/external/chromium_org/third_party/WebKit/Source/modules/webaudio/ |
H A D | AudioSourceNode.h | 39 AudioSourceNode(AudioContext* context, float sampleRate) argument 40 : AudioNode(context, 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 | AsyncAudioDecoder.cpp | 52 void AsyncAudioDecoder::decodeAsync(ArrayBuffer* audioData, float sampleRate, AudioBufferCallback* successCallback, AudioBufferCallback* errorCallback) argument 63 m_thread->postTask(new Task(WTF::bind(&AsyncAudioDecoder::decode, audioDataRef.release().leakRef(), sampleRate, successCallback, errorCallback))); 66 void AsyncAudioDecoder::decode(ArrayBuffer* audioData, float sampleRate, AudioBufferCallback* successCallback, AudioBufferCallback* errorCallback) argument 68 RefPtr<AudioBus> bus = createBusFromInMemoryAudioFile(audioData->data(), audioData->byteLength(), false, sampleRate);
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H A D | AudioBuffer.h | 46 static AudioBuffer* create(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate); 47 static AudioBuffer* create(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState&); 50 static AudioBuffer* createFromAudioFileData(const void* data, size_t dataSize, bool mixToMono, float sampleRate); 56 double duration() const { return length() / static_cast<double>(sampleRate()); } 57 float sampleRate() const { return m_sampleRate; } function in class:blink::AudioBuffer 70 AudioBuffer(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate);
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H A D | AudioDestinationNode.cpp | 39 AudioDestinationNode::AudioDestinationNode(AudioContext* context, float sampleRate) argument 40 : AudioNode(context, sampleRate)
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H A D | BiquadFilterNode.cpp | 33 BiquadFilterNode::BiquadFilterNode(AudioContext* context, float sampleRate) argument 34 : AudioBasicProcessorNode(context, sampleRate) 37 m_processor = new BiquadProcessor(context, sampleRate, 1, false);
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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 | ConvolverNode.h | 42 static ConvolverNode* create(AudioContext* context, float sampleRate) argument 44 return adoptRefCountedGarbageCollectedWillBeNoop(new ConvolverNode(context, sampleRate)); 65 ConvolverNode(AudioContext*, float sampleRate);
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H A D | DelayProcessor.cpp | 35 DelayProcessor::DelayProcessor(AudioContext* context, float sampleRate, unsigned numberOfChannels, double maxDelayTime) argument 36 : AudioDSPKernelProcessor(sampleRate, numberOfChannels)
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H A D | GainNode.cpp | 37 GainNode::GainNode(AudioContext* context, float sampleRate) argument 38 : AudioNode(context, sampleRate)
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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 | 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 | AnalyserNode.cpp | 39 AnalyserNode::AnalyserNode(AudioContext* context, float sampleRate) argument 40 : AudioBasicInspectorNode(context, sampleRate, 2)
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/external/chromium_org/third_party/WebKit/Source/platform/audio/ |
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 | 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 | 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 | 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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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 | 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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/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/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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/external/chromium_org/third_party/WebKit/Source/platform/exported/ |
H A D | WebAudioBus.cpp | 47 void WebAudioBus::initialize(unsigned numberOfChannels, size_t length, double sampleRate) argument 51 audioBus->setSampleRate(sampleRate); 112 double WebAudioBus::sampleRate() const function in class:blink::WebAudioBus 117 return m_private->sampleRate();
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/external/chromium_org/content/renderer/media/ |
H A D | renderer_webaudiodevice_impl.cc | 74 double RendererWebAudioDeviceImpl::sampleRate() { function in class:content::RendererWebAudioDeviceImpl
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