Changeset 285985 in webkit

Timestamp:

Nov 17, 2021 11:53:25 PM (8 months ago)

Author:

youenn@apple.com

Message:

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):

Location:

trunk/Source/WebCore

Files:

• ChangeLog (modified) (1 diff)
• platform/mediastream/cocoa/AudioMediaStreamTrackRendererInternalUnit.cpp (modified) (3 diffs)

trunk/Source/WebCore/ChangeLog

@@ +1 @@
+2021-11-17  Youenn Fablet  <youenn@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  <youenn@apple.com>
 

trunk/Source/WebCore/platform/mediastream/cocoa/AudioMediaStreamTrackRendererInternalUnit.cpp

@@ -163 +163 @@
         return;
 
-    CAAudioStreamDescription outputDescription;
     AudioComponentInstance remoteIOUnit { nullptr };
 
@@ -212 +211 @@
     }
 
-    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();
-
-    error = PAL::AudioUnitSetProperty(remoteIOUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &outputDescription.streamDescription(), sizeof(outputDescription.streamDescription()));
+    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);
+    }
+    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);
@@ -233 +235 @@
     }
 
-    m_outputDescription = makeUnique<CAAudioStreamDescription>(outputDescription);
     m_remoteIOUnit = remoteIOUnit;
 }