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webrtc_m130/webrtc/pc
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zstein 4b9798024f Relanding: Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP. 
BUG=webrtc:7395

Review-Url: https://codereview.webrtc.org/2888303005
Cr-Original-Commit-Position: refs/heads/master@{#18417}
Committed: 9641c13327
Review-Url: https://codereview.webrtc.org/2888303005
Cr-Commit-Position: refs/heads/master@{#18421}
2017-06-02 21:37:37 +00:00
..
test
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
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
enabling gn check on the whole WebRTC repo
2017-06-01 20:01:48 +00:00
bundlefilter_unittest.cc
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bundlefilter.cc
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bundlefilter.h
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channel_unittest.cc
Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport.
2017-06-01 20:22:42 +00:00
channel.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
channel.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
channelmanager_unittest.cc
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channelmanager.cc
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channelmanager.h
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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
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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
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iceserverparsing.cc
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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
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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 "UDP/DTLS/SCTP" and "TCP/DTLS/SCTP" profile strings.
2017-05-25 16:38:55 +00:00
mediasession.h
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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
Relanding: Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP.
2017-06-02 21:37:37 +00:00
peerconnection.h
Relanding: Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP.
2017-06-02 21:37:37 +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
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peerconnectionfactory.h
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peerconnectioninterface_unittest.cc
Relanding: Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP.
2017-06-02 21:37:37 +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
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rtpreceiver.h
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rtpsender.cc
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rtpsender.h
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rtpsenderreceiver_unittest.cc
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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
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srtpfilter.cc
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srtpfilter.h
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statscollector_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
statscollector.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
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
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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
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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
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
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
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webrtcsessiondescriptionfactory.h
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