/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
|
H A D | MockWebAudioDevice.h | 41 explicit MockWebAudioDevice(double sampleRate); 46 virtual double sampleRate();
|
/external/chromium_org/third_party/WebKit/Source/core/platform/audio/ |
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));
|
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);
|
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));
|
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(); }
|
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)
|
H A D | AudioDestination.h | 48 static PassOwnPtr<AudioDestination> create(AudioIOCallback&, const String& inputDeviceId, unsigned numberOfInputChannels, unsigned numberOfOutputChannels, float sampleRate); 57 virtual float sampleRate() const = 0;
|
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
|
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)
|
H A D | HRTFElevation.h | 53 static PassOwnPtr<HRTFElevation> createForSubject(const String& subjectName, int elevation, float sampleRate); 56 static PassOwnPtr<HRTFElevation> createByInterpolatingSlices(HRTFElevation* hrtfElevation1, HRTFElevation* hrtfElevation2, float x, float sampleRate); 64 float sampleRate() const { return m_sampleRate; } function in class:WebCore::HRTFElevation 86 static bool calculateKernelsForAzimuthElevation(int azimuth, int elevation, float sampleRate, const String& subjectName, 92 static bool calculateSymmetricKernelsForAzimuthElevation(int azimuth, int elevation, float sampleRate, const String& subjectName, 96 HRTFElevation(PassOwnPtr<HRTFKernelList> kernelListL, PassOwnPtr<HRTFKernelList> kernelListR, int elevation, float sampleRate) argument 100 , m_sampleRate(sampleRate)
|
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
|
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);
|
/external/jmonkeyengine/engine/src/terrain/com/jme3/terrain/noise/modulator/ |
H A D | CatRom2.java | 39 private int sampleRate = 100;
field in class:CatRom2 45 public CatRom2(final int sampleRate) {
argument 46 this.sampleRate = sampleRate;
47 this.table = new float[4 * sampleRate + 1];
48 for (int i = 0; i < 4 * sampleRate + 1; i++) {
49 float x = i / (float) sampleRate;
59 public static CatRom2 getInstance(final int sampleRate) {
argument 60 if (!CatRom2.instances.containsKey(sampleRate)) {
61 CatRom2.instances.put(sampleRate, ne [all...] |
/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...] |
/external/jmonkeyengine/engine/src/core/com/jme3/audio/ |
H A D | AudioData.java | 47 protected int sampleRate; field in class:AudioData 92 return sampleRate; 99 * @param sampleRate Sample rate, 44100, 22050, etc. 101 public void setupFormat(int channels, int bitsPerSample, int sampleRate){ argument 107 this.sampleRate = sampleRate;
|
/external/chromium_org/third_party/WebKit/Source/modules/webaudio/ |
H A D | PeriodicWave.h | 45 static PassRefPtr<PeriodicWave> createSine(float sampleRate); 46 static PassRefPtr<PeriodicWave> createSquare(float sampleRate); 47 static PassRefPtr<PeriodicWave> createSawtooth(float sampleRate); 48 static PassRefPtr<PeriodicWave> createTriangle(float sampleRate); 51 static PassRefPtr<PeriodicWave> create(float sampleRate, Float32Array* real, Float32Array* imag); 65 float sampleRate() const { return m_sampleRate; } function in class:WebCore::PeriodicWave 68 explicit PeriodicWave(float sampleRate);
|
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));
|
H A D | AudioSourceNode.h | 38 AudioSourceNode(AudioContext* context, float sampleRate) argument 39 : AudioNode(context, sampleRate)
|
H A D | DelayProcessor.cpp | 35 DelayProcessor::DelayProcessor(AudioContext* context, float sampleRate, unsigned numberOfChannels, double maxDelayTime) argument 36 : AudioDSPKernelProcessor(sampleRate, numberOfChannels)
|
H A D | OfflineAudioContext.h | 36 static PassRefPtr<OfflineAudioContext> create(ScriptExecutionContext*, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState&); 41 OfflineAudioContext(Document*, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate);
|
H A D | OfflineAudioContext.idl | 28 Constructor(unsigned long numberOfChannels, unsigned long numberOfFrames, float sampleRate),
|
H A D | DelayDSPKernel.cpp | 46 ASSERT(processor && processor->sampleRate() > 0); 47 if (!(processor && processor->sampleRate() > 0)) 55 m_buffer.allocate(bufferLengthForDelay(m_maxDelayTime, processor->sampleRate())); 58 m_smoothingRate = AudioUtilities::discreteTimeConstantForSampleRate(SmoothingTimeConstant, processor->sampleRate()); 61 DelayDSPKernel::DelayDSPKernel(double maxDelayTime, float sampleRate) argument 62 : AudioDSPKernel(sampleRate) 71 size_t bufferLength = bufferLengthForDelay(maxDelayTime, sampleRate); 79 m_smoothingRate = AudioUtilities::discreteTimeConstantForSampleRate(SmoothingTimeConstant, sampleRate); 82 size_t DelayDSPKernel::bufferLengthForDelay(double maxDelayTime, double sampleRate) const 86 return 1 + AudioUtilities::timeToSampleFrame(maxDelayTime, sampleRate); 102 float sampleRate = this->sampleRate(); local [all...] |
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);
|
H A D | OfflineAudioContext.cpp | 38 PassRefPtr<OfflineAudioContext> OfflineAudioContext::create(ScriptExecutionContext* context, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState& es) argument 48 if (numberOfChannels > 10 || !isSampleRateRangeGood(sampleRate)) { 53 RefPtr<OfflineAudioContext> audioContext(adoptRef(new OfflineAudioContext(document, numberOfChannels, numberOfFrames, sampleRate))); 58 OfflineAudioContext::OfflineAudioContext(Document* document, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate) argument 59 : AudioContext(document, numberOfChannels, numberOfFrames, sampleRate)
|