FAAC源碼分析之faacEncEncode

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FAAC編碼代碼流程圖

通用的AAC編碼系統框圖（偷來的圖）

對比可以發現，其實FAAC的編碼和一般的AAC編碼大致一致，主要包括心理聲學模型處理部分和量化處理兩部分，還包括一些最佳化處理的過程。實現的源碼+注釋：

int FAACAPI faacEncEncode(faacEncHandle hEncoder,int32_t *inputBuffer,unsigned int samplesInput,unsigned char *outputBuffer,unsigned int bufferSize){unsigned int channel, i;int sb, frameBytes;unsigned int offset;BitStream *bitStream; /* bitstream used for writing the frame to */TnsInfo *tnsInfo_for_LTP;TnsInfo *tnsDecInfo;#ifdef DRMint desbits, diff;double fix;#endif/* local copy's of parameters */ChannelInfo *channelInfo = hEncoder->channelInfo;CoderInfo *coderInfo = hEncoder->coderInfo;unsigned int numChannels = hEncoder->numChannels;unsigned int sampleRate = hEncoder->sampleRate;unsigned int aacObjectType = hEncoder->config.aacObjectType;unsigned int mpegVersion = hEncoder->config.mpegVersion;unsigned int useLfe = hEncoder->config.useLfe;unsigned int useTns = hEncoder->config.useTns;unsigned int allowMidside = hEncoder->config.allowMidside;unsigned int bandWidth = hEncoder->config.bandWidth;unsigned int shortctl = hEncoder->config.shortctl;/* Increase frame number */hEncoder->frameNum++;if (samplesInput == 0)hEncoder->flushFrame++;/* After 4 flush frames all samples have been encoded,return 0 bytes written */if (hEncoder->flushFrame > 4)return 0;/* Determine the channel configuration */GetChannelInfo(channelInfo, numChannels, useLfe);/* Update current sample buffers */for (channel = 0; channel < numChannels; channel++) {double *tmp;if (hEncoder->sampleBuff[channel]) {for(i = 0; i < FRAME_LEN; i++) {hEncoder->ltpTimeBuff[channel][i] = hEncoder->sampleBuff[channel][i];}}if (hEncoder->nextSampleBuff[channel]) {for(i = 0; i < FRAME_LEN; i++) {hEncoder->ltpTimeBuff[channel][FRAME_LEN + i] =hEncoder->nextSampleBuff[channel][i];}}if (!hEncoder->sampleBuff[channel])hEncoder->sampleBuff[channel] = (double*)AllocMemory(FRAME_LEN*sizeof(double));tmp = hEncoder->sampleBuff[channel];hEncoder->sampleBuff[channel]= hEncoder->nextSampleBuff[channel];hEncoder->nextSampleBuff[channel]= hEncoder->next2SampleBuff[channel];hEncoder->next2SampleBuff[channel]= hEncoder->next3SampleBuff[channel];hEncoder->next3SampleBuff[channel]= tmp;if (samplesInput == 0){/* start flushing*/for (i = 0; i < FRAME_LEN; i++)hEncoder->next3SampleBuff[channel][i] = 0.0;}else{int samples_per_channel = samplesInput/numChannels;/* handle the various input formats and channel remapping */switch( hEncoder->config.inputFormat ){case FAAC_INPUT_16BIT:{short *input_channel = (short*)inputBuffer + hEncoder->config.channel_map[channel];for (i = 0; i < samples_per_channel; i++){hEncoder->next3SampleBuff[channel][i] = (double)*input_channel;input_channel += numChannels;}}break;case FAAC_INPUT_32BIT:{int32_t *input_channel = (int32_t*)inputBuffer + hEncoder->config.channel_map[channel];for (i = 0; i < samples_per_channel; i++){hEncoder->next3SampleBuff[channel][i] = (1.0/256) * (double)*input_channel;input_channel += numChannels;}}break;case FAAC_INPUT_FLOAT:{float *input_channel = (float*)inputBuffer + hEncoder->config.channel_map[channel];for (i = 0; i < samples_per_channel; i++){hEncoder->next3SampleBuff[channel][i] = (double)*input_channel;input_channel += numChannels;}}break;default:return -1; /* invalid input format */break;}for (i = (int)(samplesInput/numChannels); i < FRAME_LEN; i++)hEncoder->next3SampleBuff[channel][i] = 0.0;}/* Psychoacoustics *//* Update buffers and run FFT on new samples *//* LFE psychoacoustic can run without it */if (!channelInfo[channel].lfe || channelInfo[channel].cpe){// 心理聲學模型的緩衝區更新, 計算當前幀能量值hEncoder->psymodel->PsyBufferUpdate( &hEncoder->fft_tables, &hEncoder->gpsyInfo, &hEncoder->psyInfo[channel],hEncoder->next3SampleBuff[channel], bandWidth,hEncoder->srInfo->cb_width_short,hEncoder->srInfo->num_cb_short);}}if (hEncoder->frameNum <= 3) /* Still filling up the buffers */return 0;// 內部調用實現檢測瞬變訊號, 判斷長短塊/* Psychoacoustics */hEncoder->psymodel->PsyCalculate(channelInfo, &hEncoder->gpsyInfo, hEncoder->psyInfo,hEncoder->srInfo->cb_width_long, hEncoder->srInfo->num_cb_long,hEncoder->srInfo->cb_width_short,hEncoder->srInfo->num_cb_short, numChannels);// 長短塊切換hEncoder->psymodel->BlockSwitch(coderInfo, hEncoder->psyInfo, numChannels);/* force block type */if (shortctl == SHORTCTL_NOSHORT){for (channel = 0; channel < numChannels; channel++){coderInfo[channel].block_type = ONLY_LONG_WINDOW;}}if (shortctl == SHORTCTL_NOLONG){for (channel = 0; channel < numChannels; channel++){coderInfo[channel].block_type = ONLY_SHORT_WINDOW;}}/* AAC Filterbank, MDCT with overlap and add */for (channel = 0; channel < numChannels; channel++) {int k;FilterBank(hEncoder,&coderInfo[channel],hEncoder->sampleBuff[channel],hEncoder->freqBuff[channel],hEncoder->overlapBuff[channel],MOVERLAPPED);if (coderInfo[channel].block_type == ONLY_SHORT_WINDOW) {for (k = 0; k < 8; k++) {specFilter(hEncoder->freqBuff[channel]+k*BLOCK_LEN_SHORT,sampleRate, bandWidth, BLOCK_LEN_SHORT);}} else {specFilter(hEncoder->freqBuff[channel], sampleRate,bandWidth, BLOCK_LEN_LONG);}}/* TMP: Build sfb offset table and other stuff */for (channel = 0; channel < numChannels; channel++) {channelInfo[channel].msInfo.is_present = 0;if (coderInfo[channel].block_type == ONLY_SHORT_WINDOW) {coderInfo[channel].max_sfb = hEncoder->srInfo->num_cb_short;coderInfo[channel].nr_of_sfb = hEncoder->srInfo->num_cb_short;coderInfo[channel].num_window_groups = 1;coderInfo[channel].window_group_length[0] = 8;coderInfo[channel].window_group_length[1] = 0;coderInfo[channel].window_group_length[2] = 0;coderInfo[channel].window_group_length[3] = 0;coderInfo[channel].window_group_length[4] = 0;coderInfo[channel].window_group_length[5] = 0;coderInfo[channel].window_group_length[6] = 0;coderInfo[channel].window_group_length[7] = 0;offset = 0;for (sb = 0; sb < coderInfo[channel].nr_of_sfb; sb++) {coderInfo[channel].sfb_offset[sb] = offset;offset += hEncoder->srInfo->cb_width_short[sb];}coderInfo[channel].sfb_offset[coderInfo[channel].nr_of_sfb] = offset;} else {coderInfo[channel].max_sfb = hEncoder->srInfo->num_cb_long;coderInfo[channel].nr_of_sfb = hEncoder->srInfo->num_cb_long;coderInfo[channel].num_window_groups = 1;coderInfo[channel].window_group_length[0] = 1;offset = 0;for (sb = 0; sb < coderInfo[channel].nr_of_sfb; sb++) {coderInfo[channel].sfb_offset[sb] = offset;offset += hEncoder->srInfo->cb_width_long[sb];}coderInfo[channel].sfb_offset[coderInfo[channel].nr_of_sfb] = offset;}}/* Perform TNS analysis and filtering */for (channel = 0; channel < numChannels; channel++) {if ((!channelInfo[channel].lfe) && (useTns)) {TnsEncode(&(coderInfo[channel].tnsInfo),coderInfo[channel].max_sfb,coderInfo[channel].max_sfb,(WINDOW_TYPE)coderInfo[channel].block_type,coderInfo[channel].sfb_offset,hEncoder->freqBuff[channel]);} else {coderInfo[channel].tnsInfo.tnsDataPresent = 0;      /* TNS not used for LFE */}}for(channel = 0; channel < numChannels; channel++){if((coderInfo[channel].tnsInfo.tnsDataPresent != 0) && (useTns))tnsInfo_for_LTP = &(coderInfo[channel].tnsInfo);elsetnsInfo_for_LTP = NULL;if(channelInfo[channel].present && (!channelInfo[channel].lfe) &&(coderInfo[channel].block_type != ONLY_SHORT_WINDOW) &&(mpegVersion == MPEG4) && (aacObjectType == LTP)){LtpEncode(hEncoder,&coderInfo[channel],&(coderInfo[channel].ltpInfo),tnsInfo_for_LTP,hEncoder->freqBuff[channel],hEncoder->ltpTimeBuff[channel]);} else {coderInfo[channel].ltpInfo.global_pred_flag = 0;}}for(channel = 0; channel < numChannels; channel++){if ((aacObjectType == MAIN) && (!channelInfo[channel].lfe)) {int numPredBands = min(coderInfo[channel].max_pred_sfb, coderInfo[channel].nr_of_sfb);PredCalcPrediction(hEncoder->freqBuff[channel],coderInfo[channel].requantFreq,coderInfo[channel].block_type,numPredBands,(coderInfo[channel].block_type==ONLY_SHORT_WINDOW)?hEncoder->srInfo->cb_width_short:hEncoder->srInfo->cb_width_long,coderInfo,channelInfo,channel);} else {coderInfo[channel].pred_global_flag = 0;}}for (channel = 0; channel < numChannels; channel++) {if (coderInfo[channel].block_type == ONLY_SHORT_WINDOW) {SortForGrouping(&coderInfo[channel],&hEncoder->psyInfo[channel],&channelInfo[channel],hEncoder->srInfo->cb_width_short,hEncoder->freqBuff[channel]);}CalcAvgEnrg(&coderInfo[channel], hEncoder->freqBuff[channel]);// reduce LFE bandwidthif (!channelInfo[channel].cpe && channelInfo[channel].lfe){coderInfo[channel].nr_of_sfb = coderInfo[channel].max_sfb = 3;}}MSEncode(coderInfo, channelInfo, hEncoder->freqBuff, numChannels, allowMidside);for (channel = 0; channel < numChannels; channel++){CalcAvgEnrg(&coderInfo[channel], hEncoder->freqBuff[channel]);}#ifdef DRM/* loop the quantization until the desired bit-rate is reached */diff = 1; /* to enter while loop */hEncoder->aacquantCfg.quality = 120; /* init quality setting */while (diff > 0) { /* if too many bits, do it again */#endif/* Quantize and code the signal */for (channel = 0; channel < numChannels; channel++) {if (coderInfo[channel].block_type == ONLY_SHORT_WINDOW) {AACQuantize(&coderInfo[channel], &hEncoder->psyInfo[channel],&channelInfo[channel], hEncoder->srInfo->cb_width_short,hEncoder->srInfo->num_cb_short, hEncoder->freqBuff[channel],&(hEncoder->aacquantCfg));} else {AACQuantize(&coderInfo[channel], &hEncoder->psyInfo[channel],&channelInfo[channel], hEncoder->srInfo->cb_width_long,hEncoder->srInfo->num_cb_long, hEncoder->freqBuff[channel],&(hEncoder->aacquantCfg));}}#ifdef DRM/* Write the AAC bitstream */bitStream = OpenBitStream(bufferSize, outputBuffer);WriteBitstream(hEncoder, coderInfo, channelInfo, bitStream, numChannels);/* Close the bitstream and return the number of bytes written */frameBytes = CloseBitStream(bitStream);/* now calculate desired bits and compare with actual encoded bits */desbits = (int) ((double) numChannels * (hEncoder->config.bitRate * FRAME_LEN)/ hEncoder->sampleRate);diff = ((frameBytes - 1 /* CRC */) * 8) - desbits;/* do linear correction according to relative difference */fix = (double) desbits / ((frameBytes - 1 /* CRC */) * 8);/* speed up convergence. A value of 0.92 gives approx up to 10 iterations */if (fix > 0.92)fix = 0.92;hEncoder->aacquantCfg.quality *= fix;/* quality should not go lower than 1, set diff to exit loop */if (hEncoder->aacquantCfg.quality <= 1)diff = -1;}#endif// fix max_sfb in CPE modefor (channel = 0; channel < numChannels; channel++){if (channelInfo[channel].present&& (channelInfo[channel].cpe)&& (channelInfo[channel].ch_is_left)){CoderInfo *cil, *cir;cil = &coderInfo[channel];cir = &coderInfo[channelInfo[channel].paired_ch];cil->max_sfb = cir->max_sfb = max(cil->max_sfb, cir->max_sfb);cil->nr_of_sfb = cir->nr_of_sfb = cil->max_sfb;}}MSReconstruct(coderInfo, channelInfo, numChannels);for (channel = 0; channel < numChannels; channel++){/* If short window, reconstruction not needed for prediction */if ((coderInfo[channel].block_type == ONLY_SHORT_WINDOW)) {int sind;for (sind = 0; sind < BLOCK_LEN_LONG; sind++) {coderInfo[channel].requantFreq[sind] = 0.0;}} else {if((coderInfo[channel].tnsInfo.tnsDataPresent != 0) && (useTns))tnsDecInfo = &(coderInfo[channel].tnsInfo);elsetnsDecInfo = NULL;if ((!channelInfo[channel].lfe) && (aacObjectType == LTP)) {  /* no reconstruction needed for LFE channel*/LtpReconstruct(&coderInfo[channel], &(coderInfo[channel].ltpInfo),coderInfo[channel].requantFreq);if(tnsDecInfo != NULL)TnsDecodeFilterOnly(&(coderInfo[channel].tnsInfo), coderInfo[channel].nr_of_sfb,coderInfo[channel].max_sfb, (WINDOW_TYPE)coderInfo[channel].block_type,coderInfo[channel].sfb_offset, coderInfo[channel].requantFreq);IFilterBank(hEncoder, &coderInfo[channel],coderInfo[channel].requantFreq,coderInfo[channel].ltpInfo.time_buffer,coderInfo[channel].ltpInfo.ltp_overlap_buffer,MOVERLAPPED);LtpUpdate(&(coderInfo[channel].ltpInfo),coderInfo[channel].ltpInfo.time_buffer,coderInfo[channel].ltpInfo.ltp_overlap_buffer,BLOCK_LEN_LONG);}}}#ifndef DRM/* Write the AAC bitstream */bitStream = OpenBitStream(bufferSize, outputBuffer);WriteBitstream(hEncoder, coderInfo, channelInfo, bitStream, numChannels);/* Close the bitstream and return the number of bytes written */frameBytes = CloseBitStream(bitStream);/* Adjust quality to get correct average bitrate */if (hEncoder->config.bitRate){double fix;int desbits = numChannels * (hEncoder->config.bitRate * FRAME_LEN)/ hEncoder->sampleRate;int diff = (frameBytes * 8) - desbits;hEncoder->bitDiff += diff;fix = (double)hEncoder->bitDiff / desbits;fix *= 0.01;fix = max(fix, -0.2);fix = min(fix, 0.2);if (((diff > 0) && (fix > 0.0)) || ((diff < 0) && (fix < 0.0))){hEncoder->aacquantCfg.quality *= (1.0 - fix);if (hEncoder->aacquantCfg.quality > 300)hEncoder->aacquantCfg.quality = 300;if (hEncoder->aacquantCfg.quality < 50)hEncoder->aacquantCfg.quality = 50;}}#endifreturn frameBytes;}

FAAC源碼分析之faacEncEncode