[GStreamer][WebRTC] Crash when setting video encoder bitrate per gccb…

…we request

https://bugs.webkit.org/show_bug.cgi?id=271517

Reviewed by Adrian Perez de Castro.

* Source/WebCore/platform/graphics/gstreamer/GStreamerRegistryScanner.h: Disable the
ssrc-audio-level RTP extension.
* Source/WebCore/platform/gstreamer/VideoEncoderPrivateGStreamer.cpp:
(videoEncoderConstructed): Fix typo, our encoder bitrate property is not named target-property.

Canonical link: https://commits.webkit.org/276688@main

LayoutTests/platform/glib/TestExpectations

@@ -2028,8 +2028,7 @@ webkit.org/b/235885 webrtc/ephemeral-certificates-and-cnames.html [ Failure ]
 webkit.org/b/235885 webrtc/filtering-ice-candidate-after-reload.html [ Failure ]
 webkit.org/b/235885 webrtc/peer-connection-track-end.html [ Failure ]
 webkit.org/b/235885 webrtc/peerconnection-page-cache.html [ Timeout ]
-webkit.org/b/271517 webrtc/video-stats.html [ Crash ]
-#webkit.org/b/235885 webrtc/video-stats.html [ Failure ]
+webkit.org/b/235885 webrtc/video-stats.html [ Failure ]
 webkit.org/b/235885 webrtc/vp8-then-h264.html [ Failure Timeout Crash ]
 webkit.org/b/252878 fast/mediastream/mediastreamtrack-video-clone.html [ Crash Failure Timeout Pass ]
 webkit.org/b/213202 fast/mediastream/getUserMedia-grant-persistency3.html [ Failure Pass ]
@@ -2049,8 +2048,7 @@ webrtc/captureCanvas-webrtc.html [ Pass Timeout ]
 webkit.org/b/252878 webrtc/audio-peer-connection-webaudio.html [ Failure Pass Timeout ]
 webkit.org/b/252878 webrtc/audio-samplerate-change.html [ Pass Timeout ]
 webkit.org/b/252878 webrtc/datachannel/bufferedAmountLowThreshold-default.html [ Failure Pass ]
-webkit.org/b/271517 webrtc/video-h264.html [ Crash ]
-#webkit.org/b/252878 webrtc/video-h264.html [ Pass Timeout ]
+webkit.org/b/252878 webrtc/video-h264.html [ Pass Timeout ]
 
 imported/w3c/web-platform-tests/mediacapture-streams/GUM-required-constraint-with-ideal-value.https.html [ Pass Failure ]
 imported/w3c/web-platform-tests/mediacapture-streams/MediaStream-MediaElement-preload-none.https.html [ Pass Failure ]

LayoutTests/platform/glib/fast/mediastream/RTCPeerConnection-inspect-answer-expected.txt

@@ -49,7 +49,6 @@ a=msid:{media-stream-id:OK} {media-stream-track-id:OK}
 a=fingerprint:sha-256 {fingerprint:OK}
 a=extmap:1 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
 a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:mid
-a=extmap:3 urn:ietf:params:rtp-hdrext:ssrc-audio-level
 a=fmtp:96 minptime=10;useinbandfec=1
 a=recvonly
 a=fingerprint:sha-256 {fingerprint:OK}

LayoutTests/platform/glib/fast/mediastream/RTCPeerConnection-inspect-offer-expected.txt

@@ -24,7 +24,6 @@ a=rtcp-fb:96 transport-cc
 a=msid:{media-stream-id:OK} {media-stream-track-id:OK}
 a=extmap:1 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
 a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:mid
-a=extmap:3 urn:ietf:params:rtp-hdrext:ssrc-audio-level
 a=fmtp:96 minptime=10;useinbandfec=1
 a=rtpmap:9 G722/8000
 a=rtcp-fb:9 transport-cc
@@ -71,7 +70,6 @@ a=rtcp-fb:96 transport-cc
 a=msid:{media-stream-id:OK} {media-stream-track-id:OK}
 a=extmap:1 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
 a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:mid
-a=extmap:3 urn:ietf:params:rtp-hdrext:ssrc-audio-level
 a=fmtp:96 minptime=10;useinbandfec=1
 a=rtpmap:9 G722/8000
 a=rtcp-fb:9 transport-cc

Source/WebCore/platform/graphics/gstreamer/GStreamerRegistryScanner.h

@@ -177,8 +177,9 @@ class GStreamerRegistryScanner {
 
     Vector<const char*> m_allAudioRtpExtensions { "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01",
         "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time",
-        "urn:ietf:params:rtp-hdrext:sdes:mid",
-        "urn:ietf:params:rtp-hdrext:ssrc-audio-level"
+        "urn:ietf:params:rtp-hdrext:sdes:mid"
+        // This extension triggers caps negotiation issues. See https://bugs.webkit.org/show_bug.cgi?id=271519.
+        // "urn:ietf:params:rtp-hdrext:ssrc-audio-level"
     };
     Vector<const char*> m_allVideoRtpExtensions { "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01",
         "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time",

Source/WebCore/platform/gstreamer/VideoEncoderPrivateGStreamer.cpp

@@ -579,7 +579,7 @@ static void videoEncoderConstructed(GObject* encoder)
             if (gst_structure_has_name(structure, "encoder-bitrate-change-request")) {
                 uint32_t bitrate;
                 gst_structure_get_uint(structure, "bitrate", &bitrate);
-                g_object_set(parent, "target-bitrate", bitrate, nullptr);
+                g_object_set(parent, "bitrate", bitrate, nullptr);
                 return TRUE;
             }
         }