Changeset 284689 in webkit

Timestamp:

Oct 22, 2021, 9:31:34 AM (5 years ago)

Author:

youenn@apple.com

Message:

REGRESSION (Safari 15 - iOS15): [WebRTC] Increased audio latency while playing webrtc audio stream over audio element
​https://bugs.webkit.org/show_bug.cgi?id=230903
<rdar://problem/83692944>

Reviewed by Eric Carlson.

We used to render audio tracks in process and are now doing rendering in GPU process.
Current implementation requests data with a fixed chunk size, that we were setting as the max of preferred buffer size and web audio chunk size.
If we are reading too close to the end of data in AudioSampleDataSource, we will delay reading the data by this buffer size. This triggers delay but allows getting a consistent rendering.
To reduce delay, we are now using the maximum of web audio chunk size and 10 ms chunk size, as WebRTC tracks are usually manipulating 10 ms chunks.

Manually tested.

• GPUProcess/webrtc/RemoteAudioMediaStreamTrackRendererInternalUnitManager.cpp:

Location:

trunk/Source/WebKit

Files:

• ChangeLog (modified) (1 diff)
• GPUProcess/webrtc/RemoteAudioMediaStreamTrackRendererInternalUnitManager.cpp (modified) (1 diff)

trunk/Source/WebKit/ChangeLog

@@ +1 @@
+2021-10-22  Youenn Fablet  <youenn@apple.com>
+
+        REGRESSION (Safari 15 - iOS15): [WebRTC] Increased audio latency while playing webrtc audio stream over audio element
+        https://bugs.webkit.org/show_bug.cgi?id=230903
+        <rdar://problem/83692944>
+
+        Reviewed by Eric Carlson.
+
+        We used to render audio tracks in process and are now doing rendering in GPU process.
+        Current implementation requests data with a fixed chunk size, that we were setting as the max of preferred buffer size and web audio chunk size.
+        If we are reading too close to the end of data in AudioSampleDataSource, we will delay reading the data by this buffer size. This triggers delay but allows getting a consistent rendering.
+        To reduce delay, we are now using the maximum of web audio chunk size and 10 ms chunk size, as WebRTC tracks are usually manipulating 10 ms chunks.
+
+        Manually tested.
+
+        * GPUProcess/webrtc/RemoteAudioMediaStreamTrackRendererInternalUnitManager.cpp:
+
 2021-10-22  Per Arne Vollan <pvollan@apple.com>
 

trunk/Source/WebKit/GPUProcess/webrtc/RemoteAudioMediaStreamTrackRendererInternalUnitManager.cpp

@@ -143 +143 @@
             return;
         }
-        m_frameChunkSize = std::max(WebCore::AudioUtilities::renderQuantumSize, WebCore::AudioSession::sharedSession().preferredBufferSize());
+        size_t tenMsSampleSize = description->sampleRate() * 10 / 1000;
+        m_frameChunkSize = std::max(WebCore::AudioUtilities::renderQuantumSize, tenMsSampleSize);
         callback(*description, m_frameChunkSize);
     });