probleme d'utilisation avec les lib speex et portaudio

Bonjour,

dans le but de creer un Voip, donc dans un premier il va y avoir la creation d'un

client/serveur, puis via portaudio(http://www.portaudio.com/) je peux communiquer

par la voix, et enfin grace a speex, j'encode la voix enregistrée par le micro

pour qu'il pèse moins lourd puis je l'envoie de client a un client2, et a la

reception le client2 decode les données encodées puis le lire.
Le problème:
je suis en train de tester un encodeur audio appelé speex(http://www.speex.org/),

mais le probleme est, que quand j'encode l'audio puis je decode l'audio, je ne

retrouve pas les memes données dans le fichier!
Et puis pour enregistrer j'utilise portaudio.


Je post ci-joint les morceaux de code necessaires:
*********************************************************************
Pour portaudio(il faut telecharger la librairie dynamique portaudio et le deposer

dans le meme repertoire que le main.cpp, et faire de meme pour le fichier

portaudio.h):

Code:

---

1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201 202 203 204 205 206 207 208 209 210 211 212 213 214 215 216 217 218 219 220 221 222 223 224 225 226 227 228 229 230 231 232 233 234 235 236 237 238 239 240 241 242 243 244 245 246 247 248 249 250 251 252 253 254 255 256 257 258 259 260 261 262 263 264 265 266 267 268 269 270 271 272 273 274 275 276 277 278 279 280 281 282 283 284 285 286 287 288 289 290 291 292 293 294 295 296 297 298 299 300 301 302 303 304 305 306 307 308 309 310 311 312 313 314 315 316 317 318 319 320 321 322 323 324 325 326 327 328 329 330 331 332 333 334 335 336 337 338 339 340 341 342 343 344 345 346 347 348 349 350 351 352 353 354 355 356 357 358 359 360 361 362 363 364 365 366 367 368 369 370 371 372 373 374 375 376 377 378 379 380 381 382 383 384 385 386 387 388 389 390 391 392 393 394 395 396 397 398 399 400 401 402 403 404 405 406 407 408 409 410 411 412 413 414 415 416 417 418 419 420 421 422 423 424 425 426 427 428 429 430 431 432 433 434 435 436 437 438 439 440 441 442 443 444 445 446 447 448 449 450 451 452 453 454 455 456 457 458 459 460 461 462 463 464 465 466 467 468 469 470 471 472 473 474 475 476 477 478 479 480 481 482 483 484 485 486 487 488 489 490 491 492 493 494 495 496 497 498 499 500 501 502 503 504 505 506 507 508 509 510 511 512 513 514 515 516 517 518 519 520 521 522 523 524 525

/*  * $Id$  * patest_record.c  * Record input into an array.  * Save array to a file.  * Playback recorded data.  *  * Author: Phil Burk <a href="http://www.softsynth.com" target="_blank">http://www.softsynth.com</a>  *  * This program uses the PortAudio Portable Audio Library.  * For more information see: <a href="http://www.portaudio.com" target="_blank">http://www.portaudio.com</a>  * Copyright (c) 1999-2000 Ross Bencina and Phil Burk  *  * Permission is hereby granted, free of charge, to any person obtaining  * a copy of this software and associated documentation files  * (the "Software"), to deal in the Software without restriction,  * including without limitation the rights to use, copy, modify, merge,  * publish, distribute, sublicense, and/or sell copies of the Software,  * and to permit persons to whom the Software is furnished to do so,  * subject to the following conditions:  *  * The above copyright notice and this permission notice shall be  * included in all copies or substantial portions of the Software.  *  * Any person wishing to distribute modifications to the Software is  * requested to send the modifications to the original developer so that  * they can be incorporated into the canonical version.  *  * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,  * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF  * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.  * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR  * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF  * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION  * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.  *  */ #include <stdio.h> #include <stdlib.h> #include "portaudio.h"   /* #define SAMPLE_RATE (17932) /* Test failure to open with this value. */ #define SAMPLE_RATE 44100 #define NUM_SECONDS 5 #define NUM_CHANNELS 2 /* #define DITHER_FLAG (paDitherOff) /**/ #define DITHER_FLAG (0) /**/   /* Select sample format. */ #define PA_SAMPLE_TYPE paFloat32 typedef float SAMPLE; #define SAMPLE_SILENCE (0.0f)   typedef struct { int frameIndex; /* Index into sample array. */ int maxFrameIndex; SAMPLE *recordedSamples; } paTestData; /* This routine will be called by the PortAudio engine when audio is needed. ** It may be called at interrupt level on some machines so don't do anything ** that could mess up the system like calling malloc() or free(). */   static int recordCallback( const void *inputBuffer, void *outputBuffer, unsigned long framesPerBuffer, const PaStreamCallbackTimeInfo* timeInfo, PaStreamCallbackFlags statusFlags, void *userData) { paTestData *data = (paTestData*)userData; SAMPLE *rptr = (SAMPLE*)inputBuffer; SAMPLE *wptr = &data->recordedSamples[data->frameIndex * NUM_CHANNELS]; long framesToCalc; long i; int finished; unsigned long framesLeft = data->maxFrameIndex - data->frameIndex;   (void) outputBuffer; /* Prevent unused variable warnings. */ (void) timeInfo;   if( framesLeft < framesPerBuffer ) { framesToCalc = framesLeft; finished = 1; } else { framesToCalc = framesPerBuffer; finished = 0; } if( inputBuffer == NULL ) { for( i=0; i<framesToCalc; i++ ) { *wptr++ = SAMPLE_SILENCE; /* left */ if( NUM_CHANNELS == 2 ) *wptr++ = SAMPLE_SILENCE; /* right */ } } else { for( i=0; i<framesToCalc; i++ ) { *wptr++ = *rptr++; /* left */ if( NUM_CHANNELS == 2 ) *wptr++ = *rptr++; /* right */ } } data->frameIndex += framesToCalc; return finished; }   static int liveCallback( const void *inputBuffer, void *outputBuffer, unsigned long framesPerBuffer, const PaStreamCallbackTimeInfo* timeInfo, PaStreamCallbackFlags statusFlags, void *userData) { SAMPLE *rptr = (SAMPLE*)inputBuffer; SAMPLE *wptr = (SAMPLE*)outputBuffer;   (void) outputBuffer; /* Prevent unused variable warnings. */ (void) timeInfo;   for( int i=0; i<framesPerBuffer; i++ ) { *wptr++ = *rptr++; /* left */ if( NUM_CHANNELS == 2 ) *wptr++ = *rptr++; /* right */ }   return 0; }   /* This routine will be called by the PortAudio engine when audio is needed. ** It may be called at interrupt level on some machines so don't do anything ** that could mess up the system like calling malloc() or free(). */ static int playCallback( const void *inputBuffer, void *outputBuffer, unsigned long framesPerBuffer, const PaStreamCallbackTimeInfo* timeInfo, PaStreamCallbackFlags statusFlags, void *userData ) { paTestData *data = (paTestData*)userData; SAMPLE *rptr = &data->recordedSamples[data->frameIndex * NUM_CHANNELS]; SAMPLE *wptr = (SAMPLE*)outputBuffer; unsigned int i; int finished; unsigned int framesLeft = data->maxFrameIndex - data->frameIndex; (void) inputBuffer; /* Prevent unused variable warnings. */ (void) timeInfo;   if( framesLeft < framesPerBuffer ) { /* final buffer... */ for( i=0; i<framesLeft; i++ ) { *wptr++ = *rptr++; /* left */ if( NUM_CHANNELS == 2 ) *wptr++ = *rptr++; /* right */ } for( ; i<framesPerBuffer; i++ ) { *wptr++ = 0; /* left */ if( NUM_CHANNELS == 2 ) *wptr++ = 0; /* right */ } data->frameIndex += framesLeft; finished = 1; } else { for( i=0; i<framesPerBuffer; i++ ) { *wptr++ = *rptr++; /* left */ if( NUM_CHANNELS == 2 ) *wptr++ = *rptr++; /* right */ } data->frameIndex += framesPerBuffer; finished = 0; } return finished; }   /*******************************************************************/ int main(void) { PaStream *stream; PaError err; paTestData data; int i; int totalFrames; int numSamples; int numBytes;   data.maxFrameIndex = totalFrames = NUM_SECONDS * SAMPLE_RATE; /* Record for a   few seconds. */ data.frameIndex = 0; numSamples = totalFrames * NUM_CHANNELS;   numBytes = numSamples * sizeof(SAMPLE); data.recordedSamples = (SAMPLE *) malloc( numBytes ); if( data.recordedSamples == NULL ) { printf("Could not allocate record array.\n"); exit(1); } for( i=0; i<numSamples; i++ ) data.recordedSamples[i] = 0;   err = Pa_Initialize(); if( err != paNoError ) goto error;   /* Record some audio. -------------------------------------------- */ err = Pa_OpenDefaultStream( &stream, 2, /* no input channels */ 2, /* stereo output */ PA_SAMPLE_TYPE, /* 32 bit floating point output   */ SAMPLE_RATE, paFramesPerBufferUnspecified, /* frames   per buffer, i.e. the number   of sample frames that   PortAudio will   request from the callback.   Many apps   may want to use   paFramesPerBufferUnspecified,   which   tells PortAudio to pick the   best,   possibly changing, buffer   size.*/     liveCallback, /* this is your callback function */ 0 ); /*This is a pointer that will be passed to   //&data your callback*/ if( err != paNoError ) goto error;   err = Pa_StartStream( stream ); if( err != paNoError ) goto error; printf("Now recording!!\n"); fflush(stdout);   while (1); //Pa_Sleep(1000 * NUM_SECONDS);   err = Pa_CloseStream( stream ); if( err != paNoError ) goto error;   /* Write recorded data to a file. */ #if 0 { FILE *fid; fid = fopen("recorded.raw", "wb"); if( fid == NULL ) { printf("Could not open file."); } else { fwrite( data.recordedSamples, NUM_CHANNELS * sizeof(SAMPLE),   totalFrames, fid ); fclose( fid ); printf("Wrote data to 'recorded.raw'\n"); } } #endif   /* Playback recorded data. -------------------------------------------- */ data.frameIndex = 0; printf("Begin playback.\n"); fflush(stdout); err =Pa_OpenDefaultStream( &stream, 0, /* no input channels */ 2, /* stereo output */ PA_SAMPLE_TYPE, /* 32 bit floating point output   */ SAMPLE_RATE, paFramesPerBufferUnspecified, /* frames   per buffer, i.e. the number   of sample frames that   PortAudio will   request from the callback.   Many apps   may want to use   paFramesPerBufferUnspecified,   which   tells PortAudio to pick the   best,   possibly changing, buffer   size.*/     playCallback, /* this is your callback function */ &data ); /*This is a pointer that will be passed   to   your callback*/ if( err != paNoError ) goto error;   if( stream ) { err = Pa_StartStream( stream ); if( err != paNoError ) goto error; printf("Waiting for playback to finish.\n"); fflush(stdout);   Pa_Sleep(1000 * NUM_SECONDS);   err = Pa_CloseStream( stream ); if( err != paNoError ) goto error; printf("Done.\n"); fflush(stdout); } free( data.recordedSamples );   Pa_Terminate(); return 0;   error: Pa_Terminate(); fprintf( stderr, "An error occured while using the portaudio stream\n" ); fprintf( stderr, "Error number: %d\n", err ); fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) ); return -1; } ********************************************************************** Pour l'encodage(fichier sampleenc.c qui se trouve dans la doc de speex.org),   faites en ligne de commande: "gcc -lspeex -o encode sampleenc.c":       #include <speex/speex.h> #include <stdio.h>   /*The frame size in hardcoded for this sample code but it doesn't have to be*/ #define FRAME_SIZE 160   int main(int argc, char **argv) { char *inFile; FILE *fin; FILE *fin2; short in[FRAME_SIZE]; float input[FRAME_SIZE]; printf("FRAME_SIZE: %i\n", FRAME_SIZE); char cbits[4096]; int nbBytes; /*Holds the state of the encoder*/ void *state; /*Holds bits so they can be read and written to by the Speex routines*/ SpeexBits bits; int i, tmp;   if (argc != 2) { printf("Warning: 2 arguments needed!\n"); return 0; }   /*Create a new encoder state in narrowband mode*/ state = speex_encoder_init(&speex_nb_mode);   /*Set the quality to 8 (15 kbps)*/ tmp=8; speex_encoder_ctl(state, SPEEX_SET_QUALITY, &tmp);   inFile = argv[1]; //fin = fopen(inFile, "r"); if ((fin = fopen(inFile, "r")) == NULL) { printf("Warning: Cannot open file!\n"); return 0; }   //fin2 = fopen("test.flux", "w"); if ((fin2 = fopen("test.flux", "w")) == NULL) { printf("Warning: Cannot open file!\n"); return 0; }   /*Initialization of the structure that holds the bits*/ speex_bits_init(&bits); while (1) { /*Read a 16 bits/sample audio frame*/ fread(in, sizeof(short), FRAME_SIZE, fin); if (feof(fin)) break; /*Copy the 16 bits values to float so Speex can work on them*/ for (i=0;i<FRAME_SIZE;i++) input[i]=in[i];   /*Flush all the bits in the struct so we can encode a new frame*/ speex_bits_reset(&bits);   /*Encode the frame*/ speex_encode(state, input, &bits); /*Copy the bits to an array of char that can be written*/ nbBytes = speex_bits_write(&bits, cbits, 200);   /*Write the size of the frame first. This is what sampledec expects but it's likely to be different in your own application*/ fwrite(&nbBytes, sizeof(int), 1, fin2); /*Write the compressed data*/ fwrite(cbits, 1, nbBytes, fin2);   }   /*Destroy the encoder state*/ speex_encoder_destroy(state); /*Destroy the bit-packing struct*/ speex_bits_destroy(&bits); fclose(fin); return 0; }     ********************************************************************************* Pour le decodage(fichier sampledec.c qui se trouve dans la doc de speex.org),   faites en ligne de commande: "gcc -lspeex -o decode sampledec.c":       #include <speex/speex.h> #include <stdio.h> /*The frame size in hardcoded for this sample code but it doesn't have to be*/ #define FRAME_SIZE 160   int main(int argc, char **argv) { char *outFile; FILE *fout; FILE *fout2; /*Holds the audio that will be written to file (16 bits per sample)*/ short out[FRAME_SIZE]; /*Speex handle samples as float, so we need an array of floats*/ float output[FRAME_SIZE]; char cbits[4096]; int nbBytes; /*Holds the state of the decoder*/ void *state; /*Holds bits so they can be read and written to by the Speex routines*/ SpeexBits bits; int i, tmp;   if (argc != 2) { printf("Warning: 2 arguments needed!\n"); return 0; } /*Create a new decoder state in narrowband mode*/ state = speex_decoder_init(&speex_nb_mode);   /*Set the perceptual enhancement on*/ tmp=8; speex_decoder_ctl(state, SPEEX_SET_ENH, &tmp);   outFile = argv[1]; if ((fout = fopen(outFile, "r+")) == NULL) { printf("Warning: Cannot open file!\n"); return 0; } if ((fout2 = fopen("newFile.mp3", "w+")) == NULL) { printf("Warning: Cannot open file!\n"); return 0; }   /*Initialization of the structure that holds the bits*/ speex_bits_init(&bits); //printf("FRAME SIZE: %i\n", FRAME_SIZE); while (1) { /*Read the size encoded by sampleenc, this part will likely be   different in your application*/ fread(&nbBytes, sizeof(int), 1, fout); fprintf (stderr, "nbBytes: %d\n", nbBytes);   if (feof(fout)) break;   /*Read the "packet" encoded by sampleenc*/ fread(cbits, 1, nbBytes, fout);   /*Copy the data into the bit-stream struct*/ speex_bits_read_from(&bits, cbits, nbBytes);   /*Decode the data*/ speex_decode(state, &bits, output);   /*Copy from float to short (16 bits) for output*/ for (i=0;i<FRAME_SIZE;i++) out[i]=output[i];   /*Write the decoded audio to file*/ fwrite(out, sizeof(short), FRAME_SIZE, fout2);   }   /*Destroy the decoder state*/ speex_decoder_destroy(state); /*Destroy the bit-stream truct*/ speex_bits_destroy(&bits); fclose(fout); fclose(fout2); return 0; }

---

****************************************************
****************************************************

Les questions sont:

Quand j'encode puis je decode un fichier mp3, le fichier de donnees est illisible

avec vlc. Alors pourquoi?

Le samplecode de portaudio marche normalement, le test marche chez moi en tout

cas. Du coup je me pose la question qui est: comment je vais passer les donnees

,stockees par portaudio dans un void*, a speex pour qu' il m'encode et qu'il me

decode ces donnes?

Voila mes questions actuellement.
Si quelqu'un a deja travaille sur ces 2 technologies ou quelqu'un qui a capte le

truc, j'aimerai que cette personne m'aide et me donner la demarche dans la

realisation de ce Voip.

Bien cordialement.

Je tenterai de lire ton code tranquillement demain en matinée, mais ptite suggestion en passant en coup de vent : quitte à passer par portaudio et speex, pourquoi pas passer par la librairie iaxclient (portaudio + libiax + speex + gsm + ulaw/alaw) disponible LGPL (t'a donc les sources), et qui est avec le support video aussi ... c'est vrai que si tu cherches à coder un client SIP ma suggestion te sera inutile, mais si au final c asterisk que tu as en ipbx, ptre que tu peux passer au protocole IAX, et la librairie iaxclient te permettera de plancher sur les fonctionnalités de ton softphone que sur le reste. Suggestion comme une autre ...