| .. |
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test
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Add support for send_encodings parameters in addTransceiver
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2018-10-01 22:56:30 +00:00 |
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audiotrack.cc
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Remove "using rtc::scoped_ptr" from audiotrack.cc.
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2017-11-28 21:29:18 +00:00 |
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audiotrack.h
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BUILD.gn
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Remove runtime_enabled_feature.
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2018-10-01 07:03:25 +00:00 |
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builtin_video_bitrate_allocator_factory.cc
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The default logic for creating video bitrate allocator.
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2018-08-23 20:50:32 +00:00 |
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builtin_video_bitrate_allocator_factory.h
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The default logic for creating video bitrate allocator.
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2018-08-23 20:50:32 +00:00 |
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bundlefilter_unittest.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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bundlefilter.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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bundlefilter.h
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Revert "Reland "Replace BundleFilter with RtpDemuxer in RtpTransport.""
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2018-03-29 02:45:17 +00:00 |
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channel_unittest.cc
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Cleanup networkroute.h
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2018-07-23 19:35:28 +00:00 |
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channel.cc
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Remove simple stringstream usages.
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2018-09-06 12:53:19 +00:00 |
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channel.h
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Move sigslot to proper third_party directory
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2018-07-25 14:53:33 +00:00 |
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channelmanager_unittest.cc
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Use absl::make_unique and absl::WrapUnique directly
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2018-07-05 10:59:49 +00:00 |
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channelmanager.cc
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Use absl::make_unique and absl::WrapUnique directly
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2018-07-05 10:59:49 +00:00 |
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channelmanager.h
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Use new TransportController implementation in PeerConnection.
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2018-03-30 18:41:19 +00:00 |
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createpeerconnectionfactory.cc
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Allows injection of network controller factory into peer connection factory.
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2018-05-18 17:07:16 +00:00 |
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datachannel_unittest.cc
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Removing warning suppression flags from pc/.
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2018-07-04 10:35:27 +00:00 |
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datachannel.cc
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Use AsyncInvoker in DataChannel instead of MessageHandler
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2018-08-31 22:48:20 +00:00 |
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datachannel.h
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Use AsyncInvoker in DataChannel instead of MessageHandler
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2018-08-31 22:48:20 +00:00 |
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DEPS
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Allows injection of network controller factory into peer connection factory.
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2018-05-18 17:07:16 +00:00 |
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dtlssrtptransport_unittest.cc
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Use absl::make_unique and absl::WrapUnique directly
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2018-07-05 10:59:49 +00:00 |
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dtlssrtptransport.cc
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Add a flag to actively reset the SRTP parameters
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2018-06-12 20:32:00 +00:00 |
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dtlssrtptransport.h
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Replace rtc::Optional with absl::optional in pc
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2018-06-19 20:55:07 +00:00 |
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dtmfsender_unittest.cc
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Add "tones remaining" argument to DTMF ontonechange callback
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2018-09-07 17:29:37 +00:00 |
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dtmfsender.cc
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Add "tones remaining" argument to DTMF ontonechange callback
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2018-09-07 17:29:37 +00:00 |
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dtmfsender.h
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Use AsyncInvoker in DtmfSender instead of MessageHandler
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2018-08-31 22:49:26 +00:00 |
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externalhmac.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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externalhmac.h
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[CFI] Fix cfi-icall failures
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2017-12-12 09:27:07 +00:00 |
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iceserverparsing_unittest.cc
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Revert "Add SSLConfig object to IceServer."
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2018-09-12 10:46:04 +00:00 |
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iceserverparsing.cc
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Revert "Add SSLConfig object to IceServer."
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2018-09-12 10:46:04 +00:00 |
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iceserverparsing.h
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jsepicecandidate.cc
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Add a missing #include from jsepicecandidate.cc to absl/memory/memory.h
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2018-08-15 15:19:07 +00:00 |
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jsepsessiondescription_unittest.cc
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Use absl::make_unique and absl::WrapUnique directly
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2018-07-05 10:59:49 +00:00 |
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jsepsessiondescription.cc
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Add CreateSessionDescription overload which takes a cricket::SessionDescription
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2018-07-25 18:03:05 +00:00 |
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jseptransport_unittest.cc
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Use absl::make_unique and absl::WrapUnique directly
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2018-07-05 10:59:49 +00:00 |
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jseptransport.cc
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Use absl::make_unique and absl::WrapUnique directly
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2018-07-05 10:59:49 +00:00 |
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jseptransport.h
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Move sigslot to proper third_party directory
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2018-07-25 14:53:33 +00:00 |
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jseptransportcontroller_unittest.cc
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Add AsyncResolverFactory interface and basic implementation.
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2018-08-02 21:20:15 +00:00 |
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jseptransportcontroller.cc
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Use AsyncInvoker in JsepTransportController instead of MessageHandler
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2018-08-31 22:12:45 +00:00 |
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jseptransportcontroller.h
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Use AsyncInvoker in JsepTransportController instead of MessageHandler
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2018-08-31 22:12:45 +00:00 |
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localaudiosource_unittest.cc
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Delete almost all use of MediaConstraintsInterface in the PeerConnection API
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2018-08-23 07:14:37 +00:00 |
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localaudiosource.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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localaudiosource.h
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Delete deprecated CreateAudioSource method, with constraints.
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2018-06-04 08:19:30 +00:00 |
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mediaconstraintsinterface_unittest.cc
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Rename a few MediaConfig::Video flags for consistency.
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2018-01-22 17:32:58 +00:00 |
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mediasession_unittest.cc
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Make fewer copies when using StringBuilder.
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2018-09-24 09:39:19 +00:00 |
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mediasession.cc
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Parse two-byte header extensions.
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2018-09-28 08:32:17 +00:00 |
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mediasession.h
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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mediastream_unittest.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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mediastream.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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mediastream.h
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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mediastreamobserver.cc
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mediastreamobserver.h
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Move sigslot to proper third_party directory
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2018-07-25 14:53:33 +00:00 |
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mediastreamtrack.h
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OWNERS
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Adding shampson (me) as an owner to pc/ & api/.
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2018-07-03 20:39:17 +00:00 |
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peerconnection_bundle_unittest.cc
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Add CreateIceCandidate overload which takes a cricket::Candidate
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2018-07-23 23:49:44 +00:00 |
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peerconnection_crypto_unittest.cc
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Delete almost all use of MediaConstraintsInterface in the PeerConnection API
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2018-08-23 07:14:37 +00:00 |
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peerconnection_datachannel_unittest.cc
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Use absl::make_unique and absl::WrapUnique directly
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2018-07-05 10:59:49 +00:00 |
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peerconnection_histogram_unittest.cc
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Remove deprecated field_trial_default and metrics_default.
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2018-09-28 07:21:07 +00:00 |
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peerconnection_ice_unittest.cc
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Remove deprecated field_trial_default and metrics_default.
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2018-09-28 07:21:07 +00:00 |
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peerconnection_integrationtest.cc
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Remove deprecated field_trial_default and metrics_default.
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2018-09-28 07:21:07 +00:00 |
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peerconnection_jsep_unittest.cc
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Use absl::make_unique and absl::WrapUnique directly
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2018-07-05 10:59:49 +00:00 |
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peerconnection_media_unittest.cc
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Add support for enabling simulcast in "Plan B" using MediaConstraints.
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2018-08-24 09:55:59 +00:00 |
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peerconnection_rampup_tests.cc
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Delete almost all use of MediaConstraintsInterface in the PeerConnection API
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2018-08-23 07:14:37 +00:00 |
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peerconnection_rtp_unittest.cc
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Add support for send_encodings parameters in addTransceiver
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2018-10-01 22:56:30 +00:00 |
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peerconnection_signaling_unittest.cc
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Use absl::make_unique and absl::WrapUnique directly
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2018-07-05 10:59:49 +00:00 |
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peerconnection.cc
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Add support for send_encodings parameters in addTransceiver
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2018-10-01 22:56:30 +00:00 |
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peerconnection.h
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Add support for send_encodings parameters in addTransceiver
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2018-10-01 22:56:30 +00:00 |
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peerconnectionendtoend_unittest.cc
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Reland "Delete leftover includes and declarations for MediaConstraintsInterface"
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2018-09-03 09:00:01 +00:00 |
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peerconnectionfactory_unittest.cc
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Delete almost all use of MediaConstraintsInterface in the PeerConnection API
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2018-08-23 07:14:37 +00:00 |
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peerconnectionfactory.cc
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Delete almost all use of MediaConstraintsInterface in the PeerConnection API
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2018-08-23 07:14:37 +00:00 |
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peerconnectionfactory.h
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Delete almost all use of MediaConstraintsInterface in the PeerConnection API
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2018-08-23 07:14:37 +00:00 |
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peerconnectioninterface_unittest.cc
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Reland "Delete leftover includes and declarations for MediaConstraintsInterface"
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2018-09-03 09:00:01 +00:00 |
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peerconnectioninternal.h
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Replace rtc::Optional with absl::optional in pc
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2018-06-19 20:55:07 +00:00 |
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peerconnectionwrapper.cc
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Reland "Reland "Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*.""
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2018-07-12 21:35:47 +00:00 |
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peerconnectionwrapper.h
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Reland "Reland "Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*.""
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2018-07-12 21:35:47 +00:00 |
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proxy_unittest.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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remoteaudiosource.cc
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Use absl::make_unique and absl::WrapUnique directly
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2018-07-05 10:59:49 +00:00 |
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remoteaudiosource.h
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Configure media flow correctly with Unified Plan
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2018-01-18 19:01:38 +00:00 |
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rtcpmuxfilter_unittest.cc
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rtcpmuxfilter.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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rtcpmuxfilter.h
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Move sessiondescription.h/cc from p2p/base to pc/
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2017-12-20 00:21:52 +00:00 |
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rtcstats_integrationtest.cc
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getStats: add relayProtocol
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2018-10-02 08:43:06 +00:00 |
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rtcstatscollector_unittest.cc
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getStats: add relayProtocol
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2018-10-02 08:43:06 +00:00 |
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rtcstatscollector.cc
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getStats: add relayProtocol
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2018-10-02 08:43:06 +00:00 |
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rtcstatscollector.h
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Move sigslot to proper third_party directory
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2018-07-25 14:53:33 +00:00 |
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rtcstatstraversal_unittest.cc
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Stats traversal algorithm added.
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2018-03-12 10:54:09 +00:00 |
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rtcstatstraversal.cc
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RTCStatsCollector::GetStatsReport() with optional selector argument.
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2018-03-19 15:32:16 +00:00 |
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rtcstatstraversal.h
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Stats traversal algorithm added.
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2018-03-12 10:54:09 +00:00 |
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rtpmediautils_unittest.cc
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Implement legacy offer_to_receive options for Unified Plan
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2018-01-27 02:20:29 +00:00 |
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rtpmediautils.cc
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Implement legacy offer_to_receive options for Unified Plan
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2018-01-27 02:20:29 +00:00 |
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rtpmediautils.h
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Reland "Reland "Remove our stream << overloads from non-test build targets.""
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2018-05-03 10:41:41 +00:00 |
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rtpparametersconversion_unittest.cc
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Parse two-byte header extensions.
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2018-09-28 08:32:17 +00:00 |
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rtpparametersconversion.cc
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Add PeerConnection GetRtpSender/ReceiverCapabilities
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2018-06-28 12:40:07 +00:00 |
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rtpparametersconversion.h
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Add PeerConnection GetRtpSender/ReceiverCapabilities
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2018-06-28 12:40:07 +00:00 |
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rtpreceiver.cc
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Add GetSources to VideoRtpReceiver
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2018-09-27 10:00:40 +00:00 |
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rtpreceiver.h
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Add GetSources to VideoRtpReceiver
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2018-09-27 10:00:40 +00:00 |
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rtpsender.cc
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Add support for send_encodings parameters in addTransceiver
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2018-10-01 22:56:30 +00:00 |
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rtpsender.h
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Add support for send_encodings parameters in addTransceiver
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2018-10-01 22:56:30 +00:00 |
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rtpsenderreceiver_unittest.cc
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Add support for send_encodings parameters in addTransceiver
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2018-10-01 22:56:30 +00:00 |
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rtptransceiver.cc
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[Unified Plan] Implement FiredDirection for RtpTransceiver
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2018-07-17 23:56:04 +00:00 |
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rtptransceiver.h
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[Unified Plan] Implement FiredDirection for RtpTransceiver
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2018-07-17 23:56:04 +00:00 |
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rtptransport_unittest.cc
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Replace rtc::Optional with absl::optional in pc
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2018-06-19 20:55:07 +00:00 |
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rtptransport.cc
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Replace rtc::Optional with absl::optional in pc
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2018-06-19 20:55:07 +00:00 |
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rtptransport.h
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Delete struct webrtc::PacketTime.
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2018-08-07 10:07:15 +00:00 |
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rtptransportinternal.h
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Move sigslot to proper third_party directory
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2018-07-25 14:53:33 +00:00 |
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rtptransportinternaladapter.h
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Remove MetricsObserverInterface.
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2018-07-19 23:00:20 +00:00 |
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rtptransporttestutil.h
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Move sigslot to proper third_party directory
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2018-07-25 14:53:33 +00:00 |
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sctputils_unittest.cc
|
Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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sctputils.cc
|
Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
|
sctputils.h
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sdputils.cc
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Use absl::make_unique and absl::WrapUnique directly
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2018-07-05 10:59:49 +00:00 |
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sdputils.h
|
Move sessiondescription.h/cc from p2p/base to pc/
|
2017-12-20 00:21:52 +00:00 |
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sessiondescription.cc
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Use the SDP ContentInfo helpers to avoid downcasting
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2017-12-22 00:17:53 +00:00 |
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sessiondescription.h
|
Add MSID signaling compatibility for Unified Plan endpoints
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2018-02-01 21:17:41 +00:00 |
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srtpfilter_unittest.cc
|
Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
|
srtpfilter.cc
|
Move base64.h to the proper location.
|
2018-07-23 15:40:36 +00:00 |
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srtpfilter.h
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Replace rtc::Optional with absl::optional in pc
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2018-06-19 20:55:07 +00:00 |
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srtpsession_unittest.cc
|
Remove deprecated field_trial_default and metrics_default.
|
2018-09-28 07:21:07 +00:00 |
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srtpsession.cc
|
Remove MetricsObserverInterface.
|
2018-07-19 23:00:20 +00:00 |
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srtpsession.h
|
Remove MetricsObserverInterface.
|
2018-07-19 23:00:20 +00:00 |
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srtptestutil.h
|
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srtptransport_unittest.cc
|
Use absl::make_unique and absl::WrapUnique directly
|
2018-07-05 10:59:49 +00:00 |
|
srtptransport.cc
|
Move base64.h to the proper location.
|
2018-07-23 15:40:36 +00:00 |
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srtptransport.h
|
Remove MetricsObserverInterface.
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2018-07-19 23:00:20 +00:00 |
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statscollector_unittest.cc
|
Reimplement rtc::ToString and rtc::FromString without streams.
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2018-08-16 16:14:01 +00:00 |
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statscollector.cc
|
Reimplement rtc::ToString and rtc::FromString without streams.
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2018-08-16 16:14:01 +00:00 |
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statscollector.h
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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streamcollection.h
|
Reformat the WebRTC code base
|
2018-06-19 14:00:39 +00:00 |
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trackmediainfomap_unittest.cc
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Replace rtc::Optional with absl::optional in pc
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2018-06-19 20:55:07 +00:00 |
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trackmediainfomap.cc
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Replace rtc::Optional with absl::optional in pc
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2018-06-19 20:55:07 +00:00 |
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trackmediainfomap.h
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Replace rtc::Optional with absl::optional in pc
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2018-06-19 20:55:07 +00:00 |
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transportstats.cc
|
Create the JsepTransportController and JsepTransport2.
|
2018-02-23 00:13:45 +00:00 |
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transportstats.h
|
Create the JsepTransportController and JsepTransport2.
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2018-02-23 00:13:45 +00:00 |
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videocapturertracksource_unittest.cc
|
Reimplement rtc::ToString and rtc::FromString without streams.
|
2018-08-16 16:14:01 +00:00 |
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videocapturertracksource.cc
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Replace rtc::Optional with absl::optional in pc
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2018-06-19 20:55:07 +00:00 |
|
videocapturertracksource.h
|
Move sigslot to proper third_party directory
|
2018-07-25 14:53:33 +00:00 |
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videotrack_unittest.cc
|
Refactor VideoTrackSource, without raw pointer injection.
|
2018-05-23 15:42:10 +00:00 |
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videotrack.cc
|
Enable cpplint in pc/
|
2017-10-30 18:08:29 +00:00 |
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videotrack.h
|
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videotracksource.cc
|
Delete deprecated VideoTrackSource constructor.
|
2018-06-11 19:15:10 +00:00 |
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videotracksource.h
|
Replace rtc::Optional with absl::optional in pc
|
2018-06-19 20:55:07 +00:00 |
|
webrtcsdp_unittest.cc
|
Make HasAttribute handle partial matching of attribute names.
|
2018-09-10 08:43:52 +00:00 |
|
webrtcsdp.cc
|
Make HasAttribute handle partial matching of attribute names.
|
2018-09-10 08:43:52 +00:00 |
|
webrtcsdp.h
|
Add MSID signaling compatibility for Unified Plan endpoints
|
2018-02-01 21:17:41 +00:00 |
|
webrtcsessiondescriptionfactory.cc
|
Reland "Delete leftover includes and declarations for MediaConstraintsInterface"
|
2018-09-03 09:00:01 +00:00 |
|
webrtcsessiondescriptionfactory.h
|
Reformat the WebRTC code base
|
2018-06-19 14:00:39 +00:00 |
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