Searched refs:sampleRate (Results 1 - 25 of 217) sorted by relevance

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/external/chromium_org/third_party/WebKit/Source/platform/audio/
H A DAudioUtilities.cpp54 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();
H A DAudioUtilities.h39 // 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);
H A DPanner.cpp40 Panner* Panner::create(PanningModel model, float sampleRate, HRTFDatabaseLoader* databaseLoader) argument
46 panner = new EqualPowerPanner(sampleRate);
50 panner = new HRTFPanner(sampleRate, databaseLoader);
H A DAudioDSPKernel.h44 , 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(); }
H A DHRTFKernel.h54 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)
H A DAudioFileReader.h40 // 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
H A DHRTFDatabaseLoader.cpp50 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)
H A DAudioDelayDSPKernel.cpp47 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
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H A DAudioProcessor.h47 AudioProcessor(float sampleRate, unsigned numberOfChannels) argument
50 , m_sampleRate(sampleRate)
72 float sampleRate() const { return m_sampleRate; } function in class:blink::AudioProcessor
H A DHRTFDatabase.h46 static PassOwnPtr<HRTFDatabase> create(float sampleRate);
57 float sampleRate() const { return m_sampleRate; } function in class:blink::HRTFDatabase
63 explicit HRTFDatabase(float sampleRate);
/external/chromium_org/third_party/WebKit/Source/modules/webaudio/
H A DPeriodicWave.h43 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);
H A DAudioSourceNode.h39 AudioSourceNode(AudioContext* context, float sampleRate) argument
40 : AudioNode(context, sampleRate) { }
H A DOfflineAudioContext.h37 static OfflineAudioContext* create(ExecutionContext*, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState&);
42 OfflineAudioContext(Document*, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate);
H A DOfflineAudioContext.idl27 Constructor(unsigned long numberOfChannels, unsigned long numberOfFrames, float sampleRate),
H A DOfflineAudioContext.cpp40 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)
H A DDelayNode.cpp40 DelayNode::DelayNode(AudioContext* context, float sampleRate, double maxDelayTime, ExceptionState& exceptionState) argument
41 : AudioBasicProcessorNode(context, sampleRate)
51 m_processor = new DelayProcessor(context, sampleRate, 1, maxDelayTime);
H A DDelayNode.h40 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&);
H A DGainNode.h43 static GainNode* create(AudioContext* context, float sampleRate) argument
45 return adoptRefCountedGarbageCollectedWillBeNoop(new GainNode(context, sampleRate));
63 GainNode(AudioContext*, float sampleRate);
H A DChannelSplitterNode.cpp37 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)
H A DDelayDSPKernel.cpp42 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());
H A DChannelMergerNode.h42 static ChannelMergerNode* create(AudioContext*, float sampleRate, unsigned numberOfInputs);
56 ChannelMergerNode(AudioContext*, float sampleRate, unsigned numberOfInputs);
/external/aac/libAACenc/src/
H A Dbandwidth.h101 INT sampleRate,
/external/chromium_org/content/shell/renderer/test_runner/
H A Dmock_web_audio_device.cc18 double MockWebAudioDevice::sampleRate() { function in class:content::MockWebAudioDevice
/external/chromium_org/third_party/WebKit/public/platform/
H A DWebAudioDestinationConsumer.h36 virtual void setFormat(size_t numberOfChannels, float sampleRate) = 0;
H A DWebAudioSourceProviderClient.h32 virtual void setFormat(size_t numberOfChannels, float sampleRate) = 0;

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