Cherry-pick 276688@main (635fa17). https://bugs.webkit.org/show_bug.c…

…gi?id=271517 [GStreamer][WebRTC] Crash when setting video encoder bitrate per gccbwe 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
WebKit · Mar 27, 2024 · 8e32d00 · 8e32d00
1 parent 28f0194
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Showing 4 changed files with 4 additions and 6 deletions.
diff --git a/LayoutTests/platform/glib/fast/mediastream/RTCPeerConnection-inspect-answer-expected.txt b/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}

diff --git a/LayoutTests/platform/glib/fast/mediastream/RTCPeerConnection-inspect-offer-expected.txt b/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

diff --git a/Source/WebCore/platform/graphics/gstreamer/GStreamerRegistryScanner.h b/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",

diff --git a/Source/WebCore/platform/gstreamer/VideoEncoderPrivateGStreamer.cpp b/Source/WebCore/platform/gstreamer/VideoEncoderPrivateGStreamer.cpp
@@ -577,7 +577,7 @@ 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;
             }
         }