third_party/blink/renderer/modules/webaudio/realtime_analyser.cc - 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 <limits.h>
 #include <algorithm>
 #include <complex>
 #include "third_party/blink/renderer/modules/webaudio/realtime_analyser.h"
 #include "third_party/blink/renderer/platform/audio/audio_bus.h"
 #include "third_party/blink/renderer/platform/audio/audio_utilities.h"
 #include "third_party/blink/renderer/platform/audio/vector_math.h"
 #include "third_party/blink/renderer/platform/wtf/math_extras.h"

 namespace blink {

 const double RealtimeAnalyser::kDefaultSmoothingTimeConstant = 0.8;
 const double RealtimeAnalyser::kDefaultMinDecibels = -100;
 const double RealtimeAnalyser::kDefaultMaxDecibels = -30;

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

 RealtimeAnalyser::RealtimeAnalyser()
     : input_buffer_(kInputBufferSize),
       down_mix_bus_(AudioBus::Create(1, audio_utilities::kRenderQuantumFrames)),
       fft_size_(kDefaultFFTSize),
       magnitude_buffer_(kDefaultFFTSize / 2),
       smoothing_time_constant_(kDefaultSmoothingTimeConstant),
       min_decibels_(kDefaultMinDecibels),
       max_decibels_(kDefaultMaxDecibels),
       last_analysis_time_(-1) {
   analysis_frame_ = std::make_unique<FFTFrame>(kDefaultFFTSize);
 }

 bool RealtimeAnalyser::SetFftSize(uint32_t size) {
   DCHECK(IsMainThread());

   // Only allow powers of two within the allowed range.
   if (size > kMaxFFTSize || size < kMinFFTSize ||
       !audio_utilities::IsPowerOfTwo(size))
     return false;

   if (fft_size_ != size) {
     analysis_frame_ = std::make_unique<FFTFrame>(size);
     // m_magnitudeBuffer has size = fftSize / 2 because it contains floats
     // reduced from complex values in m_analysisFrame.
     magnitude_buffer_.Allocate(size / 2);
     fft_size_ = size;
   }

   return true;
 }

 void RealtimeAnalyser::WriteInput(AudioBus* bus, uint32_t frames_to_process) {
   bool is_bus_good = bus && bus->NumberOfChannels() > 0 &&
                      bus->Channel(0)->length() >= frames_to_process;
   DCHECK(is_bus_good);
   if (!is_bus_good)
     return;

   unsigned write_index = GetWriteIndex();
   // FIXME : allow to work with non-FFTSize divisible chunking
   bool is_destination_good =
       write_index < input_buffer_.size() &&
       write_index + frames_to_process <= input_buffer_.size();
   DCHECK(is_destination_good);
   if (!is_destination_good)
     return;

   // Perform real-time analysis
   float* dest = input_buffer_.Data() + write_index;

   // 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.
   down_mix_bus_->Zero();
   down_mix_bus_->SumFrom(*bus);
   memcpy(dest, down_mix_bus_->Channel(0)->Data(),
          frames_to_process * sizeof(*dest));

   write_index += frames_to_process;
   if (write_index >= kInputBufferSize)
     write_index = 0;
   SetWriteIndex(write_index);
 }

 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(kTwoPiDouble * x) + a2 * cos(kTwoPiDouble * 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.
   uint32_t fft_size = this->FftSize();

   AudioFloatArray temporary_buffer(fft_size);
   float* input_buffer = input_buffer_.Data();
   float* temp_p = temporary_buffer.Data();

   // Take the previous fftSize values from the input buffer and copy into the
   // temporary buffer.
   unsigned write_index = GetWriteIndex();
   if (write_index < fft_size) {
     memcpy(temp_p, input_buffer + write_index - fft_size + kInputBufferSize,
            sizeof(*temp_p) * (fft_size - write_index));
     memcpy(temp_p + fft_size - write_index, input_buffer,
            sizeof(*temp_p) * write_index);
   } else {
     memcpy(temp_p, input_buffer + write_index - fft_size,
            sizeof(*temp_p) * fft_size);
   }

   // Window the input samples.
   ApplyWindow(temp_p, fft_size);

   // Do the analysis.
   analysis_frame_->DoFFT(temp_p);

   const float* real_p = analysis_frame_->RealData();
   float* imag_p = analysis_frame_->ImagData();

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

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

   // A value of 0 does no averaging with the previous result.  Larger values
   // produce slower, but smoother changes.
   double k = smoothing_time_constant_;
   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(real_p[i], imag_p[i]);
     double scalar_magnitude = abs(c) * magnitude_scale;
     destination[i] = float(k * destination[i] + (1 - k) * scalar_magnitude);
   }
 }

 void RealtimeAnalyser::ConvertFloatToDb(DOMFloat32Array* destination_array) {
   // Convert from linear magnitude to floating-point decibels.
   unsigned source_length = MagnitudeBuffer().size();
   size_t len = std::min(source_length, destination_array->length());
   if (len > 0) {
     const float* source = MagnitudeBuffer().Data();
     float* destination = destination_array->Data();

     for (unsigned i = 0; i < len; ++i) {
       float linear_value = source[i];
       double db_mag = audio_utilities::LinearToDecibels(linear_value);
       destination[i] = float(db_mag);
     }
   }
 }

 void RealtimeAnalyser::GetFloatFrequencyData(DOMFloat32Array* destination_array,
                                              double current_time) {
   DCHECK(IsMainThread());
   DCHECK(destination_array);

   if (current_time <= last_analysis_time_) {
     ConvertFloatToDb(destination_array);
     return;
   }

   // Time has advanced since the last call; update the FFT data.
   last_analysis_time_ = current_time;
   DoFFTAnalysis();

   ConvertFloatToDb(destination_array);
 }

 void RealtimeAnalyser::ConvertToByteData(DOMUint8Array* destination_array) {
   // Convert from linear magnitude to unsigned-byte decibels.
   unsigned source_length = MagnitudeBuffer().size();
   size_t len = std::min(source_length, destination_array->length());
   if (len > 0) {
     const double range_scale_factor = max_decibels_ == min_decibels_
                                           ? 1
                                           : 1 / (max_decibels_ - min_decibels_);
     const double min_decibels = min_decibels_;

     const float* source = MagnitudeBuffer().Data();
     unsigned char* destination = destination_array->Data();

     for (unsigned i = 0; i < len; ++i) {
       float linear_value = source[i];
       double db_mag = audio_utilities::LinearToDecibels(linear_value);

       // The range m_minDecibels to m_maxDecibels will be scaled to byte values
       // from 0 to UCHAR_MAX.
       double scaled_value =
           UCHAR_MAX * (db_mag - min_decibels) * range_scale_factor;

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

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

 void RealtimeAnalyser::GetByteFrequencyData(DOMUint8Array* destination_array,
                                             double current_time) {
   DCHECK(IsMainThread());
   DCHECK(destination_array);

   if (current_time <= last_analysis_time_) {
     // 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(destination_array);
     return;
   }

   // Time has advanced since the last call; update the FFT data.
   last_analysis_time_ = current_time;
   DoFFTAnalysis();

   ConvertToByteData(destination_array);
 }

 void RealtimeAnalyser::GetFloatTimeDomainData(
     DOMFloat32Array* destination_array) {
   DCHECK(IsMainThread());
   DCHECK(destination_array);

   unsigned fft_size = this->FftSize();
   size_t len = std::min(fft_size, destination_array->length());
   if (len > 0) {
     bool is_input_buffer_good = input_buffer_.size() == kInputBufferSize &&
                                 input_buffer_.size() > fft_size;
     DCHECK(is_input_buffer_good);
     if (!is_input_buffer_good)
       return;

     float* input_buffer = input_buffer_.Data();
     float* destination = destination_array->Data();

     unsigned write_index = GetWriteIndex();

     for (unsigned i = 0; i < len; ++i) {
       // Buffer access is protected due to modulo operation.
       float value =
           input_buffer[(i + write_index - fft_size + kInputBufferSize) %
                        kInputBufferSize];

       destination[i] = value;
     }
   }
 }

 void RealtimeAnalyser::GetByteTimeDomainData(DOMUint8Array* destination_array) {
   DCHECK(IsMainThread());
   DCHECK(destination_array);

   unsigned fft_size = this->FftSize();
   size_t len = std::min(fft_size, destination_array->length());
   if (len > 0) {
     bool is_input_buffer_good = input_buffer_.size() == kInputBufferSize &&
                                 input_buffer_.size() > fft_size;
     DCHECK(is_input_buffer_good);
     if (!is_input_buffer_good)
       return;

     float* input_buffer = input_buffer_.Data();
     unsigned char* destination = destination_array->Data();

     unsigned write_index = GetWriteIndex();

     for (unsigned i = 0; i < len; ++i) {
       // Buffer access is protected due to modulo operation.
       float value =
           input_buffer[(i + write_index - fft_size + kInputBufferSize) %
                        kInputBufferSize];

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

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

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

 }  // 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 <limits.h>
	#include <algorithm>
	#include <complex>
	#include "third_party/blink/renderer/modules/webaudio/realtime_analyser.h"
	#include "third_party/blink/renderer/platform/audio/audio_bus.h"
	#include "third_party/blink/renderer/platform/audio/audio_utilities.h"
	#include "third_party/blink/renderer/platform/audio/vector_math.h"
	#include "third_party/blink/renderer/platform/wtf/math_extras.h"

	namespace blink {

	const double RealtimeAnalyser::kDefaultSmoothingTimeConstant = 0.8;
	const double RealtimeAnalyser::kDefaultMinDecibels = -100;
	const double RealtimeAnalyser::kDefaultMaxDecibels = -30;

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

	RealtimeAnalyser::RealtimeAnalyser()
	: input_buffer_(kInputBufferSize),
	down_mix_bus_(AudioBus::Create(1, audio_utilities::kRenderQuantumFrames)),
	fft_size_(kDefaultFFTSize),
	magnitude_buffer_(kDefaultFFTSize / 2),
	smoothing_time_constant_(kDefaultSmoothingTimeConstant),
	min_decibels_(kDefaultMinDecibels),
	max_decibels_(kDefaultMaxDecibels),
	last_analysis_time_(-1) {
	analysis_frame_ = std::make_unique<FFTFrame>(kDefaultFFTSize);
	}

	bool RealtimeAnalyser::SetFftSize(uint32_t size) {
	DCHECK(IsMainThread());

	// Only allow powers of two within the allowed range.
	if (size > kMaxFFTSize \|\| size < kMinFFTSize \|\|
	!audio_utilities::IsPowerOfTwo(size))
	return false;

	if (fft_size_ != size) {
	analysis_frame_ = std::make_unique<FFTFrame>(size);
	// m_magnitudeBuffer has size = fftSize / 2 because it contains floats
	// reduced from complex values in m_analysisFrame.
	magnitude_buffer_.Allocate(size / 2);
	fft_size_ = size;
	}

	return true;
	}

	void RealtimeAnalyser::WriteInput(AudioBus* bus, uint32_t frames_to_process) {
	bool is_bus_good = bus && bus->NumberOfChannels() > 0 &&
	bus->Channel(0)->length() >= frames_to_process;
	DCHECK(is_bus_good);
	if (!is_bus_good)
	return;

	unsigned write_index = GetWriteIndex();
	// FIXME : allow to work with non-FFTSize divisible chunking
	bool is_destination_good =
	write_index < input_buffer_.size() &&
	write_index + frames_to_process <= input_buffer_.size();
	DCHECK(is_destination_good);
	if (!is_destination_good)
	return;

	// Perform real-time analysis
	float* dest = input_buffer_.Data() + write_index;

	// 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.
	down_mix_bus_->Zero();
	down_mix_bus_->SumFrom(*bus);
	memcpy(dest, down_mix_bus_->Channel(0)->Data(),
	frames_to_process * sizeof(*dest));

	write_index += frames_to_process;
	if (write_index >= kInputBufferSize)
	write_index = 0;
	SetWriteIndex(write_index);
	}

	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(kTwoPiDouble * x) + a2 * cos(kTwoPiDouble * 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.
	uint32_t fft_size = this->FftSize();

	AudioFloatArray temporary_buffer(fft_size);
	float* input_buffer = input_buffer_.Data();
	float* temp_p = temporary_buffer.Data();

	// Take the previous fftSize values from the input buffer and copy into the
	// temporary buffer.
	unsigned write_index = GetWriteIndex();
	if (write_index < fft_size) {
	memcpy(temp_p, input_buffer + write_index - fft_size + kInputBufferSize,
	sizeof(temp_p) (fft_size - write_index));
	memcpy(temp_p + fft_size - write_index, input_buffer,
	sizeof(temp_p) write_index);
	} else {
	memcpy(temp_p, input_buffer + write_index - fft_size,
	sizeof(temp_p) fft_size);
	}

	// Window the input samples.
	ApplyWindow(temp_p, fft_size);

	// Do the analysis.
	analysis_frame_->DoFFT(temp_p);

	const float* real_p = analysis_frame_->RealData();
	float* imag_p = analysis_frame_->ImagData();

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

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

	// A value of 0 does no averaging with the previous result. Larger values
	// produce slower, but smoother changes.
	double k = smoothing_time_constant_;
	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(real_p[i], imag_p[i]);
	double scalar_magnitude = abs(c) * magnitude_scale;
	destination[i] = float(k * destination[i] + (1 - k) * scalar_magnitude);
	}
	}

	void RealtimeAnalyser::ConvertFloatToDb(DOMFloat32Array* destination_array) {
	// Convert from linear magnitude to floating-point decibels.
	unsigned source_length = MagnitudeBuffer().size();
	size_t len = std::min(source_length, destination_array->length());
	if (len > 0) {
	const float* source = MagnitudeBuffer().Data();
	float* destination = destination_array->Data();

	for (unsigned i = 0; i < len; ++i) {
	float linear_value = source[i];
	double db_mag = audio_utilities::LinearToDecibels(linear_value);
	destination[i] = float(db_mag);
	}
	}
	}

	void RealtimeAnalyser::GetFloatFrequencyData(DOMFloat32Array* destination_array,
	double current_time) {
	DCHECK(IsMainThread());
	DCHECK(destination_array);

	if (current_time <= last_analysis_time_) {
	ConvertFloatToDb(destination_array);
	return;
	}

	// Time has advanced since the last call; update the FFT data.
	last_analysis_time_ = current_time;
	DoFFTAnalysis();

	ConvertFloatToDb(destination_array);
	}

	void RealtimeAnalyser::ConvertToByteData(DOMUint8Array* destination_array) {
	// Convert from linear magnitude to unsigned-byte decibels.
	unsigned source_length = MagnitudeBuffer().size();
	size_t len = std::min(source_length, destination_array->length());
	if (len > 0) {
	const double range_scale_factor = max_decibels_ == min_decibels_
	? 1
	: 1 / (max_decibels_ - min_decibels_);
	const double min_decibels = min_decibels_;

	const float* source = MagnitudeBuffer().Data();
	unsigned char* destination = destination_array->Data();

	for (unsigned i = 0; i < len; ++i) {
	float linear_value = source[i];
	double db_mag = audio_utilities::LinearToDecibels(linear_value);

	// The range m_minDecibels to m_maxDecibels will be scaled to byte values
	// from 0 to UCHAR_MAX.
	double scaled_value =
	UCHAR_MAX * (db_mag - min_decibels) * range_scale_factor;

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

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

	void RealtimeAnalyser::GetByteFrequencyData(DOMUint8Array* destination_array,
	double current_time) {
	DCHECK(IsMainThread());
	DCHECK(destination_array);

	if (current_time <= last_analysis_time_) {
	// 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(destination_array);
	return;
	}

	// Time has advanced since the last call; update the FFT data.
	last_analysis_time_ = current_time;
	DoFFTAnalysis();

	ConvertToByteData(destination_array);
	}

	void RealtimeAnalyser::GetFloatTimeDomainData(
	DOMFloat32Array* destination_array) {
	DCHECK(IsMainThread());
	DCHECK(destination_array);

	unsigned fft_size = this->FftSize();
	size_t len = std::min(fft_size, destination_array->length());
	if (len > 0) {
	bool is_input_buffer_good = input_buffer_.size() == kInputBufferSize &&
	input_buffer_.size() > fft_size;
	DCHECK(is_input_buffer_good);
	if (!is_input_buffer_good)
	return;

	float* input_buffer = input_buffer_.Data();
	float* destination = destination_array->Data();

	unsigned write_index = GetWriteIndex();

	for (unsigned i = 0; i < len; ++i) {
	// Buffer access is protected due to modulo operation.
	float value =
	input_buffer[(i + write_index - fft_size + kInputBufferSize) %
	kInputBufferSize];

	destination[i] = value;
	}
	}
	}

	void RealtimeAnalyser::GetByteTimeDomainData(DOMUint8Array* destination_array) {
	DCHECK(IsMainThread());
	DCHECK(destination_array);

	unsigned fft_size = this->FftSize();
	size_t len = std::min(fft_size, destination_array->length());
	if (len > 0) {
	bool is_input_buffer_good = input_buffer_.size() == kInputBufferSize &&
	input_buffer_.size() > fft_size;
	DCHECK(is_input_buffer_good);
	if (!is_input_buffer_good)
	return;

	float* input_buffer = input_buffer_.Data();
	unsigned char* destination = destination_array->Data();

	unsigned write_index = GetWriteIndex();

	for (unsigned i = 0; i < len; ++i) {
	// Buffer access is protected due to modulo operation.
	float value =
	input_buffer[(i + write_index - fft_size + kInputBufferSize) %
	kInputBufferSize];

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

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

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

	} // namespace blink