/external/aac/libMpegTPEnc/src/ |
H A D | tpenc_adif.cpp | 109 INT sampleRate = adif->samplingRate; local 147 transportEnc_writePCE(hBs, adif->cm, sampleRate, adif->instanceTag, adif->profile, 0, 0, alignAnchor);
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H A D | tpenc_asc.cpp | 221 INT sampleRate, 233 sampleRateIndex = getSamplingRateIndex(sampleRate); 357 static void writeSampleRate(HANDLE_FDK_BITSTREAM hBitstreamBuffer, int sampleRate) argument 359 int sampleRateIndex = getSamplingRateIndex(sampleRate); 363 FDKwriteBits( hBitstreamBuffer, sampleRate, 24 ); 219 transportEnc_writePCE(HANDLE_FDK_BITSTREAM hBs, CHANNEL_MODE channelMode, INT sampleRate, int instanceTagPCE, int profile, int matrixMixdownA, int pseudoSurroundEnable, UINT alignAnchor) argument
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/external/sonic/ |
H A D | Main.java | 29 int sampleRate, 32 Sonic sonic = new Sonic(sampleRate, numChannels); 72 int sampleRate = (int)format.getSampleRate(); 80 sampleRate, numChannels); 20 runSonic( AudioInputStream audioStream, SourceDataLine line, float speed, float pitch, float rate, float volume, boolean emulateChordPitch, int quality, int sampleRate, int numChannels) argument
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H A D | main.c | 26 int sampleRate, 29 sonicStream stream = sonicCreateStream(sampleRate, numChannels); 83 int sampleRate, numChannels; local 125 inFile = openInputWaveFile(inFileName, &sampleRate, &numChannels); 129 outFile = openOutputWaveFile(outFileName, sampleRate, numChannels); 135 sampleRate, numChannels); 17 runSonic( waveFile inFile, waveFile outFile, float speed, float pitch, float rate, float volume, int emulateChordPitch, int quality, int sampleRate, int numChannels) argument
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H A D | wave.c | 21 int sampleRate; member in struct:waveFileStruct 169 int sampleRate) 181 writeInt(file, sampleRate); /* 24 - samples per second (numbers per second) */ 182 writeInt(file, sampleRate * 2); /* 28 - bytes per second */ 210 file->sampleRate = readInt(file); /* 24 - samples per second (numbers per second) */ 242 int *sampleRate, 259 *sampleRate = file->sampleRate; 267 int sampleRate, 279 file->sampleRate 167 writeHeader( waveFile file, int sampleRate) argument 240 openInputWaveFile( char *fileName, int *sampleRate, int *numChannels) argument 265 openOutputWaveFile( char *fileName, int sampleRate, int numChannels) argument [all...] |
/external/sonivox/arm-wt-22k/lib_src/ |
H A D | eas_pcm.h | 45 EAS_U32 sampleRate; member in struct:s_pcm_open_params_tag
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H A D | eas_pcm.c | 102 static EAS_U32 CalcBaseFreq (EAS_U32 sampleRate); 371 pState->sampleRate = (EAS_U16) pParams->sampleRate; 374 pState->basefreq = (SRC_RATE_MULTIPLER * (EAS_U32) pParams->sampleRate) >> 15; 879 * sampleRate - sample rate in samples/sec 888 static EAS_U32 CalcBaseFreq (EAS_U32 sampleRate) argument 895 if (srcConvRate[i][0] == sampleRate) 900 { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Sample rate %u not in table, calculating by division\n", sampleRate); */ } 902 return (SRC_RATE_MULTIPLER * (EAS_U32) sampleRate) >> 15; 1374 temp = (msecs * pState->sampleRate); [all...] |
H A D | eas_pcmdata.h | 113 EAS_U16 sampleRate; /* input sample rate */ member in struct:s_pcm_state_tag
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/external/aac/libAACenc/src/ |
H A D | bandwidth.cpp | 198 const INT sampleRate, 213 switch (sampleRate) { 289 INT sampleRate, 334 *bandWidth = FDKmin(proposedBandWidth, FDKmin(20000, sampleRate>>1)); 362 sampleRate, 376 *bandWidth = FDKmin(*bandWidth, sampleRate/2); 196 GetBandwidthEntry( const INT frameLength, const INT sampleRate, const INT chanBitRate, const INT entryNo) argument 285 FDKaacEnc_DetermineBandWidth(INT* bandWidth, INT proposedBandWidth, INT bitrate, AACENC_BITRATE_MODE bitrateMode, INT sampleRate, INT frameLength, CHANNEL_MAPPING* cm, CHANNEL_MODE encoderMode) argument
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H A D | aacenc_pns.cpp | 134 INT sampleRate, 146 sampleRate, 132 FDKaacEnc_InitPnsConfiguration(PNS_CONFIG *pnsConf, INT bitRate, INT sampleRate, INT usePns, INT sfbCnt, const INT *sfbOffset, const INT numChan, const INT isLC) argument
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H A D | pnsparam.cpp | 190 int FDKaacEnc_lookUpPnsUse (int bitRate, int sampleRate, int numChan, const int isLC) { argument 215 switch (sampleRate) { 245 INT sampleRate, 270 hUsePns = FDKaacEnc_lookUpPnsUse (bitRate, sampleRate, numChan, isLC); 282 sampleRate, 243 FDKaacEnc_GetPnsParam(NOISEPARAMS *np, INT bitRate, INT sampleRate, INT sfbCnt, const INT *sfbOffset, INT *usePns, INT numChan, const int isLC) argument
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H A D | aacenc.cpp | 147 INT sampleRate); 437 switch (config->sampleRate) 465 config->sampleRate, 516 config->sampleRate); 522 config->ancDataBitRate += ( (hAacEnc->ancillaryBitsPerFrame * config->sampleRate) / config->framelength ); 530 FIXP_DBL tmp = fDivNorm(config->framelength, config->sampleRate, &q_res); 554 config->sampleRate, 578 config->sampleRate, 624 qcInit.sampleRate = config->sampleRate; 1024 FDKaacEnc_InitCheckAncillary(INT bitRate, INT framelength, INT ancillaryRate, INT *ancillaryBitsPerFrame, INT sampleRate) argument [all...] |
H A D | aacenc.h | 188 INT sampleRate; /* encoder sample rate */ member in struct:AACENC_CONFIG
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H A D | metadata_main.cpp | 407 const UINT sampleRate, 470 sampleRate, 401 FDK_MetadataEnc_Init( HANDLE_FDK_METADATA_ENCODER hMetaData, const INT resetStates, const INT metadataMode, const INT audioDelay, const UINT frameLength, const UINT sampleRate, const UINT nChannels, const CHANNEL_MODE channelMode, const CHANNEL_ORDER channelOrder ) argument
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/external/aac/libSYS/include/ |
H A D | wav_file.h | 147 UINT sampleRate; member in struct:WAV_HEADER 200 * \param sampleRate Desired samplerate of the resulting WAV file. 206 INT WAV_OutputOpen(HANDLE_WAV *pWav, const char *outputFilename, INT sampleRate, INT numChannels, INT bitsPerSample);
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/external/webrtc/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/ |
H A D | WebRtcAudioRecord.java | 153 private int initRecording(int sampleRate, int channels) { argument 154 Logging.d(TAG, "initRecording(sampleRate=" + sampleRate + ", channels=" + 166 final int framesPerBuffer = sampleRate / BUFFERS_PER_SECOND; 178 sampleRate, 196 sampleRate,
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H A D | WebRtcAudioTrack.java | 155 private void initPlayout(int sampleRate, int channels) { argument 156 Logging.d(TAG, "initPlayout(sampleRate=" + sampleRate + ", channels=" 160 bytesPerFrame * (sampleRate / BUFFERS_PER_SECOND)); 172 sampleRate, 187 sampleRate,
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H A D | WebRtcAudioManager.java | 84 private int sampleRate; field in class:WebRtcAudioManager 100 sampleRate, channels, hardwareAEC, hardwareAGC, hardwareNS, 140 sampleRate = getNativeOutputSampleRate(); 147 getMinOutputFrameSize(sampleRate, channels); 149 inputBufferSize = getMinInputFrameSize(sampleRate, channels); 289 int sampleRate, int channels, boolean hardwareAEC, boolean hardwareAGC, 288 nativeCacheAudioParameters( int sampleRate, int channels, boolean hardwareAEC, boolean hardwareAGC, boolean hardwareNS, boolean lowLatencyOutput, int outputBufferSize, int inputBufferSize, long nativeAudioManager) argument
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/external/aac/libPCMutils/src/ |
H A D | limiter.cpp | 100 unsigned int sampleRate, maxSampleRate; member in struct:TDLimiter 164 limiter->sampleRate = maxSampleRate; 406 TDLIMITER_ERROR setLimiterSampleRate(TDLimiterPtr limiter, unsigned int sampleRate) argument 414 if (sampleRate > limiter->maxSampleRate) return TDLIMIT_INVALID_PARAMETER; 417 attack = (unsigned int)(limiter->attackMs * sampleRate / 1000); 418 release = (unsigned int)(limiter->releaseMs * sampleRate / 1000); 433 limiter->sampleRate = sampleRate; 453 attack = (unsigned int)(attackMs * limiter->sampleRate / 1000); 477 release = (unsigned int)(releaseMs * limiter->sampleRate / 100 [all...] |
/external/aac/libSYS/src/ |
H A D | wav_file.cpp | 163 FDKfread_EL(&(wav->header.sampleRate), 4, 1, wav->fp); 378 * \param sampleRate desired samplerate of the resulting WAV file 385 INT WAV_OutputOpen(HANDLE_WAV *pWav, const char *outputFilename, INT sampleRate, INT numChannels, INT bitsPerSample) argument 419 wav->header.sampleRate = LittleEndian32(sampleRate); 420 wav->header.bytesPerSecond = LittleEndian32(sampleRate * wav->header.blockAlign);
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/external/mp4parser/isoparser/src/main/java/com/coremedia/iso/boxes/apple/ |
H A D | AppleLosslessSpecificBox.java | 32 private long sampleRate; // 32bit
field in class:AppleLosslessSpecificBox 115 return sampleRate;
118 public void setSampleRate(int sampleRate) {
argument 119 this.sampleRate = sampleRate;
136 sampleRate = IsoTypeReader.readUInt32(content);
152 IsoTypeWriter.writeUInt32(byteBuffer, sampleRate);
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/external/sonivox/arm-wt-22k/host_src/ |
H A D | eas.h | 55 EAS_I32 sampleRate; member in struct:__anon17371
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/external/sonivox/arm-wt-22k/include/libsonivox/ |
H A D | eas.h | 55 EAS_I32 sampleRate; member in struct:__anon17393
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/external/mp4parser/isoparser/src/main/java/com/coremedia/iso/boxes/sampleentry/ |
H A D | AudioSampleEntry.java | 55 private long sampleRate; field in class:AudioSampleEntry 82 return sampleRate; 133 public void setSampleRate(long sampleRate) { argument 134 this.sampleRate = sampleRate; 198 //sampleRate = in.readFixedPoint1616(); 199 sampleRate = IsoTypeReader.readUInt32(content); 201 sampleRate = sampleRate >>> 16; 242 ", sampleRate [all...] |
/external/aac/libAACdec/include/ |
H A D | aacdecoder_lib.h | 220 While the members sampleRate, frameSize and numChannels might be quite self explaining, 536 INT sampleRate; /*!< The samplerate in Hz of the fully decoded PCM audio signal (after SBR processing). */ member in struct:__anon177
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