Add m-esr52 at 52.6.0

1 files changed, 226 insertions, 0 deletions

diff --git a/dom/media/webaudio/FFTBlock.cpp b/dom/media/webaudio/FFTBlock.cpp
new file mode 100644
index 000000000..f517ef283
--- /dev/null
+++ b/dom/media/webaudio/FFTBlock.cpp
@@ -0,0 +1,226 @@
+/* -*- Mode: C++; tab-width: 4; indent-tabs-mode: nil; c-basic-offset: 4 -*- */
+/* vim:set ts=4 sw=4 sts=4 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;
+
+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