Searched defs:sampleRate (Results 26 - 50 of 129) sorted by relevance

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/external/chromium_org/third_party/WebKit/Source/modules/webaudio/
H A DAnalyserNode.h39 static AnalyserNode* create(AudioContext* context, float sampleRate) argument
41 return adoptRefCountedGarbageCollectedWillBeNoop(new AnalyserNode(context, sampleRate));
73 AnalyserNode(AudioContext*, float sampleRate);
H A DAudioBasicInspectorNode.cpp37 AudioBasicInspectorNode::AudioBasicInspectorNode(AudioContext* context, float sampleRate, unsigned outputChannelCount) argument
38 : AudioNode(context, sampleRate)
H A DBiquadFilterNode.h50 static BiquadFilterNode* create(AudioContext* context, float sampleRate) argument
52 return adoptRefCountedGarbageCollectedWillBeNoop(new BiquadFilterNode(context, sampleRate));
70 BiquadFilterNode(AudioContext*, float sampleRate);
H A DChannelMergerNode.cpp43 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 DDynamicsCompressorNode.h39 static DynamicsCompressorNode* create(AudioContext* context, float sampleRate) argument
41 return adoptRefCountedGarbageCollectedWillBeNoop(new DynamicsCompressorNode(context, sampleRate));
68 DynamicsCompressorNode(AudioContext*, float sampleRate);
H A DWaveShaperProcessor.cpp35 WaveShaperProcessor::WaveShaperProcessor(float sampleRate, size_t numberOfChannels) argument
36 : AudioDSPKernelProcessor(sampleRate, numberOfChannels)
H A DAudioBasicProcessorNode.cpp39 AudioBasicProcessorNode::AudioBasicProcessorNode(AudioContext* context, float sampleRate) argument
40 : AudioNode(context, sampleRate)
H A DAudioBuffer.cpp46 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
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H A DAudioListener.cpp82 void AudioListener::createAndLoadHRTFDatabaseLoader(float sampleRate) argument
85 m_hrtfDatabaseLoader = HRTFDatabaseLoader::createAndLoadAsynchronouslyIfNecessary(sampleRate);
H A DAudioParam.cpp158 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 DAudioScheduledSourceNode.cpp44 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 DBiquadProcessor.cpp35 BiquadProcessor::BiquadProcessor(AudioContext* context, float sampleRate, size_t numberOfChannels, bool autoInitialize) argument
36 : AudioDSPKernelProcessor(sampleRate, numberOfChannels)
H A DConvolverNode.cpp50 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 DDynamicsCompressorNode.cpp41 DynamicsCompressorNode::DynamicsCompressorNode(AudioContext* context, float sampleRate) argument
42 : AudioNode(context, sampleRate)
99 m_dynamicsCompressor = adoptPtr(new DynamicsCompressor(sampleRate(), defaultNumberOfOutputChannels));
H A DScriptProcessorNode.cpp59 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 DAudioDSPKernelProcessor.cpp43 AudioDSPKernelProcessor::AudioDSPKernelProcessor(float sampleRate, unsigned numberOfChannels) argument
44 : AudioProcessor(sampleRate, numberOfChannels)
H A DAudioDestination.h50 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 DHRTFElevation.h53 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 DHRTFKernel.cpp68 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 DAudioBus.h90 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 DDynamicsCompressor.cpp43 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 Deas_pcm.h45 EAS_U32 sampleRate; member in struct:s_pcm_open_params_tag
/external/sonivox/arm-hybrid-22k/lib_src/
H A Deas_pcm.h45 EAS_U32 sampleRate; member in struct:s_pcm_open_params_tag
/external/sonivox/arm-wt-22k/lib_src/
H A Deas_pcm.h45 EAS_U32 sampleRate; member in struct:s_pcm_open_params_tag
/external/chromium_org/media/base/android/java/src/org/chromium/media/
H A DWebAudioMediaCodecBridge.java71 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

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