[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 #4 from Alicia Boya García <aboya at igalia.com> ---
RealtimeOutgoingAudioSourceLibWebRTC::pullAudioData() is the most suspicious part of the code, although I couldn't quickly find a blatant memory error. The input buffer is ref'ed and locked during conversion, and the output buffer (m_audioBuffer) is protected by [protectedThis = makeRef(*this)] in the calling lambda.

I can only think of audio being non interleaved in either and therefore in/out expecting an array of more than one pointer, and an invalid pointer being read. But if that was the case that would happen more consistently.

Catching the error with asan would be helpful.

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