Modified: trunk/Source/WebCore/ChangeLog (212768 => 212769)
--- 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)
@@ -1,3 +1,20 @@
+2017-02-21 Youenn Fablet <[email protected]>
+
+ [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:
+
2017-02-21 Simon Fraser <[email protected]>
Fix ImageBitmap comment to not insert a <canvas>.
Modified: trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSource.cpp (212768 => 212769)
--- 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)
@@ -33,7 +33,11 @@
#if USE(LIBWEBRTC)
-#include "RealtimeMediaSourceSettings.h"
+#include "AudioStreamDescription.h"
+#include "CAAudioStreamDescription.h"
+#include "LibWebRTCAudioFormat.h"
+#include "MediaTimeAVFoundation.h"
+#include "WebAudioBufferList.h"
#include "WebAudioSourceProviderAVFObjC.h"
#include "CoreMediaSoftLink.h"
@@ -42,7 +46,9 @@
Ref<RealtimeIncomingAudioSource> RealtimeIncomingAudioSource::create(rtc::scoped_refptr<webrtc::AudioTrackInterface>&& audioTrack, String&& audioTrackId)
{
- return adoptRef(*new RealtimeIncomingAudioSource(WTFMove(audioTrack), WTFMove(audioTrackId)));
+ auto source = adoptRef(*new RealtimeIncomingAudioSource(WTFMove(audioTrack), WTFMove(audioTrackId)));
+ source->startProducingData();
+ return source;
}
RealtimeIncomingAudioSource::RealtimeIncomingAudioSource(rtc::scoped_refptr<webrtc::AudioTrackInterface>&& audioTrack, String&& audioTrackId)
@@ -59,14 +65,39 @@
}
}
+
+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;
+}
+
void RealtimeIncomingAudioSource::OnData(const void* audioData, int bitsPerSample, int sampleRate, size_t numberOfChannels, size_t numberOfFrames)
{
- // FIXME: Implement this.
- UNUSED_PARAM(audioData);
- UNUSED_PARAM(bitsPerSample);
- UNUSED_PARAM(sampleRate);
- UNUSED_PARAM(numberOfChannels);
- UNUSED_PARAM(numberOfFrames);
+ // 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);
}
void RealtimeIncomingAudioSource::startProducingData()
@@ -107,13 +138,8 @@
AudioSourceProvider* RealtimeIncomingAudioSource::audioSourceProvider()
{
- if (!m_audioSourceProvider) {
- m_audioSourceProvider = WebAudioSourceProviderAVFObjC::create(*this);
- const auto* description = CMAudioFormatDescriptionGetStreamBasicDescription(m_formatDescription.get());
- m_audioSourceProvider->prepare(description);
- }
-
- return m_audioSourceProvider.get();
+ // FIXME: Create the audioSourceProvider
+ return nullptr;
}
} // namespace WebCore
Modified: trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSource.h (212768 => 212769)
--- 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)
@@ -77,7 +77,8 @@
rtc::scoped_refptr<webrtc::AudioTrackInterface> m_audioTrack;
RefPtr<WebAudioSourceProviderAVFObjC> m_audioSourceProvider;
- RetainPtr<CMFormatDescriptionRef> m_formatDescription;
+ AudioStreamBasicDescription m_streamFormat;
+ uint64_t m_numberOfFrames;
};
} // namespace WebCore
