- 9c68613 Update gn files to support Mozilla build by Dan Minor · 7 years ago
- a3dab84 Refactor stream stats generation by Harald Alvestrand · 7 years ago
- c72af93 Reland "Move stats ID generation from SSRC to local ID" by Harald Alvestrand · 7 years ago
- 02879f9 Reland "Make freeNativePeerConnectionObserver generic." by Sami Kalliomäki · 7 years ago
- c0092c3 Revert "Move stats ID generation from SSRC to local ID" by Erik Språng · 7 years ago
- 34814c7 Stop using public_deps in api/. by Mirko Bonadei · 7 years ago
- e357a4d Move stats ID generation from SSRC to local ID by Harald Alvestrand · 7 years ago
- a7f2d84 Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" by Per Kjellander · 7 years ago
- c73e1f4 Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" by Per Kjellander · 7 years ago
- 588c548 GN rtc_* templates: Set default visibility to webrtc_root + "/*" by Karl Wiberg · 7 years ago
- 825249f Add final missing header files before enabling orphans check. by Patrik Höglund · 7 years ago
- 24722b3 Reland "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Seth Hampson · 7 years ago
- c9e74f5 Removing API forwarding headers. by Mirko Bonadei · 7 years ago
- 2d6c76a Switch to using AddTrack with stream labels by Steve Anton · 7 years ago
- 9e19403 Move videosourceinterface to api. by Patrik Höglund · 7 years ago
- be214a2 Move videosinkinterface.h to common_video to solve a circular dep. by Patrik Höglund · 7 years ago
- 8906187 Pivot generation of stats to iterate senders/receivers by Harald Alvestrand · 7 years ago
- dcc3c02 Add support for JSEP offer/answer with transceivers by Steve Anton · 7 years ago
- b96d62d Revert "Make freeNativePeerConnectionObserver generic." by Sami Kalliomäki · 7 years ago
- 046f78c Make freeNativePeerConnectionObserver generic. by Sami Kalliomäki · 7 years ago
- 8b77aea Revert "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Lu Liu · 7 years ago
- d2b912a Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator. by Seth Hampson · 7 years ago
- f32795e Updates to video config to allow changes in google3 tests, in order to not break anything. by Seth Hampson · 7 years ago
- 30bd03b Clarify NACL dependency from peerconnection API. by Patrik Höglund · 7 years ago
- 4b9e6ba Add missing mock headers to api. by Patrik Höglund · 7 years ago
- ecb5e2a Removing deprecated //api:libjingle_peerconnection. by Mirko Bonadei · 7 years ago
- 76df0df Add missing files to rtc_base. by Patrik Höglund · 7 years ago
- 7c5597a Remove unused enum (kStatsValueNameEchoCancellationQualityMin). by Gustaf Ullberg · 7 years ago
- f9381f0 Implement PeerConnection::AddTrack/RemoveTrack for Unified Plan by Steve Anton · 7 years ago
- 3e11343 Fix circular dependencies in webrtc_common. by Patrik Höglund · 7 years ago
- 401d056 Removing $rtc_libyuv_dir and removing useless dependencies on libyuv. by Mirko Bonadei · 7 years ago
- 21eb9fc Make the old GetStats interface on AudioProcessorInterface impure. by Ivo Creusen · 7 years ago
- a8005cf Fix circular dependencies between optional, array_view, and rtc_base. by Patrik Höglund · 7 years ago
- d37709b Revert "Fix circular dependencies between optional, array_view, and rtc_base." by Patrik Höglund · 7 years ago
- a9e0924 Fix circular dependencies between optional, array_view, and rtc_base. by Patrik Höglund · 7 years ago
- 719487e Generate signed packets_lost in WebRTC-stats by Harald Alvestrand · 7 years ago
- 6543206 Including libyuv headers using fully qualified paths. by Mirko Bonadei · 7 years ago
- 3a23374 Reland "Remove the aec_quality_min metric." by Gustaf Ullberg · 7 years ago
- a3fad93 Revert "Remove the aec_quality_min metric." by Mirko Bonadei · 7 years ago
- 99b1bd1 Remove the aec_quality_min metric. by Gustaf Ullberg · 7 years ago
- 8a13874 Remove deprecated datachannel, mediastream, rtpsend and streamcollection headers by Oskar Sundbom · 7 years ago
- 845bb73 Fix clang style errors in api/jsep.h by Steve Anton · 7 years ago
- 88f2cb9 Introduce webrtc::SdpType, the chosen enum for offer/pranswer/answer by Steve Anton · 7 years ago
- 0250be5 Stop using public_deps to depend on libyuv. by Mirko Bonadei · 7 years ago
- 7eaf2f9 Require audio codec API types to have a Config type member by Karl Wiberg · 7 years ago
- b874a35 Complete moving i420 out from video_frame_api. by Patrik Höglund · 7 years ago
- 9158ef6 Reland "Add AddTransceiver and GetTransceivers to PeerConnection" by Steve Anton · 7 years ago
- 8b13f96 Revert "Add AddTransceiver and GetTransceivers to PeerConnection" by Steve Anton · 7 years ago
- f93d280 Add AddTransceiver and GetTransceivers to PeerConnection by Steve Anton · 7 years ago
- 0710965 Old SetRemoteDescription signature default implementation. by Henrik Boström · 7 years ago
- 56d4609 Use the new AudioProcessing statistics everywhere. by Ivo Creusen · 7 years ago
- 0e1d798 Optional: Use nullopt and implicit construction in /api by Oskar Sundbom · 7 years ago
- 3ecdd0f Android: Generate JNI code for MediaConstraints by Magnus Jedvert · 7 years ago
- 3163867 Reland "SetRemoteDescriptionObserverInterface added." by Henrik Boström · 7 years ago
- a4ecf55 Revert "SetRemoteDescriptionObserverInterface added." by Henrik Boström · 7 years ago
- 6c7ec32 SetRemoteDescriptionObserverInterface added. by Henrik Boström · 7 years ago
- e40468b Move some numeric utility code from rtc_base/ to rtc_base/numerics/ by Karl Wiberg · 7 years ago
- 9e6fd2b Add streams() to RtpReceiverInterface and implementations. by Henrik Boström · 7 years ago
- 37e489c Add network_type to local RTCIceCandidateStats by Gary Liu · 7 years ago
- 872cf38 Fix some dependencies for peerconnection_and_implicit_call_api. by Edward Lemur · 7 years ago
- de93943 Revert "Revert "Encode log events periodically instead of for every event."" by Bjorn Terelius · 7 years ago
- 79e7960 Add SDP semantics option to RTCConfiguration by Steve Anton · 7 years ago
- ae02609 Add parallel stats interface with optional stats to APM. by Ivo Creusen · 7 years ago
- 33c5c7f Revert "Encode log events periodically instead of for every event." by Zhi Huang · 7 years ago
- b154c27 Encode log events periodically instead of for every event. by Bjorn Terelius · 7 years ago
- c3ed630 Add stats googHasEnteredLowResolution. by Åsa Persson · 7 years ago
- 9065730 Optional: Use nullopt and implicit construction in /api/audio_codecs by Oskar Sundbom · 7 years ago
- e403212 Added nullopt and implicit construction to rtc::Optional by Oskar Sundbom · 7 years ago
- 7aee3d5 Fix ortc_api circular deps. by Patrik Höglund · 7 years ago
- 6e634bf Add RtpTransceiverInterface and implementing class by Steve Anton · 7 years ago
- 61a7b14 Removing conditional visibility. by Mirko Bonadei · 7 years ago
- b5b5bce Separate i420 and i444 implementations to separate targets. by Patrik Höglund · 7 years ago
- 8f91f1e Make hbos@webrtc.org OWNER of peerconnection*. by Henrik Boström · 7 years ago
- 3e9e5b3 OrtcFactoryInterface::Create(): Require caller to supply audio codec factories by Karl Wiberg · 7 years ago
- 574eaa4 Add alpha channel to VideoFrameBuffer containers by Emircan Uysaler · 7 years ago
- fb65b10 Removing forward declarations that are never referenced. by Mirko Bonadei · 7 years ago
- c241af9 Add timing info to all encoded frames. by Niels Möller · 7 years ago
- 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
- 32df86e Remove deprecated CreatePeerConnectionFactory() overloads by Karl Wiberg · 7 years ago
- eb254b4 Don't select audio codecs depending on GN vars `build_with_{chromium|mozilla}` by Karl Wiberg · 7 years ago
- 5f6bf24 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II) by henrika · 7 years ago
- 990d6b8 Revert "Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API" by Mirko Bonadei · 7 years ago
- f2737d2 Enable the clang style plugin in primary p2p/ target by Steve Anton · 7 years ago
- 90bace0 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API by henrika · 7 years ago
- 996eb9e Fix typo in VideoSendTiming header extension structure by Danil Chapovalov · 7 years ago
- dbcac7f Add StereoCodecAdapter classes by Emircan Uysaler · 7 years ago
- 36b28db Fix clang style warnings in api/candidate.h by Steve Anton · 7 years ago
- 7275e18 Hide the internal AudioEncoderOpus class by giving it an "Impl" suffix by Karl Wiberg · 7 years ago
- d79314f Reland "Add fine grained dropped video frames counters on sending side" by Ilya Nikolaevskiy · 7 years ago
- c4faa9c Remove QUIC transport/data channel by Steve Anton · 7 years ago
- 9155e49 New classes RefCounter and RefCountedBase. by Niels Möller · 7 years ago
- cb423c4 Adding comment about ice_connection_state really being ICE+DTLS. by Taylor Brandstetter · 7 years ago
- 1c1a681 Revert "Add fine grained dropped video frames counters on sending side" by Ilya Nikolaevskiy · 7 years ago
- 78609d5 Reland of BWE allocation strategy by Alex Narest · 7 years ago
- dc9ca93 Revert "BWE allocation strategy" by Alex Narest · 7 years ago
- 1c66f9f Fix chrome tests by making SetBitrateAllocationStrategy not pure virtual by Alex Narest · 7 years ago
- 4b1a363 Add fine grained dropped video frames counters on sending side by Ilya Nikolaevskiy · 7 years ago
- a5fbc23 BWE allocation strategy by Alex Narest · 7 years ago
- 90d46c4 Add phoglund@ to various OWNERS and remove kjellander@ by Henrik Kjellander · 7 years ago
- 39260c4 Revert "BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic." by Lu Liu · 7 years ago