Merge WebKit at r75315: Initial merge by git.

Change-Id: I570314b346ce101c935ed22a626b48c2af266b84

1 files changed, 365 insertions, 0 deletions

diff --git a/Source/WebCore/platform/audio/AudioBus.cpp b/Source/WebCore/platform/audio/AudioBus.cpp
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+++ b/Source/WebCore/platform/audio/AudioBus.cpp
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+/*
+ * 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)