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

/*
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#include "platform/audio/AudioResamplerKernel.h"
#include "platform/audio/AudioResampler.h"
#include "wtf/MathExtras.h"

namespace blink {

const size_t AudioResamplerKernel::MaxFramesToProcess = 128;

AudioResamplerKernel::AudioResamplerKernel(AudioResampler* resampler)
    : m_resampler(resampler),
      // The buffer size must be large enough to hold up to two extra sample
      // frames for the linear interpolation.
      m_sourceBuffer(
          2 + static_cast<int>(MaxFramesToProcess * AudioResampler::MaxRate)),
      m_virtualReadIndex(0.0),
      m_fillIndex(0) {
  m_lastValues[0] = 0.0f;
  m_lastValues[1] = 0.0f;
}

float* AudioResamplerKernel::getSourcePointer(
    size_t framesToProcess,
    size_t* numberOfSourceFramesNeededP) {
  ASSERT(framesToProcess <= MaxFramesToProcess);

  // Calculate the next "virtual" index.  After process() is called,
  // m_virtualReadIndex will equal this value.
  double nextFractionalIndex = m_virtualReadIndex + framesToProcess * rate();

  // Because we're linearly interpolating between the previous and next sample
  // we need to round up so we include the next sample.
  int endIndex = static_cast<int>(nextFractionalIndex +
                                  1.0);  // round up to next integer index

  // Determine how many input frames we'll need.
  // We need to fill the buffer up to and including endIndex (so add 1) but
  // we've already buffered m_fillIndex frames from last time.
  size_t framesNeeded = 1 + endIndex - m_fillIndex;
  if (numberOfSourceFramesNeededP)
    *numberOfSourceFramesNeededP = framesNeeded;

  // Do bounds checking for the source buffer.
  bool isGood = m_fillIndex < m_sourceBuffer.size() &&
                m_fillIndex + framesNeeded <= m_sourceBuffer.size();
  ASSERT(isGood);
  if (!isGood)
    return 0;

  return m_sourceBuffer.data() + m_fillIndex;
}

void AudioResamplerKernel::process(float* destination, size_t framesToProcess) {
  ASSERT(framesToProcess <= MaxFramesToProcess);

  float* source = m_sourceBuffer.data();

  double rate = this->rate();
  rate = clampTo(rate, 0.0, AudioResampler::MaxRate);

  // Start out with the previous saved values (if any).
  if (m_fillIndex > 0) {
    source[0] = m_lastValues[0];
    source[1] = m_lastValues[1];
  }

  // Make a local copy.
  double virtualReadIndex = m_virtualReadIndex;

  // Sanity check source buffer access.
  ASSERT(framesToProcess > 0);
  ASSERT(virtualReadIndex >= 0 &&
         1 + static_cast<unsigned>(virtualReadIndex +
                                   (framesToProcess - 1) * rate) <
             m_sourceBuffer.size());

  // Do the linear interpolation.
  int n = framesToProcess;
  while (n--) {
    unsigned readIndex = static_cast<unsigned>(virtualReadIndex);
    double interpolationFactor = virtualReadIndex - readIndex;

    double sample1 = source[readIndex];
    double sample2 = source[readIndex + 1];

    double sample =
        (1.0 - interpolationFactor) * sample1 + interpolationFactor * sample2;

    *destination++ = static_cast<float>(sample);

    virtualReadIndex += rate;
  }

  // Save the last two sample-frames which will later be used at the beginning
  // of the source buffer the next time around.
  int readIndex = static_cast<int>(virtualReadIndex);
  m_lastValues[0] = source[readIndex];
  m_lastValues[1] = source[readIndex + 1];
  m_fillIndex = 2;

  // Wrap the virtual read index back to the start of the buffer.
  virtualReadIndex -= readIndex;

  // Put local copy back into member variable.
  m_virtualReadIndex = virtualReadIndex;
}

void AudioResamplerKernel::reset() {
  m_virtualReadIndex = 0.0;
  m_fillIndex = 0;
  m_lastValues[0] = 0.0f;
  m_lastValues[1] = 0.0f;
}

double AudioResamplerKernel::rate() const {
  return m_resampler->rate();
}

}  // namespace blink