- 74097fd Delete unused file screencastid.h. by nisse · 8 years ago
- 2675274 Remove cricket::VideoCodec with, height and framerate properties by perkj · 8 years ago
- e40a7ee GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. by kjellander · 8 years ago
- 8ff52cc Remove useless debugging code by mattdr · 8 years ago
- 8cab52d Fix externalhmac.h/.cc to compile with libsrtp 1 and 2 by mattdr · 8 years ago
- 9fa4975 - Filter data channel codecs based on codec name instead of payload type, which may have been remapped. by solenberg · 8 years ago
- 11a9cbf Refactoring: move ownership of RtcEventLog from Call to PeerConnection by skvlad · 8 years ago
- 7ba3051 Delete unused class cricket::MediaSinkInterface, and mediasink.h. by nisse · 8 years ago
- 51f2919 Update WebRTC to build against libsrtp 2.0 by mattdr · 9 years ago
- b62dbbe GN: Change rtc_source_set targets --> rtc_static_library by kjellander · 9 years ago
- 17f008b GYP: Remove targets inside include_tests==1 that are converted to GN. by kjellander · 9 years ago
- 705ecc5 GN: Change group deps to public_deps. by kjellander · 9 years ago
- a41c13e OWNERS: Make everyone able to change *.gn,*.gni files. by Henrik Kjellander · 9 years ago
- e9cc686 GN Templates: Move common_inherited_config to the template. by ehmaldonado · 9 years ago
- 7a2ce0b GN Templates: Move common_config to the template. by ehmaldonado · 9 years ago
- 38a2132 GN: Introduce templates. by ehmaldonado · 9 years ago
- 4cedf2b Add signaling to support ICE renomination. by Honghai Zhang · 9 years ago
- a69d973 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 9 years ago
- 062ce9f Combining "SetTransportChannel" and "SetRtcpTransportChannel". by deadbeef · 9 years ago
- bad33bf Renaming BaseChannel methods and adding comments for added clarity. by Taylor Brandstetter · 9 years ago
- 1d7a637 Fixing off-by-one error with max SCTP id. by Taylor Brandstetter · 9 years ago
- 23d947d Some cleanup in BaseChannel RTCP mux code. by deadbeef · 9 years ago
- e131ea5 Adding deadbeef and honghaiz as owners of webrtc/pc. by deadbeef · 9 years ago
- 72333d2 Add kjellander@webrtc.org to more BUILD.gn OWNERS files. by kjellander · 9 years ago
- cb56065 Add support for GCM cipher suites from RFC 7714. by jbauch · 9 years ago
- 8853289 Un-flaking TestSrtpError by using a fake clock. by Taylor Brandstetter · 9 years ago
- c309e0e Don't stop sending media on EWOULDBLOCK by skvlad · 9 years ago
- 14d5dbe Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop functions for the Rtc Eventlog non-virtual." and "Fix for RtcEventLog ObjC interface" by ivoc · 9 years ago
- 9e03c3b Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ ) by ivoc · 9 years ago
- 1895526 Move RtcEventLog object from inside VoiceEngine to Call. by Ivo Creusen · 9 years ago
- 37bb54e Reland: Remove global list of SRTP sessions. by Joachim Bauch · 9 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
- b00dc38 Delete GetExecutablePath and related unused code. by Niels Möller · 9 years ago
- 184a3fd Forward the SignalFirstPacketReceived to RtpReceiver. by zhihuang · 9 years ago
- 05b9803 Removed unused GetOutputVolume() and SetOutputVolume() from MediaEngineInterface. by solenberg · 9 years ago
- dedfd28 Support for two audio codec lists down into WebRtcVoiceEngine. by ossu · 9 years ago
- 075af92 Initial asymmetric codec support in MediaSessionDescription by ossu · 9 years ago
- 6379793 Removing obsolete method from channel.h. by deadbeef · 9 years ago
- 142f8c5 GN: Add rtc_pc_unittests by kjellander · 9 years ago
- 5d97a9a Adding more detail to MessageQueue::Dispatch logging. by Taylor Brandstetter · 9 years ago
- c76dc95 Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} by kjellander · 9 years ago
- 5a4a75a Combining SetVideoSend and SetSource into one method. by deadbeef · 9 years ago
- 98bba39 Remove metrics_default from rtc_media dependencies. by kjellander · 9 years ago
- 4d167e5 Revert of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} (patchset #5 id:80001 of https://codereview.webrtc.org/1979933002/ ) by kjellander · 9 years ago
- 164e978 Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} by kjellander · 9 years ago
- a1c548b Add RtpHeaderExtension to avoid client breakage by isheriff · 9 years ago
- 6f8d686 Remove use of RtpHeaderExtension and clean up by isheriff · 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
- fd8be34 Remove webrtc/base/scoped_ptr.h by kwiberg · 9 years ago
- dae07ba Fix BaseChannel destructor when network thread differ from worker thread by Danil Chapovalov · 9 years ago
- fb1dd43 Revert of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} (patchset #2 id:20001 of https://codereview.webrtc.org/1973313002/ ) by kjellander · 9 years ago
- c8d848b Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} by kjellander · 9 years ago
- c1513ee Add a parameter to set a maximum file size when starting an RTC event log on the PeerConnectionFactory API. by ivoc · 9 years ago
- 8744cf6 Revert of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} (patchset #2 id:140001 of https://codereview.webrtc.org/1929633002/ ) by kjellander · 9 years ago
- 4d02a35 GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} by kjellander · 9 years ago
- dc4eb8c Refactoring some tests in peerconnectioninterface_unittest.cc. by Taylor Brandstetter · 9 years ago
- 7f216b7 Renames TransportController worker_thread to network_thread. by Danil Chapovalov · 9 years ago
- 3fe372d Fix all -Wnon-virtual-dtor warnings. by Henrik Kjellander · 9 years ago
- 33b01f2 Adds network thread to rtc::BaseChannel by Danil Chapovalov · 9 years ago
- 6ab3db2 Revert of Remove webrtc/base/scoped_ptr.h (patchset #3 id:100001 of https://codereview.webrtc.org/1942823002/ ) by kwiberg · 9 years ago
- 65fc62e Remove webrtc/base/scoped_ptr.h by kwiberg · 9 years ago
- 8f65cdf Only generate one CNAME per PeerConnection. by zhihuang · 9 years ago
- 82d7862 Change default timestamp to 64 bits in all webrtc directories. by Honghai Zhang · 9 years ago
- cf5b37c Accept all the media profiles required by JSEP. by zhihuang · 9 years ago
- bfefb03 Replace scoped_ptr with unique_ptr everywhere by kwiberg · 9 years ago
- 4485ffb #include "webrtc/base/constructormagic.h" where appropriate by kwiberg · 9 years ago
- 8c011e5 Simple lint fixes by terelius · 9 years ago
- 555604a Replace scoped_ptr with unique_ptr in webrtc/base/ by jbauch · 9 years ago
- 7bc7c06 Revert of Remove the rtc_relative_path GYP variable and similar defines (patchset #1 id:1 of https://codereview.webrtc.org/1903553003/ ) by kjellander · 9 years ago
- e19cf59 Remove the rtc_relative_path GYP variable and similar defines by kjellander · 9 years ago
- 0e533ef Update the call when the network route changes by Honghai Zhang · 9 years ago
- d713e86 Revert of Accept all the media profiles required by JSEP. (patchset #5 id:80001 of https://codereview.webrtc.org/1880913002/ ) by zhihuang · 9 years ago
- 67cf2c1 Removing `preference` field from `cricket::Codec`. by deadbeef · 9 years ago
- b7f425a Accept all the media profiles required by JSEP. by zhihuang · 9 years ago
- 2ded9b1 Replace SetCapturer and SetCaptureDevice by SetSource. Drop return value. by nisse · 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
- 00984ff Reland of move {media,p2p,pc,xmllite,xmpp}_tests.gypi files. (patchset #1 id:1 of https://codereview.webrtc.org/1846693002/ ) by kjellander · 9 years ago
- ff97631 - Add temporary VoEBase::audio_device_module() method. by solenberg · 9 years ago
- 72644d2 Revert of Remove {media,p2p,pc,xmllite,xmpp}_tests.gypi files. (patchset #1 id:1 of https://codereview.webrtc.org/1839763004/ ) by kjellander · 9 years ago
- 231b69f Remove {media,p2p,pc,xmllite,xmpp}_tests.gypi files. by kjellander · 9 years ago
- cc411c0 Reset the BWE when the network changes. by Honghai Zhang · 9 years ago
- b252856 Remove all uses of the HAVE_CONFIG_H define. by Henrik Kjellander · 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
- 5f0b83b Enabling rtcp-rsize negotiation and fixing some issues with it. by Taylor Brandstetter · 9 years ago
- 94a23f0 Reland "Add check_deps rules in DEPS files." by kjellander@webrtc.org · 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
- 0510331 Drop VideoOptions from VideoSendParameters. by nisse · 9 years ago
- 56cf60e Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ ) by kjellander · 9 years ago
- 086f851 Add check_deps rules in DEPS files. by kjellander@webrtc.org · 9 years ago
- 3102294 Replace scoped_ptr with unique_ptr in webrtc/pc/ by kwiberg · 9 years ago
- ca8b404 Add tracing to interesting media-related methods. by Peter Boström · 9 years ago
- 1a018dc Prevent a voice channel from sending data before a source is set. by Taylor Brandstetter · 9 years ago
