diff options
| author | Martin Peach <mrpeach@users.sourceforge.net> | 2010-01-11 14:25:48 +0000 |
|---|---|---|
| committer | Martin Peach <mrpeach@users.sourceforge.net> | 2010-01-11 14:25:48 +0000 |
| commit | d141cb156f5088c3e6b99eca0b86694e957100a8 (patch) | |
| tree | 47c9cf8a2b66790ce7e44ee39330fb814064345d | |
| parent | 0f2f021304f0d8abac28f6126b3cea1c6fe0e5a1 (diff) | |
Versions of netsend~ for udp.
svn path=/trunk/externals/mrpeach/; revision=12953
| -rw-r--r-- | net/float_cast.h | 203 | ||||
| -rw-r--r-- | net/udpreceive~.c | 749 | ||||
| -rw-r--r-- | net/udpsend~.c | 648 | ||||
| -rw-r--r-- | net/udpsend~.h | 156 |
4 files changed, 1756 insertions, 0 deletions
diff --git a/net/float_cast.h b/net/float_cast.h new file mode 100644 index 0000000..0ebc4bb --- /dev/null +++ b/net/float_cast.h @@ -0,0 +1,203 @@ +/*
+** Copyright (C) 2001-2003 Erik de Castro Lopo <erikd@mega-nerd.com>
+**
+** This program 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.
+*/
+
+/* Version 1.3 */
+
+
+/*============================================================================
+** On Intel Pentium processors (especially PIII and probably P4), converting
+** from float to int is very slow. To meet the C specs, the code produced by
+** most C compilers targeting Pentium needs to change the FPU rounding mode
+** before the float to int conversion is performed.
+**
+** Changing the FPU rounding mode causes the FPU pipeline to be flushed. It
+** is this flushing of the pipeline which is so slow.
+**
+** Fortunately the ISO C99 specifications define the functions lrint, lrintf,
+** llrint and llrintf which fix this problem as a side effect.
+**
+** On Unix-like systems, the configure process should have detected the
+** presence of these functions. If they weren't found we have to replace them
+** here with a standard C cast.
+*/
+
+/*
+** The C99 prototypes for lrint and lrintf are as follows:
+**
+** long int lrintf (float x) ;
+** long int lrint (double x) ;
+*/
+
+// #include "config.h"
+
+/*
+** The presence of the required functions are detected during the configure
+** process and the values HAVE_LRINT and HAVE_LRINTF are set accordingly in
+** the config.h file.
+*/
+
+#define HAVE_LRINT_REPLACEMENT 0
+
+#if (HAVE_LRINT && HAVE_LRINTF)
+
+ /*
+ ** These defines enable functionality introduced with the 1999 ISO C
+ ** standard. They must be defined before the inclusion of math.h to
+ ** engage them. If optimisation is enabled, these functions will be
+ ** inlined. With optimisation switched off, you have to link in the
+ ** maths library using -lm.
+ */
+
+ #define _ISOC9X_SOURCE 1
+ #define _ISOC99_SOURCE 1
+
+ #define __USE_ISOC9X 1
+ #define __USE_ISOC99 1
+
+ #include <math.h>
+
+#elif (defined (WIN32) || defined (_WIN32))
+
+ #undef HAVE_LRINT_REPLACEMENT
+ #define HAVE_LRINT_REPLACEMENT 1
+ #include <math.h>
+
+ /*
+ ** Win32 doesn't seem to have these functions.
+ ** Therefore implement inline versions of these functions here.
+ */
+
+ __inline long int
+ lrint (double flt)
+ { int intgr;
+
+ _asm
+ { fld flt
+ fistp intgr
+ } ;
+
+ return intgr ;
+ }
+
+ __inline long int
+ lrintf (float flt)
+ { int intgr;
+
+ _asm
+ { fld flt
+ fistp intgr
+ } ;
+
+ return intgr ;
+ }
+
+#elif (defined (__MWERKS__) && defined (macintosh))
+
+ /* This MacOS 9 solution was provided by Stephane Letz */
+
+ #undef HAVE_LRINT_REPLACEMENT
+ #define HAVE_LRINT_REPLACEMENT 1
+ #include <math.h>
+
+ #undef lrint
+ #undef lrintf
+
+ #define lrint double2int
+ #define lrintf float2int
+
+ inline int
+ float2int (register float in)
+ { long res [2] ;
+
+ asm
+ { fctiw in,in
+ stfd in,res
+ }
+ return res [1] ;
+ } /* float2int */
+
+ inline int
+ double2int (register double in)
+ { long res [2] ;
+
+ asm
+ { fctiw in,in
+ stfd in,res
+ }
+ return res [1] ;
+ } /* double2int */
+
+#elif (defined (__MACH__) && defined (__APPLE__))
+
+ /* For Apple MacOSX. */
+
+ #undef HAVE_LRINT_REPLACEMENT
+ #define HAVE_LRINT_REPLACEMENT 1
+ #include <math.h>
+
+ #undef lrint
+ #undef lrintf
+
+ #define lrint double2int
+ #define lrintf float2int
+
+ inline static long int
+ float2int (register float in)
+ { int res [2] ;
+
+ __asm__ __volatile__
+ ( "fctiw %1, %1\n\t"
+ "stfd %1, %0"
+ : "=m" (res) /* Output */
+ : "f" (in) /* Input */
+ : "memory"
+ ) ;
+
+ return res [1] ;
+ } /* lrintf */
+
+ inline static long int
+ double2int (register double in)
+ { int res [2] ;
+
+ __asm__ __volatile__
+ ( "fctiw %1, %1\n\t"
+ "stfd %1, %0"
+ : "=m" (res) /* Output */
+ : "f" (in) /* Input */
+ : "memory"
+ ) ;
+
+ return res [1] ;
+ } /* lrint */
+
+#else
+ #ifndef __sgi
+ #warning "Don't have the functions lrint() and lrintf()."
+ #warning "Replacing these functions with a standard C cast."
+ #endif
+
+ #include <math.h>
+
+ #define lrint(dbl) ((int) (dbl))
+ #define lrintf(flt) ((int) (flt))
+
+#endif
+
+
+
diff --git a/net/udpreceive~.c b/net/udpreceive~.c new file mode 100644 index 0000000..cbf8040 --- /dev/null +++ b/net/udpreceive~.c @@ -0,0 +1,749 @@ +/* udpreceive~ started 20100110 by Martin Peach based on netreceive~: */
+/* ------------------------ netreceive~ --------------------------------------- */
+/* */
+/* Tilde object to receive uncompressed audio data from netsend~. */
+/* Written by Olaf Matthes <olaf.matthes@gmx.de>. */
+/* Based on streamin~ by Guenter Geiger. */
+/* Get source at http://www.akustische-kunst.org/ */
+/* */
+/* This program 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. */
+/* */
+/* See file LICENSE for further informations on licensing terms. */
+/* */
+/* 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. */
+/* */
+/* Based on PureData by Miller Puckette and others. */
+/* */
+/* This project was commissioned by the Society for Arts and Technology [SAT], */
+/* Montreal, Quebec, Canada, http://www.sat.qc.ca/. */
+/* */
+/* ---------------------------------------------------------------------------- */
+
+
+#include "m_pd.h"
+
+#include "udpsend~.h"
+
+#include <sys/types.h>
+#include <string.h>
+#ifdef UNIX
+#include <sys/socket.h>
+#include <errno.h>
+#include <netinet/in.h>
+#include <netinet/tcp.h>
+#include <arpa/inet.h>
+#include <netdb.h>
+#include <sys/time.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <stdio.h>
+#define SOCKET_ERROR -1
+#else
+#include <winsock.h>
+#endif
+
+#ifndef SOL_IP
+#define SOL_IP IPPROTO_IP
+#endif
+
+#define DEFAULT_AUDIO_BUFFER_FRAMES 16 /* a small circ. buffer for 16 frames */
+#define DEFAULT_AVERAGE_NUMBER 10 /* number of values we store for average history */
+#define DEFAULT_NETWORK_POLLTIME 1 /* interval in ms for polling for input data (Max/MSP only) */
+#define DEFAULT_QUEUE_LENGTH 3 /* min. number of buffers that can be used reliably on your hardware */
+
+#ifdef UNIX
+#define CLOSESOCKET(fd) close(fd)
+#endif
+#ifdef _WIN32
+#define CLOSESOCKET(fd) closesocket(fd)
+#endif
+
+/* ------------------------ udpreceive~ ----------------------------- */
+
+typedef struct _udpreceive_tilde
+{
+ t_object x_obj;
+ t_outlet *x_outlet1;
+ t_outlet *x_outlet2;
+ int x_socket;
+ int x_connectsocket;
+ int x_nconnections;
+ int x_ndrops;
+ int x_tcp;
+ t_symbol *x_hostname;
+
+ /* buffering */
+ int x_framein;
+ int x_frameout;
+ t_frame x_frames[DEFAULT_AUDIO_BUFFER_FRAMES];
+ int x_maxframes;
+ long x_framecount;
+ int x_blocksize;
+ int x_blocksperrecv;
+ int x_blockssincerecv;
+
+ int x_nbytes;
+ int x_counter;
+ int x_average[DEFAULT_AVERAGE_NUMBER];
+ int x_averagecur;
+ int x_underflow;
+ int x_overflow;
+
+ long x_samplerate;
+ int x_noutlets;
+ int x_vecsize;
+ t_int **x_myvec; /* vector we pass on to the DSP routine */
+} t_udpreceive_tilde;
+
+/* function prototypes */
+static void udpreceive_tilde_closesocket(t_udpreceive_tilde* x);
+static void udpreceive_tilde_reset(t_udpreceive_tilde* x, t_floatarg buffer);
+static void udpreceive_tilde_datapoll(t_udpreceive_tilde *x);
+static void udpreceive_tilde_connectpoll(t_udpreceive_tilde *x);
+static int udpreceive_tilde_createsocket(t_udpreceive_tilde* x, int portno);
+static t_int *udpreceive_tilde_perform(t_int *w);
+static void udpreceive_tilde_dsp(t_udpreceive_tilde *x, t_signal **sp);
+static void udpreceive_tilde_info(t_udpreceive_tilde *x);
+static void udpreceive_tilde_print(t_udpreceive_tilde* x);
+static void *udpreceive_tilde_new(t_floatarg fportno, t_floatarg outlets);
+static void udpreceive_tilde_free(t_udpreceive_tilde *x);
+void udpreceive_tilde_setup(void);
+static int udpreceive_tilde_sockerror(char *s);
+static int udpreceive_tilde_setsocketoptions(int sockfd);
+/* these would require to include some headers that are different
+ between pd 0.36 and later, so it's easier to do it like this! */
+EXTERN void sys_rmpollfn(int fd);
+EXTERN void sys_addpollfn(int fd, void* fn, void *ptr);
+
+static t_class *udpreceive_tilde_class;
+static t_symbol *ps_format, *ps_channels, *ps_framesize, *ps_overflow, *ps_underflow,
+ *ps_queuesize, *ps_average, *ps_sf_float, *ps_sf_16bit, *ps_sf_8bit,
+ *ps_sf_mp3, *ps_sf_aac, *ps_sf_unknown, *ps_bitrate, *ps_hostname, *ps_nothing;
+
+/* remove all pollfunctions and close socket */
+static void udpreceive_tilde_closesocket(t_udpreceive_tilde* x)
+{
+ sys_rmpollfn(x->x_socket);
+ outlet_float(x->x_outlet1, 0);
+ CLOSESOCKET(x->x_socket);
+ x->x_socket = -1;
+}
+
+static void udpreceive_tilde_reset(t_udpreceive_tilde* x, t_floatarg buffer)
+{
+ int i;
+
+ x->x_counter = 0;
+ x->x_nbytes = 0;
+ x->x_framein = 0;
+ x->x_frameout = 0;
+ x->x_blockssincerecv = 0;
+ x->x_blocksperrecv = x->x_blocksize / x->x_vecsize;
+
+ for (i = 0; i < DEFAULT_AVERAGE_NUMBER; i++)
+ x->x_average[i] = x->x_maxframes;
+ x->x_averagecur = 0;
+
+ if (buffer == 0.0) /* set default */
+ x->x_maxframes = DEFAULT_QUEUE_LENGTH;
+ else
+ {
+ buffer = (float)CLIP((float)buffer, 0., 1.);
+ x->x_maxframes = (int)(DEFAULT_AUDIO_BUFFER_FRAMES * buffer);
+ x->x_maxframes = CLIP(x->x_maxframes, 1, DEFAULT_AUDIO_BUFFER_FRAMES - 1);
+ post("udpreceive~: set buffer to %g (%d frames)", buffer, x->x_maxframes);
+ }
+ x->x_underflow = 0;
+ x->x_overflow = 0;
+}
+
+static void udpreceive_tilde_datapoll(t_udpreceive_tilde *x)
+{
+ int ret;
+ int n;
+
+ n = x->x_nbytes;
+
+ if (x->x_nbytes == 0) /* we ate all the samples and need a new header tag */
+ {
+ /* receive header tag */
+ ret = recv(x->x_socket, (char*)&x->x_frames[x->x_framein].tag, sizeof(t_tag), 0);
+ if (ret <= 0) /* error */
+ {
+ if (udpreceive_tilde_sockerror("recv tag"))
+ goto bail;
+ udpreceive_tilde_reset(x, 0);
+ x->x_counter = 0;
+ return;
+ }
+ else if (ret != sizeof(t_tag))
+ {
+ /* incomplete header tag: return and try again later */
+ /* in the hope that more data will be available */
+ error("udpreceive~: got incomplete header tag");
+ return;
+ }
+ /* adjust byte order if neccessarry */
+ if (x->x_frames[x->x_framein].tag.version != SF_BYTE_NATIVE)
+ {
+ x->x_frames[x->x_framein].tag.count = netsend_long(x->x_frames[x->x_framein].tag.count);
+ x->x_frames[x->x_framein].tag.framesize = netsend_long(x->x_frames[x->x_framein].tag.framesize);
+ }
+ /* get info from header tag */
+ if (x->x_frames[x->x_framein].tag.channels > x->x_noutlets)
+ {
+ error("udpreceive~: incoming stream has too many channels (%d)", x->x_frames[x->x_framein].tag.channels);
+ x->x_counter = 0;
+ return;
+ }
+ x->x_nbytes = n = x->x_frames[x->x_framein].tag.framesize;
+ }
+ else /* we already have header tag or some data and need more */
+ {
+ ret = recv(x->x_socket, (char*)x->x_frames[x->x_framein].data + x->x_frames[x->x_framein].tag.framesize - n, n, 0);
+ if (ret > 0)
+ {
+ n -= ret;
+ }
+ else if (ret < 0) /* error */
+ {
+ if (udpreceive_tilde_sockerror("recv data"))
+ goto bail;
+ udpreceive_tilde_reset(x, 0);
+ x->x_counter = 0;
+ return;
+ }
+
+ x->x_nbytes = n;
+ if (n == 0) /* a complete packet is received */
+ {
+ if (x->x_frames[x->x_framein].tag.format == SF_AAC)
+ {
+ error("udpreceive~: don't know how to decode AAC format");
+ return;
+ }
+ x->x_counter++;
+ x->x_framein++;
+ x->x_framein %= DEFAULT_AUDIO_BUFFER_FRAMES;
+
+ /* check for buffer overflow */
+ if (x->x_framein == x->x_frameout)
+ {
+ x->x_overflow++;
+ }
+ }
+ }
+bail:
+;
+}
+
+static void udpreceive_tilde_connectpoll(t_udpreceive_tilde *x)
+{
+ int sockaddrl = (int)sizeof(struct sockaddr);
+ struct sockaddr_in incomer_address;
+ int fd = accept(x->x_connectsocket, (struct sockaddr*)&incomer_address, &sockaddrl);
+
+ if (fd < 0)
+ {
+ post("udpreceive~: accept failed");
+ return;
+ }
+#ifdef O_NONBLOCK
+ fcntl(fd, F_SETFL, O_NONBLOCK);
+#endif
+ if (x->x_socket != -1)
+ {
+ post("udpreceive~: new connection");
+ udpreceive_tilde_closesocket(x);
+ }
+
+ udpreceive_tilde_reset(x, 0);
+ x->x_socket = fd;
+ x->x_nbytes = 0;
+ x->x_hostname = gensym(inet_ntoa(incomer_address.sin_addr));
+ sys_addpollfn(fd, udpreceive_tilde_datapoll, x);
+ outlet_float(x->x_outlet1, 1);
+}
+
+static int udpreceive_tilde_createsocket(t_udpreceive_tilde* x, int portno)
+{
+ struct sockaddr_in server;
+ int sockfd;
+
+ /* create a socket */
+ sockfd = socket(AF_INET, SOCK_DGRAM, 0);
+
+ if (sockfd < 0)
+ {
+ udpreceive_tilde_sockerror("socket");
+ return 0;
+ }
+ server.sin_family = AF_INET;
+ server.sin_addr.s_addr = INADDR_ANY;
+
+ /* assign server port number */
+
+ server.sin_port = htons((u_short)portno);
+ post("listening to port number %d", portno);
+
+ udpreceive_tilde_setsocketoptions(sockfd);
+
+ /* name the socket */
+ if (bind(sockfd, (struct sockaddr *)&server, sizeof(server)) < 0)
+ {
+ udpreceive_tilde_sockerror("bind");
+ CLOSESOCKET(sockfd);
+ return 0;
+ }
+
+ x->x_socket = sockfd;
+ x->x_nbytes = 0;
+ sys_addpollfn(sockfd, udpreceive_tilde_datapoll, x);
+ return 1;
+}
+
+/* Queue is 1 to 16 frames long */
+#define QUEUESIZE (int)((x->x_framein + DEFAULT_AUDIO_BUFFER_FRAMES - x->x_frameout) % DEFAULT_AUDIO_BUFFER_FRAMES)
+/* Block is a set of sample vectors inside a frame, one vector per channel */
+#define BLOCKOFFSET (x->x_blockssincerecv * x->x_vecsize * x->x_frames[x->x_frameout].tag.channels)
+
+static t_int *udpreceive_tilde_perform(t_int *w)
+{
+ t_udpreceive_tilde *x = (t_udpreceive_tilde*) (w[1]);
+ int n = (int)(w[2]);
+ t_float *out[DEFAULT_AUDIO_CHANNELS];
+ const int offset = 3;
+ const int channels = x->x_frames[x->x_frameout].tag.channels;
+ int i = 0;
+
+ for (i = 0; i < x->x_noutlets; i++)
+ {
+ out[i] = (t_float *)(w[offset + i]);
+ }
+
+ /* set our vector size to the local vector size */
+ if (n != x->x_vecsize)
+ {
+ x->x_vecsize = n;
+ x->x_blocksperrecv = x->x_blocksize / x->x_vecsize;
+ x->x_blockssincerecv = 0;
+ }
+
+ /* check whether there is enough data in buffer */
+ if (x->x_counter < x->x_maxframes)
+ {
+ goto bail;
+ }
+
+ /* check for buffer underflow */
+ if (x->x_framein == x->x_frameout)
+ {
+ x->x_underflow++;
+ goto bail;
+ }
+
+ /* queue balancing */
+ x->x_average[x->x_averagecur] = QUEUESIZE;
+ if (++x->x_averagecur >= DEFAULT_AVERAGE_NUMBER)
+ x->x_averagecur = 0;
+
+ switch (x->x_frames[x->x_frameout].tag.format)
+ {
+ case SF_FLOAT:
+ {
+ t_float* buf = (t_float *)x->x_frames[x->x_frameout].data + BLOCKOFFSET;
+
+ if (x->x_frames[x->x_frameout].tag.version == SF_BYTE_NATIVE)
+ {
+ while (n--)
+ {
+ for (i = 0; i < channels; i++)
+ {
+ *out[i]++ = *buf++;
+ }
+ for (i = channels; i < x->x_noutlets; i++)
+ {
+ *out[i]++ = 0.;
+ }
+ }
+ }
+ else /* swap bytes */
+ {
+ while (n--)
+ {
+ for (i = 0; i < channels; i++)
+ {
+ *out[i]++ = netsend_float(*buf++);
+ }
+ for (i = channels; i < x->x_noutlets; i++)
+ {
+ *out[i]++ = 0.;
+ }
+ }
+ }
+ break;
+ }
+ case SF_16BIT:
+ {
+ short* buf = (short *)x->x_frames[x->x_frameout].data + BLOCKOFFSET;
+
+ if (x->x_frames[x->x_frameout].tag.version == SF_BYTE_NATIVE)
+ {
+ while (n--)
+ {
+ for (i = 0; i < channels; i++)
+ {
+ *out[i]++ = (t_float)(*buf++ * 3.051850e-05);
+ }
+ for (i = channels; i < x->x_noutlets; i++)
+ {
+ *out[i]++ = 0.;
+ }
+ }
+ }
+ else /* swap bytes */
+ {
+ while (n--)
+ {
+ for (i = 0; i < channels; i++)
+ {
+ *out[i]++ = (t_float)(netsend_short(*buf++) * 3.051850e-05);
+ }
+ for (i = channels; i < x->x_noutlets; i++)
+ {
+ *out[i]++ = 0.;
+ }
+ }
+ }
+ break;
+ }
+ case SF_8BIT:
+ {
+ unsigned char* buf = (char *)x->x_frames[x->x_frameout].data + BLOCKOFFSET;
+
+ while (n--)
+ {
+ for (i = 0; i < channels; i++)
+ {
+ *out[i]++ = (t_float)((0.0078125 * (*buf++)) - 1.0);
+ }
+ for (i = channels; i < x->x_noutlets; i++)
+ {
+ *out[i]++ = 0.;
+ }
+ }
+ break;
+ }
+ case SF_MP3:
+ {
+ post("udpreceive~: mp3 format not supported");
+ }
+ case SF_AAC:
+ {
+ post("udpreceive~: aac format not supported");
+ break;
+ }
+ default:
+ post("udpreceive~: unknown format (%d)",x->x_frames[x->x_frameout].tag.format);
+ break;
+ }
+
+ if (!(x->x_blockssincerecv < x->x_blocksperrecv - 1))
+ {
+ x->x_blockssincerecv = 0;
+ x->x_frameout++;
+ x->x_frameout %= DEFAULT_AUDIO_BUFFER_FRAMES;
+ }
+ else
+ {
+ x->x_blockssincerecv++;
+ }
+
+ return (w + offset + x->x_noutlets);
+
+bail:
+ /* set output to zero */
+ while (n--)
+ {
+ for (i = 0; i < x->x_noutlets; i++)
+ {
+ *(out[i]++) = 0.;
+ }
+ }
+ return (w + offset + x->x_noutlets);
+}
+
+static void udpreceive_tilde_dsp(t_udpreceive_tilde *x, t_signal **sp)
+{
+ int i;
+
+ x->x_myvec[0] = (t_int*)x;
+ x->x_myvec[1] = (t_int*)sp[0]->s_n;
+
+ x->x_samplerate = (long)sp[0]->s_sr;
+
+ if (DEFAULT_AUDIO_BUFFER_SIZE % sp[0]->s_n)
+ {
+ error("netsend~: signal vector size too large (needs to be even divisor of %d)", DEFAULT_AUDIO_BUFFER_SIZE);
+ }
+ else
+ {
+ for (i = 0; i < x->x_noutlets; i++)
+ {
+ x->x_myvec[2 + i] = (t_int*)sp[i + 1]->s_vec;
+ }
+ dsp_addv(udpreceive_tilde_perform, x->x_noutlets + 2, (t_int*)x->x_myvec);
+ }
+}
+
+/* send stream info */
+static void udpreceive_tilde_info(t_udpreceive_tilde *x)
+{
+ t_atom list[2];
+ t_symbol *sf_format;
+ t_float bitrate;
+ int i, avg = 0;
+
+ for (i = 0; i < DEFAULT_AVERAGE_NUMBER; i++)
+ avg += x->x_average[i];
+
+ bitrate = (t_float)((SF_SIZEOF(x->x_frames[x->x_frameout].tag.format) * x->x_samplerate * 8 * x->x_frames[x->x_frameout].tag.channels) / 1000.);
+
+ switch (x->x_frames[x->x_frameout].tag.format)
+ {
+ case SF_FLOAT:
+ {
+ sf_format = ps_sf_float;
+ break;
+ }
+ case SF_16BIT:
+ {
+ sf_format = ps_sf_16bit;
+ break;
+ }
+ case SF_8BIT:
+ {
+ sf_format = ps_sf_8bit;
+ break;
+ }
+ case SF_MP3:
+ {
+ sf_format = ps_sf_mp3;
+ break;
+ }
+ case SF_AAC:
+ {
+ sf_format = ps_sf_aac;
+ break;
+ }
+ default:
+ {
+ sf_format = ps_sf_unknown;
+ break;
+ }
+ }
+
+ /* --- stream information (t_tag) --- */
+ /* audio format */
+ SETSYMBOL(list, (t_symbol *)sf_format);
+ outlet_anything(x->x_outlet2, ps_format, 1, list);
+
+ /* channels */
+ SETFLOAT(list, (t_float)x->x_frames[x->x_frameout].tag.channels);
+ outlet_anything(x->x_outlet2, ps_channels, 1, list);
+
+ /* framesize */
+ SETFLOAT(list, (t_float)x->x_frames[x->x_frameout].tag.framesize);
+ outlet_anything(x->x_outlet2, ps_framesize, 1, list);
+
+ /* bitrate */
+ SETFLOAT(list, (t_float)bitrate);
+ outlet_anything(x->x_outlet2, ps_bitrate, 1, list);
+
+ /* --- internal info (buffer and network) --- */
+ /* overflow */
+ SETFLOAT(list, (t_float)x->x_overflow);
+ outlet_anything(x->x_outlet2, ps_overflow, 1, list);
+
+ /* underflow */
+ SETFLOAT(list, (t_float)x->x_underflow);
+ outlet_anything(x->x_outlet2, ps_underflow, 1, list);
+
+ /* queuesize */
+ SETFLOAT(list, (t_float)QUEUESIZE);
+ outlet_anything(x->x_outlet2, ps_queuesize, 1, list);
+
+ /* average queuesize */
+ SETFLOAT(list, (t_float)((t_float)avg / (t_float)DEFAULT_AVERAGE_NUMBER));
+ outlet_anything(x->x_outlet2, ps_average, 1, list);
+}
+
+static void udpreceive_tilde_print(t_udpreceive_tilde* x)
+{
+ int i, avg = 0;
+
+ for (i = 0; i < DEFAULT_AVERAGE_NUMBER; i++)
+ avg += x->x_average[i];
+ post("udpreceive~: last size = %d, avg size = %g, %d underflows, %d overflows", QUEUESIZE, (float)((float)avg / (float)DEFAULT_AVERAGE_NUMBER), x->x_underflow, x->x_overflow);
+ post("udpreceive~: channels = %d, framesize = %d, packets = %d", x->x_frames[x->x_framein].tag.channels, x->x_frames[x->x_framein].tag.framesize, x->x_counter);
+}
+
+static void *udpreceive_tilde_new(t_floatarg fportno, t_floatarg outlets)
+{
+ t_udpreceive_tilde *x;
+ int i;
+
+ if (fportno == 0) fportno = DEFAULT_PORT;
+
+ x = (t_udpreceive_tilde *)pd_new(udpreceive_tilde_class);
+ if (x)
+ {
+ for (i = sizeof(t_object); i < (int)sizeof(t_udpreceive_tilde); i++)
+ ((char *)x)[i] = 0;
+
+ x->x_noutlets = CLIP((int)outlets, 1, DEFAULT_AUDIO_CHANNELS);
+ for (i = 0; i < x->x_noutlets; i++)
+ outlet_new(&x->x_obj, &s_signal);
+ x->x_outlet2 = outlet_new(&x->x_obj, &s_anything);
+ x->x_myvec = (t_int **)t_getbytes(sizeof(t_int *) * (x->x_noutlets + 3));
+ if (!x->x_myvec)
+ {
+ error("udpreceive~: out of memory");
+ return NULL;
+ }
+
+ x->x_connectsocket = -1;
+ x->x_socket = -1;
+ x->x_nconnections = 0;
+ x->x_ndrops = 0;
+ x->x_underflow = 0;
+ x->x_overflow = 0;
+ x->x_hostname = ps_nothing;
+/* allocate space for 16 frames of 1024 X numchannels floats*/
+ for (i = 0; i < DEFAULT_AUDIO_BUFFER_FRAMES; i++)
+ {
+ x->x_frames[i].data = (char *)t_getbytes(DEFAULT_AUDIO_BUFFER_SIZE * x->x_noutlets * sizeof(t_float));
+ }
+ x->x_framein = 0;
+ x->x_frameout = 0;
+ x->x_maxframes = DEFAULT_QUEUE_LENGTH;
+ x->x_vecsize = 64; /* we'll update this later */
+ x->x_blocksize = DEFAULT_AUDIO_BUFFER_SIZE; /* LATER make this dynamic */
+ x->x_blockssincerecv = 0;
+ x->x_blocksperrecv = x->x_blocksize / x->x_vecsize;
+
+ if (!udpreceive_tilde_createsocket(x, (int)fportno))
+ {
+ error("udpreceive~: failed to create listening socket");
+ return (NULL);
+ }
+ }
+ return (x);
+}
+
+static void udpreceive_tilde_free(t_udpreceive_tilde *x)
+{
+ int i;
+
+ if (x->x_connectsocket != -1)
+ {
+ sys_rmpollfn(x->x_connectsocket);
+ CLOSESOCKET(x->x_connectsocket);
+ }
+ if (x->x_socket != -1)
+ {
+ sys_rmpollfn(x->x_socket);
+ CLOSESOCKET(x->x_socket);
+ }
+
+ /* free memory */
+ t_freebytes(x->x_myvec, sizeof(t_int *) * (x->x_noutlets + 3));
+ for (i = 0; i < DEFAULT_AUDIO_BUFFER_FRAMES; i++)
+ {
+ t_freebytes(x->x_frames[i].data, DEFAULT_AUDIO_BUFFER_SIZE * x->x_noutlets * sizeof(t_float));
+ }
+}
+
+void udpreceive_tilde_setup(void)
+{
+ udpreceive_tilde_class = class_new(gensym("udpreceive~"),
+ (t_newmethod) udpreceive_tilde_new, (t_method) udpreceive_tilde_free,
+ sizeof(t_udpreceive_tilde), 0, A_DEFFLOAT, A_DEFFLOAT, A_DEFFLOAT, A_NULL);
+
+ class_addmethod(udpreceive_tilde_class, nullfn, gensym("signal"), 0);
+ class_addmethod(udpreceive_tilde_class, (t_method)udpreceive_tilde_info, gensym("info"), 0);
+ class_addmethod(udpreceive_tilde_class, (t_method)udpreceive_tilde_dsp, gensym("dsp"), 0);
+ class_addmethod(udpreceive_tilde_class, (t_method)udpreceive_tilde_print, gensym("print"), 0);
+ class_addmethod(udpreceive_tilde_class, (t_method)udpreceive_tilde_reset, gensym("reset"), A_DEFFLOAT, 0);
+ class_addmethod(udpreceive_tilde_class, (t_method)udpreceive_tilde_reset, gensym("buffer"), A_DEFFLOAT, 0);
+ post("udpreceive~ v%s, (c) 2004 Olaf Matthes, 2010 Martin Peach", VERSION);
+
+ ps_format = gensym("format");
+ ps_channels = gensym("channels");
+ ps_framesize = gensym("framesize");
+ ps_bitrate = gensym("bitrate");
+ ps_overflow = gensym("overflow");
+ ps_underflow = gensym("underflow");
+ ps_queuesize = gensym("queuesize");
+ ps_average = gensym("average");
+ ps_hostname = gensym("ipaddr");
+ ps_sf_float = gensym("_float_");
+ ps_sf_16bit = gensym("_16bit_");
+ ps_sf_8bit = gensym("_8bit_");
+ ps_sf_mp3 = gensym("_mp3_");
+ ps_sf_aac = gensym("_aac_");
+ ps_sf_unknown = gensym("_unknown_");
+ ps_nothing = gensym("");
+}
+
+/* error handlers */
+static int udpreceive_tilde_sockerror(char *s)
+{
+#ifdef _WIN32
+ int err = WSAGetLastError();
+ if (err == 10054) return 1;
+ else if (err == 10040) post("netsend~: %s: message too long (%d)", s, err);
+ else if (err == 10053) post("netsend~: %s: software caused connection abort (%d)", s, err);
+ else if (err == 10055) post("netsend~: %s: no buffer space available (%d)", s, err);
+ else if (err == 10060) post("netsend~: %s: connection timed out (%d)", s, err);
+ else if (err == 10061) post("netsend~: %s: connection refused (%d)", s, err);
+ else post("udpreceive~: %s: %s (%d)", s, strerror(err), err);
+#else
+ int err = errno;
+ post("udpreceive~: %s: %s (%d)", s, strerror(err), err);
+#endif
+#ifdef _WIN32
+ if (err == WSAEWOULDBLOCK)
+#endif
+#ifdef UNIX
+ if (err == EAGAIN)
+#endif
+ {
+ return 1; /* recoverable error */
+ }
+ return 0; /* indicate non-recoverable error */
+}
+
+static int udpreceive_tilde_setsocketoptions(int sockfd)
+{
+ int sockopt = 1;
+ if (setsockopt(sockfd, SOL_IP, TCP_NODELAY, (const char*)&sockopt, sizeof(int)) < 0)
+ post("setsockopt NODELAY failed");
+
+ sockopt = 1;
+ if (setsockopt(sockfd, SOL_SOCKET, SO_REUSEADDR, (const char*)&sockopt, sizeof(int)) < 0)
+ post("udpreceive~: setsockopt REUSEADDR failed");
+ return 0;
+}
+
+/* fin udpreceive~.c */
diff --git a/net/udpsend~.c b/net/udpsend~.c new file mode 100644 index 0000000..ce9bcad --- /dev/null +++ b/net/udpsend~.c @@ -0,0 +1,648 @@ +/* udpsend~ started by Martin Peach on 20100110, based on netsend~ */
+/* udpsend~ sends audio via udp only.*/
+/* It is a PD external, all Max stuff has been removed from the source */
+/* ------------------------ netsend~ ------------------------------------------ */
+/* */
+/* Tilde object to send uncompressed audio data to netreceive~. */
+/* Written by Olaf Matthes <olaf.matthes@gmx.de>. */
+/* Based on streamout~ by Guenter Geiger. */
+/* Get source at http://www.akustische-kunst.org/ */
+/* */
+/* This program 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. */
+/* */
+/* See file LICENSE for further informations on licensing terms. */
+/* */
+/* 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. */
+/* */
+/* Based on PureData by Miller Puckette and others. */
+/* */
+/* This project was commissioned by the Society for Arts and Technology [SAT], */
+/* Montreal, Quebec, Canada, http://www.sat.qc.ca/. */
+/* */
+/* ---------------------------------------------------------------------------- */
+
+#include "m_pd.h"
+
+#include "udpsend~.h"
+#include "float_cast.h" /* tools for fast conversion from float to int */
+
+#include <sys/types.h>
+#include <string.h>
+#include <stdlib.h>
+#ifdef UNIX
+#include <sys/socket.h>
+#include <errno.h>
+#include <netinet/in.h>
+#include <netinet/tcp.h>
+#include <arpa/inet.h>
+#include <netdb.h>
+#include <sys/time.h>
+#include <signal.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <stdio.h>
+#include <pthread.h>
+#define SOCKET_ERROR -1
+#endif
+#ifdef _WIN32
+#include <winsock.h>
+#include "pthread.h"
+#endif
+
+#ifdef MSG_NOSIGNAL
+#define SEND_FLAGS /*MSG_DONTWAIT|*/MSG_NOSIGNAL
+#else
+#define SEND_FLAGS 0
+#endif
+
+#ifndef SOL_IP
+#define SOL_IP IPPROTO_IP
+#endif
+
+
+/* ------------------------ udpsend~ ----------------------------- */
+
+static t_class *udpsend_tilde_class;
+
+static t_symbol *ps_nothing, *ps_localhost;
+static t_symbol *ps_format, *ps_channels, *ps_framesize, *ps_overflow, *ps_underflow;
+static t_symbol *ps_queuesize, *ps_average, *ps_sf_float, *ps_sf_16bit, *ps_sf_8bit;
+static t_symbol *ps_sf_mp3, *ps_sf_aac, *ps_sf_unknown, *ps_bitrate, *ps_hostname;
+
+typedef struct _udpsend_tilde
+{
+ t_object x_obj;
+ t_outlet *x_outlet;
+ t_outlet *x_outlet2;
+ t_clock *x_clock;
+ int x_fd;
+ t_tag x_tag;
+ t_symbol* x_hostname;
+ int x_portno;
+ int x_connectstate;
+ char *x_cbuf;
+ int x_cbufsize;
+ int x_blocksize; /* set to DEFAULT_AUDIO_BUFFER_SIZE in udpsend_tilde_new() */
+ int x_blockspersend; /* set to x->x_blocksize / x->x_vecsize in udpsend_tilde_perform() */
+ int x_blockssincesend;
+
+ long x_samplerate; /* samplerate we're running at */
+ int x_vecsize; /* current DSP signal vector size */
+ int x_ninlets; /* number of inlets */
+ int x_channels; /* number of channels we want to stream */
+ int x_format; /* format of streamed audio data */
+ int x_bitrate; /* specifies bitrate for compressed formats */
+ int x_count; /* total number of audio frames */
+ t_int **x_myvec; /* vector we pass on in the DSP routine */
+
+ pthread_mutex_t x_mutex;
+ pthread_cond_t x_requestcondition;
+ pthread_cond_t x_answercondition;
+ pthread_t x_childthread;
+} t_udpsend_tilde;
+
+/* function prototypes */
+static int udpsend_tilde_sockerror(char *s);
+static void udpsend_tilde_closesocket(int fd);
+static void udpsend_tilde_notify(t_udpsend_tilde *x);
+static void udpsend_tilde_disconnect(t_udpsend_tilde *x);
+static void *udpsend_tilde_doconnect(void *zz);
+static void udpsend_tilde_connect(t_udpsend_tilde *x, t_symbol *host, t_floatarg fportno);
+static t_int *udpsend_tilde_perform(t_int *w);
+static void udpsend_tilde_dsp(t_udpsend_tilde *x, t_signal **sp);
+static void udpsend_tilde_channels(t_udpsend_tilde *x, t_floatarg channels);
+static void udpsend_tilde_format(t_udpsend_tilde *x, t_symbol* form, t_floatarg bitrate);
+static void udpsend_tilde_float(t_udpsend_tilde* x, t_floatarg arg);
+static void udpsend_tilde_info(t_udpsend_tilde *x);
+static void *udpsend_tilde_new(t_floatarg inlets, t_floatarg prot);
+static void udpsend_tilde_free(t_udpsend_tilde* x);
+void udpsend_tilde_setup(void);
+
+/* functions */
+static void udpsend_tilde_notify(t_udpsend_tilde *x)
+{
+ pthread_mutex_lock(&x->x_mutex);
+ x->x_childthread = 0;
+ outlet_float(x->x_outlet, x->x_connectstate);
+ pthread_mutex_unlock(&x->x_mutex);
+}
+
+static void udpsend_tilde_disconnect(t_udpsend_tilde *x)
+{
+ pthread_mutex_lock(&x->x_mutex);
+ if (x->x_fd != -1)
+ {
+ udpsend_tilde_closesocket(x->x_fd);
+ x->x_fd = -1;
+ x->x_connectstate = 0;
+ outlet_float(x->x_outlet, 0);
+ }
+ pthread_mutex_unlock(&x->x_mutex);
+}
+
+static void *udpsend_tilde_doconnect(void *zz)
+{
+ t_udpsend_tilde *x = (t_udpsend_tilde *)zz;
+ struct sockaddr_in server;
+ struct hostent *hp;
+ int intarg = 1;
+ int sockfd;
+ int portno;
+ t_symbol *hostname;
+
+ pthread_mutex_lock(&x->x_mutex);
+ hostname = x->x_hostname;
+ portno = x->x_portno;
+ pthread_mutex_unlock(&x->x_mutex);
+
+ /* create a socket */
+ sockfd = socket(AF_INET, SOCK_DGRAM, 0);
+ if (sockfd < 0)
+ {
+ post("udpsend~: connection to %s on port %d failed", hostname->s_name,portno);
+ udpsend_tilde_sockerror("socket");
+ x->x_childthread = 0;
+ return (0);
+ }
+
+ /* connect socket using hostname provided in command line */
+ server.sin_family = AF_INET;
+ hp = gethostbyname(x->x_hostname->s_name);
+ if (hp == 0)
+ {
+ post("udpsend~: bad host?");
+ x->x_childthread = 0;
+ return (0);
+ }
+
+#ifdef SO_PRIORITY
+ /* set high priority, LINUX only */
+ intarg = 6; /* select a priority between 0 and 7 */
+ if (setsockopt(sockfd, SOL_SOCKET, SO_PRIORITY, (const char*)&intarg, sizeof(int)) < 0)
+ {
+ error("udpsend~: setsockopt(SO_PRIORITY) failed");
+ }
+#endif
+
+ memcpy((char *)&server.sin_addr, (char *)hp->h_addr, hp->h_length);
+
+ /* assign client port number */
+ server.sin_port = htons((unsigned short)portno);
+
+ /* try to connect */
+ if (connect(sockfd, (struct sockaddr *) &server, sizeof (server)) < 0)
+ {
+ udpsend_tilde_sockerror("connecting stream socket");
+ udpsend_tilde_closesocket(sockfd);
+ x->x_childthread = 0;
+ return (0);
+ }
+
+ post("udpsend~: connected host %s on port %d", hostname->s_name, portno);
+
+ pthread_mutex_lock(&x->x_mutex);
+ x->x_fd = sockfd;
+ x->x_connectstate = 1;
+ clock_delay(x->x_clock, 0);
+ pthread_mutex_unlock(&x->x_mutex);
+ return (0);
+}
+
+static void udpsend_tilde_connect(t_udpsend_tilde *x, t_symbol *host, t_floatarg fportno)
+{
+ pthread_mutex_lock(&x->x_mutex);
+ if (x->x_childthread != 0)
+ {
+ pthread_mutex_unlock(&x->x_mutex);
+ post("udpsend~: already trying to connect");
+ return;
+ }
+ if (x->x_fd != -1)
+ {
+ pthread_mutex_unlock(&x->x_mutex);
+ post("udpsend~: already connected");
+ return;
+ }
+
+ if (host != ps_nothing)
+ x->x_hostname = host;
+ else
+ x->x_hostname = ps_localhost; /* default host */
+
+ if (!fportno)
+ x->x_portno = DEFAULT_PORT;
+ else
+ x->x_portno = (int)fportno;
+ x->x_count = 0;
+
+ /* start child thread to connect */
+ pthread_create(&x->x_childthread, 0, udpsend_tilde_doconnect, x);
+ pthread_mutex_unlock(&x->x_mutex);
+}
+
+static t_int *udpsend_tilde_perform(t_int *w)
+{
+ t_udpsend_tilde* x = (t_udpsend_tilde*) (w[1]);
+ int n = (int)(w[2]);
+ t_float *in[DEFAULT_AUDIO_CHANNELS];
+ const int offset = 3;
+ char* bp = NULL;
+ int i, length = x->x_blocksize * SF_SIZEOF(x->x_tag.format) * x->x_tag.channels;
+ int sent = 0;
+
+ pthread_mutex_lock(&x->x_mutex);
+
+ for (i = 0; i < x->x_ninlets; i++)
+ in[i] = (t_float *)(w[offset + i]);
+
+ if (n != x->x_vecsize) /* resize buffer */
+ {
+ x->x_vecsize = n;
+ x->x_blockspersend = x->x_blocksize / x->x_vecsize;
+ x->x_blockssincesend = 0;
+ length = x->x_blocksize * SF_SIZEOF(x->x_tag.format) * x->x_tag.channels;
+ }
+
+ /* format the buffer */
+ switch (x->x_tag.format)
+ {
+ case SF_FLOAT:
+ {
+ t_float* fbuf = (t_float *)x->x_cbuf + (x->x_blockssincesend * x->x_vecsize * x->x_tag.channels);
+ while (n--)
+ for (i = 0; i < x->x_tag.channels; i++)
+ *fbuf++ = *(in[i]++);
+ break;
+ }
+ case SF_16BIT:
+ {
+ short* cibuf = (short *)x->x_cbuf + (x->x_blockssincesend * x->x_vecsize * x->x_tag.channels);
+ while (n--)
+ for (i = 0; i < x->x_tag.channels; i++)
+ *cibuf++ = (short)lrint(32767.0 * *(in[i]++));
+ break;
+ }
+ case SF_8BIT:
+ {
+ unsigned char* cbuf = (unsigned char*)x->x_cbuf + (x->x_blockssincesend * x->x_vecsize * x->x_tag.channels);
+ while (n--)
+ for (i = 0; i < x->x_tag.channels; i++)
+ *cbuf++ = (unsigned char)(128. * (1.0 + *(in[i]++)));
+ break;
+ }
+ default:
+ break;
+ }
+
+ if (!(x->x_blockssincesend < x->x_blockspersend - 1)) /* time to send the buffer */
+ {
+ x->x_blockssincesend = 0;
+ x->x_count++; /* count data packet we're going to send */
+
+ if (x->x_fd != -1)
+ {
+ bp = (char *)x->x_cbuf;
+ /* fill in the header tag */
+ x->x_tag.framesize = length;
+ x->x_tag.count = x->x_count;
+ /* send the format tag */
+ if (send(x->x_fd, (char*)&x->x_tag, sizeof(t_tag), SEND_FLAGS) < 0)
+ {
+ udpsend_tilde_sockerror("send tag");
+ pthread_mutex_unlock(&x->x_mutex);
+ udpsend_tilde_disconnect(x);
+ return (w + offset + x->x_ninlets);
+ }
+/* UDP: max. packet size is 64k (incl. headers) so we have to split */
+ {
+#ifdef __APPLE__
+ /* WARNING: due to a 'bug' (maybe Apple would call it a feature?) in OS X
+ send calls with data packets larger than 16k fail with error number 40!
+ Thus we have to split the data packets into several packets that are
+ 16k in size. The other side will reassemble them again. */
+ int size = DEFAULT_UDP_PACKT_SIZE;
+ if (length < size) /* maybe data fits into one packet? */
+ size = length;
+ /* send the buffer */
+ for (sent = 0; sent < length;)
+ {
+ int ret = 0;
+ ret = send(x->x_fd, bp, size, SEND_FLAGS);
+ if (ret <= 0)
+ {
+ udpsend_tilde_sockerror("send data");
+ pthread_mutex_unlock(&x->x_mutex);
+ udpsend_tilde_disconnect(x);
+ return (w + offset + x->x_ninlets);
+ }
+ else
+ {
+ bp += ret;
+ sent += ret;
+ if ((length - sent) < size)
+ size = length - sent;
+ }
+ }
+#else
+ /* send the buffer, the OS might segment it into smaller packets */
+ int ret = send(x->x_fd, bp, length, SEND_FLAGS);
+ if (ret <= 0)
+ {
+ udpsend_tilde_sockerror("send data");
+ pthread_mutex_unlock(&x->x_mutex);
+ udpsend_tilde_disconnect(x);
+ return (w + offset + x->x_ninlets);
+ }
+#endif
+ }
+ }
+
+/* check whether user has updated any parameters */
+ if (x->x_tag.channels != x->x_channels)
+ {
+ x->x_tag.channels = x->x_channels;
+ }
+ if (x->x_tag.format != x->x_format)
+ {
+ x->x_tag.format = x->x_format;
+ }
+ }
+ else
+ {
+ x->x_blockssincesend++;
+ }
+ pthread_mutex_unlock(&x->x_mutex);
+ return (w + offset + x->x_ninlets);
+}
+
+static void udpsend_tilde_dsp(t_udpsend_tilde *x, t_signal **sp)
+{
+ int i;
+
+ pthread_mutex_lock(&x->x_mutex);
+
+ x->x_myvec[0] = (t_int*)x;
+ x->x_myvec[1] = (t_int*)sp[0]->s_n;
+
+ x->x_samplerate = sp[0]->s_sr;
+
+ for (i = 0; i < x->x_ninlets; i++)
+ {
+ x->x_myvec[2 + i] = (t_int*)sp[i]->s_vec;
+ }
+
+ pthread_mutex_unlock(&x->x_mutex);
+
+ if (DEFAULT_AUDIO_BUFFER_SIZE % sp[0]->s_n)
+ {
+ error("udpsend~: signal vector size too large (needs to be even divisor of %d)", DEFAULT_AUDIO_BUFFER_SIZE);
+ }
+ else
+ {
+ dsp_addv(udpsend_tilde_perform, x->x_ninlets + 2, (t_int*)x->x_myvec);
+ }
+}
+
+static void udpsend_tilde_channels(t_udpsend_tilde *x, t_floatarg channels)
+{
+ pthread_mutex_lock(&x->x_mutex);
+ if (channels >= 0 && channels <= DEFAULT_AUDIO_CHANNELS)
+ {
+ x->x_channels = (int)channels;
+ post("udpsend~: channels set to %d", (int)channels);
+ }
+ pthread_mutex_unlock(&x->x_mutex);
+}
+
+static void udpsend_tilde_format(t_udpsend_tilde *x, t_symbol* form, t_floatarg bitrate)
+{
+ pthread_mutex_lock(&x->x_mutex);
+ if (!strncmp(form->s_name,"float", 5) && x->x_tag.format != SF_FLOAT)
+ {
+ x->x_format = (int)SF_FLOAT;
+ }
+ else if (!strncmp(form->s_name,"16bit", 5) && x->x_tag.format != SF_16BIT)
+ {
+ x->x_format = (int)SF_16BIT;
+ }
+ else if (!strncmp(form->s_name,"8bit", 4) && x->x_tag.format != SF_8BIT)
+ {
+ x->x_format = (int)SF_8BIT;
+ }
+
+ post("udpsend~: format set to %s", form->s_name);
+ pthread_mutex_unlock(&x->x_mutex);
+}
+
+static void udpsend_tilde_float(t_udpsend_tilde* x, t_floatarg arg)
+{
+ if (arg == 0.0)
+ udpsend_tilde_disconnect(x);
+ else
+ udpsend_tilde_connect(x,x->x_hostname,(float) x->x_portno);
+}
+
+/* send stream info */
+static void udpsend_tilde_info(t_udpsend_tilde *x)
+{
+ t_atom list[2];
+ t_symbol *sf_format;
+ t_float bitrate;
+
+ bitrate = (t_float)((SF_SIZEOF(x->x_tag.format) * x->x_samplerate * 8 * x->x_tag.channels) / 1000.);
+
+ switch (x->x_tag.format)
+ {
+ case SF_FLOAT:
+ {
+ sf_format = ps_sf_float;
+ break;
+ }
+ case SF_16BIT:
+ {
+ sf_format = ps_sf_16bit;
+ break;
+ }
+ case SF_8BIT:
+ {
+ sf_format = ps_sf_8bit;
+ break;
+ }
+ default:
+ {
+ sf_format = ps_sf_unknown;
+ break;
+ }
+ }
+
+ /* --- stream information (t_tag) --- */
+
+ /* audio format */
+ SETSYMBOL(list, (t_symbol *)sf_format);
+ outlet_anything(x->x_outlet2, ps_format, 1, list);
+
+ /* channels */
+ SETFLOAT(list, (t_float)x->x_tag.channels);
+ outlet_anything(x->x_outlet2, ps_channels, 1, list);
+
+ /* framesize */
+ SETFLOAT(list, (t_float)x->x_tag.framesize);
+ outlet_anything(x->x_outlet2, ps_framesize, 1, list);
+
+ /* bitrate */
+ SETFLOAT(list, (t_float)bitrate);
+ outlet_anything(x->x_outlet2, ps_bitrate, 1, list);
+
+ /* IP address */
+ SETSYMBOL(list, (t_symbol *)x->x_hostname);
+ outlet_anything(x->x_outlet2, ps_hostname, 1, list);
+}
+
+static void *udpsend_tilde_new(t_floatarg inlets, t_floatarg prot)
+{
+ int i;
+
+ t_udpsend_tilde *x = (t_udpsend_tilde *)pd_new(udpsend_tilde_class);
+ if (x)
+ {
+ for (i = sizeof(t_object); i < (int)sizeof(t_udpsend_tilde); i++)
+ ((char *)x)[i] = 0;
+ }
+
+ x->x_ninlets = CLIP((int)inlets, 1, DEFAULT_AUDIO_CHANNELS);
+ for (i = 1; i < x->x_ninlets; i++)
+ inlet_new(&x->x_obj, &x->x_obj.ob_pd, &s_signal, &s_signal);
+
+ x->x_outlet = outlet_new(&x->x_obj, &s_float);
+ x->x_outlet2 = outlet_new(&x->x_obj, &s_list);
+ x->x_clock = clock_new(x, (t_method)udpsend_tilde_notify);
+
+ x->x_myvec = (t_int **)t_getbytes(sizeof(t_int *) * (x->x_ninlets + 3));
+ if (!x->x_myvec)
+ {
+ error("udpsend~: out of memory");
+ return NULL;
+ }
+
+ pthread_mutex_init(&x->x_mutex, 0);
+ pthread_cond_init(&x->x_requestcondition, 0);
+ pthread_cond_init(&x->x_answercondition, 0);
+
+ x->x_hostname = ps_localhost;
+ x->x_portno = DEFAULT_PORT;
+ x->x_connectstate = 0;
+ x->x_childthread = 0;
+ x->x_fd = -1;
+
+ x->x_tag.format = x->x_format = SF_FLOAT;
+ x->x_tag.channels = x->x_channels = x->x_ninlets;
+ x->x_tag.version = SF_BYTE_NATIVE; /* native endianness */
+ x->x_vecsize = 64; /* this is updated in the perform routine udpsend_tilde_perform */
+ x->x_bitrate = 0; /* not specified, use default */
+ x->x_cbuf = NULL;
+ x->x_blocksize = DEFAULT_AUDIO_BUFFER_SIZE; /* <-- the only place blocksize is set */
+ x->x_blockspersend = x->x_blocksize / x->x_vecsize; /* 1024/64 = 16 blocks */
+ x->x_blockssincesend = 0;
+ x->x_cbufsize = x->x_blocksize * sizeof(t_float) * x->x_ninlets;
+ x->x_cbuf = (char *)t_getbytes(x->x_cbufsize);
+
+#ifdef UNIX
+ /* we don't want to get signaled in case send() fails */
+ signal(SIGPIPE, SIG_IGN);
+#endif
+
+ return (x);
+}
+
+static void udpsend_tilde_free(t_udpsend_tilde* x)
+{
+ udpsend_tilde_disconnect(x);
+
+ /* free the memory */
+ if (x->x_cbuf)t_freebytes(x->x_cbuf, x->x_cbufsize);
+ if (x->x_myvec)t_freebytes(x->x_myvec, sizeof(t_int) * (x->x_ninlets + 3));
+
+ clock_free(x->x_clock);
+
+ pthread_cond_destroy(&x->x_requestcondition);
+ pthread_cond_destroy(&x->x_answercondition);
+ pthread_mutex_destroy(&x->x_mutex);
+}
+
+void udpsend_tilde_setup(void)
+{
+ udpsend_tilde_class = class_new(gensym("udpsend~"), (t_newmethod)udpsend_tilde_new, (t_method)udpsend_tilde_free,
+ sizeof(t_udpsend_tilde), 0, A_DEFFLOAT, A_DEFFLOAT, A_NULL);
+ class_addmethod(udpsend_tilde_class, nullfn, gensym("signal"), 0);
+ class_addmethod(udpsend_tilde_class, (t_method)udpsend_tilde_dsp, gensym("dsp"), 0);
+ class_addfloat(udpsend_tilde_class, udpsend_tilde_float);
+ class_addmethod(udpsend_tilde_class, (t_method)udpsend_tilde_info, gensym("info"), 0);
+ class_addmethod(udpsend_tilde_class, (t_method)udpsend_tilde_connect, gensym("connect"), A_DEFSYM, A_DEFFLOAT, 0);
+ class_addmethod(udpsend_tilde_class, (t_method)udpsend_tilde_disconnect, gensym("disconnect"), 0);
+ class_addmethod(udpsend_tilde_class, (t_method)udpsend_tilde_channels, gensym("channels"), A_FLOAT, 0);
+ class_addmethod(udpsend_tilde_class, (t_method)udpsend_tilde_format, gensym("format"), A_SYMBOL, A_DEFFLOAT, 0);
+ class_sethelpsymbol(udpsend_tilde_class, gensym("udpsend~"));
+ post("udpsend~ v%s, (c) 2004-2005 Olaf Matthes, 2010 Martin Peach", VERSION);
+
+ ps_nothing = gensym("");
+ ps_localhost = gensym("localhost");
+ ps_hostname = gensym("ipaddr");
+ ps_format = gensym("format");
+ ps_channels = gensym("channels");
+ ps_framesize = gensym("framesize");
+ ps_bitrate = gensym("bitrate");
+ ps_sf_float = gensym("_float_");
+ ps_sf_16bit = gensym("_16bit_");
+ ps_sf_8bit = gensym("_8bit_");
+ ps_sf_unknown = gensym("_unknown_");
+}
+
+/* Utility functions */
+
+static int udpsend_tilde_sockerror(char *s)
+{
+#ifdef _WIN32
+ int err = WSAGetLastError();
+ if (err == 10054) return 1;
+ else if (err == 10053) post("udpsend~: %s: software caused connection abort (%d)", s, err);
+ else if (err == 10055) post("udpsend~: %s: no buffer space available (%d)", s, err);
+ else if (err == 10060) post("udpsend~: %s: connection timed out (%d)", s, err);
+ else if (err == 10061) post("udpsend~: %s: connection refused (%d)", s, err);
+ else post("udpsend~: %s: %s (%d)", s, strerror(err), err);
+#else
+ int err = errno;
+ post("udpsend~: %s: %s (%d)", s, strerror(err), err);
+#endif
+#ifdef _WIN32
+ if (err == WSAEWOULDBLOCK)
+#endif
+#ifdef UNIX
+ if (err == EAGAIN)
+#endif
+ {
+ return 1; /* recoverable error */
+ }
+ return 0; /* indicate non-recoverable error */
+}
+
+static void udpsend_tilde_closesocket(int fd)
+{
+#ifdef UNIX
+ close(fd);
+#endif
+#ifdef _WIN32
+ closesocket(fd);
+#endif
+}
+
+/* fin udpsend~.c */
diff --git a/net/udpsend~.h b/net/udpsend~.h new file mode 100644 index 0000000..6894e49 --- /dev/null +++ b/net/udpsend~.h @@ -0,0 +1,156 @@ +/* udpsend~.h based on netsend~.h */
+/* ------------------------ netsend~ ------------------------------------------ */
+/* */
+/* Tilde object to send uncompressed audio data to netreceive~. */
+/* Written by Olaf Matthes <olaf.matthes@gmx.de>. */
+/* Based on streamout~ by Guenter Geiger. */
+/* Get source at http://www.akustische-kunst.org/ */
+/* */
+/* This program 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. */
+/* */
+/* See file LICENSE for further informations on licensing terms. */
+/* */
+/* 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. */
+/* */
+/* Based on PureData by Miller Puckette and others. */
+/* */
+/* This project was commissioned by the Society for Arts and Technology [SAT], */
+/* Montreal, Quebec, Canada, http://www.sat.qc.ca/. */
+/* */
+/* ---------------------------------------------------------------------------- */
+
+
+/* This file is based on and inspired by stream.h (C) Guenter Geiger 1999. */
+/* Some enhancements have been made with the goal of keeping compatibility */
+/* between the stream formats of streamout~/in~ and netsend~/receive~. */
+
+#define VERSION "0.1"
+
+#define DEFAULT_AUDIO_CHANNELS 32 /* nax. number of audio channels we support */
+#define DEFAULT_AUDIO_BUFFER_SIZE 1024 /* number of samples in one audio block */
+#define DEFAULT_UDP_PACKT_SIZE 8192 /* number of bytes we send in one UDP datagram (OS X only) */
+#define DEFAULT_PORT 8000 /* default network port number */
+
+#ifdef _WIN32
+#ifndef HAVE_INT32_T
+typedef int int32_t;
+#define HAVE_INT32_T
+#endif
+#ifndef HAVE_INT16_T
+typedef short int16_t;
+#define HAVE_INT16_T
+#endif
+#ifndef HAVE_U_INT32_T
+typedef unsigned int u_int32_t;
+#define HAVE_U_INT32_T
+#endif
+#ifndef HAVE_U_INT16_T
+typedef unsigned short u_int16_t;
+#define HAVE_U_INT16_T
+#endif
+#endif
+
+#ifndef CLIP
+#define CLIP(a, lo, hi) ( (a)>(lo)?( (a)<(hi)?(a):(hi) ):(lo) )
+#endif
+
+
+/* swap 32bit t_float. Is there a better way to do that???? */
+#ifdef _WIN32
+__inline static float netsend_float(float f)
+#else
+inline static float netsend_float(float f)
+#endif
+{
+ union
+ {
+ float f;
+ unsigned char b[4];
+ } dat1, dat2;
+
+ dat1.f = f;
+ dat2.b[0] = dat1.b[3];
+ dat2.b[1] = dat1.b[2];
+ dat2.b[2] = dat1.b[1];
+ dat2.b[3] = dat1.b[0];
+ return dat2.f;
+}
+
+/* swap 32bit long int */
+#ifdef _WIN32
+__inline static long netsend_long(long n)
+#else
+inline static long netsend_long(long n)
+#endif
+{
+ return (((n & 0xff) << 24) | ((n & 0xff00) << 8) |
+ ((n & 0xff0000) >> 8) | ((n & 0xff000000) >> 24));
+}
+
+/* swap 16bit short int */
+#ifdef _WIN32
+__inline static long netsend_short(long n)
+#else
+inline static short netsend_short(short n)
+#endif
+{
+ return (((n & 0xff) << 8) | ((n & 0xff00) >> 8));
+}
+
+
+/* format specific stuff */
+
+#define SF_FLOAT 1
+#define SF_DOUBLE 2 /* not implemented */
+#define SF_8BIT 10
+#define SF_16BIT 11
+#define SF_32BIT 12 /* not implemented */
+#define SF_ALAW 20 /* not implemented */
+#define SF_MP3 30 /* not implemented */
+#define SF_AAC 31 /* AAC encoding using FAAC */
+#define SF_VORBIS 40 /* not implemented */
+#define SF_FLAC 50 /* not implemented */
+
+#define SF_SIZEOF(a) (a == SF_FLOAT ? sizeof(t_float) : \
+ a == SF_16BIT ? sizeof(short) : 1)
+
+
+/* version / byte-endian specific stuff */
+
+#define SF_BYTE_LE 1 /* little endian */
+#define SF_BYTE_BE 2 /* big endian */
+
+#if defined(_WIN32) || defined(__linux__) || defined(IRIX)
+#define SF_BYTE_NATIVE SF_BYTE_LE
+#else /* must be __APPLE__ */
+#define SF_BYTE_NATIVE SF_BYTE_BE
+#endif
+
+typedef struct _tag
+{ /* size (bytes) */
+ char version; /* 1 */
+ char format; /* 1 */
+ long count; /* 4 */
+ char channels; /* 1 */
+ long framesize; /* 4 */
+ char extension[5]; /* 5 */
+} t_tag; /*--------------*/
+ /* 16 */
+
+typedef struct _frame
+{
+ t_tag tag;
+ char *data;
+} t_frame;
+
+/* fin udpsend~.h */
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