From dad636821f6e7d3ead02c157f308c0ceeba9af3d Mon Sep 17 00:00:00 2001 From: Miller Puckette Date: Mon, 6 Sep 2004 19:45:05 +0000 Subject: Removed files which appear to be unused. svn path=/trunk/; revision=2009 --- pd/extra/bonk~/README.txt | 30 - pd/extra/choice/README.txt | 12 - pd/extra/fiddle~/README.txt | 33 - pd/portaudio/pa_linux_alsa/blocking_calls.o | Bin 1180 -> 0 bytes pd/portaudio/pa_linux_alsa/callback_thread.o | Bin 4180 -> 0 bytes pd/portaudio/pablio/pablio_pd.o | Bin 3500 -> 0 bytes pd/portaudio/pablio/ringbuffer_pd.o | Bin 1824 -> 0 bytes pd/portaudio_v18/VERSION.txt | 2 - pd/portaudio_v18/pablio/pablio_pd.c | 340 ---------- pd/portaudio_v18/pablio/pablio_pd.h | 110 ---- pd/portaudio_v18/pablio/ringbuffer_pd.c | 217 ------ pd/portaudio_v18/pablio/test_w_saw_pd.c | 108 --- pd/portmidi_osx/MSP-README.txt | 3 - pd/src/makefile.mingw | 133 ---- pd/src/makefile.nt.pa19 | 185 ------ pd/src/s_audio_alsa.c.old | 946 --------------------------- 16 files changed, 2119 deletions(-) delete mode 100644 pd/extra/bonk~/README.txt delete mode 100644 pd/extra/choice/README.txt delete mode 100644 pd/extra/fiddle~/README.txt delete mode 100644 pd/portaudio/pa_linux_alsa/blocking_calls.o delete mode 100644 pd/portaudio/pa_linux_alsa/callback_thread.o delete mode 100644 pd/portaudio/pablio/pablio_pd.o delete mode 100644 pd/portaudio/pablio/ringbuffer_pd.o delete mode 100644 pd/portaudio_v18/VERSION.txt delete mode 100644 pd/portaudio_v18/pablio/pablio_pd.c delete mode 100644 pd/portaudio_v18/pablio/pablio_pd.h delete mode 100644 pd/portaudio_v18/pablio/ringbuffer_pd.c delete mode 100644 pd/portaudio_v18/pablio/test_w_saw_pd.c delete mode 100644 pd/portmidi_osx/MSP-README.txt delete mode 100755 pd/src/makefile.mingw delete mode 100644 pd/src/makefile.nt.pa19 delete mode 100644 pd/src/s_audio_alsa.c.old (limited to 'pd') diff --git a/pd/extra/bonk~/README.txt b/pd/extra/bonk~/README.txt deleted file mode 100644 index 6b0017cd..00000000 --- a/pd/extra/bonk~/README.txt +++ /dev/null @@ -1,30 +0,0 @@ -Bonk is copyright (C) 1997 Regents of the University of California. -Permission is granted to use this software for any purpose, commercial -or noncommercial, as long as this notice is included with all copies. - -UC MAKES NO WARRANTY, EXPRESS OR IMPLIED, IN CONNECTION WITH THIS SOFTWARE! - ----------------------------------------------------------------------------- - -This is the README file for the "bonk" percussion detector. This software -is available from http://man104nfs.ucsd.edu/~mpuckett in versions for -IRIX 5.x and for NT on Intel boxes. - -Bonk will soon be available separately for Max/MSP on Macintoshes. - -TO INSTALL FOR IRIX 5.x: download from the Web page, which will give you a -file named "bonk-x.xx.tar.Z" (where x.xx is the version number). Unpack this -by typing "zcat bonk-x.xx.tar.Z | tar xf -" which will give you a directory -named "bonk" containing the source, the object code, and the "help patch." - -TO INSTALL FOR NT: download from the Web page, which will give you a file -named "bonk-x.xx.zip" (where x.xx is the version number). Unpack this by -typing "unzip bonk-x.xx.zip" to a terminal window which will give you a -directory named "bonk". The source, the object code, and the "help patch" -will be there. - -Pd currently has no search path facility; the object file (bonk.pd_irix5 or -bonk.dll) should be copied to the directories containing whatever patches you -want to use it with. - --Miller Puckette (msp@ucsd.edu) diff --git a/pd/extra/choice/README.txt b/pd/extra/choice/README.txt deleted file mode 100644 index 3c3ed132..00000000 --- a/pd/extra/choice/README.txt +++ /dev/null @@ -1,12 +0,0 @@ -Choice is copyright (C) 1999 Miller Puckette. -Permission is granted to use this software for any purpose, commercial -or noncommercial, as long as this notice is included with all copies. - -NEITHER THE AUTHORS NOR THEIR EMPLOYERS MAKE ANY WARRANTY, EXPRESS OR IMPLIED, -IN CONNECTION WITH THIS SOFTWARE! - ----------------------------------------------------------------------------- - -This is the README file for the "choice" object. This software -is available from http://www.crca.ucsd.edu/~msp as part of the "toys" -library. - msp@ucsd.edu diff --git a/pd/extra/fiddle~/README.txt b/pd/extra/fiddle~/README.txt deleted file mode 100644 index 82d2ad68..00000000 --- a/pd/extra/fiddle~/README.txt +++ /dev/null @@ -1,33 +0,0 @@ -Fiddle is copyright (C) 1997 Regents of the University of California. -Permission is granted to use this software for any noncommercial purpose. -For commercial licensing contact the UCSD Technology Transfer Office. - -UC MAKES NO WARRANTY, EXPRESS OR IMPLIED, IN CONNECTION WITH THIS SOFTWARE! - ----------------------------------------------------------------------------- - -This is the README file for the "fiddle" audio pitch detector. This software -is available from http://man104nfs.ucsd.edu/~mpuckett in versions for -IRIX 5.x and for NT on Intel boxes. - -Fiddle will soon be available separately for Max/MSP on Macintoshes. - -TO INSTALL FOR IRIX 5.x: download from the Web page, which will give you a -file named "fiddle-x.xx.tar.Z" (where x.xx is the version number). Unpack this -by typing "zcat fiddle-x.xx.tar.Z | tar xf -" which will give you a directory -named "fiddle" containing the source, the object code, and the "help patch." - -TO INSTALL FOR NT: download from the Web page, which will give you a file -named "fiddle-x.xx.zip" (where x.xx is the version number). Unpack this by -typing "unzip fiddle-x.xx.zip" which will give you a directory named "fiddle". -The source, the object code, and the "help patch" will be there. - -Pd currently has no search path facility; the object file (fiddle.pd_irix5 or -fiddle.dll) should be copied to the directorys containing whatever patches you -want to use it with. - -Please note that the copyright notice on Fiddle is more restrictive than for -Pd; this is to prevent Fiddle from being incorporated in any commercial product -that could compete with IRCAM's audio analysis tools. - --Miller Puckette (msp@ucsd.edu) diff --git a/pd/portaudio/pa_linux_alsa/blocking_calls.o b/pd/portaudio/pa_linux_alsa/blocking_calls.o deleted file mode 100644 index d4bc2108..00000000 Binary files a/pd/portaudio/pa_linux_alsa/blocking_calls.o and /dev/null differ diff --git a/pd/portaudio/pa_linux_alsa/callback_thread.o b/pd/portaudio/pa_linux_alsa/callback_thread.o deleted file mode 100644 index a242d490..00000000 Binary files a/pd/portaudio/pa_linux_alsa/callback_thread.o and /dev/null differ diff --git a/pd/portaudio/pablio/pablio_pd.o b/pd/portaudio/pablio/pablio_pd.o deleted file mode 100644 index 359a8c49..00000000 Binary files a/pd/portaudio/pablio/pablio_pd.o and /dev/null differ diff --git a/pd/portaudio/pablio/ringbuffer_pd.o b/pd/portaudio/pablio/ringbuffer_pd.o deleted file mode 100644 index a348b1ed..00000000 Binary files a/pd/portaudio/pablio/ringbuffer_pd.o and /dev/null differ diff --git a/pd/portaudio_v18/VERSION.txt b/pd/portaudio_v18/VERSION.txt deleted file mode 100644 index 350a736a..00000000 --- a/pd/portaudio_v18/VERSION.txt +++ /dev/null @@ -1,2 +0,0 @@ -Portaudio version 18.1, packaged 03.05.28, downloaded Aug. 1: --rw------- 1 msp 550130 Aug 1 19:47 portaudio_v18_1.zip diff --git a/pd/portaudio_v18/pablio/pablio_pd.c b/pd/portaudio_v18/pablio/pablio_pd.c deleted file mode 100644 index 24b12ff1..00000000 --- a/pd/portaudio_v18/pablio/pablio_pd.c +++ /dev/null @@ -1,340 +0,0 @@ -/* - * $Id: pablio_pd.c,v 1.2 2004-02-22 16:21:33 ggeiger 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. - * - */ - - /* changes by Miller Puckette to support Pd: device selection, - settable audio buffer size, and settable number of channels. - LATER also fix it to poll for input and output fifo fill points. */ -#include -#include -#include -#include "portaudio.h" -#include "ringbuffer.h" -#include "pablio_pd.h" /* MSP */ -#include - - /* MSP -- FRAMES_PER_BUFFER constant removed */ -static void NPa_Sleep(int n) /* MSP wrapper to check we never stall... */ -{ -#if 0 - fprintf(stderr, "sleep\n"); -#endif - Pa_Sleep(n); -} - -/************************************************************************/ -/******** 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; ibuffer ) 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) NPa_Sleep(10); /* MSP */ - } - 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) NPa_Sleep(10); /* MSP */ - } - 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<samplesPerFrame = nchannels; /* MSP */ - 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. */ /* MSP */ - minNumBuffers = Pa_GetMinNumBuffers(framesperbuf, sampleRate); - if (minNumBuffers > nbuffers) - fprintf(stderr, - "warning: number of buffers %d less than recommended minimum %d\n", - (int)nbuffers, (int)minNumBuffers); - numFrames = nbuffers * framesperbuf; - /* fprintf(stderr, "numFrames %d\n", 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 ? indeviceno : paNoDevice), /* MSP */ - (doRead ? aStream->samplesPerFrame : 0 ), - format, - NULL, - (doWrite ? outdeviceno : paNoDevice), /* MSP */ - (doWrite ? aStream->samplesPerFrame : 0 ), - format, - NULL, - sampleRate, - framesperbuf, /* MSP */ - nbuffers, /* MSP */ - paNoFlag, /* MSP -- portaudio will clip for us */ - blockingIOCallback, - aStream ); - if( err != paNoError ) goto error; - - err = Pa_StartStream( aStream->stream ); - if( err != paNoError ) /* MSP */ - { - fprintf(stderr, "Pa_StartStream failed; closing audio stream...\n"); - CloseAudioStream( aStream ); - goto error; - } - - *rwblPtr = aStream; - return paNoError; - -error: - *rwblPtr = NULL; - return err; -} - -/************************************************************/ -PaError CloseAudioStream( PABLIO_Stream *aStream ) -{ - PaError err; - int bytesEmpty; - int byteSize = aStream->outFIFO.bufferSize; - - /* If we are writing data, make sure we play everything written. */ - if( byteSize > 0 ) - { - bytesEmpty = RingBuffer_GetWriteAvailable( &aStream->outFIFO ); - while( bytesEmpty < byteSize ) - { - NPa_Sleep( 10 ); /* MSP */ - bytesEmpty = RingBuffer_GetWriteAvailable( &aStream->outFIFO ); - } - } - - err = Pa_StopStream( aStream->stream ); - if( err != paNoError ) goto error; - err = Pa_CloseStream( aStream->stream ); - if( err != paNoError ) goto error; - Pa_Terminate(); - -error: - PABLIO_TermFIFO( &aStream->inFIFO ); - PABLIO_TermFIFO( &aStream->outFIFO ); - free( aStream ); - return err; -} diff --git a/pd/portaudio_v18/pablio/pablio_pd.h b/pd/portaudio_v18/pablio/pablio_pd.h deleted file mode 100644 index 7f01dc91..00000000 --- a/pd/portaudio_v18/pablio/pablio_pd.h +++ /dev/null @@ -1,110 +0,0 @@ -#ifndef _PABLIO_H -#define _PABLIO_H - -#ifdef __cplusplus -extern "C" -{ -#endif /* __cplusplus */ - -/* - * $Id: pablio_pd.h,v 1.2 2004-02-22 16:21:33 ggeiger 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 -#include -#include -#include "portaudio.h" -#include "ringbuffer.h" -#include - -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, - */ -PaError OpenAudioStream( PABLIO_Stream **rwblPtr, double sampleRate, - PaSampleFormat format, long flags, int nchannels, - int framesperbuf, int nbuffers, - int indeviceno, int outdeviceno); /* MSP */ - -PaError CloseAudioStream( PABLIO_Stream *aStream ); - -#ifdef __cplusplus -} -#endif /* __cplusplus */ -#endif /* _PABLIO_H */ diff --git a/pd/portaudio_v18/pablio/ringbuffer_pd.c b/pd/portaudio_v18/pablio/ringbuffer_pd.c deleted file mode 100644 index a008ca3f..00000000 --- a/pd/portaudio_v18/pablio/ringbuffer_pd.c +++ /dev/null @@ -1,217 +0,0 @@ -/* - * $Id: ringbuffer_pd.c,v 1.2 2004-02-22 16:21:33 ggeiger 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. - * - */ -/* - * modified 2002/07/13 by olaf.matthes@gmx.de to allow any number if channels - * - */ - -#include -#include -#include -#include "ringbuffer.h" -#include - -/*************************************************************************** - * Initialize FIFO. - */ -long RingBuffer_Init( RingBuffer *rbuf, long numBytes, void *dataPtr ) -{ - rbuf->bufferSize = numBytes; - rbuf->buffer = (char *)dataPtr; - RingBuffer_Flush( rbuf ); - return 0; -} -/*************************************************************************** -** Return number of bytes available for reading. */ -long RingBuffer_GetReadAvailable( RingBuffer *rbuf ) -{ - long ret = rbuf->writeIndex - rbuf->readIndex; - if (ret < 0) - ret += 2 * rbuf->bufferSize; - if (ret < 0 || ret > rbuf->bufferSize) - fprintf(stderr, - "consistency check failed: RingBuffer_GetReadAvailable\n"); - return ( ret ); -} -/*************************************************************************** -** 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; - while (index >= rbuf->bufferSize) - index -= rbuf->bufferSize; - 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 ) -{ - long ret = (rbuf->writeIndex + numBytes); - if ( ret >= 2 * rbuf->bufferSize) - ret -= 2 * rbuf->bufferSize; /* check for end of buffer */ - return rbuf->writeIndex = ret; -} - -/*************************************************************************** -** 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; - while (index >= rbuf->bufferSize) - index -= rbuf->bufferSize; - - 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 ) -{ - long ret = (rbuf->readIndex + numBytes); - if( ret >= 2 * rbuf->bufferSize) - ret -= 2 * rbuf->bufferSize; - return rbuf->readIndex = ret; -} - -/*************************************************************************** -** 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/test_w_saw_pd.c b/pd/portaudio_v18/pablio/test_w_saw_pd.c deleted file mode 100644 index 73a4d9c4..00000000 --- a/pd/portaudio_v18/pablio/test_w_saw_pd.c +++ /dev/null @@ -1,108 +0,0 @@ -/* - * $Id: test_w_saw_pd.c,v 1.2 2004-02-22 16:21:33 ggeiger 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 -#include -#include -#include "pablio_pd.h" -#include - -#define SAMPLE_RATE (44100) -#define NUM_SECONDS (6) -#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, 2, 512, 8, -1, -1 ); - 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 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/portmidi_osx/MSP-README.txt b/pd/portmidi_osx/MSP-README.txt deleted file mode 100644 index c48e8c8e..00000000 --- a/pd/portmidi_osx/MSP-README.txt +++ /dev/null @@ -1,3 +0,0 @@ -This is from a PortMidi pre-release for OSX. - --MSP diff --git a/pd/src/makefile.mingw b/pd/src/makefile.mingw deleted file mode 100755 index 1d93cdcd..00000000 --- a/pd/src/makefile.mingw +++ /dev/null @@ -1,133 +0,0 @@ -VPATH = ../obj:./ -INSTDIR = .. -OBJ_DIR = $(INSTDIR)/src -BIN_DIR = $(INSTDIR)/bin - -PDEXEC = pd.exe -GUINAME= pdtcl.dll -DLLWRAP= dllwrap - -GFLAGS = -DINSTALL_PREFIX=\"$(INSTDIR)\" - -MORECFLAGS = -g - -PADIR = $(INSTDIR)/portaudio -INCPA = -I$(PADIR) -I$(PADIR)/pa_common -I$(PADIR)/pablio -I../lib/asio -INCLUDE = -GINCLUDE = $(INCLUDE) -I../tcl/include - -LDFLAGS = -lpthreadGC -lwsock32 -lportaudio -lwinmm /lib/tcl85.lib /lib/tk85.lib -LIB = -lm - -OPT_CFLAGS = -WARN_CFLAGS = -Wall -W -Wstrict-prototypes \ - -Wno-unused-parameter -Wno-parentheses -Wno-switch -ARCH_CFLAGS = -DPD -DPD_INTERNAL -DMSW -DUSEAPI_MMIO -DUSEAPI_PORTAUDIO \ - -CFLAGS += $(ARCH_CFLAGS) $(WARN_CFLAGS) $(OPT_CFLAGS) $(MORECFLAGS) - -# the sources - -SYSSRC += s_audio_pa.c s_audio_mmio.c s_midi_mmio.c - -PASRC = $(PADIR)/pa_common/pa_lib.c $(PADIR)/pa_common/pa_trace.c \ - $(PADIR)/pablio/pablio_pd.c $(PADIR)/pablio/ringbuffer_pd.c - -GSRC = t_main.c t_tkcmd.c - -SRC = g_canvas.c g_graph.c g_text.c g_rtext.c g_array.c g_template.c g_io.c \ - g_scalar.c g_traversal.c g_guiconnect.c g_readwrite.c g_editor.c \ - g_all_guis.c g_bang.c g_hdial.c g_hslider.c g_mycanvas.c g_numbox.c \ - g_toggle.c g_vumeter.c m_pd.c m_class.c m_obj.c m_atom.c m_memory.c \ - m_binbuf.c m_conf.c m_glob.c m_sched.c s_main.c s_inter.c s_file.c \ - s_print.c s_loader.c s_path.c s_entry.c s_audio.c s_midi.c \ - d_ugen.c d_ctl.c d_arithmetic.c d_osc.c d_filter.c d_dac.c d_misc.c \ - d_math.c d_fft.c d_mayer_fft.c d_fftroutine.c d_array.c d_global.c \ - d_delay.c d_resample.c x_arithmetic.c x_connective.c x_interface.c \ - x_midi.c x_misc.c x_time.c x_acoustics.c x_net.c x_qlist.c x_gui.c \ - d_soundfile.c g_vslider.c g_vdial.c $(SYSSRC) - -SRSRC = u_pdsend.c u_pdreceive.c - -OBJ = $(SRC:.c=.o) -GOBJ = $(GSRC:.c=.o) -SROBJ = $(SRSRC:.c=.o) -PAOBJ = $(PASRC:.c=.o) -OBJC = $(OBJ) $(PAOBJ) - -# -# ------------------ targets ------------------------------------ -# - -.PHONY: all - -all: $(OBJ_DIR)/$(GUINAME) $(OBJ_DIR)/$(PDDLL) $(OBJ_DIR)/$(PDEXEC) \ - $(OBJ_DIR)/pdsend.exe $(OBJ_DIR)/pdreceive.exe $(OBJ_DIR)/pd.tk - -$(OBJ) : %.o : %.c - $(CC) $(CFLAGS) $(GFLAGS) $(INCLUDE) -c -o $(OBJ_DIR)/$*.o $*.c - -$(GOBJ) : %.o : %.c - $(CC) $(CFLAGS) $(GFLAGS) $(GINCLUDE) -c -o $(OBJ_DIR)/$*.o $*.c - -$(SROBJ) : %.o : %.c - $(CC) $(CFLAGS) $(GFLAGS) $(INCLUDE) -c -o $(OBJ_DIR)/$*.o $*.c - -$(PAOBJ) : %.o : %.c - $(CC) $(CFLAGS) $(GFLAGS) $(INCPA) -c -o $(OBJ_DIR)/$*.o $*.c - -$(OBJ_DIR)/pdsend.exe: u_pdsend.o - cd $(OBJ_DIR); $(CC) $(CFLAGS) $(STRIPFLAG) -o pdsend.exe u_pdsend.o $(LDFLAGS) - -$(OBJ_DIR)/pdreceive.exe: u_pdreceive.o - cd $(OBJ_DIR); $(CC) $(CFLAGS) $(STRIPFLAG) -o pdreceive.exe u_pdreceive.o $(LDFLAGS) - -$(OBJ_DIR)/$(PDEXEC): s_entry.o - cd $(OBJ_DIR); $(CC) -o $(PDEXEC) s_entry.o $(LIB) pd.a $(LDFLAGS) - -$(OBJ_DIR)/$(PDDLL): $(OBJC) - cd $(OBJ_DIR); $(DLLWRAP) --export-all-symbols --output-def pd.def \ - --output-lib=pd.a --dllname=pd.dll $(OBJC) pdtcl.a $(LDFLAGS) - -$(OBJ_DIR)/$(GUINAME): t_tkcmd.o - cd $(OBJ_DIR); $(DLLWRAP) --export-all-symbols --output-def pdtcl.def \ - --output-lib=pdtcl.a --dllname=$(GUINAME) t_tkcmd.o $(LDFLAGS) - -$(OBJ_DIR)/pd.tk: u_main.tk - echo set pd_nt 1 > $(OBJ_DIR)/pd.tk - grep -v "set pd_nt" < u_main.tk >> $(OBJ_DIR)/pd.tk - -install: all - install -d $(INSTDIR)/bin - install $(OBJ_DIR)/pd*.exe $(BIN_DIR) - install $(OBJ_DIR)/pd*.dll $(BIN_DIR) - install -m644 $(OBJ_DIR)/pd.tk $(BIN_DIR)/pd.tk - install -m644 ../extra/*/*.pd $(INSTDIR)/doc/5.reference/ - -clean: - -rm -f $(OBJ_DIR)/*.o $(OBJ_DIR)/*.a $(OBJ_DIR)/*.def - -rm -f $(OBJ_DIR)/pd*.exe $(OBJ_DIR)/pd*.dll - -rm -f $(PADIR)/pablio/*.o $(PADIR)/pa_common/*.o - -rm -f $(OBJ_DIR)/pd.tk - -rm -f makefile.dependencies - touch makefile.dependencies - chmod 666 makefile.dependencies - -distclean: clean - rm -rf config.cache config.log config.status makefile tags \ - autom4te-*.cache - echo all: > makefile - echo -e "\t./configure" >> makefile - echo -e "\tmake" >> makefile - -tags: $(SRC) $(GSRC); ctags *.[ch] - -depend: - $(CC) $(INCLUDE) $(CFLAGS) -M $(SRC) > makefile.dependencies - -uninstall: - -rm $(INSTDIR)/bin/pd*.exe - -rm $(INSTDIR)/bin/pd*.dll - -rm $(INSTDIR)/bin/*.tk - -include makefile.dependencies diff --git a/pd/src/makefile.nt.pa19 b/pd/src/makefile.nt.pa19 deleted file mode 100644 index 9105d734..00000000 --- a/pd/src/makefile.nt.pa19 +++ /dev/null @@ -1,185 +0,0 @@ -# Makefile for PD on MSW - -all: pd gui ..\bin\pd.tk ..\bin\pdsend.exe ..\bin\pdreceive.exe - -VC = "C:\Program Files\Microsoft Visual Studio\VC98" -#VC="\Program Files\DevStudio\Vc" -INCLUDE = -I.\ -I..\Tcl\include -I$(VC)\include - -LDIR = $(VC)\lib - -LIB = /NODEFAULTLIB:libc /NODEFAULTLIB:oldnames /NODEFAULTLIB:kernel \ - /NODEFAULTLIB:uuid \ - $(LDIR)\libc.lib $(LDIR)\oldnames.lib $(LDIR)\kernel32.lib \ - $(LDIR)\wsock32.lib $(LDIR)\winmm.lib ..\bin\pthreadVC.lib - -GLIB = $(LIB) ..\lib\tcl83.lib ..\lib\tk83.lib -CFLAGS = /nologo /W3 /DMSW /DNT /DPD /DPD_INTERNAL /DWIN32 /DWINDOWS /Ox \ - -DPA_LITTLE_ENDIAN -DUSEAPI_MMIO -DUSEAPI_PORTAUDIO -LFLAGS = /nologo - -SYSSRC = s_audio_pa.c s_audio_mmio.c s_midi_pm.c - -SRC = g_canvas.c g_graph.c g_text.c g_rtext.c g_array.c g_template.c g_io.c \ - g_scalar.c g_traversal.c g_guiconnect.c g_readwrite.c g_editor.c \ - g_all_guis.c g_bang.c g_hdial.c g_hslider.c g_mycanvas.c g_numbox.c \ - g_toggle.c g_vdial.c g_vslider.c g_vumeter.c \ - m_pd.c m_class.c m_obj.c m_atom.c m_memory.c m_binbuf.c \ - m_conf.c m_glob.c m_sched.c \ - s_main.c s_inter.c s_file.c s_print.c \ - s_loader.c s_path.c s_entry.c s_audio.c s_midi.c \ - d_ugen.c d_ctl.c d_arithmetic.c d_osc.c d_filter.c d_dac.c d_misc.c \ - d_math.c d_fft.c d_mayer_fft.c d_fftroutine.c d_array.c d_global.c \ - d_delay.c d_resample.c \ - x_arithmetic.c x_connective.c x_interface.c x_midi.c x_misc.c \ - x_time.c x_acoustics.c x_net.c x_qlist.c x_gui.c d_soundfile.c \ - $(SYSSRC) - -PADIR = ..\portaudio -INCPA = -I$(PADIR) -I$(PADIR)\pa_common -I$(PADIR)\pablio -I..\lib\asio -SRCPA = \ - $(PADIR)/pa_common/pa_allocation.c \ - $(PADIR)/pa_common/pa_converters.c \ - $(PADIR)/pa_common/pa_cpuload.c \ - $(PADIR)/pa_common/pa_dither.c \ - $(PADIR)/pa_common/pa_front.c \ - $(PADIR)/pa_common/pa_process.c \ - $(PADIR)/pa_common/pa_skeleton.c \ - $(PADIR)/pa_common/pa_stream.c \ - $(PADIR)/pa_common/pa_trace.c \ - $(PADIR)/pablio/pablio_pd.c \ - $(PADIR)/pablio/ringbuffer_pd.c \ - $(PADIR)/pa_win/pa_win_hostapis.c \ - $(PADIR)/pa_win/pa_win_util.c \ - $(PADIR)/pa_win/pa_x86_plain_converters.c \ - $(PADIR)/pa_win_wmme/pa_win_wmme.c - -# $(PADIR)/pa_win_ds/dsound_wrapper.c \ -# $(PADIR)/pa_win_ds/pa_dsound.c \ -# $(PADIR)/pa_win_ds/pa_win_ds.c \ - -PAOBJ = \ - pa_allocation.obj pa_converters.obj pa_cpuload.obj pa_dither.obj pa_front.obj \ - pa_process.obj pa_skeleton.obj pa_stream.obj pa_trace.obj pablio_pd.obj \ - ringbuffer_pd.obj pa_win_hostapis.obj pa_win_util.obj \ - pa_x86_plain_converters.obj \ - pa_win_wmme.obj pa_asio.obj \ - -# dsound_wrapper.obj pa_dsound.obj pa_win_ds.obj - - -SRCASIO = $(PADIR)/pa_asio/pa_asio.cpp -ASIOLIB = $(LDIR)\user32.lib $(LDIR)\gdi32.lib $(LDIR)\winspool.lib \ - $(LDIR)\comdlg32.lib \ -$(LDIR)\advapi32.lib $(LDIR)\shell32.lib $(LDIR)\ole32.lib $(LDIR)\oleaut32.lib $(LDIR)\uuid.lib \ -$(LDIR)\odbc32.lib $(LDIR)\odbccp32.lib ..\lib\asio\asiolib.lib - -PMDIR = ..\portmidi -INCPM = -I$(PMDIR)\pm_common -I$(PMDIR)\pm_win -I$(PMDIR)\porttime -SRCPM = \ - $(PMDIR)/pm_common/portmidi.c \ - $(PMDIR)/pm_common/pmutil.c \ - $(PMDIR)/pm_win/pmwin.c \ - $(PMDIR)/pm_win/pmwinmm.c \ - $(PMDIR)/porttime/porttime.c \ - $(PMDIR)/porttime/ptwinmm.c \ - -PMOBJ = portmidi.obj pmutil.obj pmwin.obj pmwinmm.obj porttime.obj ptwinmm.obj - -OBJC = $(SRC:.c=.obj) $(PAOBJ) $(PMOBJ) - -GSRC = t_main.c t_tkcmd.c - -GOBJ = $(GSRC:.c=.obj) -.PHONY: pd gui - -ALLCF = $(CFLAGS) $(INCLUDE) $(INCASIO) $(INCPA) $(INCPM) /D_WINDOWS /DPA_NO_DS - -.c.obj: - cl /c $(ALLCF) /Tc$*.c - -pd: ..\bin\pd.exe - -gui: ..\bin\pdtcl.dll - -..\bin\pd.exe: s_entry.obj ..\bin\pd.lib - link $(LFLAGS) /out:..\bin\pd.exe /INCREMENTAL:NO s_entry.obj \ - ..\bin\pd.lib $(LIB) $(ASIOLIB) - -..\bin\pd.dll ..\bin\pd.lib: $(OBJC) $(OBJASIO) - link $(LFLAGS) /dll /export:sys_main /out:..\bin\pd.dll $(OBJC) \ - $(OBJASIO) $(LIB) $(ASIOLIB) - -..\bin\pdtcl.dll: t_tkcmd.obj - link $(LFLAGS) /dll /export:Pdtcl_Init /out:..\bin\pdtcl.dll \ - t_tkcmd.obj $(GLIB) - -..\bin\pd.tk: u_main.tk; copy u_main.tk ..\bin\pd.tk - -..\bin\pdsend.exe: u_pdsend.obj - link $(LFLAGS) /out:..\bin\pdsend.exe /INCREMENTAL:NO u_pdsend.obj \ - $(LIB) - -..\bin\pdreceive.exe: u_pdreceive.obj - link $(LFLAGS) /out:..\bin\pdreceive.exe /INCREMENTAL:NO u_pdreceive.obj \ - $(LIB) - -# explicit rules to compile portaudio and portmidi sources: -pa_allocation.obj: $(PADIR)\pa_common\pa_allocation.c - cl /c $(ALLCF) $(PADIR)\pa_common\pa_allocation.c -pa_converters.obj: $(PADIR)\pa_common\pa_converters.c - cl /c $(ALLCF) $(PADIR)\pa_common\pa_converters.c -pa_cpuload.obj: $(PADIR)\pa_common\pa_cpuload.c - cl /c $(ALLCF) $(PADIR)\pa_common\pa_cpuload.c -pa_dither.obj: $(PADIR)\pa_common\pa_dither.c - cl /c $(ALLCF) $(PADIR)\pa_common\pa_dither.c -pa_front.obj: $(PADIR)\pa_common\pa_front.c - cl /c $(ALLCF) $(PADIR)\pa_common\pa_front.c -pa_process.obj: $(PADIR)\pa_common\pa_process.c - cl /c $(ALLCF) $(PADIR)\pa_common\pa_process.c -pa_skeleton.obj: $(PADIR)\pa_common\pa_skeleton.c - cl /c $(ALLCF) $(PADIR)\pa_common\pa_skeleton.c -pa_stream.obj: $(PADIR)\pa_common\pa_stream.c - cl /c $(ALLCF) $(PADIR)\pa_common\pa_stream.c -pa_trace.obj: $(PADIR)\pa_common\pa_trace.c - cl /c $(ALLCF) $(PADIR)\pa_common\pa_trace.c -pablio_pd.obj: $(PADIR)\pablio\pablio_pd.c - cl /c $(ALLCF) $(PADIR)\pablio\pablio_pd.c -ringbuffer_pd.obj: $(PADIR)\pablio\ringbuffer_pd.c - cl /c $(ALLCF) $(PADIR)\pablio\ringbuffer_pd.c -pa_win_hostapis.obj: $(PADIR)\pa_win\pa_win_hostapis.c - cl /c $(ALLCF) $(PADIR)\pa_win\pa_win_hostapis.c -pa_win_util.obj: $(PADIR)\pa_win\pa_win_util.c - cl /c $(ALLCF) $(PADIR)\pa_win\pa_win_util.c -pa_x86_plain_converters.obj: $(PADIR)\pa_win\pa_x86_plain_converters.c - cl /c $(ALLCF) $(PADIR)\pa_win\pa_x86_plain_converters.c -dsound_wrapper.obj: $(PADIR)\pa_win_ds\dsound_wrapper.c - cl /c $(ALLCF) $(PADIR)\pa_win_ds\dsound_wrapper.c -pa_dsound.obj: $(PADIR)\pa_win_ds\pa_dsound.c - cl /c $(ALLCF) $(PADIR)\pa_win_ds\pa_dsound.c -pa_win_ds.obj: $(PADIR)\pa_win_ds\pa_win_ds.c - cl /c $(ALLCF) $(PADIR)\pa_win_ds\pa_win_ds.c -pa_win_wmme.obj: $(PADIR)\pa_win_wmme\pa_win_wmme.c - cl /c $(ALLCF) $(PADIR)\pa_win_wmme\pa_win_wmme.c - -pa_asio.obj: $(PADIR)\pa_asio\pa_asio.cpp - cl /c $(ALLCF) $(PADIR)\pa_asio\pa_asio.cpp - -portmidi.obj: $(PMDIR)\pm_common\portmidi.c - cl /c $(ALLCF) $(PMDIR)\pm_common\portmidi.c -pmutil.obj: $(PMDIR)\pm_common\pmutil.c - cl /c $(ALLCF) $(PMDIR)\pm_common\pmutil.c -pmwin.obj: $(PMDIR)\pm_win\pmwin.c - cl /c $(ALLCF) $(PMDIR)\pm_win\pmwin.c -pmwinmm.obj: $(PMDIR)\pm_win\pmwinmm.c - cl /c $(ALLCF) $(PMDIR)\pm_win\pmwinmm.c -porttime.obj: $(PMDIR)\porttime\porttime.c - cl /c $(ALLCF) $(PMDIR)\porttime\porttime.c -ptwinmm.obj: $(PMDIR)\porttime\ptwinmm.c - cl /c $(ALLCF) $(PMDIR)\porttime\ptwinmm.c - -# the following should also clean up "bin" but it doesn't because "bin" holds -# precious stuff from elsewhere. -clean: - del *.obj - diff --git a/pd/src/s_audio_alsa.c.old b/pd/src/s_audio_alsa.c.old deleted file mode 100644 index 99ca3403..00000000 --- a/pd/src/s_audio_alsa.c.old +++ /dev/null @@ -1,946 +0,0 @@ -/* Copyright (c) 1997-2003 Guenter Geiger, Miller Puckette, Larry Troxler, -* Winfried Ritsch, Karl MacMillan, and others. -* For information on usage and redistribution, and for a DISCLAIMER OF ALL -* WARRANTIES, see the file, "LICENSE.txt," in this distribution. */ - -/* this file inputs and outputs audio using the ALSA API available on linux. */ - -#include - -#include "m_pd.h" -#include "s_stuff.h" -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include - -typedef int16_t t_alsa_sample16; -typedef int32_t t_alsa_sample32; -#define ALSA_SAMPLEWIDTH_16 sizeof(t_alsa_sample16) -#define ALSA_SAMPLEWIDTH_32 sizeof(t_alsa_sample32) -#define ALSA_XFERSIZE16 (signed int)(sizeof(t_alsa_sample16) * DEFDACBLKSIZE) -#define ALSA_XFERSIZE32 (signed int)(sizeof(t_alsa_sample32) * DEFDACBLKSIZE) -#define ALSA_MAXDEV 1 -#define ALSA_JITTER 1024 -#define ALSA_EXTRABUFFER 2048 -#define ALSA_DEFFRAGSIZE 64 -#define ALSA_DEFNFRAG 12 - -#ifndef INT32_MAX -#define INT32_MAX 0x7fffffff -#endif - -#if (SND_LIB_MAJOR < 1) -#define ALSAAPI9 -#endif - -typedef struct _alsa_dev -{ - snd_pcm_t *inhandle; - snd_pcm_t *outhandle; - int innoninterleave; /* true if we're set for noninterleaved read */ - int outnoninterleave; /* same for write */ -} t_alsa_dev; - -t_alsa_dev alsa_device; -static void *alsa_snd_buf = 0; -static void **alsa_buf_ptrs; -static int alsa_samplewidth; -static snd_pcm_status_t* in_status; -static snd_pcm_status_t* out_status; - -static int alsa_mode; -static int alsa_buf_samps; /* believed actual ALSA bufsize in sample frames */ -static int alsa_inchannels; -static int alsa_outchannels; - -/* Defines */ -#define DEBUG(x) x -#define DEBUG2(x) {x;} - -static void alsa_checkiosync( void); -static void alsa_numbertoname(int devno, char *devname, int nchar); - - /* don't assume we can turn all 31 bits when doing float-to-fix; - otherwise some audio drivers (e.g. Midiman/ALSA) wrap around. */ -#define FMAX 0x7ffff000 -#define CLIP32(x) (((x)>FMAX)?FMAX:((x) < -FMAX)?-FMAX:(x)) - -/* support for ALSA pcmv2 api by Karl MacMillan */ - -static void check_error(int err, const char *why) -{ - if (err < 0) - fprintf(stderr, "%s: %s\n", why, snd_strerror(err)); -} - -/* was: alsa_open_audio(int wantinchans, int wantoutchans, int srate) */ - -int alsa_open_audio(int naudioindev, int *audioindev, int nchindev, - int *chindev, int naudiooutdev, int *audiooutdev, int nchoutdev, - int *choutdev, int rate) -{ - int err, inchans = 0, outchans = 0, subunitdir; - char devname[512]; - snd_pcm_hw_params_t* hw_params; - snd_pcm_sw_params_t* sw_params; - snd_output_t* out; - int frag_size = (sys_blocksize ? sys_blocksize : ALSA_DEFFRAGSIZE); - int nfrags, i; - short* tmp_buf; - unsigned int tmp_uint; - snd_pcm_uframes_t tmp_snd_pcm_uframes; - int wantinchans, wantoutchans, devno; - - if (naudioindev >= 2 || naudiooutdev >= 2) - post("alsa: only one input and output device allowed (extras ignored"); - if (naudioindev >= 1 && naudiooutdev >= 1 && - audioindev[0] != audiooutdev[0]) - post("alsa: changing output device to agree with input device"); - if (nchindev) - wantinchans = chindev[0]; - else wantinchans = (naudioindev ? 2 : 0); - if (nchoutdev) - wantoutchans = choutdev[0]; - else wantoutchans = (naudiooutdev ? 2 : 0); - devno = (naudioindev > 0 ? audioindev[0] : - (naudiooutdev > 0 ? audiooutdev[0] : 0)); - - alsa_numbertoname(devno, devname, 512); - - if (sys_verbose) - post("device name %s; channels in %d, out %d", devname, wantinchans, - wantoutchans); - - nfrags = sys_schedadvance * (float)rate / (1e6 * frag_size); - /* save our belief as to ALSA's buffer size for later */ - alsa_buf_samps = nfrags * frag_size; - - if (sys_verbose) - post("audio buffer set to %d", (int)(0.001 * sys_schedadvance)); - - alsa_device.innoninterleave = alsa_device.outnoninterleave = 0; - if (wantinchans) - { - err = snd_pcm_open(&alsa_device.inhandle, devname, - SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK); - - check_error(err, "snd_pcm_open (input)"); - if (err < 0) - inchans = 0; - else - { - inchans = wantinchans; - snd_pcm_nonblock(alsa_device.inhandle, 1); - } - } - if (wantoutchans) - { - err = snd_pcm_open(&alsa_device.outhandle, devname, - SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); - - check_error(err, "snd_pcm_open (output)"); - if (err < 0) - outchans = 0; - else - { - outchans = wantoutchans; - snd_pcm_nonblock(alsa_device.outhandle, 1); - } - } - if (inchans) - { - if (sys_verbose) - post("opening sound input..."); - err = snd_pcm_hw_params_malloc(&hw_params); - check_error(err, "snd_pcm_hw_params_malloc (input)"); - - // get the default params - err = snd_pcm_hw_params_any(alsa_device.inhandle, hw_params); - check_error(err, "snd_pcm_hw_params_any (input)"); - - /* try to set interleaved access */ - err = snd_pcm_hw_params_set_access(alsa_device.inhandle, - hw_params, SND_PCM_ACCESS_RW_INTERLEAVED); - if (err < 0) - { - /* OK, so try non-interleaved */ - err = snd_pcm_hw_params_set_access(alsa_device.inhandle, - hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED); - if (err >= 0) - { - post("using non-interleaved audio input"); - alsa_device.innoninterleave = 1; - } - } - check_error(err, "snd_pcm_hw_params_set_access (input)"); - // Try to set 32 bit format first - err = snd_pcm_hw_params_set_format(alsa_device.inhandle, hw_params, - SND_PCM_FORMAT_S32); - if (err < 0) - { - /* fprintf(stderr, - "PD-ALSA: 32 bit format not available - using 16\n"); */ - err = snd_pcm_hw_params_set_format(alsa_device.inhandle, hw_params, - SND_PCM_FORMAT_S16); - check_error(err, "snd_pcm_hw_params_set_format (input)"); - alsa_samplewidth = 2; - } - else - { - alsa_samplewidth = 4; - } - post("Sample width set to %d bytes", alsa_samplewidth); - // set the subformat - err = snd_pcm_hw_params_set_subformat(alsa_device.inhandle, hw_params, - SND_PCM_SUBFORMAT_STD); - check_error(err, "snd_pcm_hw_params_set_subformat (input)"); - // set the number of channels - tmp_uint = inchans; - err = snd_pcm_hw_params_set_channels_min(alsa_device.inhandle, - hw_params, &tmp_uint); - check_error(err, "snd_pcm_hw_params_set_channels (input)"); - if (tmp_uint != (unsigned)inchans) - post("ALSA: set input channels to %d", tmp_uint); - inchans = tmp_uint; - // set the sampling rate - err = snd_pcm_hw_params_set_rate_min(alsa_device.inhandle, hw_params, - &rate, 0); - check_error(err, "snd_pcm_hw_params_set_rate_min (input)"); -#if 0 - err = snd_pcm_hw_params_get_rate(hw_params, &subunitdir); - post("input sample rate %d", err); -#endif - // set the period - ie frag size - // post("fragsize a %d", frag_size); - - /* LATER try this to get a recommended period size... - right now, it trips an assertion failure in ALSA lib */ -#if 0 - post("input period was %d, min %d, max %d\n", - snd_pcm_hw_params_get_period_size(hw_params, 0), - snd_pcm_hw_params_get_period_size_min(hw_params, 0), - snd_pcm_hw_params_get_period_size_max(hw_params, 0)); -#endif -#ifdef ALSAAPI9 - err = snd_pcm_hw_params_set_period_size_near(alsa_device.inhandle, - hw_params, - (snd_pcm_uframes_t) - frag_size, 0); -#else - tmp_snd_pcm_uframes = frag_size; - err = snd_pcm_hw_params_set_period_size_near(alsa_device.inhandle, - hw_params, &tmp_snd_pcm_uframes, 0); -#endif - check_error(err, "snd_pcm_hw_params_set_period_size_near (input)"); - // post("fragsize b %d", frag_size); - // set the number of periods - ie numfrags - // post("nfrags a %d", nfrags); -#ifdef ALSAAPI9 - err = snd_pcm_hw_params_set_periods_near(alsa_device.inhandle, - hw_params, nfrags, 0); -#else - tmp_uint = nfrags; - err = snd_pcm_hw_params_set_periods_near(alsa_device.inhandle, - hw_params, &tmp_uint, 0); -#endif - check_error(err, "snd_pcm_hw_params_set_periods_near (input)"); - // set the buffer size -#ifdef ALSAAPI9 - err = snd_pcm_hw_params_set_buffer_size_near(alsa_device.inhandle, - hw_params, nfrags * frag_size); -#else - tmp_snd_pcm_uframes = nfrags * frag_size; - err = snd_pcm_hw_params_set_buffer_size_near(alsa_device.inhandle, - hw_params, &tmp_snd_pcm_uframes); -#endif - check_error(err, "snd_pcm_hw_params_set_buffer_size_near (input)"); - - err = snd_pcm_hw_params(alsa_device.inhandle, hw_params); - check_error(err, "snd_pcm_hw_params (input)"); - - snd_pcm_hw_params_free(hw_params); - - err = snd_pcm_sw_params_malloc(&sw_params); - check_error(err, "snd_pcm_sw_params_malloc (input)"); - err = snd_pcm_sw_params_current(alsa_device.inhandle, sw_params); - check_error(err, "snd_pcm_sw_params_current (input)"); - err = snd_pcm_sw_params_set_start_threshold(alsa_device.inhandle, - sw_params, nfrags * frag_size); - check_error(err, "snd_pcm_sw_params_set_start_threshold (input)"); - err = snd_pcm_sw_params_set_stop_threshold(alsa_device.inhandle, - sw_params, 0x7fffffff); - check_error(err, "snd_pcm_sw_params_set_stop_threshold (input)"); - err = snd_pcm_sw_params_set_avail_min(alsa_device.inhandle, sw_params, - frag_size); - check_error(err, "snd_pcm_sw_params_set_avail_min (input)"); - err = snd_pcm_sw_params(alsa_device.inhandle, sw_params); - check_error(err, "snd_pcm_sw_params (input)"); - - snd_pcm_sw_params_free(sw_params); - - snd_output_stdio_attach(&out, stderr, 0); -#if 0 - if (sys_verbose) - { - snd_pcm_dump_hw_setup(alsa_device.inhandle, out); - snd_pcm_dump_sw_setup(alsa_device.inhandle, out); - } -#endif - } - - if (outchans) - { - int foo; - if (sys_verbose) - post("opening sound output..."); - err = snd_pcm_hw_params_malloc(&hw_params); - check_error(err, "snd_pcm_sw_params (output)"); - - // get the default params - err = snd_pcm_hw_params_any(alsa_device.outhandle, hw_params); - check_error(err, "snd_pcm_hw_params_any (output)"); - // set interleaved access - FIXME deal with other access types - err = snd_pcm_hw_params_set_access(alsa_device.outhandle, hw_params, - SND_PCM_ACCESS_RW_INTERLEAVED); - check_error(err, "snd_pcm_hw_params_set_access (output)"); - - /* try to set interleaved access */ - err = snd_pcm_hw_params_set_access(alsa_device.outhandle, - hw_params, SND_PCM_ACCESS_RW_INTERLEAVED); - if (err < 0) - { - /* OK, so try non-interleaved */ - err = snd_pcm_hw_params_set_access(alsa_device.outhandle, - hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED); - if (err >= 0) - { - post("using non-interleaved audio"); - alsa_device.outnoninterleave = 1; - } - } - check_error(err, "snd_pcm_hw_params_set_access (output)"); - - - // Try to set 32 bit format first - err = snd_pcm_hw_params_set_format(alsa_device.outhandle, hw_params, - SND_PCM_FORMAT_S32); - if (err < 0) - { - err = snd_pcm_hw_params_set_format(alsa_device.outhandle, - hw_params,SND_PCM_FORMAT_S16); - check_error(err, "snd_pcm_hw_params_set_format (output)"); - /* fprintf(stderr, - "PD-ALSA: 32 bit format not available - using 16\n"); */ - alsa_samplewidth = 2; - } - else - { - alsa_samplewidth = 4; - } - // set the subformat - err = snd_pcm_hw_params_set_subformat(alsa_device.outhandle, hw_params, - SND_PCM_SUBFORMAT_STD); - check_error(err, "snd_pcm_hw_params_set_subformat (output)"); - // set the number of channels - tmp_uint = outchans; - err = snd_pcm_hw_params_set_channels_min(alsa_device.outhandle, - hw_params, &tmp_uint); - check_error(err, "snd_pcm_hw_params_set_channels (output)"); - if (tmp_uint != (unsigned)outchans) - post("alsa: set output channels to %d", tmp_uint); - outchans = tmp_uint; - // set the sampling rate - err = snd_pcm_hw_params_set_rate_min(alsa_device.outhandle, hw_params, - &rate, 0); - check_error(err, "snd_pcm_hw_params_set_rate_min (output)"); -#if 0 - err = snd_pcm_hw_params_get_rate(hw_params, &subunitdir); - post("output sample rate %d", err); -#endif - // set the period - ie frag size -#if 0 - post("output period was %d, min %d, max %d\n", - snd_pcm_hw_params_get_period_size(hw_params, 0), - snd_pcm_hw_params_get_period_size_min(hw_params, 0), - snd_pcm_hw_params_get_period_size_max(hw_params, 0)); -#endif - // post("fragsize c %d", frag_size); -#ifdef ALSAAPI9 - err = snd_pcm_hw_params_set_period_size_near(alsa_device.outhandle, - hw_params, - (snd_pcm_uframes_t) - frag_size, 0); -#else - tmp_snd_pcm_uframes = frag_size; - err = snd_pcm_hw_params_set_period_size_near(alsa_device.outhandle, - hw_params, &tmp_snd_pcm_uframes, 0); -#endif - // post("fragsize d %d", frag_size); - check_error(err, "snd_pcm_hw_params_set_period_size_near (output)"); - // set the number of periods - ie numfrags -#ifdef ALSAAPI9 - err = snd_pcm_hw_params_set_periods_near(alsa_device.outhandle, - hw_params, nfrags, 0); -#else - tmp_uint = nfrags; - err = snd_pcm_hw_params_set_periods_near(alsa_device.outhandle, - hw_params, &tmp_uint, 0); -#endif - check_error(err, "snd_pcm_hw_params_set_periods_near (output)"); - // set the buffer size -#ifdef ALSAAPI9 - err = snd_pcm_hw_params_set_buffer_size_near(alsa_device.outhandle, - hw_params, nfrags * frag_size); -#else - tmp_snd_pcm_uframes = nfrags * frag_size; - err = snd_pcm_hw_params_set_buffer_size_near(alsa_device.outhandle, - hw_params, &tmp_snd_pcm_uframes); -#endif - check_error(err, "snd_pcm_hw_params_set_buffer_size_near (output)"); - - err = snd_pcm_hw_params(alsa_device.outhandle, hw_params); - check_error(err, "snd_pcm_hw_params (output)"); - - snd_pcm_hw_params_free(hw_params); - - err = snd_pcm_sw_params_malloc(&sw_params); - check_error(err, "snd_pcm_sw_params_malloc (output)"); - err = snd_pcm_sw_params_current(alsa_device.outhandle, sw_params); - check_error(err, "snd_pcm_sw_params_current (output)"); - err = snd_pcm_sw_params_set_start_threshold(alsa_device.outhandle, - sw_params, nfrags * frag_size); - check_error(err, "snd_pcm_sw_params_set_start_threshold (output)"); - err = snd_pcm_sw_params_set_stop_threshold(alsa_device.outhandle, - sw_params, 0x7fffffff); - check_error(err, "snd_pcm_sw_params_set_stop_threshold (output)"); - err = snd_pcm_sw_params_set_avail_min(alsa_device.outhandle, sw_params, - frag_size); - check_error(err, "snd_pcm_sw_params_set_avail_min (output)"); - err = snd_pcm_sw_params(alsa_device.outhandle, sw_params); - check_error(err, "snd_pcm_sw_params (output)"); - snd_pcm_sw_params_free(sw_params); - - snd_output_stdio_attach(&out, stderr, 0); -#if 0 - if (sys_verbose) - { - snd_pcm_dump_hw_setup(alsa_device.outhandle, out); - snd_pcm_dump_sw_setup(alsa_device.outhandle, out); - } -#endif - } - - if (inchans) - snd_pcm_prepare(alsa_device.inhandle); - if (outchans) - snd_pcm_prepare(alsa_device.outhandle); - - // if duplex we can link the channels so they start together - if (inchans && outchans) - snd_pcm_link(alsa_device.inhandle, alsa_device.outhandle); - - // set up the status variables - err = snd_pcm_status_malloc(&in_status); - check_error(err, "snd_pcm_status_malloc"); - err = snd_pcm_status_malloc(&out_status); - check_error(err, "snd_pcm_status_malloc"); - - // set up the buffer - if (alsa_snd_buf) - free(alsa_snd_buf); - alsa_snd_buf = (void *)malloc( - sizeof(char) * alsa_samplewidth * DEFDACBLKSIZE * - (outchans > inchans ? outchans : inchans)); - memset(alsa_snd_buf, 0, sizeof(char) * alsa_samplewidth * DEFDACBLKSIZE * - (outchans > inchans ? outchans : inchans)); - /* make an array of pointers too in case we need them */ - if (alsa_buf_ptrs) - free(alsa_buf_ptrs); - alsa_buf_ptrs = (void **)malloc( - sizeof(void *) * (outchans > inchans ? outchans : inchans)); - for (i = 0; i < (outchans > inchans ? outchans : inchans); i++) - alsa_buf_ptrs[i] = (t_alsa_sample32 *)alsa_snd_buf + i * DEFDACBLKSIZE; - - // fill the buffer with silence - if (outchans) - { - i = (frag_size * nfrags)/DEFDACBLKSIZE + 1; - while (i--) - { - if (alsa_device.outnoninterleave) - snd_pcm_writen(alsa_device.outhandle, alsa_buf_ptrs, - DEFDACBLKSIZE); - else snd_pcm_writei(alsa_device.outhandle, alsa_snd_buf, - DEFDACBLKSIZE); - } - /* confused about this: */ - /* if ((err = snd_pcm_start(alsa_device.outhandle) < 0)) - check_error(err, "output start failed\n"); */ - } - else if (inchans) - { - if (snd_pcm_start(alsa_device.inhandle) < 0) - check_error(err, "input start failed\n"); - } - alsa_outchannels = outchans; - alsa_inchannels = inchans; - - return (!(inchans || outchans)); -} - -void alsa_close_audio(void) -{ - int err; - if (alsa_inchannels) - { - err = snd_pcm_close(alsa_device.inhandle); - check_error(err, "snd_pcm_close (input)"); - } - if (alsa_outchannels) - { - err = snd_pcm_close(alsa_device.outhandle); - check_error(err, "snd_pcm_close (output)"); - } -} - -// #define DEBUG_ALSA_XFER - -int alsa_send_dacs(void) -{ - static int16_t *sp; - static int xferno = 0; - static int callno = 0; - static double timenow; - double timelast; - t_sample *fp, *fp1, *fp2; - int i, j, k, err, devno = 0; - int inputcount = 0, outputcount = 0, inputlate = 0, outputlate = 0; - int result; - int inchannels = (sys_inchannels > alsa_inchannels ? - alsa_inchannels : sys_inchannels); - int outchannels = (sys_outchannels > alsa_outchannels ? - alsa_outchannels : sys_outchannels); - unsigned int intransfersize = DEFDACBLKSIZE; - unsigned int outtransfersize = DEFDACBLKSIZE; - - // get the status - if (!inchannels && !outchannels) - { - return SENDDACS_NO; - } - - timelast = timenow; - timenow = sys_getrealtime(); - -#ifdef DEBUG_ALSA_XFER - if (timenow - timelast > 0.050) - fprintf(stderr, "(%d)", - (int)(1000 * (timenow - timelast))), fflush(stderr); -#endif - - callno++; - - alsa_checkiosync(); /* check I/O are in sync and data not late */ - - if (alsa_inchannels) - { - snd_pcm_status(alsa_device.inhandle, in_status); - if (snd_pcm_status_get_avail(in_status) < intransfersize) - return SENDDACS_NO; - } - if (alsa_outchannels) - { - snd_pcm_status(alsa_device.outhandle, out_status); - if (snd_pcm_status_get_avail(out_status) < outtransfersize) - return SENDDACS_NO; - } - - /* do output */ - if (alsa_outchannels) - { - fp = sys_soundout; - if (alsa_samplewidth == 4) - { - if (alsa_device.outnoninterleave) - { - int n = outchannels * DEFDACBLKSIZE; - for (i = 0, fp1 = fp; i < n; i++) - { - float s1 = *fp1 * INT32_MAX; - ((t_alsa_sample32 *)alsa_snd_buf)[i] = CLIP32(s1); - } - n = alsa_outchannels * DEFDACBLKSIZE; - for (; i < n; i++) - ((t_alsa_sample32 *)alsa_snd_buf)[i] = 0; - } - else - { - for (i = 0, fp1 = fp; i < outchannels; i++, - fp1 += DEFDACBLKSIZE) - { - for (j = i, k = DEFDACBLKSIZE, fp2 = fp1; k--; - j += alsa_outchannels, fp2++) - { - float s1 = *fp2 * INT32_MAX; - ((t_alsa_sample32 *)alsa_snd_buf)[j] = CLIP32(s1); - } - } - } - } - else - { - for (i = 0, fp1 = fp; i < outchannels; i++, fp1 += DEFDACBLKSIZE) - { - for (j = i, k = DEFDACBLKSIZE, fp2 = fp1; k--; - j += alsa_outchannels, fp2++) - { - int s = *fp2 * 32767.; - if (s > 32767) - s = 32767; - else if (s < -32767) - s = -32767; - ((t_alsa_sample16 *)alsa_snd_buf)[j] = s; - } - } - } - - if (alsa_device.outnoninterleave) - result = snd_pcm_writen(alsa_device.outhandle, alsa_buf_ptrs, - outtransfersize); - else result = snd_pcm_writei(alsa_device.outhandle, alsa_snd_buf, - outtransfersize); - - if (result != (int)outtransfersize) - { - #ifdef DEBUG_ALSA_XFER - if (result >= 0 || errno == EAGAIN) - fprintf(stderr, "ALSA: write returned %d of %d\n", - result, outtransfersize); - else fprintf(stderr, "ALSA: write: %s\n", - snd_strerror(errno)); - fprintf(stderr, - "inputcount %d, outputcount %d, outbufsize %d\n", - inputcount, outputcount, - (ALSA_EXTRABUFFER + sys_advance_samples) - * alsa_samplewidth * outchannels); - #endif - sys_log_error(ERR_DACSLEPT); - return (SENDDACS_NO); - } - - /* zero out the output buffer */ - memset(sys_soundout, 0, DEFDACBLKSIZE * sizeof(*sys_soundout) * - sys_outchannels); - if (sys_getrealtime() - timenow > 0.002) - { - #ifdef DEBUG_ALSA_XFER - fprintf(stderr, "output %d took %d msec\n", - callno, (int)(1000 * (timenow - timelast))), fflush(stderr); - #endif - timenow = sys_getrealtime(); - sys_log_error(ERR_DACSLEPT); - } - } - /* do input */ - if (alsa_inchannels) - { - if (alsa_device.innoninterleave) - result = snd_pcm_readn(alsa_device.inhandle, alsa_buf_ptrs, - intransfersize); - else result = snd_pcm_readi(alsa_device.inhandle, alsa_snd_buf, - intransfersize); - if (result < (int)intransfersize) - { -#ifdef DEBUG_ALSA_XFER - if (result < 0) - fprintf(stderr, - "snd_pcm_read %d %d: %s\n", - callno, xferno, snd_strerror(errno)); - else fprintf(stderr, - "snd_pcm_read %d %d returned only %d\n", - callno, xferno, result); - fprintf(stderr, - "inputcount %d, outputcount %d, inbufsize %d\n", - inputcount, outputcount, - (ALSA_EXTRABUFFER + sys_advance_samples) - * alsa_samplewidth * inchannels); -#endif - sys_log_error(ERR_ADCSLEPT); - return (SENDDACS_NO); - } - fp = sys_soundin; - if (alsa_samplewidth == 4) - { - if (alsa_device.innoninterleave) - { - int n = inchannels * DEFDACBLKSIZE; - for (i = 0, fp1 = fp; i < n; i++) - *fp1 = (float) ((t_alsa_sample32 *)alsa_snd_buf)[i] - * (1./ INT32_MAX); - } - else - { - for (i = 0, fp1 = fp; i < inchannels; - i++, fp1 += DEFDACBLKSIZE) - { - for (j = i, k = DEFDACBLKSIZE, fp2 = fp1; k--; - j += alsa_inchannels, fp2++) - *fp2 = (float) ((t_alsa_sample32 *)alsa_snd_buf)[j] - * (1./ INT32_MAX); - } - } - } - else - { - for (i = 0, fp1 = fp; i < inchannels; i++, fp1 += DEFDACBLKSIZE) - { - for (j = i, k = DEFDACBLKSIZE, fp2 = fp1; k--; - j += alsa_inchannels, fp2++) - *fp2 = (float) ((t_alsa_sample16 *)alsa_snd_buf)[j] - * 3.051850e-05; - } - } - } - xferno++; - if (sys_getrealtime() - timenow > 0.002) - { -#ifdef DEBUG_ALSA_XFER - fprintf(stderr, "routine took %d msec\n", - (int)(1000 * (sys_getrealtime() - timenow))); -#endif - sys_log_error(ERR_ADCSLEPT); - } - return SENDDACS_YES; -} - -void alsa_printstate( void) -{ - int i, result; - snd_pcm_sframes_t indelay, outdelay; - if (sys_audioapi != API_ALSA) - { - error("restart-audio: implemented for ALSA only."); - return; - } - if (sys_inchannels) - { - result = snd_pcm_delay(alsa_device.inhandle, &indelay); - if (result < 0) - post("snd_pcm_delay 1 failed"); - else post("in delay %d", indelay); - } - if (sys_outchannels) - { - result = snd_pcm_delay(alsa_device.outhandle, &outdelay); - if (result < 0) - post("snd_pcm_delay 2 failed"); - else post("out delay %d", outdelay); - } - post("sum %d (%d mod 64)\n", indelay + outdelay, (indelay+outdelay)%64); - - post("buf samples %d", alsa_buf_samps); -} - - -void alsa_resync( void) -{ - int i, result; - if (sys_audioapi != API_ALSA) - { - error("restart-audio: implemented for ALSA only."); - return; - } - memset(alsa_snd_buf, 0, - sizeof(char) * alsa_samplewidth * DEFDACBLKSIZE * sys_outchannels); - for (i = 0; i < 1000000; i++) - { - if (alsa_device.outnoninterleave) - result = snd_pcm_writen(alsa_device.outhandle, alsa_buf_ptrs, - DEFDACBLKSIZE); - else result = snd_pcm_writei(alsa_device.outhandle, alsa_snd_buf, - DEFDACBLKSIZE); - if (result != (int)DEFDACBLKSIZE) - break; - } - post("%d written", i); -} - -void alsa_putzeros(int n) -{ - int i, result; - memset(alsa_snd_buf, 0, - sizeof(char) * alsa_samplewidth * DEFDACBLKSIZE * alsa_outchannels); - for (i = 0; i < n; i++) - { - if (alsa_device.outnoninterleave) - result = snd_pcm_writen(alsa_device.outhandle, alsa_buf_ptrs, - DEFDACBLKSIZE); - else result = snd_pcm_writei(alsa_device.outhandle, alsa_snd_buf, - DEFDACBLKSIZE); -#if 0 - if (result != DEFDACBLKSIZE) - post("result %d", result); -#endif - } -} - -void alsa_getzeros(int n) -{ - int i, result; - for (i = 0; i < n; i++) - { - result = snd_pcm_readi(alsa_device.inhandle, alsa_snd_buf, - DEFDACBLKSIZE); -#if 0 - if (result != DEFDACBLKSIZE) - post("result %d", result); -#endif - } -} - - /* call this only if both input and output are open */ -static void alsa_checkiosync( void) -{ - int i, result, checkit = 1, giveup = 1000, alreadylogged = 0; - snd_pcm_sframes_t indelay, outdelay, defect; - - if (!(alsa_outchannels && alsa_inchannels)) - return; - while (checkit) - { - checkit = 0; - if (giveup-- <= 0) - return; - result = snd_pcm_delay(alsa_device.outhandle, &outdelay); - if (result < 0) - { - post("output snd_pcm_delay failed: %s", snd_strerror(result)); - if (snd_pcm_status(alsa_device.outhandle, out_status) < 0) - post("output snd_pcm_status failed"); - else post("astate %d", - snd_pcm_status_get_state(out_status)); - return; - } - if (outdelay < 0) - sys_log_error(ERR_DATALATE), alreadylogged = 1; - - if (sys_inchannels) - { - result = snd_pcm_delay(alsa_device.inhandle, &indelay); - if (result < 0) - { - post("input snd_pcm_delay failed"); - return; - } - defect = indelay + outdelay - alsa_buf_samps; - if (defect < -DEFDACBLKSIZE) - { - checkit = 1; - alsa_putzeros(1); - if (!alreadylogged) - sys_log_error(ERR_RESYNC), alreadylogged = 1; - } - else if (defect > 0) - { - checkit = 1; - alsa_getzeros(1); - if (!alreadylogged) - sys_log_error(ERR_RESYNC), alreadylogged = 1; - } - } - } -} - -void alsa_listdevs( void) -{ - post("device listing not implemented in ALSA yet\n"); -} - -static int alsa_nnames = 0; -static char **alsa_names = 0; - -void alsa_adddev(char *name) -{ - if (alsa_nnames) - alsa_names = (char **)t_resizebytes(alsa_names, - alsa_nnames * sizeof(char *), - (alsa_nnames+1) * sizeof(char *)); - else alsa_names = (char **)t_getbytes(sizeof(char *)); - alsa_names[alsa_nnames] = gensym(name)->s_name; - alsa_nnames++; -} - -static void alsa_numbertoname(int devno, char *devname, int nchar) -{ - int ndev = 0, cardno = -1; - while (!snd_card_next(&cardno) && cardno >= 0) - ndev++; - if (devno < 2*ndev) - { - if (devno & 1) - snprintf(devname, nchar, "plughw:%d", devno/2); - else snprintf(devname, nchar, "hw:%d", devno/2); - } - else if (devno <2*ndev + alsa_nnames) - snprintf(devname, nchar, "%s", alsa_names[devno - 2*ndev]); - else snprintf(devname, nchar, "???"); -} - - /* For each hardware card found, we list two devices, the "hard" and - "plug" one. The card scan is derived from portaudio code. */ -void alsa_getdevs(char *indevlist, int *nindevs, - char *outdevlist, int *noutdevs, int *canmulti, - int maxndev, int devdescsize) -{ - int ndev = 0, cardno = -1, i, j; - *canmulti = 0; /* only one device; must be the same for input&output */ - while (!snd_card_next(&cardno) && cardno >= 0) - { - snd_ctl_t *ctl; - snd_ctl_card_info_t *info; - char devname[80]; - const char *desc; - if (2 * ndev + 2 > maxndev) - break; - /* apparently, "cardno" is just a counter; but check that here */ - if (ndev != cardno) - fprintf(stderr, "oops: ALSA cards not reported in order?\n"); - sprintf(devname, "hw:%d", cardno ); - /* fprintf(stderr, "\ntry %s...\n", devname); */ - if (snd_ctl_open(&ctl, devname, 0) >= 0) - { - snd_ctl_card_info_malloc(&info); - snd_ctl_card_info(ctl, info); - desc = snd_ctl_card_info_get_name(info); - snd_ctl_card_info_free(info); - } - else - { - fprintf(stderr, "ALSA card scan error\n"); - desc = "???"; - } - /* fprintf(stderr, "name: %s\n", snd_ctl_card_info_get_name(info)); */ - sprintf(indevlist + 2*ndev * devdescsize, "%s (hardware)", desc); - sprintf(indevlist + (2*ndev + 1) * devdescsize, "%s (plug-in)", desc); - sprintf(outdevlist + 2*ndev * devdescsize, "%s (hardware)", desc); - sprintf(outdevlist + (2*ndev + 1) * devdescsize, "%s (plug-in)", desc); - ndev++; - } - for (i = 0, j = 2*ndev; i < alsa_nnames; i++, j++) - { - if (j >= maxndev) - break; - snprintf(indevlist + j * devdescsize, devdescsize, "%s", - alsa_names[i]); - } - *nindevs = *noutdevs = j; -} -- cgit v1.2.1