/external/chromium_org/third_party/WebKit/Source/modules/webaudio/ |
H A D | AudioSourceNode.h | 38 AudioSourceNode(AudioContext* context, float sampleRate) argument 39 : AudioNode(context, sampleRate)
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H A D | DelayNode.cpp | 38 DelayNode::DelayNode(AudioContext* context, float sampleRate, double maxDelayTime, ExceptionState& es) argument 39 : AudioBasicProcessorNode(context, sampleRate) 46 m_processor = adoptPtr(new DelayProcessor(context, sampleRate, 1, maxDelayTime));
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H A D | DelayNode.h | 39 static PassRefPtr<DelayNode> create(AudioContext* context, float sampleRate, double maxDelayTime, ExceptionState& es) argument 41 return adoptRef(new DelayNode(context, sampleRate, maxDelayTime, es)); 47 DelayNode(AudioContext*, float sampleRate, double maxDelayTime, ExceptionState&);
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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.h | 42 static PassRefPtr<GainNode> create(AudioContext* context, float sampleRate) argument 44 return adoptRef(new GainNode(context, sampleRate)); 61 GainNode(AudioContext*, float sampleRate);
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H A D | AnalyserNode.cpp | 38 AnalyserNode::AnalyserNode(AudioContext* context, float sampleRate) argument 39 : AudioBasicInspectorNode(context, sampleRate, 2)
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H A D | AudioBuffer.h | 46 static PassRefPtr<AudioBuffer> create(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate); 49 static PassRefPtr<AudioBuffer> createFromAudioFileData(const void* data, size_t dataSize, bool mixToMono, float sampleRate); 53 double duration() const { return length() / sampleRate(); } 54 float sampleRate() const { return m_sampleRate; } function in class:WebCore::AudioBuffer 72 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) 38 m_processor = adoptPtr(new BiquadProcessor(context, sampleRate, 1, false));
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H A D | ChannelSplitterNode.cpp | 37 PassRefPtr<ChannelSplitterNode> ChannelSplitterNode::create(AudioContext* context, float sampleRate, unsigned numberOfOutputs) argument 42 return adoptRef(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 | 40 static PassRefPtr<ConvolverNode> create(AudioContext* context, float sampleRate) argument 42 return adoptRef(new ConvolverNode(context, sampleRate)); 61 ConvolverNode(AudioContext*, float sampleRate);
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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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/external/chromium_org/third_party/WebKit/Source/core/platform/audio/ |
H A D | Panner.cpp | 41 PassOwnPtr<Panner> Panner::create(PanningModel model, float sampleRate, HRTFDatabaseLoader* databaseLoader) argument 47 panner = adoptPtr(new EqualPowerPanner(sampleRate)); 51 panner = adoptPtr(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:WebCore::AudioDSPKernel 61 double nyquist() const { return 0.5 * sampleRate(); }
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H A D | AudioDSPKernelProcessor.cpp | 43 AudioDSPKernelProcessor::AudioDSPKernelProcessor(float sampleRate, unsigned numberOfChannels) argument 44 : AudioProcessor(sampleRate, numberOfChannels)
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H A D | AudioProcessor.h | 44 AudioProcessor(float sampleRate, unsigned numberOfChannels) argument 47 , m_sampleRate(sampleRate) 68 float sampleRate() const { return m_sampleRate; } function in class:WebCore::AudioProcessor
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H A D | AudioUtilities.cpp | 52 double discreteTimeConstantForSampleRate(double timeConstant, double sampleRate) argument 54 return 1 - exp(-1 / (sampleRate * timeConstant)); 57 size_t timeToSampleFrame(double time, double sampleRate) argument 59 return static_cast<size_t>(round(time * sampleRate));
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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:WebCore::HRTFDatabase 63 explicit HRTFDatabase(float sampleRate);
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H A D | HRTFDatabaseLoader.cpp | 43 PassRefPtr<HRTFDatabaseLoader> HRTFDatabaseLoader::createAndLoadAsynchronouslyIfNecessary(float sampleRate) argument 52 loader = s_loaderMap->get(sampleRate); 54 ASSERT(sampleRate == loader->databaseSampleRate()); 58 loader = adoptRef(new HRTFDatabaseLoader(sampleRate)); 59 s_loaderMap->add(sampleRate, loader.get()); 66 HRTFDatabaseLoader::HRTFDatabaseLoader(float sampleRate) argument 68 , 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:WebCore::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/third_party/WebKit/Source/testing/runner/ |
H A D | MockWebAudioDevice.cpp | 37 MockWebAudioDevice::MockWebAudioDevice(double sampleRate) argument 38 : m_sampleRate(sampleRate) 54 double MockWebAudioDevice::sampleRate() function in class:WebTestRunner::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/core/platform/audio/chromium/ |
H A D | AudioBusChromium.cpp | 36 PassRefPtr<AudioBus> decodeAudioFileData(const char* data, size_t size, double sampleRate) argument 39 if (WebKit::Platform::current()->loadAudioResource(&webAudioBus, data, size, sampleRate)) 44 PassRefPtr<AudioBus> AudioBus::loadPlatformResource(const char* name, float sampleRate) argument 50 // FIXME: the sampleRate parameter is ignored. It should be removed from the API. 51 RefPtr<AudioBus> audioBus = decodeAudioFileData(resource.data(), resource.size(), sampleRate); 57 if (audioBus->sampleRate() == sampleRate) 60 return AudioBus::createBySampleRateConverting(audioBus.get(), false, sampleRate); 63 PassRefPtr<AudioBus> createBusFromInMemoryAudioFile(const void* data, size_t dataSize, bool mixToMono, float sampleRate) argument 65 // FIXME: the sampleRate paramete [all...] |
H A D | AudioDestinationChromium.h | 48 AudioDestinationChromium(AudioIOCallback&, const String& inputDeviceId, unsigned numberOfInputChannels, unsigned numberOfOutputChannels, float sampleRate); 55 float sampleRate() const { return m_sampleRate; } function in class:WebCore::AudioDestinationChromium
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/external/chromium_org/third_party/WebKit/Source/core/platform/chromium/support/ |
H A D | WebAudioBus.cpp | 49 void WebAudioBus::initialize(unsigned numberOfChannels, size_t length, double sampleRate) argument 53 audioBus->setSampleRate(sampleRate); 114 double WebAudioBus::sampleRate() const function in class:WebKit::WebAudioBus 119 return m_private->sampleRate();
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