[webkit-changes] [WebKit/WebKit] ee440b: Cherry-pick 287268 at main (fd33a1d49820). https://bu...

[Philippe Normand]

  Branch: refs/heads/webkitglib/2.46
  Home:   https://github.com/WebKit/WebKit
  Commit: ee440b19b29f72950a1b5dbe109c8bed25fe5d91
      https://github.com/WebKit/WebKit/commit/ee440b19b29f72950a1b5dbe109c8bed25fe5d91
  Author: Philippe Normand <philn at igalia.com>
  Date:   2024-12-04 (Wed, 04 Dec 2024)

  Changed paths:
    M Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioCapturer.cpp
    M Source/WebCore/platform/mediastream/libwebrtc/gstreamer/GStreamerVideoFrameLibWebRTC.cpp
    M Source/WebCore/platform/mediastream/libwebrtc/gstreamer/RealtimeIncomingAudioSourceLibWebRTC.cpp
    M Source/WebCore/platform/mediastream/libwebrtc/gstreamer/RealtimeIncomingAudioSourceLibWebRTC.h
    M Source/WebCore/platform/mediastream/libwebrtc/gstreamer/RealtimeIncomingVideoSourceLibWebRTC.cpp

  Log Message:
  -----------
  Cherry-pick 287268 at main (fd33a1d49820). https://bugs.webkit.org/show_bug.cgi?id=283896

    [GStreamer][LibWebRTC] A/V sync issues
    https://bugs.webkit.org/show_bug.cgi?id=283896

    Reviewed by Xabier Rodriguez-Calvar.

    Since the pipeline now synchronize on the system clock the timestamps for incoming audio/video
    buffers should be based on that time. It was the case for video already (though with incorrect
    units) and for audio we now apply a static base time base on the monotonic time.

    LibWebRTC ifdefs were also removed from the audio capturer because they were leading to a caps
    mismatch on the generated samples.

    * Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioCapturer.cpp:
    (WebCore::GStreamerAudioCapturer::GStreamerAudioCapturer):
    * Source/WebCore/platform/mediastream/libwebrtc/gstreamer/GStreamerVideoFrameLibWebRTC.cpp:
    (WebCore::convertLibWebRTCVideoFrameToGStreamerSample):
    * Source/WebCore/platform/mediastream/libwebrtc/gstreamer/RealtimeIncomingAudioSourceLibWebRTC.cpp:
    (WebCore::RealtimeIncomingAudioSourceLibWebRTC::OnData):
    * Source/WebCore/platform/mediastream/libwebrtc/gstreamer/RealtimeIncomingAudioSourceLibWebRTC.h:
    * Source/WebCore/platform/mediastream/libwebrtc/gstreamer/RealtimeIncomingVideoSourceLibWebRTC.cpp:
    (WebCore::RealtimeIncomingVideoSourceLibWebRTC::OnFrame):

    Canonical link: https://commits.webkit.org/287268@main

Canonical link: https://commits.webkit.org/282416.335@webkitglib/2.46



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