[Webkit-unassigned] [Bug 213506] [GStreamer][WebRTC] SIGSEGV at _mm_mul_pd() during audio resampling

bugzilla-daemon at webkit.org bugzilla-daemon at webkit.org
Thu Dec 30 22:09:03 PST 2021


Diego Pino <dpino at igalia.com> changed:

           What    |Removed                     |Added
             Status|NEW                         |RESOLVED
                 CC|                            |dpino at igalia.com
         Resolution|---                         |FIXED

--- Comment #9 from Diego Pino <dpino at igalia.com> ---
There were 2 tests left filed under this bug:

  webrtc/audio-video-element-playing.html [ Crash Pass ]
  webrtc/remove-track.html [ Crash Pass ]

The tests have been constantly passing for the last 4 months so I'm marking this bug as resolved.


Tests removed from test expectations in r287490.

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