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path: root/talk/app/webrtc/webrtcsession.cc
AgeCommit message (Expand)Author
2016-01-13Revert of Storing raw audio sink for default audio track. (patchset #7 id:120...deadbeef
2016-01-13Storing raw audio sink for default audio track.deadbeef
2016-01-08Properly handle different transports having different SSL roles.Taylor Brandstetter
2015-12-17Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedV...kwiberg
2015-12-12Support for unmixed remote audio into tracks.Tommi
2015-12-11Discard old-generation candidates when ICE restartsHonghai Zhang
2015-12-11Remove cricket::VideoEncoderConfig.Peter Boström
2015-12-08Add tracing to public PeerConnection methods.Peter Boström
2015-12-08Add tracing to upper-level WebRTC calls.Peter Boström
2015-12-04Ping backup connection at a slower rateHonghai Zhang
2015-12-02- Remove cricket::VoiceChannel::PressDtmf(); AFAICT unused.solenberg
2015-11-25Reland of Adding the ability to create an RtpSender without a track.deadbeef
2015-11-25Remove frame time scheduing in IncomingVideoStreamqiangchen
2015-11-20Revert of Adding the ability to create an RtpSender without a track. (patchse...deadbeef
2015-11-20Reland of Adding the ability to create an RtpSender without a track. (patchse...deadbeef
2015-11-19Removed dummy "mediastreamsignaling.h"perkj
2015-11-19Reland Convert internal representation of Srtp cryptos from string to intGuo-wei Shieh
2015-11-19Revert of Convert internal representation of Srtp cryptos from string to int....guoweis
2015-11-19Convert internal representation of Srtp cryptos from string to int.guoweis
2015-11-10WebRTC should generate default private address even when adapter enumeration ...Guo-wei Shieh
2015-11-10Rename Maybe to OptionalKarl Wiberg
2015-10-30Replace rtc::cricket::Settable with rtc::Maybekwiberg
2015-10-26Revert of Adding the ability to create an RtpSender without a track. (patchse...deadbeef
2015-10-26Adding the ability to create an RtpSender without a track.deadbeef
2015-10-16Remove simulcast bitrate modes.pbos
2015-10-15Wire up packet_id / send time callbacks to webrtc via libjingle.stefan
2015-10-14Merging BaseSession code into WebRtcSession.deadbeef
2015-10-14Reland of Moving MediaStreamSignaling logic into PeerConnection. (patchset #1...deadbeef
2015-10-13Revert of Moving MediaStreamSignaling logic into PeerConnection. (patchset #1...deadbeef
2015-10-13Moving MediaStreamSignaling logic into PeerConnection.deadbeef
2015-10-09Remove MediaChannel::SetRemoteRenderer().solenberg
2015-10-09Change SetOutputScaling to set a single level, not left/right levels.solenberg
2015-10-07Use suffixed {uint,int}{8,16,32,64}_t types.Peter Boström
2015-10-05Convert uint16_t to int for WebRTC cipher/crypto suite.Guo-wei Shieh
2015-10-01Change 'mute' parameter of MediaChannel::SetAudioSend()/SetVideoSend() to 'en...solenberg
2015-10-01Reland Change WebRTC SslCipher to be exposed as number onlyGuo-wei Shieh
2015-10-01Revert of Change WebRTC SslCipher to be exposed as number only. (patchset #20...guoweis
2015-10-01Change WebRTC SslCipher to be exposed as number only.guoweis
2015-09-28Passing the new policy from PeerConnection RTCConfiguration tohonghaiz
2015-09-23Reland of TransportController refactoring. (patchset #1 id:1 of https://coder...deadbeef
2015-09-23Revert of TransportController refactoring. (patchset #6 id:100001 of https://...torbjorng
2015-09-22TransportController refactoring.deadbeef
2015-09-18Revert "TransportController refactoring."Guo-wei Shieh
2015-09-18TransportController refactoring.deadbeef
2015-09-17Add RTC_ prefix to (D)CHECKs and related macros.henrikg
2015-09-15Move instantiation of webrtc::Call into a MediaController class so that it ca...Fredrik Solenberg
2015-09-10- Rename VoiceChannel::MuteStream() -> SetAudioSend() (incl. media channel)solenberg
2015-09-07Remove GetOutputScaling from VoiceMediaChannel.solenberg
2015-09-04Add more IceCandidatePairType for host-host CandidatePairGuo-wei Shieh
2015-09-04Cleanup: Remove duplicated functionsGuo-wei Shieh
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