Add m-esr52 at 52.6.0

1 files changed, 328 insertions, 0 deletions

diff --git a/dom/media/webaudio/blink/HRTFElevation.cpp b/dom/media/webaudio/blink/HRTFElevation.cpp
new file mode 100644
index 000000000..2300872f3
--- /dev/null
+++ b/dom/media/webaudio/blink/HRTFElevation.cpp
@@ -0,0 +1,328 @@
+/*
+ * 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 "HRTFElevation.h"
+
+#include <speex/speex_resampler.h>
+#include "mozilla/PodOperations.h"
+#include "AudioSampleFormat.h"
+
+#include "IRC_Composite_C_R0195-incl.cpp"
+
+using namespace std;
+using namespace mozilla;
+ 
+namespace WebCore {
+
+const int elevationSpacing = irc_composite_c_r0195_elevation_interval;
+const int firstElevation = irc_composite_c_r0195_first_elevation;
+const int numberOfElevations = MOZ_ARRAY_LENGTH(irc_composite_c_r0195);
+
+const unsigned HRTFElevation::NumberOfTotalAzimuths = 360 / 15 * 8;
+
+const int rawSampleRate = irc_composite_c_r0195_sample_rate;
+
+// Number of frames in an individual impulse response.
+const size_t ResponseFrameSize = 256;
+
+size_t HRTFElevation::sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const
+{
+    size_t amount = aMallocSizeOf(this);
+
+    amount += m_kernelListL.ShallowSizeOfExcludingThis(aMallocSizeOf);
+    for (size_t i = 0; i < m_kernelListL.Length(); i++) {
+        amount += m_kernelListL[i]->sizeOfIncludingThis(aMallocSizeOf);
+    }
+
+    return amount;
+}
+
+size_t HRTFElevation::fftSizeForSampleRate(float sampleRate)
+{
+    // The IRCAM HRTF impulse responses were 512 sample-frames @44.1KHz,
+    // but these have been truncated to 256 samples.
+    // An FFT-size of twice impulse response size is used (for convolution).
+    // So for sample rates of 44.1KHz an FFT size of 512 is good.
+    // We double the FFT-size only for sample rates at least double this.
+    // If the FFT size is too large then the impulse response will be padded
+    // with zeros without the fade-out provided by HRTFKernel.
+    MOZ_ASSERT(sampleRate > 1.0 && sampleRate < 1048576.0);
+
+    // This is the size if we were to use all raw response samples.
+    unsigned resampledLength =
+        floorf(ResponseFrameSize * sampleRate / rawSampleRate);
+    // Keep things semi-sane, with max FFT size of 1024.
+    unsigned size = min(resampledLength, 1023U);
+    // Ensure a minimum of 2 * WEBAUDIO_BLOCK_SIZE (with the size++ below) for
+    // FFTConvolver and set the 8 least significant bits for rounding up to
+    // the next power of 2 below.
+    size |= 2 * WEBAUDIO_BLOCK_SIZE - 1;
+    // Round up to the next power of 2, making the FFT size no more than twice
+    // the impulse response length.  This doubles size for values that are
+    // already powers of 2.  This works by filling in alls bit to right of the
+    // most significant bit.  The most significant bit is no greater than
+    // 1 << 9, and the least significant 8 bits were already set above, so
+    // there is at most one bit to add.
+    size |= (size >> 1);
+    size++;
+    MOZ_ASSERT((size & (size - 1)) == 0);
+
+    return size;
+}
+
+nsReturnRef<HRTFKernel> HRTFElevation::calculateKernelForAzimuthElevation(int azimuth, int elevation, SpeexResamplerState* resampler, float sampleRate)
+{
+    int elevationIndex = (elevation - firstElevation) / elevationSpacing;
+    MOZ_ASSERT(elevationIndex >= 0 && elevationIndex <= numberOfElevations);
+
+    int numberOfAzimuths = irc_composite_c_r0195[elevationIndex].count;
+    int azimuthSpacing = 360 / numberOfAzimuths;
+    MOZ_ASSERT(numberOfAzimuths * azimuthSpacing == 360);
+
+    int azimuthIndex = azimuth / azimuthSpacing;
+    MOZ_ASSERT(azimuthIndex * azimuthSpacing == azimuth);
+
+    const int16_t (&impulse_response_data)[ResponseFrameSize] =
+        irc_composite_c_r0195[elevationIndex].azimuths[azimuthIndex];
+
+    // When libspeex_resampler is compiled with FIXED_POINT, samples in
+    // speex_resampler_process_float are rounded directly to int16_t, which
+    // only works well if the floats are in the range +/-32767.  On such
+    // platforms it's better to resample before converting to float anyway.
+#ifdef MOZ_SAMPLE_TYPE_S16
+#  define RESAMPLER_PROCESS speex_resampler_process_int
+    const int16_t* response = impulse_response_data;
+    const int16_t* resampledResponse;
+#else
+#  define RESAMPLER_PROCESS speex_resampler_process_float
+    float response[ResponseFrameSize];
+    ConvertAudioSamples(impulse_response_data, response, ResponseFrameSize);
+    float* resampledResponse;
+#endif
+
+    // Note that depending on the fftSize returned by the panner, we may be truncating the impulse response.
+    const size_t resampledResponseLength = fftSizeForSampleRate(sampleRate) / 2;
+
+    AutoTArray<AudioDataValue, 2 * ResponseFrameSize> resampled;
+    if (sampleRate == rawSampleRate) {
+        resampledResponse = response;
+        MOZ_ASSERT(resampledResponseLength == ResponseFrameSize);
+    } else {
+        resampled.SetLength(resampledResponseLength);
+        resampledResponse = resampled.Elements();
+        speex_resampler_skip_zeros(resampler);
+
+        // Feed the input buffer into the resampler.
+        spx_uint32_t in_len = ResponseFrameSize;
+        spx_uint32_t out_len = resampled.Length();
+        RESAMPLER_PROCESS(resampler, 0, response, &in_len,
+                          resampled.Elements(), &out_len);
+
+        if (out_len < resampled.Length()) {
+            // The input should have all been processed.
+            MOZ_ASSERT(in_len == ResponseFrameSize);
+            // Feed in zeros get the data remaining in the resampler.
+            spx_uint32_t out_index = out_len;
+            in_len = speex_resampler_get_input_latency(resampler);
+            out_len = resampled.Length() - out_index;
+            RESAMPLER_PROCESS(resampler, 0, nullptr, &in_len,
+                              resampled.Elements() + out_index, &out_len);
+            out_index += out_len;
+            // There may be some uninitialized samples remaining for very low
+            // sample rates.
+            PodZero(resampled.Elements() + out_index,
+                    resampled.Length() - out_index);
+        }
+
+        speex_resampler_reset_mem(resampler);
+    }
+
+#ifdef MOZ_SAMPLE_TYPE_S16
+    AutoTArray<float, 2 * ResponseFrameSize> floatArray;
+    floatArray.SetLength(resampledResponseLength);
+    float *floatResponse = floatArray.Elements();
+    ConvertAudioSamples(resampledResponse,
+                        floatResponse, resampledResponseLength);
+#else
+    float *floatResponse = resampledResponse;
+#endif
+#undef RESAMPLER_PROCESS
+
+    return HRTFKernel::create(floatResponse, resampledResponseLength, sampleRate);
+}
+
+// The range of elevations for the IRCAM impulse responses varies depending on azimuth, but the minimum elevation appears to always be -45.
+//
+// Here's how it goes:
+static int maxElevations[] = {
+        //  Azimuth
+        //
+    90, // 0  
+    45, // 15 
+    60, // 30 
+    45, // 45 
+    75, // 60 
+    45, // 75 
+    60, // 90 
+    45, // 105 
+    75, // 120 
+    45, // 135 
+    60, // 150 
+    45, // 165 
+    75, // 180 
+    45, // 195 
+    60, // 210 
+    45, // 225 
+    75, // 240 
+    45, // 255 
+    60, // 270 
+    45, // 285 
+    75, // 300 
+    45, // 315 
+    60, // 330 
+    45 //  345 
+};
+
+nsReturnRef<HRTFElevation> HRTFElevation::createBuiltin(int elevation, float sampleRate)
+{
+    if (elevation < firstElevation ||
+        elevation > firstElevation + numberOfElevations * elevationSpacing ||
+        (elevation / elevationSpacing) * elevationSpacing != elevation)
+        return nsReturnRef<HRTFElevation>();
+        
+    // Spacing, in degrees, between every azimuth loaded from resource.
+    // Some elevations do not have data for all these intervals.
+    // See maxElevations.
+    static const unsigned AzimuthSpacing = 15;
+    static const unsigned NumberOfRawAzimuths = 360 / AzimuthSpacing;
+    static_assert(AzimuthSpacing * NumberOfRawAzimuths == 360,
+                  "Not a multiple");
+    static const unsigned InterpolationFactor =
+        NumberOfTotalAzimuths / NumberOfRawAzimuths;
+    static_assert(NumberOfTotalAzimuths ==
+                  NumberOfRawAzimuths * InterpolationFactor, "Not a multiple");
+
+    HRTFKernelList kernelListL;
+    kernelListL.SetLength(NumberOfTotalAzimuths);
+
+    SpeexResamplerState* resampler = sampleRate == rawSampleRate ? nullptr :
+        speex_resampler_init(1, rawSampleRate, sampleRate,
+                             SPEEX_RESAMPLER_QUALITY_MIN, nullptr);
+
+    // Load convolution kernels from HRTF files.
+    int interpolatedIndex = 0;
+    for (unsigned rawIndex = 0; rawIndex < NumberOfRawAzimuths; ++rawIndex) {
+        // Don't let elevation exceed maximum for this azimuth.
+        int maxElevation = maxElevations[rawIndex];
+        int actualElevation = min(elevation, maxElevation);
+
+        kernelListL[interpolatedIndex] = calculateKernelForAzimuthElevation(rawIndex * AzimuthSpacing, actualElevation, resampler, sampleRate);
+            
+        interpolatedIndex += InterpolationFactor;
+    }
+
+    if (resampler)
+        speex_resampler_destroy(resampler);
+
+    // Now go back and interpolate intermediate azimuth values.
+    for (unsigned i = 0; i < NumberOfTotalAzimuths; i += InterpolationFactor) {
+        int j = (i + InterpolationFactor) % NumberOfTotalAzimuths;
+
+        // Create the interpolated convolution kernels and delays.
+        for (unsigned jj = 1; jj < InterpolationFactor; ++jj) {
+            float x = float(jj) / float(InterpolationFactor); // interpolate from 0 -> 1
+
+            kernelListL[i + jj] = HRTFKernel::createInterpolatedKernel(kernelListL[i], kernelListL[j], x);
+        }
+    }
+    
+    return nsReturnRef<HRTFElevation>(new HRTFElevation(&kernelListL, elevation, sampleRate));
+}
+
+nsReturnRef<HRTFElevation> HRTFElevation::createByInterpolatingSlices(HRTFElevation* hrtfElevation1, HRTFElevation* hrtfElevation2, float x, float sampleRate)
+{
+    MOZ_ASSERT(hrtfElevation1 && hrtfElevation2);
+    if (!hrtfElevation1 || !hrtfElevation2)
+        return nsReturnRef<HRTFElevation>();
+        
+    MOZ_ASSERT(x >= 0.0 && x < 1.0);
+    
+    HRTFKernelList kernelListL;
+    kernelListL.SetLength(NumberOfTotalAzimuths);
+
+    const HRTFKernelList& kernelListL1 = hrtfElevation1->kernelListL();
+    const HRTFKernelList& kernelListL2 = hrtfElevation2->kernelListL();
+    
+    // Interpolate kernels of corresponding azimuths of the two elevations.
+    for (unsigned i = 0; i < NumberOfTotalAzimuths; ++i) {
+        kernelListL[i] = HRTFKernel::createInterpolatedKernel(kernelListL1[i], kernelListL2[i], x);
+    }
+
+    // Interpolate elevation angle.
+    double angle = (1.0 - x) * hrtfElevation1->elevationAngle() + x * hrtfElevation2->elevationAngle();
+    
+    return nsReturnRef<HRTFElevation>(new HRTFElevation(&kernelListL, static_cast<int>(angle), sampleRate));
+}
+
+void HRTFElevation::getKernelsFromAzimuth(double azimuthBlend, unsigned azimuthIndex, HRTFKernel* &kernelL, HRTFKernel* &kernelR, double& frameDelayL, double& frameDelayR)
+{
+    bool checkAzimuthBlend = azimuthBlend >= 0.0 && azimuthBlend < 1.0;
+    MOZ_ASSERT(checkAzimuthBlend);
+    if (!checkAzimuthBlend)
+        azimuthBlend = 0.0;
+    
+    unsigned numKernels = m_kernelListL.Length();
+
+    bool isIndexGood = azimuthIndex < numKernels;
+    MOZ_ASSERT(isIndexGood);
+    if (!isIndexGood) {
+        kernelL = 0;
+        kernelR = 0;
+        return;
+    }
+    
+    // Return the left and right kernels,
+    // using symmetry to produce the right kernel.
+    kernelL = m_kernelListL[azimuthIndex];
+    int azimuthIndexR = (numKernels - azimuthIndex) % numKernels;
+    kernelR = m_kernelListL[azimuthIndexR];
+
+    frameDelayL = kernelL->frameDelay();
+    frameDelayR = kernelR->frameDelay();
+
+    int azimuthIndex2L = (azimuthIndex + 1) % numKernels;
+    double frameDelay2L = m_kernelListL[azimuthIndex2L]->frameDelay();
+    int azimuthIndex2R = (numKernels - azimuthIndex2L) % numKernels;
+    double frameDelay2R = m_kernelListL[azimuthIndex2R]->frameDelay();
+
+    // Linearly interpolate delays.
+    frameDelayL = (1.0 - azimuthBlend) * frameDelayL + azimuthBlend * frameDelay2L;
+    frameDelayR = (1.0 - azimuthBlend) * frameDelayR + azimuthBlend * frameDelay2R;
+}
+
+} // namespace WebCore