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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.
* 3. Neither the name of Apple Computer, Inc. ("Apple") nor the names of
* its contributors may be used to endorse or promote products derived
* from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY APPLE 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 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 "HRTFKernel.h"
#include "AudioChannel.h"
#include "Biquad.h"
#include "FFTFrame.h"
#include "FloatConversion.h"
#include <wtf/MathExtras.h>
using namespace std;
namespace WebCore {
// 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 = narrowPrecisionToFloat(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.
size_t truncatedResponseLength = min(responseLength, fftSize / 2); // truncate if necessary to max impulse response length allowed by FFT
// 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 = adoptPtr(new FFTFrame(fftSize));
m_fftFrame->doPaddedFFT(impulseResponse, truncatedResponseLength);
}
size_t HRTFKernel::fftSize() const
{
return m_fftFrame->fftSize();
}
PassOwnPtr<AudioChannel> HRTFKernel::createImpulseResponse()
{
OwnPtr<AudioChannel> channel = adoptPtr(new AudioChannel(fftSize()));
FFTFrame fftFrame(*m_fftFrame);
// Add leading delay back in.
fftFrame.addConstantGroupDelay(m_frameDelay);
fftFrame.doInverseFFT(channel->mutableData());
return channel.release();
}
// Interpolates two kernels with x: 0 -> 1 and returns the result.
PassRefPtr<HRTFKernel> HRTFKernel::createInterpolatedKernel(HRTFKernel* kernel1, HRTFKernel* kernel2, float x)
{
ASSERT(kernel1 && kernel2);
if (!kernel1 || !kernel2)
return 0;
ASSERT(x >= 0.0 && x < 1.0);
x = min(1.0f, max(0.0f, x));
float sampleRate1 = kernel1->sampleRate();
float sampleRate2 = kernel2->sampleRate();
ASSERT(sampleRate1 == sampleRate2);
if (sampleRate1 != sampleRate2)
return 0;
float frameDelay = (1 - x) * kernel1->frameDelay() + x * kernel2->frameDelay();
OwnPtr<FFTFrame> interpolatedFrame = FFTFrame::createInterpolatedFrame(*kernel1->fftFrame(), *kernel2->fftFrame(), x);
return HRTFKernel::create(interpolatedFrame.release(), frameDelay, sampleRate1);
}
} // namespace WebCore
#endif // ENABLE(WEB_AUDIO)
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