1 /* 2 * Copyright (c) 2011 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 /* 12 * encode.c 13 * 14 * Encoding function for the iSAC coder. 15 * 16 */ 17 18 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h" 19 20 #include <assert.h> 21 #include <stdio.h> 22 23 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routins.h" 24 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h" 25 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h" 26 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h" 27 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_tables.h" 28 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h" 29 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.h" 30 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.h" 31 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/structs.h" 32 33 34 int WebRtcIsacfix_EncodeImpl(int16_t *in, 35 IsacFixEncoderInstance *ISACenc_obj, 36 BwEstimatorstr *bw_estimatordata, 37 int16_t CodingMode) 38 { 39 int16_t stream_length = 0; 40 int16_t usefulstr_len = 0; 41 int k; 42 int16_t BWno; 43 44 int16_t lofilt_coefQ15[(ORDERLO)*SUBFRAMES]; 45 int16_t hifilt_coefQ15[(ORDERHI)*SUBFRAMES]; 46 int32_t gain_lo_hiQ17[2*SUBFRAMES]; 47 48 int16_t LPandHP[FRAMESAMPLES/2 + QLOOKAHEAD]; 49 int16_t LP16a[FRAMESAMPLES/2 + QLOOKAHEAD]; 50 int16_t HP16a[FRAMESAMPLES/2 + QLOOKAHEAD]; 51 52 int16_t PitchLags_Q7[PITCH_SUBFRAMES]; 53 int16_t PitchGains_Q12[PITCH_SUBFRAMES]; 54 int16_t AvgPitchGain_Q12; 55 56 int16_t frame_mode; /* 0 for 30ms, 1 for 60ms */ 57 int16_t processed_samples; 58 int status; 59 60 int32_t bits_gainsQ11; 61 int16_t MinBytes; 62 int16_t bmodel; 63 64 transcode_obj transcodingParam; 65 int16_t payloadLimitBytes; 66 int16_t arithLenBeforeEncodingDFT; 67 int16_t iterCntr; 68 69 /* copy new frame length and bottle neck rate only for the first 10 ms data */ 70 if (ISACenc_obj->buffer_index == 0) { 71 /* set the framelength for the next packet */ 72 ISACenc_obj->current_framesamples = ISACenc_obj->new_framelength; 73 } 74 75 frame_mode = ISACenc_obj->current_framesamples/MAX_FRAMESAMPLES; /* 0 (30 ms) or 1 (60 ms) */ 76 processed_samples = ISACenc_obj->current_framesamples/(frame_mode+1); /* 480 (30, 60 ms) */ 77 78 /* buffer speech samples (by 10ms packet) until the framelength is reached (30 or 60 ms) */ 79 /**************************************************************************************/ 80 /* fill the buffer with 10ms input data */ 81 for(k=0; k<FRAMESAMPLES_10ms; k++) { 82 ISACenc_obj->data_buffer_fix[k + ISACenc_obj->buffer_index] = in[k]; 83 } 84 /* if buffersize is not equal to current framesize, and end of file is not reached yet, */ 85 /* increase index and go back to main to get more speech samples */ 86 if (ISACenc_obj->buffer_index + FRAMESAMPLES_10ms != processed_samples) { 87 ISACenc_obj->buffer_index = ISACenc_obj->buffer_index + FRAMESAMPLES_10ms; 88 return 0; 89 } 90 /* if buffer reached the right size, reset index and continue with encoding the frame */ 91 ISACenc_obj->buffer_index = 0; 92 93 /* end of buffer function */ 94 /**************************/ 95 96 /* encoding */ 97 /************/ 98 99 if (frame_mode == 0 || ISACenc_obj->frame_nb == 0 ) 100 { 101 /* reset bitstream */ 102 ISACenc_obj->bitstr_obj.W_upper = 0xFFFFFFFF; 103 ISACenc_obj->bitstr_obj.streamval = 0; 104 ISACenc_obj->bitstr_obj.stream_index = 0; 105 ISACenc_obj->bitstr_obj.full = 1; 106 107 if (CodingMode == 0) { 108 ISACenc_obj->BottleNeck = WebRtcIsacfix_GetUplinkBandwidth(bw_estimatordata); 109 ISACenc_obj->MaxDelay = WebRtcIsacfix_GetUplinkMaxDelay(bw_estimatordata); 110 } 111 if (CodingMode == 0 && frame_mode == 0 && (ISACenc_obj->enforceFrameSize == 0)) { 112 ISACenc_obj->new_framelength = WebRtcIsacfix_GetNewFrameLength(ISACenc_obj->BottleNeck, 113 ISACenc_obj->current_framesamples); 114 } 115 116 // multiply the bottleneck by 0.88 before computing SNR, 0.88 is tuned by experimenting on TIMIT 117 // 901/1024 is 0.87988281250000 118 ISACenc_obj->s2nr = WebRtcIsacfix_GetSnr( 119 (int16_t)(ISACenc_obj->BottleNeck * 901 >> 10), 120 ISACenc_obj->current_framesamples); 121 122 /* encode frame length */ 123 status = WebRtcIsacfix_EncodeFrameLen(ISACenc_obj->current_framesamples, &ISACenc_obj->bitstr_obj); 124 if (status < 0) 125 { 126 /* Wrong frame size */ 127 if (frame_mode == 1 && ISACenc_obj->frame_nb == 1) 128 { 129 // If this is the second 30ms of a 60ms frame reset this such that in the next call 130 // encoder starts fresh. 131 ISACenc_obj->frame_nb = 0; 132 } 133 return status; 134 } 135 136 /* Save framelength for multiple packets memory */ 137 if (ISACenc_obj->SaveEnc_ptr != NULL) { 138 (ISACenc_obj->SaveEnc_ptr)->framelength=ISACenc_obj->current_framesamples; 139 } 140 141 /* bandwidth estimation and coding */ 142 BWno = WebRtcIsacfix_GetDownlinkBwIndexImpl(bw_estimatordata); 143 status = WebRtcIsacfix_EncodeReceiveBandwidth(&BWno, &ISACenc_obj->bitstr_obj); 144 if (status < 0) 145 { 146 if (frame_mode == 1 && ISACenc_obj->frame_nb == 1) 147 { 148 // If this is the second 30ms of a 60ms frame reset this such that in the next call 149 // encoder starts fresh. 150 ISACenc_obj->frame_nb = 0; 151 } 152 return status; 153 } 154 } 155 156 /* split signal in two bands */ 157 WebRtcIsacfix_SplitAndFilter1(ISACenc_obj->data_buffer_fix, LP16a, HP16a, &ISACenc_obj->prefiltbankstr_obj ); 158 159 /* estimate pitch parameters and pitch-filter lookahead signal */ 160 WebRtcIsacfix_PitchAnalysis(LP16a+QLOOKAHEAD, LPandHP, 161 &ISACenc_obj->pitchanalysisstr_obj, PitchLags_Q7, PitchGains_Q12); /* LPandHP = LP_lookahead_pfQ0, */ 162 163 /* Set where to store data in multiple packets memory */ 164 if (ISACenc_obj->SaveEnc_ptr != NULL) { 165 if (frame_mode == 0 || ISACenc_obj->frame_nb == 0) 166 { 167 (ISACenc_obj->SaveEnc_ptr)->startIdx = 0; 168 } 169 else 170 { 171 (ISACenc_obj->SaveEnc_ptr)->startIdx = 1; 172 } 173 } 174 175 /* quantize & encode pitch parameters */ 176 status = WebRtcIsacfix_EncodePitchGain(PitchGains_Q12, &ISACenc_obj->bitstr_obj, ISACenc_obj->SaveEnc_ptr); 177 if (status < 0) 178 { 179 if (frame_mode == 1 && ISACenc_obj->frame_nb == 1) 180 { 181 // If this is the second 30ms of a 60ms frame reset this such that in the next call 182 // encoder starts fresh. 183 ISACenc_obj->frame_nb = 0; 184 } 185 return status; 186 } 187 status = WebRtcIsacfix_EncodePitchLag(PitchLags_Q7 , PitchGains_Q12, &ISACenc_obj->bitstr_obj, ISACenc_obj->SaveEnc_ptr); 188 if (status < 0) 189 { 190 if (frame_mode == 1 && ISACenc_obj->frame_nb == 1) 191 { 192 // If this is the second 30ms of a 60ms frame reset this such that in the next call 193 // encoder starts fresh. 194 ISACenc_obj->frame_nb = 0; 195 } 196 return status; 197 } 198 AvgPitchGain_Q12 = (PitchGains_Q12[0] + PitchGains_Q12[1] + 199 PitchGains_Q12[2] + PitchGains_Q12[3]) >> 2; 200 201 /* find coefficients for perceptual pre-filters */ 202 WebRtcIsacfix_GetLpcCoef(LPandHP, HP16a+QLOOKAHEAD, &ISACenc_obj->maskfiltstr_obj, 203 ISACenc_obj->s2nr, PitchGains_Q12, 204 gain_lo_hiQ17, lofilt_coefQ15, hifilt_coefQ15); /*LPandHP = LP_lookahead_pfQ0*/ 205 206 // record LPC Gains for possible bit-rate reduction 207 for(k = 0; k < KLT_ORDER_GAIN; k++) 208 { 209 transcodingParam.lpcGains[k] = gain_lo_hiQ17[k]; 210 } 211 212 /* code LPC model and shape - gains not quantized yet */ 213 status = WebRtcIsacfix_EncodeLpc(gain_lo_hiQ17, lofilt_coefQ15, hifilt_coefQ15, 214 &bmodel, &bits_gainsQ11, &ISACenc_obj->bitstr_obj, ISACenc_obj->SaveEnc_ptr, &transcodingParam); 215 if (status < 0) 216 { 217 if (frame_mode == 1 && ISACenc_obj->frame_nb == 1) 218 { 219 // If this is the second 30ms of a 60ms frame reset this such that in the next call 220 // encoder starts fresh. 221 ISACenc_obj->frame_nb = 0; 222 } 223 return status; 224 } 225 arithLenBeforeEncodingDFT = (ISACenc_obj->bitstr_obj.stream_index << 1) + (1-ISACenc_obj->bitstr_obj.full); 226 227 /* low-band filtering */ 228 WebRtcIsacfix_NormLatticeFilterMa(ORDERLO, ISACenc_obj->maskfiltstr_obj.PreStateLoGQ15, 229 LP16a, lofilt_coefQ15, gain_lo_hiQ17, 0, LPandHP);/* LPandHP = LP16b */ 230 231 /* pitch filter */ 232 WebRtcIsacfix_PitchFilter(LPandHP, LP16a, &ISACenc_obj->pitchfiltstr_obj, PitchLags_Q7, PitchGains_Q12, 1);/* LPandHP = LP16b */ 233 234 /* high-band filtering */ 235 WebRtcIsacfix_NormLatticeFilterMa(ORDERHI, ISACenc_obj->maskfiltstr_obj.PreStateHiGQ15, 236 HP16a, hifilt_coefQ15, gain_lo_hiQ17, 1, LPandHP);/*LPandHP = HP16b*/ 237 238 /* transform */ 239 WebRtcIsacfix_Time2Spec(LP16a, LPandHP, LP16a, LPandHP); /*LPandHP = HP16b*/ 240 241 /* Save data for multiple packets memory */ 242 if (ISACenc_obj->SaveEnc_ptr != NULL) { 243 for (k = 0; k < FRAMESAMPLES_HALF; k++) { 244 (ISACenc_obj->SaveEnc_ptr)->fre[k + (ISACenc_obj->SaveEnc_ptr)->startIdx*FRAMESAMPLES_HALF] = LP16a[k]; 245 (ISACenc_obj->SaveEnc_ptr)->fim[k + (ISACenc_obj->SaveEnc_ptr)->startIdx*FRAMESAMPLES_HALF] = LPandHP[k]; 246 } 247 (ISACenc_obj->SaveEnc_ptr)->AvgPitchGain[(ISACenc_obj->SaveEnc_ptr)->startIdx] = AvgPitchGain_Q12; 248 } 249 250 /* quantization and lossless coding */ 251 status = WebRtcIsacfix_EncodeSpec(LP16a, LPandHP, &ISACenc_obj->bitstr_obj, AvgPitchGain_Q12); 252 if((status <= -1) && (status != -ISAC_DISALLOWED_BITSTREAM_LENGTH)) /*LPandHP = HP16b*/ 253 { 254 if (frame_mode == 1 && ISACenc_obj->frame_nb == 1) 255 { 256 // If this is the second 30ms of a 60ms frame reset this such that in the next call 257 // encoder starts fresh. 258 ISACenc_obj->frame_nb = 0; 259 } 260 return status; 261 } 262 263 if((frame_mode == 1) && (ISACenc_obj->frame_nb == 0)) 264 { 265 // it is a 60ms and we are in the first 30ms 266 // then the limit at this point should be half of the assigned value 267 payloadLimitBytes = ISACenc_obj->payloadLimitBytes60 >> 1; 268 } 269 else if (frame_mode == 0) 270 { 271 // it is a 30ms frame 272 payloadLimitBytes = (ISACenc_obj->payloadLimitBytes30) - 3; 273 } 274 else 275 { 276 // this is the second half of a 60ms frame. 277 payloadLimitBytes = ISACenc_obj->payloadLimitBytes60 - 3; // subract 3 because termination process may add 3 bytes 278 } 279 280 iterCntr = 0; 281 while((((ISACenc_obj->bitstr_obj.stream_index) << 1) > payloadLimitBytes) || 282 (status == -ISAC_DISALLOWED_BITSTREAM_LENGTH)) 283 { 284 int16_t arithLenDFTByte; 285 int16_t bytesLeftQ5; 286 int16_t ratioQ5[8] = {0, 6, 9, 12, 16, 19, 22, 25}; 287 288 // According to experiments on TIMIT the following is proper for audio, but it is not agressive enough for tonal inputs 289 // such as DTMF, sweep-sine, ... 290 // 291 // (0.55 - (0.8 - ratio[i]/32) * 5 / 6) * 2^14 292 // int16_t scaleQ14[8] = {0, 648, 1928, 3208, 4915, 6195, 7475, 8755}; 293 294 295 // This is a supper-agressive scaling passed the tests (tonal inputs) tone with one iteration for payload limit 296 // of 120 (32kbps bottleneck), number of frames needed a rate-reduction was 58403 297 // 298 int16_t scaleQ14[8] = {0, 348, 828, 1408, 2015, 3195, 3500, 3500}; 299 int16_t idx; 300 301 if(iterCntr >= MAX_PAYLOAD_LIMIT_ITERATION) 302 { 303 // We were not able to limit the payload size 304 305 if((frame_mode == 1) && (ISACenc_obj->frame_nb == 0)) 306 { 307 // This was the first 30ms of a 60ms frame. Although the payload is larger than it 308 // should be but we let the second 30ms be encoded. Maybe togetehr we won't exceed 309 // the limit. 310 ISACenc_obj->frame_nb = 1; 311 return 0; 312 } 313 else if((frame_mode == 1) && (ISACenc_obj->frame_nb == 1)) 314 { 315 ISACenc_obj->frame_nb = 0; 316 } 317 318 if(status != -ISAC_DISALLOWED_BITSTREAM_LENGTH) 319 { 320 return -ISAC_PAYLOAD_LARGER_THAN_LIMIT; 321 } 322 else 323 { 324 return status; 325 } 326 } 327 if(status != -ISAC_DISALLOWED_BITSTREAM_LENGTH) 328 { 329 arithLenDFTByte = (ISACenc_obj->bitstr_obj.stream_index << 1) + (1-ISACenc_obj->bitstr_obj.full) - arithLenBeforeEncodingDFT; 330 bytesLeftQ5 = (payloadLimitBytes - arithLenBeforeEncodingDFT) << 5; 331 332 // bytesLeft / arithLenDFTBytes indicates how much scaling is required a rough estimate (agressive) 333 // scale = 0.55 - (0.8 - bytesLeft / arithLenDFTBytes) * 5 / 6 334 // bytesLeft / arithLenDFTBytes below 0.2 will have a scale of zero and above 0.8 are treated as 0.8 335 // to avoid division we do more simplification. 336 // 337 // values of (bytesLeft / arithLenDFTBytes)*32 between ratioQ5[i] and ratioQ5[i+1] are rounded to ratioQ5[i] 338 // and the corresponding scale is chosen 339 340 // we compare bytesLeftQ5 with ratioQ5[]*arithLenDFTByte; 341 idx = 4; 342 idx += (bytesLeftQ5 >= ratioQ5[idx] * arithLenDFTByte) ? 2 : -2; 343 idx += (bytesLeftQ5 >= ratioQ5[idx] * arithLenDFTByte) ? 1 : -1; 344 idx += (bytesLeftQ5 >= ratioQ5[idx] * arithLenDFTByte) ? 0 : -1; 345 } 346 else 347 { 348 // we are here because the bit-stream did not fit into the buffer, in this case, the stream_index is not 349 // trustable, especially if the is the first 30ms of a packet. Thereforem, we will go for the most agressive 350 // case. 351 idx = 0; 352 } 353 // scale FFT coefficients to reduce the bit-rate 354 for(k = 0; k < FRAMESAMPLES_HALF; k++) 355 { 356 LP16a[k] = (int16_t)(LP16a[k] * scaleQ14[idx] >> 14); 357 LPandHP[k] = (int16_t)(LPandHP[k] * scaleQ14[idx] >> 14); 358 } 359 360 // Save data for multiple packets memory 361 if (ISACenc_obj->SaveEnc_ptr != NULL) 362 { 363 for(k = 0; k < FRAMESAMPLES_HALF; k++) 364 { 365 (ISACenc_obj->SaveEnc_ptr)->fre[k + (ISACenc_obj->SaveEnc_ptr)->startIdx*FRAMESAMPLES_HALF] = LP16a[k]; 366 (ISACenc_obj->SaveEnc_ptr)->fim[k + (ISACenc_obj->SaveEnc_ptr)->startIdx*FRAMESAMPLES_HALF] = LPandHP[k]; 367 } 368 } 369 370 // scale the unquantized LPC gains and save the scaled version for the future use 371 for(k = 0; k < KLT_ORDER_GAIN; k++) 372 { 373 gain_lo_hiQ17[k] = WEBRTC_SPL_MUL_16_32_RSFT14(scaleQ14[idx], transcodingParam.lpcGains[k]);//transcodingParam.lpcGains[k]; // 374 transcodingParam.lpcGains[k] = gain_lo_hiQ17[k]; 375 } 376 377 // reset the bit-stream object to the state which it had before encoding LPC Gains 378 ISACenc_obj->bitstr_obj.full = transcodingParam.full; 379 ISACenc_obj->bitstr_obj.stream_index = transcodingParam.stream_index; 380 ISACenc_obj->bitstr_obj.streamval = transcodingParam.streamval; 381 ISACenc_obj->bitstr_obj.W_upper = transcodingParam.W_upper; 382 ISACenc_obj->bitstr_obj.stream[transcodingParam.stream_index-1] = transcodingParam.beforeLastWord; 383 ISACenc_obj->bitstr_obj.stream[transcodingParam.stream_index] = transcodingParam.lastWord; 384 385 386 // quantize and encode LPC gain 387 WebRtcIsacfix_EstCodeLpcGain(gain_lo_hiQ17, &ISACenc_obj->bitstr_obj, ISACenc_obj->SaveEnc_ptr); 388 arithLenBeforeEncodingDFT = (ISACenc_obj->bitstr_obj.stream_index << 1) + (1-ISACenc_obj->bitstr_obj.full); 389 status = WebRtcIsacfix_EncodeSpec(LP16a, LPandHP, &ISACenc_obj->bitstr_obj, AvgPitchGain_Q12); 390 if((status <= -1) && (status != -ISAC_DISALLOWED_BITSTREAM_LENGTH)) /*LPandHP = HP16b*/ 391 { 392 if (frame_mode == 1 && ISACenc_obj->frame_nb == 1) 393 { 394 // If this is the second 30ms of a 60ms frame reset this such that in the next call 395 // encoder starts fresh. 396 ISACenc_obj->frame_nb = 0; 397 } 398 return status; 399 } 400 iterCntr++; 401 } 402 403 if (frame_mode == 1 && ISACenc_obj->frame_nb == 0) 404 /* i.e. 60 ms framesize and just processed the first 30ms, */ 405 /* go back to main function to buffer the other 30ms speech frame */ 406 { 407 ISACenc_obj->frame_nb = 1; 408 return 0; 409 } 410 else if (frame_mode == 1 && ISACenc_obj->frame_nb == 1) 411 { 412 ISACenc_obj->frame_nb = 0; 413 /* also update the framelength for next packet, in Adaptive mode only */ 414 if (CodingMode == 0 && (ISACenc_obj->enforceFrameSize == 0)) { 415 ISACenc_obj->new_framelength = WebRtcIsacfix_GetNewFrameLength(ISACenc_obj->BottleNeck, 416 ISACenc_obj->current_framesamples); 417 } 418 } 419 420 421 /* complete arithmetic coding */ 422 stream_length = WebRtcIsacfix_EncTerminate(&ISACenc_obj->bitstr_obj); 423 /* can this be negative? */ 424 425 if(CodingMode == 0) 426 { 427 428 /* update rate model and get minimum number of bytes in this packet */ 429 MinBytes = WebRtcIsacfix_GetMinBytes(&ISACenc_obj->rate_data_obj, (int16_t) stream_length, 430 ISACenc_obj->current_framesamples, ISACenc_obj->BottleNeck, ISACenc_obj->MaxDelay); 431 432 /* if bitstream is too short, add garbage at the end */ 433 434 /* Store length of coded data */ 435 usefulstr_len = stream_length; 436 437 /* Make sure MinBytes does not exceed packet size limit */ 438 if ((ISACenc_obj->frame_nb == 0) && (MinBytes > ISACenc_obj->payloadLimitBytes30)) { 439 MinBytes = ISACenc_obj->payloadLimitBytes30; 440 } else if ((ISACenc_obj->frame_nb == 1) && (MinBytes > ISACenc_obj->payloadLimitBytes60)) { 441 MinBytes = ISACenc_obj->payloadLimitBytes60; 442 } 443 444 /* Make sure we don't allow more than 255 bytes of garbage data. 445 We store the length of the garbage data in 8 bits in the bitstream, 446 255 is the max garbage lenght we can signal using 8 bits. */ 447 if( MinBytes > usefulstr_len + 255 ) { 448 MinBytes = usefulstr_len + 255; 449 } 450 451 /* Save data for creation of multiple bitstreams */ 452 if (ISACenc_obj->SaveEnc_ptr != NULL) { 453 (ISACenc_obj->SaveEnc_ptr)->minBytes = MinBytes; 454 } 455 456 while (stream_length < MinBytes) 457 { 458 assert(stream_length >= 0); 459 if (stream_length & 0x0001){ 460 ISACenc_obj->bitstr_seed = WEBRTC_SPL_RAND( ISACenc_obj->bitstr_seed ); 461 ISACenc_obj->bitstr_obj.stream[stream_length / 2] |= 462 (uint16_t)(ISACenc_obj->bitstr_seed & 0xFF); 463 } else { 464 ISACenc_obj->bitstr_seed = WEBRTC_SPL_RAND( ISACenc_obj->bitstr_seed ); 465 ISACenc_obj->bitstr_obj.stream[stream_length / 2] = 466 ((uint16_t)ISACenc_obj->bitstr_seed << 8); 467 } 468 stream_length++; 469 } 470 471 /* to get the real stream_length, without garbage */ 472 if (usefulstr_len & 0x0001) { 473 ISACenc_obj->bitstr_obj.stream[usefulstr_len>>1] &= 0xFF00; 474 ISACenc_obj->bitstr_obj.stream[usefulstr_len>>1] += (MinBytes - usefulstr_len) & 0x00FF; 475 } 476 else { 477 ISACenc_obj->bitstr_obj.stream[usefulstr_len>>1] &= 0x00FF; 478 ISACenc_obj->bitstr_obj.stream[usefulstr_len >> 1] += 479 ((uint16_t)((MinBytes - usefulstr_len) & 0x00FF) << 8); 480 } 481 } 482 else 483 { 484 /* update rate model */ 485 WebRtcIsacfix_UpdateRateModel(&ISACenc_obj->rate_data_obj, (int16_t) stream_length, 486 ISACenc_obj->current_framesamples, ISACenc_obj->BottleNeck); 487 } 488 return stream_length; 489 } 490 491 /* This function is used to create a new bitstream with new BWE. 492 The same data as previously encoded with the fucntion WebRtcIsacfix_EncodeImpl() 493 is used. The data needed is taken from the struct, where it was stored 494 when calling the encoder. */ 495 int WebRtcIsacfix_EncodeStoredData(IsacFixEncoderInstance *ISACenc_obj, 496 int BWnumber, 497 float scale) 498 { 499 int ii; 500 int status; 501 int16_t BWno = (int16_t)BWnumber; 502 int stream_length = 0; 503 504 int16_t model; 505 const uint16_t *Q_PitchGain_cdf_ptr[1]; 506 const uint16_t **cdf; 507 const IsacSaveEncoderData *SaveEnc_str; 508 int32_t tmpLPCcoeffs_g[KLT_ORDER_GAIN<<1]; 509 int16_t tmpLPCindex_g[KLT_ORDER_GAIN<<1]; 510 int16_t tmp_fre[FRAMESAMPLES]; 511 int16_t tmp_fim[FRAMESAMPLES]; 512 513 SaveEnc_str = ISACenc_obj->SaveEnc_ptr; 514 515 /* Check if SaveEnc memory exists */ 516 if (SaveEnc_str == NULL) { 517 return (-1); 518 } 519 520 /* Sanity Check - possible values for BWnumber is 0 - 23 */ 521 if ((BWnumber < 0) || (BWnumber > 23)) { 522 return -ISAC_RANGE_ERROR_BW_ESTIMATOR; 523 } 524 525 /* reset bitstream */ 526 ISACenc_obj->bitstr_obj.W_upper = 0xFFFFFFFF; 527 ISACenc_obj->bitstr_obj.streamval = 0; 528 ISACenc_obj->bitstr_obj.stream_index = 0; 529 ISACenc_obj->bitstr_obj.full = 1; 530 531 /* encode frame length */ 532 status = WebRtcIsacfix_EncodeFrameLen(SaveEnc_str->framelength, &ISACenc_obj->bitstr_obj); 533 if (status < 0) { 534 /* Wrong frame size */ 535 return status; 536 } 537 538 /* encode bandwidth estimate */ 539 status = WebRtcIsacfix_EncodeReceiveBandwidth(&BWno, &ISACenc_obj->bitstr_obj); 540 if (status < 0) { 541 return status; 542 } 543 544 /* Transcoding */ 545 /* If scale < 1, rescale data to produce lower bitrate signal */ 546 if ((0.0 < scale) && (scale < 1.0)) { 547 /* Compensate LPC gain */ 548 for (ii = 0; ii < (KLT_ORDER_GAIN*(1+SaveEnc_str->startIdx)); ii++) { 549 tmpLPCcoeffs_g[ii] = (int32_t) ((scale) * (float) SaveEnc_str->LPCcoeffs_g[ii]); 550 } 551 552 /* Scale DFT */ 553 for (ii = 0; ii < (FRAMESAMPLES_HALF*(1+SaveEnc_str->startIdx)); ii++) { 554 tmp_fre[ii] = (int16_t) ((scale) * (float) SaveEnc_str->fre[ii]) ; 555 tmp_fim[ii] = (int16_t) ((scale) * (float) SaveEnc_str->fim[ii]) ; 556 } 557 } else { 558 for (ii = 0; ii < (KLT_ORDER_GAIN*(1+SaveEnc_str->startIdx)); ii++) { 559 tmpLPCindex_g[ii] = SaveEnc_str->LPCindex_g[ii]; 560 } 561 562 for (ii = 0; ii < (FRAMESAMPLES_HALF*(1+SaveEnc_str->startIdx)); ii++) { 563 tmp_fre[ii] = SaveEnc_str->fre[ii]; 564 tmp_fim[ii] = SaveEnc_str->fim[ii]; 565 } 566 } 567 568 /* Loop over number of 30 msec */ 569 for (ii = 0; ii <= SaveEnc_str->startIdx; ii++) 570 { 571 572 /* encode pitch gains */ 573 *Q_PitchGain_cdf_ptr = WebRtcIsacfix_kPitchGainCdf; 574 status = WebRtcIsacfix_EncHistMulti(&ISACenc_obj->bitstr_obj, &SaveEnc_str->pitchGain_index[ii], 575 Q_PitchGain_cdf_ptr, 1); 576 if (status < 0) { 577 return status; 578 } 579 580 /* entropy coding of quantization pitch lags */ 581 /* voicing classificiation */ 582 if (SaveEnc_str->meanGain[ii] <= 819) { 583 cdf = WebRtcIsacfix_kPitchLagPtrLo; 584 } else if (SaveEnc_str->meanGain[ii] <= 1638) { 585 cdf = WebRtcIsacfix_kPitchLagPtrMid; 586 } else { 587 cdf = WebRtcIsacfix_kPitchLagPtrHi; 588 } 589 status = WebRtcIsacfix_EncHistMulti(&ISACenc_obj->bitstr_obj, 590 &SaveEnc_str->pitchIndex[PITCH_SUBFRAMES*ii], cdf, PITCH_SUBFRAMES); 591 if (status < 0) { 592 return status; 593 } 594 595 /* LPC */ 596 /* entropy coding of model number */ 597 model = 0; 598 status = WebRtcIsacfix_EncHistMulti(&ISACenc_obj->bitstr_obj, &model, 599 WebRtcIsacfix_kModelCdfPtr, 1); 600 if (status < 0) { 601 return status; 602 } 603 604 /* entropy coding of quantization indices - LPC shape only */ 605 status = WebRtcIsacfix_EncHistMulti(&ISACenc_obj->bitstr_obj, &SaveEnc_str->LPCindex_s[KLT_ORDER_SHAPE*ii], 606 WebRtcIsacfix_kCdfShapePtr[0], KLT_ORDER_SHAPE); 607 if (status < 0) { 608 return status; 609 } 610 611 /* If transcoding, get new LPC gain indices */ 612 if (scale < 1.0) { 613 WebRtcIsacfix_TranscodeLpcCoef(&tmpLPCcoeffs_g[KLT_ORDER_GAIN*ii], &tmpLPCindex_g[KLT_ORDER_GAIN*ii]); 614 } 615 616 /* entropy coding of quantization indices - LPC gain */ 617 status = WebRtcIsacfix_EncHistMulti(&ISACenc_obj->bitstr_obj, &tmpLPCindex_g[KLT_ORDER_GAIN*ii], 618 WebRtcIsacfix_kCdfGainPtr[0], KLT_ORDER_GAIN); 619 if (status < 0) { 620 return status; 621 } 622 623 /* quantization and lossless coding */ 624 status = WebRtcIsacfix_EncodeSpec(&tmp_fre[ii*FRAMESAMPLES_HALF], &tmp_fim[ii*FRAMESAMPLES_HALF], 625 &ISACenc_obj->bitstr_obj, SaveEnc_str->AvgPitchGain[ii]); 626 if (status < 0) { 627 return status; 628 } 629 } 630 631 /* complete arithmetic coding */ 632 stream_length = WebRtcIsacfix_EncTerminate(&ISACenc_obj->bitstr_obj); 633 634 return stream_length; 635 } 636
