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ebbe4f2ed5fb1a1d065ccc4e7c9d56bb2291e8fd
platform-external-webrtc/webrtc/api
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zhihuang ebbe4f2ed5 Set the preferred DSCP value for Rtp data channel to be DSCP_AF41. 
BUG=b/31996729

Review-Url: https://codereview.webrtc.org/2539813003
Cr-Commit-Position: refs/heads/master@{#15449}
2016-12-06 18:45:47 +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
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
stats
Collecting RTCIceCandidatePairStats.transport_id and improved unittests.
2016-11-30 09:50:36 +00:00
test
Move /webrtc/api/android files to /webrtc/sdk/android
2016-12-06 12:29:45 +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 /webrtc/api/android files to /webrtc/sdk/android
2016-12-06 12:29:45 +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
Refactoring: move ownership of RtcEventLog from Call to PeerConnection
2016-10-07 18:53:15 +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
Added a level controller initialization value to MediaConstraints.
2016-10-20 08:53:30 +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
Added a level controller initialization value to MediaConstraints.
2016-10-20 08:53:30 +00:00
mediaconstraintsinterface.h
Added a level controller initialization value to MediaConstraints.
2016-10-20 08:53:30 +00:00
mediacontroller.cc
Refactoring: move ownership of RtcEventLog from Call to PeerConnection
2016-10-07 18:53:15 +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
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
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
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
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
Revert of Disable P2PTestConductor.LocalP2PTestDtlsBundleInIceRestart under msan (patchset #1 id:1 of https://codereview.webrtc.org/2546913003/ )
2016-12-03 02:09:09 +00:00
peerconnection.cc
Added a callback function OnAddTrack to PeerConnectionObserver
2016-11-17 20:06:37 +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
WebRTC: Fix and enable -Woverloaded-virtual warnings.
2016-11-07 09:18:14 +00:00
peerconnectionfactory.cc
Refactoring: move ownership of RtcEventLog from Call to PeerConnection
2016-10-07 18:53:15 +00:00
peerconnectionfactory.h
Refactoring: move ownership of RtcEventLog from Call to PeerConnection
2016-10-07 18:53:15 +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
Modify the parameter type of PeerConnectionObserver callback OnAddTrack.
2016-12-02 23:41:15 +00:00
peerconnectioninterface.h
Modify the parameter type of PeerConnectionObserver callback OnAddTrack.
2016-12-02 23:41:15 +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
Collecting RTCIceCandidatePairStats.transport_id and improved unittests.
2016-11-30 09:50:36 +00:00
rtcstatscollector_unittest.cc
Collecting RTCIceCandidatePairStats.transport_id and improved unittests.
2016-11-30 09:50:36 +00:00
rtcstatscollector.cc
Collecting RTCIceCandidatePairStats.transport_id and improved unittests.
2016-11-30 09:50:36 +00:00
rtcstatscollector.h
RTCCodecStats[1] added.
2016-11-23 10:32:14 +00:00
rtpparameters.h
Pass VideoDecoderParams to VideoDecoderFactory and add SSRC to RtpEncodingParameters.
2016-06-22 07:46:19 +00:00
rtpreceiver.cc
Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
2016-06-27 23:30:45 +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
Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
2016-06-27 23:30:45 +00:00
rtpsender.h
Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
2016-06-27 23:30:45 +00:00
rtpsenderinterface.h
Separating internal and external methods of RtpSender/RtpReceiver.
2016-06-06 21:27:43 +00:00
rtpsenderreceiver_unittest.cc
Refactoring: move ownership of RtcEventLog from Call to PeerConnection
2016-10-07 18:53:15 +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
Implement qpSum stat for video send ssrc stats.
2016-10-31 13:53:51 +00:00
statscollector.cc
Return false if PeerConnection::GetStats() is called on invalid tracks
2016-11-04 18:38:19 +00:00
statscollector.h
Return false if PeerConnection::GetStats() is called on invalid tracks
2016-11-04 18:38:19 +00:00
statstypes.cc
Implement qpSum stat for video send ssrc stats.
2016-10-31 13:53:51 +00:00
statstypes.h
Implement qpSum stat for video send ssrc stats.
2016-10-31 13:53:51 +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
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.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
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
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ )
2016-11-18 09:34:14 +00:00
webrtcsdp.cc
Optimize FindCodecById and ReferencedCodecsMatch
2016-11-11 12:00:20 +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
Make the default value of rtcp-mux policy to required.
2016-11-23 18:30:21 +00:00
webrtcsession.cc
Set the preferred DSCP value for Rtp data channel to be DSCP_AF41.
2016-12-06 18:45:47 +00:00
webrtcsession.h
Add UMA metrics for ICE regathering reasons.
2016-10-05 18:47:39 +00:00
webrtcsessiondescriptionfactory.cc
Adding more detail to MessageQueue::Dispatch logging.
2016-06-10 21:17:33 +00:00
webrtcsessiondescriptionfactory.h
Remove DtlsIdentityStoreInterface, it is no longer used.
2016-07-11 11:10:14 +00:00