1/*
2 * Copyright (C) 2010 Google Inc. All rights reserved.
3 *
4 * Redistribution and use in source and binary forms, with or without
5 * modification, are permitted provided that the following conditions
6 * are met:
7 *
8 * 1.  Redistributions of source code must retain the above copyright
9 *     notice, this list of conditions and the following disclaimer.
10 * 2.  Redistributions in binary form must reproduce the above copyright
11 *     notice, this list of conditions and the following disclaimer in the
12 *     documentation and/or other materials provided with the distribution.
13 * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
14 *     its contributors may be used to endorse or promote products derived
15 *     from this software without specific prior written permission.
16 *
17 * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
18 * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
19 * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
20 * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
21 * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
22 * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
23 * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
24 * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
25 * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
26 * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
27 */
28
29#include "config.h"
30
31#if ENABLE(WEB_AUDIO)
32
33#include "platform/audio/HRTFKernel.h"
34
35#include "platform/audio/AudioChannel.h"
36#include "platform/FloatConversion.h"
37#include "wtf/MathExtras.h"
38
39namespace blink {
40
41// Takes the input AudioChannel as an input impulse response and calculates the average group delay.
42// This represents the initial delay before the most energetic part of the impulse response.
43// The sample-frame delay is removed from the impulseP impulse response, and this value  is returned.
44// the length of the passed in AudioChannel must be a power of 2.
45static float extractAverageGroupDelay(AudioChannel* channel, size_t analysisFFTSize)
46{
47    ASSERT(channel);
48
49    float* impulseP = channel->mutableData();
50
51    bool isSizeGood = channel->length() >= analysisFFTSize;
52    ASSERT(isSizeGood);
53    if (!isSizeGood)
54        return 0;
55
56    // Check for power-of-2.
57    ASSERT(1UL << static_cast<unsigned>(log2(analysisFFTSize)) == analysisFFTSize);
58
59    FFTFrame estimationFrame(analysisFFTSize);
60    estimationFrame.doFFT(impulseP);
61
62    float frameDelay = narrowPrecisionToFloat(estimationFrame.extractAverageGroupDelay());
63    estimationFrame.doInverseFFT(impulseP);
64
65    return frameDelay;
66}
67
68HRTFKernel::HRTFKernel(AudioChannel* channel, size_t fftSize, float sampleRate)
69    : m_frameDelay(0)
70    , m_sampleRate(sampleRate)
71{
72    ASSERT(channel);
73
74    // Determine the leading delay (average group delay) for the response.
75    m_frameDelay = extractAverageGroupDelay(channel, fftSize / 2);
76
77    float* impulseResponse = channel->mutableData();
78    size_t responseLength = channel->length();
79
80    // We need to truncate to fit into 1/2 the FFT size (with zero padding) in order to do proper convolution.
81    size_t truncatedResponseLength = std::min(responseLength, fftSize / 2); // truncate if necessary to max impulse response length allowed by FFT
82
83    // Quick fade-out (apply window) at truncation point
84    unsigned numberOfFadeOutFrames = static_cast<unsigned>(sampleRate / 4410); // 10 sample-frames @44.1KHz sample-rate
85    ASSERT(numberOfFadeOutFrames < truncatedResponseLength);
86    if (numberOfFadeOutFrames < truncatedResponseLength) {
87        for (unsigned i = truncatedResponseLength - numberOfFadeOutFrames; i < truncatedResponseLength; ++i) {
88            float x = 1.0f - static_cast<float>(i - (truncatedResponseLength - numberOfFadeOutFrames)) / numberOfFadeOutFrames;
89            impulseResponse[i] *= x;
90        }
91    }
92
93    m_fftFrame = adoptPtr(new FFTFrame(fftSize));
94    m_fftFrame->doPaddedFFT(impulseResponse, truncatedResponseLength);
95}
96
97PassOwnPtr<AudioChannel> HRTFKernel::createImpulseResponse()
98{
99    OwnPtr<AudioChannel> channel = adoptPtr(new AudioChannel(fftSize()));
100    FFTFrame fftFrame(*m_fftFrame);
101
102    // Add leading delay back in.
103    fftFrame.addConstantGroupDelay(m_frameDelay);
104    fftFrame.doInverseFFT(channel->mutableData());
105
106    return channel.release();
107}
108
109// Interpolates two kernels with x: 0 -> 1 and returns the result.
110PassRefPtr<HRTFKernel> HRTFKernel::createInterpolatedKernel(HRTFKernel* kernel1, HRTFKernel* kernel2, float x)
111{
112    ASSERT(kernel1 && kernel2);
113    if (!kernel1 || !kernel2)
114        return nullptr;
115
116    ASSERT(x >= 0.0 && x < 1.0);
117    x = std::min(1.0f, std::max(0.0f, x));
118
119    float sampleRate1 = kernel1->sampleRate();
120    float sampleRate2 = kernel2->sampleRate();
121    ASSERT(sampleRate1 == sampleRate2);
122    if (sampleRate1 != sampleRate2)
123        return nullptr;
124
125    float frameDelay = (1 - x) * kernel1->frameDelay() + x * kernel2->frameDelay();
126
127    OwnPtr<FFTFrame> interpolatedFrame = FFTFrame::createInterpolatedFrame(*kernel1->fftFrame(), *kernel2->fftFrame(), x);
128    return HRTFKernel::create(interpolatedFrame.release(), frameDelay, sampleRate1);
129}
130
131} // namespace blink
132
133#endif // ENABLE(WEB_AUDIO)
134