Modified: trunk/Source/WebCore/ChangeLog (198034 => 198035)
--- trunk/Source/WebCore/ChangeLog 2016-03-11 19:30:55 UTC (rev 198034)
+++ trunk/Source/WebCore/ChangeLog 2016-03-11 19:35:15 UTC (rev 198035)
@@ -1,3 +1,27 @@
+2016-03-10 Jer Noble <[email protected]>
+
+ Web Audio becomes distorted after sample rate changes
+ https://bugs.webkit.org/show_bug.cgi?id=154538
+ <rdar://problem/24771292>
+
+ Reviewed by Darin Adler.
+
+ 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:
+
2016-03-11 Yusuke Suzuki <[email protected]>
Unreviewed build fix after r198023.
Modified: trunk/Source/WebCore/platform/audio/AudioBus.cpp (198034 => 198035)
--- trunk/Source/WebCore/platform/audio/AudioBus.cpp 2016-03-11 19:30:55 UTC (rev 198034)
+++ trunk/Source/WebCore/platform/audio/AudioBus.cpp 2016-03-11 19:35:15 UTC (rev 198035)
@@ -213,6 +213,33 @@
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)
Modified: trunk/Source/WebCore/platform/audio/AudioBus.h (198034 => 198035)
--- trunk/Source/WebCore/platform/audio/AudioBus.h 2016-03-11 19:30:55 UTC (rev 198034)
+++ trunk/Source/WebCore/platform/audio/AudioBus.h 2016-03-11 19:35:15 UTC (rev 198035)
@@ -122,6 +122,9 @@
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);
Modified: trunk/Source/WebCore/platform/audio/ios/AudioDestinationIOS.cpp (198034 => 198035)
--- trunk/Source/WebCore/platform/audio/ios/AudioDestinationIOS.cpp 2016-03-11 19:30:55 UTC (rev 198034)
+++ trunk/Source/WebCore/platform/audio/ios/AudioDestinationIOS.cpp 2016-03-11 19:35:15 UTC (rev 198035)
@@ -101,6 +101,7 @@
: 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 +200,47 @@
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 = reinterpret_cast<float*>(reinterpret_cast<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, numberOfBuffers, numberOfFrames, frameOffset, framesThisTime);
+ m_renderBus->copyFromRange(*m_spareBus, m_firstSpareFrame, m_lastSpareFrame);
+ frameOffset += framesThisTime;
+ framesRemaining -= framesThisTime;
+ m_firstSpareFrame = 0;
+ m_lastSpareFrame = 0;
}
- m_callback.render(0, m_renderBus.get(), remainingFrames);
+
+ UInt32 framesThisTime = std::min<UInt32>(kRenderBufferSize, framesRemaining);
+ assignAudioBuffersToBus(buffers, *m_renderBus, 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;
Modified: trunk/Source/WebCore/platform/audio/ios/AudioDestinationIOS.h (198034 => 198035)
--- trunk/Source/WebCore/platform/audio/ios/AudioDestinationIOS.h 2016-03-11 19:30:55 UTC (rev 198034)
+++ trunk/Source/WebCore/platform/audio/ios/AudioDestinationIOS.h 2016-03-11 19:35:15 UTC (rev 198035)
@@ -65,6 +65,9 @@
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;
