/frameworks/av/media/libstagefright/codecs/aacenc/inc/ |
H A D | stat_bits.h | 31 Word16 nChannels,
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H A D | channel_map.h | 29 Word16 InitElementInfo (Word16 nChannels, ELEMENT_INFO* elInfo);
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H A D | sf_estim.h | 37 const Word16 nChannels); 45 const Word16 nChannels);
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H A D | bitenc.h | 33 Word16 nChannels; member in struct:BITSTREAMENCODER_INIT
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H A D | line_pe.h | 61 const Word16 nChannels, 70 const Word16 nChannels);
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H A D | qc_main.h | 32 Word16 QCOutNew(QC_OUT *hQC, Word16 nChannels, VO_MEM_OPERATOR *pMemOP); 51 Word16 nChannels,
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H A D | adj_thr.h | 50 const Word16 nChannels,
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H A D | block_switch.h | 59 const Word32 bitRate, const Word16 nChannels);
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/frameworks/av/media/libstagefright/codecs/common/include/ |
H A D | voAAC.h | 47 short nChannels; /*! number of channels on input (1,2) */ member in struct:__anon658
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/frameworks/av/media/libeffects/testlibs/ |
H A D | AudioPeakingFilter.cpp | 44 AudioPeakingFilter::AudioPeakingFilter(int nChannels, int sampleRate) argument 45 : mBiquad(nChannels, sampleRate) { 46 configure(nChannels, sampleRate); 50 void AudioPeakingFilter::configure(int nChannels, int sampleRate) { argument 53 mBiquad.configure(nChannels, sampleRate);
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H A D | AudioShelvingFilter.cpp | 49 AudioShelvingFilter::AudioShelvingFilter(ShelfType type, int nChannels, argument 52 mBiquad(nChannels, sampleRate) { 53 configure(nChannels, sampleRate); 56 void AudioShelvingFilter::configure(int nChannels, int sampleRate) { argument 59 mBiquad.configure(nChannels, sampleRate);
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H A D | AudioShelvingFilter.h | 49 // nChannels Number of input/output channels (interlaced). 51 AudioShelvingFilter(ShelfType type, int nChannels, int sampleRate); 55 // nChannels Number of input/output channels (interlaced). 57 void configure(int nChannels, int sampleRate); 93 // frameCount * nChannels interlaced samples. Processing can be done
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H A D | AudioEqualizer.h | 69 // nChannels Number of input/output channels (interlaced). 80 int nChannels, 87 // nChannels Number of input/output channels (interlaced). 89 void configure(int nChannels, int sampleRate); 180 // frameCount * nChannels interlaced samples. Processing can be done 231 // nChannels Number of input/output channels (interlaced). 240 AudioEqualizer(void * pMem, int nBands, int nChannels, int sampleRate,
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H A D | AudioEqualizer.cpp | 39 int nChannels, int sampleRate, 42 ALOGV("AudioEqualizer::CreateInstance(pMem=%p, nBands=%d, nChannels=%d, " 44 pMem, nBands, nChannels, sampleRate, nPresets); 54 return new (pMem) AudioEqualizer(pMem, nBands, nChannels, sampleRate, 58 void AudioEqualizer::configure(int nChannels, int sampleRate) { argument 59 ALOGV("AudioEqualizer::configure(nChannels=%d, sampleRate=%d)", nChannels, 61 mpLowShelf->configure(nChannels, sampleRate); 63 mpPeakingFilters[i].configure(nChannels, sampleRate); 65 mpHighShelf->configure(nChannels, sampleRat 38 CreateInstance(void * pMem, int nBands, int nChannels, int sampleRate, const PresetConfig * presets, int nPresets) argument 287 AudioEqualizer(void * pMem, int nBands, int nChannels, int sampleRate, bool ownMem, const PresetConfig * presets, int nPresets) argument [all...] |
H A D | AudioBiquadFilter.h | 43 // nChannels Number of input/output channels. 45 AudioBiquadFilter(int nChannels, int sampleRate); 48 // nChannels Number of input/output channels. 50 void configure(int nChannels, int sampleRate); 75 // in The input buffer. Should be of size frameCount * nChannels. 76 // out The output buffer. Should be of size frameCount * nChannels.
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H A D | AudioPeakingFilter.h | 42 // nChannels Number of input/output channels (interlaced). 44 AudioPeakingFilter(int nChannels, int sampleRate); 48 // nChannels Number of input/output channels (interlaced). 50 void configure(int nChannels, int sampleRate); 99 // frameCount * nChannels interlaced samples. Processing can be done
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H A D | AudioFormatAdapter.h | 54 // nChannels Number of input and output channels. The adapter does not do 59 void configure(T & processor, int nChannels, uint8_t pcmFormat, argument 62 mNumChannels = nChannels; 65 mMaxSamplesPerCall = bufSize / nChannels;
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H A D | AudioBiquadFilter.cpp | 28 AudioBiquadFilter::AudioBiquadFilter(int nChannels, int sampleRate) { argument 29 configure(nChannels, sampleRate); 33 void AudioBiquadFilter::configure(int nChannels, int sampleRate) { argument 34 assert(nChannels > 0 && nChannels <= MAX_CHANNELS); 36 mNumChannels = nChannels;
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/frameworks/native/include/media/openmax/ |
H A D | OMX_Audio.h | 181 OMX_U32 nChannels; /**< Number of channels (e.g. 2 for stereo) */ member in struct:OMX_AUDIO_PARAM_PCMMODETYPE 226 OMX_U32 nChannels; /**< Number of channels */ member in struct:OMX_AUDIO_PARAM_MP3TYPE 297 OMX_U32 nChannels; /**< Number of channels */ member in struct:OMX_AUDIO_PARAM_AACPROFILETYPE 320 OMX_U32 nChannels; /**< Number of channels */ member in struct:OMX_AUDIO_PARAM_VORBISTYPE 349 OMX_U32 nChannels; /**< Number of channels */ member in struct:OMX_AUDIO_PARAM_FLACTYPE 386 OMX_U16 nChannels; /**< Number of channels */ member in struct:OMX_AUDIO_PARAM_WMATYPE 419 OMX_U32 nChannels; /**< Number of channels */ member in struct:OMX_AUDIO_PARAM_RATYPE 445 OMX_U32 nChannels; /**< Number of channels */ member in struct:OMX_AUDIO_PARAM_SBCTYPE 464 OMX_U32 nChannels; /**< Number of channels in the data stream (not member in struct:OMX_AUDIO_PARAM_ADPCMTYPE 489 OMX_U32 nChannels; /**< Numbe member in struct:OMX_AUDIO_PARAM_G723TYPE 517 OMX_U32 nChannels; /**< Number of channels in the data stream (not member in struct:OMX_AUDIO_PARAM_G726TYPE 541 OMX_U32 nChannels; /**< Number of channels in the data stream (not member in struct:OMX_AUDIO_PARAM_G729TYPE 617 OMX_U32 nChannels; /**< Number of channels */ member in struct:OMX_AUDIO_PARAM_AMRTYPE 660 OMX_U32 nChannels; /**< Number of channels in the data stream (not member in struct:OMX_AUDIO_PARAM_TDMAFRTYPE 673 OMX_U32 nChannels; /**< Number of channels in the data stream (not member in struct:OMX_AUDIO_PARAM_TDMAEFRTYPE 686 OMX_U32 nChannels; /**< Number of channels in the data stream (not member in struct:OMX_AUDIO_PARAM_PDCFRTYPE 699 OMX_U32 nChannels; /**< Number of channels in the data stream (not member in struct:OMX_AUDIO_PARAM_PDCEFRTYPE 711 OMX_U32 nChannels; /**< Number of channels in the data stream (not member in struct:OMX_AUDIO_PARAM_PDCHRTYPE 738 OMX_U32 nChannels; /**< Number of channels in the data stream (not member in struct:OMX_AUDIO_PARAM_QCELP8TYPE 754 OMX_U32 nChannels; /**< Number of channels in the data stream (not member in struct:OMX_AUDIO_PARAM_QCELP13TYPE 768 OMX_U32 nChannels; /**< Number of channels in the data stream (not member in struct:OMX_AUDIO_PARAM_EVRCTYPE 786 OMX_U32 nChannels; /**< Number of channels in the data stream (not member in struct:OMX_AUDIO_PARAM_SMVTYPE [all...] |
/frameworks/av/media/libstagefright/codecs/aacenc/src/ |
H A D | adj_thr.c | 72 const Word16 nChannels) 76 for (ch=0; ch<nChannels; ch++) { 97 const Word16 nChannels) 105 for (ch=0; ch<nChannels; ch++) { 167 const Word16 nChannels, 174 for (ch=0; ch<nChannels; ch++) { 199 for(ch=0; ch<nChannels; ch++) { 259 if (nChannels == 2) { 299 for(ch=0; ch<nChannels; ch++) { 333 const Word16 nChannels) 70 calcThreshExp(Word32 thrExp[MAX_CHANNELS][MAX_GROUPED_SFB], PSY_OUT_CHANNEL psyOutChannel[MAX_CHANNELS], const Word16 nChannels) argument 94 adaptMinSnr(PSY_OUT_CHANNEL psyOutChannel[MAX_CHANNELS], Word16 logSfbEnergy[MAX_CHANNELS][MAX_GROUPED_SFB], MINSNR_ADAPT_PARAM *msaParam, const Word16 nChannels) argument 164 initAvoidHoleFlag(Word16 ahFlag[MAX_CHANNELS][MAX_GROUPED_SFB], PSY_OUT_CHANNEL psyOutChannel[MAX_CHANNELS], PSY_OUT_ELEMENT* psyOutElement, const Word16 nChannels, AH_PARAM *ahParam) argument 327 calcPeNoAH(Word16 *pe, Word16 *constPart, Word16 *nActiveLines, PE_DATA *peData, Word16 ahFlag[MAX_CHANNELS][MAX_GROUPED_SFB], PSY_OUT_CHANNEL psyOutChannel[MAX_CHANNELS], const Word16 nChannels) argument 367 reduceThresholds(PSY_OUT_CHANNEL psyOutChannel[MAX_CHANNELS], Word16 ahFlag[MAX_CHANNELS][MAX_GROUPED_SFB], Word32 thrExp[MAX_CHANNELS][MAX_GROUPED_SFB], const Word16 nChannels, const Word32 redVal) argument 414 correctThresh(PSY_OUT_CHANNEL psyOutChannel[MAX_CHANNELS], Word16 ahFlag[MAX_CHANNELS][MAX_GROUPED_SFB], PE_DATA *peData, Word32 thrExp[MAX_CHANNELS][MAX_GROUPED_SFB], const Word32 redVal, const Word16 nChannels, const Word32 deltaPe) argument 528 reduceMinSnr(PSY_OUT_CHANNEL psyOutChannel[MAX_CHANNELS], PE_DATA *peData, Word16 ahFlag[MAX_CHANNELS][MAX_GROUPED_SFB], const Word16 nChannels, const Word16 desiredPe) argument 578 allowMoreHoles(PSY_OUT_CHANNEL psyOutChannel[MAX_CHANNELS], PSY_OUT_ELEMENT *psyOutElement, PE_DATA *peData, Word16 ahFlag[MAX_CHANNELS][MAX_GROUPED_SFB], const AH_PARAM *ahParam, const Word16 nChannels, const Word16 desiredPe) argument 733 adaptThresholdsToPe(PSY_OUT_CHANNEL psyOutChannel[MAX_CHANNELS], PSY_OUT_ELEMENT *psyOutElement, Word16 logSfbEnergy[MAX_CHANNELS][MAX_GROUPED_SFB], PE_DATA *peData, const Word16 nChannels, const Word16 desiredPe, AH_PARAM *ahParam, MINSNR_ADAPT_PARAM *msaParam) argument 1124 AdjustThresholds(ADJ_THR_STATE *adjThrState, ATS_ELEMENT *AdjThrStateElement, PSY_OUT_CHANNEL psyOutChannel[MAX_CHANNELS], PSY_OUT_ELEMENT *psyOutElement, Word16 *chBitDistribution, Word16 logSfbEnergy[MAX_CHANNELS][MAX_GROUPED_SFB], Word16 sfbNRelevantLines[MAX_CHANNELS][MAX_GROUPED_SFB], QC_OUT_ELEMENT *qcOE, ELEMENT_BITS *elBits, const Word16 nChannels, const Word16 maxBitFac) argument [all...] |
H A D | channel_map.c | 64 Word16 InitElementInfo (Word16 nChannels, ELEMENT_INFO* elInfo) argument 69 switch(nChannels) {
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H A D | qc_main.c | 121 Word16 QCOutNew(QC_OUT *hQC, Word16 nChannels, VO_MEM_OPERATOR *pMemOP) argument 128 quantSpec = (Word16 *)mem_malloc(pMemOP, nChannels * FRAME_LEN_LONG * sizeof(Word16), 32, VO_INDEX_ENC_AAC); 131 scf = (Word16 *)mem_malloc(pMemOP, nChannels * MAX_GROUPED_SFB * sizeof(Word16), 32, VO_INDEX_ENC_AAC); 136 maxValueInSfb = (UWord16 *)mem_malloc(pMemOP, nChannels * MAX_GROUPED_SFB * sizeof(UWord16), 32, VO_INDEX_ENC_AAC); 142 for (i=0; i<nChannels; i++) { 224 hQC->nChannels = init->elInfo->nChannelsInEl; 264 Word16 nChannels, 281 nChannels, 295 CalcFormFactor(hQC->logSfbFormFactor, hQC->sfbNRelevantLines, hQC->logSfbEnergy, psyOutChannel, nChannels); 307 nChannels, 257 QCMain(QC_STATE* hQC, ELEMENT_BITS* elBits, ATS_ELEMENT* adjThrStateElement, PSY_OUT_CHANNEL psyOutChannel[MAX_CHANNELS], PSY_OUT_ELEMENT* psyOutElement, QC_OUT_CHANNEL qcOutChannel[MAX_CHANNELS], QC_OUT_ELEMENT* qcOutElement, Word16 nChannels, Word16 ancillaryDataBytes) argument [all...] |
H A D | line_pe.c | 44 const Word16 nChannels, 50 for(ch=0; ch<nChannels; ch++) { 73 const Word16 nChannels) 84 for(ch=0; ch<nChannels; ch++) { 40 prepareSfbPe(PE_DATA *peData, PSY_OUT_CHANNEL psyOutChannel[MAX_CHANNELS], Word16 logSfbEnergy[MAX_CHANNELS][MAX_GROUPED_SFB], Word16 sfbNRelevantLines[MAX_CHANNELS][MAX_GROUPED_SFB], const Word16 nChannels, const Word16 peOffset) argument 71 calcSfbPe(PE_DATA *peData, PSY_OUT_CHANNEL psyOutChannel[MAX_CHANNELS], const Word16 nChannels) argument
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/frameworks/av/media/libstagefright/ |
H A D | FLACExtractor.cpp | 125 void (*mCopy)(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels); 383 static void copyMono8(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels) argument 390 static void copyStereo8(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels) argument 398 static void copyMultiCh8(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels) argument 401 for (unsigned c = 0; c < nChannels; ++c) { 407 static void copyMono16(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels) argument 414 static void copyStereo16(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels) argument 422 static void copyMultiCh16(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels) argument 425 for (unsigned c = 0; c < nChannels; ++c) { 433 static void copyMono24(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels) argument 440 copyStereo24(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels) argument 448 copyMultiCh24(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels) argument 457 copyTrespass(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels) argument [all...] |
/frameworks/av/media/libstagefright/codecs/aacenc/SampleCode/ |
H A D | AAC_E_SAMPLES.c | 51 // bitRate/nChannels > 8000 52 // bitRate/nChannels < 160000 53 // bitRate/nChannels < sampleRate*6 56 param->nChannels = 2; 90 param->nChannels = atoi(*argv); 118 param->bitRate = 640*param->nChannels*param->sampleRate/scale;
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