[webkit-changes] [WebKit/WebKit] d94431: [GStreamer][WebRTC] Outgoing tracks are missing SSRCs

[Philippe Normand]

  Branch: refs/heads/main
  Home:   https://github.com/WebKit/WebKit
  Commit: d944311ad4940392fcb8ef736cbd8ffb978d1c50
      https://github.com/WebKit/WebKit/commit/d944311ad4940392fcb8ef736cbd8ffb978d1c50
  Author: Philippe Normand <philn at igalia.com>
  Date:   2023-04-11 (Tue, 11 Apr 2023)

  Changed paths:
    M LayoutTests/fast/mediastream/resources/sdp-utils.js
    M LayoutTests/platform/glib/fast/mediastream/RTCPeerConnection-inspect-answer-expected.txt
    M LayoutTests/platform/glib/fast/mediastream/RTCPeerConnection-inspect-offer-expected.txt
    M Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.cpp
    M Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.h
    M Source/WebCore/Modules/mediastream/gstreamer/GStreamerWebRTCUtils.cpp
    M Source/WebCore/Modules/mediastream/gstreamer/GStreamerWebRTCUtils.h
    M Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceGStreamer.cpp
    M Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceGStreamer.h
    M Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingMediaSourceGStreamer.cpp
    M Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingMediaSourceGStreamer.h
    M Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingVideoSourceGStreamer.cpp
    M Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingVideoSourceGStreamer.h

  Log Message:
  -----------
  [GStreamer][WebRTC] Outgoing tracks are missing SSRCs
https://bugs.webkit.org/show_bug.cgi?id=255099

Reviewed by Xabier Rodriguez-Calvar.

According to RFC 3550 section 8, the SSRC identifier is required to be globally unique within an RTP
session. So we create a dedicated generator per end-point, in charge of tracking the generated
identifiers in order to avoid collisions.

In the updated RTCPeerConnection-inspect-offer.html expected baselines the ssrc attributes expose an
invalid mediastream track ID, webrtcbin uses the transceiver name instead. There is a FIXME in their
code about this, these attributes might not be necessary anymore.

* Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.cpp:
(WebCore::GStreamerMediaEndpoint::GStreamerMediaEndpoint):
(WebCore::GStreamerMediaEndpoint::addTrack):
(WebCore::GStreamerMediaEndpoint::createTransceiverBackends):
(WebCore::GStreamerMediaEndpoint::createSourceForTrack):
* Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.h:
* Source/WebCore/Modules/mediastream/gstreamer/GStreamerWebRTCUtils.cpp:
(WebCore::UniqueSSRCGenerator::generateSSRC):
(WebCore::capsFromRtpCapabilities):
* Source/WebCore/Modules/mediastream/gstreamer/GStreamerWebRTCUtils.h:
(WebCore::UniqueSSRCGenerator::create):
* Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceGStreamer.cpp:
(WebCore::RealtimeOutgoingAudioSourceGStreamer::RealtimeOutgoingAudioSourceGStreamer):
* Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceGStreamer.h:
* Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingMediaSourceGStreamer.cpp:
(WebCore::RealtimeOutgoingMediaSourceGStreamer::RealtimeOutgoingMediaSourceGStreamer):
(WebCore::RealtimeOutgoingMediaSourceGStreamer::allowedCaps const):
* Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingMediaSourceGStreamer.h:
* Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingVideoSourceGStreamer.cpp:
(WebCore::RealtimeOutgoingVideoSourceGStreamer::RealtimeOutgoingVideoSourceGStreamer):
* Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingVideoSourceGStreamer.h:

Canonical link: https://commits.webkit.org/262826@main