| /external/webkit/Source/WebCore/platform/audio/ |
| FFTConvolver.h | 39 // fftSize must be a power of two
40 FFTConvolver(size_t fftSize);
42 // For now, with multiple calls to Process(), framesToProcess MUST add up EXACTLY to fftSize / 2
46 // The input to output latency is equal to fftSize / 2
53 size_t fftSize() const { return m_frame.fftSize(); }
58 // Buffer input until we get fftSize / 2 samples then do an FFT
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| FFTConvolver.cpp | 41 FFTConvolver::FFTConvolver(size_t fftSize)
42 : m_frame(fftSize)
44 , m_inputBuffer(fftSize) // 2nd half of buffer is always zeroed
45 , m_outputBuffer(fftSize)
46 , m_lastOverlapBuffer(fftSize / 2)
52 // FIXME: make so framesToProcess is not required to fit evenly into fftSize/2
79 size_t halfSize = fftSize() / 2;
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| FFTFrame.h | 60 FFTFrame(unsigned fftSize);
82 void doPaddedFFT(float* data, size_t dataSize); // zero-padding with dataSize <= fftSize
86 unsigned fftSize() const { return m_FFTSize; }
99 static FFTSetup fftSetupForSize(unsigned fftSize);
117 static DFTI_DESCRIPTOR_HANDLE descriptorHandleForSize(unsigned fftSize);
146 static fftwf_plan fftwPlanForSize(unsigned fftSize, Direction,
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| HRTFKernel.h | 54 static PassRefPtr<HRTFKernel> create(AudioChannel* channel, size_t fftSize, double sampleRate, bool bassBoost)
56 return adoptRef(new HRTFKernel(channel, fftSize, sampleRate, bassBoost));
69 size_t fftSize() const { return m_fftFrame->fftSize(); }
80 HRTFKernel(AudioChannel* channel, size_t fftSize, double sampleRate, bool bassBoost);
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| ReverbConvolverStage.cpp | 47 size_t fftSize, size_t renderPhase, size_t renderSliceSize, ReverbAccumulationBuffer* accumulationBuffer)
48 : m_fftKernel(fftSize)
58 m_convolver = adoptPtr(new FFTConvolver(fftSize));
64 // But, the FFT convolution itself incurs fftSize / 2 latency, so subtract this out...
65 size_t halfSize = fftSize / 2;
81 m_preDelayBuffer.resize(m_preDelayLength < fftSize ? fftSize : m_preDelayLength);
133 // An expensive FFT will happen every fftSize / 2 frames.
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| ReverbConvolver.cpp | 93 size_t fftSize = m_minFFTSize;
95 size_t stageSize = fftSize / 2;
105 OwnPtr<ReverbConvolverStage> stage(new ReverbConvolverStage(response, totalResponseLength, reverbTotalLatency, stageOffset, stageSize, fftSize, renderPhase, renderSliceSize, &m_accumulationBuffer));
119 fftSize *= 2;
120 if (hasRealtimeConstraint && !isBackgroundStage && fftSize > m_maxRealtimeFFTSize)
121 fftSize = m_maxRealtimeFFTSize;
122 if (fftSize > m_maxFFTSize)
123 fftSize = m_maxFFTSize;
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| FFTFrame.cpp | 48 AudioFloatArray paddedResponse(fftSize()); // zero-initialized
57 OwnPtr<FFTFrame> newFrame = adoptPtr(new FFTFrame(frame1.fftSize()));
62 int fftSize = newFrame->fftSize();
63 AudioFloatArray buffer(fftSize);
65 buffer.zeroRange(fftSize / 2, fftSize);
85 m_FFTSize = frame1.fftSize();
184 int halfSize = fftSize() / 2;
186 const double kSamplePhaseDelay = (2.0 * piDouble) / double(fftSize());
[all...] |
| HRTFKernel.cpp | 68 HRTFKernel::HRTFKernel(AudioChannel* channel, size_t fftSize, double sampleRate, bool bassBoost)
75 m_frameDelay = extractAverageGroupDelay(channel, fftSize / 2);
90 size_t truncatedResponseLength = min(responseLength, fftSize / 2); // truncate if necessary to max impulse response length allowed by FFT
102 m_fftFrame = adoptPtr(new FFTFrame(fftSize));
108 OwnPtr<AudioChannel> channel = adoptPtr(new AudioChannel(fftSize()));
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| HRTFPanner.h | 43 size_t fftSize() { return fftSizeForSampleRate(m_sampleRate); }
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| ReverbConvolverStage.h | 49 size_t fftSize, size_t renderPhase, size_t renderSliceSize, ReverbAccumulationBuffer* accumulationBuffer);
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| HRTFElevation.cpp | 121 // Note that depending on the fftSize returned by the panner, we may be truncating the impulse response we just loaded in.
122 const size_t fftSize = HRTFPanner::fftSizeForSampleRate(sampleRate);
123 kernelL = HRTFKernel::create(leftEarImpulseResponse, fftSize, sampleRate, true);
124 kernelR = HRTFKernel::create(rightEarImpulseResponse, fftSize, sampleRate, true);
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| /cts/apps/CtsVerifier/jni/audioquality/ |
| CompareSpectra.cpp | 70 /* Return a pointer to 1+(fftSize/2) spectrum magnitude values
74 fftSize points is used to compute the spectra, The overall signal
77 double* getAverageSpectrum(short* pcm, int numSamples, int fftSize,
79 if (numSamples < fftSize) return NULL;
80 int numFrames = 1 + ((2 * (numSamples - fftSize)) / fftSize);
81 int numMag = 1 + (fftSize / 2);
82 float* re = new float[fftSize];
83 float* im = new float[fftSize];
88 Window wind(fftSize);
[all...] |
| /external/webkit/Source/WebCore/platform/audio/fftw/ |
| FFTFrameFFTW.cpp | 49 unsigned unpackedFFTWDataSize(unsigned fftSize)
51 return fftSize / 2 + 1;
57 // Normal constructor: allocates for a given fftSize.
58 FFTFrame::FFTFrame(unsigned fftSize)
59 : m_FFTSize(fftSize)
60 , m_log2FFTSize(static_cast<unsigned>(log2(fftSize)))
63 , m_data(2 * (3 + unpackedFFTWDataSize(fftSize))) // enough space for real and imaginary data plus 16-byte alignment padding
81 m_forwardPlan = fftwPlanForSize(fftSize, Forward,
83 m_backwardPlan = fftwPlanForSize(fftSize, Backward,
102 , m_data(2 * (3 + unpackedFFTWDataSize(fftSize()))) // enough space for real and imaginary data plus 16-byte alignment paddin
[all...] |
| /external/webkit/Source/WebCore/platform/audio/mkl/ |
| FFTFrameMKL.cpp | 42 DFTI_DESCRIPTOR_HANDLE createDescriptorHandle(int fftSize)
47 MKL_LONG status = DftiCreateDescriptor(&handle, DFTI_SINGLE, DFTI_REAL, 1, fftSize);
67 double scale = 1.0 / (2.0 * fftSize);
88 // Normal constructor: allocates for a given fftSize.
89 FFTFrame::FFTFrame(unsigned fftSize)
90 : m_FFTSize(fftSize)
91 , m_log2FFTSize(static_cast<unsigned>(log2(fftSize)))
93 , m_complexData(fftSize)
94 , m_realData(fftSize / 2)
95 , m_imagData(fftSize / 2
[all...] |
| /external/webkit/Source/WebCore/platform/audio/mac/ |
| FFTFrameMac.cpp | 43 // Normal constructor: allocates for a given fftSize
44 FFTFrame::FFTFrame(unsigned fftSize)
45 : m_realData(fftSize)
46 , m_imagData(fftSize)
48 m_FFTSize = fftSize;
49 m_log2FFTSize = static_cast<unsigned>(log2(fftSize));
55 m_FFTSetup = fftSetupForSize(fftSize);
150 FFTSetup FFTFrame::fftSetupForSize(unsigned fftSize)
157 int pow2size = static_cast<int>(log2(fftSize));
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| /external/webkit/Source/WebCore/webaudio/ |
| RealtimeAnalyserNode.idl | 30 attribute unsigned long fftSize;
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| RealtimeAnalyser.cpp | 108 // FIXME : allow to work with non-FFTSize divisible chunking
153 size_t fftSize = this->fftSize();
155 AudioFloatArray temporaryBuffer(fftSize);
159 // Take the previous fftSize values from the input buffer and copy into the temporary buffer.
162 for (unsigned i = 0; i < fftSize; ++i)
163 tempP[i] = inputBuffer[(i + writeIndex - fftSize + InputBufferSize) % InputBufferSize];
166 applyWindow(tempP, fftSize);
266 unsigned fftSize = this->fftSize();
[all...] |
| RealtimeAnalyserNode.h | 48 unsigned int fftSize() const { return m_analyser.fftSize(); }
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| RealtimeAnalyser.h | 50 size_t fftSize() const { return m_fftSize; }
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| /frameworks/base/include/media/ |
| Visualizer.h | 87 // fftSize) is 0.
91 uint32_t fftSize,
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| /frameworks/base/media/libmedia/ |
| Visualizer.cpp | 259 uint32_t fftSize = 0;
266 fftSize = mCaptureSize;
268 mCaptureCallBack(mCaptureCbkUser, waveSize, wavePtr, fftSize, fftPtr, mSampleRate);
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| /frameworks/base/media/jni/audioeffect/ |
| android_media_Visualizer.cpp | 99 uint32_t fftSize,
138 if (fftSize != 0 && fft != NULL) {
139 jbyteArray jArray = env->NewByteArray(fftSize);
142 memcpy(nArray, fft, fftSize);
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