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webrtc_m130/webrtc/pc
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Alex Narest 42308f615c Fix uploading of available send bitrate statistics. 
BUG=webrtc:5079
R=stefan@webrtc.org

Review-Url: https://codereview.webrtc.org/2943073002 .
Cr-Commit-Position: refs/heads/master@{#18664}
2017-06-19 15:58:15 +00:00
..
test
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
audiomonitor.cc
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audiomonitor.h
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audiotrack.cc
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audiotrack.h
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BUILD.gn
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
bundlefilter_unittest.cc
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bundlefilter.cc
…
bundlefilter.h
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channel_unittest.cc
Delete SignalSrtpError.
2017-06-16 07:57:21 +00:00
channel.cc
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
channel.h
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
channelmanager_unittest.cc
Delete MediaController class, move Call ownership to PeerConnection.
2017-05-05 09:23:02 +00:00
channelmanager.cc
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
channelmanager.h
Delete MediaController class, move Call ownership to PeerConnection.
2017-05-05 09:23:02 +00:00
createpeerconnectionfactory.cc
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
currentspeakermonitor_unittest.cc
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currentspeakermonitor.cc
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currentspeakermonitor.h
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datachannel_unittest.cc
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datachannel.cc
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datachannel.h
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DEPS
Injectable audio encoders: BuiltinAudioEncoderFactory
2017-04-06 17:03:21 +00:00
dtmfsender_unittest.cc
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dtmfsender.cc
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dtmfsender.h
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externalhmac.cc
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externalhmac.h
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iceserverparsing_unittest.cc
Enable SNI in ssl adapter.
2017-06-16 23:30:48 +00:00
iceserverparsing.cc
Enable SNI in ssl adapter.
2017-06-16 23:30:48 +00:00
iceserverparsing.h
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jsepicecandidate.cc
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jsepsessiondescription_unittest.cc
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jsepsessiondescription.cc
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localaudiosource_unittest.cc
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localaudiosource.cc
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localaudiosource.h
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mediaconstraintsinterface_unittest.cc
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mediamonitor.cc
…
mediamonitor.h
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mediasession_unittest.cc
Support "UDP/DTLS/SCTP" and "TCP/DTLS/SCTP" profile strings.
2017-05-25 16:38:55 +00:00
mediasession.cc
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
mediasession.h
Negotiate the same SRTP crypto suites for every DTLS association formed.
2017-04-21 10:23:33 +00:00
mediastream_unittest.cc
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mediastream.cc
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mediastream.h
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mediastreamobserver.cc
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mediastreamobserver.h
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mediastreamtrack.h
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OWNERS
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peerconnection_integrationtest.cc
Support "UDP/DTLS/SCTP" and "TCP/DTLS/SCTP" profile strings.
2017-05-25 16:38:55 +00:00
peerconnection.cc
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
peerconnection.h
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
peerconnectionendtoend_unittest.cc
Remove temporary include of builtin_audio_encoder_factory.h.
2017-05-17 11:39:36 +00:00
peerconnectionfactory_unittest.cc
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peerconnectionfactory.cc
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
peerconnectionfactory.h
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
peerconnectioninterface_unittest.cc
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
proxy_unittest.cc
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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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rtcpmuxfilter_unittest.cc
Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport.
2017-06-01 20:22:42 +00:00
rtcpmuxfilter.cc
Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport.
2017-06-01 20:22:42 +00:00
rtcpmuxfilter.h
Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport.
2017-06-01 20:22:42 +00:00
rtcstats_integrationtest.cc
Remove temporary include of builtin_audio_encoder_factory.h.
2017-05-17 11:39:36 +00:00
rtcstatscollector_unittest.cc
Revert "Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ )"
2017-06-02 13:44:03 +00:00
rtcstatscollector.cc
Revert "Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ )"
2017-06-02 13:44:03 +00:00
rtcstatscollector.h
Revert "Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ )"
2017-06-02 13:44:03 +00:00
rtpreceiver.cc
Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ )
2017-04-10 14:39:05 +00:00
rtpreceiver.h
s/WebRtcVideoChannel2/WebRtcVideoChannel and s/WebRtcVideoEngine2/WebRtcVideoEngine
2017-06-12 08:16:46 +00:00
rtpsender.cc
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rtpsender.h
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
rtpsenderreceiver_unittest.cc
Delete MediaController class, move Call ownership to PeerConnection.
2017-05-05 09:23:02 +00:00
rtptransport_unittest.cc
Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport.
2017-06-01 20:22:42 +00:00
rtptransport.cc
Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport.
2017-06-01 20:22:42 +00:00
rtptransport.h
Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport.
2017-06-01 20:22:42 +00:00
sctputils_unittest.cc
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sctputils.cc
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sctputils.h
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srtpfilter_unittest.cc
Delete SignalSrtpError.
2017-06-16 07:57:21 +00:00
srtpfilter.cc
Delete SignalSrtpError.
2017-06-16 07:57:21 +00:00
srtpfilter.h
Delete SignalSrtpError.
2017-06-16 07:57:21 +00:00
statscollector_unittest.cc
Fix uploading of available send bitrate statistics.
2017-06-19 15:58:15 +00:00
statscollector.cc
Fix uploading of available send bitrate statistics.
2017-06-19 15:58:15 +00:00
statscollector.h
Revert "Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ )"
2017-06-02 13:44:03 +00:00
streamcollection.h
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trackmediainfomap_unittest.cc
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trackmediainfomap.cc
Add media related stats (audio level etc.) to unsignaled streams.
2017-05-19 20:09:47 +00:00
trackmediainfomap.h
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videocapturertracksource_unittest.cc
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videocapturertracksource.cc
Fix compilation issues of std::unique_ptr
2017-04-09 16:09:06 +00:00
videocapturertracksource.h
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videotrack_unittest.cc
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videotrack.cc
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videotrack.h
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videotracksource.cc
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videotracksource.h
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voicechannel.h
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webrtcsdp_unittest.cc
s/WebRtcVideoChannel2/WebRtcVideoChannel and s/WebRtcVideoEngine2/WebRtcVideoEngine
2017-06-12 08:16:46 +00:00
webrtcsdp.cc
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webrtcsdp.h
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webrtcsession_unittest.cc
Remove VirtualSocketServer's dependency on PhysicalSocketServer.
2017-05-17 01:00:06 +00:00
webrtcsession.cc
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
webrtcsession.h
Revert "Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ )"
2017-06-02 13:44:03 +00:00
webrtcsessiondescriptionfactory.cc
Fix comment about remote restart being requested in createOffer
2017-04-25 19:57:14 +00:00
webrtcsessiondescriptionfactory.h
…