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<dt>Revision</dt> <dd><a href="http://trac.webkit.org/projects/webkit/changeset/212928">212928</a></dd>
<dt>Author</dt> <dd>commit-queue@webkit.org</dd>
<dt>Date</dt> <dd>2017-02-23 15:05:45 -0800 (Thu, 23 Feb 2017)</dd>
</dl>
<h3>Log Message</h3>
<pre>[WebRTC] RealtimeOutgoingAudioSource does not need to upsample audio buffers
https://bugs.webkit.org/show_bug.cgi?id=168796
Patch by Youenn Fablet <youenn@apple.com> on 2017-02-23
Reviewed by Jer Noble.
Covered by manual testing.
Limiting RealtimeOutgoingAudioSource conversion to interleaving and float-to-integer.
Removed the sample rate conversion.
* platform/mediastream/mac/RealtimeOutgoingAudioSource.cpp:
(WebCore::libwebrtcAudioFormat):
(WebCore::RealtimeOutgoingAudioSource::audioSamplesAvailable):
(WebCore::RealtimeOutgoingAudioSource::pullAudioData):</pre>
<h3>Modified Paths</h3>
<ul>
<li><a href="#trunkSourceWebCoreChangeLog">trunk/Source/WebCore/ChangeLog</a></li>
<li><a href="#trunkSourceWebCoreplatformmediastreammacRealtimeOutgoingAudioSourcecpp">trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.cpp</a></li>
</ul>
</div>
<div id="patch">
<h3>Diff</h3>
<a id="trunkSourceWebCoreChangeLog"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/ChangeLog (212927 => 212928)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/ChangeLog 2017-02-23 22:52:55 UTC (rev 212927)
+++ trunk/Source/WebCore/ChangeLog 2017-02-23 23:05:45 UTC (rev 212928)
</span><span class="lines">@@ -1,3 +1,19 @@
</span><ins>+2017-02-23 Youenn Fablet <youenn@apple.com>
+
+ [WebRTC] RealtimeOutgoingAudioSource does not need to upsample audio buffers
+ https://bugs.webkit.org/show_bug.cgi?id=168796
+
+ Reviewed by Jer Noble.
+
+ Covered by manual testing.
+ Limiting RealtimeOutgoingAudioSource conversion to interleaving and float-to-integer.
+ Removed the sample rate conversion.
+
+ * platform/mediastream/mac/RealtimeOutgoingAudioSource.cpp:
+ (WebCore::libwebrtcAudioFormat):
+ (WebCore::RealtimeOutgoingAudioSource::audioSamplesAvailable):
+ (WebCore::RealtimeOutgoingAudioSource::pullAudioData):
+
</ins><span class="cx"> 2017-02-23 Alex Christensen <achristensen@webkit.org>
</span><span class="cx">
</span><span class="cx"> Re-soft-link CoreVideo after r212906
</span></span></pre></div>
<a id="trunkSourceWebCoreplatformmediastreammacRealtimeOutgoingAudioSourcecpp"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.cpp (212927 => 212928)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.cpp 2017-02-23 22:52:55 UTC (rev 212927)
+++ trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.cpp 2017-02-23 23:05:45 UTC (rev 212928)
</span><span class="lines">@@ -37,10 +37,10 @@
</span><span class="cx">
</span><span class="cx"> namespace WebCore {
</span><span class="cx">
</span><del>-static inline AudioStreamBasicDescription libwebrtcAudioFormat(size_t channelCount)
</del><ins>+static inline AudioStreamBasicDescription libwebrtcAudioFormat(Float64 sampleRate, size_t channelCount)
</ins><span class="cx"> {
</span><span class="cx"> AudioStreamBasicDescription streamFormat;
</span><del>- FillOutASBDForLPCM(streamFormat, LibWebRTCAudioFormat::sampleRate, channelCount, LibWebRTCAudioFormat::sampleSize, LibWebRTCAudioFormat::sampleSize, LibWebRTCAudioFormat::isFloat, LibWebRTCAudioFormat::isBigEndian, LibWebRTCAudioFormat::isNonInterleaved);
</del><ins>+ FillOutASBDForLPCM(streamFormat, sampleRate, channelCount, LibWebRTCAudioFormat::sampleSize, LibWebRTCAudioFormat::sampleSize, LibWebRTCAudioFormat::isFloat, LibWebRTCAudioFormat::isBigEndian, LibWebRTCAudioFormat::isNonInterleaved);
</ins><span class="cx"> return streamFormat;
</span><span class="cx"> }
</span><span class="cx">
</span><span class="lines">@@ -65,12 +65,14 @@
</span><span class="cx">
</span><span class="cx"> void RealtimeOutgoingAudioSource::audioSamplesAvailable(const MediaTime& time, const PlatformAudioData& audioData, const AudioStreamDescription& streamDescription, size_t sampleCount)
</span><span class="cx"> {
</span><ins>+ ASSERT(streamDescription.numberOfChannels() <= 2);
+
</ins><span class="cx"> if (m_inputStreamDescription != streamDescription) {
</span><span class="cx"> m_inputStreamDescription = toCAAudioStreamDescription(streamDescription);
</span><span class="cx"> auto status = m_sampleConverter->setInputFormat(m_inputStreamDescription);
</span><span class="cx"> ASSERT_UNUSED(status, !status);
</span><span class="cx">
</span><del>- status = m_sampleConverter->setOutputFormat(libwebrtcAudioFormat(streamDescription.numberOfChannels()));
</del><ins>+ status = m_sampleConverter->setOutputFormat(libwebrtcAudioFormat(streamDescription.sampleRate(), streamDescription.numberOfChannels()));
</ins><span class="cx"> ASSERT(!status);
</span><span class="cx"> }
</span><span class="cx"> m_sampleConverter->pushSamples(time, audioData, sampleCount);
</span><span class="lines">@@ -82,7 +84,9 @@
</span><span class="cx">
</span><span class="cx"> void RealtimeOutgoingAudioSource::pullAudioData()
</span><span class="cx"> {
</span><del>- size_t bufferSize = LibWebRTCAudioFormat::chunkSampleCount * LibWebRTCAudioFormat::sampleByteSize * m_inputStreamDescription.numberOfChannels();
</del><ins>+ // libwebrtc expects 10 ms chunks.
+ size_t chunkSampleCount = m_inputStreamDescription.sampleRate() / 100;
+ size_t bufferSize = chunkSampleCount * LibWebRTCAudioFormat::sampleByteSize * m_inputStreamDescription.numberOfChannels();
</ins><span class="cx"> m_audioBuffer.reserveCapacity(bufferSize);
</span><span class="cx">
</span><span class="cx"> AudioBufferList bufferList;
</span><span class="lines">@@ -91,10 +95,10 @@
</span><span class="cx"> bufferList.mBuffers[0].mDataByteSize = bufferSize;
</span><span class="cx"> bufferList.mBuffers[0].mData = m_audioBuffer.data();
</span><span class="cx">
</span><del>- m_sampleConverter->pullAvalaibleSamplesAsChunks(bufferList, LibWebRTCAudioFormat::chunkSampleCount, m_startFrame, [this] {
- m_startFrame += LibWebRTCAudioFormat::chunkSampleCount;
</del><ins>+ m_sampleConverter->pullAvalaibleSamplesAsChunks(bufferList, chunkSampleCount, m_startFrame, [this, chunkSampleCount] {
+ m_startFrame += chunkSampleCount;
</ins><span class="cx"> for (auto sink : m_sinks)
</span><del>- sink->OnData(m_audioBuffer.data(), LibWebRTCAudioFormat::sampleSize, LibWebRTCAudioFormat::sampleRate, m_inputStreamDescription.numberOfChannels(), LibWebRTCAudioFormat::chunkSampleCount);
</del><ins>+ sink->OnData(m_audioBuffer.data(), LibWebRTCAudioFormat::sampleSize, m_inputStreamDescription.sampleRate(), m_inputStreamDescription.numberOfChannels(), chunkSampleCount);
</ins><span class="cx"> });
</span><span class="cx"> }
</span><span class="cx">
</span></span></pre>
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