/external/chromium_org/third_party/WebKit/Source/modules/webaudio/ |
H A D | OscillatorNode.cpp | 44 OscillatorNode* OscillatorNode::create(AudioContext* context, float sampleRate) argument 46 return adoptRefCountedGarbageCollectedWillBeNoop(new OscillatorNode(context, sampleRate)); 49 OscillatorNode::OscillatorNode(AudioContext* context, float sampleRate) argument 50 : AudioScheduledSourceNode(context, sampleRate) 118 float sampleRate = this->sampleRate(); local 122 DEFINE_STATIC_LOCAL(Persistent<PeriodicWave>, periodicWaveSine, (PeriodicWave::createSine(sampleRate))); 127 DEFINE_STATIC_LOCAL(Persistent<PeriodicWave>, periodicWaveSquare, (PeriodicWave::createSquare(sampleRate))); 132 DEFINE_STATIC_LOCAL(Persistent<PeriodicWave>, periodicWaveSawtooth, (PeriodicWave::createSawtooth(sampleRate))); 137 DEFINE_STATIC_LOCAL(Persistent<PeriodicWave>, periodicWaveTriangle, (PeriodicWave::createTriangle(sampleRate))); [all...] |
H A D | PeriodicWave.cpp | 48 PeriodicWave* PeriodicWave::create(float sampleRate, Float32Array* real, Float32Array* imag) argument 53 PeriodicWave* periodicWave = new PeriodicWave(sampleRate); 61 PeriodicWave* PeriodicWave::createSine(float sampleRate) argument 63 PeriodicWave* periodicWave = new PeriodicWave(sampleRate); 68 PeriodicWave* PeriodicWave::createSquare(float sampleRate) argument 70 PeriodicWave* periodicWave = new PeriodicWave(sampleRate); 75 PeriodicWave* PeriodicWave::createSawtooth(float sampleRate) argument 77 PeriodicWave* periodicWave = new PeriodicWave(sampleRate); 82 PeriodicWave* PeriodicWave::createTriangle(float sampleRate) argument 84 PeriodicWave* periodicWave = new PeriodicWave(sampleRate); 89 PeriodicWave(float sampleRate) argument [all...] |
H A D | AudioBufferSourceNode.cpp | 50 AudioBufferSourceNode* AudioBufferSourceNode::create(AudioContext* context, float sampleRate) argument 52 return adoptRefCountedGarbageCollectedWillBeNoop(new AudioBufferSourceNode(context, sampleRate)); 55 AudioBufferSourceNode::AudioBufferSourceNode(AudioContext* context, float sampleRate) argument 56 : AudioScheduledSourceNode(context, sampleRate) 203 double bufferSampleRate = buffer()->sampleRate(); 228 double loopStartFrame = m_loopStart * buffer()->sampleRate(); 229 double loopEndFrame = m_loopEnd * buffer()->sampleRate(); 435 m_virtualReadIndex = AudioUtilities::timeToSampleFrame(m_grainOffset, buffer()->sampleRate()); 450 sampleRateFactor = buffer()->sampleRate() / sampleRate(); [all...] |
/external/chromium_org/third_party/WebKit/Source/platform/audio/ |
H A D | HRTFElevation.cpp | 104 bool HRTFElevation::calculateKernelsForAzimuthElevation(int azimuth, int elevation, float sampleRate, const String& subjectName, argument 163 RefPtr<AudioBus> response(AudioBus::createBySampleRateConverting(preSampleRateConvertedResponse.get(), false, sampleRate)); 169 RefPtr<AudioBus> impulseResponse(AudioBus::loadPlatformResource(resourceName.utf8().data(), sampleRate)); 176 size_t expectedLength = static_cast<size_t>(256 * (sampleRate / 44100.0)); 189 const size_t fftSize = HRTFPanner::fftSizeForSampleRate(sampleRate); 190 kernelL = HRTFKernel::create(leftEarImpulseResponse, fftSize, sampleRate); 191 kernelR = HRTFKernel::create(rightEarImpulseResponse, fftSize, sampleRate); 228 PassOwnPtr<HRTFElevation> HRTFElevation::createForSubject(const String& subjectName, int elevation, float sampleRate) 245 bool success = calculateKernelsForAzimuthElevation(rawIndex * AzimuthSpacing, actualElevation, sampleRate, subjectName, kernelListL->at(interpolatedIndex), kernelListR->at(interpolatedIndex)); 265 OwnPtr<HRTFElevation> hrtfElevation = adoptPtr(new HRTFElevation(kernelListL.release(), kernelListR.release(), elevation, sampleRate)); 269 createByInterpolatingSlices(HRTFElevation* hrtfElevation1, HRTFElevation* hrtfElevation2, float x, float sampleRate) argument [all...] |
H A D | HRTFPanner.cpp | 47 HRTFPanner::HRTFPanner(float sampleRate, HRTFDatabaseLoader* databaseLoader) argument 50 , m_sampleRate(sampleRate) 58 , m_convolverL1(fftSizeForSampleRate(sampleRate)) 59 , m_convolverR1(fftSizeForSampleRate(sampleRate)) 60 , m_convolverL2(fftSizeForSampleRate(sampleRate)) 61 , m_convolverR2(fftSizeForSampleRate(sampleRate)) 62 , m_delayLineL(MaxDelayTimeSeconds, sampleRate) 63 , m_delayLineR(MaxDelayTimeSeconds, sampleRate) 76 size_t HRTFPanner::fftSizeForSampleRate(float sampleRate) argument 86 ASSERT(AudioUtilities::isValidAudioBufferSampleRate(sampleRate)); [all...] |
/external/chromium_org/third_party/WebKit/Source/platform/exported/ |
H A D | WebMediaStreamSource.cpp | 176 virtual void setFormat(size_t numberOfChannels, float sampleRate) OVERRIDE; 188 void ConsumerWrapper::setFormat(size_t numberOfChannels, float sampleRate) argument 190 m_consumer->setFormat(numberOfChannels, sampleRate);
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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-fm-22k/host_src/ |
H A D | eas.h | 55 EAS_I32 sampleRate; member in struct:__anon30830
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/external/sonivox/arm-hybrid-22k/host_src/ |
H A D | eas.h | 55 EAS_I32 sampleRate; member in struct:__anon30880
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/external/sonivox/arm-wt-22k/host_src/ |
H A D | eas.h | 55 EAS_I32 sampleRate; member in struct:__anon30931
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/external/srec/srec/include/ |
H A D | utteranc.h | 221 unsigned long sampleRate; member in struct:_UttHeader
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/external/aac/libAACenc/src/ |
H A D | aacenc.cpp | 114 INT sampleRate); 401 switch (config->sampleRate) 429 config->sampleRate, 478 config->sampleRate); 484 config->ancDataBitRate += ( (hAacEnc->ancillaryBitsPerFrame * config->sampleRate) / config->framelength ); 492 FIXP_DBL tmp = fDivNorm(config->framelength, config->sampleRate, &q_res); 516 config->sampleRate, 540 config->sampleRate, 581 qcInit.sampleRate = config->sampleRate; 999 FDKaacEnc_InitCheckAncillary(INT bitRate, INT framelength, INT ancillaryRate, INT *ancillaryBitsPerFrame, INT sampleRate) argument [all...] |
H A D | aacenc.h | 183 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/libMpegTPDec/src/ |
H A D | tpdec_asc.cpp | 984 INT sampleRate; local 992 sampleRate = FDKreadBits(bs,24); 994 sampleRate = SamplingRateTable[idx]; 999 return sampleRate;
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/external/chromium_org/third_party/WebKit/Source/web/ |
H A D | WebMediaPlayerClientImpl.cpp | 301 void WebMediaPlayerClientImpl::AudioClientImpl::setFormat(size_t numberOfChannels, float sampleRate) argument 304 m_client->setFormat(numberOfChannels, sampleRate);
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/external/chromium_org/third_party/webrtc/modules/audio_device/test/ |
H A D | audio_device_test_api.cc | 91 const uint32_t sampleRate, 117 const uint32_t sampleRate, 1722 uint32_t sampleRate(0); 1725 EXPECT_EQ(0, audio_device_->RecordingSampleRate(&sampleRate)); 1727 EXPECT_EQ(48000, sampleRate); 1729 TEST_LOG("Recording sample rate is %u\n\n", sampleRate); 1730 EXPECT_TRUE((sampleRate == 44000) || (sampleRate == 16000)); 1732 TEST_LOG("Recording sample rate is %u\n\n", sampleRate); 1733 EXPECT_TRUE((sampleRate 86 RecordedDataIsAvailable( const void* audioSamples, const uint32_t nSamples, const uint8_t nBytesPerSample, const uint8_t nChannels, const uint32_t sampleRate, const uint32_t totalDelay, const int32_t clockSkew, const uint32_t currentMicLevel, const bool keyPressed, uint32_t& newMicLevel) argument 113 NeedMorePlayData( const uint32_t nSamples, const uint8_t nBytesPerSample, const uint8_t nChannels, const uint32_t sampleRate, void* audioSamples, uint32_t& nSamplesOut, int64_t* elapsed_time_ms, int64_t* ntp_time_ms) argument [all...] |
/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/sonivox/arm-fm-22k/lib_src/ |
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/sonivox/arm-hybrid-22k/lib_src/ |
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/sonivox/arm-wt-22k/lib_src/ |
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/libAACdec/include/ |
H A D | aacdecoder_lib.h | 220 While the members sampleRate, frameSize and numChannels might be quite self explaining, 535 INT sampleRate; /*!< The samplerate in Hz of the fully decoded PCM audio signal (after SBR processing). */ member in struct:__anon3
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