Modified: trunk/Source/WebCore/ChangeLog (285984 => 285985)
--- trunk/Source/WebCore/ChangeLog 2021-11-18 07:46:05 UTC (rev 285984)
+++ trunk/Source/WebCore/ChangeLog 2021-11-18 07:53:25 UTC (rev 285985)
@@ -1,5 +1,24 @@
2021-11-17 Youenn Fablet <you...@apple.com>
+ Audio Rate Gets Messed up With Safari WebRTC and Bluetooth Switching
+ https://bugs.webkit.org/show_bug.cgi?id=232822
+ <rdar://problem/85418545>
+
+ Reviewed by Eric Carlson.
+
+ We usually create the unit, start the unit and stop the unit.
+ After that cycle, if we restart the unit, we expect the configuration to stay the same in WebProcess.
+ But, in LocalAudioMediaStreamTrackRendererInternalUnit::stop, we were disposing the audio unit and recreating it if necesary in LocalAudioMediaStreamTrackRendererInternalUnit::start.
+ If the sample rate changed, the new audio unit would use the new sample rate while WebProcess will use the old sample rate.
+ To prevent this, we now always reuse the same description to initialize the audio unit.
+
+ Manually tested.
+
+ * platform/mediastream/cocoa/AudioMediaStreamTrackRendererInternalUnit.cpp:
+ (WebCore::LocalAudioMediaStreamTrackRendererInternalUnit::createAudioUnitIfNeeded):
+
+2021-11-17 Youenn Fablet <you...@apple.com>
+
Add support for more rvfc metadata
https://bugs.webkit.org/show_bug.cgi?id=233185
Modified: trunk/Source/WebCore/platform/mediastream/cocoa/AudioMediaStreamTrackRendererInternalUnit.cpp (285984 => 285985)
--- trunk/Source/WebCore/platform/mediastream/cocoa/AudioMediaStreamTrackRendererInternalUnit.cpp 2021-11-18 07:46:05 UTC (rev 285984)
+++ trunk/Source/WebCore/platform/mediastream/cocoa/AudioMediaStreamTrackRendererInternalUnit.cpp 2021-11-18 07:53:25 UTC (rev 285985)
@@ -162,7 +162,6 @@
if (m_remoteIOUnit)
return;
- CAAudioStreamDescription outputDescription;
AudioComponentInstance remoteIOUnit { nullptr };
AudioComponentDescription ioUnitDescription { kAudioUnitType_Output, 0, kAudioUnitManufacturer_Apple, 0, 0 };
@@ -211,16 +210,19 @@
return;
}
- UInt32 size = sizeof(outputDescription.streamDescription());
- error = PAL::AudioUnitGetProperty(remoteIOUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &outputDescription.streamDescription(), &size);
- if (error) {
- RELEASE_LOG_ERROR(WebRTC, "AudioMediaStreamTrackRendererInternalUnit::createAudioUnit unable to get input stream format, error = %d", error);
- return;
+ if (!m_outputDescription) {
+ CAAudioStreamDescription outputDescription;
+ UInt32 size = sizeof(outputDescription.streamDescription());
+ error = PAL::AudioUnitGetProperty(remoteIOUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &outputDescription.streamDescription(), &size);
+ if (error) {
+ RELEASE_LOG_ERROR(WebRTC, "AudioMediaStreamTrackRendererInternalUnit::createAudioUnit unable to get input stream format, error = %d", error);
+ return;
+ }
+
+ outputDescription.streamDescription().mSampleRate = AudioSession::sharedSession().sampleRate();
+ m_outputDescription = makeUnique<CAAudioStreamDescription>(outputDescription);
}
-
- outputDescription.streamDescription().mSampleRate = AudioSession::sharedSession().sampleRate();
-
- error = PAL::AudioUnitSetProperty(remoteIOUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &outputDescription.streamDescription(), sizeof(outputDescription.streamDescription()));
+ error = PAL::AudioUnitSetProperty(remoteIOUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &m_outputDescription->streamDescription(), sizeof(m_outputDescription->streamDescription()));
if (error) {
RELEASE_LOG_ERROR(WebRTC, "AudioMediaStreamTrackRendererInternalUnit::createAudioUnit unable to set input stream format, error = %d", error);
return;
@@ -232,7 +234,6 @@
return;
}
- m_outputDescription = makeUnique<CAAudioStreamDescription>(outputDescription);
m_remoteIOUnit = remoteIOUnit;
}
