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webrtc_m130/webrtc/api
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hbos 3168c7a04b Rename RTCOutboundRTPStreamStats *_rtt members to *_round_trip_time. 
The spec renamed these recently:
https://w3c.github.io/webrtc-stats/#since-21-sep-2016*

BUG=chromium:627816

Review-Url: https://codereview.webrtc.org/2576383002
Cr-Commit-Position: refs/heads/master@{#15630}
2016-12-15 14:17:15 +00:00
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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
Generalize FlexfecReceiveStream::Config.
2016-12-08 12:18:05 +00:00
stats
Rename RTCOutboundRTPStreamStats *_rtt members to *_round_trip_time.
2016-12-15 14:17:15 +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
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audiotrack.h
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BUILD.gn
Refactor webrtc/{api,audio} and modules/audio_coding for GN check
2016-12-07 16:23:35 +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
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
dtmfsender_unittest.cc
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dtmfsender.cc
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dtmfsender.h
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dtmfsenderinterface.h
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fakemediacontroller.h
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fakemetricsobserver.cc
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fakemetricsobserver.h
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jsep.h
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jsepicecandidate.cc
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jsepicecandidate.h
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jsepsessiondescription_unittest.cc
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jsepsessiondescription.cc
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jsepsessiondescription.h
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localaudiosource_unittest.cc
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localaudiosource.cc
Adding googAudioNetworkAdaptorConfig to MediaConstraintsInterface.
2016-12-11 10:18:00 +00:00
localaudiosource.h
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mediaconstraintsinterface_unittest.cc
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mediaconstraintsinterface.cc
Adding googAudioNetworkAdaptorConfig to MediaConstraintsInterface.
2016-12-11 10:18:00 +00:00
mediaconstraintsinterface.h
Adding googAudioNetworkAdaptorConfig to MediaConstraintsInterface.
2016-12-11 10:18:00 +00:00
mediacontroller.cc
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
mediacontroller.h
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mediastream_unittest.cc
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mediastream.cc
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mediastream.h
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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
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mediastreamobserver.h
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mediastreamproxy.h
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mediastreamtrack.h
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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
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OWNERS
Move androidvideotracksource from api under api/android/jni.
2016-11-22 09:26:22 +00:00
peerconnection_unittest.cc
Refactor "secure bool" into explicit PROTO_TLS.
2016-12-13 13:17:31 +00:00
peerconnection.cc
Refactor "secure bool" into explicit PROTO_TLS.
2016-12-13 13:17:31 +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
Refactor "secure bool" into explicit PROTO_TLS.
2016-12-13 13:17:31 +00:00
peerconnectionfactory.cc
Support external audio mixer in webrtc 2.
2016-12-13 22:06:35 +00:00
peerconnectionfactory.h
Support external audio mixer in webrtc 2.
2016-12-13 22:06:35 +00:00
peerconnectionfactoryproxy.h
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peerconnectioninterface_unittest.cc
Don't allow changing ICE pool size after SetLocalDescription.
2016-12-13 02:49:40 +00:00
peerconnectioninterface.h
Support external audio mixer in webrtc 2.
2016-12-13 22:06:35 +00:00
peerconnectionproxy.h
Adding GetConfiguration to PeerConnection.
2016-11-17 03:42:09 +00:00
proxy_unittest.cc
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proxy.h
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quicdatachannel_unittest.cc
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quicdatachannel.cc
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quicdatachannel.h
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quicdatatransport_unittest.cc
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quicdatatransport.cc
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quicdatatransport.h
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remoteaudiosource.cc
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remoteaudiosource.h
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rtcstats_integrationtest.cc
Rename RTCOutboundRTPStreamStats *_rtt members to *_round_trip_time.
2016-12-15 14:17:15 +00:00
rtcstatscollector_unittest.cc
Rename RTCOutboundRTPStreamStats *_rtt members to *_round_trip_time.
2016-12-15 14:17:15 +00:00
rtcstatscollector.cc
Rename RTCOutboundRTPStreamStats *_rtt members to *_round_trip_time.
2016-12-15 14:17:15 +00:00
rtcstatscollector.h
RTCCodecStats[1] added.
2016-11-23 10:32:14 +00:00
rtpparameters.h
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rtpreceiver.cc
Create the Java Wrapper of RtpReceiverObserverInterface.
2016-12-07 18:36:49 +00:00
rtpreceiver.h
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rtpreceiverinterface.h
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rtpsender.cc
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rtpsender.h
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rtpsenderinterface.h
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rtpsenderreceiver_unittest.cc
Removing "crypto_required" from MediaContentDescription.
2016-12-13 19:29:16 +00:00
sctputils_unittest.cc
Reinstated sctputils_unittest.cc
2016-11-08 10:05:36 +00:00
sctputils.cc
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sctputils.h
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statscollector_unittest.cc
Removing "crypto_required" from MediaContentDescription.
2016-12-13 19:29:16 +00:00
statscollector.cc
remove googViewLimitedResolution stat
2016-12-12 12:46:27 +00:00
statscollector.h
Return false if PeerConnection::GetStats() is called on invalid tracks
2016-11-04 18:38:19 +00:00
statstypes.cc
remove googViewLimitedResolution stat
2016-12-12 12:46:27 +00:00
statstypes.h
Revert of Delete rtc::linked_ptr. Only use, in statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2581663002/ )
2016-12-15 11:54:52 +00:00
streamcollection.h
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umametrics.h
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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
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videocapturertracksource.h
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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
Fixing integer overflow when parsing bandwidth attribute.
2016-12-14 00:37:16 +00:00
webrtcsdp.cc
Fixing integer overflow when parsing bandwidth attribute.
2016-12-14 00:37:16 +00:00
webrtcsdp.h
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webrtcsession_unittest.cc
Removing "crypto_required" from MediaContentDescription.
2016-12-13 19:29:16 +00:00
webrtcsession.cc
Removing "crypto_required" from MediaContentDescription.
2016-12-13 19:29:16 +00:00
webrtcsession.h
Removing "crypto_required" from MediaContentDescription.
2016-12-13 19:29:16 +00:00
webrtcsessiondescriptionfactory.cc
Implement the "needs-ice-restart" logic for SetConfiguration.
2016-12-10 21:15:39 +00:00
webrtcsessiondescriptionfactory.h
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