| /* |
| * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| /* |
| * encode.c |
| * |
| * Encoding function for the iSAC coder. |
| * |
| */ |
| |
| #include "rtc_base/checks.h" |
| #include "modules/audio_coding/codecs/isac/fix/source/codec.h" |
| |
| #include <stdio.h> |
| |
| #include "modules/audio_coding/codecs/isac/fix/source/arith_routins.h" |
| #include "modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h" |
| #include "modules/audio_coding/codecs/isac/fix/source/entropy_coding.h" |
| #include "modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h" |
| #include "modules/audio_coding/codecs/isac/fix/source/lpc_tables.h" |
| #include "modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h" |
| #include "modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.h" |
| #include "modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.h" |
| #include "modules/audio_coding/codecs/isac/fix/source/structs.h" |
| |
| |
| int WebRtcIsacfix_EncodeImpl(int16_t *in, |
| IsacFixEncoderInstance *ISACenc_obj, |
| BwEstimatorstr *bw_estimatordata, |
| int16_t CodingMode) |
| { |
| int16_t stream_length = 0; |
| int16_t usefulstr_len = 0; |
| int k; |
| int16_t BWno; |
| |
| int16_t lofilt_coefQ15[(ORDERLO)*SUBFRAMES]; |
| int16_t hifilt_coefQ15[(ORDERHI)*SUBFRAMES]; |
| int32_t gain_lo_hiQ17[2*SUBFRAMES]; |
| |
| int16_t LPandHP[FRAMESAMPLES/2 + QLOOKAHEAD]; |
| int16_t LP16a[FRAMESAMPLES/2 + QLOOKAHEAD]; |
| int16_t HP16a[FRAMESAMPLES/2 + QLOOKAHEAD]; |
| |
| int16_t PitchLags_Q7[PITCH_SUBFRAMES]; |
| int16_t PitchGains_Q12[PITCH_SUBFRAMES]; |
| int16_t AvgPitchGain_Q12; |
| |
| int16_t frame_mode; /* 0 for 30ms, 1 for 60ms */ |
| int16_t processed_samples; |
| int status; |
| |
| int32_t bits_gainsQ11; |
| int16_t MinBytes; |
| int16_t bmodel; |
| |
| transcode_obj transcodingParam; |
| int16_t payloadLimitBytes; |
| int16_t arithLenBeforeEncodingDFT; |
| int16_t iterCntr; |
| |
| /* copy new frame length and bottle neck rate only for the first 10 ms data */ |
| if (ISACenc_obj->buffer_index == 0) { |
| /* set the framelength for the next packet */ |
| ISACenc_obj->current_framesamples = ISACenc_obj->new_framelength; |
| } |
| |
| frame_mode = ISACenc_obj->current_framesamples/MAX_FRAMESAMPLES; /* 0 (30 ms) or 1 (60 ms) */ |
| processed_samples = ISACenc_obj->current_framesamples/(frame_mode+1); /* 480 (30, 60 ms) */ |
| |
| /* buffer speech samples (by 10ms packet) until the framelength is reached (30 or 60 ms) */ |
| /**************************************************************************************/ |
| /* fill the buffer with 10ms input data */ |
| for(k=0; k<FRAMESAMPLES_10ms; k++) { |
| ISACenc_obj->data_buffer_fix[k + ISACenc_obj->buffer_index] = in[k]; |
| } |
| /* if buffersize is not equal to current framesize, and end of file is not reached yet, */ |
| /* increase index and go back to main to get more speech samples */ |
| if (ISACenc_obj->buffer_index + FRAMESAMPLES_10ms != processed_samples) { |
| ISACenc_obj->buffer_index = ISACenc_obj->buffer_index + FRAMESAMPLES_10ms; |
| return 0; |
| } |
| /* if buffer reached the right size, reset index and continue with encoding the frame */ |
| ISACenc_obj->buffer_index = 0; |
| |
| /* end of buffer function */ |
| /**************************/ |
| |
| /* encoding */ |
| /************/ |
| |
| if (frame_mode == 0 || ISACenc_obj->frame_nb == 0 ) |
| { |
| /* reset bitstream */ |
| ISACenc_obj->bitstr_obj.W_upper = 0xFFFFFFFF; |
| ISACenc_obj->bitstr_obj.streamval = 0; |
| ISACenc_obj->bitstr_obj.stream_index = 0; |
| ISACenc_obj->bitstr_obj.full = 1; |
| |
| if (CodingMode == 0) { |
| ISACenc_obj->BottleNeck = WebRtcIsacfix_GetUplinkBandwidth(bw_estimatordata); |
| ISACenc_obj->MaxDelay = WebRtcIsacfix_GetUplinkMaxDelay(bw_estimatordata); |
| } |
| if (CodingMode == 0 && frame_mode == 0 && (ISACenc_obj->enforceFrameSize == 0)) { |
| ISACenc_obj->new_framelength = WebRtcIsacfix_GetNewFrameLength(ISACenc_obj->BottleNeck, |
| ISACenc_obj->current_framesamples); |
| } |
| |
| // multiply the bottleneck by 0.88 before computing SNR, 0.88 is tuned by experimenting on TIMIT |
| // 901/1024 is 0.87988281250000 |
| ISACenc_obj->s2nr = WebRtcIsacfix_GetSnr( |
| (int16_t)(ISACenc_obj->BottleNeck * 901 >> 10), |
| ISACenc_obj->current_framesamples); |
| |
| /* encode frame length */ |
| status = WebRtcIsacfix_EncodeFrameLen(ISACenc_obj->current_framesamples, &ISACenc_obj->bitstr_obj); |
| if (status < 0) |
| { |
| /* Wrong frame size */ |
| if (frame_mode == 1 && ISACenc_obj->frame_nb == 1) |
| { |
| // If this is the second 30ms of a 60ms frame reset this such that in the next call |
| // encoder starts fresh. |
| ISACenc_obj->frame_nb = 0; |
| } |
| return status; |
| } |
| |
| /* Save framelength for multiple packets memory */ |
| if (ISACenc_obj->SaveEnc_ptr != NULL) { |
| (ISACenc_obj->SaveEnc_ptr)->framelength=ISACenc_obj->current_framesamples; |
| } |
| |
| /* bandwidth estimation and coding */ |
| BWno = WebRtcIsacfix_GetDownlinkBwIndexImpl(bw_estimatordata); |
| status = WebRtcIsacfix_EncodeReceiveBandwidth(&BWno, &ISACenc_obj->bitstr_obj); |
| if (status < 0) |
| { |
| if (frame_mode == 1 && ISACenc_obj->frame_nb == 1) |
| { |
| // If this is the second 30ms of a 60ms frame reset this such that in the next call |
| // encoder starts fresh. |
| ISACenc_obj->frame_nb = 0; |
| } |
| return status; |
| } |
| } |
| |
| /* split signal in two bands */ |
| WebRtcIsacfix_SplitAndFilter1(ISACenc_obj->data_buffer_fix, LP16a, HP16a, &ISACenc_obj->prefiltbankstr_obj ); |
| |
| /* estimate pitch parameters and pitch-filter lookahead signal */ |
| WebRtcIsacfix_PitchAnalysis(LP16a+QLOOKAHEAD, LPandHP, |
| &ISACenc_obj->pitchanalysisstr_obj, PitchLags_Q7, PitchGains_Q12); /* LPandHP = LP_lookahead_pfQ0, */ |
| |
| /* Set where to store data in multiple packets memory */ |
| if (ISACenc_obj->SaveEnc_ptr != NULL) { |
| if (frame_mode == 0 || ISACenc_obj->frame_nb == 0) |
| { |
| (ISACenc_obj->SaveEnc_ptr)->startIdx = 0; |
| } |
| else |
| { |
| (ISACenc_obj->SaveEnc_ptr)->startIdx = 1; |
| } |
| } |
| |
| /* quantize & encode pitch parameters */ |
| status = WebRtcIsacfix_EncodePitchGain(PitchGains_Q12, &ISACenc_obj->bitstr_obj, ISACenc_obj->SaveEnc_ptr); |
| if (status < 0) |
| { |
| if (frame_mode == 1 && ISACenc_obj->frame_nb == 1) |
| { |
| // If this is the second 30ms of a 60ms frame reset this such that in the next call |
| // encoder starts fresh. |
| ISACenc_obj->frame_nb = 0; |
| } |
| return status; |
| } |
| status = WebRtcIsacfix_EncodePitchLag(PitchLags_Q7 , PitchGains_Q12, &ISACenc_obj->bitstr_obj, ISACenc_obj->SaveEnc_ptr); |
| if (status < 0) |
| { |
| if (frame_mode == 1 && ISACenc_obj->frame_nb == 1) |
| { |
| // If this is the second 30ms of a 60ms frame reset this such that in the next call |
| // encoder starts fresh. |
| ISACenc_obj->frame_nb = 0; |
| } |
| return status; |
| } |
| AvgPitchGain_Q12 = (PitchGains_Q12[0] + PitchGains_Q12[1] + |
| PitchGains_Q12[2] + PitchGains_Q12[3]) >> 2; |
| |
| /* find coefficients for perceptual pre-filters */ |
| WebRtcIsacfix_GetLpcCoef(LPandHP, HP16a+QLOOKAHEAD, &ISACenc_obj->maskfiltstr_obj, |
| ISACenc_obj->s2nr, PitchGains_Q12, |
| gain_lo_hiQ17, lofilt_coefQ15, hifilt_coefQ15); /*LPandHP = LP_lookahead_pfQ0*/ |
| |
| // record LPC Gains for possible bit-rate reduction |
| for(k = 0; k < KLT_ORDER_GAIN; k++) |
| { |
| transcodingParam.lpcGains[k] = gain_lo_hiQ17[k]; |
| } |
| |
| /* code LPC model and shape - gains not quantized yet */ |
| status = WebRtcIsacfix_EncodeLpc(gain_lo_hiQ17, lofilt_coefQ15, hifilt_coefQ15, |
| &bmodel, &bits_gainsQ11, &ISACenc_obj->bitstr_obj, ISACenc_obj->SaveEnc_ptr, &transcodingParam); |
| if (status < 0) |
| { |
| if (frame_mode == 1 && ISACenc_obj->frame_nb == 1) |
| { |
| // If this is the second 30ms of a 60ms frame reset this such that in the next call |
| // encoder starts fresh. |
| ISACenc_obj->frame_nb = 0; |
| } |
| return status; |
| } |
| arithLenBeforeEncodingDFT = (ISACenc_obj->bitstr_obj.stream_index << 1) + (1-ISACenc_obj->bitstr_obj.full); |
| |
| /* low-band filtering */ |
| WebRtcIsacfix_NormLatticeFilterMa(ORDERLO, ISACenc_obj->maskfiltstr_obj.PreStateLoGQ15, |
| LP16a, lofilt_coefQ15, gain_lo_hiQ17, 0, LPandHP);/* LPandHP = LP16b */ |
| |
| /* pitch filter */ |
| WebRtcIsacfix_PitchFilter(LPandHP, LP16a, &ISACenc_obj->pitchfiltstr_obj, PitchLags_Q7, PitchGains_Q12, 1);/* LPandHP = LP16b */ |
| |
| /* high-band filtering */ |
| WebRtcIsacfix_NormLatticeFilterMa(ORDERHI, ISACenc_obj->maskfiltstr_obj.PreStateHiGQ15, |
| HP16a, hifilt_coefQ15, gain_lo_hiQ17, 1, LPandHP);/*LPandHP = HP16b*/ |
| |
| /* transform */ |
| WebRtcIsacfix_Time2Spec(LP16a, LPandHP, LP16a, LPandHP); /*LPandHP = HP16b*/ |
| |
| /* Save data for multiple packets memory */ |
| if (ISACenc_obj->SaveEnc_ptr != NULL) { |
| for (k = 0; k < FRAMESAMPLES_HALF; k++) { |
| (ISACenc_obj->SaveEnc_ptr)->fre[k + (ISACenc_obj->SaveEnc_ptr)->startIdx*FRAMESAMPLES_HALF] = LP16a[k]; |
| (ISACenc_obj->SaveEnc_ptr)->fim[k + (ISACenc_obj->SaveEnc_ptr)->startIdx*FRAMESAMPLES_HALF] = LPandHP[k]; |
| } |
| (ISACenc_obj->SaveEnc_ptr)->AvgPitchGain[(ISACenc_obj->SaveEnc_ptr)->startIdx] = AvgPitchGain_Q12; |
| } |
| |
| /* quantization and lossless coding */ |
| status = WebRtcIsacfix_EncodeSpec(LP16a, LPandHP, &ISACenc_obj->bitstr_obj, AvgPitchGain_Q12); |
| if((status <= -1) && (status != -ISAC_DISALLOWED_BITSTREAM_LENGTH)) /*LPandHP = HP16b*/ |
| { |
| if (frame_mode == 1 && ISACenc_obj->frame_nb == 1) |
| { |
| // If this is the second 30ms of a 60ms frame reset this such that in the next call |
| // encoder starts fresh. |
| ISACenc_obj->frame_nb = 0; |
| } |
| return status; |
| } |
| |
| if((frame_mode == 1) && (ISACenc_obj->frame_nb == 0)) |
| { |
| // it is a 60ms and we are in the first 30ms |
| // then the limit at this point should be half of the assigned value |
| payloadLimitBytes = ISACenc_obj->payloadLimitBytes60 >> 1; |
| } |
| else if (frame_mode == 0) |
| { |
| // it is a 30ms frame |
| payloadLimitBytes = (ISACenc_obj->payloadLimitBytes30) - 3; |
| } |
| else |
| { |
| // this is the second half of a 60ms frame. |
| payloadLimitBytes = ISACenc_obj->payloadLimitBytes60 - 3; // subract 3 because termination process may add 3 bytes |
| } |
| |
| iterCntr = 0; |
| while((((ISACenc_obj->bitstr_obj.stream_index) << 1) > payloadLimitBytes) || |
| (status == -ISAC_DISALLOWED_BITSTREAM_LENGTH)) |
| { |
| int16_t arithLenDFTByte; |
| int16_t bytesLeftQ5; |
| int16_t ratioQ5[8] = {0, 6, 9, 12, 16, 19, 22, 25}; |
| |
| // According to experiments on TIMIT the following is proper for audio, but it is not agressive enough for tonal inputs |
| // such as DTMF, sweep-sine, ... |
| // |
| // (0.55 - (0.8 - ratio[i]/32) * 5 / 6) * 2^14 |
| // int16_t scaleQ14[8] = {0, 648, 1928, 3208, 4915, 6195, 7475, 8755}; |
| |
| |
| // This is a supper-agressive scaling passed the tests (tonal inputs) tone with one iteration for payload limit |
| // of 120 (32kbps bottleneck), number of frames needed a rate-reduction was 58403 |
| // |
| int16_t scaleQ14[8] = {0, 348, 828, 1408, 2015, 3195, 3500, 3500}; |
| int16_t idx; |
| |
| if(iterCntr >= MAX_PAYLOAD_LIMIT_ITERATION) |
| { |
| // We were not able to limit the payload size |
| |
| if((frame_mode == 1) && (ISACenc_obj->frame_nb == 0)) |
| { |
| // This was the first 30ms of a 60ms frame. Although the payload is larger than it |
| // should be but we let the second 30ms be encoded. Maybe togetehr we won't exceed |
| // the limit. |
| ISACenc_obj->frame_nb = 1; |
| return 0; |
| } |
| else if((frame_mode == 1) && (ISACenc_obj->frame_nb == 1)) |
| { |
| ISACenc_obj->frame_nb = 0; |
| } |
| |
| if(status != -ISAC_DISALLOWED_BITSTREAM_LENGTH) |
| { |
| return -ISAC_PAYLOAD_LARGER_THAN_LIMIT; |
| } |
| else |
| { |
| return status; |
| } |
| } |
| if(status != -ISAC_DISALLOWED_BITSTREAM_LENGTH) |
| { |
| arithLenDFTByte = (ISACenc_obj->bitstr_obj.stream_index << 1) + (1-ISACenc_obj->bitstr_obj.full) - arithLenBeforeEncodingDFT; |
| bytesLeftQ5 = (payloadLimitBytes - arithLenBeforeEncodingDFT) << 5; |
| |
| // bytesLeft / arithLenDFTBytes indicates how much scaling is required a rough estimate (agressive) |
| // scale = 0.55 - (0.8 - bytesLeft / arithLenDFTBytes) * 5 / 6 |
| // bytesLeft / arithLenDFTBytes below 0.2 will have a scale of zero and above 0.8 are treated as 0.8 |
| // to avoid division we do more simplification. |
| // |
| // values of (bytesLeft / arithLenDFTBytes)*32 between ratioQ5[i] and ratioQ5[i+1] are rounded to ratioQ5[i] |
| // and the corresponding scale is chosen |
| |
| // we compare bytesLeftQ5 with ratioQ5[]*arithLenDFTByte; |
| idx = 4; |
| idx += (bytesLeftQ5 >= ratioQ5[idx] * arithLenDFTByte) ? 2 : -2; |
| idx += (bytesLeftQ5 >= ratioQ5[idx] * arithLenDFTByte) ? 1 : -1; |
| idx += (bytesLeftQ5 >= ratioQ5[idx] * arithLenDFTByte) ? 0 : -1; |
| } |
| else |
| { |
| // we are here because the bit-stream did not fit into the buffer, in this case, the stream_index is not |
| // trustable, especially if the is the first 30ms of a packet. Thereforem, we will go for the most agressive |
| // case. |
| idx = 0; |
| } |
| // scale FFT coefficients to reduce the bit-rate |
| for(k = 0; k < FRAMESAMPLES_HALF; k++) |
| { |
| LP16a[k] = (int16_t)(LP16a[k] * scaleQ14[idx] >> 14); |
| LPandHP[k] = (int16_t)(LPandHP[k] * scaleQ14[idx] >> 14); |
| } |
| |
| // Save data for multiple packets memory |
| if (ISACenc_obj->SaveEnc_ptr != NULL) |
| { |
| for(k = 0; k < FRAMESAMPLES_HALF; k++) |
| { |
| (ISACenc_obj->SaveEnc_ptr)->fre[k + (ISACenc_obj->SaveEnc_ptr)->startIdx*FRAMESAMPLES_HALF] = LP16a[k]; |
| (ISACenc_obj->SaveEnc_ptr)->fim[k + (ISACenc_obj->SaveEnc_ptr)->startIdx*FRAMESAMPLES_HALF] = LPandHP[k]; |
| } |
| } |
| |
| // scale the unquantized LPC gains and save the scaled version for the future use |
| for(k = 0; k < KLT_ORDER_GAIN; k++) |
| { |
| gain_lo_hiQ17[k] = WEBRTC_SPL_MUL_16_32_RSFT14(scaleQ14[idx], transcodingParam.lpcGains[k]);//transcodingParam.lpcGains[k]; // |
| transcodingParam.lpcGains[k] = gain_lo_hiQ17[k]; |
| } |
| |
| // reset the bit-stream object to the state which it had before encoding LPC Gains |
| ISACenc_obj->bitstr_obj.full = transcodingParam.full; |
| ISACenc_obj->bitstr_obj.stream_index = transcodingParam.stream_index; |
| ISACenc_obj->bitstr_obj.streamval = transcodingParam.streamval; |
| ISACenc_obj->bitstr_obj.W_upper = transcodingParam.W_upper; |
| ISACenc_obj->bitstr_obj.stream[transcodingParam.stream_index-1] = transcodingParam.beforeLastWord; |
| ISACenc_obj->bitstr_obj.stream[transcodingParam.stream_index] = transcodingParam.lastWord; |
| |
| |
| // quantize and encode LPC gain |
| WebRtcIsacfix_EstCodeLpcGain(gain_lo_hiQ17, &ISACenc_obj->bitstr_obj, ISACenc_obj->SaveEnc_ptr); |
| arithLenBeforeEncodingDFT = (ISACenc_obj->bitstr_obj.stream_index << 1) + (1-ISACenc_obj->bitstr_obj.full); |
| status = WebRtcIsacfix_EncodeSpec(LP16a, LPandHP, &ISACenc_obj->bitstr_obj, AvgPitchGain_Q12); |
| if((status <= -1) && (status != -ISAC_DISALLOWED_BITSTREAM_LENGTH)) /*LPandHP = HP16b*/ |
| { |
| if (frame_mode == 1 && ISACenc_obj->frame_nb == 1) |
| { |
| // If this is the second 30ms of a 60ms frame reset this such that in the next call |
| // encoder starts fresh. |
| ISACenc_obj->frame_nb = 0; |
| } |
| return status; |
| } |
| iterCntr++; |
| } |
| |
| if (frame_mode == 1 && ISACenc_obj->frame_nb == 0) |
| /* i.e. 60 ms framesize and just processed the first 30ms, */ |
| /* go back to main function to buffer the other 30ms speech frame */ |
| { |
| ISACenc_obj->frame_nb = 1; |
| return 0; |
| } |
| else if (frame_mode == 1 && ISACenc_obj->frame_nb == 1) |
| { |
| ISACenc_obj->frame_nb = 0; |
| /* also update the framelength for next packet, in Adaptive mode only */ |
| if (CodingMode == 0 && (ISACenc_obj->enforceFrameSize == 0)) { |
| ISACenc_obj->new_framelength = WebRtcIsacfix_GetNewFrameLength(ISACenc_obj->BottleNeck, |
| ISACenc_obj->current_framesamples); |
| } |
| } |
| |
| |
| /* complete arithmetic coding */ |
| stream_length = WebRtcIsacfix_EncTerminate(&ISACenc_obj->bitstr_obj); |
| /* can this be negative? */ |
| |
| if(CodingMode == 0) |
| { |
| |
| /* update rate model and get minimum number of bytes in this packet */ |
| MinBytes = WebRtcIsacfix_GetMinBytes(&ISACenc_obj->rate_data_obj, (int16_t) stream_length, |
| ISACenc_obj->current_framesamples, ISACenc_obj->BottleNeck, ISACenc_obj->MaxDelay); |
| |
| /* if bitstream is too short, add garbage at the end */ |
| |
| /* Store length of coded data */ |
| usefulstr_len = stream_length; |
| |
| /* Make sure MinBytes does not exceed packet size limit */ |
| if ((ISACenc_obj->frame_nb == 0) && (MinBytes > ISACenc_obj->payloadLimitBytes30)) { |
| MinBytes = ISACenc_obj->payloadLimitBytes30; |
| } else if ((ISACenc_obj->frame_nb == 1) && (MinBytes > ISACenc_obj->payloadLimitBytes60)) { |
| MinBytes = ISACenc_obj->payloadLimitBytes60; |
| } |
| |
| /* Make sure we don't allow more than 255 bytes of garbage data. |
| We store the length of the garbage data in 8 bits in the bitstream, |
| 255 is the max garbage lenght we can signal using 8 bits. */ |
| if( MinBytes > usefulstr_len + 255 ) { |
| MinBytes = usefulstr_len + 255; |
| } |
| |
| /* Save data for creation of multiple bitstreams */ |
| if (ISACenc_obj->SaveEnc_ptr != NULL) { |
| (ISACenc_obj->SaveEnc_ptr)->minBytes = MinBytes; |
| } |
| |
| while (stream_length < MinBytes) |
| { |
| RTC_DCHECK_GE(stream_length, 0); |
| if (stream_length & 0x0001){ |
| ISACenc_obj->bitstr_seed = WEBRTC_SPL_RAND( ISACenc_obj->bitstr_seed ); |
| ISACenc_obj->bitstr_obj.stream[stream_length / 2] |= |
| (uint16_t)(ISACenc_obj->bitstr_seed & 0xFF); |
| } else { |
| ISACenc_obj->bitstr_seed = WEBRTC_SPL_RAND( ISACenc_obj->bitstr_seed ); |
| ISACenc_obj->bitstr_obj.stream[stream_length / 2] = |
| ((uint16_t)ISACenc_obj->bitstr_seed << 8); |
| } |
| stream_length++; |
| } |
| |
| /* to get the real stream_length, without garbage */ |
| if (usefulstr_len & 0x0001) { |
| ISACenc_obj->bitstr_obj.stream[usefulstr_len>>1] &= 0xFF00; |
| ISACenc_obj->bitstr_obj.stream[usefulstr_len>>1] += (MinBytes - usefulstr_len) & 0x00FF; |
| } |
| else { |
| ISACenc_obj->bitstr_obj.stream[usefulstr_len>>1] &= 0x00FF; |
| ISACenc_obj->bitstr_obj.stream[usefulstr_len >> 1] += |
| ((uint16_t)((MinBytes - usefulstr_len) & 0x00FF) << 8); |
| } |
| } |
| else |
| { |
| /* update rate model */ |
| WebRtcIsacfix_UpdateRateModel(&ISACenc_obj->rate_data_obj, (int16_t) stream_length, |
| ISACenc_obj->current_framesamples, ISACenc_obj->BottleNeck); |
| } |
| return stream_length; |
| } |
| |
| /* This function is used to create a new bitstream with new BWE. |
| The same data as previously encoded with the fucntion WebRtcIsacfix_EncodeImpl() |
| is used. The data needed is taken from the struct, where it was stored |
| when calling the encoder. */ |
| int WebRtcIsacfix_EncodeStoredData(IsacFixEncoderInstance *ISACenc_obj, |
| int BWnumber, |
| float scale) |
| { |
| int ii; |
| int status; |
| int16_t BWno = (int16_t)BWnumber; |
| int stream_length = 0; |
| |
| int16_t model; |
| const uint16_t *Q_PitchGain_cdf_ptr[1]; |
| const uint16_t **cdf; |
| const IsacSaveEncoderData *SaveEnc_str; |
| int32_t tmpLPCcoeffs_g[KLT_ORDER_GAIN<<1]; |
| int16_t tmpLPCindex_g[KLT_ORDER_GAIN<<1]; |
| int16_t tmp_fre[FRAMESAMPLES]; |
| int16_t tmp_fim[FRAMESAMPLES]; |
| |
| SaveEnc_str = ISACenc_obj->SaveEnc_ptr; |
| |
| /* Check if SaveEnc memory exists */ |
| if (SaveEnc_str == NULL) { |
| return (-1); |
| } |
| |
| /* Sanity Check - possible values for BWnumber is 0 - 23 */ |
| if ((BWnumber < 0) || (BWnumber > 23)) { |
| return -ISAC_RANGE_ERROR_BW_ESTIMATOR; |
| } |
| |
| /* reset bitstream */ |
| ISACenc_obj->bitstr_obj.W_upper = 0xFFFFFFFF; |
| ISACenc_obj->bitstr_obj.streamval = 0; |
| ISACenc_obj->bitstr_obj.stream_index = 0; |
| ISACenc_obj->bitstr_obj.full = 1; |
| |
| /* encode frame length */ |
| status = WebRtcIsacfix_EncodeFrameLen(SaveEnc_str->framelength, &ISACenc_obj->bitstr_obj); |
| if (status < 0) { |
| /* Wrong frame size */ |
| return status; |
| } |
| |
| /* encode bandwidth estimate */ |
| status = WebRtcIsacfix_EncodeReceiveBandwidth(&BWno, &ISACenc_obj->bitstr_obj); |
| if (status < 0) { |
| return status; |
| } |
| |
| /* Transcoding */ |
| /* If scale < 1, rescale data to produce lower bitrate signal */ |
| if ((0.0 < scale) && (scale < 1.0)) { |
| /* Compensate LPC gain */ |
| for (ii = 0; ii < (KLT_ORDER_GAIN*(1+SaveEnc_str->startIdx)); ii++) { |
| tmpLPCcoeffs_g[ii] = (int32_t) ((scale) * (float) SaveEnc_str->LPCcoeffs_g[ii]); |
| } |
| |
| /* Scale DFT */ |
| for (ii = 0; ii < (FRAMESAMPLES_HALF*(1+SaveEnc_str->startIdx)); ii++) { |
| tmp_fre[ii] = (int16_t) ((scale) * (float) SaveEnc_str->fre[ii]) ; |
| tmp_fim[ii] = (int16_t) ((scale) * (float) SaveEnc_str->fim[ii]) ; |
| } |
| } else { |
| for (ii = 0; ii < (KLT_ORDER_GAIN*(1+SaveEnc_str->startIdx)); ii++) { |
| tmpLPCindex_g[ii] = SaveEnc_str->LPCindex_g[ii]; |
| } |
| |
| for (ii = 0; ii < (FRAMESAMPLES_HALF*(1+SaveEnc_str->startIdx)); ii++) { |
| tmp_fre[ii] = SaveEnc_str->fre[ii]; |
| tmp_fim[ii] = SaveEnc_str->fim[ii]; |
| } |
| } |
| |
| /* Loop over number of 30 msec */ |
| for (ii = 0; ii <= SaveEnc_str->startIdx; ii++) |
| { |
| |
| /* encode pitch gains */ |
| *Q_PitchGain_cdf_ptr = WebRtcIsacfix_kPitchGainCdf; |
| status = WebRtcIsacfix_EncHistMulti(&ISACenc_obj->bitstr_obj, &SaveEnc_str->pitchGain_index[ii], |
| Q_PitchGain_cdf_ptr, 1); |
| if (status < 0) { |
| return status; |
| } |
| |
| /* entropy coding of quantization pitch lags */ |
| /* voicing classificiation */ |
| if (SaveEnc_str->meanGain[ii] <= 819) { |
| cdf = WebRtcIsacfix_kPitchLagPtrLo; |
| } else if (SaveEnc_str->meanGain[ii] <= 1638) { |
| cdf = WebRtcIsacfix_kPitchLagPtrMid; |
| } else { |
| cdf = WebRtcIsacfix_kPitchLagPtrHi; |
| } |
| status = WebRtcIsacfix_EncHistMulti(&ISACenc_obj->bitstr_obj, |
| &SaveEnc_str->pitchIndex[PITCH_SUBFRAMES*ii], cdf, PITCH_SUBFRAMES); |
| if (status < 0) { |
| return status; |
| } |
| |
| /* LPC */ |
| /* entropy coding of model number */ |
| model = 0; |
| status = WebRtcIsacfix_EncHistMulti(&ISACenc_obj->bitstr_obj, &model, |
| WebRtcIsacfix_kModelCdfPtr, 1); |
| if (status < 0) { |
| return status; |
| } |
| |
| /* entropy coding of quantization indices - LPC shape only */ |
| status = WebRtcIsacfix_EncHistMulti(&ISACenc_obj->bitstr_obj, &SaveEnc_str->LPCindex_s[KLT_ORDER_SHAPE*ii], |
| WebRtcIsacfix_kCdfShapePtr[0], KLT_ORDER_SHAPE); |
| if (status < 0) { |
| return status; |
| } |
| |
| /* If transcoding, get new LPC gain indices */ |
| if (scale < 1.0) { |
| WebRtcIsacfix_TranscodeLpcCoef(&tmpLPCcoeffs_g[KLT_ORDER_GAIN*ii], &tmpLPCindex_g[KLT_ORDER_GAIN*ii]); |
| } |
| |
| /* entropy coding of quantization indices - LPC gain */ |
| status = WebRtcIsacfix_EncHistMulti(&ISACenc_obj->bitstr_obj, &tmpLPCindex_g[KLT_ORDER_GAIN*ii], |
| WebRtcIsacfix_kCdfGainPtr[0], KLT_ORDER_GAIN); |
| if (status < 0) { |
| return status; |
| } |
| |
| /* quantization and lossless coding */ |
| status = WebRtcIsacfix_EncodeSpec(&tmp_fre[ii*FRAMESAMPLES_HALF], &tmp_fim[ii*FRAMESAMPLES_HALF], |
| &ISACenc_obj->bitstr_obj, SaveEnc_str->AvgPitchGain[ii]); |
| if (status < 0) { |
| return status; |
| } |
| } |
| |
| /* complete arithmetic coding */ |
| stream_length = WebRtcIsacfix_EncTerminate(&ISACenc_obj->bitstr_obj); |
| |
| return stream_length; |
| } |
