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authorwuchengli@chromium.org <wuchengli@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d>2014-03-19 03:44:20 +0000
committerwuchengli@chromium.org <wuchengli@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d>2014-03-19 03:44:20 +0000
commit76cd2f7dec281c9842ebc5417eeea376fa54746c (patch)
treeb3e965ba3d74b04475c401a4c6b1df86f39c10a2
parentdd6cf04628fd4772e543b5008b5328f6c7988f5a (diff)
downloadwebrtc-76cd2f7dec281c9842ebc5417eeea376fa54746c.tar.gz
Fix a deadlock in ViEEncoder::DeliverFrame.
The deadlock can happen when using HW encoder. HW encoder calls the encode complete callback on libjingle worker thread instead of ViECaptureThread. The capture thread can hold VieEncoder::|data_cs_| and wait for ModuleRtpRtcpImpl::|critical_section_module_ptrs_|. When libjingle worker thread runs encode complete callback, it can hold ModuleRtpRtcpImpl::|critical_section_module_ptrs_| and wait for VieEncoder::|data_cs_|. |default_rtp_rtcp_| is not guarded by |data_cs|. So move it out of the critical section to avoid the deadlock. BUG=chromium:352567 TEST=Run apprtc loopback on CrOS. Run apprtc between CrOS and Linux. Run vie_auto_test. R=henrik.lundin@webrtc.org, pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10039004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5721 4adac7df-926f-26a2-2b94-8c16560cd09d
-rw-r--r--video_engine/vie_encoder.cc8
1 files changed, 4 insertions, 4 deletions
diff --git a/video_engine/vie_encoder.cc b/video_engine/vie_encoder.cc
index 22a74fb7..5583c758 100644
--- a/video_engine/vie_encoder.cc
+++ b/video_engine/vie_encoder.cc
@@ -575,13 +575,13 @@ void ViEEncoder::DeliverFrame(int id,
ViEId(engine_id_, channel_id_),
"%s: %llu", __FUNCTION__,
video_frame->timestamp());
+ if (default_rtp_rtcp_->SendingMedia() == false) {
+ // We've paused or we have no channels attached, don't encode.
+ return;
+ }
{
CriticalSectionScoped cs(data_cs_.get());
time_of_last_incoming_frame_ms_ = TickTime::MillisecondTimestamp();
- if (default_rtp_rtcp_->SendingMedia() == false) {
- // We've paused or we have no channels attached, don't encode.
- return;
- }
if (EncoderPaused()) {
if (!encoder_paused_and_dropped_frame_) {
TRACE_EVENT_ASYNC_BEGIN0("webrtc", "EncoderPaused", this);