1 files changed, 0 insertions, 229 deletions
diff --git a/WebCore/platform/audio/HRTFPanner.cpp b/WebCore/platform/audio/HRTFPanner.cpp
deleted file mode 100644
index 68bc505..0000000
--- a/WebCore/platform/audio/HRTFPanner.cpp
+++ /dev/null
@@ -1,229 +0,0 @@
-/*
- * 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.
- *
- * THIS SOFTWARE IS PROVIDED BY APPLE INC. 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 INC. 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 "HRTFPanner.h"
-
-#include "AudioBus.h"
-#include "FFTConvolver.h"
-#include "HRTFDatabase.h"
-#include "HRTFDatabaseLoader.h"
-#include <algorithm>
-#include <wtf/MathExtras.h>
-#include <wtf/RefPtr.h>
-
-using namespace std;
- 
-namespace WebCore {
-
-// The value of 2 milliseconds is larger than the largest delay which exists in any HRTFKernel from the default HRTFDatabase (0.0136 seconds).
-// We ASSERT the delay values used in process() with this value.
-const double MaxDelayTimeSeconds = 0.002;
-
-HRTFPanner::HRTFPanner(double sampleRate)
-    : Panner(PanningModelHRTF)
-    , m_sampleRate(sampleRate)
-    , m_isFirstRender(true)
-    , m_azimuthIndex(0)
-    , m_convolverL(fftSizeForSampleRate(sampleRate))
-    , m_convolverR(fftSizeForSampleRate(sampleRate))
-    , m_delayLineL(MaxDelayTimeSeconds, sampleRate)
-    , m_delayLineR(MaxDelayTimeSeconds, sampleRate)
-{ 
-}
-
-HRTFPanner::~HRTFPanner()
-{
-}
-
-size_t HRTFPanner::fftSizeForSampleRate(double sampleRate)
-{
-    // The HRTF impulse responses (loaded as audio resources) are 512 sample-frames @44.1KHz.
-    // Currently, we truncate the impulse responses to half this size, but an FFT-size of twice impulse response size is needed (for convolution).
-    // So for sample rates around 44.1KHz an FFT size of 512 is good.  We double that size for higher sample rates.
-    ASSERT(sampleRate >= 44100 && sampleRate <= 96000.0);
-    return (sampleRate <= 48000.0) ? 512 : 1024;
-}
-
-void HRTFPanner::reset()
-{
-    m_isFirstRender = true;
-    m_convolverL.reset();
-    m_convolverR.reset();
-    m_delayLineL.reset();
-    m_delayLineR.reset();
-}
-
-static bool wrapDistance(int i, int j, int length)
-{
-    int directDistance = abs(i - j);
-    int indirectDistance = length - directDistance;
-
-    return indirectDistance < directDistance;
-}
-
-int HRTFPanner::calculateDesiredAzimuthIndexAndBlend(double azimuth, double& azimuthBlend)
-{
-    // Convert the azimuth angle from the range -180 -> +180 into the range 0 -> 360.
-    // The azimuth index may then be calculated from this positive value.
-    if (azimuth < 0)
-        azimuth += 360.0;
-    
-    HRTFDatabase* database = HRTFDatabaseLoader::defaultHRTFDatabase();    
-    ASSERT(database);
-    
-    int numberOfAzimuths = database->numberOfAzimuths();
-    const double angleBetweenAzimuths = 360.0 / numberOfAzimuths;
-
-    // Calculate the azimuth index and the blend (0 -> 1) for interpolation.
-    double desiredAzimuthIndexFloat = azimuth / angleBetweenAzimuths;
-    int desiredAzimuthIndex = static_cast<int>(desiredAzimuthIndexFloat);
-    azimuthBlend = desiredAzimuthIndexFloat - static_cast<double>(desiredAzimuthIndex);
-    
-    // We don't immediately start using this azimuth index, but instead approach this index from the last index we rendered at.
-    // This minimizes the clicks and graininess for moving sources which occur otherwise.
-    desiredAzimuthIndex = max(0, desiredAzimuthIndex);
-    desiredAzimuthIndex = min(numberOfAzimuths - 1, desiredAzimuthIndex);
-    return desiredAzimuthIndex;
-}
-
-void HRTFPanner::pan(double desiredAzimuth, double elevation, AudioBus* inputBus, AudioBus* outputBus, size_t framesToProcess)
-{
-    unsigned numInputChannels = inputBus ? inputBus->numberOfChannels() : 0;
-
-    bool isInputGood = inputBus &&  numInputChannels >= 1 && numInputChannels <= 2;
-    ASSERT(isInputGood);
-
-    bool isOutputGood = outputBus && outputBus->numberOfChannels() == 2 && framesToProcess <= outputBus->length();
-    ASSERT(isOutputGood);
-
-    if (!isInputGood || !isOutputGood) {
-        if (outputBus)
-            outputBus->zero();
-        return;
-    }
-
-    // This code only runs as long as the context is alive and after database has been loaded.
-    HRTFDatabase* database = HRTFDatabaseLoader::defaultHRTFDatabase();    
-    ASSERT(database);
-    if (!database) {
-        outputBus->zero();
-        return;
-    }
-
-    // IRCAM HRTF azimuths values from the loaded database is reversed from the panner's notion of azimuth.
-    double azimuth = -desiredAzimuth;
-
-    bool isAzimuthGood = azimuth >= -180.0 && azimuth <= 180.0;
-    ASSERT(isAzimuthGood);
-    if (!isAzimuthGood) {
-        outputBus->zero();
-        return;
-    }
-
-    // Normally, we'll just be dealing with mono sources.
-    // If we have a stereo input, implement stereo panning with left source processed by left HRTF, and right source by right HRTF.
-    AudioChannel* inputChannelL = inputBus->channelByType(AudioBus::ChannelLeft);
-    AudioChannel* inputChannelR = numInputChannels > 1 ? inputBus->channelByType(AudioBus::ChannelRight) : 0;
-
-    // Get source and destination pointers.
-    float* sourceL = inputChannelL->data();
-    float* sourceR = numInputChannels > 1 ? inputChannelR->data() : sourceL;
-    float* destinationL = outputBus->channelByType(AudioBus::ChannelLeft)->data();
-    float* destinationR = outputBus->channelByType(AudioBus::ChannelRight)->data();
-
-    double azimuthBlend;
-    int desiredAzimuthIndex = calculateDesiredAzimuthIndexAndBlend(azimuth, azimuthBlend);
-
-    // This algorithm currently requires that we process in power-of-two size chunks at least 128.
-    ASSERT(1UL << static_cast<int>(log2(framesToProcess)) == framesToProcess);
-    ASSERT(framesToProcess >= 128);
-    
-    const unsigned framesPerSegment = 128;
-    const unsigned numberOfSegments = framesToProcess / framesPerSegment;
-
-    for (unsigned segment = 0; segment < numberOfSegments; ++segment) {
-        if (m_isFirstRender) {
-            // Snap exactly to desired position (first time and after reset()).
-            m_azimuthIndex = desiredAzimuthIndex;
-            m_isFirstRender = false;
-        } else {
-            // Each segment renders with an azimuth index closer by one to the desired azimuth index.
-            // Because inter-aural time delay is mostly a factor of azimuth and the delay is where the clicks and graininess come from,
-            // we don't bother smoothing the elevations.
-            int numberOfAzimuths = database->numberOfAzimuths();
-            bool wrap = wrapDistance(m_azimuthIndex, desiredAzimuthIndex, numberOfAzimuths);
-            if (wrap) {
-                if (m_azimuthIndex < desiredAzimuthIndex)
-                    m_azimuthIndex = (m_azimuthIndex - 1 + numberOfAzimuths) % numberOfAzimuths;
-                else if (m_azimuthIndex > desiredAzimuthIndex)
-                    m_azimuthIndex = (m_azimuthIndex + 1) % numberOfAzimuths;
-            } else {
-                if (m_azimuthIndex < desiredAzimuthIndex)
-                    m_azimuthIndex = (m_azimuthIndex + 1) % numberOfAzimuths;
-                else if (m_azimuthIndex > desiredAzimuthIndex)
-                    m_azimuthIndex = (m_azimuthIndex - 1 + numberOfAzimuths) % numberOfAzimuths;
-            }
-        }
-        
-        // Get the HRTFKernels and interpolated delays.    
-        HRTFKernel* kernelL;
-        HRTFKernel* kernelR;
-        double frameDelayL;
-        double frameDelayR;
-        database->getKernelsFromAzimuthElevation(azimuthBlend, m_azimuthIndex, elevation, kernelL, kernelR, frameDelayL, frameDelayR);
-
-        ASSERT(kernelL && kernelR);
-        if (!kernelL || !kernelR) {
-            outputBus->zero();
-            return;
-        }
-        
-        ASSERT(frameDelayL / sampleRate() < MaxDelayTimeSeconds && frameDelayR / sampleRate() < MaxDelayTimeSeconds);
-            
-        // Calculate the source and destination pointers for the current segment.
-        unsigned offset = segment * framesPerSegment;
-        float* segmentSourceL = sourceL + offset;
-        float* segmentSourceR = sourceR + offset;
-        float* segmentDestinationL = destinationL + offset;
-        float* segmentDestinationR = destinationR + offset;
-
-        // First run through delay lines for inter-aural time difference.
-        m_delayLineL.setDelayFrames(frameDelayL);
-        m_delayLineR.setDelayFrames(frameDelayR);
-        m_delayLineL.process(segmentSourceL, segmentDestinationL, framesPerSegment);
-        m_delayLineR.process(segmentSourceR, segmentDestinationR, framesPerSegment);
-
-        // Now do the convolutions in-place.
-        m_convolverL.process(kernelL->fftFrame(), segmentDestinationL, segmentDestinationL, framesPerSegment);
-        m_convolverR.process(kernelR->fftFrame(), segmentDestinationR, segmentDestinationR, framesPerSegment);
-    }
-}
-
-} // namespace WebCore
-
-#endif // ENABLE(WEB_AUDIO)