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webrtc_m130/pc
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Harald Alvestrand 1979384e40 Ensure that PC usage is recorded if a PC is alive for 60 seconds. 
Bug: chromium:718508
Change-Id: Id2cbcb370b56cb8a6a6c821e0f89c51089cc8e6b
Reviewed-on: https://webrtc-review.googlesource.com/83140
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23723}
2018-06-25 10:25:38 +00:00
..
test
Remove unused/deprecated DTMF methods
2018-06-20 21:00:10 +00:00
audiotrack.cc
…
audiotrack.h
…
BUILD.gn
Ensure that PC usage is recorded if a PC is alive for 60 seconds.
2018-06-25 10:25:38 +00:00
bundlefilter_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
bundlefilter.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
bundlefilter.h
…
channel_unittest.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
channel.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
channel.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
channelmanager_unittest.cc
Delete unused include of fakevideocapturer.h.
2018-05-17 07:23:01 +00:00
channelmanager.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
channelmanager.h
…
createpeerconnectionfactory.cc
Allows injection of network controller factory into peer connection factory.
2018-05-18 17:07:16 +00:00
datachannel_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
datachannel.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
datachannel.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
DEPS
Allows injection of network controller factory into peer connection factory.
2018-05-18 17:07:16 +00:00
dtlssrtptransport_unittest.cc
Add a flag to actively reset the SRTP parameters
2018-06-12 20:32:00 +00:00
dtlssrtptransport.cc
Add a flag to actively reset the SRTP parameters
2018-06-12 20:32:00 +00:00
dtlssrtptransport.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
dtmfsender_unittest.cc
Remove unused/deprecated DTMF methods
2018-06-20 21:00:10 +00:00
dtmfsender.cc
Remove unused/deprecated DTMF methods
2018-06-20 21:00:10 +00:00
dtmfsender.h
Remove unused/deprecated DTMF methods
2018-06-20 21:00:10 +00:00
externalhmac.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
externalhmac.h
…
iceserverparsing_unittest.cc
…
iceserverparsing.cc
…
iceserverparsing.h
…
jsepicecandidate.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
jsepsessiondescription_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
jsepsessiondescription.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
jseptransport_unittest.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
jseptransport.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
jseptransport.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
jseptransportcontroller_unittest.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
jseptransportcontroller.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
jseptransportcontroller.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
localaudiosource_unittest.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
localaudiosource.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
localaudiosource.h
Delete deprecated CreateAudioSource method, with constraints.
2018-06-04 08:19:30 +00:00
mediaconstraintsinterface_unittest.cc
…
mediasession_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mediasession.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
mediasession.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mediastream_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mediastream.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mediastream.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mediastreamobserver.cc
…
mediastreamobserver.h
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mediastreamtrack.h
…
OWNERS
…
peerconnection_bundle_unittest.cc
Built in video codec factories.
2018-05-03 11:49:42 +00:00
peerconnection_crypto_unittest.cc
Built in video codec factories.
2018-05-03 11:49:42 +00:00
peerconnection_datachannel_unittest.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
peerconnection_histogram_unittest.cc
Ensure that PC usage is recorded if a PC is alive for 60 seconds.
2018-06-25 10:25:38 +00:00
peerconnection_ice_unittest.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
peerconnection_integrationtest.cc
Fixing flakiness in PeerConnectionIntegrationTest.
2018-06-19 22:34:53 +00:00
peerconnection_jsep_unittest.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
peerconnection_media_unittest.cc
Replace BundleFilter with RtpDemuxer in RtpTransport.
2018-04-14 00:57:11 +00:00
peerconnection_rtp_unittest.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
peerconnection_signaling_unittest.cc
Built in video codec factories.
2018-05-03 11:49:42 +00:00
peerconnection.cc
Ensure that PC usage is recorded if a PC is alive for 60 seconds.
2018-06-25 10:25:38 +00:00
peerconnection.h
Ensure that PC usage is recorded if a PC is alive for 60 seconds.
2018-06-25 10:25:38 +00:00
peerconnectionendtoend_unittest.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
peerconnectionfactory_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
peerconnectionfactory.cc
Ensure that PC usage is recorded if a PC is alive for 60 seconds.
2018-06-25 10:25:38 +00:00
peerconnectionfactory.h
Ensure that PC usage is recorded if a PC is alive for 60 seconds.
2018-06-25 10:25:38 +00:00
peerconnectioninterface_unittest.cc
Remove unused/deprecated DTMF methods
2018-06-20 21:00:10 +00:00
peerconnectioninternal.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
peerconnectionwrapper.cc
Convert PeerConnectionWrapper from FakeVideoCapturer to FakeVideoTrackSource.
2018-05-30 09:17:17 +00:00
peerconnectionwrapper.h
Convert PeerConnectionWrapper from FakeVideoCapturer to FakeVideoTrackSource.
2018-05-30 09:17:17 +00:00
proxy_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
remoteaudiosource.cc
…
remoteaudiosource.h
…
rtcpmuxfilter_unittest.cc
…
rtcpmuxfilter.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtcpmuxfilter.h
…
rtcstats_integrationtest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtcstatscollector_unittest.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
rtcstatscollector.cc
Remove stringstreams from pc/
2018-06-11 15:20:59 +00:00
rtcstatscollector.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
rtcstatstraversal_unittest.cc
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rtcstatstraversal.cc
…
rtcstatstraversal.h
…
rtpmediautils_unittest.cc
…
rtpmediautils.cc
…
rtpmediautils.h
Reland "Reland "Remove our stream << overloads from non-test build targets.""
2018-05-03 10:41:41 +00:00
rtpreceiver.cc
Refactor VideoTrackSource, without raw pointer injection.
2018-05-23 15:42:10 +00:00
rtpreceiver.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
rtpsender.cc
Add support for content-hint value "text"
2018-06-21 11:29:20 +00:00
rtpsender.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
rtpsenderreceiver_unittest.cc
Add support for content-hint value "text"
2018-06-21 11:29:20 +00:00
rtptransceiver.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
rtptransceiver.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
rtptransport_unittest.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
rtptransport.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
rtptransport.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
rtptransportinternal.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
rtptransportinternaladapter.h
Replace BundleFilter with RtpDemuxer in RtpTransport.
2018-04-14 00:57:11 +00:00
rtptransporttestutil.h
Replace BundleFilter with RtpDemuxer in RtpTransport.
2018-04-14 00:57:11 +00:00
sctputils_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
sctputils.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
sctputils.h
…
sdputils.cc
…
sdputils.h
…
sessiondescription.cc
…
sessiondescription.h
…
srtpfilter_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
srtpfilter.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
srtpfilter.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
srtpsession_unittest.cc
…
srtpsession.cc
Use RTC_HISTOGRAM_ENUMERATION to report SRTP/SRTCP unprotect error.
2018-06-14 18:35:11 +00:00
srtpsession.h
Delete unneeded includes of basictypes.h.
2018-05-21 19:35:08 +00:00
srtptestutil.h
…
srtptransport_unittest.cc
Replace BundleFilter with RtpDemuxer in RtpTransport.
2018-04-14 00:57:11 +00:00
srtptransport.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
srtptransport.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
statscollector_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
statscollector.cc
Return SSRC stats with the old stats API when SSRCs are unsignaled.
2018-06-19 17:28:25 +00:00
statscollector.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
streamcollection.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
trackmediainfomap_unittest.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
trackmediainfomap.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
trackmediainfomap.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
transportstats.cc
…
transportstats.h
…
videocapturertracksource_unittest.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
videocapturertracksource.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
videocapturertracksource.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
videotrack_unittest.cc
Refactor VideoTrackSource, without raw pointer injection.
2018-05-23 15:42:10 +00:00
videotrack.cc
…
videotrack.h
…
videotracksource.cc
Delete deprecated VideoTrackSource constructor.
2018-06-11 19:15:10 +00:00
videotracksource.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
webrtcsdp_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
webrtcsdp.cc
Convert video quality test from a TEST_F to a TEST fixture.
2018-06-21 15:49:43 +00:00
webrtcsdp.h
…
webrtcsessiondescriptionfactory.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
webrtcsessiondescriptionfactory.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00