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/* -*- Mode: C++; tab-width: 4; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
/* vim:set ts=4 sw=2 sts=2 et cindent: */
/*
* 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
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* 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
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* (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
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*/
#include "FFTBlock.h"
#include <complex>
namespace mozilla {
typedef std::complex<double> Complex;
#ifdef MOZ_LIBAV_FFT
FFmpegRDFTFuncs FFTBlock::sRDFTFuncs;
#endif
FFTBlock* FFTBlock::CreateInterpolatedBlock(const FFTBlock& block0,
const FFTBlock& block1,
double interp) {
FFTBlock* newBlock = new FFTBlock(block0.FFTSize());
newBlock->InterpolateFrequencyComponents(block0, block1, interp);
// In the time-domain, the 2nd half of the response must be zero, to avoid
// circular convolution aliasing...
int fftSize = newBlock->FFTSize();
AlignedTArray<float> buffer(fftSize);
newBlock->GetInverseWithoutScaling(buffer.Elements());
AudioBufferInPlaceScale(buffer.Elements(), 1.0f / fftSize, fftSize / 2);
PodZero(buffer.Elements() + fftSize / 2, fftSize / 2);
// Put back into frequency domain.
newBlock->PerformFFT(buffer.Elements());
return newBlock;
}
void FFTBlock::InterpolateFrequencyComponents(const FFTBlock& block0,
const FFTBlock& block1,
double interp) {
// FIXME : with some work, this method could be optimized
ComplexU* dft = mOutputBuffer.Elements();
const ComplexU* dft1 = block0.mOutputBuffer.Elements();
const ComplexU* dft2 = block1.mOutputBuffer.Elements();
MOZ_ASSERT(mFFTSize == block0.FFTSize());
MOZ_ASSERT(mFFTSize == block1.FFTSize());
double s1base = (1.0 - interp);
double s2base = interp;
double phaseAccum = 0.0;
double lastPhase1 = 0.0;
double lastPhase2 = 0.0;
int n = mFFTSize / 2;
dft[0].r = static_cast<float>(s1base * dft1[0].r + s2base * dft2[0].r);
dft[n].r = static_cast<float>(s1base * dft1[n].r + s2base * dft2[n].r);
for (int i = 1; i < n; ++i) {
Complex c1(dft1[i].r, dft1[i].i);
Complex c2(dft2[i].r, dft2[i].i);
double mag1 = abs(c1);
double mag2 = abs(c2);
// Interpolate magnitudes in decibels
double mag1db = 20.0 * log10(mag1);
double mag2db = 20.0 * log10(mag2);
double s1 = s1base;
double s2 = s2base;
double magdbdiff = mag1db - mag2db;
// Empirical tweak to retain higher-frequency zeroes
double threshold = (i > 16) ? 5.0 : 2.0;
if (magdbdiff < -threshold && mag1db < 0.0) {
s1 = pow(s1, 0.75);
s2 = 1.0 - s1;
} else if (magdbdiff > threshold && mag2db < 0.0) {
s2 = pow(s2, 0.75);
s1 = 1.0 - s2;
}
// Average magnitude by decibels instead of linearly
double magdb = s1 * mag1db + s2 * mag2db;
double mag = pow(10.0, 0.05 * magdb);
// Now, deal with phase
double phase1 = arg(c1);
double phase2 = arg(c2);
double deltaPhase1 = phase1 - lastPhase1;
double deltaPhase2 = phase2 - lastPhase2;
lastPhase1 = phase1;
lastPhase2 = phase2;
// Unwrap phase deltas
if (deltaPhase1 > M_PI) deltaPhase1 -= 2.0 * M_PI;
if (deltaPhase1 < -M_PI) deltaPhase1 += 2.0 * M_PI;
if (deltaPhase2 > M_PI) deltaPhase2 -= 2.0 * M_PI;
if (deltaPhase2 < -M_PI) deltaPhase2 += 2.0 * M_PI;
// Blend group-delays
double deltaPhaseBlend;
if (deltaPhase1 - deltaPhase2 > M_PI)
deltaPhaseBlend = s1 * deltaPhase1 + s2 * (2.0 * M_PI + deltaPhase2);
else if (deltaPhase2 - deltaPhase1 > M_PI)
deltaPhaseBlend = s1 * (2.0 * M_PI + deltaPhase1) + s2 * deltaPhase2;
else
deltaPhaseBlend = s1 * deltaPhase1 + s2 * deltaPhase2;
phaseAccum += deltaPhaseBlend;
// Unwrap
if (phaseAccum > M_PI) phaseAccum -= 2.0 * M_PI;
if (phaseAccum < -M_PI) phaseAccum += 2.0 * M_PI;
dft[i].r = static_cast<float>(mag * cos(phaseAccum));
dft[i].i = static_cast<float>(mag * sin(phaseAccum));
}
}
double FFTBlock::ExtractAverageGroupDelay() {
ComplexU* dft = mOutputBuffer.Elements();
double aveSum = 0.0;
double weightSum = 0.0;
double lastPhase = 0.0;
int halfSize = FFTSize() / 2;
const double kSamplePhaseDelay = (2.0 * M_PI) / double(FFTSize());
// Remove DC offset
dft[0].r = 0.0f;
// Calculate weighted average group delay
for (int i = 1; i < halfSize; i++) {
Complex c(dft[i].r, dft[i].i);
double mag = abs(c);
double phase = arg(c);
double deltaPhase = phase - lastPhase;
lastPhase = phase;
// Unwrap
if (deltaPhase < -M_PI) deltaPhase += 2.0 * M_PI;
if (deltaPhase > M_PI) deltaPhase -= 2.0 * M_PI;
aveSum += mag * deltaPhase;
weightSum += mag;
}
// Note how we invert the phase delta wrt frequency since this is how group
// delay is defined
double ave = aveSum / weightSum;
double aveSampleDelay = -ave / kSamplePhaseDelay;
// Leave 20 sample headroom (for leading edge of impulse)
aveSampleDelay -= 20.0;
if (aveSampleDelay <= 0.0) return 0.0;
// Remove average group delay (minus 20 samples for headroom)
AddConstantGroupDelay(-aveSampleDelay);
return aveSampleDelay;
}
void FFTBlock::AddConstantGroupDelay(double sampleFrameDelay) {
int halfSize = FFTSize() / 2;
ComplexU* dft = mOutputBuffer.Elements();
const double kSamplePhaseDelay = (2.0 * M_PI) / double(FFTSize());
double phaseAdj = -sampleFrameDelay * kSamplePhaseDelay;
// Add constant group delay
for (int i = 1; i < halfSize; i++) {
Complex c(dft[i].r, dft[i].i);
double mag = abs(c);
double phase = arg(c);
phase += i * phaseAdj;
dft[i].r = static_cast<float>(mag * cos(phase));
dft[i].i = static_cast<float>(mag * sin(phase));
}
}
} // namespace mozilla