1 /* 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 // SwitchingSampRate.cpp : Defines the entry point for the console 12 // application. 13 // 14 15 #include <iostream> 16 #include "isac.h" 17 #include "utility.h" 18 #include "signal_processing_library.h" 19 20 #define MAX_FILE_NAME 500 21 #define MAX_NUM_CLIENTS 2 22 23 24 #define NUM_CLIENTS 2 25 26 using namespace std; 27 28 int main(int argc, char* argv[]) 29 { 30 char fileNameWB[MAX_FILE_NAME]; 31 char fileNameSWB[MAX_FILE_NAME]; 32 33 char outFileName[MAX_NUM_CLIENTS][MAX_FILE_NAME]; 34 35 FILE* inFile[MAX_NUM_CLIENTS]; 36 FILE* outFile[MAX_NUM_CLIENTS]; 37 38 ISACStruct* codecInstance[MAX_NUM_CLIENTS]; 39 int32_t resamplerState[MAX_NUM_CLIENTS][8]; 40 41 int encoderSampRate[MAX_NUM_CLIENTS]; 42 43 int minBn = 16000; 44 int maxBn = 56000; 45 46 int bnWB = 32000; 47 int bnSWB = 56000; 48 49 strcpy(outFileName[0], "switchSampRate_out1.pcm"); 50 strcpy(outFileName[1], "switchSampRate_out2.pcm"); 51 52 short clientCntr; 53 54 unsigned int lenEncodedInBytes[MAX_NUM_CLIENTS]; 55 unsigned int lenAudioIn10ms[MAX_NUM_CLIENTS]; 56 unsigned int lenEncodedInBytesTmp[MAX_NUM_CLIENTS]; 57 unsigned int lenAudioIn10msTmp[MAX_NUM_CLIENTS]; 58 BottleNeckModel* packetData[MAX_NUM_CLIENTS]; 59 60 char versionNumber[100]; 61 short samplesIn10ms[MAX_NUM_CLIENTS]; 62 int bottleneck[MAX_NUM_CLIENTS]; 63 64 printf("\n\n"); 65 printf("____________________________________________\n\n"); 66 WebRtcIsac_version(versionNumber); 67 printf(" iSAC-swb version %s\n", versionNumber); 68 printf("____________________________________________\n"); 69 70 71 fileNameWB[0] = '\0'; 72 fileNameSWB[0] = '\0'; 73 74 char myFlag[20]; 75 strcpy(myFlag, "-wb"); 76 // READ THE WIDEBAND AND SUPER-WIDEBAND FILE NAMES 77 if(readParamString(argc, argv, myFlag, fileNameWB, MAX_FILE_NAME) <= 0) 78 { 79 printf("No wideband file is specified"); 80 } 81 82 strcpy(myFlag, "-swb"); 83 if(readParamString(argc, argv, myFlag, fileNameSWB, MAX_FILE_NAME) <= 0) 84 { 85 printf("No super-wideband file is specified"); 86 } 87 88 // THE FIRST CLIENT STARTS IN WIDEBAND 89 encoderSampRate[0] = 16000; 90 OPEN_FILE_RB(inFile[0], fileNameWB); 91 92 // THE SECOND CLIENT STARTS IN SUPER-WIDEBAND 93 encoderSampRate[1] = 32000; 94 OPEN_FILE_RB(inFile[1], fileNameSWB); 95 96 strcpy(myFlag, "-I"); 97 short codingMode = readSwitch(argc, argv, myFlag); 98 99 for(clientCntr = 0; clientCntr < NUM_CLIENTS; clientCntr++) 100 { 101 codecInstance[clientCntr] = NULL; 102 103 printf("\n"); 104 printf("Client %d\n", clientCntr + 1); 105 printf("---------\n"); 106 printf("Starting %s", 107 (encoderSampRate[clientCntr] == 16000) 108 ? "wideband":"super-wideband"); 109 110 // Open output File Name 111 OPEN_FILE_WB(outFile[clientCntr], outFileName[clientCntr]); 112 printf("Output File...................... %s\n", outFileName[clientCntr]); 113 114 samplesIn10ms[clientCntr] = encoderSampRate[clientCntr] * 10; 115 116 if(codingMode == 1) 117 { 118 bottleneck[clientCntr] = (clientCntr)? bnSWB:bnWB; 119 } 120 else 121 { 122 bottleneck[clientCntr] = (clientCntr)? minBn:maxBn; 123 } 124 125 printf("Bottleneck....................... %0.3f kbits/sec \n", 126 bottleneck[clientCntr] / 1000.0); 127 128 // coding-mode 129 printf("Encoding Mode.................... %s\n", 130 (codingMode == 1)? "Channel-Independent (Instantaneous)":"Adaptive"); 131 132 lenEncodedInBytes[clientCntr] = 0; 133 lenAudioIn10ms[clientCntr] = 0; 134 lenEncodedInBytesTmp[clientCntr] = 0; 135 lenAudioIn10msTmp[clientCntr] = 0; 136 137 packetData[clientCntr] = (BottleNeckModel*)new(BottleNeckModel); 138 if(packetData[clientCntr] == NULL) 139 { 140 printf("Could not allocate memory for packetData \n"); 141 return -1; 142 } 143 memset(packetData[clientCntr], 0, sizeof(BottleNeckModel)); 144 memset(resamplerState[clientCntr], 0, sizeof(int32_t) * 8); 145 } 146 147 for(clientCntr = 0; clientCntr < NUM_CLIENTS; clientCntr++) 148 { 149 // Create 150 if(WebRtcIsac_Create(&codecInstance[clientCntr])) 151 { 152 printf("Could not creat client %d\n", clientCntr + 1); 153 return -1; 154 } 155 156 WebRtcIsac_SetEncSampRate(codecInstance[clientCntr], encoderSampRate[clientCntr]); 157 158 WebRtcIsac_SetDecSampRate(codecInstance[clientCntr], 159 encoderSampRate[clientCntr + (1 - ((clientCntr & 1)<<1))]); 160 161 // Initialize Encoder 162 if(WebRtcIsac_EncoderInit(codecInstance[clientCntr], 163 codingMode) < 0) 164 { 165 printf("Could not initialize client, %d\n", clientCntr + 1); 166 return -1; 167 } 168 169 // Initialize Decoder 170 if(WebRtcIsac_DecoderInit(codecInstance[clientCntr]) < 0) 171 { 172 printf("Could not initialize decoder of client %d\n", 173 clientCntr + 1); 174 return -1; 175 } 176 177 // setup Rate if in Instantaneous mode 178 if(codingMode != 0) 179 { 180 // ONLY Clients who are not in Adaptive mode 181 if(WebRtcIsac_Control(codecInstance[clientCntr], 182 bottleneck[clientCntr], 30) < 0) 183 { 184 printf("Could not setup bottleneck and frame-size for client %d\n", 185 clientCntr + 1); 186 return -1; 187 } 188 } 189 } 190 191 192 short streamLen; 193 short numSamplesRead; 194 short lenDecodedAudio; 195 short senderIdx; 196 short receiverIdx; 197 198 printf("\n"); 199 short num10ms[MAX_NUM_CLIENTS]; 200 memset(num10ms, 0, sizeof(short)*MAX_NUM_CLIENTS); 201 FILE* arrivalTimeFile1 = fopen("arrivalTime1.dat", "wb"); 202 FILE* arrivalTimeFile2 = fopen("arrivalTime2.dat", "wb"); 203 short numPrint[MAX_NUM_CLIENTS]; 204 memset(numPrint, 0, sizeof(short) * MAX_NUM_CLIENTS); 205 206 // Audio Buffers 207 short silence10ms[10 * 32]; 208 memset(silence10ms, 0, 320 * sizeof(short)); 209 short audioBuff10ms[10 * 32]; 210 short audioBuff60ms[60 * 32]; 211 short resampledAudio60ms[60 * 32]; 212 213 unsigned short bitStream[600+600]; 214 short speechType[1]; 215 216 short numSampFreqChanged = 0; 217 while(numSampFreqChanged < 10) 218 { 219 for(clientCntr = 0; clientCntr < NUM_CLIENTS; clientCntr++) 220 { 221 // Encoding/decoding for this pair of clients, if there is 222 // audio for any of them 223 //if(audioLeft[clientCntr] || audioLeft[clientCntr + 1]) 224 //{ 225 //for(pairCntr = 0; pairCntr < 2; pairCntr++) 226 //{ 227 senderIdx = clientCntr; // + pairCntr; 228 receiverIdx = 1 - clientCntr;// + (1 - pairCntr); 229 230 //if(num10ms[senderIdx] > 6) 231 //{ 232 // printf("Too many frames read for client %d", 233 // senderIdx + 1); 234 // return -1; 235 //} 236 237 numSamplesRead = (short)fread(audioBuff10ms, sizeof(short), 238 samplesIn10ms[senderIdx], inFile[senderIdx]); 239 if(numSamplesRead != samplesIn10ms[senderIdx]) 240 { 241 // file finished switch encoder sampling frequency. 242 printf("Changing Encoder Sampling frequency in client %d to ", senderIdx+1); 243 fclose(inFile[senderIdx]); 244 numSampFreqChanged++; 245 if(encoderSampRate[senderIdx] == 16000) 246 { 247 printf("super-wideband.\n"); 248 OPEN_FILE_RB(inFile[senderIdx], fileNameSWB); 249 encoderSampRate[senderIdx] = 32000; 250 } 251 else 252 { 253 printf("wideband.\n"); 254 OPEN_FILE_RB(inFile[senderIdx], fileNameWB); 255 encoderSampRate[senderIdx] = 16000; 256 } 257 WebRtcIsac_SetEncSampRate(codecInstance[senderIdx], encoderSampRate[senderIdx]); 258 WebRtcIsac_SetDecSampRate(codecInstance[receiverIdx], encoderSampRate[senderIdx]); 259 260 samplesIn10ms[clientCntr] = encoderSampRate[clientCntr] * 10; 261 262 numSamplesRead = (short)fread(audioBuff10ms, sizeof(short), 263 samplesIn10ms[senderIdx], inFile[senderIdx]); 264 if(numSamplesRead != samplesIn10ms[senderIdx]) 265 { 266 printf(" File %s for client %d has not enough audio\n", 267 (encoderSampRate[senderIdx]==16000)? "wideband":"super-wideband", 268 senderIdx + 1); 269 return -1; 270 } 271 } 272 num10ms[senderIdx]++; 273 274 // sanity check 275 //if(num10ms[senderIdx] > 6) 276 //{ 277 // printf("Client %d has got more than 60 ms audio and encoded no packet.\n", 278 // senderIdx); 279 // return -1; 280 //} 281 282 // Encode 283 284 285 streamLen = WebRtcIsac_Encode(codecInstance[senderIdx], 286 audioBuff10ms, (short*)bitStream); 287 int16_t ggg; 288 if (streamLen > 0) { 289 if(( WebRtcIsac_ReadFrameLen(codecInstance[receiverIdx], 290 (short *) bitStream, &ggg))<0) 291 printf("ERROR\n"); 292 } 293 294 // Sanity check 295 if(streamLen < 0) 296 { 297 printf(" Encoder error in client %d \n", senderIdx + 1); 298 return -1; 299 } 300 301 302 if(streamLen > 0) 303 { 304 // Packet generated; model sending through a channel, do bandwidth 305 // estimation at the receiver and decode. 306 lenEncodedInBytes[senderIdx] += streamLen; 307 lenAudioIn10ms[senderIdx] += (unsigned int)num10ms[senderIdx]; 308 lenEncodedInBytesTmp[senderIdx] += streamLen; 309 lenAudioIn10msTmp[senderIdx] += (unsigned int)num10ms[senderIdx]; 310 311 // Print after ~5 sec. 312 if(lenAudioIn10msTmp[senderIdx] >= 100) 313 { 314 numPrint[senderIdx]++; 315 printf(" %d, %6.3f => %6.3f ", senderIdx+1, 316 bottleneck[senderIdx] / 1000.0, 317 lenEncodedInBytesTmp[senderIdx] * 0.8 / 318 lenAudioIn10msTmp[senderIdx]); 319 320 if(codingMode == 0) 321 { 322 int32_t bn; 323 WebRtcIsac_GetUplinkBw(codecInstance[senderIdx], &bn); 324 printf("[%d] ", bn); 325 } 326 //int16_t rateIndexLB; 327 //int16_t rateIndexUB; 328 //WebRtcIsac_GetDownLinkBwIndex(codecInstance[receiverIdx], 329 // &rateIndexLB, &rateIndexUB); 330 //printf(" (%2d, %2d) ", rateIndexLB, rateIndexUB); 331 332 cout << flush; 333 lenEncodedInBytesTmp[senderIdx] = 0; 334 lenAudioIn10msTmp[senderIdx] = 0; 335 //if(senderIdx == (NUM_CLIENTS - 1)) 336 //{ 337 printf(" %0.1f \n", lenAudioIn10ms[senderIdx] * 10. /1000); 338 //} 339 340 // After ~20 sec change the bottleneck. 341 // if((numPrint[senderIdx] == 4) && (codingMode == 0)) 342 // { 343 // numPrint[senderIdx] = 0; 344 // if(codingMode == 0) 345 // { 346 // int newBottleneck = bottleneck[senderIdx] + 347 // (bottleneckChange[senderIdx] * 1000); 348 349 // if(bottleneckChange[senderIdx] > 0) 350 // { 351 // if(newBottleneck >maxBn) 352 // { 353 // bottleneckChange[senderIdx] = -1; 354 // newBottleneck = bottleneck[senderIdx] + 355 // (bottleneckChange[senderIdx] * 1000); 356 // if(newBottleneck > minBn) 357 // { 358 // bottleneck[senderIdx] = newBottleneck; 359 // } 360 // } 361 // else 362 // { 363 // bottleneck[senderIdx] = newBottleneck; 364 // } 365 // } 366 // else 367 // { 368 // if(newBottleneck < minBn) 369 // { 370 // bottleneckChange[senderIdx] = 1; 371 // newBottleneck = bottleneck[senderIdx] + 372 // (bottleneckChange[senderIdx] * 1000); 373 // if(newBottleneck < maxBn) 374 // { 375 // bottleneck[senderIdx] = newBottleneck; 376 // } 377 // } 378 // else 379 // { 380 // bottleneck[senderIdx] = newBottleneck; 381 // } 382 // } 383 // } 384 // } 385 } 386 387 // model a channel of given bottleneck, to get the receive timestamp 388 get_arrival_time(num10ms[senderIdx] * samplesIn10ms[senderIdx], 389 streamLen, bottleneck[senderIdx], packetData[senderIdx], 390 encoderSampRate[senderIdx]*1000, encoderSampRate[senderIdx]*1000); 391 392 // Write the arrival time. 393 if(senderIdx == 0) 394 { 395 if (fwrite(&(packetData[senderIdx]->arrival_time), 396 sizeof(unsigned int), 397 1, arrivalTimeFile1) != 1) { 398 return -1; 399 } 400 } 401 else 402 { 403 if (fwrite(&(packetData[senderIdx]->arrival_time), 404 sizeof(unsigned int), 405 1, arrivalTimeFile2) != 1) { 406 return -1; 407 } 408 } 409 410 // BWE 411 if(WebRtcIsac_UpdateBwEstimate(codecInstance[receiverIdx], 412 bitStream, streamLen, packetData[senderIdx]->rtp_number, 413 packetData[senderIdx]->sample_count, 414 packetData[senderIdx]->arrival_time) < 0) 415 { 416 printf(" BWE Error at client %d \n", receiverIdx + 1); 417 return -1; 418 } 419 /**/ 420 // Decode 421 lenDecodedAudio = WebRtcIsac_Decode( 422 codecInstance[receiverIdx], bitStream, streamLen, 423 audioBuff60ms, speechType); 424 if(lenDecodedAudio < 0) 425 { 426 printf(" Decoder error in client %d \n", receiverIdx + 1); 427 return -1; 428 } 429 430 431 if(encoderSampRate[senderIdx] == 16000) 432 { 433 WebRtcSpl_UpsampleBy2(audioBuff60ms, lenDecodedAudio, resampledAudio60ms, 434 resamplerState[receiverIdx]); 435 if (fwrite(resampledAudio60ms, sizeof(short), lenDecodedAudio << 1, 436 outFile[receiverIdx]) != 437 static_cast<size_t>(lenDecodedAudio << 1)) { 438 return -1; 439 } 440 } 441 else 442 { 443 if (fwrite(audioBuff60ms, sizeof(short), lenDecodedAudio, 444 outFile[receiverIdx]) != 445 static_cast<size_t>(lenDecodedAudio)) { 446 return -1; 447 } 448 } 449 num10ms[senderIdx] = 0; 450 } 451 //} 452 //} 453 } 454 } 455 } 456
