- bad33bf Renaming BaseChannel methods and adding comments for added clarity. by Taylor Brandstetter · 8 years ago
- cb56065 Add support for GCM cipher suites from RFC 7714. by jbauch · 8 years ago
- 8853289 Un-flaking TestSrtpError by using a fake clock. by Taylor Brandstetter · 8 years ago
- 6bb1ef2 Fixing bug where Connection drops packets when presumed writable. by Taylor Brandstetter · 9 years ago
- 059e183 Reland of "Revert of Update the BWE when the network route changes. (patchset #5 id:180001 of https://… (patchset #1 id:1 of https://codereview.webrtc.org/2098703004/ ) by honghaiz · 9 years ago
- ae4d0d9 Revert of Revert "Revert of Update the BWE when the network route changes. (patchset #5 id:180001 of https://… (patchset #5 id:120001 of https://codereview.webrtc.org/2041593002/ ) by honghaiz · 9 years ago
- 5b5d2cd Revert "Revert of Update the BWE when the network route changes. (patchset #5 id:180001 of https://codereview.webrtc.org/2000063003/ )" by Honghai Zhang · 9 years ago
- 5d97a9a Adding more detail to MessageQueue::Dispatch logging. by Taylor Brandstetter · 9 years ago
- 5a4a75a Combining SetVideoSend and SetSource into one method. by deadbeef · 9 years ago
- 6c87a67 Do not create a temporary transport channel when using max-bundle by skvlad · 9 years ago
- db0cd9e Adding getParameters/setParameters APIs to RtpReceiver. by Taylor Brandstetter · 9 years ago
- dae07ba Fix BaseChannel destructor when network thread differ from worker thread by Danil Chapovalov · 9 years ago
- 7f216b7 Renames TransportController worker_thread to network_thread. by Danil Chapovalov · 9 years ago
- 33b01f2 Adds network thread to rtc::BaseChannel by Danil Chapovalov · 9 years ago
- 555604a Replace scoped_ptr with unique_ptr in webrtc/base/ by jbauch · 9 years ago
- 0e533ef Update the call when the network route changes by Honghai Zhang · 9 years ago
- 67cf2c1 Removing `preference` field from `cricket::Codec`. by deadbeef · 9 years ago
- e0d4637 Allow applications to control audio send bitrate through RtpParameters. by skvlad · 9 years ago
- 52dce73f Add the last_sent_packet_id to the candidate pair change signal by Honghai Zhang · 9 years ago
- cc411c0 Reset the BWE when the network changes. by Honghai Zhang · 9 years ago
- eec21bd Reland Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 9 years ago
- 194e3bc Revert of Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. (patchset #4 id:60001 of https://codereview.webrtc.org/1785713005/ ) by kjellander · 9 years ago
- 944c390 Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 9 years ago
- 292d658 Fix for intermittent tsan2 errors from SendRtpToRtpOnThread and SendSrtpToSrtpOnThread. by ossu · 9 years ago
- dc1c62c Enable setting the maximum bitrate limit in RtpSender. by skvlad · 9 years ago
- 3102294 Replace scoped_ptr with unique_ptr in webrtc/pc/ by kwiberg · 9 years ago
- c11b184 Remove CaptureManager and related calls in ChannelManager. by perkj · 9 years ago
- 65c7f67 Fix license headers in webrtc/pc by kjellander · 9 years ago
- 9b8df25 Move talk/session/media -> webrtc/pc by kjellander@webrtc.org · 9 years ago[Renamed (99%) from talk/session/media/channel_unittest.cc]
- a96e2d7 Move talk/media to webrtc/media by kjellander · 9 years ago
- ce23bee Remove SendStreamFormat and ViewRequests. by Peter Boström · 9 years ago
- 0eb15ed Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector by kwiberg · 9 years ago
- f888bb5 Support for unmixed remote audio into tracks. by Tommi · 9 years ago
- 1d63dd0 - Remove cricket::VoiceChannel::PressDtmf(); AFAICT unused. by solenberg · 9 years ago
- 482b12e Remove BundleFilter filtering of RTCP. by pbos · 9 years ago
- 5237aaf Convert usage of ARRAY_SIZE to arraysize. by tfarina · 9 years ago
- c1aeaf0 Wire up packet_id / send time callbacks to webrtc via libjingle. by stefan · 9 years ago
- 4bac9c5 Change SetOutputScaling to set a single level, not left/right levels. by solenberg · 9 years ago
- 0c4e06b Use suffixed {uint,int}{8,16,32,64}_t types. by Peter Boström · 9 years ago
- 5629a1d Fix flaky test TestSrtpError, introduced in https://codereview.webrtc.org/1362913004. by solenberg · 9 years ago
- 5b14b42 Remove unused SignalMediaError and infrastructure. by solenberg · 9 years ago
- dfc8f4f Change 'mute' parameter of MediaChannel::SetAudioSend()/SetVideoSend() to 'enable'. by solenberg · 9 years ago
- 456696a Reland Change WebRTC SslCipher to be exposed as number only by Guo-wei Shieh · 9 years ago
- 27dc29b Revert of Change WebRTC SslCipher to be exposed as number only. (patchset #20 id:750001 of https://codereview.webrtc.org/1337673002/ ) by guoweis · 9 years ago
- 4fe3c9a Change WebRTC SslCipher to be exposed as number only. by guoweis · 9 years ago
- 34fbfff Remove VideoMediaChannel::SetRender(). by Peter Boström · 9 years ago
- cbecd35 Reland of TransportController refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/1358413003/ ) by deadbeef · 9 years ago
- a81a42f Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ ) by torbjorng · 9 years ago
- 47ee2f3 TransportController refactoring. by deadbeef · 9 years ago
- 22011c1 Remove Channel::SetRingbackTone() and Channel::PlayRingbackTone(), and the code beneath it (within libjingle). by solenberg · 9 years ago
- 8902433 Revert "TransportController refactoring." by Guo-wei Shieh · 9 years ago
- 9af63f4 TransportController refactoring. by deadbeef · 9 years ago
- b071a19 Full use of NnChannel::SetSendParameters and NnChannel::SetRecvParameters. by Fredrik Solenberg · 9 years ago
- 1dd98f3 - Rename VoiceChannel::MuteStream() -> SetAudioSend() (incl. media channel) by solenberg · 9 years ago
- 66f4339 Remove [Voice|Video]MediaChannel::GetOptions(). by solenberg · 9 years ago
- 8006f07 Remove unused TypingMonitor class. by solenberg · 9 years ago
- d828198 Replaces SSLIdentity* with scoped_refptr<RTCCertificate> in the cricket::Transport layer. by Henrik Boström · 9 years ago
- b6d4ec4 Support generation of EC keys using P256 curve and support ECDSA certs. by Torbjorn Granlund · 9 years ago
- 0c02264 Get rid of media_engine_ from BaseChannel; only VoiceChannel needs it. by Fredrik Solenberg · 9 years ago
- a9b4c32 Nuke buffered latency mode. It's not actually working, and it's not used. It's just dead code complexity. by Peter Thatcher · 9 years ago
- a6d2444 Remove BaseSession::SignalNewDescription. It was only used by GTP and now just clutters the code. by Peter Thatcher · 10 years ago
- 3b1e647b6 Remove media sinks from Channel. by pbos · 10 years ago
- af55ccc Add RtcpMuxPolicy support to PeerConnection. by Peter Thatcher · 10 years ago
- 7fb711f Remove unused voice channel argument from cricket::VideoChannel ctor and corresponding field in class. by Fredrik Solenberg · 10 years ago
- 0e81fdf Avoid implicit type truncations by inserting explicit casts or modifying prototypes to avoid needless up- and then down-casting. by pkasting@chromium.org · 10 years ago
- 5f93d0a Update libjingle license statements at top of talk files for consistency by jlmiller@webrtc.org · 10 years ago
- aacc234 Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository. by pthatcher@webrtc.org · 10 years ago
- 4cb3856 Revert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository." by pthatcher@webrtc.org · 10 years ago
- 536f999 Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository. by pthatcher@webrtc.org · 10 years ago
- f050791 Revert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository." by pthatcher@webrtc.org · 10 years ago
- 4afb599 Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository. by pthatcher@webrtc.org · 10 years ago
- e2b7585 Move ViewRequest and MediaStreams to streamparams.h, and remove dependency on mediasessionclient.h and mediamessages.h. This is part of the effort to remove Jingle-specific code from WebRTC and into its own repository. by pthatcher@webrtc.org · 10 years ago
- 18a3896 Revert r7886:7887. by pbos@webrtc.org · 10 years ago
- dee76f3 Move the obvious/easy Jingle-specific code into webrtc/libjingle. by pthatcher@webrtc.org · 10 years ago
- 269fb4b move xmpp and p2p to webrtc by henrike@webrtc.org · 10 years ago
- 28100cb Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p." by henrike@webrtc.org · 10 years ago
- d1ba6d9 Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p. by henrike@webrtc.org · 10 years ago
- 81ddc78 (Auto)update libjingle 77701902-> 77709729 by buildbot@webrtc.org · 10 years ago
- 34f2a9e Initialize SSL in unittest_main.cc. by pbos@webrtc.org · 10 years ago
- a09a999 (Auto)update libjingle 73222930-> 73226398 by buildbot@webrtc.org · 10 years ago
- 65b98d1 (Auto)update libjingle 72839629-> 72847605 by buildbot@webrtc.org · 10 years ago
- 5b1ebac (Auto)update libjingle 72820109-> 72822008 by buildbot@webrtc.org · 10 years ago
- d509678 (Auto)update libjingle 72819313-> 72820109 by buildbot@webrtc.org · 10 years ago
- 94b996c (Auto)update libjingle 72785516-> 72819313 by buildbot@webrtc.org · 10 years ago
- 476efa2 (Auto)update libjingle 72785180-> 72785516 by buildbot@webrtc.org · 10 years ago
- e0d03f1 (Auto)update libjingle 72443101-> 72446860 by buildbot@webrtc.org · 10 years ago
- 6e203d5 (Auto)update libjingle 72442050-> 72443101 by buildbot@webrtc.org · 10 years ago
- 52148c2 (Auto)update libjingle 72430895-> 72442050 by buildbot@webrtc.org · 10 years ago
- 7cb60cc (Auto)update libjingle 72407428-> 72430895 by buildbot@webrtc.org · 10 years ago
- d4e598d (Auto)update libjingle 72097588-> 72159069 by buildbot@webrtc.org · 10 years ago
- b5348c6 Minor refactoring of the session classes. by tommi@webrtc.org · 11 years ago
- 910473b Fix C++11 -Wnarrowing in channel_unittest.cc. by pbos@webrtc.org · 11 years ago
- 6bfd619 (Auto)update libjingle 67052073-> 67134648 by buildbot@webrtc.org · 11 years ago
- 3e92468 (Auto)update libjingle 67037200-> 67043374 by buildbot@webrtc.org · 11 years ago
- 5ee0f05 (Auto)update libjingle 66138442-> 66236292 by buildbot@webrtc.org · 11 years ago
- f5bebd4 (Auto)update libjingle 64247466-> 64326665 by henrike@webrtc.org · 11 years ago
- 4b26e2e Update libjingle to 59676287 by sergeyu@chromium.org · 11 years ago
- a989080 Update talk to 58127566 together with by wu@webrtc.org · 11 years ago
- 2018269 Revert 5274 "Update talk to 58113193 together with https://webrt..." by wu@webrtc.org · 11 years ago
- a129b6c Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. by wu@webrtc.org · 11 years ago