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

[bugzilla-daemon at webkit.org]

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

            Bug ID: 213506
           Summary: [GStreamer][WebRTC] SIGSEGV at _mm_mul_pd() during
                    audio resampling
           Product: WebKit
           Version: WebKit Nightly Build
          Hardware: Unspecified
                OS: Unspecified
            Status: NEW
          Severity: Normal
          Priority: P2
         Component: WebKitGTK
          Assignee: webkit-unassigned at lists.webkit.org
          Reporter: aboya at igalia.com
                CC: bugs-noreply at webkitgtk.org

Hit once on fast/mediastream/RTCPeerConnection-page-cache.html, but not easy to reproduce (can't reproduce after >8000 iterations).

Program terminated with signal SIGSEGV, Segmentation fault.
#0  0x00007f278686c480 in _mm_mul_pd (__B=..., __A=...) at /usr/lib/gcc/x86_64-unknown-linux-gnu/9.3.0/include/emmintrin.h:272
272       return (__m128d) ((__v2df)__A * (__v2df)__B);
[Current thread is 1 (Thread 0x7f27292fe700 (LWP 1014))]

Thread 1 (Thread 0x7f27292fe700 (LWP 1014)):
#0  0x00007f278686c480 in _mm_mul_pd (__B=..., __A=...) at /usr/lib/gcc/x86_64-unknown-linux-gnu/9.3.0/include/emmintrin.h:272
#1  0x00007f278686c480 in inner_product_gdouble_full_1_sse2 (icoeff=<optimized out>, bstride=<optimized out>, len=<optimized out>, b=<optimized out>, a=<optimized out>, o=<optimized out>) at ../gst-libs/gst/audio/audio-resampler-x86-sse2.c:189
#2  0x00007f278686c480 in resample_gdouble_full_1_sse2 (resampler=0x7f270409fea0, in=0x7f26f4036570, in_len=464, out=0x7f27080301e0, out_len=480, consumed=0x7f27292fd6f0) at ../gst-libs/gst/audio/audio-resampler-x86-sse2.c:264
#3  0x00007f278683b64c in gst_audio_resampler_resample (resampler=0x7f270409fea0, in=in at entry=0x7f270802f3f0, in_frames=<optimized out>, out=out at entry=0x7f27080301e0, out_frames=out_frames at entry=480) at ../gst-libs/gst/audio/audio-resampler.c:1786
#4  0x00007f2786830081 in do_resample (chain=0x7f26f4007570, user_data=0x7f26f40080a0) at ../gst-libs/gst/audio/audio-converter.c:546
#5  0x00007f278682f562 in audio_chain_get_samples (avail=<synthetic pointer>, chain=0x7f26f4007570) at ../gst-libs/gst/audio/audio-converter.c:257
#6  0x00007f278682f562 in do_convert_out (chain=0x7f26f40075e0, user_data=0x7f26f40080a0) at ../gst-libs/gst/audio/audio-converter.c:562
#7  0x00007f27868301d2 in audio_chain_get_samples (avail=<synthetic pointer>, chain=0x7f26f40075e0) at ../gst-libs/gst/audio/audio-converter.c:257
#8  0x00007f27868301d2 in do_quantize (chain=0x7f26f40076c0, user_data=0x7f26f40080a0) at ../gst-libs/gst/audio/audio-converter.c:581
#9  0x00007f278682ed8a in audio_chain_get_samples (avail=<synthetic pointer>, chain=0x7f26f40076c0) at ../gst-libs/gst/audio/audio-converter.c:257
#10 0x00007f278682ed8a in converter_generic (convert=0x7f26f40080a0, flags=<optimized out>, in=<optimized out>, in_frames=<optimized out>, out=0x7f27292fd8b0, out_frames=<optimized out>) at ../gst-libs/gst/audio/audio-converter.c:1275
#11 0x00007f279fa19f3c in WebCore::RealtimeOutgoingAudioSourceLibWebRTC::pullAudioData() (this=0x7f270ee0a450) at ../../Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.cpp:120
#12 0x00007f279fa19893 in WebCore::RealtimeOutgoingAudioSourceLibWebRTC::<lambda()>::operator()(void) const (__closure=0x7f270ee16038) at ../../Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.cpp:91
#13 0x00007f279fa1aba6 in WTF::Detail::CallableWrapper<WebCore::RealtimeOutgoingAudioSourceLibWebRTC::audioSamplesAvailable(const WTF::MediaTime&, const WebCore::PlatformAudioData&, const WebCore::AudioStreamDescription&, size_t)::<lambda()>, void>::call(void) (this=0x7f270ee16030) at DerivedSources/ForwardingHeaders/wtf/Function.h:52
#14 0x00007f279ae1957b in WTF::Function<void ()>::operator()() const (this=0x7f26f4036508) at DerivedSources/ForwardingHeaders/wtf/Function.h:84
#15 0x00007f279eb02def in WebCore::PeerConnectionFactoryAndThreads::OnMessage(rtc::Message*) (this=0x7f27a69b4100 <WebCore::staticFactoryAndThreads()::factoryAndThreads>, message=0x7f27292fdbc0) at ../../Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.cpp:219
#16 0x00007f279c37fae0 in rtc::Thread::Dispatch(rtc::Message*) (this=0x5623f5798eb0, pmsg=0x7f27292fdbc0) at ../../Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread.cc:664
#17 0x00007f279c381989 in rtc::Thread::ProcessMessages(int) (this=0x5623f5798eb0, cmsLoop=-1) at ../../Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread.cc:1000
#18 0x00007f279c380cbb in rtc::Thread::Run() (this=0x5623f5798eb0) at ../../Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread.cc:842
#19 0x00007f279c380c5d in rtc::Thread::PreRun(void*) (pv=0x5623f5798eb0) at ../../Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread.cc:831
#20 0x00007f27870b95e2 in start_thread (arg=<optimized out>) at pthread_create.c:479
#21 0x00007f2784d4a473 in clone () at ../sysdeps/unix/sysv/linux/x86_64/clone.S:95

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