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AgeCommit message (Expand)Author
2015-08-05Get rid of media_engine_ from BaseChannel; only VoiceChannel needs it.Fredrik Solenberg
2015-07-16Nuke buffered latency mode. It's not actually working, and it's not used. I...Peter Thatcher
2015-07-16Use std::string references instead of copying contents.jbauch
2015-07-15Use "UDP/TLS/RTP/SAVPF" profile in offer when DTLS-SRTP is used.deadbeef
2015-07-10Remove BaseSession::SignalNewDescription. It was only used by GTP and now ju...Peter Thatcher
2015-07-09Remove media sinks from Channel.pbos
2015-05-29VoE: Initialize WebRtcVoiceMediaChannel with AudioOptions during creationJelena Marusic
2015-05-28Protect access to shared list of SRTP sessions.Joachim Bauch
2015-05-27Prevent potential double-free if srtp_create fails.Joachim Bauch
2015-05-23Make Config::default_value leak instead of having an exit-time destructor.Andrew MacDonald
2015-05-21Add RtcpMuxPolicy support to PeerConnection.Peter Thatcher
2015-05-19Remove Soundclip handling from libjingle.Fredrik Solenberg
2015-05-18Ensure mediasession generated offers with RTX contain an RTX ssrc for each vi...Noah Richards
2015-05-07Hook up libjingle WebRtcVoiceEngine to Call API for combined A/V BWE.Fredrik Solenberg
2015-05-07Remove WebRtcVideoEngine.Peter Boström
2015-05-04rtc::Buffer: Remove backwards compatibility band-aidsKarl Wiberg
2015-04-30Revert "rtc::Buffer: Remove backwards compatibility band-aids"Karl Wiberg
2015-04-30rtc::Buffer: Remove backwards compatibility band-aidsKarl Wiberg
2015-04-22Remove unused voice channel argument from cricket::VideoChannel ctor and corr...Fredrik Solenberg
2015-04-22Enable more Clang warnings for talk/Henrik Kjellander
2015-04-20rtc::Buffer improvementsKarl Wiberg
2015-04-15Roll chromium_revision 8af41b3..dcb0929 (324854:325030)Magnus Jedvert
2015-04-14Remove SignalCaptureStateChange from MediaEngine.Peter Thatcher
2015-04-13Remove GetStartCaptureFormat and some related code.Peter Thatcher
2015-03-31Use WebRTC API to convert byteorder in srtpfilter.Jiayang Liu
2015-03-24Delete NullVideoRendererMagnus Jedvert
2015-03-24rtc::Buffer: Rename length to size, for conformance with the STLkwiberg@webrtc.org
2015-03-23Update libsrtp includes in preparation of roll into Chromium.jiayl@webrtc.org
2015-03-16Remove a dependency of BaseChannel on WebRtcSession by having WebRtcSession p...pthatcher@webrtc.org
2015-03-16Refactor how the TransportChannels are set in the BaseChannel to rely lesson ...pthatcher@webrtc.org
2015-03-16Remove a hacky dependency of BaseChannel on BaseSession by moving the handlin...pthatcher@webrtc.org
2015-03-16Check associated payload type when negotiate RTX codecs.changbin.shao@webrtc.org
2015-03-13Use a NULL session in unit tests that don't actually use the session.pthatcher@webrtc.org
2015-03-13Cleanup SocketMonitor a little so that it can handle a change in transport ch...pthatcher@webrtc.org
2015-03-13Remove unused transport code.pthatcher@webrtc.org
2015-03-12Socket options are only applied when first setting TransportChannelImpl.guoweis@webrtc.org
2015-03-04Remove GetReceiveBandwidthEstimatorStats.pbos@webrtc.org
2015-02-25Add thread checks to the CaptureManager.hbos@webrtc.org
2015-02-25Thread-safe ChannelManager.GetSupportedFormats, used by VideoSourcehbos@webrtc.org
2015-02-24After another round of reviews.lally@webrtc.org
2015-02-24Attempt on read-only acceptance of -12.lally@webrtc.org
2015-02-17Removing CELT.minyue@webrtc.org
2015-02-11Refactoring WebRTC Java/JNI audio recording in C++ and Java.henrika@webrtc.org
2015-02-02Avoid implicit type truncations by inserting explicit casts or modifying prot...pkasting@chromium.org
2015-01-28Only report the first rtp packet because it indicates the media has started f...honghaiz@google.com
2015-01-22Change GetStreamBySsrc to not copy StreamParams.tommi@webrtc.org
2015-01-20Update libjingle license statements at top of talk files for consistencyjlmiller@webrtc.org
2015-01-02Parallelize MediaRecorder unittests.pbos@webrtc.org
2014-12-23Move the Jingle-specific network code into webrtc/libjingle.pthatcher@webrtc.org
2014-12-19Move Jingle-specific files from talk/session/media to webrtc/libjingle/sessio...pthatcher@webrtc.org
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