| /* |
| * 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 |
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| * 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->data(); |
| |
| 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, bool bassBoost) |
| : 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->data(); |
| size_t responseLength = channel->length(); |
| |
| if (bassBoost) { |
| // Run through some post-processing to boost the bass a little -- the HRTF's seem to be a little bass-deficient. |
| // FIXME: this post-processing should have already been applied to the HRTF file resources. Once the files are put into this form, |
| // then this code path can be removed along with the bassBoost parameter. |
| Biquad filter; |
| filter.setLowShelfParams(700.0 / nyquist(), 6.0); // boost 6dB at 700Hz |
| filter.process(impulseResponse, impulseResponse, responseLength); |
| } |
| |
| // 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); |
| } |
| |
| 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->data()); |
| |
| 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) |