[webkit-changes] [274808] trunk/Source/WebCore

Mon, 22 Mar 2021 15:10:44 -0700

Title: [274808] trunk/Source/WebCore
Revision
274808
Author
[email protected]
Date
2021-03-22 15:09:55 -0700 (Mon, 22 Mar 2021)

Log Message

Reduce number of heap allocations on the audio thread in AudioSampleDataSource
https://bugs.webkit.org/show_bug.cgi?id=223549

Reviewed by Darin Adler.

Reduce number of heap allocations on the audio thread in AudioSampleDataSource
for performance reasons. I got rid of the heap allocations that I could easily
address. Some trickier ones remain.

* platform/audio/cocoa/AudioSampleDataSource.h:
(WebCore::AudioSampleDataSource::inputDescription const):
- Use Optional<> type for m_inputDescription & m_outputDescription data members instead of
std::unique_ptr<> so that we can initialize them without heap allocation.
- Use UniqueRef<> instead of std::unique_ptr<> for m_ringBuffer now that it gets
  initialized in the constructor (on the main thread).

* platform/audio/cocoa/AudioSampleDataSource.mm:
(WebCore::AudioSampleDataSource::AudioSampleDataSource):
Initialize m_ringBuffer in the constructor which runs on the main thread, instead of
doing it lazily later on, on the audio rendering thread.

(WebCore::AudioSampleDataSource::~AudioSampleDataSource):
Stop nulling data members out unnecessarily in the destructor.

(WebCore::AudioSampleDataSource::setInputFormat):
Initialize m_inputDescription without doing a heap allocation.

(WebCore::AudioSampleDataSource::setOutputFormat):
- Initialize m_outputDescription without doing a heap allocation.
- Drop initialization of m_ringBuffer now that it is done eagerly in the constructor.
- Drop RunLoop::main().dispatch() just to do logging since this was causing heap allocations.

(WebCore::AudioSampleDataSource::pushSamplesInternal):
(WebCore::AudioSampleDataSource::pushSamples):
(WebCore::AudioSampleDataSource::pullSamplesInternal):
(WebCore::AudioSampleDataSource::pullAvalaibleSamplesAsChunks):
(WebCore::AudioSampleDataSource::pullSamples):
Drop null checks for m_ringBuffer now that it is initialized eagerly in the constructor
and thus can never be null.

Modified Paths

Diff

Modified: trunk/Source/WebCore/ChangeLog (274807 => 274808)


--- trunk/Source/WebCore/ChangeLog	2021-03-22 22:08:37 UTC (rev 274807)
+++ trunk/Source/WebCore/ChangeLog	2021-03-22 22:09:55 UTC (rev 274808)
@@ -1,3 +1,45 @@
+2021-03-22  Chris Dumez  <[email protected]>
+
+        Reduce number of heap allocations on the audio thread in AudioSampleDataSource
+        https://bugs.webkit.org/show_bug.cgi?id=223549
+
+        Reviewed by Darin Adler.
+
+        Reduce number of heap allocations on the audio thread in AudioSampleDataSource
+        for performance reasons. I got rid of the heap allocations that I could easily
+        address. Some trickier ones remain.
+
+        * platform/audio/cocoa/AudioSampleDataSource.h:
+        (WebCore::AudioSampleDataSource::inputDescription const):
+        - Use Optional<> type for m_inputDescription & m_outputDescription data members instead of
+        std::unique_ptr<> so that we can initialize them without heap allocation.
+        - Use UniqueRef<> instead of std::unique_ptr<> for m_ringBuffer now that it gets
+          initialized in the constructor (on the main thread).
+
+        * platform/audio/cocoa/AudioSampleDataSource.mm:
+        (WebCore::AudioSampleDataSource::AudioSampleDataSource):
+        Initialize m_ringBuffer in the constructor which runs on the main thread, instead of
+        doing it lazily later on, on the audio rendering thread.
+
+        (WebCore::AudioSampleDataSource::~AudioSampleDataSource):
+        Stop nulling data members out unnecessarily in the destructor.
+
+        (WebCore::AudioSampleDataSource::setInputFormat):
+        Initialize m_inputDescription without doing a heap allocation.
+
+        (WebCore::AudioSampleDataSource::setOutputFormat):
+        - Initialize m_outputDescription without doing a heap allocation.
+        - Drop initialization of m_ringBuffer now that it is done eagerly in the constructor.
+        - Drop RunLoop::main().dispatch() just to do logging since this was causing heap allocations.
+
+        (WebCore::AudioSampleDataSource::pushSamplesInternal):
+        (WebCore::AudioSampleDataSource::pushSamples):
+        (WebCore::AudioSampleDataSource::pullSamplesInternal):
+        (WebCore::AudioSampleDataSource::pullAvalaibleSamplesAsChunks):
+        (WebCore::AudioSampleDataSource::pullSamples):
+        Drop null checks for m_ringBuffer now that it is initialized eagerly in the constructor
+        and thus can never be null.
+
 2021-03-22  Devin Rousso  <[email protected]>
 
         Fix incorrect fallback values in `PlaybackSessionModelMediaElement` after r274249

Modified: trunk/Source/WebCore/platform/audio/cocoa/AudioSampleDataSource.h (274807 => 274808)


--- trunk/Source/WebCore/platform/audio/cocoa/AudioSampleDataSource.h	2021-03-22 22:08:37 UTC (rev 274807)
+++ trunk/Source/WebCore/platform/audio/cocoa/AudioSampleDataSource.h	2021-03-22 22:09:55 UTC (rev 274808)
@@ -61,7 +61,7 @@
     bool pullSamples(AudioSampleBufferList&, size_t, uint64_t, double, PullMode);
     bool pullSamples(AudioBufferList&, size_t, uint64_t, double, PullMode);
 
-    bool pullAvalaibleSamplesAsChunks(AudioBufferList&, size_t frameCount, uint64_t timeStamp, Function<void()>&&);
+    bool pullAvailableSamplesAsChunks(AudioBufferList&, size_t frameCount, uint64_t timeStamp, Function<void()>&&);
 
     void setVolume(float volume) { m_volume = volume; }
     float volume() const { return m_volume; }
@@ -69,7 +69,7 @@
     void setMuted(bool muted) { m_muted = muted; }
     bool muted() const { return m_muted; }
 
-    const CAAudioStreamDescription* inputDescription() const { return m_inputDescription.get(); }
+    const CAAudioStreamDescription* inputDescription() const { return m_inputDescription ? &m_inputDescription.value() : nullptr; }
 
 #if !RELEASE_LOG_DISABLED
     const Logger& logger() const final { return m_logger; }
@@ -87,8 +87,8 @@
 
     void pushSamplesInternal(const AudioBufferList&, const MediaTime&, size_t frameCount);
 
-    std::unique_ptr<CAAudioStreamDescription> m_inputDescription;
-    std::unique_ptr<CAAudioStreamDescription> m_outputDescription;
+    Optional<CAAudioStreamDescription> m_inputDescription;
+    Optional<CAAudioStreamDescription> m_outputDescription;
 
     MediaTime hostTime() const;
 
@@ -106,7 +106,7 @@
     AudioConverterRef m_converter;
     RefPtr<AudioSampleBufferList> m_scratchBuffer;
 
-    std::unique_ptr<CARingBuffer> m_ringBuffer;
+    UniqueRef<CARingBuffer> m_ringBuffer;
     size_t m_maximumSampleCount { 0 };
 
     float m_volume { 1.0 };

Modified: trunk/Source/WebCore/platform/audio/cocoa/AudioSampleDataSource.mm (274807 => 274808)


--- trunk/Source/WebCore/platform/audio/cocoa/AudioSampleDataSource.mm	2021-03-22 22:08:37 UTC (rev 274807)
+++ trunk/Source/WebCore/platform/audio/cocoa/AudioSampleDataSource.mm	2021-03-22 22:09:55 UTC (rev 274808)
@@ -52,6 +52,7 @@
 
 AudioSampleDataSource::AudioSampleDataSource(size_t maximumSampleCount, LoggerHelper& loggerHelper)
     : m_inputSampleOffset(MediaTime::invalidTime())
+    , m_ringBuffer(makeUniqueRef<CARingBuffer>())
     , m_maximumSampleCount(maximumSampleCount)
 #if !RELEASE_LOG_DISABLED
     , m_logger(loggerHelper.logger())
@@ -65,13 +66,8 @@
 
 AudioSampleDataSource::~AudioSampleDataSource()
 {
-    m_inputDescription = nullptr;
-    m_outputDescription = nullptr;
-    m_ringBuffer = nullptr;
-    if (m_converter) {
+    if (m_converter)
         AudioConverterDispose(m_converter);
-        m_converter = nullptr;
-    }
 }
 
 OSStatus AudioSampleDataSource::setupConverter()
@@ -101,7 +97,7 @@
 {
     ASSERT(format.sampleRate() >= 0);
 
-    m_inputDescription = makeUnique<CAAudioStreamDescription>(format);
+    m_inputDescription = CAAudioStreamDescription { format };
     if (m_outputDescription)
         return setupConverter();
 
@@ -113,9 +109,7 @@
     ASSERT(m_inputDescription);
     ASSERT(format.sampleRate() >= 0);
 
-    m_outputDescription = makeUnique<CAAudioStreamDescription>(format);
-    if (!m_ringBuffer)
-        m_ringBuffer = makeUnique<CARingBuffer>();
+    m_outputDescription = CAAudioStreamDescription { format };
 
     m_ringBuffer->allocate(format, static_cast<size_t>(m_maximumSampleCount));
     m_scratchBuffer = AudioSampleBufferList::create(m_outputDescription->streamDescription(), m_maximumSampleCount);
@@ -160,12 +154,8 @@
         sampleTime = m_expectedNextPushedSampleTime;
     m_expectedNextPushedSampleTime = sampleTime + MediaTime(sampleCount, sampleTime.timeScale());
 
-    if (m_inputSampleOffset == MediaTime::invalidTime()) {
+    if (m_inputSampleOffset == MediaTime::invalidTime())
         m_inputSampleOffset = MediaTime(1 - sampleTime.timeValue(), sampleTime.timeScale());
-        RunLoop::main().dispatch([logIdentifier = LOGIDENTIFIER, inputSampleOffset = m_inputSampleOffset.timeValue(), maximumSampleCount = m_maximumSampleCount, this, protectedThis = makeRefPtr(*this)] {
-            ALWAYS_LOG(logIdentifier, "input sample offset is ", inputSampleOffset, ", maximumSampleCount is ", maximumSampleCount);
-        });
-    }
     sampleTime += m_inputSampleOffset;
 
 #if !LOG_DISABLED
@@ -181,8 +171,7 @@
 void AudioSampleDataSource::pushSamples(const AudioStreamBasicDescription& sampleDescription, CMSampleBufferRef sampleBuffer)
 {
     ASSERT_UNUSED(sampleDescription, *m_inputDescription == sampleDescription);
-    ASSERT(m_ringBuffer);
-    
+
     WebAudioBufferList list(*m_inputDescription, sampleBuffer);
     pushSamplesInternal(list, PAL::toMediaTime(PAL::CMSampleBufferGetPresentationTimeStamp(sampleBuffer)), PAL::CMSampleBufferGetNumSamples(sampleBuffer));
 }
@@ -218,7 +207,7 @@
         return false;
     }
 
-    if (!m_ringBuffer || m_muted || m_inputSampleOffset == MediaTime::invalidTime()) {
+    if (m_muted || m_inputSampleOffset == MediaTime::invalidTime()) {
         AudioSampleBufferList::zeroABL(buffer, byteCount);
         return false;
     }
@@ -270,7 +259,7 @@
         return true;
     }
 
-    if (m_scratchBuffer->copyFrom(*m_ringBuffer.get(), sampleCount, timeStamp, CARingBuffer::Copy))
+    if (m_scratchBuffer->copyFrom(m_ringBuffer.get(), sampleCount, timeStamp, CARingBuffer::Copy))
         return false;
 
     m_scratchBuffer->applyGain(m_volume);
@@ -281,11 +270,8 @@
     return true;
 }
 
-bool AudioSampleDataSource::pullAvalaibleSamplesAsChunks(AudioBufferList& buffer, size_t sampleCountPerChunk, uint64_t timeStamp, Function<void()>&& consumeFilledBuffer)
+bool AudioSampleDataSource::pullAvailableSamplesAsChunks(AudioBufferList& buffer, size_t sampleCountPerChunk, uint64_t timeStamp, Function<void()>&& consumeFilledBuffer)
 {
-    if (!m_ringBuffer)
-        return false;
-
     ASSERT(buffer.mNumberBuffers == m_ringBuffer->channelCount());
     if (buffer.mNumberBuffers != m_ringBuffer->channelCount())
         return false;
@@ -324,22 +310,11 @@
 
 bool AudioSampleDataSource::pullSamples(AudioBufferList& buffer, size_t sampleCount, uint64_t timeStamp, double hostTime, PullMode mode)
 {
-    if (!m_ringBuffer) {
-        size_t byteCount = sampleCount * m_outputDescription->bytesPerFrame();
-        AudioSampleBufferList::zeroABL(buffer, byteCount);
-        return false;
-    }
-
     return pullSamplesInternal(buffer, sampleCount, timeStamp, hostTime, mode);
 }
 
 bool AudioSampleDataSource::pullSamples(AudioSampleBufferList& buffer, size_t sampleCount, uint64_t timeStamp, double hostTime, PullMode mode)
 {
-    if (!m_ringBuffer) {
-        buffer.zero();
-        return false;
-    }
-
     if (!pullSamplesInternal(buffer.bufferList(), sampleCount, timeStamp, hostTime, mode))
         return false;

Modified: trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSourceCocoa.cpp (274807 => 274808)


--- trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSourceCocoa.cpp	2021-03-22 22:08:37 UTC (rev 274807)
+++ trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSourceCocoa.cpp	2021-03-22 22:09:55 UTC (rev 274808)
@@ -139,7 +139,7 @@
     if (isSilenced() !=  m_sampleConverter->muted())
         m_sampleConverter->setMuted(isSilenced());
 
-    m_sampleConverter->pullAvalaibleSamplesAsChunks(bufferList, chunkSampleCount, m_readCount, [this, chunkSampleCount] {
+    m_sampleConverter->pullAvailableSamplesAsChunks(bufferList, chunkSampleCount, m_readCount, [this, chunkSampleCount] {
         m_readCount += chunkSampleCount;
         sendAudioFrames(m_audioBuffer.data(), LibWebRTCAudioFormat::sampleSize, m_outputStreamDescription.sampleRate(), m_outputStreamDescription.numberOfChannels(), chunkSampleCount);
     });

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