Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 231 insertions, 0 deletions

diff --git a/dom/media/webaudio/FFTBlock.cpp b/dom/media/webaudio/FFTBlock.cpp
new file mode 100644
index 0000000000..6789ca0264
--- /dev/null
+++ b/dom/media/webaudio/FFTBlock.cpp
@@ -0,0 +1,231 @@
+/* -*- 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
+ *     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 "FFTBlock.h"
+
+#include <complex>
+
+namespace mozilla {
+
+typedef std::complex<double> Complex;
+
+#ifdef MOZ_LIBAV_FFT
+FFmpegRDFTFuncs FFTBlock::sRDFTFuncs;
+#endif
+
+static double fdlibm_cabs(const Complex& z) {
+  return fdlibm_hypot(real(z), imag(z));
+}
+
+static double fdlibm_carg(const Complex& z) {
+  return fdlibm_atan2(imag(z), real(z));
+}
+
+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 = fdlibm_cabs(c1);
+    double mag2 = fdlibm_cabs(c2);
+
+    // Interpolate magnitudes in decibels
+    double mag1db = 20.0 * fdlibm_log10(mag1);
+    double mag2db = 20.0 * fdlibm_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 = fdlibm_pow(s1, 0.75);
+      s2 = 1.0 - s1;
+    } else if (magdbdiff > threshold && mag2db < 0.0) {
+      s2 = fdlibm_pow(s2, 0.75);
+      s1 = 1.0 - s2;
+    }
+
+    // Average magnitude by decibels instead of linearly
+    double magdb = s1 * mag1db + s2 * mag2db;
+    double mag = fdlibm_pow(10.0, 0.05 * magdb);
+
+    // Now, deal with phase
+    double phase1 = fdlibm_carg(c1);
+    double phase2 = fdlibm_carg(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 * fdlibm_cos(phaseAccum));
+    dft[i].i = static_cast<float>(mag * fdlibm_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 = fdlibm_cabs(c);
+    double phase = fdlibm_carg(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 = fdlibm_cabs(c);
+    double phase = fdlibm_carg(c);
+
+    phase += i * phaseAdj;
+
+    dft[i].r = static_cast<float>(mag * fdlibm_cos(phase));
+    dft[i].i = static_cast<float>(mag * fdlibm_sin(phase));
+  }
+}
+
+}  // namespace mozilla