/macosx-10.10/WebCore-7600.1.25/platform/audio/ |
H A D | AudioUtilities.h | 38 // discreteTimeConstantForSampleRate() will return the discrete time-constant for the specific sampleRate. 39 double discreteTimeConstantForSampleRate(double timeConstant, double sampleRate); 42 size_t timeToSampleFrame(double time, double sampleRate);
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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 | 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 | Panner.cpp | 40 std::unique_ptr<Panner> Panner::create(PanningModel model, float sampleRate, HRTFDatabaseLoader* databaseLoader) argument 46 panner = std::make_unique<EqualPowerPanner>(sampleRate); 50 panner = std::make_unique<HRTFPanner>(sampleRate, databaseLoader);
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H A D | HRTFKernel.h | 53 static PassRefPtr<HRTFKernel> create(AudioChannel* channel, size_t fftSize, float sampleRate) argument 55 return adoptRef(new HRTFKernel(channel, fftSize, sampleRate)); 58 static PassRefPtr<HRTFKernel> create(std::unique_ptr<FFTFrame> fftFrame, float frameDelay, float sampleRate) argument 60 return adoptRef(new HRTFKernel(WTF::move(fftFrame), frameDelay, sampleRate)); 71 float sampleRate() const { return m_sampleRate; } function in class:WebCore::HRTFKernel 72 double nyquist() const { return 0.5 * sampleRate(); } 79 HRTFKernel(AudioChannel*, size_t fftSize, float sampleRate); 81 HRTFKernel(std::unique_ptr<FFTFrame> fftFrame, float frameDelay, float sampleRate) argument 84 , m_sampleRate(sampleRate)
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H A D | HRTFDatabaseLoader.cpp | 48 PassRefPtr<HRTFDatabaseLoader> HRTFDatabaseLoader::createAndLoadAsynchronouslyIfNecessary(float sampleRate) argument 54 loader = loaderMap().get(sampleRate); 56 ASSERT(sampleRate == loader->databaseSampleRate()); 60 loader = adoptRef(new HRTFDatabaseLoader(sampleRate)); 61 loaderMap().add(sampleRate, loader.get()); 68 HRTFDatabaseLoader::HRTFDatabaseLoader(float sampleRate) argument 70 , m_databaseSampleRate(sampleRate)
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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 | AudioSourceProviderClient.h | 32 virtual void setFormat(size_t numberOfChannels, float sampleRate) = 0;
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H A D | AudioDestination.h | 47 static std::unique_ptr<AudioDestination> create(AudioIOCallback&, const String& inputDeviceId, unsigned numberOfInputChannels, unsigned numberOfOutputChannels, float sampleRate); 56 virtual float sampleRate() const = 0;
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H A D | AudioFileReader.h | 41 // Pass in 0.0 for sampleRate to use the file's sample-rate, otherwise a sample-rate conversion to the requested 42 // sampleRate will be made (if it doesn't already match the file's sample-rate). 45 PassRefPtr<AudioBus> createBusFromInMemoryAudioFile(const void* data, size_t dataSize, bool mixToMono, float sampleRate); 47 PassRefPtr<AudioBus> createBusFromAudioFile(const char* filePath, bool mixToMono, float sampleRate); 49 // May pass in 0.0 for sampleRate in which case it will use the AudioBus's sampleRate
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H A D | HRTFElevation.h | 48 HRTFElevation(std::unique_ptr<HRTFKernelList> kernelListL, std::unique_ptr<HRTFKernelList> kernelListR, int elevation, float sampleRate) argument 52 , m_sampleRate(sampleRate) 60 static std::unique_ptr<HRTFElevation> createForSubject(const String& subjectName, int elevation, float sampleRate); 63 static std::unique_ptr<HRTFElevation> createByInterpolatingSlices(HRTFElevation* hrtfElevation1, HRTFElevation* hrtfElevation2, float x, float sampleRate); 71 float sampleRate() const { return m_sampleRate; } function in class:WebCore::HRTFElevation 93 static bool calculateKernelsForAzimuthElevation(int azimuth, int elevation, float sampleRate, const String& subjectName, 99 static bool calculateSymmetricKernelsForAzimuthElevation(int azimuth, int elevation, float sampleRate, const String& subjectName,
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H A D | HRTFPanner.cpp | 48 HRTFPanner::HRTFPanner(float sampleRate, HRTFDatabaseLoader* databaseLoader) argument 51 , m_sampleRate(sampleRate) 59 , m_convolverL1(fftSizeForSampleRate(sampleRate)) 60 , m_convolverR1(fftSizeForSampleRate(sampleRate)) 61 , m_convolverL2(fftSizeForSampleRate(sampleRate)) 62 , m_convolverR2(fftSizeForSampleRate(sampleRate)) 63 , m_delayLineL(MaxDelayTimeSeconds, sampleRate) 64 , m_delayLineR(MaxDelayTimeSeconds, sampleRate) 77 size_t HRTFPanner::fftSizeForSampleRate(float sampleRate) argument 82 ASSERT(sampleRate > [all...] |
/macosx-10.10/WebCore-7600.1.25/Modules/webaudio/ |
H A D | DelayNode.cpp | 35 DelayNode::DelayNode(AudioContext* context, float sampleRate, double maxDelayTime, ExceptionCode& ec) argument 36 : AudioBasicProcessorNode(context, sampleRate) 42 m_processor = std::make_unique<DelayProcessor>(context, sampleRate, 1, maxDelayTime);
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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 | OfflineAudioContext.h | 34 static PassRefPtr<OfflineAudioContext> create(ScriptExecutionContext&, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionCode&); 39 OfflineAudioContext(Document&, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate);
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H A D | DelayDSPKernel.cpp | 44 ASSERT(processor && processor->sampleRate() > 0); 45 if (!(processor && processor->sampleRate() > 0)) 53 m_buffer.allocate(bufferLengthForDelay(m_maxDelayTime, processor->sampleRate())); 56 m_smoothingRate = AudioUtilities::discreteTimeConstantForSampleRate(SmoothingTimeConstant, processor->sampleRate()); 59 DelayDSPKernel::DelayDSPKernel(double maxDelayTime, float sampleRate) argument 60 : AudioDSPKernel(sampleRate) 69 size_t bufferLength = bufferLengthForDelay(maxDelayTime, sampleRate); 77 m_smoothingRate = AudioUtilities::discreteTimeConstantForSampleRate(SmoothingTimeConstant, sampleRate); 80 size_t DelayDSPKernel::bufferLengthForDelay(double maxDelayTime, double sampleRate) const 84 return 1 + AudioUtilities::timeToSampleFrame(maxDelayTime, sampleRate); 100 float sampleRate = this->sampleRate(); local [all...] |
H A D | OfflineAudioContext.cpp | 37 PassRefPtr<OfflineAudioContext> OfflineAudioContext::create(ScriptExecutionContext& context, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionCode& ec) argument 47 if (numberOfChannels > 10 || !isSampleRateRangeGood(sampleRate)) { 52 RefPtr<OfflineAudioContext> audioContext(adoptRef(new OfflineAudioContext(document, numberOfChannels, numberOfFrames, sampleRate))); 57 OfflineAudioContext::OfflineAudioContext(Document& document, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate) argument 58 : AudioContext(document, numberOfChannels, numberOfFrames, sampleRate)
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H A D | PeriodicWave.h | 44 static PassRefPtr<PeriodicWave> createSine(float sampleRate); 45 static PassRefPtr<PeriodicWave> createSquare(float sampleRate); 46 static PassRefPtr<PeriodicWave> createSawtooth(float sampleRate); 47 static PassRefPtr<PeriodicWave> createTriangle(float sampleRate); 50 static PassRefPtr<PeriodicWave> create(float sampleRate, Float32Array* real, Float32Array* imag); 64 float sampleRate() const { return m_sampleRate; } function in class:WebCore::PeriodicWave 67 explicit PeriodicWave(float sampleRate);
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H A D | DelayNode.h | 39 static PassRefPtr<DelayNode> create(AudioContext* context, float sampleRate, double maxDelayTime, ExceptionCode& ec) argument 41 return adoptRef(new DelayNode(context, sampleRate, maxDelayTime, ec)); 47 DelayNode(AudioContext*, float sampleRate, double maxDelayTime, ExceptionCode&);
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H A D | ChannelSplitterNode.h | 37 static PassRefPtr<ChannelSplitterNode> create(AudioContext*, float sampleRate, unsigned numberOfOutputs); 47 ChannelSplitterNode(AudioContext*, float sampleRate, unsigned numberOfOutputs);
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H A D | OfflineAudioContext.idl | 29 Constructor(unsigned long numberOfChannels, unsigned long numberOfFrames, unrestricted float sampleRate),
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H A D | AsyncAudioDecoder.cpp | 54 void AsyncAudioDecoder::decodeAsync(ArrayBuffer* audioData, float sampleRate, PassRefPtr<AudioBufferCallback> successCallback, PassRefPtr<AudioBufferCallback> errorCallback) argument 61 auto decodingTask = DecodingTask::create(audioData, sampleRate, successCallback, errorCallback); 90 std::unique_ptr<AsyncAudioDecoder::DecodingTask> AsyncAudioDecoder::DecodingTask::create(ArrayBuffer* audioData, float sampleRate, PassRefPtr<AudioBufferCallback> successCallback, PassRefPtr<AudioBufferCallback> errorCallback) argument 92 return std::unique_ptr<DecodingTask>(new DecodingTask(audioData, sampleRate, successCallback, errorCallback)); 95 AsyncAudioDecoder::DecodingTask::DecodingTask(ArrayBuffer* audioData, float sampleRate, PassRefPtr<AudioBufferCallback> successCallback, PassRefPtr<AudioBufferCallback> errorCallback) argument 97 , m_sampleRate(sampleRate) 110 m_audioBuffer = AudioBuffer::createFromAudioFileData(m_audioData->data(), m_audioData->byteLength(), false, sampleRate());
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H A D | AudioBuffer.cpp | 46 PassRefPtr<AudioBuffer> AudioBuffer::create(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate) argument 48 if (sampleRate < 22050 || sampleRate > 96000 || numberOfChannels > AudioContext::maxNumberOfChannels() || !numberOfFrames) 51 return adoptRef(new AudioBuffer(numberOfChannels, numberOfFrames, sampleRate)); 54 PassRefPtr<AudioBuffer> AudioBuffer::createFromAudioFileData(const void* data, size_t dataSize, bool mixToMono, float sampleRate) argument 56 RefPtr<AudioBus> bus = createBusFromInMemoryAudioFile(data, dataSize, mixToMono, sampleRate); 63 AudioBuffer::AudioBuffer(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate) argument 65 , m_sampleRate(sampleRate) 79 , m_sampleRate(bus->sampleRate())
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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 74 AudioBuffer(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate);
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/macosx-10.10/WebCore-7600.1.25/platform/audio/efl/ |
H A D | AudioBusEfl.cpp | 31 PassRefPtr<AudioBus> AudioBus::loadPlatformResource(const char* name, float sampleRate) argument 37 return createBusFromAudioFile(absoluteFilename.utf8().data(), false, sampleRate);
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