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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
2015-12-21Deleted VideoCapturer::screencast_max_pixels, together withnisse
2015-12-19Revert of Reland "Added option to specify a maximum file size when recording ...ivoc
2015-12-19Reland "Added option to specify a maximum file size when recording an AEC dum...ivoc
2015-12-18Revert of Added option to specify a maximum file size when recording an AEC d...ivoc
2015-12-18Added option to specify a maximum file size when recording an AEC dump.ivoc
2015-12-18Fix build break in google3 import caused by https://codereview.webrtc.org/153...guoweis
2015-12-18DTLS-SRTP set up is bypassed when the channel has been writable.guoweis
2015-12-17Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedV...kwiberg
2015-12-16Fixing bug where "mid" wasn't preserved across re-offers.deadbeef
2015-12-12Support for unmixed remote audio into tracks.Tommi
2015-12-11Remove cricket::VideoEncoderConfig.Peter Boström
2015-12-09Adding reduced size RTCP configuration down to the video stream level.deadbeef
2015-12-08Add tracing to public PeerConnection methods.Peter Boström
2015-12-08Refactor handling of AudioOptions.solenberg
2015-12-08Add tracing to upper-level WebRTC calls.Peter Boström
2015-12-07Revert of "Create rtc::AtomicInt POD struct." (patchset #3 id:40001 of https:...pbos
2015-12-07Implement standalone event tracing in AppRTCDemo.Peter Boström
2015-12-07Reland of "Create rtc::AtomicInt POD struct."Peter Boström
2015-12-07Return a copy of the supported RTP header extensions instead of a reference.Stefan Holmer
2015-12-05Allow remote fingerprint update during a callGuo-wei Shieh
2015-12-05Revert "Allow remote fingerprint update during a call"Guo-wei Shieh
2015-12-05Allow remote fingerprint update during a callGuo-wei Shieh
2015-12-02- Remove cricket::VoiceChannel::PressDtmf(); AFAICT unused.solenberg
2015-11-24Fixing direction attribute in answer for non-RTP protocols.deadbeef
2015-11-21Remove SetVideoLogging/SetAudioLogging from ChannelManager and down the stack.solenberg
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
2015-11-16Remove BundleFilter filtering of RTCP.pbos
2015-11-13Revert of Remove global list of SRTP sessions. (patchset #4 id:60001 of https...phoglund
2015-11-11Convert usage of ARRAY_SIZE to arraysize.tfarina
2015-11-10Remove global list of SRTP sessions.jbauch
2015-11-10Rename Maybe to OptionalKarl Wiberg
2015-10-30Replace rtc::cricket::Settable with rtc::Maybekwiberg
2015-10-27Added override keyword to overridden methods to stop compiler warnings.rlester
2015-10-22talk: Use NDEBUG macro.tfarina
2015-10-22Fixing some issues with the direction attribute of m-lines in offers.deadbeef
2015-10-22Added StopAecDump function to PeerConnectionFactory.ivoc
2015-10-16Added functions on libjingle API to start and stop the recording of an RtcEve...ivoc
2015-10-15Wire up packet_id / send time callbacks to webrtc via libjingle.stefan
2015-10-14Merging BaseSession code into WebRtcSession.deadbeef
2015-10-13Remove default receive channel from WVoE; baby step 3.solenberg
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
2015-10-01Fix flaky test TestSrtpError, introduced in https://codereview.webrtc.org/136...solenberg
2015-10-01Remove unused SignalMediaError and infrastructure.solenberg
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