Attachment 272280 Details for Bug 154538 – Patch

[patch] Patch

bug-154538-20160225193212.patch (text/plain), 7.87 KB, created by Jer Noble on 2016-02-25 19:32:29 PST

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Creator: Jer Noble

Created: 2016-02-25 19:32:29 PST

Size: 7.87 KB

patch

obsolete

>Subversion Revision: 196961
>diff --git a/Source/WebCore/ChangeLog b/Source/WebCore/ChangeLog
>index 218b3e6e50b7060922c8607252662c20b6e82689..b99ef8d222eb89fec44e738bd90a8fadbb974a71 100644
>--- a/Source/WebCore/ChangeLog
>+++ b/Source/WebCore/ChangeLog
>@@ -1,3 +1,27 @@
>+2016-02-25  Jer Noble  <jer.noble@apple.com>
>+
>+        Web Audio becomes distorted after sample rate changes
>+        https://bugs.webkit.org/show_bug.cgi?id=154538
>+        <rdar://problem/24771292>
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        When the underlying audio hardware sample rate changes, the AudioUnit render callback will begin asking
>+        for fewer or more frames. For example, when the sample rate goes from 44.1kHz to 48kHz, it will ask for
>+        118 samples instead of 128. (And vice-versa, 140 samples instead of 128.) But the Web Audio engine can only
>+        really handle requests in multiples of 128 samples. In the case where there are requests for < 128 samples,
>+        actually render 128, but save off the unrequested samples in a separate bus. Then fill that bus during the
>+        next request.
>+
>+        * platform/audio/AudioBus.cpp:
>+        (WebCore::AudioBus::copyFromRange): Added utility method.
>+        * platform/audio/AudioBus.h:
>+        * platform/audio/ios/AudioDestinationIOS.cpp:
>+        (WebCore::AudioDestinationIOS::AudioDestinationIOS): Create a "spare" bus.
>+        (WebCore::assignAudioBuffersToBus): Moved from inside render.
>+        (WebCore::AudioDestinationIOS::render): Save off extra samples to the "spare" bus.
>+        * platform/audio/ios/AudioDestinationIOS.h:
>+
> 2016-02-19  Jer Noble  <jer.noble@apple.com>
> 
>         Adopt CachedRawResourceClient::shouldCacheResponse() in MediaResourceLoader and WebCoreNSURLSession
>diff --git a/Source/WebCore/platform/audio/AudioBus.cpp b/Source/WebCore/platform/audio/AudioBus.cpp
>index faa96170a8c410df8f2dc8e8caf48354760f3b4c..ec02ac48771f860384512e6e782fc2940b9ad6db 100644
>--- a/Source/WebCore/platform/audio/AudioBus.cpp
>+++ b/Source/WebCore/platform/audio/AudioBus.cpp
>@@ -213,6 +213,33 @@ void AudioBus::scale(float scale)
>         channel(i)->scale(scale);
> }
> 
>+void AudioBus::copyFromRange(const AudioBus& sourceBus, unsigned startFrame, unsigned endFrame)
>+{
>+    if (!topologyMatches(sourceBus)) {
>+        ASSERT_NOT_REACHED();
>+        zero();
>+        return;
>+    }
>+
>+    size_t numberOfSourceFrames = sourceBus.length();
>+    bool isRangeSafe = startFrame < endFrame && endFrame <= numberOfSourceFrames;
>+    ASSERT(isRangeSafe);
>+    if (!isRangeSafe) {
>+        zero();
>+        return;
>+    }
>+
>+    unsigned numberOfChannels = this->numberOfChannels();
>+    ASSERT(numberOfChannels <= MaxBusChannels);
>+    if (numberOfChannels > MaxBusChannels) {
>+        zero();
>+        return;
>+    }
>+
>+    for (unsigned i = 0; i < numberOfChannels; ++i)
>+        channel(i)->copyFromRange(sourceBus.channel(i), startFrame, endFrame);
>+}
>+
> void AudioBus::copyFrom(const AudioBus& sourceBus, ChannelInterpretation channelInterpretation)
> {
>     if (&sourceBus == this)
>diff --git a/Source/WebCore/platform/audio/AudioBus.h b/Source/WebCore/platform/audio/AudioBus.h
>index 8d6839f012110dbfd7c0f501b7d0433e4340640e..4f9b5f55cc843f4ab87de355439f86d9786d16ea 100644
>--- a/Source/WebCore/platform/audio/AudioBus.h
>+++ b/Source/WebCore/platform/audio/AudioBus.h
>@@ -122,6 +122,9 @@ public:
>     void reset() { m_isFirstTime = true; } // for de-zippering
> 
>     // Copies the samples from the source bus to this one.
>+    void copyFromRange(const AudioBus& sourceBus, unsigned startFrame, unsigned endFrame);
>+
>+    // Copies the samples from the source bus to this one.
>     // This is just a simple per-channel copy if the number of channels match, otherwise an up-mix or down-mix is done.
>     void copyFrom(const AudioBus& sourceBus, ChannelInterpretation = Speakers);
> 
>diff --git a/Source/WebCore/platform/audio/ios/AudioDestinationIOS.cpp b/Source/WebCore/platform/audio/ios/AudioDestinationIOS.cpp
>index addb524f017a17ea91211235993220d8c9b1419c..76ce942f1ca632d4e9a6123904778a1ebe585552 100644
>--- a/Source/WebCore/platform/audio/ios/AudioDestinationIOS.cpp
>+++ b/Source/WebCore/platform/audio/ios/AudioDestinationIOS.cpp
>@@ -101,6 +101,7 @@ AudioDestinationIOS::AudioDestinationIOS(AudioIOCallback& callback, double sampl
>     : m_outputUnit(0)
>     , m_callback(callback)
>     , m_renderBus(AudioBus::create(2, kRenderBufferSize, false))
>+    , m_spareBus(AudioBus::create(2, kRenderBufferSize, true))
>     , m_sampleRate(sampleRate)
>     , m_isPlaying(false)
> {
>@@ -199,19 +200,46 @@ void AudioDestinationIOS::stop()
>         setIsPlaying(false);
> }
> 
>+static void assignAudioBuffersToBus(AudioBuffer* buffers, AudioBus* bus, UInt32 numberOfBuffers, UInt32 numberOfFrames, UInt32 frameOffset, UInt32 framesThisTime)
>+{
>+    for (UInt32 i = 0; i < numberOfBuffers; ++i) {
>+        UInt32 bytesPerFrame = buffers[i].mDataByteSize / numberOfFrames;
>+        UInt32 byteOffset = frameOffset * bytesPerFrame;
>+        float* memory = (float*)((char*)buffers[i].mData + byteOffset);
>+        bus->setChannelMemory(i, memory, framesThisTime);
>+    }
>+}
>+
> // Pulls on our provider to get rendered audio stream.
> OSStatus AudioDestinationIOS::render(UInt32 numberOfFrames, AudioBufferList* ioData)
> {
>     AudioBuffer* buffers = ioData->mBuffers;
>-    for (UInt32 frameOffset = 0; frameOffset + kRenderBufferSize <= numberOfFrames; frameOffset += kRenderBufferSize) {
>-        UInt32 remainingFrames = std::min<UInt32>(kRenderBufferSize, numberOfFrames - frameOffset);
>-        for (UInt32 i = 0; i < ioData->mNumberBuffers; ++i) {
>-            UInt32 bytesPerFrame = buffers[i].mDataByteSize / numberOfFrames;
>-            UInt32 byteOffset = frameOffset * bytesPerFrame;
>-            float* memory = (float*)((char*)buffers[i].mData + byteOffset);
>-            m_renderBus->setChannelMemory(i, memory, remainingFrames);
>+    UInt32 numberOfBuffers = ioData->mNumberBuffers;
>+    UInt32 framesRemaining = numberOfFrames;
>+    UInt32 frameOffset = 0;
>+    while (framesRemaining > 0) {
>+        if (m_firstSpareFrame && m_lastSpareFrame) {
>+            ASSERT(m_firstSpareFrame < m_lastSpareFrame);
>+            UInt32 framesThisTime = m_lastSpareFrame - m_firstSpareFrame;
>+            assignAudioBuffersToBus(buffers, m_renderBus.get(), numberOfBuffers, numberOfFrames, frameOffset, framesThisTime);
>+            m_renderBus->copyFromRange(*m_spareBus, m_firstSpareFrame, m_lastSpareFrame);
>+            frameOffset += framesThisTime;
>+            framesRemaining -= framesThisTime;
>+            m_firstSpareFrame = m_lastSpareFrame = 0;
>         }
>-        m_callback.render(0, m_renderBus.get(), remainingFrames);
>+
>+        UInt32 framesThisTime = std::min<UInt32>(kRenderBufferSize, framesRemaining);
>+        assignAudioBuffersToBus(buffers, m_renderBus.get(), numberOfBuffers, numberOfFrames, frameOffset, framesThisTime);
>+
>+        if (framesThisTime < kRenderBufferSize) {
>+            m_callback.render(0, m_spareBus.get(), kRenderBufferSize);
>+            m_renderBus->copyFromRange(*m_spareBus, 0, framesThisTime);
>+            m_firstSpareFrame = framesThisTime;
>+            m_lastSpareFrame = kRenderBufferSize - 1;
>+        } else
>+            m_callback.render(0, m_renderBus.get(), framesThisTime);
>+        frameOffset += framesThisTime;
>+        framesRemaining -= framesThisTime;
>     }
> 
>     return noErr;
>diff --git a/Source/WebCore/platform/audio/ios/AudioDestinationIOS.h b/Source/WebCore/platform/audio/ios/AudioDestinationIOS.h
>index 812487a33ff0a8c63aea4a0cc7d4789676cfe3a3..1a4598ecb375b11ba2ee11cb6a5ffa58c73ba705 100644
>--- a/Source/WebCore/platform/audio/ios/AudioDestinationIOS.h
>+++ b/Source/WebCore/platform/audio/ios/AudioDestinationIOS.h
>@@ -65,6 +65,9 @@ private:
>     AudioUnit m_outputUnit;
>     AudioIOCallback& m_callback;
>     RefPtr<AudioBus> m_renderBus;
>+    RefPtr<AudioBus> m_spareBus;
>+    unsigned m_firstSpareFrame { 0 };
>+    unsigned m_lastSpareFrame { 0 };
> 
>     double m_sampleRate;
>     bool m_isPlaying;

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darin: review+

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