Changeset 292563 in webkit

Timestamp:

Apr 7, 2022 1:58:35 PM (3 months ago)

Author:

youenn@apple.com

Message:

(Safari 15 - iOS15): Increased audio latency on streaming via webrtc
​https://bugs.webkit.org/show_bug.cgi?id=236363
<rdar://problem/88969850>

Reviewed by Eric Carlson.

On macOS 12.3, the default preferred buffer size is roughly 100 ms.
This is ok for regular audio playback but is not desirable when playing realtime audio.
To reduce the perceived latency, we now reduce the preferred buffer size to 20ms
whenever playing an audio MediaStreamTrack, similarly to when capturing audio.

Manually tested.

• platform/audio/PlatformMediaSession.cpp:
• platform/audio/PlatformMediaSession.h:
• platform/audio/cocoa/MediaSessionManagerCocoa.mm:

Location:

trunk/Source/WebCore

Files:

• ChangeLog (modified) (1 diff)
• platform/audio/PlatformMediaSession.cpp (modified) (1 diff)
• platform/audio/PlatformMediaSession.h (modified) (1 diff)
• platform/audio/cocoa/MediaSessionManagerCocoa.mm (modified) (4 diffs)

trunk/Source/WebCore/ChangeLog

@@ +1 @@
+2022-04-07  Youenn Fablet  <youenn@apple.com>
+
+        (Safari 15 - iOS15):  Increased audio latency on streaming via webrtc
+        https://bugs.webkit.org/show_bug.cgi?id=236363
+        <rdar://problem/88969850>
+
+        Reviewed by Eric Carlson.
+
+        On macOS 12.3, the default preferred buffer size is roughly 100 ms.
+        This is ok for regular audio playback but is not desirable when playing realtime audio.
+        To reduce the perceived latency, we now reduce the preferred buffer size to 20ms
+        whenever playing an audio MediaStreamTrack, similarly to when capturing audio.
+
+        Manually tested.
+
+        * platform/audio/PlatformMediaSession.cpp:
+        * platform/audio/PlatformMediaSession.h:
+        * platform/audio/cocoa/MediaSessionManagerCocoa.mm:
+
 2022-04-07  Antoine Quint  <graouts@webkit.org>
 

trunk/Source/WebCore/platform/audio/PlatformMediaSession.cpp

@@ -366 +366 @@
 }
 
+bool PlatformMediaSession::hasMediaStreamSource() const
+{
+    return m_client.hasMediaStreamSource();
+}
+
 void PlatformMediaSession::canProduceAudioChanged()
 {

trunk/Source/WebCore/platform/audio/PlatformMediaSession.h

@@ -180 +180 @@
     bool activeAudioSessionRequired() const;
     bool canProduceAudio() const;
+    bool hasMediaStreamSource() const;
     void canProduceAudioChanged();
 

trunk/Source/WebCore/platform/audio/cocoa/MediaSessionManagerCocoa.mm

@@ -114 +114 @@
     int audioCount = 0;
     int webAudioCount = 0;
+    int audioMediaStreamTrackCount = 0;
     int captureCount = countActiveAudioCaptureSources();
     bool hasAudibleAudioOrVideoMediaType = false;
@@ -127 +128 @@
         case PlatformMediaSession::MediaType::VideoAudio:
             ++videoAudioCount;
+            if (session.canProduceAudio() && session.hasMediaStreamSource())
+                ++audioMediaStreamTrackCount;
             break;
         case PlatformMediaSession::MediaType::Audio:
             ++audioCount;
+            if (session.canProduceAudio() && session.hasMediaStreamSource())
+                ++audioMediaStreamTrackCount;
             break;
         case PlatformMediaSession::MediaType::WebAudio:
@@ -153 +158 @@
     ALWAYS_LOG(LOGIDENTIFIER, "types: "
         "AudioCapture(", captureCount, "), "
+        "AudioTrack(", audioMediaStreamTrackCount, "), "
         "Video(", videoCount, "), "
         "Audio(", audioCount, "), "
@@ -161 +167 @@
     if (webAudioCount)
         bufferSize = AudioUtilities::renderQuantumSize;
-    else if (captureCount) {
-        // In case of audio capture, we want to grab 20 ms chunks to limit the latency so that it is not noticeable by users
+    else if (captureCount || audioMediaStreamTrackCount) {
+        // In case of audio capture or audio MediaStreamTrack playing, we want to grab 20 ms chunks to limit the latency so that it is not noticeable by users
         // while having a large enough buffer so that the audio rendering remains stable, hence a computation based on sample rate.
         bufferSize = AudioSession::sharedSession().sampleRate() / 50;