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<dt>Revision</dt> <dd><a href="http://trac.webkit.org/projects/webkit/changeset/212769">212769</a></dd>
<dt>Author</dt> <dd>commit-queue@webkit.org</dd>
<dt>Date</dt> <dd>2017-02-21 15:16:02 -0800 (Tue, 21 Feb 2017)</dd>
</dl>
<h3>Log Message</h3>
<pre>[WebRTC] Implement Incoming libwebrtc audio source support.
https://bugs.webkit.org/show_bug.cgi?id=167961
Patch by Youenn Fablet <youenn@apple.com> on 2017-02-21
Reviewed by Eric Carlson.
Hook libwebrtc incoming audio source into WebCore audio rendering path.
Manually testing that muted sources produce data with zeros and unmuted sources provide data with non zeros.
* platform/mediastream/mac/RealtimeIncomingAudioSource.cpp:
(WebCore::RealtimeIncomingAudioSource::create):
(WebCore::streamDescription):
(WebCore::RealtimeIncomingAudioSource::OnData):
(WebCore::RealtimeIncomingAudioSource::audioSourceProvider):
* platform/mediastream/mac/RealtimeIncomingAudioSource.h:</pre>
<h3>Modified Paths</h3>
<ul>
<li><a href="#trunkSourceWebCoreChangeLog">trunk/Source/WebCore/ChangeLog</a></li>
<li><a href="#trunkSourceWebCoreplatformmediastreammacRealtimeIncomingAudioSourcecpp">trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSource.cpp</a></li>
<li><a href="#trunkSourceWebCoreplatformmediastreammacRealtimeIncomingAudioSourceh">trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSource.h</a></li>
</ul>
</div>
<div id="patch">
<h3>Diff</h3>
<a id="trunkSourceWebCoreChangeLog"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/ChangeLog (212768 => 212769)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/ChangeLog 2017-02-21 23:07:51 UTC (rev 212768)
+++ trunk/Source/WebCore/ChangeLog 2017-02-21 23:16:02 UTC (rev 212769)
</span><span class="lines">@@ -1,3 +1,20 @@
</span><ins>+2017-02-21 Youenn Fablet <youenn@apple.com>
+
+ [WebRTC] Implement Incoming libwebrtc audio source support.
+ https://bugs.webkit.org/show_bug.cgi?id=167961
+
+ Reviewed by Eric Carlson.
+
+ Hook libwebrtc incoming audio source into WebCore audio rendering path.
+ Manually testing that muted sources produce data with zeros and unmuted sources provide data with non zeros.
+
+ * platform/mediastream/mac/RealtimeIncomingAudioSource.cpp:
+ (WebCore::RealtimeIncomingAudioSource::create):
+ (WebCore::streamDescription):
+ (WebCore::RealtimeIncomingAudioSource::OnData):
+ (WebCore::RealtimeIncomingAudioSource::audioSourceProvider):
+ * platform/mediastream/mac/RealtimeIncomingAudioSource.h:
+
</ins><span class="cx"> 2017-02-21 Simon Fraser <simon.fraser@apple.com>
</span><span class="cx">
</span><span class="cx"> Fix ImageBitmap comment to not insert a <canvas>.
</span></span></pre></div>
<a id="trunkSourceWebCoreplatformmediastreammacRealtimeIncomingAudioSourcecpp"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSource.cpp (212768 => 212769)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSource.cpp 2017-02-21 23:07:51 UTC (rev 212768)
+++ trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSource.cpp 2017-02-21 23:16:02 UTC (rev 212769)
</span><span class="lines">@@ -33,7 +33,11 @@
</span><span class="cx">
</span><span class="cx"> #if USE(LIBWEBRTC)
</span><span class="cx">
</span><del>-#include "RealtimeMediaSourceSettings.h"
</del><ins>+#include "AudioStreamDescription.h"
+#include "CAAudioStreamDescription.h"
+#include "LibWebRTCAudioFormat.h"
+#include "MediaTimeAVFoundation.h"
+#include "WebAudioBufferList.h"
</ins><span class="cx"> #include "WebAudioSourceProviderAVFObjC.h"
</span><span class="cx">
</span><span class="cx"> #include "CoreMediaSoftLink.h"
</span><span class="lines">@@ -42,7 +46,9 @@
</span><span class="cx">
</span><span class="cx"> Ref<RealtimeIncomingAudioSource> RealtimeIncomingAudioSource::create(rtc::scoped_refptr<webrtc::AudioTrackInterface>&& audioTrack, String&& audioTrackId)
</span><span class="cx"> {
</span><del>- return adoptRef(*new RealtimeIncomingAudioSource(WTFMove(audioTrack), WTFMove(audioTrackId)));
</del><ins>+ auto source = adoptRef(*new RealtimeIncomingAudioSource(WTFMove(audioTrack), WTFMove(audioTrackId)));
+ source->startProducingData();
+ return source;
</ins><span class="cx"> }
</span><span class="cx">
</span><span class="cx"> RealtimeIncomingAudioSource::RealtimeIncomingAudioSource(rtc::scoped_refptr<webrtc::AudioTrackInterface>&& audioTrack, String&& audioTrackId)
</span><span class="lines">@@ -59,14 +65,39 @@
</span><span class="cx"> }
</span><span class="cx"> }
</span><span class="cx">
</span><ins>+
+static inline AudioStreamBasicDescription streamDescription(size_t sampleRate, size_t channelCount)
+{
+ AudioStreamBasicDescription streamFormat;
+ FillOutASBDForLPCM(streamFormat, sampleRate, channelCount, LibWebRTCAudioFormat::sampleSize, LibWebRTCAudioFormat::sampleSize, LibWebRTCAudioFormat::isFloat, LibWebRTCAudioFormat::isBigEndian, LibWebRTCAudioFormat::isNonInterleaved);
+ return streamFormat;
+}
+
</ins><span class="cx"> void RealtimeIncomingAudioSource::OnData(const void* audioData, int bitsPerSample, int sampleRate, size_t numberOfChannels, size_t numberOfFrames)
</span><span class="cx"> {
</span><del>- // FIXME: Implement this.
- UNUSED_PARAM(audioData);
- UNUSED_PARAM(bitsPerSample);
- UNUSED_PARAM(sampleRate);
- UNUSED_PARAM(numberOfChannels);
- UNUSED_PARAM(numberOfFrames);
</del><ins>+ // We may receive OnData calls with empty sound data (mono, samples equal to zero and sampleRate equal to 16000) when starting the call.
+ // FIXME: For the moment we skip them, we should find a better solution at libwebrtc level to not be called until getting some real data.
+ if (sampleRate == 16000 && numberOfChannels == 1)
+ return;
+
+ ASSERT(bitsPerSample == 16);
+ ASSERT(numberOfChannels == 2);
+ ASSERT(sampleRate == 48000);
+
+ CMTime startTime = CMTimeMake(m_numberOfFrames, sampleRate);
+ auto mediaTime = toMediaTime(startTime);
+ m_numberOfFrames += numberOfFrames;
+
+ m_streamFormat = streamDescription(sampleRate, numberOfChannels);
+
+ WebAudioBufferList audioBufferList { CAAudioStreamDescription(m_streamFormat), WTF::safeCast<uint32_t>(numberOfFrames) };
+ audioBufferList.buffer(0)->mDataByteSize = numberOfChannels * numberOfFrames * bitsPerSample / 8;
+ audioBufferList.buffer(0)->mNumberChannels = numberOfChannels;
+ // FIXME: We should not need to do the extra memory allocation and copy.
+ // Instead, we should be able to directly pass audioData pointer.
+ memcpy(audioBufferList.buffer(0)->mData, audioData, audioBufferList.buffer(0)->mDataByteSize);
+
+ audioSamplesAvailable(mediaTime, audioBufferList, CAAudioStreamDescription(m_streamFormat), numberOfFrames);
</ins><span class="cx"> }
</span><span class="cx">
</span><span class="cx"> void RealtimeIncomingAudioSource::startProducingData()
</span><span class="lines">@@ -107,13 +138,8 @@
</span><span class="cx">
</span><span class="cx"> AudioSourceProvider* RealtimeIncomingAudioSource::audioSourceProvider()
</span><span class="cx"> {
</span><del>- if (!m_audioSourceProvider) {
- m_audioSourceProvider = WebAudioSourceProviderAVFObjC::create(*this);
- const auto* description = CMAudioFormatDescriptionGetStreamBasicDescription(m_formatDescription.get());
- m_audioSourceProvider->prepare(description);
- }
-
- return m_audioSourceProvider.get();
</del><ins>+ // FIXME: Create the audioSourceProvider
+ return nullptr;
</ins><span class="cx"> }
</span><span class="cx">
</span><span class="cx"> } // namespace WebCore
</span></span></pre></div>
<a id="trunkSourceWebCoreplatformmediastreammacRealtimeIncomingAudioSourceh"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSource.h (212768 => 212769)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSource.h 2017-02-21 23:07:51 UTC (rev 212768)
+++ trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSource.h 2017-02-21 23:16:02 UTC (rev 212769)
</span><span class="lines">@@ -77,7 +77,8 @@
</span><span class="cx"> rtc::scoped_refptr<webrtc::AudioTrackInterface> m_audioTrack;
</span><span class="cx">
</span><span class="cx"> RefPtr<WebAudioSourceProviderAVFObjC> m_audioSourceProvider;
</span><del>- RetainPtr<CMFormatDescriptionRef> m_formatDescription;
</del><ins>+ AudioStreamBasicDescription m_streamFormat;
+ uint64_t m_numberOfFrames;
</ins><span class="cx"> };
</span><span class="cx">
</span><span class="cx"> } // namespace WebCore
</span></span></pre>
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