/external/chromium_org/third_party/WebKit/Source/platform/audio/ |
H A D | AudioUtilities.cpp | 54 double discreteTimeConstantForSampleRate(double timeConstant, double sampleRate) argument 56 return 1 - exp(-1 / (sampleRate * timeConstant)); 59 size_t timeToSampleFrame(double time, double sampleRate) argument 61 return static_cast<size_t>(round(time * sampleRate)); 64 bool isValidAudioBufferSampleRate(float sampleRate) argument 66 return sampleRate >= minAudioBufferSampleRate() && sampleRate <= maxAudioBufferSampleRate();
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H A D | AudioUtilities.h | 39 // discreteTimeConstantForSampleRate() will return the discrete time-constant for the specific sampleRate. 40 PLATFORM_EXPORT double discreteTimeConstantForSampleRate(double timeConstant, double sampleRate); 43 PLATFORM_EXPORT size_t timeToSampleFrame(double time, double sampleRate); 45 // Check that |sampleRate| is a valid rate for AudioBuffers. 46 PLATFORM_EXPORT bool isValidAudioBufferSampleRate(float sampleRate);
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H A D | Panner.cpp | 40 Panner* Panner::create(PanningModel model, float sampleRate, HRTFDatabaseLoader* databaseLoader) argument 46 panner = new EqualPowerPanner(sampleRate); 50 panner = new HRTFPanner(sampleRate, databaseLoader);
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H A D | AudioDSPKernel.h | 44 , m_sampleRate(kernelProcessor->sampleRate()) 48 AudioDSPKernel(float sampleRate) argument 50 , m_sampleRate(sampleRate) 60 float sampleRate() const { return m_sampleRate; } function in class:blink::AudioDSPKernel 61 double nyquist() const { return 0.5 * sampleRate(); }
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H A D | HRTFKernel.h | 54 static PassRefPtr<HRTFKernel> create(AudioChannel* channel, size_t fftSize, float sampleRate) argument 56 return adoptRef(new HRTFKernel(channel, fftSize, sampleRate)); 59 static PassRefPtr<HRTFKernel> create(PassOwnPtr<FFTFrame> fftFrame, float frameDelay, float sampleRate) argument 61 return adoptRef(new HRTFKernel(fftFrame, frameDelay, sampleRate)); 72 float sampleRate() const { return m_sampleRate; } function in class:blink::HRTFKernel 73 double nyquist() const { return 0.5 * sampleRate(); } 80 HRTFKernel(AudioChannel*, size_t fftSize, float sampleRate); 82 HRTFKernel(PassOwnPtr<FFTFrame> fftFrame, float frameDelay, float sampleRate) argument 85 , m_sampleRate(sampleRate)
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H A D | AudioFileReader.h | 40 // Pass in 0.0 for sampleRate to use the file's sample-rate, otherwise a sample-rate conversion to the requested 41 // sampleRate will be made (if it doesn't already match the file's sample-rate). 44 PLATFORM_EXPORT PassRefPtr<AudioBus> createBusFromInMemoryAudioFile(const void* data, size_t dataSize, bool mixToMono, float sampleRate); 46 PLATFORM_EXPORT PassRefPtr<AudioBus> createBusFromAudioFile(const char* filePath, bool mixToMono, float sampleRate); 48 // May pass in 0.0 for sampleRate in which case it will use the AudioBus's sampleRate
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H A D | HRTFDatabaseLoader.cpp | 50 HRTFDatabaseLoader* HRTFDatabaseLoader::createAndLoadAsynchronouslyIfNecessary(float sampleRate) argument 54 HRTFDatabaseLoader* loader = loaderMap().get(sampleRate); 56 ASSERT(sampleRate == loader->databaseSampleRate()); 60 loader = new HRTFDatabaseLoader(sampleRate); 61 loaderMap().add(sampleRate, loader); 66 HRTFDatabaseLoader::HRTFDatabaseLoader(float sampleRate) argument 67 : m_databaseSampleRate(sampleRate)
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H A D | AudioDelayDSPKernel.cpp | 47 AudioDelayDSPKernel::AudioDelayDSPKernel(double maxDelayTime, float sampleRate) argument 48 : AudioDSPKernel(sampleRate) 57 size_t bufferLength = bufferLengthForDelay(maxDelayTime, sampleRate); 65 m_smoothingRate = AudioUtilities::discreteTimeConstantForSampleRate(SmoothingTimeConstant, sampleRate); 68 size_t AudioDelayDSPKernel::bufferLengthForDelay(double maxDelayTime, double sampleRate) const 72 return 1 + AudioUtilities::timeToSampleFrame(maxDelayTime, sampleRate); 85 double AudioDelayDSPKernel::delayTime(float sampleRate) argument 87 return m_desiredDelayFrames / sampleRate; 103 float sampleRate = this->sampleRate(); local [all...] |
H A D | AudioProcessor.h | 47 AudioProcessor(float sampleRate, unsigned numberOfChannels) argument 50 , m_sampleRate(sampleRate) 72 float sampleRate() const { return m_sampleRate; } function in class:blink::AudioProcessor
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H A D | HRTFDatabase.h | 46 static PassOwnPtr<HRTFDatabase> create(float sampleRate); 57 float sampleRate() const { return m_sampleRate; } function in class:blink::HRTFDatabase 63 explicit HRTFDatabase(float sampleRate);
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/external/chromium_org/third_party/WebKit/Source/modules/webaudio/ |
H A D | PeriodicWave.h | 43 static PeriodicWave* createSine(float sampleRate); 44 static PeriodicWave* createSquare(float sampleRate); 45 static PeriodicWave* createSawtooth(float sampleRate); 46 static PeriodicWave* createTriangle(float sampleRate); 49 static PeriodicWave* create(float sampleRate, Float32Array* real, Float32Array* imag); 67 explicit PeriodicWave(float sampleRate);
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H A D | AudioSourceNode.h | 39 AudioSourceNode(AudioContext* context, float sampleRate) argument 40 : AudioNode(context, sampleRate) { }
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H A D | OfflineAudioContext.h | 37 static OfflineAudioContext* create(ExecutionContext*, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState&); 42 OfflineAudioContext(Document*, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate);
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H A D | OfflineAudioContext.idl | 27 Constructor(unsigned long numberOfChannels, unsigned long numberOfFrames, float sampleRate),
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H A D | OfflineAudioContext.cpp | 40 OfflineAudioContext* OfflineAudioContext::create(ExecutionContext* context, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState& exceptionState) argument 70 if (!AudioUtilities::isValidAudioBufferSampleRate(sampleRate)) { 74 "sampleRate", sampleRate, 80 OfflineAudioContext* audioContext = adoptRefCountedGarbageCollectedWillBeNoop(new OfflineAudioContext(document, numberOfChannels, numberOfFrames, sampleRate)); 87 + ", " + String::number(sampleRate) 95 OfflineAudioContext::OfflineAudioContext(Document* document, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate) argument 96 : AudioContext(document, numberOfChannels, numberOfFrames, sampleRate)
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H A D | DelayNode.cpp | 40 DelayNode::DelayNode(AudioContext* context, float sampleRate, double maxDelayTime, ExceptionState& exceptionState) argument 41 : AudioBasicProcessorNode(context, sampleRate) 51 m_processor = new DelayProcessor(context, sampleRate, 1, maxDelayTime);
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H A D | DelayNode.h | 40 static DelayNode* create(AudioContext* context, float sampleRate, double maxDelayTime, ExceptionState& exceptionState) argument 42 return adoptRefCountedGarbageCollectedWillBeNoop(new DelayNode(context, sampleRate, maxDelayTime, exceptionState)); 48 DelayNode(AudioContext*, float sampleRate, double maxDelayTime, ExceptionState&);
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H A D | GainNode.h | 43 static GainNode* create(AudioContext* context, float sampleRate) argument 45 return adoptRefCountedGarbageCollectedWillBeNoop(new GainNode(context, sampleRate)); 63 GainNode(AudioContext*, float sampleRate);
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H A D | ChannelSplitterNode.cpp | 37 ChannelSplitterNode* ChannelSplitterNode::create(AudioContext* context, float sampleRate, unsigned numberOfOutputs) argument 42 return adoptRefCountedGarbageCollectedWillBeNoop(new ChannelSplitterNode(context, sampleRate, numberOfOutputs)); 45 ChannelSplitterNode::ChannelSplitterNode(AudioContext* context, float sampleRate, unsigned numberOfOutputs) argument 46 : AudioNode(context, sampleRate)
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H A D | DelayDSPKernel.cpp | 42 ASSERT(processor && processor->sampleRate() > 0); 43 if (!(processor && processor->sampleRate() > 0)) 51 m_buffer.allocate(bufferLengthForDelay(m_maxDelayTime, processor->sampleRate())); 54 m_smoothingRate = AudioUtilities::discreteTimeConstantForSampleRate(SmoothingTimeConstant, processor->sampleRate());
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H A D | ChannelMergerNode.h | 42 static ChannelMergerNode* create(AudioContext*, float sampleRate, unsigned numberOfInputs); 56 ChannelMergerNode(AudioContext*, float sampleRate, unsigned numberOfInputs);
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/external/aac/libAACenc/src/ |
H A D | bandwidth.h | 101 INT sampleRate,
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/external/chromium_org/content/shell/renderer/test_runner/ |
H A D | mock_web_audio_device.cc | 18 double MockWebAudioDevice::sampleRate() { function in class:content::MockWebAudioDevice
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/external/chromium_org/third_party/WebKit/public/platform/ |
H A D | WebAudioDestinationConsumer.h | 36 virtual void setFormat(size_t numberOfChannels, float sampleRate) = 0;
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H A D | WebAudioSourceProviderClient.h | 32 virtual void setFormat(size_t numberOfChannels, float sampleRate) = 0;
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