third_party/WebKit/Source/platform/audio/HRTFKernel.cpp

/*
 * Copyright (C) 2010 Google Inc. All rights reserved.
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 *     from this software without specific prior written permission.
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#include "platform/audio/HRTFKernel.h"

#include "platform/audio/AudioChannel.h"
#include "wtf/MathExtras.h"
#include "wtf/PtrUtil.h"
#include <algorithm>
#include <memory>

namespace blink {

// Takes the input AudioChannel as an input impulse response and calculates the
// average group delay.  This represents the initial delay before the most
// energetic part of the impulse response.  The sample-frame delay is removed
// from the impulseP impulse response, and this value  is returned.  The length
// of the passed in AudioChannel must be a power of 2.
static float extractAverageGroupDelay(AudioChannel* channel,
                                      size_t analysisFFTSize) {
  ASSERT(channel);

  float* impulseP = channel->mutableData();

  bool isSizeGood = channel->length() >= analysisFFTSize;
  ASSERT(isSizeGood);
  if (!isSizeGood)
    return 0;

  // Check for power-of-2.
  ASSERT(1UL << static_cast<unsigned>(log2(analysisFFTSize)) ==
         analysisFFTSize);

  FFTFrame estimationFrame(analysisFFTSize);
  estimationFrame.doFFT(impulseP);

  float frameDelay = clampTo<float>(estimationFrame.extractAverageGroupDelay());
  estimationFrame.doInverseFFT(impulseP);

  return frameDelay;
}

HRTFKernel::HRTFKernel(AudioChannel* channel, size_t fftSize, float sampleRate)
    : m_frameDelay(0), m_sampleRate(sampleRate) {
  ASSERT(channel);

  // Determine the leading delay (average group delay) for the response.
  m_frameDelay = extractAverageGroupDelay(channel, fftSize / 2);

  float* impulseResponse = channel->mutableData();
  size_t responseLength = channel->length();

  // We need to truncate to fit into 1/2 the FFT size (with zero padding) in
  // order to do proper convolution.
  // Truncate if necessary to max impulse response length allowed by FFT.
  size_t truncatedResponseLength = std::min(responseLength, fftSize / 2);

  // Quick fade-out (apply window) at truncation point
  unsigned numberOfFadeOutFrames = static_cast<unsigned>(
      sampleRate / 4410);  // 10 sample-frames @44.1KHz sample-rate
  ASSERT(numberOfFadeOutFrames < truncatedResponseLength);
  if (numberOfFadeOutFrames < truncatedResponseLength) {
    for (unsigned i = truncatedResponseLength - numberOfFadeOutFrames;
         i < truncatedResponseLength; ++i) {
      float x = 1.0f -
                static_cast<float>(
                    i - (truncatedResponseLength - numberOfFadeOutFrames)) /
                    numberOfFadeOutFrames;
      impulseResponse[i] *= x;
    }
  }

  m_fftFrame = wrapUnique(new FFTFrame(fftSize));
  m_fftFrame->doPaddedFFT(impulseResponse, truncatedResponseLength);
}

std::unique_ptr<AudioChannel> HRTFKernel::createImpulseResponse() {
  std::unique_ptr<AudioChannel> channel =
      wrapUnique(new AudioChannel(fftSize()));
  FFTFrame fftFrame(*m_fftFrame);

  // Add leading delay back in.
  fftFrame.addConstantGroupDelay(m_frameDelay);
  fftFrame.doInverseFFT(channel->mutableData());

  return channel;
}

// Interpolates two kernels with x: 0 -> 1 and returns the result.
std::unique_ptr<HRTFKernel> HRTFKernel::createInterpolatedKernel(
    HRTFKernel* kernel1,
    HRTFKernel* kernel2,
    float x) {
  ASSERT(kernel1 && kernel2);
  if (!kernel1 || !kernel2)
    return nullptr;

  ASSERT(x >= 0.0 && x < 1.0);
  x = clampTo(x, 0.0f, 1.0f);

  float sampleRate1 = kernel1->sampleRate();
  float sampleRate2 = kernel2->sampleRate();
  ASSERT(sampleRate1 == sampleRate2);
  if (sampleRate1 != sampleRate2)
    return nullptr;

  float frameDelay =
      (1 - x) * kernel1->frameDelay() + x * kernel2->frameDelay();

  std::unique_ptr<FFTFrame> interpolatedFrame =
      FFTFrame::createInterpolatedFrame(*kernel1->fftFrame(),
                                        *kernel2->fftFrame(), x);
  return HRTFKernel::create(std::move(interpolatedFrame), frameDelay,
                            sampleRate1);
}

}  // namespace blink