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Steve Anton 24efa72e54 Fix RTCP transport not destroyed when channel creation fails 
Bug: None
Change-Id: Ic2f1b7899307eff9b2c98805a5a0eb22ca2e062d
Reviewed-on: https://chromium-review.googlesource.com/642458
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19611}
2017-08-30 22:35:18 +00:00
..
test
Removes unused WaveOut APIs from ADM.
2017-08-29 13:02:10 +00:00
audiomonitor.cc
Adding more detail to MessageQueue::Dispatch logging.
2016-06-10 21:17:33 +00:00
audiomonitor.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
audiotrack.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
audiotrack.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
BUILD.gn
Introduce RtpTransportInternal and SrtpTransport.
2017-07-19 20:38:02 +00:00
bundlefilter_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
bundlefilter.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
bundlefilter.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
channel_unittest.cc
Fix places that trigger no-unused-lambda-capture - change to using static-constexpr.
2017-08-24 14:40:16 +00:00
channel.cc
Introduce RtpTransportInternal and SrtpTransport.
2017-07-19 20:38:02 +00:00
channel.h
Introduce RtpTransportInternal and SrtpTransport.
2017-07-19 20:38:02 +00:00
channelmanager_unittest.cc
Reland of Make the default ctor of rtc::Thread, protected
2017-07-14 21:44:46 +00:00
channelmanager.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
channelmanager.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
createpeerconnectionfactory.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
currentspeakermonitor_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
currentspeakermonitor.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
currentspeakermonitor.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
datachannel_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
datachannel.cc
Replace CHECK(x && y) with two separate CHECK() calls
2017-08-10 00:22:01 +00:00
datachannel.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
DEPS
Recently we moved webrtc/base to webrtc/rtc_base, so these
2017-08-30 11:45:58 +00:00
dtmfsender_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
dtmfsender.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
dtmfsender.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
externalhmac.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
externalhmac.h
Remove HAVE_SRTP define and unmaintained code.
2017-03-06 19:32:22 +00:00
iceserverparsing_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
iceserverparsing.cc
Support a user-provided string for the TLS ALPN extension.
2017-08-29 20:11:16 +00:00
iceserverparsing.h
Rewrite PeerConnection integration tests using better testing practices.
2017-03-30 04:08:16 +00:00
jsepicecandidate.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
jsepsessiondescription_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
jsepsessiondescription.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
localaudiosource_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
localaudiosource.cc
Remove PC factory options param from LocalAudioSource::Create.
2017-02-11 05:26:48 +00:00
localaudiosource.h
Remove PC factory options param from LocalAudioSource::Create.
2017-02-11 05:26:48 +00:00
mediaconstraintsinterface_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
mediamonitor.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
mediamonitor.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
mediasession_unittest.cc
Relanding: Adding support for Unified Plan offer/answer negotiation to the mediasession layer.
2017-08-17 21:10:50 +00:00
mediasession.cc
Relanding: Adding support for Unified Plan offer/answer negotiation to the mediasession layer.
2017-08-17 21:10:50 +00:00
mediasession.h
Relanding: Adding support for Unified Plan offer/answer negotiation to the mediasession layer.
2017-08-17 21:10:50 +00:00
mediastream_unittest.cc
Reland of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt
2017-08-01 06:22:01 +00:00
mediastream.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
mediastream.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
mediastreamobserver.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
mediastreamobserver.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
mediastreamtrack.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
OWNERS
Make hbos and hta rtcstats* OWNERS of webrtc/pc, not webrtc/api.
2017-02-28 14:46:44 +00:00
peerconnection_integrationtest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
peerconnection.cc
Fix the Chromium crash when creating an answer without a remote description.
2017-08-29 20:23:53 +00:00
peerconnection.h
Relanding: Adding support for Unified Plan offer/answer negotiation to the mediasession layer.
2017-08-17 21:10:50 +00:00
peerconnectionendtoend_unittest.cc
Reland of Make the default ctor of rtc::Thread, protected
2017-07-14 21:44:46 +00:00
peerconnectionfactory_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
peerconnectionfactory.cc
Revert of Add logging of host lookups made by TurnPort to the RtcEventLog. (patchset #11 id:200001 of https://codereview.webrtc.org/2996933003/ )
2017-08-29 11:49:00 +00:00
peerconnectionfactory.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
peerconnectioninterface_unittest.cc
Fix the Chromium crash when creating an answer without a remote description.
2017-08-29 20:23:53 +00:00
proxy_unittest.cc
Reland of Make the default ctor of rtc::Thread, protected
2017-07-14 21:44:46 +00:00
quicdatachannel_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
quicdatachannel.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
quicdatachannel.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
quicdatatransport_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
quicdatatransport.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
quicdatatransport.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
remoteaudiosource.cc
Replace CHECK(x && y) with two separate CHECK() calls
2017-08-10 00:22:01 +00:00
remoteaudiosource.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtcpmuxfilter_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtcpmuxfilter.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtcpmuxfilter.h
Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport.
2017-06-01 20:22:42 +00:00
rtcstats_integrationtest.cc
Add stats totalSamplesReceived and concealedSamples
2017-08-25 00:50:42 +00:00
rtcstatscollector_unittest.cc
Add stats totalSamplesReceived and concealedSamples
2017-08-25 00:50:42 +00:00
rtcstatscollector.cc
Add stats totalSamplesReceived and concealedSamples
2017-08-25 00:50:42 +00:00
rtcstatscollector.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtpreceiver.cc
Replace CHECK(x && y) with two separate CHECK() calls
2017-08-10 00:22:01 +00:00
rtpreceiver.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtpsender.cc
Replace CHECK(x && y) with two separate CHECK() calls
2017-08-10 00:22:01 +00:00
rtpsender.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtpsenderreceiver_unittest.cc
Reland of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt
2017-08-01 06:22:01 +00:00
rtptransport_unittest.cc
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
rtptransport.cc
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
rtptransport.h
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
rtptransportinternal.h
Introduce RtpTransportInternal and SrtpTransport.
2017-07-19 20:38:02 +00:00
rtptransporttestutil.h
Introduce RtpTransportInternal and SrtpTransport.
2017-07-19 20:38:02 +00:00
sctputils_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
sctputils.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
sctputils.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
srtpfilter_unittest.cc
Move SrtpSession and tests to their own files.
2017-07-07 21:26:25 +00:00
srtpfilter.cc
Move SrtpSession and tests to their own files.
2017-07-07 21:26:25 +00:00
srtpfilter.h
Move SrtpSession and tests to their own files.
2017-07-07 21:26:25 +00:00
srtpsession_unittest.cc
Move SrtpSession and tests to their own files.
2017-07-07 21:26:25 +00:00
srtpsession.cc
Remove libsrtp 2.0.0 compatibility code.
2017-07-27 22:04:20 +00:00
srtpsession.h
Move SrtpSession and tests to their own files.
2017-07-07 21:26:25 +00:00
srtptestutil.h
Move SrtpSession and tests to their own files.
2017-07-07 21:26:25 +00:00
srtptransport_unittest.cc
Introduce RtpTransportInternal and SrtpTransport.
2017-07-19 20:38:02 +00:00
srtptransport.cc
Introduce RtpTransportInternal and SrtpTransport.
2017-07-19 20:38:02 +00:00
srtptransport.h
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
statscollector_unittest.cc
Wiring discard rate of audio FEC/RED packets up to StatsReport.
2017-08-28 13:17:55 +00:00
statscollector.cc
Wiring discard rate of audio FEC/RED packets up to StatsReport.
2017-08-28 13:17:55 +00:00
statscollector.h
Revert "Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ )"
2017-06-02 13:44:03 +00:00
streamcollection.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
trackmediainfomap_unittest.cc
Reland of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt
2017-08-01 06:22:01 +00:00
trackmediainfomap.cc
Add media related stats (audio level etc.) to unsignaled streams.
2017-05-19 20:09:47 +00:00
trackmediainfomap.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
videocapturertracksource_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
videocapturertracksource.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
videocapturertracksource.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
videotrack_unittest.cc
Reland of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt
2017-08-01 06:22:01 +00:00
videotrack.cc
Reland of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt
2017-08-01 06:22:01 +00:00
videotrack.h
Reland of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt
2017-08-01 06:22:01 +00:00
videotracksource.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
videotracksource.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
voicechannel.h
Simple lint fixes
2016-04-26 12:28:18 +00:00
webrtcsdp_unittest.cc
Ignore "b=AS:-1" instead of treating as a hard error.
2017-08-04 00:49:30 +00:00
webrtcsdp.cc
Ignore "b=AS:-1" instead of treating as a hard error.
2017-08-04 00:49:30 +00:00
webrtcsdp.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
webrtcsession_unittest.cc
Fix places that trigger no-unused-lambda-capture
2017-08-23 11:15:18 +00:00
webrtcsession.cc
Fix RTCP transport not destroyed when channel creation fails
2017-08-30 22:35:18 +00:00
webrtcsession.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
webrtcsessiondescriptionfactory.cc
Relanding: Adding support for Unified Plan offer/answer negotiation to the mediasession layer.
2017-08-17 21:10:50 +00:00
webrtcsessiondescriptionfactory.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00