/external/chromium_org/third_party/WebKit/Source/modules/webaudio/ |
H A D | AnalyserNode.h | 39 static AnalyserNode* create(AudioContext* context, float sampleRate) argument 41 return adoptRefCountedGarbageCollectedWillBeNoop(new AnalyserNode(context, sampleRate)); 73 AnalyserNode(AudioContext*, float sampleRate);
|
H A D | AudioBasicInspectorNode.cpp | 37 AudioBasicInspectorNode::AudioBasicInspectorNode(AudioContext* context, float sampleRate, unsigned outputChannelCount) argument 38 : AudioNode(context, sampleRate)
|
H A D | BiquadFilterNode.h | 50 static BiquadFilterNode* create(AudioContext* context, float sampleRate) argument 52 return adoptRefCountedGarbageCollectedWillBeNoop(new BiquadFilterNode(context, sampleRate)); 70 BiquadFilterNode(AudioContext*, float sampleRate);
|
H A D | ChannelMergerNode.cpp | 43 ChannelMergerNode* ChannelMergerNode::create(AudioContext* context, float sampleRate, unsigned numberOfInputs) argument 48 return adoptRefCountedGarbageCollectedWillBeNoop(new ChannelMergerNode(context, sampleRate, numberOfInputs)); 51 ChannelMergerNode::ChannelMergerNode(AudioContext* context, float sampleRate, unsigned numberOfInputs) argument 52 : AudioNode(context, sampleRate)
|
H A D | DynamicsCompressorNode.h | 39 static DynamicsCompressorNode* create(AudioContext* context, float sampleRate) argument 41 return adoptRefCountedGarbageCollectedWillBeNoop(new DynamicsCompressorNode(context, sampleRate)); 68 DynamicsCompressorNode(AudioContext*, float sampleRate);
|
H A D | WaveShaperProcessor.cpp | 35 WaveShaperProcessor::WaveShaperProcessor(float sampleRate, size_t numberOfChannels) argument 36 : AudioDSPKernelProcessor(sampleRate, numberOfChannels)
|
H A D | AudioBasicProcessorNode.cpp | 39 AudioBasicProcessorNode::AudioBasicProcessorNode(AudioContext* context, float sampleRate) argument 40 : AudioNode(context, sampleRate)
|
H A D | AudioBuffer.cpp | 46 AudioBuffer* AudioBuffer::create(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate) argument 48 if (!AudioUtilities::isValidAudioBufferSampleRate(sampleRate) || numberOfChannels > AudioContext::maxNumberOfChannels() || !numberOfChannels || !numberOfFrames) 51 AudioBuffer* buffer = new AudioBuffer(numberOfChannels, numberOfFrames, sampleRate); 58 AudioBuffer* AudioBuffer::create(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState& exceptionState) argument 73 if (!AudioUtilities::isValidAudioBufferSampleRate(sampleRate)) { 78 sampleRate, 96 AudioBuffer* audioBuffer = create(numberOfChannels, numberOfFrames, sampleRate); 104 + String::number(sampleRate) 111 AudioBuffer* AudioBuffer::createFromAudioFileData(const void* data, size_t dataSize, bool mixToMono, float sampleRate) argument 113 RefPtr<AudioBus> bus = createBusFromInMemoryAudioFile(data, dataSize, mixToMono, sampleRate); 138 AudioBuffer(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate) argument [all...] |
H A D | AudioListener.cpp | 82 void AudioListener::createAndLoadHRTFDatabaseLoader(float sampleRate) argument 85 m_hrtfDatabaseLoader = HRTFDatabaseLoader::createAndLoadAsynchronouslyIfNecessary(sampleRate);
|
H A D | AudioParam.cpp | 158 double sampleRate = context()->sampleRate(); local 160 double endTime = startTime + numberOfValues / sampleRate; 164 m_value = m_timeline.valuesForTimeRange(startTime, endTime, narrowPrecisionToFloat(m_value), values, numberOfValues, sampleRate, sampleRate);
|
H A D | AudioScheduledSourceNode.cpp | 44 AudioScheduledSourceNode::AudioScheduledSourceNode(AudioContext* context, float sampleRate) argument 45 : AudioSourceNode(context, sampleRate) 66 double sampleRate = this->sampleRate(); local 74 size_t startFrame = AudioUtilities::timeToSampleFrame(m_startTime, sampleRate); 75 size_t endFrame = m_endTime == UnknownTime ? 0 : AudioUtilities::timeToSampleFrame(m_endTime, sampleRate);
|
H A D | BiquadProcessor.cpp | 35 BiquadProcessor::BiquadProcessor(AudioContext* context, float sampleRate, size_t numberOfChannels, bool autoInitialize) argument 36 : AudioDSPKernelProcessor(sampleRate, numberOfChannels)
|
H A D | ConvolverNode.cpp | 50 ConvolverNode::ConvolverNode(AudioContext* context, float sampleRate) argument 51 : AudioNode(context, sampleRate) 124 if (buffer->sampleRate() != context()->sampleRate()) { 127 "The buffer sample rate of " + String::number(buffer->sampleRate()) 128 + " does not match the context rate of " + String::number(context()->sampleRate()) 147 bufferBus->setSampleRate(buffer->sampleRate()); 171 return m_reverb ? m_reverb->impulseResponseLength() / static_cast<double>(sampleRate()) : 0; 181 return m_reverb ? m_reverb->latencyFrames() / static_cast<double>(sampleRate()) : 0;
|
H A D | DynamicsCompressorNode.cpp | 41 DynamicsCompressorNode::DynamicsCompressorNode(AudioContext* context, float sampleRate) argument 42 : AudioNode(context, sampleRate) 99 m_dynamicsCompressor = adoptPtr(new DynamicsCompressor(sampleRate(), defaultNumberOfOutputChannels));
|
H A D | ScriptProcessorNode.cpp | 59 ScriptProcessorNode* ScriptProcessorNode::create(AudioContext* context, float sampleRate, size_t bufferSize, unsigned numberOfInputChannels, unsigned numberOfOutputChannels) argument 87 return adoptRefCountedGarbageCollectedWillBeNoop(new ScriptProcessorNode(context, sampleRate, bufferSize, numberOfInputChannels, numberOfOutputChannels)); 90 ScriptProcessorNode::ScriptProcessorNode(AudioContext* context, float sampleRate, size_t bufferSize, unsigned numberOfInputChannels, unsigned numberOfOutputChannels) argument 91 : AudioNode(context, sampleRate) 130 float sampleRate = context()->sampleRate(); local 135 AudioBuffer* inputBuffer = m_numberOfInputChannels ? AudioBuffer::create(m_numberOfInputChannels, bufferSize(), sampleRate) : 0; 136 AudioBuffer* outputBuffer = m_numberOfOutputChannels ? AudioBuffer::create(m_numberOfOutputChannels, bufferSize(), sampleRate) : 0; 259 double playbackTime = (context()->currentSampleFrame() + m_bufferSize) / static_cast<double>(context()->sampleRate());
|
/external/chromium_org/third_party/WebKit/Source/platform/audio/ |
H A D | AudioDSPKernelProcessor.cpp | 43 AudioDSPKernelProcessor::AudioDSPKernelProcessor(float sampleRate, unsigned numberOfChannels) argument 44 : AudioProcessor(sampleRate, numberOfChannels)
|
H A D | AudioDestination.h | 50 AudioDestination(AudioIOCallback&, const String& inputDeviceId, unsigned numberOfInputChannels, unsigned numberOfOutputChannels, float sampleRate); 55 static PassOwnPtr<AudioDestination> create(AudioIOCallback&, const String& inputDeviceId, unsigned numberOfInputChannels, unsigned numberOfOutputChannels, float sampleRate); 61 float sampleRate() const { return m_sampleRate; } function in class:blink::AudioDestination
|
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:blink::HRTFElevation 86 static bool calculateKernelsForAzimuthElevation(int azimuth, int elevation, float sampleRate, const String& subjectName, 90 HRTFElevation(PassOwnPtr<HRTFKernelList> kernelListL, PassOwnPtr<HRTFKernelList> kernelListR, int elevation, float sampleRate) argument 94 , m_sampleRate(sampleRate)
|
H A D | HRTFKernel.cpp | 68 HRTFKernel::HRTFKernel(AudioChannel* channel, size_t fftSize, float sampleRate) argument 70 , m_sampleRate(sampleRate) 84 unsigned numberOfFadeOutFrames = static_cast<unsigned>(sampleRate / 4410); // 10 sample-frames @44.1KHz sample-rate 119 float sampleRate1 = kernel1->sampleRate(); 120 float sampleRate2 = kernel2->sampleRate();
|
H A D | AudioBus.h | 90 float sampleRate() const { return m_sampleRate; } function in class:blink::AudioBus 91 void setSampleRate(float sampleRate) { m_sampleRate = sampleRate; } argument 147 static PassRefPtr<AudioBus> loadPlatformResource(const char* name, float sampleRate);
|
H A D | DynamicsCompressor.cpp | 43 DynamicsCompressor::DynamicsCompressor(float sampleRate, unsigned numberOfChannels) argument 45 , m_sampleRate(sampleRate) 46 , m_compressor(sampleRate, numberOfChannels)
|
/external/sonivox/arm-fm-22k/lib_src/ |
H A D | eas_pcm.h | 45 EAS_U32 sampleRate; member in struct:s_pcm_open_params_tag
|
/external/sonivox/arm-hybrid-22k/lib_src/ |
H A D | eas_pcm.h | 45 EAS_U32 sampleRate; member in struct:s_pcm_open_params_tag
|
/external/sonivox/arm-wt-22k/lib_src/ |
H A D | eas_pcm.h | 45 EAS_U32 sampleRate; member in struct:s_pcm_open_params_tag
|
/external/chromium_org/media/base/android/java/src/org/chromium/media/ |
H A D | WebAudioMediaCodecBridge.java | 71 int sampleRate = format.getInteger(MediaFormat.KEY_SAMPLE_RATE); 159 Log.d(LOG_TAG, "Final: Rate: " + sampleRate + 166 sampleRate, 187 sampleRate = newFormat.getInteger(MediaFormat.KEY_SAMPLE_RATE); 208 int sampleRate, 205 nativeInitializeDestination( long nativeWebAudioMediaCodecBridge, int inputChannelCount, int sampleRate, long durationMicroseconds) argument
|