/* ----------------------------------------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------------------------------------- */ /*************************** Fraunhofer IIS FDK Tools *********************** Author(s): Andreas Ehret, Tobias Chalupka Description: SBR encoder top level processing. ******************************************************************************/ #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" #define SBRENCODER_LIB_VL0 3 #define SBRENCODER_LIB_VL1 3 #define SBRENCODER_LIB_VL2 12 /***************************************************************************/ /* * SBR Delay balancing definitions. */ /* input buffer (1ch) |------------ 1537 -------------|-----|---------- 2048 -------------| (core2sbr delay ) ds (read, core and ds area) */ #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 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,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 (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: 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 */ /***************************************************************************/ #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 ****************************************************************************/ #define DISTANCE_CEIL_VALUE 5000000 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, UINT *pBitRateClosest ) { int i, bitRateClosestLowerIndex=-1, bitRateClosestUpperIndex=-1, found = 0; UINT bitRateClosestUpper = 0, bitRateClosestLower=DISTANCE_CEIL_VALUE; #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=0; i < sbrTuningTableSize ; i++) { if ( isForThisCore(i) ) /* tuning table is for this core codec */ { 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 (pBitRateClosest != NULL) { /* 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; if (bitRateClosestLowerIndex >= 0) { distanceLower = sbrTuningTable [bitRateClosestLowerIndex].bitrateFrom - bitrate; } if (bitRateClosestUpperIndex >= 0) { distanceUpper = bitrate - sbrTuningTable [bitRateClosestUpperIndex].bitrateTo; } if ( distanceUpper < distanceLower ) { *pBitRateClosest = bitRateClosestUpper; } else { *pBitRateClosest = bitRateClosestLower; } } else { *pBitRateClosest = 0; } } 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, UINT *pBitRateClosest){ INT i, paramSets = sizeof (psTuningTable) / sizeof (psTuningTable [0]); int bitRateClosestLowerIndex=-1, bitRateClosestUpperIndex=-1; UINT bitRateClosestUpper = 0, bitRateClosestLower=DISTANCE_CEIL_VALUE; for (i = 0 ; i < paramSets ; i++) { if ((bitrate >= psTuningTable [i].bitrateFrom) && (bitrate < psTuningTable [i].bitrateTo)) { return i ; } else { if ( psTuningTable [i].bitrateFrom > bitrate ) { if (psTuningTable [i].bitrateFrom < bitRateClosestLower) { bitRateClosestLower = psTuningTable [i].bitrateFrom; bitRateClosestLowerIndex = i; } } if ( psTuningTable [i].bitrateTo <= bitrate ) { if (psTuningTable [i].bitrateTo > bitRateClosestUpper) { bitRateClosestUpper = psTuningTable [i].bitrateTo-1; bitRateClosestUpperIndex = i; } } } } if (pBitRateClosest != NULL) { int distanceUpper=DISTANCE_CEIL_VALUE, distanceLower=DISTANCE_CEIL_VALUE; if (bitRateClosestLowerIndex >= 0) { distanceLower = sbrTuningTable [bitRateClosestLowerIndex].bitrateFrom - bitrate; } if (bitRateClosestUpperIndex >= 0) { distanceUpper = bitrate - sbrTuningTable [bitRateClosestUpperIndex].bitrateTo; } if ( distanceUpper < distanceLower ) { *pBitRateClosest = bitRateClosestUpper; } else { *pBitRateClosest = bitRateClosestLower; } } 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; } /***************************************************************************/ /*! \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] */ UINT sampleRateCore, /*! the core's sampling rate */ AUDIO_OBJECT_TYPE core ) { INT idx = INVALID_TABLE_IDX; 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; } idx = getSbrTuningTableIndex(bitrate, numOutputChannels, sampleRateCore, core, NULL); 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 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*/ 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; /* set the core codec settings */ config->codecSettings.bitRate = bitRate; config->codecSettings.nChannels = numChannels; config->codecSettings.sampleFreq = sampleRateCore; config->codecSettings.transFac = transFac; config->codecSettings.standardBitrate = standardBitrate; if (bitRate < 28000) { config->threshold_AmpRes_FF_m = (FIXP_DBL)MAXVAL_DBL; config->threshold_AmpRes_FF_e = 7; } else if (bitRate >= 28000 && bitRate <= 48000) { /* The float threshold is 75 0.524288f is fractional part of RELAXATION, the quotaMatrix and therefore tonality are scaled by this 2/3 is because the original implementation divides the tonality values by 3, here it's divided by 2 128 compensates the necessary shiftfactor of 7 */ config->threshold_AmpRes_FF_m = FL2FXCONST_DBL(75.0f*0.524288f/(2.0f/3.0f)/128.0f); config->threshold_AmpRes_FF_e = 7; } else if (bitRate > 48000) { config->threshold_AmpRes_FF_m = FL2FXCONST_DBL(0); config->threshold_AmpRes_FF_e = 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 *= numChannels; /* fix to enable mono vbrMode<40 @ 44.1 of 48kHz */ if (numChannels==1) { if (sampleRateSbr==44100 || sampleRateSbr==48000) { if (vbrMode<40) bitRate = 32000; } } } idx = getSbrTuningTableIndex(bitRate,numChannels,sampleRateCore, core, NULL); 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; } /* 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; config->noiseFloorOffset= sbrTuningTable[idx].noiseFloorOffset; config->ana_max_level = sbrTuningTable[idx].noiseMaxLevel ; config->stereoMode = sbrTuningTable[idx].stereoMode ; config->freqScale = sbrTuningTable[idx].freqScale ; if (numChannels == 1) { /* stereo case */ switch (core) { case AOT_AAC_LC: if (bitRate <= (useSpeechConfig?24000U:20000U)) { config->freq_res_fixfix[0] = FREQ_RES_LOW; /* set low frequency resolution for non-split frames */ config->freq_res_fixfix[1] = FREQ_RES_LOW; /* set low frequency resolution for split frames */ } break; case AOT_ER_AAC_ELD: if (bitRate < 36000) config->freq_res_fixfix[1] = FREQ_RES_LOW; /* set low frequency resolution for split frames */ if (bitRate < 26000) { config->freq_res_fixfix[0] = FREQ_RES_LOW; /* set low frequency resolution for non-split frames */ config->fResTransIsLow = 1; /* for transient frames, set low frequency resolution */ } break; default: break; } } else { /* stereo case */ switch (core) { case AOT_AAC_LC: if (bitRate <= 28000) { config->freq_res_fixfix[0] = FREQ_RES_LOW; /* set low frequency resolution for non-split frames */ config->freq_res_fixfix[1] = FREQ_RES_LOW; /* set low frequency resolution for split frames */ } break; case AOT_ER_AAC_ELD: if (bitRate < 72000) { config->freq_res_fixfix[1] = FREQ_RES_LOW; /* set low frequency resolution for split frames */ } if (bitRate < 52000) { config->freq_res_fixfix[0] = FREQ_RES_LOW; /* set low frequency resolution for non-split frames */ config->fResTransIsLow = 1; /* for transient frames, set low frequency resolution */ } break; default: break; } if (bitRate <= 28000) { /* additionally restrict frequency resolution in FIXFIX frames to further reduce SBR payload size */ config->freq_res_fixfix[0] = FREQ_RES_LOW; config->freq_res_fixfix[1] = FREQ_RES_LOW; } } /* 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, - factor of SBR to core frame length, - core frame length output: initialized SBR configuration *****************************************************************************/ static UINT FDKsbrEnc_InitializeSbrDefaults (sbrConfigurationPtr config, INT downSampleFactor, UINT codecGranuleLen ,const INT isLowDelay ) { if ( (downSampleFactor < 1 || downSampleFactor > 2) || (codecGranuleLen*downSampleFactor > QMF_CHANNELS*QMF_MAX_TIME_SLOTS) ) return(0); /* error */ config->SendHeaderDataTime = 1000; config->useWaveCoding = 0; config->crcSbr = 0; config->dynBwSupported = 1; if (isLowDelay) config->tran_thr = 6000; else config->tran_thr = 13000; config->parametricCoding = 1; config->sbrFrameSize = codecGranuleLen * downSampleFactor; config->downSampleFactor = downSampleFactor; /* 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; /* 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 */ config->freq_res_fixfix[0] = FREQ_RES_HIGH; /* non-split case */ config->freq_res_fixfix[1] = FREQ_RES_HIGH; /* split case */ config->fResTransIsLow = 0; /* for transient frames, set variable frequency resolution according to freqResTable */ /* 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) { FDKsbrEnc_DeleteTonCorrParamExtr(&hEnvCut->TonCorr); FDKsbrEnc_deleteExtractSbrEnvelope (&hEnvCut->sbrExtractEnvelope); } } /***************************************************************************** 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<(8); el++) { if (hSbrEncoder->sbrElement[el]!=NULL) { sbrEncoder_ElementClose(&hSbrEncoder->sbrElement[el]); } } /* Close sbr Channels */ for (ch=0; ch<(8); 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->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, const INT downSampleFactor ) { INT 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 ) ) 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 ) ) 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 / sbrConfigData->noQmfBands+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 = NULL; 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, hEnvEncoder->downSampleFactor )) 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); /* 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) { QMF_SCALE_FACTOR tmpScale; FIXP_DBL **pQmfReal, **pQmfImag; C_AALLOC_SCRATCH_START(qmfWorkBuffer, FIXP_DBL, QMF_CHANNELS*2) /* Obtain pointers to QMF buffers. */ pQmfReal = sbrExtrEnv->rBuffer; pQmfImag = sbrExtrEnv->iBuffer; qmfAnalysisFiltering( hSbrElement->hQmfAnalysis[ch], pQmfReal, pQmfImag, &tmpScale, samples + hSbrElement->elInfo.ChannelIndex[ch], timeInStride, qmfWorkBuffer ); h_envChan->qmfScale = tmpScale.lb_scale + 7; C_AALLOC_SCRATCH_END(qmfWorkBuffer, FIXP_DBL, QMF_CHANNELS*2) } /* fParametricStereo == 0 */ /* Parametric Stereo processing */ if (hSbrElement->elInfo.fParametricStereo) { INT error = noError; /* 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; INT_PCM* pSamples[2] = {samples + hSbrElement->elInfo.ChannelIndex[0],samples + hSbrElement->elInfo.ChannelIndex[1]}; error = FDKsbrEnc_PSEnc_ParametricStereoProcessing( hEnvEncoder->hParametricStereo, pSamples, timeInStride, hSbrElement->hQmfAnalysis, sbrExtrEnv->rBuffer, sbrExtrEnv->iBuffer, 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[0] = params->freq_res_fixfix[0]; hEnv->encEnvData.freq_res_fixfix[1] = params->freq_res_fixfix[1]; hEnv->encEnvData.fResTransIsLow = params->fResTransIsLow; 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: case 512: timeSlots = 16; break; case 1920: case 960: case 480: 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,params->codecSettings.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 (timeSlots) { /* The factor of 2 is by definition. */ 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, 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.fResTransIsLow, hEnv->encEnvData.ldGrid ); if(sbrConfigData->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) { INT bandwidth_qmf_slot = (sbrConfigData->sampleFreq>>1) / (sbrConfigData->noQmfBands); if(FDKsbrEnc_InitSbrFastTransientDetector( &hEnv->sbrFastTransientDetector, sbrConfigData->noQmfSlots, bandwidth_qmf_slot, sbrConfigData->noQmfBands, sbrConfigData->freqBandTable[0][0] )) return(1); } /* The transient detector has to be initialized also if the fast transient detector was active, because the values from the transient detector structure are used. */ if(FDKsbrEnc_InitSbrTransientDetector (&hEnv->sbrTransientDetector, sbrConfigData->sbrSyntaxFlags, 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) { 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[(8)], 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, const int headerPeriod, ULONG statesInitFlag, int fTimeDomainDownsampling ,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_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>>(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); /* 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; hSbrElement->sbrConfigData.sampleFreq = params->downSampleFactor * params->codecSettings.sampleFreq; hSbrElement->sbrBitstreamData.CountSendHeaderData = 0; if (params->SendHeaderDataTime > 0 ) { if (headerPeriod==-1) { hSbrElement->sbrBitstreamData.NrSendHeaderData = (INT)(params->SendHeaderDataTime * hSbrElement->sbrConfigData.sampleFreq / (1000 * hSbrElement->sbrConfigData.frameSize)); 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; } 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; hSbrElement->sbrConfigData.thresholdAmpResFF_m = params->threshold_AmpRes_FF_m; hSbrElement->sbrConfigData.thresholdAmpResFF_e = params->threshold_AmpRes_FF_e; /* init freq band table */ if(updateFreqBandTable(&hSbrElement->sbrConfigData, &hSbrElement->sbrHeaderData, params->downSampleFactor )) { 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 ); if (0!=err) { return err; } } /* */ 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 = (2048); 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; } UINT sbrEncoder_LimitBitRate(UINT bitRate, UINT numChannels, UINT coreSampleRate, AUDIO_OBJECT_TYPE aot) { UINT newBitRate; INT index; FDK_ASSERT(numChannels > 0 && numChannels <= 2); if (aot == AOT_PS) { if (numChannels == 2) { index = getPsTuningTableIndex(bitRate, &newBitRate); if (index == INVALID_TABLE_IDX) { bitRate = newBitRate; } /* Set numChannels to 1 because for PS we need a SBR SCE (mono) element. */ numChannels = 1; } else { return 0; } } index = getSbrTuningTableIndex(bitRate, numChannels, coreSampleRate, aot, &newBitRate); if (index != INVALID_TABLE_IDX) { newBitRate = bitRate; } 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[(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[(8)]; INT error = 0; INT lowestBandwidth; /* Save input parameters */ INT inputSampleRate = *coreSampleRate; int coreFrameLength = *frameLength; int inputBandWidth = *coreBandwidth; int inputChannels = *numChannels; int downsampledOffset = 0; int sbrOffset = 0; int downsamplerDelay = 0; int timeDomainDownsample = 0; int nBitstrDelay = 0; int highestSbrStartFreq, highestSbrStopFreq; int lowDelay = 0; int usePs = 0; /* check whether SBR setting is available for the current encoder configuration (bitrate, samplerate) */ if (!sbrEncoder_IsSingleRatePossible(aot)) { *downSampleFactor = 2; } if ( aot==AOT_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 { error = 1; goto bail; } } /* usePs */ /* 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; /* 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 * *downSampleFactor; /* Add one frame delay to SBR path */ delayDiff -= coreFrameLength * *downSampleFactor; } 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. */ FDK_ASSERT(*downSampleFactor>0 && *downSampleFactor<=2); downsampledOffset = (delayDiff*(*numChannels))>>(*downSampleFactor-1); sbrOffset = 0; } else { /* Delay SBR input */ if ( delayDiff > (int)coreFrameLength * (int)*downSampleFactor ) { /* Do bitstream frame-wise delay balancing if we have more than SBR framelength samples delay difference */ delayDiff -= coreFrameLength * *downSampleFactor; nBitstrDelay = 1; } /* Multiply input offset by input channels */ sbrOffset = delayDiff*(*numChannels); downsampledOffset = 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; highestSbrStartFreq = highestSbrStopFreq = 0; 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 = highestSbrStartFreq; sbrConfig[el].stopFreq = highestSbrStopFreq; /* initialize SBR element, and get core bandwidth */ error = FDKsbrEnc_EnvInit(hSbrEncoder->sbrElement[el], &sbrConfig[el], &bandwidth, aot, nBitstrDelay, el, headerPeriod, statesInitFlag, hSbrEncoder->fTimeDomainDownsampling ,hSbrEncoder->dynamicRam ); if (error != 0) { error = 2; 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, *downSampleFactor); FDK_ASSERT (hSbrEl->sbrChannel[ch]->downSampler.delay <=MAX_DS_FILTER_DELAY); } downsamplerDelay = hSbrEl->sbrChannel[0]->downSampler.delay; } /* third element loop */ /* lfe */ 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))>>(*downSampleFactor-1))) { sbrOffset += (downsamplerDelay - downsampledOffset) * (*numChannels) ; *delay += downsamplerDelay - downsampledOffset; downsampledOffset = 0; } else { downsampledOffset -= (downsamplerDelay * (*numChannels))>>(*downSampleFactor-1); 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) { PSENC_CONFIG psEncConfig; FDK_ASSERT(hSbrEncoder->noElements == 1); INT psTuningTableIdx = getPsTuningTableIndex(elInfo[0].bitRate, NULL); psEncConfig.frameSize = coreFrameLength; //sbrConfig.sbrFrameSize; psEncConfig.qmfFilterMode = 0; psEncConfig.sbrPsDelay = 0; /* tuning parameters */ if (psTuningTableIdx != INVALID_TABLE_IDX) { psEncConfig.nStereoBands = psTuningTable[psTuningTableIdx].nStereoBands; psEncConfig.maxEnvelopes = psTuningTable[psTuningTableIdx].nEnvelopes; psEncConfig.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 += ( (((*coreSampleRate) * 5 * psEncConfig.nStereoBands * psEncConfig.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 */ psEncConfig.sbrPsDelay = FDKsbrEnc_GetEnvEstDelay(&hSbrEncoder->sbrElement[0]->sbrChannel[0]->hEnvChannel.sbrExtractEnvelope); if(noError != (errorInfo = PSEnc_Init( hSbrEncoder->hParametricStereo, &psEncConfig, hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfSlots, hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfBands ,hSbrEncoder->dynamicRam ))) { 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->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 * *downSampleFactor; /* Input buffer offset */ *inputBufferOffset = fixMax(sbrOffset, downsampledOffset); } return error; bail: /* Restore input settings */ *coreSampleRate = inputSampleRate; *frameLength = coreFrameLength; *numChannels = inputChannels; *coreBandwidth = inputBandWidth; return error; } INT sbrEncoder_EncodeFrame( HANDLE_SBR_ENCODER hSbrEncoder, INT_PCM *samples, UINT timeInStride, UINT sbrDataBits[(8)], UCHAR sbrData[(8)][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->downSampleFactor > 1) ) { /* lfe downsampler */ INT nOutSamples; FDKaacEnc_Downsample(&hSbrEncoder->lfeDownSampler, samples + hSbrEncoder->downsampledOffset + hSbrEncoder->bufferOffset + hSbrEncoder->lfeChIdx, hSbrEncoder->frameSize, timeInStride, samples + hSbrEncoder->downsampledOffset + hSbrEncoder->lfeChIdx, &nOutSamples, hSbrEncoder->nChannels); } 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); #ifdef __ANDROID__ info->build_date = ""; info->build_time = ""; #else info->build_date = __DATE__; info->build_time = __TIME__; #endif info->title = "SBR Encoder"; /* Set flags */ info->flags = 0 | CAPF_SBR_HQ | CAPF_SBR_PS_MPEG ; /* End of flags */ return 0; }