Source/WebCore/platform/audio/FFTFrame.cpp - WebKit - Git at Google

 /*
  * 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 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 "FFTFrame.h"

 #include "Logging.h"
 #include <complex>
 #include <wtf/MathExtras.h>

 #ifndef NDEBUG
 #include <stdio.h>
 #endif

 namespace WebCore {

 void FFTFrame::doPaddedFFT(const float* data, size_t dataSize)
 {
     // Zero-pad the impulse response
     AudioFloatArray paddedResponse(fftSize()); // zero-initialized
     paddedResponse.copyToRange(data, 0, dataSize);

     // Get the frequency-domain version of padded response
     doFFT(paddedResponse.data());
 }

 std::unique_ptr<FFTFrame> FFTFrame::createInterpolatedFrame(const FFTFrame& frame1, const FFTFrame& frame2, double x)
 {
     auto newFrame = makeUnique<FFTFrame>(frame1.fftSize());

     newFrame->interpolateFrequencyComponents(frame1, frame2, x);

     // In the time-domain, the 2nd half of the response must be zero, to avoid circular convolution aliasing...
     int fftSize = newFrame->fftSize();
     AudioFloatArray buffer(fftSize);
     newFrame->doInverseFFT(buffer.data());
     buffer.zeroRange(fftSize / 2, fftSize);

     // Put back into frequency domain.
     newFrame->doFFT(buffer.data());

     return newFrame;
 }

 void FFTFrame::interpolateFrequencyComponents(const FFTFrame& frame1, const FFTFrame& frame2, double interp)
 {
     // FIXME : with some work, this method could be optimized

     float* realP = realData();
     float* imagP = imagData();

     const float* realP1 = frame1.realData();
     const float* imagP1 = frame1.imagData();
     const float* realP2 = frame2.realData();
     const float* imagP2 = frame2.imagData();

     m_FFTSize = frame1.fftSize();
     m_log2FFTSize = frame1.log2FFTSize();

     double s1base = (1.0 - interp);
     double s2base = interp;

     double phaseAccum = 0.0;
     double lastPhase1 = 0.0;
     double lastPhase2 = 0.0;

     realP[0] = static_cast<float>(s1base * realP1[0] + s2base * realP2[0]);
     imagP[0] = static_cast<float>(s1base * imagP1[0] + s2base * imagP2[0]);

     int n = m_FFTSize / 2;

     for (int i = 1; i < n; ++i) {
         std::complex<double> c1(realP1[i], imagP1[i]);
         std::complex<double> c2(realP2[i], imagP2[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 > piDouble)
             deltaPhase1 -= 2.0 * piDouble;
         if (deltaPhase1 < -piDouble)
             deltaPhase1 += 2.0 * piDouble;
         if (deltaPhase2 > piDouble)
             deltaPhase2 -= 2.0 * piDouble;
         if (deltaPhase2 < -piDouble)
             deltaPhase2 += 2.0 * piDouble;

         // Blend group-delays
         double deltaPhaseBlend;

         if (deltaPhase1 - deltaPhase2 > piDouble)
             deltaPhaseBlend = s1 * deltaPhase1 + s2 * (2.0 * piDouble + deltaPhase2);
         else if (deltaPhase2 - deltaPhase1 > piDouble)
             deltaPhaseBlend = s1 * (2.0 * piDouble + deltaPhase1) + s2 * deltaPhase2;
         else
             deltaPhaseBlend = s1 * deltaPhase1 + s2 * deltaPhase2;

         phaseAccum += deltaPhaseBlend;

         // Unwrap
         if (phaseAccum > piDouble)
             phaseAccum -= 2.0 * piDouble;
         if (phaseAccum < -piDouble)
             phaseAccum += 2.0 * piDouble;

         std::complex<double> c = std::polar(mag, phaseAccum);

         realP[i] = static_cast<float>(c.real());
         imagP[i] = static_cast<float>(c.imag());
     }
 }

 double FFTFrame::extractAverageGroupDelay()
 {
     float* realP = realData();
     float* imagP = imagData();

     double aveSum = 0.0;
     double weightSum = 0.0;
     double lastPhase = 0.0;

     int halfSize = fftSize() / 2;

     const double kSamplePhaseDelay = (2.0 * piDouble) / double(fftSize());

     // Calculate weighted average group delay
     for (int i = 0; i < halfSize; i++) {
         std::complex<double> c(realP[i], imagP[i]);
         double mag = abs(c);
         double phase = arg(c);

         double deltaPhase = phase - lastPhase;
         lastPhase = phase;

         // Unwrap
         if (deltaPhase < -piDouble)
             deltaPhase += 2.0 * piDouble;
         if (deltaPhase > piDouble)
             deltaPhase -= 2.0 * piDouble;

         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)
     if (aveSampleDelay > 20.0)
         aveSampleDelay -= 20.0;

     // Remove average group delay (minus 20 samples for headroom)
     addConstantGroupDelay(-aveSampleDelay);

     // Remove DC offset
     realP[0] = 0.0f;

     return aveSampleDelay;
 }

 void FFTFrame::addConstantGroupDelay(double sampleFrameDelay)
 {
     int halfSize = fftSize() / 2;

     float* realP = realData();
     float* imagP = imagData();

     const double kSamplePhaseDelay = (2.0 * piDouble) / double(fftSize());

     double phaseAdj = -sampleFrameDelay * kSamplePhaseDelay;

     // Add constant group delay
     for (int i = 1; i < halfSize; i++) {
         std::complex<double> c(realP[i], imagP[i]);
         double mag = abs(c);
         double phase = arg(c);

         phase += i * phaseAdj;

         std::complex<double> c2 = std::polar(mag, phase);

         realP[i] = static_cast<float>(c2.real());
         imagP[i] = static_cast<float>(c2.imag());
     }
 }

 #ifndef NDEBUG
 void FFTFrame::print()
 {
     FFTFrame& frame = *this;
     float* realP = frame.realData();
     float* imagP = frame.imagData();
     LOG(WebAudio, "**** \n");
     LOG(WebAudio, "DC = %f : nyquist = %f\n", realP[0], imagP[0]);

     int n = m_FFTSize / 2;

     for (int i = 1; i < n; i++) {
         double mag = sqrt(realP[i] * realP[i] + imagP[i] * imagP[i]);
         double phase = atan2(realP[i], imagP[i]);

         LOG(WebAudio, "[%d] (%f %f)\n", i, mag, phase);
     }
     LOG(WebAudio, "****\n");
 }
 #endif // NDEBUG

 } // namespace WebCore

 #endif // ENABLE(WEB_AUDIO)
	/*
	* 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 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 "FFTFrame.h"

	#include "Logging.h"
	#include <complex>
	#include <wtf/MathExtras.h>

	#ifndef NDEBUG
	#include <stdio.h>
	#endif

	namespace WebCore {

	void FFTFrame::doPaddedFFT(const float* data, size_t dataSize)
	{
	// Zero-pad the impulse response
	AudioFloatArray paddedResponse(fftSize()); // zero-initialized
	paddedResponse.copyToRange(data, 0, dataSize);

	// Get the frequency-domain version of padded response
	doFFT(paddedResponse.data());
	}

	std::unique_ptr<FFTFrame> FFTFrame::createInterpolatedFrame(const FFTFrame& frame1, const FFTFrame& frame2, double x)
	{
	auto newFrame = makeUnique<FFTFrame>(frame1.fftSize());

	newFrame->interpolateFrequencyComponents(frame1, frame2, x);

	// In the time-domain, the 2nd half of the response must be zero, to avoid circular convolution aliasing...
	int fftSize = newFrame->fftSize();
	AudioFloatArray buffer(fftSize);
	newFrame->doInverseFFT(buffer.data());
	buffer.zeroRange(fftSize / 2, fftSize);

	// Put back into frequency domain.
	newFrame->doFFT(buffer.data());

	return newFrame;
	}

	void FFTFrame::interpolateFrequencyComponents(const FFTFrame& frame1, const FFTFrame& frame2, double interp)
	{
	// FIXME : with some work, this method could be optimized

	float* realP = realData();
	float* imagP = imagData();

	const float* realP1 = frame1.realData();
	const float* imagP1 = frame1.imagData();
	const float* realP2 = frame2.realData();
	const float* imagP2 = frame2.imagData();

	m_FFTSize = frame1.fftSize();
	m_log2FFTSize = frame1.log2FFTSize();

	double s1base = (1.0 - interp);
	double s2base = interp;

	double phaseAccum = 0.0;
	double lastPhase1 = 0.0;
	double lastPhase2 = 0.0;

	realP[0] = static_cast<float>(s1base * realP1[0] + s2base * realP2[0]);
	imagP[0] = static_cast<float>(s1base * imagP1[0] + s2base * imagP2[0]);

	int n = m_FFTSize / 2;

	for (int i = 1; i < n; ++i) {
	std::complex<double> c1(realP1[i], imagP1[i]);
	std::complex<double> c2(realP2[i], imagP2[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 > piDouble)
	deltaPhase1 -= 2.0 * piDouble;
	if (deltaPhase1 < -piDouble)
	deltaPhase1 += 2.0 * piDouble;
	if (deltaPhase2 > piDouble)
	deltaPhase2 -= 2.0 * piDouble;
	if (deltaPhase2 < -piDouble)
	deltaPhase2 += 2.0 * piDouble;

	// Blend group-delays
	double deltaPhaseBlend;

	if (deltaPhase1 - deltaPhase2 > piDouble)
	deltaPhaseBlend = s1 * deltaPhase1 + s2 * (2.0 * piDouble + deltaPhase2);
	else if (deltaPhase2 - deltaPhase1 > piDouble)
	deltaPhaseBlend = s1 * (2.0 * piDouble + deltaPhase1) + s2 * deltaPhase2;
	else
	deltaPhaseBlend = s1 * deltaPhase1 + s2 * deltaPhase2;

	phaseAccum += deltaPhaseBlend;

	// Unwrap
	if (phaseAccum > piDouble)
	phaseAccum -= 2.0 * piDouble;
	if (phaseAccum < -piDouble)
	phaseAccum += 2.0 * piDouble;

	std::complex<double> c = std::polar(mag, phaseAccum);

	realP[i] = static_cast<float>(c.real());
	imagP[i] = static_cast<float>(c.imag());
	}
	}

	double FFTFrame::extractAverageGroupDelay()
	{
	float* realP = realData();
	float* imagP = imagData();

	double aveSum = 0.0;
	double weightSum = 0.0;
	double lastPhase = 0.0;

	int halfSize = fftSize() / 2;

	const double kSamplePhaseDelay = (2.0 * piDouble) / double(fftSize());

	// Calculate weighted average group delay
	for (int i = 0; i < halfSize; i++) {
	std::complex<double> c(realP[i], imagP[i]);
	double mag = abs(c);
	double phase = arg(c);

	double deltaPhase = phase - lastPhase;
	lastPhase = phase;

	// Unwrap
	if (deltaPhase < -piDouble)
	deltaPhase += 2.0 * piDouble;
	if (deltaPhase > piDouble)
	deltaPhase -= 2.0 * piDouble;

	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)
	if (aveSampleDelay > 20.0)
	aveSampleDelay -= 20.0;

	// Remove average group delay (minus 20 samples for headroom)
	addConstantGroupDelay(-aveSampleDelay);

	// Remove DC offset
	realP[0] = 0.0f;

	return aveSampleDelay;
	}

	void FFTFrame::addConstantGroupDelay(double sampleFrameDelay)
	{
	int halfSize = fftSize() / 2;

	float* realP = realData();
	float* imagP = imagData();

	const double kSamplePhaseDelay = (2.0 * piDouble) / double(fftSize());

	double phaseAdj = -sampleFrameDelay * kSamplePhaseDelay;

	// Add constant group delay
	for (int i = 1; i < halfSize; i++) {
	std::complex<double> c(realP[i], imagP[i]);
	double mag = abs(c);
	double phase = arg(c);

	phase += i * phaseAdj;

	std::complex<double> c2 = std::polar(mag, phase);

	realP[i] = static_cast<float>(c2.real());
	imagP[i] = static_cast<float>(c2.imag());
	}
	}

	#ifndef NDEBUG
	void FFTFrame::print()
	{
	FFTFrame& frame = *this;
	float* realP = frame.realData();
	float* imagP = frame.imagData();
	LOG(WebAudio, "**** \n");
	LOG(WebAudio, "DC = %f : nyquist = %f\n", realP[0], imagP[0]);

	int n = m_FFTSize / 2;

	for (int i = 1; i < n; i++) {
	double mag = sqrt(realP[i] * realP[i] + imagP[i] * imagP[i]);
	double phase = atan2(realP[i], imagP[i]);

	LOG(WebAudio, "[%d] (%f %f)\n", i, mag, phase);
	}
	LOG(WebAudio, "****\n");
	}
	#endif // NDEBUG

	} // namespace WebCore

	#endif // ENABLE(WEB_AUDIO)