != noErr ) - { - if (numChannels > (maxChannels + 4)) // Try several possibilities above current max - { - gotMax = true; - } - } - else - { - // This value worked so we have a new candidate for maxChannels. - if (formatDesc.mChannelsPerFrame > numChannels) - { - maxChannels = formatDesc.mChannelsPerFrame; - } - else if(formatDesc.mChannelsPerFrame < numChannels) - { - if (numChannels > (maxChannels + 4)) // Try several possibilities above current max - { - gotMax = true; - } - } - else - { - maxChannels = numChannels; - } - } - } - return maxChannels; -} - - - -/************************************************************************* -** Determine the maximum number of channels a device will support. -** It is not clear at this point which the better technique so -** we do both and use the biggest result. -** -** @return maxChannels or negative error. -*/ -static int PaOSX_GetMaxChannels( AudioDeviceID devID, Boolean isInput ) -{ - int maxChannelsFormat; - int maxChannelsConfig; - maxChannelsFormat = PaOSX_GetMaxChannels_Format( devID, isInput ); - maxChannelsConfig = PaOSX_GetMaxChannels_Config( devID, isInput ); - return (maxChannelsFormat > maxChannelsConfig) ? maxChannelsFormat : maxChannelsConfig; -} - -/************************************************************************* -** Try to fill in the device info for this device. -** Return 1 if a good device that PA can use. -** Return 0 if not appropriate -** or return negative error. -** -*/ -static int PaOSX_QueryDeviceInfo( PaHostDeviceInfo *hostDeviceInfo, int coreDeviceIndex, Boolean isInput ) -{ - OSStatus err; - UInt32 outSize; - AudioStreamBasicDescription formatDesc; - AudioDeviceID devID; - PaDeviceInfo *deviceInfo = &hostDeviceInfo->paInfo; - int maxChannels; - - deviceInfo->structVersion = 1; - deviceInfo->maxInputChannels = 0; - deviceInfo->maxOutputChannels = 0; - - deviceInfo->sampleRates = supportedSampleRateRange; // because we use sample rate converter to get continuous rates - deviceInfo->numSampleRates = -1; - - devID = sCoreDeviceIDs[ coreDeviceIndex ]; - hostDeviceInfo->audioDeviceID = devID; - DBUG(("PaOSX_QueryDeviceInfo: coreDeviceIndex = %d, devID = %d, isInput = %d\n", - coreDeviceIndex, (int) devID, isInput )); - - // Get data format info from the device. - outSize = sizeof(formatDesc); - err = AudioDeviceGetProperty(devID, 0, isInput, kAudioDevicePropertyStreamFormat, &outSize, &formatDesc); - // This just may not be an appropriate device for input or output so leave quietly. - if( (err != noErr) || (formatDesc.mChannelsPerFrame == 0) ) goto error; - - DBUG(("PaOSX_QueryDeviceInfo: mFormatID = 0x%x\n", (unsigned int) formatDesc.mFormatID)); - DBUG(("PaOSX_QueryDeviceInfo: mFormatFlags = 0x%x\n",(unsigned int) formatDesc.mFormatFlags)); - - // Right now the Core Audio headers only define one formatID: LinearPCM - // Apparently LinearPCM must be Float32 for now. - if( (formatDesc.mFormatID == kAudioFormatLinearPCM) && - ((formatDesc.mFormatFlags & kLinearPCMFormatFlagIsFloat) != 0) ) - { - deviceInfo->nativeSampleFormats = paFloat32; - } - else - { - PRINT(("PaOSX_QueryDeviceInfo: ERROR - not LinearPCM & Float32!!!\n")); - return paSampleFormatNotSupported; - } - - maxChannels = PaOSX_GetMaxChannels( devID, isInput ); - if( maxChannels <= 0 ) goto error; - if( isInput ) - { - deviceInfo->maxInputChannels = maxChannels; - } - else - { - deviceInfo->maxOutputChannels = maxChannels; - } - - // Get the device name - deviceInfo->name = PaOSX_DeviceNameFromID( devID, isInput ); - DBUG(("PaOSX_QueryDeviceInfo: name = %s\n", deviceInfo->name )); - return 1; - -error: - return 0; -} - -/**********************************************************************/ -static PaError PaOSX_MaybeQueryDevices( void ) -{ - if( sNumPaDevices == 0 ) - { - return PaOSX_QueryDevices(); - } - return 0; -} - -static char zeroPad[256] = { 0 }; - -/********************************************************************** -** This is the proc that supplies the data to the AudioConverterFillBuffer call. -** We can pass back arbitrarily sized blocks so if the FIFO region is split -** just pass back the first half. -*/ -static OSStatus PaOSX_InputConverterCallbackProc (AudioConverterRef inAudioConverter, - UInt32* outDataSize, - void** outData, - void* inUserData) -{ - internalPortAudioStream *past = (internalPortAudioStream *) inUserData; - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - void *dataPtr1; - long size1; - void *dataPtr2; - long size2; - - /* Pass contiguous region from FIFO directly to converter. */ - RingBuffer_GetReadRegions( &pahsc->ringBuffer, *outDataSize, - &dataPtr1, &size1, &dataPtr2, &size2 ); - - if( size1 > 0 ) - { - *outData = dataPtr1; - *outDataSize = size1; - RingBuffer_AdvanceReadIndex( &pahsc->ringBuffer, size1 ); - DBUGX(("PaOSX_InputConverterCallbackProc: read %ld bytes from FIFO.\n", size1 )); - } - else - { - DBUGBACK(("PaOSX_InputConverterCallbackProc: got no data!\n")); - *outData = zeroPad; /* Give it zero data to keep it happy. */ - *outDataSize = sizeof(zeroPad); - } - return noErr; -} - -/***************************************************************************** -** Get audio input, if any, from passed in buffer, or from converter or from FIFO, -** then run PA callback and output data. -*/ -static OSStatus PaOSX_LoadAndProcess( internalPortAudioStream *past, - void *inputBuffer, void *outputBuffer ) -{ - OSStatus err = noErr; - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - - if( past->past_StopSoon ) - { - if( outputBuffer ) - { - /* Clear remainder of audio buffer if we are waiting for stop. */ - memset( outputBuffer, 0, pahsc->output.bytesPerUserNativeBuffer ); - } - } - else - { - /* Do we need data from the converted input? */ - if( PA_USING_INPUT ) - { - UInt32 size = pahsc->input.bytesPerUserNativeBuffer; - err = AudioConverterFillBuffer( - pahsc->input.converter, - PaOSX_InputConverterCallbackProc, - past, - &size, - pahsc->input.converterBuffer); - if( err != noErr ) return err; - inputBuffer = pahsc->input.converterBuffer; - } - - /* Measure CPU load. */ -#if PA_ENABLE_LOAD_MEASUREMENT - Pa_StartUsageCalculation( past ); -#endif - - /* Fill part of audio converter buffer by converting input to user format, - * calling user callback, then converting output to native format. */ - if( PaConvert_Process( past, inputBuffer, outputBuffer )) - { - past->past_StopSoon = 1; - } - -#if PA_ENABLE_LOAD_MEASUREMENT - Pa_EndUsageCalculation( past ); -#endif - - } - return err; -} - -/***************************************************************************** -** This is the proc that supplies the data to the AudioConverterFillBuffer call -*/ -static OSStatus PaOSX_OutputConverterCallbackProc (AudioConverterRef inAudioConverter, - UInt32* outDataSize, - void** outData, - void* inUserData) -{ - internalPortAudioStream *past = (internalPortAudioStream *) inUserData; - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - - *outData = pahsc->output.converterBuffer; - *outDataSize = pahsc->output.bytesPerUserNativeBuffer; - - return PaOSX_LoadAndProcess ( past, pahsc->input.converterBuffer, pahsc->output.converterBuffer ); -} - -/********************************************************************** -** If data available, write it to the Ring Buffer so we can -** pull it from the other side. -*/ -static OSStatus PaOSX_WriteInputRingBuffer( internalPortAudioStream *past, - const AudioBufferList* inInputData ) -{ - int numBytes = 0; - int currentInterleavedChannelIndex; - int numFramesInInputBuffer; - int numInterleavedChannels; - int numChannelsRemaining; - int i; - long writeRoom; - char *inputNativeBufferfPtr = NULL; - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - - /* Do we need to deinterleave the buffers first? */ - if( past->past_NumInputChannels != inInputData->mBuffers[0].mNumberChannels ) - { - numFramesInInputBuffer = inInputData->mBuffers[0].mDataByteSize / - (sizeof(float) * inInputData->mBuffers[0].mNumberChannels); - - numBytes = numFramesInInputBuffer * sizeof(float) * past->past_NumInputChannels; - - /* Allocate temporary buffer if needed. */ - if ( (pahsc->input.streamInterleavingBuffer != NULL) && - (pahsc->input.streamInterleavingBufferLen < numBytes) ) - { - PaHost_FreeFastMemory( pahsc->input.streamInterleavingBuffer, pahsc->input.streamInterleavingBufferLen ); - pahsc->input.streamInterleavingBuffer = NULL; - } - if ( pahsc->input.streamInterleavingBuffer == NULL ) - { - pahsc->input.streamInterleavingBufferLen = numBytes; - pahsc->input.streamInterleavingBuffer = (float *)PaHost_AllocateFastMemory( pahsc->input.streamInterleavingBufferLen ); - } - - /* Perform interleaving by writing to temp buffer. */ - currentInterleavedChannelIndex = 0; - numInterleavedChannels = past->past_NumInputChannels; - numChannelsRemaining = numInterleavedChannels; - - for( i=0; i<inInputData->mNumberBuffers; i++ ) - { - int j; - int numBufChannels = inInputData->mBuffers[i].mNumberChannels; - /* Don't use more than we need or more than we have. */ - int numChannelsUsedInThisBuffer = (numChannelsRemaining < numBufChannels ) ? - numChannelsRemaining : numBufChannels; - for( j=0; j<numChannelsUsedInThisBuffer; j++ ) - { - int k; - float *dest = &pahsc->input.streamInterleavingBuffer[ currentInterleavedChannelIndex ]; - float *src = &((float *)inInputData->mBuffers[i].mData)[ j ]; - /* Move one channel from CoreAudio buffer to interleaved buffer. */ - for( k=0; k<numFramesInInputBuffer; k++ ) - { - *dest = *src; - src += numBufChannels; - dest += numInterleavedChannels; - } - currentInterleavedChannelIndex++; - } - numChannelsRemaining -= numChannelsUsedInThisBuffer; - if( numChannelsRemaining <= 0 ) break; - } - - inputNativeBufferfPtr = (char *)pahsc->input.streamInterleavingBuffer; - } - else - { - inputNativeBufferfPtr = (char*)inInputData->mBuffers[0].mData; - numBytes = inInputData->mBuffers[0].mDataByteSize; - } - - writeRoom = RingBuffer_GetWriteAvailable( &pahsc->ringBuffer ); - - if( numBytes <= writeRoom ) - { - RingBuffer_Write( &pahsc->ringBuffer, inputNativeBufferfPtr, numBytes ); - DBUGBACK(("PaOSX_WriteInputRingBuffer: wrote %ld bytes to FIFO.\n", numBytes)); - } // FIXME else drop samples on floor, remember overflow??? - - return noErr; -} - -/********************************************************************** -** Use any available input buffers by writing to RingBuffer. -** Process input if PA_MODE_INPUT_ONLY. -*/ -static OSStatus PaOSX_HandleInput( internalPortAudioStream *past, - const AudioBufferList* inInputData ) -{ - OSStatus err = noErr; - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - - if( inInputData == NULL ) - { - DBUG(("PaOSX_HandleInput: inInputData == NULL\n")); - return noErr; - } - - if( inInputData->mNumberBuffers > 0 ) - { - /* Write to FIFO here if we are only using this callback. */ - if( (pahsc->mode == PA_MODE_INPUT_ONLY) || (pahsc->mode == PA_MODE_IO_ONE_DEVICE) ) - { - err = PaOSX_WriteInputRingBuffer( past, inInputData ); - if( err != noErr ) goto error; - } - } - - if( pahsc->mode == PA_MODE_INPUT_ONLY ) - { - /* Generate user buffers as long as we have a half full input FIFO. */ - long halfSize = pahsc->ringBuffer.bufferSize / 2; - while( (RingBuffer_GetReadAvailable( &pahsc->ringBuffer ) >= halfSize) && - (past->past_StopSoon == 0) ) - { - err = PaOSX_LoadAndProcess ( past, NULL, NULL ); - if( err != noErr ) goto error; - } - } - -error: - return err; -} - -/********************************************************************** -** Fill any available output buffers. -*/ -static OSStatus PaOSX_HandleOutput( internalPortAudioStream *past, - AudioBufferList* outOutputData ) -{ - OSStatus err = noErr; - void *outputNativeBufferfPtr = NULL; - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - UInt32 numBytes = 0; - int numChannelsRemaining; - Boolean deinterleavingNeeded; - int numFramesInOutputBuffer; - - if( outOutputData == NULL ) - { - DBUG(("PaOSX_HandleOutput: outOutputData == NULL\n")); - return noErr; - } - - deinterleavingNeeded = past->past_NumOutputChannels != outOutputData->mBuffers[0].mNumberChannels; - - numFramesInOutputBuffer = outOutputData->mBuffers[0].mDataByteSize / - (sizeof(float) * outOutputData->mBuffers[0].mNumberChannels); - - if( pahsc->mode != PA_MODE_INPUT_ONLY ) - { - /* If we are using output, then we need an empty output buffer. */ - if( outOutputData->mNumberBuffers > 0 ) - { - - /* If we have multiple CoreAudio buffers, then we will need to deinterleave after conversion. */ - if( deinterleavingNeeded ) - { - numBytes = numFramesInOutputBuffer * sizeof(float) * past->past_NumOutputChannels; - - /* Free old buffer if we are allocating new one. */ - if ( (pahsc->output.streamInterleavingBuffer != NULL) && - (pahsc->output.streamInterleavingBufferLen < numBytes) ) - { - PaHost_FreeFastMemory( pahsc->output.streamInterleavingBuffer, pahsc->output.streamInterleavingBufferLen ); - pahsc->output.streamInterleavingBuffer = NULL; - } - /* Allocate interleaving buffer if needed. */ - if ( pahsc->output.streamInterleavingBuffer == NULL ) - { - pahsc->output.streamInterleavingBufferLen = numBytes; - pahsc->output.streamInterleavingBuffer = (float *)PaHost_AllocateFastMemory( pahsc->output.streamInterleavingBufferLen ); - } - - outputNativeBufferfPtr = (void*)pahsc->output.streamInterleavingBuffer; - } - else - { - numBytes = outOutputData->mBuffers[0].mDataByteSize; - outputNativeBufferfPtr = (void*)outOutputData->mBuffers[0].mData; - } - - /* Pull data from PA user through converter. */ - err = AudioConverterFillBuffer( - pahsc->output.converter, - PaOSX_OutputConverterCallbackProc, - past, - &numBytes, - outputNativeBufferfPtr); - if( err != noErr ) - { - PRINT_ERR("PaOSX_HandleOutput: AudioConverterFillBuffer failed", err); - goto error; - } - - /* Deinterleave data from PortAudio and write to multiple CoreAudio buffers. */ - if( deinterleavingNeeded ) - { - int numInterleavedChannels = past->past_NumOutputChannels; - int i, currentInterleavedChannelIndex = 0; - numChannelsRemaining = numInterleavedChannels; - - for( i=0; i<outOutputData->mNumberBuffers; i++ ) - { - int numBufChannels = outOutputData->mBuffers[i].mNumberChannels; - int j; - /* Don't use more than we need or more than we have. */ - int numChannelsUsedInThisBuffer = (numChannelsRemaining < numBufChannels ) ? - numChannelsRemaining : numBufChannels; - - for( j=0; j<numChannelsUsedInThisBuffer; j++ ) - { - int k; - float *dest = &((float *)outOutputData->mBuffers[i].mData)[ j ]; - float *src = &pahsc->output.streamInterleavingBuffer[ currentInterleavedChannelIndex ]; - /* Move one channel from interleaved buffer to CoreAudio buffer. */ - for( k=0; k<numFramesInOutputBuffer; k++ ) - { - *dest = *src; - dest += numBufChannels; - src += numInterleavedChannels; - } - currentInterleavedChannelIndex++; - } - - numChannelsRemaining -= numChannelsUsedInThisBuffer; - if( numChannelsRemaining <= 0 ) break; - } - } - } - } - -error: - return err; -} - -/****************************************************************** - * This callback is used when two separate devices are used for input and output. - * This often happens when using USB devices which present as two devices: input and output. - * It just writes its data to a FIFO so that it can be read by the main callback - * proc PaOSX_CoreAudioIOCallback(). - */ -static OSStatus PaOSX_CoreAudioInputCallback (AudioDeviceID inDevice, const AudioTimeStamp* inNow, - const AudioBufferList* inInputData, const AudioTimeStamp* inInputTime, - AudioBufferList* outOutputData, const AudioTimeStamp* inOutputTime, - void* contextPtr) -{ - OSStatus err = noErr; - internalPortAudioStream *past = (internalPortAudioStream *) contextPtr; - PaHostSoundControl *pahsc; - pahsc = (PaHostSoundControl *) past->past_DeviceData; - - /* If there is a FIFO for input then write to it. */ - if( (pahsc->ringBufferData != NULL) && (inInputData != NULL) ) - { - err = PaOSX_WriteInputRingBuffer( past, inInputData ); - if( err != noErr ) goto error; - } -error: - return err; -} - -/****************************************************************** - * This is the primary callback for CoreAudio. - * It can handle input and/or output for a single device. - * It takes input from CoreAudio, converts it and passes it to the - * PortAudio user callback. Then takes the PA results and passes it - * back to CoreAudio. - */ -static OSStatus PaOSX_CoreAudioIOCallback (AudioDeviceID inDevice, const AudioTimeStamp* inNow, - const AudioBufferList* inInputData, const AudioTimeStamp* inInputTime, - AudioBufferList* outOutputData, const AudioTimeStamp* inOutputTime, - void* contextPtr) -{ - OSStatus err = noErr; - internalPortAudioStream *past; - PaHostSoundControl *pahsc; - past = (internalPortAudioStream *) contextPtr; - pahsc = (PaHostSoundControl *) past->past_DeviceData; - - /* Has someone asked us to abort by calling Pa_AbortStream()? */ - if( past->past_StopNow ) - { - past->past_IsActive = 0; /* Will cause thread to return. */ - } - /* Has someone asked us to stop by calling Pa_StopStream() - * OR has a user callback returned '1' to indicate finished. - */ - else if( past->past_StopSoon ) - { - // FIXME - Pretend all done. Should wait for audio to play out but CoreAudio latency very low. - past->past_IsActive = 0; /* Will cause thread to return. */ - } - else - { - /* use time stamp from CoreAudio if valid */ - if( inOutputTime->mFlags & kAudioTimeStampSampleTimeValid) - { - past->past_FrameCount = inOutputTime->mSampleTime; - } - else if( inInputTime->mFlags & kAudioTimeStampSampleTimeValid) - { - past->past_FrameCount = inInputTime->mSampleTime; - } - - past->past_NumCallbacks += 1; - - /* Process full input buffer. */ - err = PaOSX_HandleInput( past, inInputData ); - if( err != 0 ) goto error; - - /* Fill up empty output buffers. */ - err = PaOSX_HandleOutput( past, outOutputData ); - if( err != 0 ) goto error; - } - - if( err != 0 ) DBUG(("PaOSX_CoreAudioIOCallback: returns %ld.\n", err )); - -error: - return err; -} - -/*******************************************************************/ -/** Attempt to set device sample rate. - * This is not critical because we use an AudioConverter but we may - * get better fidelity if we can avoid resampling. - * - * Only set format once because some devices take time to settle. - * Return flag indicating whether format changed so we know whether to wait - * for DevicePropertyListener to get called. - * - * @return negative error, zero if no change, or one if changed successfully. - */ -static PaError PaOSX_SetFormat( AudioDeviceID devID, Boolean isInput, - double desiredRate, int desiredNumChannels ) -{ - AudioStreamBasicDescription formatDesc; - PaError result = 0; - OSStatus err; - UInt32 dataSize; - Float64 originalRate; - int originalChannels; - - /* Get current device format. This is critical because if we pass - * zeros for unspecified fields then the iMic device gets switched to a 16 bit - * integer format!!! I don't know if this is a Mac bug or not. But it only - * started happening when I upgraded from OS X V10.1 to V10.2 (Jaguar). - */ - dataSize = sizeof(formatDesc); - err = AudioDeviceGetProperty( devID, 0, isInput, - kAudioDevicePropertyStreamFormat, &dataSize, &formatDesc); - if( err != noErr ) - { - PRINT_ERR("PaOSX_SetFormat: Could not get format.", err); - sSavedHostError = err; - return paHostError; - } - - originalRate = formatDesc.mSampleRate; - originalChannels = formatDesc.mChannelsPerFrame; - - // Changing the format can mess up other apps. - // So only change the format if the original format - // has a lower sample rate, or fewer channels, than the desired format. - if( (originalRate < desiredRate) || (originalChannels < desiredNumChannels) ) - { - DBUG(("PaOSX_SetFormat: try to change sample rate from %f to %f.\n", originalRate, desiredRate )); - DBUG(("PaOSX_SetFormat: try to set number of channels %d to %d\n", originalChannels, desiredNumChannels)); - - formatDesc.mSampleRate = desiredRate; - formatDesc.mChannelsPerFrame = desiredNumChannels; - formatDesc.mBytesPerFrame = formatDesc.mChannelsPerFrame * sizeof(float); - formatDesc.mBytesPerPacket = formatDesc.mBytesPerFrame * formatDesc.mFramesPerPacket; - - err = AudioDeviceSetProperty( devID, 0, 0, - isInput, kAudioDevicePropertyStreamFormat, sizeof(formatDesc), &formatDesc); - if (err != noErr) - { - /* Could not set to desired rate so query for closest match. */ - dataSize = sizeof(formatDesc); - err = AudioDeviceGetProperty( devID, 0, - isInput, kAudioDevicePropertyStreamFormatMatch, &dataSize, &formatDesc); - - DBUG(("PaOSX_SetFormat: closest rate is %f.\n", formatDesc.mSampleRate )); - DBUG(("PaOSX_SetFormat: closest numChannels is %d.\n", (int)formatDesc.mChannelsPerFrame )); - // Set to closest if different from original. - if( (err == noErr) && - ((originalRate != formatDesc.mSampleRate) || - (originalChannels != formatDesc.mChannelsPerFrame)) ) - { - err = AudioDeviceSetProperty( devID, 0, 0, - isInput, kAudioDevicePropertyStreamFormat, sizeof(formatDesc), &formatDesc); - if( err == noErr ) result = 1; - } - } - else result = 1; - } - - return result; -} - -#if 0 -static void PaOSX_DumpDeviceInfo( AudioDeviceID devID, Boolean isInput ) -{ - OSStatus err = noErr; - UInt32 dataSize; - UInt32 uidata32; - Float32 fdata32; - AudioValueRange audioRange; - - dataSize = sizeof( uidata32 ); - err = AudioDeviceGetProperty( devID, 0, isInput, - kAudioDevicePropertyLatency, &dataSize, &uidata32 ); - if( err != noErr ) - { - PRINT_ERR("Error reading kAudioDevicePropertyLatency", err); - return; - } - PRINT(("kAudioDevicePropertyLatency = %d\n", (int)uidata32 )); - - dataSize = sizeof( fdata32 ); - err = AudioDeviceGetProperty( devID, 1, isInput, - kAudioDevicePropertyVolumeScalar, &dataSize, &fdata32 ); - if( err != noErr ) - { - PRINT_ERR("Error reading kAudioDevicePropertyVolumeScalar", err); - return; - } - PRINT(("kAudioDevicePropertyVolumeScalar = %f\n", fdata32 )); - - dataSize = sizeof( uidata32 ); - err = AudioDeviceGetProperty( devID, 0, isInput, - kAudioDevicePropertyBufferSize, &dataSize, &uidata32 ); - if( err != noErr ) - { - PRINT_ERR("Error reading buffer size", err); - return; - } - PRINT(("kAudioDevicePropertyBufferSize = %d bytes\n", (int)uidata32 )); - - dataSize = sizeof( audioRange ); - err = AudioDeviceGetProperty( devID, 0, isInput, - kAudioDevicePropertyBufferSizeRange, &dataSize, &audioRange ); - if( err != noErr ) - { - PRINT_ERR("Error reading buffer size range", err); - return; - } - PRINT(("kAudioDevicePropertyBufferSizeRange = %g to %g bytes\n", audioRange.mMinimum, audioRange.mMaximum )); - - dataSize = sizeof( uidata32 ); - err = AudioDeviceGetProperty( devID, 0, isInput, - kAudioDevicePropertyBufferFrameSize, &dataSize, &uidata32 ); - if( err != noErr ) - { - PRINT_ERR("Error reading buffer size", err); - return; - } - PRINT(("kAudioDevicePropertyBufferFrameSize = %d frames\n", (int)uidata32 )); - - dataSize = sizeof( audioRange ); - err = AudioDeviceGetProperty( devID, 0, isInput, - kAudioDevicePropertyBufferFrameSizeRange, &dataSize, &audioRange ); - if( err != noErr ) - { - PRINT_ERR("Error reading buffer size range", err); - return; - } - PRINT(("kAudioDevicePropertyBufferFrameSizeRange = %g to %g frames\n", audioRange.mMinimum, audioRange.mMaximum )); - - return; -} -#endif - -/*******************************************************************/ -static OSStatus PAOSX_DevicePropertyListener (AudioDeviceID inDevice, - UInt32 inChannel, - Boolean isInput, - AudioDevicePropertyID inPropertyID, - void* inClientData) -{ - PaHostSoundControl *pahsc; - internalPortAudioStream *past; - UInt32 dataSize; - OSStatus err = noErr; - AudioStreamBasicDescription userStreamFormat, hardwareStreamFormat; - PaHostInOut *hostInOut; - AudioStreamBasicDescription *destFormatPtr, *srcFormatPtr; - AudioConverterRef oldConverter = NULL; // PLB 20031208 - Declare here for standard 'C'. - - past = (internalPortAudioStream *) inClientData; - pahsc = (PaHostSoundControl *) past->past_DeviceData; - - DBUG(("PAOSX_DevicePropertyListener: called with propertyID = 0x%0X\n", (unsigned int) inPropertyID )); - - if(inPropertyID == kAudioDevicePropertyStreamFormat) - { - /* Get target device format */ - dataSize = sizeof(hardwareStreamFormat); - err = AudioDeviceGetProperty(inDevice, 0, isInput, - kAudioDevicePropertyStreamFormat, &dataSize, &hardwareStreamFormat); - if( err != noErr ) - { - PRINT_ERR("PAOSX_DevicePropertyListener: Could not get device format", err); - sSavedHostError = err; - goto error; - } - - DBUG(("PAOSX_DevicePropertyListener: HW mChannelsPerFrame = %d\n", (int)hardwareStreamFormat.mChannelsPerFrame )); - - /* Set source user format. */ - userStreamFormat = hardwareStreamFormat; - userStreamFormat.mSampleRate = past->past_SampleRate; // sample rate of the user synthesis code - userStreamFormat.mChannelsPerFrame = (isInput) ? past->past_NumInputChannels : past->past_NumOutputChannels; // the number of channels in each frame - DBUG(("PAOSX_DevicePropertyListener: User mChannelsPerFrame = %d\n", (int)userStreamFormat.mChannelsPerFrame )); - - userStreamFormat.mBytesPerFrame = userStreamFormat.mChannelsPerFrame * sizeof(float); - userStreamFormat.mBytesPerPacket = userStreamFormat.mBytesPerFrame * userStreamFormat.mFramesPerPacket; - - /* Don't use AudioConverter for merging or splitting channels. */ - hardwareStreamFormat.mChannelsPerFrame = userStreamFormat.mChannelsPerFrame; - hardwareStreamFormat.mBytesPerFrame = userStreamFormat.mBytesPerFrame; - hardwareStreamFormat.mBytesPerPacket = userStreamFormat.mBytesPerPacket; - - if( isInput ) - { - hostInOut = &pahsc->input; - srcFormatPtr = &hardwareStreamFormat; - destFormatPtr = &userStreamFormat; - } - else - { - hostInOut = &pahsc->output; - srcFormatPtr = &userStreamFormat; - destFormatPtr = &hardwareStreamFormat; - } - DBUG(("PAOSX_DevicePropertyListener: source rate = %f\n", srcFormatPtr->mSampleRate )); - DBUG(("PAOSX_DevicePropertyListener: dest rate = %f\n", destFormatPtr->mSampleRate )); - - // Don't delete old converter until we create new one so we don't pull - // the rug out from under other audio threads. - oldConverter = hostInOut->converter; - - // Make converter to change sample rate. - err = AudioConverterNew ( - srcFormatPtr, - destFormatPtr, - &hostInOut->converter ); - if( err != noErr ) - { - PRINT_ERR("Could not create format converter", err); - sSavedHostError = err; - goto error; - } - - if( oldConverter != NULL ) - { - verify_noerr( AudioConverterDispose( oldConverter ) ); - } - } - -error: - pahsc->formatListenerCalled = true; - return err; -} - -/* Allocate FIFO between Device callback and Converter callback so that device can push data -* and converter can pull data. -*/ -static PaError PaOSX_CreateInputRingBuffer( internalPortAudioStream *past ) -{ - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - OSStatus err = noErr; - UInt32 dataSize; - double sampleRateRatio; - long numBytes; - UInt32 framesPerHostBuffer; - UInt32 bytesForDevice; - UInt32 bytesForUser; - UInt32 bytesPerFrame; - AudioStreamBasicDescription formatDesc; - - dataSize = sizeof(formatDesc); - err = AudioDeviceGetProperty( pahsc->input.audioDeviceID, 0, IS_INPUT, - kAudioDevicePropertyStreamFormat, &dataSize, &formatDesc); - if( err != noErr ) - { - PRINT_ERR("PaOSX_CreateInputRingBuffer: Could not get input format.\n", err); - sSavedHostError = err; - return paHostError; - } - - // If device is delivering audio faster than being consumed then buffer must be bigger. - sampleRateRatio = formatDesc.mSampleRate / past->past_SampleRate; - - // Get size of CoreAudio IO buffers. - dataSize = sizeof(framesPerHostBuffer); - err = AudioDeviceGetProperty( pahsc->input.audioDeviceID, 0, IS_INPUT, - kAudioDevicePropertyBufferFrameSize, &dataSize, - &framesPerHostBuffer); - if( err != noErr ) - { - PRINT_ERR("PaOSX_CreateInputRingBuffer: Could not get input buffer size.\n", err); - sSavedHostError = err; - return paHostError; - } - - bytesPerFrame = past->past_NumInputChannels * sizeof(Float32); - - bytesForDevice = framesPerHostBuffer * bytesPerFrame * 2; - bytesForUser = past->past_FramesPerUserBuffer * bytesPerFrame * 3 * sampleRateRatio; - - // Ring buffer should be large enough to consume audio input from device, - // and to deliver a complete user buffer. - numBytes = (bytesForDevice > bytesForUser) ? bytesForDevice : bytesForUser; - - numBytes = RoundUpToNextPowerOf2( numBytes ); - - DBUG(("PaOSX_CreateInputRingBuffer: FIFO numBytes = %ld\n", numBytes)); - pahsc->ringBufferData = PaHost_AllocateFastMemory( numBytes ); - if( pahsc->ringBufferData == NULL ) - { - return paInsufficientMemory; - } - RingBuffer_Init( &pahsc->ringBuffer, numBytes, pahsc->ringBufferData ); - // Make it look almost full at beginning. We must advance by an integral number of frames - // so that the channels don't get scrambled when numChannels is not a power of 2. - { - int numZeroFrames = numBytes / bytesPerFrame; - int numZeroBytes = numZeroFrames * bytesPerFrame; - RingBuffer_AdvanceWriteIndex( &pahsc->ringBuffer, numZeroBytes ); - } - - return paNoError; -} - -/****************************************************************** - * Try to set the I/O bufferSize of the device. - * Scale the size by the ratio of the sample rates so that the converter will have - * enough data to operate on. - */ -static OSStatus PaOSX_SetDeviceBufferSize( AudioDeviceID devID, Boolean isInput, int framesPerUserBuffer, Float64 sampleRateRatio ) -{ - UInt32 dataSize; - UInt32 ioBufferSize; - int scaler; - - scaler = (int) sampleRateRatio; - if( sampleRateRatio > (Float64) scaler ) scaler += 1; - DBUG(("PaOSX_SetDeviceBufferSize: buffer size scaler = %d\n", scaler )); - ioBufferSize = framesPerUserBuffer * scaler; - - // Limit buffer size to reasonable value. - if( ioBufferSize < 128 ) ioBufferSize = 128; - - dataSize = sizeof(ioBufferSize); - return AudioDeviceSetProperty( devID, 0, 0, isInput, - kAudioDevicePropertyBufferFrameSize, dataSize, - &ioBufferSize); -} - - -/*******************************************************************/ -static PaError PaOSX_OpenCommonDevice( internalPortAudioStream *past, - PaHostInOut *inOut, Boolean isInput ) -{ - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - PaError result = paNoError; - OSStatus err = noErr; - Float64 deviceRate; - - - // The HW device format changes are asynchronous. - // So we don't know when or if the PAOSX_DevicePropertyListener() will - // get called. To be safe, call the listener now to forcibly create the converter. - if( inOut->converter == NULL ) - { - err = PAOSX_DevicePropertyListener (inOut->audioDeviceID, - 0, isInput, kAudioDevicePropertyStreamFormat, past); - if (err != kAudioHardwareNoError) - { - PRINT_ERR("PaOSX_OpenCommonDevice: PAOSX_DevicePropertyListener failed.\n", err); - sSavedHostError = err; - return paHostError; - } - } - - // Add listener for when format changed by other apps. - DBUG(("PaOSX_OpenCommonDevice: call AudioDeviceAddPropertyListener()\n" )); - err = AudioDeviceAddPropertyListener( inOut->audioDeviceID, 0, isInput, - kAudioDevicePropertyStreamFormat, - (AudioDevicePropertyListenerProc) PAOSX_DevicePropertyListener, past ); - if (err != noErr) - { - return -1; // FIXME - } - - // Only change format if current HW format is different. - // Don't bother to check result because we are going to use an AudioConverter anyway. - pahsc->formatListenerCalled = false; - result = PaOSX_SetFormat( inOut->audioDeviceID, isInput, past->past_SampleRate, inOut->numChannels ); - // Synchronize with device because format changes put some devices into unusable mode. - if( result > 0 ) - { - const int sleepDurMsec = 50; - int spinCount = MIN_TIMEOUT_MSEC / sleepDurMsec; - while( !pahsc->formatListenerCalled && (spinCount > 0) ) - { - Pa_Sleep( sleepDurMsec ); // FIXME - use a semaphore or signal - spinCount--; - } - if( !pahsc->formatListenerCalled ) - { - PRINT(("PaOSX_OpenCommonDevice: timed out waiting for device format to settle.\n")); - } - result = 0; - } - -#if SET_DEVICE_BUFFER_SIZE - // Try to set the I/O bufferSize of the device. - { - Float64 ratio; - deviceRate = PaOSX_GetDeviceSampleRate( inOut->audioDeviceID, isInput ); - if( deviceRate <= 0.0 ) deviceRate = past->past_SampleRate; - ratio = deviceRate / past->past_SampleRate ; - err = PaOSX_SetDeviceBufferSize( inOut->audioDeviceID, isInput, - past->past_FramesPerUserBuffer, ratio ); - if( err != noErr ) - { - DBUG(("PaOSX_OpenCommonDevice: Could not set I/O buffer size.\n")); - } - } -#endif - - /* Allocate an input buffer because we need it between the user callback and the converter. */ - inOut->converterBuffer = PaHost_AllocateFastMemory( inOut->bytesPerUserNativeBuffer ); - if( inOut->converterBuffer == NULL ) - { - return paInsufficientMemory; - } - - return result; -} - -/*******************************************************************/ -static PaError PaOSX_OpenInputDevice( internalPortAudioStream *past ) -{ - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - const PaHostDeviceInfo *hostDeviceInfo; - PaError result = paNoError; - - DBUG(("PaOSX_OpenInputDevice: -------------\n")); - - if( (past->past_InputDeviceID < LOWEST_INPUT_DEVID) || - (past->past_InputDeviceID > HIGHEST_INPUT_DEVID) ) - { - return paInvalidDeviceId; - } - hostDeviceInfo = &sDeviceInfos[past->past_InputDeviceID]; - - if( past->past_NumInputChannels > hostDeviceInfo->paInfo.maxInputChannels ) - { - return paInvalidChannelCount; /* Too many channels! */ - } - pahsc->input.numChannels = past->past_NumInputChannels; - - // setup PA conversion procedure - result = PaConvert_SetupInput( past, paFloat32 ); - - result = PaOSX_OpenCommonDevice( past, &pahsc->input, IS_INPUT ); - if( result != paNoError ) goto error; - - // Allocate a ring buffer so we can push in data from device, and pull through AudioConverter. - result = PaOSX_CreateInputRingBuffer( past ); - if( result != paNoError ) goto error; - -error: - return result; -} - -/*******************************************************************/ -static PaError PaOSX_OpenOutputDevice( internalPortAudioStream *past ) -{ - PaHostSoundControl *pahsc; - const PaHostDeviceInfo *hostDeviceInfo; - PaError result = paNoError; - - DBUG(("PaOSX_OpenOutputDevice: -------------\n")); - pahsc = (PaHostSoundControl *) past->past_DeviceData; - - // Validate DeviceID - DBUG(("PaOSX_OpenOutputDevice: deviceID = 0x%x\n", past->past_OutputDeviceID)); - if( (past->past_OutputDeviceID < LOWEST_OUTPUT_DEVID) || - (past->past_OutputDeviceID > HIGHEST_OUTPUT_DEVID) ) - { - return paInvalidDeviceId; - } - hostDeviceInfo = &sDeviceInfos[past->past_OutputDeviceID]; - - // Validate number of channels. - if( past->past_NumOutputChannels > hostDeviceInfo->paInfo.maxOutputChannels ) - { - return paInvalidChannelCount; /* Too many channels! */ - } - pahsc->output.numChannels = past->past_NumOutputChannels; - - // setup conversion procedure - result = PaConvert_SetupOutput( past, paFloat32 ); - if( result != paNoError ) goto error; - - result = PaOSX_OpenCommonDevice( past, &pahsc->output, IS_OUTPUT ); - if( result != paNoError ) goto error; - -error: - return result; -} - -/******************************************************************* -* Determine how many User Buffers we can put into our CoreAudio stream buffer. -* Uses: -* past->past_FramesPerUserBuffer, etc. -* Sets: -* past->past_NumUserBuffers -* pahsc->input.bytesPerUserNativeBuffer -* pahsc->output.bytesPerUserNativeBuffer -*/ -static void PaOSX_CalcHostBufferSize( internalPortAudioStream *past ) -{ - PaHostSoundControl *pahsc = ( PaHostSoundControl *)past->past_DeviceData; - - // Determine number of user buffers based strictly on minimum reasonable buffer size. - // We ignore the Pa_OpenStream numBuffer parameter because CoreAudio has a big - // mix buffer and handles latency automatically. - past->past_NumUserBuffers = Pa_GetMinNumBuffers( past->past_FramesPerUserBuffer, past->past_SampleRate ); - - // calculate buffer sizes in bytes - pahsc->input.bytesPerUserNativeBuffer = past->past_FramesPerUserBuffer * - Pa_GetSampleSize(paFloat32) * past->past_NumInputChannels; - pahsc->output.bytesPerUserNativeBuffer = past->past_FramesPerUserBuffer * - Pa_GetSampleSize(paFloat32) * past->past_NumOutputChannels; - - DBUG(("PaOSX_CalcNumHostBuffers: past_NumUserBuffers = %ld\n", past->past_NumUserBuffers )); - DBUG(("PaOSX_CalcNumHostBuffers: input.bytesPerUserNativeBuffer = %d\n", pahsc->input.bytesPerUserNativeBuffer )); - DBUG(("PaOSX_CalcNumHostBuffers: output.bytesPerUserNativeBuffer = %d\n", pahsc->output.bytesPerUserNativeBuffer )); -} - -/*****************************************************************************/ -/************** Internal Host API ********************************************/ -/*****************************************************************************/ -PaError PaHost_OpenStream( internalPortAudioStream *past ) -{ - PaError result = paNoError; - PaHostSoundControl *pahsc; - Boolean useInput; - Boolean useOutput; - - assert( past->past_Magic == PA_MAGIC ); - - /* Allocate and initialize host data. */ - pahsc = (PaHostSoundControl *) malloc(sizeof(PaHostSoundControl)); - if( pahsc == NULL ) - { - result = paInsufficientMemory; - goto error; - } - memset( pahsc, 0, sizeof(PaHostSoundControl) ); - past->past_DeviceData = (void *) pahsc; - pahsc->primaryDeviceID = kAudioDeviceUnknown; - pahsc->input.audioDeviceID = kAudioDeviceUnknown; - pahsc->output.audioDeviceID = kAudioDeviceUnknown; - - PaOSX_CalcHostBufferSize( past ); - - useOutput = (past->past_OutputDeviceID != paNoDevice) && (past->past_NumOutputChannels > 0); - useInput = (past->past_InputDeviceID != paNoDevice) && (past->past_NumInputChannels > 0); - - // Set device IDs and determine IO Device mode. - if( useOutput ) - { - pahsc->output.audioDeviceID = sDeviceInfos[past->past_OutputDeviceID].audioDeviceID; - pahsc->primaryDeviceID = pahsc->output.audioDeviceID; - if( useInput ) - { - pahsc->input.audioDeviceID = sDeviceInfos[past->past_InputDeviceID].audioDeviceID; - pahsc->mode = ( pahsc->input.audioDeviceID != pahsc->primaryDeviceID ) ? - PA_MODE_IO_TWO_DEVICES : PA_MODE_IO_ONE_DEVICE; - } - else - { - pahsc->mode = PA_MODE_OUTPUT_ONLY; - } - } - else - { - /* Just input, not output. */ - pahsc->input.audioDeviceID = sDeviceInfos[past->past_InputDeviceID].audioDeviceID; - pahsc->primaryDeviceID = pahsc->input.audioDeviceID; - pahsc->mode = PA_MODE_INPUT_ONLY; - } - - DBUG(("outputDeviceID = %ld\n", pahsc->output.audioDeviceID )); - DBUG(("inputDeviceID = %ld\n", pahsc->input.audioDeviceID )); - DBUG(("primaryDeviceID = %ld\n", pahsc->primaryDeviceID )); - - /* ------------------ OUTPUT */ - if( useOutput ) - { - result = PaOSX_OpenOutputDevice( past ); - if( result < 0 ) goto error; - } - - /* ------------------ INPUT */ - if( useInput ) - { - result = PaOSX_OpenInputDevice( past ); - if( result < 0 ) goto error; - } - -#if PA_ENABLE_LOAD_MEASUREMENT - pahsc->inverseHostTicksPerBuffer = past->past_SampleRate / - (AudioGetHostClockFrequency() * past->past_FramesPerUserBuffer); - DBUG(("inverseHostTicksPerBuffer based on buffer size of %d frames.\n", past->past_FramesPerUserBuffer )); -#else - PRINT(("WARNING - Pa_GetCPULoad() mesaurement disabled in pa_mac_core.c.\n")); -#endif - - return result; - -error: - PaHost_CloseStream( past ); - return result; -} - -/*************************************************************************/ -PaError PaHost_StartOutput( internalPortAudioStream *past ) -{ - return 0; -} - -/*************************************************************************/ -PaError PaHost_StartInput( internalPortAudioStream *past ) -{ - return 0; -} - -/*************************************************************************/ -PaError PaHost_StartEngine( internalPortAudioStream *past ) -{ - OSStatus err = noErr; - PaError result = paNoError; - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - - past->past_StopSoon = 0; - past->past_StopNow = 0; - past->past_IsActive = 1; - -/* If full duplex and using two separate devices then start input device. */ - if( pahsc->mode == PA_MODE_IO_TWO_DEVICES ) - { - // Associate an IO proc with the device and pass a pointer to the audio data context - err = AudioDeviceAddIOProc(pahsc->input.audioDeviceID, (AudioDeviceIOProc)PaOSX_CoreAudioInputCallback, past); - if (err != noErr) - { - PRINT_ERR("PaHost_StartEngine: AudioDeviceAddIOProc secondary failed", err ); - goto error; - } - - // start playing sound through the device - err = AudioDeviceStart(pahsc->input.audioDeviceID, (AudioDeviceIOProc)PaOSX_CoreAudioInputCallback); - if (err != noErr) - { - PRINT_ERR("PaHost_StartEngine: AudioDeviceStart secondary failed", err ); - PRINT(("The program may succeed if you run it again!\n")); - goto error; - } - } - - // Associate an IO proc with the device and pass a pointer to the audio data context - err = AudioDeviceAddIOProc(pahsc->primaryDeviceID, (AudioDeviceIOProc)PaOSX_CoreAudioIOCallback, past); - if (err != noErr) - { - PRINT_ERR("PaHost_StartEngine: AudioDeviceAddIOProc primary failed", err ); - goto error; - } - - // start playing sound through the device - err = AudioDeviceStart(pahsc->primaryDeviceID, (AudioDeviceIOProc)PaOSX_CoreAudioIOCallback); - if (err != noErr) - { - PRINT_ERR("PaHost_StartEngine: AudioDeviceStart primary failed", err ); - PRINT(("The program may succeed if you run it again!\n")); - goto error; - } - - return result; - -error: - sSavedHostError = err; - return paHostError; -} - -/*************************************************************************/ -PaError PaHost_StopEngine( internalPortAudioStream *past, int abort ) -{ - OSStatus err = noErr; - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - if( pahsc == NULL ) return paNoError; - (void) abort; - - /* Tell background thread to stop generating more data and to let current data play out. */ - past->past_StopSoon = 1; - /* If aborting, tell background thread to stop NOW! */ - if( abort ) past->past_StopNow = 1; - past->past_IsActive = 0; - -#if PA_TRACE_START_STOP - AddTraceMessage( "PaHost_StopOutput: pahsc_HWaveOut ", (int) pahsc->pahsc_HWaveOut ); -#endif - - // FIXME - we should ask proc to stop instead of stopping abruptly - err = AudioDeviceStop(pahsc->primaryDeviceID, (AudioDeviceIOProc)PaOSX_CoreAudioIOCallback); - if (err != noErr) - { - goto error; - } - - err = AudioDeviceRemoveIOProc(pahsc->primaryDeviceID, (AudioDeviceIOProc)PaOSX_CoreAudioIOCallback); - if (err != noErr) goto error; - -/* If full duplex and using two separate devices then stop second input device. */ - if( pahsc->mode == PA_MODE_IO_TWO_DEVICES ) - { - err = AudioDeviceStop(pahsc->input.audioDeviceID, (AudioDeviceIOProc)PaOSX_CoreAudioInputCallback); - if (err != noErr) goto error; - err = AudioDeviceRemoveIOProc(pahsc->input.audioDeviceID, (AudioDeviceIOProc)PaOSX_CoreAudioInputCallback); - if (err != noErr) goto error; - } - - return paNoError; - -error: - sSavedHostError = err; - return paHostError; -} - -/*************************************************************************/ -PaError PaHost_StopInput( internalPortAudioStream *past, int abort ) -{ - return paNoError; -} - -/*************************************************************************/ -PaError PaHost_StopOutput( internalPortAudioStream *past, int abort ) -{ - return paNoError; -} - -/*******************************************************************/ -PaError PaHost_CloseStream( internalPortAudioStream *past ) -{ - PaHostSoundControl *pahsc; - - if( past == NULL ) return paBadStreamPtr; - pahsc = (PaHostSoundControl *) past->past_DeviceData; - if( pahsc == NULL ) return paNoError; - - //PaOSX_DumpDeviceInfo( sDeviceInfos[past->past_OutputDeviceID].audioDeviceID, IS_OUTPUT ); - -#if PA_TRACE_START_STOP - AddTraceMessage( "PaHost_CloseStream: pahsc_HWaveOut ", (int) pahsc->pahsc_HWaveOut ); -#endif - // Stop Listener callbacks ASAP before dismantling stream. - if( PA_USING_INPUT ) - { - AudioDeviceRemovePropertyListener( pahsc->input.audioDeviceID, 0, IS_INPUT, - kAudioDevicePropertyStreamFormat, - (AudioDevicePropertyListenerProc) PAOSX_DevicePropertyListener ); - } - - if( PA_USING_OUTPUT ) - { - AudioDeviceRemovePropertyListener( pahsc->output.audioDeviceID, 0, IS_OUTPUT, - kAudioDevicePropertyStreamFormat, - (AudioDevicePropertyListenerProc) PAOSX_DevicePropertyListener ); - } - - if( pahsc->output.converterBuffer != NULL ) - { - PaHost_FreeFastMemory( pahsc->output.converterBuffer, pahsc->output.bytesPerUserNativeBuffer ); - } - if( pahsc->input.converterBuffer != NULL ) - { - PaHost_FreeFastMemory( pahsc->input.converterBuffer, pahsc->input.bytesPerUserNativeBuffer ); - } - if( pahsc->ringBufferData != NULL ) - { - PaHost_FreeFastMemory( pahsc->ringBufferData, pahsc->ringBuffer.bufferSize ); - } - if( pahsc->output.converter != NULL ) - { - verify_noerr(AudioConverterDispose (pahsc->output.converter)); - } - if( pahsc->input.converter != NULL ) - { - verify_noerr(AudioConverterDispose (pahsc->input.converter)); - } - if ( pahsc->input.streamInterleavingBuffer != NULL ) - { - PaHost_FreeFastMemory( pahsc->input.streamInterleavingBuffer, pahsc->input.streamInterleavingBufferLen ); - } - if ( pahsc->output.streamInterleavingBuffer != NULL ) - { - PaHost_FreeFastMemory( pahsc->output.streamInterleavingBuffer, pahsc->output.streamInterleavingBufferLen ); - } - - free( pahsc ); - past->past_DeviceData = NULL; - - return paNoError; -} - -/********************************************************************** -** Initialize Host dependant part of API. -*/ -PaError PaHost_Init( void ) -{ - return PaOSX_MaybeQueryDevices(); -} - -/************************************************************************* -** Cleanup device info. -*/ -PaError PaHost_Term( void ) -{ - int i; - - if( sDeviceInfos != NULL ) - { - for( i=0; i<sNumPaDevices; i++ ) - { - if( sDeviceInfos[i].paInfo.name != NULL ) - { - free( (char*)sDeviceInfos[i].paInfo.name ); - } - } - free( sDeviceInfos ); - sDeviceInfos = NULL; - } - - sNumPaDevices = 0; - return paNoError; -} - -/************************************************************************* - * Allocate memory that can be accessed in real-time. - * This may need to be held in physical memory so that it is not - * paged to virtual memory. - * This call MUST be balanced with a call to PaHost_FreeFastMemory(). - */ -void *PaHost_AllocateFastMemory( long numBytes ) -{ - void *addr = malloc( numBytes ); /* FIXME - do we need physical memory, not virtual memory? */ - if( addr != NULL ) memset( addr, 0, numBytes ); - return addr; -} - -/************************************************************************* - * Free memory that could be accessed in real-time. - * This call MUST be balanced with a call to PaHost_AllocateFastMemory(). - */ -void PaHost_FreeFastMemory( void *addr, long numBytes ) -{ - if( addr != NULL ) free( addr ); -} - - -/***********************************************************************/ -PaError PaHost_StreamActive( internalPortAudioStream *past ) -{ - PaHostSoundControl *pahsc; - if( past == NULL ) return paBadStreamPtr; - pahsc = (PaHostSoundControl *) past->past_DeviceData; - if( pahsc == NULL ) return paInternalError; - return (PaError) past->past_IsActive; -} - -/*****************************************************************************/ -/************** External User API ********************************************/ -/*****************************************************************************/ - -/********************************************************************** -** Query devices and use result. -*/ -PaDeviceID Pa_GetDefaultInputDeviceID( void ) -{ - PaError result = PaOSX_MaybeQueryDevices(); - if( result < 0 ) return result; - return sDefaultInputDeviceID; -} - -PaDeviceID Pa_GetDefaultOutputDeviceID( void ) -{ - PaError result = PaOSX_MaybeQueryDevices(); - if( result < 0 ) return result; - return sDefaultOutputDeviceID; -} - - -/************************************************************************* -** Determine minimum number of buffers required for this host based -** on minimum latency. Because CoreAudio manages latency, this just selects -** a reasonably small number of buffers. -*/ -int Pa_GetMinNumBuffers( int framesPerBuffer, double framesPerSecond ) -{ - int minBuffers; - double denominator; - int minLatencyMsec = PA_MIN_LATENCY_MSEC; - denominator = 1000.0 * framesPerBuffer; - minBuffers = (int) (((minLatencyMsec * framesPerSecond) + denominator - 1) / denominator ); - if( minBuffers < 1 ) minBuffers = 1; - return minBuffers; -} - -/*************************************************************************/ -void Pa_Sleep( long msec ) -{ - usleep( msec * 1000 ); -} - -/*************************************************************************/ -PaTimestamp Pa_StreamTime( PortAudioStream *stream ) -{ - AudioTimeStamp timeStamp; - PaTimestamp streamTime; - PaHostSoundControl *pahsc; - internalPortAudioStream *past = (internalPortAudioStream *) stream; - if( past == NULL ) return paBadStreamPtr; - pahsc = (PaHostSoundControl *) past->past_DeviceData; - - AudioDeviceGetCurrentTime(pahsc->primaryDeviceID, &timeStamp); - - streamTime = ( timeStamp.mFlags & kAudioTimeStampSampleTimeValid) ? - timeStamp.mSampleTime : past->past_FrameCount; - - return streamTime; -} - -/************************************************************************************/ -long Pa_GetHostError() -{ - return sSavedHostError; -} - -/*************************************************************************/ -int Pa_CountDevices() -{ - if( sNumPaDevices <= 0 ) Pa_Initialize(); - return sNumPaDevices; -} - -/************************************************************************* -** PaDeviceInfo structures have already been created -** so just return the pointer. -** -*/ -const PaDeviceInfo* Pa_GetDeviceInfo( PaDeviceID id ) -{ - if( id < 0 || id >= sNumPaDevices ) - return NULL; - - return &sDeviceInfos[id].paInfo; -} - - diff --git a/pd/portaudio_v18/pa_sgi/Makefile b/pd/portaudio_v18/pa_sgi/Makefile deleted file mode 100644 index 0cecda2d..00000000 --- a/pd/portaudio_v18/pa_sgi/Makefile +++ /dev/null @@ -1,63 +0,0 @@ -# Makefile for pa_sgi. PortAudio for Silicon Graphics IRIX 6.2-6.5. -# Pieter suurmond, march 15, 2003. (pa-V18-patch). -# Tested under IRIX 6.5 with both GCC and MIPS compilers. -# Based on SGI-specific sproc()-method to spawn children, not on POSIX-threads. - -# Choose compiler in combination with options: - -CC = cc -CFLAGS = -O2 - -# Possible options with MIPSpro compiler are: -32, -o32, -n32, -64, -# -mips1, -mips2, -mips3, -mips4, etc. Use -g, -g2, -g3 for debugging. -# For GCC, use -Wall. And use for example -O2 or -O3 for better optimization: - -#CC = gcc -#CFLAGS = -O2 -Wall - -# Instead of "-lpthread", as with linux, just the audio- and math-library for SGI: - -LIBS = -laudio -lm - -# So sourcefiles can find included headerfiles in pa_common: -CDEFINES = -I../pa_common - -PASRC = ../pa_common/pa_lib.c pa_sgi.c -PAINC = ../pa_common/portaudio.h - -# Tests performed on SGI Indy with R5000 @ 180MHz running IRIX 6.5: -# Used GCC compiler version 3.0.4. and MIPS. - -TESTC = $(PASRC) ../pa_tests/patest_record.c ## OK GCC march 2003 (MIPS sees errors in patest_record.c, refuses compilation). -#TESTC = $(PASRC) ../pa_tests/patest_latency.c ## OK GCC march 2003. (MIPS doesn't like //) -#TESTC = $(PASRC) ../pa_tests/patest_longsine.c ## OK GCC march 2003. -#TESTC = $(PASRC) ../pa_tests/patest_wire.c ## OK GCC Click free now. march 2003. -#TESTC = $(PASRC) ../pa_tests/pa_fuzz.c ## OK GCC march 2003. - -#TESTC = $(PASRC) ../pa_tests/pa_devs.c # Never knew my Indy had 16 input- and output-channels! -#TESTC = $(PASRC) ../pa_tests/patest_saw.c ## - Pa_sleep doesn't work anymore?! -#TESTC = $(PASRC) ../pa_tests/patest_sine.c # - Pa_sleep doesn't work anymore?! -#TESTC = $(PASRC) ../pa_tests/patest_many.c ## - Pa_sleep doesn't work anymore?! -#TESTC = $(PASRC) ../pa_tests/patest_sine_time.c ## Silence for 2 seconds doesn't work, sine sounds ok. - -#TESTC = $(PASRC) ../pa_tests/patest_sine8.c ## Silence for 2 seconds doesn't work, sine sounds ok. -#TESTC = $(PASRC) ../pa_tests/patest_leftright.c # OK GCC and MIPS-CC march 2003. -#TESTC = $(PASRC) ../pa_tests/patest_pink.c ## OK GCC march 2003 -#TESTC = $(PASRC) ../pa_tests/patest_clip.c # -#TESTC = $(PASRC) ../pa_tests/patest_stop.c # MIPS doesn't like patest_stop.c Worked before but now - # error AL_BAD_QSIZE on IRIX 6.5 (with GCC). -#TESTC = $(PASRC) ../pa_tests/patest_dither.c # -#TESTC = $(PASRC) ../pa_tests/patest_sync.c # I don't hear the 6th beep, not really in sync @500ms lat. -#TESTC = $(PASRC) ../pa_tests/paqa_devs.c # A lot of error messages but no coredump. -#TESTC = $(PASRC) ../pa_tests/paqa_errs.c # Segmentation fault (core dumped)! - -TESTH = $(PAINC) - -all: patest - -patest: $(TESTC) $(TESTH) Makefile - $(CC) $(CFLAGS) $(TESTC) $(CDEFINES) $(LIBS) -o patest - -run: patest - ./patest - diff --git a/pd/portaudio_v18/pa_sgi/pa_sgi.c b/pd/portaudio_v18/pa_sgi/pa_sgi.c deleted file mode 100644 index fa2978b5..00000000 --- a/pd/portaudio_v18/pa_sgi/pa_sgi.c +++ /dev/null @@ -1,1069 +0,0 @@ -/* - * $Id: pa_sgi.c,v 1.2.4.2 2003/03/13 00:56:47 pieter Exp $ - * PortAudio Portable Real-Time Audio Library. Copyright (c) 1999-2001 Phil Burk. - * Latest Version at: http://www.portaudio.com - * - * Silicon Graphics (SGI) IRIX implementation by Pieter Suurmond. - * This implementation uses sproc()-spawning, not the POSIX-threads. - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * -MODIFICATIONS: - 8/12/2001 - Pieter Suurmond - took the v15 pa_linux_oss.c file and started to adapt for IRIX 6.2. - 8/17/2001 - v15, first unstable alpha release for IRIX, sent to Phil & Ross. - 9/23/2001 - Many fixes and changes: POLLIN for input, not POLLOUT! - 7/04/2002 - Implemented multiple user buffers per host buffer to allow clients that - request smaller buffersizes. - 3/13/2003 - Fixed clicks in full-duplex (wire) mode. Fixed some uninitialised vars, got rid of - all GCC-warnings (-Wall). Tested with MIPS compiler and GCC 3.0.4. on IRIX 6.5 (AL v7). -TODO: - - Dynamically switch to 32 bit float as native format when appropriate (let SGI do the conversion), - and maybe also the other natively supported formats? (might increase performance) - - Implement fancy callback block adapter as described in the PDF by Stephane Letz in the ASIO dir. - -REFERENCES: - - IRIX 6.2 man pages regarding SGI AL library. - - IRIS Digital Media Programming Guide (online books and man-pages come - with IRIX 6.2 and may not be publically available on the internet). -*/ - -#include <stdio.h> /* Standard libraries. */ -#include <stdlib.h> - -#include "../pa_common/portaudio.h" /* (Makefile fails to find in subdirs, -I doesn't work?). */ -#include "../pa_common/pa_host.h" -#include "../pa_common/pa_trace.h" - -#include <errno.h> /* Needed for int oserror(void);. */ -#include <sys/time.h> -#include <sys/types.h> -#include <sys/prctl.h> -#include <sys/schedctl.h> /* For schedctl(NDPRI, NDPHIMIN). */ -#include <fcntl.h> /* fcntl.h needed. */ -#include <unistd.h> /* For streams, ioctl(), etc. */ -#include <ulocks.h> -#include <poll.h> -#include <dmedia/audio.h> /* System specific (IRIX 6.2-6.5). */ - -/*----------------- MACROS --------------------*/ -#define PRINT(x) { printf x; fflush(stdout); } -#define ERR_RPT(x) PRINT(x) -#define DBUG(x) /* PRINT(x) */ -#define DBUGX(x) /* PRINT(x) */ - -#define MAX_CHARS_DEVNAME (16) -#define MAX_SAMPLE_RATES (8) /* Known from SGI AL there are 7. */ - /* Constants used in 'Pa_GetMinNumBuffers()' below: */ -#define MIN_LATENCY_MSEC (200) /* Used if 'getenv("PA_MIN_LATENCY_MSEC")' fails. */ -#define PA_LATENCY_ENV_NAME ("PA_MIN_LATENCY_MSEC") /* Same names as in file pa_unix.h. */ - -/*------------------------------- IRIX AL specific device info: --------------------------------------*/ -typedef struct internalPortAudioDevice -{ - PaDeviceID pad_DeviceID; /* THIS "ID" IS NEW HERE. */ - long pad_ALdevice; /* SGI-number! */ - double pad_SampleRates[MAX_SAMPLE_RATES]; /* For pointing to from pad_Info */ - char pad_DeviceName[MAX_CHARS_DEVNAME+1]; /* +1 for \0, one more than OSS. */ - PaDeviceInfo pad_Info; /* pad_Info (v15) contains: */ - struct internalPortAudioDevice* pad_Next; /* Singly linked list (NULL=end). */ -} internalPortAudioDevice; - -/*----------------- Structure containing all SGI IRIX specific data: ---------------------------------------*/ -typedef struct PaHostSoundControl -{ - ALconfig pahsc_ALconfigIN, /* IRIX-audio-library-datatype. Configuration */ - pahsc_ALconfigOUT; /* stucts separate for input and output ports. */ - ALport pahsc_ALportIN, /* IRIX-audio-library-datatype. ALports can only be */ - pahsc_ALportOUT; /* unidirectional, so we sometimes need 2 of them. */ - int pahsc_threadPID; /* Sproc()-result, written by PaHost_StartEngine(). */ - - unsigned int pahsc_UserBuffersPerHostBuffer, - pahsc_SamplesPerInputHostBuffer, /* Channels per frame are accounted for. */ - pahsc_SamplesPerOutputHostBuffer, - pahsc_BytesPerInputHostBuffer, /* Size per sample are accounted for. */ - pahsc_BytesPerOutputHostBuffer; - short *pahsc_InputHostBuffer, /* Allocated here, in this file, if necessary. */ - *pahsc_OutputHostBuffer; - - struct itimerval pahsc_EntryTime, /* For measuring CPU utilization (same as linux). */ - pahsc_LastExitTime; - long pahsc_InsideCountSum, - pahsc_TotalCountSum; -} PaHostSoundControl; - -/*-------------------------------------------------------- Shared Data -------------------------------*/ -static internalPortAudioDevice* sDeviceList = NULL; /* FIXME - put Mutex around this shared data. */ -static int sPaHostError = 0; /* Maybe more than one process writing errs!? */ -usema_t *SendSema, /* These variables are shared between the */ - *RcvSema; /* audio handling process and main process. */ -/*--------------------------*/ -long Pa_GetHostError(void) -{ - return (long)sPaHostError; -} - -/*----------------------------- BEGIN CPU UTILIZATION MEASUREMENT -----------------*/ -/* (copied from source pa_linux_oss/pa_linux_oss.c) */ -static void Pa_StartUsageCalculation( internalPortAudioStream *past ) -{ - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - if( pahsc == NULL ) return; - /* Query system timer for usage analysis and to prevent overuse of CPU. */ - getitimer( ITIMER_REAL, &pahsc->pahsc_EntryTime ); -} - -static long SubtractTime_AminusB( struct itimerval *timeA, struct itimerval *timeB ) -{ - long secs = timeA->it_value.tv_sec - timeB->it_value.tv_sec; - long usecs = secs * 1000000; - usecs += (timeA->it_value.tv_usec - timeB->it_value.tv_usec); - return usecs; -} - -static void Pa_EndUsageCalculation( internalPortAudioStream *past ) -{ - struct itimerval currentTime; - long insideCount; - long totalCount; /* Measure CPU utilization during this callback. */ - -#define LOWPASS_COEFFICIENT_0 (0.95) -#define LOWPASS_COEFFICIENT_1 (0.99999 - LOWPASS_COEFFICIENT_0) - - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - if (pahsc == NULL) - return; - if (getitimer( ITIMER_REAL, ¤tTime ) == 0 ) - { - if (past->past_IfLastExitValid) - { - insideCount = SubtractTime_AminusB( &pahsc->pahsc_EntryTime, ¤tTime ); - pahsc->pahsc_InsideCountSum += insideCount; - totalCount = SubtractTime_AminusB( &pahsc->pahsc_LastExitTime, ¤tTime ); - pahsc->pahsc_TotalCountSum += totalCount; - /* DBUG(("insideCount = %d, totalCount = %d\n", insideCount, totalCount )); */ - /* Low pass filter the result because sometimes we get called several times in a row. */ - /* That can cause the TotalCount to be very low which can cause the usage to appear */ - /* unnaturally high. So we must filter numerator and denominator separately!!! */ - if (pahsc->pahsc_InsideCountSum > 0) - { - past->past_AverageInsideCount = ((LOWPASS_COEFFICIENT_0 * past->past_AverageInsideCount) + - (LOWPASS_COEFFICIENT_1 * pahsc->pahsc_InsideCountSum)); - past->past_AverageTotalCount = ((LOWPASS_COEFFICIENT_0 * past->past_AverageTotalCount) + - (LOWPASS_COEFFICIENT_1 * pahsc->pahsc_TotalCountSum)); - past->past_Usage = past->past_AverageInsideCount / past->past_AverageTotalCount; - pahsc->pahsc_InsideCountSum = 0; - pahsc->pahsc_TotalCountSum = 0; - } - } - past->past_IfLastExitValid = 1; - } - pahsc->pahsc_LastExitTime.it_value.tv_sec = 100; - pahsc->pahsc_LastExitTime.it_value.tv_usec = 0; - setitimer( ITIMER_REAL, &pahsc->pahsc_LastExitTime, NULL ); - past->past_IfLastExitValid = 1; -} /*----------- END OF CPU UTILIZATION CODE (from pa_linux_oss/pa_linux_oss.c v15)--------------------*/ - - -/*--------------------------------------------------------------------------------------*/ -PaError translateSGIerror(void) /* Calls oserror(), may be used after an SGI AL-library */ -{ /* call to report via ERR_RPT(), yields a PaError-num. */ - const char* a = "SGI AL "; /* (Not absolutely sure errno came from THIS thread! */ - switch(oserror()) /* Read IRIX man-pages about the _SGI_MP_SOURCE macro.) */ - { - case AL_BAD_OUT_OF_MEM: - ERR_RPT(("%sout of memory.\n", a)); - return paInsufficientMemory; /* Known PaError. */ - case AL_BAD_CONFIG: - ERR_RPT(("%sconfiguration invalid or NULL.\n", a)); - return paHostError; /* Generic PaError. */ - case AL_BAD_CHANNELS: - ERR_RPT(("%schannels not 1,2 or 4.\n", a)); - return paHostError; /* Generic PaError. */ - case AL_BAD_NO_PORTS: - ERR_RPT(("%sout of audio ports.\n", a)); - return paHostError; /* Generic PaError. */ - case AL_BAD_DEVICE: - ERR_RPT(("%swrong device number.\n", a)); - return paHostError; /* Generic PaError. */ - case AL_BAD_DEVICE_ACCESS: - ERR_RPT(("%swrong device access.\n", a)); - return paHostError; /* Generic PaError. */ - case AL_BAD_DIRECTION: - ERR_RPT(("%sinvalid direction.\n", a)); - return paHostError; /* Generic PaError. */ - case AL_BAD_SAMPFMT: - ERR_RPT(("%sdoesn't accept sampleformat.\n", a)); - return paHostError; /* Generic PaError. */ - case AL_BAD_FLOATMAX: - ERR_RPT(("%smax float value is zero.\n", a)); - return paHostError; /* Generic PaError. */ - case AL_BAD_WIDTH: - ERR_RPT(("%sunsupported samplewidth.\n", a)); - return paHostError; /* Generic PaError. */ - case AL_BAD_QSIZE: - ERR_RPT(("%sinvalid queue size.\n", a)); - return paHostError; /* Generic PaError. */ - case AL_BAD_PVBUFFER: - ERR_RPT(("%sPVbuffer null.\n", a)); - return paHostError; /* Generic PaError. */ - case AL_BAD_BUFFERLENGTH_NEG: - ERR_RPT(("%snegative bufferlength.\n", a)); - return paHostError; /* Generic PaError. */ - case AL_BAD_BUFFERLENGTH_ODD: - ERR_RPT(("%sodd bufferlength.\n", a)); - return paHostError; /* Generic PaError. */ - case AL_BAD_PARAM: - ERR_RPT(("%sparameter not valid for device.\n", a)); - return paHostError; /* Generic PaError. */ - default: - ERR_RPT(("%sunknown error.\n", a)); - return paHostError; /* Generic PaError. */ - } -} - -/*------------------------------------------------------------------------------------------*/ -/* Tries to set various rates and formats and fill in the device info structure. */ -static PaError Pa_sgiQueryDevice(long ALdev, /* (AL_DEFAULT_DEVICE) */ - PaDeviceID id, /* (DefaultI|ODeviceID()) */ - char* name, /* (for example "SGI AL") */ - internalPortAudioDevice* pad) /* Result written to pad. */ -{ - long min, max; /* To catch hardware characteristics. */ - ALseterrorhandler(0); /* 0 = turn off the default error handler. */ - /*--------------------------------------------------------------------------------------*/ - pad->pad_ALdevice = ALdev; /* Set the AL device number. */ - pad->pad_DeviceID = id; /* Set the PA device number. */ - if (strlen(name) > MAX_CHARS_DEVNAME) /* MAX_CHARS defined above. */ - { - ERR_RPT(("Pa_QueryDevice(): name too long (%s).\n", name)); - return paHostError; - } - strcpy(pad->pad_DeviceName, name); /* Write name-string. */ - pad->pad_Info.name = pad->pad_DeviceName; /* Set pointer,..hmmm. */ - /*--------------------------------- natively supported sample formats: -----------------*/ - pad->pad_Info.nativeSampleFormats = paInt16; /* Later also include paFloat32 | ..| etc. */ - /* Then also choose other CallConvertXX()! */ - /*--------------------------------- number of available i/o channels: ------------------*/ - if (ALgetminmax(ALdev, AL_INPUT_COUNT, &min, &max)) - return translateSGIerror(); - pad->pad_Info.maxInputChannels = max; - DBUG(("Pa_QueryDevice: maxInputChannels = %d\n", pad->pad_Info.maxInputChannels)) - if (ALgetminmax(ALdev, AL_OUTPUT_COUNT, &min, &max)) - return translateSGIerror(); - pad->pad_Info.maxOutputChannels = max; - DBUG(("Pa_QueryDevice: maxOutputChannels = %d\n", pad->pad_Info.maxOutputChannels)) - /*--------------------------------- supported samplerates: ----------------------*/ - pad->pad_Info.numSampleRates = 7; - pad->pad_Info.sampleRates = pad->pad_SampleRates; - pad->pad_SampleRates[0] = (double)AL_RATE_8000; /* long -> double. */ - pad->pad_SampleRates[1] = (double)AL_RATE_11025; - pad->pad_SampleRates[2] = (double)AL_RATE_16000; - pad->pad_SampleRates[3] = (double)AL_RATE_22050; - pad->pad_SampleRates[4] = (double)AL_RATE_32000; - pad->pad_SampleRates[5] = (double)AL_RATE_44100; - pad->pad_SampleRates[6] = (double)AL_RATE_48000; - if (ALgetminmax(ALdev, AL_INPUT_RATE, &min, &max)) /* Ask INPUT rate-max. */ - return translateSGIerror(); /* double -> long. */ - if (max != (long)(0.5 + pad->pad_SampleRates[6])) /* FP-compare not recommndd. */ - goto weird; - if (ALgetminmax(ALdev, AL_OUTPUT_RATE, &min, &max)) /* Ask OUTPUT rate-max. */ - return translateSGIerror(); - if (max != (long)(0.5 + pad->pad_SampleRates[6])) - { -weird: ERR_RPT(("Pa_sgiQueryDevice() did not confirm max samplerate (%ld)\n",max)); - return paHostError; /* Or make it a warning and just carry on... */ - } - /*-------------------------------------------------------------------------------*/ - return paNoError; -} - - -/*--------------------------------------------------------------------------------*/ -int Pa_CountDevices() /* Name of this function suggests it only counts and */ -{ /* is NOT destructive, it however resets whole PA ! */ - int numDevices = 0; /* Let 's not do that here. */ - internalPortAudioDevice* currentDevice = sDeviceList; /* COPY GLOBAL VAR. */ -#if 0 /* Remains from linux_oss v15: Pa_Initialize(), on */ - if (!currentDevice) /* its turn, calls PaHost_Init() via file pa_lib.c. */ - Pa_Initialize(); /* Isn't that a bit too 'rude'? Don't be too */ -#endif /* friendly to clients that forgot to initialize PA. */ - while (currentDevice) /* Slower but more elegant than the sNumDevices-way: */ - { - numDevices++; - currentDevice = currentDevice->pad_Next; - } - return numDevices; -} - -/*-------------------------------------------------------------------------------*/ -static internalPortAudioDevice *Pa_GetInternalDevice(PaDeviceID id) -{ - int numDevices = 0; - internalPortAudioDevice *res = (internalPortAudioDevice*)NULL; - internalPortAudioDevice *pad = sDeviceList; /* COPY GLOBAL VAR. */ - while (pad) /* pad may be NULL, that's ok, return 0. */ - { /* (Added ->pad_DeviceID field to the pad-struct, Pieter, 2001.) */ - if (pad->pad_DeviceID == id) /* This the device we were looking for? */ - res = pad; /* But keep on(!) counting so we don't */ - numDevices++; /* have to call Pa_CountDevices() later. */ - pad = pad->pad_Next; /* Advance to the next device or NULL. */ - } /* No assumptions about order of ID's in */ - if (!res) /* the list. */ - ERR_RPT(("Pa_GetInternalDevice() could not find specified ID (%d).\n",id)); - if ((id < 0) || (id >= numDevices)) - { - ERR_RPT(("Pa_GetInternalDevice() supplied with an illegal ID (%d).\n",id)); -#if 1 /* Be strict, even when found, */ - res = (internalPortAudioDevice*)NULL; /* do not accept illegal ID's. */ -#endif - } - return res; -} - -/*----------------------------------------------------------------------*/ -const PaDeviceInfo* Pa_GetDeviceInfo(PaDeviceID id) -{ - PaDeviceInfo* res = (PaDeviceInfo*)NULL; - internalPortAudioDevice* pad = Pa_GetInternalDevice(id); /* Call. */ - if (pad) - res = &pad->pad_Info; /* Not finding the specified ID is not */ - if (!res) /* the same as &pad->pad_Info == NULL. */ - ERR_RPT(("Pa_GetDeviceInfo() could not find it (ID=%d).\n", id)); - return res; /* So (maybe) a second/third ERR_RPT(). */ -} - -/*------------------------------------------------*/ -PaDeviceID Pa_GetDefaultInputDeviceID(void) -{ - return 0; /* 0 is the default device ID. */ -} -/*------------------------------------------------*/ -PaDeviceID Pa_GetDefaultOutputDeviceID(void) -{ - return 0; -} - -/*-------------------------------------------------------------------------------------------------*/ -/* Build linked a list with all the available audio devices on this SGI machine (only 1 for now). */ -PaError PaHost_Init(void) /* Called by Pa_Initialize() from pa_lib.c. */ -{ - internalPortAudioDevice* pad; - PaError r = paNoError; - int audioLibFileID; /* To test for the presence of audio. */ - - if (sDeviceList) /* Allow re-init, only warn, no error. */ - { - ERR_RPT(("Warning: PaHost_Init() did not really re-init PA.\n")); - return r; - } - /*------------- ADD THE SGI DEFAULT DEVICE TO THE LIST: ---------------------------------------*/ - audioLibFileID = open("/dev/hdsp/hdsp0master", O_RDONLY); /* Try to open Indigo style audio */ - if (audioLibFileID < 0) /* IO port. On failure, machine */ - { /* has no audio ability. */ - ERR_RPT(("PaHost_Init(): This machine has no (Indigo-style) audio abilities.\n")); - return paHostError; - } - close(audioLibFileID); /* Allocate fast mem to hold device info. */ - pad = PaHost_AllocateFastMemory(sizeof(internalPortAudioDevice)); - if (pad == NULL) - return paInsufficientMemory; - memset(pad, 0, sizeof(internalPortAudioDevice)); /* "pad->pad_Next = NULL" is more elegant. */ - r = Pa_sgiQueryDevice(AL_DEFAULT_DEVICE, /* Set AL device num (AL_DEFAULT_DEVICE). */ - Pa_GetDefaultOutputDeviceID(),/* Set PA device num (or InputDeviceID()). */ - "AL default", /* A suitable name. */ - pad); /* Write args and queried info into pad. */ - if (r != paNoError) - { - ERR_RPT(("Pa_QueryDevice for '%s' returned: %d\n", pad->pad_DeviceName, r)); - PaHost_FreeFastMemory(pad, sizeof(internalPortAudioDevice)); /* sDeviceList still NULL ! */ - } - else - sDeviceList = pad; /* First element in linked list. pad->pad_Next already NULL. */ - /*------------- QUERY AND ADD MORE POSSIBLE SGI DEVICES TO THE LINKED LIST: -------------------*/ - /*---------------------------------------------------------------------------------------------*/ - return r; -} - -/*--------------------------------------------------------------------------------------------*/ -#define MIN(a,b) ((a)<(b)?(a):(b)) /* MIN()-function is used below. */ -#define kPollSEMA 0 /* To index the pollfd-array, reads nicer than just */ -#define kPollOUT 1 /* numbers. */ -#define kPollIN 2 -void Pa_SgiAudioProcess(void *v) /* This function is sproc-ed by PaHost_StartEngine() */ -{ /* as a separate thread. (Argument must be void*). */ - short evtLoop; /* Reset by parent indirectly, or at local errors. */ - PaError result; - struct pollfd PollFD[3]; /* To catch kPollSEMA-, kPollOUT- and kPollIN-events. */ - internalPortAudioStream *past = (internalPortAudioStream*)v; /* Copy void-ptr-argument. */ - PaHostSoundControl *pahsc; - short n, inputEvent, outputEvent, ioEvent, semaEvent = 0; - short *inBuffer, *outBuffer; /* Only 16 bit for now, may change... */ - unsigned int samplesPerInputUserBuffer, samplesPerOutputUserBuffer; - - DBUG(("Entering sproc-thread.\n")); - if (!past) - { - sPaHostError = paInternalError; /* Or paBadStreamPtr ? */ - ERR_RPT(("argument NULL!\n")); - goto noPast; - } - pahsc = (PaHostSoundControl*)past->past_DeviceData; - if (!pahsc) - { - sPaHostError = paInternalError; /* The only way is to signal error to shared area?! */ - ERR_RPT(("past_DeviceData NULL!\n")); - goto noPahsc; /* Sproc-ed threads MAY NOT RETURN paInternalError. */ - } - /*----------------------------- open AL-ports here, after sproc(): -----------------------*/ - if (past->past_NumInputChannels > 0) /* Open input port. */ - { - pahsc->pahsc_ALportIN = ALopenport("PA sgi in", "r", pahsc->pahsc_ALconfigIN); - if (!pahsc->pahsc_ALportIN) - { - ERR_RPT(("Failed to open AL input port.\n")); - sPaHostError = paInternalError; - goto skip; - } - DBUG(("Opened %d input channel(s).\n", past->past_NumInputChannels)); - samplesPerInputUserBuffer = pahsc->pahsc_SamplesPerInputHostBuffer / - pahsc->pahsc_UserBuffersPerHostBuffer; - } - else - samplesPerInputUserBuffer = 0; /* Added 2003. */ - if (past->past_NumOutputChannels > 0) /* Open output port. */ - { - pahsc->pahsc_ALportOUT = ALopenport("PA sgi out", "w", pahsc->pahsc_ALconfigOUT); - if (!pahsc->pahsc_ALportOUT) - { - ERR_RPT(("Failed to open AL output port.\n")); - sPaHostError = paInternalError; /* Assume pahsc_ALconfigs are the */ - goto skip; /* same for IN and OUT in case */ - } /* both ports are opened (bidir). */ - DBUG(("Opened %d output channel(s).\n", past->past_NumOutputChannels)); - samplesPerOutputUserBuffer = pahsc->pahsc_SamplesPerOutputHostBuffer / - pahsc->pahsc_UserBuffersPerHostBuffer; - DBUG(("samplesPerOutputUserBuffer = %d\n", samplesPerOutputUserBuffer)); - } - else - samplesPerOutputUserBuffer = 0; /* Added 2003. */ - /*-----------------------------------------------------------------------*/ - past->past_IsActive = 1; /* Wasn't this already done by the calling parent?! */ - PollFD[kPollIN].fd = ALgetfd(pahsc->pahsc_ALportIN); /* ALgetfd returns -1 on failures */ - PollFD[kPollIN].events = POLLIN; /* such as ALport not there. */ - PollFD[kPollOUT].fd = ALgetfd(pahsc->pahsc_ALportOUT); - PollFD[kPollOUT].events = POLLOUT; /* .events = POLLOUT is OK. */ - schedctl(NDPRI, NDPHIMIN); /* Sets non-degrading priority for this process. */ - PollFD[kPollSEMA].fd = usopenpollsema(SendSema, 0777); /* To communicate with parent. */ - PollFD[kPollSEMA].events = POLLIN; /* .events = POLLIN is OK. */ - uspsema(SendSema); /* Blocks until ... MUST be here, this uspsema(). */ - evtLoop = ((past->past_StopNow | past->past_StopSoon) == 0); - while (evtLoop) - { - /*----------------------------- SET FILLPOINTS AND WAIT UNTIL SOMETHING HAPPENS: ---------*/ - if (pahsc->pahsc_InputHostBuffer) /* Then pahsc_ALportIN should also be there. */ - { - /* For input port, fill point is number of locations in the sample queue that must be */ - /* filled in order to trigger a return from select(). (or poll()) */ - /* Notice IRIX docs mention number of samples as argument, not number of sampleframes.*/ - if (ALsetfillpoint(pahsc->pahsc_ALportIN, pahsc->pahsc_SamplesPerInputHostBuffer)) - { /* Multiple amount as transferred per time. */ - ERR_RPT(("ALsetfillpoint() for ALportIN failed.\n")); - sPaHostError = paInternalError; /* (Using exit(-1) would be a bit rude.) */ - goto skip; - } - } - /* 'else' added march 2003: set only one of both fillpoints: input or output. When */ - /* setting both fillpoints (as in earlier version) clicks occur at full duplex-mode. */ - else if (pahsc->pahsc_OutputHostBuffer) /* Then pahsc_ALportOUT should also be there. */ - { - /* For output port, fill point is number of locations that must be free in order to */ - /* wake up from select(). (or poll()) */ - if (ALsetfillpoint(pahsc->pahsc_ALportOUT, pahsc->pahsc_SamplesPerOutputHostBuffer)) - { - ERR_RPT(("ALsetfillpoint() for ALportOUT failed.\n")); - sPaHostError = paInternalError; - goto skip; - } - } /* poll() with timeout=-1 makes it block until a requested */ - poll(PollFD, 3, -1); /* event occurs or until call is interrupted. If fd-value in */ - /* array <0, events is ignored and revents is set to 0. */ - /*---------------------------- MESSAGE-EVENT FROM PARENT THREAD: -------------------------*/ - semaEvent = PollFD[kPollSEMA].revents & POLLIN; - if (semaEvent) - { - if (past->past_StopSoon) - evtLoop = 0; - if (past->past_StopNow) - goto skip; - } - /*------------------------------------- FILLED-EVENT FROM INPUT BUFFER: --------------------------*/ - inputEvent = PollFD[kPollIN].revents & POLLIN; - if (inputEvent) /* Don't need to check (pahsc->pahsc_InputHostBuffer): */ - { /* if buffer was not there, ALport not there, no events! */ - if (ALreadsamps(pahsc->pahsc_ALportIN, (void*)pahsc->pahsc_InputHostBuffer, - pahsc->pahsc_SamplesPerInputHostBuffer)) - { /* Here again: number of samples instead of number of frames. */ - ERR_RPT(("ALreadsamps() failed.\n")); - sPaHostError = paInternalError; - goto skip; - } - } - outputEvent = PollFD[kPollOUT].revents & POLLOUT; - ioEvent = (inputEvent | outputEvent); /* Binary or is ok. */ - /*------------------------------------- USER-CALLBACK-ROUTINE: -----------------------------------*/ - if (ioEvent) /* Always true? Or can some other system-event awaken the */ - { /* poll? Sure it wasn't just a "sema"- (i.e. user)-event? */ - Pa_StartUsageCalculation(past); /* Convert 16 bit native data to */ - /* user data and call user routine. */ - inBuffer = pahsc->pahsc_InputHostBuffer; /* Short pointers for now, care! */ - outBuffer = pahsc->pahsc_OutputHostBuffer; - n = pahsc->pahsc_UserBuffersPerHostBuffer; /* 'n' may never start at NULL ! */ - do { - result = Pa_CallConvertInt16(past, inBuffer, outBuffer); - if (result) /* This is apparently NOT an error! Just letting the userCallBack stop us. */ - { DBUG(("Pa_CallConvertInt16() returned %d, stopping...\n", result)); goto skip; } - inBuffer += samplesPerInputUserBuffer; /* Num channels is accounted for. */ - outBuffer += samplesPerOutputUserBuffer; - } while (--n); - Pa_EndUsageCalculation(past); - } - /*------------------------------------ FREE-EVENT FROM OUTPUT BUFFER: ---------------------------*/ - if (pahsc->pahsc_OutputHostBuffer && ioEvent) /* Don't wait for outputEvent solely (that may cause clicks). */ - { /* Just assume it's time to write, outputEvent may not yet be there. */ - if (ALwritesamps(pahsc->pahsc_ALportOUT, (void*)pahsc->pahsc_OutputHostBuffer, - pahsc->pahsc_SamplesPerOutputHostBuffer)) - { - ERR_RPT(("ALwritesamps() failed.\n")); /* Better use SEMAS for messaging back to parent! */ - sPaHostError = paInternalError; - goto skip; - } - } - } -skip: - /*------------------------------- close AL-ports ----------------------------*/ - if (pahsc->pahsc_ALportIN) - { - if (ALcloseport(pahsc->pahsc_ALportIN)) - translateSGIerror(); /* Translates SGI AL-code to PA-code and ERR_RPTs string. */ - else /* But go on anyway... to release other stuff... */ - pahsc->pahsc_ALportIN = (ALport)0; - } - if (pahsc->pahsc_ALportOUT) - { - if (ALcloseport(pahsc->pahsc_ALportOUT)) - translateSGIerror(); - else - pahsc->pahsc_ALportOUT = (ALport)0; - } -noPahsc: - past->past_IsActive = 0; - if (semaEvent) - { - uspsema(SendSema); /* StopEngine() was still waiting for this acknowledgement. */ - usvsema(RcvSema); /* (semaEvent initialized with 0.) */ - } -noPast: - DBUG(("Leaving sproc-thread.\n")); -} - - -/*--------------------------------------------------------------------------------------*/ -PaError PaHost_OpenStream(internalPortAudioStream *past) -{ - PaError result = paNoError; - PaHostSoundControl *pahsc; - unsigned int minNumBuffers; - internalPortAudioDevice *padIN, *padOUT; /* For looking up native AL-numbers. */ - long pvbuf[8], sr, alq; /* To get/set hardware configs. */ - - DBUG(("PaHost_OpenStream() called.\n")); /* Alloc FASTMEM and init host data. */ - if (!past) - { - ERR_RPT(("Streampointer NULL!\n")); - result = paBadStreamPtr; goto done; - } - pahsc = (PaHostSoundControl*)PaHost_AllocateFastMemory(sizeof(PaHostSoundControl)); - if (pahsc == NULL) - { - ERR_RPT(("FAST Memory allocation failed.\n")); /* Pass trough some ERR_RPT-exit- */ - result = paInsufficientMemory; goto done; /* code (nothing will be freed). */ - } - memset(pahsc, 0, sizeof(PaHostSoundControl)); - pahsc->pahsc_threadPID = -1; /* Should pahsc_threadPID be inited to */ - past->past_DeviceData = (void*)pahsc; /* -1 instead of 0 ?? */ - /*------------------------------------------ Manipulate hardware if necessary and allowed: --*/ - ALseterrorhandler(0); /* 0 = turn off the default error handler. */ - pvbuf[0] = AL_INPUT_RATE; - pvbuf[2] = AL_INPUT_COUNT; - pvbuf[4] = AL_OUTPUT_RATE; /* TO FIX: rates may be logically, not always in Hz! */ - pvbuf[6] = AL_OUTPUT_COUNT; - sr = (long)(past->past_SampleRate + 0.5); /* Common for both input and output :-) */ - /*-----------------------------------------------------------------------------*/ - /* OVERWRITE 'past_NumUserBuffers'-field in the struct supplied by the caller. */ - /* This field may be set to zero by a client application to ask for minimum */ - /* latency. It is used below, to set both input- and output-AL-queuesizes. */ - minNumBuffers = Pa_GetMinNumBuffers(past->past_FramesPerUserBuffer, - past->past_SampleRate); /* Take biggest. */ - past->past_NumUserBuffers = (minNumBuffers > past->past_NumUserBuffers) ? - minNumBuffers : past->past_NumUserBuffers; - DBUG(("past->past_NumUserBuffers=%d\n", past->past_NumUserBuffers)); - /*----------------------------------------------------------------------------------*/ - pahsc->pahsc_UserBuffersPerHostBuffer = past->past_NumUserBuffers >> 1; - DBUG(("pahsc_UserBuffersPerHostBuffer=%d\n",pahsc->pahsc_UserBuffersPerHostBuffer)); - /* 1 is minimum because Pa_GetMinNumBuffers() returns >= 2. - Callback will be called 'pahsc_UserBuffersPerHostBuffer' times (with 'past_FramesPerUserBuffer') - per host transfer. */ - /*---------------------------------------------------- SET INPUT CONFIGURATION: ---------------------*/ - if (past->past_NumInputChannels > 0) /* We need to lookup the corre- */ - { /* sponding native AL-number(s). */ - /*--------------------------------------------------- Allocate native buffers: --------------*/ - pahsc->pahsc_SamplesPerInputHostBuffer = pahsc->pahsc_UserBuffersPerHostBuffer * - past->past_FramesPerUserBuffer * /* Needed by the */ - past->past_NumInputChannels; /* audio-thread. */ - DBUG(("pahsc_SamplesPerInputHostBuffer=%d\n", pahsc->pahsc_SamplesPerInputHostBuffer)); - pahsc->pahsc_BytesPerInputHostBuffer = pahsc->pahsc_SamplesPerInputHostBuffer * sizeof(short); - pahsc->pahsc_InputHostBuffer = (short*)PaHost_AllocateFastMemory(pahsc->pahsc_BytesPerInputHostBuffer); - if (!pahsc->pahsc_InputHostBuffer) - { - ERR_RPT(("Fast memory allocation failed (in).\n")); - result = paInsufficientMemory; - goto done; - } - padIN = Pa_GetInternalDevice(past->past_InputDeviceID); - if (!padIN) - { - ERR_RPT(("Pa_GetInternalDevice() for input failed.\n")); - result = paHostError; - goto done; - } - if (ALgetparams(padIN->pad_ALdevice, &pvbuf[0], 4)) /* Although input and output will both be on */ - goto sgiError; /* the same AL-device, the AL-library might */ - if (pvbuf[1] != sr) /* contain more than AL_DEFAULT_DEVICE in */ - { /* Rate different from current harware-rate? the future. Therefore 2 seperate queries. */ - if (pvbuf[3] > 0) /* Means, there's other clients using AL-input-ports */ - { - ERR_RPT(("Sorry, not allowed to switch input-hardware to %ld Hz because \ -another process is currently using input at %ld Hz.\n", sr, pvbuf[1])); - result = paHostError; - goto done; - } - pvbuf[1] = sr; /* Then set input-rate. */ - if (ALsetparams(padIN->pad_ALdevice, &pvbuf[0], 2)) - goto sgiError; /* WHETHER THIS SAMPLERATE WAS REALLY PRESENT IN OUR ARRAY OF RATES, */ - } /* IS NOT CHECKED, AT LEAST NOT BY ME, WITHIN THIS FILE! Does PA do? */ - pahsc->pahsc_ALconfigIN = ALnewconfig(); /* Released at PaHost_CloseStream(). */ - if (pahsc->pahsc_ALconfigIN == (ALconfig)0) - goto sgiError; - if (ALsetsampfmt(pahsc->pahsc_ALconfigIN, AL_SAMPFMT_TWOSCOMP))/* Choose paInt16 as native i/o-format. */ - goto sgiError; - if (ALsetwidth (pahsc->pahsc_ALconfigIN, AL_SAMPLE_16)) /* Only meaningful when sample format for */ - goto sgiError; /* config is set to two's complement format. */ - /************************ Future versions might (dynamically) switch to 32-bit floats? ******* - if (ALsetsampfmt(pahsc_ALconfigIN, AL_SAMPFMT_FLOAT)) (Then also call another CallConvert-func.) - goto sgiError; - if (ALsetfloatmax (pahsc_ALconfigIN, 1.0)) Only meaningful when sample format for config - goto sgiError; is set to AL_SAMPFMT_FLOAT or AL_SAMPFMT_DOUBLE. */ - /*--------- Set internal AL queuesize (in samples, not in frames!) -------------------------------*/ - alq = (long)past->past_NumUserBuffers * past->past_FramesPerUserBuffer * past->past_NumInputChannels; - DBUG(("AL input queuesize = %ld samples.\n", alq)); - if (ALsetqueuesize(pahsc->pahsc_ALconfigIN, alq)) - goto sgiError; - if (ALsetchannels (pahsc->pahsc_ALconfigIN, (long)(past->past_NumInputChannels))) - goto sgiError; /* Returns 0 on success, -1 on failure. */ - } - else - pahsc->pahsc_InputHostBuffer = (short*)NULL; /* Added 2003! Is checked in callback-routine. */ - /*---------------------------------------------------- SET OUTPUT CONFIGURATION: ------------------------*/ - if (past->past_NumOutputChannels > 0) /* CARE: padOUT/IN may NOT be NULL if Channels <= 0! */ - { /* We use padOUT/IN later on, or at least 1 of both. */ - pahsc->pahsc_SamplesPerOutputHostBuffer = pahsc->pahsc_UserBuffersPerHostBuffer * - past->past_FramesPerUserBuffer * /* Needed by the */ - past->past_NumOutputChannels; /* audio-thread. */ - DBUG(("pahsc_SamplesPerOutputHostBuffer=%d\n", pahsc->pahsc_SamplesPerOutputHostBuffer)); - pahsc->pahsc_BytesPerOutputHostBuffer = pahsc->pahsc_SamplesPerOutputHostBuffer * sizeof(short); - pahsc->pahsc_OutputHostBuffer = (short*)PaHost_AllocateFastMemory(pahsc->pahsc_BytesPerOutputHostBuffer); - if (!pahsc->pahsc_OutputHostBuffer) - { - ERR_RPT(("Fast memory allocation failed (out).\n")); - result = paInsufficientMemory; - goto done; - } - padOUT = Pa_GetInternalDevice(past->past_OutputDeviceID); - if (!padOUT) - { - ERR_RPT(("Pa_GetInternalDevice() for output failed.\n")); - result = paHostError; - goto done; - } - if (ALgetparams(padOUT->pad_ALdevice,&pvbuf[4], 4)) - goto sgiError; - if (pvbuf[5] != sr) - { /* Output needed and rate different from current harware-rate. */ - if (pvbuf[7] > 0) /* Means, there's other clients using AL-output-ports */ - { - ERR_RPT(("Sorry, not allowed to switch output-hardware to %ld Hz because \ -another process is currently using output at %ld Hz.\n", sr, pvbuf[5])); - result = paHostError; - goto done; /* Will free again the inputbuffer */ - } /* that was just created above. */ - pvbuf[5] = sr; /* Then set output-rate. */ - if (ALsetparams(padOUT->pad_ALdevice, &pvbuf[4], 2)) - goto sgiError; - } - pahsc->pahsc_ALconfigOUT = ALnewconfig(); /* Released at PaHost_CloseStream(). */ - if (pahsc->pahsc_ALconfigOUT == (ALconfig)0) - goto sgiError; - if (ALsetsampfmt(pahsc->pahsc_ALconfigOUT, AL_SAMPFMT_TWOSCOMP)) /* Choose paInt16 as native i/o-format. */ - goto sgiError; - if (ALsetwidth (pahsc->pahsc_ALconfigOUT, AL_SAMPLE_16)) /* Only meaningful when sample format for */ - goto sgiError; /* config is set to two's complement format. */ - /** Future versions might (dynamically) switch to 32-bit floats. **/ - alq = (long)past->past_NumUserBuffers * past->past_FramesPerUserBuffer * past->past_NumOutputChannels; - DBUG(("AL output queuesize = %ld samples.\n", alq)); - if (ALsetqueuesize(pahsc->pahsc_ALconfigOUT, alq)) - goto sgiError; - if (ALsetchannels (pahsc->pahsc_ALconfigOUT, (long)(past->past_NumOutputChannels))) - goto sgiError; - } - else - pahsc->pahsc_OutputHostBuffer = (short*)NULL; - /*----------------------------------------------- TEST DEVICE ID's: --------------------*/ - if ((past->past_OutputDeviceID != past->past_InputDeviceID) && /* Who SETS these devive-numbers? */ - (past->past_NumOutputChannels > 0) && (past->past_NumInputChannels > 0)) - { - ERR_RPT(("Cannot setup bidirectional stream between different devices.\n")); - result = paHostError; - goto done; - } - goto done; /* (no errors occured) */ -sgiError: - result = translateSGIerror(); /* Translates SGI AL-code to PA-code and ERR_RPTs string. */ -done: - if (result != paNoError) - PaHost_CloseStream(past); /* Frees memory (only if really allocated!). */ - return result; -} - -/*-----------------------------------------------------*/ -PaError PaHost_StartOutput(internalPortAudioStream *past) -{ - return paNoError; /* Hmm, not implemented yet? */ -} -PaError PaHost_StartInput(internalPortAudioStream *past) -{ - return paNoError; -} - -/*------------------------------------------------------------------------------*/ -PaError PaHost_StartEngine(internalPortAudioStream *past) -{ - PaHostSoundControl *pahsc; - usptr_t *arena; - if (!past) /* Test argument. */ - { - ERR_RPT(("PaHost_StartEngine(NULL)!\n")); - return paBadStreamPtr; - } - pahsc = (PaHostSoundControl*)past->past_DeviceData; - if (!pahsc) - { - ERR_RPT(("PaHost_StartEngine(arg): arg->past_DeviceData = NULL!\n")); - return paHostError; - } - past->past_StopSoon = 0; /* Assume SGI ALport is already opened! */ - past->past_StopNow = 0; /* Why don't we check pahsc for NULL? */ - past->past_IsActive = 1; - - /* Although the pthread_create() function, as well as <pthread.h>, may be */ - /* available in IRIX, use sproc() on SGI to create audio-background-thread. */ - /* (Linux/oss uses pthread_create() instead of __clone() because: */ - /* - pthread_create also works for other UNIX systems like Solaris, */ - /* - Java HotSpot VM crashes in pthread_setcanceltype() using __clone().) */ - - usconfig(CONF_ARENATYPE, US_SHAREDONLY); /* (From SGI-AL-examples, file */ - arena = usinit(tmpnam(0)); /* motifexample.c, function */ - SendSema = usnewpollsema(arena, 0); /* InitializeAudioProcess().) */ - RcvSema = usnewsema(arena, 1); /* 1= common mutual exclusion semaphore, where 1 and only 1 process - will be permitted through a semaphore at a time. Values > 1 - imply that up to val resources may be simultaneously used, but requests - for more than val resources cause the calling process to block until a - resource comes free (by a process holding a resource performing a - usvsema(). IS THIS usnewsema() TOO PLATFORM SPECIFIC? */ - prctl(PR_SETEXITSIG, 0); /* No not (void*)9, but 0, which doesn't kill the parent! */ - /* PR_SETEXITSIG controls whether all members of a share group will be - signaled if any one of them leaves the share group (either via exit() - or exec()). If 2nd arg, interpreted as an int is 0, then normal IRIX - process termination rules apply, namely that the parent is sent a - SIGCLD upon death of child, but no indication of death of parent is - given. If the second argument is a valid signal number then if any - member of a share group leaves the share group, a signal is - sent to ALL surviving members of the share group. */ - /* SPAWN AUDIO-CHILD: */ - pahsc->pahsc_threadPID = sproc(Pa_SgiAudioProcess, /* Returns process ID of */ - PR_SALL, /* new process, or -1. */ - (void*)past); /* Pass past as optional */ /* IS THIS SAFE, will past never */ - if (pahsc->pahsc_threadPID == -1) /* third void-ptr-arg. */ /* be moved around in memory???? */ - { - ERR_RPT(("PaHost_StartEngine() failed to spawn audio-thread.\n")); - sPaHostError = oserror(); /* Pass native error-number to shared area. */ - return paHostError; /* But return the generic error-number. */ - } - return paNoError; /* Hmmm, errno may come from other threads in same group! */ -} /* ("man sproc" in IRIX6.2 to read about _SGI_MP_SOURCE.) */ - -/*------------------------------------------------------------------------------*/ -PaError PaHost_StopEngine(internalPortAudioStream *past, int abort) -{ - PaError result = paNoError; - PaHostSoundControl *pahsc; - - DBUG(("PaHost_StopEngine() called.\n")); - if (!past) - return paBadStreamPtr; - pahsc = (PaHostSoundControl*)past->past_DeviceData; - /* Prevent from doing this twice!! */ - if ((!pahsc) || /* Some tests call this CLOSE twice!! */ - (!past->past_IsActive) || - past->past_StopSoon || past->past_StopNow) - return result; /* paNoError (already stopped, no err?). */ - past->past_StopSoon = 1; /* Tell background thread to stop generating */ - if (abort) /* more and to let current data play out. If */ - past->past_StopNow = 1; /* aborting, tell backgrnd thread to stop NOW! */ - /*---- USE SEMAPHORE LOCK TO COMMUNICATE: -----*/ - usvsema(SendSema); /* Increments count associated with SendSema. */ - /* Wait for the response. */ - uspsema(RcvSema); /* Decrements count of previously allocated */ - /* semaphore specified by RcvSema. */ - while (past->past_IsActive) /* REALLY WAIT. */ - { - /* DBUG(("wait 1 ms for audio-thread to stop.\n")); */ - Pa_Sleep(1); - } - -#if 0 /* We don't need to KILL(), just COMMUNICATE and be patient... */ - if (pahsc->pahsc_threadPID != -1) /* Did we really init it to -1 somewhere? */ - { - DBUG(("PaHost_StopEngine() is about to kill(SIGKILL) audio-thread.\n")); - if (kill(pahsc->pahsc_threadPID, SIGKILL)) /* Or SIGTERM or SIGQUIT(core) */ - { /* Returns -1 in case of error. */ - result = paHostError; - sPaHostError = oserror(); /* Hmmm, other threads may also write here! */ - ERR_RPT(("PaHost_StopEngine() failed to kill audio-thread.\n")); - } - else - pahsc->pahsc_threadPID = -1; /* Notify that we've killed this thread. */ - } -#endif - past->past_IsActive = 0; /* Even when kill() failed and pahsc_threadPID still there??? */ - return result; -} - -/*---------------------------------------------------------------*/ -PaError PaHost_StopOutput(internalPortAudioStream *past, int abort) -{ - return paNoError; /* Not implemented yet? */ -} -PaError PaHost_StopInput(internalPortAudioStream *past, int abort ) -{ - return paNoError; -} - -/*******************************************************************/ -PaError PaHost_CloseStream(internalPortAudioStream *past) -{ - PaHostSoundControl *pahsc; - PaError result = paNoError; - - DBUG(("PaHost_CloseStream() called.\n")); - if (!past) - return paBadStreamPtr; - pahsc = (PaHostSoundControl *) past->past_DeviceData; - if (!pahsc) /* If pahsc not NULL, past_DeviceData will be freed, and set to NULL. */ - return result; /* This test prevents from freeing NULL-pointers. */ - - if (pahsc->pahsc_ALconfigIN) - { /* Release configuration structs, only if allocated. */ - ALfreeconfig(pahsc->pahsc_ALconfigIN); - pahsc->pahsc_ALconfigIN = NULL; - } - if (pahsc->pahsc_ALconfigOUT) - { - ALfreeconfig(pahsc->pahsc_ALconfigOUT); /* (Al-ports were already closed by audioProcess). */ - pahsc->pahsc_ALconfigOUT = NULL; - } - if (pahsc->pahsc_InputHostBuffer) - { - PaHost_FreeFastMemory(pahsc->pahsc_InputHostBuffer, pahsc->pahsc_BytesPerInputHostBuffer); - pahsc->pahsc_InputHostBuffer = NULL; - } - if (pahsc->pahsc_OutputHostBuffer) - { - PaHost_FreeFastMemory(pahsc->pahsc_OutputHostBuffer, pahsc->pahsc_BytesPerOutputHostBuffer); - pahsc->pahsc_OutputHostBuffer = NULL; - } - PaHost_FreeFastMemory(pahsc, sizeof(PaHostSoundControl)); - past->past_DeviceData = NULL; /* PaHost_OpenStream() allocated FAST MEM. */ - return result; -} - - -/*------------------------------------------------------------------------*/ -/* Determine minimum number of buffers required for (SGI) host based on */ -/* minimum latency. Latency can be optionally set by user by setting an */ -/* environment variable. For example, to set my latency to 200 msec, I've */ -/* put this line in my '.cshrc' file: setenv PA_MIN_LATENCY_MSEC 200 */ -/* It always calls the 'PRINT' macro. */ -/* The minimum number that is returned is 2. */ -/* This number is directly proportional to the AL-queue sizes to set up. */ -/* It is one more than the number of user buffers per host buffer - in */ -/* case minimum is returned, or, twice the user buffers per host buffer. */ -/*------------------------------------------------------------------------*/ -int Pa_GetMinNumBuffers(int framesPerUserBuffer, double framesPerSecond) -{ - int minBuffers, minLatencyMsec; - char *minLatencyText; - double actualLatency; - - minLatencyText = getenv(PA_LATENCY_ENV_NAME); /* Defined at top of file. */ - if (minLatencyText) - { - minLatencyMsec = atoi(minLatencyText); - if (minLatencyMsec < 10) - { /* 10 is the minimum. */ - minLatencyMsec = 10; - PRINT (("Environment variable 'PA_MIN_LATENCY_MSEC' below minimum of %d milliseconds.\n", - minLatencyMsec)); - } - else if (minLatencyMsec > 4000) - { /* 4000 is the maximum. */ - minLatencyMsec = 4000; - PRINT (("Environment variable 'PA_MIN_LATENCY_MSEC' above maximum of %d milliseconds.\n", - minLatencyMsec)); - } - else - PRINT (("Using environment variable 'PA_MIN_LATENCY_MSEC' (set to %d milliseconds).\n", - minLatencyMsec)); - } - else - { - minLatencyMsec = MIN_LATENCY_MSEC; /* Defined at top of this file. */ - PRINT (("Environment variable 'PA_MIN_LATENCY_MSEC' not found.\nUsing default of %d milliseconds\n", - minLatencyMsec)); - } - minBuffers = (int)((minLatencyMsec * framesPerSecond) / - (1000.0 * framesPerUserBuffer)); - if (minBuffers < 2) - minBuffers = 2; - actualLatency = 1000.0 * minBuffers * framesPerUserBuffer / framesPerSecond; - PRINT (("Actual AL latency set to %.2f milliseconds\n", actualLatency)); - return minBuffers; -} - -/*---------------------------------------------------------------------*/ -PaError PaHost_Term(void) /* Frees all of the linked audio-devices. */ -{ /* Called by Pa_Terminate() from pa_lib.c. */ - internalPortAudioDevice *pad = sDeviceList, - *nxt; - while (pad) - { - DBUG(("PaHost_Term: freeing %s\n", pad->pad_DeviceName)); - nxt = pad->pad_Next; - PaHost_FreeFastMemory(pad, sizeof(internalPortAudioDevice)); - pad = nxt; /* PaHost_Init allocated this fast mem.*/ - } - sDeviceList = (internalPortAudioDevice*)NULL; - return 0; -} - -/***********************************************************************/ -void Pa_Sleep( long msec ) /* Sleep requested number of milliseconds. */ -{ -#if 0 - struct timeval timeout; - timeout.tv_sec = msec / 1000; - timeout.tv_usec = (msec % 1000) * 1000; - select(0, NULL, NULL, NULL, &timeout); -#else - long usecs = msec * 1000; - usleep( usecs ); -#endif -} - -/*---------------------------------------------------------------------------------------*/ -/* Allocate memory that can be accessed in real-time. This may need to be held in physi- */ -/* cal memory so that it is not paged to virtual memory. This call MUST be balanced with */ -/* a call to PaHost_FreeFastMemory(). */ -void *PaHost_AllocateFastMemory(long numBytes) -{ - void *addr = malloc(numBytes); /* mpin() reads into memory all pages over the given */ - if (addr) /* range and locks the pages into memory. A counter */ - { /* is incremented each time the page is locked. The */ - if (mpin(addr, numBytes)) /* superuser can lock as many pages as it wishes, */ - { /* others are limited to the configurable PLOCK_MA. */ - ERR_RPT(("PaHost_AllocateFastMemory() failed to mpin() memory.\n")); -#if 1 - free(addr); /* You MAY cut out these 2 lines to be less strict, */ - addr = NULL; /* you then only get the warning but PA goes on... */ -#endif /* Only problem then may be corresponding munpin() */ - } /* call at PaHost_FreeFastMemory(), below. */ - memset(addr, 0, numBytes); /* Locks established with mlock are not inherited by */ - } /* a child process after a fork. Furthermore, IRIX- */ - return addr; /* man-pages warn against mixing both mpin and mlock */ -} /* in 1 piece of code, so stick to mpin()/munpin() ! */ - - -/*---------------------------------------------------------------------------------------*/ -/* Free memory that could be accessed in real-time. This call MUST be balanced with a */ -/* call to PaHost_AllocateFastMemory(). */ -void PaHost_FreeFastMemory(void *addr, long numBytes) -{ - if (addr) - { - if (munpin(addr, numBytes)) /* Will munpin() fail when it was never mpinned? */ - ERR_RPT(("WARNING: PaHost_FreeFastMemory() failed to munpin() memory.\n")); - free(addr); /* But go on, try to release it, just warn... */ - } -} - -/*----------------------------------------------------------*/ -PaError PaHost_StreamActive( internalPortAudioStream *past ) -{ - PaHostSoundControl *pahsc; - if (past == NULL) - return paBadStreamPtr; - pahsc = (PaHostSoundControl *) past->past_DeviceData; - if (pahsc == NULL) - return paInternalError; - return (PaError)(past->past_IsActive != 0); -} - -/*-------------------------------------------------------------------*/ -PaTimestamp Pa_StreamTime( PortAudioStream *stream ) -{ - internalPortAudioStream *past = (internalPortAudioStream *) stream; -/* FIXME - return actual frames played, not frames generated. -** Need to query the output device somehow. -*/ - return past->past_FrameCount; -} diff --git a/pd/portaudio_v18/pa_sgi/pthread-Makefile b/pd/portaudio_v18/pa_sgi/pthread-Makefile deleted file mode 100644 index 527677a8..00000000 --- a/pd/portaudio_v18/pa_sgi/pthread-Makefile +++ /dev/null @@ -1,52 +0,0 @@ -# Make PortAudio for Silicon Graphics IRIX (6.2) -# Pieter suurmond, september 22, 2001. (v15 pa_sgi sub-version #0.18) - -# pthread, math (as with linux) and SGI audio library: -# SGI-books say -lpthread should be the last on the line. -LIBS = -lm -laudio -lpthread - -CDEFINES = -I../pa_common - -# Possible CFLAGS with MIPSpro compiler are: -32, -o32, -n32, -64, -# -mips1, -mips2, -mips3, -mips4, etc. Use -g, -g2, -g3 for debugging. -# And use for example -O2 or -O3 for better optimization: -CFLAGS = -O2 -PASRC = ../pa_common/pa_lib.c pa_sgi.c -PAINC = ../pa_common/portaudio.h - -# Tests that work (SGI Indy with R5000 @ 180MHz running IRIX 6.2). -#TESTC = $(PASRC) ../pa_tests/patest_record.c # OK -#TESTC = $(PASRC) ../pa_tests/patest_many.c # OK -#TESTC = $(PASRC) ../pa_tests/patest_latency.c # OK -#TESTC = $(PASRC) ../pa_tests/patest_longsine.c # OK but needs more than 4 buffers to do without glitches. -TESTC = $(PASRC) ../pa_tests/patest_saw.c # Seems OK (does it gracefully exit?). -#TESTC = $(PASRC) ../pa_tests/patest_wire.c # OK -#TESTC = $(PASRC) ../pa_tests/pa_devs.c # OK -#TESTC = $(PASRC) ../pa_tests/patest_sine.c # OK -#TESTC = $(PASRC) ../pa_tests/patest_sine_time.c # OK -#TESTC = $(PASRC) ../pa_tests/patest_sine8.c # OK -#TESTC = $(PASRC) ../pa_tests/patest_leftright.c # OK -#TESTC = $(PASRC) ../pa_tests/patest_pink.c # OK -#TESTC = $(PASRC) ../pa_tests/patest_clip.c # OK -#TESTC = $(PASRC) ../pa_tests/patest_stop.c # OK -#TESTC = $(PASRC) ../pa_tests/patest_dither.c # OK -#TESTC = $(PASRC) ../pa_tests/patest_sync.c # BEEPS and delta's, no crashes anymore. - -# Tests that do not yet work. -#TESTC = $(PASRC) ../pa_tests/paqa_devs.c # Heavy crash with core-dump after PaHost_OpenStream: opening 1 output channel(s) on AL default -#TESTC = $(PASRC) ../pa_tests/paqa_errs.c # Heavy crash with core-dump after PaHost_OpenStream: opening 2 output channel(s) on AL default -#TESTC = $(PASRC) ../pa_tests/pa_fuzz.c # THIS FUZZ CRASHED MY WHOLE IRIX SYSTEM after "ENTER"! :-( - # PROCESS IN ITSELF RUNS OK, WITH LARGER BUFFSIZES THOUGH. - # OH MAYBE IT WAS BECAUSE DEBUGGING WAS ON??? - # ANYWAY, I'M NOT GONNA RUN THAT AGAIN... WAY TOO DANGEROUS! -TESTH = $(PAINC) - -all: patest - -# "cc" for the MIPSpro compiler, may be changed to "gcc": -patest: $(TESTC) $(TESTH) Makefile - cc $(CFLAGS) $(TESTC) $(CDEFINES) $(LIBS) -o patest - -run: patest - ./patest - diff --git a/pd/portaudio_v18/pa_sgi/pthread-pa_sgi.c b/pd/portaudio_v18/pa_sgi/pthread-pa_sgi.c deleted file mode 100644 index e9ab273c..00000000 --- a/pd/portaudio_v18/pa_sgi/pthread-pa_sgi.c +++ /dev/null @@ -1,908 +0,0 @@ -/* - * PortAudio Portable Real-Time Audio Library - * Latest Version at: http://www.portaudio.com - * SGI IRIX implementation by Pieter Suurmond, september 22, 2001 (#0.18). - * - * Copyright (c) 1999-2001 Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - */ -/* -Modfication History: - 8/12/2001 - Pieter Suurmond - took the v15 pa_linux_oss.c file and started to adapt for IRIX 6.2. - 8/17/2001 - alpha release with IRIX sproc()-method, may sometimes let IRIX6.2 crash at closing audiostream. - 9/22/2001 - #0.18 pthread starts to work a bit: - BUT UNDER IRIX6.2, I DON'T GET IT TO WORK REALLY CORRECTLY, - this POSIX-attempt, - DON'T USE THIS FILE FOR RELIABLE OPERATION, IT IS HERE JUST - FOR DOCUMENTATION/ARCHIVE... OR FOR ANYONE WHO WANTS TO FIX...... -TODO: - - Test under IRIX 6.5. - - Dynamically switch to 32 bit float as native format when appropriate (let SGI do the conversion), - and maybe also the other natively supported formats? (might increase performance) - - Not sure whether CPU UTILIZATION MEASUREMENT (from OSS/linux) really works. Changed nothing yet, - seems ok, but I've not yet tested it thoroughly. (maybe utilization-code may be made _unix_common_ then?) - - The minimal number of buffers setting... I do not yet fully understand it.. I now just take *4. -REFERENCES: - - IRIX 6.2 man pages regarding SGI AL library. - - IRIS Digital MediaProgramming Guide (online books as well as man-pages come with IRIX 6.2 and - may not be publically available on the internet). -*/ - -#include <stdio.h> /* Standard libraries. */ -#include <stdlib.h> - -#include "../pa_common/portaudio.h" /* Portaudio headers. */ -#include "../pa_common/pa_host.h" -#include "../pa_common/pa_trace.h" - -/* -#include <malloc.h> -#include <memory.h> -#include <sys/prctl.h> Not needed -#include <sys/types.h> -#include <sys/schedctl.h> -#include <signal.h> -#include <sys/ioctl.h> Needed? -#include <sys/time.h> -#include <sched.h> sched_param struct and related functions - used in setting thread priorities. -#include <limits.h> Some POSIX constants such as _POSIX_THREAD_THREADS_MAX -*/ - -#include <pthread.h> /* Pthreads are supported by IRIX 6.2 after */ - /* patches 1361, 1367, and 1429 are applied. */ - -#include <fcntl.h> /* fcntl.h needed for "O_RDONLY". */ -#include <unistd.h> /* For usleep() and constants used when calling sysconf() */ - /* to query POSIX limits (see the sysconf(3) ref. page. */ - -#include <dmedia/audio.h> /* SGI-specific audio library. */ - - -/*--------------------------------------------*/ -#define PRINT(x) { printf x; fflush(stdout); } -#define ERR_RPT(x) PRINT(x) -#define DBUG(x) PRINT(x) -#define DBUGX(x) /* PRINT(x) */ - -#define MAX_CHARS_DEVNAME (16) /* Was 32 in OSS (20 for AL but "in"/"out" is concat. */ -#define MAX_SAMPLE_RATES (8) /* Known from SGI AL there are 7 (was 10 in OSS v15). */ - -typedef struct internalPortAudioDevice /* IRIX specific device info: */ -{ - PaDeviceID /* NEW: */ pad_DeviceID; /* THIS "ID" IS NEW HERE (Pieter)! */ - long pad_ALdevice; /* SGI-number! */ - double pad_SampleRates[MAX_SAMPLE_RATES]; /* for pointing to from pad_Info */ - char pad_DeviceName[MAX_CHARS_DEVNAME+1]; /* +1 for \0, one more than OSS. */ - PaDeviceInfo pad_Info; /* pad_Info (v15) contains: */ - /* int structVersion; */ - /* const char* name; */ - /* int maxInputChannels, maxOutputChannels; */ - /* int numSampleRates; Num rates, or -1 if range supprtd. */ - /* const double* sampleRates; Array of supported sample rates, */ - /* PaSampleFormat nativeSampleFormats; or {min,max} if range supported. */ - struct internalPortAudioDevice* pad_Next; /* Singly linked list, (NULL=end). */ -} internalPortAudioDevice; - -typedef struct PaHostSoundControl /* Structure to contain all SGI IRIX specific data. */ -{ - ALport pahsc_ALportIN, /* IRIX-audio-library-datatype. ALports can only be */ - pahsc_ALportOUT; /* unidirectional, so we sometimes need 2 of them. */ - pthread_t pahsc_ThreadPID; - short *pahsc_NativeInputBuffer, /* Allocated here, in this file, if necessary. */ - *pahsc_NativeOutputBuffer; - unsigned int pahsc_BytesPerInputBuffer, /* Native buffer sizes in bytes, really needed here */ - pahsc_BytesPerOutputBuffer; /* to free FAST memory, if buffs were alloctd FAST. */ - unsigned int pahsc_SamplesPerInputBuffer, /* These amounts are needed again and again in the */ - pahsc_SamplesPerOutputBuffer; /* audio-thread (don't need to be kept globally). */ - struct itimerval pahsc_EntryTime, /* For measuring CPU utilization (same as linux). */ - pahsc_LastExitTime; - long pahsc_InsideCountSum, - pahsc_TotalCountSum; -} PaHostSoundControl; - -/*----------------------------- Shared Data ------------------------------------------------------*/ -static internalPortAudioDevice* sDeviceList = NULL; /* FIXME - put Mutex around this shared data. */ -static int sPaHostError = 0; /* Maybe more than one process writing errs!? */ - -long Pa_GetHostError(void) -{ - return (long)sPaHostError; -} - -/*----------------------------- BEGIN CPU UTILIZATION MEASUREMENT -----------------*/ -/* (copied from source pa_linux_oss/pa_linux_oss.c) */ -static void Pa_StartUsageCalculation( internalPortAudioStream *past ) -{ - struct itimerval itimer; - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - if( pahsc == NULL ) return; -/* Query system timer for usage analysis and to prevent overuse of CPU. */ - getitimer( ITIMER_REAL, &pahsc->pahsc_EntryTime ); -} - -static long SubtractTime_AminusB( struct itimerval *timeA, struct itimerval *timeB ) -{ - long secs = timeA->it_value.tv_sec - timeB->it_value.tv_sec; - long usecs = secs * 1000000; - usecs += (timeA->it_value.tv_usec - timeB->it_value.tv_usec); - return usecs; -} - -static void Pa_EndUsageCalculation( internalPortAudioStream *past ) -{ - struct itimerval currentTime; - long insideCount; - long totalCount; /* Measure CPU utilization during this callback. */ - -#define LOWPASS_COEFFICIENT_0 (0.95) -#define LOWPASS_COEFFICIENT_1 (0.99999 - LOWPASS_COEFFICIENT_0) - - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - if (pahsc == NULL) - return; - if (getitimer( ITIMER_REAL, ¤tTime ) == 0 ) - { - if (past->past_IfLastExitValid) - { - insideCount = SubtractTime_AminusB( &pahsc->pahsc_EntryTime, ¤tTime ); - pahsc->pahsc_InsideCountSum += insideCount; - totalCount = SubtractTime_AminusB( &pahsc->pahsc_LastExitTime, ¤tTime ); - pahsc->pahsc_TotalCountSum += totalCount; - /* DBUG(("insideCount = %d, totalCount = %d\n", insideCount, totalCount )); */ - /* Low pass filter the result because sometimes we get called several times in a row. */ - /* That can cause the TotalCount to be very low which can cause the usage to appear */ - /* unnaturally high. So we must filter numerator and denominator separately!!! */ - if (pahsc->pahsc_InsideCountSum > 0) - { - past->past_AverageInsideCount = ((LOWPASS_COEFFICIENT_0 * past->past_AverageInsideCount) + - (LOWPASS_COEFFICIENT_1 * pahsc->pahsc_InsideCountSum)); - past->past_AverageTotalCount = ((LOWPASS_COEFFICIENT_0 * past->past_AverageTotalCount) + - (LOWPASS_COEFFICIENT_1 * pahsc->pahsc_TotalCountSum)); - past->past_Usage = past->past_AverageInsideCount / past->past_AverageTotalCount; - pahsc->pahsc_InsideCountSum = 0; - pahsc->pahsc_TotalCountSum = 0; - } - } - past->past_IfLastExitValid = 1; - } - pahsc->pahsc_LastExitTime.it_value.tv_sec = 100; - pahsc->pahsc_LastExitTime.it_value.tv_usec = 0; - setitimer( ITIMER_REAL, &pahsc->pahsc_LastExitTime, NULL ); - past->past_IfLastExitValid = 1; -} /*----------- END OF CPU UTILIZATION CODE (from pa_linux_oss/pa_linux_oss.c v15)--------------------*/ - - -/*--------------------------------------------------------------------------------------*/ -PaError translateSGIerror(void) /* Calls oserror(), may be used after an SGI AL-library */ -{ /* call to report via ERR_RPT(), yields a PaError-num. */ - const char* a = "SGI AL "; /* (Not absolutely sure errno came from THIS thread! */ - switch(oserror()) /* Read IRIX man-pages about the _SGI_MP_SOURCE macro.) */ - { - case AL_BAD_OUT_OF_MEM: - ERR_RPT(("%sout of memory.\n", a)); - return paInsufficientMemory; /* Known PaError. */ - case AL_BAD_CONFIG: - ERR_RPT(("%sconfiguration invalid or NULL.\n", a)); - return paHostError; /* Generic PaError. */ - case AL_BAD_CHANNELS: - ERR_RPT(("%schannels not 1,2 or 4.\n", a)); - return paHostError; /* Generic PaError. */ - case AL_BAD_NO_PORTS: - ERR_RPT(("%sout of audio ports.\n", a)); - return paHostError; /* Generic PaError. */ - case AL_BAD_DEVICE: - ERR_RPT(("%swrong device number.\n", a)); - return paHostError; /* Generic PaError. */ - case AL_BAD_DEVICE_ACCESS: - ERR_RPT(("%swrong device access.\n", a)); - return paHostError; /* Generic PaError. */ - case AL_BAD_DIRECTION: - ERR_RPT(("%sinvalid direction.\n", a)); - return paHostError; /* Generic PaError. */ - case AL_BAD_SAMPFMT: - ERR_RPT(("%sdoesn't accept sampleformat.\n", a)); - return paHostError; /* Generic PaError. */ - case AL_BAD_FLOATMAX: - ERR_RPT(("%smax float value is zero.\n", a)); - return paHostError; /* Generic PaError. */ - case AL_BAD_WIDTH: - ERR_RPT(("%sunsupported samplewidth.\n", a)); - return paHostError; /* Generic PaError. */ - case AL_BAD_QSIZE: - ERR_RPT(("%sinvalid queue size.\n", a)); - return paHostError; /* Generic PaError. */ - case AL_BAD_PVBUFFER: - ERR_RPT(("%sPVbuffer null.\n", a)); - return paHostError; /* Generic PaError. */ - case AL_BAD_BUFFERLENGTH_NEG: - ERR_RPT(("%snegative bufferlength.\n", a)); - return paHostError; /* Generic PaError. */ - case AL_BAD_BUFFERLENGTH_ODD: - ERR_RPT(("%sodd bufferlength.\n", a)); - return paHostError; /* Generic PaError. */ - case AL_BAD_PARAM: - ERR_RPT(("%sparameter not valid for device.\n", a)); - return paHostError; /* Generic PaError. */ - default: - ERR_RPT(("%sunknown error.\n", a)); - return paHostError; /* Generic PaError. */ - } -} - -/*------------------------------------------------------------------------------------------*/ -/* Tries to set various rates and formats and fill in the device info structure. */ -static PaError Pa_sgiQueryDevice(long ALdev, /* (AL_DEFAULT_DEVICE) */ - PaDeviceID id, /* (DefaultI|ODeviceID()) */ - char* name, /* (for example "SGI AL") */ - internalPortAudioDevice* pad) /* Result written to pad. */ -{ - int format; - long min, max; /* To catch hardware characteristics. */ - ALseterrorhandler(0); /* 0 = turn off the default error handler. */ - /*--------------------------------------------------------------------------------------*/ - pad->pad_ALdevice = ALdev; /* Set the AL device number. */ - pad->pad_DeviceID = id; /* Set the PA device number. */ - if (strlen(name) > MAX_CHARS_DEVNAME) /* MAX_CHARS defined above. */ - { - ERR_RPT(("Pa_QueryDevice(): name too long (%s).\n", name)); - return paHostError; - } - strcpy(pad->pad_DeviceName, name); /* Write name-string. */ - pad->pad_Info.name = pad->pad_DeviceName; /* Set pointer,..hmmm. */ - /*--------------------------------- natively supported sample formats: -----------------*/ - pad->pad_Info.nativeSampleFormats = paInt16; /* Later also include paFloat32 | ..| etc. */ - /* Then also choose other CallConvertXX()! */ - /*--------------------------------- number of available i/o channels: ------------------*/ - if (ALgetminmax(ALdev, AL_INPUT_COUNT, &min, &max)) - return translateSGIerror(); - pad->pad_Info.maxInputChannels = max; - DBUG(("Pa_QueryDevice: maxInputChannels = %d\n", pad->pad_Info.maxInputChannels)) - if (ALgetminmax(ALdev, AL_OUTPUT_COUNT, &min, &max)) - return translateSGIerror(); - pad->pad_Info.maxOutputChannels = max; - DBUG(("Pa_QueryDevice: maxOutputChannels = %d\n", pad->pad_Info.maxOutputChannels)) - /*--------------------------------- supported samplerates: ----------------------*/ - pad->pad_Info.numSampleRates = 7; - pad->pad_Info.sampleRates = pad->pad_SampleRates; - pad->pad_SampleRates[0] = (double)AL_RATE_8000; /* long -> double. */ - pad->pad_SampleRates[1] = (double)AL_RATE_11025; - pad->pad_SampleRates[2] = (double)AL_RATE_16000; - pad->pad_SampleRates[3] = (double)AL_RATE_22050; - pad->pad_SampleRates[4] = (double)AL_RATE_32000; - pad->pad_SampleRates[5] = (double)AL_RATE_44100; - pad->pad_SampleRates[6] = (double)AL_RATE_48000; - if (ALgetminmax(ALdev, AL_INPUT_RATE, &min, &max)) /* Ask INPUT rate-max. */ - return translateSGIerror(); /* double -> long. */ - if (max != (long)(0.5 + pad->pad_SampleRates[6])) /* FP-compare not recommndd. */ - goto weird; - if (ALgetminmax(ALdev, AL_OUTPUT_RATE, &min, &max)) /* Ask OUTPUT rate-max. */ - return translateSGIerror(); - if (max != (long)(0.5 + pad->pad_SampleRates[6])) - { -weird: ERR_RPT(("Pa_sgiQueryDevice() did not confirm max samplerate (%ld)\n",max)); - return paHostError; /* Or make it a warning and just carry on... */ - } - /*-------------------------------------------------------------------------------*/ - return paNoError; -} - - -/*--------------------------------------------------------------------------------*/ -int Pa_CountDevices() /* Name of this function suggests it only counts and */ -{ /* is NOT destructive, it however resets whole PA ! */ - int numDevices = 0; /* Let 's not do that here. */ - internalPortAudioDevice* currentDevice = sDeviceList; /* COPY GLOBAL VAR. */ -#if 0 /* Remains from linux_oss v15: Pa_Initialize(), on */ - if (!currentDevice) /* its turn, calls PaHost_Init() via file pa_lib.c. */ - Pa_Initialize(); /* Isn't that a bit too 'rude'? Don't be too */ -#endif /* friendly to clients that forgot to initialize PA. */ - while (currentDevice) /* Slower but more elegant than the sNumDevices-way: */ - { - numDevices++; - currentDevice = currentDevice->pad_Next; - } - return numDevices; -} - -/*-------------------------------------------------------------------------------*/ -static internalPortAudioDevice *Pa_GetInternalDevice(PaDeviceID id) -{ - int numDevices = 0; - internalPortAudioDevice *res = (internalPortAudioDevice*)NULL; - internalPortAudioDevice *pad = sDeviceList; /* COPY GLOBAL VAR. */ - while (pad) /* pad may be NULL, that's ok, return 0. */ - { /* (Added ->pad_DeviceID field to the pad-struct, Pieter, 2001.) */ - if (pad->pad_DeviceID == id) /* This the device we were looking for? */ - res = pad; /* But keep on(!) counting so we don't */ - numDevices++; /* have to call Pa_CountDevices() later. */ - pad = pad->pad_Next; /* Advance to the next device or NULL. */ - } /* No assumptions about order of ID's in */ - if (!res) /* the list. */ - ERR_RPT(("Pa_GetInternalDevice() could not find specified ID (%d).\n",id)); - if ((id < 0) || (id >= numDevices)) - { - ERR_RPT(("Pa_GetInternalDevice() supplied with an illegal ID (%d).\n",id)); -#if 1 /* Be strict, even when found, */ - res = (internalPortAudioDevice*)NULL; /* do not accept illegal ID's. */ -#endif - } - return res; -} - -/*----------------------------------------------------------------------*/ -const PaDeviceInfo* Pa_GetDeviceInfo(PaDeviceID id) -{ - PaDeviceInfo* res = (PaDeviceInfo*)NULL; - internalPortAudioDevice* pad = Pa_GetInternalDevice(id); /* Call. */ - if (pad) - res = &pad->pad_Info; /* Not finding the specified ID is not */ - if (!res) /* the same as &pad->pad_Info == NULL. */ - ERR_RPT(("Pa_GetDeviceInfo() could not find it (ID=%d).\n", id)); - return res; /* So (maybe) a second/third ERR_RPT(). */ -} - -/*------------------------------------------------*/ -PaDeviceID Pa_GetDefaultInputDeviceID(void) -{ - return 0; /* 0 is the default device ID. */ -} -/*------------------------------------------------*/ -PaDeviceID Pa_GetDefaultOutputDeviceID(void) -{ - return 0; -} - -/*-------------------------------------------------------------------------------------------------*/ -/* Build linked a list with all the available audio devices on this SGI machine (only 1 for now). */ -PaError PaHost_Init(void) /* Called by Pa_Initialize() from pa_lib.c. */ -{ - internalPortAudioDevice* pad; - PaError r = paNoError; - int audioLibFileID; /* To test for the presence of audio. */ - - if (sDeviceList) /* Allow re-init, only warn, no error. */ - { - ERR_RPT(("Warning: PaHost_Init() did not really re-init PA.\n")); - return r; - } - /*------------- ADD THE SGI DEFAULT DEVICE TO THE LIST: ---------------------------------------*/ - audioLibFileID = open("/dev/hdsp/hdsp0master", O_RDONLY); /* Try to open Indigo style audio */ - if (audioLibFileID < 0) /* IO port. On failure, machine */ - { /* has no audio ability. */ - ERR_RPT(("PaHost_Init(): This machine has no (Indigo-style) audio abilities.\n")); - return paHostError; - } - close(audioLibFileID); /* Allocate fast mem to hold device info. */ - pad = PaHost_AllocateFastMemory(sizeof(internalPortAudioDevice)); - if (pad == NULL) - return paInsufficientMemory; - memset(pad, 0, sizeof(internalPortAudioDevice)); /* "pad->pad_Next = NULL" is more elegant. */ - r = Pa_sgiQueryDevice(AL_DEFAULT_DEVICE, /* Set AL device num (AL_DEFAULT_DEVICE). */ - Pa_GetDefaultOutputDeviceID(),/* Set PA device num (or InputDeviceID()). */ - "AL default", /* A suitable name. */ - pad); /* Write args and queried info into pad. */ - if (r != paNoError) - { - ERR_RPT(("Pa_QueryDevice for '%s' returned: %d\n", pad->pad_DeviceName, r)); - PaHost_FreeFastMemory(pad, sizeof(internalPortAudioDevice)); /* sDeviceList still NULL ! */ - } - else - sDeviceList = pad; /* First element in linked list. pad->pad_Next already NULL. */ - /*------------- QUERY AND ADD MORE POSSIBLE SGI DEVICES TO THE LINKED LIST: -------------------*/ - /*---------------------------------------------------------------------------------------------*/ - return r; -} - -/*---------------------------------------------------------------------------------------------------*/ -static PaError Pa_SgiAudioProcess(internalPortAudioStream *past) /* Spawned by PaHost_StartEngine(). */ -{ - PaError result = paNoError; - PaHostSoundControl *pahsc; - - if (!past) - return paBadStreamPtr; - pahsc = (PaHostSoundControl*)past->past_DeviceData; - if (!pahsc) - return paInternalError; - past->past_IsActive = 1; /* Wasn't this already done by the calling parent?! */ - DBUG(("entering thread.\n")); - - while (!past->past_StopSoon) /* OR-ing StopSoon and StopNow here gives problems! */ - { - /*---------------------------------------- INPUT: ------------------------------------*/ - if (pahsc->pahsc_NativeInputBuffer) /* Then pahsc_ALportIN should also be there! */ - { - while (ALgetfilled(pahsc->pahsc_ALportIN) < pahsc->pahsc_SamplesPerInputBuffer) - { - /* Trying sginap(1); and usleep(); here... things get blocked under IRIX6.2. */ - if (past->past_StopNow) /* Don't let ALreadsamps() block */ - goto done; - } - if (ALreadsamps(pahsc->pahsc_ALportIN, (void*)pahsc->pahsc_NativeInputBuffer, - pahsc->pahsc_SamplesPerInputBuffer)) /* Number of samples instead */ - { /* of number of frames. */ - ERR_RPT(("ALreadsamps() failed.\n")); - result = paInternalError; - goto done; - } - } - /*---------------------------------------------------- USER CALLBACK ROUTINE: ----------*/ - /* DBUG(("Calling Pa_CallConvertInt16()...\n")); */ - Pa_StartUsageCalculation(past); /* Convert 16 bit native data to */ - result = Pa_CallConvertInt16(past, /* user data and call user routine. */ - pahsc->pahsc_NativeInputBuffer, pahsc->pahsc_NativeOutputBuffer); - Pa_EndUsageCalculation(past); - if (result) - { - DBUG(("Pa_CallConvertInt16() returned %d, stopping...\n", result)); - goto done; /* This is apparently NOT an error! */ - } /* Just letting the userCallBack stop us. */ - /*---------------------------------------- OUTPUT: ------------------------------------*/ - if (pahsc->pahsc_NativeOutputBuffer) /* Then pahsc_ALportOUT should also be there! */ - { - while (ALgetfillable(pahsc->pahsc_ALportOUT) < pahsc->pahsc_SamplesPerOutputBuffer) - { - /* Trying sginap(1); and usleep(); here... things get blocked under IRIX6.2. */ - if (past->past_StopNow) /* Don't let ALwritesamps() block */ - goto done; - } - if (ALwritesamps(pahsc->pahsc_ALportOUT, (void*)pahsc->pahsc_NativeOutputBuffer, - pahsc->pahsc_SamplesPerOutputBuffer)) - { - ERR_RPT(("ALwritesamps() failed.\n")); - result = paInternalError; - goto done; - } - } - /*-------------------------------------------------------------------------------------*/ - } -done: - /* pahsc->pahsc_ThreadPID = -1; Hu? doesn't help!! (added by Pieter) */ - past->past_IsActive = 0; - DBUG(("leaving thread.\n")); - return result; -} - - -/*--------------------------------------------------------------------------------------*/ -PaError PaHost_OpenStream(internalPortAudioStream *past) -{ - PaError result = paNoError; - PaHostSoundControl *pahsc; - unsigned int minNumBuffers; - internalPortAudioDevice *padIN, *padOUT; /* For looking up native AL-numbers. */ - ALconfig sgiALconfig = NULL; /* IRIX-datatype. */ - long pvbuf[8]; /* To get/set hardware configs. */ - long sr, ALqsize; - DBUG(("PaHost_OpenStream() called.\n")); /* Alloc FASTMEM and init host data. */ - if (!past) - { - ERR_RPT(("Streampointer NULL!\n")); - result = paBadStreamPtr; goto done; - } - pahsc = (PaHostSoundControl*)PaHost_AllocateFastMemory(sizeof(PaHostSoundControl)); - if (pahsc == NULL) - { - ERR_RPT(("FAST Memory allocation failed.\n")); /* Pass trough some ERR_RPT-exit- */ - result = paInsufficientMemory; goto done; /* code (nothing will be freed). */ - } - memset(pahsc, 0, sizeof(PaHostSoundControl)); -/* pahsc->pahsc_threadPID = -1; Should pahsc_threadPID be inited to */ - past->past_DeviceData = (void*)pahsc; /* -1 instead of 0 ?? */ - /*--------------------------------------------------- Allocate native buffers: --------*/ - pahsc->pahsc_SamplesPerInputBuffer = past->past_FramesPerUserBuffer * /* Needed by the */ - past->past_NumInputChannels; /* audio-thread. */ - pahsc->pahsc_BytesPerInputBuffer = pahsc->pahsc_SamplesPerInputBuffer * sizeof(short); - if (past->past_NumInputChannels > 0) /* Assumes short = 16 bits! */ - { - pahsc->pahsc_NativeInputBuffer = (short*)PaHost_AllocateFastMemory(pahsc->pahsc_BytesPerInputBuffer); - if( pahsc->pahsc_NativeInputBuffer == NULL ) - { - ERR_RPT(("Fast memory allocation for input-buffer failed.\n")); - result = paInsufficientMemory; goto done; - } - } - pahsc->pahsc_SamplesPerOutputBuffer = past->past_FramesPerUserBuffer * /* Needed by the */ - past->past_NumOutputChannels; /* audio-thread. */ - pahsc->pahsc_BytesPerOutputBuffer = pahsc->pahsc_SamplesPerOutputBuffer * sizeof(short); - if (past->past_NumOutputChannels > 0) /* Assumes short = 16 bits! */ - { - pahsc->pahsc_NativeOutputBuffer = (short*)PaHost_AllocateFastMemory(pahsc->pahsc_BytesPerOutputBuffer); - if (pahsc->pahsc_NativeOutputBuffer == NULL) - { - ERR_RPT(("Fast memory allocation for output-buffer failed.\n")); - result = paInsufficientMemory; goto done; - } - } - /*------------------------------------------ Manipulate hardware if necessary and allowed: --*/ - ALseterrorhandler(0); /* 0 = turn off the default error handler. */ - pvbuf[0] = AL_INPUT_RATE; - pvbuf[2] = AL_INPUT_COUNT; - pvbuf[4] = AL_OUTPUT_RATE; /* TO FIX: rates may be logically, not always in Hz! */ - pvbuf[6] = AL_OUTPUT_COUNT; - sr = (long)(past->past_SampleRate + 0.5); /* Common for input and output :-) */ - if (past->past_NumInputChannels > 0) /* We need to lookup the corre- */ - { /* sponding native AL-number(s). */ - padIN = Pa_GetInternalDevice(past->past_InputDeviceID); - if (!padIN) - { - ERR_RPT(("Pa_GetInternalDevice() for input failed.\n")); - result = paHostError; goto done; - } - if (ALgetparams(padIN->pad_ALdevice, &pvbuf[0], 4)) /* Although input and output will both be on */ - goto sgiError; /* the same AL-device, the AL-library might */ - if (pvbuf[1] != sr) /* contain more than AL_DEFAULT_DEVICE in */ - { /* Rate different from current harware-rate? the future. Therefore 2 seperate queries. */ - if (pvbuf[3] > 0) /* Means, there's other clients using AL-input-ports */ - { - ERR_RPT(("Sorry, not allowed to switch input-hardware to %ld Hz because \ -another process is currently using input at %ld kHz.\n", sr, pvbuf[1])); - result = paHostError; goto done; - } - pvbuf[1] = sr; /* Then set input-rate. */ - if (ALsetparams(padIN->pad_ALdevice, &pvbuf[0], 2)) - goto sgiError; /* WHETHER THIS SAMPLERATE WAS REALLY PRESENT IN OUR ARRAY OF RATES, */ - } /* IS NOT CHECKED, AT LEAST NOT BY ME, WITHIN THIS FILE! Does PA do? */ - } - if (past->past_NumOutputChannels > 0) /* CARE: padOUT/IN may NOT be NULL if Channels <= 0! */ - { /* We use padOUT/IN later on, or at least 1 of both. */ - padOUT = Pa_GetInternalDevice(past->past_OutputDeviceID); - if (!padOUT) - { - ERR_RPT(("Pa_GetInternalDevice() for output failed.\n")); - result = paHostError; goto done; - } - if (ALgetparams(padOUT->pad_ALdevice,&pvbuf[4], 4)) - goto sgiError; - if ((past->past_NumOutputChannels > 0) && (pvbuf[5] != sr)) - { /* Output needed and rate different from current harware-rate. */ - if (pvbuf[7] > 0) /* Means, there's other clients using AL-output-ports */ - { - ERR_RPT(("Sorry, not allowed to switch output-hardware to %ld Hz because \ -another process is currently using output at %ld kHz.\n", sr, pvbuf[5])); - result = paHostError; goto done; /* Will free again the inputbuffer */ - } /* that was just created above. */ - pvbuf[5] = sr; /* Then set output-rate. */ - if (ALsetparams(padOUT->pad_ALdevice, &pvbuf[4], 2)) - goto sgiError; - } - } - /*------------------------------------------ Construct an audio-port-configuration ----------*/ - sgiALconfig = ALnewconfig(); /* Change the SGI-AL-default-settings. */ - if (sgiALconfig == (ALconfig)0) /* sgiALconfig is released here after use! */ - goto sgiError; /* See that sgiALconfig is NOT released! */ - if (ALsetsampfmt(sgiALconfig, AL_SAMPFMT_TWOSCOMP)) /* Choose paInt16 as native i/o-format. */ - goto sgiError; - if (ALsetwidth (sgiALconfig, AL_SAMPLE_16)) /* Only meaningful when sample format for */ - goto sgiError; /* config is set to two's complement format. */ - /************************ Future versions might (dynamically) switch to 32-bit floats? ******* - if (ALsetsampfmt(sgiALconfig, AL_SAMPFMT_FLOAT)) (Then also call another CallConvert-func.) - goto sgiError; - if (ALsetfloatmax (sgiALconfig, 1.0)) Only meaningful when sample format for config - goto sgiError; is set to AL_SAMPFMT_FLOAT or AL_SAMPFMT_DOUBLE. */ - /*---------- ?? --------------------*/ - /* DBUG(("PaHost_OpenStream: pahsc_MinFramesPerHostBuffer = %d\n", pahsc->pahsc_MinFramesPerHostBuffer )); */ - minNumBuffers = Pa_GetMinNumBuffers(past->past_FramesPerUserBuffer, past->past_SampleRate); - past->past_NumUserBuffers = (minNumBuffers > past->past_NumUserBuffers) ? - minNumBuffers : past->past_NumUserBuffers; - /*------------------------------------------------ Set internal AL queuesize (in samples) ----*/ - if (pahsc->pahsc_SamplesPerInputBuffer >= pahsc->pahsc_SamplesPerOutputBuffer) - ALqsize = (long)pahsc->pahsc_SamplesPerInputBuffer; - else /* Take the largest of the two amounts. */ - ALqsize = (long)pahsc->pahsc_SamplesPerOutputBuffer; - ALqsize *= 4; /* 4 times as large as amount per transfer! */ - if (ALsetqueuesize(sgiALconfig, ALqsize)) /* Or should we use past_NumUserBuffers here? */ - goto sgiError; /* Using 2 times may give glitches... */ - /* Have to work on ALsetqueuesize() above. */ - - /* Do ALsetchannels() later, apart per input and/or output. */ - /*----------------------------------------------- OPEN 1 OR 2 AL-DEVICES: --------------------*/ - if (past->past_OutputDeviceID == past->past_InputDeviceID) /* Who SETS these devive-numbers? */ - { - if ((past->past_NumOutputChannels > 0) && (past->past_NumInputChannels > 0)) - { - DBUG(("PaHost_OpenStream: opening both input and output channels.\n")); - /*------------------------- open output port: ----------------------------------*/ - if (ALsetchannels (sgiALconfig, (long)(past->past_NumOutputChannels))) - goto sgiError; /* Returns 0 on success, -1 on failure. */ - pahsc->pahsc_ALportOUT = ALopenport("PA sgi out", "w", sgiALconfig); - if (pahsc->pahsc_ALportOUT == (ALport)0) - goto sgiError; - /*------------------------- open input port: -----------------------------------*/ - if (ALsetchannels (sgiALconfig, (long)(past->past_NumInputChannels))) - goto sgiError; /* Returns 0 on success, -1 on failure. */ - pahsc->pahsc_ALportIN = ALopenport("PA sgi in", "r", sgiALconfig); - if (pahsc->pahsc_ALportIN == (ALport)0) - goto sgiError; /* For some reason the "patest_wire.c"-test crashes! */ - } /* Probably due to too small buffersizes?.... */ - else - { - ERR_RPT(("Cannot setup bidirectional stream between different devices.\n")); - result = paHostError; - goto done; - } - } - else /* (OutputDeviceID != InputDeviceID) */ - { - if (past->past_NumOutputChannels > 0) /* WRITE-ONLY: */ - { - /*------------------------- open output port: ----------------------------------*/ - DBUG(("PaHost_OpenStream: opening %d output channel(s) on %s.\n", - past->past_NumOutputChannels, padOUT->pad_DeviceName)); - if (ALsetchannels (sgiALconfig, (long)(past->past_NumOutputChannels))) - goto sgiError; /* Returns 0 on success, -1 on failure. */ - pahsc->pahsc_ALportOUT = ALopenport("PA sgi out", "w", sgiALconfig); - if (pahsc->pahsc_ALportOUT == (ALport)0) - goto sgiError; - } - if (past->past_NumInputChannels > 0) /* READ-ONLY: */ - { - /*------------------------- open input port: -----------------------------------*/ - DBUG(("PaHost_OpenStream: opening %d input channel(s) on %s.\n", - past->past_NumInputChannels, padIN->pad_DeviceName)); - if (ALsetchannels (sgiALconfig, (long)(past->past_NumInputChannels))) - goto sgiError; /* Returns 0 on success, -1 on failure. */ - pahsc->pahsc_ALportIN = ALopenport("PA sgi in", "r", sgiALconfig); - if (pahsc->pahsc_ALportIN == (ALport)0) - goto sgiError; - } - } - DBUG(("PaHost_OpenStream() succeeded.\n")); - goto done; /* (no errors occured) */ -sgiError: - result = translateSGIerror(); /* Translates SGI AL-code to PA-code and ERR_RPTs string. */ -done: - if (sgiALconfig) - ALfreeconfig(sgiALconfig); /* We don't need that struct anymore. */ - if (result != paNoError) - PaHost_CloseStream(past); /* Frees memory (only if really allocated!). */ - return result; -} - -/*-----------------------------------------------------*/ -PaError PaHost_StartOutput(internalPortAudioStream *past) -{ - return paNoError; /* Hmm, not implemented yet? */ -} -PaError PaHost_StartInput(internalPortAudioStream *past) -{ - return paNoError; -} - -/*------------------------------------------------------------------------------*/ -PaError PaHost_StartEngine(internalPortAudioStream *past) -{ - PaHostSoundControl *pahsc; - int hres; - PaError result = paNoError; - - if (!past) /* Test argument. */ - { - ERR_RPT(("PaHost_StartEngine(NULL)!\n")); - return paBadStreamPtr; - } - pahsc = (PaHostSoundControl*)past->past_DeviceData; - if (!pahsc) - { - ERR_RPT(("PaHost_StartEngine(arg): arg->past_DeviceData == NULL!\n")); - return paHostError; - } - past->past_StopSoon = 0; /* Assume SGI ALport is already opened! */ - past->past_StopNow = 0; - past->past_IsActive = 1; - DBUG(("PaHost_StartEngine() called.\n")); - /* Use pthread_create() instead of __clone() because: */ - /* - pthread_create also works for other UNIX systems like Solaris, */ - /* - Java HotSpot VM crashes in pthread_setcanceltype() when using __clone(). */ - hres = pthread_create(&(pahsc->pahsc_ThreadPID), /* SPAWN AUDIO-CHILD. */ - NULL, /* pthread_attr_t * attr */ - (void*)Pa_SgiAudioProcess, - past); - if (hres) - { - result = paHostError; - sPaHostError = hres; - ERR_RPT(("PaHost_StartEngine() failed to spawn audio-thread.\n")); - } - return result; -} - -/*------------------------------------------------------------------------------*/ -PaError PaHost_StopEngine(internalPortAudioStream *past, int abort) -{ - int hres; - PaError result = paNoError; - PaHostSoundControl *pahsc; - - DBUG(("PaHost_StopEngine() called.\n")); - if (!past) - return paBadStreamPtr; - pahsc = (PaHostSoundControl*)past->past_DeviceData; - if (pahsc == NULL) - return result; /* paNoError (already stopped, no err?). */ - past->past_StopSoon = 1; /* Tell background thread to stop generating */ - if (abort) /* more and to let current data play out. If */ - past->past_StopNow = 1; /* aborting, tell backgrnd thread to stop NOW! */ - if (pahsc->pahsc_ThreadPID != -1) /* Join thread to recover memory resources. */ - { - DBUG(("pthread_join() called.\n")); - hres = pthread_join(pahsc->pahsc_ThreadPID, NULL); - if (hres) - { - result = paHostError; - sPaHostError = hres; - ERR_RPT(("PaHost_StopEngine() failed pthread_join().\n")); - } - pahsc->pahsc_ThreadPID = -1; - } - past->past_IsActive = 0; - return result; -} - -/*---------------------------------------------------------------*/ -PaError PaHost_StopOutput(internalPortAudioStream *past, int abort) -{ - return paNoError; /* Not implemented yet? */ -} -PaError PaHost_StopInput(internalPortAudioStream *past, int abort ) -{ - return paNoError; -} - -/*******************************************************************/ -PaError PaHost_CloseStream(internalPortAudioStream *past) -{ - PaHostSoundControl *pahsc; - PaError result = paNoError; - - DBUG(("PaHost_CloseStream() called.\n")); - if (past == NULL) - return paBadStreamPtr; - pahsc = (PaHostSoundControl *) past->past_DeviceData; - if (pahsc == NULL) /* If pahsc not NULL, past_DeviceData will be freed, and set to NULL. */ - return result; /* This test prevents from freeing NULL-pointers. */ - - if (pahsc->pahsc_ALportIN) - { - if (ALcloseport(pahsc->pahsc_ALportIN)) - result = translateSGIerror(); /* Translates SGI AL-code to PA-code and ERR_RPTs string. */ - else /* But go on anyway... to release other stuff... */ - pahsc->pahsc_ALportIN = (ALport)0; - } - if (pahsc->pahsc_ALportOUT) - { - if (ALcloseport(pahsc->pahsc_ALportOUT)) - result = translateSGIerror(); - else - pahsc->pahsc_ALportOUT = (ALport)0; - } - if (pahsc->pahsc_NativeInputBuffer) - { - PaHost_FreeFastMemory(pahsc->pahsc_NativeInputBuffer, pahsc->pahsc_BytesPerInputBuffer); - pahsc->pahsc_NativeInputBuffer = NULL; - } - if (pahsc->pahsc_NativeOutputBuffer) - { - PaHost_FreeFastMemory(pahsc->pahsc_NativeOutputBuffer, pahsc->pahsc_BytesPerOutputBuffer); - pahsc->pahsc_NativeOutputBuffer = NULL; - } - PaHost_FreeFastMemory(pahsc, sizeof(PaHostSoundControl)); - past->past_DeviceData = NULL; /* PaHost_OpenStream() allocated FAST. */ - return result; -} - -/************************************************************************* -** Determine minimum number of buffers required for this host based -** on minimum latency. Latency can be optionally set by user by setting -** an environment variable. For example, to set latency to 200 msec, put: -** set PA_MIN_LATENCY_MSEC=200 -** in the AUTOEXEC.BAT file and reboot. -** If the environment variable is not set, then the latency will be -** determined based on the OS. Windows NT has higher latency than Win95. -*/ -#define PA_LATENCY_ENV_NAME ("PA_MIN_LATENCY_MSEC") - -int Pa_GetMinNumBuffers( int framesPerBuffer, double sampleRate ) -{ - return 2; -} -/* Hmmm, the note above isn't appropriate for SGI I'm afraid... */ -/* Do we HAVE to do it this way under IRIX???.... */ -/*--------------------------------------------------------------*/ - - -/*---------------------------------------------------------------------*/ -PaError PaHost_Term(void) /* Frees all of the linked audio-devices. */ -{ /* Called by Pa_Terminate() from pa_lib.c. */ - internalPortAudioDevice *pad = sDeviceList, - *nxt; - while (pad) - { - DBUG(("PaHost_Term: freeing %s\n", pad->pad_DeviceName)); - nxt = pad->pad_Next; - PaHost_FreeFastMemory(pad, sizeof(internalPortAudioDevice)); - pad = nxt; /* PaHost_Init allocated this FAST MEM.*/ - } - sDeviceList = (internalPortAudioDevice*)NULL; - return 0; /* Got rid of sNumDevices=0; */ -} - -/***********************************************************************/ -void Pa_Sleep( long msec ) /* Sleep requested number of milliseconds. */ -{ -#if 0 - struct timeval timeout; - timeout.tv_sec = msec / 1000; - timeout.tv_usec = (msec % 1000) * 1000; - select(0, NULL, NULL, NULL, &timeout); -#else - long usecs = msec * 1000; - usleep( usecs ); -#endif -} - -/*---------------------------------------------------------------------------------------*/ -/* Allocate memory that can be accessed in real-time. This may need to be held in physi- */ -/* cal memory so that it is not paged to virtual memory. This call MUST be balanced with */ -/* a call to PaHost_FreeFastMemory(). */ -void *PaHost_AllocateFastMemory(long numBytes) -{ - void *addr = malloc(numBytes); - if (addr) - memset(addr, 0, numBytes); - return addr; -} - -/*---------------------------------------------------------------------------------------*/ -/* Free memory that could be accessed in real-time. This call MUST be balanced with a */ -/* call to PaHost_AllocateFastMemory(). */ -void PaHost_FreeFastMemory(void *addr, long numBytes) -{ - if (addr) - free(addr); -} - -/*----------------------------------------------------------*/ -PaError PaHost_StreamActive (internalPortAudioStream *past) -{ - PaHostSoundControl *pahsc; - if (past == NULL) - return paBadStreamPtr; - pahsc = (PaHostSoundControl *) past->past_DeviceData; - if (pahsc == NULL) - return paInternalError; - return (PaError)(past->past_IsActive != 0); -} - -/*-------------------------------------------------------------------*/ -PaTimestamp Pa_StreamTime( PortAudioStream *stream ) -{ - internalPortAudioStream *past = (internalPortAudioStream *) stream; -/* FIXME - return actual frames played, not frames generated. -** Need to query the output device somehow. -*/ - return past->past_FrameCount; -} diff --git a/pd/portaudio_v18/pa_tests/debug_convert.c b/pd/portaudio_v18/pa_tests/debug_convert.c deleted file mode 100644 index 11d25d9a..00000000 --- a/pd/portaudio_v18/pa_tests/debug_convert.c +++ /dev/null @@ -1,131 +0,0 @@ -/* - * $Id: debug_convert.c,v 1.1 2002/03/21 00:44:35 philburk Exp $ - * Convert tagged values. - * - * Author: Phil Burk <philburk@softsynth.com> - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.portaudio.com - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <math.h> -#include "portaudio.h" -#define OUTPUT_DEVICE (Pa_GetDefaultOutputDeviceID()) -//#define OUTPUT_DEVICE (11) -#define NUM_SECONDS (8) -#define SLEEP_DUR (800) -#define SAMPLE_RATE (44100) -#define FRAMES_PER_BUFFER (256) - -#define NUM_BUFFERS (0) - -typedef struct -{ - unsigned int framesToGo; -} -paTestData; -/* This routine will be called by the PortAudio engine when audio is needed. -** It may called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int patestCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData*)userData; - short *out = (short*)outputBuffer; - int i; - int finished = 0; - (void) outTime; /* Prevent unused variable warnings. */ - (void) inputBuffer; - - if( data->framesToGo < framesPerBuffer ) finished = 1; - - for( i=0; i<framesPerBuffer; i++ ) - { - *out++ = 0x0000 + i; /* left */ - *out++ = 0x1000 + i; /* right */ - } - return finished; -} -/*******************************************************************/ -int main(void); -int main(void) -{ - PortAudioStream *stream; - PaError err; - paTestData data; - int i; - int totalSamps; - printf("PortAudio Test: output debug values\n" ); - data.framesToGo = totalSamps = NUM_SECONDS * SAMPLE_RATE; /* Play for a few seconds. */ - printf("totalSamps = %d\n", totalSamps ); - err = Pa_Initialize(); - if( err != paNoError ) goto error; - printf("PortAudio Test: output device = %d\n", OUTPUT_DEVICE ); - - err = Pa_OpenStream( - &stream, - paNoDevice, - 0, /* no input */ - paFloat32, /* 32 bit floating point input */ - NULL, - OUTPUT_DEVICE, - 2, /* stereo output */ - paInt16, /* 32 bit floating point output */ - NULL, - SAMPLE_RATE, - FRAMES_PER_BUFFER, - NUM_BUFFERS, /* number of buffers, if zero then use default minimum */ - paClipOff|paDitherOff, /* we won't output out of range samples so don't bother clipping them */ - patestCallback, - &data ); - if( err != paNoError ) goto error; - - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - printf("Is callback being called?\n"); - for( i=0; i<((NUM_SECONDS+1)*1000); i+=SLEEP_DUR ) - { - printf("data.framesToGo = %d\n", data.framesToGo ); fflush(stdout); - Pa_Sleep( SLEEP_DUR ); - } - /* Stop sound until ENTER hit. */ - printf("Call Pa_StopStream()\n"); - err = Pa_StopStream( stream ); - if( err != paNoError ) goto error; - Pa_Terminate(); - printf("Test finished.\n"); - return err; -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return err; -} diff --git a/pd/portaudio_v18/pa_tests/debug_dither_calc.c b/pd/portaudio_v18/pa_tests/debug_dither_calc.c deleted file mode 100644 index 793c3990..00000000 --- a/pd/portaudio_v18/pa_tests/debug_dither_calc.c +++ /dev/null @@ -1,55 +0,0 @@ -/* - * $Id: debug_dither_calc.c,v 1.1 2002/03/21 00:44:35 philburk Exp $ - * Test Dither calculations. - * - * Author: Phil Burk http://www.softsynth.com - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.portaudio.com - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <math.h> -#include "portaudio.h" -#include "pa_host.h" - -/*******************************************************************/ -int main(void); -int main(void) -{ - long max,min; - int i; - - for( i=0; i<10000; i++ ) - { - long dither = PaConvert_TriangularDither(); - // printf("dither = 0x%08X\n", dither ); - if( dither < min ) min = dither; - else if( dither > max ) max = dither; - } - printf("min = 0x%08X = %d, max = 0x%08X = %d\n", min, min, max, max ); -} diff --git a/pd/portaudio_v18/pa_tests/debug_dual.c b/pd/portaudio_v18/pa_tests/debug_dual.c deleted file mode 100644 index bdb3c413..00000000 --- a/pd/portaudio_v18/pa_tests/debug_dual.c +++ /dev/null @@ -1,183 +0,0 @@ -/* - * $Id: debug_dual.c,v 1.1.1.1 2002/01/22 00:52:27 phil Exp $ - * debug_dual.c - * Try to open TWO streams on separate cards. - * Play a sine sweep using the Portable Audio api for several seconds. - * Hacked test for debugging PA. - * - * Author: Phil Burk <philburk@softsynth.com> - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.portaudio.com - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <math.h> -#include "portaudio.h" -#define DEV_ID_1 (13) -#define DEV_ID_2 (15) -#define NUM_SECONDS (8) -#define SLEEP_DUR (800) -#define SAMPLE_RATE (44100) -#define FRAMES_PER_BUFFER (256) -#if 0 -#define MIN_LATENCY_MSEC (200) -#define NUM_BUFFERS ((MIN_LATENCY_MSEC * SAMPLE_RATE) / (FRAMES_PER_BUFFER * 1000)) -#else -#define NUM_BUFFERS (0) -#endif -#define MIN_FREQ (100.0f) -#define MAX_FREQ (4000.0f) -#define FREQ_SCALAR (1.00002f) -#define CalcPhaseIncrement(freq) (freq/SAMPLE_RATE) -#ifndef M_PI -#define M_PI (3.14159265) -#endif -#define TABLE_SIZE (400) -typedef struct -{ - float sine[TABLE_SIZE + 1]; // add one for guard point for interpolation - float phase_increment; - float left_phase; - float right_phase; -} -paTestData; -/* Convert phase between and 1.0 to sine value - * using linear interpolation. - */ -float LookupSine( paTestData *data, float phase ); -float LookupSine( paTestData *data, float phase ) -{ - float fIndex = phase*TABLE_SIZE; - int index = (int) fIndex; - float fract = fIndex - index; - float lo = data->sine[index]; - float hi = data->sine[index+1]; - float val = lo + fract*(hi-lo); - return val; -} -/* This routine will be called by the PortAudio engine when audio is needed. -** It may called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int patestCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData*)userData; - float *out = (float*)outputBuffer; - unsigned long i; - int finished = 0; - (void) outTime; /* Prevent unused variable warnings. */ - (void) inputBuffer; - - - for( i=0; i<framesPerBuffer; i++ ) - { - *out++ = LookupSine(data, data->left_phase); /* left */ - *out++ = LookupSine(data, data->right_phase); /* right */ - data->left_phase += data->phase_increment; - if( data->left_phase >= 1.0f ) data->left_phase -= 1.0f; - data->right_phase += (data->phase_increment * 1.5f); /* fifth above */ - if( data->right_phase >= 1.0f ) data->right_phase -= 1.0f; - /* sweep frequency then start over. */ - data->phase_increment *= FREQ_SCALAR; - if( data->phase_increment > CalcPhaseIncrement(MAX_FREQ) ) data->phase_increment = CalcPhaseIncrement(MIN_FREQ); - } - return 0; -} - -PaError TestStart( PortAudioStream **streamPtr, PaDeviceID devID, - paTestData *data ); -/*******************************************************************/ -int main(void); -int main(void) -{ - PortAudioStream *stream1, *stream2; - PaError err; - paTestData DATA1, DATA2; - printf("PortAudio Test: DUAL sine sweep. ask for %d buffers\n", NUM_BUFFERS ); - err = Pa_Initialize(); - if( err != paNoError ) goto error; - err = TestStart( &stream1, DEV_ID_1, &DATA1 ); - if( err != paNoError ) goto error; - err = TestStart( &stream2, DEV_ID_2, &DATA2 ); - if( err != paNoError ) goto error; - printf("Hit ENTER\n"); - getchar(); - err = Pa_StopStream( stream1 ); - if( err != paNoError ) goto error; - err = Pa_StopStream( stream2 ); - if( err != paNoError ) goto error; - Pa_Terminate(); - printf("Test finished.\n"); - return err; -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return err; -} -PaError TestStart( PortAudioStream **streamPtr, PaDeviceID devID, paTestData *data ) -{ - PortAudioStream *stream; - PaError err; - int i; - /* initialise sinusoidal wavetable */ - for( i=0; i<TABLE_SIZE; i++ ) - { - data->sine[i] = (float) sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. ); - } - data->sine[TABLE_SIZE] = data->sine[0]; // set guard point - data->left_phase = data->right_phase = 0.0; - data->phase_increment = CalcPhaseIncrement(MIN_FREQ); - printf("PortAudio Test: output device = %d\n", devID ); - err = Pa_OpenStream( - &stream, - paNoDevice, - 0, /* no input */ - paFloat32, /* 32 bit floating point input */ - NULL, - devID, - 2, /* stereo output */ - paFloat32, /* 32 bit floating point output */ - NULL, - SAMPLE_RATE, - FRAMES_PER_BUFFER, - NUM_BUFFERS, /* number of buffers, if zero then use default minimum */ - paClipOff|paDitherOff, /* we won't output out of range samples so don't bother clipping them */ - patestCallback, - data ); - if( err != paNoError ) goto error; - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - *streamPtr = stream; - return 0; -error: - return err; -} diff --git a/pd/portaudio_v18/pa_tests/debug_multi_in.c b/pd/portaudio_v18/pa_tests/debug_multi_in.c deleted file mode 100644 index 9e376ce1..00000000 --- a/pd/portaudio_v18/pa_tests/debug_multi_in.c +++ /dev/null @@ -1,187 +0,0 @@ -/* - * $Id: debug_multi_in.c,v 1.1.1.1.4.3 2003/04/16 19:07:56 philburk Exp $ - * debug_multi_in.c - * Pass output from each of multiple channels - * to a stereo output using the Portable Audio api. - * Hacked test for debugging PA. - * - * Author: Phil Burk http://www.softsynth.com - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.portaudio.com - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <math.h> -#include <string.h> -#include "portaudio.h" -//#define INPUT_DEVICE_NAME ("EWS88 MT Interleaved Rec") -#define OUTPUT_DEVICE (Pa_GetDefaultOutputDeviceID()) -//#define OUTPUT_DEVICE (18) -#define SAMPLE_RATE (44100) -#define FRAMES_PER_BUFFER (256) -#define MIN_LATENCY_MSEC (400) -#define MAX_INPUT_CHANNELS (9999) -#define NUM_BUFFERS ((MIN_LATENCY_MSEC * SAMPLE_RATE) / (FRAMES_PER_BUFFER * 1000)) -#ifndef M_PI -#define M_PI (3.14159265) -#endif -typedef struct -{ - int liveChannel; - int numChannels; -} -paTestData; -/* This routine will be called by the PortAudio engine when audio is needed. -** It may called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int patestCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData*)userData; - float *out = (float*)outputBuffer; - float *in = (float*)inputBuffer; - int i; - (void) outTime; /* Prevent unused variable warnings. */ - (void) inputBuffer; - - if( in == NULL ) return 0; - for( i=0; i<(int)framesPerBuffer; i++ ) - { - /* Copy one channel of input to stereo output. */ - *out++ = in[data->liveChannel]; - *out++ = in[data->liveChannel]; - in += data->numChannels; - } - return 0; -} -/*******************************************************************/ -int PaFindDeviceByName( const char *name ) -{ - int i; - int numDevices; - const PaDeviceInfo *pdi; - int len = strlen( name ); - PaDeviceID result = paNoDevice; - numDevices = Pa_CountDevices(); - for( i=0; i<numDevices; i++ ) - { - pdi = Pa_GetDeviceInfo( i ); - if( strncmp( name, pdi->name, len ) == 0 ) - { - result = i; - break; - } - } - return result; -} -/*******************************************************************/ -int main(void); -int main(void) -{ - PortAudioStream *stream; - PaError err; - paTestData data; - int i; - PaDeviceID inputDevice; - const PaDeviceInfo *pdi; - printf("PortAudio Test: input signal from each channel. %d buffers\n", NUM_BUFFERS ); - data.liveChannel = 0; - err = Pa_Initialize(); - if( err != paNoError ) goto error; -#ifdef INPUT_DEVICE_NAME - printf("Try to use device: %s\n", INPUT_DEVICE_NAME ); - inputDevice = PaFindDeviceByName(INPUT_DEVICE_NAME); - if( inputDevice == paNoDevice ) - { - printf("Could not find %s. Using default instead.\n", INPUT_DEVICE_NAME ); - inputDevice = Pa_GetDefaultInputDeviceID(); - } -#else - printf("Using default input device.\n"); - inputDevice = Pa_GetDefaultInputDeviceID(); -#endif - pdi = Pa_GetDeviceInfo( inputDevice ); - if( pdi == NULL ) - { - printf("Could not get device info!\n"); - goto error; - } - printf("Input Device name is %s\n", pdi->name ); - printf("Input Device has %d channels.\n", pdi->maxInputChannels); - if( pdi->maxInputChannels <= MAX_INPUT_CHANNELS ) - { - data.numChannels = pdi->maxInputChannels; - } - else - { - data.numChannels = MAX_INPUT_CHANNELS; - printf("Only use %d channels.\n", MAX_INPUT_CHANNELS ); - } - err = Pa_OpenStream( - &stream, - inputDevice, - data.numChannels, - paFloat32, /* 32 bit floating point input */ - NULL, - OUTPUT_DEVICE, - 2, - paFloat32, /* 32 bit floating point output */ - NULL, - SAMPLE_RATE, - FRAMES_PER_BUFFER, /* frames per buffer */ - NUM_BUFFERS, /* number of buffers, if zero then use default minimum */ - paClipOff, /* we won't output out of range samples so don't bother clipping them */ - patestCallback, - &data ); - if( err != paNoError ) goto error; - data.liveChannel = 0; - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - for( i=0; i<data.numChannels; i++ ) - { - data.liveChannel = i; - printf("Channel %d being sent to output. Hit ENTER for next channel.", i ); - fflush(stdout); - getchar(); - } - err = Pa_StopStream( stream ); - if( err != paNoError ) goto error; - - err = Pa_CloseStream( stream ); - Pa_Terminate(); - printf("Test finished.\n"); - return err; -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return err; -} diff --git a/pd/portaudio_v18/pa_tests/debug_multi_out.c b/pd/portaudio_v18/pa_tests/debug_multi_out.c deleted file mode 100644 index 7568d1d5..00000000 --- a/pd/portaudio_v18/pa_tests/debug_multi_out.c +++ /dev/null @@ -1,139 +0,0 @@ -/* - * $Id: debug_multi_out.c,v 1.3.4.1 2003/04/07 20:00:57 philburk Exp $ - * debug_multi_out.c - * Output different numbers on each channels for step debugging, - * using the Portable Audio api. - * - * Author: Phil Burk http://www.softsynth.com - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.portaudio.com - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <math.h> -#include "portaudio.h" - -#define OUTPUT_DEVICE (Pa_GetDefaultOutputDeviceID()) -#define SAMPLE_RATE (44100) -#define FRAMES_PER_BUFFER (256) -#define FREQ_INCR (300.0 / SAMPLE_RATE) -#define MAX_CHANNELS (64) - -#ifndef M_PI -#define M_PI (3.14159265) -#endif - -typedef struct -{ - int numChannels; -} -paTestData; - -/* This routine will be called by the PortAudio engine when audio is needed. -** It may called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int patestCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData*)userData; - float *out = (float*)outputBuffer; - int frameIndex, channelIndex; - int finished = 0; - (void) outTime; /* Prevent unused variable warnings. */ - (void) inputBuffer; - - for( frameIndex=0; frameIndex<(int)framesPerBuffer; frameIndex++ ) - { - for( channelIndex=0; channelIndex<data->numChannels; channelIndex++ ) - { - /* Output sine wave on every channel. */ - *out++ = (float) ((channelIndex + 1) * 0.1); - } - } - - return 0; -} -/*******************************************************************/ -int main(void); -int main(void) -{ - PortAudioStream *stream; - PaError err; - const PaDeviceInfo *pdi; - paTestData data = {0}; - printf("PortAudio Test: output channel number on each channel.\n" ); - - err = Pa_Initialize(); - if( err != paNoError ) goto error; - - pdi = Pa_GetDeviceInfo( OUTPUT_DEVICE ); - data.numChannels = pdi->maxOutputChannels; - if( data.numChannels > MAX_CHANNELS ) data.numChannels = MAX_CHANNELS; - printf("Number of Channels = %d\n", data.numChannels ); - - err = Pa_OpenStream( - &stream, - paNoDevice, /* default input device */ - 0, /* no input */ - paFloat32, /* 32 bit floating point input */ - NULL, - OUTPUT_DEVICE, - data.numChannels, - paFloat32, /* 32 bit floating point output */ - NULL, - SAMPLE_RATE, - FRAMES_PER_BUFFER, /* frames per buffer */ - 0, /* number of buffers, if zero then use default minimum */ - paClipOff, /* we won't output out of range samples so don't bother clipping them */ - patestCallback, - &data ); - if( err != paNoError ) goto error; - - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - - printf("Hit ENTER to stop sound.\n"); - fflush(stdout); - getchar(); - - err = Pa_StopStream( stream ); - if( err != paNoError ) goto error; - - Pa_CloseStream( stream ); - Pa_Terminate(); - printf("Test finished.\n"); - return err; -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return err; -} diff --git a/pd/portaudio_v18/pa_tests/debug_record.c b/pd/portaudio_v18/pa_tests/debug_record.c deleted file mode 100644 index f9359254..00000000 --- a/pd/portaudio_v18/pa_tests/debug_record.c +++ /dev/null @@ -1,338 +0,0 @@ -/* - * $Id: debug_record.c,v 1.3.4.3 2003/03/06 06:09:20 philburk Exp $ - * patest_record.c - * Record input into an array. - * Save array to a file. - * Playback recorded data. - * - * Author: Phil Burk http://www.softsynth.com - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.portaudio.com - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <stdlib.h> -#include "portaudio.h" - -#define SAMPLE_RATE (44100) -#define NUM_SECONDS (6) -#define NUM_CHANNELS (2) -#define FRAMES_PER_BUFFER (64) -/* #define DITHER_FLAG (paDitherOff) */ -#define DITHER_FLAG (0) - -/* Select sample format. */ -#if 1 -#define PA_SAMPLE_TYPE paFloat32 -typedef float SAMPLE; -#define SAMPLE_SILENCE (0.0f) - -#elif 0 -#define PA_SAMPLE_TYPE paInt32 -typedef long SAMPLE; -#define SAMPLE_SILENCE (0) - -#elif 0 -#define PA_SAMPLE_TYPE paInt16 -typedef short SAMPLE; -#define SAMPLE_SILENCE (0) - -#elif 0 -#define PA_SAMPLE_TYPE paInt8 -typedef char SAMPLE; -#define SAMPLE_SILENCE (0) - -#else -#define PA_SAMPLE_TYPE paUInt8 -typedef unsigned char SAMPLE; -#define SAMPLE_SILENCE (128) - -#endif - -typedef struct -{ - int frameIndex; /* Index into sample array. */ - int maxFrameIndex; - SAMPLE *recordedSamples; -} -paTestData; -/* This routine will be called by the PortAudio engine when audio is needed. -** It may be called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int recordCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData*)userData; - SAMPLE *rptr = (SAMPLE*)inputBuffer; - SAMPLE *wptr = &data->recordedSamples[data->frameIndex * NUM_CHANNELS]; - long framesToCalc; - long i; - int finished; - unsigned long framesLeft = data->maxFrameIndex - data->frameIndex; - - (void) outputBuffer; /* Prevent unused variable warnings. */ - (void) outTime; - - if( framesLeft < framesPerBuffer ) - { - framesToCalc = framesLeft; - finished = 1; - } - else - { - framesToCalc = framesPerBuffer; - finished = 0; - } - if( inputBuffer == NULL ) - { - for( i=0; i<framesToCalc; i++ ) - { - *wptr++ = SAMPLE_SILENCE; /* left */ - if( NUM_CHANNELS == 2 ) *wptr++ = SAMPLE_SILENCE; /* right */ - } - } - else - { - for( i=0; i<framesToCalc; i++ ) - { - *wptr++ = *rptr++; /* left */ - if( NUM_CHANNELS == 2 ) *wptr++ = *rptr++; /* right */ - } - } - data->frameIndex += framesToCalc; - return finished; -} - -/* This routine will be called by the PortAudio engine when audio is needed. -** It may be called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int playCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData*)userData; - SAMPLE *rptr = &data->recordedSamples[data->frameIndex * NUM_CHANNELS]; - SAMPLE *wptr = (SAMPLE*)outputBuffer; - unsigned int i; - int finished; - unsigned int framesLeft = data->maxFrameIndex - data->frameIndex; - (void) inputBuffer; /* Prevent unused variable warnings. */ - (void) outTime; - - if( framesLeft < framesPerBuffer ) - { - /* final buffer... */ - for( i=0; i<framesLeft; i++ ) - { - *wptr++ = *rptr++; /* left */ - if( NUM_CHANNELS == 2 ) *wptr++ = *rptr++; /* right */ - } - for( ; i<framesPerBuffer; i++ ) - { - *wptr++ = 0; /* left */ - if( NUM_CHANNELS == 2 ) *wptr++ = 0; /* right */ - } - data->frameIndex += framesLeft; - finished = 1; - } - else - { - for( i=0; i<framesPerBuffer; i++ ) - { - *wptr++ = *rptr++; /* left */ - if( NUM_CHANNELS == 2 ) *wptr++ = *rptr++; /* right */ - } - data->frameIndex += framesPerBuffer; - finished = 0; - } - return finished; -} - -/*******************************************************************/ -int main(void); -int main(void) -{ - PortAudioStream *stream; - PaError err; - paTestData data; - int i; - int totalFrames; - int numSamples; - int numBytes; - SAMPLE max, average, val; - - printf("debug_record.c, sampleRate = %d, numChannels = %d\n", - SAMPLE_RATE, NUM_CHANNELS ); - fflush(stdout); - - data.maxFrameIndex = totalFrames = NUM_SECONDS * SAMPLE_RATE; /* Record for a few seconds. */ - data.frameIndex = 0; - numSamples = totalFrames * NUM_CHANNELS; - - numBytes = numSamples * sizeof(SAMPLE); - data.recordedSamples = (SAMPLE *) malloc( numBytes ); - if( data.recordedSamples == NULL ) - { - printf("Could not allocate record array.\n"); - exit(1); - } - for( i=0; i<numSamples; i++ ) data.recordedSamples[i] = 0; - - err = Pa_Initialize(); - if( err != paNoError ) goto error; - - /* Record some audio. -------------------------------------------- */ - err = Pa_OpenStream( - &stream, - Pa_GetDefaultInputDeviceID(), - NUM_CHANNELS, /* stereo input */ - PA_SAMPLE_TYPE, - NULL, - paNoDevice, - 0, - PA_SAMPLE_TYPE, - NULL, - SAMPLE_RATE, - FRAMES_PER_BUFFER, /* frames per buffer */ - 0, /* number of buffers, if zero then use default minimum */ - 0, //paDitherOff, /* flags */ - recordCallback, - &data ); - if( err != paNoError ) goto error; - - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - printf("Start recording!!\n"); fflush(stdout); - - while( Pa_StreamActive( stream ) ) - { - Pa_Sleep(1000); - printf("index = %d\n", data.frameIndex ); fflush(stdout); - } - printf("Stop recording!!\n"); fflush(stdout); - err = Pa_CloseStream( stream ); - if( err != paNoError ) goto error; - - /* Measure maximum peak amplitude. */ - max = 0; - average = 0; - for( i=0; i<numSamples; i++ ) - { - val = data.recordedSamples[i]; - if( val < 0 ) val = -val; /* ABS */ - if( val > max ) - { - max = val; - } - average += val; - } - - average = average / numSamples; - - if( PA_SAMPLE_TYPE == paFloat32 ) - { - printf("sample max amplitude = %f\n", (double) max ); - printf("sample average = %f\n", (double) average ); - } - else - { - printf("sample max amplitude = %d\n", (int) max ); - printf("sample average = %d\n", (int) average ); - } - - /* Write recorded data to a file. */ -#if 0 - { - FILE *fid; - fid = fopen("recorded.raw", "wb"); - if( fid == NULL ) - { - printf("Could not open file."); - } - else - { - fwrite( data.recordedSamples, NUM_CHANNELS * sizeof(SAMPLE), totalFrames, fid ); - fclose( fid ); - printf("Wrote data to 'recorded.raw'\n"); - } - } -#endif - - /* Playback recorded data. -------------------------------------------- */ - data.frameIndex = 0; - printf("Begin playback.\n"); fflush(stdout); - err = Pa_OpenStream( - &stream, - paNoDevice, - 0, /* NO input */ - PA_SAMPLE_TYPE, - NULL, - Pa_GetDefaultOutputDeviceID(), - NUM_CHANNELS, /* stereo output */ - PA_SAMPLE_TYPE, - NULL, - SAMPLE_RATE, - FRAMES_PER_BUFFER, /* frames per buffer */ - 0, /* number of buffers, if zero then use default minimum */ - paClipOff, /* we won't output out of range samples so don't bother clipping them */ - playCallback, - &data ); - if( err != paNoError ) goto error; - - if( stream ) - { - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - printf("Start playback!!\n"); fflush(stdout); - - while( Pa_StreamActive( stream ) ) - { - Pa_Sleep(1000); - printf("index = %d\n", data.frameIndex ); fflush(stdout); - } - - printf("Stop playback!!\n"); fflush(stdout); - err = Pa_CloseStream( stream ); - if( err != paNoError ) goto error; - printf("Done.\n"); fflush(stdout); - } - free( data.recordedSamples ); - - Pa_Terminate(); - return 0; - -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return -1; -} diff --git a/pd/portaudio_v18/pa_tests/debug_record_reuse.c b/pd/portaudio_v18/pa_tests/debug_record_reuse.c deleted file mode 100644 index e0670009..00000000 --- a/pd/portaudio_v18/pa_tests/debug_record_reuse.c +++ /dev/null @@ -1,351 +0,0 @@ -/* - * $Id: debug_record_reuse.c,v 1.1 2002/05/02 20:16:29 philburk Exp $ - * debug_record_reuse.c - * Record input into an array. - * Save array to a file. - * Based on patest_record.c but with various ugly debug hacks thrown in. - * Loop twice and reuse same streams. - * - * Author: Phil Burk http://www.softsynth.com - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.portaudio.com - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <stdlib.h> -#include <memory.h> -#include "portaudio.h" -#define SAMPLE_RATE (22050) -#define NUM_SECONDS (4) -#define SLEEP_DUR_MSEC (200) -#define FRAMES_PER_BUFFER (256) -#define NUM_REC_BUFS (0) - -#if 1 -#define PA_SAMPLE_TYPE paFloat32 -typedef float SAMPLE; -#else -#define PA_SAMPLE_TYPE paInt16 -typedef short SAMPLE; -#endif - -typedef struct -{ - long frameIndex; /* Index into sample array. */ - long maxFrameIndex; - long samplesPerFrame; - long numSamples; - PortAudioStream *outputStream; - PortAudioStream *inputStream; - SAMPLE *recordedSamples; -} -paTestData; - -/* This routine will be called by the PortAudio engine when audio is needed. -** It may be called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int recordCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData*)userData; - SAMPLE *rptr = (SAMPLE*)inputBuffer; - SAMPLE *wptr = &data->recordedSamples[data->frameIndex * data->samplesPerFrame]; - long framesToCalc; - unsigned long i; - int finished; - unsigned long framesLeft = data->maxFrameIndex - data->frameIndex; - - (void) outputBuffer; /* Prevent unused variable warnings. */ - (void) outTime; - - if( framesLeft < framesPerBuffer ) - { - framesToCalc = framesLeft; - finished = 1; - } - else - { - framesToCalc = framesPerBuffer; - finished = 0; - } - if( inputBuffer == NULL ) - { - for( i=0; i<framesToCalc; i++ ) - { - *wptr++ = 0; /* left */ - *wptr++ = 0; /* right */ - } - } - else - { - for( i=0; i<framesToCalc; i++ ) - { - *wptr++ = *rptr++; /* left */ - *wptr++ = *rptr++; /* right */ - } - } - data->frameIndex += framesToCalc; - return finished; -} - -/* This routine will be called by the PortAudio engine when audio is needed. -** It may be called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int playCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData*)userData; - SAMPLE *rptr = &data->recordedSamples[data->frameIndex * data->samplesPerFrame]; - SAMPLE *wptr = (SAMPLE*)outputBuffer; - unsigned long i; - int finished; - unsigned int framesLeft = data->maxFrameIndex - data->frameIndex; - if( outputBuffer == NULL ) return 0; - (void) inputBuffer; /* Prevent unused variable warnings. */ - (void) outTime; - - if( framesLeft < framesPerBuffer ) - { - /* final buffer... */ - for( i=0; i<framesLeft; i++ ) - { - *wptr++ = *rptr++; /* left */ - *wptr++ = *rptr++; /* right */ - } - for( ; i<framesPerBuffer; i++ ) - { - *wptr++ = 0; /* left */ - *wptr++ = 0; /* right */ - } - data->frameIndex += framesLeft; - finished = 1; - } - else - { - for( i=0; i<framesPerBuffer; i++ ) - { - *wptr++ = *rptr++; /* left */ - *wptr++ = *rptr++; /* right */ - } - data->frameIndex += framesPerBuffer; - finished = 0; - } - return finished; -} - -/****************************************************************/ -PaError TestRecording( paTestData *dataPtr ) -{ - PaError err; - int i; - int lastIndex = 0; - -/* Open input stream if not already open. */ - if( dataPtr->inputStream == NULL ) - { - /* Record some audio. */ - err = Pa_OpenStream( - &dataPtr->inputStream, - Pa_GetDefaultInputDeviceID(), - dataPtr->samplesPerFrame, /* stereo input */ - PA_SAMPLE_TYPE, - NULL, - paNoDevice, - 0, - PA_SAMPLE_TYPE, - NULL, - SAMPLE_RATE, - FRAMES_PER_BUFFER, /* frames per buffer */ - NUM_REC_BUFS, /* number of buffers, if zero then use default minimum */ - paClipOff, /* we won't output out of range samples so don't bother clipping them */ - recordCallback, - dataPtr ); - if( err != paNoError ) goto error; - } - - dataPtr->frameIndex = 0; - - err = Pa_StartStream( dataPtr->inputStream ); - if( err != paNoError ) goto error; - - printf("Now recording!\n"); fflush(stdout); - for( i=0; i<(NUM_SECONDS*1000/SLEEP_DUR_MSEC); i++ ) - { - int frameIndex, delta; - Pa_Sleep(SLEEP_DUR_MSEC); - - frameIndex = dataPtr->frameIndex; - if( Pa_StreamActive( dataPtr->inputStream ) <= 0) - { - printf("Stream inactive!\n"); - break; - } - if( dataPtr->maxFrameIndex <= frameIndex ) - { - printf("Buffer recording complete.\n"); - break; - } - - delta = frameIndex - lastIndex; - lastIndex = frameIndex; - printf("index = %d, delta = %d\n", frameIndex, delta ); fflush(stdout); - } - - err = Pa_StopStream( dataPtr->inputStream ); - if( err != paNoError ) goto error; - - printf("Done.\n"); fflush(stdout); - -error: - return err; -} - -/****************************************************************/ -PaError TestPlayback( paTestData *dataPtr ) -{ - PaError err; - int i; - int lastIndex = 0; - - /* Playback recorded data. */ - dataPtr->frameIndex = 0; - printf("Begin playback.\n"); fflush(stdout); - -/* Open output stream if not already open. */ - if( dataPtr->outputStream == NULL ) - { - err = Pa_OpenStream( - &dataPtr->outputStream, - paNoDevice, - 0, /* NO input */ - PA_SAMPLE_TYPE, - NULL, - Pa_GetDefaultOutputDeviceID(), - dataPtr->samplesPerFrame, /* stereo output */ - PA_SAMPLE_TYPE, - NULL, - SAMPLE_RATE, - FRAMES_PER_BUFFER, /* frames per buffer */ - 0, /* number of buffers, if zero then use default minimum */ - paClipOff, /* we won't output out of range samples so don't bother clipping them */ - playCallback, - dataPtr ); - if( err != paNoError ) goto error; - } - - err = Pa_StartStream( dataPtr->outputStream ); - if( err != paNoError ) goto error; - - printf("Waiting for playback to finish.\n"); fflush(stdout); - for( i=0; i<(NUM_SECONDS*1000/SLEEP_DUR_MSEC); i++ ) - { - int frameIndex, delta; - Pa_Sleep(SLEEP_DUR_MSEC); - frameIndex = dataPtr->frameIndex; - delta = frameIndex - lastIndex; - lastIndex = frameIndex; - printf("index = %d, delta = %d\n", frameIndex, delta ); fflush(stdout); - } - - err = Pa_StopStream( dataPtr->outputStream ); - if( err != paNoError ) goto error; - -error: - return err; -} -/*******************************************************************/ -int main(void); -int main(void) -{ - PaError err; - paTestData data = { 0 }; - long totalFrames; - long numBytes; - long i; - printf("patest_record.c\n"); fflush(stdout); - -/* Set up test data structure and sample array. */ - data.frameIndex = 0; - data.samplesPerFrame = 2; - data.maxFrameIndex = totalFrames = NUM_SECONDS*SAMPLE_RATE; - - printf("totalFrames = %d\n", totalFrames ); fflush(stdout); - data.numSamples = totalFrames * data.samplesPerFrame; - - numBytes = data.numSamples * sizeof(SAMPLE); - data.recordedSamples = (SAMPLE *) malloc( numBytes ); - if( data.recordedSamples == NULL ) - { - printf("Could not allocate record array.\n"); - exit(1); - } - for( i=0; i<data.numSamples; i++ ) data.recordedSamples[i] = 0; - - err = Pa_Initialize(); - if( err != paNoError ) goto error; - -/* Record and playback multiple times. */ - for( i=0; i<2; i++ ) - { - err = TestRecording( &data ); - if( err != paNoError ) goto error; - - err = TestPlayback( &data ); - if( err != paNoError ) goto error; - } - -/* Clean up. */ - err = Pa_CloseStream( data.inputStream ); - if( err != paNoError ) goto error; - - err = Pa_CloseStream( data.outputStream ); - if( err != paNoError ) goto error; - - if( err != paNoError ) goto error; - - free( data.recordedSamples ); - Pa_Terminate(); - - printf("Test complete.\n"); fflush(stdout); - return 0; - -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - if( err == paHostError ) - { - fprintf( stderr, "Host Error number: %d\n", Pa_GetHostError() ); - } - return -1; -} diff --git a/pd/portaudio_v18/pa_tests/debug_sine.c b/pd/portaudio_v18/pa_tests/debug_sine.c deleted file mode 100644 index c6e4bbbf..00000000 --- a/pd/portaudio_v18/pa_tests/debug_sine.c +++ /dev/null @@ -1,201 +0,0 @@ -/* - * $Id: debug_sine.c,v 1.2.4.2 2003/03/06 06:09:20 philburk Exp $ - * debug_sine.c - * Play a sine sweep using the Portable Audio api for several seconds. - * Hacked test for debugging PA. - * - * Author: Phil Burk <philburk@softsynth.com> - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.portaudio.com - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <math.h> -#include "portaudio.h" -#define OUTPUT_DEVICE (Pa_GetDefaultOutputDeviceID()) -//#define OUTPUT_DEVICE (11) -#define NUM_SECONDS (8) -#define SLEEP_DUR (800) -#define SAMPLE_RATE (44100) -#define FRAMES_PER_BUFFER (256) - -#define MSEC_PER_BUFFER (1000 * FRAMES_PER_BUFFER / SAMPLE_RATE) - -#if 0 -#define MIN_LATENCY_MSEC (200) -#define NUM_BUFFERS ((MIN_LATENCY_MSEC * SAMPLE_RATE) / (FRAMES_PER_BUFFER * 1000)) -#else -#define NUM_BUFFERS (0) -#endif - -#define MIN_FREQ (100.0f) -#define MAX_FREQ (4000.0f) -#define FREQ_SCALAR (1.00002f) -#define CalcPhaseIncrement(freq) (freq/SAMPLE_RATE) -#ifndef M_PI -#define M_PI (3.14159265) -#endif -#define TABLE_SIZE (400) -typedef struct -{ - float sine[TABLE_SIZE + 1]; // add one for guard point for interpolation - float phase_increment; - float left_phase; - float right_phase; - unsigned int framesToGo; -} -paTestData; -/* Convert phase between and 1.0 to sine value - * using linear interpolation. - */ -float LookupSine( paTestData *data, float phase ); -float LookupSine( paTestData *data, float phase ) -{ - float fIndex = phase*TABLE_SIZE; - int index = (int) fIndex; - float fract = fIndex - index; - float lo = data->sine[index]; - float hi = data->sine[index+1]; - float val = lo + fract*(hi-lo); - return val; -} -/* This routine will be called by the PortAudio engine when audio is needed. -** It may called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int patestCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData*)userData; - float *out = (float*)outputBuffer; - int framesToCalc; - int i; - int finished = 0; - (void) outTime; /* Prevent unused variable warnings. */ - (void) inputBuffer; - - if( data->framesToGo < framesPerBuffer ) - { - framesToCalc = data->framesToGo; - data->framesToGo = 0; - finished = 1; - } - else - { - framesToCalc = framesPerBuffer; - data->framesToGo -= framesPerBuffer; - } - - for( i=0; i<framesToCalc; i++ ) - { - *out++ = LookupSine(data, data->left_phase); /* left */ - *out++ = LookupSine(data, data->right_phase); /* right */ - data->left_phase += data->phase_increment; - if( data->left_phase >= 1.0f ) data->left_phase -= 1.0f; - data->right_phase += (data->phase_increment * 1.5f); /* fifth above */ - if( data->right_phase >= 1.0f ) data->right_phase -= 1.0f; - /* sweep frequency then start over. */ - data->phase_increment *= FREQ_SCALAR; - if( data->phase_increment > CalcPhaseIncrement(MAX_FREQ) ) data->phase_increment = CalcPhaseIncrement(MIN_FREQ); - } - /* zero remainder of final buffer */ - for( ; i<(int)framesPerBuffer; i++ ) - { - *out++ = 0; /* left */ - *out++ = 0; /* right */ - } - // Pa_Sleep( 3 * MSEC_PER_BUFFER / 4 ); - // Pa_Sleep( MSEC_PER_BUFFER / 3 ); - - return finished; -} -/*******************************************************************/ -int main(void); -int main(void) -{ - PortAudioStream *stream; - PaError err; - paTestData data; - int i; - int totalSamps; - printf("PortAudio Test: output sine sweep. ask for %d buffers\n", NUM_BUFFERS ); - printf("MSEC_PER_BUFFER = %d\n", MSEC_PER_BUFFER ); - - /* initialise sinusoidal wavetable */ - for( i=0; i<TABLE_SIZE; i++ ) - { - data.sine[i] = (float) sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. ); - } - data.sine[TABLE_SIZE] = data.sine[0]; // set guard point - data.left_phase = data.right_phase = 0.0; - data.phase_increment = CalcPhaseIncrement(MIN_FREQ); - data.framesToGo = totalSamps = NUM_SECONDS * SAMPLE_RATE; /* Play for a few seconds. */ - printf("totalSamps = %d\n", totalSamps ); - err = Pa_Initialize(); - if( err != paNoError ) goto error; - printf("PortAudio Test: output device = %d\n", OUTPUT_DEVICE ); - err = Pa_OpenStream( - &stream, - paNoDevice, - 0, /* no input */ - paFloat32, /* 32 bit floating point input */ - NULL, - OUTPUT_DEVICE, - 2, /* stereo output */ - paFloat32, /* 32 bit floating point output */ - NULL, - SAMPLE_RATE, - FRAMES_PER_BUFFER, - NUM_BUFFERS, /* number of buffers, if zero then use default minimum */ - paClipOff|paDitherOff, /* we won't output out of range samples so don't bother clipping them */ - patestCallback, - &data ); - if( err != paNoError ) goto error; - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - printf("Is callback being called?\n"); - for( i=0; i<((NUM_SECONDS+1)*1000); i+=SLEEP_DUR ) - { - printf("data.framesToGo = %d\n", data.framesToGo ); fflush(stdout); - Pa_Sleep( SLEEP_DUR ); - } - /* Stop sound until ENTER hit. */ - printf("Call Pa_StopStream()\n"); - err = Pa_StopStream( stream ); - if( err != paNoError ) goto error; - Pa_Terminate(); - printf("Test finished.\n"); - return err; -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return err; -} diff --git a/pd/portaudio_v18/pa_tests/debug_sine_amp.c b/pd/portaudio_v18/pa_tests/debug_sine_amp.c deleted file mode 100644 index 3eeca007..00000000 --- a/pd/portaudio_v18/pa_tests/debug_sine_amp.c +++ /dev/null @@ -1,157 +0,0 @@ -/* - * $Id: debug_sine_amp.c,v 1.1 2002/03/21 00:44:35 philburk Exp $ - * Play a different sine wave on each channels, - * using the Portable Audio api. - * Allos amplitude to be set interactively. - * - * Author: Phil Burk http://www.softsynth.com - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.portaudio.com - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <math.h> -#include "portaudio.h" - -#define OUTPUT_DEVICE (Pa_GetDefaultOutputDeviceID()) -#define SAMPLE_RATE (44100) -#define FRAMES_PER_BUFFER (256) -#define FREQ_INCR (300.0 / SAMPLE_RATE) -#define MAX_CHANNELS (64) - -#ifndef M_PI -#define M_PI (3.14159265) -#endif - -typedef struct -{ - int numChannels; - double phases[MAX_CHANNELS]; - float amplitude; -} -paTestData; - -/* This routine will be called by the PortAudio engine when audio is needed. -** It may called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int patestCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData*)userData; - float *out = (float*)outputBuffer; - int frameIndex, channelIndex; - (void) outTime; /* Prevent unused variable warnings. */ - (void) inputBuffer; - - for( frameIndex=0; frameIndex<(int)framesPerBuffer; frameIndex++ ) - { - for( channelIndex=0; channelIndex<data->numChannels; channelIndex++ ) - { - /* Output sine wave on every channel. */ - *out++ = (float) ( data->amplitude * sin(data->phases[channelIndex]) ); - - /* Play each channel at a higher frequency. */ - data->phases[channelIndex] += FREQ_INCR * (4 + channelIndex); - if( data->phases[channelIndex] >= (2.0 * M_PI) ) data->phases[channelIndex] -= (2.0 * M_PI); - } - } - - return 0; -} -/*******************************************************************/ -int main(void); -int main(void) -{ - char pad[256]; - PortAudioStream *stream; - PaError err; - const PaDeviceInfo *pdi; - paTestData data = {0}; - printf("PortAudio Test: output sine wave on each channel.\n" ); - - err = Pa_Initialize(); - if( err != paNoError ) goto error; - - pdi = Pa_GetDeviceInfo( OUTPUT_DEVICE ); - data.numChannels = pdi->maxOutputChannels; - if( data.numChannels > MAX_CHANNELS ) data.numChannels = MAX_CHANNELS; - printf("Number of Channels = %d\n", data.numChannels ); - data.amplitude = 1.0; - - err = Pa_OpenStream( - &stream, - paNoDevice, /* default input device */ - 0, /* no input */ - paFloat32, /* 32 bit floating point input */ - NULL, - OUTPUT_DEVICE, - data.numChannels, - paFloat32, /* 32 bit floating point output */ - NULL, - SAMPLE_RATE, - FRAMES_PER_BUFFER, /* frames per buffer */ - 0, /* number of buffers, if zero then use default minimum */ - paClipOff, /* we won't output out of range samples so don't bother clipping them */ - patestCallback, - &data ); - if( err != paNoError ) goto error; - - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - - do - { - printf("Current amplitude = %f\n", data.amplitude ); - printf("Enter new amplitude or 'q' to quit.\n"); - fflush(stdout); - gets( pad ); - if( pad[0] != 'q' ) - { - // I tried to use atof but it seems to be broken on Mac OS X 10.1 - float amp; - sscanf( pad, "%f", & ); - data.amplitude = amp; - } - } while( pad[0] != 'q' ); - - err = Pa_StopStream( stream ); - if( err != paNoError ) goto error; - - Pa_CloseStream( stream ); - Pa_Terminate(); - printf("Test finished.\n"); - return err; -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return err; -} diff --git a/pd/portaudio_v18/pa_tests/debug_sine_formats.c b/pd/portaudio_v18/pa_tests/debug_sine_formats.c deleted file mode 100644 index c331dde3..00000000 --- a/pd/portaudio_v18/pa_tests/debug_sine_formats.c +++ /dev/null @@ -1,202 +0,0 @@ -/* - * $Id: debug_sine_formats.c,v 1.1 2002/03/21 00:44:35 philburk Exp $ - * patest_sine_formats.c - * Play a sine wave using the Portable Audio api for several seconds. - * Test various data formats. - * - * Author: Phil Burk - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.audiomulch.com/portaudio/ - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <math.h> -#include "portaudio.h" - -#define NUM_SECONDS (10) -#define SAMPLE_RATE (44100) -#define FRAMES_PER_BUFFER (256) - -#define LEFT_FREQ (SAMPLE_RATE/512.0) /* So we hit 1.0 */ -#define RIGHT_FREQ (500.0) - -#define AMPLITUDE (1.0) - -/* Select ONE format for testing. */ -#define TEST_UINT8 (1) -#define TEST_INT8 (0) -#define TEST_INT16 (0) -#define TEST_FLOAT32 (0) - -#if TEST_UINT8 -#define TEST_FORMAT paUInt8 -typedef unsigned char SAMPLE_t; -#define SAMPLE_ZERO (0x80) -#define DOUBLE_TO_SAMPLE(x) (SAMPLE_ZERO + (SAMPLE_t)(127.0 * (x))) -#define FORMAT_NAME "Unsigned 8 Bit" - -#elif TEST_INT8 -#define TEST_FORMAT paInt8 -typedef char SAMPLE_t; -#define SAMPLE_ZERO (0) -#define DOUBLE_TO_SAMPLE(x) (SAMPLE_ZERO + (SAMPLE_t)(127.0 * (x))) -#define FORMAT_NAME "Signed 8 Bit" - -#elif TEST_INT16 -#define TEST_FORMAT paInt16 -typedef short SAMPLE_t; -#define SAMPLE_ZERO (0) -#define DOUBLE_TO_SAMPLE(x) (SAMPLE_ZERO + (SAMPLE_t)(32767 * (x))) -#define FORMAT_NAME "Signed 16 Bit" - -#elif TEST_FLOAT32 -#define TEST_FORMAT paFloat32 -typedef float SAMPLE_t; -#define SAMPLE_ZERO (0.0) -#define DOUBLE_TO_SAMPLE(x) ((SAMPLE_t)(x)) -#define FORMAT_NAME "Float 32 Bit" -#endif - -#ifndef M_PI -#define M_PI (3.14159265) -#endif - - -typedef struct -{ - double left_phase; - double right_phase; - unsigned int framesToGo; -} -paTestData; -/* This routine will be called by the PortAudio engine when audio is needed. -** It may called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int patestCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData*)userData; - SAMPLE_t *out = (SAMPLE_t *)outputBuffer; - SAMPLE_t sample; - int i; - int framesToCalc; - int finished = 0; - (void) outTime; /* Prevent unused variable warnings. */ - (void) inputBuffer; - - if( data->framesToGo < framesPerBuffer ) - { - framesToCalc = data->framesToGo; - data->framesToGo = 0; - finished = 1; - } - else - { - framesToCalc = framesPerBuffer; - data->framesToGo -= framesPerBuffer; - } - - for( i=0; i<framesToCalc; i++ ) - { - data->left_phase += (LEFT_FREQ / SAMPLE_RATE); - if( data->left_phase > 1.0) data->left_phase -= 1.0; - sample = DOUBLE_TO_SAMPLE( AMPLITUDE * sin( (data->left_phase * M_PI * 2. ))); /**/ - *out++ = sample; -/* *out++ = sample; /**/ -/* *out++ = 0; /**/ - - data->right_phase += (RIGHT_FREQ / SAMPLE_RATE); - if( data->right_phase > 1.0) data->right_phase -= 1.0; - *out++ = DOUBLE_TO_SAMPLE( AMPLITUDE * sin( (data->right_phase * M_PI * 2. ))); /**/ -/* *out++ = 0; /* */ - } - /* zero remainder of final buffer */ - for( ; i<(int)framesPerBuffer; i++ ) - { - *out++ = SAMPLE_ZERO; /* left */ - *out++ = SAMPLE_ZERO; /* right */ - } - return finished; -} -/*******************************************************************/ -int main(void); -int main(void) -{ - PortAudioStream *stream; - PaError err; - paTestData data; - int totalSamps; - - printf("PortAudio Test: output " FORMAT_NAME "\n"); - - - data.left_phase = data.right_phase = 0.0; - data.framesToGo = totalSamps = NUM_SECONDS * SAMPLE_RATE; /* Play for a few seconds. */ - err = Pa_Initialize(); - if( err != paNoError ) goto error; - - err = Pa_OpenStream( - &stream, - paNoDevice,/* default input device */ - 0, /* no input */ - TEST_FORMAT, - NULL, - Pa_GetDefaultOutputDeviceID(), /* default output device */ - 2, /* stereo output */ - TEST_FORMAT, - NULL, - SAMPLE_RATE, - FRAMES_PER_BUFFER, - 0, /* number of buffers, if zero then use default minimum */ - paClipOff, /* we won't output out of range samples so don't bother clipping them */ - patestCallback, - &data ); - if( err != paNoError ) goto error; - - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - - printf("Waiting %d seconds for sound to finish.\n", NUM_SECONDS ); - while( Pa_StreamActive( stream ) ) Pa_Sleep(10); - - err = Pa_CloseStream( stream ); - if( err != paNoError ) goto error; - Pa_Terminate(); - - printf("PortAudio Test Finished: " FORMAT_NAME "\n"); - - return err; -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return err; -} diff --git a/pd/portaudio_v18/pa_tests/debug_sine_getchar.c b/pd/portaudio_v18/pa_tests/debug_sine_getchar.c deleted file mode 100644 index 54c0cba5..00000000 --- a/pd/portaudio_v18/pa_tests/debug_sine_getchar.c +++ /dev/null @@ -1,140 +0,0 @@ -/* - * $Id: debug_sine_getchar.c,v 1.1.2.2 2003/04/10 23:09:40 philburk Exp $ - * - * Play a sine wave using the Portable Audio api until ENTER hit. - * - * Authors: - * Ross Bencina <rossb@audiomulch.com> - * Phil Burk <philburk@softsynth.com> - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.audiomulch.com/portaudio/ - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <math.h> -#include "portaudio.h" - -#define SAMPLE_RATE (48000) -#define AMPLITUDE (0.3) -#define FRAMES_PER_BUFFER (64) -#define OUTPUT_DEVICE Pa_GetDefaultOutputDeviceID() -//#define OUTPUT_DEVICE (2) - -#ifndef M_PI -#define M_PI (3.14159265) -#endif - -#define TABLE_SIZE (200) -typedef struct -{ - float sine[TABLE_SIZE]; - int left_phase; - int right_phase; -} -paTestData; - -/* This routine will be called by the PortAudio engine when audio is needed. -** It may called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int patestCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData*)userData; - float *out = (float*)outputBuffer; - unsigned long i; - int finished = 0; - (void) outTime; /* Prevent unused variable warnings. */ - (void) inputBuffer; - for( i=0; i<framesPerBuffer; i++ ) - { - *out++ = data->sine[data->left_phase]; /* left */ - *out++ = data->sine[data->right_phase]; /* right */ - data->left_phase += 1; - if( data->left_phase >= TABLE_SIZE ) data->left_phase -= TABLE_SIZE; - data->right_phase += 3; /* higher pitch so we can distinguish left and right. */ - if( data->right_phase >= TABLE_SIZE ) data->right_phase -= TABLE_SIZE; - } - return finished; -} - -/*******************************************************************/ -int main(void); -int main(void) -{ - PortAudioStream *stream; - PaError err; - paTestData data; - int i; - printf("PortAudio Test: output sine wave. SR = %d, BufSize = %d, devID = %d\n", - SAMPLE_RATE, FRAMES_PER_BUFFER, OUTPUT_DEVICE); - /* initialise sinusoidal wavetable */ - for( i=0; i<TABLE_SIZE; i++ ) - { - data.sine[i] = (float) (AMPLITUDE * sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. )); - } - data.left_phase = data.right_phase = 0; - err = Pa_Initialize(); - if( err != paNoError ) goto error; - err = Pa_OpenStream( - &stream, - paNoDevice,/* default input device */ - 0, /* no input */ - paFloat32, /* 32 bit floating point input */ - NULL, - OUTPUT_DEVICE, - 2, /* stereo output */ - paFloat32, /* 32 bit floating point output */ - NULL, - SAMPLE_RATE, - FRAMES_PER_BUFFER, - 0, /* number of buffers, if zero then use default minimum */ - paClipOff, /* we won't output out of range samples so don't bother clipping them */ - patestCallback, - &data ); - if( err != paNoError ) goto error; - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - printf("Press ENTER to stop.\n" ); fflush(stdout); - getchar(); - - err = Pa_StopStream( stream ); - if( err != paNoError ) goto error; - err = Pa_CloseStream( stream ); - if( err != paNoError ) goto error; - Pa_Terminate(); - printf("Test finished.\n"); - return err; -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return err; -} diff --git a/pd/portaudio_v18/pa_tests/debug_srate.c b/pd/portaudio_v18/pa_tests/debug_srate.c deleted file mode 100644 index ae42611b..00000000 --- a/pd/portaudio_v18/pa_tests/debug_srate.c +++ /dev/null @@ -1,265 +0,0 @@ -/* - * $Id: debug_srate.c,v 1.1 2002/05/02 20:16:29 philburk Exp $ - * debug_record_reuse.c - * Record input into an array. - * Save array to a file. - * Based on patest_record.c but with various ugly debug hacks thrown in. - * Loop twice and reuse same streams. - * - * Author: Phil Burk http://www.softsynth.com - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.portaudio.com - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <stdlib.h> -#include <memory.h> -#include "portaudio.h" - -#define EWS88MT_12_REC (1) -#define EWS88MT_12_PLAY (10) -#define SBLIVE_REC (2) -#define SBLIVE_PLAY (11) - -#if 0 -#define INPUT_DEVICE_ID Pa_GetDefaultInputDeviceID() -#define OUTPUT_DEVICE_ID Pa_GetDefaultOutputDeviceID() -#else -#define INPUT_DEVICE_ID (EWS88MT_12_REC) -#define OUTPUT_DEVICE_ID (SBLIVE_PLAY) -#endif - -#define INPUT_SAMPLE_RATE (22050.0) -#define OUTPUT_SAMPLE_RATE (22050.0) -#define NUM_SECONDS (4) -#define SLEEP_DUR_MSEC (1000) -#define FRAMES_PER_BUFFER (64) -#define NUM_REC_BUFS (0) -#define SAMPLES_PER_FRAME (2) - -#define PA_SAMPLE_TYPE paInt16 -typedef short SAMPLE; - -typedef struct -{ - long frameIndex; /* Index into sample array. */ -} -paTestData; - -/* This routine will be called by the PortAudio engine when audio is needed. -** It may be called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int recordCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData *) userData; - (void) outputBuffer; /* Prevent unused variable warnings. */ - (void) outTime; - - if( inputBuffer != NULL ) - { - data->frameIndex += framesPerBuffer; - } - return 0; -} - -/* This routine will be called by the PortAudio engine when audio is needed. -** It may be called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int playCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData *) userData; - (void) inputBuffer; /* Prevent unused variable warnings. */ - (void) outTime; - - if( outputBuffer != NULL ) - { - data->frameIndex += framesPerBuffer; - } - return 0; -} - -/****************************************************************/ -PaError MeasureStreamRate( PortAudioStream *stream, paTestData *dataPtr, double *ratePtr ) -{ - PaError err; - int i; - int totalFrames = 0; - int totalMSec = 0; - - dataPtr->frameIndex = 0; - - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - - for( i=0; i<(NUM_SECONDS*1000/SLEEP_DUR_MSEC); i++ ) - { - int delta, endIndex; - - int startIndex = dataPtr->frameIndex; - Pa_Sleep(SLEEP_DUR_MSEC); - endIndex = dataPtr->frameIndex; - - delta = endIndex - startIndex; - totalFrames += delta; - totalMSec += SLEEP_DUR_MSEC; - - printf("index = %d, delta = %d\n", endIndex, delta ); fflush(stdout); - } - - err = Pa_StopStream( stream ); - if( err != paNoError ) goto error; - - *ratePtr = (totalFrames * 1000.0) / totalMSec; - -error: - return err; -} - -void ReportRate( double measuredRate, double expectedRate ) -{ - double error; - - error = (measuredRate - expectedRate) / expectedRate; - error = (error < 0 ) ? -error : error; - - printf("Measured rate = %6.1f, expected rate = %6.1f\n", - measuredRate, expectedRate ); - if( error > 0.1 ) - { - printf("ERROR: unexpected rate! --------------------- ERROR!\n"); - } - else - { - printf("SUCCESS: rate within tolerance!\n"); - } -} - -/*******************************************************************/ -int main(void); -int main(void) -{ - PaError err; - paTestData data = { 0 }; - long i; - double rate; - const PaDeviceInfo *pdi; - - PortAudioStream *outputStream; - PortAudioStream *inputStream; - - err = Pa_Initialize(); - if( err != paNoError ) goto error; - - - pdi = Pa_GetDeviceInfo( INPUT_DEVICE_ID ); - printf("Input device = %s\n", pdi->name ); - pdi = Pa_GetDeviceInfo( OUTPUT_DEVICE_ID ); - printf("Output device = %s\n", pdi->name ); - -/* Open input stream. */ - err = Pa_OpenStream( - &inputStream, - INPUT_DEVICE_ID, - SAMPLES_PER_FRAME, /* stereo input */ - PA_SAMPLE_TYPE, - NULL, - paNoDevice, - 0, - PA_SAMPLE_TYPE, - NULL, - INPUT_SAMPLE_RATE, - FRAMES_PER_BUFFER, /* frames per buffer */ - NUM_REC_BUFS, /* number of buffers, if zero then use default minimum */ - paClipOff, /* we won't output out of range samples so don't bother clipping them */ - recordCallback, - &data ); - if( err != paNoError ) goto error; - - err = Pa_OpenStream( - &outputStream, - paNoDevice, - 0, /* NO input */ - PA_SAMPLE_TYPE, - NULL, - OUTPUT_DEVICE_ID, - SAMPLES_PER_FRAME, /* stereo output */ - PA_SAMPLE_TYPE, - NULL, - OUTPUT_SAMPLE_RATE, - FRAMES_PER_BUFFER, /* frames per buffer */ - 0, /* number of buffers, if zero then use default minimum */ - paClipOff, /* we won't output out of range samples so don't bother clipping them */ - playCallback, - &data ); - if( err != paNoError ) goto error; - -/* Record and playback multiple times. */ - for( i=0; i<2; i++ ) - { - printf("Measuring INPUT ------------------------- \n"); - err = MeasureStreamRate( inputStream, &data, &rate ); - if( err != paNoError ) goto error; - ReportRate( rate, INPUT_SAMPLE_RATE ); - - printf("Measuring OUTPUT ------------------------- \n"); - err = MeasureStreamRate( outputStream, &data, &rate ); - if( err != paNoError ) goto error; - ReportRate( rate, OUTPUT_SAMPLE_RATE ); - } - -/* Clean up. */ - err = Pa_CloseStream( inputStream ); - if( err != paNoError ) goto error; - - err = Pa_CloseStream( outputStream ); - if( err != paNoError ) goto error; - - if( err != paNoError ) goto error; - - Pa_Terminate(); - - printf("Test complete.\n"); fflush(stdout); - return 0; - -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - if( err == paHostError ) - { - fprintf( stderr, "Host Error number: %d\n", Pa_GetHostError() ); - } - return -1; -} diff --git a/pd/portaudio_v18/pa_tests/debug_test1.c b/pd/portaudio_v18/pa_tests/debug_test1.c deleted file mode 100644 index 187d0f5c..00000000 --- a/pd/portaudio_v18/pa_tests/debug_test1.c +++ /dev/null @@ -1,114 +0,0 @@ -/* - * $Id: debug_test1.c,v 1.1.1.1 2002/01/22 00:52:30 phil Exp $ - patest1.c - Ring modulate the audio input with a 441hz sine wave for 20 seconds - using the Portable Audio api - Author: Ross Bencina <rossb@audiomulch.com> - Modifications: - April 5th, 2001 - PLB - Check for NULL inputBuffer. -*/ -#include <stdio.h> -#include <math.h> -#include "portaudio.h" -#ifndef M_PI -#define M_PI (3.14159265) -#endif -typedef struct -{ - float sine[100]; - int phase; - int sampsToGo; -} -patest1data; -static int patest1Callback( void *inputBuffer, void *outputBuffer, - unsigned long bufferFrames, - PaTimestamp outTime, void *userData ) -{ - patest1data *data = (patest1data*)userData; - float *in = (float*)inputBuffer; - float *out = (float*)outputBuffer; - int framesToCalc = bufferFrames; - unsigned long i; - int finished = 0; - if(inputBuffer == NULL) return 0; - if( data->sampsToGo < bufferFrames ) - { - finished = 1; - } - for( i=0; i<bufferFrames; i++ ) - { - *out++ = *in++; - *out++ = *in++; - if( data->phase >= 100 ) - data->phase = 0; - } - data->sampsToGo -= bufferFrames; - /* zero remainder of final buffer if not already done */ - for( ; i<bufferFrames; i++ ) - { - *out++ = 0; /* left */ - *out++ = 0; /* right */ - } - return finished; -} -int main(int argc, char* argv[]); -int main(int argc, char* argv[]) -{ - PaStream *stream; - PaError err; - patest1data data; - int i; - int inputDevice = Pa_GetDefaultInputDeviceID(); - int outputDevice = Pa_GetDefaultOutputDeviceID(); - /* initialise sinusoidal wavetable */ - for( i=0; i<100; i++ ) - data.sine[i] = sin( ((double)i/100.) * M_PI * 2. ); - data.phase = 0; - data.sampsToGo = 44100 * 4; // 20 seconds - /* initialise portaudio subsytem */ - Pa_Initialize(); - err = Pa_OpenStream( - &stream, - inputDevice, - 2, /* stereo input */ - paFloat32, /* 32 bit floating point input */ - NULL, - outputDevice, - 2, /* stereo output */ - paFloat32, /* 32 bit floating point output */ - NULL, - 44100., - // 22050, /* half second buffers */ - // 4, /* four buffers */ - 512, /* half second buffers */ - 0, /* four buffers */ - paClipOff, /* we won't output out of range samples so don't bother clipping them */ - patest1Callback, - &data ); - if( err == paNoError ) - { - err = Pa_StartStream( stream ); - // printf( "Press any key to end.\n" ); - // getc( stdin ); //wait for input before exiting - // Pa_AbortStream( stream ); - - printf( "Waiting for stream to complete...\n" ); - - while( Pa_StreamActive( stream ) ) - Pa_Sleep(1000); /* sleep until playback has finished */ - - err = Pa_CloseStream( stream ); - } - else - { - fprintf( stderr, "An error occured while opening the portaudio stream\n" ); - if( err == paHostError ) - fprintf( stderr, "Host error number: %d\n", Pa_GetHostError() ); - else - fprintf( stderr, "Error number: %d\n", err ); - } - Pa_Terminate(); - printf( "bye\n" ); - - return 0; -} diff --git a/pd/portaudio_v18/pa_tests/pa_devs.c b/pd/portaudio_v18/pa_tests/pa_devs.c deleted file mode 100644 index 66302395..00000000 --- a/pd/portaudio_v18/pa_tests/pa_devs.c +++ /dev/null @@ -1,99 +0,0 @@ -/* - * $Id: pa_devs.c,v 1.1.1.1.4.1 2003/02/11 21:41:32 philburk Exp $ - * pa_devs.c - * List available devices. - * - * Author: Phil Burk http://www.softsynth.com - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.portaudio.com - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <math.h> -#include "portaudio.h" - -/*******************************************************************/ -int main(void); -int main(void) -{ - int i,j; - int numDevices; - const PaDeviceInfo *pdi; - PaError err; - Pa_Initialize(); - numDevices = Pa_CountDevices(); - if( numDevices < 0 ) - { - printf("ERROR: Pa_CountDevices returned 0x%x\n", numDevices ); - err = numDevices; - goto error; - } - printf("Number of devices = %d\n", numDevices ); - for( i=0; i<numDevices; i++ ) - { - pdi = Pa_GetDeviceInfo( i ); - printf("---------------------------------------------- #%d", i ); - if( i == Pa_GetDefaultInputDeviceID() ) printf(" DefaultInput"); - if( i == Pa_GetDefaultOutputDeviceID() ) printf(" DefaultOutput"); - printf("\nName = %s\n", pdi->name ); - printf("Max Inputs = %d", pdi->maxInputChannels ); - printf(", Max Outputs = %d\n", pdi->maxOutputChannels ); - if( pdi->numSampleRates == -1 ) - { - printf("Sample Rate Range = %f to %f\n", pdi->sampleRates[0], pdi->sampleRates[1] ); - } - else - { - printf("Sample Rates ="); - for( j=0; j<pdi->numSampleRates; j++ ) - { - printf(" %8.2f,", pdi->sampleRates[j] ); - } - printf("\n"); - } - printf("Native Sample Formats = "); - if( pdi->nativeSampleFormats & paInt8 ) printf("paInt8, "); - if( pdi->nativeSampleFormats & paUInt8 ) printf("paUInt8, "); - if( pdi->nativeSampleFormats & paInt16 ) printf("paInt16, "); - if( pdi->nativeSampleFormats & paInt32 ) printf("paInt32, "); - if( pdi->nativeSampleFormats & paFloat32 ) printf("paFloat32, "); - if( pdi->nativeSampleFormats & paInt24 ) printf("paInt24, "); - if( pdi->nativeSampleFormats & paPackedInt24 ) printf("paPackedInt24, "); - printf("\n"); - } - Pa_Terminate(); - - printf("----------------------------------------------\n"); - return 0; -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return err; -} diff --git a/pd/portaudio_v18/pa_tests/pa_fuzz.c b/pd/portaudio_v18/pa_tests/pa_fuzz.c deleted file mode 100644 index 3f0cdb56..00000000 --- a/pd/portaudio_v18/pa_tests/pa_fuzz.c +++ /dev/null @@ -1,156 +0,0 @@ -/* - * $Id: pa_fuzz.c,v 1.1.1.1.4.1 2003/02/11 21:41:32 philburk Exp $ - * pa_fuzz.c - * Distort input like a fuzz boz. - * - * Author: Phil Burk http://www.softsynth.com - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.portaudio.com - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <math.h> -#include "portaudio.h" -/* -** Note that many of the older ISA sound cards on PCs do NOT support -** full duplex audio (simultaneous record and playback). -** And some only support full duplex at lower sample rates. -*/ -#define SAMPLE_RATE (44100) -#define PA_SAMPLE_TYPE paFloat32 -#define FRAMES_PER_BUFFER (64) - -typedef float SAMPLE; - -float CubicAmplifier( float input ); -static int fuzzCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ); - -/* Non-linear amplifier with soft distortion curve. */ -float CubicAmplifier( float input ) -{ - float output, temp; - if( input < 0.0 ) - { - temp = input + 1.0f; - output = (temp * temp * temp) - 1.0f; - } - else - { - temp = input - 1.0f; - output = (temp * temp * temp) + 1.0f; - } - - return output; -} -#define FUZZ(x) CubicAmplifier(CubicAmplifier(CubicAmplifier(CubicAmplifier(x)))) - -static int gNumNoInputs = 0; -/* This routine will be called by the PortAudio engine when audio is needed. -** It may be called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int fuzzCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - SAMPLE *out = (SAMPLE*)outputBuffer; - SAMPLE *in = (SAMPLE*)inputBuffer; - unsigned int i; - (void) outTime; /* Prevent unused variable warnings. */ - (void) userData; - - if( inputBuffer == NULL ) - { - for( i=0; i<framesPerBuffer; i++ ) - { - *out++ = 0; /* left - silent */ - *out++ = 0; /* right - silent */ - } - gNumNoInputs += 1; - } - else - { - for( i=0; i<framesPerBuffer; i++ ) - { - *out++ = FUZZ(*in++); /* left - distorted */ - *out++ = *in++; /* right - clean */ - } - } - return 0; -} - -/*******************************************************************/ -int main(void); -int main(void) -{ - PortAudioStream *stream; - PaError err; - - err = Pa_Initialize(); - if( err != paNoError ) goto error; - - err = Pa_OpenStream( - &stream, - Pa_GetDefaultInputDeviceID(), /* default output device */ - 2, /* stereo input */ - PA_SAMPLE_TYPE, - NULL, - Pa_GetDefaultOutputDeviceID(), /* default output device */ - 2, /* stereo output */ - PA_SAMPLE_TYPE, - NULL, - SAMPLE_RATE, - FRAMES_PER_BUFFER, - 0, /* number of buffers, if zero then use default minimum */ - 0, // paClipOff, /* we won't output out of range samples so don't bother clipping them */ - fuzzCallback, - NULL ); - if( err != paNoError ) goto error; - - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - - printf("Hit ENTER to stop program.\n"); - fflush(stdout); - getchar(); - err = Pa_CloseStream( stream ); - if( err != paNoError ) goto error; - - printf("Finished. gNumNoInputs = %d\n", gNumNoInputs ); - Pa_Terminate(); - return 0; - -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return -1; -} diff --git a/pd/portaudio_v18/pa_tests/pa_minlat.c b/pd/portaudio_v18/pa_tests/pa_minlat.c deleted file mode 100644 index 7256cc00..00000000 --- a/pd/portaudio_v18/pa_tests/pa_minlat.c +++ /dev/null @@ -1,172 +0,0 @@ -/* - * $Id: pa_minlat.c,v 1.4 2002/04/30 18:19:00 philburk Exp $ - * paminlat.c - * Experiment with different numbers of buffers to determine the - * minimum latency for a computer. - * - * Author: Phil Burk http://www.softsynth.com - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.portaudio.com - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <stdlib.h> -#include <math.h> -#include "portaudio.h" - -#ifndef M_PI -#define M_PI (3.14159265) -#endif -#define TWOPI (M_PI * 2.0) - -#define DEFAULT_BUFFER_SIZE (64) - -typedef struct -{ - double left_phase; - double right_phase; -} -paTestData; - -/* Very simple synthesis routine to generate two sine waves. */ -static int paminlatCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData*)userData; - float *out = (float*)outputBuffer; - unsigned int i; - double left_phaseInc = 0.02; - double right_phaseInc = 0.06; - - double left_phase = data->left_phase; - double right_phase = data->right_phase; - - for( i=0; i<framesPerBuffer; i++ ) - { - left_phase += left_phaseInc; - if( left_phase > TWOPI ) left_phase -= TWOPI; - *out++ = (float) sin( left_phase ); - - right_phase += right_phaseInc; - if( right_phase > TWOPI ) right_phase -= TWOPI; - *out++ = (float) sin( right_phase ); - } - - data->left_phase = left_phase; - data->right_phase = right_phase; - return 0; -} -void main( int argc, char **argv ); -void main( int argc, char **argv ) -{ - PortAudioStream *stream; - PaError err; - paTestData data; - int go; - int numBuffers = 0; - int minBuffers = 0; - int framesPerBuffer; - double sampleRate = 44100.0; - char str[256]; - printf("paminlat - Determine minimum latency for your computer.\n"); - printf(" usage: paminlat {framesPerBuffer}\n"); - printf(" for example: paminlat 256\n"); - printf("Adjust your stereo until you hear a smooth tone in each speaker.\n"); - printf("Then try to find the smallest number of buffers that still sounds smooth.\n"); - printf("Note that the sound will stop momentarily when you change the number of buffers.\n"); - /* Get bufferSize from command line. */ - framesPerBuffer = ( argc > 1 ) ? atol( argv[1] ) : DEFAULT_BUFFER_SIZE; - printf("Frames per buffer = %d\n", framesPerBuffer ); - - data.left_phase = data.right_phase = 0.0; - err = Pa_Initialize(); - if( err != paNoError ) goto error; - /* Ask PortAudio for the recommended minimum number of buffers. */ - numBuffers = minBuffers = Pa_GetMinNumBuffers( framesPerBuffer, sampleRate ); - printf("NumBuffers set to %d based on a call to Pa_GetMinNumBuffers()\n", numBuffers ); - /* Try different numBuffers in a loop. */ - go = 1; - while( go ) - { - - printf("Latency = framesPerBuffer * numBuffers = %d * %d = %d frames = %d msecs.\n", - framesPerBuffer, numBuffers, framesPerBuffer*numBuffers, - (int)((1000 * framesPerBuffer * numBuffers) / sampleRate) ); - err = Pa_OpenStream( - &stream, - paNoDevice, - 0, /* no input */ - paFloat32, /* 32 bit floating point input */ - NULL, - Pa_GetDefaultOutputDeviceID(), /* default output device */ - 2, /* stereo output */ - paFloat32, /* 32 bit floating point output */ - NULL, - sampleRate, - framesPerBuffer, - numBuffers, /* number of buffers */ - paClipOff, /* we won't output out of range samples so don't bother clipping them */ - paminlatCallback, - &data ); - if( err != paNoError ) goto error; - if( stream == NULL ) goto error; - /* Start audio. */ - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - /* Ask user for a new number of buffers. */ - printf("\nMove windows around to see if the sound glitches.\n"); - printf("NumBuffers currently %d, enter new number, or 'q' to quit: ", numBuffers ); - gets( str ); - if( str[0] == 'q' ) go = 0; - else - { - numBuffers = atol( str ); - if( numBuffers < minBuffers ) - { - printf( "numBuffers below minimum of %d! Set to minimum!!!\n", minBuffers ); - numBuffers = minBuffers; - } - } - /* Stop sound until ENTER hit. */ - err = Pa_StopStream( stream ); - if( err != paNoError ) goto error; - err = Pa_CloseStream( stream ); - if( err != paNoError ) goto error; - } - printf("A good setting for latency would be somewhat higher than\n"); - printf("the minimum latency that worked.\n"); - printf("PortAudio: Test finished.\n"); - Pa_Terminate(); - return; -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); -} diff --git a/pd/portaudio_v18/pa_tests/paqa_devs.c b/pd/portaudio_v18/pa_tests/paqa_devs.c deleted file mode 100644 index 92eb6c7a..00000000 --- a/pd/portaudio_v18/pa_tests/paqa_devs.c +++ /dev/null @@ -1,322 +0,0 @@ -/* - * $Id: paqa_devs.c,v 1.1.1.1.4.1 2003/02/11 21:41:32 philburk Exp $ - * paqa_devs.c - * Self Testing Quality Assurance app for PortAudio - * Try to open each device and run through all the - * possible configurations. - * - * Author: Phil Burk http://www.softsynth.com - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.portaudio.com - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <math.h> -#include "portaudio.h" -#include "pa_trace.h" -/****************************************** Definitions ***********/ -#define MODE_INPUT (0) -#define MODE_OUTPUT (1) -typedef struct PaQaData -{ - unsigned long framesLeft; - int numChannels; - int bytesPerSample; - int mode; - short sawPhase; - PaSampleFormat format; -} PaQaData; - -/****************************************** Prototypes ***********/ -static void TestDevices( int mode ); -static void TestFormats( int mode, PaDeviceID deviceID, double sampleRate, - int numChannels ); -static int TestAdvance( int mode, PaDeviceID deviceID, double sampleRate, - int numChannels, PaSampleFormat format ); -static int QaCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ); - -/****************************************** Globals ***********/ -static int gNumPassed = 0; -static int gNumFailed = 0; - -/****************************************** Macros ***********/ -/* Print ERROR if it fails. Tally success or failure. */ -/* Odd do-while wrapper seems to be needed for some compilers. */ -#define EXPECT(_exp) \ - do \ - { \ - if ((_exp)) {\ - /* printf("SUCCESS for %s\n", #_exp ); */ \ - gNumPassed++; \ - } \ - else { \ - printf("ERROR - 0x%x - %s for %s\n", result, \ - ((result == 0) ? "-" : Pa_GetErrorText(result)), \ - #_exp ); \ - gNumFailed++; \ - goto error; \ - } \ - } while(0) -/*******************************************************************/ -/* This routine will be called by the PortAudio engine when audio is needed. -** It may be called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int QaCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - unsigned long i; - short phase; - PaQaData *data = (PaQaData *) userData; - (void) inputBuffer; - (void) outTime; - - /* Play simple sawtooth wave. */ - if( data->mode == MODE_OUTPUT ) - { - phase = data->sawPhase; - switch( data->format ) - { - case paFloat32: - { - float *out = (float *) outputBuffer; - for( i=0; i<framesPerBuffer; i++ ) - { - phase += 0x123; - *out++ = (float) (phase * (1.0 / 32768.0)); - if( data->numChannels == 2 ) - { - *out++ = (float) (phase * (1.0 / 32768.0)); - } - } - } - break; - - case paInt32: - { - int *out = (int *) outputBuffer; - for( i=0; i<framesPerBuffer; i++ ) - { - phase += 0x123; - *out++ = ((int) phase ) << 16; - if( data->numChannels == 2 ) - { - *out++ = ((int) phase ) << 16; - } - } - } - break; - case paInt16: - { - short *out = (short *) outputBuffer; - for( i=0; i<framesPerBuffer; i++ ) - { - phase += 0x123; - *out++ = phase; - if( data->numChannels == 2 ) - { - *out++ = phase; - } - } - } - break; - - default: - { - unsigned char *out = (unsigned char *) outputBuffer; - unsigned long numBytes = framesPerBuffer * data->numChannels * data->bytesPerSample; - for( i=0; i<numBytes; i++ ) - { - *out++ = 0; - } - } - break; - } - data->sawPhase = phase; - } - /* Are we through yet? */ - if( data->framesLeft > framesPerBuffer ) - { - AddTraceMessage("QaCallback: running. framesLeft", data->framesLeft ); - data->framesLeft -= framesPerBuffer; - return 0; - } - else - { - AddTraceMessage("QaCallback: DONE! framesLeft", data->framesLeft ); - data->framesLeft = 0; - return 1; - } -} -/*******************************************************************/ -int main(void); -int main(void) -{ - PaError result; - EXPECT( ((result=Pa_Initialize()) == 0) ); - printf("Test OUTPUT ---------------\n"); - TestDevices( MODE_OUTPUT ); - printf("Test INPUT ---------------\n"); - TestDevices( MODE_INPUT ); -error: - Pa_Terminate(); - printf("QA Report: %d passed, %d failed.\n", gNumPassed, gNumFailed ); -} -/******************************************************************* -* Try each output device, through its full range of capabilities. */ -static void TestDevices( int mode ) -{ - int id,jc,kr; - int maxChannels; - const PaDeviceInfo *pdi; - int numDevices = Pa_CountDevices(); - /* Iterate through all devices. */ - for( id=0; id<numDevices; id++ ) - { - pdi = Pa_GetDeviceInfo( id ); - /* Try 1 to maxChannels on each device. */ - maxChannels = ( mode == MODE_INPUT ) ? pdi->maxInputChannels : pdi->maxOutputChannels; - for( jc=1; jc<=maxChannels; jc++ ) - { - printf("Name = %s\n", pdi->name ); - /* Try each legal sample rate. */ - if( pdi->numSampleRates == -1 ) - { - double low, high; - low = pdi->sampleRates[0]; - high = pdi->sampleRates[1]; - if( low < 8000.0 ) low = 8000.0; - TestFormats( mode, id, low, jc ); -#define TESTSR(sr) {if(((sr)>=low) && ((sr)<=high)) TestFormats( mode, id, (sr), jc ); } - - TESTSR(11025.0); - TESTSR(22050.0); - TESTSR(34567.0); - TESTSR(44100.0); - TestFormats( mode, id, high, jc ); - } - else - { - for( kr=0; kr<pdi->numSampleRates; kr++ ) - { - TestFormats( mode, id, pdi->sampleRates[kr], jc ); - } - } - } - } -} -/*******************************************************************/ -static void TestFormats( int mode, PaDeviceID deviceID, double sampleRate, - int numChannels ) -{ - TestAdvance( mode, deviceID, sampleRate, numChannels, paFloat32 ); /* */ - TestAdvance( mode, deviceID, sampleRate, numChannels, paInt16 ); /* */ - TestAdvance( mode, deviceID, sampleRate, numChannels, paInt32 ); /* */ -} -/*******************************************************************/ -static int TestAdvance( int mode, PaDeviceID deviceID, double sampleRate, - int numChannels, PaSampleFormat format ) -{ - PortAudioStream *stream = NULL; - PaError result; - PaQaData myData; -#define FRAMES_PER_BUFFER (64) - printf("------ TestAdvance: %s, device = %d, rate = %g, numChannels = %d, format = %d -------\n", - ( mode == MODE_INPUT ) ? "INPUT" : "OUTPUT", - deviceID, sampleRate, numChannels, format); - fflush(stdout); - /* Setup data for synthesis thread. */ - myData.framesLeft = (unsigned long) (sampleRate * 100); /* 100 seconds */ - myData.numChannels = numChannels; - myData.mode = mode; - myData.format = format; - switch( format ) - { - case paFloat32: - case paInt32: - case paInt24: - myData.bytesPerSample = 4; - break; - case paPackedInt24: - myData.bytesPerSample = 3; - break; - default: - myData.bytesPerSample = 2; - break; - } - EXPECT( ((result = Pa_OpenStream( - &stream, - ( mode == MODE_INPUT ) ? deviceID : paNoDevice, - ( mode == MODE_INPUT ) ? numChannels : 0, - format, - NULL, - ( mode == MODE_OUTPUT ) ? deviceID : paNoDevice, - ( mode == MODE_OUTPUT ) ? numChannels : 0, - format, - NULL, - sampleRate, - FRAMES_PER_BUFFER, /* frames per buffer */ - 0, /* number of buffers, if zero then use default minimum */ - paClipOff, /* we won't output out of range samples so don't bother clipping them */ - QaCallback, - &myData ) - ) == 0) ); - if( stream ) - { - PaTimestamp oldStamp, newStamp; - unsigned long oldFrames; - int minDelay = ( mode == MODE_INPUT ) ? 1000 : 400; - int minNumBuffers = Pa_GetMinNumBuffers( FRAMES_PER_BUFFER, sampleRate ); - int msec = (int) ((minNumBuffers * 3 * 1000.0 * FRAMES_PER_BUFFER) / sampleRate); - if( msec < minDelay ) msec = minDelay; - printf("msec = %d\n", msec); /**/ - EXPECT( ((result=Pa_StartStream( stream )) == 0) ); - /* Check to make sure PortAudio is advancing timeStamp. */ - result = paNoError; - oldStamp = Pa_StreamTime(stream); - fflush(stdout); - Pa_Sleep(msec); - newStamp = Pa_StreamTime(stream); - printf("oldStamp = %g,newStamp = %g\n", oldStamp, newStamp ); /**/ - EXPECT( (oldStamp < newStamp) ); - /* Check to make sure callback is decrementing framesLeft. */ - oldFrames = myData.framesLeft; - Pa_Sleep(msec); - printf("oldFrames = %d, myData.framesLeft = %d\n", oldFrames, myData.framesLeft ); /**/ - EXPECT( (oldFrames > myData.framesLeft) ); - EXPECT( ((result=Pa_CloseStream( stream )) == 0) ); - stream = NULL; - } -error: - if( stream != NULL ) Pa_CloseStream( stream ); - fflush(stdout); - return result; -} diff --git a/pd/portaudio_v18/pa_tests/paqa_errs.c b/pd/portaudio_v18/pa_tests/paqa_errs.c deleted file mode 100644 index a26779cc..00000000 --- a/pd/portaudio_v18/pa_tests/paqa_errs.c +++ /dev/null @@ -1,330 +0,0 @@ -/* - * $Id: paqa_errs.c,v 1.2.4.1 2003/02/11 21:41:32 philburk Exp $ - * paqa_devs.c - * Self Testing Quality Assurance app for PortAudio - * Do lots of bad things to test error reporting. - * - * Author: Phil Burk http://www.softsynth.com - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.portaudio.com - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <math.h> -#include "portaudio.h" -/****************************************** Definitions ***********/ -#define MODE_INPUT (0) -#define MODE_OUTPUT (1) -#define FRAMES_PER_BUFFER (64) -#define SAMPLE_RATE (44100.0) -#define NUM_BUFFERS (0) -typedef struct PaQaData -{ - unsigned long framesLeft; - int numChannels; - int bytesPerSample; - int mode; -} -PaQaData; -/****************************************** Prototypes ***********/ -static void TestDevices( int mode ); -static void TestFormats( int mode, PaDeviceID deviceID, double sampleRate, - int numChannels ); -static int TestBadOpens( void ); -static int TestBadActions( void ); -static int QaCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ); -/****************************************** Globals ***********/ -static int gNumPassed = 0; -static int gNumFailed = 0; -/****************************************** Macros ***********/ -/* Print ERROR if it fails. Tally success or failure. */ -/* Odd do-while wrapper seems to be needed for some compilers. */ -#define EXPECT( msg, _exp) \ - do \ - { \ - if ((_exp)) {\ - gNumPassed++; \ - } \ - else { \ - printf("\nERROR %s\n - 0x%x - %s for %s\n", (msg), result, Pa_GetErrorText(result), #_exp ); \ - gNumFailed++; \ - goto error; \ - } \ - } while(0) -#define HOPEFOR( msg, _exp) \ - do \ - { \ - if ((_exp)) {\ - gNumPassed++; \ - } \ - else { \ - printf("\nERROR %s\n - 0x%x - %s for %s\n", (msg), result, Pa_GetErrorText(result), #_exp ); \ - gNumFailed++; \ - } \ - } while(0) -/*******************************************************************/ -/* This routine will be called by the PortAudio engine when audio is needed. -** It may be called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int QaCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - unsigned long i; - unsigned char *out = (unsigned char *) outputBuffer; - PaQaData *data = (PaQaData *) userData; - (void) inputBuffer; /* Prevent "unused variable" warnings. */ - (void) outTime; - - /* Zero out buffer so we don't hear terrible noise. */ - if( data->mode == MODE_OUTPUT ) - { - unsigned long numBytes = framesPerBuffer * data->numChannels * data->bytesPerSample; - for( i=0; i<numBytes; i++ ) - { - *out++ = 0; - } - } - /* Are we through yet? */ - if( data->framesLeft > framesPerBuffer ) - { - data->framesLeft -= framesPerBuffer; - return 0; - } - else - { - data->framesLeft = 0; - return 1; - } -} -/*******************************************************************/ -int main(void); -int main(void) -{ - PaError result; - EXPECT( "init", ((result=Pa_Initialize()) == 0) ); - TestBadActions(); - TestBadOpens(); -error: - Pa_Terminate(); - printf("QA Report: %d passed, %d failed.\n", gNumPassed, gNumFailed ); - return 0; -} -/*******************************************************************/ -static int TestBadOpens( void ) -{ - PortAudioStream *stream = NULL; - PaError result; - PaQaData myData; - /* Setup data for synthesis thread. */ - myData.framesLeft = (unsigned long) (SAMPLE_RATE * 100); /* 100 seconds */ - myData.numChannels = 1; - myData.mode = MODE_OUTPUT; - HOPEFOR( "No devices specified.",( - (result = Pa_OpenStream( - &stream, - paNoDevice, 0, paFloat32, NULL, - paNoDevice, 0, paFloat32, NULL, - SAMPLE_RATE, FRAMES_PER_BUFFER, NUM_BUFFERS, - paClipOff, - QaCallback, - &myData ) - ) == paInvalidDeviceId) ); - HOPEFOR( "Out of range input device specified.",( - (result = Pa_OpenStream( - &stream, - Pa_CountDevices(), 0, paFloat32, NULL, - paNoDevice, 0, paFloat32, NULL, - SAMPLE_RATE, FRAMES_PER_BUFFER, NUM_BUFFERS, - paClipOff, - QaCallback, - &myData ) - ) == paInvalidDeviceId) ); - - HOPEFOR( "Out of range output device specified.",( - (result = Pa_OpenStream( - &stream, - paNoDevice, 0, paFloat32, NULL, - Pa_CountDevices(), 0, paFloat32, NULL, - SAMPLE_RATE, FRAMES_PER_BUFFER, NUM_BUFFERS, - paClipOff, - QaCallback, - &myData ) - ) == paInvalidDeviceId) ); - HOPEFOR( "Zero input channels.",( - (result = Pa_OpenStream( - &stream, - Pa_GetDefaultInputDeviceID(), 0, paFloat32, NULL, - paNoDevice, 0, paFloat32, NULL, - SAMPLE_RATE, FRAMES_PER_BUFFER, NUM_BUFFERS, - paClipOff, - QaCallback, - &myData ) - ) == paInvalidChannelCount) ); - HOPEFOR( "Zero output channels.",( - (result = Pa_OpenStream( - &stream, - paNoDevice, 0, paFloat32, NULL, - Pa_GetDefaultOutputDeviceID(), 0, paFloat32, NULL, - SAMPLE_RATE, FRAMES_PER_BUFFER, NUM_BUFFERS, - paClipOff, - QaCallback, - &myData ) - ) == paInvalidChannelCount) ); - HOPEFOR( "Nonzero input channels but no device.",( - (result = Pa_OpenStream( - &stream, - Pa_GetDefaultInputDeviceID(), 2, paFloat32, NULL, - paNoDevice, 2, paFloat32, NULL, - SAMPLE_RATE, FRAMES_PER_BUFFER, NUM_BUFFERS, - paClipOff, - QaCallback, - &myData ) - ) == paInvalidChannelCount) ); - - HOPEFOR( "Nonzero output channels but no device.",( - (result = Pa_OpenStream( - &stream, - paNoDevice, 2, paFloat32, NULL, - Pa_GetDefaultOutputDeviceID(), 2, paFloat32, NULL, - SAMPLE_RATE, FRAMES_PER_BUFFER, NUM_BUFFERS, - paClipOff, - QaCallback, - &myData ) - ) == paInvalidChannelCount) ); - HOPEFOR( "NULL stream pointer.",( - (result = Pa_OpenStream( - NULL, - paNoDevice, 0, paFloat32, NULL, - Pa_GetDefaultOutputDeviceID(), 2, paFloat32, NULL, - SAMPLE_RATE, FRAMES_PER_BUFFER, NUM_BUFFERS, - paClipOff, - QaCallback, - &myData ) - ) == paBadStreamPtr) ); - HOPEFOR( "Low sample rate.",( - (result = Pa_OpenStream( - &stream, - paNoDevice, 0, paFloat32, NULL, - Pa_GetDefaultOutputDeviceID(), 2, paFloat32, NULL, - 1.0, FRAMES_PER_BUFFER, NUM_BUFFERS, - paClipOff, - QaCallback, - &myData ) - ) == paInvalidSampleRate) ); - HOPEFOR( "High sample rate.",( - (result = Pa_OpenStream( - &stream, - paNoDevice, 0, paFloat32, NULL, - Pa_GetDefaultOutputDeviceID(), 2, paFloat32, NULL, - 10000000.0, FRAMES_PER_BUFFER, NUM_BUFFERS, - paClipOff, - QaCallback, - &myData ) - ) == paInvalidSampleRate) ); - HOPEFOR( "NULL callback.",( - (result = Pa_OpenStream( - &stream, - paNoDevice, 0, paFloat32, NULL, - Pa_GetDefaultOutputDeviceID(), 2, paFloat32, NULL, - SAMPLE_RATE, FRAMES_PER_BUFFER, NUM_BUFFERS, - paClipOff, - NULL, - &myData ) - ) == paNullCallback) ); - HOPEFOR( "Bad flag.",( - (result = Pa_OpenStream( - &stream, - paNoDevice, 0, paFloat32, NULL, - Pa_GetDefaultOutputDeviceID(), 2, paFloat32, NULL, - SAMPLE_RATE, FRAMES_PER_BUFFER, NUM_BUFFERS, - (1<<3), - QaCallback, - &myData ) - ) == paInvalidFlag) ); - -#if 1 /* FIXME - this is legal for some implementations. */ - HOPEFOR( "Use input device as output device.",( - (result = Pa_OpenStream( - &stream, - paNoDevice, 0, paFloat32, NULL, - Pa_GetDefaultInputDeviceID(), 2, paFloat32, NULL, - SAMPLE_RATE, FRAMES_PER_BUFFER, NUM_BUFFERS, - paClipOff, - QaCallback, - &myData ) - ) == paInvalidDeviceId) ); - - HOPEFOR( "Use output device as input device.",( - (result = Pa_OpenStream( - &stream, - Pa_GetDefaultOutputDeviceID(), 2, paFloat32, NULL, - paNoDevice, 0, paFloat32, NULL, - SAMPLE_RATE, FRAMES_PER_BUFFER, NUM_BUFFERS, - paClipOff, - QaCallback, - &myData ) - ) == paInvalidDeviceId) ); -#endif - - if( stream != NULL ) Pa_CloseStream( stream ); - return result; -} -/*******************************************************************/ -static int TestBadActions( void ) -{ - PortAudioStream *stream = NULL; - PaError result; - PaQaData myData; - /* Setup data for synthesis thread. */ - myData.framesLeft = (unsigned long) (SAMPLE_RATE * 100); /* 100 seconds */ - myData.numChannels = 1; - myData.mode = MODE_OUTPUT; - /* Default output. */ - EXPECT( "TestBadActions", ((result = Pa_OpenStream( - &stream, - paNoDevice, 0, paFloat32, NULL, - Pa_GetDefaultOutputDeviceID(), 2, paFloat32, NULL, - SAMPLE_RATE, FRAMES_PER_BUFFER, NUM_BUFFERS, - paClipOff, - QaCallback, - &myData ) - ) == 0) ); - HOPEFOR( "start", ((result = Pa_StartStream( NULL )) == paBadStreamPtr) ); - HOPEFOR( "stop", ((result = Pa_StopStream( NULL )) == paBadStreamPtr) ); - HOPEFOR( "active?", ((result = Pa_StreamActive( NULL )) == paBadStreamPtr) ); - HOPEFOR( "close", ((result = Pa_CloseStream( NULL )) == paBadStreamPtr) ); - HOPEFOR( "time?", ((result = (PaError)Pa_StreamTime( NULL )) != 0) ); - HOPEFOR( "CPULoad?", ((result = (PaError)Pa_GetCPULoad( NULL )) != 0) ); -error: - if( stream != NULL ) Pa_CloseStream( stream ); - return result; -} diff --git a/pd/portaudio_v18/pa_tests/patest1.c b/pd/portaudio_v18/pa_tests/patest1.c deleted file mode 100644 index 8ac45ad1..00000000 --- a/pd/portaudio_v18/pa_tests/patest1.c +++ /dev/null @@ -1,114 +0,0 @@ -/* - $Id: patest1.c,v 1.1.1.1 2002/01/22 00:52:33 phil Exp $ - patest1.c - Ring modulate the audio input with a sine wave for 20 seconds - using the Portable Audio api - Author: Ross Bencina <rossb@audiomulch.com> - Modifications: - April 5th, 2001 - PLB - Check for NULL inputBuffer. -*/ -#include <stdio.h> -#include <math.h> -#include "portaudio.h" -#ifndef M_PI -#define M_PI (3.14159265) -#endif -typedef struct -{ - float sine[100]; - int phase; - int sampsToGo; -} -patest1data; -static int patest1Callback( void *inputBuffer, void *outputBuffer, - unsigned long bufferFrames, - PaTimestamp outTime, void *userData ) -{ - patest1data *data = (patest1data*)userData; - float *in = (float*)inputBuffer; - float *out = (float*)outputBuffer; - int framesToCalc = bufferFrames; - unsigned long i; - int finished = 0; - /* Check to see if any input data is available. */ - if(inputBuffer == NULL) return 0; - if( data->sampsToGo < bufferFrames ) - { - framesToCalc = data->sampsToGo; - finished = 1; - } - for( i=0; i<framesToCalc; i++ ) - { - *out++ = *in++ * data->sine[data->phase]; /* left */ - *out++ = *in++ * data->sine[data->phase++]; /* right */ - if( data->phase >= 100 ) - data->phase = 0; - } - data->sampsToGo -= framesToCalc; - /* zero remainder of final buffer if not already done */ - for( ; i<bufferFrames; i++ ) - { - *out++ = 0; /* left */ - *out++ = 0; /* right */ - } - return finished; -} -int main(int argc, char* argv[]); -int main(int argc, char* argv[]) -{ - PaStream *stream; - PaError err; - patest1data data; - int i; - int inputDevice = Pa_GetDefaultInputDeviceID(); - int outputDevice = Pa_GetDefaultOutputDeviceID(); - /* initialise sinusoidal wavetable */ - for( i=0; i<100; i++ ) - data.sine[i] = sin( ((double)i/100.) * M_PI * 2. ); - data.phase = 0; - data.sampsToGo = 44100 * 20; // 20 seconds - /* initialise portaudio subsytem */ - Pa_Initialize(); - err = Pa_OpenStream( - &stream, - inputDevice, - 2, /* stereo input */ - paFloat32, /* 32 bit floating point input */ - NULL, - outputDevice, - 2, /* stereo output */ - paFloat32, /* 32 bit floating point output */ - NULL, - 44100., - 512, /* small buffers */ - 0, /* let PA determine number of buffers */ - paClipOff, /* we won't output out of range samples so don't bother clipping them */ - patest1Callback, - &data ); - if( err == paNoError ) - { - err = Pa_StartStream( stream ); - printf( "Press any key to end.\n" ); - getc( stdin ); //wait for input before exiting - Pa_AbortStream( stream ); - - printf( "Waiting for stream to complete...\n" ); - - while( Pa_StreamActive( stream ) ) - Pa_Sleep(1000); /* sleep until playback has finished */ - - err = Pa_CloseStream( stream ); - } - else - { - fprintf( stderr, "An error occured while opening the portaudio stream\n" ); - if( err == paHostError ) - fprintf( stderr, "Host error number: %d\n", Pa_GetHostError() ); - else - fprintf( stderr, "Error number: %d\n", err ); - } - Pa_Terminate(); - printf( "bye\n" ); - - return 0; -} diff --git a/pd/portaudio_v18/pa_tests/patest_buffer.c b/pd/portaudio_v18/pa_tests/patest_buffer.c deleted file mode 100644 index 8d61f4f7..00000000 --- a/pd/portaudio_v18/pa_tests/patest_buffer.c +++ /dev/null @@ -1,180 +0,0 @@ -/* - * $Id: patest_buffer.c,v 1.1.1.1.4.1 2003/02/11 21:41:32 philburk Exp $ - * patest_buffer.c - * Test opening streams with different buffer sizes. - * - * Author: Phil Burk http://www.softsynth.com - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.portaudio.com - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <stdlib.h> -#include <math.h> -#include "portaudio.h" -#define NUM_SECONDS (1) -#define SAMPLE_RATE (44100) -#ifndef M_PI -#define M_PI (3.14159265) -#endif -#define TABLE_SIZE (200) - -#define BUFFER_TABLE 9 -long buffer_table[] = {200,256,500,512,600, 723, 1000, 1024, 2345}; - -typedef struct -{ - short sine[TABLE_SIZE]; - int left_phase; - int right_phase; - unsigned int sampsToGo; -} -paTestData; -PaError TestOnce( int buffersize ); - -static int paSineCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ); -/* This routine will be called by the PortAudio engine when audio is needed. -** It may called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int paSineCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData*)userData; - short *out = (short*)outputBuffer; - unsigned int i; - int finished = 0; - (void) inputBuffer; /* Prevent "unused variable" warnings. */ - (void) outTime; - - if( data->sampsToGo < framesPerBuffer ) - { - for( i=0; i<data->sampsToGo; i++ ) - { - *out++ = data->sine[data->left_phase]; /* left */ - *out++ = data->sine[data->right_phase]; /* right */ - data->left_phase += 1; - if( data->left_phase >= TABLE_SIZE ) data->left_phase -= TABLE_SIZE; - data->right_phase += 3; /* higher pitch so we can distinguish left and right. */ - if( data->right_phase >= TABLE_SIZE ) data->right_phase -= TABLE_SIZE; - } - /* zero remainder of final buffer */ - for( ; i<framesPerBuffer; i++ ) - { - *out++ = 0; /* left */ - *out++ = 0; /* right */ - } - - finished = 1; - } - else - { - for( i=0; i<framesPerBuffer; i++ ) - { - *out++ = data->sine[data->left_phase]; /* left */ - *out++ = data->sine[data->right_phase]; /* right */ - data->left_phase += 1; - if( data->left_phase >= TABLE_SIZE ) data->left_phase -= TABLE_SIZE; - data->right_phase += 3; /* higher pitch so we can distinguish left and right. */ - if( data->right_phase >= TABLE_SIZE ) data->right_phase -= TABLE_SIZE; - } - data->sampsToGo -= framesPerBuffer; - } - return finished; -} -/*******************************************************************/ -int main(void); -int main(void) -{ - int i; - PaError err; - printf("Test opening streams with different buffer sizes\n\n"); - - for (i = 0 ; i < BUFFER_TABLE; i++) - { - printf("Buffer size %d\n", buffer_table[i]); - err = TestOnce(buffer_table[i]); - if( err < 0 ) return 0; - - } -} - - -PaError TestOnce( int buffersize ) -{ - PortAudioStream *stream; - PaError err; - paTestData data; - int i; - int totalSamps; - /* initialise sinusoidal wavetable */ - for( i=0; i<TABLE_SIZE; i++ ) - { - data.sine[i] = (short) (30000.0 * sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. )); - } - data.left_phase = data.right_phase = 0; - data.sampsToGo = totalSamps = NUM_SECONDS * SAMPLE_RATE; /* Play for a few seconds. */ - err = Pa_Initialize(); - if( err != paNoError ) goto error; - err = Pa_OpenStream( - &stream, - paNoDevice,/* default input device */ - 0, /* no input */ - paInt16, /* sample format */ - NULL, - Pa_GetDefaultOutputDeviceID(), /* default output device */ - 2, /* stereo output */ - paInt16, /* sample format */ - NULL, - SAMPLE_RATE, - buffersize, /* frames per buffer */ - 0, /* number of buffers, if zero then use default minimum */ - paClipOff, /* we won't output out of range samples so don't bother clipping them */ - paSineCallback, - &data ); - if( err != paNoError ) goto error; - - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - printf("Waiting for sound to finish.\n"); - fflush(stdout); - Pa_Sleep(1000); - err = Pa_CloseStream( stream ); - if( err != paNoError ) goto error; - Pa_Terminate(); - return paNoError; -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return err; -} diff --git a/pd/portaudio_v18/pa_tests/patest_clip.c b/pd/portaudio_v18/pa_tests/patest_clip.c deleted file mode 100644 index 3fb30d40..00000000 --- a/pd/portaudio_v18/pa_tests/patest_clip.c +++ /dev/null @@ -1,156 +0,0 @@ -/* - * $Id: patest_clip.c,v 1.1.1.1 2002/01/22 00:52:34 phil Exp $ - * patest_clip.c - * Play a sine wave using the Portable Audio api for several seconds - * at an amplitude that would require clipping. - * - * Author: Phil Burk http://www.softsynth.com - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.portaudio.com - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <math.h> -#include "portaudio.h" -#define NUM_SECONDS (4) -#define SAMPLE_RATE (44100) -#ifndef M_PI -#define M_PI (3.14159265) -#endif -#define TABLE_SIZE (200) -typedef struct paTestData -{ - float sine[TABLE_SIZE]; - float amplitude; - int left_phase; - int right_phase; -} -paTestData; -PaError PlaySine( paTestData *data, unsigned long flags, float amplitude ); -/* This routine will be called by the PortAudio engine when audio is needed. -** It may called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int sineCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData*)userData; - float *out = (float*)outputBuffer; - float amplitude = data->amplitude; - unsigned int i; - (void) inputBuffer; /* Prevent "unused variable" warnings. */ - (void) outTime; - - for( i=0; i<framesPerBuffer; i++ ) - { - *out++ = amplitude * data->sine[data->left_phase]; /* left */ - *out++ = amplitude * data->sine[data->right_phase]; /* right */ - data->left_phase += 1; - if( data->left_phase >= TABLE_SIZE ) data->left_phase -= TABLE_SIZE; - data->right_phase += 3; /* higher pitch so we can distinguish left and right. */ - if( data->right_phase >= TABLE_SIZE ) data->right_phase -= TABLE_SIZE; - } - return 0; -} -/*******************************************************************/ -int main(void); -int main(void) -{ - PaError err; - paTestData DATA; - int i; - printf("PortAudio Test: output sine wave with and without clipping.\n"); - /* initialise sinusoidal wavetable */ - for( i=0; i<TABLE_SIZE; i++ ) - { - DATA.sine[i] = (float) sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. ); - } - printf("\nHalf amplitude. Should sound like sine wave.\n"); fflush(stdout); - err = PlaySine( &DATA, paClipOff | paDitherOff, 0.5f ); - if( err < 0 ) goto error; - printf("\nFull amplitude. Should sound like sine wave.\n"); fflush(stdout); - err = PlaySine( &DATA, paClipOff | paDitherOff, 0.999f ); - if( err < 0 ) goto error; - printf("\nOver range with clipping and dithering turned OFF. Should sound very nasty.\n"); - fflush(stdout); - err = PlaySine( &DATA, paClipOff | paDitherOff, 1.1f ); - if( err < 0 ) goto error; - printf("\nOver range with clipping and dithering turned ON. Should sound smoother than previous.\n"); - fflush(stdout); - err = PlaySine( &DATA, paNoFlag, 1.1f ); - if( err < 0 ) goto error; - printf("\nOver range with paClipOff but dithering ON.\n" - "That forces clipping ON so it should sound the same as previous.\n"); - fflush(stdout); - err = PlaySine( &DATA, paClipOff, 1.1f ); - if( err < 0 ) goto error; - return 0; -error: - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return 1; -} -/*****************************************************************************/ -PaError PlaySine( paTestData *data, unsigned long flags, float amplitude ) -{ - PortAudioStream *stream; - PaError err; - data->left_phase = data->right_phase = 0; - data->amplitude = amplitude; - err = Pa_Initialize(); - if( err != paNoError ) goto error; - err = Pa_OpenStream( - &stream, - paNoDevice,/* default input device */ - 0, /* no input */ - paFloat32, /* 32 bit floating point input */ - NULL, - Pa_GetDefaultOutputDeviceID(), /* default output device */ - 2, /* stereo output */ - paFloat32, /* 32 bit floating point output */ - NULL, - SAMPLE_RATE, - 1024, - 0, /* number of buffers, if zero then use default minimum */ - flags, /* we won't output out of range samples so don't bother clipping them */ - sineCallback, - data ); - if( err != paNoError ) goto error; - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - Pa_Sleep( NUM_SECONDS * 1000 ); - printf("CPULoad = %8.6f\n", Pa_GetCPULoad( stream ) ); - err = Pa_CloseStream( stream ); - if( err != paNoError ) goto error; - Pa_Terminate(); - return paNoError; -error: - return err; -} diff --git a/pd/portaudio_v18/pa_tests/patest_dither.c b/pd/portaudio_v18/pa_tests/patest_dither.c deleted file mode 100644 index c58b9cea..00000000 --- a/pd/portaudio_v18/pa_tests/patest_dither.c +++ /dev/null @@ -1,152 +0,0 @@ -/* - * $Id: patest_dither.c,v 1.2 2002/03/21 00:58:45 philburk Exp $ - * patest_dither.c - * Attempt to hear difference between dithered and non-dithered signal. - * This only has an effect if the native format is 16 bit. - * - * Author: Phil Burk http://www.softsynth.com - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.portaudio.com - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <math.h> -#include "portaudio.h" -#define NUM_SECONDS (4) -#define SAMPLE_RATE (44100) -#ifndef M_PI -#define M_PI (3.14159265) -#endif -#define TABLE_SIZE (200) -typedef struct paTestData -{ - float sine[TABLE_SIZE]; - float amplitude; - int left_phase; - int right_phase; -} -paTestData; -PaError PlaySine( paTestData *data, PaStreamFlags flags, float amplitude ); -static int sineCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ); -/* This routine will be called by the PortAudio engine when audio is needed. -** It may called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int sineCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData*)userData; - float *out = (float*)outputBuffer; - float amplitude = data->amplitude; - unsigned int i; - (void) outTime; - (void) inputBuffer; - for( i=0; i<framesPerBuffer; i++ ) - { - *out++ = amplitude * data->sine[data->left_phase]; /* left */ - *out++ = amplitude * data->sine[data->right_phase]; /* right */ - data->left_phase += 1; - if( data->left_phase >= TABLE_SIZE ) data->left_phase -= TABLE_SIZE; - data->right_phase += 3; /* higher pitch so we can distinguish left and right. */ - if( data->right_phase >= TABLE_SIZE ) data->right_phase -= TABLE_SIZE; - } - return 0; -} -/*******************************************************************/ -int main(void); -int main(void) -{ - PaError err; - paTestData DATA; - int i; - float amplitude = 32.0 / (1<<15); - printf("PortAudio Test: output EXTREMELY QUIET sine wave with and without dithering.\n"); - /* initialise sinusoidal wavetable */ - for( i=0; i<TABLE_SIZE; i++ ) - { - DATA.sine[i] = (float) sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. ); - } - printf("\nNo treatment..\n"); fflush(stdout); - err = PlaySine( &DATA, paClipOff | paDitherOff, amplitude ); - if( err < 0 ) goto error; - printf("\nClip.\n"); - fflush(stdout); - err = PlaySine( &DATA, paDitherOff, amplitude ); - if( err < 0 ) goto error; - printf("\nClip and Dither.\n"); - fflush(stdout); - err = PlaySine( &DATA, paNoFlag, amplitude ); - if( err < 0 ) goto error; - return 0; -error: - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return -1; -} -/*****************************************************************************/ -PaError PlaySine( paTestData *data, PaStreamFlags flags, float amplitude ) -{ - PortAudioStream *stream; - PaError err; - data->left_phase = data->right_phase = 0; - data->amplitude = amplitude; - err = Pa_Initialize(); - if( err != paNoError ) goto error; - err = Pa_OpenStream( - &stream, - paNoDevice,/* default input device */ - 0, /* no input */ - paFloat32, /* 32 bit floating point input */ - NULL, - Pa_GetDefaultOutputDeviceID(), /* default output device */ - 2, /* stereo output */ - paFloat32, /* 32 bit floating point output */ - NULL, - SAMPLE_RATE, - 1024, - 0, /* number of buffers, if zero then use default minimum */ - flags, /* we won't output out of range samples so don't bother clipping them */ - sineCallback, - (void *)data ); - if( err != paNoError ) goto error; - - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - Pa_Sleep( NUM_SECONDS * 1000 ); - printf("CPULoad = %8.6f\n", Pa_GetCPULoad( stream ) ); - err = Pa_CloseStream( stream ); - if( err != paNoError ) goto error; - Pa_Terminate(); - return paNoError; -error: - return err; -} diff --git a/pd/portaudio_v18/pa_tests/patest_hang.c b/pd/portaudio_v18/pa_tests/patest_hang.c deleted file mode 100644 index d1d67199..00000000 --- a/pd/portaudio_v18/pa_tests/patest_hang.c +++ /dev/null @@ -1,153 +0,0 @@ -/* - * $Id: patest_hang.c,v 1.1.4.1 2003/02/11 21:41:32 philburk Exp $ - * Play a sine then hang audio callback to test watchdog. - * - * Authors: - * Ross Bencina <rossb@audiomulch.com> - * Phil Burk <philburk@softsynth.com> - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.audiomulch.com/portaudio/ - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <math.h> -#include "portaudio.h" - - -#define SAMPLE_RATE (44100) -#define FRAMES_PER_BUFFER (1024) -#ifndef M_PI -#define M_PI (3.14159265) -#endif -#define TWOPI (M_PI * 2.0) - -typedef struct paTestData -{ - int sleepFor; - double phase; -} -paTestData; - -/* This routine will be called by the PortAudio engine when audio is needed. -** It may called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int patestCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData*)userData; - float *out = (float*)outputBuffer; - unsigned long i; - int finished = 0; - double phaseInc = 0.02; - double phase = data->phase; - - (void) outTime; /* Prevent unused variable warnings. */ - (void) inputBuffer; - - for( i=0; i<framesPerBuffer; i++ ) - { - phase += phaseInc; - if( phase > TWOPI ) phase -= TWOPI; - /* This is not a very efficient way to calc sines. */ - *out++ = (float) sin( phase ); /* mono */ - } - - if( data->sleepFor > 0 ) - { - Pa_Sleep( data->sleepFor ); - } - - data->phase = phase; - return finished; -} - -/*******************************************************************/ -int main(void); -int main(void) -{ - PortAudioStream *stream; - PaError err; - int i; - paTestData data = {0}; - - printf("PortAudio Test: output sine wave. SR = %d, BufSize = %d\n", - SAMPLE_RATE, FRAMES_PER_BUFFER ); - - err = Pa_Initialize(); - if( err != paNoError ) goto error; - - err = Pa_OpenStream( - &stream, - paNoDevice,/* default input device */ - 0, /* no input */ - paFloat32, /* 32 bit floating point input */ - NULL, - Pa_GetDefaultOutputDeviceID(), /* default output device */ - 1, /* mono output */ - paFloat32, /* 32 bit floating point output */ - NULL, - SAMPLE_RATE, - FRAMES_PER_BUFFER, /* frames per buffer */ - 0, /* number of buffers, if zero then use default minimum */ - paClipOff, /* we won't output out of range samples so don't bother clipping them */ - patestCallback, - &data ); - if( err != paNoError ) goto error; - - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - -/* Gradually increase sleep time. */ - for( i=0; i<10000; i+= 1000 ) - { - printf("Sleep for %d milliseconds in audio callback.\n", i ); - data.sleepFor = i; - fflush(stdout); - Pa_Sleep( ((i<1000) ? 1000 : i) ); - } - - printf("Suffer for 10 seconds.\n"); - fflush(stdout); - Pa_Sleep( 10000 ); - - err = Pa_StopStream( stream ); - if( err != paNoError ) goto error; - err = Pa_CloseStream( stream ); - if( err != paNoError ) goto error; - Pa_Terminate(); - printf("Test finished.\n"); - return err; -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return err; -} diff --git a/pd/portaudio_v18/pa_tests/patest_latency.c b/pd/portaudio_v18/pa_tests/patest_latency.c deleted file mode 100644 index 39ede0a7..00000000 --- a/pd/portaudio_v18/pa_tests/patest_latency.c +++ /dev/null @@ -1,176 +0,0 @@ -/* - * $Id: patest_latency.c,v 1.4 2002/03/21 00:58:45 philburk Exp $ - * Hear the latency caused by big buffers. - * Play a sine wave and change frequency based on letter input. - * - * Author: Phil Burk <philburk@softsynth.com>, and Darren Gibbs - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.portaudio.com - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <math.h> -#include "portaudio.h" - -#define OUTPUT_DEVICE (Pa_GetDefaultOutputDeviceID()) -#define SAMPLE_RATE (44100) -#define FRAMES_PER_BUFFER (64) - -#if 0 -#define MIN_LATENCY_MSEC (2000) -#define NUM_BUFFERS ((MIN_LATENCY_MSEC * SAMPLE_RATE) / (FRAMES_PER_BUFFER * 1000)) -#else -#define NUM_BUFFERS (0) -#endif - -#define MIN_FREQ (100.0f) -#define CalcPhaseIncrement(freq) ((freq)/SAMPLE_RATE) -#ifndef M_PI -#define M_PI (3.14159265) -#endif -#define TABLE_SIZE (400) -typedef struct -{ - float sine[TABLE_SIZE + 1]; // add one for guard point for interpolation - float phase_increment; - float left_phase; - float right_phase; -} -paTestData; -float LookupSine( paTestData *data, float phase ); -/* Convert phase between and 1.0 to sine value - * using linear interpolation. - */ -float LookupSine( paTestData *data, float phase ) -{ - float fIndex = phase*TABLE_SIZE; - int index = (int) fIndex; - float fract = fIndex - index; - float lo = data->sine[index]; - float hi = data->sine[index+1]; - float val = lo + fract*(hi-lo); - return val; -} -/* This routine will be called by the PortAudio engine when audio is needed. -** It may called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int patestCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData*)userData; - float *out = (float*)outputBuffer; - int i; - int finished = 0; - (void) outTime; /* Prevent unused variable warnings. */ - (void) inputBuffer; - - for( i=0; i<framesPerBuffer; i++ ) - { - *out++ = LookupSine(data, data->left_phase); /* left */ - *out++ = LookupSine(data, data->right_phase); /* right */ - data->left_phase += data->phase_increment; - if( data->left_phase >= 1.0f ) data->left_phase -= 1.0f; - data->right_phase += (data->phase_increment * 1.5f); /* fifth above */ - if( data->right_phase >= 1.0f ) data->right_phase -= 1.0f; - } - return 0; -} -/*******************************************************************/ -int main(void); -int main(void) -{ - PortAudioStream *stream; - PaError err; - paTestData data; - int i; - int done = 0; - printf("PortAudio Test: enter letter then hit ENTER. numBuffers = %d\n", NUM_BUFFERS ); - /* initialise sinusoidal wavetable */ - for( i=0; i<TABLE_SIZE; i++ ) - { - data.sine[i] = 0.90f * (float) sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. ); - } - data.sine[TABLE_SIZE] = data.sine[0]; // set guard point - data.left_phase = data.right_phase = 0.0; - data.phase_increment = CalcPhaseIncrement(MIN_FREQ); - - err = Pa_Initialize(); - if( err != paNoError ) goto error; - printf("PortAudio Test: output device = %d\n", OUTPUT_DEVICE ); - err = Pa_OpenStream( - &stream, - paNoDevice, - 0, /* no input */ - paFloat32, /* 32 bit floating point input */ - NULL, - OUTPUT_DEVICE, - 2, /* stereo output */ - paFloat32, /* 32 bit floating point output */ - NULL, - SAMPLE_RATE, - FRAMES_PER_BUFFER, - NUM_BUFFERS, /* number of buffers, if zero then use default minimum */ - paClipOff|paDitherOff, /* we won't output out of range samples so don't bother clipping them */ - patestCallback, - &data ); - if( err != paNoError ) goto error; - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - printf("Play ASCII keyboard. Hit 'q' to stop. (Use RETURN key on Mac)\n"); - fflush(stdout); - while ( !done ) - { - float freq; - int index; - char c; - do - { - c = getchar(); - } - while( c < ' '); /* Strip white space and control chars. */ - - if( c == 'q' ) done = 1; - index = c % 26; - freq = MIN_FREQ + (index * 40.0); - data.phase_increment = CalcPhaseIncrement(freq); - } - printf("Call Pa_StopStream()\n"); - err = Pa_StopStream( stream ); - if( err != paNoError ) goto error; - Pa_Terminate(); - printf("Test finished.\n"); - return err; -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return err; -} diff --git a/pd/portaudio_v18/pa_tests/patest_leftright.c b/pd/portaudio_v18/pa_tests/patest_leftright.c deleted file mode 100644 index 6e6172ac..00000000 --- a/pd/portaudio_v18/pa_tests/patest_leftright.c +++ /dev/null @@ -1,168 +0,0 @@ -/* - * $Id: patest_leftright.c,v 1.2 2002/02/22 21:46:18 philburk Exp $ - * patest_leftright.c - * Play different tone sine waves that alternate between left and right channel. - * The low tone should be on the left channel. - * - * Authors: - * Ross Bencina <rossb@audiomulch.com> - * Phil Burk <philburk@softsynth.com> - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.audiomulch.com/portaudio/ - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <math.h> -#include "portaudio.h" -#define NUM_SECONDS (8) -#define SAMPLE_RATE (44100) -#define FRAMES_PER_BUFFER (512) -#ifndef M_PI -#define M_PI (3.14159265) -#endif -#define TABLE_SIZE (200) -typedef struct -{ - float sine[TABLE_SIZE]; - int left_phase; - int right_phase; - int toggle; - int countDown; -} -paTestData; -/* This routine will be called by the PortAudio engine when audio is needed. -** It may called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int patestCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData*)userData; - float *out = (float*)outputBuffer; - unsigned long i; - int finished = 0; - (void) outTime; /* Prevent unused variable warnings. */ - (void) inputBuffer; - - for( i=0; i<framesPerBuffer; i++ ) - { - if( data->toggle ) - { - *out++ = data->sine[data->left_phase]; /* left */ - *out++ = 0; /* right */ - } - else - { - *out++ = 0; /* left */ - *out++ = data->sine[data->right_phase]; /* right */ - } - - data->left_phase += 1; - if( data->left_phase >= TABLE_SIZE ) data->left_phase -= TABLE_SIZE; - data->right_phase += 3; /* higher pitch so we can distinguish left and right. */ - if( data->right_phase >= TABLE_SIZE ) data->right_phase -= TABLE_SIZE; - } - - if( data->countDown < 0 ) - { - data->countDown = SAMPLE_RATE; - data->toggle = !data->toggle; - } - data->countDown -= framesPerBuffer; - - return finished; -} - -/*******************************************************************/ -int main(void); -int main(void) -{ - PortAudioStream *stream; - PaError err; - paTestData data; - int i; - int timeout; - - printf("Play different tone sine waves that alternate between left and right channel.\n"); - printf("The low tone should be on the left channel.\n"); - - /* initialise sinusoidal wavetable */ - for( i=0; i<TABLE_SIZE; i++ ) - { - data.sine[i] = (float) sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. ); - } - data.left_phase = data.right_phase = data.toggle = 0; - data.countDown = SAMPLE_RATE; - - err = Pa_Initialize(); - if( err != paNoError ) goto error; - - err = Pa_OpenStream( - &stream, - paNoDevice,/* default input device */ - 0, /* no input */ - paFloat32, /* 32 bit floating point input */ - NULL, - Pa_GetDefaultOutputDeviceID(), /* default output device */ - 2, /* stereo output */ - paFloat32, /* 32 bit floating point output */ - NULL, - SAMPLE_RATE, - FRAMES_PER_BUFFER, /* frames per buffer */ - 0, /* number of buffers, if zero then use default minimum */ - paClipOff, /* we won't output out of range samples so don't bother clipping them */ - patestCallback, - &data ); - if( err != paNoError ) goto error; - - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - - printf("Play for several seconds.\n"); - timeout = NUM_SECONDS * 4; - while( timeout > 0 ) - { - Pa_Sleep( 300 ); - timeout -= 1; - } - - err = Pa_StopStream( stream ); - if( err != paNoError ) goto error; - err = Pa_CloseStream( stream ); - if( err != paNoError ) goto error; - Pa_Terminate(); - printf("Test finished.\n"); - return err; -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return err; -} diff --git a/pd/portaudio_v18/pa_tests/patest_longsine.c b/pd/portaudio_v18/pa_tests/patest_longsine.c deleted file mode 100644 index 7dc8ae56..00000000 --- a/pd/portaudio_v18/pa_tests/patest_longsine.c +++ /dev/null @@ -1,137 +0,0 @@ -/* - * $Id: patest_longsine.c,v 1.2 2002/04/30 21:21:30 philburk Exp $ - * patest_longsine.c - * Play a sine wave using the Portable Audio api until ENTER hit. - * - * Author: Phil Burk http://www.softsynth.com - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.portaudio.com - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <math.h> -#include "portaudio.h" - -#define SAMPLE_RATE (44100) - -#ifndef M_PI -#define M_PI (3.14159265) -#endif - -#define TABLE_SIZE (200) -typedef struct -{ - float sine[TABLE_SIZE]; - int left_phase; - int right_phase; -} -paTestData; - -/* This routine will be called by the PortAudio engine when audio is needed. -** It may called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int patestCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData*)userData; - float *out = (float*)outputBuffer; - unsigned int i; - (void) outTime; /* Prevent unused variable warnings. */ - (void) inputBuffer; - for( i=0; i<framesPerBuffer; i++ ) - { - *out++ = data->sine[data->left_phase]; /* left */ - *out++ = data->sine[data->right_phase]; /* right */ - data->left_phase += 1; - if( data->left_phase >= TABLE_SIZE ) data->left_phase -= TABLE_SIZE; - data->right_phase += 3; /* higher pitch so we can distinguish left and right. */ - if( data->right_phase >= TABLE_SIZE ) data->right_phase -= TABLE_SIZE; - } - return 0; -} - -/*******************************************************************/ -int main(void); -int main(void) -{ - PortAudioStream *stream; - PaError err; - paTestData data; - int i; - printf("PortAudio Test: output sine wave.\n"); - - /* initialise sinusoidal wavetable */ - for( i=0; i<TABLE_SIZE; i++ ) - { - data.sine[i] = (float) sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. ); - } - data.left_phase = data.right_phase = 0; - - err = Pa_Initialize(); - if( err != paNoError ) goto error; - - err = Pa_OpenStream( - &stream, - paNoDevice,/* default input device */ - 0, /* no input */ - paFloat32, /* 32 bit floating point input */ - NULL, - Pa_GetDefaultOutputDeviceID(), /* default output device */ - 2, /* stereo output */ - paFloat32, /* 32 bit floating point output */ - NULL, - SAMPLE_RATE, - 256, /* frames per buffer */ - 0, /* number of buffers, if zero then use default minimum */ - paClipOff, /* we won't output out of range samples so don't bother clipping them */ - patestCallback, - &data ); - if( err != paNoError ) goto error; - - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - - printf("Hit ENTER to stop program.\n"); - getchar(); - - err = Pa_CloseStream( stream ); - if( err != paNoError ) goto error; - Pa_Terminate(); - - printf("Test finished.\n"); - return err; - -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return err; -} diff --git a/pd/portaudio_v18/pa_tests/patest_many.c b/pd/portaudio_v18/pa_tests/patest_many.c deleted file mode 100644 index 99f12ec8..00000000 --- a/pd/portaudio_v18/pa_tests/patest_many.c +++ /dev/null @@ -1,195 +0,0 @@ -/* - * $Id: patest_many.c,v 1.1.1.1.4.1 2003/02/11 21:41:32 philburk Exp $ - * patest_many.c - * Start and stop the PortAudio Driver multiple times. - * - * Author: Phil Burk http://www.softsynth.com - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.portaudio.com - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <stdlib.h> -#include <math.h> -#include "portaudio.h" -#define NUM_SECONDS (1) -#define SAMPLE_RATE (44100) -#ifndef M_PI -#define M_PI (3.14159265) -#endif -#define TABLE_SIZE (200) -typedef struct -{ - short sine[TABLE_SIZE]; - int left_phase; - int right_phase; - unsigned int sampsToGo; -} -paTestData; -PaError TestOnce( void ); -static int patest1Callback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ); -/* This routine will be called by the PortAudio engine when audio is needed. -** It may called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int patest1Callback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData*)userData; - short *out = (short*)outputBuffer; - unsigned int i; - int finished = 0; - (void) inputBuffer; /* Prevent "unused variable" warnings. */ - (void) outTime; - - if( data->sampsToGo < framesPerBuffer ) - { - /* final buffer... */ - - for( i=0; i<data->sampsToGo; i++ ) - { - *out++ = data->sine[data->left_phase]; /* left */ - *out++ = data->sine[data->right_phase]; /* right */ - data->left_phase += 1; - if( data->left_phase >= TABLE_SIZE ) data->left_phase -= TABLE_SIZE; - data->right_phase += 3; /* higher pitch so we can distinguish left and right. */ - if( data->right_phase >= TABLE_SIZE ) data->right_phase -= TABLE_SIZE; - } - /* zero remainder of final buffer */ - for( ; i<framesPerBuffer; i++ ) - { - *out++ = 0; /* left */ - *out++ = 0; /* right */ - } - - finished = 1; - } - else - { - for( i=0; i<framesPerBuffer; i++ ) - { - *out++ = data->sine[data->left_phase]; /* left */ - *out++ = data->sine[data->right_phase]; /* right */ - data->left_phase += 1; - if( data->left_phase >= TABLE_SIZE ) data->left_phase -= TABLE_SIZE; - data->right_phase += 3; /* higher pitch so we can distinguish left and right. */ - if( data->right_phase >= TABLE_SIZE ) data->right_phase -= TABLE_SIZE; - } - data->sampsToGo -= framesPerBuffer; - } - return finished; -} -/*******************************************************************/ -#ifdef MACINTOSH -int main(void); -int main(void) -{ - int i; - PaError err; - int numLoops = 10; - printf("Loop %d times.\n", numLoops ); - for( i=0; i<numLoops; i++ ) - { - printf("Loop %d out of %d.\n", i+1, numLoops ); - err = TestOnce(); - if( err < 0 ) return 0; - } -} -#else -int main(int argc, char **argv); -int main(int argc, char **argv) -{ - PaError err; - int i, numLoops = 10; - if( argc > 1 ) - { - numLoops = atoi(argv[1]); - } - for( i=0; i<numLoops; i++ ) - { - printf("Loop %d out of %d.\n", i+1, numLoops ); - err = TestOnce(); - if( err < 0 ) return 1; - } - printf("Test complete.\n"); - return 0; -} -#endif -PaError TestOnce( void ) -{ - PortAudioStream *stream; - PaError err; - paTestData data; - int i; - int totalSamps; - /* initialise sinusoidal wavetable */ - for( i=0; i<TABLE_SIZE; i++ ) - { - data.sine[i] = (short) (32767.0 * sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. )); - } - data.left_phase = data.right_phase = 0; - data.sampsToGo = totalSamps = NUM_SECONDS * SAMPLE_RATE; /* Play for a few seconds. */ - err = Pa_Initialize(); - if( err != paNoError ) goto error; - err = Pa_OpenStream( - &stream, - paNoDevice,/* default input device */ - 0, /* no input */ - paInt16, /* sample format */ - NULL, - Pa_GetDefaultOutputDeviceID(), /* default output device */ - 2, /* stereo output */ - paInt16, /* sample format */ - NULL, - SAMPLE_RATE, - 1024, /* frames per buffer */ - 8, /* number of buffers, if zero then use default minimum */ - paClipOff, /* we won't output out of range samples so don't bother clipping them */ - patest1Callback, - &data ); - if( err != paNoError ) goto error; - - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - printf("Waiting for sound to finish.\n"); - fflush(stdout); - Pa_Sleep(1000); - err = Pa_CloseStream( stream ); - if( err != paNoError ) goto error; - Pa_Terminate(); - return paNoError; -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return err; -} diff --git a/pd/portaudio_v18/pa_tests/patest_maxsines.c b/pd/portaudio_v18/pa_tests/patest_maxsines.c deleted file mode 100644 index e352715f..00000000 --- a/pd/portaudio_v18/pa_tests/patest_maxsines.c +++ /dev/null @@ -1,201 +0,0 @@ -/* - * $Id: patest_maxsines.c,v 1.4.4.1 2003/04/10 23:09:40 philburk Exp $ - * patest_maxsines.c - * How many sine waves can we calculate and play in less than 80% CPU Load. - * - * Authors: - * Ross Bencina <rossb@audiomulch.com> - * Phil Burk <philburk@softsynth.com> - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.audiomulch.com/portaudio/ - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <math.h> -#include "portaudio.h" - -#define MAX_SINES (500) -#define MAX_USAGE (0.8) -#define SAMPLE_RATE (44100) -#define FREQ_TO_PHASE_INC(freq) (freq/(float)SAMPLE_RATE) - -#define MIN_PHASE_INC FREQ_TO_PHASE_INC(200.0f) -#define MAX_PHASE_INC (MIN_PHASE_INC * (1 << 5)) - -#define FRAMES_PER_BUFFER (512) -#ifndef M_PI -#define M_PI (3.14159265) -#endif -#define TWOPI (M_PI * 2.0) - -#define TABLE_SIZE (512) - -typedef struct paTestData -{ - int numSines; - float sine[TABLE_SIZE + 1]; /* add one for guard point for interpolation */ - float phases[MAX_SINES]; -} -paTestData; - -/* Convert phase between and 1.0 to sine value - * using linear interpolation. - */ -float LookupSine( paTestData *data, float phase ); -float LookupSine( paTestData *data, float phase ) -{ - float fIndex = phase*TABLE_SIZE; - int index = (int) fIndex; - float fract = fIndex - index; - float lo = data->sine[index]; - float hi = data->sine[index+1]; - float val = lo + fract*(hi-lo); - return val; -} - -/* This routine will be called by the PortAudio engine when audio is needed. -** It may called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int patestCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData*)userData; - float *out = (float*)outputBuffer; - float outSample; - float scaler; - int numForScale; - unsigned long i; - int j; - int finished = 0; - (void) outTime; /* Prevent unused variable warnings. */ - (void) inputBuffer; - -/* Detemine amplitude scaling factor */ - numForScale = data->numSines; - if( numForScale < 8 ) numForScale = 8; /* prevent pops at beginning */ - scaler = 1.0f / numForScale; - - for( i=0; i<framesPerBuffer; i++ ) - { - float output = 0.0; - float phaseInc = MIN_PHASE_INC; - float phase; - for( j=0; j<data->numSines; j++ ) - { - /* Advance phase of next oscillator. */ - phase = data->phases[j]; - phase += phaseInc; - if( phase >= 1.0 ) phase -= 1.0; - - output += LookupSine(data, phase); - data->phases[j] = phase; - - phaseInc *= 1.02f; - if( phaseInc > MAX_PHASE_INC ) phaseInc = MIN_PHASE_INC; - } - - outSample = (float) (output * scaler); - *out++ = outSample; /* Left */ - *out++ = outSample; /* Right */ - } - return finished; -} - -/*******************************************************************/ -int main(void); -int main(void) -{ - int i; - PortAudioStream *stream; - PaError err; - paTestData data = {0}; - double load; - printf("PortAudio Test: output sine wave. SR = %d, BufSize = %d\n", SAMPLE_RATE, FRAMES_PER_BUFFER); - - /* initialise sinusoidal wavetable */ - for( i=0; i<TABLE_SIZE; i++ ) - { - data.sine[i] = (float) sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. ); - } - data.sine[TABLE_SIZE] = data.sine[0]; /* set guard point */ - - err = Pa_Initialize(); - if( err != paNoError ) goto error; - err = Pa_OpenStream( - &stream, - paNoDevice, - 0, /* no input */ - paFloat32, - NULL, - Pa_GetDefaultOutputDeviceID(), /* default output device */ - 2, /* stereo output */ - paFloat32, /* 32 bit floating point output */ - NULL, - SAMPLE_RATE, - FRAMES_PER_BUFFER, - 0, /* number of buffers, if zero then use default minimum */ - paClipOff, /* we won't output out of range samples so don't bother clipping them */ - patestCallback, - &data ); - if( err != paNoError ) goto error; - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - -/* Play an increasing number of sine waves until we hit MAX_USAGE */ - do - { - data.numSines++; - Pa_Sleep( 200 ); - - load = Pa_GetCPULoad( stream ); - printf("numSines = %d, CPU load = %f\n", data.numSines, load ); - fflush( stdout ); - } - while( (load < MAX_USAGE) && (data.numSines < MAX_SINES) ); - - printf("Press ENTER to stop.\n" ); fflush(stdout); - getchar(); - - printf("CPU load = %f\n", Pa_GetCPULoad( stream ) ); - err = Pa_StopStream( stream ); - if( err != paNoError ) goto error; - err = Pa_CloseStream( stream ); - if( err != paNoError ) goto error; - Pa_Terminate(); - printf("Test finished.\n"); - fflush( stdout ); - return err; -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return err; -} diff --git a/pd/portaudio_v18/pa_tests/patest_mono.c b/pd/portaudio_v18/pa_tests/patest_mono.c deleted file mode 100644 index 18682b5d..00000000 --- a/pd/portaudio_v18/pa_tests/patest_mono.c +++ /dev/null @@ -1,136 +0,0 @@ -/* - * $Id: patest_mono.c,v 1.1.2.3 2003/04/10 23:09:40 philburk Exp $ - * patest_sine.c - * Play a monophonic sine wave using the Portable Audio api for several seconds. - * - * Authors: - * Ross Bencina <rossb@audiomulch.com> - * Phil Burk <philburk@softsynth.com> - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.audiomulch.com/portaudio/ - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <math.h> -#include "portaudio.h" - -#define NUM_SECONDS (10) -#define SAMPLE_RATE (44100) -#define AMPLITUDE (0.8) -#define FRAMES_PER_BUFFER (64) -#define OUTPUT_DEVICE Pa_GetDefaultOutputDeviceID() - -#ifndef M_PI -#define M_PI (3.14159265) -#endif - -#define TABLE_SIZE (200) -typedef struct -{ - float sine[TABLE_SIZE]; - int phase; -} -paTestData; - -/* This routine will be called by the PortAudio engine when audio is needed. -** It may called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int patestCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData*)userData; - float *out = (float*)outputBuffer; - unsigned long i; - int finished = 0; - (void) outTime; /* Prevent unused variable warnings. */ - (void) inputBuffer; - for( i=0; i<framesPerBuffer; i++ ) - { - *out++ = data->sine[data->phase]; /* left */ - data->phase += 1; - if( data->phase >= TABLE_SIZE ) data->phase -= TABLE_SIZE; - } - return finished; -} - -/*******************************************************************/ -int main(void); -int main(void) -{ - PortAudioStream *stream; - PaError err; - paTestData data; - int i; - printf("PortAudio Test: output MONO sine wave. SR = %d, BufSize = %d\n", SAMPLE_RATE, FRAMES_PER_BUFFER); - /* initialise sinusoidal wavetable */ - for( i=0; i<TABLE_SIZE; i++ ) - { - data.sine[i] = (float) (AMPLITUDE * sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. )); - } - data.phase = 0; - - err = Pa_Initialize(); - if( err != paNoError ) goto error; - err = Pa_OpenStream( - &stream, - paNoDevice,/* default input device */ - 0, /* no input */ - paFloat32, /* 32 bit floating point input */ - NULL, - OUTPUT_DEVICE, - 1, /* MONO output */ - paFloat32, /* 32 bit floating point output */ - NULL, - SAMPLE_RATE, - FRAMES_PER_BUFFER, - 0, /* number of buffers, if zero then use default minimum */ - paClipOff, /* we won't output out of range samples so don't bother clipping them */ - patestCallback, - &data ); - if( err != paNoError ) goto error; - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - printf("Play for %d seconds.\n", NUM_SECONDS ); fflush(stdout); - Pa_Sleep( NUM_SECONDS * 1000 ); - - err = Pa_StopStream( stream ); - if( err != paNoError ) goto error; - err = Pa_CloseStream( stream ); - if( err != paNoError ) goto error; - Pa_Terminate(); - printf("Test finished.\n"); - return err; -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return err; -} diff --git a/pd/portaudio_v18/pa_tests/patest_multi_sine.c b/pd/portaudio_v18/pa_tests/patest_multi_sine.c deleted file mode 100644 index d005287a..00000000 --- a/pd/portaudio_v18/pa_tests/patest_multi_sine.c +++ /dev/null @@ -1,144 +0,0 @@ -/* - * $Id: patest_multi_sine.c,v 1.1.4.2 2003/02/13 18:05:30 philburk Exp $ - * patest_multi_out.c - * Play a different sine wave on each channels, - * using the Portable Audio api. - * - * Author: Phil Burk http://www.softsynth.com - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.portaudio.com - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <math.h> -#include "portaudio.h" - -#define OUTPUT_DEVICE (Pa_GetDefaultOutputDeviceID()) -#define SAMPLE_RATE (44100) -#define FRAMES_PER_BUFFER (256) -#define FREQ_INCR (300.0 / SAMPLE_RATE) -#define MAX_CHANNELS (64) - -#ifndef M_PI -#define M_PI (3.14159265) -#endif - -typedef struct -{ - int numChannels; - double phases[MAX_CHANNELS]; -} -paTestData; - -/* This routine will be called by the PortAudio engine when audio is needed. -** It may called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int patestCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData*)userData; - float *out = (float*)outputBuffer; - int frameIndex, channelIndex; - (void) outTime; /* Prevent unused variable warnings. */ - (void) inputBuffer; - - for( frameIndex=0; frameIndex<(int)framesPerBuffer; frameIndex++ ) - { - for( channelIndex=0; channelIndex<data->numChannels; channelIndex++ ) - { - /* Output sine wave on every channel. */ - *out++ = (float) (0.7 * sin(data->phases[channelIndex])); - - /* Play each channel at a higher frequency. */ - data->phases[channelIndex] += FREQ_INCR * (4 + channelIndex); - if( data->phases[channelIndex] >= (2.0 * M_PI) ) data->phases[channelIndex] -= (2.0 * M_PI); - } - } - - return 0; -} - -/*******************************************************************/ -int main(void); -int main(void) -{ - PortAudioStream *stream; - PaError err; - const PaDeviceInfo *pdi; - paTestData data = {0}; - printf("PortAudio Test: output sine wave on each channel.\n" ); - - err = Pa_Initialize(); - if( err != paNoError ) goto error; - - pdi = Pa_GetDeviceInfo( OUTPUT_DEVICE ); - data.numChannels = pdi->maxOutputChannels; - if( data.numChannels > MAX_CHANNELS ) data.numChannels = MAX_CHANNELS; - printf("Number of Channels = %d\n", data.numChannels ); - - err = Pa_OpenStream( - &stream, - paNoDevice, /* default input device */ - 0, /* no input */ - paFloat32, /* 32 bit floating point input */ - NULL, - OUTPUT_DEVICE, - data.numChannels, - paFloat32, /* 32 bit floating point output */ - NULL, - SAMPLE_RATE, - FRAMES_PER_BUFFER, /* frames per buffer */ - 0, /* number of buffers, if zero then use default minimum */ - paClipOff, /* we won't output out of range samples so don't bother clipping them */ - patestCallback, - &data ); - if( err != paNoError ) goto error; - - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - - printf("Hit ENTER to stop sound.\n"); - fflush(stdout); - getchar(); - - err = Pa_StopStream( stream ); - if( err != paNoError ) goto error; - - Pa_CloseStream( stream ); - Pa_Terminate(); - printf("Test finished.\n"); - return err; -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return err; -} diff --git a/pd/portaudio_v18/pa_tests/patest_pink.c b/pd/portaudio_v18/pa_tests/patest_pink.c deleted file mode 100644 index f3bfdadb..00000000 --- a/pd/portaudio_v18/pa_tests/patest_pink.c +++ /dev/null @@ -1,245 +0,0 @@ -/* - * $Id: patest_pink.c,v 1.1.1.1 2002/01/22 00:52:36 phil Exp $ - patest_pink.c - Generate Pink Noise using Gardner method. - Optimization suggested by James McCartney uses a tree - to select which random value to replace. - x x x x x x x x x x x x x x x x - x x x x x x x x - x x x x - x x - x - Tree is generated by counting trailing zeros in an increasing index. - When the index is zero, no random number is selected. - * - * Author: Phil Burk http://www.softsynth.com - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.portaudio.com - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <math.h> -#include "portaudio.h" -#define PINK_MAX_RANDOM_ROWS (30) -#define PINK_RANDOM_BITS (24) -#define PINK_RANDOM_SHIFT ((sizeof(long)*8)-PINK_RANDOM_BITS) -typedef struct -{ - long pink_Rows[PINK_MAX_RANDOM_ROWS]; - long pink_RunningSum; /* Used to optimize summing of generators. */ - int pink_Index; /* Incremented each sample. */ - int pink_IndexMask; /* Index wrapped by ANDing with this mask. */ - float pink_Scalar; /* Used to scale within range of -1.0 to +1.0 */ -} -PinkNoise; -/* Prototypes */ -static unsigned long GenerateRandomNumber( void ); -void InitializePinkNoise( PinkNoise *pink, int numRows ); -float GeneratePinkNoise( PinkNoise *pink ); -/************************************************************/ -/* Calculate pseudo-random 32 bit number based on linear congruential method. */ -static unsigned long GenerateRandomNumber( void ) -{ - /* Change this seed for different random sequences. */ - static unsigned long randSeed = 22222; - randSeed = (randSeed * 196314165) + 907633515; - return randSeed; -} -/************************************************************/ -/* Setup PinkNoise structure for N rows of generators. */ -void InitializePinkNoise( PinkNoise *pink, int numRows ) -{ - int i; - long pmax; - pink->pink_Index = 0; - pink->pink_IndexMask = (1<<numRows) - 1; - /* Calculate maximum possible signed random value. Extra 1 for white noise always added. */ - pmax = (numRows + 1) * (1<<(PINK_RANDOM_BITS-1)); - pink->pink_Scalar = 1.0f / pmax; - /* Initialize rows. */ - for( i=0; i<numRows; i++ ) pink->pink_Rows[i] = 0; - pink->pink_RunningSum = 0; -} -#define PINK_MEASURE -#ifdef PINK_MEASURE -float pinkMax = -999.0; -float pinkMin = 999.0; -#endif -/* Generate Pink noise values between -1.0 and +1.0 */ -float GeneratePinkNoise( PinkNoise *pink ) -{ - long newRandom; - long sum; - float output; - /* Increment and mask index. */ - pink->pink_Index = (pink->pink_Index + 1) & pink->pink_IndexMask; - /* If index is zero, don't update any random values. */ - if( pink->pink_Index != 0 ) - { - /* Determine how many trailing zeros in PinkIndex. */ - /* This algorithm will hang if n==0 so test first. */ - int numZeros = 0; - int n = pink->pink_Index; - while( (n & 1) == 0 ) - { - n = n >> 1; - numZeros++; - } - /* Replace the indexed ROWS random value. - * Subtract and add back to RunningSum instead of adding all the random - * values together. Only one changes each time. - */ - pink->pink_RunningSum -= pink->pink_Rows[numZeros]; - newRandom = ((long)GenerateRandomNumber()) >> PINK_RANDOM_SHIFT; - pink->pink_RunningSum += newRandom; - pink->pink_Rows[numZeros] = newRandom; - } - - /* Add extra white noise value. */ - newRandom = ((long)GenerateRandomNumber()) >> PINK_RANDOM_SHIFT; - sum = pink->pink_RunningSum + newRandom; - /* Scale to range of -1.0 to 0.9999. */ - output = pink->pink_Scalar * sum; -#ifdef PINK_MEASURE - /* Check Min/Max */ - if( output > pinkMax ) pinkMax = output; - else if( output < pinkMin ) pinkMin = output; -#endif - return output; -} -/*******************************************************************/ -#define PINK_TEST -#ifdef PINK_TEST -/* Context for callback routine. */ -typedef struct -{ - PinkNoise leftPink; - PinkNoise rightPink; - unsigned int sampsToGo; -} -paTestData; -/* This routine will be called by the PortAudio engine when audio is needed. -** It may called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int patestCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - int finished; - int i; - int numFrames; - paTestData *data = (paTestData*)userData; - float *out = (float*)outputBuffer; - (void) inputBuffer; /* Prevent "unused variable" warnings. */ - (void) outTime; - - /* Are we almost at end. */ - if( data->sampsToGo < framesPerBuffer ) - { - numFrames = data->sampsToGo; - finished = 1; - } - else - { - numFrames = framesPerBuffer; - finished = 0; - } - for( i=0; i<numFrames; i++ ) - { - *out++ = GeneratePinkNoise( &data->leftPink ); - *out++ = GeneratePinkNoise( &data->rightPink ); - } - data->sampsToGo -= numFrames; - return finished; -} -/*******************************************************************/ -int main(void); -int main(void) -{ - PortAudioStream *stream; - PaError err; - paTestData data; - int totalSamps; - /* Initialize two pink noise signals with different numbers of rows. */ - InitializePinkNoise( &data.leftPink, 12 ); - InitializePinkNoise( &data.rightPink, 16 ); - /* Look at a few values. */ - { - int i; - float pink; - for( i=0; i<20; i++ ) - { - pink = GeneratePinkNoise( &data.leftPink ); - printf("Pink = %f\n", pink ); - } - } - data.sampsToGo = totalSamps = 8*44100; /* Play for a few seconds. */ - err = Pa_Initialize(); - if( err != paNoError ) goto error; - /* Open a stereo PortAudio stream so we can hear the result. */ - err = Pa_OpenStream( - &stream, - paNoDevice, - 0, /* no input */ - paFloat32, /* 32 bit floating point input */ - NULL, - Pa_GetDefaultOutputDeviceID(), /* default output device */ - 2, /* stereo output */ - paFloat32, /* 32 bit floating point output */ - NULL, - 44100., - 2048, /* 46 msec buffers */ - 0, /* number of buffers, if zero then use default minimum */ - paClipOff, /* we won't output out of range samples so don't bother clipping them */ - patestCallback, - &data ); - if( err != paNoError ) goto error; - - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - printf("Waiting for sound to finish.\n"); - while( Pa_StreamActive( stream ) ) - { - Pa_Sleep(100); /* SPIN! */ - } - err = Pa_CloseStream( stream ); - if( err != paNoError ) goto error; -#ifdef PINK_MEASURE - printf("Pink min = %f, max = %f\n", pinkMin, pinkMax ); -#endif - Pa_Terminate(); - return 0; -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return 0; -} -#endif /* PINK_TEST */ diff --git a/pd/portaudio_v18/pa_tests/patest_record.c b/pd/portaudio_v18/pa_tests/patest_record.c deleted file mode 100644 index f7f79bd3..00000000 --- a/pd/portaudio_v18/pa_tests/patest_record.c +++ /dev/null @@ -1,325 +0,0 @@ -/* - * $Id: patest_record.c,v 1.2.4.4 2003/04/16 19:07:56 philburk Exp $ - * patest_record.c - * Record input into an array. - * Optionally save array to a file. - * Playback recorded data. - * - * Author: Phil Burk http://www.softsynth.com - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.portaudio.com - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <stdlib.h> -#include "portaudio.h" - -/* #define SAMPLE_RATE (17932) // Test failure to open with this value. */ -#define SAMPLE_RATE (44100) -#define NUM_SECONDS (5) -#define NUM_CHANNELS (2) -/* #define DITHER_FLAG (paDitherOff) */ -#define DITHER_FLAG (0) /**/ -#define FRAMES_PER_BUFFER (1024) - -/* Select sample format. */ -#if 1 -#define PA_SAMPLE_TYPE paFloat32 -typedef float SAMPLE; -#define SAMPLE_SILENCE (0.0f) -#elif 0 -#define PA_SAMPLE_TYPE paInt16 -typedef short SAMPLE; -#define SAMPLE_SILENCE (0) -#elif 0 -#define PA_SAMPLE_TYPE paInt8 -typedef char SAMPLE; -#define SAMPLE_SILENCE (0) -#else -#define PA_SAMPLE_TYPE paUInt8 -typedef unsigned char SAMPLE; -#define SAMPLE_SILENCE (128) - -#endif - -typedef struct -{ - int frameIndex; /* Index into sample array. */ - int maxFrameIndex; - SAMPLE *recordedSamples; -} -paTestData; -/* This routine will be called by the PortAudio engine when audio is needed. -** It may be called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int recordCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData*)userData; - SAMPLE *rptr = (SAMPLE*)inputBuffer; - SAMPLE *wptr = &data->recordedSamples[data->frameIndex * NUM_CHANNELS]; - long framesToRecord; - long i; - int finished; - unsigned long framesLeft = data->maxFrameIndex - data->frameIndex; - int samplesToRecord; - - (void) outputBuffer; /* Prevent unused variable warnings. */ - (void) outTime; - - if( framesLeft < framesPerBuffer ) - { - framesToRecord = framesLeft; - finished = 1; - } - else - { - framesToRecord = framesPerBuffer; - finished = 0; - } - - samplesToRecord = framesToRecord * NUM_CHANNELS; - - if( inputBuffer == NULL ) - { - for( i=0; i<samplesToRecord; i++ ) - { - *wptr++ = SAMPLE_SILENCE; - } - } - else - { - for( i=0; i<samplesToRecord; i++ ) - { - *wptr++ = *rptr++; - } - } - data->frameIndex += framesToRecord; - return finished; -} - -/* This routine will be called by the PortAudio engine when audio is needed. -** It may be called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int playCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData*)userData; - SAMPLE *rptr = &data->recordedSamples[data->frameIndex * NUM_CHANNELS]; - SAMPLE *wptr = (SAMPLE*)outputBuffer; - unsigned int i; - int finished; - unsigned int framesLeft = data->maxFrameIndex - data->frameIndex; - (void) inputBuffer; /* Prevent unused variable warnings. */ - (void) outTime; - int framesToPlay, samplesToPlay, samplesPerBuffer; - - if( framesLeft < framesPerBuffer ) - { - framesToPlay = framesLeft; - finished = 1; - } - else - { - framesToPlay = framesPerBuffer; - finished = 0; - } - - samplesToPlay = framesToPlay * NUM_CHANNELS; - samplesPerBuffer = framesPerBuffer * NUM_CHANNELS; - - for( i=0; i<samplesToPlay; i++ ) - { - *wptr++ = *rptr++; - } - for( ; i<framesPerBuffer; i++ ) - { - *wptr++ = 0; /* left */ - if( NUM_CHANNELS == 2 ) *wptr++ = 0; /* right */ - } - data->frameIndex += framesToPlay; - - return finished; -} - -/*******************************************************************/ -int main(void); -int main(void) -{ - PortAudioStream *stream; - PaError err; - paTestData data; - int i; - int totalFrames; - int numSamples; - int numBytes; - SAMPLE max, average, val; - printf("patest_record.c\n"); fflush(stdout); - - data.maxFrameIndex = totalFrames = NUM_SECONDS * SAMPLE_RATE; /* Record for a few seconds. */ - data.frameIndex = 0; - numSamples = totalFrames * NUM_CHANNELS; - - numBytes = numSamples * sizeof(SAMPLE); - data.recordedSamples = (SAMPLE *) malloc( numBytes ); - if( data.recordedSamples == NULL ) - { - printf("Could not allocate record array.\n"); - exit(1); - } - for( i=0; i<numSamples; i++ ) data.recordedSamples[i] = 0; - - err = Pa_Initialize(); - if( err != paNoError ) goto error; - - /* Record some audio. -------------------------------------------- */ - err = Pa_OpenStream( - &stream, - Pa_GetDefaultInputDeviceID(), - NUM_CHANNELS, - PA_SAMPLE_TYPE, - NULL, - paNoDevice, - 0, - PA_SAMPLE_TYPE, - NULL, - SAMPLE_RATE, - FRAMES_PER_BUFFER, /* frames per buffer */ - 0, /* number of buffers, if zero then use default minimum */ - 0, /* paDitherOff, // flags */ - recordCallback, - &data ); - if( err != paNoError ) goto error; - - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - printf("Now recording!!\n"); fflush(stdout); - - while( Pa_StreamActive( stream ) ) - { - Pa_Sleep(1000); - printf("index = %d\n", data.frameIndex ); fflush(stdout); - } - - err = Pa_CloseStream( stream ); - if( err != paNoError ) goto error; - - /* Measure maximum peak amplitude. */ - max = 0; - average = 0; - for( i=0; i<numSamples; i++ ) - { - val = data.recordedSamples[i]; - if( val < 0 ) val = -val; /* ABS */ - if( val > max ) - { - max = val; - } - average += val; - } - - average = average / numSamples; - - if( PA_SAMPLE_TYPE == paFloat32 ) /* This should be done at compile-time with "#if" ?? */ - { /* MIPS-compiler warns at the int-version below. */ - printf("sample max amplitude = %f\n", max ); - printf("sample average = %f\n", average ); - } - else - { - printf("sample max amplitude = %d\n", max ); /* <-- This IS compiled anyhow. */ - printf("sample average = %d\n", average ); - } - - /* Write recorded data to a file. */ -#if 0 - { - FILE *fid; - fid = fopen("recorded.raw", "wb"); - if( fid == NULL ) - { - printf("Could not open file."); - } - else - { - fwrite( data.recordedSamples, NUM_CHANNELS * sizeof(SAMPLE), totalFrames, fid ); - fclose( fid ); - printf("Wrote data to 'recorded.raw'\n"); - } - } -#endif - - /* Playback recorded data. -------------------------------------------- */ - data.frameIndex = 0; - printf("Begin playback.\n"); fflush(stdout); - err = Pa_OpenStream( - &stream, - paNoDevice, - 0, /* NO input */ - PA_SAMPLE_TYPE, - NULL, - Pa_GetDefaultOutputDeviceID(), - NUM_CHANNELS, - PA_SAMPLE_TYPE, - NULL, - SAMPLE_RATE, - FRAMES_PER_BUFFER, /* frames per buffer */ - 0, /* number of buffers, if zero then use default minimum */ - paClipOff, /* we won't output out of range samples so don't bother clipping them */ - playCallback, - &data ); - if( err != paNoError ) goto error; - - if( stream ) - { - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - printf("Waiting for playback to finish.\n"); fflush(stdout); - - while( Pa_StreamActive( stream ) ) Pa_Sleep(100); - - err = Pa_CloseStream( stream ); - if( err != paNoError ) goto error; - printf("Done.\n"); fflush(stdout); - } - free( data.recordedSamples ); - - Pa_Terminate(); - return 0; - -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return -1; -} diff --git a/pd/portaudio_v18/pa_tests/patest_ringmix.c b/pd/portaudio_v18/pa_tests/patest_ringmix.c deleted file mode 100644 index 9d78ea13..00000000 --- a/pd/portaudio_v18/pa_tests/patest_ringmix.c +++ /dev/null @@ -1,41 +0,0 @@ -/* $Id: patest_ringmix.c,v 1.1.1.1 2002/01/22 00:52:37 phil Exp $ */ - -#include "stdio.h" -#include "portaudio.h" -/* This will be called asynchronously by the PortAudio engine. */ -static int myCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, PaTimestamp outTime, void *userData ) -{ - float *out = (float *) outputBuffer; - float *in = (float *) inputBuffer; - float leftInput, rightInput; - unsigned int i; - if( inputBuffer == NULL ) return 0; - /* Read input buffer, process data, and fill output buffer. */ - for( i=0; i<framesPerBuffer; i++ ) - { - leftInput = *in++; /* Get interleaved samples from input buffer. */ - rightInput = *in++; - *out++ = leftInput * rightInput; /* ring modulation */ - *out++ = 0.5f * (leftInput + rightInput); /* mix */ - } - return 0; -} -/* Open a PortAudioStream to input and output audio data. */ -int main(void) -{ - PortAudioStream *stream; - Pa_Initialize(); - Pa_OpenDefaultStream( - &stream, - 2, 2, /* stereo input and output */ - paFloat32, 44100.0, - 64, 0, /* 64 frames per buffer, let PA determine numBuffers */ - myCallback, NULL ); - Pa_StartStream( stream ); - Pa_Sleep( 10000 ); /* Sleep for 10 seconds while processing. */ - Pa_StopStream( stream ); - Pa_CloseStream( stream ); - Pa_Terminate(); - return 0; -} diff --git a/pd/portaudio_v18/pa_tests/patest_saw.c b/pd/portaudio_v18/pa_tests/patest_saw.c deleted file mode 100644 index 187db354..00000000 --- a/pd/portaudio_v18/pa_tests/patest_saw.c +++ /dev/null @@ -1,118 +0,0 @@ -/* - * $Id: patest_saw.c,v 1.1.1.1 2002/01/22 00:52:38 phil Exp $ - * patest_saw.c - * Play a simple sawtooth wave. - * - * Author: Phil Burk http://www.softsynth.com - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.portaudio.com - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <math.h> -#include "portaudio.h" -#define NUM_SECONDS (4) -#define SAMPLE_RATE (44100) -typedef struct -{ - float left_phase; - float right_phase; -} -paTestData; -/* This routine will be called by the PortAudio engine when audio is needed. -** It may called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int patestCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - /* Cast data passed through stream to our structure. */ - paTestData *data = (paTestData*)userData; - float *out = (float*)outputBuffer; - unsigned int i; - (void) outTime; /* Prevent unused variable warnings. */ - (void) inputBuffer; - - for( i=0; i<framesPerBuffer; i++ ) - { - *out++ = data->left_phase; /* left */ - *out++ = data->right_phase; /* right */ - /* Generate simple sawtooth phaser that ranges between -1.0 and 1.0. */ - data->left_phase += 0.01f; - /* When signal reaches top, drop back down. */ - if( data->left_phase >= 1.0f ) data->left_phase -= 2.0f; - /* higher pitch so we can distinguish left and right. */ - data->right_phase += 0.03f; - if( data->right_phase >= 1.0f ) data->right_phase -= 2.0f; - } - return 0; -} -/*******************************************************************/ -static paTestData data; -int main(void); -int main(void) -{ - PortAudioStream *stream; - PaError err; - printf("PortAudio Test: output sawtooth wave.\n"); - /* Initialize our data for use by callback. */ - data.left_phase = data.right_phase = 0.0; - /* Initialize library before making any other calls. */ - err = Pa_Initialize(); - if( err != paNoError ) goto error; - /* Open an audio I/O stream. */ - err = Pa_OpenDefaultStream( - &stream, - 0, /* no input channels */ - 2, /* stereo output */ - paFloat32, /* 32 bit floating point output */ - SAMPLE_RATE, - 256, /* frames per buffer */ - 0, /* number of buffers, if zero then use default minimum */ - patestCallback, - &data ); - if( err != paNoError ) goto error; - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - /* Sleep for several seconds. */ - Pa_Sleep(NUM_SECONDS*1000); - err = Pa_StopStream( stream ); - if( err != paNoError ) goto error; - err = Pa_CloseStream( stream ); - if( err != paNoError ) goto error; - Pa_Terminate(); - printf("Test finished.\n"); - return err; -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return err; -} diff --git a/pd/portaudio_v18/pa_tests/patest_sine.c b/pd/portaudio_v18/pa_tests/patest_sine.c deleted file mode 100644 index 162fac02..00000000 --- a/pd/portaudio_v18/pa_tests/patest_sine.c +++ /dev/null @@ -1,141 +0,0 @@ -/* - * $Id: patest_sine.c,v 1.2.4.1 2003/02/11 21:41:32 philburk Exp $ - * patest_sine.c - * Play a sine wave using the Portable Audio api for several seconds. - * - * Authors: - * Ross Bencina <rossb@audiomulch.com> - * Phil Burk <philburk@softsynth.com> - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.audiomulch.com/portaudio/ - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <math.h> -#include "portaudio.h" - -#define NUM_SECONDS (10) -#define SAMPLE_RATE (44100) -#define AMPLITUDE (0.9) -#define FRAMES_PER_BUFFER (64) -#define OUTPUT_DEVICE Pa_GetDefaultOutputDeviceID() -//#define OUTPUT_DEVICE (2) - -#ifndef M_PI -#define M_PI (3.14159265) -#endif - -#define TABLE_SIZE (200) -typedef struct -{ - float sine[TABLE_SIZE]; - int left_phase; - int right_phase; -} -paTestData; - -/* This routine will be called by the PortAudio engine when audio is needed. -** It may called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int patestCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData*)userData; - float *out = (float*)outputBuffer; - unsigned long i; - int finished = 0; - (void) outTime; /* Prevent unused variable warnings. */ - (void) inputBuffer; - for( i=0; i<framesPerBuffer; i++ ) - { - *out++ = data->sine[data->left_phase]; /* left */ - *out++ = data->sine[data->right_phase]; /* right */ - data->left_phase += 1; - if( data->left_phase >= TABLE_SIZE ) data->left_phase -= TABLE_SIZE; - data->right_phase += 3; /* higher pitch so we can distinguish left and right. */ - if( data->right_phase >= TABLE_SIZE ) data->right_phase -= TABLE_SIZE; - } - return finished; -} - -/*******************************************************************/ -int main(void); -int main(void) -{ - PortAudioStream *stream; - PaError err; - paTestData data; - int i; - printf("PortAudio Test: output sine wave. SR = %d, BufSize = %d, devID = %d\n", - SAMPLE_RATE, FRAMES_PER_BUFFER, OUTPUT_DEVICE); - /* initialise sinusoidal wavetable */ - for( i=0; i<TABLE_SIZE; i++ ) - { - data.sine[i] = (float) (AMPLITUDE * sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. )); - } - data.left_phase = data.right_phase = 0; - err = Pa_Initialize(); - if( err != paNoError ) goto error; - err = Pa_OpenStream( - &stream, - paNoDevice,/* default input device */ - 0, /* no input */ - paFloat32, /* 32 bit floating point input */ - NULL, - OUTPUT_DEVICE, - 2, /* stereo output */ - paFloat32, /* 32 bit floating point output */ - NULL, - SAMPLE_RATE, - FRAMES_PER_BUFFER, - 0, /* number of buffers, if zero then use default minimum */ - paClipOff, /* we won't output out of range samples so don't bother clipping them */ - patestCallback, - &data ); - if( err != paNoError ) goto error; - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - printf("Play for %d seconds.\n", NUM_SECONDS ); fflush(stdout); - Pa_Sleep( NUM_SECONDS * 1000 ); - - err = Pa_StopStream( stream ); - if( err != paNoError ) goto error; - err = Pa_CloseStream( stream ); - if( err != paNoError ) goto error; - Pa_Terminate(); - printf("Test finished.\n"); - return err; -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return err; -} diff --git a/pd/portaudio_v18/pa_tests/patest_sine8.c b/pd/portaudio_v18/pa_tests/patest_sine8.c deleted file mode 100644 index 36cf863a..00000000 --- a/pd/portaudio_v18/pa_tests/patest_sine8.c +++ /dev/null @@ -1,184 +0,0 @@ -/* - * $Id: patest_sine8.c,v 1.1.1.1 2002/01/22 00:52:38 phil Exp $ - * patest_sine8.c - * Play a sine wave using the Portable Audio api for several seconds. - * Test 8 bit data. - * - * Author: Ross Bencina <rossb@audiomulch.com> - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.audiomulch.com/portaudio/ - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <math.h> -#include "portaudio.h" -#define NUM_SECONDS (8) -#define SAMPLE_RATE (44100) -#define TEST_UNSIGNED (1) -#if TEST_UNSIGNED -#define TEST_FORMAT paUInt8 -#else -#define TEST_FORMAT paInt8 -#endif -#ifndef M_PI -#define M_PI (3.14159265) -#endif -#define TABLE_SIZE (200) -typedef struct -{ -#if TEST_UNSIGNED - unsigned char sine[TABLE_SIZE]; -#else - char sine[TABLE_SIZE]; -#endif - int left_phase; - int right_phase; - unsigned int framesToGo; -} -paTestData; -/* This routine will be called by the PortAudio engine when audio is needed. -** It may called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int patestCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData*)userData; - char *out = (char*)outputBuffer; - int i; - int framesToCalc; - int finished = 0; - (void) outTime; /* Prevent unused variable warnings. */ - (void) inputBuffer; - - if( data->framesToGo < framesPerBuffer ) - { - framesToCalc = data->framesToGo; - data->framesToGo = 0; - finished = 1; - } - else - { - framesToCalc = framesPerBuffer; - data->framesToGo -= framesPerBuffer; - } - - for( i=0; i<framesToCalc; i++ ) - { - *out++ = data->sine[data->left_phase]; /* left */ - *out++ = data->sine[data->right_phase]; /* right */ - data->left_phase += 1; - if( data->left_phase >= TABLE_SIZE ) data->left_phase -= TABLE_SIZE; - data->right_phase += 3; /* higher pitch so we can distinguish left and right. */ - if( data->right_phase >= TABLE_SIZE ) data->right_phase -= TABLE_SIZE; - } - /* zero remainder of final buffer */ - for( ; i<(int)framesPerBuffer; i++ ) - { -#if TEST_UNSIGNED - *out++ = (unsigned char) 0x80; /* left */ - *out++ = (unsigned char) 0x80; /* right */ -#else - *out++ = 0; /* left */ - *out++ = 0; /* right */ -#endif - - } - return finished; -} -/*******************************************************************/ -int main(void); -int main(void) -{ - PortAudioStream *stream; - PaError err; - paTestData data; - int i; - int totalSamps; -#if TEST_UNSIGNED - printf("PortAudio Test: output UNsigned 8 bit sine wave.\n"); -#else - printf("PortAudio Test: output signed 8 bit sine wave.\n"); -#endif - /* initialise sinusoidal wavetable */ - for( i=0; i<TABLE_SIZE; i++ ) - { - data.sine[i] = (char) (127.0 * sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. )); -#if TEST_UNSIGNED - data.sine[i] += (unsigned char) 0x80; -#endif - - } - data.left_phase = data.right_phase = 0; - data.framesToGo = totalSamps = NUM_SECONDS * SAMPLE_RATE; /* Play for a few seconds. */ - err = Pa_Initialize(); - if( err != paNoError ) goto error; - err = Pa_OpenStream( - &stream, - paNoDevice,/* default input device */ - 0, /* no input */ - TEST_FORMAT, - NULL, - Pa_GetDefaultOutputDeviceID(), /* default output device */ - 2, /* stereo output */ - TEST_FORMAT, - NULL, - SAMPLE_RATE, - 256, /* frames per buffer */ - 0, /* number of buffers, if zero then use default minimum */ - paClipOff, /* we won't output out of range samples so don't bother clipping them */ - patestCallback, - &data ); - if( err != paNoError ) goto error; - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - /* Watch until sound is halfway finished. */ - while( Pa_StreamTime( stream ) < (totalSamps/2) ) Pa_Sleep(10); - /* Stop sound until ENTER hit. */ - err = Pa_StopStream( stream ); - if( err != paNoError ) goto error; - printf("Pause for 2 seconds.\n"); - Pa_Sleep( 2000 ); - - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - printf("Waiting for sound to finish.\n"); - while( Pa_StreamActive( stream ) ) Pa_Sleep(10); - err = Pa_CloseStream( stream ); - if( err != paNoError ) goto error; - Pa_Terminate(); - printf("Test finished.\n"); - return err; -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return err; -} diff --git a/pd/portaudio_v18/pa_tests/patest_sine_formats.c b/pd/portaudio_v18/pa_tests/patest_sine_formats.c deleted file mode 100644 index 6be0708b..00000000 --- a/pd/portaudio_v18/pa_tests/patest_sine_formats.c +++ /dev/null @@ -1,203 +0,0 @@ -/* - * $Id: patest_sine_formats.c,v 1.2.4.2 2003/02/12 01:39:29 philburk Exp $ - * patest_sine_formats.c - * Play a sine wave using the Portable Audio api for several seconds. - * Test various data formats. - * - * Author: Phil Burk - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.audiomulch.com/portaudio/ - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <math.h> -#include "portaudio.h" - -#define NUM_SECONDS (5) -#define SAMPLE_RATE (44100) -#define FRAMES_PER_BUFFER (512) -#define LEFT_FREQ ((2 * SAMPLE_RATE)/FRAMES_PER_BUFFER) /* So we hit 1.0 */ -#define RIGHT_FREQ (500.0) -#define AMPLITUDE (0.9) - -/* Select ONE format for testing. */ -#define TEST_UINT8 (0) -#define TEST_INT8 (0) -#define TEST_INT16 (0) -#define TEST_INT32 (1) -#define TEST_FLOAT32 (0) - -#if TEST_UINT8 -#define TEST_FORMAT paUInt8 -typedef unsigned char SAMPLE_t; -#define SAMPLE_ZERO (0x80) -#define DOUBLE_TO_SAMPLE(x) (SAMPLE_ZERO + (SAMPLE_t)(127.0 * (x))) -#define FORMAT_NAME "Unsigned 8 Bit" - -#elif TEST_INT8 -#define TEST_FORMAT paInt8 -typedef char SAMPLE_t; -#define SAMPLE_ZERO (0) -#define DOUBLE_TO_SAMPLE(x) (SAMPLE_ZERO + (SAMPLE_t)(127.0 * (x))) -#define FORMAT_NAME "Signed 8 Bit" - -#elif TEST_INT16 -#define TEST_FORMAT paInt16 -typedef short SAMPLE_t; -#define SAMPLE_ZERO (0) -#define DOUBLE_TO_SAMPLE(x) (SAMPLE_ZERO + (SAMPLE_t)(32767 * (x))) -#define FORMAT_NAME "Signed 16 Bit" - -#elif TEST_INT32 -#define TEST_FORMAT paInt32 -typedef long SAMPLE_t; -#define SAMPLE_ZERO (0) -#define DOUBLE_TO_SAMPLE(x) (SAMPLE_ZERO + (SAMPLE_t)(0x7FFFFFFF * (x))) -#define FORMAT_NAME "Signed 32 Bit" - -#elif TEST_FLOAT32 -#define TEST_FORMAT paFloat32 -typedef float SAMPLE_t; -#define SAMPLE_ZERO (0.0) -#define DOUBLE_TO_SAMPLE(x) ((SAMPLE_t)(x)) -#define FORMAT_NAME "Float 32 Bit" -#endif - -#ifndef M_PI -#define M_PI (3.14159265) -#endif - - -typedef struct -{ - double left_phase; - double right_phase; - unsigned int framesToGo; -} -paTestData; -/* This routine will be called by the PortAudio engine when audio is needed. -** It may called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int patestCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData*)userData; - SAMPLE_t *out = (SAMPLE_t *)outputBuffer; - int i; - int framesToCalc; - int finished = 0; - (void) outTime; /* Prevent unused variable warnings. */ - (void) inputBuffer; - - if( data->framesToGo < framesPerBuffer ) - { - framesToCalc = data->framesToGo; - data->framesToGo = 0; - finished = 1; - } - else - { - framesToCalc = framesPerBuffer; - data->framesToGo -= framesPerBuffer; - } - - for( i=0; i<framesToCalc; i++ ) - { - data->left_phase += (LEFT_FREQ / SAMPLE_RATE); - if( data->left_phase > 1.0) data->left_phase -= 1.0; - *out++ = DOUBLE_TO_SAMPLE( AMPLITUDE * sin( (data->left_phase * M_PI * 2. ))); - - data->right_phase += (RIGHT_FREQ / SAMPLE_RATE); - if( data->right_phase > 1.0) data->right_phase -= 1.0; - *out++ = DOUBLE_TO_SAMPLE( AMPLITUDE * sin( (data->right_phase * M_PI * 2. ))); - } - /* zero remainder of final buffer */ - for( ; i<(int)framesPerBuffer; i++ ) - { - *out++ = SAMPLE_ZERO; /* left */ - *out++ = SAMPLE_ZERO; /* right */ - } - return finished; -} -/*******************************************************************/ -int main(void); -int main(void) -{ - PortAudioStream *stream; - PaError err; - paTestData data; - int totalSamps; - - printf("PortAudio Test: output " FORMAT_NAME "\n"); - - - data.left_phase = data.right_phase = 0.0; - data.framesToGo = totalSamps = NUM_SECONDS * SAMPLE_RATE; /* Play for a few seconds. */ - err = Pa_Initialize(); - if( err != paNoError ) goto error; - - err = Pa_OpenStream( - &stream, - paNoDevice,/* default input device */ - 0, /* no input */ - TEST_FORMAT, - NULL, - Pa_GetDefaultOutputDeviceID(), /* default output device */ - 2, /* stereo output */ - TEST_FORMAT, - NULL, - SAMPLE_RATE, - FRAMES_PER_BUFFER, - 0, /* number of buffers, if zero then use default minimum */ - paClipOff, /* we won't output out of range samples so don't bother clipping them */ - patestCallback, - &data ); - if( err != paNoError ) goto error; - - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - - printf("Waiting %d seconds for sound to finish.\n", NUM_SECONDS ); fflush(stdout); - while( Pa_StreamActive( stream ) ) Pa_Sleep(10); - - err = Pa_CloseStream( stream ); - if( err != paNoError ) goto error; - Pa_Terminate(); - - printf("PortAudio Test Finished: " FORMAT_NAME "\n"); - - return err; -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return err; -} diff --git a/pd/portaudio_v18/pa_tests/patest_sine_time.c b/pd/portaudio_v18/pa_tests/patest_sine_time.c deleted file mode 100644 index e9bc33da..00000000 --- a/pd/portaudio_v18/pa_tests/patest_sine_time.c +++ /dev/null @@ -1,205 +0,0 @@ -/* - * $Id: patest_sine_time.c,v 1.2 2002/03/21 00:58:45 philburk Exp $ - * patest_sine_time.c - * Play a sine wave using the Portable Audio api for several seconds. - * Pausing in the middle. - * use the Pa_StreamTime() and Pa_StreamActive() calls. - * - * Authors: - * Ross Bencina <rossb@audiomulch.com> - * Phil Burk <philburk@softsynth.com> - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.audiomulch.com/portaudio/ - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <math.h> -#include "portaudio.h" -#define NUM_SECONDS (8) -#define SAMPLE_RATE (44100) -#define FRAMES_PER_BUFFER (64) -#define NUM_BUFFERS (0) -#ifndef M_PI -#define M_PI (3.14159265) -#endif -#define TABLE_SIZE (200) -typedef struct -{ - float sine[TABLE_SIZE]; - int left_phase; - int right_phase; - int framesToGo; - volatile PaTimestamp outTime; -} -paTestData; -/* This routine will be called by the PortAudio engine when audio is needed. -** It may called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int patestCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData*)userData; - float *out = (float*)outputBuffer; - int i; - int framesToCalc; - int finished = 0; - (void) outTime; /* Prevent unused variable warnings. */ - (void) inputBuffer; - - data->outTime = outTime; - - if( data->framesToGo < framesPerBuffer ) - { - framesToCalc = data->framesToGo; - data->framesToGo = 0; - finished = 1; - } - else - { - framesToCalc = framesPerBuffer; - data->framesToGo -= framesPerBuffer; - } - - for( i=0; i<framesToCalc; i++ ) - { - *out++ = data->sine[data->left_phase]; /* left */ - *out++ = data->sine[data->right_phase]; /* right */ - data->left_phase += 1; - if( data->left_phase >= TABLE_SIZE ) data->left_phase -= TABLE_SIZE; - data->right_phase += 3; /* higher pitch so we can distinguish left and right. */ - if( data->right_phase >= TABLE_SIZE ) data->right_phase -= TABLE_SIZE; - } - /* zero remainder of final buffer */ - for( ; i<(int)framesPerBuffer; i++ ) - { - *out++ = 0; /* left */ - *out++ = 0; /* right */ - } - return finished; -} -/*******************************************************************/ -static void ReportStreamTime( PortAudioStream *stream, paTestData *data ); -static void ReportStreamTime( PortAudioStream *stream, paTestData *data ) -{ - PaTimestamp streamTime, latency, outTime; - - streamTime = Pa_StreamTime( stream ); - outTime = data->outTime; - if( outTime < 0.0 ) - { - printf("Stream time = %8.1f\n", streamTime ); - } - else - { - latency = outTime - streamTime; - printf("Stream time = %8.1f, outTime = %8.1f, latency = %8.1f\n", - streamTime, outTime, latency ); - } - fflush(stdout); -} - -/*******************************************************************/ -int main(void); -int main(void) -{ - PortAudioStream *stream; - PaError err; - paTestData DATA; - int i; - int totalSamps; - printf("PortAudio Test: output sine wave. SR = %d, BufSize = %d\n", SAMPLE_RATE, FRAMES_PER_BUFFER); - /* initialise sinusoidal wavetable */ - for( i=0; i<TABLE_SIZE; i++ ) - { - DATA.sine[i] = (float) sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. ); - } - DATA.left_phase = DATA.right_phase = 0; - DATA.framesToGo = totalSamps = NUM_SECONDS * SAMPLE_RATE; /* Play for a few seconds. */ - err = Pa_Initialize(); - if( err != paNoError ) goto error; - err = Pa_OpenStream( - &stream, - paNoDevice,/* default input device */ - 0, /* no input */ - paFloat32, /* 32 bit floating point input */ - NULL, - Pa_GetDefaultOutputDeviceID(), /* default output device */ - 2, /* stereo output */ - paFloat32, /* 32 bit floating point output */ - NULL, - SAMPLE_RATE, - FRAMES_PER_BUFFER, /* frames per buffer */ - NUM_BUFFERS, /* number of buffers, if zero then use default minimum */ - paClipOff, /* we won't output out of range samples so don't bother clipping them */ - patestCallback, - &DATA ); - if( err != paNoError ) goto error; - - DATA.outTime = -1.0; // mark time for callback as undefined - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - - /* Watch until sound is halfway finished. */ - printf("Play for %d seconds.\n", NUM_SECONDS/2 ); fflush(stdout); - do - { - ReportStreamTime( stream, &DATA ); - Pa_Sleep(100); - } while( Pa_StreamTime( stream ) < (totalSamps/2) ); - - /* Stop sound until ENTER hit. */ - err = Pa_StopStream( stream ); - if( err != paNoError ) goto error; - printf("Pause for 2 seconds.\n"); fflush(stdout); - Pa_Sleep( 2000 ); - - DATA.outTime = -1.0; // mark time for callback as undefined - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - - printf("Play until sound is finished.\n"); fflush(stdout); - do - { - ReportStreamTime( stream, &DATA ); - Pa_Sleep(100); - } while( Pa_StreamActive( stream ) ); - - err = Pa_CloseStream( stream ); - if( err != paNoError ) goto error; - Pa_Terminate(); - printf("Test finished.\n"); - return err; -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return err; -} diff --git a/pd/portaudio_v18/pa_tests/patest_stop.c b/pd/portaudio_v18/pa_tests/patest_stop.c deleted file mode 100644 index 2fd9895c..00000000 --- a/pd/portaudio_v18/pa_tests/patest_stop.c +++ /dev/null @@ -1,285 +0,0 @@ -/* - * $Id: patest_stop.c,v 1.1.1.1 2002/01/22 00:52:39 phil Exp $ - * patest_stop.c - * - * Test the three ways of stopping audio: - * calling Pa_StopStream(), - * calling Pa_AbortStream(), - * and returning a 1 from the callback function. - * - * A long latency is set up so that you can hear the difference. - * Then a simple 8 note sequence is repeated twice. - * The program will print what you should hear. - * - * Author: Phil Burk <philburk@softsynth.com> - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.portaudio.com - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <math.h> -#include "portaudio.h" -#define OUTPUT_DEVICE (Pa_GetDefaultOutputDeviceID()) -#define SLEEP_DUR (200) -#define SAMPLE_RATE (44100) -#define FRAMES_PER_BUFFER (256) -#define LATENCY_MSEC (3000) -#define NUM_BUFFERS ((LATENCY_MSEC * SAMPLE_RATE) / (FRAMES_PER_BUFFER * 1000)) -#define FRAMES_PER_NOTE (SAMPLE_RATE/2) -#define MAX_REPEATS (2) -#define FUNDAMENTAL (400.0f / SAMPLE_RATE) -#define NOTE_0 (FUNDAMENTAL * 1.0f / 1.0f) -#define NOTE_1 (FUNDAMENTAL * 5.0f / 4.0f) -#define NOTE_2 (FUNDAMENTAL * 4.0f / 3.0f) -#define NOTE_3 (FUNDAMENTAL * 3.0f / 2.0f) -#define NOTE_4 (FUNDAMENTAL * 2.0f / 1.0f) -#define MODE_FINISH (0) -#define MODE_STOP (1) -#define MODE_ABORT (2) -#ifndef M_PI -#define M_PI (3.14159265) -#endif -#define TABLE_SIZE (400) -typedef struct -{ - float waveform[TABLE_SIZE + 1]; // add one for guard point for interpolation - float phase_increment; - float phase; - float *tune; - int notesPerTune; - int frameCounter; - int noteCounter; - int repeatCounter; - PaTimestamp outTime; - int stopMode; - int done; -} -paTestData; -/************* Prototypes *****************************/ -int TestStopMode( paTestData *data ); -float LookupWaveform( paTestData *data, float phase ); -/****************************************************** - * Convert phase between 0.0 and 1.0 to waveform value - * using linear interpolation. - */ -float LookupWaveform( paTestData *data, float phase ) -{ - float fIndex = phase*TABLE_SIZE; - int index = (int) fIndex; - float fract = fIndex - index; - float lo = data->waveform[index]; - float hi = data->waveform[index+1]; - float val = lo + fract*(hi-lo); - return val; -} -/* This routine will be called by the PortAudio engine when audio is needed. -** It may called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int patestCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData*)userData; - float *out = (float*)outputBuffer; - float value; - unsigned int i = 0; - int finished = 0; - (void) outTime; /* Prevent unused variable warnings. */ - (void) inputBuffer; - - data->outTime = outTime; - if( !data->done ) - { - for( i=0; i<framesPerBuffer; i++ ) - { - /* Are we done with this note? */ - if( data->frameCounter >= FRAMES_PER_NOTE ) - { - data->noteCounter += 1; - data->frameCounter = 0; - /* Are we done with this tune? */ - if( data->noteCounter >= data->notesPerTune ) - { - data->noteCounter = 0; - data->repeatCounter += 1; - /* Are we totally done? */ - if( data->repeatCounter >= MAX_REPEATS ) - { - data->done = 1; - if( data->stopMode == MODE_FINISH ) - { - finished = 1; - break; - } - } - } - data->phase_increment = data->tune[data->noteCounter]; - } - value = LookupWaveform(data, data->phase); - *out++ = value; /* left */ - *out++ = value; /* right */ - data->phase += data->phase_increment; - if( data->phase >= 1.0f ) data->phase -= 1.0f; - - data->frameCounter += 1; - } - } - /* zero remainder of final buffer */ - for( ; i<framesPerBuffer; i++ ) - { - *out++ = 0; /* left */ - *out++ = 0; /* right */ - } - return finished; -} -/*******************************************************************/ -int main(void); -int main(void) -{ - paTestData DATA; - int i; - float simpleTune[] = { NOTE_0, NOTE_1, NOTE_2, NOTE_3, NOTE_4, NOTE_3, NOTE_2, NOTE_1 }; - printf("PortAudio Test: play song and test stopping. ask for %d buffers\n", NUM_BUFFERS ); - /* initialise sinusoidal wavetable */ - for( i=0; i<TABLE_SIZE; i++ ) - { - DATA.waveform[i] = (float) ( - (0.2 * sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. )) + - (0.2 * sin( ((double)(3*i)/(double)TABLE_SIZE) * M_PI * 2. )) + - (0.1 * sin( ((double)(5*i)/(double)TABLE_SIZE) * M_PI * 2. )) - ); - } - DATA.waveform[TABLE_SIZE] = DATA.waveform[0]; // set guard point - DATA.tune = &simpleTune[0]; - DATA.notesPerTune = sizeof(simpleTune) / sizeof(float); - printf("Test MODE_FINISH - callback returns 1.\n"); - printf("Should hear entire %d note tune repeated twice.\n", DATA.notesPerTune); - DATA.stopMode = MODE_FINISH; - if( TestStopMode( &DATA ) != paNoError ) - { - printf("Test of MODE_FINISH failed!\n"); - goto error; - } - printf("Test MODE_STOP - stop when song is done.\n"); - printf("Should hear entire %d note tune repeated twice.\n", DATA.notesPerTune); - DATA.stopMode = MODE_STOP; - if( TestStopMode( &DATA ) != paNoError ) - { - printf("Test of MODE_STOP failed!\n"); - goto error; - } - - printf("Test MODE_ABORT - abort immediately.\n"); - printf("Should hear last repetition cut short by %d msec.\n", LATENCY_MSEC); - DATA.stopMode = MODE_ABORT; - if( TestStopMode( &DATA ) != paNoError ) - { - printf("Test of MODE_ABORT failed!\n"); - goto error; - } - return 0; -error: - return 1; -} - -int TestStopMode( paTestData *data ) -{ - PortAudioStream *stream; - PaError err; - data->done = 0; - data->phase = 0.0; - data->frameCounter = 0; - data->noteCounter = 0; - data->repeatCounter = 0; - data->phase_increment = data->tune[data->noteCounter]; - err = Pa_Initialize(); - if( err != paNoError ) goto error; - err = Pa_OpenStream( - &stream, - paNoDevice,/* default input device */ - 0, /* no input */ - paFloat32, /* 32 bit floating point input */ - NULL, - OUTPUT_DEVICE, - 2, /* stereo output */ - paFloat32, /* 32 bit floating point output */ - NULL, - SAMPLE_RATE, - FRAMES_PER_BUFFER, /* frames per buffer */ - NUM_BUFFERS, /* number of buffers, if zero then use default minimum */ - paClipOff, /* we won't output out of range samples so don't bother clipping them */ - patestCallback, - data ); - if( err != paNoError ) goto error; - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - if( data->stopMode == MODE_FINISH ) - { - while( Pa_StreamActive( stream ) ) - { - /*printf("outTime = %g, note# = %d, repeat# = %d\n", data->outTime, - data->noteCounter, data->repeatCounter ); - fflush(stdout); /**/ - Pa_Sleep( SLEEP_DUR ); - } - } - else - { - while( data->repeatCounter < MAX_REPEATS ) - { - /*printf("outTime = %g, note# = %d, repeat# = %d\n", data->outTime, - data->noteCounter, data->repeatCounter ); - fflush(stdout); /**/ - Pa_Sleep( SLEEP_DUR ); - } - } - if( data->stopMode == MODE_ABORT ) - { - printf("Call Pa_AbortStream()\n"); - err = Pa_AbortStream( stream ); - } - else - { - printf("Call Pa_StopStream()\n"); - err = Pa_StopStream( stream ); - } - if( err != paNoError ) goto error; - printf("Call Pa_CloseStream()\n"); fflush(stdout); - err = Pa_CloseStream( stream ); - if( err != paNoError ) goto error; - Pa_Terminate(); - printf("Test finished.\n"); - return err; -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return err; -} diff --git a/pd/portaudio_v18/pa_tests/patest_sync.c b/pd/portaudio_v18/pa_tests/patest_sync.c deleted file mode 100644 index c70fbb35..00000000 --- a/pd/portaudio_v18/pa_tests/patest_sync.c +++ /dev/null @@ -1,227 +0,0 @@ -/* - * $Id: patest_sync.c,v 1.1.1.1 2002/01/22 00:52:40 phil Exp $ - * patest_sync.c - * Test time stamping and synchronization of audio and video. - * A high latency is used so we can hear the difference in time. - * Random durations are used so we know we are hearing the right beep - * and not the one before or after. - * - * Sequence of events: - * Foreground requests a beep. - * Background randomly schedules a beep. - * Foreground waits for the beep to be heard based on Pa_StreamTime(). - * Foreground outputs video (printf) in sync with audio. - * Repeat. - * - * Author: Phil Burk http://www.softsynth.com - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.portaudio.com - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <math.h> -#include "portaudio.h" -#define NUM_BEEPS (6) -#define SAMPLE_RATE (44100) -#define FRAMES_PER_BUFFER (256) -#define BEEP_DURATION (1000) -#define LATENCY_MSEC (2000) -#define SLEEP_MSEC (10) -#define TIMEOUT_MSEC ((3 * LATENCY_MSEC) / (2 * SLEEP_MSEC)) -#define NUM_BUFFERS ((LATENCY_MSEC * SAMPLE_RATE) / (FRAMES_PER_BUFFER * 1000)) -#define STATE_BKG_IDLE (0) -#define STATE_BKG_PENDING (1) -#define STATE_BKG_BEEPING (2) -typedef struct -{ - float left_phase; - float right_phase; - int state; - int requestBeep; /* Set by foreground, cleared by background. */ - PaTimestamp beepTime; - int beepCount; -} -paTestData; -static unsigned long GenerateRandomNumber( void ); -/************************************************************/ -/* Calculate pseudo-random 32 bit number based on linear congruential method. */ -static unsigned long GenerateRandomNumber( void ) -{ - static unsigned long randSeed = 22222; /* Change this for different random sequences. */ - randSeed = (randSeed * 196314165) + 907633515; - return randSeed; -} -/* This routine will be called by the PortAudio engine when audio is needed. -** It may called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int patestCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - /* Cast data passed through stream to our structure. */ - paTestData *data = (paTestData*)userData; - float *out = (float*)outputBuffer; - unsigned int i; - (void) inputBuffer; - - for( i=0; i<framesPerBuffer; i++ ) - { - switch( data->state ) - { - case STATE_BKG_IDLE: - /* Schedule beep at some random time in the future. */ - if( data->requestBeep ) - { - int random = GenerateRandomNumber() >> 14; - data->beepTime = outTime + (i + random + (SAMPLE_RATE/4)); - data->state = STATE_BKG_PENDING; - data->requestBeep = 0; - data->left_phase = data->right_phase = 0.0; - } - *out++ = 0.0; /* left */ - *out++ = 0.0; /* right */ - break; - case STATE_BKG_PENDING: - if( (outTime + i) >= data->beepTime ) - { - data->state = STATE_BKG_BEEPING; - data->beepCount = BEEP_DURATION; - } - *out++ = 0.0; /* left */ - *out++ = 0.0; /* right */ - break; - case STATE_BKG_BEEPING: - if( data->beepCount <= 0 ) - { - data->state = STATE_BKG_IDLE; - *out++ = 0.0; /* left */ - *out++ = 0.0; /* right */ - } - else - { - /* Play sawtooth wave. */ - *out++ = data->left_phase; /* left */ - *out++ = data->right_phase; /* right */ - /* Generate simple sawtooth phaser that ranges between -1.0 and 1.0. */ - data->left_phase += 0.01f; - /* When signal reaches top, drop back down. */ - if( data->left_phase >= 1.0f ) data->left_phase -= 2.0f; - /* higher pitch so we can distinguish left and right. */ - data->right_phase += 0.03f; - if( data->right_phase >= 1.0f ) data->right_phase -= 2.0f; - } - data->beepCount -= 1; - break; - default: - data->state = STATE_BKG_IDLE; - break; - } - } - return 0; -} -/*******************************************************************/ -int main(void); -int main(void) -{ - PortAudioStream *stream; - PaError err; - paTestData DATA; - int i, timeout; - PaTimestamp previousTime; - printf("PortAudio Test: you should see BEEP at the same time you hear it.\n"); - printf("Wait for a few seconds random delay between BEEPs.\n"); - printf("BEEP %d times.\n", NUM_BEEPS ); - /* Initialize our DATA for use by callback. */ - DATA.left_phase = DATA.right_phase = 0.0; - DATA.state = STATE_BKG_IDLE; - DATA.requestBeep = 0; - /* Initialize library before making any other calls. */ - err = Pa_Initialize(); - if( err != paNoError ) goto error; - /* Open an audio I/O stream. */ - err = Pa_OpenDefaultStream( - &stream, - 0, /* no input channels */ - 2, /* stereo output */ - paFloat32, /* 32 bit floating point output */ - SAMPLE_RATE, - FRAMES_PER_BUFFER, - NUM_BUFFERS, - patestCallback, - &DATA ); - if( err != paNoError ) goto error; - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - previousTime = Pa_StreamTime( stream ); - for( i=0; i<NUM_BEEPS; i++ ) - { - /* Request a beep from background. */ - DATA.requestBeep = 1; - /* Wait for background to acknowledge request. */ - timeout = TIMEOUT_MSEC; - while( (DATA.requestBeep == 1) && (timeout-- > 0 ) ) Pa_Sleep(SLEEP_MSEC); - if( timeout <= 0 ) - { - fprintf( stderr, "Timed out waiting for background to acknowledge request.\n" ); - goto error; - } - /* Wait for scheduled beep time. */ - timeout = TIMEOUT_MSEC + (10000/SLEEP_MSEC); - while( (Pa_StreamTime( stream ) < DATA.beepTime) && (timeout-- > 0 ) ) - { - Pa_Sleep(SLEEP_MSEC); - } - if( timeout <= 0 ) - { - fprintf( stderr, "Timed out waiting for time. Now = %g, Beep for %g.\n", - Pa_StreamTime( stream ), DATA.beepTime ); - goto error; - } - /* Beep should be sounding now so print synchronized BEEP. */ - printf("BEEP"); - fflush(stdout); - printf(" at %d, delta = %d\n", - (long) DATA.beepTime, (long) (DATA.beepTime - previousTime) ); - fflush(stdout); - previousTime = DATA.beepTime; - } - err = Pa_StopStream( stream ); - if( err != paNoError ) goto error; - err = Pa_CloseStream( stream ); - if( err != paNoError ) goto error; - Pa_Terminate(); - printf("Test finished.\n"); - return err; -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return err; -} diff --git a/pd/portaudio_v18/pa_tests/patest_toomanysines.c b/pd/portaudio_v18/pa_tests/patest_toomanysines.c deleted file mode 100644 index 53ca8950..00000000 --- a/pd/portaudio_v18/pa_tests/patest_toomanysines.c +++ /dev/null @@ -1,175 +0,0 @@ -/* - * $Id: patest_toomanysines.c,v 1.2.4.1 2003/02/11 21:41:32 philburk Exp $ - * Play more sine waves than we can handle in real time as a stress test, - * - * Authors: - * Ross Bencina <rossb@audiomulch.com> - * Phil Burk <philburk@softsynth.com> - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.audiomulch.com/portaudio/ - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <math.h> -#include "portaudio.h" - -#define MAX_SINES (500) -#define MAX_LOAD (1.2) -#define SAMPLE_RATE (44100) -#define FRAMES_PER_BUFFER (512) -#ifndef M_PI -#define M_PI (3.14159265) -#endif -#define TWOPI (M_PI * 2.0) - -typedef struct paTestData -{ - int numSines; - double phases[MAX_SINES]; -} -paTestData; - -/* This routine will be called by the PortAudio engine when audio is needed. -** It may called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int patestCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData*)userData; - float *out = (float*)outputBuffer; - unsigned long i; - int j; - int finished = 0; - (void) outTime; /* Prevent unused variable warnings. */ - (void) inputBuffer; - - for( i=0; i<framesPerBuffer; i++ ) - { - float output = 0.0; - double phaseInc = 0.02; - double phase; - for( j=0; j<data->numSines; j++ ) - { - /* Advance phase of next oscillator. */ - phase = data->phases[j]; - phase += phaseInc; - if( phase > TWOPI ) phase -= TWOPI; - - phaseInc *= 1.02; - if( phaseInc > 0.5 ) phaseInc *= 0.5; - - /* This is not a very efficient way to calc sines. */ - output += (float) sin( phase ); - data->phases[j] = phase; - } - - - *out++ = (float) (output / data->numSines); - } - return finished; -} - -/*******************************************************************/ -int main(void); -int main(void) -{ - PortAudioStream *stream; - PaError err; - int numStress; - paTestData data = {0}; - double load; - printf("PortAudio Test: output sine wave. SR = %d, BufSize = %d. MAX_LOAD = %f\n", - SAMPLE_RATE, FRAMES_PER_BUFFER, MAX_LOAD ); - - err = Pa_Initialize(); - if( err != paNoError ) goto error; - err = Pa_OpenStream( - &stream, - paNoDevice,/* default input device */ - 0, /* no input */ - paFloat32, /* 32 bit floating point input */ - NULL, - Pa_GetDefaultOutputDeviceID(), /* default output device */ - 1, /* mono output */ - paFloat32, /* 32 bit floating point output */ - NULL, - SAMPLE_RATE, - FRAMES_PER_BUFFER, /* frames per buffer */ - 0, /* number of buffers, if zero then use default minimum */ - paClipOff, /* we won't output out of range samples so don't bother clipping them */ - patestCallback, - &data ); - if( err != paNoError ) goto error; - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - - /* Determine number of sines required to get to 50% */ - do - { - data.numSines++; - Pa_Sleep( 100 ); - - load = Pa_GetCPULoad( stream ); - printf("numSines = %d, CPU load = %f\n", data.numSines, load ); - fflush(stdout); - } - while( load < 0.5 ); - - /* Calculate target stress value then ramp up to that level*/ - numStress = (int) (2.0 * data.numSines * MAX_LOAD ); - for( ; data.numSines < numStress; data.numSines++ ) - { - Pa_Sleep( 200 ); - load = Pa_GetCPULoad( stream ); - printf("STRESSING: numSines = %d, CPU load = %f\n", data.numSines, load ); - fflush(stdout); - - } - - printf("Suffer for 5 seconds.\n"); - Pa_Sleep( 5000 ); - - printf("Stop stream.\n"); - err = Pa_StopStream( stream ); - if( err != paNoError ) goto error; - - err = Pa_CloseStream( stream ); - if( err != paNoError ) goto error; - - Pa_Terminate(); - printf("Test finished.\n"); - return err; -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return err; -} diff --git a/pd/portaudio_v18/pa_tests/patest_two_rates.c b/pd/portaudio_v18/pa_tests/patest_two_rates.c deleted file mode 100644 index 350ce043..00000000 --- a/pd/portaudio_v18/pa_tests/patest_two_rates.c +++ /dev/null @@ -1,168 +0,0 @@ -/* - * $Id: patest_two_rates.c,v 1.1.2.1 2003/02/11 21:42:24 philburk Exp $ - * patest_two_rates.c - * Play two streams at different rates to make sure they don't interfere. - * - * Author: Phil Burk http://www.softsynth.com - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.portaudio.com - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <math.h> -#include "portaudio.h" - -#define OUTPUT_DEVICE (Pa_GetDefaultOutputDeviceID()) -#define SAMPLE_RATE_1 (44100) -#define SAMPLE_RATE_2 (44100) -#define FRAMES_PER_BUFFER (256) -#define FREQ_INCR (0.1) - -#ifndef M_PI -#define M_PI (3.14159265) -#endif - -typedef struct -{ - double phase; - double numFrames; -} paTestData; - -/* This routine will be called by the PortAudio engine when audio is needed. -** It may called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int patestCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData*)userData; - float *out = (float*)outputBuffer; - int frameIndex, channelIndex; - int finished = 0; - (void) outTime; /* Prevent unused variable warnings. */ - (void) inputBuffer; - - for( frameIndex=0; frameIndex<(int)framesPerBuffer; frameIndex++ ) - { - /* Generate sine wave. */ - float value = (float) 0.3 * sin(data->phase); - /* Stereo - two channels. */ - *out++ = value; - *out++ = value; - - data->phase += FREQ_INCR; - if( data->phase >= (2.0 * M_PI) ) data->phase -= (2.0 * M_PI); - } - - return 0; -} - -/*******************************************************************/ -int main(void); -int main(void) -{ - PaError err; - PortAudioStream *stream1; - PortAudioStream *stream2; - paTestData data1 = {0}; - paTestData data2 = {0}; - printf("PortAudio Test: two rates.\n" ); - - err = Pa_Initialize(); - if( err != paNoError ) goto error; - - /* Start first stream. **********************/ - err = Pa_OpenStream( - &stream1, - paNoDevice, /* default input device */ - 0, /* no input */ - paFloat32, /* 32 bit floating point input */ - NULL, - OUTPUT_DEVICE, - 2, /* Stereo */ - paFloat32, /* 32 bit floating point output */ - NULL, - SAMPLE_RATE_1, - FRAMES_PER_BUFFER, /* frames per buffer */ - 0, /* number of buffers, if zero then use default minimum */ - paClipOff, /* we won't output out of range samples so don't bother clipping them */ - patestCallback, - &data1 ); - if( err != paNoError ) goto error; - - err = Pa_StartStream( stream1 ); - if( err != paNoError ) goto error; - - Pa_Sleep( 3 * 1000 ); - - /* Start second stream. **********************/ - err = Pa_OpenStream( - &stream2, - paNoDevice, /* default input device */ - 0, /* no input */ - paFloat32, /* 32 bit floating point input */ - NULL, - OUTPUT_DEVICE, - 2, /* Stereo */ - paFloat32, /* 32 bit floating point output */ - NULL, - SAMPLE_RATE_2, - FRAMES_PER_BUFFER, /* frames per buffer */ - 0, /* number of buffers, if zero then use default minimum */ - paClipOff, /* we won't output out of range samples so don't bother clipping them */ - patestCallback, - &data2 ); - if( err != paNoError ) goto error; - - err = Pa_StartStream( stream2 ); - if( err != paNoError ) goto error; - - Pa_Sleep( 3 * 1000 ); - - err = Pa_StopStream( stream2 ); - if( err != paNoError ) goto error; - - Pa_Sleep( 3 * 1000 ); - - err = Pa_StopStream( stream1 ); - if( err != paNoError ) goto error; - - Pa_CloseStream( stream2 ); - Pa_CloseStream( stream1 ); - - Pa_Terminate(); - printf("Test finished.\n"); - return err; -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return err; -} diff --git a/pd/portaudio_v18/pa_tests/patest_underflow.c b/pd/portaudio_v18/pa_tests/patest_underflow.c deleted file mode 100644 index 1547b2e4..00000000 --- a/pd/portaudio_v18/pa_tests/patest_underflow.c +++ /dev/null @@ -1,151 +0,0 @@ -/* - * $Id: patest_underflow.c,v 1.1.1.1 2002/01/22 00:52:40 phil Exp $ - * patest_underflow.c - * Simulate an output buffer underflow condition. - * Tests whether the stream can be stopped when underflowing buffers. - * - * Authors: - * Ross Bencina <rossb@audiomulch.com> - * Phil Burk <philburk@softsynth.com> - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.audiomulch.com/portaudio/ - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <math.h> -#include "portaudio.h" - -#define NUM_SECONDS (20) -#define SAMPLE_RATE (44100) -#define FRAMES_PER_BUFFER (2048) -#define MSEC_PER_BUFFER ( (FRAMES_PER_BUFFER * 1000) / SAMPLE_RATE ) - -#ifndef M_PI -#define M_PI (3.14159265) -#endif - -#define TABLE_SIZE (200) -typedef struct -{ - float sine[TABLE_SIZE]; - int left_phase; - int right_phase; - int sleepTime; -} -paTestData; - -/* This routine will be called by the PortAudio engine when audio is needed. -** It may called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int patestCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - paTestData *data = (paTestData*)userData; - float *out = (float*)outputBuffer; - unsigned long i; - int finished = 0; - (void) outTime; /* Prevent unused variable warnings. */ - (void) inputBuffer; - for( i=0; i<framesPerBuffer; i++ ) - { - *out++ = data->sine[data->left_phase]; /* left */ - *out++ = data->sine[data->right_phase]; /* right */ - data->left_phase += 1; - if( data->left_phase >= TABLE_SIZE ) data->left_phase -= TABLE_SIZE; - data->right_phase += 3; /* higher pitch so we can distinguish left and right. */ - if( data->right_phase >= TABLE_SIZE ) data->right_phase -= TABLE_SIZE; - } - - /* Cause underflow to occur. */ - if( data->sleepTime > 0 ) Pa_Sleep( data->sleepTime ); - data->sleepTime += 1; - - return finished; -} - -/*******************************************************************/ -int main(void); -int main(void) -{ - PortAudioStream *stream; - PaError err; - paTestData data; - int i; - printf("PortAudio Test: output sine wave. SR = %d, BufSize = %d\n", SAMPLE_RATE, FRAMES_PER_BUFFER); - /* initialise sinusoidal wavetable */ - for( i=0; i<TABLE_SIZE; i++ ) - { - data.sine[i] = (float) sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. ); - } - data.left_phase = data.right_phase = data.sleepTime = 0; - err = Pa_Initialize(); - if( err != paNoError ) goto error; - err = Pa_OpenStream( - &stream, - paNoDevice,/* default input device */ - 0, /* no input */ - paFloat32, /* 32 bit floating point input */ - NULL, - Pa_GetDefaultOutputDeviceID(), /* default output device */ - 2, /* stereo output */ - paFloat32, /* 32 bit floating point output */ - NULL, - SAMPLE_RATE, - FRAMES_PER_BUFFER, - 0, /* number of buffers, if zero then use default minimum */ - paClipOff, /* we won't output out of range samples so don't bother clipping them */ - patestCallback, - &data ); - if( err != paNoError ) goto error; - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - - while( data.sleepTime < (2 * MSEC_PER_BUFFER) ) - { - printf("SleepTime = %d\n", data.sleepTime ); - Pa_Sleep( data.sleepTime ); - } - - printf("Try to stop stream.\n"); - err = Pa_StopStream( stream ); /* */ - err = Pa_AbortStream( stream ); /* */ - if( err != paNoError ) goto error; - err = Pa_CloseStream( stream ); - if( err != paNoError ) goto error; - Pa_Terminate(); - printf("Test finished.\n"); - return err; -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return err; -} diff --git a/pd/portaudio_v18/pa_tests/patest_wire.c b/pd/portaudio_v18/pa_tests/patest_wire.c deleted file mode 100644 index 7ea64bb4..00000000 --- a/pd/portaudio_v18/pa_tests/patest_wire.c +++ /dev/null @@ -1,176 +0,0 @@ -/* - * $Id: patest_wire.c,v 1.2.4.3 2003/04/10 23:09:40 philburk Exp $ - * patest_wire.c - * - * Pass input directly to output. - * Note that some HW devices, for example many ISA audio cards - * on PCs, do NOT support full duplex! For a PC, you normally need - * a PCI based audio card such as the SBLive. - * - * Author: Phil Burk http://www.softsynth.com - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.portaudio.com - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <math.h> -#include "portaudio.h" - -#define INPUT_DEVICE Pa_GetDefaultInputDeviceID() -#define OUTPUT_DEVICE Pa_GetDefaultOutputDeviceID() - -/* -** Note that many of the older ISA sound cards on PCs do NOT support -** full duplex audio (simultaneous record and playback). -** And some only support full duplex at lower sample rates. -*/ -#define SAMPLE_RATE (44100) -#define FRAMES_PER_BUFFER (64) - -#if 1 -#define PA_SAMPLE_TYPE paFloat32 -typedef float SAMPLE; -#else -#define PA_SAMPLE_TYPE paInt16 -typedef short SAMPLE; -#endif -static int wireCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ); - -/* This routine will be called by the PortAudio engine when audio is needed. -** It may be called at interrupt level on some machines so don't do anything -** that could mess up the system like calling malloc() or free(). -*/ -static int wireCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - SAMPLE *out = (SAMPLE*)outputBuffer; - SAMPLE *in = (SAMPLE*)inputBuffer; - unsigned int i; - (void) outTime; - int samplesPerFrame; - int numSamples; - - samplesPerFrame = (int) userData; - numSamples = framesPerBuffer * samplesPerFrame; - - /* This may get called with NULL inputBuffer during initial setup. */ - if( inputBuffer == NULL ) - { - for( i=0; i<numSamples; i++ ) - { - *out++ = 0; - } - } - else - { - for( i=0; i<numSamples; i++ ) - { - *out++ = *in++; - } - } - - return 0; -} - - -/*******************************************************************/ -int main(void); -int main(void) -{ - PortAudioStream *stream; - PaError err; - const PaDeviceInfo *inputInfo; - const PaDeviceInfo *outputInfo; - int numChannels; - - err = Pa_Initialize(); - if( err != paNoError ) goto error; - - printf("PortAudio Test: input device ID = %d\n", INPUT_DEVICE ); - printf("PortAudio Test: output device ID = %d\n", OUTPUT_DEVICE ); - - /* Use as many channels aspossible. */ - inputInfo = Pa_GetDeviceInfo( INPUT_DEVICE ); - outputInfo = Pa_GetDeviceInfo( OUTPUT_DEVICE ); - /* Use smaller count. */ - numChannels = (inputInfo->maxInputChannels < outputInfo->maxOutputChannels) ? - inputInfo->maxInputChannels : outputInfo->maxOutputChannels; - - printf("maxInputChannels channels = %d\n", inputInfo->maxInputChannels ); - printf("maxOutputChannels channels = %d\n", outputInfo->maxOutputChannels ); - if( numChannels > 0 ) - { - printf("Using %d channels.\n", numChannels ); - - err = Pa_OpenStream( - &stream, - INPUT_DEVICE, - numChannels, - PA_SAMPLE_TYPE, - NULL, - OUTPUT_DEVICE, - numChannels, - PA_SAMPLE_TYPE, - NULL, - SAMPLE_RATE, - FRAMES_PER_BUFFER, /* frames per buffer */ - 0, /* number of buffers, if zero then use default minimum */ - paClipOff, /* we won't output out of range samples so don't bother clipping them */ - wireCallback, - (void *) numChannels ); /* pass numChannels to callback */ - if( err != paNoError ) goto error; - - err = Pa_StartStream( stream ); - if( err != paNoError ) goto error; - - printf("Full duplex sound test in progress.\n"); - printf("Hit ENTER to exit test.\n"); fflush(stdout); - getchar(); - - printf("Closing stream.\n"); - err = Pa_CloseStream( stream ); - if( err != paNoError ) goto error; - } - else - { - printf("Sorry, not enough channels.\n"); - } - Pa_Terminate(); - - printf("Full duplex sound test complete.\n"); fflush(stdout); - return 0; -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return -1; -} diff --git a/pd/portaudio_v18/pa_unix_oss/Makefile b/pd/portaudio_v18/pa_unix_oss/Makefile deleted file mode 100644 index 3603816a..00000000 --- a/pd/portaudio_v18/pa_unix_oss/Makefile +++ /dev/null @@ -1,32 +0,0 @@ -# Make PortAudio for Linux -# -# by Phil Burk -# -# To compile a test program, make a target with a .app suffix. -# For example to compile "../pa_tests/pa_devs.c", enter: -# gmake pa_devs.app -# -# To compile and execute a test program, make a target with a .run suffix. -# For example to build and run "../pa_tests/pa_devs.c", enter: -# gmake pa_devs.run - -VPATH = ../pa_common ../pa_tests -CFLAGS = -g -Wall -I../pa_common -CC = gcc - -PAOBJS = pa_lib.o pa_unix_oss.o pa_unix.o -PAINC = portaudio.h pa_unix.h -PALIBS = -lm -lpthread - -all: patest_sine.run - -%.app: %.o $(PAOBJS) $(PAINC) Makefile - gcc $(CFLAGS) $*.o $(PAOBJS) $(PALIBS) -o $@ - -%.run: %.app - ./$*.app - -clean: - -rm *.app - -rm *.o - diff --git a/pd/portaudio_v18/pa_unix_oss/Makefile_freebsd b/pd/portaudio_v18/pa_unix_oss/Makefile_freebsd deleted file mode 100644 index fc8b14db..00000000 --- a/pd/portaudio_v18/pa_unix_oss/Makefile_freebsd +++ /dev/null @@ -1,36 +0,0 @@ -# Make PortAudio for FreeBSD - -LIBS = -lm -pthread - -CDEFINES = -I../pa_common -CFLAGS = -g -PASRC = ../pa_common/pa_lib.c pa_freebsd.c -PAINC = ../pa_common/portaudio.h - -# Tests that work. -#TESTC = $(PASRC) ../pa_tests/patest_sine.c -TESTC = $(PASRC) ../pa_tests/patest_sine_time.c -#TESTC = $(PASRC) ../pa_tests/patest_stop.c -#TESTC = $(PASRC) ../pa_tests/patest_sync.c -#TESTC = $(PASRC) ../pa_tests/patest_pink.c -#TESTC = $(PASRC) ../pa_tests/patest_leftright.c -#TESTC = $(PASRC) ../pa_tests/patest_clip.c -#TESTC = $(PASRC) ../pa_tests/patest_dither.c -#TESTC = $(PASRC) ../pa_tests/pa_devs.c -#TESTC = $(PASRC) ../pa_tests/patest_many.c -#TESTC = $(PASRC) ../pa_tests/patest_record.c -#TESTC = $(PASRC) ../pa_tests/patest_wire.c -#TESTC = $(PASRC) ../pa_tests/paqa_devs.c - -# Tests that do not yet work. - -TESTH = $(PAINC) - -all: patest - -patest: $(TESTC) $(TESTH) Makefile - gcc $(CFLAGS) $(TESTC) $(CDEFINES) $(LIBS) -o patest - -run: patest - ./patest - diff --git a/pd/portaudio_v18/pa_unix_oss/low_latency_tip.txt b/pd/portaudio_v18/pa_unix_oss/low_latency_tip.txt Binary files differdeleted file mode 100644 index 2d982b79..00000000 --- a/pd/portaudio_v18/pa_unix_oss/low_latency_tip.txt +++ /dev/null diff --git a/pd/portaudio_v18/pa_unix_oss/pa_unix.c b/pd/portaudio_v18/pa_unix_oss/pa_unix.c deleted file mode 100644 index 8d652b73..00000000 --- a/pd/portaudio_v18/pa_unix_oss/pa_unix.c +++ /dev/null @@ -1,1122 +0,0 @@ -/* - * PortAudio Portable Real-Time Audio Library - * Latest Version at: http://www.portaudio.com - * Linux OSS Implementation by douglas repetto and Phil Burk - * - * Copyright (c) 1999-2000 Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ - -/* -Modification History - 1/2001 - Phil Burk - initial hack for Linux - 2/2001 - Douglas Repetto - many improvements, initial query support - 4/2/2001 - Phil - stop/abort thread control, separate in/out native buffers - 5/28/2001 - Phil - use pthread_create() instead of clone(). Thanks Stephen Brandon! - use pthread_join() after thread shutdown. - 5/29/2001 - Phil - query for multiple devices, multiple formats, - input mode and input+output mode working, - Pa_GetCPULoad() implemented. - PLB20010817 - Phil & Janos Haber - don't halt if test of sample rate fails. - SB20010904 - Stephen Brandon - mods needed for GNUSTEP and SndKit - JH20010905 - Janos Haber - FreeBSD mods - 2001-09-22 - Heiko - (i.e. Heiko Purnhagen <purnhage@tnt.uni-hannover.de> ;-) - added 24k and 16k to ratesToTry[] - fixed Pa_GetInternalDevice() - changed DEVICE_NAME_BASE from /dev/audio to /dev/dsp - handled SNDCTL_DSP_SPEED in Pq_QueryDevice() more graceful - fixed Pa_StreamTime() for paqa_errs.c - fixed numCannel=2 oddity and error handling in Pa_SetupDeviceFormat() - grep also for HP20010922 ... - PLB20010924 - Phil - merged Heiko's changes - removed sNumDevices and potential related bugs, - use getenv("PA_MIN_LATENCY_MSEC") to set desired latency, - simplify CPU Load calculation by comparing real-time to framesPerBuffer, - always close device when querying even if error occurs, - PLB20010927 - Phil - Improved negotiation for numChannels. - SG20011005 - Stewart Greenhill - set numChannels back to reasonable value after query. - DH20010115 - David Herring - fixed uninitialized handle. - - DM20020218 - Dominic Mazzoni - Try to open in nonblocking mode first, in case - the device is already open. New implementation of - Pa_StreamTime that uses SNDCTL_DSP_GETOPTR but - uses our own counter to avoid wraparound. - PLB20020222 - Phil Burk - Added WatchDog proc if audio running at high priority. - Check error return from read() and write(). - Check CPU endianness instead of assuming Little Endian. - 20020621 - pa_unix_oss.c split into pa_unix.c, pa_unix.h, pa_unix_oss.c by - Augustus Saunders. Return values from usleep() ignored by Sam Bayer - because not cross-platform compatible (at least until we get configure - going). Pa_SetupDeviceFormat split into input and output sides to - reflect capabilities of Solaris. - - 20030206 - Martin Rohrbach - various mods for Solaris - - 20030410 - Bjorn Dittmer-Roche - fixed numerous problems associated with pthread_t - - 20030630 - Thomas Richter - eliminated unused variable warnings. - -TODO -O- put semaphore lock around shared data? -O- handle native formats better -O- handle stereo-only device better ??? -O- what if input and output of a device capabilities differ (e.g. es1371) ??? -*/ - - -#include "pa_unix.h" - -typedef void *(*pthread_function_t)(void *); - -/************************************************* Shared Data ********/ -/* FIXME - put Mutex around this shared data. */ -static internalPortAudioDevice *sDeviceList = NULL; -static int sDefaultInputDeviceID = paNoDevice; -static int sDefaultOutputDeviceID = paNoDevice; -static int sPaHostError = 0; - -/********************************* BEGIN CPU UTILIZATION MEASUREMENT ****/ -static void Pa_StartUsageCalculation( internalPortAudioStream *past ) -{ - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - if( pahsc == NULL ) return; - /* Query system timer for usage analysis and to prevent overuse of CPU. */ - gettimeofday( &pahsc->pahsc_EntryTime, NULL ); -} - -static long SubtractTime_AminusB( struct timeval *timeA, struct timeval *timeB ) -{ - long secs = timeA->tv_sec - timeB->tv_sec; - long usecs = secs * 1000000; - usecs += (timeA->tv_usec - timeB->tv_usec); - return usecs; -} - -/****************************************************************************** -** Measure fractional CPU load based on real-time it took to calculate -** buffers worth of output. -*/ -static void Pa_EndUsageCalculation( internalPortAudioStream *past ) -{ - struct timeval currentTime; - long usecsElapsed; - double newUsage; - -#define LOWPASS_COEFFICIENT_0 (0.95) -#define LOWPASS_COEFFICIENT_1 (0.99999 - LOWPASS_COEFFICIENT_0) - - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - if( pahsc == NULL ) return; - - if( gettimeofday( ¤tTime, NULL ) == 0 ) - { - usecsElapsed = SubtractTime_AminusB( ¤tTime, &pahsc->pahsc_EntryTime ); - /* Use inverse because it is faster than the divide. */ - newUsage = usecsElapsed * pahsc->pahsc_InverseMicrosPerBuffer; - - past->past_Usage = (LOWPASS_COEFFICIENT_0 * past->past_Usage) + - (LOWPASS_COEFFICIENT_1 * newUsage); - - } -} -/****************************************** END CPU UTILIZATION *******/ - -/********************************************************************* - * Determines the number of available devices by trying to open - * each "/dev/dsp#" or "/dsp/audio#" in order until it fails. - * Add each working device to a singly linked list of devices. - */ -PaError Pa_QueryDevices( void ) -{ - internalPortAudioDevice *pad, *lastPad; - int go = 1; - int numDevices = 0; - PaError testResult; - PaError result = paNoError; - char *envdev; - - sDefaultInputDeviceID = paNoDevice; - sDefaultOutputDeviceID = paNoDevice; - - lastPad = NULL; - - while( go ) - { - /* Allocate structure to hold device info. */ - pad = (internalPortAudioDevice *) - PaHost_AllocateFastMemory( sizeof(internalPortAudioDevice) ); - if( pad == NULL ) return paInsufficientMemory; - memset( pad, 0, sizeof(internalPortAudioDevice) ); - - /* Build name for device. */ - if( numDevices == 0 ) - { - sprintf( pad->pad_DeviceName, DEVICE_NAME_BASE); - } - else - { - sprintf( pad->pad_DeviceName, DEVICE_NAME_BASE "%d", numDevices ); - } - - DBUG(("Try device %s\n", pad->pad_DeviceName )); - testResult = Pa_QueryDevice( pad->pad_DeviceName, pad ); - DBUG(("Pa_QueryDevice returned %d\n", testResult )); - if( testResult != paNoError ) - { - if( lastPad == NULL ) - { - result = testResult; /* No good devices! */ - } - go = 0; - PaHost_FreeFastMemory( pad, sizeof(internalPortAudioDevice) ); - } - else - { - numDevices += 1; - /* Add to linked list of devices. */ - if( lastPad ) - { - lastPad->pad_Next = pad; - } - else - { - sDeviceList = pad; /* First element in linked list. */ - } - lastPad = pad; - } - } - - /* I'm sitting at a SunRay1 and I neither have /dev/audio# nor /dev/dsp#. - Instead, the correct audio device is stored in the environment variable - AUDIODEV and/or UTAUDIODEV, so check these devices as well if we haven't - checked them yet above - MR */ - - DBUG(("Checking for AUDIODEV and UTAUDIODEV\n")); - envdev = getenv("AUDIODEV"); - if (envdev != NULL && !strstr(envdev, DEVICE_NAME_BASE)) { - result = paNoError; - - /* Allocate structure to hold device info. */ - pad = (internalPortAudioDevice *) - PaHost_AllocateFastMemory( sizeof(internalPortAudioDevice) ); - if( pad == NULL ) return paInsufficientMemory; - memset( pad, 0, sizeof(internalPortAudioDevice) ); - - /* Build name for device. */ - strcpy(pad->pad_DeviceName, envdev); - - DBUG(("Try device %s\n", pad->pad_DeviceName )); - testResult = Pa_QueryDevice( pad->pad_DeviceName, pad ); - DBUG(("Pa_QueryDevice returned %d\n", testResult )); - if( testResult != paNoError ) - { - if( lastPad == NULL ) - { - result = testResult; /* No good devices! */ - } - PaHost_FreeFastMemory( pad, sizeof(internalPortAudioDevice) ); - } - else - { - numDevices += 1; - /* Add to linked list of devices. */ - if( lastPad ) - { - lastPad->pad_Next = pad; - } - else - { - sDeviceList = pad; /* First element in linked list. */ - } - lastPad = pad; - } - } - - envdev = getenv("UTAUDIODEV"); - if (envdev != NULL && !strstr(envdev, DEVICE_NAME_BASE) && getenv("AUDIODEV") != NULL && strcmp(envdev, getenv("AUDIODEV"))) { - result = paNoError; - - /* Allocate structure to hold device info. */ - pad = (internalPortAudioDevice *) - PaHost_AllocateFastMemory( sizeof(internalPortAudioDevice) ); - if( pad == NULL ) return paInsufficientMemory; - memset( pad, 0, sizeof(internalPortAudioDevice) ); - - /* Build name for device. */ - strcpy(pad->pad_DeviceName, envdev); - - DBUG(("Try device %s\n", pad->pad_DeviceName )); - testResult = Pa_QueryDevice( pad->pad_DeviceName, pad ); - DBUG(("Pa_QueryDevice returned %d\n", testResult )); - if( testResult != paNoError ) - { - if( lastPad == NULL ) - { - result = testResult; /* No good devices! */ - } - PaHost_FreeFastMemory( pad, sizeof(internalPortAudioDevice) ); - } - else - { - numDevices += 1; - /* Add to linked list of devices. */ - if( lastPad ) - { - lastPad->pad_Next = pad; - } - else - { - sDeviceList = pad; /* First element in linked list. */ - } - lastPad = pad; - } - } - - return result; -} - -/*************************************************************************/ -int Pa_CountDevices() -{ - int numDevices = 0; - internalPortAudioDevice *pad; - - if( sDeviceList == NULL ) Pa_Initialize(); - /* Count devices in list. */ - pad = sDeviceList; - while( pad != NULL ) - { - pad = pad->pad_Next; - numDevices++; - } - - return numDevices; -} - -/*************************************************************************/ -internalPortAudioDevice *Pa_GetInternalDevice( PaDeviceID id ) -{ - internalPortAudioDevice *pad; - if( (id < 0) || ( id >= Pa_CountDevices()) ) return NULL; - pad = sDeviceList; - while( id > 0 ) - { - pad = pad->pad_Next; - id--; - } - return pad; -} - -/*************************************************************************/ -const PaDeviceInfo* Pa_GetDeviceInfo( PaDeviceID id ) -{ - internalPortAudioDevice *pad; - if( (id < 0) || ( id >= Pa_CountDevices()) ) return NULL; - pad = Pa_GetInternalDevice( id ); - return &pad->pad_Info ; -} - -static PaError Pa_MaybeQueryDevices( void ) -{ - if( sDeviceList == NULL ) - { - return Pa_QueryDevices(); - } - return 0; -} - -PaDeviceID Pa_GetDefaultInputDeviceID( void ) -{ - /* return paNoDevice; */ - return 0; -} - -PaDeviceID Pa_GetDefaultOutputDeviceID( void ) -{ - return 0; -} - -/********************************************************************** -** Make sure that we have queried the device capabilities. -*/ - -PaError PaHost_Init( void ) -{ - return Pa_MaybeQueryDevices(); -} - -/******************************************************************************************* - * The ol' Canary in a Coal Mine trick. - * Just update the time periodically. - * Runs at low priority so if audio thread runs wild, this thread will get starved - * and the watchdog will detect it. - */ - -#define SCHEDULER_POLICY SCHED_RR -#define WATCHDOG_MAX_SECONDS (3) -#define WATCHDOG_INTERVAL_USEC (1000000) - -static int PaHost_CanaryProc( PaHostSoundControl *pahsc ) -{ - int result = 0; - -#ifdef GNUSTEP - GSRegisterCurrentThread(); /* SB20010904 */ -#endif - - while( pahsc->pahsc_CanaryRun) { - usleep( WATCHDOG_INTERVAL_USEC ); - gettimeofday( &pahsc->pahsc_CanaryTime, NULL ); - } - - DBUG(("PaHost_CanaryProc: exiting.\n")); - -#ifdef GNUSTEP - GSUnregisterCurrentThread(); /* SB20010904 */ -#endif - - return result; -} - -/******************************************************************************************* - * Monitor audio thread and lower its it if it hogs the CPU. - * To prevent getting killed, the audio thread must update a - * variable with a timer value. - * If the value is not recent enough, then the - * thread will get killed. - */ - -static PaError PaHost_WatchDogProc( PaHostSoundControl *pahsc ) -{ - struct sched_param schp = { 0 }; - int maxPri; - -#ifdef GNUSTEP - GSRegisterCurrentThread(); /* SB20010904 */ -#endif - -/* Run at a priority level above audio thread so we can still run if it hangs. */ -/* Rise more than 1 because of rumored off-by-one scheduler bugs. */ - schp.sched_priority = pahsc->pahsc_AudioPriority + 4; - maxPri = sched_get_priority_max(SCHEDULER_POLICY); - if( schp.sched_priority > maxPri ) schp.sched_priority = maxPri; - - if (sched_setscheduler(0, SCHEDULER_POLICY, &schp) != 0) - { - ERR_RPT(("PaHost_WatchDogProc: cannot set watch dog priority!\n")); - goto killAudio; - } - - /* Compare watchdog time with audio and canary thread times. */ - /* Sleep for a while or until thread cancelled. */ - while( pahsc->pahsc_WatchDogRun ) - { - - int delta; - struct timeval currentTime; - - usleep( WATCHDOG_INTERVAL_USEC ); - gettimeofday( ¤tTime, NULL ); - - /* If audio thread is not advancing, then it must be hung so kill it. */ - delta = currentTime.tv_sec - pahsc->pahsc_EntryTime.tv_sec; - DBUG(("PaHost_WatchDogProc: audio delta = %d\n", delta )); - if( delta > WATCHDOG_MAX_SECONDS ) - { - goto killAudio; - } - - /* If canary died, then lower audio priority and halt canary. */ - delta = currentTime.tv_sec - pahsc->pahsc_CanaryTime.tv_sec; - if( delta > WATCHDOG_MAX_SECONDS ) - { - ERR_RPT(("PaHost_WatchDogProc: canary died!\n")); - goto lowerAudio; - } - } - - DBUG(("PaHost_WatchDogProc: exiting.\n")); -#ifdef GNUSTEP - GSUnregisterCurrentThread(); /* SB20010904 */ -#endif - return 0; - -lowerAudio: - { - struct sched_param schat = { 0 }; - if( sched_setscheduler(pahsc->pahsc_AudioThreadPID, SCHED_OTHER, &schat) != 0) - { - ERR_RPT(("PaHost_WatchDogProc: failed to lower audio priority. errno = %d\n", errno )); - /* Fall through into killing audio thread. */ - } - else - { - ERR_RPT(("PaHost_WatchDogProc: lowered audio priority to prevent hogging of CPU.\n")); - goto cleanup; - } - } - -killAudio: - ERR_RPT(("PaHost_WatchDogProc: killing hung audio thread!\n")); - pthread_kill( pahsc->pahsc_AudioThread, SIGKILL ); - -cleanup: - pahsc->pahsc_CanaryRun = 0; - DBUG(("PaHost_WatchDogProc: cancel Canary\n")); - pthread_cancel( pahsc->pahsc_CanaryThread ); - DBUG(("PaHost_WatchDogProc: join Canary\n")); - pthread_join( pahsc->pahsc_CanaryThread, NULL ); - DBUG(("PaHost_WatchDogProc: forget Canary\n")); - pahsc->pahsc_IsCanaryThreadValid = 0; - -#ifdef GNUSTEP - GSUnregisterCurrentThread(); /* SB20010904 */ -#endif - return 0; -} - -/*******************************************************************************************/ -static void PaHost_StopWatchDog( PaHostSoundControl *pahsc ) -{ -/* Cancel WatchDog thread if there is one. */ - if( pahsc->pahsc_IsWatchDogThreadValid ) - { - pahsc->pahsc_WatchDogRun = 0; - DBUG(("PaHost_StopWatchDog: cancel WatchDog\n")); - pthread_cancel( pahsc->pahsc_WatchDogThread ); - pthread_join( pahsc->pahsc_WatchDogThread, NULL ); - pahsc->pahsc_IsWatchDogThreadValid = 0; - } -/* Cancel Canary thread if there is one. */ - if( pahsc->pahsc_IsCanaryThreadValid ) - { - pahsc->pahsc_CanaryRun = 0; - DBUG(("PaHost_StopWatchDog: cancel Canary\n")); - pthread_cancel( pahsc->pahsc_CanaryThread ); - DBUG(("PaHost_StopWatchDog: join Canary\n")); - pthread_join( pahsc->pahsc_CanaryThread, NULL ); - pahsc->pahsc_IsCanaryThreadValid = 0; - } -} - -/*******************************************************************************************/ -static PaError PaHost_StartWatchDog( PaHostSoundControl *pahsc ) -{ - int hres; - PaError result = 0; - - /* The watch dog watches for these timer updates */ - gettimeofday( &pahsc->pahsc_EntryTime, NULL ); - gettimeofday( &pahsc->pahsc_CanaryTime, NULL ); - - /* Launch a canary thread to detect priority abuse. */ - pahsc->pahsc_CanaryRun = 1; - hres = pthread_create(&(pahsc->pahsc_CanaryThread), - NULL /*pthread_attr_t * attr*/, - (pthread_function_t)PaHost_CanaryProc, pahsc); - if( hres != 0 ) - { - pahsc->pahsc_IsCanaryThreadValid = 0; - result = paHostError; - sPaHostError = hres; - goto error; - } - pahsc->pahsc_IsCanaryThreadValid = 1; - - /* Launch a watchdog thread to prevent runaway audio thread. */ - pahsc->pahsc_WatchDogRun = 1; - hres = pthread_create(&(pahsc->pahsc_WatchDogThread), - NULL /*pthread_attr_t * attr*/, - (pthread_function_t)PaHost_WatchDogProc, pahsc); - if( hres != 0 ) - { - pahsc->pahsc_IsWatchDogThreadValid = 0; - result = paHostError; - sPaHostError = hres; - goto error; - } - pahsc->pahsc_IsWatchDogThreadValid = 1; - return result; - -error: - PaHost_StopWatchDog( pahsc ); - return result; -} - -/******************************************************************************************* - * Bump priority of audio thread if running with superuser priveledges. - * if priority bumped then launch a watchdog. - */ -static PaError PaHost_BoostPriority( internalPortAudioStream *past ) -{ - PaHostSoundControl *pahsc; - PaError result = paNoError; - struct sched_param schp = { 0 }; - - pahsc = (PaHostSoundControl *) past->past_DeviceData; - if( pahsc == NULL ) return paInternalError; - - pahsc->pahsc_AudioThreadPID = getpid(); - DBUG(("PaHost_BoostPriority: audio PID = %d\n", pahsc->pahsc_AudioThreadPID )); - - /* Choose a priority in the middle of the range. */ - pahsc->pahsc_AudioPriority = (sched_get_priority_max(SCHEDULER_POLICY) - - sched_get_priority_min(SCHEDULER_POLICY)) / 2; - schp.sched_priority = pahsc->pahsc_AudioPriority; - - if (sched_setscheduler(0, SCHEDULER_POLICY, &schp) != 0) - { - DBUG(("PortAudio: only superuser can use real-time priority.\n")); - } - else - { - DBUG(("PortAudio: audio callback priority set to level %d!\n", schp.sched_priority)); - /* We are running at high priority so we should have a watchdog in case audio goes wild. */ - result = PaHost_StartWatchDog( pahsc ); - } - - return result; -} - -/*******************************************************************************************/ -static PaError Pa_AudioThreadProc( internalPortAudioStream *past ) -{ - PaError result; - PaHostSoundControl *pahsc; - ssize_t bytes_read, bytes_written; - - pahsc = (PaHostSoundControl *) past->past_DeviceData; - if( pahsc == NULL ) return paInternalError; - -#ifdef GNUSTEP - GSRegisterCurrentThread(); /* SB20010904 */ -#endif - - result = PaHost_BoostPriority( past ); - if( result < 0 ) goto error; - - past->past_IsActive = 1; - DBUG(("entering thread.\n")); - - while( (past->past_StopNow == 0) && (past->past_StopSoon == 0) ) - { - /* Read data from device */ - if(pahsc->pahsc_NativeInputBuffer) - { - unsigned int totalread = 0; - DBUG(("Pa_AudioThreadProc: attempt to read %d bytes\n", pahsc->pahsc_BytesPerInputBuffer)); - do - { - bytes_read = read(pahsc->pahsc_InputHandle, - (char *)pahsc->pahsc_NativeInputBuffer + totalread, - pahsc->pahsc_BytesPerInputBuffer - totalread); - - if (bytes_read < 0) - { - ERR_RPT(("PortAudio: read interrupted!\n")); - break; - } - - totalread += bytes_read; - } while( totalread < pahsc->pahsc_BytesPerInputBuffer); - } - - /* Convert 16 bit native data to user data and call user routine. */ - DBUG(("converting...\n")); - Pa_StartUsageCalculation( past ); - result = Pa_CallConvertInt16( past, - pahsc->pahsc_NativeInputBuffer, - pahsc->pahsc_NativeOutputBuffer ); - Pa_EndUsageCalculation( past ); - if( result != 0) - { - DBUG(("hmm, Pa_CallConvertInt16() says: %d. i'm bailing.\n", - result)); - break; - } - - /* Write data to device. */ - if( pahsc->pahsc_NativeOutputBuffer ) - { - unsigned int totalwritten = 0; - do - { - bytes_written = write(pahsc->pahsc_OutputHandle, - (void *)pahsc->pahsc_NativeOutputBuffer, - pahsc->pahsc_BytesPerOutputBuffer); - if( bytes_written < 0 ) - { - ERR_RPT(("PortAudio: write interrupted!")); - break; - } - - totalwritten += bytes_written; - } while( totalwritten < pahsc->pahsc_BytesPerOutputBuffer); - } - - Pa_UpdateStreamTime(pahsc); - } - DBUG(("Pa_AudioThreadProc: left audio loop.\n")); - - past->past_IsActive = 0; - PaHost_StopWatchDog( pahsc ); - -error: - DBUG(("leaving audio thread.\n")); -#ifdef GNUSTEP - GSUnregisterCurrentThread(); /* SB20010904 */ -#endif - return result; -} - -/************************************************************************* -** Determine minimum number of buffers required for this host based -** on minimum latency. Latency can be optionally set by user by setting -** an environment variable. For example, to set latency to 200 msec, put: -** -** set PA_MIN_LATENCY_MSEC=200 -** -** in the cshrc file. -*/ -#define PA_LATENCY_ENV_NAME ("PA_MIN_LATENCY_MSEC") - -int Pa_GetMinNumBuffers( int framesPerBuffer, double framesPerSecond ) -{ - int minBuffers; - int minLatencyMsec = MIN_LATENCY_MSEC; - char *minLatencyText = getenv(PA_LATENCY_ENV_NAME); - if( minLatencyText != NULL ) - { - PRINT(("PA_MIN_LATENCY_MSEC = %s\n", minLatencyText )); - minLatencyMsec = atoi( minLatencyText ); - if( minLatencyMsec < 1 ) minLatencyMsec = 1; - else if( minLatencyMsec > 5000 ) minLatencyMsec = 5000; - } - - minBuffers = (int) ((minLatencyMsec * framesPerSecond) / ( 1000.0 * framesPerBuffer )); - if( minBuffers < 2 ) minBuffers = 2; - return minBuffers; -} - -/*******************************************************************/ -PaError PaHost_OpenStream( internalPortAudioStream *past ) -{ - PaError result = paNoError; - PaHostSoundControl *pahsc; - unsigned int minNumBuffers; - internalPortAudioDevice *pad; - DBUG(("PaHost_OpenStream() called.\n" )); - - /* Allocate and initialize host data. */ - pahsc = (PaHostSoundControl *) malloc(sizeof(PaHostSoundControl)); - if( pahsc == NULL ) - { - result = paInsufficientMemory; - goto error; - } - memset( pahsc, 0, sizeof(PaHostSoundControl) ); - past->past_DeviceData = (void *) pahsc; - - pahsc->pahsc_OutputHandle = BAD_DEVICE_ID; /* No device currently opened. */ - pahsc->pahsc_InputHandle = BAD_DEVICE_ID; - pahsc->pahsc_IsAudioThreadValid = 0; - pahsc->pahsc_IsWatchDogThreadValid = 0; - - /* Allocate native buffers. */ - pahsc->pahsc_BytesPerInputBuffer = past->past_FramesPerUserBuffer * - past->past_NumInputChannels * sizeof(short); - if( past->past_NumInputChannels > 0) - { - pahsc->pahsc_NativeInputBuffer = (short *) malloc(pahsc->pahsc_BytesPerInputBuffer); - if( pahsc->pahsc_NativeInputBuffer == NULL ) - { - result = paInsufficientMemory; - goto error; - } - } - pahsc->pahsc_BytesPerOutputBuffer = past->past_FramesPerUserBuffer * - past->past_NumOutputChannels * sizeof(short); - if( past->past_NumOutputChannels > 0) - { - pahsc->pahsc_NativeOutputBuffer = (short *) malloc(pahsc->pahsc_BytesPerOutputBuffer); - if( pahsc->pahsc_NativeOutputBuffer == NULL ) - { - result = paInsufficientMemory; - goto error; - } - } - - /* DBUG(("PaHost_OpenStream: pahsc_MinFramesPerHostBuffer = %d\n", pahsc->pahsc_MinFramesPerHostBuffer )); */ - minNumBuffers = Pa_GetMinNumBuffers( past->past_FramesPerUserBuffer, past->past_SampleRate ); - past->past_NumUserBuffers = ( minNumBuffers > past->past_NumUserBuffers ) ? minNumBuffers : past->past_NumUserBuffers; - - pahsc->pahsc_InverseMicrosPerBuffer = past->past_SampleRate / (1000000.0 * past->past_FramesPerUserBuffer); - DBUG(("past_SampleRate = %g\n", past->past_SampleRate )); - DBUG(("past_FramesPerUserBuffer = %d\n", past->past_FramesPerUserBuffer )); - DBUG(("pahsc_InverseMicrosPerBuffer = %g\n", pahsc->pahsc_InverseMicrosPerBuffer )); - - /* ------------------------- OPEN DEVICE -----------------------*/ - - /* just output */ - if (past->past_OutputDeviceID == past->past_InputDeviceID) - { - - if ((past->past_NumOutputChannels > 0) && (past->past_NumInputChannels > 0) ) - { - pad = Pa_GetInternalDevice( past->past_OutputDeviceID ); - DBUG(("PaHost_OpenStream: attempt to open %s for O_RDWR\n", pad->pad_DeviceName )); - - /* dmazzoni: test it first in nonblocking mode to - make sure the device is not busy */ - pahsc->pahsc_InputHandle = open(pad->pad_DeviceName,O_RDWR|O_NONBLOCK); - if(pahsc->pahsc_InputHandle==-1) - { - ERR_RPT(("PaHost_OpenStream: could not open %s for O_RDWR\n", pad->pad_DeviceName )); - result = paHostError; - goto error; - } - close(pahsc->pahsc_InputHandle); - - pahsc->pahsc_OutputHandle = pahsc->pahsc_InputHandle = - open(pad->pad_DeviceName,O_RDWR); - if(pahsc->pahsc_InputHandle==-1) - { - ERR_RPT(("PaHost_OpenStream: could not open %s for O_RDWR\n", pad->pad_DeviceName )); - result = paHostError; - goto error; - } - Pa_SetLatency( pahsc->pahsc_OutputHandle, - past->past_NumUserBuffers, past->past_FramesPerUserBuffer, - past->past_NumOutputChannels ); - result = Pa_SetupDeviceFormat( pahsc->pahsc_OutputHandle, - past->past_NumOutputChannels, (int)past->past_SampleRate ); - } - } - else - { - if (past->past_NumOutputChannels > 0) - { - pad = Pa_GetInternalDevice( past->past_OutputDeviceID ); - DBUG(("PaHost_OpenStream: attempt to open %s for O_WRONLY\n", pad->pad_DeviceName )); - /* dmazzoni: test it first in nonblocking mode to - make sure the device is not busy */ - pahsc->pahsc_OutputHandle = open(pad->pad_DeviceName,O_WRONLY|O_NONBLOCK); - if(pahsc->pahsc_OutputHandle==-1) - { - ERR_RPT(("PaHost_OpenStream: could not open %s for O_WRONLY\n", pad->pad_DeviceName )); - result = paHostError; - goto error; - } - close(pahsc->pahsc_OutputHandle); - - pahsc->pahsc_OutputHandle = open(pad->pad_DeviceName,O_WRONLY); - if(pahsc->pahsc_OutputHandle==-1) - { - ERR_RPT(("PaHost_OpenStream: could not open %s for O_WRONLY\n", pad->pad_DeviceName )); - result = paHostError; - goto error; - } - Pa_SetLatency( pahsc->pahsc_OutputHandle, - past->past_NumUserBuffers, past->past_FramesPerUserBuffer, - past->past_NumOutputChannels ); - result = Pa_SetupOutputDeviceFormat( pahsc->pahsc_OutputHandle, - past->past_NumOutputChannels, (int)past->past_SampleRate ); - } - - if (past->past_NumInputChannels > 0) - { - pad = Pa_GetInternalDevice( past->past_InputDeviceID ); - DBUG(("PaHost_OpenStream: attempt to open %s for O_RDONLY\n", pad->pad_DeviceName )); - /* dmazzoni: test it first in nonblocking mode to - make sure the device is not busy */ - pahsc->pahsc_InputHandle = open(pad->pad_DeviceName,O_RDONLY|O_NONBLOCK); - if(pahsc->pahsc_InputHandle==-1) - { - ERR_RPT(("PaHost_OpenStream: could not open %s for O_RDONLY\n", pad->pad_DeviceName )); - result = paHostError; - goto error; - } - close(pahsc->pahsc_InputHandle); - - pahsc->pahsc_InputHandle = open(pad->pad_DeviceName,O_RDONLY); - if(pahsc->pahsc_InputHandle==-1) - { - ERR_RPT(("PaHost_OpenStream: could not open %s for O_RDONLY\n", pad->pad_DeviceName )); - result = paHostError; - goto error; - } - Pa_SetLatency( pahsc->pahsc_InputHandle, /* DH20010115 - was OutputHandle! */ - past->past_NumUserBuffers, past->past_FramesPerUserBuffer, - past->past_NumInputChannels ); - result = Pa_SetupInputDeviceFormat( pahsc->pahsc_InputHandle, - past->past_NumInputChannels, (int)past->past_SampleRate ); - } - } - - - DBUG(("PaHost_OpenStream: SUCCESS - result = %d\n", result )); - return result; - -error: - ERR_RPT(("PaHost_OpenStream: ERROR - result = %d\n", result )); - PaHost_CloseStream( past ); - return result; -} - -/*************************************************************************/ -PaError PaHost_StartOutput( internalPortAudioStream *past ) -{ - past = past; /* unused */ - return paNoError; -} - -/*************************************************************************/ -PaError PaHost_StartInput( internalPortAudioStream *past ) -{ - past = past; /* unused */ - return paNoError; -} - -/*************************************************************************/ -PaError PaHost_StartEngine( internalPortAudioStream *past ) -{ - PaHostSoundControl *pahsc; - PaError result = paNoError; - int hres; - - pahsc = (PaHostSoundControl *) past->past_DeviceData; - - past->past_StopSoon = 0; - past->past_StopNow = 0; - past->past_IsActive = 1; - - /* Use pthread_create() instead of __clone() because: - * - pthread_create also works for other UNIX systems like Solaris, - * - the Java HotSpot VM crashes in pthread_setcanceltype() when using __clone() - */ - hres = pthread_create(&(pahsc->pahsc_AudioThread), - NULL /*pthread_attr_t * attr*/, - (pthread_function_t)Pa_AudioThreadProc, past); - if( hres != 0 ) - { - result = paHostError; - sPaHostError = hres; - pahsc->pahsc_IsAudioThreadValid = 0; - goto error; - } - pahsc->pahsc_IsAudioThreadValid = 1; - -error: - return result; -} - -/*************************************************************************/ -PaError PaHost_StopEngine( internalPortAudioStream *past, int abort ) -{ - int hres; - PaError result = paNoError; - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - - if( pahsc == NULL ) return paNoError; - - /* Tell background thread to stop generating more data and to let current data play out. */ - past->past_StopSoon = 1; - /* If aborting, tell background thread to stop NOW! */ - if( abort ) past->past_StopNow = 1; - - /* Join thread to recover memory resources. */ - if( pahsc->pahsc_IsAudioThreadValid ) - { - /* This check is needed for GNUSTEP - SB20010904 */ - if ( !pthread_equal( pahsc->pahsc_AudioThread, pthread_self() ) ) - { - hres = pthread_join( pahsc->pahsc_AudioThread, NULL ); - } - else - { - DBUG(("Play thread was stopped from itself - can't do pthread_join()\n")); - hres = 0; - } - - if( hres != 0 ) - { - result = paHostError; - sPaHostError = hres; - } - pahsc->pahsc_IsAudioThreadValid = 0; - } - - past->past_IsActive = 0; - - return result; -} - -/*************************************************************************/ -PaError PaHost_StopInput( internalPortAudioStream *past, int abort ) -{ - past = past; /* unused */ - abort = abort; /* unused */ - return paNoError; -} - -/*************************************************************************/ -PaError PaHost_StopOutput( internalPortAudioStream *past, int abort ) -{ - past = past; /* unused */ - abort = abort; /* unused */ - return paNoError; -} - -/*******************************************************************/ -PaError PaHost_CloseStream( internalPortAudioStream *past ) -{ - PaHostSoundControl *pahsc; - - if( past == NULL ) return paBadStreamPtr; - pahsc = (PaHostSoundControl *) past->past_DeviceData; - if( pahsc == NULL ) return paNoError; - - if( pahsc->pahsc_OutputHandle != BAD_DEVICE_ID ) - { - int err = 0; - DBUG(("PaHost_CloseStream: attempt to close output device handle = %d\n", - pahsc->pahsc_OutputHandle )); - - Pa_FlushStream(pahsc->pahsc_OutputHandle); - - err = close(pahsc->pahsc_OutputHandle); - if( err < 0 ) - { - ERR_RPT(("PaHost_CloseStream: warning, closing output device failed.\n")); - } - } - - if( (pahsc->pahsc_InputHandle != BAD_DEVICE_ID) && - (pahsc->pahsc_InputHandle != pahsc->pahsc_OutputHandle) ) - { - int err = 0; - DBUG(("PaHost_CloseStream: attempt to close input device handle = %d\n", - pahsc->pahsc_InputHandle )); - - Pa_FlushStream(pahsc->pahsc_InputHandle); - - err = close(pahsc->pahsc_InputHandle); - if( err < 0 ) - { - ERR_RPT(("PaHost_CloseStream: warning, closing input device failed.\n")); - } - } - pahsc->pahsc_OutputHandle = BAD_DEVICE_ID; - pahsc->pahsc_InputHandle = BAD_DEVICE_ID; - - if( pahsc->pahsc_NativeInputBuffer ) - { - free( pahsc->pahsc_NativeInputBuffer ); - pahsc->pahsc_NativeInputBuffer = NULL; - } - if( pahsc->pahsc_NativeOutputBuffer ) - { - free( pahsc->pahsc_NativeOutputBuffer ); - pahsc->pahsc_NativeOutputBuffer = NULL; - } - - free( pahsc ); - past->past_DeviceData = NULL; - return paNoError; -} - -/*************************************************************************/ -PaError PaHost_Term( void ) -{ - /* Free all of the linked devices. */ - internalPortAudioDevice *pad, *nextPad; - pad = sDeviceList; - while( pad != NULL ) - { - nextPad = pad->pad_Next; - DBUG(("PaHost_Term: freeing %s\n", pad->pad_DeviceName )); - PaHost_FreeFastMemory( pad, sizeof(internalPortAudioDevice) ); - pad = nextPad; - } - sDeviceList = NULL; - return 0; -} - -/************************************************************************* - * Sleep for the requested number of milliseconds. - */ -void Pa_Sleep( long msec ) -{ -#if 0 - struct timeval timeout; - timeout.tv_sec = msec / 1000; - timeout.tv_usec = (msec % 1000) * 1000; - select( 0, NULL, NULL, NULL, &timeout ); -#else - long usecs = msec * 1000; - usleep( usecs ); -#endif -} - -/************************************************************************* - * Allocate memory that can be accessed in real-time. - * This may need to be held in physical memory so that it is not - * paged to virtual memory. - * This call MUST be balanced with a call to PaHost_FreeFastMemory(). - */ -void *PaHost_AllocateFastMemory( long numBytes ) -{ - void *addr = malloc( numBytes ); /* FIXME - do we need physical, wired, non-virtual memory? */ - if( addr != NULL ) memset( addr, 0, numBytes ); - return addr; -} - -/************************************************************************* - * Free memory that could be accessed in real-time. - * This call MUST be balanced with a call to PaHost_AllocateFastMemory(). - */ -void PaHost_FreeFastMemory( void *addr, long numBytes ) -{ - numBytes = numBytes; /* unused */ - if( addr != NULL ) free( addr ); -} - - -/***********************************************************************/ -PaError PaHost_StreamActive( internalPortAudioStream *past ) -{ - PaHostSoundControl *pahsc; - if( past == NULL ) return paBadStreamPtr; - pahsc = (PaHostSoundControl *) past->past_DeviceData; - if( pahsc == NULL ) return paInternalError; - return (PaError) (past->past_IsActive != 0); -} - -/***********************************************************************/ -long Pa_GetHostError( void ) -{ - return (long) sPaHostError; -} diff --git a/pd/portaudio_v18/pa_unix_oss/pa_unix.h b/pd/portaudio_v18/pa_unix_oss/pa_unix.h deleted file mode 100644 index 55b16d50..00000000 --- a/pd/portaudio_v18/pa_unix_oss/pa_unix.h +++ /dev/null @@ -1,141 +0,0 @@ -/* - * PortAudio Portable Real-Time Audio Library - * Latest Version at: http://www.portaudio.com - * Linux OSS Implementation by douglas repetto and Phil Burk - * - * Copyright (c) 1999-2000 Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ - -/* Modification history: - 20020621: pa_unix_oss.c split into pa_unix.c, pa_unix.h, pa_unix_oss.c by - Augustus Saunders. See pa_unix.c for previous history. */ - -/* - PROPOSED - should we add this to "portaudio.h". Problem with - Pa_QueryDevice() not having same driver name os Pa_OpenStream(). - - A PaDriverInfo structure can be passed to the underlying device - on the Pa_OpenStream() call. The contents and interpretation of - the structure is determined by the PA implementation. -*/ -typedef struct PaDriverInfo /* PROPOSED */ -{ - /* Size of structure. Allows driver to extend the structure without breaking existing applications. */ - int size; - /* Can be used to request a specific device name. */ - const char *name; - unsigned long data; -} -PaDriverInfo; - -#include <stdio.h> -#include <stdlib.h> -//#include <malloc.h> -#include <memory.h> -#include <math.h> -#include <sys/ioctl.h> -#include <sys/time.h> -#include <fcntl.h> -#include <unistd.h> -#include <signal.h> -#include <sched.h> -#include <pthread.h> -#include <errno.h> - -#include "portaudio.h" -#include "pa_host.h" -#include "pa_trace.h" - -#define PRINT(x) { printf x; fflush(stdout); } -#define ERR_RPT(x) PRINT(x) -#define DBUG(x) /* PRINT(x) */ -#define DBUGX(x) /* PRINT(x) */ - -#define BAD_DEVICE_ID (-1) - -#define MIN_LATENCY_MSEC (100) -#define MIN_TIMEOUT_MSEC (100) -#define MAX_TIMEOUT_MSEC (1000) - -/************************************************* Definitions ********/ -#ifdef __linux__ - #define DEVICE_NAME_BASE "/dev/dsp" -#else - #define DEVICE_NAME_BASE "/dev/audio" -#endif - -#define MAX_CHARS_DEVNAME (32) -#define MAX_SAMPLE_RATES (10) -typedef struct internalPortAudioDevice -{ - struct internalPortAudioDevice *pad_Next; /* Singly linked list. */ - double pad_SampleRates[MAX_SAMPLE_RATES]; /* for pointing to from pad_Info */ - char pad_DeviceName[MAX_CHARS_DEVNAME]; - PaDeviceInfo pad_Info; -} -internalPortAudioDevice; - -/* Define structure to contain all OSS and Linux specific data. */ -typedef struct PaHostSoundControl -{ - int pahsc_OutputHandle; - int pahsc_InputHandle; - int pahsc_AudioPriority; /* priority of background audio thread */ - pthread_t pahsc_AudioThread; /* background audio thread */ - int pahsc_IsAudioThreadValid; /* Is pahsc_AudioThread valid?*/ pid_t pahsc_AudioThreadPID; /* background audio thread */ - pthread_t pahsc_WatchDogThread; /* highest priority thread that protects system */ - int pahsc_IsWatchDogThreadValid; /* Is pahsc_WatchDogThread valid?*/ - int pahsc_WatchDogRun; /* Ask WatchDog to stop. */ - pthread_t pahsc_CanaryThread; /* low priority thread that detects abuse by audio */ - int pahsc_IsCanaryThreadValid; /* Is pahsc_CanaryThread valid?*/ - struct timeval pahsc_CanaryTime; - int pahsc_CanaryRun; /* Ask Canary to stop. */ - short *pahsc_NativeInputBuffer; - short *pahsc_NativeOutputBuffer; - unsigned int pahsc_BytesPerInputBuffer; /* native buffer size in bytes */ - unsigned int pahsc_BytesPerOutputBuffer; /* native buffer size in bytes */ - /* For measuring CPU utilization. */ - struct timeval pahsc_EntryTime; - double pahsc_InverseMicrosPerBuffer; /* 1/Microseconds of real-time audio per user buffer. */ - - /* For calculating stream time */ - int pahsc_LastPosPtr; - double pahsc_LastStreamBytes; -} -PaHostSoundControl; - -/************************************************* Prototypes **********/ - -internalPortAudioDevice *Pa_GetInternalDevice( PaDeviceID id ); -PaError Pa_QueryDevices( void ); -PaError Pa_QueryDevice( const char *deviceName, internalPortAudioDevice *pad ); -PaError Pa_SetupDeviceFormat( int devHandle, int numChannels, int sampleRate ); -PaError Pa_SetupInputDeviceFormat( int devHandle, int numChannels, int sampleRate ); -PaError Pa_SetupOutputDeviceFormat( int devHandle, int numChannels, int sampleRate ); -void Pa_SetLatency( int devHandle, int numBuffers, int framesPerBuffer, int channelsPerFrame ); -void Pa_UpdateStreamTime(PaHostSoundControl *pahsc); -int Pa_FlushStream(int devHandle); diff --git a/pd/portaudio_v18/pa_unix_oss/pa_unix_oss.c b/pd/portaudio_v18/pa_unix_oss/pa_unix_oss.c deleted file mode 100644 index 386cd75b..00000000 --- a/pd/portaudio_v18/pa_unix_oss/pa_unix_oss.c +++ /dev/null @@ -1,385 +0,0 @@ -/* - * PortAudio Portable Real-Time Audio Library - * Latest Version at: http://www.portaudio.com - * Linux OSS Implementation by douglas repetto and Phil Burk - * - * Copyright (c) 1999-2000 Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ - -/* Modification history: - 20020621: pa_unix_oss.c split into pa_unix.c, pa_unix.h, pa_unix_oss.c by - Augustus Saunders. See pa_unix.c for previous history. Pa_FlushStream - added by Augustus Saunders for Solaris compatibility. - PLB20021018 - Fill device info table with actual sample rates instead of wished for rates. - - Allow stream to open if sample rate within 10% of desired rate. - 20030630 - Thomas Richter - eliminated unused variable warnings. -*/ - -#include "pa_unix.h" - -#ifdef __linux__ -#include <linux/soundcard.h> -#else -#include <machine/soundcard.h> /* JH20010905 */ -#endif - - -#ifndef AFMT_S16_NE -#define AFMT_S16_NE Get_AFMT_S16_NE() -/********************************************************************* - * Some versions of OSS do not define AFMT_S16_NE. So check CPU. - * PowerPC is Big Endian. X86 is Little Endian. - */ -int Get_AFMT_S16_NE( void ) -{ - long testData = 1; - char *ptr = (char *) &testData; - int isLittle = ( *ptr == 1 ); /* Does address point to least significant byte? */ - return isLittle ? AFMT_S16_LE : AFMT_S16_BE; -} -#endif /* AFMT_S16_NE */ - - -/********************************************************************* - * Try to open the named device. - * If it opens, try to set various rates and formats and fill in - * the device info structure. - */ -PaError Pa_QueryDevice( const char *deviceName, internalPortAudioDevice *pad ) -{ - int result = paHostError; - int tempDevHandle; - int numChannels, maxNumChannels; - int format; - int numSampleRates; - int sampleRate; - int numRatesToTry; - int lastRate; - int ratesToTry[9] = {96000, 48000, 44100, 32000, 24000, 22050, 16000, 11025, 8000}; - int i; - - /* douglas: - we have to do this querying in a slightly different order. apparently - some sound cards will give you different info based on their settings. - e.g. a card might give you stereo at 22kHz but only mono at 44kHz. - the correct order for OSS is: format, channels, sample rate - - */ - if ( (tempDevHandle = open(deviceName,O_WRONLY|O_NONBLOCK)) == -1 ) - { - DBUG(("Pa_QueryDevice: could not open %s\n", deviceName )); - return paHostError; - } - - /* Ask OSS what formats are supported by the hardware. */ - pad->pad_Info.nativeSampleFormats = 0; - - if (ioctl(tempDevHandle, SNDCTL_DSP_GETFMTS, &format) == -1) - { - ERR_RPT(("Pa_QueryDevice: could not get format info\n" )); - goto error; - } - if( format & AFMT_U8 ) pad->pad_Info.nativeSampleFormats |= paUInt8; - if( format & AFMT_S16_NE ) pad->pad_Info.nativeSampleFormats |= paInt16; - - /* Negotiate for the maximum number of channels for this device. PLB20010927 - * Consider up to 16 as the upper number of channels. - * Variable numChannels should contain the actual upper limit after the call. - * Thanks to John Lazzaro and Heiko Purnhagen for suggestions. - */ - maxNumChannels = 0; - for( numChannels = 1; numChannels <= 16; numChannels++ ) - { - int temp = numChannels; - DBUG(("Pa_QueryDevice: use SNDCTL_DSP_CHANNELS, numChannels = %d\n", numChannels )) - if(ioctl(tempDevHandle, SNDCTL_DSP_CHANNELS, &temp) < 0 ) - { - /* ioctl() failed so bail out if we already have stereo */ - if( numChannels > 2 ) break; - } - else - { - /* ioctl() worked but bail out if it does not support numChannels. - * We don't want to leave gaps in the numChannels supported. - */ - if( (numChannels > 2) && (temp != numChannels) ) break; - DBUG(("Pa_QueryDevice: temp = %d\n", temp )) - if( temp > maxNumChannels ) maxNumChannels = temp; /* Save maximum. */ - } - } - - /* The above negotiation may fail for an old driver so try this older technique. */ - if( maxNumChannels < 1 ) - { - int stereo = 1; - if(ioctl(tempDevHandle, SNDCTL_DSP_STEREO, &stereo) < 0) - { - maxNumChannels = 1; - } - else - { - maxNumChannels = (stereo) ? 2 : 1; - } - DBUG(("Pa_QueryDevice: use SNDCTL_DSP_STEREO, maxNumChannels = %d\n", maxNumChannels )) - } - - pad->pad_Info.maxOutputChannels = maxNumChannels; - DBUG(("Pa_QueryDevice: maxNumChannels = %d\n", maxNumChannels)) - - /* During channel negotiation, the last ioctl() may have failed. This can - * also cause sample rate negotiation to fail. Hence the following, to return - * to a supported number of channels. SG20011005 */ - { - int temp = maxNumChannels; - if( temp > 2 ) temp = 2; /* use most reasonable default value */ - ioctl(tempDevHandle, SNDCTL_DSP_CHANNELS, &temp); - } - - /* FIXME - for now, assume maxInputChannels = maxOutputChannels. - * Eventually do separate queries for O_WRONLY and O_RDONLY - */ - pad->pad_Info.maxInputChannels = pad->pad_Info.maxOutputChannels; - - DBUG(("Pa_QueryDevice: maxInputChannels = %d\n", - pad->pad_Info.maxInputChannels)) - - - /* Determine available sample rates by trying each one and seeing result. - * OSS often supports funky rates such as 44188 instead of 44100! - */ - numSampleRates = 0; - lastRate = 0; - numRatesToTry = sizeof(ratesToTry)/sizeof(int); - for (i = 0; i < numRatesToTry; i++) - { - sampleRate = ratesToTry[i]; - - if (ioctl(tempDevHandle, SNDCTL_DSP_SPEED, &sampleRate) >= 0 ) /* PLB20010817 */ - { - /* Use whatever rate OSS tells us. PLB20021018 */ - if (sampleRate != lastRate) - { - DBUG(("Pa_QueryDevice: adding sample rate: %d\n", sampleRate)) - pad->pad_SampleRates[numSampleRates] = (float)sampleRate; - numSampleRates++; - lastRate = sampleRate; - } - else - { - DBUG(("Pa_QueryDevice: dang - got sample rate %d again!\n", sampleRate)) - } - } - } - - DBUG(("Pa_QueryDevice: final numSampleRates = %d\n", numSampleRates)) - if (numSampleRates==0) /* HP20010922 */ - { - /* Desparate attempt to keep running even though no good rates found! */ - ERR_RPT(("Pa_QueryDevice: no supported sample rate (or SNDCTL_DSP_SPEED ioctl call failed). Force 44100 Hz\n" )); - pad->pad_SampleRates[numSampleRates++] = 44100; - } - - pad->pad_Info.numSampleRates = numSampleRates; - pad->pad_Info.sampleRates = pad->pad_SampleRates; /* use pointer to embedded array */ - - pad->pad_Info.name = deviceName; - - result = paNoError; - -error: - /* We MUST close the handle here or we won't be able to reopen it later!!! */ - close(tempDevHandle); - - return result; -} - -/*******************************************************************************************/ -PaError Pa_SetupDeviceFormat( int devHandle, int numChannels, int sampleRate ) -{ - PaError result = paNoError; - int tmp; - - /* Set format, channels, and rate in this order to keep OSS happy. */ - /* Set data format. FIXME - handle more native formats. */ - tmp = AFMT_S16_NE; - if( ioctl(devHandle,SNDCTL_DSP_SETFMT,&tmp) == -1) - { - ERR_RPT(("Pa_SetupDeviceFormat: could not SNDCTL_DSP_SETFMT\n" )); - return paHostError; - } - if( tmp != AFMT_S16_NE ) - { - ERR_RPT(("Pa_SetupDeviceFormat: HW does not support AFMT_S16_NE\n" )); - return paHostError; - } - - - /* Set number of channels. */ - tmp = numChannels; - if (ioctl(devHandle, SNDCTL_DSP_CHANNELS, &numChannels) == -1) - { - ERR_RPT(("Pa_SetupDeviceFormat: could not SNDCTL_DSP_CHANNELS\n" )); - return paHostError; - } - if( tmp != numChannels) - { - ERR_RPT(("Pa_SetupDeviceFormat: HW does not support %d channels\n", numChannels )); - return paHostError; - } - - /* Set playing frequency. */ - tmp = sampleRate; - if( ioctl(devHandle,SNDCTL_DSP_SPEED,&tmp) == -1) - { - ERR_RPT(("Pa_SetupDeviceFormat: could not SNDCTL_DSP_SPEED\n" )); - return paHostError; - } - else if( tmp != sampleRate ) - { - int percentError = abs( (100 * (sampleRate - tmp)) / sampleRate ); - PRINT(("Pa_SetupDeviceFormat: warning - requested sample rate = %d Hz - closest = %d\n", - sampleRate, tmp )); - /* Allow sample rate within 10% off of requested rate. PLB20021018 - * Sometimes OSS uses a funky rate like 44188 instead of 44100. - */ - if( percentError > 10 ) - { - ERR_RPT(("Pa_SetupDeviceFormat: HW does not support %d Hz sample rate\n",sampleRate )); - return paHostError; - } - } - - return result; -} - -PaError Pa_SetupOutputDeviceFormat( int devHandle, int numChannels, int sampleRate ) -{ - return Pa_SetupDeviceFormat(devHandle, numChannels, sampleRate); -} - -PaError Pa_SetupInputDeviceFormat( int devHandle, int numChannels, int sampleRate ) -{ - return Pa_SetupDeviceFormat(devHandle, numChannels, sampleRate); -} - - -/******************************************************************************************* -** Set number of fragments and size of fragments to achieve desired latency. -*/ - -static int CalcHigherLogTwo( int n ) -{ - int log2 = 0; - while( (1<<log2) < n ) log2++; - return log2; -} - -void Pa_SetLatency( int devHandle, int numBuffers, int framesPerBuffer, int channelsPerFrame ) -{ - int tmp; - int bufferSize, powerOfTwo; - - /* Increase size of buffers and reduce number of buffers to reduce latency inside driver. */ - while( numBuffers > 8 ) - { - numBuffers = (numBuffers + 1) >> 1; - framesPerBuffer = framesPerBuffer << 1; - } - - /* calculate size of buffers in bytes */ - bufferSize = framesPerBuffer * channelsPerFrame * sizeof(short); /* FIXME - other sizes? */ - - /* Calculate next largest power of two */ - powerOfTwo = CalcHigherLogTwo( bufferSize ); - DBUG(("Pa_SetLatency: numBuffers = %d, framesPerBuffer = %d, powerOfTwo = %d\n", - numBuffers, framesPerBuffer, powerOfTwo )); - - /* Encode info into a single int */ - tmp=(numBuffers<<16) + powerOfTwo; - - if(ioctl(devHandle,SNDCTL_DSP_SETFRAGMENT,&tmp) == -1) - { - ERR_RPT(("Pa_SetLatency: could not SNDCTL_DSP_SETFRAGMENT\n" )); - /* Don't return an error. Best to just continue and hope for the best. */ - ERR_RPT(("Pa_SetLatency: numBuffers = %d, framesPerBuffer = %d, powerOfTwo = %d\n", - numBuffers, framesPerBuffer, powerOfTwo )); - } -} - -/***********************************************************************/ -PaTimestamp Pa_StreamTime( PortAudioStream *stream ) -{ - internalPortAudioStream *past = (internalPortAudioStream *) stream; - PaHostSoundControl *pahsc; - - count_info info; - int delta; - - if( past == NULL ) return paBadStreamPtr; - - pahsc = (PaHostSoundControl *) past->past_DeviceData; - - if( pahsc->pahsc_NativeOutputBuffer ) - { - ioctl(pahsc->pahsc_OutputHandle, SNDCTL_DSP_GETOPTR, &info); - delta = (info.bytes - pahsc->pahsc_LastPosPtr) & 0x000FFFFF; - return (pahsc->pahsc_LastStreamBytes + delta) / (past->past_NumOutputChannels * sizeof(short)); - } - else - { - ioctl(pahsc->pahsc_InputHandle, SNDCTL_DSP_GETIPTR, &info); - delta = (info.bytes - pahsc->pahsc_LastPosPtr) & 0x000FFFFF; - return (pahsc->pahsc_LastStreamBytes + delta) / (past->past_NumInputChannels * sizeof(short)); - } -} - -void Pa_UpdateStreamTime(PaHostSoundControl *pahsc) -{ - count_info info; - int delta; - - /* Update current stream time (using a double so that - we don't wrap around like info.bytes does) */ - if( pahsc->pahsc_NativeOutputBuffer ) - { - ioctl(pahsc->pahsc_OutputHandle, SNDCTL_DSP_GETOPTR, &info); - } - else - { - ioctl(pahsc->pahsc_InputHandle, SNDCTL_DSP_GETIPTR, &info); - } - delta = (info.bytes - pahsc->pahsc_LastPosPtr) & 0x000FFFFF; - pahsc->pahsc_LastStreamBytes += delta; - pahsc->pahsc_LastPosPtr = info.bytes; -} - -PaError Pa_FlushStream(int devHandle) -{ - /* AS: This doesn't do anything under OSS; it was added for Solaris.*/ - devHandle = devHandle; /* unused */ - return paNoError; -} diff --git a/pd/portaudio_v18/pa_unix_oss/pa_unix_solaris.c b/pd/portaudio_v18/pa_unix_oss/pa_unix_solaris.c deleted file mode 100644 index 1e1846b3..00000000 --- a/pd/portaudio_v18/pa_unix_oss/pa_unix_solaris.c +++ /dev/null @@ -1,397 +0,0 @@ -/* - * PortAudio Portable Real-Time Audio Library - * Latest Version at: http://www.portaudio.com - * Linux OSS Implementation by douglas repetto and Phil Burk - * - * Copyright (c) 1999-2000 Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ - -/* Modification history: - 20020621: Initial cut at Solaris modifications jointly by Sam Bayer - and Augustus Saunders. - 20030206 - Martin Rohrbach - various mods for Solaris - */ - -#define __solaris_native__ - -#include "pa_unix.h" - -/* SAM 6/2/02: Docs say we should include sys/audio.h, but - that doesn't exist pre Solaris 2.8. These headers work fine. */ - -#include <sys/audioio.h> -#include <sys/stropts.h> - -/********************************************************************* - * Try to open the named device. - * If it opens, try to set various rates and formats and fill in - * the device info structure. - */ -PaError Pa_QueryDevice( const char *deviceName, internalPortAudioDevice *pad ) -{ - int result = paHostError; - int tempDevHandle; - int numChannels, maxNumChannels; - int numSampleRates; - int sampleRate; - int numRatesToTry; - int ratesToTry[9] = {96000, 48000, 44100, 32000, 24000, 22050, 16000, 11025, 8000}; - int i; - audio_info_t solaris_info; - audio_device_t device_info; - - /* douglas: - we have to do this querying in a slightly different order. apparently - some sound cards will give you different info based on their settins. - e.g. a card might give you stereo at 22kHz but only mono at 44kHz. - the correct order for OSS is: format, channels, sample rate - - */ - /* - to check a device for it's capabilities, it's probably better to use the - equivalent "-ctl"-descriptor - MR - */ - char devname[strlen(deviceName) + 4]; - if ( (tempDevHandle = open(strcat(strcpy(devname, deviceName), "ctl"), O_WRONLY|O_NONBLOCK)) == -1 ) - { - DBUG(("Pa_QueryDevice: could not open %s\n", deviceName )); - return paHostError; - } - - /* Ask OSS what formats are supported by the hardware. */ - pad->pad_Info.nativeSampleFormats = 0; - AUDIO_INITINFO(&solaris_info); - - /* SAM 12/31/01: Sparc native does mulaw, alaw and PCM. - I think PCM is signed. */ - - for (i = 8; i <= 32; i += 8) { - solaris_info.play.precision = i; - solaris_info.play.encoding = AUDIO_ENCODING_LINEAR; - /* If there are no errors, add the format. */ - if (ioctl(tempDevHandle, AUDIO_SETINFO, &solaris_info) > -1) { - switch (i) { - case 8: - pad->pad_Info.nativeSampleFormats |= paInt8; - break; - case 16: - pad->pad_Info.nativeSampleFormats |= paInt16; - break; - case 24: - pad->pad_Info.nativeSampleFormats |= paInt24; - break; - case 32: - pad->pad_Info.nativeSampleFormats |= paInt32; - break; - } - } - } - - maxNumChannels = 0; - for( numChannels = 1; numChannels <= 16; numChannels++ ) - { - int temp = numChannels; - DBUG(("Pa_QueryDevice: use SNDCTL_DSP_CHANNELS, numChannels = %d\n", numChannels )) - AUDIO_INITINFO(&solaris_info); - solaris_info.play.channels = temp; - if (ioctl(tempDevHandle, AUDIO_SETINFO, &solaris_info) < 0) - { - /* ioctl() failed so bail out if we already have stereo */ - if( numChannels > 2 ) break; - } - else - { - /* ioctl() worked but bail out if it does not support numChannels. - * We don't want to leave gaps in the numChannels supported. - */ - if( (numChannels > 2) && (temp != numChannels) ) break; - DBUG(("Pa_QueryDevice: temp = %d\n", temp )) - if( temp > maxNumChannels ) maxNumChannels = temp; /* Save maximum. */ - } - } - - pad->pad_Info.maxOutputChannels = maxNumChannels; - DBUG(("Pa_QueryDevice: maxNumChannels = %d\n", maxNumChannels)) - - /* FIXME - for now, assume maxInputChannels = maxOutputChannels. - * Eventually do separate queries for O_WRONLY and O_RDONLY - */ - pad->pad_Info.maxInputChannels = pad->pad_Info.maxOutputChannels; - - DBUG(("Pa_QueryDevice: maxInputChannels = %d\n", - pad->pad_Info.maxInputChannels)) - - - /* Determine available sample rates by trying each one and seeing result. - */ - numSampleRates = 0; - - AUDIO_INITINFO(&solaris_info); - - numRatesToTry = sizeof(ratesToTry)/sizeof(int); - for (i = 0; i < numRatesToTry; i++) - { - sampleRate = ratesToTry[i]; - - solaris_info.play.sample_rate = sampleRate; /* AS: We opened for Write, so set play */ - if (ioctl(tempDevHandle, AUDIO_SETINFO, &solaris_info) >= 0 ) /* PLB20010817 */ - { - if (sampleRate == ratesToTry[i]) - { - DBUG(("Pa_QueryDevice: got sample rate: %d\n", sampleRate)) - pad->pad_SampleRates[numSampleRates] = (float)ratesToTry[i]; - numSampleRates++; - } - } - } - - DBUG(("Pa_QueryDevice: final numSampleRates = %d\n", numSampleRates)) - if (numSampleRates==0) /* HP20010922 */ - { - ERR_RPT(("Pa_QueryDevice: no supported sample rate (or SNDCTL_DSP_SPEED ioctl call failed).\n" )); - goto error; - } - - pad->pad_Info.numSampleRates = numSampleRates; - pad->pad_Info.sampleRates = pad->pad_SampleRates; - - /* query for the device name instead of using the filesystem-device - MR */ - if (ioctl(tempDevHandle, AUDIO_GETDEV, &device_info) == -1) { - pad->pad_Info.name = deviceName; - } else { - char *pt = (char *)PaHost_AllocateFastMemory(strlen(device_info.name)); - strcpy(pt, device_info.name); - pad->pad_Info.name = pt; - } - - result = paNoError; - -error: - /* We MUST close the handle here or we won't be able to reopen it later!!! */ - close(tempDevHandle); - - return result; -} - -/*******************************************************************************************/ - -PaError Pa_SetupInputDeviceFormat( int devHandle, int numChannels, int sampleRate ) -{ - audio_info_t solaris_info; - AUDIO_INITINFO(&solaris_info); - - /* Sam Bayer/Bryan George 1/10/02: Various folks have - reported that on Solaris Ultra II, the not-right thing - happens on read unless you make sure the audio device is - flushed. The folks who wrote the Robust Audio Tool say: - + XXX driver issue - on Ultra II's if you don't drain - * the device before reading commences then the device - * reads in blocks of 500ms irrespective of the - * blocksize set. After a minute or so it flips into the - * correct mode, but obviously this is too late to be + * useful for most apps. grrr. - */ - /* AS: And the Solaris man audio pages say you should flush before changing formats - anyway. So there you go. */ - if (Pa_FlushStream(devHandle) != paNoError) - return paHostError; - - solaris_info.record.encoding = AUDIO_ENCODING_LINEAR; - solaris_info.record.sample_rate = sampleRate; - solaris_info.record.precision = 16; - solaris_info.record.channels = numChannels; - - if (ioctl(devHandle, AUDIO_SETINFO, &solaris_info) == -1) - { - ERR_RPT(("Pa_SetupDeviceFormat: could not set audio info\n" )); - return paHostError; - } - - return paNoError; -} - -PaError Pa_SetupOutputDeviceFormat( int devHandle, int numChannels, int sampleRate ) -{ - audio_info_t solaris_info; - AUDIO_INITINFO(&solaris_info); - - /* Sam Bayer/Bryan George 1/10/02: Various folks have - reported that on Solaris Ultra II, the not-right thing - happens on read unless you make sure the audio device is - flushed. The folks who wrote the Robust Audio Tool say: - + XXX driver issue - on Ultra II's if you don't drain - * the device before reading commences then the device - * reads in blocks of 500ms irrespective of the - * blocksize set. After a minute or so it flips into the - * correct mode, but obviously this is too late to be + * useful for most apps. grrr. - */ - /* AS: And the Solaris man audio pages say you should flush before changing formats - anyway. So there you go. */ - if (Pa_FlushStream(devHandle) != paNoError) - return paHostError; - - solaris_info.play.encoding = AUDIO_ENCODING_LINEAR; - solaris_info.play.sample_rate = sampleRate; - solaris_info.play.precision = 16; - solaris_info.play.channels = numChannels; - - if (ioctl(devHandle, AUDIO_SETINFO, &solaris_info) == -1) - { - ERR_RPT(("Pa_SetupDeviceFormat: could not set audio info\n" )); - return paHostError; - } - - return paNoError; -} - -PaError Pa_SetupDeviceFormat( int devHandle, int numChannels, int sampleRate ) -{ - PaError result = paNoError; - - result = Pa_SetupOutputDeviceFormat(devHandle, numChannels, sampleRate); - if (result != paNoError) - return result; - return Pa_SetupInputDeviceFormat(devHandle, numChannels, sampleRate); -} - -/******************************************************************************************* -** Set number of fragments and size of fragments to achieve desired latency. -*/ - -static PaError Pa_Unpause(int devHandle); -static PaError Pa_PauseAndFlush(int devHandle); - -void Pa_SetLatency( int devHandle, int numBuffers, int framesPerBuffer, int channelsPerFrame ) -{ - int bufferSize; - audio_info_t solaris_info; - - /* Increase size of buffers and reduce number of buffers to reduce latency inside driver. */ - while( numBuffers > 8 ) - { - numBuffers = (numBuffers + 1) >> 1; - framesPerBuffer = framesPerBuffer << 1; - } - - /* calculate size of buffers in bytes */ - bufferSize = framesPerBuffer * channelsPerFrame * sizeof(short); /* FIXME - other sizes? */ - - DBUG(("Pa_SetLatency: numBuffers = %d, framesPerBuffer = %d\n", - numBuffers, framesPerBuffer)); - - /* SAM 6/6/02: Documentation says to pause and flush before - changing buffer size. */ - - if (Pa_PauseAndFlush(devHandle) != paNoError) { - ERR_RPT(("Pa_SetLatency: could not pause audio\n" )); - return; - } - - AUDIO_INITINFO(&solaris_info); - - /* AS: Doesn't look like solaris has multiple buffers, - so I'm being conservative and - making one buffer. Might not be what we want... */ - - solaris_info.play.buffer_size = solaris_info.record.buffer_size = bufferSize; - - if (ioctl(devHandle, AUDIO_SETINFO, &solaris_info) == -1) - { - ERR_RPT(("Pa_SetLatency: could not set audio info\n" )); - } - Pa_Unpause(devHandle); -} - -/***********************************************************************/ -PaTimestamp Pa_StreamTime( PortAudioStream *stream ) -{ - internalPortAudioStream *past = (internalPortAudioStream *) stream; - PaHostSoundControl *pahsc; - audio_info_t solaris_info; - - if( past == NULL ) return paBadStreamPtr; - - pahsc = (PaHostSoundControl *) past->past_DeviceData; - - ioctl(pahsc->pahsc_OutputHandle, AUDIO_GETINFO, &solaris_info); - return solaris_info.play.samples; -} - -void Pa_UpdateStreamTime(PaHostSoundControl *pahsc) -{ - /* AS: Don't need to do anytying for this under Solaris.*/ -} - -static PaError Pa_PauseAndFlush(int devHandle) -{ - audio_info_t solaris_info; - AUDIO_INITINFO(&solaris_info); - - solaris_info.play.pause = solaris_info.record.pause = 1; - - if (ioctl(devHandle, AUDIO_SETINFO, &solaris_info) == -1) - { - ERR_RPT(("Pa_FlushStream failed.\n")); - return paHostError; - } - - if (ioctl(devHandle, I_FLUSH, FLUSHRW) == -1) - { - ERR_RPT(("Pa_FlushStream failed.\n")); - - /* Unpause! */ - AUDIO_INITINFO(&solaris_info); - solaris_info.play.pause = solaris_info.record.pause = 0; - ioctl(devHandle, AUDIO_SETINFO, &solaris_info); - - return paHostError; - } - return paNoError; -} - -static PaError Pa_Unpause(int devHandle) -{ - audio_info_t solaris_info; - AUDIO_INITINFO(&solaris_info); - - solaris_info.play.pause = solaris_info.record.pause = 0; - - if (ioctl(devHandle, AUDIO_SETINFO, &solaris_info) == -1) - { - ERR_RPT(("Pa_FlushStream failed.\n")); - return paHostError; - } - - return paNoError; -} - -PaError Pa_FlushStream(int devHandle) -{ - PaError res = Pa_PauseAndFlush(devHandle); - if (res == paNoError) - return Pa_Unpause(devHandle); - else return res; -} diff --git a/pd/portaudio_v18/pa_unix_oss/recplay.c b/pd/portaudio_v18/pa_unix_oss/recplay.c deleted file mode 100644 index 9d4c78cf..00000000 --- a/pd/portaudio_v18/pa_unix_oss/recplay.c +++ /dev/null @@ -1,114 +0,0 @@ -/* - * recplay.c - * Phil Burk - * Minimal record and playback test. - * - */ -#include <stdio.h> -#include <unistd.h> -#include <stdlib.h> -#ifndef __STDC__ -/* #include <getopt.h> */ -#endif /* __STDC__ */ -#include <fcntl.h> -#ifdef __STDC__ -#include <string.h> -#else /* __STDC__ */ -#include <strings.h> -#endif /* __STDC__ */ -#include <sys/soundcard.h> - -#define NUM_BYTES (64*1024) -#define BLOCK_SIZE (4*1024) - -#define AUDIO "/dev/dsp" - -char buffer[NUM_BYTES]; - -int audioDev = 0; - -main (int argc, char *argv[]) -{ - int numLeft; - char *ptr; - int num; - int samplesize; - - /********** RECORD ********************/ - /* Open audio device. */ - audioDev = open (AUDIO, O_RDONLY, 0); - if (audioDev == -1) - { - perror (AUDIO); - exit (-1); - } - - /* Set to 16 bit samples. */ - samplesize = 16; - ioctl(audioDev, SNDCTL_DSP_SAMPLESIZE, &samplesize); - if (samplesize != 16) - { - perror("Unable to set the sample size."); - exit(-1); - } - - /* Record in blocks */ - printf("Begin recording.\n"); - numLeft = NUM_BYTES; - ptr = buffer; - while( numLeft >= BLOCK_SIZE ) - { - if ( (num = read (audioDev, ptr, BLOCK_SIZE)) < 0 ) - { - perror (AUDIO); - exit (-1); - } - else - { - printf("Read %d bytes\n", num); - ptr += num; - numLeft -= num; - } - } - - close( audioDev ); - - /********** PLAYBACK ********************/ - /* Open audio device for writing. */ - audioDev = open (AUDIO, O_WRONLY, 0); - if (audioDev == -1) - { - perror (AUDIO); - exit (-1); - } - - /* Set to 16 bit samples. */ - samplesize = 16; - ioctl(audioDev, SNDCTL_DSP_SAMPLESIZE, &samplesize); - if (samplesize != 16) - { - perror("Unable to set the sample size."); - exit(-1); - } - - /* Play in blocks */ - printf("Begin playing.\n"); - numLeft = NUM_BYTES; - ptr = buffer; - while( numLeft >= BLOCK_SIZE ) - { - if ( (num = write (audioDev, ptr, BLOCK_SIZE)) < 0 ) - { - perror (AUDIO); - exit (-1); - } - else - { - printf("Wrote %d bytes\n", num); - ptr += num; - numLeft -= num; - } - } - - close( audioDev ); -} diff --git a/pd/portaudio_v18/pa_win_ds/dsound_wrapper.c b/pd/portaudio_v18/pa_win_ds/dsound_wrapper.c deleted file mode 100644 index 527f0eb3..00000000 --- a/pd/portaudio_v18/pa_win_ds/dsound_wrapper.c +++ /dev/null @@ -1,466 +0,0 @@ -/* - * $Id: dsound_wrapper.c,v 1.1.1.1 2002/01/22 00:52:45 phil Exp $ - * Simplified DirectSound interface. - * - * Author: Phil Burk & Robert Marsanyi - * - * PortAudio Portable Real-Time Audio Library - * For more information see: http://www.softsynth.com/portaudio/ - * DirectSound Implementation - * Copyright (c) 1999-2000 Phil Burk & Robert Marsanyi - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <stdlib.h> -#include <math.h> -#define INITGUID // Needed to build IID_IDirectSoundNotify. See objbase.h for info. -#include <objbase.h> -#include <unknwn.h> -#include "dsound_wrapper.h" -#include "pa_trace.h" - -/************************************************************************************/ -void DSW_Term( DSoundWrapper *dsw ) -{ - // Cleanup the sound buffers - if (dsw->dsw_OutputBuffer) - { - IDirectSoundBuffer_Stop( dsw->dsw_OutputBuffer ); - IDirectSoundBuffer_Release( dsw->dsw_OutputBuffer ); - dsw->dsw_OutputBuffer = NULL; - } -#if SUPPORT_AUDIO_CAPTURE - if (dsw->dsw_InputBuffer) - { - IDirectSoundCaptureBuffer_Stop( dsw->dsw_InputBuffer ); - IDirectSoundCaptureBuffer_Release( dsw->dsw_InputBuffer ); - dsw->dsw_InputBuffer = NULL; - } - if (dsw->dsw_pDirectSoundCapture) - { - IDirectSoundCapture_Release( dsw->dsw_pDirectSoundCapture ); - dsw->dsw_pDirectSoundCapture = NULL; - } -#endif /* SUPPORT_AUDIO_CAPTURE */ - if (dsw->dsw_pDirectSound) - { - IDirectSound_Release( dsw->dsw_pDirectSound ); - dsw->dsw_pDirectSound = NULL; - } -} -/************************************************************************************/ -HRESULT DSW_Init( DSoundWrapper *dsw ) -{ - memset( dsw, 0, sizeof(DSoundWrapper) ); - return 0; -} -/************************************************************************************/ -HRESULT DSW_InitOutputDevice( DSoundWrapper *dsw, LPGUID lpGUID ) -{ - // Create the DS object - HRESULT hr = DirectSoundCreate( lpGUID, &dsw->dsw_pDirectSound, NULL ); - if( hr != DS_OK ) return hr; - return hr; -} - -/************************************************************************************/ -HRESULT DSW_InitOutputBuffer( DSoundWrapper *dsw, unsigned long nFrameRate, int nChannels, int bytesPerBuffer ) -{ - DWORD dwDataLen; - DWORD playCursor; - HRESULT result; - LPDIRECTSOUNDBUFFER pPrimaryBuffer; - HWND hWnd; - HRESULT hr; - WAVEFORMATEX wfFormat; - DSBUFFERDESC primaryDesc; - DSBUFFERDESC secondaryDesc; - unsigned char* pDSBuffData; - LARGE_INTEGER counterFrequency; - dsw->dsw_OutputSize = bytesPerBuffer; - dsw->dsw_OutputRunning = FALSE; - dsw->dsw_OutputUnderflows = 0; - dsw->dsw_FramesWritten = 0; - dsw->dsw_BytesPerFrame = nChannels * sizeof(short); - // We were using getForegroundWindow() but sometimes the ForegroundWindow may not be the - // applications's window. Also if that window is closed before the Buffer is closed - // then DirectSound can crash. (Thanks for Scott Patterson for reporting this.) - // So we will use GetDesktopWindow() which was suggested by Miller Puckette. - // hWnd = GetForegroundWindow(); - hWnd = GetDesktopWindow(); - // Set cooperative level to DSSCL_EXCLUSIVE so that we can get 16 bit output, 44.1 KHz. - // Exclusize also prevents unexpected sounds from other apps during a performance. - if ((hr = IDirectSound_SetCooperativeLevel( dsw->dsw_pDirectSound, - hWnd, DSSCL_EXCLUSIVE)) != DS_OK) - { - return hr; - } - // ----------------------------------------------------------------------- - // Create primary buffer and set format just so we can specify our custom format. - // Otherwise we would be stuck with the default which might be 8 bit or 22050 Hz. - // Setup the primary buffer description - ZeroMemory(&primaryDesc, sizeof(DSBUFFERDESC)); - primaryDesc.dwSize = sizeof(DSBUFFERDESC); - primaryDesc.dwFlags = DSBCAPS_PRIMARYBUFFER; // all panning, mixing, etc done by synth - primaryDesc.dwBufferBytes = 0; - primaryDesc.lpwfxFormat = NULL; - // Create the buffer - if ((result = IDirectSound_CreateSoundBuffer( dsw->dsw_pDirectSound, - &primaryDesc, &pPrimaryBuffer, NULL)) != DS_OK) return result; - // Define the buffer format - wfFormat.wFormatTag = WAVE_FORMAT_PCM; - wfFormat.nChannels = nChannels; - wfFormat.nSamplesPerSec = nFrameRate; - wfFormat.wBitsPerSample = 8 * sizeof(short); - wfFormat.nBlockAlign = wfFormat.nChannels * wfFormat.wBitsPerSample / 8; - wfFormat.nAvgBytesPerSec = wfFormat.nSamplesPerSec * wfFormat.nBlockAlign; - wfFormat.cbSize = 0; /* No extended format info. */ - // Set the primary buffer's format - if((result = IDirectSoundBuffer_SetFormat( pPrimaryBuffer, &wfFormat)) != DS_OK) return result; - // ---------------------------------------------------------------------- - // Setup the secondary buffer description - ZeroMemory(&secondaryDesc, sizeof(DSBUFFERDESC)); - secondaryDesc.dwSize = sizeof(DSBUFFERDESC); - secondaryDesc.dwFlags = DSBCAPS_GLOBALFOCUS | DSBCAPS_GETCURRENTPOSITION2; - secondaryDesc.dwBufferBytes = bytesPerBuffer; - secondaryDesc.lpwfxFormat = &wfFormat; - // Create the secondary buffer - if ((result = IDirectSound_CreateSoundBuffer( dsw->dsw_pDirectSound, - &secondaryDesc, &dsw->dsw_OutputBuffer, NULL)) != DS_OK) return result; - // Lock the DS buffer - if ((result = IDirectSoundBuffer_Lock( dsw->dsw_OutputBuffer, 0, dsw->dsw_OutputSize, (LPVOID*)&pDSBuffData, - &dwDataLen, NULL, 0, 0)) != DS_OK) return result; - // Zero the DS buffer - ZeroMemory(pDSBuffData, dwDataLen); - // Unlock the DS buffer - if ((result = IDirectSoundBuffer_Unlock( dsw->dsw_OutputBuffer, pDSBuffData, dwDataLen, NULL, 0)) != DS_OK) return result; - if( QueryPerformanceFrequency( &counterFrequency ) ) - { - int framesInBuffer = bytesPerBuffer / (nChannels * sizeof(short)); - dsw->dsw_CounterTicksPerBuffer.QuadPart = (counterFrequency.QuadPart * framesInBuffer) / nFrameRate; - AddTraceMessage("dsw_CounterTicksPerBuffer = %d\n", dsw->dsw_CounterTicksPerBuffer.LowPart ); - } - else - { - dsw->dsw_CounterTicksPerBuffer.QuadPart = 0; - } - // Let DSound set the starting write position because if we set it to zero, it looks like the - // buffer is full to begin with. This causes a long pause before sound starts when using large buffers. - hr = IDirectSoundBuffer_GetCurrentPosition( dsw->dsw_OutputBuffer, &playCursor, &dsw->dsw_WriteOffset ); - if( hr != DS_OK ) - { - return hr; - } - dsw->dsw_FramesWritten = dsw->dsw_WriteOffset / dsw->dsw_BytesPerFrame; - /* printf("DSW_InitOutputBuffer: playCursor = %d, writeCursor = %d\n", playCursor, dsw->dsw_WriteOffset ); */ - return DS_OK; -} - -/************************************************************************************/ -HRESULT DSW_StartOutput( DSoundWrapper *dsw ) -{ - HRESULT hr; - QueryPerformanceCounter( &dsw->dsw_LastPlayTime ); - dsw->dsw_LastPlayCursor = 0; - dsw->dsw_FramesPlayed = 0; - hr = IDirectSoundBuffer_SetCurrentPosition( dsw->dsw_OutputBuffer, 0 ); - if( hr != DS_OK ) - { - return hr; - } - // Start the buffer playback in a loop. - if( dsw->dsw_OutputBuffer != NULL ) - { - hr = IDirectSoundBuffer_Play( dsw->dsw_OutputBuffer, 0, 0, DSBPLAY_LOOPING ); - if( hr != DS_OK ) - { - return hr; - } - dsw->dsw_OutputRunning = TRUE; - } - - return 0; -} -/************************************************************************************/ -HRESULT DSW_StopOutput( DSoundWrapper *dsw ) -{ - // Stop the buffer playback - if( dsw->dsw_OutputBuffer != NULL ) - { - dsw->dsw_OutputRunning = FALSE; - return IDirectSoundBuffer_Stop( dsw->dsw_OutputBuffer ); - } - else return 0; -} - -/************************************************************************************/ -HRESULT DSW_QueryOutputSpace( DSoundWrapper *dsw, long *bytesEmpty ) -{ - HRESULT hr; - DWORD playCursor; - DWORD writeCursor; - long numBytesEmpty; - long playWriteGap; - // Query to see how much room is in buffer. - // Note: Even though writeCursor is not used, it must be passed to prevent DirectSound from dieing - // under WinNT. The Microsoft documentation says we can pass NULL but apparently not. - // Thanks to Max Rheiner for the fix. - hr = IDirectSoundBuffer_GetCurrentPosition( dsw->dsw_OutputBuffer, &playCursor, &writeCursor ); - if( hr != DS_OK ) - { - return hr; - } - AddTraceMessage("playCursor", playCursor); - AddTraceMessage("dsw_WriteOffset", dsw->dsw_WriteOffset); - // Determine size of gap between playIndex and WriteIndex that we cannot write into. - playWriteGap = writeCursor - playCursor; - if( playWriteGap < 0 ) playWriteGap += dsw->dsw_OutputSize; // unwrap - /* DirectSound doesn't have a large enough playCursor so we cannot detect wrap-around. */ - /* Attempt to detect playCursor wrap-around and correct it. */ - if( dsw->dsw_OutputRunning && (dsw->dsw_CounterTicksPerBuffer.QuadPart != 0) ) - { - /* How much time has elapsed since last check. */ - LARGE_INTEGER currentTime; - LARGE_INTEGER elapsedTime; - long bytesPlayed; - long bytesExpected; - long buffersWrapped; - QueryPerformanceCounter( ¤tTime ); - elapsedTime.QuadPart = currentTime.QuadPart - dsw->dsw_LastPlayTime.QuadPart; - dsw->dsw_LastPlayTime = currentTime; - /* How many bytes does DirectSound say have been played. */ - bytesPlayed = playCursor - dsw->dsw_LastPlayCursor; - if( bytesPlayed < 0 ) bytesPlayed += dsw->dsw_OutputSize; // unwrap - dsw->dsw_LastPlayCursor = playCursor; - /* Calculate how many bytes we would have expected to been played by now. */ - bytesExpected = (long) ((elapsedTime.QuadPart * dsw->dsw_OutputSize) / dsw->dsw_CounterTicksPerBuffer.QuadPart); - buffersWrapped = (bytesExpected - bytesPlayed) / dsw->dsw_OutputSize; - if( buffersWrapped > 0 ) - { - AddTraceMessage("playCursor wrapped! bytesPlayed", bytesPlayed ); - AddTraceMessage("playCursor wrapped! bytesExpected", bytesExpected ); - playCursor += (buffersWrapped * dsw->dsw_OutputSize); - bytesPlayed += (buffersWrapped * dsw->dsw_OutputSize); - } - /* Maintain frame output cursor. */ - dsw->dsw_FramesPlayed += (bytesPlayed / dsw->dsw_BytesPerFrame); - } - numBytesEmpty = playCursor - dsw->dsw_WriteOffset; - if( numBytesEmpty < 0 ) numBytesEmpty += dsw->dsw_OutputSize; // unwrap offset - /* Have we underflowed? */ - if( numBytesEmpty > (dsw->dsw_OutputSize - playWriteGap) ) - { - if( dsw->dsw_OutputRunning ) - { - dsw->dsw_OutputUnderflows += 1; - AddTraceMessage("underflow detected! numBytesEmpty", numBytesEmpty ); - } - dsw->dsw_WriteOffset = writeCursor; - numBytesEmpty = dsw->dsw_OutputSize - playWriteGap; - } - *bytesEmpty = numBytesEmpty; - return hr; -} - -/************************************************************************************/ -HRESULT DSW_ZeroEmptySpace( DSoundWrapper *dsw ) -{ - HRESULT hr; - LPBYTE lpbuf1 = NULL; - LPBYTE lpbuf2 = NULL; - DWORD dwsize1 = 0; - DWORD dwsize2 = 0; - long bytesEmpty; - hr = DSW_QueryOutputSpace( dsw, &bytesEmpty ); // updates dsw_FramesPlayed - if (hr != DS_OK) return hr; - if( bytesEmpty == 0 ) return DS_OK; - // Lock free space in the DS - hr = IDirectSoundBuffer_Lock( dsw->dsw_OutputBuffer, dsw->dsw_WriteOffset, bytesEmpty, (void **) &lpbuf1, &dwsize1, - (void **) &lpbuf2, &dwsize2, 0); - if (hr == DS_OK) - { - // Copy the buffer into the DS - ZeroMemory(lpbuf1, dwsize1); - if(lpbuf2 != NULL) - { - ZeroMemory(lpbuf2, dwsize2); - } - // Update our buffer offset and unlock sound buffer - dsw->dsw_WriteOffset = (dsw->dsw_WriteOffset + dwsize1 + dwsize2) % dsw->dsw_OutputSize; - IDirectSoundBuffer_Unlock( dsw->dsw_OutputBuffer, lpbuf1, dwsize1, lpbuf2, dwsize2); - dsw->dsw_FramesWritten += bytesEmpty / dsw->dsw_BytesPerFrame; - } - return hr; -} - -/************************************************************************************/ -HRESULT DSW_WriteBlock( DSoundWrapper *dsw, char *buf, long numBytes ) -{ - HRESULT hr; - LPBYTE lpbuf1 = NULL; - LPBYTE lpbuf2 = NULL; - DWORD dwsize1 = 0; - DWORD dwsize2 = 0; - // Lock free space in the DS - hr = IDirectSoundBuffer_Lock( dsw->dsw_OutputBuffer, dsw->dsw_WriteOffset, numBytes, (void **) &lpbuf1, &dwsize1, - (void **) &lpbuf2, &dwsize2, 0); - if (hr == DS_OK) - { - // Copy the buffer into the DS - CopyMemory(lpbuf1, buf, dwsize1); - if(lpbuf2 != NULL) - { - CopyMemory(lpbuf2, buf+dwsize1, dwsize2); - } - // Update our buffer offset and unlock sound buffer - dsw->dsw_WriteOffset = (dsw->dsw_WriteOffset + dwsize1 + dwsize2) % dsw->dsw_OutputSize; - IDirectSoundBuffer_Unlock( dsw->dsw_OutputBuffer, lpbuf1, dwsize1, lpbuf2, dwsize2); - dsw->dsw_FramesWritten += numBytes / dsw->dsw_BytesPerFrame; - } - return hr; -} - -/************************************************************************************/ -DWORD DSW_GetOutputStatus( DSoundWrapper *dsw ) -{ - DWORD status; - if (IDirectSoundBuffer_GetStatus( dsw->dsw_OutputBuffer, &status ) != DS_OK) - return( DSERR_INVALIDPARAM ); - else - return( status ); -} - -#if SUPPORT_AUDIO_CAPTURE -/* These routines are used to support audio input. - * Do NOT compile these calls when using NT4 because it does - * not support the entry points. - */ -/************************************************************************************/ -HRESULT DSW_InitInputDevice( DSoundWrapper *dsw, LPGUID lpGUID ) -{ - HRESULT hr = DirectSoundCaptureCreate( lpGUID, &dsw->dsw_pDirectSoundCapture, NULL ); - if( hr != DS_OK ) return hr; - return hr; -} -/************************************************************************************/ -HRESULT DSW_InitInputBuffer( DSoundWrapper *dsw, unsigned long nFrameRate, int nChannels, int bytesPerBuffer ) -{ - DSCBUFFERDESC captureDesc; - WAVEFORMATEX wfFormat; - HRESULT result; - // Define the buffer format - wfFormat.wFormatTag = WAVE_FORMAT_PCM; - wfFormat.nChannels = nChannels; - wfFormat.nSamplesPerSec = nFrameRate; - wfFormat.wBitsPerSample = 8 * sizeof(short); - wfFormat.nBlockAlign = wfFormat.nChannels * (wfFormat.wBitsPerSample / 8); - wfFormat.nAvgBytesPerSec = wfFormat.nSamplesPerSec * wfFormat.nBlockAlign; - wfFormat.cbSize = 0; /* No extended format info. */ - dsw->dsw_InputSize = bytesPerBuffer; - // ---------------------------------------------------------------------- - // Setup the secondary buffer description - ZeroMemory(&captureDesc, sizeof(DSCBUFFERDESC)); - captureDesc.dwSize = sizeof(DSCBUFFERDESC); - captureDesc.dwFlags = 0; - captureDesc.dwBufferBytes = bytesPerBuffer; - captureDesc.lpwfxFormat = &wfFormat; - // Create the capture buffer - if ((result = IDirectSoundCapture_CreateCaptureBuffer( dsw->dsw_pDirectSoundCapture, - &captureDesc, &dsw->dsw_InputBuffer, NULL)) != DS_OK) return result; - dsw->dsw_ReadOffset = 0; // reset last read position to start of buffer - return DS_OK; -} - -/************************************************************************************/ -HRESULT DSW_StartInput( DSoundWrapper *dsw ) -{ - // Start the buffer playback - if( dsw->dsw_InputBuffer != NULL ) - { - return IDirectSoundCaptureBuffer_Start( dsw->dsw_InputBuffer, DSCBSTART_LOOPING ); - } - else return 0; -} - -/************************************************************************************/ -HRESULT DSW_StopInput( DSoundWrapper *dsw ) -{ - // Stop the buffer playback - if( dsw->dsw_InputBuffer != NULL ) - { - return IDirectSoundCaptureBuffer_Stop( dsw->dsw_InputBuffer ); - } - else return 0; -} - -/************************************************************************************/ -HRESULT DSW_QueryInputFilled( DSoundWrapper *dsw, long *bytesFilled ) -{ - HRESULT hr; - DWORD capturePos; - DWORD readPos; - long filled; - // Query to see how much data is in buffer. - // We don't need the capture position but sometimes DirectSound doesn't handle NULLS correctly - // so let's pass a pointer just to be safe. - hr = IDirectSoundCaptureBuffer_GetCurrentPosition( dsw->dsw_InputBuffer, &capturePos, &readPos ); - if( hr != DS_OK ) - { - return hr; - } - filled = readPos - dsw->dsw_ReadOffset; - if( filled < 0 ) filled += dsw->dsw_InputSize; // unwrap offset - *bytesFilled = filled; - return hr; -} - -/************************************************************************************/ -HRESULT DSW_ReadBlock( DSoundWrapper *dsw, char *buf, long numBytes ) -{ - HRESULT hr; - LPBYTE lpbuf1 = NULL; - LPBYTE lpbuf2 = NULL; - DWORD dwsize1 = 0; - DWORD dwsize2 = 0; - // Lock free space in the DS - hr = IDirectSoundCaptureBuffer_Lock ( dsw->dsw_InputBuffer, dsw->dsw_ReadOffset, numBytes, (void **) &lpbuf1, &dwsize1, - (void **) &lpbuf2, &dwsize2, 0); - if (hr == DS_OK) - { - // Copy from DS to the buffer - CopyMemory( buf, lpbuf1, dwsize1); - if(lpbuf2 != NULL) - { - CopyMemory( buf+dwsize1, lpbuf2, dwsize2); - } - // Update our buffer offset and unlock sound buffer - dsw->dsw_ReadOffset = (dsw->dsw_ReadOffset + dwsize1 + dwsize2) % dsw->dsw_InputSize; - IDirectSoundCaptureBuffer_Unlock ( dsw->dsw_InputBuffer, lpbuf1, dwsize1, lpbuf2, dwsize2); - } - return hr; -} - -#endif /* SUPPORT_AUDIO_CAPTURE */ diff --git a/pd/portaudio_v18/pa_win_ds/dsound_wrapper.h b/pd/portaudio_v18/pa_win_ds/dsound_wrapper.h deleted file mode 100644 index b04b0020..00000000 --- a/pd/portaudio_v18/pa_win_ds/dsound_wrapper.h +++ /dev/null @@ -1,106 +0,0 @@ -#ifndef __DSOUND_WRAPPER_H -#define __DSOUND_WRAPPER_H -/* - * $Id: dsound_wrapper.h,v 1.1.1.1 2002/01/22 00:52:45 phil Exp $ - * Simplified DirectSound interface. - * - * Author: Phil Burk & Robert Marsanyi - * - * For PortAudio Portable Real-Time Audio Library - * For more information see: http://www.softsynth.com/portaudio/ - * DirectSound Implementation - * Copyright (c) 1999-2000 Phil Burk & Robert Marsanyi - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <DSound.h> -#if !defined(BOOL) -#define BOOL short -#endif -#ifndef SUPPORT_AUDIO_CAPTURE -#define SUPPORT_AUDIO_CAPTURE (1) -#endif - -#ifdef __cplusplus -extern "C" -{ -#endif /* __cplusplus */ - -#define DSW_NUM_POSITIONS (4) -#define DSW_NUM_EVENTS (5) -#define DSW_TERMINATION_EVENT (DSW_NUM_POSITIONS) - -typedef struct -{ - /* Output */ - LPDIRECTSOUND dsw_pDirectSound; - LPDIRECTSOUNDBUFFER dsw_OutputBuffer; - DWORD dsw_WriteOffset; /* last write position */ - INT dsw_OutputSize; - INT dsw_BytesPerFrame; - /* Try to detect play buffer underflows. */ - LARGE_INTEGER dsw_CounterTicksPerBuffer; /* counter ticks it should take to play a full buffer */ - LARGE_INTEGER dsw_LastPlayTime; - UINT dsw_LastPlayCursor; - UINT dsw_OutputUnderflows; - BOOL dsw_OutputRunning; - /* use double which lets us can play for several thousand years with enough precision */ - double dsw_FramesWritten; - double dsw_FramesPlayed; -#if SUPPORT_AUDIO_CAPTURE - /* Input */ - LPDIRECTSOUNDCAPTURE dsw_pDirectSoundCapture; - LPDIRECTSOUNDCAPTUREBUFFER dsw_InputBuffer; - UINT dsw_ReadOffset; /* last read position */ - UINT dsw_InputSize; -#endif /* SUPPORT_AUDIO_CAPTURE */ - -} -DSoundWrapper; -HRESULT DSW_Init( DSoundWrapper *dsw ); -void DSW_Term( DSoundWrapper *dsw ); -HRESULT DSW_InitOutputBuffer( DSoundWrapper *dsw, unsigned long nFrameRate, - int nChannels, int bufSize ); -HRESULT DSW_StartOutput( DSoundWrapper *dsw ); -HRESULT DSW_StopOutput( DSoundWrapper *dsw ); -DWORD DSW_GetOutputStatus( DSoundWrapper *dsw ); -HRESULT DSW_WriteBlock( DSoundWrapper *dsw, char *buf, long numBytes ); -HRESULT DSW_ZeroEmptySpace( DSoundWrapper *dsw ); -HRESULT DSW_QueryOutputSpace( DSoundWrapper *dsw, long *bytesEmpty ); -HRESULT DSW_Enumerate( DSoundWrapper *dsw ); - -#if SUPPORT_AUDIO_CAPTURE -HRESULT DSW_InitInputBuffer( DSoundWrapper *dsw, unsigned long nFrameRate, - int nChannels, int bufSize ); -HRESULT DSW_StartInput( DSoundWrapper *dsw ); -HRESULT DSW_StopInput( DSoundWrapper *dsw ); -HRESULT DSW_ReadBlock( DSoundWrapper *dsw, char *buf, long numBytes ); -HRESULT DSW_QueryInputFilled( DSoundWrapper *dsw, long *bytesFilled ); -#endif /* SUPPORT_AUDIO_CAPTURE */ - -#ifdef __cplusplus -} -#endif /* __cplusplus */ -#endif /* __DSOUND_WRAPPER_H */ diff --git a/pd/portaudio_v18/pa_win_ds/pa_dsound.c b/pd/portaudio_v18/pa_win_ds/pa_dsound.c deleted file mode 100644 index 52532625..00000000 --- a/pd/portaudio_v18/pa_win_ds/pa_dsound.c +++ /dev/null @@ -1,1042 +0,0 @@ -/* - * $Id: pa_dsound.c,v 1.2.4.1 2002/11/19 20:46:06 philburk Exp $ - * PortAudio Portable Real-Time Audio Library - * Latest Version at: http://www.softsynth.com/portaudio/ - * DirectSound Implementation - * - * Copyright (c) 1999-2000 Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -/* Modifications - * 7/19/01 Mike Berry - casts for compiling with __MWERKS__ CodeWarrior - * 9/27/01 Phil Burk - use number of frames instead of real-time for CPULoad calculation. - * 4/19/02 Phil Burk - Check for Win XP for system latency calculation. - */ -/* Compiler flags: - SUPPORT_AUDIO_CAPTURE - define this flag if you want to SUPPORT_AUDIO_CAPTURE - */ - -#include <stdio.h> -#include <stdlib.h> -#ifndef __MWERKS__ -#include <malloc.h> -#include <memory.h> -#endif //__MWERKS__ -#include <math.h> -#include "portaudio.h" -#include "pa_host.h" -#include "pa_trace.h" -#include "dsound_wrapper.h" - -#define PRINT(x) { printf x; fflush(stdout); } -#define ERR_RPT(x) PRINT(x) -#define DBUG(x) /* PRINT(x) */ -#define DBUGX(x) /* PRINT(x) */ - -#define PA_USE_HIGH_LATENCY (0) -#if PA_USE_HIGH_LATENCY -#define PA_WIN_9X_LATENCY (500) -#define PA_WIN_NT_LATENCY (600) -#else -#define PA_WIN_9X_LATENCY (140) -#define PA_WIN_NT_LATENCY (280) -#endif - -#define PA_WIN_WDM_LATENCY (120) - -/* Trigger an underflow for testing purposes. Should normally be (0). */ -#define PA_SIMULATE_UNDERFLOW (0) -#if PA_SIMULATE_UNDERFLOW -static gUnderCallbackCounter = 0; -#define UNDER_START_GAP (10) -#define UNDER_STOP_GAP (UNDER_START_GAP + 4) -#endif - -/************************************************* Definitions ********/ -typedef struct internalPortAudioStream internalPortAudioStream; -typedef struct internalPortAudioDevice -{ - GUID pad_GUID; - GUID *pad_lpGUID; - double pad_SampleRates[10]; /* for pointing to from pad_Info FIXME?!*/ - PaDeviceInfo pad_Info; -} -internalPortAudioDevice; - -/* Define structure to contain all DirectSound and Windows specific data. */ -typedef struct PaHostSoundControl -{ - DSoundWrapper pahsc_DSoundWrapper; - MMRESULT pahsc_TimerID; - BOOL pahsc_IfInsideCallback; /* Test for reentrancy. */ - short *pahsc_NativeBuffer; - unsigned int pahsc_BytesPerBuffer; /* native buffer size in bytes */ - double pahsc_ValidFramesWritten; - int pahsc_FramesPerDSBuffer; - /* For measuring CPU utilization. */ - LARGE_INTEGER pahsc_EntryCount; - double pahsc_InverseTicksPerUserBuffer; -} -PaHostSoundControl; - -/************************************************* Shared Data ********/ -/* FIXME - put Mutex around this shared data. */ -static int sNumDevices = 0; -static int sDeviceIndex = 0; -static internalPortAudioDevice *sDevices = NULL; -static int sDefaultInputDeviceID = paNoDevice; -static int sDefaultOutputDeviceID = paNoDevice; -static int sEnumerationError; -static int sPaHostError = 0; -/************************************************* Prototypes **********/ -static internalPortAudioDevice *Pa_GetInternalDevice( PaDeviceID id ); -static BOOL CALLBACK Pa_EnumProc(LPGUID lpGUID, - LPCTSTR lpszDesc, - LPCTSTR lpszDrvName, - LPVOID lpContext ); -static BOOL CALLBACK Pa_CountDevProc(LPGUID lpGUID, - LPCTSTR lpszDesc, - LPCTSTR lpszDrvName, - LPVOID lpContext ); -static Pa_QueryDevices( void ); -static void CALLBACK Pa_TimerCallback(UINT uID, UINT uMsg, - DWORD dwUser, DWORD dw1, DWORD dw2); - -/********************************* BEGIN CPU UTILIZATION MEASUREMENT ****/ -static void Pa_StartUsageCalculation( internalPortAudioStream *past ) -{ - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - if( pahsc == NULL ) return; - /* Query system timer for usage analysis and to prevent overuse of CPU. */ - QueryPerformanceCounter( &pahsc->pahsc_EntryCount ); -} - -static void Pa_EndUsageCalculation( internalPortAudioStream *past ) -{ - LARGE_INTEGER CurrentCount = { 0, 0 }; - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - if( pahsc == NULL ) return; - /* - ** Measure CPU utilization during this callback. Note that this calculation - ** assumes that we had the processor the whole time. - */ -#define LOWPASS_COEFFICIENT_0 (0.9) -#define LOWPASS_COEFFICIENT_1 (0.99999 - LOWPASS_COEFFICIENT_0) - if( QueryPerformanceCounter( &CurrentCount ) ) - { - LONGLONG InsideCount = CurrentCount.QuadPart - pahsc->pahsc_EntryCount.QuadPart; - double newUsage = InsideCount * pahsc->pahsc_InverseTicksPerUserBuffer; - past->past_Usage = (LOWPASS_COEFFICIENT_0 * past->past_Usage) + - (LOWPASS_COEFFICIENT_1 * newUsage); - } -} - -/****************************************** END CPU UTILIZATION *******/ -static PaError Pa_QueryDevices( void ) -{ - int numBytes; - sDefaultInputDeviceID = paNoDevice; - sDefaultOutputDeviceID = paNoDevice; - /* Enumerate once just to count devices. */ - sNumDevices = 0; // for default device - DirectSoundEnumerate( (LPDSENUMCALLBACK)Pa_CountDevProc, NULL ); -#if SUPPORT_AUDIO_CAPTURE - DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK)Pa_CountDevProc, NULL ); -#endif /* SUPPORT_AUDIO_CAPTURE */ - /* Allocate structures to hold device info. */ - numBytes = sNumDevices * sizeof(internalPortAudioDevice); - sDevices = (internalPortAudioDevice *)PaHost_AllocateFastMemory( numBytes ); /* MEM */ - if( sDevices == NULL ) return paInsufficientMemory; - /* Enumerate again to fill in structures. */ - sDeviceIndex = 0; - sEnumerationError = 0; - DirectSoundEnumerate( (LPDSENUMCALLBACK)Pa_EnumProc, (void *)0 ); -#if SUPPORT_AUDIO_CAPTURE - if( sEnumerationError != paNoError ) return sEnumerationError; - sEnumerationError = 0; - DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK)Pa_EnumProc, (void *)1 ); -#endif /* SUPPORT_AUDIO_CAPTURE */ - return sEnumerationError; -} -/************************************************************************************/ -long Pa_GetHostError() -{ - return sPaHostError; -} -/************************************************************************************ -** Just count devices so we know how much memory to allocate. -*/ -static BOOL CALLBACK Pa_CountDevProc(LPGUID lpGUID, - LPCTSTR lpszDesc, - LPCTSTR lpszDrvName, - LPVOID lpContext ) -{ - sNumDevices++; - return TRUE; -} -/************************************************************************************ -** Extract capabilities info from each device. -*/ -static BOOL CALLBACK Pa_EnumProc(LPGUID lpGUID, - LPCTSTR lpszDesc, - LPCTSTR lpszDrvName, - LPVOID lpContext ) -{ - HRESULT hr; - LPDIRECTSOUND lpDirectSound; -#if SUPPORT_AUDIO_CAPTURE - LPDIRECTSOUNDCAPTURE lpDirectSoundCapture; -#endif /* SUPPORT_AUDIO_CAPTURE */ - int isInput = (int) lpContext; /* Passed from Pa_CountDevices() */ - internalPortAudioDevice *pad; - - if( sDeviceIndex >= sNumDevices ) - { - sEnumerationError = paInternalError; - return FALSE; - } - pad = &sDevices[sDeviceIndex]; - /* Copy GUID to static array. Set pointer. */ - if( lpGUID == NULL ) - { - pad->pad_lpGUID = NULL; - } - else - { - memcpy( &pad->pad_GUID, lpGUID, sizeof(GUID) ); - pad->pad_lpGUID = &pad->pad_GUID; - } - pad->pad_Info.sampleRates = pad->pad_SampleRates; /* Point to array. */ - /* Allocate room for descriptive name. */ - if( lpszDesc != NULL ) - { - int len = strlen(lpszDesc); - pad->pad_Info.name = (char *)malloc( len+1 ); - if( pad->pad_Info.name == NULL ) - { - sEnumerationError = paInsufficientMemory; - return FALSE; - } - memcpy( (void *) pad->pad_Info.name, lpszDesc, len+1 ); - } -#if SUPPORT_AUDIO_CAPTURE - if( isInput ) - { - /********** Input ******************************/ - DSCCAPS caps; - if( lpGUID == NULL ) sDefaultInputDeviceID = sDeviceIndex; - hr = DirectSoundCaptureCreate( lpGUID, &lpDirectSoundCapture, NULL ); - if( hr != DS_OK ) - { - pad->pad_Info.maxInputChannels = 0; - DBUG(("Cannot create Capture for %s. Result = 0x%x\n", lpszDesc, hr )); - } - else - { - /* Query device characteristics. */ - caps.dwSize = sizeof(caps); - IDirectSoundCapture_GetCaps( lpDirectSoundCapture, &caps ); - /* printf("caps.dwFormats = 0x%x\n", caps.dwFormats ); */ - pad->pad_Info.maxInputChannels = caps.dwChannels; - /* Determine sample rates from flags. */ - if( caps.dwChannels == 2 ) - { - int index = 0; - if( caps.dwFormats & WAVE_FORMAT_1S16) pad->pad_SampleRates[index++] = 11025.0; - if( caps.dwFormats & WAVE_FORMAT_2S16) pad->pad_SampleRates[index++] = 22050.0; - if( caps.dwFormats & WAVE_FORMAT_4S16) pad->pad_SampleRates[index++] = 44100.0; - pad->pad_Info.numSampleRates = index; - } - else if( caps.dwChannels == 1 ) - { - int index = 0; - if( caps.dwFormats & WAVE_FORMAT_1M16) pad->pad_SampleRates[index++] = 11025.0; - if( caps.dwFormats & WAVE_FORMAT_2M16) pad->pad_SampleRates[index++] = 22050.0; - if( caps.dwFormats & WAVE_FORMAT_4M16) pad->pad_SampleRates[index++] = 44100.0; - pad->pad_Info.numSampleRates = index; - } - else pad->pad_Info.numSampleRates = 0; - IDirectSoundCapture_Release( lpDirectSoundCapture ); - } - } - else -#endif /* SUPPORT_AUDIO_CAPTURE */ - - { - /********** Output ******************************/ - DSCAPS caps; - if( lpGUID == NULL ) sDefaultOutputDeviceID = sDeviceIndex; - /* Create interfaces for each object. */ - hr = DirectSoundCreate( lpGUID, &lpDirectSound, NULL ); - if( hr != DS_OK ) - { - pad->pad_Info.maxOutputChannels = 0; - DBUG(("Cannot create dsound for %s. Result = 0x%x\n", lpszDesc, hr )); - } - else - { - /* Query device characteristics. */ - caps.dwSize = sizeof(caps); - IDirectSound_GetCaps( lpDirectSound, &caps ); - pad->pad_Info.maxOutputChannels = ( caps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1; - /* Get sample rates. */ - pad->pad_SampleRates[0] = (double) caps.dwMinSecondarySampleRate; - pad->pad_SampleRates[1] = (double) caps.dwMaxSecondarySampleRate; - if( caps.dwFlags & DSCAPS_CONTINUOUSRATE ) pad->pad_Info.numSampleRates = -1; - else if( caps.dwMinSecondarySampleRate == caps.dwMaxSecondarySampleRate ) - { - if( caps.dwMinSecondarySampleRate == 0 ) - { - /* - ** On my Thinkpad 380Z, DirectSoundV6 returns min-max=0 !! - ** But it supports continuous sampling. - ** So fake range of rates, and hope it really supports it. - */ - pad->pad_SampleRates[0] = 11025.0f; - pad->pad_SampleRates[1] = 48000.0f; - pad->pad_Info.numSampleRates = -1; /* continuous range */ - - DBUG(("PA - Reported rates both zero. Setting to fake values for device #%d\n", sDeviceIndex )); - } - else - { - pad->pad_Info.numSampleRates = 1; - } - } - else if( (caps.dwMinSecondarySampleRate < 1000.0) && (caps.dwMaxSecondarySampleRate > 50000.0) ) - { - /* The EWS88MT drivers lie, lie, lie. The say they only support two rates, 100 & 100000. - ** But we know that they really support a range of rates! - ** So when we see a ridiculous set of rates, assume it is a range. - */ - pad->pad_Info.numSampleRates = -1; - DBUG(("PA - Sample rate range used instead of two odd values for device #%d\n", sDeviceIndex )); - } - else pad->pad_Info.numSampleRates = 2; - IDirectSound_Release( lpDirectSound ); - } - } - pad->pad_Info.nativeSampleFormats = paInt16; - sDeviceIndex++; - return( TRUE ); -} -/*************************************************************************/ -int Pa_CountDevices() -{ - if( sNumDevices <= 0 ) Pa_Initialize(); - return sNumDevices; -} -static internalPortAudioDevice *Pa_GetInternalDevice( PaDeviceID id ) -{ - if( (id < 0) || ( id >= Pa_CountDevices()) ) return NULL; - return &sDevices[id]; -} -/*************************************************************************/ -const PaDeviceInfo* Pa_GetDeviceInfo( PaDeviceID id ) -{ - internalPortAudioDevice *pad; - if( (id < 0) || ( id >= Pa_CountDevices()) ) return NULL; - pad = Pa_GetInternalDevice( id ); - return &pad->pad_Info ; -} -static PaError Pa_MaybeQueryDevices( void ) -{ - if( sNumDevices == 0 ) - { - return Pa_QueryDevices(); - } - return 0; -} -/************************************************************************* -** Returns recommended device ID. -** On the PC, the recommended device can be specified by the user by -** setting an environment variable. For example, to use device #1. -** -** set PA_RECOMMENDED_OUTPUT_DEVICE=1 -** -** The user should first determine the available device ID by using -** the supplied application "pa_devs". -*/ -#define PA_ENV_BUF_SIZE (32) -#define PA_REC_IN_DEV_ENV_NAME ("PA_RECOMMENDED_INPUT_DEVICE") -#define PA_REC_OUT_DEV_ENV_NAME ("PA_RECOMMENDED_OUTPUT_DEVICE") -static PaDeviceID PaHost_GetEnvDefaultDeviceID( char *envName ) -{ - DWORD hresult; - char envbuf[PA_ENV_BUF_SIZE]; - PaDeviceID recommendedID = paNoDevice; - /* Let user determine default device by setting environment variable. */ - hresult = GetEnvironmentVariable( envName, envbuf, PA_ENV_BUF_SIZE ); - if( (hresult > 0) && (hresult < PA_ENV_BUF_SIZE) ) - { - recommendedID = atoi( envbuf ); - } - return recommendedID; -} -PaDeviceID Pa_GetDefaultInputDeviceID( void ) -{ - PaError result; - result = PaHost_GetEnvDefaultDeviceID( PA_REC_IN_DEV_ENV_NAME ); - if( result < 0 ) - { - result = Pa_MaybeQueryDevices(); - if( result < 0 ) return result; - result = sDefaultInputDeviceID; - } - return result; -} -PaDeviceID Pa_GetDefaultOutputDeviceID( void ) -{ - PaError result; - result = PaHost_GetEnvDefaultDeviceID( PA_REC_OUT_DEV_ENV_NAME ); - if( result < 0 ) - { - result = Pa_MaybeQueryDevices(); - if( result < 0 ) return result; - result = sDefaultOutputDeviceID; - } - return result; -} -/********************************************************************** -** Make sure that we have queried the device capabilities. -*/ -PaError PaHost_Init( void ) -{ -#if PA_SIMULATE_UNDERFLOW - PRINT(("WARNING - Underflow Simulation Enabled - Expect a Big Glitch!!!\n")); -#endif - return Pa_MaybeQueryDevices(); -} -static PaError Pa_TimeSlice( internalPortAudioStream *past ) -{ - PaError result = 0; - long bytesEmpty = 0; - long bytesFilled = 0; - long bytesToXfer = 0; - long numChunks; - HRESULT hresult; - PaHostSoundControl *pahsc; - short *nativeBufPtr; - past->past_NumCallbacks += 1; - pahsc = (PaHostSoundControl *) past->past_DeviceData; - if( pahsc == NULL ) return paInternalError; - /* How much input data is available? */ -#if SUPPORT_AUDIO_CAPTURE - if( past->past_NumInputChannels > 0 ) - { - DSW_QueryInputFilled( &pahsc->pahsc_DSoundWrapper, &bytesFilled ); - bytesToXfer = bytesFilled; - } -#endif /* SUPPORT_AUDIO_CAPTURE */ - /* How much output room is available? */ - if( past->past_NumOutputChannels > 0 ) - { - DSW_QueryOutputSpace( &pahsc->pahsc_DSoundWrapper, &bytesEmpty ); - bytesToXfer = bytesEmpty; - } - AddTraceMessage( "bytesEmpty ", bytesEmpty ); - /* Choose smallest value if both are active. */ - if( (past->past_NumInputChannels > 0) && (past->past_NumOutputChannels > 0) ) - { - bytesToXfer = ( bytesFilled < bytesEmpty ) ? bytesFilled : bytesEmpty; - } - /* printf("bytesFilled = %d, bytesEmpty = %d, bytesToXfer = %d\n", - bytesFilled, bytesEmpty, bytesToXfer); - */ - /* Quantize to multiples of a buffer. */ - numChunks = bytesToXfer / pahsc->pahsc_BytesPerBuffer; - if( numChunks > (long)(past->past_NumUserBuffers/2) ) - { - numChunks = (long)past->past_NumUserBuffers/2; - } - else if( numChunks < 0 ) - { - numChunks = 0; - } - AddTraceMessage( "numChunks ", numChunks ); - nativeBufPtr = pahsc->pahsc_NativeBuffer; - if( numChunks > 0 ) - { - while( numChunks-- > 0 ) - { - /* Measure usage based on time to process one user buffer. */ - Pa_StartUsageCalculation( past ); -#if SUPPORT_AUDIO_CAPTURE - /* Get native data from DirectSound. */ - if( past->past_NumInputChannels > 0 ) - { - hresult = DSW_ReadBlock( &pahsc->pahsc_DSoundWrapper, (char *) nativeBufPtr, pahsc->pahsc_BytesPerBuffer ); - if( hresult < 0 ) - { - ERR_RPT(("DirectSound ReadBlock failed, hresult = 0x%x\n",hresult)); - sPaHostError = hresult; - break; - } - } -#endif /* SUPPORT_AUDIO_CAPTURE */ - /* Convert 16 bit native data to user data and call user routine. */ - result = Pa_CallConvertInt16( past, nativeBufPtr, nativeBufPtr ); - if( result != 0) break; - /* Pass native data to DirectSound. */ - if( past->past_NumOutputChannels > 0 ) - { - /* static short DEBUGHACK = 0; - DEBUGHACK += 0x0049; - nativeBufPtr[0] = DEBUGHACK; /* Make buzz to see if DirectSound still running. */ - hresult = DSW_WriteBlock( &pahsc->pahsc_DSoundWrapper, (char *) nativeBufPtr, pahsc->pahsc_BytesPerBuffer ); - if( hresult < 0 ) - { - ERR_RPT(("DirectSound WriteBlock failed, result = 0x%x\n",hresult)); - sPaHostError = hresult; - break; - } - } - Pa_EndUsageCalculation( past ); - } - } - return result; -} -/*******************************************************************/ -static void CALLBACK Pa_TimerCallback(UINT uID, UINT uMsg, DWORD dwUser, DWORD dw1, DWORD dw2) -{ - internalPortAudioStream *past; - PaHostSoundControl *pahsc; -#if PA_SIMULATE_UNDERFLOW - gUnderCallbackCounter++; - if( (gUnderCallbackCounter >= UNDER_START_GAP) && - (gUnderCallbackCounter <= UNDER_STOP_GAP) ) - { - if( gUnderCallbackCounter == UNDER_START_GAP) - { - AddTraceMessage("Begin stall: gUnderCallbackCounter =======", gUnderCallbackCounter ); - } - if( gUnderCallbackCounter == UNDER_STOP_GAP) - { - AddTraceMessage("End stall: gUnderCallbackCounter =======", gUnderCallbackCounter ); - } - return; - } -#endif - past = (internalPortAudioStream *) dwUser; - if( past == NULL ) return; - pahsc = (PaHostSoundControl *) past->past_DeviceData; - if( pahsc == NULL ) return; - if( !pahsc->pahsc_IfInsideCallback && past->past_IsActive ) - { - if( past->past_StopNow ) - { - past->past_IsActive = 0; - } - else if( past->past_StopSoon ) - { - DSoundWrapper *dsw = &pahsc->pahsc_DSoundWrapper; - if( past->past_NumOutputChannels > 0 ) - { - DSW_ZeroEmptySpace( dsw ); - AddTraceMessage("Pa_TimerCallback: waiting - written ", (int) dsw->dsw_FramesWritten ); - AddTraceMessage("Pa_TimerCallback: waiting - played ", (int) dsw->dsw_FramesPlayed ); - /* clear past_IsActive when all sound played */ - if( dsw->dsw_FramesPlayed >= past->past_FrameCount ) - { - past->past_IsActive = 0; - } - } - else - { - past->past_IsActive = 0; - } - } - else - { - pahsc->pahsc_IfInsideCallback = 1; - if( Pa_TimeSlice( past ) != 0) /* Call time slice independant of timing method. */ - { - past->past_StopSoon = 1; - } - pahsc->pahsc_IfInsideCallback = 0; - } - } -} -/*******************************************************************/ -PaError PaHost_OpenStream( internalPortAudioStream *past ) -{ - HRESULT hr; - PaError result = paNoError; - PaHostSoundControl *pahsc; - int numBytes, maxChannels; - unsigned int minNumBuffers; - internalPortAudioDevice *pad; - DSoundWrapper *dsw; - /* Allocate and initialize host data. */ - pahsc = (PaHostSoundControl *) PaHost_AllocateFastMemory(sizeof(PaHostSoundControl)); /* MEM */ - if( pahsc == NULL ) - { - result = paInsufficientMemory; - goto error; - } - memset( pahsc, 0, sizeof(PaHostSoundControl) ); - past->past_DeviceData = (void *) pahsc; - pahsc->pahsc_TimerID = 0; - dsw = &pahsc->pahsc_DSoundWrapper; - DSW_Init( dsw ); - /* Allocate native buffer. */ - maxChannels = ( past->past_NumOutputChannels > past->past_NumInputChannels ) ? - past->past_NumOutputChannels : past->past_NumInputChannels; - pahsc->pahsc_BytesPerBuffer = past->past_FramesPerUserBuffer * maxChannels * sizeof(short); - if( maxChannels > 0 ) - { - pahsc->pahsc_NativeBuffer = (short *) PaHost_AllocateFastMemory(pahsc->pahsc_BytesPerBuffer); /* MEM */ - if( pahsc->pahsc_NativeBuffer == NULL ) - { - result = paInsufficientMemory; - goto error; - } - } - else - { - result = paInvalidChannelCount; - goto error; - } - - DBUG(("PaHost_OpenStream: pahsc_MinFramesPerHostBuffer = %d\n", pahsc->pahsc_MinFramesPerHostBuffer )); - minNumBuffers = Pa_GetMinNumBuffers( past->past_FramesPerUserBuffer, past->past_SampleRate ); - past->past_NumUserBuffers = ( minNumBuffers > past->past_NumUserBuffers ) ? minNumBuffers : past->past_NumUserBuffers; - numBytes = pahsc->pahsc_BytesPerBuffer * past->past_NumUserBuffers; - if( numBytes < DSBSIZE_MIN ) - { - result = paBufferTooSmall; - goto error; - } - if( numBytes > DSBSIZE_MAX ) - { - result = paBufferTooBig; - goto error; - } - pahsc->pahsc_FramesPerDSBuffer = past->past_FramesPerUserBuffer * past->past_NumUserBuffers; - { - int msecLatency = (int) ((pahsc->pahsc_FramesPerDSBuffer * 1000) / past->past_SampleRate); - PRINT(("PortAudio on DirectSound - Latency = %d frames, %d msec\n", pahsc->pahsc_FramesPerDSBuffer, msecLatency )); - } - /* ------------------ OUTPUT */ - if( (past->past_OutputDeviceID >= 0) && (past->past_NumOutputChannels > 0) ) - { - DBUG(("PaHost_OpenStream: deviceID = 0x%x\n", past->past_OutputDeviceID)); - pad = Pa_GetInternalDevice( past->past_OutputDeviceID ); - hr = DirectSoundCreate( pad->pad_lpGUID, &dsw->dsw_pDirectSound, NULL ); - /* If this fails, then try each output device until we find one that works. */ - if( hr != DS_OK ) - { - int i; - ERR_RPT(("Creation of requested Audio Output device '%s' failed.\n", - ((pad->pad_lpGUID == NULL) ? "Default" : pad->pad_Info.name) )); - for( i=0; i<Pa_CountDevices(); i++ ) - { - pad = Pa_GetInternalDevice( i ); - if( pad->pad_Info.maxOutputChannels >= past->past_NumOutputChannels ) - { - DBUG(("Try device '%s' instead.\n", pad->pad_Info.name )); - hr = DirectSoundCreate( pad->pad_lpGUID, &dsw->dsw_pDirectSound, NULL ); - if( hr == DS_OK ) - { - ERR_RPT(("Using device '%s' instead.\n", pad->pad_Info.name )); - break; - } - } - } - } - if( hr != DS_OK ) - { - ERR_RPT(("PortAudio: DirectSoundCreate() failed!\n")); - result = paHostError; - sPaHostError = hr; - goto error; - } - hr = DSW_InitOutputBuffer( dsw, - (unsigned long) (past->past_SampleRate + 0.5), - past->past_NumOutputChannels, numBytes ); - DBUG(("DSW_InitOutputBuffer() returns %x\n", hr)); - if( hr != DS_OK ) - { - result = paHostError; - sPaHostError = hr; - goto error; - } - past->past_FrameCount = pahsc->pahsc_DSoundWrapper.dsw_FramesWritten; - } -#if SUPPORT_AUDIO_CAPTURE - /* ------------------ INPUT */ - if( (past->past_InputDeviceID >= 0) && (past->past_NumInputChannels > 0) ) - { - pad = Pa_GetInternalDevice( past->past_InputDeviceID ); - hr = DirectSoundCaptureCreate( pad->pad_lpGUID, &dsw->dsw_pDirectSoundCapture, NULL ); - /* If this fails, then try each input device until we find one that works. */ - if( hr != DS_OK ) - { - int i; - ERR_RPT(("Creation of requested Audio Capture device '%s' failed.\n", - ((pad->pad_lpGUID == NULL) ? "Default" : pad->pad_Info.name) )); - for( i=0; i<Pa_CountDevices(); i++ ) - { - pad = Pa_GetInternalDevice( i ); - if( pad->pad_Info.maxInputChannels >= past->past_NumInputChannels ) - { - PRINT(("Try device '%s' instead.\n", pad->pad_Info.name )); - hr = DirectSoundCaptureCreate( pad->pad_lpGUID, &dsw->dsw_pDirectSoundCapture, NULL ); - if( hr == DS_OK ) break; - } - } - } - if( hr != DS_OK ) - { - ERR_RPT(("PortAudio: DirectSoundCaptureCreate() failed!\n")); - result = paHostError; - sPaHostError = hr; - goto error; - } - hr = DSW_InitInputBuffer( dsw, - (unsigned long) (past->past_SampleRate + 0.5), - past->past_NumInputChannels, numBytes ); - DBUG(("DSW_InitInputBuffer() returns %x\n", hr)); - if( hr != DS_OK ) - { - ERR_RPT(("PortAudio: DSW_InitInputBuffer() returns %x\n", hr)); - result = paHostError; - sPaHostError = hr; - goto error; - } - } -#endif /* SUPPORT_AUDIO_CAPTURE */ - /* Calculate scalar used in CPULoad calculation. */ - { - LARGE_INTEGER frequency; - if( QueryPerformanceFrequency( &frequency ) == 0 ) - { - pahsc->pahsc_InverseTicksPerUserBuffer = 0.0; - } - else - { - pahsc->pahsc_InverseTicksPerUserBuffer = past->past_SampleRate / - ( (double)frequency.QuadPart * past->past_FramesPerUserBuffer ); - DBUG(("pahsc_InverseTicksPerUserBuffer = %g\n", pahsc->pahsc_InverseTicksPerUserBuffer )); - } - } - return result; -error: - PaHost_CloseStream( past ); - return result; -} -/*************************************************************************/ -PaError PaHost_StartOutput( internalPortAudioStream *past ) -{ - HRESULT hr; - PaHostSoundControl *pahsc; - PaError result = paNoError; - pahsc = (PaHostSoundControl *) past->past_DeviceData; - /* Give user callback a chance to pre-fill buffer. */ - result = Pa_TimeSlice( past ); - if( result != paNoError ) return result; // FIXME - what if finished? - hr = DSW_StartOutput( &pahsc->pahsc_DSoundWrapper ); - DBUG(("PaHost_StartOutput: DSW_StartOutput returned = 0x%X.\n", hr)); - if( hr != DS_OK ) - { - result = paHostError; - sPaHostError = hr; - goto error; - } -error: - return result; -} -/*************************************************************************/ -PaError PaHost_StartInput( internalPortAudioStream *past ) -{ - PaError result = paNoError; -#if SUPPORT_AUDIO_CAPTURE - HRESULT hr; - PaHostSoundControl *pahsc; - pahsc = (PaHostSoundControl *) past->past_DeviceData; - hr = DSW_StartInput( &pahsc->pahsc_DSoundWrapper ); - DBUG(("Pa_StartStream: DSW_StartInput returned = 0x%X.\n", hr)); - if( hr != DS_OK ) - { - result = paHostError; - sPaHostError = hr; - goto error; - } -error: -#endif /* SUPPORT_AUDIO_CAPTURE */ - return result; -} -/*************************************************************************/ -PaError PaHost_StartEngine( internalPortAudioStream *past ) -{ - PaHostSoundControl *pahsc; - PaError result = paNoError; - pahsc = (PaHostSoundControl *) past->past_DeviceData; - past->past_StopNow = 0; - past->past_StopSoon = 0; - past->past_IsActive = 1; - /* Create timer that will wake us up so we can fill the DSound buffer. */ - { - int msecPerBuffer; - int resolution; - int bufsPerInterrupt; - - DBUG(("PaHost_StartEngine: past_NumUserBuffers = %d\n", past->past_NumUserBuffers)); - /* Decide how often to wake up and fill the buffers. */ - if( past->past_NumUserBuffers == 2 ) - { - /* Generate two timer interrupts per user buffer. */ - msecPerBuffer = (500 * past->past_FramesPerUserBuffer) / (int) past->past_SampleRate; - } - else - { - if ( past->past_NumUserBuffers >= 16 ) bufsPerInterrupt = past->past_NumUserBuffers/8; - else if ( past->past_NumUserBuffers >= 8 ) bufsPerInterrupt = 2; - else bufsPerInterrupt = 1; - - msecPerBuffer = 1000 * (bufsPerInterrupt * past->past_FramesPerUserBuffer) / (int) past->past_SampleRate; - - DBUG(("PaHost_StartEngine: bufsPerInterrupt = %d\n", bufsPerInterrupt)); - } - - DBUG(("PaHost_StartEngine: msecPerBuffer = %d\n", msecPerBuffer)); - - if( msecPerBuffer < 10 ) msecPerBuffer = 10; - else if( msecPerBuffer > 100 ) msecPerBuffer = 100; - DBUG(("PaHost_StartEngine: clipped msecPerBuffer = %d\n", msecPerBuffer)); - - resolution = msecPerBuffer/4; - pahsc->pahsc_TimerID = timeSetEvent( msecPerBuffer, resolution, (LPTIMECALLBACK) Pa_TimerCallback, - (DWORD) past, TIME_PERIODIC ); - } - if( pahsc->pahsc_TimerID == 0 ) - { - past->past_IsActive = 0; - result = paHostError; - sPaHostError = 0; - goto error; - } -error: - return result; -} -/*************************************************************************/ -PaError PaHost_StopEngine( internalPortAudioStream *past, int abort ) -{ - int timeoutMsec; - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - if( pahsc == NULL ) return paNoError; - if( abort ) past->past_StopNow = 1; - past->past_StopSoon = 1; - /* Set timeout at 20% beyond maximum time we might wait. */ - timeoutMsec = (int) (1200.0 * pahsc->pahsc_FramesPerDSBuffer / past->past_SampleRate); - while( past->past_IsActive && (timeoutMsec > 0) ) - { - Sleep(10); - timeoutMsec -= 10; - } - if( pahsc->pahsc_TimerID != 0 ) - { - timeKillEvent(pahsc->pahsc_TimerID); /* Stop callback timer. */ - pahsc->pahsc_TimerID = 0; - } - return paNoError; -} -/*************************************************************************/ -PaError PaHost_StopInput( internalPortAudioStream *past, int abort ) -{ -#if SUPPORT_AUDIO_CAPTURE - HRESULT hr; - PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - if( pahsc == NULL ) return paNoError; - (void) abort; - hr = DSW_StopInput( &pahsc->pahsc_DSoundWrapper ); - DBUG(("DSW_StopInput() result is %x\n", hr)); -#endif /* SUPPORT_AUDIO_CAPTURE */ - return paNoError; -} -/*************************************************************************/ -PaError PaHost_StopOutput( internalPortAudioStream *past, int abort ) -{ - HRESULT hr; - PaHostSoundControl *pahsc; - pahsc = (PaHostSoundControl *) past->past_DeviceData; - if( pahsc == NULL ) return paNoError; - (void) abort; - hr = DSW_StopOutput( &pahsc->pahsc_DSoundWrapper ); - DBUG(("DSW_StopOutput() result is %x\n", hr)); - return paNoError; -} -/*******************************************************************/ -PaError PaHost_CloseStream( internalPortAudioStream *past ) -{ - PaHostSoundControl *pahsc; - if( past == NULL ) return paBadStreamPtr; - pahsc = (PaHostSoundControl *) past->past_DeviceData; - if( pahsc == NULL ) return paNoError; - DSW_Term( &pahsc->pahsc_DSoundWrapper ); - if( pahsc->pahsc_NativeBuffer ) - { - PaHost_FreeFastMemory( pahsc->pahsc_NativeBuffer, pahsc->pahsc_BytesPerBuffer ); /* MEM */ - pahsc->pahsc_NativeBuffer = NULL; - } - PaHost_FreeFastMemory( pahsc, sizeof(PaHostSoundControl) ); /* MEM */ - past->past_DeviceData = NULL; - return paNoError; -} - -/* Set minimal latency based on whether NT or Win95. - * NT has higher latency. - */ -static int PaHost_GetMinSystemLatency( void ) -{ - int minLatencyMsec; - /* Set minimal latency based on whether NT or other OS. - * NT has higher latency. - */ - OSVERSIONINFO osvi; - osvi.dwOSVersionInfoSize = sizeof( osvi ); - GetVersionEx( &osvi ); - DBUG(("PA - PlatformId = 0x%x\n", osvi.dwPlatformId )); - DBUG(("PA - MajorVersion = 0x%x\n", osvi.dwMajorVersion )); - DBUG(("PA - MinorVersion = 0x%x\n", osvi.dwMinorVersion )); - /* Check for NT */ - if( (osvi.dwMajorVersion == 4) && (osvi.dwPlatformId == 2) ) - { - minLatencyMsec = PA_WIN_NT_LATENCY; - } - else if(osvi.dwMajorVersion >= 5) - { - minLatencyMsec = PA_WIN_WDM_LATENCY; - } - else - { - minLatencyMsec = PA_WIN_9X_LATENCY; - } - return minLatencyMsec; -} - -/************************************************************************* -** Determine minimum number of buffers required for this host based -** on minimum latency. Latency can be optionally set by user by setting -** an environment variable. For example, to set latency to 200 msec, put: -** -** set PA_MIN_LATENCY_MSEC=200 -** -** in the AUTOEXEC.BAT file and reboot. -** If the environment variable is not set, then the latency will be determined -** based on the OS. Windows NT has higher latency than Win95. -*/ -#define PA_LATENCY_ENV_NAME ("PA_MIN_LATENCY_MSEC") -int Pa_GetMinNumBuffers( int framesPerBuffer, double sampleRate ) -{ - char envbuf[PA_ENV_BUF_SIZE]; - DWORD hresult; - int minLatencyMsec = 0; - double msecPerBuffer = (1000.0 * framesPerBuffer) / sampleRate; - int minBuffers; - /* Let user determine minimal latency by setting environment variable. */ - hresult = GetEnvironmentVariable( PA_LATENCY_ENV_NAME, envbuf, PA_ENV_BUF_SIZE ); - if( (hresult > 0) && (hresult < PA_ENV_BUF_SIZE) ) - { - minLatencyMsec = atoi( envbuf ); - } - else - { - minLatencyMsec = PaHost_GetMinSystemLatency(); -#if PA_USE_HIGH_LATENCY - PRINT(("PA - Minimum Latency set to %d msec!\n", minLatencyMsec )); -#endif - - } - minBuffers = (int) (1.0 + ((double)minLatencyMsec / msecPerBuffer)); - if( minBuffers < 2 ) minBuffers = 2; - return minBuffers; -} -/*************************************************************************/ -PaError PaHost_Term( void ) -{ - int i; - /* Free names allocated during enumeration. */ - for( i=0; i<sNumDevices; i++ ) - { - if( sDevices[i].pad_Info.name != NULL ) - { - free( (void *) sDevices[i].pad_Info.name ); - sDevices[i].pad_Info.name = NULL; - } - } - if( sDevices != NULL ) - { - PaHost_FreeFastMemory( sDevices, sNumDevices * sizeof(internalPortAudioDevice) ); /* MEM */ - sDevices = NULL; - sNumDevices = 0; - } - return 0; -} -void Pa_Sleep( long msec ) -{ - Sleep( msec ); -} -/************************************************************************* - * Allocate memory that can be accessed in real-time. - * This may need to be held in physical memory so that it is not - * paged to virtual memory. - * This call MUST be balanced with a call to PaHost_FreeFastMemory(). - * Memory will be set to zero. - */ -void *PaHost_AllocateFastMemory( long numBytes ) -{ - void *addr = GlobalAlloc( GPTR, numBytes ); /* FIXME - do we need physical memory? Use VirtualLock() */ /* MEM */ - return addr; -} -/************************************************************************* - * Free memory that could be accessed in real-time. - * This call MUST be balanced with a call to PaHost_AllocateFastMemory(). - */ -void PaHost_FreeFastMemory( void *addr, long numBytes ) -{ - if( addr != NULL ) GlobalFree( addr ); /* MEM */ -} -/***********************************************************************/ -PaError PaHost_StreamActive( internalPortAudioStream *past ) -{ - PaHostSoundControl *pahsc; - if( past == NULL ) return paBadStreamPtr; - pahsc = (PaHostSoundControl *) past->past_DeviceData; - if( pahsc == NULL ) return paInternalError; - return (PaError) (past->past_IsActive); -} -/*************************************************************************/ -PaTimestamp Pa_StreamTime( PortAudioStream *stream ) -{ - DSoundWrapper *dsw; - internalPortAudioStream *past = (internalPortAudioStream *) stream; - PaHostSoundControl *pahsc; - if( past == NULL ) return paBadStreamPtr; - pahsc = (PaHostSoundControl *) past->past_DeviceData; - dsw = &pahsc->pahsc_DSoundWrapper; - return dsw->dsw_FramesPlayed; -} diff --git a/pd/portaudio_v18/pa_win_ds/portaudio.def b/pd/portaudio_v18/pa_win_ds/portaudio.def deleted file mode 100644 index 8012b99e..00000000 --- a/pd/portaudio_v18/pa_win_ds/portaudio.def +++ /dev/null @@ -1,28 +0,0 @@ -LIBRARY PortAudio -DESCRIPTION 'PortAudio Portable interface to audio HW' - -EXPORTS - ; Explicit exports can go here - Pa_Initialize @1 - Pa_Terminate @2 - Pa_GetHostError @3 - Pa_GetErrorText @4 - Pa_CountDevices @5 - Pa_GetDefaultInputDeviceID @6 - Pa_GetDefaultOutputDeviceID @7 - Pa_GetDeviceInfo @8 - Pa_OpenStream @9 - Pa_OpenDefaultStream @10 - Pa_CloseStream @11 - Pa_StartStream @12 - Pa_StopStream @13 - Pa_StreamActive @14 - Pa_StreamTime @15 - Pa_GetCPULoad @16 - Pa_GetMinNumBuffers @17 - Pa_Sleep @18 - - ;123456789012345678901234567890123456 - ;000000000111111111122222222223333333 - - diff --git a/pd/portaudio_v18/pa_win_wmme/Makefile.cygwin b/pd/portaudio_v18/pa_win_wmme/Makefile.cygwin deleted file mode 100644 index 5cb4acef..00000000 --- a/pd/portaudio_v18/pa_win_wmme/Makefile.cygwin +++ /dev/null @@ -1,34 +0,0 @@ - -# Makefile for PortAudio on cygwin -# Contributed by Bill Eldridge on 6/13/2001 - -ARCH= pa_win_wmme - -TESTS:= $(wildcard pa_tests/pa*.c pa_tests/debug*.c) - -.c.o: - -gcc -c -I./pa_common $< -o $*.o - -gcc $*.o -o $*.exe -L/usr/local/lib -L$(ARCH) -lportaudio.dll -lwinmm - -all: sharedlib tests - -sharedlib: ./pa_common/pa_lib.c - gcc -c -I./pa_common pa_common/pa_lib.c -o pa_common/pa_lib.o - gcc -c -I./pa_common pa_win_wmme/pa_win_wmme.c -o pa_win_wmme/pa_win_wmme.o - dlltool --export-all --output-def pa_win_wmme/pa_lib.def pa_common/pa_lib.o pa_win_wmme/pa_win_wmme.o - gcc -shared -Wl,--enable-auto-image-base -o pa_win_wmme/portaudio.dll -Wl,--out-implib=pa_win_wmme/libportaudio.dll.a pa_win_wmme/pa_lib.def pa_common/pa_lib.o pa_win_wmme/pa_win_wmme.o -L/usr/lib/w32api -lwinmm - cp pa_win_wmme/portaudio.dll /usr/local/bin - -tests: $(TESTS:.c=.o) - -sine: - gcc -c -I./pa_common pa_tests/patest_sine.c -o pa_tests/patest_sine.o - gcc pa_tests/patest_sine.o -o pa_tests/patest_sine.exe -L/usr/local/lib -lportaudio.dll -lwinmm - -clean: - -rm ./pa_tests/*.exe - -nothing: - gcc pa_tests/patest_sine.o -L/usr/lib/w32api -L./pa_win_wmme -lportaudio.dll -lwinmm - - diff --git a/pd/portaudio_v18/pa_win_wmme/pa_win_wmme.c b/pd/portaudio_v18/pa_win_wmme/pa_win_wmme.c deleted file mode 100644 index 9dc03826..00000000 --- a/pd/portaudio_v18/pa_win_wmme/pa_win_wmme.c +++ /dev/null @@ -1,1714 +0,0 @@ -/* - * $Id: pa_win_wmme.c,v 1.6.4.3 2003/04/28 17:43:48 philburk Exp $ - * pa_win_wmme.c - * Implementation of PortAudio for Windows MultiMedia Extensions (WMME) - * - * PortAudio Portable Real-Time Audio Library - * Latest Version at: http://www.portaudio.com - * - * Authors: Ross Bencina and Phil Burk - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -/* - All memory allocations and frees are marked with MEM for quick review. -*/ - -/* Modification History: - PLB = Phil Burk - JM = Julien Maillard - RDB = Ross Bencina - PLB20010402 - sDevicePtrs now allocates based on sizeof(pointer) - PLB20010413 - check for excessive numbers of channels - PLB20010422 - apply Mike Berry's changes for CodeWarrior on PC - including condition including of memory.h, - and explicit typecasting on memory allocation - PLB20010802 - use GlobalAlloc for sDevicesPtr instead of PaHost_AllocFastMemory - PLB20010816 - pass process instead of thread to SetPriorityClass() - PLB20010927 - use number of frames instead of real-time for CPULoad calculation. - JM20020118 - prevent hung thread when buffers underflow. - PLB20020321 - detect Win XP versus NT, 9x; fix DBUG typo; removed init of CurrentCount - RDB20020411 - various renaming cleanups, factored streamData alloc and cpu usage init - RDB20020417 - stopped counting WAVE_MAPPER when there were no real devices - refactoring, renaming and fixed a few edge case bugs - PLB20020612 - added 8000.0 Hz to custom sampling rates array -*/ -#pragma warning (disable: 4115) -#include <stdio.h> -#include <stdlib.h> -#include <math.h> -#include <windows.h> -#include <mmsystem.h> -#include <process.h> -/* PLB20010422 - "memory.h" doesn't work on CodeWarrior for PC. Thanks Mike Berry for the mod. */ -#ifndef __MWERKS__ -#include <malloc.h> -#include <memory.h> -#endif /* __MWERKS__ */ -#include "portaudio.h" -#include "pa_host.h" -#include "pa_trace.h" - -/************************************************* Constants ********/ -#define PA_TRACK_MEMORY (0) - -#define PA_USE_TIMER_CALLBACK (0) /* Select between two options for background task. 0=thread, 1=timer */ -/* Switches for debugging. */ -#define PA_SIMULATE_UNDERFLOW (0) /* Set to one to force an underflow of the output buffer. */ - -/* To trace program, enable TRACE_REALTIME_EVENTS in pa_trace.h */ -#define PA_TRACE_RUN (0) -#define PA_TRACE_START_STOP (1) - -#define PA_USE_HIGH_LATENCY (0) /* For debugging glitches. */ - -#if PA_USE_HIGH_LATENCY - #define PA_MIN_MSEC_PER_HOST_BUFFER (100) - #define PA_MAX_MSEC_PER_HOST_BUFFER (300) /* Do not exceed unless user buffer exceeds */ - #define PA_MIN_NUM_HOST_BUFFERS (4) - #define PA_MAX_NUM_HOST_BUFFERS (16) /* OK to exceed if necessary */ - #define PA_WIN_9X_LATENCY (400) -#else - #define PA_MIN_MSEC_PER_HOST_BUFFER (10) - #define PA_MAX_MSEC_PER_HOST_BUFFER (100) /* Do not exceed unless user buffer exceeds */ - #define PA_MIN_NUM_HOST_BUFFERS (3) - #define PA_MAX_NUM_HOST_BUFFERS (16) /* OK to exceed if necessary */ - #define PA_WIN_9X_LATENCY (200) -#endif -#define MIN_TIMEOUT_MSEC (1000) - -/* -** Use higher latency for NT because it is even worse at real-time -** operation than Win9x. -*/ -#define PA_WIN_NT_LATENCY (PA_WIN_9X_LATENCY * 2) -#define PA_WIN_WDM_LATENCY (PA_WIN_9X_LATENCY) - -#if PA_SIMULATE_UNDERFLOW -static gUnderCallbackCounter = 0; -#define UNDER_SLEEP_AT (40) -#define UNDER_SLEEP_FOR (500) -#endif - -#define PRINT(x) { printf x; fflush(stdout); } -#define ERR_RPT(x) PRINT(x) -#define DBUG(x) /* PRINT(x) */ -#define DBUGX(x) /* PRINT(x) */ -/************************************************* Definitions ********/ -/************************************************************** - * Structure for internal host specific stream data. - * This is allocated on a per stream basis. - */ -typedef struct PaWMMEStreamData -{ - /* Input -------------- */ - HWAVEIN hWaveIn; - WAVEHDR *inputBuffers; - int currentInputBuffer; - int bytesPerHostInputBuffer; - int bytesPerUserInputBuffer; /* native buffer size in bytes */ - /* Output -------------- */ - HWAVEOUT hWaveOut; - WAVEHDR *outputBuffers; - int currentOutputBuffer; - int bytesPerHostOutputBuffer; - int bytesPerUserOutputBuffer; /* native buffer size in bytes */ - /* Run Time -------------- */ - PaTimestamp framesPlayed; - long lastPosition; /* used to track frames played. */ - /* For measuring CPU utilization. */ - LARGE_INTEGER entryCount; - double inverseTicksPerHostBuffer; - /* Init Time -------------- */ - int numHostBuffers; - int framesPerHostBuffer; - int userBuffersPerHostBuffer; - CRITICAL_SECTION streamLock; /* Mutext to prevent threads from colliding. */ - INT streamLockInited; -#if PA_USE_TIMER_CALLBACK - BOOL ifInsideCallback; /* Test for reentrancy. */ - MMRESULT timerID; -#else - HANDLE abortEvent; - int abortEventInited; - HANDLE bufferEvent; - int bufferEventInited; - HANDLE engineThread; - DWORD engineThreadID; -#endif -} -PaWMMEStreamData; -/************************************************* Shared Data ********/ -/* FIXME - put Mutex around this shared data. */ -static int sNumInputDevices = 0; -static int sNumOutputDevices = 0; -static int sNumDevices = 0; -static PaDeviceInfo **sDevicePtrs = NULL; -static int sDefaultInputDeviceID = paNoDevice; -static int sDefaultOutputDeviceID = paNoDevice; -static int sPaHostError = 0; -static const char sMapperSuffixInput[] = " - Input"; -static const char sMapperSuffixOutput[] = " - Output"; - -#if PA_TRACK_MEMORY -static int sNumAllocations = 0; -#endif - -/************************************************* Macros ********/ -/* Convert external PA ID to an internal ID that includes WAVE_MAPPER */ -#define PaDeviceIdToWinId(id) (((id) < sNumInputDevices) ? (id - 1) : (id - sNumInputDevices - 1)) -/************************************************* Prototypes **********/ - -void Pa_InitializeNumDevices( void ); -PaError Pa_AllocateDevicePtrs( void ); - -static void CALLBACK Pa_TimerCallback(UINT uID, UINT uMsg, - DWORD dwUser, DWORD dw1, DWORD dw2); -PaError PaHost_GetTotalBufferFrames( internalPortAudioStream *past ); -static PaError PaHost_UpdateStreamTime( PaWMMEStreamData *wmmeStreamData ); -static PaError PaHost_BackgroundManager( internalPortAudioStream *past ); - -static void *PaHost_AllocateTrackedMemory( long numBytes ); -static void PaHost_FreeTrackedMemory( void *addr ); - -/*******************************************************************/ -static PaError PaHost_AllocateWMMEStreamData( internalPortAudioStream *stream ) -{ - PaError result = paNoError; - PaWMMEStreamData *wmmeStreamData; - - wmmeStreamData = (PaWMMEStreamData *) PaHost_AllocateFastMemory(sizeof(PaWMMEStreamData)); /* MEM */ - if( wmmeStreamData == NULL ) - { - result = paInsufficientMemory; - goto error; - } - memset( wmmeStreamData, 0, sizeof(PaWMMEStreamData) ); - stream->past_DeviceData = (void *) wmmeStreamData; - - return result; - -error: - return result; -} - -/*******************************************************************/ -static void PaHost_FreeWMMEStreamData( internalPortAudioStream *internalStream ) -{ - PaWMMEStreamData *wmmeStreamData = (PaWMMEStreamData *) internalStream->past_DeviceData; - - PaHost_FreeFastMemory( wmmeStreamData, sizeof(PaWMMEStreamData) ); /* MEM */ - internalStream->past_DeviceData = NULL; -} -/*************************************************************************/ -static PaWMMEStreamData* PaHost_GetWMMEStreamData( internalPortAudioStream* internalStream ) -{ - PaWMMEStreamData *result = NULL; - - if( internalStream != NULL ) - { - result = (PaWMMEStreamData *) internalStream->past_DeviceData; - } - return result; -} -/********************************* BEGIN CPU UTILIZATION MEASUREMENT ****/ -/* FIXME: the cpu usage code should be factored out into a common module */ -static void Pa_InitializeCpuUsageScalar( internalPortAudioStream *stream ) -{ - PaWMMEStreamData *wmmeStreamData = (PaWMMEStreamData *) stream->past_DeviceData; - - LARGE_INTEGER frequency; - if( QueryPerformanceFrequency( &frequency ) == 0 ) - { - wmmeStreamData->inverseTicksPerHostBuffer = 0.0; - } - else - { - wmmeStreamData->inverseTicksPerHostBuffer = stream->past_SampleRate / - ( (double)frequency.QuadPart * stream->past_FramesPerUserBuffer * wmmeStreamData->userBuffersPerHostBuffer ); - DBUG(("inverseTicksPerHostBuffer = %g\n", wmmeStreamData->inverseTicksPerHostBuffer )); - } -} -static void Pa_StartUsageCalculation( internalPortAudioStream *stream ) -{ - PaWMMEStreamData *wmmeStreamData = (PaWMMEStreamData *) stream->past_DeviceData; - - if( wmmeStreamData == NULL ) return; - /* Query system timer for usage analysis and to prevent overuse of CPU. */ - QueryPerformanceCounter( &wmmeStreamData->entryCount ); -} -static void Pa_EndUsageCalculation( internalPortAudioStream *stream ) -{ - LARGE_INTEGER CurrentCount; - PaWMMEStreamData *wmmeStreamData = (PaWMMEStreamData *) stream->past_DeviceData; - - if( wmmeStreamData == NULL ) return; - /* - * Measure CPU utilization during this callback. Note that this calculation - * assumes that we had the processor the whole time. - */ -#define LOWPASS_COEFFICIENT_0 (0.9) -#define LOWPASS_COEFFICIENT_1 (0.99999 - LOWPASS_COEFFICIENT_0) - if( QueryPerformanceCounter( &CurrentCount ) ) - { - LONGLONG InsideCount = CurrentCount.QuadPart - wmmeStreamData->entryCount.QuadPart; - double newUsage = InsideCount * wmmeStreamData->inverseTicksPerHostBuffer; - stream->past_Usage = (LOWPASS_COEFFICIENT_0 * stream->past_Usage) + - (LOWPASS_COEFFICIENT_1 * newUsage); - } -} -/****************************************** END CPU UTILIZATION *******/ - -static void Pa_InitializeNumDevices( void ) -{ - sNumInputDevices = waveInGetNumDevs(); - if( sNumInputDevices > 0 ) - { - sNumInputDevices += 1; /* add one extra for the WAVE_MAPPER */ - sDefaultInputDeviceID = 0; - } - else - { - sDefaultInputDeviceID = paNoDevice; - } - - sNumOutputDevices = waveOutGetNumDevs(); - if( sNumOutputDevices > 0 ) - { - sNumOutputDevices += 1; /* add one extra for the WAVE_MAPPER */ - sDefaultOutputDeviceID = sNumInputDevices; - } - else - { - sDefaultOutputDeviceID = paNoDevice; - } - - sNumDevices = sNumInputDevices + sNumOutputDevices; -} - -static PaError Pa_AllocateDevicePtrs( void ) -{ - int numBytes; - int i; - - /* Allocate structures to hold device info. */ - /* PLB20010402 - was allocating too much memory. */ - /* numBytes = sNumDevices * sizeof(PaDeviceInfo); // PLB20010402 */ - - if( sNumDevices > 0 ) - { - numBytes = sNumDevices * sizeof(PaDeviceInfo *); /* PLB20010402 */ - sDevicePtrs = (PaDeviceInfo **) PaHost_AllocateTrackedMemory( numBytes ); /* MEM */ - if( sDevicePtrs == NULL ) return paInsufficientMemory; - - for( i = 0; i < sNumDevices; i++ ) - sDevicePtrs[i] = NULL; /* RDB20020417 explicitly set each ptr to NULL */ - } - else - { - sDevicePtrs = NULL; - } - - return paNoError; -} -/*************************************************************************/ -long Pa_GetHostError() -{ - return sPaHostError; -} -/*************************************************************************/ -int Pa_CountDevices() -{ - if( PaHost_IsInitialized() ) - return sNumDevices; - else - return 0; -} -/************************************************************************* - * If a PaDeviceInfo structure has not already been created, - * then allocate one and fill it in for the selected device. - * - * We create one extra input and one extra output device for the WAVE_MAPPER. - * [Does anyone know how to query the default device and get its name?] - */ -const PaDeviceInfo* Pa_GetDeviceInfo( PaDeviceID id ) -{ -#define NUM_STANDARDSAMPLINGRATES 3 /* 11025, 22050, 44100 */ - static DWORD customSamplingRates[] = { 8000, 32000, 48000, 64000, 88200, 96000 }; -#define NUM_CUSTOMSAMPLINGRATES (sizeof(customSamplingRates)/sizeof(DWORD)) -#define MAX_NUMSAMPLINGRATES (NUM_STANDARDSAMPLINGRATES+NUM_CUSTOMSAMPLINGRATES) - - PaDeviceInfo *deviceInfo; - double *sampleRates; /* non-const ptr */ - int i; - char *s; - - DBUG(( "Pa_GetDeviceInfo( %d )\n", id )); - if( id < 0 || id >= sNumDevices ) - return NULL; - if( sDevicePtrs[ id ] != NULL ) - { - return sDevicePtrs[ id ]; - } - deviceInfo = (PaDeviceInfo *)PaHost_AllocateTrackedMemory( sizeof(PaDeviceInfo) ); /* MEM */ - if( deviceInfo == NULL ) return NULL; - deviceInfo->structVersion = 1; - deviceInfo->maxInputChannels = 0; - deviceInfo->maxOutputChannels = 0; - deviceInfo->numSampleRates = 0; - sampleRates = (double*)PaHost_AllocateTrackedMemory( MAX_NUMSAMPLINGRATES * sizeof(double) ); /* MEM */ - deviceInfo->sampleRates = sampleRates; - deviceInfo->nativeSampleFormats = paInt16; /* should query for higher bit depths below */ - if( id < sNumInputDevices ) - { - /* input device */ - int inputMmID = PaDeviceIdToWinId(id); - WAVEINCAPS wic; - if( waveInGetDevCaps( inputMmID, &wic, sizeof( WAVEINCAPS ) ) != MMSYSERR_NOERROR ) - goto error; - - /* Append I/O suffix to WAVE_MAPPER device. */ - if( inputMmID == WAVE_MAPPER ) - { - s = (char *) PaHost_AllocateTrackedMemory( strlen( wic.szPname ) + 1 + sizeof(sMapperSuffixInput) ); /* MEM */ - strcpy( s, wic.szPname ); - strcat( s, sMapperSuffixInput ); - } - else - { - s = (char *) PaHost_AllocateTrackedMemory( strlen( wic.szPname ) + 1 ); /* MEM */ - strcpy( s, wic.szPname ); - } - deviceInfo->name = s; - deviceInfo->maxInputChannels = wic.wChannels; - DBUG(( "Pa_GetDeviceInfo: input %s, maxChannels = %d\n", deviceInfo->name, deviceInfo->maxInputChannels )); - /* Sometimes a device can return a rediculously large number of channels. - * This happened with an SBLive card on a Windows ME box. - * If that happens, then force it to 2 channels. PLB20010413 - */ - if( (deviceInfo->maxInputChannels < 1) || (deviceInfo->maxInputChannels > 256) ) - { - ERR_RPT(("Pa_GetDeviceInfo: Num input channels reported as %d! Changed to 2.\n", deviceInfo->maxOutputChannels )); - deviceInfo->maxInputChannels = 2; - } - /* Add a sample rate to the list if we can do stereo 16 bit at that rate - * based on the format flags. */ - if( wic.dwFormats & WAVE_FORMAT_1M16 ||wic.dwFormats & WAVE_FORMAT_1S16 ) - sampleRates[ deviceInfo->numSampleRates++ ] = 11025.; - if( wic.dwFormats & WAVE_FORMAT_2M16 ||wic.dwFormats & WAVE_FORMAT_2S16 ) - sampleRates[ deviceInfo->numSampleRates++ ] = 22050.; - if( wic.dwFormats & WAVE_FORMAT_4M16 ||wic.dwFormats & WAVE_FORMAT_4S16 ) - sampleRates[ deviceInfo->numSampleRates++ ] = 44100.; - /* Add a sample rate to the list if we can do stereo 16 bit at that rate - * based on opening the device successfully. */ - for( i=0; i < NUM_CUSTOMSAMPLINGRATES; i++ ) - { - WAVEFORMATEX wfx; - wfx.wFormatTag = WAVE_FORMAT_PCM; - wfx.nSamplesPerSec = customSamplingRates[i]; - wfx.wBitsPerSample = 16; - wfx.cbSize = 0; /* ignored */ - wfx.nChannels = (WORD)deviceInfo->maxInputChannels; - wfx.nAvgBytesPerSec = wfx.nChannels * wfx.nSamplesPerSec * sizeof(short); - wfx.nBlockAlign = (WORD)(wfx.nChannels * sizeof(short)); - if( waveInOpen( NULL, inputMmID, &wfx, 0, 0, WAVE_FORMAT_QUERY ) == MMSYSERR_NOERROR ) - { - sampleRates[ deviceInfo->numSampleRates++ ] = customSamplingRates[i]; - } - } - - } - else if( id - sNumInputDevices < sNumOutputDevices ) - { - /* output device */ - int outputMmID = PaDeviceIdToWinId(id); - WAVEOUTCAPS woc; - if( waveOutGetDevCaps( outputMmID, &woc, sizeof( WAVEOUTCAPS ) ) != MMSYSERR_NOERROR ) - goto error; - /* Append I/O suffix to WAVE_MAPPER device. */ - if( outputMmID == WAVE_MAPPER ) - { - s = (char *) PaHost_AllocateTrackedMemory( strlen( woc.szPname ) + 1 + sizeof(sMapperSuffixOutput) ); /* MEM */ - strcpy( s, woc.szPname ); - strcat( s, sMapperSuffixOutput ); - } - else - { - s = (char *) PaHost_AllocateTrackedMemory( strlen( woc.szPname ) + 1 ); /* MEM */ - strcpy( s, woc.szPname ); - } - deviceInfo->name = s; - deviceInfo->maxOutputChannels = woc.wChannels; - DBUG(( "Pa_GetDeviceInfo: output %s, maxChannels = %d\n", deviceInfo->name, deviceInfo->maxOutputChannels )); - /* Sometimes a device can return a rediculously large number of channels. - * This happened with an SBLive card on a Windows ME box. - * It also happens on Win XP! - */ - if( (deviceInfo->maxOutputChannels < 1) || (deviceInfo->maxOutputChannels > 256) ) - { -#if 1 - deviceInfo->maxOutputChannels = 2; -#else - /* If channel max is goofy, then query for max channels. PLB20020228 - * This doesn't seem to help. Disable code for now. Remove it later. - */ - ERR_RPT(("Pa_GetDeviceInfo: Num output channels reported as %d!", deviceInfo->maxOutputChannels )); - deviceInfo->maxOutputChannels = 0; - /* Attempt to find the correct maximum by querying the device. */ - for( i=2; i<16; i += 2 ) - { - WAVEFORMATEX wfx; - wfx.wFormatTag = WAVE_FORMAT_PCM; - wfx.nSamplesPerSec = 44100; - wfx.wBitsPerSample = 16; - wfx.cbSize = 0; /* ignored */ - wfx.nChannels = (WORD) i; - wfx.nAvgBytesPerSec = wfx.nChannels * wfx.nSamplesPerSec * sizeof(short); - wfx.nBlockAlign = (WORD)(wfx.nChannels * sizeof(short)); - if( waveOutOpen( NULL, outputMmID, &wfx, 0, 0, WAVE_FORMAT_QUERY ) == MMSYSERR_NOERROR ) - { - deviceInfo->maxOutputChannels = i; - } - else - { - break; - } - } -#endif - ERR_RPT((" Changed to %d.\n", deviceInfo->maxOutputChannels )); - } - - /* Add a sample rate to the list if we can do stereo 16 bit at that rate - * based on the format flags. */ - if( woc.dwFormats & WAVE_FORMAT_1M16 ||woc.dwFormats & WAVE_FORMAT_1S16 ) - sampleRates[ deviceInfo->numSampleRates++ ] = 11025.; - if( woc.dwFormats & WAVE_FORMAT_2M16 ||woc.dwFormats & WAVE_FORMAT_2S16 ) - sampleRates[ deviceInfo->numSampleRates++ ] = 22050.; - if( woc.dwFormats & WAVE_FORMAT_4M16 ||woc.dwFormats & WAVE_FORMAT_4S16 ) - sampleRates[ deviceInfo->numSampleRates++ ] = 44100.; - - /* Add a sample rate to the list if we can do stereo 16 bit at that rate - * based on opening the device successfully. */ - for( i=0; i < NUM_CUSTOMSAMPLINGRATES; i++ ) - { - WAVEFORMATEX wfx; - wfx.wFormatTag = WAVE_FORMAT_PCM; - wfx.nSamplesPerSec = customSamplingRates[i]; - wfx.wBitsPerSample = 16; - wfx.cbSize = 0; /* ignored */ - wfx.nChannels = (WORD)deviceInfo->maxOutputChannels; - wfx.nAvgBytesPerSec = wfx.nChannels * wfx.nSamplesPerSec * sizeof(short); - wfx.nBlockAlign = (WORD)(wfx.nChannels * sizeof(short)); - DBUG(( "Pa_GetDeviceInfo: waveOutOpen( ... WAVE_FORMAT_QUERY at SR = %d\n", customSamplingRates[i] )); - if( waveOutOpen( NULL, outputMmID, &wfx, 0, 0, WAVE_FORMAT_QUERY ) == MMSYSERR_NOERROR ) - { - sampleRates[ deviceInfo->numSampleRates++ ] = customSamplingRates[i]; - } - } - } - DBUG(( "Pa_GetDeviceInfo: done.\n" )); - sDevicePtrs[ id ] = deviceInfo; - return deviceInfo; - -error: - PaHost_FreeTrackedMemory( sampleRates ); /* MEM */ - PaHost_FreeTrackedMemory( deviceInfo ); /* MEM */ - - return NULL; -} -/************************************************************************* - * Returns recommended device ID. - * On the PC, the recommended device can be specified by the user by - * setting an environment variable. For example, to use device #1. - * - * set PA_RECOMMENDED_OUTPUT_DEVICE=1 - * - * The user should first determine the available device ID by using - * the supplied application "pa_devs". - */ -#define PA_ENV_BUF_SIZE (32) -#define PA_REC_IN_DEV_ENV_NAME ("PA_RECOMMENDED_INPUT_DEVICE") -#define PA_REC_OUT_DEV_ENV_NAME ("PA_RECOMMENDED_OUTPUT_DEVICE") -static PaDeviceID PaHost_GetEnvDefaultDeviceID( char *envName ) -{ - DWORD hresult; - char envbuf[PA_ENV_BUF_SIZE]; - PaDeviceID recommendedID = paNoDevice; - - /* Let user determine default device by setting environment variable. */ - hresult = GetEnvironmentVariable( envName, envbuf, PA_ENV_BUF_SIZE ); - if( (hresult > 0) && (hresult < PA_ENV_BUF_SIZE) ) - { - recommendedID = atoi( envbuf ); - } - return recommendedID; -} -/********************************************************************** - * Check for environment variable, else query devices and use result. - */ -PaDeviceID Pa_GetDefaultInputDeviceID( void ) -{ - PaDeviceID result; - - result = PaHost_GetEnvDefaultDeviceID( PA_REC_IN_DEV_ENV_NAME ); - if( result == paNoDevice || result < 0 || result >= sNumInputDevices ) - { - result = sDefaultInputDeviceID; - } - return result; -} -PaDeviceID Pa_GetDefaultOutputDeviceID( void ) -{ - PaDeviceID result; - - result = PaHost_GetEnvDefaultDeviceID( PA_REC_OUT_DEV_ENV_NAME ); - if( result == paNoDevice || result < sNumInputDevices || result >= sNumDevices ) - { - result = sDefaultOutputDeviceID; - } - return result; -} -/********************************************************************** - * Initialize Host dependant part of API. - */ -PaError PaHost_Init( void ) -{ - -#if PA_TRACK_MEMORY - PRINT(("PaHost_Init: sNumAllocations = %d\n", sNumAllocations )); -#endif - -#if PA_SIMULATE_UNDERFLOW - PRINT(("WARNING - Underflow Simulation Enabled - Expect a Big Glitch!!!\n")); -#endif - - - Pa_InitializeNumDevices(); - - return Pa_AllocateDevicePtrs(); -} - -/********************************************************************** - * Check WAVE buffers to see if they are done. - * Fill any available output buffers and use any available - * input buffers by calling user callback. - * - * This routine will loop until: - * user callback returns !=0 OR - * all output buffers are filled OR - * past->past_StopSoon is set OR - * an error occurs when calling WMME. - * - * Returns >0 when user requests a stop, <0 on error. - * - */ -static PaError Pa_TimeSlice( internalPortAudioStream *stream ) -{ - PaError result = paNoError; - MMRESULT mmresult; - char *inBufPtr; - char *outBufPtr; - int gotInput = 0; - int gotOutput = 0; - int i; - int buffersProcessed = 0; - int done = 0; - PaWMMEStreamData *wmmeStreamData = (PaWMMEStreamData *) stream->past_DeviceData; - - if( wmmeStreamData == NULL ) return paInternalError; - - stream->past_NumCallbacks += 1; -#if PA_TRACE_RUN - AddTraceMessage("Pa_TimeSlice: past_NumCallbacks ", stream->past_NumCallbacks ); -#endif - - /* JM20020118 - prevent hung thread when buffers underflow. */ - /* while( !done ) /* BAD */ - while( !done && !stream->past_StopSoon ) /* GOOD */ - { -#if PA_SIMULATE_UNDERFLOW - if(gUnderCallbackCounter++ == UNDER_SLEEP_AT) - { - Sleep(UNDER_SLEEP_FOR); - } -#endif - - /* If we are using output, then we need an empty output buffer. */ - gotOutput = 0; - outBufPtr = NULL; - if( stream->past_NumOutputChannels > 0 ) - { - if((wmmeStreamData->outputBuffers[ wmmeStreamData->currentOutputBuffer ].dwFlags & WHDR_DONE) == 0) - { - break; /* If none empty then bail and try again later. */ - } - else - { - outBufPtr = wmmeStreamData->outputBuffers[ wmmeStreamData->currentOutputBuffer ].lpData; - gotOutput = 1; - } - } - /* Use an input buffer if one is available. */ - gotInput = 0; - inBufPtr = NULL; - if( ( stream->past_NumInputChannels > 0 ) && - (wmmeStreamData->inputBuffers[ wmmeStreamData->currentInputBuffer ].dwFlags & WHDR_DONE) ) - { - inBufPtr = wmmeStreamData->inputBuffers[ wmmeStreamData->currentInputBuffer ].lpData; - gotInput = 1; -#if PA_TRACE_RUN - AddTraceMessage("Pa_TimeSlice: got input buffer at ", (int)inBufPtr ); - AddTraceMessage("Pa_TimeSlice: got input buffer # ", wmmeStreamData->currentInputBuffer ); -#endif - - } - /* If we can't do anything then bail out. */ - if( !gotInput && !gotOutput ) break; - buffersProcessed += 1; - /* Each Wave buffer contains multiple user buffers so do them all now. */ - /* Base Usage on time it took to process one host buffer. */ - Pa_StartUsageCalculation( stream ); - for( i=0; i<wmmeStreamData->userBuffersPerHostBuffer; i++ ) - { - if( done ) - { - if( gotOutput ) - { - /* Clear remainder of wave buffer if we are waiting for stop. */ - AddTraceMessage("Pa_TimeSlice: zero rest of wave buffer ", i ); - memset( outBufPtr, 0, wmmeStreamData->bytesPerUserOutputBuffer ); - } - } - else - { - /* Convert 16 bit native data to user data and call user routine. */ - result = Pa_CallConvertInt16( stream, (short *) inBufPtr, (short *) outBufPtr ); - if( result != 0) done = 1; - } - if( gotInput ) inBufPtr += wmmeStreamData->bytesPerUserInputBuffer; - if( gotOutput) outBufPtr += wmmeStreamData->bytesPerUserOutputBuffer; - } - Pa_EndUsageCalculation( stream ); - /* Send WAVE buffer to Wave Device to be refilled. */ - if( gotInput ) - { - mmresult = waveInAddBuffer( wmmeStreamData->hWaveIn, - &wmmeStreamData->inputBuffers[ wmmeStreamData->currentInputBuffer ], - sizeof(WAVEHDR) ); - if( mmresult != MMSYSERR_NOERROR ) - { - sPaHostError = mmresult; - result = paHostError; - break; - } - wmmeStreamData->currentInputBuffer = (wmmeStreamData->currentInputBuffer+1 >= wmmeStreamData->numHostBuffers) ? - 0 : wmmeStreamData->currentInputBuffer+1; - } - /* Write WAVE buffer to Wave Device. */ - if( gotOutput ) - { -#if PA_TRACE_START_STOP - AddTraceMessage( "Pa_TimeSlice: writing buffer ", wmmeStreamData->currentOutputBuffer ); -#endif - mmresult = waveOutWrite( wmmeStreamData->hWaveOut, - &wmmeStreamData->outputBuffers[ wmmeStreamData->currentOutputBuffer ], - sizeof(WAVEHDR) ); - if( mmresult != MMSYSERR_NOERROR ) - { - sPaHostError = mmresult; - result = paHostError; - break; - } - wmmeStreamData->currentOutputBuffer = (wmmeStreamData->currentOutputBuffer+1 >= wmmeStreamData->numHostBuffers) ? - 0 : wmmeStreamData->currentOutputBuffer+1; - } - - } - -#if PA_TRACE_RUN - AddTraceMessage("Pa_TimeSlice: buffersProcessed ", buffersProcessed ); -#endif - return (result != 0) ? result : done; -} - -/*******************************************************************/ -static PaError PaHost_BackgroundManager( internalPortAudioStream *stream ) -{ - PaError result = paNoError; - int i; - int numQueuedoutputBuffers = 0; - PaWMMEStreamData *wmmeStreamData = (PaWMMEStreamData *) stream->past_DeviceData; - - /* Has someone asked us to abort by calling Pa_AbortStream()? */ - if( stream->past_StopNow ) - { - stream->past_IsActive = 0; /* Will cause thread to return. */ - } - /* Has someone asked us to stop by calling Pa_StopStream() - * OR has a user callback returned '1' to indicate finished. - */ - else if( stream->past_StopSoon ) - { - /* Poll buffer and when all have played then exit thread. */ - /* Count how many output buffers are queued. */ - numQueuedoutputBuffers = 0; - if( stream->past_NumOutputChannels > 0 ) - { - for( i=0; i<wmmeStreamData->numHostBuffers; i++ ) - { - if( !( wmmeStreamData->outputBuffers[ i ].dwFlags & WHDR_DONE) ) - { -#if PA_TRACE_START_STOP - AddTraceMessage( "PaHost_BackgroundManager: waiting for buffer ", i ); -#endif - numQueuedoutputBuffers++; - } - } - } -#if PA_TRACE_START_STOP - AddTraceMessage( "PaHost_BackgroundManager: numQueuedoutputBuffers ", numQueuedoutputBuffers ); -#endif - if( numQueuedoutputBuffers == 0 ) - { - stream->past_IsActive = 0; /* Will cause thread to return. */ - } - } - else - { - /* Process full input buffer and fill up empty output buffers. */ - if( (result = Pa_TimeSlice( stream )) != 0) - { - /* User callback has asked us to stop. */ -#if PA_TRACE_START_STOP - AddTraceMessage( "PaHost_BackgroundManager: TimeSlice() returned ", result ); -#endif - stream->past_StopSoon = 1; /* Request that audio play out then stop. */ - result = paNoError; - } - } - - PaHost_UpdateStreamTime( wmmeStreamData ); - return result; -} - -#if PA_USE_TIMER_CALLBACK -/*******************************************************************/ -static void CALLBACK Pa_TimerCallback(UINT uID, UINT uMsg, DWORD dwUser, DWORD dw1, DWORD dw2) -{ - internalPortAudioStream *stream; - PaWMMEStreamData *wmmeStreamData; - PaError result; - - stream = (internalPortAudioStream *) dwUser; - if( stream == NULL ) return; - wmmeStreamData = (PaWMMEStreamData *) stream->past_DeviceData; - if( wmmeStreamData == NULL ) return; - if( wmmeStreamData->ifInsideCallback ) - { - if( wmmeStreamData->timerID != 0 ) - { - timeKillEvent(wmmeStreamData->timerID); /* Stop callback timer. */ - wmmeStreamData->timerID = 0; - } - return; - } - wmmeStreamData->ifInsideCallback = 1; - /* Manage flags and audio processing. */ - result = PaHost_BackgroundManager( stream ); - if( result != paNoError ) - { - stream->past_IsActive = 0; - } - wmmeStreamData->ifInsideCallback = 0; -} -#else /* PA_USE_TIMER_CALLBACK */ -/*******************************************************************/ -static DWORD WINAPI WinMMPa_OutputThreadProc( void *pArg ) -{ - internalPortAudioStream *stream; - PaWMMEStreamData *wmmeStreamData; - HANDLE events[2]; - int numEvents = 0; - DWORD result = 0; - DWORD waitResult; - DWORD numTimeouts = 0; - DWORD timeOut; - stream = (internalPortAudioStream *) pArg; - wmmeStreamData = (PaWMMEStreamData *) stream->past_DeviceData; -#if PA_TRACE_START_STOP - AddTraceMessage( "WinMMPa_OutputThreadProc: timeoutPeriod", timeoutPeriod ); - AddTraceMessage( "WinMMPa_OutputThreadProc: past_NumUserBuffers", stream->past_NumUserBuffers ); -#endif - /* Calculate timeOut as half the time it would take to play all buffers. */ - timeOut = (DWORD) (500.0 * PaHost_GetTotalBufferFrames( stream ) / stream->past_SampleRate); - /* Get event(s) ready for wait. */ - events[numEvents++] = wmmeStreamData->bufferEvent; - if( wmmeStreamData->abortEventInited ) events[numEvents++] = wmmeStreamData->abortEvent; - /* Stay in this thread as long as we are "active". */ - while( stream->past_IsActive ) - { - /*******************************************************************/ - /******** WAIT here for an event from WMME or PA *******************/ - /*******************************************************************/ - waitResult = WaitForMultipleObjects( numEvents, events, FALSE, timeOut ); - /* Error? */ - if( waitResult == WAIT_FAILED ) - { - sPaHostError = GetLastError(); - result = paHostError; - stream->past_IsActive = 0; - } - /* Timeout? Don't stop. Just keep polling for DONE.*/ - else if( waitResult == WAIT_TIMEOUT ) - { -#if PA_TRACE_START_STOP - AddTraceMessage( "WinMMPa_OutputThreadProc: timed out ", numQueuedoutputBuffers ); -#endif - numTimeouts += 1; - } - /* Manage flags and audio processing. */ - result = PaHost_BackgroundManager( stream ); - if( result != paNoError ) - { - stream->past_IsActive = 0; - } - } - return result; -} -#endif - -/*******************************************************************/ -PaError PaHost_OpenInputStream( internalPortAudioStream *stream ) -{ - PaError result = paNoError; - MMRESULT mmresult; - PaWMMEStreamData *wmmeStreamData; - int i; - int inputMmId; - int bytesPerInputFrame; - WAVEFORMATEX wfx; - const PaDeviceInfo *deviceInfo; - - wmmeStreamData = (PaWMMEStreamData *) stream->past_DeviceData; - DBUG(("PaHost_OpenStream: deviceID = 0x%x\n", stream->past_InputDeviceID)); - deviceInfo = Pa_GetDeviceInfo( stream->past_InputDeviceID ); - if( deviceInfo == NULL ) return paInternalError; - - switch( deviceInfo->nativeSampleFormats ) - { - case paInt32: - case paFloat32: - bytesPerInputFrame = sizeof(float) * stream->past_NumInputChannels; - break; - default: - bytesPerInputFrame = sizeof(short) * stream->past_NumInputChannels; - break; - } - wfx.wFormatTag = WAVE_FORMAT_PCM; - wfx.nChannels = (WORD) stream->past_NumInputChannels; - wfx.nSamplesPerSec = (DWORD) stream->past_SampleRate; - wfx.nAvgBytesPerSec = (DWORD)(bytesPerInputFrame * stream->past_SampleRate); - wfx.nBlockAlign = (WORD)bytesPerInputFrame; - wfx.wBitsPerSample = (WORD)((bytesPerInputFrame/stream->past_NumInputChannels) * 8); - wfx.cbSize = 0; - inputMmId = PaDeviceIdToWinId( stream->past_InputDeviceID ); -#if PA_USE_TIMER_CALLBACK - mmresult = waveInOpen( &wmmeStreamData->hWaveIn, inputMmId, &wfx, - 0, 0, CALLBACK_NULL ); -#else - mmresult = waveInOpen( &wmmeStreamData->hWaveIn, inputMmId, &wfx, - (DWORD)wmmeStreamData->bufferEvent, (DWORD) stream, CALLBACK_EVENT ); -#endif - if( mmresult != MMSYSERR_NOERROR ) - { - ERR_RPT(("PortAudio: PaHost_OpenInputStream() failed!\n")); - result = paHostError; - sPaHostError = mmresult; - goto error; - } - /* Allocate an array to hold the buffer pointers. */ - wmmeStreamData->inputBuffers = (WAVEHDR *) PaHost_AllocateTrackedMemory( sizeof(WAVEHDR)*wmmeStreamData->numHostBuffers ); /* MEM */ - if( wmmeStreamData->inputBuffers == NULL ) - { - result = paInsufficientMemory; - goto error; - } - /* Allocate each buffer. */ - for( i=0; i<wmmeStreamData->numHostBuffers; i++ ) - { - wmmeStreamData->inputBuffers[i].lpData = (char *)PaHost_AllocateTrackedMemory( wmmeStreamData->bytesPerHostInputBuffer ); /* MEM */ - if( wmmeStreamData->inputBuffers[i].lpData == NULL ) - { - result = paInsufficientMemory; - goto error; - } - wmmeStreamData->inputBuffers[i].dwBufferLength = wmmeStreamData->bytesPerHostInputBuffer; - wmmeStreamData->inputBuffers[i].dwUser = i; - if( ( mmresult = waveInPrepareHeader( wmmeStreamData->hWaveIn, &wmmeStreamData->inputBuffers[i], sizeof(WAVEHDR) )) != MMSYSERR_NOERROR ) - { - result = paHostError; - sPaHostError = mmresult; - goto error; - } - } - return result; - -error: - return result; -} -/*******************************************************************/ -PaError PaHost_OpenOutputStream( internalPortAudioStream *stream ) -{ - PaError result = paNoError; - MMRESULT mmresult; - PaWMMEStreamData *wmmeStreamData; - int i; - int outputMmID; - int bytesPerOutputFrame; - WAVEFORMATEX wfx; - const PaDeviceInfo *deviceInfo; - - wmmeStreamData = (PaWMMEStreamData *) stream->past_DeviceData; - DBUG(("PaHost_OpenStream: deviceID = 0x%x\n", stream->past_OutputDeviceID)); - - deviceInfo = Pa_GetDeviceInfo( stream->past_OutputDeviceID ); - if( deviceInfo == NULL ) return paInternalError; - - switch( deviceInfo->nativeSampleFormats ) - { - case paInt32: - case paFloat32: - bytesPerOutputFrame = sizeof(float) * stream->past_NumOutputChannels; - break; - default: - bytesPerOutputFrame = sizeof(short) * stream->past_NumOutputChannels; - break; - } - wfx.wFormatTag = WAVE_FORMAT_PCM; - wfx.nChannels = (WORD) stream->past_NumOutputChannels; - wfx.nSamplesPerSec = (DWORD) stream->past_SampleRate; - wfx.nAvgBytesPerSec = (DWORD)(bytesPerOutputFrame * stream->past_SampleRate); - wfx.nBlockAlign = (WORD)bytesPerOutputFrame; - wfx.wBitsPerSample = (WORD)((bytesPerOutputFrame/stream->past_NumOutputChannels) * 8); - wfx.cbSize = 0; - outputMmID = PaDeviceIdToWinId( stream->past_OutputDeviceID ); -#if PA_USE_TIMER_CALLBACK - mmresult = waveOutOpen( &wmmeStreamData->hWaveOut, outputMmID, &wfx, - 0, 0, CALLBACK_NULL ); -#else - - wmmeStreamData->abortEvent = CreateEvent( NULL, TRUE, FALSE, NULL ); - if( wmmeStreamData->abortEvent == NULL ) - { - result = paHostError; - sPaHostError = GetLastError(); - goto error; - } - wmmeStreamData->abortEventInited = 1; - mmresult = waveOutOpen( &wmmeStreamData->hWaveOut, outputMmID, &wfx, - (DWORD)wmmeStreamData->bufferEvent, (DWORD) stream, CALLBACK_EVENT ); -#endif - if( mmresult != MMSYSERR_NOERROR ) - { - ERR_RPT(("PortAudio: PaHost_OpenOutputStream() failed!\n")); - result = paHostError; - sPaHostError = mmresult; - goto error; - } - /* Allocate an array to hold the buffer pointers. */ - wmmeStreamData->outputBuffers = (WAVEHDR *) PaHost_AllocateTrackedMemory( sizeof(WAVEHDR)*wmmeStreamData->numHostBuffers ); /* MEM */ - if( wmmeStreamData->outputBuffers == NULL ) - { - result = paInsufficientMemory; - goto error; - } - /* Allocate each buffer. */ - for( i=0; i<wmmeStreamData->numHostBuffers; i++ ) - { - wmmeStreamData->outputBuffers[i].lpData = (char *) PaHost_AllocateTrackedMemory( wmmeStreamData->bytesPerHostOutputBuffer ); /* MEM */ - if( wmmeStreamData->outputBuffers[i].lpData == NULL ) - { - result = paInsufficientMemory; - goto error; - } - wmmeStreamData->outputBuffers[i].dwBufferLength = wmmeStreamData->bytesPerHostOutputBuffer; - wmmeStreamData->outputBuffers[i].dwUser = i; - if( (mmresult = waveOutPrepareHeader( wmmeStreamData->hWaveOut, &wmmeStreamData->outputBuffers[i], sizeof(WAVEHDR) )) != MMSYSERR_NOERROR ) - { - result = paHostError; - sPaHostError = mmresult; - goto error; - } - } - return result; - -error: - return result; -} -/*******************************************************************/ -PaError PaHost_GetTotalBufferFrames( internalPortAudioStream *stream ) -{ - PaWMMEStreamData *wmmeStreamData = (PaWMMEStreamData *) stream->past_DeviceData; - return wmmeStreamData->numHostBuffers * wmmeStreamData->framesPerHostBuffer; -} -/******************************************************************* - * Determine number of WAVE Buffers - * and how many User Buffers we can put into each WAVE buffer. - */ -static void PaHost_CalcNumHostBuffers( internalPortAudioStream *stream ) -{ - PaWMMEStreamData *wmmeStreamData = (PaWMMEStreamData *) stream->past_DeviceData; - unsigned int minNumBuffers; - int minframesPerHostBuffer; - int maxframesPerHostBuffer; - int minTotalFrames; - int userBuffersPerHostBuffer; - int framesPerHostBuffer; - int numHostBuffers; - - /* Calculate minimum and maximum sizes based on timing and sample rate. */ - minframesPerHostBuffer = (int) (PA_MIN_MSEC_PER_HOST_BUFFER * stream->past_SampleRate * 0.001); - minframesPerHostBuffer = (minframesPerHostBuffer + 7) & ~7; - DBUG(("PaHost_CalcNumHostBuffers: minframesPerHostBuffer = %d\n", minframesPerHostBuffer )); - maxframesPerHostBuffer = (int) (PA_MAX_MSEC_PER_HOST_BUFFER * stream->past_SampleRate * 0.001); - maxframesPerHostBuffer = (maxframesPerHostBuffer + 7) & ~7; - DBUG(("PaHost_CalcNumHostBuffers: maxframesPerHostBuffer = %d\n", maxframesPerHostBuffer )); - /* Determine number of user buffers based on minimum latency. */ - minNumBuffers = Pa_GetMinNumBuffers( stream->past_FramesPerUserBuffer, stream->past_SampleRate ); - stream->past_NumUserBuffers = ( minNumBuffers > stream->past_NumUserBuffers ) ? minNumBuffers : stream->past_NumUserBuffers; - DBUG(("PaHost_CalcNumHostBuffers: min past_NumUserBuffers = %d\n", stream->past_NumUserBuffers )); - minTotalFrames = stream->past_NumUserBuffers * stream->past_FramesPerUserBuffer; - /* We cannot make the WAVE buffers too small because they may not get serviced quickly enough. */ - if( (int) stream->past_FramesPerUserBuffer < minframesPerHostBuffer ) - { - userBuffersPerHostBuffer = - (minframesPerHostBuffer + stream->past_FramesPerUserBuffer - 1) / - stream->past_FramesPerUserBuffer; - } - else - { - userBuffersPerHostBuffer = 1; - } - framesPerHostBuffer = stream->past_FramesPerUserBuffer * userBuffersPerHostBuffer; - /* Calculate number of WAVE buffers needed. Round up to cover minTotalFrames. */ - numHostBuffers = (minTotalFrames + framesPerHostBuffer - 1) / framesPerHostBuffer; - /* Make sure we have anough WAVE buffers. */ - if( numHostBuffers < PA_MIN_NUM_HOST_BUFFERS) - { - numHostBuffers = PA_MIN_NUM_HOST_BUFFERS; - } - else if( (numHostBuffers > PA_MAX_NUM_HOST_BUFFERS) && - ((int) stream->past_FramesPerUserBuffer < (maxframesPerHostBuffer/2) ) ) - { - /* If we have too many WAVE buffers, try to put more user buffers in a wave buffer. */ - while(numHostBuffers > PA_MAX_NUM_HOST_BUFFERS) - { - userBuffersPerHostBuffer += 1; - framesPerHostBuffer = stream->past_FramesPerUserBuffer * userBuffersPerHostBuffer; - numHostBuffers = (minTotalFrames + framesPerHostBuffer - 1) / framesPerHostBuffer; - /* If we have gone too far, back up one. */ - if( (framesPerHostBuffer > maxframesPerHostBuffer) || - (numHostBuffers < PA_MAX_NUM_HOST_BUFFERS) ) - { - userBuffersPerHostBuffer -= 1; - framesPerHostBuffer = stream->past_FramesPerUserBuffer * userBuffersPerHostBuffer; - numHostBuffers = (minTotalFrames + framesPerHostBuffer - 1) / framesPerHostBuffer; - break; - } - } - } - - wmmeStreamData->userBuffersPerHostBuffer = userBuffersPerHostBuffer; - wmmeStreamData->framesPerHostBuffer = framesPerHostBuffer; - wmmeStreamData->numHostBuffers = numHostBuffers; - DBUG(("PaHost_CalcNumHostBuffers: userBuffersPerHostBuffer = %d\n", wmmeStreamData->userBuffersPerHostBuffer )); - DBUG(("PaHost_CalcNumHostBuffers: numHostBuffers = %d\n", wmmeStreamData->numHostBuffers )); - DBUG(("PaHost_CalcNumHostBuffers: framesPerHostBuffer = %d\n", wmmeStreamData->framesPerHostBuffer )); - DBUG(("PaHost_CalcNumHostBuffers: past_NumUserBuffers = %d\n", stream->past_NumUserBuffers )); -} -/*******************************************************************/ -PaError PaHost_OpenStream( internalPortAudioStream *stream ) -{ - PaError result = paNoError; - PaWMMEStreamData *wmmeStreamData; - - result = PaHost_AllocateWMMEStreamData( stream ); - if( result != paNoError ) return result; - - wmmeStreamData = PaHost_GetWMMEStreamData( stream ); - - /* Figure out how user buffers fit into WAVE buffers. */ - PaHost_CalcNumHostBuffers( stream ); - { - int msecLatency = (int) ((PaHost_GetTotalBufferFrames(stream) * 1000) / stream->past_SampleRate); - DBUG(("PortAudio on WMME - Latency = %d frames, %d msec\n", PaHost_GetTotalBufferFrames(stream), msecLatency )); - } - InitializeCriticalSection( &wmmeStreamData->streamLock ); - wmmeStreamData->streamLockInited = 1; - -#if (PA_USE_TIMER_CALLBACK == 0) - wmmeStreamData->bufferEventInited = 0; - wmmeStreamData->bufferEvent = CreateEvent( NULL, FALSE, FALSE, NULL ); - if( wmmeStreamData->bufferEvent == NULL ) - { - result = paHostError; - sPaHostError = GetLastError(); - goto error; - } - wmmeStreamData->bufferEventInited = 1; -#endif /* (PA_USE_TIMER_CALLBACK == 0) */ - /* ------------------ OUTPUT */ - wmmeStreamData->bytesPerUserOutputBuffer = stream->past_FramesPerUserBuffer * stream->past_NumOutputChannels * sizeof(short); - wmmeStreamData->bytesPerHostOutputBuffer = wmmeStreamData->userBuffersPerHostBuffer * wmmeStreamData->bytesPerUserOutputBuffer; - if( (stream->past_OutputDeviceID != paNoDevice) && (stream->past_NumOutputChannels > 0) ) - { - result = PaHost_OpenOutputStream( stream ); - if( result < 0 ) goto error; - } - /* ------------------ INPUT */ - wmmeStreamData->bytesPerUserInputBuffer = stream->past_FramesPerUserBuffer * stream->past_NumInputChannels * sizeof(short); - wmmeStreamData->bytesPerHostInputBuffer = wmmeStreamData->userBuffersPerHostBuffer * wmmeStreamData->bytesPerUserInputBuffer; - if( (stream->past_InputDeviceID != paNoDevice) && (stream->past_NumInputChannels > 0) ) - { - result = PaHost_OpenInputStream( stream ); - if( result < 0 ) goto error; - } - - Pa_InitializeCpuUsageScalar( stream ); - - return result; - -error: - PaHost_CloseStream( stream ); - return result; -} -/*************************************************************************/ -PaError PaHost_StartOutput( internalPortAudioStream *stream ) -{ - PaError result = paNoError; - MMRESULT mmresult; - int i; - PaWMMEStreamData *wmmeStreamData = PaHost_GetWMMEStreamData( stream ); - - if( wmmeStreamData == NULL ) return paInternalError; - - if( stream->past_OutputDeviceID != paNoDevice ) - { - if( (mmresult = waveOutPause( wmmeStreamData->hWaveOut )) != MMSYSERR_NOERROR ) - { - result = paHostError; - sPaHostError = mmresult; - goto error; - } - for( i=0; i<wmmeStreamData->numHostBuffers; i++ ) - { - ZeroMemory( wmmeStreamData->outputBuffers[i].lpData, wmmeStreamData->outputBuffers[i].dwBufferLength ); - mmresult = waveOutWrite( wmmeStreamData->hWaveOut, &wmmeStreamData->outputBuffers[i], sizeof(WAVEHDR) ); - if( mmresult != MMSYSERR_NOERROR ) - { - result = paHostError; - sPaHostError = mmresult; - goto error; - } - stream->past_FrameCount += wmmeStreamData->framesPerHostBuffer; - } - wmmeStreamData->currentOutputBuffer = 0; - if( (mmresult = waveOutRestart( wmmeStreamData->hWaveOut )) != MMSYSERR_NOERROR ) - { - result = paHostError; - sPaHostError = mmresult; - goto error; - } - } - -error: - DBUG(("PaHost_StartOutput: wave returned mmresult = 0x%X.\n", mmresult)); - return result; -} -/*************************************************************************/ -PaError PaHost_StartInput( internalPortAudioStream *internalStream ) -{ - PaError result = paNoError; - MMRESULT mmresult; - int i; - PaWMMEStreamData *wmmeStreamData = PaHost_GetWMMEStreamData( internalStream ); - - if( wmmeStreamData == NULL ) return paInternalError; - - if( internalStream->past_InputDeviceID != paNoDevice ) - { - for( i=0; i<wmmeStreamData->numHostBuffers; i++ ) - { - mmresult = waveInAddBuffer( wmmeStreamData->hWaveIn, &wmmeStreamData->inputBuffers[i], sizeof(WAVEHDR) ); - if( mmresult != MMSYSERR_NOERROR ) - { - result = paHostError; - sPaHostError = mmresult; - goto error; - } - } - wmmeStreamData->currentInputBuffer = 0; - mmresult = waveInStart( wmmeStreamData->hWaveIn ); - DBUG(("Pa_StartStream: waveInStart returned = 0x%X.\n", mmresult)); - if( mmresult != MMSYSERR_NOERROR ) - { - result = paHostError; - sPaHostError = mmresult; - goto error; - } - } - -error: - return result; -} -/*************************************************************************/ -PaError PaHost_StartEngine( internalPortAudioStream *stream ) -{ - PaError result = paNoError; - PaWMMEStreamData *wmmeStreamData = PaHost_GetWMMEStreamData( stream ); -#if PA_USE_TIMER_CALLBACK - int resolution; - int bufsPerTimerCallback; - int msecPerBuffer; -#endif /* PA_USE_TIMER_CALLBACK */ - - if( wmmeStreamData == NULL ) return paInternalError; - - stream->past_StopSoon = 0; - stream->past_StopNow = 0; - stream->past_IsActive = 1; - wmmeStreamData->framesPlayed = 0.0; - wmmeStreamData->lastPosition = 0; -#if PA_TRACE_START_STOP - AddTraceMessage( "PaHost_StartEngine: TimeSlice() returned ", result ); -#endif -#if PA_USE_TIMER_CALLBACK - /* Create timer that will wake us up so we can fill the DSound buffer. */ - bufsPerTimerCallback = wmmeStreamData->numHostBuffers/4; - if( bufsPerTimerCallback < 1 ) bufsPerTimerCallback = 1; - if( bufsPerTimerCallback < 1 ) bufsPerTimerCallback = 1; - msecPerBuffer = (1000 * bufsPerTimerCallback * - wmmeStreamData->userBuffersPerHostBuffer * - internalStream->past_FramesPerUserBuffer ) / (int) internalStream->past_SampleRate; - if( msecPerBuffer < 10 ) msecPerBuffer = 10; - else if( msecPerBuffer > 100 ) msecPerBuffer = 100; - resolution = msecPerBuffer/4; - wmmeStreamData->timerID = timeSetEvent( msecPerBuffer, resolution, - (LPTIMECALLBACK) Pa_TimerCallback, - (DWORD) stream, TIME_PERIODIC ); - if( wmmeStreamData->timerID == 0 ) - { - result = paHostError; - sPaHostError = GetLastError();; - goto error; - } -#else /* PA_USE_TIMER_CALLBACK */ - ResetEvent( wmmeStreamData->abortEvent ); - /* Create thread that waits for audio buffers to be ready for processing. */ - wmmeStreamData->engineThread = CreateThread( 0, 0, WinMMPa_OutputThreadProc, stream, 0, &wmmeStreamData->engineThreadID ); - if( wmmeStreamData->engineThread == NULL ) - { - result = paHostError; - sPaHostError = GetLastError();; - goto error; - } -#if PA_TRACE_START_STOP - AddTraceMessage( "PaHost_StartEngine: thread ", (int) wmmeStreamData->engineThread ); -#endif - /* I used to pass the thread which was failing. I now pass GetCurrentProcess(). - * This fix could improve latency for some applications. It could also result in CPU - * starvation if the callback did too much processing. - * I also added result checks, so we might see more failures at initialization. - * Thanks to Alberto di Bene for spotting this. - */ - if( !SetPriorityClass( GetCurrentProcess(), HIGH_PRIORITY_CLASS ) ) /* PLB20010816 */ - { - result = paHostError; - sPaHostError = GetLastError();; - goto error; - } - if( !SetThreadPriority( wmmeStreamData->engineThread, THREAD_PRIORITY_HIGHEST ) ) - { - result = paHostError; - sPaHostError = GetLastError();; - goto error; - } -#endif - -error: - return result; -} -/*************************************************************************/ -PaError PaHost_StopEngine( internalPortAudioStream *internalStream, int abort ) -{ - int timeOut; - PaWMMEStreamData *wmmeStreamData = PaHost_GetWMMEStreamData( internalStream ); - - if( wmmeStreamData == NULL ) return paNoError; - - /* Tell background thread to stop generating more data and to let current data play out. */ - internalStream->past_StopSoon = 1; - /* If aborting, tell background thread to stop NOW! */ - if( abort ) internalStream->past_StopNow = 1; - - /* Calculate timeOut longer than longest time it could take to play all buffers. */ - timeOut = (DWORD) (1500.0 * PaHost_GetTotalBufferFrames( internalStream ) / internalStream->past_SampleRate); - if( timeOut < MIN_TIMEOUT_MSEC ) timeOut = MIN_TIMEOUT_MSEC; - -#if PA_USE_TIMER_CALLBACK - if( (internalStream->past_OutputDeviceID != paNoDevice) && - internalStream->past_IsActive && - (wmmeStreamData->timerID != 0) ) - { - /* Wait for IsActive to drop. */ - while( (internalStream->past_IsActive) && (timeOut > 0) ) - { - Sleep(10); - timeOut -= 10; - } - timeKillEvent( wmmeStreamData->timerID ); /* Stop callback timer. */ - wmmeStreamData->timerID = 0; - } -#else /* PA_USE_TIMER_CALLBACK */ -#if PA_TRACE_START_STOP - AddTraceMessage( "PaHost_StopEngine: thread ", (int) wmmeStreamData->engineThread ); -#endif - if( (internalStream->past_OutputDeviceID != paNoDevice) && - (internalStream->past_IsActive) && - (wmmeStreamData->engineThread != NULL) ) - { - DWORD got; - /* Tell background thread to stop generating more data and to let current data play out. */ - DBUG(("PaHost_StopEngine: waiting for background thread.\n")); - got = WaitForSingleObject( wmmeStreamData->engineThread, timeOut ); - if( got == WAIT_TIMEOUT ) - { - ERR_RPT(("PaHost_StopEngine: timed out while waiting for background thread to finish.\n")); - return paTimedOut; - } - CloseHandle( wmmeStreamData->engineThread ); - wmmeStreamData->engineThread = NULL; - } -#endif /* PA_USE_TIMER_CALLBACK */ - - internalStream->past_IsActive = 0; - return paNoError; -} -/*************************************************************************/ -PaError PaHost_StopInput( internalPortAudioStream *stream, int abort ) -{ - MMRESULT mmresult; - PaWMMEStreamData *wmmeStreamData = PaHost_GetWMMEStreamData( stream ); - - if( wmmeStreamData == NULL ) return paNoError; /* FIXME: why return paNoError? */ - (void) abort; /* unused parameter */ - - if( wmmeStreamData->hWaveIn != NULL ) - { - mmresult = waveInReset( wmmeStreamData->hWaveIn ); - if( mmresult != MMSYSERR_NOERROR ) - { - sPaHostError = mmresult; - return paHostError; - } - } - return paNoError; -} -/*************************************************************************/ -PaError PaHost_StopOutput( internalPortAudioStream *internalStream, int abort ) -{ - MMRESULT mmresult; - PaWMMEStreamData *wmmeStreamData = PaHost_GetWMMEStreamData( internalStream ); - - if( wmmeStreamData == NULL ) return paNoError; /* FIXME: why return paNoError? */ - (void) abort; /* unused parameter */ - -#if PA_TRACE_START_STOP - AddTraceMessage( "PaHost_StopOutput: hWaveOut ", (int) wmmeStreamData->hWaveOut ); -#endif - if( wmmeStreamData->hWaveOut != NULL ) - { - mmresult = waveOutReset( wmmeStreamData->hWaveOut ); - if( mmresult != MMSYSERR_NOERROR ) - { - sPaHostError = mmresult; - return paHostError; - } - } - return paNoError; -} -/*******************************************************************/ -PaError PaHost_CloseStream( internalPortAudioStream *stream ) -{ - int i; - PaWMMEStreamData *wmmeStreamData = PaHost_GetWMMEStreamData( stream ); - - if( stream == NULL ) return paBadStreamPtr; - if( wmmeStreamData == NULL ) return paNoError; /* FIXME: why return no error? */ - -#if PA_TRACE_START_STOP - AddTraceMessage( "PaHost_CloseStream: hWaveOut ", (int) wmmeStreamData->hWaveOut ); -#endif - /* Free data and device for output. */ - if( wmmeStreamData->hWaveOut ) - { - if( wmmeStreamData->outputBuffers ) - { - for( i=0; i<wmmeStreamData->numHostBuffers; i++ ) - { - waveOutUnprepareHeader( wmmeStreamData->hWaveOut, &wmmeStreamData->outputBuffers[i], sizeof(WAVEHDR) ); - PaHost_FreeTrackedMemory( wmmeStreamData->outputBuffers[i].lpData ); /* MEM */ - } - PaHost_FreeTrackedMemory( wmmeStreamData->outputBuffers ); /* MEM */ - } - waveOutClose( wmmeStreamData->hWaveOut ); - } - /* Free data and device for input. */ - if( wmmeStreamData->hWaveIn ) - { - if( wmmeStreamData->inputBuffers ) - { - for( i=0; i<wmmeStreamData->numHostBuffers; i++ ) - { - waveInUnprepareHeader( wmmeStreamData->hWaveIn, &wmmeStreamData->inputBuffers[i], sizeof(WAVEHDR) ); - PaHost_FreeTrackedMemory( wmmeStreamData->inputBuffers[i].lpData ); /* MEM */ - } - PaHost_FreeTrackedMemory( wmmeStreamData->inputBuffers ); /* MEM */ - } - waveInClose( wmmeStreamData->hWaveIn ); - } -#if (PA_USE_TIMER_CALLBACK == 0) - if( wmmeStreamData->abortEventInited ) CloseHandle( wmmeStreamData->abortEvent ); - if( wmmeStreamData->bufferEventInited ) CloseHandle( wmmeStreamData->bufferEvent ); -#endif - if( wmmeStreamData->streamLockInited ) - DeleteCriticalSection( &wmmeStreamData->streamLock ); - - PaHost_FreeWMMEStreamData( stream ); - - return paNoError; -} -/************************************************************************* - * Determine minimum number of buffers required for this host based - * on minimum latency. Latency can be optionally set by user by setting - * an environment variable. For example, to set latency to 200 msec, put: - * - * set PA_MIN_LATENCY_MSEC=200 - * - * in the AUTOEXEC.BAT file and reboot. - * If the environment variable is not set, then the latency will be determined - * based on the OS. Windows NT has higher latency than Win95. - */ -#define PA_LATENCY_ENV_NAME ("PA_MIN_LATENCY_MSEC") -int Pa_GetMinNumBuffers( int framesPerBuffer, double sampleRate ) -{ - char envbuf[PA_ENV_BUF_SIZE]; - DWORD hresult; - int minLatencyMsec = 0; - double msecPerBuffer = (1000.0 * framesPerBuffer) / sampleRate; - int minBuffers; - - /* Let user determine minimal latency by setting environment variable. */ - hresult = GetEnvironmentVariable( PA_LATENCY_ENV_NAME, envbuf, PA_ENV_BUF_SIZE ); - if( (hresult > 0) && (hresult < PA_ENV_BUF_SIZE) ) - { - minLatencyMsec = atoi( envbuf ); /* REVIEW: will we crash if the environment variable contains some nasty value? */ - } - else - { - /* Set minimal latency based on whether NT or other OS. - * NT has higher latency. - */ - OSVERSIONINFO osvi; - osvi.dwOSVersionInfoSize = sizeof( osvi ); - GetVersionEx( &osvi ); - DBUG(("PA - PlatformId = 0x%x\n", osvi.dwPlatformId )); - DBUG(("PA - MajorVersion = 0x%x\n", osvi.dwMajorVersion )); - DBUG(("PA - MinorVersion = 0x%x\n", osvi.dwMinorVersion )); - /* Check for NT */ - if( (osvi.dwMajorVersion == 4) && (osvi.dwPlatformId == 2) ) - { - minLatencyMsec = PA_WIN_NT_LATENCY; - } - else if(osvi.dwMajorVersion >= 5) - { - minLatencyMsec = PA_WIN_WDM_LATENCY; - } - else - { - minLatencyMsec = PA_WIN_9X_LATENCY; - } -#if PA_USE_HIGH_LATENCY - PRINT(("PA - Minimum Latency set to %d msec!\n", minLatencyMsec )); -#endif - - } - DBUG(("PA - Minimum Latency set to %d msec!\n", minLatencyMsec )); - minBuffers = (int) (1.0 + ((double)minLatencyMsec / msecPerBuffer)); - if( minBuffers < 2 ) minBuffers = 2; - return minBuffers; -} -/************************************************************************* - * Cleanup device info. - */ -PaError PaHost_Term( void ) -{ - int i; - - if( sNumDevices > 0 ) - { - if( sDevicePtrs != NULL ) - { - for( i=0; i<sNumDevices; i++ ) - { - if( sDevicePtrs[i] != NULL ) - { - PaHost_FreeTrackedMemory( (char*)sDevicePtrs[i]->name ); /* MEM */ - PaHost_FreeTrackedMemory( (void*)sDevicePtrs[i]->sampleRates ); /* MEM */ - PaHost_FreeTrackedMemory( sDevicePtrs[i] ); /* MEM */ - } - } - PaHost_FreeTrackedMemory( sDevicePtrs ); /* MEM */ - sDevicePtrs = NULL; - } - sNumDevices = 0; - } - -#if PA_TRACK_MEMORY - PRINT(("PaHost_Term: sNumAllocations = %d\n", sNumAllocations )); -#endif - - return paNoError; -} -/*************************************************************************/ -void Pa_Sleep( long msec ) -{ - Sleep( msec ); -} -/************************************************************************* -FIXME: the following memory allocation routines should not be declared here - * Allocate memory that can be accessed in real-time. - * This may need to be held in physical memory so that it is not - * paged to virtual memory. - * This call MUST be balanced with a call to PaHost_FreeFastMemory(). - * Memory will be set to zero. - */ -void *PaHost_AllocateFastMemory( long numBytes ) -{ - return PaHost_AllocateTrackedMemory( numBytes ); /* FIXME - do we need physical memory? Use VirtualLock() */ /* MEM */ -} -/************************************************************************* - * Free memory that could be accessed in real-time. - * This call MUST be balanced with a call to PaHost_AllocateFastMemory(). - */ -void PaHost_FreeFastMemory( void *addr, long numBytes ) -{ - (void) numBytes; /* unused parameter */ - - PaHost_FreeTrackedMemory( addr ); /* MEM */ -} - -/************************************************************************* - * Track memory allocations to avoid leaks. - */ -static void *PaHost_AllocateTrackedMemory( long numBytes ) -{ - void *result = GlobalAlloc( GPTR, numBytes ); /* MEM */ - -#if PA_TRACK_MEMORY - if( result != NULL ) sNumAllocations += 1; -#endif - return result; -} - -static void PaHost_FreeTrackedMemory( void *addr ) -{ - if( addr != NULL ) - { - GlobalFree( addr ); /* MEM */ -#if PA_TRACK_MEMORY - sNumAllocations -= 1; -#endif - } -} - -/***********************************************************************/ -PaError PaHost_StreamActive( internalPortAudioStream *internalStream ) -{ - if( internalStream == NULL ) return paBadStreamPtr; - - return (PaError) internalStream->past_IsActive; -} -/************************************************************************* - * This must be called periodically because mmtime.u.sample - * is a DWORD and can wrap and lose sync after a few hours. - */ -static PaError PaHost_UpdateStreamTime( PaWMMEStreamData *wmmeStreamData ) -{ - MMRESULT mmresult; - MMTIME mmtime; - mmtime.wType = TIME_SAMPLES; - - if( wmmeStreamData->hWaveOut != NULL ) - { - mmresult = waveOutGetPosition( wmmeStreamData->hWaveOut, &mmtime, sizeof(mmtime) ); - } - else - { - mmresult = waveInGetPosition( wmmeStreamData->hWaveIn, &mmtime, sizeof(mmtime) ); - } - - if( mmresult != MMSYSERR_NOERROR ) - { - sPaHostError = mmresult; - return paHostError; - } - - /* This data has two variables and is shared by foreground and background. - * So we need to make it thread safe. */ - EnterCriticalSection( &wmmeStreamData->streamLock ); - wmmeStreamData->framesPlayed += ((long)mmtime.u.sample) - wmmeStreamData->lastPosition; - wmmeStreamData->lastPosition = (long)mmtime.u.sample; - LeaveCriticalSection( &wmmeStreamData->streamLock ); - - return paNoError; -} -/*************************************************************************/ -PaTimestamp Pa_StreamTime( PortAudioStream *stream ) -{ - internalPortAudioStream *internalStream = PaHost_GetStreamRepresentation( stream ); - PaWMMEStreamData *wmmeStreamData = PaHost_GetWMMEStreamData( internalStream ); - - if( internalStream == NULL ) return paBadStreamPtr; - if( wmmeStreamData == NULL ) return paInternalError; - - PaHost_UpdateStreamTime( wmmeStreamData ); - return wmmeStreamData->framesPlayed; -} -/*************************************************************************/ - - - diff --git a/pd/portaudio_v18/pablio/README.txt b/pd/portaudio_v18/pablio/README.txt deleted file mode 100644 index 99c7d146..00000000 --- a/pd/portaudio_v18/pablio/README.txt +++ /dev/null @@ -1,39 +0,0 @@ -README for PABLIO -Portable Audio Blocking I/O Library -Author: Phil Burk - -PABLIO is a simplified interface to PortAudio that provide -read/write style blocking I/O. - -Please see the .DOC file for documentation. - -/* - * More information on PortAudio at: http://www.portaudio.com - * Copyright (c) 1999-2000 Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ - - diff --git a/pd/portaudio_v18/pablio/pablio.c b/pd/portaudio_v18/pablio/pablio.c deleted file mode 100644 index d3a1bcf2..00000000 --- a/pd/portaudio_v18/pablio/pablio.c +++ /dev/null @@ -1,327 +0,0 @@ -/* - * $Id: pablio.c,v 1.1.1.1.4.4 2003/03/13 17:28:33 philburk Exp $ - * pablio.c - * Portable Audio Blocking Input/Output utility. - * - * Author: Phil Burk, http://www.softsynth.com - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.audiomulch.com/portaudio/ - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ - -/* History: - * PLB021214 - check for valid stream in CloseAudioStream() to prevent hang. - * add timeOutMSec to CloseAudioStream() to prevent hang. - */ -#include <stdio.h> -#include <stdlib.h> -#include <math.h> -#include "portaudio.h" -#include "ringbuffer.h" -#include "pablio.h" -#include <string.h> - -/************************************************************************/ -/******** Constants *****************************************************/ -/************************************************************************/ - -#define FRAMES_PER_BUFFER (256) - -/************************************************************************/ -/******** Prototypes ****************************************************/ -/************************************************************************/ - -static int blockingIOCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ); -static PaError PABLIO_InitFIFO( RingBuffer *rbuf, long numFrames, long bytesPerFrame ); -static PaError PABLIO_TermFIFO( RingBuffer *rbuf ); - -/************************************************************************/ -/******** Functions *****************************************************/ -/************************************************************************/ - -/* Called from PortAudio. - * Read and write data only if there is room in FIFOs. - */ -static int blockingIOCallback( void *inputBuffer, void *outputBuffer, - unsigned long framesPerBuffer, - PaTimestamp outTime, void *userData ) -{ - PABLIO_Stream *data = (PABLIO_Stream*)userData; - long numBytes = data->bytesPerFrame * framesPerBuffer; - (void) outTime; - - /* This may get called with NULL inputBuffer during initial setup. */ - if( inputBuffer != NULL ) - { - RingBuffer_Write( &data->inFIFO, inputBuffer, numBytes ); - } - if( outputBuffer != NULL ) - { - int i; - int numRead = RingBuffer_Read( &data->outFIFO, outputBuffer, numBytes ); - /* Zero out remainder of buffer if we run out of data. */ - for( i=numRead; i<numBytes; i++ ) - { - ((char *)outputBuffer)[i] = 0; - } - } - - return 0; -} - -/* Allocate buffer. */ -static PaError PABLIO_InitFIFO( RingBuffer *rbuf, long numFrames, long bytesPerFrame ) -{ - long numBytes = numFrames * bytesPerFrame; - char *buffer = (char *) malloc( numBytes ); - if( buffer == NULL ) return paInsufficientMemory; - memset( buffer, 0, numBytes ); - return (PaError) RingBuffer_Init( rbuf, numBytes, buffer ); -} - -/* Free buffer. */ -static PaError PABLIO_TermFIFO( RingBuffer *rbuf ) -{ - if( rbuf->buffer ) free( rbuf->buffer ); - rbuf->buffer = NULL; - return paNoError; -} - -/************************************************************ - * Write data to ring buffer. - * Will not return until all the data has been written. - */ -long WriteAudioStream( PABLIO_Stream *aStream, void *data, long numFrames ) -{ - long bytesWritten; - char *p = (char *) data; - long numBytes = aStream->bytesPerFrame * numFrames; - while( numBytes > 0) - { - bytesWritten = RingBuffer_Write( &aStream->outFIFO, p, numBytes ); - numBytes -= bytesWritten; - p += bytesWritten; - if( numBytes > 0) Pa_Sleep(10); - } - return numFrames; -} - -/************************************************************ - * Read data from ring buffer. - * Will not return until all the data has been read. - */ -long ReadAudioStream( PABLIO_Stream *aStream, void *data, long numFrames ) -{ - long bytesRead; - char *p = (char *) data; - long numBytes = aStream->bytesPerFrame * numFrames; - while( numBytes > 0) - { - bytesRead = RingBuffer_Read( &aStream->inFIFO, p, numBytes ); - numBytes -= bytesRead; - p += bytesRead; - if( numBytes > 0) Pa_Sleep(10); - } - return numFrames; -} - -/************************************************************ - * Return the number of frames that could be written to the stream without - * having to wait. - */ -long GetAudioStreamWriteable( PABLIO_Stream *aStream ) -{ - int bytesEmpty = RingBuffer_GetWriteAvailable( &aStream->outFIFO ); - return bytesEmpty / aStream->bytesPerFrame; -} - -/************************************************************ - * Return the number of frames that are available to be read from the - * stream without having to wait. - */ -long GetAudioStreamReadable( PABLIO_Stream *aStream ) -{ - int bytesFull = RingBuffer_GetReadAvailable( &aStream->inFIFO ); - return bytesFull / aStream->bytesPerFrame; -} - -/************************************************************/ -static unsigned long RoundUpToNextPowerOf2( unsigned long n ) -{ - long numBits = 0; - if( ((n-1) & n) == 0) return n; /* Already Power of two. */ - while( n > 0 ) - { - n= n>>1; - numBits++; - } - return (1<<numBits); -} - -/************************************************************ - * Opens a PortAudio stream with default characteristics. - * Allocates PABLIO_Stream structure. - * - * flags parameter can be an ORed combination of: - * PABLIO_READ, PABLIO_WRITE, or PABLIO_READ_WRITE, - * and either PABLIO_MONO or PABLIO_STEREO - */ -PaError OpenAudioStream( PABLIO_Stream **rwblPtr, double sampleRate, - PaSampleFormat format, long flags ) -{ - long bytesPerSample; - long doRead = 0; - long doWrite = 0; - PaError err; - PABLIO_Stream *aStream; - long minNumBuffers; - long numFrames; - - /* Allocate PABLIO_Stream structure for caller. */ - aStream = (PABLIO_Stream *) malloc( sizeof(PABLIO_Stream) ); - if( aStream == NULL ) return paInsufficientMemory; - memset( aStream, 0, sizeof(PABLIO_Stream) ); - - /* Determine size of a sample. */ - bytesPerSample = Pa_GetSampleSize( format ); - if( bytesPerSample < 0 ) - { - err = (PaError) bytesPerSample; - goto error; - } - aStream->samplesPerFrame = ((flags&PABLIO_MONO) != 0) ? 1 : 2; - aStream->bytesPerFrame = bytesPerSample * aStream->samplesPerFrame; - - /* Initialize PortAudio */ - err = Pa_Initialize(); - if( err != paNoError ) goto error; - - /* Warning: numFrames must be larger than amount of data processed per interrupt - * inside PA to prevent glitches. Just to be safe, adjust size upwards. - */ - minNumBuffers = 2 * Pa_GetMinNumBuffers( FRAMES_PER_BUFFER, sampleRate ); - numFrames = minNumBuffers * FRAMES_PER_BUFFER; - /* The PortAudio callback runs in a high priority thread. But PABLIO - * runs in a normal foreground thread. So we may have much worse - * latency in PABLIO. So adjust latency to a safe level. - */ - { - const int safeLatencyMSec = 200; - int minLatencyMSec = (int) ((1000 * numFrames) / sampleRate); - if( minLatencyMSec < safeLatencyMSec ) - { - numFrames = (int) ((safeLatencyMSec * sampleRate) / 1000); - } - } - numFrames = RoundUpToNextPowerOf2( numFrames ); - - /* Initialize Ring Buffers */ - doRead = ((flags & PABLIO_READ) != 0); - doWrite = ((flags & PABLIO_WRITE) != 0); - if(doRead) - { - err = PABLIO_InitFIFO( &aStream->inFIFO, numFrames, aStream->bytesPerFrame ); - if( err != paNoError ) goto error; - } - if(doWrite) - { - long numBytes; - err = PABLIO_InitFIFO( &aStream->outFIFO, numFrames, aStream->bytesPerFrame ); - if( err != paNoError ) goto error; - /* Make Write FIFO appear full initially. */ - numBytes = RingBuffer_GetWriteAvailable( &aStream->outFIFO ); - RingBuffer_AdvanceWriteIndex( &aStream->outFIFO, numBytes ); - } - - /* Open a PortAudio stream that we will use to communicate with the underlying - * audio drivers. */ - err = Pa_OpenStream( - &aStream->stream, - (doRead ? Pa_GetDefaultInputDeviceID() : paNoDevice), - (doRead ? aStream->samplesPerFrame : 0 ), - format, - NULL, - (doWrite ? Pa_GetDefaultOutputDeviceID() : paNoDevice), - (doWrite ? aStream->samplesPerFrame : 0 ), - format, - NULL, - sampleRate, - FRAMES_PER_BUFFER, - minNumBuffers, - paClipOff, /* we won't output out of range samples so don't bother clipping them */ - blockingIOCallback, - aStream ); - if( err != paNoError ) goto error; - - err = Pa_StartStream( aStream->stream ); - if( err != paNoError ) goto error; - - *rwblPtr = aStream; - return paNoError; - -error: - CloseAudioStream( aStream ); - *rwblPtr = NULL; - return err; -} - -/************************************************************/ -PaError CloseAudioStream( PABLIO_Stream *aStream ) -{ - PaError err = paNoError; - int bytesEmpty; - int byteSize = aStream->outFIFO.bufferSize; - - if( aStream->stream != NULL ) /* Make sure stream was opened. PLB021214 */ - { - /* If we are writing data, make sure we play everything written. */ - if( byteSize > 0 ) - { - int timeOutMSec = 2000; - bytesEmpty = RingBuffer_GetWriteAvailable( &aStream->outFIFO ); - while( (bytesEmpty < byteSize) && (timeOutMSec > 0) ) - { - Pa_Sleep( 20 ); - timeOutMSec -= 20; - bytesEmpty = RingBuffer_GetWriteAvailable( &aStream->outFIFO ); - } - } - err = Pa_StopStream( aStream->stream ); - if( err != paNoError ) goto error; - err = Pa_CloseStream( aStream->stream ); - } - -error: - Pa_Terminate(); - PABLIO_TermFIFO( &aStream->inFIFO ); - PABLIO_TermFIFO( &aStream->outFIFO ); - free( aStream ); - return err; -} diff --git a/pd/portaudio_v18/pablio/pablio.def b/pd/portaudio_v18/pablio/pablio.def deleted file mode 100644 index 9e2c4e3c..00000000 --- a/pd/portaudio_v18/pablio/pablio.def +++ /dev/null @@ -1,35 +0,0 @@ -LIBRARY PABLIOV18 -DESCRIPTION 'PABLIO Portable Audio Blocking I/O' - -EXPORTS - ; Explicit exports can go here - Pa_Initialize @1 - Pa_Terminate @2 - Pa_GetHostError @3 - Pa_GetErrorText @4 - Pa_CountDevices @5 - Pa_GetDefaultInputDeviceID @6 - Pa_GetDefaultOutputDeviceID @7 - Pa_GetDeviceInfo @8 - Pa_OpenStream @9 - Pa_OpenDefaultStream @10 - Pa_CloseStream @11 - Pa_StartStream @12 - Pa_StopStream @13 - Pa_StreamActive @14 - Pa_StreamTime @15 - Pa_GetCPULoad @16 - Pa_GetMinNumBuffers @17 - Pa_Sleep @18 - - OpenAudioStream @19 - CloseAudioStream @20 - WriteAudioStream @21 - ReadAudioStream @22 - - Pa_GetSampleSize @23 - - ;123456789012345678901234567890123456 - ;000000000111111111122222222223333333 - - diff --git a/pd/portaudio_v18/pablio/pablio.h b/pd/portaudio_v18/pablio/pablio.h deleted file mode 100644 index a4871f38..00000000 --- a/pd/portaudio_v18/pablio/pablio.h +++ /dev/null @@ -1,109 +0,0 @@ -#ifndef _PABLIO_H -#define _PABLIO_H - -#ifdef __cplusplus -extern "C" -{ -#endif /* __cplusplus */ - -/* - * $Id: pablio.h,v 1.1.1.1 2002/01/22 00:52:53 phil Exp $ - * PABLIO.h - * Portable Audio Blocking read/write utility. - * - * Author: Phil Burk, http://www.softsynth.com/portaudio/ - * - * Include file for PABLIO, the Portable Audio Blocking I/O Library. - * PABLIO is built on top of PortAudio, the Portable Audio Library. - * For more information see: http://www.audiomulch.com/portaudio/ - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <stdlib.h> -#include <math.h> -#include "portaudio.h" -#include "ringbuffer.h" -#include <string.h> - -typedef struct -{ - RingBuffer inFIFO; - RingBuffer outFIFO; - PortAudioStream *stream; - int bytesPerFrame; - int samplesPerFrame; -} -PABLIO_Stream; - -/* Values for flags for OpenAudioStream(). */ -#define PABLIO_READ (1<<0) -#define PABLIO_WRITE (1<<1) -#define PABLIO_READ_WRITE (PABLIO_READ|PABLIO_WRITE) -#define PABLIO_MONO (1<<2) -#define PABLIO_STEREO (1<<3) - -/************************************************************ - * Write data to ring buffer. - * Will not return until all the data has been written. - */ -long WriteAudioStream( PABLIO_Stream *aStream, void *data, long numFrames ); - -/************************************************************ - * Read data from ring buffer. - * Will not return until all the data has been read. - */ -long ReadAudioStream( PABLIO_Stream *aStream, void *data, long numFrames ); - -/************************************************************ - * Return the number of frames that could be written to the stream without - * having to wait. - */ -long GetAudioStreamWriteable( PABLIO_Stream *aStream ); - -/************************************************************ - * Return the number of frames that are available to be read from the - * stream without having to wait. - */ -long GetAudioStreamReadable( PABLIO_Stream *aStream ); - -/************************************************************ - * Opens a PortAudio stream with default characteristics. - * Allocates PABLIO_Stream structure. - * - * flags parameter can be an ORed combination of: - * PABLIO_READ, PABLIO_WRITE, or PABLIO_READ_WRITE, - * and either PABLIO_MONO or PABLIO_STEREO - */ -PaError OpenAudioStream( PABLIO_Stream **aStreamPtr, double sampleRate, - PaSampleFormat format, long flags ); - -PaError CloseAudioStream( PABLIO_Stream *aStream ); - -#ifdef __cplusplus -} -#endif /* __cplusplus */ -#endif /* _PABLIO_H */ diff --git a/pd/portaudio_v18/pablio/ringbuffer.c b/pd/portaudio_v18/pablio/ringbuffer.c deleted file mode 100644 index e0c02890..00000000 --- a/pd/portaudio_v18/pablio/ringbuffer.c +++ /dev/null @@ -1,199 +0,0 @@ -/* - * $Id: ringbuffer.c,v 1.1.1.1 2002/01/22 00:52:53 phil Exp $ - * ringbuffer.c - * Ring Buffer utility.. - * - * Author: Phil Burk, http://www.softsynth.com - * - * This program uses the PortAudio Portable Audio Library. - * For more information see: http://www.audiomulch.com/portaudio/ - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <stdlib.h> -#include <math.h> -#include "ringbuffer.h" -#include <string.h> - -/*************************************************************************** - * Initialize FIFO. - * numBytes must be power of 2, returns -1 if not. - */ -long RingBuffer_Init( RingBuffer *rbuf, long numBytes, void *dataPtr ) -{ - if( ((numBytes-1) & numBytes) != 0) return -1; /* Not Power of two. */ - rbuf->bufferSize = numBytes; - rbuf->buffer = (char *)dataPtr; - RingBuffer_Flush( rbuf ); - rbuf->bigMask = (numBytes*2)-1; - rbuf->smallMask = (numBytes)-1; - return 0; -} -/*************************************************************************** -** Return number of bytes available for reading. */ -long RingBuffer_GetReadAvailable( RingBuffer *rbuf ) -{ - return ( (rbuf->writeIndex - rbuf->readIndex) & rbuf->bigMask ); -} -/*************************************************************************** -** Return number of bytes available for writing. */ -long RingBuffer_GetWriteAvailable( RingBuffer *rbuf ) -{ - return ( rbuf->bufferSize - RingBuffer_GetReadAvailable(rbuf)); -} - -/*************************************************************************** -** Clear buffer. Should only be called when buffer is NOT being read. */ -void RingBuffer_Flush( RingBuffer *rbuf ) -{ - rbuf->writeIndex = rbuf->readIndex = 0; -} - -/*************************************************************************** -** Get address of region(s) to which we can write data. -** If the region is contiguous, size2 will be zero. -** If non-contiguous, size2 will be the size of second region. -** Returns room available to be written or numBytes, whichever is smaller. -*/ -long RingBuffer_GetWriteRegions( RingBuffer *rbuf, long numBytes, - void **dataPtr1, long *sizePtr1, - void **dataPtr2, long *sizePtr2 ) -{ - long index; - long available = RingBuffer_GetWriteAvailable( rbuf ); - if( numBytes > available ) numBytes = available; - /* Check to see if write is not contiguous. */ - index = rbuf->writeIndex & rbuf->smallMask; - if( (index + numBytes) > rbuf->bufferSize ) - { - /* Write data in two blocks that wrap the buffer. */ - long firstHalf = rbuf->bufferSize - index; - *dataPtr1 = &rbuf->buffer[index]; - *sizePtr1 = firstHalf; - *dataPtr2 = &rbuf->buffer[0]; - *sizePtr2 = numBytes - firstHalf; - } - else - { - *dataPtr1 = &rbuf->buffer[index]; - *sizePtr1 = numBytes; - *dataPtr2 = NULL; - *sizePtr2 = 0; - } - return numBytes; -} - - -/*************************************************************************** -*/ -long RingBuffer_AdvanceWriteIndex( RingBuffer *rbuf, long numBytes ) -{ - return rbuf->writeIndex = (rbuf->writeIndex + numBytes) & rbuf->bigMask; -} - -/*************************************************************************** -** Get address of region(s) from which we can read data. -** If the region is contiguous, size2 will be zero. -** If non-contiguous, size2 will be the size of second region. -** Returns room available to be written or numBytes, whichever is smaller. -*/ -long RingBuffer_GetReadRegions( RingBuffer *rbuf, long numBytes, - void **dataPtr1, long *sizePtr1, - void **dataPtr2, long *sizePtr2 ) -{ - long index; - long available = RingBuffer_GetReadAvailable( rbuf ); - if( numBytes > available ) numBytes = available; - /* Check to see if read is not contiguous. */ - index = rbuf->readIndex & rbuf->smallMask; - if( (index + numBytes) > rbuf->bufferSize ) - { - /* Write data in two blocks that wrap the buffer. */ - long firstHalf = rbuf->bufferSize - index; - *dataPtr1 = &rbuf->buffer[index]; - *sizePtr1 = firstHalf; - *dataPtr2 = &rbuf->buffer[0]; - *sizePtr2 = numBytes - firstHalf; - } - else - { - *dataPtr1 = &rbuf->buffer[index]; - *sizePtr1 = numBytes; - *dataPtr2 = NULL; - *sizePtr2 = 0; - } - return numBytes; -} -/*************************************************************************** -*/ -long RingBuffer_AdvanceReadIndex( RingBuffer *rbuf, long numBytes ) -{ - return rbuf->readIndex = (rbuf->readIndex + numBytes) & rbuf->bigMask; -} - -/*************************************************************************** -** Return bytes written. */ -long RingBuffer_Write( RingBuffer *rbuf, void *data, long numBytes ) -{ - long size1, size2, numWritten; - void *data1, *data2; - numWritten = RingBuffer_GetWriteRegions( rbuf, numBytes, &data1, &size1, &data2, &size2 ); - if( size2 > 0 ) - { - - memcpy( data1, data, size1 ); - data = ((char *)data) + size1; - memcpy( data2, data, size2 ); - } - else - { - memcpy( data1, data, size1 ); - } - RingBuffer_AdvanceWriteIndex( rbuf, numWritten ); - return numWritten; -} - -/*************************************************************************** -** Return bytes read. */ -long RingBuffer_Read( RingBuffer *rbuf, void *data, long numBytes ) -{ - long size1, size2, numRead; - void *data1, *data2; - numRead = RingBuffer_GetReadRegions( rbuf, numBytes, &data1, &size1, &data2, &size2 ); - if( size2 > 0 ) - { - memcpy( data, data1, size1 ); - data = ((char *)data) + size1; - memcpy( data, data2, size2 ); - } - else - { - memcpy( data, data1, size1 ); - } - RingBuffer_AdvanceReadIndex( rbuf, numRead ); - return numRead; -} diff --git a/pd/portaudio_v18/pablio/ringbuffer.h b/pd/portaudio_v18/pablio/ringbuffer.h deleted file mode 100644 index 02dd40c3..00000000 --- a/pd/portaudio_v18/pablio/ringbuffer.h +++ /dev/null @@ -1,102 +0,0 @@ -#ifndef _RINGBUFFER_H -#define _RINGBUFFER_H -#ifdef __cplusplus -extern "C" -{ -#endif /* __cplusplus */ - -/* - * $Id: ringbuffer.h,v 1.1.1.1.4.2 2003/04/28 17:45:34 philburk Exp $ - * ringbuffer.h - * Ring Buffer utility.. - * - * Author: Phil Burk, http://www.softsynth.com - * - * This program is distributed with the PortAudio Portable Audio Library. - * For more information see: http://www.audiomulch.com/portaudio/ - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ -#include <stdio.h> -#include <stdlib.h> -#include <math.h> -#include "ringbuffer.h" -#include <string.h> - -typedef struct -{ - long bufferSize; /* Number of bytes in FIFO. Power of 2. Set by RingBuffer_Init. */ -/* These are declared volatile because they are written by a different thread than the reader. */ - volatile long writeIndex; /* Index of next writable byte. Set by RingBuffer_AdvanceWriteIndex. */ - volatile long readIndex; /* Index of next readable byte. Set by RingBuffer_AdvanceReadIndex. */ - long bigMask; /* Used for wrapping indices with extra bit to distinguish full/empty. */ - long smallMask; /* Used for fitting indices to buffer. */ - char *buffer; -} -RingBuffer; -/* - * Initialize Ring Buffer. - * numBytes must be power of 2, returns -1 if not. - */ -long RingBuffer_Init( RingBuffer *rbuf, long numBytes, void *dataPtr ); - -/* Clear buffer. Should only be called when buffer is NOT being read. */ -void RingBuffer_Flush( RingBuffer *rbuf ); - -/* Return number of bytes available for writing. */ -long RingBuffer_GetWriteAvailable( RingBuffer *rbuf ); -/* Return number of bytes available for read. */ -long RingBuffer_GetReadAvailable( RingBuffer *rbuf ); -/* Return bytes written. */ -long RingBuffer_Write( RingBuffer *rbuf, void *data, long numBytes ); -/* Return bytes read. */ -long RingBuffer_Read( RingBuffer *rbuf, void *data, long numBytes ); - -/* Get address of region(s) to which we can write data. -** If the region is contiguous, size2 will be zero. -** If non-contiguous, size2 will be the size of second region. -** Returns room available to be written or numBytes, whichever is smaller. -*/ -long RingBuffer_GetWriteRegions( RingBuffer *rbuf, long numBytes, - void **dataPtr1, long *sizePtr1, - void **dataPtr2, long *sizePtr2 ); -long RingBuffer_AdvanceWriteIndex( RingBuffer *rbuf, long numBytes ); - -/* Get address of region(s) from which we can read data. -** If the region is contiguous, size2 will be zero. -** If non-contiguous, size2 will be the size of second region. -** Returns room available to be read or numBytes, whichever is smaller. -*/ -long RingBuffer_GetReadRegions( RingBuffer *rbuf, long numBytes, - void **dataPtr1, long *sizePtr1, - void **dataPtr2, long *sizePtr2 ); - -long RingBuffer_AdvanceReadIndex( RingBuffer *rbuf, long numBytes ); - -#ifdef __cplusplus -} -#endif /* __cplusplus */ -#endif /* _RINGBUFFER_H */ diff --git a/pd/portaudio_v18/pablio/test_rw.c b/pd/portaudio_v18/pablio/test_rw.c deleted file mode 100644 index 27a94b43..00000000 --- a/pd/portaudio_v18/pablio/test_rw.c +++ /dev/null @@ -1,99 +0,0 @@ -/* - * $Id: test_rw.c,v 1.2 2002/02/22 22:06:23 philburk Exp $ - * test_rw.c - * Read input from one stream and write it to another. - * - * Author: Phil Burk, http://www.softsynth.com/portaudio/ - * - * This program uses PABLIO, the Portable Audio Blocking I/O Library. - * PABLIO is built on top of PortAudio, the Portable Audio Library. - * For more information see: http://www.audiomulch.com/portaudio/ - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ - -#include <stdio.h> -#include <stdlib.h> -#include "pablio.h" - -/* -** Note that many of the older ISA sound cards on PCs do NOT support -** full duplex audio (simultaneous record and playback). -** And some only support full duplex at lower sample rates. -*/ -#define SAMPLE_RATE (44100) -#define NUM_SECONDS (5) -#define SAMPLES_PER_FRAME (2) -#define FRAMES_PER_BLOCK (64) - -/* Select whether we will use floats or shorts. */ -#if 1 -#define SAMPLE_TYPE paFloat32 -typedef float SAMPLE; -#else -#define SAMPLE_TYPE paInt16 -typedef short SAMPLE; -#endif - -/*******************************************************************/ -int main(void); -int main(void) -{ - int i; - SAMPLE samples[SAMPLES_PER_FRAME * FRAMES_PER_BLOCK]; - PaError err; - PABLIO_Stream *aStream; - - printf("Full duplex sound test using PortAudio and RingBuffers\n"); - fflush(stdout); - - /* Open simplified blocking I/O layer on top of PortAudio. */ - err = OpenAudioStream( &aStream, SAMPLE_RATE, SAMPLE_TYPE, - (PABLIO_READ_WRITE | PABLIO_STEREO) ); - if( err != paNoError ) goto error; - - /* Process samples in the foreground. */ - for( i=0; i<(NUM_SECONDS * SAMPLE_RATE); i += FRAMES_PER_BLOCK ) - { - /* Read one block of data into sample array from audio input. */ - ReadAudioStream( aStream, samples, FRAMES_PER_BLOCK ); - /* Write that same block of data to output. */ - WriteAudioStream( aStream, samples, FRAMES_PER_BLOCK ); - } - - CloseAudioStream( aStream ); - - printf("Full duplex sound test complete.\n" ); - fflush(stdout); - return 0; - -error: - Pa_Terminate(); - fprintf( stderr, "An error occured while using the portaudio stream\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return -1; -} diff --git a/pd/portaudio_v18/pablio/test_rw_echo.c b/pd/portaudio_v18/pablio/test_rw_echo.c deleted file mode 100644 index 7bc4e9b4..00000000 --- a/pd/portaudio_v18/pablio/test_rw_echo.c +++ /dev/null @@ -1,123 +0,0 @@ -/* - * $Id: test_rw_echo.c,v 1.1.1.1 2002/01/22 00:52:54 phil Exp $ - * test_rw_echo.c - * Echo delayed input to output. - * - * Author: Phil Burk, http://www.softsynth.com/portaudio/ - * - * This program uses PABLIO, the Portable Audio Blocking I/O Library. - * PABLIO is built on top of PortAudio, the Portable Audio Library. - * - * Note that if you need low latency, you should not use PABLIO. - * Use the PA_OpenStream callback technique which is lower level - * than PABLIO. - * - * For more information see: http://www.audiomulch.com/portaudio/ - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ - -#include <stdio.h> -#include <stdlib.h> -#include <math.h> -#include "pablio.h" -#include <string.h> - -/* -** Note that many of the older ISA sound cards on PCs do NOT support -** full duplex audio (simultaneous record and playback). -** And some only support full duplex at lower sample rates. -*/ -#define SAMPLE_RATE (22050) -#define NUM_SECONDS (20) -#define SAMPLES_PER_FRAME (2) - -/* Select whether we will use floats or shorts. */ -#if 1 -#define SAMPLE_TYPE paFloat32 -typedef float SAMPLE; -#else -#define SAMPLE_TYPE paInt16 -typedef short SAMPLE; -#endif - -#define NUM_ECHO_FRAMES (2*SAMPLE_RATE) -SAMPLE samples[NUM_ECHO_FRAMES][SAMPLES_PER_FRAME] = {0.0}; - -/*******************************************************************/ -int main(void); -int main(void) -{ - int i; - PaError err; - PABLIO_Stream *aInStream; - PABLIO_Stream *aOutStream; - int index; - - printf("Full duplex sound test using PABLIO\n"); - fflush(stdout); - - /* Open simplified blocking I/O layer on top of PortAudio. */ - /* Open input first so it can start to fill buffers. */ - err = OpenAudioStream( &aInStream, SAMPLE_RATE, SAMPLE_TYPE, - (PABLIO_READ | PABLIO_STEREO) ); - if( err != paNoError ) goto error; - /* printf("opened input\n"); fflush(stdout); /**/ - - err = OpenAudioStream( &aOutStream, SAMPLE_RATE, SAMPLE_TYPE, - (PABLIO_WRITE | PABLIO_STEREO) ); - if( err != paNoError ) goto error; - /* printf("opened output\n"); fflush(stdout); /**/ - - /* Process samples in the foreground. */ - index = 0; - for( i=0; i<(NUM_SECONDS * SAMPLE_RATE); i++ ) - { - /* Write old frame of data to output. */ - /* samples[index][1] = (i&256) * (1.0f/256.0f); /* sawtooth */ - WriteAudioStream( aOutStream, &samples[index][0], 1 ); - - /* Read one frame of data into sample array for later output. */ - ReadAudioStream( aInStream, &samples[index][0], 1 ); - index += 1; - if( index >= NUM_ECHO_FRAMES ) index = 0; - - if( (i & 0xFFFF) == 0 ) printf("i = %d\n", i ); fflush(stdout); /**/ - } - - CloseAudioStream( aOutStream ); - CloseAudioStream( aInStream ); - - printf("R/W echo sound test complete.\n" ); - fflush(stdout); - return 0; - -error: - fprintf( stderr, "An error occured while using PortAudio\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return -1; -} diff --git a/pd/portaudio_v18/pablio/test_w_saw.c b/pd/portaudio_v18/pablio/test_w_saw.c deleted file mode 100644 index ca727aa2..00000000 --- a/pd/portaudio_v18/pablio/test_w_saw.c +++ /dev/null @@ -1,108 +0,0 @@ -/* - * $Id: test_w_saw.c,v 1.1.1.1.4.1 2003/02/12 02:22:21 philburk Exp $ - * test_w_saw.c - * Generate stereo sawtooth waveforms. - * - * Author: Phil Burk, http://www.softsynth.com - * - * This program uses PABLIO, the Portable Audio Blocking I/O Library. - * PABLIO is built on top of PortAudio, the Portable Audio Library. - * - * For more information see: http://www.audiomulch.com/portaudio/ - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ - -#include <stdio.h> -#include <stdlib.h> -#include <math.h> -#include "pablio.h" -#include <string.h> - -#define SAMPLE_RATE (44100) -#define NUM_SECONDS (15) -#define SAMPLES_PER_FRAME (2) - -#define FREQUENCY (220.0f) -#define PHASE_INCREMENT (2.0f * FREQUENCY / SAMPLE_RATE) -#define FRAMES_PER_BLOCK (100) - -float samples[FRAMES_PER_BLOCK][SAMPLES_PER_FRAME]; -float phases[SAMPLES_PER_FRAME]; - -/*******************************************************************/ -int main(void); -int main(void) -{ - int i,j; - PaError err; - PABLIO_Stream *aOutStream; - - printf("Generate sawtooth waves using PABLIO.\n"); - fflush(stdout); - - /* Open simplified blocking I/O layer on top of PortAudio. */ - err = OpenAudioStream( &aOutStream, SAMPLE_RATE, paFloat32, - (PABLIO_WRITE | PABLIO_STEREO) ); - if( err != paNoError ) goto error; - - /* Initialize oscillator phases. */ - phases[0] = 0.0; - phases[1] = 0.0; - - for( i=0; i<(NUM_SECONDS * SAMPLE_RATE); i += FRAMES_PER_BLOCK ) - { - /* Generate sawtooth waveforms in a block for efficiency. */ - for( j=0; j<FRAMES_PER_BLOCK; j++ ) - { - /* Generate a sawtooth wave by incrementing a variable. */ - phases[0] += PHASE_INCREMENT; - /* The signal range is -1.0 to +1.0 so wrap around if we go over. */ - if( phases[0] > 1.0f ) phases[0] -= 2.0f; - samples[j][0] = phases[0]; - - /* On the second channel, generate a sawtooth wave a fifth higher. */ - phases[1] += PHASE_INCREMENT * (3.0f / 2.0f); - if( phases[1] > 1.0f ) phases[1] -= 2.0f; - samples[j][1] = phases[1]; - } - - /* Write samples to output. */ - WriteAudioStream( aOutStream, samples, FRAMES_PER_BLOCK ); - } - - CloseAudioStream( aOutStream ); - - printf("Sawtooth sound test complete.\n" ); - fflush(stdout); - return 0; - -error: - fprintf( stderr, "An error occured while using PABLIO\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return -1; -} diff --git a/pd/portaudio_v18/pablio/test_w_saw8.c b/pd/portaudio_v18/pablio/test_w_saw8.c deleted file mode 100644 index 0f7e02e3..00000000 --- a/pd/portaudio_v18/pablio/test_w_saw8.c +++ /dev/null @@ -1,106 +0,0 @@ -/* - * $Id: test_w_saw8.c,v 1.1.1.1 2002/01/22 00:52:55 phil Exp $ - * test_w_saw8.c - * Generate stereo 8 bit sawtooth waveforms. - * - * Author: Phil Burk, http://www.softsynth.com - * - * This program uses PABLIO, the Portable Audio Blocking I/O Library. - * PABLIO is built on top of PortAudio, the Portable Audio Library. - * - * For more information see: http://www.audiomulch.com/portaudio/ - * Copyright (c) 1999-2000 Ross Bencina and Phil Burk - * - * Permission is hereby granted, free of charge, to any person obtaining - * a copy of this software and associated documentation files - * (the "Software"), to deal in the Software without restriction, - * including without limitation the rights to use, copy, modify, merge, - * publish, distribute, sublicense, and/or sell copies of the Software, - * and to permit persons to whom the Software is furnished to do so, - * subject to the following conditions: - * - * The above copyright notice and this permission notice shall be - * included in all copies or substantial portions of the Software. - * - * Any person wishing to distribute modifications to the Software is - * requested to send the modifications to the original developer so that - * they can be incorporated into the canonical version. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. - * - */ - -#include <stdio.h> -#include <stdlib.h> -#include <math.h> -#include "pablio.h" -#include <string.h> - -#define SAMPLE_RATE (22050) -#define NUM_SECONDS (6) -#define SAMPLES_PER_FRAME (2) - - -#define FRAMES_PER_BLOCK (100) - -unsigned char samples[FRAMES_PER_BLOCK][SAMPLES_PER_FRAME]; -unsigned char phases[SAMPLES_PER_FRAME]; - -/*******************************************************************/ -int main(void); -int main(void) -{ - int i,j; - PaError err; - PABLIO_Stream *aOutStream; - - printf("Generate unsigned 8 bit sawtooth waves using PABLIO.\n"); - fflush(stdout); - - /* Open simplified blocking I/O layer on top of PortAudio. */ - err = OpenAudioStream( &aOutStream, SAMPLE_RATE, paUInt8, - (PABLIO_WRITE | PABLIO_STEREO) ); - if( err != paNoError ) goto error; - - /* Initialize oscillator phases to "ground" level for paUInt8. */ - phases[0] = 128; - phases[1] = 128; - - for( i=0; i<(NUM_SECONDS * SAMPLE_RATE); i += FRAMES_PER_BLOCK ) - { - /* Generate sawtooth waveforms in a block for efficiency. */ - for( j=0; j<FRAMES_PER_BLOCK; j++ ) - { - /* Generate a sawtooth wave by incrementing a variable. */ - phases[0] += 1; - /* We don't have to do anything special to wrap when using paUint8 because - * 8 bit arithmetic automatically wraps. */ - samples[j][0] = phases[0]; - - /* On the second channel, generate a higher sawtooth wave. */ - phases[1] += 3; - samples[j][1] = phases[1]; - } - - /* Write samples to output. */ - WriteAudioStream( aOutStream, samples, FRAMES_PER_BLOCK ); - } - - CloseAudioStream( aOutStream ); - - printf("Sawtooth sound test complete.\n" ); - fflush(stdout); - return 0; - -error: - fprintf( stderr, "An error occured while using PABLIO\n" ); - fprintf( stderr, "Error number: %d\n", err ); - fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); - return -1; -} diff --git a/pd/portaudio_v18/testcvs/changeme.txt b/pd/portaudio_v18/testcvs/changeme.txt deleted file mode 100644 index 2866c3b7..00000000 --- a/pd/portaudio_v18/testcvs/changeme.txt +++ /dev/null @@ -1,8 +0,0 @@ -This is just a dopy little file used to test the CVS repository. -Feel free to trash this file. -Minor change. -Another tweak. -philburk tweak -stephane test -Phil changed this again on 2/21/02. Yawn... - |