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/*
* 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 "config.h"
#if ENABLE(WEB_AUDIO)
#include "AudioResamplerKernel.h"
#include "AudioResampler.h"
#include <algorithm>
using namespace std;
namespace WebCore {
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 = max(0.0, rate);
rate = min(AudioResampler::MaxRate, rate);
// 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 WebCore
#endif // ENABLE(WEB_AUDIO)
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