Kári Tristan Helgason 8b35df7277 Try re-enabling VoiceChannel::TestInit. ...

This test was disabled for being flaky on the bots.
Try reenabling it to see if something has changed.

Bug: webrtc:7247
Change-Id: I65ce2cf6ce7a3761247369255d9ba106aa3e53f9
Reviewed-on: https://webrtc-review.googlesource.com/3262
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20320}

2017-10-17 09:47:40 +00:00

..

test

Rewrite WebRtcSession ICE tests as PeerConnection tests

2017-10-13 18:42:34 +00:00

audiomonitor.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

audiomonitor.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

audiotrack.cc

Add explicit includes of refcountedobject.h where it is used.

2017-10-06 13:00:14 +00:00

audiotrack.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

BUILD.gn

Rewrite WebRtcSession ICE tests as PeerConnection tests

2017-10-13 18:42:34 +00:00

bundlefilter_unittest.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

bundlefilter.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

bundlefilter.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

channel_unittest.cc

Try re-enabling VoiceChannel::TestInit.

2017-10-17 09:47:40 +00:00

channel.cc

Change the flag when RtpTransport objects send packet.

2017-10-05 04:41:55 +00:00

channel.h

Rewrite WebRtcSession DTLS/SDES crypto tests as PeerConnection tests

2017-10-06 18:50:24 +00:00

channelmanager_unittest.cc

Move the TransportController from p2p/base to pc/.

2017-09-29 18:20:07 +00:00

channelmanager.cc

Keep count of libsrtp clients, and only deinitialize when it goes to 0.

2017-10-13 05:11:27 +00:00

channelmanager.h

Delete unneeded includes of fileutils.h

2017-09-29 12:39:09 +00:00

createpeerconnectionfactory.cc

Remove unnecessary video factory references in PeerConnectionFactory

2017-09-27 14:41:46 +00:00

currentspeakermonitor_unittest.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

currentspeakermonitor.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

currentspeakermonitor.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

datachannel_unittest.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

datachannel.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

datachannel.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

DEPS

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

dtmfsender_unittest.cc

Clean up dependencies of peerconnection_unittest.

2017-09-15 12:51:00 +00:00

dtmfsender.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

dtmfsender.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

externalhmac.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

externalhmac.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

iceserverparsing_unittest.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

iceserverparsing.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

iceserverparsing.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

jsepicecandidate.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

jsepsessiondescription_unittest.cc

Reland "Clean up libjingle API dependencies."

2017-10-05 13:51:21 +00:00

jsepsessiondescription.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

localaudiosource_unittest.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

localaudiosource.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

localaudiosource.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

mediaconstraintsinterface_unittest.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

mediamonitor.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

mediamonitor.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

mediasession_unittest.cc

Removing logging in unit test that was committed accidentally.

2017-10-04 02:32:06 +00:00

mediasession.cc

RTC_CHECK'ing content type before static_casting descriptions.

2017-10-13 16:36:28 +00:00

mediasession.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

mediastream_unittest.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

mediastream.cc

Add explicit includes of refcountedobject.h where it is used.

2017-10-06 13:00:14 +00:00

mediastream.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

mediastreamobserver.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

mediastreamobserver.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

mediastreamtrack.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

OWNERS

…

peerconnection_crypto_unittest.cc

Rewrite WebRtcSession DTLS/SDES crypto tests as PeerConnection tests

2017-10-06 18:50:24 +00:00

peerconnection_ice_unittest.cc

Rewrite WebRtcSession ICE tests as PeerConnection tests

2017-10-13 18:42:34 +00:00

peerconnection_integrationtest.cc

Rewrite WebRtcSession ICE integration tests as PeerConnection tests

2017-10-16 21:44:15 +00:00

peerconnection_rtp_unittest.cc

Reland "Added PeerConnectionObserver::OnRemoveTrack."

2017-10-10 17:06:00 +00:00

peerconnection.cc

PeerConnection::StartRtcEventLog: Improve callback memory safety

2017-10-16 21:37:14 +00:00

peerconnection.h

Add PeerConnection::StartRtcEventLog version that takes RtcEventLogOutput as parameter

2017-10-13 17:18:07 +00:00

peerconnectionendtoend_unittest.cc

PeerConnection end-to-end test with a non-builtin codec

2017-10-11 11:48:54 +00:00

peerconnectionfactory_unittest.cc

Removes fallback from Linux PulseAudio to ALSA.

2017-10-16 11:49:37 +00:00

peerconnectionfactory.cc

TurnCustomizer - an interface for modifying stun messages sent by TurnPort

2017-10-11 07:45:29 +00:00

peerconnectionfactory.h

Remove dead version of StartRtcEventLog

2017-10-06 15:18:24 +00:00

peerconnectioninterface_unittest.cc

Removes fallback from Linux PulseAudio to ALSA.

2017-10-16 11:49:37 +00:00

peerconnectionwrapper.cc

Rewrite WebRtcSession ICE tests as PeerConnection tests

2017-10-13 18:42:34 +00:00

peerconnectionwrapper.h

Rewrite WebRtcSession ICE tests as PeerConnection tests

2017-10-13 18:42:34 +00:00

proxy_unittest.cc

Clean up dependencies of peerconnection_unittest.

2017-09-15 12:51:00 +00:00

quicdatachannel_unittest.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

quicdatachannel.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

quicdatachannel.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

quicdatatransport_unittest.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

quicdatatransport.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

quicdatatransport.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

remoteaudiosource.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

remoteaudiosource.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

rtcpmuxfilter_unittest.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

rtcpmuxfilter.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

rtcpmuxfilter.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

rtcstats_integrationtest.cc

Added RTCMediaStreamTrackStats.jitterBufferDelay for audio

2017-10-02 10:47:00 +00:00

rtcstatscollector_unittest.cc

Move clients of WebRtcSession to use PeerConnection

2017-10-02 17:16:09 +00:00

rtcstatscollector.cc

Reland "Clean up libjingle API dependencies."

2017-10-05 13:51:21 +00:00

rtcstatscollector.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

rtpreceiver.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

rtpreceiver.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

rtpsender.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

rtpsender.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

rtpsenderreceiver_unittest.cc

Move the TransportController from p2p/base to pc/.

2017-09-29 18:20:07 +00:00

rtptransport_unittest.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

rtptransport.cc

Reland: Completed the functionalities of SrtpTransport.

2017-09-26 18:12:45 +00:00

rtptransport.h

Reland: Completed the functionalities of SrtpTransport.

2017-09-26 18:12:45 +00:00

rtptransportinternal.h

Reland: Completed the functionalities of SrtpTransport.

2017-09-26 18:12:45 +00:00

rtptransporttestutil.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

sctputils_unittest.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

sctputils.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

sctputils.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

sdputils.cc

Add sdputils.h with useful functions for working with session descriptions

2017-10-06 17:42:24 +00:00

sdputils.h

Add sdputils.h with useful functions for working with session descriptions

2017-10-06 17:42:24 +00:00

srtpfilter_unittest.cc

Reland: Completed the functionalities of SrtpTransport.

2017-09-26 18:12:45 +00:00

srtpfilter.cc

Keep count of libsrtp clients, and only deinitialize when it goes to 0.

2017-10-13 05:11:27 +00:00

srtpfilter.h

Keep count of libsrtp clients, and only deinitialize when it goes to 0.

2017-10-13 05:11:27 +00:00

srtpsession_unittest.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

srtpsession.cc

Keep count of libsrtp clients, and only deinitialize when it goes to 0.

2017-10-13 05:11:27 +00:00

srtpsession.h

Keep count of libsrtp clients, and only deinitialize when it goes to 0.

2017-10-13 05:11:27 +00:00

srtptestutil.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

srtptransport_unittest.cc

Reland: Completed the functionalities of SrtpTransport.

2017-09-26 18:12:45 +00:00

srtptransport.cc

Reland: Completed the functionalities of SrtpTransport.

2017-09-26 18:12:45 +00:00

srtptransport.h

Reland: Completed the functionalities of SrtpTransport.

2017-09-26 18:12:45 +00:00

statscollector_unittest.cc

Move clients of WebRtcSession to use PeerConnection

2017-10-02 17:16:09 +00:00

statscollector.cc

Move clients of WebRtcSession to use PeerConnection

2017-10-02 17:16:09 +00:00

statscollector.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

streamcollection.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

trackmediainfomap_unittest.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

trackmediainfomap.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

trackmediainfomap.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

transportcontroller_unittest.cc

Move the TransportController from p2p/base to pc/.

2017-09-29 18:20:07 +00:00

transportcontroller.cc

Rewrite WebRtcSession ICE tests as PeerConnection tests

2017-10-13 18:42:34 +00:00

transportcontroller.h

Reland "Clean up libjingle API dependencies."

2017-10-05 13:51:21 +00:00

videocapturertracksource_unittest.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

videocapturertracksource.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

videocapturertracksource.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

videotrack_unittest.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

videotrack.cc

Add explicit includes of refcountedobject.h where it is used.

2017-10-06 13:00:14 +00:00

videotrack.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

videotracksource.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

videotracksource.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

voicechannel.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

webrtcsdp_unittest.cc

Serialize "a=x-google-flag:conference".

2017-09-16 08:24:29 +00:00

webrtcsdp.cc

Reland "Clean up libjingle API dependencies."

2017-10-05 13:51:21 +00:00

webrtcsdp.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

webrtcsession_unittest.cc

Rewrite WebRtcSession ICE integration tests as PeerConnection tests

2017-10-16 21:44:15 +00:00

webrtcsession.cc

Revert "Reject the subsequent offer with fewer m= sections."

2017-10-15 21:50:06 +00:00

webrtcsession.h

Rewrite WebRtcSession DTLS/SDES crypto tests as PeerConnection tests

2017-10-06 18:50:24 +00:00

webrtcsessiondescriptionfactory.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

webrtcsessiondescriptionfactory.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00