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8a0d3573e70a19e1e7e15abb37b7b09d6c877ec0
platform-external-webrtc/webrtc/api
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hbos b78306a7d3 Fix segfault when PeerConnection is destroyed during stats collection. 
RTCStatsCollector relies on PeerConnection and its WebRtcSession. If the
PeerConnection is destroyed, reference counting keeps the
RTCStatsCollector alive until the request has completed. But the request
is using PeerConnection/WebRtcSession resources that are destroyed in
~PeerConnection().

To get around this problem, RTCStatsCollector::WaitForPendingRequest()
is added, which is invoked at ~PeerConnection().

Integration test added, it caused a segmentation fault before this
change / EXPECT failure.

BUG=chromium:627816

Review-Url: https://codereview.webrtc.org/2583613003
Cr-Commit-Position: refs/heads/master@{#15674}
2016-12-19 13:06:57 +00:00
..
android/jni
Move /webrtc/api/android files to /webrtc/sdk/android
2016-12-06 12:29:45 +00:00
audio
Changed the interface AudioMixer::RemoveSource to have a void return type.
2016-11-18 10:03:08 +00:00
call
Move FlexfecReceiveStream from api/call/ to call/.
2016-12-19 09:13:46 +00:00
stats
Rename RTCOutboundRTPStreamStats *_rtt members to *_round_trip_time.
2016-12-15 14:17:15 +00:00
test
RTCStatsCollector: Utilize network thread to minimize thread hops.
2016-12-19 12:58:02 +00:00
video
New gn target video_frame_api.
2016-12-02 12:01:21 +00:00
audiotrack.cc
Propagate MediaStreamSource state to video tracks the same way as audio.
2016-03-23 07:34:01 +00:00
audiotrack.h
Remove webrtc/base/scoped_ptr.h
2016-05-15 02:44:18 +00:00
BUILD.gn
Move FlexfecReceiveStream from api/call/ to call/.
2016-12-19 09:13:46 +00:00
datachannel_unittest.cc
Correct stats for RTCPeerConnectionStats.dataChannels[Opened/Closed].
2016-11-14 09:41:56 +00:00
datachannel.cc
Correct stats for RTCPeerConnectionStats.dataChannels[Opened/Closed].
2016-11-14 09:41:56 +00:00
datachannel.h
Correct stats for RTCPeerConnectionStats.dataChannels[Opened/Closed].
2016-11-14 09:41:56 +00:00
datachannelinterface.h
DataChannelInterface: Remove default implementation of methods.
2016-11-24 09:18:00 +00:00
DEPS
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
dtmfsender_unittest.cc
Fixing flaky DtmfSenderTest by using fake clock.
2016-10-28 20:53:17 +00:00
dtmfsender.cc
Adding more detail to MessageQueue::Dispatch logging.
2016-06-10 21:17:33 +00:00
dtmfsender.h
Enable -Winconsistent-missing-override flag.
2016-04-29 13:09:23 +00:00
dtmfsenderinterface.h
Fix license headers in webrtc/api.
2016-02-10 15:54:53 +00:00
fakemediacontroller.h
Late initialize MediaController, for less resource i.e. ProcessThread, usage by PeerConnection.
2016-03-01 20:42:08 +00:00
fakemetricsobserver.cc
Add UMA metrics for ICE regathering reasons.
2016-10-05 18:47:39 +00:00
fakemetricsobserver.h
Fix license headers in webrtc/api.
2016-02-10 15:54:53 +00:00
jsep.h
Reland the CL to remove candidates when doing continual gathering
2016-03-14 18:59:34 +00:00
jsepicecandidate.cc
Reland the CL to remove candidates when doing continual gathering
2016-03-14 18:59:34 +00:00
jsepicecandidate.h
Reland the CL to remove candidates when doing continual gathering
2016-03-14 18:59:34 +00:00
jsepsessiondescription_unittest.cc
Remove cricket::VideoCodec with, height and framerate properties
2016-10-24 08:21:24 +00:00
jsepsessiondescription.cc
Remove cricket::VideoCodec with, height and framerate properties
2016-10-24 08:21:24 +00:00
jsepsessiondescription.h
Remove cricket::VideoCodec with, height and framerate properties
2016-10-24 08:21:24 +00:00
localaudiosource_unittest.cc
This CL provides interfaces that do not use constraints for
2016-03-04 10:51:44 +00:00
localaudiosource.cc
Adding googAudioNetworkAdaptorConfig to MediaConstraintsInterface.
2016-12-11 10:18:00 +00:00
localaudiosource.h
Remove webrtc/base/scoped_ptr.h
2016-05-15 02:44:18 +00:00
mediaconstraintsinterface_unittest.cc
Embed a cricket::MediaConfig in RTCConfiguration.
2016-04-12 06:25:34 +00:00
mediaconstraintsinterface.cc
Adding googAudioNetworkAdaptorConfig to MediaConstraintsInterface.
2016-12-11 10:18:00 +00:00
mediaconstraintsinterface.h
Adding googAudioNetworkAdaptorConfig to MediaConstraintsInterface.
2016-12-11 10:18:00 +00:00
mediacontroller.cc
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
mediacontroller.h
Refactoring: move ownership of RtcEventLog from Call to PeerConnection
2016-10-07 18:53:15 +00:00
mediastream_unittest.cc
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
mediastream.cc
Fix license headers in webrtc/api.
2016-02-10 15:54:53 +00:00
mediastream.h
Enable -Winconsistent-missing-override flag.
2016-04-29 13:09:23 +00:00
mediastreaminterface.h
Add support for content hints to VideoTrack.
2016-12-16 23:39:11 +00:00
mediastreamobserver.cc
Fix license headers in webrtc/api.
2016-02-10 15:54:53 +00:00
mediastreamobserver.h
Fix license headers in webrtc/api.
2016-02-10 15:54:53 +00:00
mediastreamproxy.h
Rename BEGIN_PROXY_MAP --> BEGIN_SIGNALLING_PROXY_MAP.
2016-04-15 10:49:13 +00:00
mediastreamtrack.h
Removed MediaStreamTrackInterface::set_state
2016-03-24 10:16:23 +00:00
mediastreamtrackproxy.h
Add support for content hints to VideoTrack.
2016-12-16 23:39:11 +00:00
notifier.h
Fix license headers in webrtc/api.
2016-02-10 15:54:53 +00:00
OWNERS
Move androidvideotracksource from api under api/android/jni.
2016-11-22 09:26:22 +00:00
peerconnection_unittest.cc
Add disabled certificate check support to IceServer PeerConnection API.
2016-12-19 12:10:30 +00:00
peerconnection.cc
Fix segfault when PeerConnection is destroyed during stats collection.
2016-12-19 13:06:57 +00:00
peerconnection.h
Adding GetConfiguration to PeerConnection.
2016-11-17 03:42:09 +00:00
peerconnectionendtoend_unittest.cc
Disable PeerConnectionEndToEndTest.CallWithLegacySdp on Asan bots.
2016-11-28 15:23:20 +00:00
peerconnectionfactory_unittest.cc
Refactor "secure bool" into explicit PROTO_TLS.
2016-12-13 13:17:31 +00:00
peerconnectionfactory.cc
Support external audio mixer in webrtc 2.
2016-12-13 22:06:35 +00:00
peerconnectionfactory.h
Support external audio mixer in webrtc 2.
2016-12-13 22:06:35 +00:00
peerconnectionfactoryproxy.h
Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop functions for the Rtc Eventlog non-virtual." and "Fix for RtcEventLog ObjC interface"
2016-07-04 14:07:03 +00:00
peerconnectioninterface_unittest.cc
Don't allow changing ICE pool size after SetLocalDescription.
2016-12-13 02:49:40 +00:00
peerconnectioninterface.h
Add disabled certificate check support to IceServer PeerConnection API.
2016-12-19 12:10:30 +00:00
peerconnectionproxy.h
Adding GetConfiguration to PeerConnection.
2016-11-17 03:42:09 +00:00
proxy_unittest.cc
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
proxy.h
Adding more detail to MessageQueue::Dispatch logging.
2016-06-10 21:17:33 +00:00
quicdatachannel_unittest.cc
Refactoring on QUIC related classes.
2016-07-13 21:13:56 +00:00
quicdatachannel.cc
Refactoring on QUIC related classes.
2016-07-13 21:13:56 +00:00
quicdatachannel.h
Refactoring on QUIC related classes.
2016-07-13 21:13:56 +00:00
quicdatatransport_unittest.cc
Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage.
2016-08-05 18:14:54 +00:00
quicdatatransport.cc
Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage.
2016-08-05 18:14:54 +00:00
quicdatatransport.h
Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage.
2016-08-05 18:14:54 +00:00
remoteaudiosource.cc
Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
2016-06-27 23:30:45 +00:00
remoteaudiosource.h
Move webrtc/audio_*.h to webrtc/api/call
2016-08-31 14:33:14 +00:00
rtcstats_integrationtest.cc
Fix segfault when PeerConnection is destroyed during stats collection.
2016-12-19 13:06:57 +00:00
rtcstatscollector_unittest.cc
RTCStatsCollector: Utilize network thread to minimize thread hops.
2016-12-19 12:58:02 +00:00
rtcstatscollector.cc
Fix segfault when PeerConnection is destroyed during stats collection.
2016-12-19 13:06:57 +00:00
rtcstatscollector.h
Fix segfault when PeerConnection is destroyed during stats collection.
2016-12-19 13:06:57 +00:00
rtpparameters.h
Pass VideoDecoderParams to VideoDecoderFactory and add SSRC to RtpEncodingParameters.
2016-06-22 07:46:19 +00:00
rtpreceiver.cc
Create the Java Wrapper of RtpReceiverObserverInterface.
2016-12-07 18:36:49 +00:00
rtpreceiver.h
Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
2016-06-27 23:30:45 +00:00
rtpreceiverinterface.h
Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
2016-06-27 23:30:45 +00:00
rtpsender.cc
Add support for content hints to VideoTrack.
2016-12-16 23:39:11 +00:00
rtpsender.h
Add support for content hints to VideoTrack.
2016-12-16 23:39:11 +00:00
rtpsenderinterface.h
Separating internal and external methods of RtpSender/RtpReceiver.
2016-06-06 21:27:43 +00:00
rtpsenderreceiver_unittest.cc
Add support for content hints to VideoTrack.
2016-12-16 23:39:11 +00:00
sctputils_unittest.cc
Reinstated sctputils_unittest.cc
2016-11-08 10:05:36 +00:00
sctputils.cc
Split ByteBuffer into writer/reader objects.
2016-03-30 13:43:44 +00:00
sctputils.h
Reland Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies.
2016-03-20 13:15:48 +00:00
statscollector_unittest.cc
RTCStatsCollector: Utilize network thread to minimize thread hops.
2016-12-19 12:58:02 +00:00
statscollector.cc
RTCStatsCollector: Utilize network thread to minimize thread hops.
2016-12-19 12:58:02 +00:00
statscollector.h
Return false if PeerConnection::GetStats() is called on invalid tracks
2016-11-04 18:38:19 +00:00
statstypes.cc
remove googViewLimitedResolution stat
2016-12-12 12:46:27 +00:00
statstypes.h
Revert of Delete rtc::linked_ptr. Only use, in statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2581663002/ )
2016-12-15 11:54:52 +00:00
streamcollection.h
Fix license headers in webrtc/api.
2016-02-10 15:54:53 +00:00
umametrics.h
Add UMA metrics for ICE regathering reasons.
2016-10-05 18:47:39 +00:00
videocapturertracksource_unittest.cc
Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ )
2016-11-11 11:55:19 +00:00
videocapturertracksource.cc
Revert of Add method cricket::VideoCapturer::NeedsDenoising, use in VideoCapturerTrackSource. (patchset #5 id:80001 of https://codereview.webrtc.org/2334683002/ )
2016-10-24 07:22:39 +00:00
videocapturertracksource.h
Remove stop method from VideoTrackSourceInterface.
2016-09-06 09:25:11 +00:00
videosourceproxy.h
Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ )
2016-11-11 11:55:19 +00:00
videotrack_unittest.cc
Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ )
2016-11-11 11:55:19 +00:00
videotrack.cc
Add support for content hints to VideoTrack.
2016-12-16 23:39:11 +00:00
videotrack.h
Add support for content hints to VideoTrack.
2016-12-16 23:39:11 +00:00
videotracksource.cc
Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ )
2016-11-11 11:55:19 +00:00
videotracksource.h
Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ )
2016-11-11 11:55:19 +00:00
webrtcsdp_unittest.cc
Fixing integer overflow when parsing bandwidth attribute.
2016-12-14 00:37:16 +00:00
webrtcsdp.cc
Fixing integer overflow when parsing bandwidth attribute.
2016-12-14 00:37:16 +00:00
webrtcsdp.h
Reland the CL to remove candidates when doing continual gathering
2016-03-14 18:59:34 +00:00
webrtcsession_unittest.cc
RTCStatsCollector: Utilize network thread to minimize thread hops.
2016-12-19 12:58:02 +00:00
webrtcsession.cc
RTCStatsCollector: Utilize network thread to minimize thread hops.
2016-12-19 12:58:02 +00:00
webrtcsession.h
RTCStatsCollector: Utilize network thread to minimize thread hops.
2016-12-19 12:58:02 +00:00
webrtcsessiondescriptionfactory.cc
Implement the "needs-ice-restart" logic for SetConfiguration.
2016-12-10 21:15:39 +00:00
webrtcsessiondescriptionfactory.h
Remove DtlsIdentityStoreInterface, it is no longer used.
2016-07-11 11:10:14 +00:00