diff options
Diffstat (limited to 'libSBRenc/src/sbr_encoder.cpp')
| -rw-r--r-- | libSBRenc/src/sbr_encoder.cpp | 625 |
1 files changed, 367 insertions, 258 deletions
diff --git a/libSBRenc/src/sbr_encoder.cpp b/libSBRenc/src/sbr_encoder.cpp index e991199..3e95d6b 100644 --- a/libSBRenc/src/sbr_encoder.cpp +++ b/libSBRenc/src/sbr_encoder.cpp @@ -2,7 +2,7 @@ /* ----------------------------------------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. +© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -83,7 +83,7 @@ amm-info@iis.fraunhofer.de /*************************** Fraunhofer IIS FDK Tools *********************** - Author(s): Andreas Ehret + Author(s): Andreas Ehret, Tobias Chalupka Description: SBR encoder top level processing. ******************************************************************************/ @@ -102,8 +102,8 @@ amm-info@iis.fraunhofer.de #include "ps_main.h" #define SBRENCODER_LIB_VL0 3 -#define SBRENCODER_LIB_VL1 2 -#define SBRENCODER_LIB_VL2 2 +#define SBRENCODER_LIB_VL1 3 +#define SBRENCODER_LIB_VL2 4 @@ -119,34 +119,30 @@ amm-info@iis.fraunhofer.de (core2sbr delay ) ds (read, core and ds area) */ -#define DOWN_SMPL_FAC (2) +#define SFB(dwnsmp) (32 << (dwnsmp-1)) /* SBR Frequency bands: 64 for dual-rate, 32 for single-rate */ +#define STS(fl) (((fl)==1024)?32:30) /* SBR Time Slots: 32 for core frame length 1024, 30 for core frame length 960 */ -#define SFL(fl) (fl*DOWN_SMPL_FAC) /* SBR frame length (hardcoded to downsample factor of 2) */ -#define STS(fl) (SFL(fl)/64) /* SBR Time Slots */ - -#define DELAY_QMF_ANA (640 - 64) /* Full bandwidth */ -#define DELAY_QMF_ANAELD (32) -#define DELAY_HYB_ANA (10*64) /* + 0.5 */ -#define DELAY_HYB_SYN (6*64 - 32) -#define DELAY_QMF_SYNELD (32) -#define DELAY_DEC_QMF (6*64) /* Decoder QMF overlap */ -#define DELAY_QMF_SYN (2) /* NO_POLY/2 */ -#define DELAY_QMF_DS (32) /* QMF synthesis for downsampled time signal */ +#define DELAY_QMF_ANA(dwnsmp) ((320<<((dwnsmp)-1)) - (32<<((dwnsmp)-1))) /* Full bandwidth */ +#define DELAY_HYB_ANA (10*64) /* + 0.5 */ /* */ +#define DELAY_HYB_SYN (6*64 - 32) /* */ +#define DELAY_QMF_POSTPROC(dwnsmp) (32*(dwnsmp)) /* QMF postprocessing delay */ +#define DELAY_DEC_QMF(dwnsmp) (6 * SFB(dwnsmp) ) /* Decoder QMF overlap */ +#define DELAY_QMF_SYN (2) /* NO_POLY/2=2.5, rounded down to 2 */ +#define DELAY_QMF_DS (32) /* QMF synthesis for downsampled time signal */ /* Delay in QMF paths */ -#define DELAY_SBR(fl) (DELAY_QMF_ANA + (64*STS(fl)-1) + DELAY_QMF_SYN) -#define DELAY_PS(fl) (DELAY_QMF_ANA + DELAY_HYB_ANA + DELAY_DEC_QMF + (64*STS(fl)-1) + DELAY_HYB_SYN + DELAY_QMF_SYN) -#define DELAY_ELDSBR(fl) (DELAY_QMF_ANAELD + (((fl)+((fl)/2))*2 - 1) + DELAY_QMF_SYNELD) +#define DELAY_SBR(fl,dwnsmp) (DELAY_QMF_ANA(dwnsmp) + (SFB(dwnsmp)*STS(fl) - 1) + DELAY_QMF_SYN) +#define DELAY_PS(fl,dwnsmp) (DELAY_QMF_ANA(dwnsmp) + DELAY_HYB_ANA + DELAY_DEC_QMF(dwnsmp) + (SFB(dwnsmp)*STS(fl)-1) + DELAY_HYB_SYN + DELAY_QMF_SYN) +#define DELAY_ELDSBR(fl,dwnsmp) ( ( ((fl)/2)*(dwnsmp) ) - 1 + DELAY_QMF_POSTPROC(dwnsmp) ) /* Delay differences for SBR and SBR+PS */ - #define MAX_DS_FILTER_DELAY (34) /* the additional max downsampler filter delay (source fs) */ -#define DELAY_AAC2SBR(fl) ((/*RESAMPLER +*/ /*(DELAY_AAC(fl)*2) + */ DELAY_QMF_ANA + DELAY_DEC_QMF + DELAY_QMF_SYN) - DELAY_SBR(fl)) /* 1537 */ -#define DELAY_ELD2SBR(fl) ((/*RESAMPLER +*/ /*(DELAY_ELD(fl)*2) + */ DELAY_QMF_ANAELD + DELAY_QMF_SYNELD) - DELAY_ELDSBR(fl)) - -#define DELAY_AAC2PS(fl) ((DELAY_QMF_ANA + DELAY_QMF_DS + /*(DELAY_AAC(fl)*2)*/ + DELAY_QMF_ANA + DELAY_DEC_QMF + DELAY_HYB_SYN + DELAY_QMF_SYN) - DELAY_PS(fl)) /* 2048 - 463*2 */ +#define MAX_DS_FILTER_DELAY (5) /* the additional max downsampler filter delay (source fs) */ +#define DELAY_AAC2SBR(fl,dwnsmp) ((DELAY_QMF_ANA(dwnsmp) + DELAY_DEC_QMF(dwnsmp) + DELAY_QMF_SYN) - DELAY_SBR((fl),(dwnsmp))) +#define DELAY_ELD2SBR(fl,dwnsmp) ((DELAY_QMF_POSTPROC(dwnsmp)) - DELAY_ELDSBR(fl,dwnsmp)) +#define DELAY_AAC2PS(fl,dwnsmp) ((DELAY_QMF_ANA(dwnsmp) + DELAY_QMF_DS + /*(DELAY_AAC(fl)*2) + */ DELAY_QMF_ANA(dwnsmp) + DELAY_DEC_QMF(dwnsmp) + DELAY_HYB_SYN + DELAY_QMF_SYN) - DELAY_PS(fl,dwnsmp)) /* 2048 - 463*2 */ -/* Assumption: that the sample delay resulting of of DELAY_AAC2PS is always smaller than the sample delay implied by DELAY_AAC2SBR */ -#define MAX_SAMPLE_DELAY (DELAY_AAC2SBR(1024) + MAX_DS_FILTER_DELAY) +/* Assumption: The sample delay resulting of of DELAY_AAC2PS is always smaller than the sample delay implied by DELAY_AAC2SBR */ +#define MAX_SAMPLE_DELAY (DELAY_AAC2SBR(1024,2) + MAX_DS_FILTER_DELAY) /* maximum delay: frame length of 1024 and dual-rate sbr */ /***************************************************************************/ @@ -172,41 +168,39 @@ getSbrTuningTableIndex(UINT bitrate, /*! the total bitrate in bits/sec */ UINT *pBitRateClosest ) { - int i, paramSetTop, bitRateClosestLowerIndex=-1, bitRateClosestUpperIndex=-1, found = 0; + int i, bitRateClosestLowerIndex=-1, bitRateClosestUpperIndex=-1, found = 0; UINT bitRateClosestUpper = 0, bitRateClosestLower=DISTANCE_CEIL_VALUE; + int isforThisCodec=0; - FDK_ASSERT(SBRENC_TUNING_SIZE == sizeof(sbrTuningTable)/sizeof(sbrTuningTable[0])); - - if (core == AOT_ER_AAC_ELD) { - paramSetTop = SBRENC_TUNING_SIZE; - i = SBRENC_AACLC_TUNING_SIZE; - } else { - paramSetTop = SBRENC_AACLC_TUNING_SIZE; - i = 0; - } + #define isForThisCore(i) \ + ( ( sbrTuningTable[i].coreCoder == CODEC_AACLD && core == AOT_ER_AAC_ELD ) || \ + ( sbrTuningTable[i].coreCoder == CODEC_AAC && core != AOT_ER_AAC_ELD ) ) - for (; i < paramSetTop ; i++) { - if ( numChannels == sbrTuningTable [i].numChannels - && sampleRate == sbrTuningTable [i].sampleRate ) + for (i=0; i < sbrTuningTableSize ; i++) { + if ( isForThisCore(i) ) /* tuning table is for this core codec */ { - found = 1; - if ((bitrate >= sbrTuningTable [i].bitrateFrom) && - (bitrate < sbrTuningTable [i].bitrateTo)) { - bitRateClosestLower = bitrate; - bitRateClosestUpper = bitrate; - //FDKprintf("entry %d\n", i); - return i ; - } else { - if ( sbrTuningTable [i].bitrateFrom > bitrate ) { - if (sbrTuningTable [i].bitrateFrom < bitRateClosestLower) { - bitRateClosestLower = sbrTuningTable [i].bitrateFrom; - bitRateClosestLowerIndex = i; + if ( numChannels == sbrTuningTable [i].numChannels + && sampleRate == sbrTuningTable [i].sampleRate ) + { + found = 1; + if ((bitrate >= sbrTuningTable [i].bitrateFrom) && + (bitrate < sbrTuningTable [i].bitrateTo)) { + bitRateClosestLower = bitrate; + bitRateClosestUpper = bitrate; + //FDKprintf("entry %d\n", i); + return i ; + } else { + if ( sbrTuningTable [i].bitrateFrom > bitrate ) { + if (sbrTuningTable [i].bitrateFrom < bitRateClosestLower) { + bitRateClosestLower = sbrTuningTable [i].bitrateFrom; + bitRateClosestLowerIndex = i; + } } - } - if ( sbrTuningTable [i].bitrateTo <= bitrate ) { - if (sbrTuningTable [i].bitrateTo > bitRateClosestUpper) { - bitRateClosestUpper = sbrTuningTable [i].bitrateTo-1; - bitRateClosestUpperIndex = i; + if ( sbrTuningTable [i].bitrateTo <= bitrate ) { + if (sbrTuningTable [i].bitrateTo > bitRateClosestUpper) { + bitRateClosestUpper = sbrTuningTable [i].bitrateTo-1; + bitRateClosestUpperIndex = i; + } } } } @@ -215,7 +209,7 @@ getSbrTuningTableIndex(UINT bitrate, /*! the total bitrate in bits/sec */ if (pBitRateClosest != NULL) { - /* Is there was at least one matching tuning entry found then pick the least distance bit rate */ + /* If there was at least one matching tuning entry found then pick the least distance bit rate */ if (found) { int distanceUpper=DISTANCE_CEIL_VALUE, distanceLower=DISTANCE_CEIL_VALUE; @@ -295,6 +289,52 @@ getPsTuningTableIndex(UINT bitrate, UINT *pBitRateClosest){ return INVALID_TABLE_IDX; } +/***************************************************************************/ +/*! + + \brief In case of downsampled SBR we may need to lower the stop freq + of a tuning setting to fit into the lower half of the + spectrum ( which is sampleRate/4 ) + + \return the adapted stop frequency index (-1 -> error) + + \ingroup SbrEncCfg + +****************************************************************************/ +static INT +FDKsbrEnc_GetDownsampledStopFreq ( + const INT sampleRateCore, + const INT startFreq, + INT stopFreq, + const INT downSampleFactor + ) +{ + INT maxStopFreqRaw = sampleRateCore / 2; + INT startBand, stopBand; + HANDLE_ERROR_INFO err; + + while (stopFreq > 0 && FDKsbrEnc_getSbrStopFreqRAW(stopFreq, sampleRateCore) > maxStopFreqRaw) { + stopFreq--; + } + + if (FDKsbrEnc_getSbrStopFreqRAW( stopFreq, sampleRateCore) > maxStopFreqRaw) + return -1; + + err = FDKsbrEnc_FindStartAndStopBand ( + sampleRateCore<<(downSampleFactor-1), + sampleRateCore, + 32<<(downSampleFactor-1), + startFreq, + stopFreq, + &startBand, + &stopBand + ); + if (err) + return -1; + + return stopFreq; +} + /***************************************************************************/ /*! @@ -307,15 +347,16 @@ getPsTuningTableIndex(UINT bitrate, UINT *pBitRateClosest){ ****************************************************************************/ static UINT -FDKsbrEnc_IsSbrSettingAvail (UINT bitrate, /*! the total bitrate in bits/sec */ - UINT vbrMode, /*! the vbr paramter, 0 means constant bitrate */ - UINT numOutputChannels,/*! the number of channels for the core coder */ - UINT sampleRateInput, /*! the input sample rate [in Hz] */ - AUDIO_OBJECT_TYPE core - ) +FDKsbrEnc_IsSbrSettingAvail ( + UINT bitrate, /*! the total bitrate in bits/sec */ + UINT vbrMode, /*! the vbr paramter, 0 means constant bitrate */ + UINT numOutputChannels, /*! the number of channels for the core coder */ + UINT sampleRateInput, /*! the input sample rate [in Hz] */ + UINT sampleRateCore, /*! the core's sampling rate */ + AUDIO_OBJECT_TYPE core + ) { INT idx = INVALID_TABLE_IDX; - UINT sampleRateCore; if (sampleRateInput < 16000) return 0; @@ -335,8 +376,6 @@ FDKsbrEnc_IsSbrSettingAvail (UINT bitrate, /*! the total bitrate in bit bitrate *= numOutputChannels; } - /* try DOWN_SMPL_FAC of the input sampling rate */ - sampleRateCore = sampleRateInput/DOWN_SMPL_FAC; idx = getSbrTuningTableIndex(bitrate, numOutputChannels, sampleRateCore, core, NULL); return (idx == INVALID_TABLE_IDX ? 0 : 1); @@ -356,7 +395,8 @@ static UINT FDKsbrEnc_AdjustSbrSettings (const sbrConfigurationPtr config, /*! output, modified */ UINT bitRate, /*! the total bitrate in bits/sec */ UINT numChannels, /*! the core coder number of channels */ - UINT fsCore, /*! the core coder sampling rate in Hz */ + UINT sampleRateCore, /*! the core coder sampling rate in Hz */ + UINT sampleRateSbr, /*! the sbr coder sampling rate in Hz */ UINT transFac, /*! the short block to long block ratio */ UINT standardBitrate, /*! the standard bitrate per channel in bits/sec */ UINT vbrMode, /*! the vbr paramter, 0 poor quality .. 100 high quality*/ @@ -366,15 +406,12 @@ FDKsbrEnc_AdjustSbrSettings (const sbrConfigurationPtr config, /*! output, modif AUDIO_OBJECT_TYPE core) /* Core audio codec object type */ { INT idx = INVALID_TABLE_IDX; - UINT sampleRate; - - /* set the codec settings */ + /* set the core codec settings */ config->codecSettings.bitRate = bitRate; config->codecSettings.nChannels = numChannels; - config->codecSettings.sampleFreq = fsCore; + config->codecSettings.sampleFreq = sampleRateCore; config->codecSettings.transFac = transFac; config->codecSettings.standardBitrate = standardBitrate; - sampleRate = fsCore * DOWN_SMPL_FAC; if (bitRate==0) { /* map vbr quality to bitrate */ @@ -391,13 +428,13 @@ FDKsbrEnc_AdjustSbrSettings (const sbrConfigurationPtr config, /*! output, modif bitRate *= numChannels; /* fix to enable mono vbrMode<40 @ 44.1 of 48kHz */ if (numChannels==1) { - if (sampleRate==44100 || sampleRate==48000) { + if (sampleRateSbr==44100 || sampleRateSbr==48000) { if (vbrMode<40) bitRate = 32000; } } } - idx = getSbrTuningTableIndex(bitRate,numChannels,fsCore, core, NULL); + idx = getSbrTuningTableIndex(bitRate,numChannels,sampleRateCore, core, NULL); if (idx != INVALID_TABLE_IDX) { config->startFreq = sbrTuningTable[idx].startFreq ; @@ -407,6 +444,21 @@ FDKsbrEnc_AdjustSbrSettings (const sbrConfigurationPtr config, /*! output, modif config->stopFreq = sbrTuningTable[idx].stopFreqSpeech; } + /* Adapt stop frequency in case of downsampled SBR - only 32 bands then */ + if (1 == config->downSampleFactor) { + INT dsStopFreq = FDKsbrEnc_GetDownsampledStopFreq( + sampleRateCore, + config->startFreq, + config->stopFreq, + config->downSampleFactor + ); + if (dsStopFreq < 0) { + return 0; + } + + config->stopFreq = dsStopFreq; + } + config->sbr_noise_bands = sbrTuningTable[idx].numNoiseBands ; if (core == AOT_ER_AAC_ELD) config->init_amp_res_FF = SBR_AMP_RES_1_5; @@ -455,19 +507,20 @@ FDKsbrEnc_AdjustSbrSettings (const sbrConfigurationPtr config, /*! output, modif description: initializes the SBR confifuration returns: error status input: - core codec type, - - fac of SBR to core frame length, + - factor of SBR to core frame length, - core frame length output: initialized SBR configuration *****************************************************************************/ static UINT FDKsbrEnc_InitializeSbrDefaults (sbrConfigurationPtr config, - INT coreSbrFrameLenFac, - UINT codecGranuleLen) + INT downSampleFactor, + UINT codecGranuleLen + ) { - if ( (coreSbrFrameLenFac != 2) || - (codecGranuleLen*coreSbrFrameLenFac > QMF_CHANNELS*QMF_MAX_TIME_SLOTS) ) - return(1); + if ( (downSampleFactor < 1 || downSampleFactor > 2) || + (codecGranuleLen*downSampleFactor > QMF_CHANNELS*QMF_MAX_TIME_SLOTS) ) + return(0); /* error */ config->SendHeaderDataTime = 1000; config->useWaveCoding = 0; @@ -476,8 +529,8 @@ FDKsbrEnc_InitializeSbrDefaults (sbrConfigurationPtr config, config->tran_thr = 13000; config->parametricCoding = 1; - config->sbrFrameSize = codecGranuleLen * coreSbrFrameLenFac; - + config->sbrFrameSize = codecGranuleLen * downSampleFactor; + config->downSampleFactor = downSampleFactor; /* sbr default parameters */ config->sbr_data_extra = 0; @@ -497,7 +550,6 @@ FDKsbrEnc_InitializeSbrDefaults (sbrConfigurationPtr config, config->sbr_xpos_level = 0; config->useSaPan = 0; config->dynBwEnabled = 0; - config->bDownSampledSbr = 0; /* the following parameters are overwritten by the FDKsbrEnc_AdjustSbrSettings() function since @@ -601,7 +653,7 @@ void sbrEncoder_Close (HANDLE_SBR_ENCODER *phSbrEncoder) { int el, ch; - for (el=0; el<(6); el++) + for (el=0; el<(8); el++) { if (hSbrEncoder->sbrElement[el]!=NULL) { sbrEncoder_ElementClose(&hSbrEncoder->sbrElement[el]); @@ -609,7 +661,7 @@ void sbrEncoder_Close (HANDLE_SBR_ENCODER *phSbrEncoder) } /* Close sbr Channels */ - for (ch=0; ch<(6); ch++) + for (ch=0; ch<(8); ch++) { if (hSbrEncoder->pSbrChannel[ch]) { sbrEncoder_ChannelClose(hSbrEncoder->pSbrChannel[ch]); @@ -645,46 +697,62 @@ void sbrEncoder_Close (HANDLE_SBR_ENCODER *phSbrEncoder) output: error info *****************************************************************************/ -static INT updateFreqBandTable(HANDLE_SBR_CONFIG_DATA sbrConfigData, - HANDLE_SBR_HEADER_DATA sbrHeaderData, - INT noQmfChannels) +static INT updateFreqBandTable( + HANDLE_SBR_CONFIG_DATA sbrConfigData, + HANDLE_SBR_HEADER_DATA sbrHeaderData, + const INT downSampleFactor + ) { INT k0, k2; - if(FDKsbrEnc_FindStartAndStopBand(sbrConfigData->sampleFreq, - noQmfChannels, - sbrHeaderData->sbr_start_frequency, - sbrHeaderData->sbr_stop_frequency, - sbrHeaderData->sampleRateMode, - &k0, &k2)) + if( FDKsbrEnc_FindStartAndStopBand ( + sbrConfigData->sampleFreq, + sbrConfigData->sampleFreq >> (downSampleFactor-1), + sbrConfigData->noQmfBands, + sbrHeaderData->sbr_start_frequency, + sbrHeaderData->sbr_stop_frequency, + &k0, + &k2 + ) + ) return(1); - if(FDKsbrEnc_UpdateFreqScale(sbrConfigData->v_k_master, &sbrConfigData->num_Master, - k0, k2, sbrHeaderData->freqScale, - sbrHeaderData->alterScale)) + if( FDKsbrEnc_UpdateFreqScale( + sbrConfigData->v_k_master, + &sbrConfigData->num_Master, + k0, + k2, + sbrHeaderData->freqScale, + sbrHeaderData->alterScale + ) + ) return(1); sbrHeaderData->sbr_xover_band=0; - if(FDKsbrEnc_UpdateHiRes(sbrConfigData->freqBandTable[HI], - &sbrConfigData->nSfb[HI], - sbrConfigData->v_k_master, - sbrConfigData->num_Master , - &sbrHeaderData->sbr_xover_band, - sbrHeaderData->sampleRateMode, - noQmfChannels)) + if( FDKsbrEnc_UpdateHiRes( + sbrConfigData->freqBandTable[HI], + &sbrConfigData->nSfb[HI], + sbrConfigData->v_k_master, + sbrConfigData->num_Master, + &sbrHeaderData->sbr_xover_band + ) + ) return(1); - FDKsbrEnc_UpdateLoRes(sbrConfigData->freqBandTable[LO], - &sbrConfigData->nSfb[LO], - sbrConfigData->freqBandTable[HI], - sbrConfigData->nSfb[HI]); + FDKsbrEnc_UpdateLoRes( + sbrConfigData->freqBandTable[LO], + &sbrConfigData->nSfb[LO], + sbrConfigData->freqBandTable[HI], + sbrConfigData->nSfb[HI] + ); + - sbrConfigData->xOverFreq = (sbrConfigData->freqBandTable[LOW_RES][0] * sbrConfigData->sampleFreq / noQmfChannels+1)>>1; + sbrConfigData->xOverFreq = (sbrConfigData->freqBandTable[LOW_RES][0] * sbrConfigData->sampleFreq / sbrConfigData->noQmfBands+1)>>1; return (0); } @@ -866,7 +934,8 @@ FDKsbrEnc_EnvEncodeFrame(HANDLE_SBR_ENCODER hEnvEncoder, */ if(updateFreqBandTable(&hSbrElement->sbrConfigData, &hSbrElement->sbrHeaderData, - hSbrElement->sbrConfigData.noQmfBands)) + hEnvEncoder->downSampleFactor + )) return(1); @@ -891,8 +960,6 @@ FDKsbrEnc_EnvEncodeFrame(HANDLE_SBR_ENCODER hEnvEncoder, &crcInfo, hSbrElement->sbrConfigData.sbrSyntaxFlags); - INT error = noError; - /* Temporal Envelope Data */ SBR_FRAME_TEMP_DATA _fData; SBR_FRAME_TEMP_DATA *fData = &_fData; @@ -923,9 +990,9 @@ FDKsbrEnc_EnvEncodeFrame(HANDLE_SBR_ENCODER hEnvEncoder, if(hSbrElement->elInfo.fParametricStereo == 0) { - C_ALLOC_SCRATCH_START(qmfWorkBuffer, FIXP_DBL, QMF_CHANNELS*2); QMF_SCALE_FACTOR tmpScale; FIXP_DBL **pQmfReal, **pQmfImag; + C_AALLOC_SCRATCH_START(qmfWorkBuffer, FIXP_DBL, QMF_CHANNELS*2) /* Obtain pointers to QMF buffers. */ @@ -940,10 +1007,11 @@ FDKsbrEnc_EnvEncodeFrame(HANDLE_SBR_ENCODER hEnvEncoder, timeInStride, qmfWorkBuffer ); - C_ALLOC_SCRATCH_END(qmfWorkBuffer, FIXP_DBL, QMF_CHANNELS*2); - h_envChan->qmfScale = tmpScale.lb_scale + 7; + + C_AALLOC_SCRATCH_END(qmfWorkBuffer, FIXP_DBL, QMF_CHANNELS*2) + } /* fParametricStereo == 0 */ @@ -952,6 +1020,8 @@ FDKsbrEnc_EnvEncodeFrame(HANDLE_SBR_ENCODER hEnvEncoder, */ if (hSbrElement->elInfo.fParametricStereo) { + INT error = noError; + /* Limit Parametric Stereo to one instance */ FDK_ASSERT(ch == 0); @@ -1177,10 +1247,12 @@ initEnvChannel ( HANDLE_SBR_CONFIG_DATA sbrConfigData, break; case 2048: case 1024: + case 512: timeSlots = 16; break; case 1920: case 960: + case 480: timeSlots = 15; break; case 1152: @@ -1221,9 +1293,9 @@ initEnvChannel ( HANDLE_SBR_CONFIG_DATA sbrConfigData, tran_fc = params->tran_fc; - if (tran_fc == 0) - tran_fc = fixMin (5000, FDKsbrEnc_getSbrStartFreqRAW (sbrHeaderData->sbr_start_frequency,64,sbrConfigData->sampleFreq)); - + if (tran_fc == 0) { + tran_fc = fixMin (5000, FDKsbrEnc_getSbrStartFreqRAW (sbrHeaderData->sbr_start_frequency,params->codecSettings.sampleFreq)); + } tran_fc = (tran_fc*4*sbrConfigData->noQmfBands/sbrConfigData->sampleFreq + 1)>>1; @@ -1233,11 +1305,11 @@ initEnvChannel ( HANDLE_SBR_CONFIG_DATA sbrConfigData, } else { frameShift = 0; - switch (params->sbrFrameSize) { + switch (timeSlots) { /* The factor of 2 is by definition. */ - case 2048: tran_off = 8 + FRAME_MIDDLE_SLOT_2048 * timeStep; break; - case 1920: tran_off = 7 + FRAME_MIDDLE_SLOT_1920 * timeStep; break; - default: return 1; break; + case NUMBER_TIME_SLOTS_2048: tran_off = 8 + FRAME_MIDDLE_SLOT_2048 * timeStep; break; + case NUMBER_TIME_SLOTS_1920: tran_off = 7 + FRAME_MIDDLE_SLOT_1920 * timeStep; break; + default: return 1; } } if ( FDKsbrEnc_InitExtractSbrEnvelope (&hEnv->sbrExtractEnvelope, @@ -1330,7 +1402,6 @@ INT sbrEncoder_Open( hSbrEncoder->pSBRdynamic_RAM = (UCHAR*)GetRam_SbrDynamic_RAM(); hSbrEncoder->dynamicRam = hSbrEncoder->pSBRdynamic_RAM; - for (i=0; i<nElements; i++) { hSbrEncoder->sbrElement[i] = GetRam_SbrElement(i); if (hSbrEncoder->sbrElement[i]==NULL) { @@ -1397,7 +1468,7 @@ bail: static INT FDKsbrEnc_Reallocate( HANDLE_SBR_ENCODER hSbrEncoder, - SBR_ELEMENT_INFO elInfo[(6)], + SBR_ELEMENT_INFO elInfo[(8)], const INT noElements) { INT totalCh = 0; @@ -1462,7 +1533,9 @@ INT FDKsbrEnc_EnvInit ( AUDIO_OBJECT_TYPE aot, int nBitstrDelay, int nElement, - ULONG statesInitFlag + const int headerPeriod, + ULONG statesInitFlag, + int fTimeDomainDownsampling ,UCHAR *dynamic_RAM ) { @@ -1496,8 +1569,16 @@ INT FDKsbrEnc_EnvInit ( hSbrElement->sbrConfigData.sbrSyntaxFlags |= SBR_SYNTAX_CRC; } - hSbrElement->sbrConfigData.noQmfBands = QMF_CHANNELS; - hSbrElement->sbrConfigData.noQmfSlots = params->sbrFrameSize/hSbrElement->sbrConfigData.noQmfBands; + hSbrElement->sbrConfigData.noQmfBands = QMF_CHANNELS>>(2-params->downSampleFactor); + switch (hSbrElement->sbrConfigData.noQmfBands) + { + case 64: hSbrElement->sbrConfigData.noQmfSlots = params->sbrFrameSize>>6; + break; + case 32: hSbrElement->sbrConfigData.noQmfSlots = params->sbrFrameSize>>5; + break; + default: hSbrElement->sbrConfigData.noQmfSlots = params->sbrFrameSize>>6; + return(2); + } FDKinitBitStream(&hSbrElement->CmonData.sbrBitbuf, bitstreamBuffer, MAX_PAYLOAD_SIZE*sizeof(UCHAR), 0, BS_WRITER); @@ -1513,17 +1594,21 @@ INT FDKsbrEnc_EnvInit ( hSbrElement->sbrConfigData.frameSize = params->sbrFrameSize; - /* implicit rule for sampleRateMode */ - /* run in "multirate" mode where sbr fs is 2 * codec fs */ - hSbrElement->sbrHeaderData.sampleRateMode = DUAL_RATE; - hSbrElement->sbrConfigData.sampleFreq = 2 * params->codecSettings.sampleFreq; + hSbrElement->sbrConfigData.sampleFreq = params->downSampleFactor * params->codecSettings.sampleFreq; hSbrElement->sbrBitstreamData.CountSendHeaderData = 0; if (params->SendHeaderDataTime > 0 ) { - hSbrElement->sbrBitstreamData.NrSendHeaderData = (INT)(params->SendHeaderDataTime * hSbrElement->sbrConfigData.sampleFreq + if (headerPeriod==-1) { + + hSbrElement->sbrBitstreamData.NrSendHeaderData = (INT)(params->SendHeaderDataTime * hSbrElement->sbrConfigData.sampleFreq / (1000 * hSbrElement->sbrConfigData.frameSize)); - hSbrElement->sbrBitstreamData.NrSendHeaderData = fixMax(hSbrElement->sbrBitstreamData.NrSendHeaderData,1); + hSbrElement->sbrBitstreamData.NrSendHeaderData = fixMax(hSbrElement->sbrBitstreamData.NrSendHeaderData,1); + } + else { + /* assure header period at least once per second */ + hSbrElement->sbrBitstreamData.NrSendHeaderData = fixMin(fixMax(headerPeriod,1),(hSbrElement->sbrConfigData.sampleFreq/hSbrElement->sbrConfigData.frameSize)); + } } else { hSbrElement->sbrBitstreamData.NrSendHeaderData = 0; @@ -1584,7 +1669,8 @@ INT FDKsbrEnc_EnvInit ( /* init freq band table */ if(updateFreqBandTable(&hSbrElement->sbrConfigData, &hSbrElement->sbrHeaderData, - hSbrElement->sbrConfigData.noQmfBands)) + params->downSampleFactor + )) { return(1); } @@ -1624,6 +1710,9 @@ INT FDKsbrEnc_EnvInit ( hSbrElement->sbrConfigData.noQmfBands, hSbrElement->sbrConfigData.noQmfBands, qmfFlags ); + if (0!=err) { + return err; + } } /* */ @@ -1645,7 +1734,7 @@ INT sbrEncoder_GetInBufferSize(int noChannels) { INT temp; - temp = (1024*DOWN_SMPL_FAC); + temp = (2048); temp += 1024 + MAX_SAMPLE_DELAY; temp *= noChannels; temp *= sizeof(INT_PCM); @@ -1677,8 +1766,8 @@ INT FDKsbrEnc_DelayCompensation ( 1 )) return -1; - sbrEncoder_UpdateBuffers(hEnvEnc, timeBuffer); } + sbrEncoder_UpdateBuffers(hEnvEnc, timeBuffer); } return 0; } @@ -1709,29 +1798,36 @@ UINT sbrEncoder_LimitBitRate(UINT bitRate, UINT numChannels, UINT coreSampleRate return newBitRate; } +UINT sbrEncoder_IsSingleRatePossible(AUDIO_OBJECT_TYPE aot) +{ + UINT isPossible=(AOT_PS==aot)?0:1; + return isPossible; +} INT sbrEncoder_Init( - HANDLE_SBR_ENCODER hSbrEncoder, - SBR_ELEMENT_INFO elInfo[(6)], - int noElements, - INT_PCM *inputBuffer, - INT *coreBandwidth, - INT *inputBufferOffset, - INT *numChannels, - INT *sampleRate, - INT *frameLength, - AUDIO_OBJECT_TYPE *aot, - int *delay, - int transformFactor, - ULONG statesInitFlag - ) + HANDLE_SBR_ENCODER hSbrEncoder, + SBR_ELEMENT_INFO elInfo[(8)], + int noElements, + INT_PCM *inputBuffer, + INT *coreBandwidth, + INT *inputBufferOffset, + INT *numChannels, + INT *coreSampleRate, + UINT *downSampleFactor, + INT *frameLength, + AUDIO_OBJECT_TYPE aot, + int *delay, + int transformFactor, + const int headerPeriod, + ULONG statesInitFlag + ) { HANDLE_ERROR_INFO errorInfo = noError; - sbrConfiguration sbrConfig[(6)]; + sbrConfiguration sbrConfig[(8)]; INT error = 0; INT lowestBandwidth; /* Save input parameters */ - INT inputSampleRate = *sampleRate; + INT inputSampleRate = *coreSampleRate; int coreFrameLength = *frameLength; int inputBandWidth = *coreBandwidth; int inputChannels = *numChannels; @@ -1739,20 +1835,26 @@ INT sbrEncoder_Init( int downsampledOffset = 0; int sbrOffset = 0; int downsamplerDelay = 0; - int downsample = 0; + int timeDomainDownsample = 0; int nBitstrDelay = 0; - int lowestSbrStartFreq, lowestSbrStopFreq; + int highestSbrStartFreq, highestSbrStopFreq; int lowDelay = 0; int usePs = 0; /* check whether SBR setting is available for the current encoder configuration (bitrate, samplerate) */ - if ( (*aot==AOT_PS) || (*aot==AOT_MP2_PS) || (*aot==AOT_DABPLUS_PS) || (*aot==AOT_DRM_MPEG_PS) ) { + if (!sbrEncoder_IsSingleRatePossible(aot)) { + *downSampleFactor = 2; + } + + + + if ( (aot==AOT_PS) || (aot==AOT_MP2_PS) || (aot==AOT_DABPLUS_PS) || (aot==AOT_DRM_MPEG_PS) ) { usePs = 1; } - if ( (*aot==AOT_ER_AAC_ELD) ) { + if ( (aot==AOT_ER_AAC_ELD) ) { lowDelay = 1; } - else if ( (*aot==AOT_ER_AAC_LD) ) { + else if ( (aot==AOT_ER_AAC_LD) ) { error = 1; goto bail; } @@ -1767,25 +1869,25 @@ INT sbrEncoder_Init( /* core encoder gets downmixed mono signal */ *numChannels = 1; } else { - switch (*aot) { - case AOT_MP2_PS: - *aot = AOT_MP2_SBR; - break; - case AOT_DABPLUS_PS: - *aot = AOT_DABPLUS_SBR; - break; - case AOT_DRM_MPEG_PS: - *aot = AOT_DRM_SBR; - break; - case AOT_PS: - default: - *aot = AOT_SBR; - } - usePs = 0; + error = 1; + goto bail; } } /* usePs */ - /* check whether SBR setting is available for the current encoder configuration (bitrate, samplerate) */ + /* set the core's sample rate */ + switch (*downSampleFactor) { + case 1: + *coreSampleRate = inputSampleRate; + break; + case 2: + *coreSampleRate = inputSampleRate>>1; + break; + default: + *coreSampleRate = inputSampleRate>>1; + return 0; /* return error */ + } + + /* check whether SBR setting is available for the current encoder configuration (bitrate, coreSampleRate) */ { int delayDiff = 0; int el, coreEl; @@ -1798,54 +1900,37 @@ INT sbrEncoder_Init( continue; } /* check if desired configuration is available */ - if ( !FDKsbrEnc_IsSbrSettingAvail (elInfo[coreEl].bitRate, 0, elInfo[coreEl].nChannelsInEl, inputSampleRate, *aot) ) + if ( !FDKsbrEnc_IsSbrSettingAvail (elInfo[coreEl].bitRate, 0, elInfo[coreEl].nChannelsInEl, inputSampleRate, *coreSampleRate, aot) ) { - /* otherwise - change to AAC-LC */ - switch (*aot) { - case AOT_MP2_SBR: - case AOT_MP2_PS: - *aot = AOT_MP2_AAC_LC; - break; - case AOT_DABPLUS_SBR: - case AOT_DABPLUS_PS: - *aot = AOT_DABPLUS_AAC_LC; - break; - case AOT_DRM_SBR: - case AOT_DRM_MPEG_PS: - *aot = AOT_DRM_AAC; - break; - case AOT_ER_AAC_ELD: - break; - case AOT_SBR: - case AOT_PS: - default: - *aot = AOT_AAC_LC; - } error = 1; goto bail; } } - *sampleRate /= DOWN_SMPL_FAC; - /* Determine Delay balancing and new encoder delay */ if (lowDelay) { - downsample = 1; /* activate downsampler */ - delayDiff = (*delay*DOWN_SMPL_FAC) + DELAY_ELD2SBR(coreFrameLength); - *delay = DELAY_ELDSBR(coreFrameLength); + { + delayDiff = (*delay * *downSampleFactor) + DELAY_ELD2SBR(coreFrameLength,*downSampleFactor); + *delay = DELAY_ELDSBR(coreFrameLength,*downSampleFactor); + } } else if (usePs) { - delayDiff = (*delay*DOWN_SMPL_FAC) + DELAY_AAC2PS(coreFrameLength); - *delay = DELAY_PS(coreFrameLength); + delayDiff = (*delay * *downSampleFactor) + DELAY_AAC2PS(coreFrameLength,*downSampleFactor); + *delay = DELAY_PS(coreFrameLength,*downSampleFactor); } else { - downsample = 1; /* activate downsampler */ - delayDiff = (*delay*DOWN_SMPL_FAC) + DELAY_AAC2SBR(coreFrameLength); - *delay = DELAY_SBR(coreFrameLength); + delayDiff = DELAY_AAC2SBR(coreFrameLength,*downSampleFactor); + delayDiff += (*delay * *downSampleFactor); + *delay = DELAY_SBR(coreFrameLength,*downSampleFactor); } + if (!usePs) { + timeDomainDownsample = *downSampleFactor-1; /* activate time domain downsampler when downSampleFactor is != 1 */ + } + + /* Take care about downsampled data bound to the SBR path */ - if (!downsample && delayDiff > 0) { + if (!timeDomainDownsample && delayDiff > 0) { /* * We must tweak the balancing into a situation where the downsampled path * is the one to be delayed, because delaying the QMF domain input, also delays @@ -1854,12 +1939,15 @@ INT sbrEncoder_Init( while ( delayDiff > 0 ) { /* Encoder delay increases */ - *delay += coreFrameLength*DOWN_SMPL_FAC; - /* Add one frame delay to SBR path */ - delayDiff -= coreFrameLength*DOWN_SMPL_FAC; + { + *delay += coreFrameLength * *downSampleFactor; + /* Add one frame delay to SBR path */ + delayDiff -= coreFrameLength * *downSampleFactor; + } nBitstrDelay += 1; } - } else { + } else + { *delay += fixp_abs(delayDiff); } @@ -1867,32 +1955,33 @@ INT sbrEncoder_Init( /* Delay AAC data */ delayDiff = -delayDiff; /* Multiply downsampled offset by AAC core channels. Divide by 2 because of half samplerate of downsampled data. */ - downsampledOffset = (delayDiff*(*numChannels))/DOWN_SMPL_FAC; + FDK_ASSERT(*downSampleFactor>0 && *downSampleFactor<=2); + downsampledOffset = (delayDiff*(*numChannels))>>(*downSampleFactor-1); sbrOffset = 0; } else { /* Delay SBR input */ - if ( delayDiff > (int)coreFrameLength*DOWN_SMPL_FAC ) + if ( delayDiff > (int)coreFrameLength * (int)*downSampleFactor ) { /* Do bitstream frame-wise delay balancing if we have more than SBR framelength samples delay difference */ - delayDiff -= coreFrameLength*DOWN_SMPL_FAC; + delayDiff -= coreFrameLength * *downSampleFactor; nBitstrDelay = 1; } /* Multiply input offset by input channels */ sbrOffset = delayDiff*(*numChannels); downsampledOffset = 0; } - - hSbrEncoder->nBitstrDelay = nBitstrDelay; - hSbrEncoder->nChannels = *numChannels; - hSbrEncoder->frameSize = *frameLength*DOWN_SMPL_FAC; - hSbrEncoder->fTimeDomainDownsampling = downsample; - hSbrEncoder->estimateBitrate = 0; - hSbrEncoder->inputDataDelay = 0; + hSbrEncoder->nBitstrDelay = nBitstrDelay; + hSbrEncoder->nChannels = *numChannels; + hSbrEncoder->frameSize = coreFrameLength * *downSampleFactor; + hSbrEncoder->fTimeDomainDownsampling = timeDomainDownsample; + hSbrEncoder->downSampleFactor = *downSampleFactor; + hSbrEncoder->estimateBitrate = 0; + hSbrEncoder->inputDataDelay = 0; /* Open SBR elements */ el = -1; - lowestSbrStartFreq = lowestSbrStopFreq = 9999; + highestSbrStartFreq = highestSbrStopFreq = 0; lowestBandwidth = 99999; /* Loop through each core encoder element and get a matching SBR element config */ @@ -1915,28 +2004,38 @@ INT sbrEncoder_Init( /* * Init sbrConfig structure */ - FDKsbrEnc_InitializeSbrDefaults ( &sbrConfig[el], - DOWN_SMPL_FAC, - coreFrameLength); + if ( ! FDKsbrEnc_InitializeSbrDefaults ( &sbrConfig[el], + *downSampleFactor, + coreFrameLength + ) ) + { + error = 1; + goto bail; + } + /* * Modify sbrConfig structure according to Element parameters */ - FDKsbrEnc_AdjustSbrSettings ( &sbrConfig[el], - elInfo[coreEl].bitRate, - elInfo[coreEl].nChannelsInEl, - *sampleRate, - transformFactor, - 24000, - 0, - 0, /* useSpeechConfig */ - 0, /* lcsMode */ - usePs, /* bParametricStereo */ - *aot); + if ( ! FDKsbrEnc_AdjustSbrSettings (&sbrConfig[el], + elInfo[coreEl].bitRate, + elInfo[coreEl].nChannelsInEl, + *coreSampleRate, + inputSampleRate, + transformFactor, + 24000, + 0, + 0, /* useSpeechConfig */ + 0, /* lcsMode */ + usePs, /* bParametricStereo */ + aot) ) + { + error = 1; + goto bail; + } /* Find common frequency border for all SBR elements */ - lowestSbrStartFreq = fixMin(lowestSbrStartFreq, sbrConfig[el].startFreq); - lowestSbrStopFreq = fixMin(lowestSbrStopFreq, sbrConfig[el].stopFreq); - + highestSbrStartFreq = fixMax(highestSbrStartFreq, sbrConfig[el].startFreq); + highestSbrStopFreq = fixMax(highestSbrStopFreq, sbrConfig[el].stopFreq); } /* first element loop */ @@ -1952,21 +2051,24 @@ INT sbrEncoder_Init( int bandwidth = *coreBandwidth; /* Use lowest common bandwidth */ - sbrConfig[el].startFreq = lowestSbrStartFreq; - sbrConfig[el].stopFreq = lowestSbrStopFreq; + sbrConfig[el].startFreq = highestSbrStartFreq; + sbrConfig[el].stopFreq = highestSbrStopFreq; /* initialize SBR element, and get core bandwidth */ error = FDKsbrEnc_EnvInit(hSbrEncoder->sbrElement[el], &sbrConfig[el], &bandwidth, - *aot, + aot, nBitstrDelay, el, - statesInitFlag + headerPeriod, + statesInitFlag, + hSbrEncoder->fTimeDomainDownsampling ,hSbrEncoder->dynamicRam ); if (error != 0) { + error = 2; goto bail; } @@ -1988,30 +2090,29 @@ INT sbrEncoder_Init( for (ch=0; ch<hSbrEl->elInfo.nChannelsInEl; ch++) { - FDKaacEnc_InitDownsampler (&hSbrEl->sbrChannel[ch]->downSampler, Wc, DOWN_SMPL_FAC); + FDKaacEnc_InitDownsampler (&hSbrEl->sbrChannel[ch]->downSampler, Wc, *downSampleFactor); + FDK_ASSERT (hSbrEl->sbrChannel[ch]->downSampler.delay <=MAX_DS_FILTER_DELAY); } - FDK_ASSERT (hSbrEl->sbrChannel[0]->downSampler.delay <=MAX_DS_FILTER_DELAY && hSbrEl->sbrChannel[0]->downSampler.delay <=MAX_DS_FILTER_DELAY); downsamplerDelay = hSbrEl->sbrChannel[0]->downSampler.delay; } /* third element loop */ /* lfe */ - FDKaacEnc_InitDownsampler (&hSbrEncoder->lfeDownSampler, 0, DOWN_SMPL_FAC); + FDKaacEnc_InitDownsampler (&hSbrEncoder->lfeDownSampler, 0, *downSampleFactor); /* Add the resampler additional delay to get the final delay and buffer offset values. */ - if (sbrOffset > 0 || downsampledOffset <= ((downsamplerDelay * (*numChannels))/DOWN_SMPL_FAC)) { + if (sbrOffset > 0 || downsampledOffset <= ((downsamplerDelay * (*numChannels))>>(*downSampleFactor-1))) { sbrOffset += (downsamplerDelay - downsampledOffset) * (*numChannels) ; *delay += downsamplerDelay - downsampledOffset; downsampledOffset = 0; } else { - downsampledOffset -= (downsamplerDelay * (*numChannels))/DOWN_SMPL_FAC; + downsampledOffset -= (downsamplerDelay * (*numChannels))>>(*downSampleFactor-1); sbrOffset = 0; } hSbrEncoder->inputDataDelay = downsamplerDelay; } - /* Assign core encoder Bandwidth */ *coreBandwidth = lowestBandwidth; @@ -2025,7 +2126,7 @@ INT sbrEncoder_Init( FDK_ASSERT(hSbrEncoder->noElements == 1); INT psTuningTableIdx = getPsTuningTableIndex(elInfo[0].bitRate, NULL); - psEncConfig.frameSize = *frameLength; //sbrConfig.sbrFrameSize; + psEncConfig.frameSize = coreFrameLength; //sbrConfig.sbrFrameSize; psEncConfig.qmfFilterMode = 0; psEncConfig.sbrPsDelay = 0; @@ -2037,7 +2138,7 @@ INT sbrEncoder_Init( /* calculation is not quite linear, increased number of envelopes causes more bits */ /* assume avg. 50 bits per frame for 10 stereo bands / 1 envelope configuration */ - hSbrEncoder->estimateBitrate += ( (((*sampleRate) * 5 * psEncConfig.nStereoBands * psEncConfig.maxEnvelopes) / hSbrEncoder->frameSize)); + hSbrEncoder->estimateBitrate += ( (((*coreSampleRate) * 5 * psEncConfig.nStereoBands * psEncConfig.maxEnvelopes) / hSbrEncoder->frameSize)); } else { error = ERROR(CDI, "Invalid ps tuning table index."); @@ -2066,10 +2167,16 @@ INT sbrEncoder_Init( errorInfo = handBack(errorInfo); } } + + /* QMF analysis + Hybrid analysis + Hybrid synthesis + QMF synthesis + downsampled input buffer delay */ + hSbrEncoder->inputDataDelay = (64*10/2) + (6*64) + (0) + (64*10/2-64+1) + ((*downSampleFactor)*downsampledOffset); } hSbrEncoder->downsampledOffset = downsampledOffset; - hSbrEncoder->downmixSize = coreFrameLength*(*numChannels); + { + hSbrEncoder->downmixSize = coreFrameLength*(*numChannels); + } + hSbrEncoder->bufferOffset = sbrOffset; /* Delay Compensation: fill bitstream delay buffer with zero input signal */ if ( hSbrEncoder->nBitstrDelay > 0 ) @@ -2080,7 +2187,7 @@ INT sbrEncoder_Init( } /* Set Output frame length */ - *frameLength = coreFrameLength*DOWN_SMPL_FAC; + *frameLength = coreFrameLength * *downSampleFactor; /* Input buffer offset */ *inputBufferOffset = fixMax(sbrOffset, downsampledOffset); @@ -2091,7 +2198,7 @@ INT sbrEncoder_Init( bail: /* Restore input settings */ - *sampleRate = inputSampleRate; + *coreSampleRate = inputSampleRate; *frameLength = coreFrameLength; *numChannels = inputChannels; *coreBandwidth = inputBandWidth; @@ -2104,8 +2211,8 @@ INT sbrEncoder_EncodeFrame( HANDLE_SBR_ENCODER hSbrEncoder, INT_PCM *samples, UINT timeInStride, - UINT sbrDataBits[(6)], - UCHAR sbrData[(6)][MAX_PAYLOAD_SIZE] + UINT sbrDataBits[(8)], + UCHAR sbrData[(8)][MAX_PAYLOAD_SIZE] ) { INT error; @@ -2129,8 +2236,8 @@ sbrEncoder_EncodeFrame( HANDLE_SBR_ENCODER hSbrEncoder, } } - if ( (hSbrEncoder->lfeChIdx!=-1) && (hSbrEncoder->fTimeDomainDownsampling) ) - { + if ( ( hSbrEncoder->lfeChIdx!=-1) && (hSbrEncoder->downSampleFactor > 1) ) + { /* lfe downsampler */ INT nOutSamples; FDKaacEnc_Downsample(&hSbrEncoder->lfeDownSampler, @@ -2140,7 +2247,9 @@ sbrEncoder_EncodeFrame( HANDLE_SBR_ENCODER hSbrEncoder, samples + hSbrEncoder->downsampledOffset + hSbrEncoder->lfeChIdx, &nOutSamples, hSbrEncoder->nChannels); - } /* lfe downsampler */ + + + } return 0; } |