Merge WebKit at r75315: Initial merge by git.

Change-Id: I570314b346ce101c935ed22a626b48c2af266b84

1 files changed, 0 insertions, 365 deletions

diff --git a/WebCore/platform/audio/AudioBus.cpp b/WebCore/platform/audio/AudioBus.cpp
deleted file mode 100644
index dd4746d..0000000
--- a/WebCore/platform/audio/AudioBus.cpp
+++ /dev/null
@@ -1,365 +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.
- * 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 "config.h"
-
-#if ENABLE(WEB_AUDIO)
-
-#include "AudioBus.h"
-
-#include "VectorMath.h"
-#include <algorithm>
-#include <assert.h>
-#include <math.h>
-#include <wtf/OwnPtr.h>
-#include <wtf/PassOwnPtr.h>
-
-namespace WebCore {
-
-using namespace VectorMath;
-    
-AudioBus::AudioBus(unsigned numberOfChannels, size_t length, bool allocate)
-    : m_length(length)
-    , m_busGain(1.0)
-    , m_isFirstTime(true)
-    , m_sampleRate(0.0)
-{
-    m_channels.reserveInitialCapacity(numberOfChannels);
-
-    for (unsigned i = 0; i < numberOfChannels; ++i) {
-        PassOwnPtr<AudioChannel> channel = allocate ? adoptPtr(new AudioChannel(length)) : adoptPtr(new AudioChannel(0, length));
-        m_channels.append(channel);
-    }
-
-    m_layout = LayoutCanonical; // for now this is the only layout we define
-}
-
-void AudioBus::setChannelMemory(unsigned channelIndex, float* storage, size_t length)
-{
-    if (channelIndex < m_channels.size()) {
-        channel(channelIndex)->set(storage, length);
-        m_length = length; // FIXME: verify that this length matches all the other channel lengths
-    }
-}
-
-void AudioBus::zero()
-{
-    for (unsigned i = 0; i < m_channels.size(); ++i)
-        m_channels[i]->zero();
-}
-
-AudioChannel* AudioBus::channelByType(unsigned channelType)
-{
-    // For now we only support canonical channel layouts...
-    if (m_layout != LayoutCanonical)
-        return 0;
-
-    switch (numberOfChannels()) {
-    case 1: // mono
-        if (channelType == ChannelMono || channelType == ChannelLeft)
-            return channel(0);
-        return 0;
-
-    case 2: // stereo
-        switch (channelType) {
-        case ChannelLeft: return channel(0);
-        case ChannelRight: return channel(1);
-        default: return 0;
-        }
-
-    case 4: // quad
-        switch (channelType) {
-        case ChannelLeft: return channel(0);
-        case ChannelRight: return channel(1);
-        case ChannelSurroundLeft: return channel(2);
-        case ChannelSurroundRight: return channel(3);
-        default: return 0;
-        }
-
-    case 5: // 5.0
-        switch (channelType) {
-        case ChannelLeft: return channel(0);
-        case ChannelRight: return channel(1);
-        case ChannelCenter: return channel(2);
-        case ChannelSurroundLeft: return channel(3);
-        case ChannelSurroundRight: return channel(4);
-        default: return 0;
-        }
-
-    case 6: // 5.1
-        switch (channelType) {
-        case ChannelLeft: return channel(0);
-        case ChannelRight: return channel(1);
-        case ChannelCenter: return channel(2);
-        case ChannelLFE: return channel(3);
-        case ChannelSurroundLeft: return channel(4);
-        case ChannelSurroundRight: return channel(5);
-        default: return 0;
-        }
-    }
-    
-    ASSERT_NOT_REACHED();
-    return 0;
-}
-
-// Returns true if the channel count and frame-size match.
-bool AudioBus::topologyMatches(const AudioBus& bus) const
-{
-    if (numberOfChannels() != bus.numberOfChannels())
-        return false; // channel mismatch
-
-    // Make sure source bus has enough frames.
-    if (length() > bus.length())
-        return false; // frame-size mismatch
-
-    return true;
-}
-
-PassOwnPtr<AudioBus> AudioBus::createBufferFromRange(AudioBus* sourceBuffer, unsigned startFrame, unsigned endFrame)
-{
-    size_t numberOfSourceFrames = sourceBuffer->length();
-    unsigned numberOfChannels = sourceBuffer->numberOfChannels();
-
-    // Sanity checking
-    bool isRangeSafe = startFrame < endFrame && endFrame <= numberOfSourceFrames;
-    ASSERT(isRangeSafe);
-    if (!isRangeSafe)
-        return 0;
-
-    size_t rangeLength = endFrame - startFrame;
-
-    OwnPtr<AudioBus> audioBus = adoptPtr(new AudioBus(numberOfChannels, rangeLength));
-    audioBus->setSampleRate(sourceBuffer->sampleRate());
-
-    for (unsigned i = 0; i < numberOfChannels; ++i)
-        audioBus->channel(i)->copyFromRange(sourceBuffer->channel(i), startFrame, endFrame);
-
-    return audioBus.release();
-}
-
-float AudioBus::maxAbsValue() const
-{
-    float max = 0.0f;
-    for (unsigned i = 0; i < numberOfChannels(); ++i) {
-        const AudioChannel* channel = this->channel(i);
-        max = std::max(max, channel->maxAbsValue());
-    }
-
-    return max;
-}
-
-void AudioBus::normalize()
-{
-    float max = maxAbsValue();
-    if (max)
-        scale(1.0f / max);
-}
-
-void AudioBus::scale(double scale)
-{
-    for (unsigned i = 0; i < numberOfChannels(); ++i)
-        channel(i)->scale(scale);
-}
-
-// Just copies the samples from the source bus to this one.
-// This is just a simple copy if the number of channels match, otherwise a mixup or mixdown is done.
-// For now, we just support a mixup from mono -> stereo.
-void AudioBus::copyFrom(const AudioBus& sourceBus)
-{
-    if (&sourceBus == this)
-        return;
-
-    if (numberOfChannels() == sourceBus.numberOfChannels()) {
-        for (unsigned i = 0; i < numberOfChannels(); ++i)
-            channel(i)->copyFrom(sourceBus.channel(i));
-    } else if (numberOfChannels() == 2 && sourceBus.numberOfChannels() == 1) {
-        // Handle mono -> stereo case (for now simply copy mono channel into both left and right)
-        // FIXME: Really we should apply an equal-power scaling factor here, since we're effectively panning center...
-        const AudioChannel* sourceChannel = sourceBus.channel(0);
-        channel(0)->copyFrom(sourceChannel);
-        channel(1)->copyFrom(sourceChannel);
-    } else {
-        // Case not handled
-        ASSERT_NOT_REACHED();
-    }
-}
-
-void AudioBus::sumFrom(const AudioBus &sourceBus)
-{
-    if (numberOfChannels() == sourceBus.numberOfChannels()) {
-        for (unsigned i = 0; i < numberOfChannels(); ++i)
-            channel(i)->sumFrom(sourceBus.channel(i));
-    } else if (numberOfChannels() == 2 && sourceBus.numberOfChannels() == 1) {
-        // Handle mono -> stereo case (for now simply sum mono channel into both left and right)
-        // FIXME: Really we should apply an equal-power scaling factor here, since we're effectively panning center...
-        const AudioChannel* sourceChannel = sourceBus.channel(0);
-        channel(0)->sumFrom(sourceChannel);
-        channel(1)->sumFrom(sourceChannel);
-    } else {
-        // Case not handled
-        ASSERT_NOT_REACHED();
-    }
-}
-
-void AudioBus::processWithGainFromMonoStereo(const AudioBus &sourceBus, double* lastMixGain, double targetGain, bool sumToBus)
-{
-    // We don't want to suddenly change the gain from mixing one time slice to the next,
-    // so we "de-zipper" by slowly changing the gain each sample-frame until we've achieved the target gain.
-
-    // FIXME: optimize this method (SSE, etc.)
-    // FIXME: Need fast path here when gain has converged on targetGain. In this case, de-zippering is no longer needed.
-    // FIXME: Need fast path when this==sourceBus && lastMixGain==targetGain==1.0 && sumToBus==false (this is a NOP)
-
-    // Take master bus gain into account as well as the targetGain.
-    double totalDesiredGain = m_busGain * targetGain;
-
-    // First time, snap directly to totalDesiredGain.
-    double gain = m_isFirstTime ? totalDesiredGain : *lastMixGain;
-    m_isFirstTime = false;
-
-    int numberOfSourceChannels = sourceBus.numberOfChannels();
-    int numberOfDestinationChannels = numberOfChannels();
-
-    AudioBus& sourceBusSafe = const_cast<AudioBus&>(sourceBus);
-    const float* sourceL = sourceBusSafe.channelByType(ChannelLeft)->data();
-    const float* sourceR = numberOfSourceChannels > 1 ? sourceBusSafe.channelByType(ChannelRight)->data() : 0;
-
-    float* destinationL = channelByType(ChannelLeft)->data();
-    float* destinationR = numberOfDestinationChannels > 1 ? channelByType(ChannelRight)->data() : 0;
-
-    const double DezipperRate = 0.005;
-    int framesToProcess = length();
-
-    if (sumToBus) {
-        // Sum to our bus
-        if (sourceR && destinationR) {
-            // Stereo
-            while (framesToProcess--) {
-                float sampleL = *sourceL++;
-                float sampleR = *sourceR++;
-                *destinationL++ += static_cast<float>(gain * sampleL);
-                *destinationR++ += static_cast<float>(gain * sampleR);
-
-                // Slowly change gain to desired gain.
-                gain += (totalDesiredGain - gain) * DezipperRate;
-            }
-        } else if (destinationR) {
-            // Mono -> stereo (mix equally into L and R)
-            // FIXME: Really we should apply an equal-power scaling factor here, since we're effectively panning center...
-            while (framesToProcess--) {
-                float sample = *sourceL++;
-                *destinationL++ += static_cast<float>(gain * sample);
-                *destinationR++ += static_cast<float>(gain * sample);
-
-                // Slowly change gain to desired gain.
-                gain += (totalDesiredGain - gain) * DezipperRate;
-            }
-        } else {
-            // Mono
-            while (framesToProcess--) {
-                float sampleL = *sourceL++;
-                *destinationL++ += static_cast<float>(gain * sampleL);
-
-                // Slowly change gain to desired gain.
-                gain += (totalDesiredGain - gain) * DezipperRate;
-            }
-        }
-    } else {
-        // Process directly (without summing) to our bus
-        if (sourceR && destinationR) {
-            // Stereo
-            while (framesToProcess--) {
-                float sampleL = *sourceL++;
-                float sampleR = *sourceR++;
-                *destinationL++ = static_cast<float>(gain * sampleL);
-                *destinationR++ = static_cast<float>(gain * sampleR);
-
-                // Slowly change gain to desired gain.
-                gain += (totalDesiredGain - gain) * DezipperRate;
-            }
-        } else if (destinationR) {
-            // Mono -> stereo (mix equally into L and R)
-            // FIXME: Really we should apply an equal-power scaling factor here, since we're effectively panning center...
-            while (framesToProcess--) {
-                float sample = *sourceL++;
-                *destinationL++ = static_cast<float>(gain * sample);
-                *destinationR++ = static_cast<float>(gain * sample);
-
-                // Slowly change gain to desired gain.
-                gain += (totalDesiredGain - gain) * DezipperRate;
-            }
-        } else {
-            // Mono
-            while (framesToProcess--) {
-                float sampleL = *sourceL++;
-                *destinationL++ = static_cast<float>(gain * sampleL);
-
-                // Slowly change gain to desired gain.
-                gain += (totalDesiredGain - gain) * DezipperRate;
-            }
-        }
-    }
-
-    // Save the target gain as the starting point for next time around.
-    *lastMixGain = gain;
-}
-
-void AudioBus::processWithGainFrom(const AudioBus &sourceBus, double* lastMixGain, double targetGain, bool sumToBus)
-{
-    // Make sure we're summing from same type of bus.
-    // We *are* able to sum from mono -> stereo
-    if (sourceBus.numberOfChannels() != 1 && !topologyMatches(sourceBus))
-        return;
-
-    // Dispatch for different channel layouts
-    switch (numberOfChannels()) {
-    case 1: // mono
-    case 2: // stereo
-        processWithGainFromMonoStereo(sourceBus, lastMixGain, targetGain, sumToBus);
-        break;
-    case 4: // FIXME: implement quad
-    case 5: // FIXME: implement 5.0
-    default:
-        ASSERT_NOT_REACHED();
-        break;
-    }
-}
-
-void AudioBus::copyWithGainFrom(const AudioBus &sourceBus, double* lastMixGain, double targetGain)
-{
-    processWithGainFrom(sourceBus, lastMixGain, targetGain, false);
-}
-
-void AudioBus::sumWithGainFrom(const AudioBus &sourceBus, double* lastMixGain, double targetGain)
-{
-    processWithGainFrom(sourceBus, lastMixGain, targetGain, true);
-}
-
-} // WebCore
-
-#endif // ENABLE(WEB_AUDIO)