webrtc /
src /
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ee97e6ad88c8a5d049a511fbbbdbcdcfe1396af2 - ee97e6a Move GetSendCodec() to MediaSendChannelInterface by Florent Castelli · 1 year, 10 months ago
- d178083 Fix incorrect use of scoped enumerations in format strings by Hans Wennborg · 1 year, 10 months ago
- c4e0254 Fix capture_clock_offset_updater_ data race. by Jeremy Leconte · 1 year, 10 months ago
- f80cf81 Changing the pre echo configuration default. by Jesús de Vicente Peña · 1 year, 10 months ago
- ff58aed Process events with the same timestamp in a defined order. by Björn Terelius · 1 year, 10 months ago
- 64d384f Fix logging of unsupported video type by Philipp Hancke · 1 year, 10 months ago
- 4e0bf2e Support conversion of VP9 non-flexible mode to generic descriptor for non-layered streams only. by philipel · 1 year, 10 months ago
- 9e247b6 Add RtcEventProcessor::AddEvents overload that accepts PacketDirection. by Björn Terelius · 1 year, 10 months ago
- e5ee437 Fix candidate leak with initWithNativeCandidate. by Yury Yarashevich · 1 year, 10 months ago
- 6efbd1f fix: do not use dispatch_queue_global_t directly by Samuel Attard · 1 year, 10 months ago
- 328e7b2 Sort media/engine/webrtc_video_engine.cc by Harald Alvestrand · 1 year, 10 months ago
- 17ec056 ICE: adjust priority of non-relay candidates by Philipp Hancke · 1 year, 10 months ago
- 52f902a Update WebRTC code version (2023-06-20T04:03:02). by webrtc-version-updater · 1 year, 10 months ago
- 097a4de Make all encodedaudioframes inherit from TransformableAudioFrameI'face by Tony Herre · 1 year, 10 months ago
- bb917ac Fixes crash in WgcCaptureSession::ProcessFrame by henrika · 1 year, 10 months ago
- b4969d0 Remove unused dependencies in rtp_rtcp by Danil Chapovalov · 1 year, 10 months ago
- 9919841 Remove preprocessor definition for StatsReport::Value::id_val() by Sameer Vijaykar · 1 year, 10 months ago
- c929ab4 Reland "[Stats] Remove enum-like structs in favor of strings." by Henrik Boström · 1 year, 10 months ago
- d05967c Update WebRTC code version (2023-06-17T04:02:25). by webrtc-version-updater · 1 year, 10 months ago
- 11affdd Fix PeerConnectionDependencies leak on PC init. by Yury Yarashevich · 1 year, 10 months ago
- 4d2a219 Change RTCPReceiver::GetAndResetXrRrRtt to return TimeDelta by Danil Chapovalov · 1 year, 10 months ago
- 18aba66 Add test to ensure task deleted on TQ by Per K · 1 year, 10 months ago
- 213090b Add AbsoluteCaptureTime RTP extension to supported list in engines. by Florent Castelli · 1 year, 10 months ago
- 6c453b7 Light-weight detection of static content when using WGC by henrika · 1 year, 10 months ago
- 45666d4 Revert "[Stats] Remove enum-like structs in favor of strings." by Christoffer Jansson · 1 year, 10 months ago
- 0b86253 Update WebRTC code version (2023-06-16T04:04:20). by webrtc-version-updater · 1 year, 10 months ago
- 816dc3b Roll chromium_revision 570bae752f..8603a0cee2 (1158090:1158377) by chromium-webrtc-autoroll · 1 year, 10 months ago
- ccc87ea [Stats] Remove enum-like structs in favor of strings. by Henrik Boström · 1 year, 10 months ago
- d0b8e8e Reland "Merge the codec types" by Florent Castelli · 1 year, 10 months ago
- 45afbc1 Allow setting a custom randomness source. by Sameer Vijaykar · 1 year, 10 months ago
- 6a38a3e sdp: reject duplicate ssrcs in ssrc-groups by Philipp Hancke · 1 year, 10 months ago
- b37f864 Flexfec: add logging of received length. by Alfred E. Heggestad · 1 year, 10 months ago
- 823c702 In VideoCaptureV4L2 use requested/configured capability by Andreas Pehrson · 1 year, 10 months ago
- 51b8206 Add missing method definition for StatsReport::Value::id_val() by Sameer Vijaykar · 1 year, 10 months ago
- 36e37c7 Roll chromium_revision 47e7a37749..570bae752f (1157983:1158090) by chromium-webrtc-autoroll · 1 year, 10 months ago
- 2fec644 Fix L1Tx target bitrate bug when the standard API is used. by Henrik Boström · 1 year, 10 months ago
- bd14a73 In VideoCaptureModulePipeWire lock around started_ by Andreas Pehrson · 1 year, 10 months ago
- 6bb12e5 In VideoCaptureModulePipeWire split frameInfo_ usage by Andreas Pehrson · 1 year, 10 months ago
- e109fb5 Roll chromium_revision 8af8fba80c..47e7a37749 (1157822:1157983) by chromium-webrtc-autoroll · 1 year, 10 months ago
- a3cd0b9 Update WebRTC code version (2023-06-15T04:11:59). by webrtc-version-updater · 1 year, 10 months ago
- 31b0098 Roll chromium_revision e931919cac..8af8fba80c (1157676:1157822) by chromium-webrtc-autoroll · 1 year, 10 months ago
- aa0b410 Roll chromium_revision 6e77598937..e931919cac (1157573:1157676) by chromium-webrtc-autoroll · 1 year, 10 months ago
- 3d7ec24 Roll chromium_revision bb8855a075..6e77598937 (1157397:1157573) by chromium-webrtc-autoroll · 1 year, 10 months ago
- 8f1903e In VideoCaptureImpl lock api_lock_ when accessing apply_rotation_ by Andreas Pehrson · 1 year, 10 months ago
- b7af6b9 Revert "Merge the codec types" by Florent Castelli · 1 year, 10 months ago
- 1cb54be Delete unused killswitch flag related to scalability mode. by Henrik Boström · 1 year, 10 months ago
- 49ace8b Merge the codec types by Florent Castelli · 1 year, 10 months ago
- 0451baa Roll chromium_revision 13b44452d4..bb8855a075 (1157265:1157397) by chromium-webrtc-autoroll · 1 year, 10 months ago
- 3cf60be Update WebRTC code version (2023-06-14T04:02:08). by webrtc-version-updater · 1 year, 10 months ago
- 7d77e24 Roll chromium_revision 8d5bd97af2..13b44452d4 (1157142:1157265) by chromium-webrtc-autoroll · 1 year, 10 months ago
- eb76ed9 Roll chromium_revision 6f33bc2255..8d5bd97af2 (1156959:1157142) by chromium-webrtc-autoroll · 1 year, 10 months ago
- 1f235d6 Roll chromium_revision 2710d66105..6f33bc2255 (1156780:1156959) by chromium-webrtc-autoroll · 1 year, 10 months ago
- 61deed5 Change flexfec header writer to finalize header according to updated RFC by Yosef Twaik · 1 year, 10 months ago
- bde7c6a Change FinalizeFecHeader to receive list of streams by Yosef Twaik · 1 year, 10 months ago
- cfa0f81 Fix DEPS path for clang-format scripts by Florent Castelli · 1 year, 10 months ago
- 18f66fc [rct_tools/video_encoder] Add video encoder tool by Jianhui Dai · 1 year, 10 months ago
- febf558 Revert "Adopt EglThread in EglRenderer" by Linus Nilsson · 1 year, 10 months ago
- c8b217a Roll chromium_revision 0e945e28ee..2710d66105 (1156642:1156780) by chromium-webrtc-autoroll · 1 year, 10 months ago
- 9bb7f81 Update WebRTC code version (2023-06-13T04:07:18). by webrtc-version-updater · 1 year, 10 months ago
- 491d1d6 Roll chromium_revision a3f4dda807..0e945e28ee (1156488:1156642) by chromium-webrtc-autoroll · 1 year, 10 months ago
- e370f2f Roll chromium_revision 6f1f457b3d..a3f4dda807 (1156303:1156488) by chromium-webrtc-autoroll · 1 year, 10 months ago
- f6f642d Roll chromium_revision dd02f6b781..6f1f457b3d (1156166:1156303) by chromium-webrtc-autoroll · 1 year, 10 months ago
- c0e2418 Sort WebRtcAudio{Send,Receive}Channel implementation by Harald Alvestrand · 1 year, 10 months ago
- 4f6783c Roll chromium_revision a3756bb36c..dd02f6b781 (1154916:1156166) by Mirko Bonadei · 1 year, 10 months ago
- de92338 Update parameters' type from NSString to AVAudioSession*. by Abby Yeh · 1 year, 10 months ago
- ee58849 Make SetRTPTimestamp pure virtual in TransformableAudioFrameInterface by Palak Agarwal · 1 year, 10 months ago
- 47bdcc1 When updating audio session, update category, mode, options at once. by Abby Yeh · 1 year, 10 months ago
- e1e8b20 Update WebRTC code version (2023-06-10T04:11:03). by webrtc-version-updater · 1 year, 11 months ago
- 4133797 Remove expired histograms WebRTC.PeerConnection.SrtpCryptoSuite by Johannes Kron · 1 year, 11 months ago
- c7695a5 Document how bitrate probing works from a RTP perspective by Philipp Hancke · 1 year, 11 months ago
- 48c44e3 Ensure RtpSenderEgress run on worker queue by Per K · 1 year, 11 months ago
- 2b5beb9 Set correct absolute send time on reordered packets by Per K · 1 year, 11 months ago
- 682755e Do not support frame tracking id extension in production by Philipp Hancke · 1 year, 11 months ago
- 5bcea25 Use version-less CIPD path for android_toolchain by Prashanth Swaminathan · 1 year, 11 months ago
- f781ff7 Update WebRTC code version (2023-06-09T04:02:47). by webrtc-version-updater · 1 year, 11 months ago
- cde5354 Implement DelayVariationCalculator for events analysis. by Rasmus Brandt · 1 year, 11 months ago
- f99e0f4 Remove stale Android NDK [2/2] by Prashanth Swaminathan · 1 year, 11 months ago
- 40ad4eb Roll chromium_revision a8db252505..a3756bb36c (1153825:1154916) by Prashanth Swaminathan · 1 year, 11 months ago
- 4ee5e5f Disable VideoCaptureTest due to flakyness by Björn Terelius · 1 year, 11 months ago
- 37fb647 Disable the roll of 'android_ndk' by Prashanth Swaminathan · 1 year, 11 months ago
- 36c945b Update WebRTC code version (2023-06-08T04:11:54). by webrtc-version-updater · 1 year, 11 months ago
- 9d9c3f4 [Analysis] Remove old threshold fields by Beining Chen · 1 year, 11 months ago
- 89f64b9 Make packet info optional and only set for primary packets in NetEq. by Jakob Ivarsson · 1 year, 11 months ago
- 9e639fa Migrate Android NDK to CIPD [1/2] by Prashanth Swaminathan · 1 year, 11 months ago
- fc260a18 Add method SetTimestamp in TransformableAudioFrameInterface by Palak Agarwal · 1 year, 11 months ago
- 3403acb av1: 8 threads for >720p and tiles config by Jerome Jiang · 1 year, 11 months ago
- d615704 Enable frame dropping in libaom AV1 encoder by Sergey Silkin · 1 year, 11 months ago
- a458fe5 Update WebRTC code version (2023-06-07T04:12:21). by webrtc-version-updater · 1 year, 11 months ago
- 09e0086 Remove ImplForTesting function from MediaChannel by Harald Alvestrand · 1 year, 11 months ago
- bd66cfe Roll chromium_revision a5cd053713..a8db252505 (1153688:1153825) by chromium-webrtc-autoroll · 1 year, 11 months ago
- 847208e Remove transitional shim classes by Harald Alvestrand · 1 year, 11 months ago
- ade07ca Rename current flexfec implementation flexfec_03 by Yosef Twaik · 1 year, 11 months ago
- 43df03d Fix spelling mistake ReplaceRemoteDescriptionAndCheckE*r*or by Philipp Hancke · 1 year, 11 months ago
- 6d25e96 Roll chromium_revision 404afa6a86..a5cd053713 (1153573:1153688) by chromium-webrtc-autoroll · 1 year, 11 months ago
- d3b71c7 Update WebRTC code version (2023-06-06T04:12:09). by webrtc-version-updater · 1 year, 11 months ago
- e00a12f Roll chromium_revision 96ad22527d..404afa6a86 (1153423:1153573) by chromium-webrtc-autoroll · 1 year, 11 months ago
- 8c4b9ea Remove references to AudioCodec and VideoCodec constructors by Florent Castelli · 1 year, 11 months ago
- fd096da Roll chromium_revision 8f3397a259..96ad22527d (1153256:1153423) by chromium-webrtc-autoroll · 1 year, 11 months ago
- 77c6230 Add create functions for voice media send and receive channels. by Harald Alvestrand · 1 year, 11 months ago
- be316da Ensure that RTCErrorOr<T, E> doesn't require T to be default constructible by Florent Castelli · 1 year, 11 months ago