[Webkit-unassigned] [Bug 218787] New: REGRESSION(r269642) [GStreamer][WebRTC] Unexpected results after update to M87

bugzilla-daemon at webkit.org bugzilla-daemon at webkit.org
Tue Nov 10 21:04:49 PST 2020


https://bugs.webkit.org/show_bug.cgi?id=218787

            Bug ID: 218787
           Summary: REGRESSION(r269642) [GStreamer][WebRTC] Unexpected
                    results after update to M87
           Product: WebKit
           Version: WebKit Nightly Build
          Hardware: Unspecified
                OS: Unspecified
            Status: NEW
          Severity: Normal
          Priority: P2
         Component: WebRTC
          Assignee: webkit-unassigned at lists.webkit.org
          Reporter: lmoura at igalia.com
                CC: youennf at gmail.com

The list of tests below are crashing/failing/timing out after the libwebrtc update in r269642.

Many of these cases are emitting error messages like:

"Failed to set remote offer sdp: Failed to set remote video description send parameters for m-section with mid='1'."
"Failed to set local offer sdp: Failed to set local video description recv parameters for m-section with mid='0'."

Detailed crashlogs for the two crashes in the following comments.

Link to the first GTK-Release test run with these failures: https://build.webkit.org/results/GTK-Linux-64-bit-Release-Tests/r269646%20(16908)/results.html

# Crashes

webrtc/direction-change.html
webrtc/h264-packetization-mode.html

# Failures
webrtc/connection-state.html
webrtc/createOfferAnswer.html
webrtc/disable-encryption.html
webrtc/ephemeral-certificates-and-cnames.html
webrtc/libwebrtc/setLocalDescriptionCrash.html
webrtc/msection-recycling.html
webrtc/multi-audio.html
webrtc/release-after-getting-track.html
webrtc/remove-track.html
webrtc/utf8-sdp.html
webrtc/video-autoplay.html
webrtc/video-getParameters.html
webrtc/video-mediastreamtrack-stats.html
webrtc/video-receivers.html
webrtc/video-replace-track-to-null.html
webrtc/video-stats.html

# Timeouts
fast/mediastream/RTCPeerConnection-statsSelector.html
webrtc/concurrentVideoPlayback.html
webrtc/h264-baseline.html
webrtc/ice-candidate-sdpMLineIndex.html
webrtc/peerconnection-page-cache-long.html
webrtc/peerconnection-page-cache.html

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