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-rw-r--r--mp3streamout~.c617
1 files changed, 617 insertions, 0 deletions
diff --git a/mp3streamout~.c b/mp3streamout~.c
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+++ b/mp3streamout~.c
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+/* ------------------------ mp3streamout~ ------------------------------------- */
+/* */
+/* Tilde object to send mp3-stream to a peer using mp3streamin~. */
+/* Written by Yves Degoyon (ydegoyon@free.fr). */
+/* Tarballs and updates @ http://ydegoyon.free.fr */
+/* */
+/* 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. */
+/* Uses the LAME MPEG 1 Layer 3 encoding library (lame_enc.dll) which can */
+/* be found at http://www.cdex.n3.net. */
+/* */
+/* "I'd sell my soul to god" */
+/* "If it could take away the pain." */
+/* Theo Hakola -- */
+/* ---------------------------------------------------------------------------- */
+
+#include <sys/types.h>
+#include <string.h>
+#include <stdio.h>
+#include <errno.h>
+#include <stdlib.h>
+#include <fcntl.h>
+#include <sys/stat.h>
+#ifndef __APPLE__
+#include <malloc.h>
+#endif
+#include <ctype.h>
+#ifdef _WIN32
+#include <io.h>
+#include <windows.h>
+#include <winsock.h>
+#include <windef.h>
+#else
+#include <unistd.h>
+#include <sys/socket.h>
+#include <netinet/in.h>
+#include <netdb.h>
+#include <time.h>
+#include <sys/time.h>
+#define SOCKET_ERROR -1
+#endif
+
+#if defined(__APPLE__) || defined(_WIN32)
+#define MSG_NOSIGNAL 0
+#endif
+
+#include <lame/lame.h> /* lame encoder stuff */
+
+#include "m_pd.h" /* standard pd stuff */
+
+#include "mpg123.h" /* sub-library MPGLIB included in lame */
+/* useful debugging functions from mpglib */
+extern int decode_header( struct frame* fr, unsigned long newhead );
+extern void print_header_compact( struct frame* fr );
+extern int head_check( unsigned long head, int check_layer );
+
+#define MY_MP3_MALLOC_IN_SIZE 65536
+ /* max size taken from lame readme */
+#define MY_MP3_MALLOC_OUT_SIZE 1.25*MY_MP3_MALLOC_IN_SIZE+7200
+
+#define MAXDATARATE 320 /* maximum mp3 data rate is 320kbit/s */
+#define STRBUF_SIZE 32
+
+static char *mp3streamout_version = "mp3streamout~: mp3 peer-to-peer streamer version 0.3, written by ydegoyon@free.fr";
+
+static t_class *mp3streamout_class;
+
+typedef struct _mp3streamout
+{
+ t_object x_obj;
+
+ /* LAME stuff */
+ int x_lame; /* info about encoder status */
+ int x_lamechunk; /* chunk size for LAME encoder */
+ int x_mp3size; /* number of returned mp3 samples */
+
+ /* buffer stuff */
+ unsigned short x_inp; /* in position for buffer */
+ unsigned short x_outp; /* out position for buffer*/
+ short *x_mp3inbuf; /* data to be sent to LAME */
+ char *x_mp3outbuf; /* data returned by LAME -> our mp3 stream */
+ short *x_buffer; /* data to be buffered */
+ int x_bytesbuffered; /* number of unprocessed bytes in buffer */
+ int x_start;
+
+ /* mp3 format stuff */
+ int x_samplerate;
+ int x_bitrate; /* bitrate of mp3 stream */
+ int x_mp3mode; /* mode (mono, joint stereo, stereo, dual mono) */
+ int x_mp3quality; /* quality of encoding */
+
+ /* connection data */
+ int x_fd; /* info about connection status */
+ int x_outpackets; /* mp3 packets sent */
+
+ t_float x_f; /* float needed for signal input */
+
+ lame_global_flags* lgfp;
+} t_mp3streamout;
+
+
+ /* encode PCM data to mp3 stream */
+static void mp3streamout_encode(t_mp3streamout *x)
+{
+ unsigned short i, wp;
+ int err = -1;
+ int n = x->x_lamechunk;
+
+#ifdef UNIX
+ if(x->x_lamechunk < (int)sizeof(x->x_mp3inbuf))
+#else
+ if(x->x_lamechunk < sizeof(x->x_mp3inbuf))
+#endif
+ {
+ error("not enough memory!");
+ return;
+ }
+
+ /* on start/reconnect set outpoint that it not interferes with inpoint */
+ if(x->x_start == -1)
+ {
+ post("mp3streamout~: initializing buffers");
+ /* we try to keep 2.5 times the data the encoder needs in the buffer */
+ if(x->x_inp > (2 * x->x_lamechunk))
+ {
+ x->x_outp = (short) x->x_inp - (2.5 * x->x_lamechunk);
+ }
+ else if(x->x_inp < (2 * x->x_lamechunk))
+ {
+ x->x_outp = (short) MY_MP3_MALLOC_IN_SIZE - (2.5 * x->x_lamechunk);
+ }
+ x->x_start = 1;
+ }
+ if((unsigned short)(x->x_outp - x->x_inp) < x->x_lamechunk)error("mp3streamout~: buffers overlap!");
+
+ i = MY_MP3_MALLOC_IN_SIZE - x->x_outp;
+
+ /* read from buffer */
+ if(x->x_lamechunk <= i)
+ {
+ /* enough data until end of buffer */
+ for(n = 0; n < x->x_lamechunk; n++) /* fill encode buffer */
+ {
+ x->x_mp3inbuf[n] = x->x_buffer[n + x->x_outp];
+ }
+ x->x_outp += x->x_lamechunk;
+ }
+ else /* split data */
+ {
+ for(wp = 0; wp < i; wp++) /* data at end of buffer */
+ {
+ x->x_mp3inbuf[wp] = x->x_buffer[wp + x->x_outp];
+ }
+
+ for(wp = i; wp < x->x_lamechunk; wp++) /* write rest of data at beginning of buffer */
+ {
+ x->x_mp3inbuf[wp] = x->x_buffer[wp - i];
+ }
+ x->x_outp = x->x_lamechunk - i;
+ }
+
+ /* encode mp3 data */
+
+ x->x_mp3size = lame_encode_buffer_interleaved(x->lgfp, x->x_mp3inbuf,
+ x->x_lamechunk/lame_get_num_channels(x->lgfp),
+ x->x_mp3outbuf, MY_MP3_MALLOC_OUT_SIZE);
+ x->x_mp3size+=lame_encode_flush( x->lgfp, x->x_mp3outbuf+x->x_mp3size, MY_MP3_MALLOC_OUT_SIZE-x->x_mp3size );
+ // post( "mp3streamout~ : encoding returned %d frames", x->x_mp3size );
+
+ /* check result */
+ if(x->x_mp3size<0)
+ {
+ lame_close( x->lgfp );
+ error("mp3streamout~: lame_encode_buffer_interleaved failed (%d)", x->x_mp3size);
+ x->x_lame = -1;
+ }
+}
+
+ /* stream mp3 to the peer */
+static void mp3streamout_stream(t_mp3streamout *x)
+{
+ int count = -1, i;
+ struct frame hframe;
+
+ /* header needs to be included in each packet */
+
+ for( i=0; i<x->x_mp3size; i++ )
+ {
+ // track valid data
+ if ( head_check( *((unsigned long*)x->x_mp3outbuf+i), 3 ) )
+ {
+ // post( "valid header emitted @ %d (byte %d)", i, i*sizeof(unsigned long) );
+ }
+ }
+
+ count = send(x->x_fd, x->x_mp3outbuf, x->x_mp3size, MSG_NOSIGNAL);
+ if(count < 0)
+ {
+ error("mp3streamout~: could not send encoded data to the peer (%d)", count);
+ lame_close( x->lgfp );
+ x->x_lame = -1;
+#ifdef _WIN32
+ closesocket(x->x_fd);
+#else
+ close(x->x_fd);
+#endif
+ x->x_fd = -1;
+ outlet_float(x->x_obj.ob_outlet, 0);
+ }
+ else
+ {
+ x->x_outpackets++;
+ if ( x->x_outpackets%100 == 0 )
+ {
+ // post( "mp3streamout~ : emitted %d bytes (packets = %d)", count, x->x_outpackets );
+ }
+ }
+ if((count > 0)&&(count != x->x_mp3size))
+ {
+ error("mp3streamout~: %d bytes skipped", x->x_mp3size - count);
+ }
+}
+
+
+ /* buffer data as channel interleaved PCM */
+static t_int *mp3streamout_perform(t_int *w)
+{
+ t_float *in1 = (t_float *)(w[1]); /* left audio inlet */
+ t_float *in2 = (t_float *)(w[2]); /* right audio inlet */
+ t_mp3streamout *x = (t_mp3streamout *)(w[3]);
+ int n = (int)(w[4]); /* number of samples */
+ unsigned short i,wp;
+ float in;
+
+ /* copy the data into the buffer */
+ i = MY_MP3_MALLOC_IN_SIZE - x->x_inp; /* space left at the end of buffer */
+
+ n *= 2; /* two channels go into one buffer */
+
+ if( n <= i )
+ {
+ /* the place between inp and MY_MP3_MALLOC_IN_SIZE */
+ /* is big enough to hold the data */
+
+ for(wp = 0; wp < n; wp++)
+ {
+ if(wp%2)
+ {
+ in = *(in2++); /* right channel / inlet */
+ }
+ else
+ {
+ in = *(in1++); /* left channel / inlet */
+ }
+ if (in > 1.0) { in = 1.0; }
+ if (in < -1.0) { in = -1.0; }
+ x->x_buffer[wp + x->x_inp] = (short) (32767.0 * in);
+ }
+ x->x_inp += n; /* n more samples written to buffer */
+ }
+ else
+ {
+ /* the place between inp and MY_MP3_MALLOC_IN_SIZE is not */
+ /* big enough to hold the data */
+ /* writing will take place in two turns, one from */
+ /* x->x_inp -> MY_MP3_MALLOC_IN_SIZE, then from 0 on */
+
+ for(wp = 0; wp < i; wp++) /* fill up to end of buffer */
+ {
+ if(wp%2)
+ {
+ in = *(in2++);
+ }
+ else
+ {
+ in = *(in1++);
+ }
+ if (in > 1.0) { in = 1.0; }
+ if (in < -1.0) { in = -1.0; }
+ x->x_buffer[wp + x->x_inp] = (short) (32767.0 * in);
+ }
+ for(wp = i; wp < n; wp++) /* write rest at start of buffer */
+ {
+ if(wp%2)
+ {
+ in = *(in2++);
+ }
+ else
+ {
+ in = *(in1++);
+ }
+ if (in > 1.0) { in = 1.0; }
+ if (in < -1.0) { in = -1.0; }
+ x->x_buffer[wp - i] = (short) (32767.0 * in);
+ }
+ x->x_inp = n - i; /* new writeposition in buffer */
+ }
+
+ if((x->x_fd >= 0)&&(x->x_lame >= 0))
+ {
+ /* count buffered samples when things are running */
+ x->x_bytesbuffered += n;
+
+ /* encode and send to the peer */
+ if(x->x_bytesbuffered > x->x_lamechunk)
+ {
+ mp3streamout_encode(x); /* encode to mp3 */
+ mp3streamout_stream(x); /* stream mp3 to the peer */
+ x->x_bytesbuffered -= x->x_lamechunk;
+ }
+ }
+ else
+ {
+ x->x_start = -1;
+ }
+ return (w+5);
+}
+
+static void mp3streamout_dsp(t_mp3streamout *x, t_signal **sp)
+{
+ dsp_add(mp3streamout_perform, 4, sp[0]->s_vec, sp[1]->s_vec, x, sp[0]->s_n);
+}
+
+ /* initialize the lame library */
+static void mp3streamout_tilde_lame_init(t_mp3streamout *x)
+{
+ int ret;
+ x->lgfp = lame_init(); /* set default parameters for now */
+
+#ifdef _WIN32
+ /* load lame_enc.dll library */
+ HINSTANCE dll;
+ dll=LoadLibrary("lame_enc.dll");
+ if(dll==NULL)
+ {
+ error("mp3streamout~: error loading lame_enc.dll");
+ closesocket(x->x_fd);
+ x->x_fd = -1;
+ outlet_float(x->x_obj.ob_outlet, 0);
+ post("mp3streamout~: connection closed");
+ return;
+ }
+#endif
+ {
+ const char *lameVersion = get_lame_version();
+ post( "mp3streamout~ : using lame version : %s", lameVersion );
+ }
+
+ /* setting lame parameters */
+ lame_set_num_channels( x->lgfp, 2);
+ lame_set_in_samplerate( x->lgfp, sys_getsr() );
+ lame_set_out_samplerate( x->lgfp, x->x_samplerate );
+ lame_set_brate( x->lgfp, x->x_bitrate );
+ lame_set_mode( x->lgfp, x->x_mp3mode );
+ lame_set_quality( x->lgfp, x->x_mp3quality );
+ lame_set_emphasis( x->lgfp, 1 );
+ lame_set_original( x->lgfp, 1 );
+ lame_set_copyright( x->lgfp, 1 ); /* viva free music societies !!! */
+ lame_set_disable_reservoir( x->lgfp, 0 );
+ lame_set_padding_type( x->lgfp, PAD_NO );
+ ret = lame_init_params( x->lgfp );
+ if ( ret<0 ) {
+ post( "mp3streamout~ : error : lame params initialization returned : %d", ret );
+ } else {
+ x->x_lame=1;
+ /* magic formula copied from windows dll for MPEG-I */
+ x->x_lamechunk = 2*1152;
+
+ post( "mp3streamout~ : lame initialization done. (%d)", x->x_lame );
+ }
+ lame_init_bitstream( x->lgfp );
+}
+
+ /* connect to the peer */
+static void mp3streamout_connect(t_mp3streamout *x, t_symbol *hostname, t_floatarg fportno)
+{
+ struct sockaddr_in csocket;
+ struct hostent *hp;
+ int portno = fportno; /* get port from message box */
+
+ /* variables used for communication with the peer */
+ const char *buf = 0;
+ char resp[STRBUF_SIZE];
+ unsigned int len;
+ int sockfd;
+
+#ifdef _WIN32
+ unsigned int ret;
+#else
+ int ret;
+#endif
+
+ if (x->x_fd >= 0)
+ {
+ error("mp3streamout~: already connected");
+ return;
+ }
+
+ sockfd = socket(AF_INET, SOCK_STREAM, IPPROTO_TCP);
+ if (sockfd < 0)
+ {
+ error("mp3streamout~: internal error while attempting to open socket");
+ return;
+ }
+
+ /* connect socket using hostname provided in command line */
+ csocket.sin_family = AF_INET;
+ hp = gethostbyname(hostname->s_name);
+ if (hp == 0)
+ {
+ post("mp3streamout~: bad host?");
+#ifdef _WIN32
+ closesocket(sockfd);
+#else
+ close(sockfd);
+#endif
+ return;
+ }
+ memcpy((char *)&csocket.sin_addr, (char *)hp->h_addr, hp->h_length);
+
+ /* assign client port number */
+ csocket.sin_port = htons((unsigned short)portno);
+
+ /* try to connect. */
+ post("mp3streamout~: connecting to port %d", portno);
+ if (connect(sockfd, (struct sockaddr *) &csocket, sizeof (csocket)) < 0)
+ {
+ error("mp3streamout~: connection failed!\n");
+#ifdef _WIN32
+ closesocket(sockfd);
+#else
+ close(sockfd);
+#endif
+ return;
+ }
+
+ x->x_fd = sockfd;
+ x->x_outpackets = 0;
+ outlet_float( x->x_obj.ob_outlet, 1 );
+ post( "mp3streamout~ : connected to peer" );
+
+ mp3streamout_tilde_lame_init(x);
+
+}
+
+ /* close connection to the peer */
+static void mp3streamout_disconnect(t_mp3streamout *x)
+{
+
+ int err = -1;
+ if(x->x_lame >= 0)
+ {
+ /* ignore remaining bytes */
+ if ( x->x_mp3size = lame_encode_flush( x->lgfp, x->x_mp3outbuf, 0) < 0 ) {
+ post( "mp3streamout~ : warning : remaining encoded bytes" );
+ }
+ lame_close( x->lgfp );
+ x->x_lame = -1;
+ post("mp3streamout~: encoder stream closed");
+ }
+
+ if(x->x_fd >= 0) /* close socket */
+ {
+#ifdef _WIN32
+ closesocket(x->x_fd);
+#else
+ close(x->x_fd);
+#endif
+ x->x_fd = -1;
+ outlet_float( x->x_obj.ob_outlet, 0 );
+ post("mp3streamout~: connection closed");
+ }
+}
+
+ /* settings for mp3 encoding */
+static void mp3streamout_mpeg(t_mp3streamout *x, t_floatarg fbitrate,
+ t_floatarg fmode, t_floatarg fquality)
+{
+ if ( fbitrate != 32 && fbitrate != 40 && fbitrate != 48 && fbitrate != 56 &&
+ fbitrate != 64 && fbitrate != 80 && fbitrate != 96 && fbitrate != 112 &&
+ fbitrate != 128 && fbitrate != 160 && fbitrate != 192 && fbitrate != 224 &&
+ fbitrate != 256 && fbitrate != 320 ) {
+ post( "mp3streamout~ : wrong bitrate." );
+ return;
+ }
+ if ( fmode <0 || fmode>2 ) {
+ post( "mp3streamout~ : wrong mp3 mode." );
+ if ( fmode == 3 )
+ {
+ post( "mp3streamout~ : mone is not supported by streamout~ for now." );
+ }
+ return;
+ }
+ /* there is a bug in lame 3.92 and quality below 5 will not work */
+ /* WAIT FOR A FIX */
+ if ( fquality <5 || fquality>9 ) {
+ post( "mp3streamout~ : wrong quality." );
+ return;
+ }
+ x->x_bitrate = fbitrate;
+ x->x_mp3mode = fmode;
+ x->x_mp3quality = (int)fquality;
+ post("mp3streamout~: setting mp3 stream to %dHz, %dkbit/s, mode %d, quality %d",
+ x->x_samplerate, x->x_bitrate, x->x_mp3mode, x->x_mp3quality);
+ mp3streamout_tilde_lame_init(x);
+}
+
+ /* print settings */
+static void mp3streamout_print(t_mp3streamout *x)
+{
+ const char * buf = 0;
+
+ post(mp3streamout_version);
+ post(" LAME mp3 settings:\n"
+ " output sample rate: %d Hz\n"
+ " bitrate: %d kbit/s", x->x_samplerate, x->x_bitrate);
+ switch(x->x_mp3mode)
+ {
+ case 0 :
+ buf = "stereo";
+ break;
+ case 1 :
+ buf = "joint stereo";
+ break;
+ case 2 :
+ buf = "dual channel";
+ break;
+ case 3 :
+ buf = "mono";
+ break;
+ }
+ post(" mode: %s\n"
+ " quality: %d", buf, x->x_mp3quality);
+#ifdef _WIN32
+ if(x->x_lamechunk!=0)post(" calculated mp3 chunk size: %d", x->x_lamechunk);
+#else
+ post(" mp3 chunk size: %d", x->x_lamechunk);
+#endif
+ if(x->x_samplerate!=sys_getsr())
+ {
+ post(" resampling from %d to %d Hz!", (int)sys_getsr(), x->x_samplerate);
+ }
+}
+
+ /* clean up */
+static void mp3streamout_free(t_mp3streamout *x)
+{
+
+ if(x->x_lame >= 0)
+ lame_close( x->lgfp );
+
+ if(x->x_fd >= 0)
+#ifdef _WIN32
+ closesocket(x->x_fd);
+#else
+ close(x->x_fd);
+#endif
+ freebytes(x->x_mp3inbuf, MY_MP3_MALLOC_IN_SIZE*sizeof(short));
+ freebytes(x->x_mp3outbuf, MY_MP3_MALLOC_OUT_SIZE);
+ freebytes(x->x_buffer, MY_MP3_MALLOC_IN_SIZE*sizeof(short));
+}
+
+static void *mp3streamout_new(void)
+{
+ t_mp3streamout *x = (t_mp3streamout *)pd_new(mp3streamout_class);
+ inlet_new (&x->x_obj, &x->x_obj.ob_pd, gensym ("signal"), gensym ("signal"));
+ outlet_new( &x->x_obj, &s_float );
+ x->lgfp = NULL;
+ x->x_fd = -1;
+ x->x_outpackets = 0;
+ x->x_lame = -1;
+ x->x_samplerate = sys_getsr();
+ x->x_bitrate = 128;
+ x->x_mp3mode = 1;
+ x->x_mp3quality = 5;
+ x->x_mp3inbuf = getbytes(MY_MP3_MALLOC_IN_SIZE*sizeof(short)); /* buffer for encoder input */
+ x->x_mp3outbuf = getbytes(MY_MP3_MALLOC_OUT_SIZE); /* our mp3 stream */
+ x->x_buffer = getbytes(MY_MP3_MALLOC_IN_SIZE*sizeof(short)); /* what we get from pd, converted to PCM */
+ if ((!x->x_buffer)||(!x->x_mp3inbuf)||(!x->x_mp3outbuf)) /* check buffers... */
+ {
+ error("out of memory!");
+ }
+ x->x_bytesbuffered = 0;
+ x->x_inp = 0;
+ x->x_outp = 0;
+ x->x_start = -1;
+ return(x);
+}
+
+void mp3streamout_tilde_setup(void)
+{
+ post(mp3streamout_version);
+ mp3streamout_class = class_new(gensym("mp3streamout~"), (t_newmethod)mp3streamout_new, (t_method)mp3streamout_free,
+ sizeof(t_mp3streamout), 0, 0);
+ CLASS_MAINSIGNALIN(mp3streamout_class, t_mp3streamout, x_f );
+ class_addmethod(mp3streamout_class, (t_method)mp3streamout_dsp, gensym("dsp"), 0);
+ class_addmethod(mp3streamout_class, (t_method)mp3streamout_connect, gensym("connect"), A_SYMBOL, A_FLOAT, 0);
+ class_addmethod(mp3streamout_class, (t_method)mp3streamout_disconnect, gensym("disconnect"), 0);
+ class_addmethod(mp3streamout_class, (t_method)mp3streamout_mpeg, gensym("mpeg"), A_FLOAT, A_FLOAT, A_FLOAT, 0);
+ class_addmethod(mp3streamout_class, (t_method)mp3streamout_print, gensym("print"), 0);
+}
+