third_party/WebKit/Source/modules/webaudio/RealtimeAnalyser.cpp - chromium/src - Git at Google

 /*
  * Copyright (C) 2010, Google Inc. All rights reserved.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
  * are met:
  * 1.  Redistributions of source code must retain the above copyright
  *    notice, this list of conditions and the following disclaimer.
  * 2.  Redistributions in binary form must reproduce the above copyright
  *    notice, this list of conditions and the following disclaimer in the
  *    documentation and/or other materials provided with the distribution.
  *
  * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' AND
  * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
  * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
  * ARE DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS BE LIABLE
  * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
  * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
  * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
  * CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
  * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
  * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH
  * DAMAGE.
  */

 #include "modules/webaudio/RealtimeAnalyser.h"
 #include "platform/audio/AudioBus.h"
 #include "platform/audio/AudioUtilities.h"
 #include "platform/audio/VectorMath.h"
 #include "wtf/MathExtras.h"
 #include "wtf/PtrUtil.h"
 #include <algorithm>
 #include <complex>
 #include <limits.h>

 namespace blink {

 const double RealtimeAnalyser::DefaultSmoothingTimeConstant = 0.8;
 const double RealtimeAnalyser::DefaultMinDecibels = -100;
 const double RealtimeAnalyser::DefaultMaxDecibels = -30;

 const unsigned RealtimeAnalyser::DefaultFFTSize = 2048;
 // All FFT implementations are expected to handle power-of-two sizes
 // MinFFTSize <= size <= MaxFFTSize.
 const unsigned RealtimeAnalyser::MinFFTSize = 32;
 const unsigned RealtimeAnalyser::MaxFFTSize = 32768;
 const unsigned RealtimeAnalyser::InputBufferSize =
     RealtimeAnalyser::MaxFFTSize * 2;

 RealtimeAnalyser::RealtimeAnalyser()
     : m_inputBuffer(InputBufferSize),
       m_writeIndex(0),
       m_downMixBus(AudioBus::create(1, AudioUtilities::kRenderQuantumFrames)),
       m_fftSize(DefaultFFTSize),
       m_magnitudeBuffer(DefaultFFTSize / 2),
       m_smoothingTimeConstant(DefaultSmoothingTimeConstant),
       m_minDecibels(DefaultMinDecibels),
       m_maxDecibels(DefaultMaxDecibels),
       m_lastAnalysisTime(-1) {
   m_analysisFrame = wrapUnique(new FFTFrame(DefaultFFTSize));
 }

 bool RealtimeAnalyser::setFftSize(size_t size) {
   DCHECK(isMainThread());

   // Only allow powers of two.
   unsigned log2size = static_cast<unsigned>(log2(size));
   bool isPOT(1UL << log2size == size);

   if (!isPOT || size > MaxFFTSize || size < MinFFTSize)
     return false;

   if (m_fftSize != size) {
     m_analysisFrame = wrapUnique(new FFTFrame(size));
     // m_magnitudeBuffer has size = fftSize / 2 because it contains floats
     // reduced from complex values in m_analysisFrame.
     m_magnitudeBuffer.allocate(size / 2);
     m_fftSize = size;
   }

   return true;
 }

 void RealtimeAnalyser::writeInput(AudioBus* bus, size_t framesToProcess) {
   bool isBusGood = bus && bus->numberOfChannels() > 0 &&
                    bus->channel(0)->length() >= framesToProcess;
   DCHECK(isBusGood);
   if (!isBusGood)
     return;

   // FIXME : allow to work with non-FFTSize divisible chunking
   bool isDestinationGood =
       m_writeIndex < m_inputBuffer.size() &&
       m_writeIndex + framesToProcess <= m_inputBuffer.size();
   DCHECK(isDestinationGood);
   if (!isDestinationGood)
     return;

   // Perform real-time analysis
   float* dest = m_inputBuffer.data() + m_writeIndex;

   // Clear the bus and downmix the input according to the down mixing rules.
   // Then save the result in the m_inputBuffer at the appropriate place.
   m_downMixBus->zero();
   m_downMixBus->sumFrom(*bus);
   memcpy(dest, m_downMixBus->channel(0)->data(),
          framesToProcess * sizeof(*dest));

   m_writeIndex += framesToProcess;
   if (m_writeIndex >= InputBufferSize)
     m_writeIndex = 0;
 }

 namespace {

 void applyWindow(float* p, size_t n) {
   DCHECK(isMainThread());

   // Blackman window
   double alpha = 0.16;
   double a0 = 0.5 * (1 - alpha);
   double a1 = 0.5;
   double a2 = 0.5 * alpha;

   for (unsigned i = 0; i < n; ++i) {
     double x = static_cast<double>(i) / static_cast<double>(n);
     double window =
         a0 - a1 * cos(twoPiDouble * x) + a2 * cos(twoPiDouble * 2.0 * x);
     p[i] *= float(window);
   }
 }

 }  // namespace

 void RealtimeAnalyser::doFFTAnalysis() {
   DCHECK(isMainThread());

   // Unroll the input buffer into a temporary buffer, where we'll apply an
   // analysis window followed by an FFT.
   size_t fftSize = this->fftSize();

   AudioFloatArray temporaryBuffer(fftSize);
   float* inputBuffer = m_inputBuffer.data();
   float* tempP = temporaryBuffer.data();

   // Take the previous fftSize values from the input buffer and copy into the
   // temporary buffer.
   unsigned writeIndex = m_writeIndex;
   if (writeIndex < fftSize) {
     memcpy(tempP, inputBuffer + writeIndex - fftSize + InputBufferSize,
            sizeof(*tempP) * (fftSize - writeIndex));
     memcpy(tempP + fftSize - writeIndex, inputBuffer,
            sizeof(*tempP) * writeIndex);
   } else {
     memcpy(tempP, inputBuffer + writeIndex - fftSize, sizeof(*tempP) * fftSize);
   }

   // Window the input samples.
   applyWindow(tempP, fftSize);

   // Do the analysis.
   m_analysisFrame->doFFT(tempP);

   float* realP = m_analysisFrame->realData();
   float* imagP = m_analysisFrame->imagData();

   // Blow away the packed nyquist component.
   imagP[0] = 0;

   // Normalize so than an input sine wave at 0dBfs registers as 0dBfs (undo FFT
   // scaling factor).
   const double magnitudeScale = 1.0 / fftSize;

   // A value of 0 does no averaging with the previous result.  Larger values
   // produce slower, but smoother changes.
   double k = m_smoothingTimeConstant;
   k = std::max(0.0, k);
   k = std::min(1.0, k);

   // Convert the analysis data from complex to magnitude and average with the
   // previous result.
   float* destination = magnitudeBuffer().data();
   size_t n = magnitudeBuffer().size();
   for (size_t i = 0; i < n; ++i) {
     std::complex<double> c(realP[i], imagP[i]);
     double scalarMagnitude = abs(c) * magnitudeScale;
     destination[i] = float(k * destination[i] + (1 - k) * scalarMagnitude);
   }
 }

 void RealtimeAnalyser::convertFloatToDb(DOMFloat32Array* destinationArray) {
   // Convert from linear magnitude to floating-point decibels.
   unsigned sourceLength = magnitudeBuffer().size();
   size_t len = std::min(sourceLength, destinationArray->length());
   if (len > 0) {
     const float* source = magnitudeBuffer().data();
     float* destination = destinationArray->data();

     for (unsigned i = 0; i < len; ++i) {
       float linearValue = source[i];
       double dbMag = AudioUtilities::linearToDecibels(linearValue);
       destination[i] = float(dbMag);
     }
   }
 }

 void RealtimeAnalyser::getFloatFrequencyData(DOMFloat32Array* destinationArray,
                                              double currentTime) {
   DCHECK(isMainThread());
   DCHECK(destinationArray);

   if (currentTime <= m_lastAnalysisTime) {
     convertFloatToDb(destinationArray);
     return;
   }

   // Time has advanced since the last call; update the FFT data.
   m_lastAnalysisTime = currentTime;
   doFFTAnalysis();

   convertFloatToDb(destinationArray);
 }

 void RealtimeAnalyser::convertToByteData(DOMUint8Array* destinationArray) {
   // Convert from linear magnitude to unsigned-byte decibels.
   unsigned sourceLength = magnitudeBuffer().size();
   size_t len = std::min(sourceLength, destinationArray->length());
   if (len > 0) {
     const double rangeScaleFactor = m_maxDecibels == m_minDecibels
                                         ? 1
                                         : 1 / (m_maxDecibels - m_minDecibels);
     const double minDecibels = m_minDecibels;

     const float* source = magnitudeBuffer().data();
     unsigned char* destination = destinationArray->data();

     for (unsigned i = 0; i < len; ++i) {
       float linearValue = source[i];
       double dbMag = AudioUtilities::linearToDecibels(linearValue);

       // The range m_minDecibels to m_maxDecibels will be scaled to byte values
       // from 0 to UCHAR_MAX.
       double scaledValue = UCHAR_MAX * (dbMag - minDecibels) * rangeScaleFactor;

       // Clip to valid range.
       if (scaledValue < 0)
         scaledValue = 0;
       if (scaledValue > UCHAR_MAX)
         scaledValue = UCHAR_MAX;

       destination[i] = static_cast<unsigned char>(scaledValue);
     }
   }
 }

 void RealtimeAnalyser::getByteFrequencyData(DOMUint8Array* destinationArray,
                                             double currentTime) {
   DCHECK(isMainThread());
   DCHECK(destinationArray);

   if (currentTime <= m_lastAnalysisTime) {
     // FIXME: Is it worth caching the data so we don't have to do the conversion
     // every time?  Perhaps not, since we expect many calls in the same
     // rendering quantum.
     convertToByteData(destinationArray);
     return;
   }

   // Time has advanced since the last call; update the FFT data.
   m_lastAnalysisTime = currentTime;
   doFFTAnalysis();

   convertToByteData(destinationArray);
 }

 void RealtimeAnalyser::getFloatTimeDomainData(
     DOMFloat32Array* destinationArray) {
   DCHECK(isMainThread());
   DCHECK(destinationArray);

   unsigned fftSize = this->fftSize();
   size_t len = std::min(fftSize, destinationArray->length());
   if (len > 0) {
     bool isInputBufferGood = m_inputBuffer.size() == InputBufferSize &&
                              m_inputBuffer.size() > fftSize;
     DCHECK(isInputBufferGood);
     if (!isInputBufferGood)
       return;

     float* inputBuffer = m_inputBuffer.data();
     float* destination = destinationArray->data();

     unsigned writeIndex = m_writeIndex;

     for (unsigned i = 0; i < len; ++i) {
       // Buffer access is protected due to modulo operation.
       float value = inputBuffer[(i + writeIndex - fftSize + InputBufferSize) %
                                 InputBufferSize];

       destination[i] = value;
     }
   }
 }

 void RealtimeAnalyser::getByteTimeDomainData(DOMUint8Array* destinationArray) {
   DCHECK(isMainThread());
   DCHECK(destinationArray);

   unsigned fftSize = this->fftSize();
   size_t len = std::min(fftSize, destinationArray->length());
   if (len > 0) {
     bool isInputBufferGood = m_inputBuffer.size() == InputBufferSize &&
                              m_inputBuffer.size() > fftSize;
     DCHECK(isInputBufferGood);
     if (!isInputBufferGood)
       return;

     float* inputBuffer = m_inputBuffer.data();
     unsigned char* destination = destinationArray->data();

     unsigned writeIndex = m_writeIndex;

     for (unsigned i = 0; i < len; ++i) {
       // Buffer access is protected due to modulo operation.
       float value = inputBuffer[(i + writeIndex - fftSize + InputBufferSize) %
                                 InputBufferSize];

       // Scale from nominal -1 -> +1 to unsigned byte.
       double scaledValue = 128 * (value + 1);

       // Clip to valid range.
       if (scaledValue < 0)
         scaledValue = 0;
       if (scaledValue > UCHAR_MAX)
         scaledValue = UCHAR_MAX;

       destination[i] = static_cast<unsigned char>(scaledValue);
     }
   }
 }

 }  // namespace blink
	/*
	* Copyright (C) 2010, Google Inc. All rights reserved.
	*
	* Redistribution and use in source and binary forms, with or without
	* modification, are permitted provided that the following conditions
	* are met:
	* 1. Redistributions of source code must retain the above copyright
	* notice, this list of conditions and the following disclaimer.
	* 2. Redistributions in binary form must reproduce the above copyright
	* notice, this list of conditions and the following disclaimer in the
	* documentation and/or other materials provided with the distribution.
	*
	* THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' AND
	* ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
	* IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
	* ARE DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS BE LIABLE
	* FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
	* DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
	* SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
	* CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
	* LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
	* OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH
	* DAMAGE.
	*/

	#include "modules/webaudio/RealtimeAnalyser.h"
	#include "platform/audio/AudioBus.h"
	#include "platform/audio/AudioUtilities.h"
	#include "platform/audio/VectorMath.h"
	#include "wtf/MathExtras.h"
	#include "wtf/PtrUtil.h"
	#include <algorithm>
	#include <complex>
	#include <limits.h>

	namespace blink {

	const double RealtimeAnalyser::DefaultSmoothingTimeConstant = 0.8;
	const double RealtimeAnalyser::DefaultMinDecibels = -100;
	const double RealtimeAnalyser::DefaultMaxDecibels = -30;

	const unsigned RealtimeAnalyser::DefaultFFTSize = 2048;
	// All FFT implementations are expected to handle power-of-two sizes
	// MinFFTSize <= size <= MaxFFTSize.
	const unsigned RealtimeAnalyser::MinFFTSize = 32;
	const unsigned RealtimeAnalyser::MaxFFTSize = 32768;
	const unsigned RealtimeAnalyser::InputBufferSize =
	RealtimeAnalyser::MaxFFTSize * 2;

	RealtimeAnalyser::RealtimeAnalyser()
	: m_inputBuffer(InputBufferSize),
	m_writeIndex(0),
	m_downMixBus(AudioBus::create(1, AudioUtilities::kRenderQuantumFrames)),
	m_fftSize(DefaultFFTSize),
	m_magnitudeBuffer(DefaultFFTSize / 2),
	m_smoothingTimeConstant(DefaultSmoothingTimeConstant),
	m_minDecibels(DefaultMinDecibels),
	m_maxDecibels(DefaultMaxDecibels),
	m_lastAnalysisTime(-1) {
	m_analysisFrame = wrapUnique(new FFTFrame(DefaultFFTSize));
	}

	bool RealtimeAnalyser::setFftSize(size_t size) {
	DCHECK(isMainThread());

	// Only allow powers of two.
	unsigned log2size = static_cast<unsigned>(log2(size));
	bool isPOT(1UL << log2size == size);

	if (!isPOT \|\| size > MaxFFTSize \|\| size < MinFFTSize)
	return false;

	if (m_fftSize != size) {
	m_analysisFrame = wrapUnique(new FFTFrame(size));
	// m_magnitudeBuffer has size = fftSize / 2 because it contains floats
	// reduced from complex values in m_analysisFrame.
	m_magnitudeBuffer.allocate(size / 2);
	m_fftSize = size;
	}

	return true;
	}

	void RealtimeAnalyser::writeInput(AudioBus* bus, size_t framesToProcess) {
	bool isBusGood = bus && bus->numberOfChannels() > 0 &&
	bus->channel(0)->length() >= framesToProcess;
	DCHECK(isBusGood);
	if (!isBusGood)
	return;

	// FIXME : allow to work with non-FFTSize divisible chunking
	bool isDestinationGood =
	m_writeIndex < m_inputBuffer.size() &&
	m_writeIndex + framesToProcess <= m_inputBuffer.size();
	DCHECK(isDestinationGood);
	if (!isDestinationGood)
	return;

	// Perform real-time analysis
	float* dest = m_inputBuffer.data() + m_writeIndex;

	// Clear the bus and downmix the input according to the down mixing rules.
	// Then save the result in the m_inputBuffer at the appropriate place.
	m_downMixBus->zero();
	m_downMixBus->sumFrom(*bus);
	memcpy(dest, m_downMixBus->channel(0)->data(),
	framesToProcess * sizeof(*dest));

	m_writeIndex += framesToProcess;
	if (m_writeIndex >= InputBufferSize)
	m_writeIndex = 0;
	}

	namespace {

	void applyWindow(float* p, size_t n) {
	DCHECK(isMainThread());

	// Blackman window
	double alpha = 0.16;
	double a0 = 0.5 * (1 - alpha);
	double a1 = 0.5;
	double a2 = 0.5 * alpha;

	for (unsigned i = 0; i < n; ++i) {
	double x = static_cast<double>(i) / static_cast<double>(n);
	double window =
	a0 - a1 * cos(twoPiDouble * x) + a2 * cos(twoPiDouble * 2.0 * x);
	p[i] *= float(window);
	}
	}

	} // namespace

	void RealtimeAnalyser::doFFTAnalysis() {
	DCHECK(isMainThread());

	// Unroll the input buffer into a temporary buffer, where we'll apply an
	// analysis window followed by an FFT.
	size_t fftSize = this->fftSize();

	AudioFloatArray temporaryBuffer(fftSize);
	float* inputBuffer = m_inputBuffer.data();
	float* tempP = temporaryBuffer.data();

	// Take the previous fftSize values from the input buffer and copy into the
	// temporary buffer.
	unsigned writeIndex = m_writeIndex;
	if (writeIndex < fftSize) {
	memcpy(tempP, inputBuffer + writeIndex - fftSize + InputBufferSize,
	sizeof(tempP) (fftSize - writeIndex));
	memcpy(tempP + fftSize - writeIndex, inputBuffer,
	sizeof(tempP) writeIndex);
	} else {
	memcpy(tempP, inputBuffer + writeIndex - fftSize, sizeof(tempP) fftSize);
	}

	// Window the input samples.
	applyWindow(tempP, fftSize);

	// Do the analysis.
	m_analysisFrame->doFFT(tempP);

	float* realP = m_analysisFrame->realData();
	float* imagP = m_analysisFrame->imagData();

	// Blow away the packed nyquist component.
	imagP[0] = 0;

	// Normalize so than an input sine wave at 0dBfs registers as 0dBfs (undo FFT
	// scaling factor).
	const double magnitudeScale = 1.0 / fftSize;

	// A value of 0 does no averaging with the previous result. Larger values
	// produce slower, but smoother changes.
	double k = m_smoothingTimeConstant;
	k = std::max(0.0, k);
	k = std::min(1.0, k);

	// Convert the analysis data from complex to magnitude and average with the
	// previous result.
	float* destination = magnitudeBuffer().data();
	size_t n = magnitudeBuffer().size();
	for (size_t i = 0; i < n; ++i) {
	std::complex<double> c(realP[i], imagP[i]);
	double scalarMagnitude = abs(c) * magnitudeScale;
	destination[i] = float(k * destination[i] + (1 - k) * scalarMagnitude);
	}
	}

	void RealtimeAnalyser::convertFloatToDb(DOMFloat32Array* destinationArray) {
	// Convert from linear magnitude to floating-point decibels.
	unsigned sourceLength = magnitudeBuffer().size();
	size_t len = std::min(sourceLength, destinationArray->length());
	if (len > 0) {
	const float* source = magnitudeBuffer().data();
	float* destination = destinationArray->data();

	for (unsigned i = 0; i < len; ++i) {
	float linearValue = source[i];
	double dbMag = AudioUtilities::linearToDecibels(linearValue);
	destination[i] = float(dbMag);
	}
	}
	}

	void RealtimeAnalyser::getFloatFrequencyData(DOMFloat32Array* destinationArray,
	double currentTime) {
	DCHECK(isMainThread());
	DCHECK(destinationArray);

	if (currentTime <= m_lastAnalysisTime) {
	convertFloatToDb(destinationArray);
	return;
	}

	// Time has advanced since the last call; update the FFT data.
	m_lastAnalysisTime = currentTime;
	doFFTAnalysis();

	convertFloatToDb(destinationArray);
	}

	void RealtimeAnalyser::convertToByteData(DOMUint8Array* destinationArray) {
	// Convert from linear magnitude to unsigned-byte decibels.
	unsigned sourceLength = magnitudeBuffer().size();
	size_t len = std::min(sourceLength, destinationArray->length());
	if (len > 0) {
	const double rangeScaleFactor = m_maxDecibels == m_minDecibels
	? 1
	: 1 / (m_maxDecibels - m_minDecibels);
	const double minDecibels = m_minDecibels;

	const float* source = magnitudeBuffer().data();
	unsigned char* destination = destinationArray->data();

	for (unsigned i = 0; i < len; ++i) {
	float linearValue = source[i];
	double dbMag = AudioUtilities::linearToDecibels(linearValue);

	// The range m_minDecibels to m_maxDecibels will be scaled to byte values
	// from 0 to UCHAR_MAX.
	double scaledValue = UCHAR_MAX * (dbMag - minDecibels) * rangeScaleFactor;

	// Clip to valid range.
	if (scaledValue < 0)
	scaledValue = 0;
	if (scaledValue > UCHAR_MAX)
	scaledValue = UCHAR_MAX;

	destination[i] = static_cast<unsigned char>(scaledValue);
	}
	}
	}

	void RealtimeAnalyser::getByteFrequencyData(DOMUint8Array* destinationArray,
	double currentTime) {
	DCHECK(isMainThread());
	DCHECK(destinationArray);

	if (currentTime <= m_lastAnalysisTime) {
	// FIXME: Is it worth caching the data so we don't have to do the conversion
	// every time? Perhaps not, since we expect many calls in the same
	// rendering quantum.
	convertToByteData(destinationArray);
	return;
	}

	// Time has advanced since the last call; update the FFT data.
	m_lastAnalysisTime = currentTime;
	doFFTAnalysis();

	convertToByteData(destinationArray);
	}

	void RealtimeAnalyser::getFloatTimeDomainData(
	DOMFloat32Array* destinationArray) {
	DCHECK(isMainThread());
	DCHECK(destinationArray);

	unsigned fftSize = this->fftSize();
	size_t len = std::min(fftSize, destinationArray->length());
	if (len > 0) {
	bool isInputBufferGood = m_inputBuffer.size() == InputBufferSize &&
	m_inputBuffer.size() > fftSize;
	DCHECK(isInputBufferGood);
	if (!isInputBufferGood)
	return;

	float* inputBuffer = m_inputBuffer.data();
	float* destination = destinationArray->data();

	unsigned writeIndex = m_writeIndex;

	for (unsigned i = 0; i < len; ++i) {
	// Buffer access is protected due to modulo operation.
	float value = inputBuffer[(i + writeIndex - fftSize + InputBufferSize) %
	InputBufferSize];

	destination[i] = value;
	}
	}
	}

	void RealtimeAnalyser::getByteTimeDomainData(DOMUint8Array* destinationArray) {
	DCHECK(isMainThread());
	DCHECK(destinationArray);

	unsigned fftSize = this->fftSize();
	size_t len = std::min(fftSize, destinationArray->length());
	if (len > 0) {
	bool isInputBufferGood = m_inputBuffer.size() == InputBufferSize &&
	m_inputBuffer.size() > fftSize;
	DCHECK(isInputBufferGood);
	if (!isInputBufferGood)
	return;

	float* inputBuffer = m_inputBuffer.data();
	unsigned char* destination = destinationArray->data();

	unsigned writeIndex = m_writeIndex;

	for (unsigned i = 0; i < len; ++i) {
	// Buffer access is protected due to modulo operation.
	float value = inputBuffer[(i + writeIndex - fftSize + InputBufferSize) %
	InputBufferSize];

	// Scale from nominal -1 -> +1 to unsigned byte.
	double scaledValue = 128 * (value + 1);

	// Clip to valid range.
	if (scaledValue < 0)
	scaledValue = 0;
	if (scaledValue > UCHAR_MAX)
	scaledValue = UCHAR_MAX;

	destination[i] = static_cast<unsigned char>(scaledValue);
	}
	}
	}

	} // namespace blink