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

[bugzilla-daemon at webkit.org]

https://bugs.webkit.org/show_bug.cgi?id=213506

--- Comment #6 from Philippe Normand <pnormand at igalia.com> ---
(In reply to Lauro Moura from comment #5)
> fast/mediastream/RTCPeerConnection-inspect-offer-bundlePolicy-bundle-only.
> html
> 

This has been failing on all ports for months. Not sure it's a good indicator :)

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