/*************************** Fraunhofer IIS FDK Tools *********************** (C) Copyright Fraunhofer IIS (2004) All Rights Reserved Please be advised that this software and/or program delivery is Confidential Information of Fraunhofer and subject to and covered by the Fraunhofer IIS Software Evaluation Agreement between Google Inc. and Fraunhofer effective and in full force since March 1, 2012. You may use this software and/or program only under the terms and conditions described in the above mentioned Fraunhofer IIS Software Evaluation Agreement. Any other and/or further use requires a separate agreement. $Id$ Author(s): Andreas Ehret Description: SBR encoder top level processing. This software and/or program is protected by copyright law and international treaties. Any reproduction or distribution of this software and/or program, or any portion of it, may result in severe civil and criminal penalties, and will be prosecuted to the maximum extent possible under law. ******************************************************************************/ #include "sbr_encoder.h" #include "sbr_ram.h" #include "sbr_rom.h" #include "sbrenc_freq_sca.h" #include "env_bit.h" #include "cmondata.h" #include "sbr_misc.h" #include "sbr.h" #include "qmf.h" #include "ps_main.h" #include "psenc_hybrid.h" #define SBRENCODER_LIB_VL0 3 #define SBRENCODER_LIB_VL1 2 #define SBRENCODER_LIB_VL2 0 /***************************************************************************/ /* * SBR Delay balancing definitions. */ /* input buffer (1ch) |------------ 1537 -------------|-----|---------- 2048 -------------| (core2sbr delay ) ds (read, core and ds area) */ #define DOWN_SMPL_FAC (2) #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 */ /* 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) /* 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 */ /* 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) /***************************************************************************/ #define INVALID_TABLE_IDX -1 /***************************************************************************/ /*! \brief Selects the SBR tuning settings to use dependent on number of channels, bitrate, sample rate and core coder \return Index to the appropriate table ****************************************************************************/ static INT getSbrTuningTableIndex(UINT bitrate, /*! the total bitrate in bits/sec */ UINT numChannels,/*! the number of channels for the core coder */ UINT sampleRate, /*! the sampling rate of the core coder */ AUDIO_OBJECT_TYPE core ) { int i, paramSetTop; if (core == AOT_ER_AAC_ELD) { paramSetTop = SBRENC_TUNING_SIZE; i = 126; } else { paramSetTop = 126; i = 0; } for (; i < paramSetTop ; i++) { if (numChannels == sbrTuningTable [i].numChannels) { if ((sampleRate == sbrTuningTable [i].sampleRate) && (bitrate >= sbrTuningTable [i].bitrateFrom) && (bitrate < sbrTuningTable [i].bitrateTo)) { return i ; } } } return INVALID_TABLE_IDX; } /***************************************************************************/ /*! \brief Selects the PS tuning settings to use dependent on bitrate and core coder \return Index to the appropriate table ****************************************************************************/ static INT getPsTuningTableIndex(UINT bitrate){ INT i, paramSets = sizeof (psTuningTable) / sizeof (psTuningTable [0]); for (i = 0 ; i < paramSets ; i++) { if ((bitrate >= psTuningTable [i].bitrateFrom) && (bitrate < psTuningTable [i].bitrateTo)) { return i ; } } return INVALID_TABLE_IDX; } /***************************************************************************/ /*! \brief tells us, if for the given coreCoder, bitrate, number of channels and input sampling rate an SBR setting is available. If yes, it tells us also the core sampling rate we would need to run with \return a flag indicating success: yes (1) or no (0) ****************************************************************************/ 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 ) { INT idx = INVALID_TABLE_IDX; UINT sampleRateCore; if (sampleRateInput < 16000) return 0; if (bitrate==0) { /* map vbr quality to bitrate */ if (vbrMode < 30) bitrate = 24000; else if (vbrMode < 40) bitrate = 28000; else if (vbrMode < 60) bitrate = 32000; else if (vbrMode < 75) bitrate = 40000; else bitrate = 48000; bitrate *= numOutputChannels; } /* try DOWN_SMPL_FAC of the input sampling rate */ sampleRateCore = sampleRateInput/DOWN_SMPL_FAC; idx = getSbrTuningTableIndex(bitrate, numOutputChannels, sampleRateCore, core); return (idx == INVALID_TABLE_IDX ? 0 : 1); } /***************************************************************************/ /*! \brief Adjusts the SBR settings according to the chosen core coder settings which are accessible via config->codecSettings \return A flag indicating success: yes (1) or no (0) ****************************************************************************/ 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 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*/ UINT useSpeechConfig, /*!< adapt tuning parameters for speech ? */ UINT lcsMode, /*! the low complexity stereo mode */ UINT bParametricStereo, /*!< use parametric stereo */ AUDIO_OBJECT_TYPE core) /* Core audio codec object type */ { INT idx = INVALID_TABLE_IDX; UINT sampleRate; /* set the codec settings */ config->codecSettings.bitRate = bitRate; config->codecSettings.nChannels = numChannels; config->codecSettings.sampleFreq = fsCore; config->codecSettings.transFac = transFac; config->codecSettings.standardBitrate = standardBitrate; sampleRate = fsCore * DOWN_SMPL_FAC; if (bitRate==0) { /* map vbr quality to bitrate */ if (vbrMode < 30) bitRate = 24000; else if (vbrMode < 40) bitRate = 28000; else if (vbrMode < 60) bitRate = 32000; else if (vbrMode < 75) bitRate = 40000; else bitRate = 48000; bitRate *= numChannels; /* fix to enable mono vbrMode<40 @ 44.1 of 48kHz */ if (numChannels==1) { if (sampleRate==44100 || sampleRate==48000) { if (vbrMode<40) bitRate = 32000; } } } idx = getSbrTuningTableIndex(bitRate,numChannels,fsCore, core); if (idx != INVALID_TABLE_IDX) { config->startFreq = sbrTuningTable[idx].startFreq ; config->stopFreq = sbrTuningTable[idx].stopFreq ; if (useSpeechConfig) { config->startFreq = sbrTuningTable[idx].startFreqSpeech; config->stopFreq = sbrTuningTable[idx].stopFreqSpeech; } config->sbr_noise_bands = sbrTuningTable[idx].numNoiseBands ; if (core == AOT_ER_AAC_ELD) config->init_amp_res_FF = SBR_AMP_RES_1_5; config->noiseFloorOffset= sbrTuningTable[idx].noiseFloorOffset; config->ana_max_level = sbrTuningTable[idx].noiseMaxLevel ; config->stereoMode = sbrTuningTable[idx].stereoMode ; config->freqScale = sbrTuningTable[idx].freqScale ; /* adjust usage of parametric coding dependent on bitrate and speech config flag */ if (useSpeechConfig) config->parametricCoding = 0; if (core == AOT_ER_AAC_ELD) { if (bitRate < 28000) config->init_amp_res_FF = SBR_AMP_RES_3_0; config->SendHeaderDataTime = -1; } if (numChannels == 1) { if (bitRate < 16000) { config->parametricCoding = 0; } } else { if (bitRate < 20000) { config->parametricCoding = 0; } } config->useSpeechConfig = useSpeechConfig; /* PS settings */ config->bParametricStereo = bParametricStereo; return 1 ; } else { return 0 ; } } /***************************************************************************** functionname: FDKsbrEnc_InitializeSbrDefaults description: initializes the SBR confifuration returns: error status input: - core codec type, - fac of SBR to core frame length, - core frame length output: initialized SBR configuration *****************************************************************************/ static UINT FDKsbrEnc_InitializeSbrDefaults (sbrConfigurationPtr config, INT coreSbrFrameLenFac, UINT codecGranuleLen) { if ( (coreSbrFrameLenFac != 2) || (codecGranuleLen*coreSbrFrameLenFac > QMF_CHANNELS*QMF_MAX_TIME_SLOTS) ) return(1); config->SendHeaderDataTime = 1000; config->useWaveCoding = 0; config->crcSbr = 0; config->dynBwSupported = 1; config->tran_thr = 13000; config->parametricCoding = 1; config->sbrFrameSize = codecGranuleLen * coreSbrFrameLenFac; /* sbr default parameters */ config->sbr_data_extra = 0; config->amp_res = SBR_AMP_RES_3_0 ; config->tran_fc = 0 ; config->tran_det_mode = 1 ; config->spread = 1 ; config->stat = 0 ; config->e = 1 ; config->deltaTAcrossFrames = 1 ; config->dF_edge_1stEnv = FL2FXCONST_DBL(0.3f) ; config->dF_edge_incr = FL2FXCONST_DBL(0.3f) ; config->sbr_invf_mode = INVF_SWITCHED; config->sbr_xpos_mode = XPOS_LC; config->sbr_xpos_ctrl = SBR_XPOS_CTRL_DEFAULT; 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 they are included in the tuning table */ config->stereoMode = SBR_SWITCH_LRC; config->ana_max_level = 6; config->noiseFloorOffset = 0; config->startFreq = 5; /* 5.9 respectively 6.0 kHz at fs = 44.1/48 kHz */ config->stopFreq = 9; /* 16.2 respectively 16.8 kHz at fs = 44.1/48 kHz */ /* header_extra_1 */ config->freqScale = SBR_FREQ_SCALE_DEFAULT; config->alterScale = SBR_ALTER_SCALE_DEFAULT; config->sbr_noise_bands = SBR_NOISE_BANDS_DEFAULT; /* header_extra_2 */ config->sbr_limiter_bands = SBR_LIMITER_BANDS_DEFAULT; config->sbr_limiter_gains = SBR_LIMITER_GAINS_DEFAULT; config->sbr_interpol_freq = SBR_INTERPOL_FREQ_DEFAULT; config->sbr_smoothing_length = SBR_SMOOTHING_LENGTH_DEFAULT; return 1; } /***************************************************************************** functionname: DeleteEnvChannel description: frees memory of one SBR channel returns: - input: handle of channel output: released handle *****************************************************************************/ static void deleteEnvChannel (HANDLE_ENV_CHANNEL hEnvCut) { if (hEnvCut) { /* deleteFrameInfoGenerator (&hEnvCut->SbrEnvFrame); */ /* deleteSbrCodeEnvelope (&hEnvCut->sbrCodeEnvelope); */ /* deleteSbrCodeEnvelope (&hEnvCut->sbrCodeNoiseFloor); */ FDKsbrEnc_DeleteTonCorrParamExtr(&hEnvCut->TonCorr); FDKsbrEnc_deleteExtractSbrEnvelope (&hEnvCut->sbrExtractEnvelope); //FreeRam_EnvChannel(phEnvCut); } } /***************************************************************************** functionname: sbrEncoder_ChannelClose description: close the channel coding handle returns: input: phSbrChannel output: *****************************************************************************/ static void sbrEncoder_ChannelClose(HANDLE_SBR_CHANNEL hSbrChannel) { if (hSbrChannel != NULL) { deleteEnvChannel (&hSbrChannel->hEnvChannel); } } /***************************************************************************** functionname: sbrEncoder_ElementClose description: close the channel coding handle returns: input: phSbrChannel output: *****************************************************************************/ static void sbrEncoder_ElementClose(HANDLE_SBR_ELEMENT *phSbrElement) { HANDLE_SBR_ELEMENT hSbrElement = *phSbrElement; if (hSbrElement!=NULL) { if (hSbrElement->sbrConfigData.v_k_master) FreeRam_Sbr_v_k_master(&hSbrElement->sbrConfigData.v_k_master); if (hSbrElement->sbrConfigData.freqBandTable[LO]) FreeRam_Sbr_freqBandTableLO(&hSbrElement->sbrConfigData.freqBandTable[LO]); if (hSbrElement->sbrConfigData.freqBandTable[HI]) FreeRam_Sbr_freqBandTableHI(&hSbrElement->sbrConfigData.freqBandTable[HI]); FreeRam_SbrElement(phSbrElement); } return ; } void sbrEncoder_Close (HANDLE_SBR_ENCODER *phSbrEncoder) { HANDLE_SBR_ENCODER hSbrEncoder = *phSbrEncoder; if (hSbrEncoder != NULL) { int el, ch; for (el=0; el<(6); el++) { if (hSbrEncoder->sbrElement[el]!=NULL) { sbrEncoder_ElementClose(&hSbrEncoder->sbrElement[el]); } } /* Close sbr Channels */ for (ch=0; ch<(6); ch++) { if (hSbrEncoder->pSbrChannel[ch]) { sbrEncoder_ChannelClose(hSbrEncoder->pSbrChannel[ch]); FreeRam_SbrChannel(&hSbrEncoder->pSbrChannel[ch]); } if (hSbrEncoder->QmfAnalysis[ch].FilterStates) FreeRam_Sbr_QmfStatesAnalysis((FIXP_QAS**)&hSbrEncoder->QmfAnalysis[ch].FilterStates); } if (hSbrEncoder->hPsEncConfig) FreeRam_PsEncConf(&hSbrEncoder->hPsEncConfig); if (hSbrEncoder->hParametricStereo) PSEnc_Destroy(&hSbrEncoder->hParametricStereo); if (hSbrEncoder->qmfSynthesisPS.FilterStates) FreeRam_PsQmfStatesSynthesis((FIXP_DBL**)&hSbrEncoder->qmfSynthesisPS.FilterStates); /* Release Overlay */ FreeRam_SbrDynamic_RAM((FIXP_DBL**)&hSbrEncoder->pSBRdynamic_RAM); FreeRam_SbrEncoder(phSbrEncoder); } } /***************************************************************************** functionname: updateFreqBandTable description: updates vk_master returns: - input: config handle output: error info *****************************************************************************/ static INT updateFreqBandTable(HANDLE_SBR_CONFIG_DATA sbrConfigData, HANDLE_SBR_HEADER_DATA sbrHeaderData, INT noQmfChannels) { INT k0, k2; if(FDKsbrEnc_FindStartAndStopBand(sbrConfigData->sampleFreq, noQmfChannels, sbrHeaderData->sbr_start_frequency, sbrHeaderData->sbr_stop_frequency, sbrHeaderData->sampleRateMode, &k0, &k2)) return(1); 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)) return(1); 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; return (0); } /***************************************************************************** functionname: resetEnvChannel description: resets parameters and allocates memory returns: error status input: output: hEnv *****************************************************************************/ static INT resetEnvChannel (HANDLE_SBR_CONFIG_DATA sbrConfigData, HANDLE_SBR_HEADER_DATA sbrHeaderData, HANDLE_ENV_CHANNEL hEnv) { /* note !!! hEnv->encEnvData.noOfnoisebands will be updated later in function FDKsbrEnc_extractSbrEnvelope !!!*/ hEnv->TonCorr.sbrNoiseFloorEstimate.noiseBands = sbrHeaderData->sbr_noise_bands; if(FDKsbrEnc_ResetTonCorrParamExtr(&hEnv->TonCorr, sbrConfigData->xposCtrlSwitch, sbrConfigData->freqBandTable[HI][0], sbrConfigData->v_k_master, sbrConfigData->num_Master, sbrConfigData->sampleFreq, sbrConfigData->freqBandTable, sbrConfigData->nSfb, sbrConfigData->noQmfBands)) return(1); hEnv->sbrCodeNoiseFloor.nSfb[LO] = hEnv->TonCorr.sbrNoiseFloorEstimate.noNoiseBands; hEnv->sbrCodeNoiseFloor.nSfb[HI] = hEnv->TonCorr.sbrNoiseFloorEstimate.noNoiseBands; hEnv->sbrCodeEnvelope.nSfb[LO] = sbrConfigData->nSfb[LO]; hEnv->sbrCodeEnvelope.nSfb[HI] = sbrConfigData->nSfb[HI]; hEnv->encEnvData.noHarmonics = sbrConfigData->nSfb[HI]; hEnv->sbrCodeEnvelope.upDate = 0; hEnv->sbrCodeNoiseFloor.upDate = 0; return (0); } /* ****************************** FDKsbrEnc_SbrGetXOverFreq ******************************/ /** * @fn * @brief calculates the closest possible crossover frequency * @return the crossover frequency SBR accepts * */ static INT FDKsbrEnc_SbrGetXOverFreq(HANDLE_SBR_ELEMENT hEnv, /*!< handle to SBR encoder instance */ INT xoverFreq) /*!< from core coder suggested crossover frequency */ { INT band; INT lastDiff, newDiff; INT cutoffSb; UCHAR *RESTRICT pVKMaster = hEnv->sbrConfigData.v_k_master; /* Check if there is a matching cutoff frequency in the master table */ cutoffSb = (4*xoverFreq * hEnv->sbrConfigData.noQmfBands / hEnv->sbrConfigData.sampleFreq + 1)>>1; lastDiff = cutoffSb; for (band = 0; band < hEnv->sbrConfigData.num_Master; band++) { newDiff = fixp_abs((INT)pVKMaster[band] - cutoffSb); if(newDiff >= lastDiff) { band--; break; } lastDiff = newDiff; } return ((pVKMaster[band] * hEnv->sbrConfigData.sampleFreq/hEnv->sbrConfigData.noQmfBands+1)>>1); } /***************************************************************************** functionname: FDKsbrEnc_EnvEncodeFrame description: performs the sbr envelope calculation for one element returns: input: output: *****************************************************************************/ INT FDKsbrEnc_EnvEncodeFrame(HANDLE_SBR_ENCODER hEnvEncoder, int iElement, INT_PCM *samples, /*!< time samples, always interleaved */ UINT timeInStride, /*!< time buffer channel interleaving stride */ UINT *sbrDataBits, /*!< Size of SBR payload */ UCHAR *sbrData, /*!< SBR payload */ int clearOutput /*!< Do not consider any input signal */ ) { HANDLE_SBR_ELEMENT hSbrElement = hEnvEncoder->sbrElement[iElement]; FDK_CRCINFO crcInfo; INT crcReg; INT ch; INT band; INT cutoffSb; INT newXOver; if (hEnvEncoder == NULL) return -1; hSbrElement = hEnvEncoder->sbrElement[iElement]; if (hSbrElement == NULL) return -1; /* header bitstream handling */ HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData = &hSbrElement->sbrBitstreamData; INT psHeaderActive = 0; sbrBitstreamData->HeaderActive = 0; /* Anticipate PS header because of internal PS bitstream delay in order to be in sync with SBR header. */ if ( sbrBitstreamData->CountSendHeaderData==(sbrBitstreamData->NrSendHeaderData-1) ) { psHeaderActive = 1; } /* Signal SBR header to be written into bitstream */ if ( sbrBitstreamData->CountSendHeaderData==0 ) { sbrBitstreamData->HeaderActive = 1; } /* Increment header interval counter */ if (sbrBitstreamData->NrSendHeaderData == 0) { sbrBitstreamData->CountSendHeaderData = 1; } else { if (sbrBitstreamData->CountSendHeaderData >= 0) { sbrBitstreamData->CountSendHeaderData++; sbrBitstreamData->CountSendHeaderData %= sbrBitstreamData->NrSendHeaderData; } } if (hSbrElement->CmonData.dynBwEnabled ) { INT i; for ( i = 4; i > 0; i-- ) hSbrElement->dynXOverFreqDelay[i] = hSbrElement->dynXOverFreqDelay[i-1]; hSbrElement->dynXOverFreqDelay[0] = hSbrElement->CmonData.dynXOverFreqEnc; if (hSbrElement->dynXOverFreqDelay[1] > hSbrElement->dynXOverFreqDelay[2]) newXOver = hSbrElement->dynXOverFreqDelay[2]; else newXOver = hSbrElement->dynXOverFreqDelay[1]; /* has the crossover frequency changed? */ if ( hSbrElement->sbrConfigData.dynXOverFreq != newXOver ) { /* get corresponding master band */ cutoffSb = ((4* newXOver * hSbrElement->sbrConfigData.noQmfBands / hSbrElement->sbrConfigData.sampleFreq)+1)>>1; for ( band = 0; band < hSbrElement->sbrConfigData.num_Master; band++ ) { if ( cutoffSb == hSbrElement->sbrConfigData.v_k_master[band] ) break; } FDK_ASSERT( band < hSbrElement->sbrConfigData.num_Master ); hSbrElement->sbrConfigData.dynXOverFreq = newXOver; hSbrElement->sbrHeaderData.sbr_xover_band = band; hSbrElement->sbrBitstreamData.HeaderActive=1; psHeaderActive = 1; /* ps header is one frame delayed */ /* update vk_master table */ if(updateFreqBandTable(&hSbrElement->sbrConfigData, &hSbrElement->sbrHeaderData, hSbrElement->sbrConfigData.noQmfBands)) return(1); /* reset SBR channels */ INT nEnvCh = hSbrElement->sbrConfigData.nChannels; for ( ch = 0; ch < nEnvCh; ch++ ) { if(resetEnvChannel (&hSbrElement->sbrConfigData, &hSbrElement->sbrHeaderData, &hSbrElement->sbrChannel[ch]->hEnvChannel)) return(1); } } } /* allocate space for dummy header and crc */ crcReg = FDKsbrEnc_InitSbrBitstream(&hSbrElement->CmonData, hSbrElement->payloadDelayLine[hEnvEncoder->nBitstrDelay], MAX_PAYLOAD_SIZE*sizeof(UCHAR), &crcInfo, hSbrElement->sbrConfigData.sbrSyntaxFlags); INT error = noError; /* Temporal Envelope Data */ SBR_FRAME_TEMP_DATA _fData; SBR_FRAME_TEMP_DATA *fData = &_fData; SBR_ENV_TEMP_DATA eData[MAX_NUM_CHANNELS]; /* Init Temporal Envelope Data */ { int i; FDKmemclear(&eData[0], sizeof(SBR_ENV_TEMP_DATA)); FDKmemclear(&eData[1], sizeof(SBR_ENV_TEMP_DATA)); FDKmemclear(fData, sizeof(SBR_FRAME_TEMP_DATA)); for(i=0; ires[i] = FREQ_RES_HIGH; } if (!clearOutput) { /* * Transform audio data into QMF domain */ for(ch = 0; ch < hSbrElement->sbrConfigData.nChannels; ch++) { HANDLE_ENV_CHANNEL h_envChan = &hSbrElement->sbrChannel[ch]->hEnvChannel; HANDLE_SBR_EXTRACT_ENVELOPE sbrExtrEnv = &h_envChan->sbrExtractEnvelope; if(hSbrElement->elInfo.fParametricStereo == 0) { C_ALLOC_SCRATCH_START(qmfWorkBuffer, FIXP_DBL, QMF_CHANNELS*2); QMF_SCALE_FACTOR tmpScale; FIXP_DBL **pQmfReal, **pQmfImag; /* Obtain pointers to QMF buffers. */ pQmfReal = sbrExtrEnv->rBuffer+sbrExtrEnv->rBufferWriteOffset; pQmfImag = sbrExtrEnv->iBuffer+sbrExtrEnv->rBufferWriteOffset; qmfAnalysisFiltering( hSbrElement->hQmfAnalysis[ch], pQmfReal, pQmfImag, &tmpScale, samples + hSbrElement->elInfo.ChannelIndex[ch], timeInStride, qmfWorkBuffer ); C_ALLOC_SCRATCH_END(qmfWorkBuffer, FIXP_DBL, QMF_CHANNELS*2); h_envChan->qmfScale = tmpScale.lb_scale + 7; } /* fParametricStereo == 0 */ /* Parametric Stereo processing */ if(hSbrElement->elInfo.fParametricStereo) { int psCh; /* Parametric Stereo QMF buffer preprocessing: copy previous qmf data down */ UpdatePSQmfData_second(hEnvEncoder->hParametricStereo); for (psCh = 0; psCh<2; psCh ++) { C_ALLOC_SCRATCH_START(qmfWorkBuffer, FIXP_DBL, QMF_CHANNELS*2); HANDLE_PS_QMF_DATA hPsQmfData = hEnvEncoder->hParametricStereo->hPsChannelData[psCh]->hPsQmfData; QMF_SCALE_FACTOR tmpScale; qmfAnalysisFiltering( hSbrElement->hQmfAnalysis[psCh], hPsQmfData->rQmfData + hPsQmfData->bufferWriteOffset, hPsQmfData->iQmfData + hPsQmfData->bufferWriteOffset, &tmpScale, samples + hSbrElement->elInfo.ChannelIndex[psCh], timeInStride, qmfWorkBuffer ); C_ALLOC_SCRATCH_END(qmfWorkBuffer, FIXP_DBL, QMF_CHANNELS*2); hEnvEncoder->hParametricStereo->hPsChannelData[psCh]->psQmfScale = -tmpScale.lb_scale; } /* Limit Parametric Stereo to one instance */ FDK_ASSERT(ch == 0); if(error == noError){ /* parametric stereo processing: - input: o left and right time domain samples - processing: o stereo qmf analysis o stereo hybrid analysis o ps parameter extraction o downmix + hybrid synthesis - output: o downmixed qmf data is written to sbrExtrEnv->rBuffer and sbrExtrEnv->iBuffer */ SCHAR qmfScale; error = FDKsbrEnc_PSEnc_ParametricStereoProcessing( hEnvEncoder->hParametricStereo, sbrExtrEnv->rBuffer, sbrExtrEnv->iBuffer, sbrExtrEnv->rBufferWriteOffset, samples + hSbrElement->elInfo.ChannelIndex[ch], &hEnvEncoder->qmfSynthesisPS, &qmfScale, psHeaderActive ); if (noError != error) { error = handBack(error); } h_envChan->qmfScale = (int)qmfScale; } } /* if (hEnvEncoder->hParametricStereo) */ /* Extract Envelope relevant things from QMF data */ FDKsbrEnc_extractSbrEnvelope1( &hSbrElement->sbrConfigData, &hSbrElement->sbrHeaderData, &hSbrElement->sbrBitstreamData, h_envChan, &hSbrElement->CmonData, &eData[ch], fData ); } /* hEnvEncoder->sbrConfigData.nChannels */ } /* Process Envelope relevant things and calculate envelope data and write payload */ FDKsbrEnc_extractSbrEnvelope2( &hSbrElement->sbrConfigData, &hSbrElement->sbrHeaderData, (hSbrElement->elInfo.fParametricStereo) ? hEnvEncoder->hParametricStereo : NULL, &hSbrElement->sbrBitstreamData, &hSbrElement->sbrChannel[0]->hEnvChannel, &hSbrElement->sbrChannel[1]->hEnvChannel, &hSbrElement->CmonData, eData, fData, clearOutput ); /* format payload, calculate crc */ FDKsbrEnc_AssembleSbrBitstream(&hSbrElement->CmonData, &crcInfo, crcReg, hSbrElement->sbrConfigData.sbrSyntaxFlags); /* save new payload, set to zero length if greater than MAX_PAYLOAD_SIZE */ hSbrElement->payloadDelayLineSize[hEnvEncoder->nBitstrDelay] = FDKgetValidBits(&hSbrElement->CmonData.sbrBitbuf); if(hSbrElement->payloadDelayLineSize[hEnvEncoder->nBitstrDelay] > (MAX_PAYLOAD_SIZE<<3)) hSbrElement->payloadDelayLineSize[hEnvEncoder->nBitstrDelay]=0; /* While filling the Delay lines, sbrData is NULL */ if (sbrData) { *sbrDataBits = hSbrElement->payloadDelayLineSize[0]; FDKmemcpy(sbrData, hSbrElement->payloadDelayLine[0], (hSbrElement->payloadDelayLineSize[0]+7)>>3); } /*******************************/ if (hEnvEncoder->fTimeDomainDownsampling) { int ch; int nChannels = hSbrElement->sbrConfigData.nChannels; for (ch=0; ch < nChannels; ch++) { INT nOutSamples; FDKaacEnc_Downsample(&hSbrElement->sbrChannel[ch]->downSampler, samples + hSbrElement->elInfo.ChannelIndex[ch] + hEnvEncoder->bufferOffset, hSbrElement->sbrConfigData.frameSize, timeInStride, samples + hSbrElement->elInfo.ChannelIndex[ch], &nOutSamples, hEnvEncoder->nChannels); } } /* downsample */ return (0); } /***************************************************************************** functionname: createEnvChannel description: initializes parameters and allocates memory returns: error status input: output: hEnv *****************************************************************************/ static INT createEnvChannel (HANDLE_ENV_CHANNEL hEnv, INT channel ,UCHAR* dynamic_RAM ) { FDKmemclear(hEnv,sizeof (struct ENV_CHANNEL)); if ( FDKsbrEnc_CreateTonCorrParamExtr(&hEnv->TonCorr, channel) ) { return(1); } if ( FDKsbrEnc_CreateExtractSbrEnvelope (&hEnv->sbrExtractEnvelope, channel ,/*chan*/0 ,dynamic_RAM ) ) { return(1); } return 0; } /***************************************************************************** functionname: initEnvChannel description: initializes parameters returns: error status input: output: *****************************************************************************/ static INT initEnvChannel ( HANDLE_SBR_CONFIG_DATA sbrConfigData, HANDLE_SBR_HEADER_DATA sbrHeaderData, HANDLE_ENV_CHANNEL hEnv, sbrConfigurationPtr params, ULONG statesInitFlag ,INT chanInEl ,UCHAR* dynamic_RAM ) { int frameShift, tran_off=0; INT e; INT tran_fc; INT timeSlots, timeStep, startIndex; INT noiseBands[2] = { 3, 3 }; e = 1 << params->e; FDK_ASSERT(params->e >= 0); hEnv->encEnvData.freq_res_fixfix = 1; hEnv->fLevelProtect = 0; hEnv->encEnvData.ldGrid = (sbrConfigData->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) ? 1 : 0; hEnv->encEnvData.sbr_xpos_mode = (XPOS_MODE)params->sbr_xpos_mode; if (hEnv->encEnvData.sbr_xpos_mode == XPOS_SWITCHED) { /* no other type than XPOS_MDCT or XPOS_SPEECH allowed, but enable switching */ sbrConfigData->switchTransposers = TRUE; hEnv->encEnvData.sbr_xpos_mode = XPOS_MDCT; } else { sbrConfigData->switchTransposers = FALSE; } hEnv->encEnvData.sbr_xpos_ctrl = params->sbr_xpos_ctrl; /* extended data */ if(params->parametricCoding) { hEnv->encEnvData.extended_data = 1; } else { hEnv->encEnvData.extended_data = 0; } hEnv->encEnvData.extension_size = 0; startIndex = QMF_FILTER_PROTOTYPE_SIZE - sbrConfigData->noQmfBands; switch (params->sbrFrameSize) { case 2304: timeSlots = 18; break; case 2048: case 1024: timeSlots = 16; break; case 1920: case 960: timeSlots = 15; break; case 1152: timeSlots = 9; break; default: return (1); /* Illegal frame size */ } timeStep = sbrConfigData->noQmfSlots / timeSlots; if ( FDKsbrEnc_InitTonCorrParamExtr(params->sbrFrameSize, &hEnv->TonCorr, sbrConfigData, timeSlots, params->sbr_xpos_ctrl, params->ana_max_level, sbrHeaderData->sbr_noise_bands, params->noiseFloorOffset, params->useSpeechConfig) ) return(1); hEnv->encEnvData.noOfnoisebands = hEnv->TonCorr.sbrNoiseFloorEstimate.noNoiseBands; noiseBands[0] = hEnv->encEnvData.noOfnoisebands; noiseBands[1] = hEnv->encEnvData.noOfnoisebands; hEnv->encEnvData.sbr_invf_mode = (INVF_MODE)params->sbr_invf_mode; if (hEnv->encEnvData.sbr_invf_mode == INVF_SWITCHED) { hEnv->encEnvData.sbr_invf_mode = INVF_MID_LEVEL; hEnv->TonCorr.switchInverseFilt = TRUE; } else { hEnv->TonCorr.switchInverseFilt = FALSE; } tran_fc = params->tran_fc; if (tran_fc == 0) tran_fc = fixMin (5000, FDKsbrEnc_getSbrStartFreqRAW (sbrHeaderData->sbr_start_frequency,64,sbrConfigData->sampleFreq)); tran_fc = (tran_fc*4*sbrConfigData->noQmfBands/sbrConfigData->sampleFreq + 1)>>1; if (sbrConfigData->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) { frameShift = LD_PRETRAN_OFF; tran_off = LD_PRETRAN_OFF + FRAME_MIDDLE_SLOT_512LD*timeStep; } else { frameShift = 0; switch (params->sbrFrameSize) { /* 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; } } if ( FDKsbrEnc_InitExtractSbrEnvelope (&hEnv->sbrExtractEnvelope, sbrConfigData->noQmfSlots, sbrConfigData->noQmfBands, startIndex, timeSlots, timeStep, tran_off, statesInitFlag ,chanInEl ,dynamic_RAM ,sbrConfigData->sbrSyntaxFlags ) ) return(1); if(FDKsbrEnc_InitSbrCodeEnvelope (&hEnv->sbrCodeEnvelope, sbrConfigData->nSfb, params->deltaTAcrossFrames, params->dF_edge_1stEnv, params->dF_edge_incr)) return(1); if(FDKsbrEnc_InitSbrCodeEnvelope (&hEnv->sbrCodeNoiseFloor, noiseBands, params->deltaTAcrossFrames, 0,0)) return(1); sbrConfigData->initAmpResFF = params->init_amp_res_FF; if(FDKsbrEnc_InitSbrHuffmanTables (&hEnv->encEnvData, &hEnv->sbrCodeEnvelope, &hEnv->sbrCodeNoiseFloor, sbrHeaderData->sbr_amp_res)) return(1); FDKsbrEnc_initFrameInfoGenerator (&hEnv->SbrEnvFrame, params->spread, e, params->stat, timeSlots, hEnv->encEnvData.freq_res_fixfix ,hEnv->encEnvData.ldGrid ); if(FDKsbrEnc_InitSbrTransientDetector (&hEnv->sbrTransientDetector, sbrConfigData->frameSize, sbrConfigData->sampleFreq, params, tran_fc, sbrConfigData->noQmfSlots, sbrConfigData->noQmfBands, hEnv->sbrExtractEnvelope.YBufferWriteOffset, hEnv->sbrExtractEnvelope.YBufferSzShift, frameShift, tran_off )) return(1); sbrConfigData->xposCtrlSwitch = params->sbr_xpos_ctrl; hEnv->encEnvData.noHarmonics = sbrConfigData->nSfb[HI]; hEnv->encEnvData.addHarmonicFlag = 0; return (0); } INT sbrEncoder_Open( HANDLE_SBR_ENCODER *phSbrEncoder, INT nElements, INT nChannels, INT supportPS ) { INT i; INT errorStatus = 1; HANDLE_SBR_ENCODER hSbrEncoder = NULL; if (phSbrEncoder==NULL ) { goto bail; } hSbrEncoder = GetRam_SbrEncoder(); if (hSbrEncoder==NULL) { goto bail; } FDKmemclear(hSbrEncoder, sizeof(SBR_ENCODER)); hSbrEncoder->pSBRdynamic_RAM = (UCHAR*)GetRam_SbrDynamic_RAM(); hSbrEncoder->dynamicRam = hSbrEncoder->pSBRdynamic_RAM; for (i=0; isbrElement[i] = GetRam_SbrElement(i); if (hSbrEncoder->sbrElement[i]==NULL) { goto bail; } FDKmemclear(hSbrEncoder->sbrElement[i], sizeof(SBR_ELEMENT)); hSbrEncoder->sbrElement[i]->sbrConfigData.freqBandTable[LO] = GetRam_Sbr_freqBandTableLO(i); hSbrEncoder->sbrElement[i]->sbrConfigData.freqBandTable[HI] = GetRam_Sbr_freqBandTableHI(i); hSbrEncoder->sbrElement[i]->sbrConfigData.v_k_master = GetRam_Sbr_v_k_master(i); if ( (hSbrEncoder->sbrElement[i]->sbrConfigData.freqBandTable[LO]==NULL) || (hSbrEncoder->sbrElement[i]->sbrConfigData.freqBandTable[HI]==NULL) || (hSbrEncoder->sbrElement[i]->sbrConfigData.v_k_master==NULL) ) { goto bail; } } for (i=0; ipSbrChannel[i] = GetRam_SbrChannel(i); if (hSbrEncoder->pSbrChannel[i]==NULL) { goto bail; } if ( createEnvChannel(&hSbrEncoder->pSbrChannel[i]->hEnvChannel, i ,hSbrEncoder->dynamicRam ) ) { goto bail; } } for (i=0; iQmfAnalysis[i].FilterStates = GetRam_Sbr_QmfStatesAnalysis(i); if (hSbrEncoder->QmfAnalysis[i].FilterStates==NULL) { goto bail; } } if (supportPS) { hSbrEncoder->hPsEncConfig = GetRam_PsEncConf(); if (hSbrEncoder->hPsEncConfig==NULL) { goto bail; } if (PSEnc_Create(&hSbrEncoder->hParametricStereo)) { goto bail; } hSbrEncoder->qmfSynthesisPS.FilterStates = GetRam_PsQmfStatesSynthesis(); if (hSbrEncoder->qmfSynthesisPS.FilterStates==NULL) { goto bail; } } /* supportPS */ *phSbrEncoder = hSbrEncoder; errorStatus = 0; return errorStatus; bail: /* Close SBR encoder instance */ sbrEncoder_Close(&hSbrEncoder); return errorStatus; } static INT FDKsbrEnc_Reallocate( HANDLE_SBR_ENCODER hSbrEncoder, SBR_ELEMENT_INFO elInfo[(6)], const INT noElements) { INT totalCh = 0; INT totalQmf = 0; INT coreEl; INT el=-1; hSbrEncoder->lfeChIdx = -1; /* default value, until lfe found */ for (coreEl=0; coreEllfeChIdx = elInfo[coreEl].ChannelIndex[0]; } continue; } SBR_ELEMENT_INFO *pelInfo = &elInfo[coreEl]; HANDLE_SBR_ELEMENT hSbrElement = hSbrEncoder->sbrElement[el]; int ch; for ( ch = 0; ch < pelInfo->nChannelsInEl; ch++ ) { hSbrElement->sbrChannel[ch] = hSbrEncoder->pSbrChannel[totalCh]; totalCh++; } /* analysis QMF */ for ( ch = 0; ch < ((pelInfo->fParametricStereo)?2:pelInfo->nChannelsInEl); ch++ ) { hSbrElement->elInfo.ChannelIndex[ch] = pelInfo->ChannelIndex[ch]; hSbrElement->hQmfAnalysis[ch] = &hSbrEncoder->QmfAnalysis[totalQmf++]; } /* Copy Element info */ hSbrElement->elInfo.elType = pelInfo->elType; hSbrElement->elInfo.instanceTag = pelInfo->instanceTag; hSbrElement->elInfo.nChannelsInEl = pelInfo->nChannelsInEl; hSbrElement->elInfo.fParametricStereo = pelInfo->fParametricStereo; } /* coreEl */ return 0; } /***************************************************************************** functionname: FDKsbrEnc_EnvInit description: initializes parameters returns: error status input: output: hEnv *****************************************************************************/ static INT FDKsbrEnc_EnvInit ( HANDLE_SBR_ELEMENT hSbrElement, sbrConfigurationPtr params, INT *coreBandWith, AUDIO_OBJECT_TYPE aot, int nBitstrDelay, int nElement, ULONG statesInitFlag ,UCHAR *dynamic_RAM ) { UCHAR *bitstreamBuffer; int ch, i; if ((params->codecSettings.nChannels < 1) || (params->codecSettings.nChannels > MAX_NUM_CHANNELS)){ return(1); } /* initialize the encoder handle and structs*/ bitstreamBuffer = hSbrElement->payloadDelayLine[nBitstrDelay]; /* init and set syntax flags */ hSbrElement->sbrConfigData.sbrSyntaxFlags = 0; switch (aot) { case AOT_DRM_MPEG_PS: case AOT_DRM_SBR: hSbrElement->sbrConfigData.sbrSyntaxFlags |= SBR_SYNTAX_SCALABLE; hSbrElement->sbrConfigData.sbrSyntaxFlags |= SBR_SYNTAX_DRM_CRC; hSbrElement->sbrConfigData.sbrSyntaxFlags |= SBR_SYNTAX_CRC; break; case AOT_ER_AAC_ELD: hSbrElement->sbrConfigData.sbrSyntaxFlags |= SBR_SYNTAX_LOW_DELAY; break; default: break; } if (params->crcSbr) { hSbrElement->sbrConfigData.sbrSyntaxFlags |= SBR_SYNTAX_CRC; } hSbrElement->sbrConfigData.noQmfBands = QMF_CHANNELS; hSbrElement->sbrConfigData.noQmfSlots = params->sbrFrameSize/hSbrElement->sbrConfigData.noQmfBands; FDKinitBitStream(&hSbrElement->CmonData.sbrBitbuf, bitstreamBuffer, MAX_PAYLOAD_SIZE*sizeof(UCHAR), 0, BS_WRITER); /* now initialize sbrConfigData, sbrHeaderData and sbrBitstreamData, */ hSbrElement->sbrConfigData.nChannels = params->codecSettings.nChannels; if(params->codecSettings.nChannels == 2) hSbrElement->sbrConfigData.stereoMode = params->stereoMode; else hSbrElement->sbrConfigData.stereoMode = SBR_MONO; 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->sbrBitstreamData.CountSendHeaderData = 0; if (params->SendHeaderDataTime > 0 ) { hSbrElement->sbrBitstreamData.NrSendHeaderData = (INT)(params->SendHeaderDataTime * hSbrElement->sbrConfigData.sampleFreq / (1000 * hSbrElement->sbrConfigData.frameSize)); hSbrElement->sbrBitstreamData.NrSendHeaderData = fixMax(hSbrElement->sbrBitstreamData.NrSendHeaderData,1); } else { hSbrElement->sbrBitstreamData.NrSendHeaderData = 0; } hSbrElement->sbrHeaderData.sbr_data_extra = params->sbr_data_extra; hSbrElement->sbrBitstreamData.HeaderActive = 0; hSbrElement->sbrHeaderData.sbr_start_frequency = params->startFreq; hSbrElement->sbrHeaderData.sbr_stop_frequency = params->stopFreq; hSbrElement->sbrHeaderData.sbr_xover_band = 0; hSbrElement->sbrHeaderData.sbr_lc_stereo_mode = 0; /* data_extra */ if (params->sbr_xpos_ctrl!= SBR_XPOS_CTRL_DEFAULT) hSbrElement->sbrHeaderData.sbr_data_extra = 1; hSbrElement->sbrHeaderData.sbr_amp_res = (AMP_RES)params->amp_res; /* header_extra_1 */ hSbrElement->sbrHeaderData.freqScale = params->freqScale; hSbrElement->sbrHeaderData.alterScale = params->alterScale; hSbrElement->sbrHeaderData.sbr_noise_bands = params->sbr_noise_bands; hSbrElement->sbrHeaderData.header_extra_1 = 0; if ((params->freqScale != SBR_FREQ_SCALE_DEFAULT) || (params->alterScale != SBR_ALTER_SCALE_DEFAULT) || (params->sbr_noise_bands != SBR_NOISE_BANDS_DEFAULT)) { hSbrElement->sbrHeaderData.header_extra_1 = 1; } /* header_extra_2 */ hSbrElement->sbrHeaderData.sbr_limiter_bands = params->sbr_limiter_bands; hSbrElement->sbrHeaderData.sbr_limiter_gains = params->sbr_limiter_gains; if ((hSbrElement->sbrConfigData.sampleFreq > 48000) && (hSbrElement->sbrHeaderData.sbr_start_frequency >= 9)) { hSbrElement->sbrHeaderData.sbr_limiter_gains = SBR_LIMITER_GAINS_INFINITE; } hSbrElement->sbrHeaderData.sbr_interpol_freq = params->sbr_interpol_freq; hSbrElement->sbrHeaderData.sbr_smoothing_length = params->sbr_smoothing_length; hSbrElement->sbrHeaderData.header_extra_2 = 0; if ((params->sbr_limiter_bands != SBR_LIMITER_BANDS_DEFAULT) || (params->sbr_limiter_gains != SBR_LIMITER_GAINS_DEFAULT) || (params->sbr_interpol_freq != SBR_INTERPOL_FREQ_DEFAULT) || (params->sbr_smoothing_length != SBR_SMOOTHING_LENGTH_DEFAULT)) { hSbrElement->sbrHeaderData.header_extra_2 = 1; } /* other switches */ hSbrElement->sbrConfigData.useWaveCoding = params->useWaveCoding; hSbrElement->sbrConfigData.useParametricCoding = params->parametricCoding; /* init freq band table */ if(updateFreqBandTable(&hSbrElement->sbrConfigData, &hSbrElement->sbrHeaderData, hSbrElement->sbrConfigData.noQmfBands)) { return(1); } /* now create envelope ext and QMF for each available channel */ for ( ch = 0; ch < hSbrElement->sbrConfigData.nChannels; ch++ ) { if ( initEnvChannel(&hSbrElement->sbrConfigData, &hSbrElement->sbrHeaderData, &hSbrElement->sbrChannel[ch]->hEnvChannel, params, statesInitFlag ,ch ,dynamic_RAM ) ) { return(1); } } /* nChannels */ /* reset and intialize analysis qmf */ for ( ch = 0; ch < ((hSbrElement->elInfo.fParametricStereo)?2:hSbrElement->sbrConfigData.nChannels); ch++ ) { int err; UINT qmfFlags = (hSbrElement->sbrConfigData.sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) ? QMF_FLAG_CLDFB : 0; if (statesInitFlag) qmfFlags &= ~QMF_FLAG_KEEP_STATES; else qmfFlags |= QMF_FLAG_KEEP_STATES; err = qmfInitAnalysisFilterBank( hSbrElement->hQmfAnalysis[ch], (FIXP_QAS*)hSbrElement->hQmfAnalysis[ch]->FilterStates, hSbrElement->sbrConfigData.noQmfSlots, hSbrElement->sbrConfigData.noQmfBands, hSbrElement->sbrConfigData.noQmfBands, hSbrElement->sbrConfigData.noQmfBands, qmfFlags ); } /* */ hSbrElement->CmonData.xOverFreq = hSbrElement->sbrConfigData.xOverFreq; hSbrElement->CmonData.dynBwEnabled = (params->dynBwSupported && params->dynBwEnabled); hSbrElement->CmonData.dynXOverFreqEnc = FDKsbrEnc_SbrGetXOverFreq( hSbrElement, hSbrElement->CmonData.xOverFreq); for ( i = 0; i < 5; i++ ) hSbrElement->dynXOverFreqDelay[i] = hSbrElement->CmonData.dynXOverFreqEnc; hSbrElement->CmonData.sbrNumChannels = hSbrElement->sbrConfigData.nChannels; hSbrElement->sbrConfigData.dynXOverFreq = hSbrElement->CmonData.xOverFreq; /* Update Bandwith to be passed to the core encoder */ *coreBandWith = hSbrElement->CmonData.xOverFreq; return(0); } INT sbrEncoder_GetInBufferSize(int noChannels) { INT temp; temp = (1024*DOWN_SMPL_FAC); temp += 1024 + MAX_SAMPLE_DELAY; temp *= noChannels; temp *= sizeof(INT_PCM); return temp; } /* * Encode Dummy SBR payload frames to fill the delay lines. */ static INT FDKsbrEnc_DelayCompensation ( HANDLE_SBR_ENCODER hEnvEnc, INT_PCM *timeBuffer ) { int n, el; for (n=hEnvEnc->nBitstrDelay; n>0; n--) { for (el=0; elnoElements; el++) { if (FDKsbrEnc_EnvEncodeFrame( hEnvEnc, el, timeBuffer + hEnvEnc->downsampledOffset, hEnvEnc->sbrElement[el]->sbrConfigData.nChannels, NULL, NULL, 1 )) return -1; sbrEncoder_UpdateBuffers(hEnvEnc, timeBuffer); } } return 0; } 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_ERROR_INFO errorInfo = noError; sbrConfiguration sbrConfig[(6)]; INT error = 0; INT lowestBandwidth; /* Save input parameters */ INT inputSampleRate = *sampleRate; int coreFrameLength = *frameLength; int inputBandWidth = *coreBandwidth; int inputChannels = *numChannels; int downsampledOffset = 0; int sbrOffset = 0; int downsamplerDelay = 0; int downsample = 0; int nBitstrDelay = 0; int lowestSbrStartFreq, lowestSbrStopFreq; 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) ) { usePs = 1; } if ( (*aot==AOT_ER_AAC_ELD) ) { lowDelay = 1; } else if ( (*aot==AOT_ER_AAC_LD) ) { error = 1; goto bail; } /* Parametric Stereo */ if ( usePs ) { if ( *numChannels == 2 && noElements == 1) { /* Override Element type in case of Parametric stereo */ elInfo[0].elType = ID_SCE; elInfo[0].fParametricStereo = 1; elInfo[0].nChannelsInEl = 1; /* 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; } } /* usePs */ /* check whether SBR setting is available for the current encoder configuration (bitrate, samplerate) */ { int delayDiff = 0; int el, coreEl; /* Check if every element config is feasible */ for (coreEl=0; coreEl 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 * the downsampled audio, counteracting to the purpose of delay balancing. */ while ( delayDiff > 0 ) { /* Encoder delay increases */ *delay += coreFrameLength*DOWN_SMPL_FAC; /* Add one frame delay to SBR path */ delayDiff -= coreFrameLength*DOWN_SMPL_FAC; nBitstrDelay += 1; } } else { *delay += fixp_abs(delayDiff); } if (delayDiff < 0) { /* 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; sbrOffset = 0; } else { /* Delay SBR input */ if ( delayDiff > (int)coreFrameLength*DOWN_SMPL_FAC ) { /* Do bitstream frame-wise delay balancing if we have more than SBR framelength samples delay difference */ delayDiff -= coreFrameLength*DOWN_SMPL_FAC; 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; /* Open SBR elements */ el = -1; lowestSbrStartFreq = lowestSbrStopFreq = 9999; lowestBandwidth = 99999; /* Loop through each core encoder element and get a matching SBR element config */ for (coreEl=0; coreElnoElements = el+1; FDKsbrEnc_Reallocate(hSbrEncoder, elInfo, noElements); for (el=0; elnoElements; el++) { int bandwidth = *coreBandwidth; /* Use lowest common bandwidth */ sbrConfig[el].startFreq = lowestSbrStartFreq; sbrConfig[el].stopFreq = lowestSbrStopFreq; /* initialize SBR element, and get core bandwidth */ error = FDKsbrEnc_EnvInit(hSbrEncoder->sbrElement[el], &sbrConfig[el], &bandwidth, *aot, nBitstrDelay, el, statesInitFlag ,hSbrEncoder->dynamicRam ); if (error != 0) { goto bail; } /* Get lowest core encoder bandwidth to be returned later. */ lowestBandwidth = fixMin(lowestBandwidth, bandwidth); } /* second element loop */ /* Initialize a downsampler for each channel in each SBR element */ if (hSbrEncoder->fTimeDomainDownsampling) { for (el=0; elnoElements; el++) { HANDLE_SBR_ELEMENT hSbrEl = hSbrEncoder->sbrElement[el]; INT Wc, ch; /* Calculated required normalized cutoff frequency (Wc = 1.0 -> lowestBandwidth = inputSampleRate/2) */ Wc = (2*lowestBandwidth)*1000 / inputSampleRate; for (ch=0; chelInfo.nChannelsInEl; ch++) { FDKaacEnc_InitDownsampler (&hSbrEl->sbrChannel[ch]->downSampler, Wc, DOWN_SMPL_FAC); } 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); /* Add the resampler additional delay to get the final delay and buffer offset values. */ if (sbrOffset > 0 || downsampledOffset <= ((downsamplerDelay * (*numChannels))/DOWN_SMPL_FAC)) { sbrOffset += (downsamplerDelay - downsampledOffset) * (*numChannels) ; *delay += downsamplerDelay - downsampledOffset; downsampledOffset = 0; } else { downsampledOffset -= (downsamplerDelay * (*numChannels))/DOWN_SMPL_FAC; sbrOffset = 0; } hSbrEncoder->inputDataDelay = downsamplerDelay; } /* Assign core encoder Bandwidth */ *coreBandwidth = lowestBandwidth; /* Estimate sbr bitrate, 2.5 kBit/s per sbr channel */ hSbrEncoder->estimateBitrate += 2500 * (*numChannels); /* initialize parametric stereo */ if (usePs) { FDK_ASSERT(hSbrEncoder->noElements == 1); INT psTuningTableIdx = getPsTuningTableIndex(elInfo[0].bitRate); //sbrConfig.codecSettings.bitRate); hSbrEncoder->hPsEncConfig->frameSize = *frameLength; //sbrConfig.sbrFrameSize; hSbrEncoder->hPsEncConfig->qmfFilterMode = 0; hSbrEncoder->hPsEncConfig->sbrPsDelay = 0; /* tuning parameters */ if (psTuningTableIdx != INVALID_TABLE_IDX) { hSbrEncoder->hPsEncConfig->nStereoBands = psTuningTable[psTuningTableIdx].nStereoBands; hSbrEncoder->hPsEncConfig->maxEnvelopes = psTuningTable[psTuningTableIdx].nEnvelopes; hSbrEncoder->hPsEncConfig->iidQuantErrorThreshold = (FIXP_DBL)psTuningTable[psTuningTableIdx].iidQuantErrorThreshold; /* 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 * hSbrEncoder->hPsEncConfig->nStereoBands * hSbrEncoder->hPsEncConfig->maxEnvelopes) / hSbrEncoder->frameSize)); } else { error = ERROR(CDI, "Invalid ps tuning table index."); goto bail; } qmfInitSynthesisFilterBank(&hSbrEncoder->qmfSynthesisPS, (FIXP_DBL*)hSbrEncoder->qmfSynthesisPS.FilterStates, hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfSlots, hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfBands>>1, hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfBands>>1, hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfBands>>1, (statesInitFlag) ? 0 : QMF_FLAG_KEEP_STATES); if(errorInfo == noError){ /* update delay */ hSbrEncoder->hPsEncConfig->sbrPsDelay = FDKsbrEnc_GetEnvEstDelay(&hSbrEncoder->sbrElement[0]->sbrChannel[0]->hEnvChannel.sbrExtractEnvelope); if(noError != (errorInfo = PSEnc_Init( hSbrEncoder->hParametricStereo, hSbrEncoder->hPsEncConfig, hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfSlots, hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfBands ,hSbrEncoder->dynamicRam ))) { errorInfo = handBack(errorInfo); } } } hSbrEncoder->downsampledOffset = downsampledOffset; hSbrEncoder->downmixSize = coreFrameLength*(*numChannels); hSbrEncoder->bufferOffset = sbrOffset; /* Delay Compensation: fill bitstream delay buffer with zero input signal */ if ( hSbrEncoder->nBitstrDelay > 0 ) { error = FDKsbrEnc_DelayCompensation (hSbrEncoder, inputBuffer); if (error != 0) goto bail; } /* Set Output frame length */ *frameLength = coreFrameLength*DOWN_SMPL_FAC; /* Input buffer offset */ *inputBufferOffset = fixMax(sbrOffset, downsampledOffset); } return error; bail: /* Restore input settings */ *sampleRate = inputSampleRate; *frameLength = coreFrameLength; *numChannels = inputChannels; *coreBandwidth = inputBandWidth; return error; } INT sbrEncoder_EncodeFrame( HANDLE_SBR_ENCODER hSbrEncoder, INT_PCM *samples, UINT timeInStride, UINT sbrDataBits[(6)], UCHAR sbrData[(6)][MAX_PAYLOAD_SIZE] ) { INT error; int el; for (el=0; elnoElements; el++) { if (hSbrEncoder->sbrElement[el] != NULL) { error = FDKsbrEnc_EnvEncodeFrame( hSbrEncoder, el, samples + hSbrEncoder->downsampledOffset, timeInStride, &sbrDataBits[el], sbrData[el], 0 ); if (error) return error; } } if ( (hSbrEncoder->lfeChIdx!=-1) && (hSbrEncoder->fTimeDomainDownsampling) ) { INT nOutSamples; FDKaacEnc_Downsample(&hSbrEncoder->lfeDownSampler, samples + hSbrEncoder->downsampledOffset + hSbrEncoder->bufferOffset + hSbrEncoder->lfeChIdx, hSbrEncoder->frameSize, timeInStride, samples + hSbrEncoder->downsampledOffset + hSbrEncoder->lfeChIdx, &nOutSamples, hSbrEncoder->nChannels); } /* lfe downsampler */ return 0; } INT sbrEncoder_UpdateBuffers( HANDLE_SBR_ENCODER hSbrEncoder, INT_PCM *timeBuffer ) { if ( hSbrEncoder->downsampledOffset > 0 ) { /* Move delayed downsampled data */ FDKmemcpy ( timeBuffer, timeBuffer + hSbrEncoder->downmixSize, sizeof(INT_PCM) * (hSbrEncoder->downsampledOffset) ); } else { /* Move delayed input data */ FDKmemcpy ( timeBuffer, timeBuffer + hSbrEncoder->nChannels * hSbrEncoder->frameSize, sizeof(INT_PCM) * hSbrEncoder->bufferOffset ); } if ( hSbrEncoder->nBitstrDelay > 0 ) { int el; for (el=0; elnoElements; el++) { FDKmemmove ( hSbrEncoder->sbrElement[el]->payloadDelayLine[0], hSbrEncoder->sbrElement[el]->payloadDelayLine[1], sizeof(UCHAR) * (hSbrEncoder->nBitstrDelay*MAX_PAYLOAD_SIZE) ); FDKmemmove( &hSbrEncoder->sbrElement[el]->payloadDelayLineSize[0], &hSbrEncoder->sbrElement[el]->payloadDelayLineSize[1], sizeof(UINT) * (hSbrEncoder->nBitstrDelay) ); } } return 0; } INT sbrEncoder_GetEstimateBitrate(HANDLE_SBR_ENCODER hSbrEncoder) { INT estimateBitrate = 0; if(hSbrEncoder) { estimateBitrate += hSbrEncoder->estimateBitrate; } return estimateBitrate; } INT sbrEncoder_GetInputDataDelay(HANDLE_SBR_ENCODER hSbrEncoder) { INT delay = -1; if(hSbrEncoder) { delay = hSbrEncoder->inputDataDelay; } return delay; } INT sbrEncoder_GetLibInfo( LIB_INFO *info ) { int i; if (info == NULL) { return -1; } /* search for next free tab */ for (i = 0; i < FDK_MODULE_LAST; i++) { if (info[i].module_id == FDK_NONE) break; } if (i == FDK_MODULE_LAST) { return -1; } info += i; info->module_id = FDK_SBRENC; info->version = LIB_VERSION(SBRENCODER_LIB_VL0, SBRENCODER_LIB_VL1, SBRENCODER_LIB_VL2); LIB_VERSION_STRING(info); info->build_date = __DATE__; info->build_time = __TIME__; info->title = "SBR Encoder"; /* Set flags */ info->flags = 0 | CAPF_SBR_HQ | CAPF_SBR_PS_MPEG ; /* End of flags */ return 0; }