- 723b35f Delete legacy function to deregister rtp header extension by type by Danil Chapovalov · 3 years, 7 months ago
- 83121d4 Propagate scalability mode in CreateSimulcastOrConferenceModeScreenshareStreams. by philipel · 3 years, 7 months ago
- eb42ab7 Dont use simulcast for AV1. by philipel · 3 years, 7 months ago
- 52d97fd Both build options for symbol export can be specified by Takaaki Suzuki · 3 years, 7 months ago
- 7e7e805 video: Re-configure scalers when encoder info changed. by Shuhai Peng · 3 years, 7 months ago
- 49c59ca Fix missing return value(2) by Björn Terelius · 3 years, 7 months ago
- 294f5bc testing code-owners plugin. by Christoffer Jansson · 3 years, 7 months ago
- 5df7b47 Fix missing return value in rtc_event_field_parser by Björn Terelius · 3 years, 7 months ago
- 88ae708 Roll chromium_revision e2c8b6114c..2895dbef35 (925965:929709) by Björn Terelius · 3 years, 7 months ago
- 585aad7 AGC2: VadWithLevel -> VoiceActivityDetectorWrapper 1/2 by Alessio Bazzica · 3 years, 7 months ago
- 5a79d28 Require 16x16 alignment when using HardwareVideoEncoder for encoding. by Jiwon Jung · 3 years, 7 months ago
- 3179fb2 dcsctp: Avoid integer overflow in HEARTBEAT-ACK v2 by Victor Boivie · 3 years, 7 months ago
- 2bcdb5d Remove phoglund from ENG_REVIEW_OWNERS by Christoffer Jansson · 3 years, 7 months ago
- dd410c0 Update WebRTC code version (2021-10-08T04:04:59). by webrtc-version-updater · 3 years, 7 months ago
- 7284bd4 Use GCD instead of Detached Thread in Async Resolver when on MacOS/iOS by Byoungchan Lee · 3 years, 7 months ago
- 40abb7d Default the behavior allowing fast rampup when REMB cap is lifted. by Christoffer Rodbro · 3 years, 7 months ago
- 5c3ae49 AudioFrameView: size_t -> int by Alessio Bazzica · 3 years, 7 months ago
- 82ea4ee AGC2 refactoring: better names for `GainController2` members by Alessio Bazzica · 3 years, 7 months ago
- 9b2a746 Use fallback encoder if primary can't be created by Sergey Silkin · 3 years, 7 months ago
- bcef6e1 Update WebRTC code version (2021-10-07T04:02:51). by webrtc-version-updater · 3 years, 7 months ago
- 5c7d5c9 AudioProcessingImpl: Move analog gain change check by Hanna Silen · 3 years, 7 months ago
- bde89ab win: Disable deprecation warning for one call of GetVersionEx by Nico Weber · 3 years, 7 months ago
- fc5a4f7 Revert "Use AsyncDnsResolver API in bindings and tests" by Björn Terelius · 3 years, 7 months ago
- 3695640 PipeWire capturer: copy content from PW buffer directly to DesktopFrame by Jan Grulich · 3 years, 7 months ago
- 6d19d14 Add AsyncListenSocket, as alias for AsyncPacketSocket by Niels Möller · 3 years, 7 months ago
- a057760 Use AsyncDnsResolver API in bindings and tests by Harald Alvestrand · 3 years, 7 months ago
- 408e4da Pipewire: Do not typecheck the portal session_handle by Robert Mader · 3 years, 7 months ago
- e1e05af Reland "PipeWire capturer: implement proper DMA-BUFs support"" by Jan Grulich · 3 years, 7 months ago
- 404cd60 Fix weird socket member naming in AsyncStunTCPSocketTest by Niels Möller · 3 years, 7 months ago
- 04c911c Revert "Turn on WebRTC-TaskQueuePacer by defualt." by Erik Språng · 3 years, 7 months ago
- 2bfa5b2 Default sending capture clock offset in abs-capture-time header extension. by Minyue Li · 3 years, 7 months ago
- c9f43f8 Use AsyncDnsResolver in TurnPort class by Harald Alvestrand · 3 years, 7 months ago
- b7b306b Use AsyncDnsResolver in UDPPort class by Harald Alvestrand · 3 years, 7 months ago
- 79bd4f1 win: Consolidate on a single version checking API by Nico Weber · 3 years, 7 months ago
- b251145 Turn on WebRTC-TaskQueuePacer by defualt. by Erik Språng · 3 years, 7 months ago
- 7b80d44 AGC2: SIMD allowed config flags to field trials by Alessio Bazzica · 3 years, 7 months ago
- 41bbc3d Fix bug in dynamic pacer causing slightly inaccurate pacing rate. by Erik Språng · 3 years, 7 months ago
- b8ffdc4 APM: fix CaptureLevelAdjustment::operator== by Alessio Bazzica · 3 years, 7 months ago
- 958772e Update WebRTC code version (2021-10-05T04:03:48). by webrtc-version-updater · 3 years, 7 months ago
- a850e6c AGC2 config: allow tuning of headroom, max gain and initial gain by Alessio Bazzica · 3 years, 7 months ago
- 41b4397 Use accumulate to calculate recv_delta_size by Vojin Ilic · 3 years, 7 months ago
- 606d3cb VideoStreamEncoderTest: Use DataRate for some constants. by Asa Persson · 3 years, 7 months ago
- c895601 Update WebRTC code version (2021-10-04T04:04:13). by webrtc-version-updater · 3 years, 7 months ago
- 75d0de3 Roll src/third_party/libjpeg_turbo/ ff19e5b2e..49836d72b (1 commit) by Keiichi Enomoto · 3 years, 7 months ago
- cc99299 Remove use_xcode_clang=true from iOS packaging script. by Mirko Bonadei · 3 years, 7 months ago
- 54c90f2 [-Wshadow] - Fix some warnings. by Mirko Bonadei · 3 years, 7 months ago
- e3d26f5 Update WebRTC code version (2021-10-03T04:05:03). by webrtc-version-updater · 3 years, 7 months ago
- db94869 Make CroppedWindowCapturer more resilient by Ilya Nikolaevskiy · 3 years, 7 months ago
- 985310e Add CreateAsyncDnsResolver to PacketSocketFactory API by Harald Alvestrand · 3 years, 7 months ago
- ae566cd audio/red: provide default fmtp line by Philipp Hancke · 3 years, 7 months ago
- 1d4aa36 Allow encoding string fields in new event log format. by Björn Terelius · 3 years, 7 months ago
- ba4d870 Update WebRTC code version (2021-10-01T04:03:55). by webrtc-version-updater · 3 years, 7 months ago
- 5755f3e dcsctp: Add sequence checker to socket by Victor Boivie · 3 years, 7 months ago
- fe903d5 Add encoding for numeric RTC event fields. by Björn Terelius · 3 years, 7 months ago
- a654e07 Eliminate a temporary std::string in ParsedFailed helper. by Niels Möller · 3 years, 7 months ago
- 82ccdd3 dcsctp: Add network/throughput tests by Victor Boivie · 3 years, 7 months ago
- ee4c930 ice: fix comment about relay preference by Philipp Hancke · 3 years, 7 months ago
- d4aa3a3 Use absl::string_view in SDP-related utilities by Niels Möller · 3 years, 7 months ago
- 8afd22f Update WebRTC code version (2021-09-30T04:03:50). by webrtc-version-updater · 3 years, 7 months ago
- f270770 video: Implement bandwidth based scaler by Shuhai Peng · 3 years, 7 months ago
- 23bfff3 Change default parameters for the low-latency video pipeline by Johannes Kron · 3 years, 7 months ago
- aa37316 Pass a SocketFactory to BasicNetworkManager constructor by Niels Möller · 3 years, 7 months ago
- 15a0c88 dcsctp: Ensure callbacks are always triggered by Victor Boivie · 3 years, 7 months ago
- 0081f1c dcsctp: Refactor CallbackDeferrer by Victor Boivie · 3 years, 7 months ago
- 43651f5 AdaptiveDigitalGainApplierTest parametric test fixed by Alessio Bazzica · 3 years, 7 months ago
- 8581f3d Update WebRTC code version (2021-09-29T04:04:04). by webrtc-version-updater · 3 years, 7 months ago
- 79326ea Implement missing candidate pair packets/bytes sent/received stats. by Taylor Brandstetter · 3 years, 7 months ago
- cc4deae Roll chromium_revision 6c17347f8a..e2c8b6114c (925841:925965) by chromium-webrtc-autoroll · 3 years, 7 months ago
- 2eef9c9 Roll chromium_revision 232a45ecaf..6c17347f8a (925740:925841) by chromium-webrtc-autoroll · 3 years, 7 months ago
- 47126fa Force -Wno-shadow to avoid variable shadowing warnings. by Peter Kasting · 3 years, 7 months ago
- 5da581b AGC2: use only one headroom parameter by Alessio Bazzica · 3 years, 7 months ago
- 355495a Roll chromium_revision cb9ab6b9e1..232a45ecaf (925640:925740) by chromium-webrtc-autoroll · 3 years, 7 months ago
- 7145a14 red: fix fmtp payload type collision handling by Philipp Hancke · 3 years, 7 months ago
- 4a1c2c4 Delete wiring of SignalAddressReady for TCP ports by Niels Möller · 3 years, 7 months ago
- 2f40988 Roll chromium_revision ac131c0a6e..cb9ab6b9e1 (925516:925640) by chromium-webrtc-autoroll · 3 years, 7 months ago
- 5377267 Update WebRTC code version (2021-09-28T04:02:20). by webrtc-version-updater · 3 years, 7 months ago
- f4fa166 dcsctp: Detect the peer SCTP implementation by Victor Boivie · 3 years, 7 months ago
- 1180689 Roll chromium_revision f1f05ac6a6..ac131c0a6e (925390:925516) by chromium-webrtc-autoroll · 3 years, 7 months ago
- 3efea37 Roll chromium_revision 714d043f97..f1f05ac6a6 (925247:925390) by chromium-webrtc-autoroll · 3 years, 7 months ago
- 8fb41a3 Add Direction indicator to TransformableFrames by Tony Herre · 3 years, 7 months ago
- 6ee9734 AGC2: update adaptive digital test by Alessio Bazzica · 3 years, 7 months ago
- 8d9395d Roll chromium_revision efa07aa310..714d043f97 (925137:925247) by chromium-webrtc-autoroll · 3 years, 7 months ago
- acf4f55 Delete unused FifoBuffer methods by Niels Möller · 3 years, 7 months ago
- 1ac4f2a2 AGC2: Remove unused parameters by Alessio Bazzica · 3 years, 7 months ago
- f75f9c2 dcsctp: Avoid integer overflow in HEARTBEAT-ACK by Victor Boivie · 3 years, 7 months ago
- 29b4049 VideoStreamEncoder: Remove check for zero VideoCodec.maxBitrate. by Åsa Persson · 3 years, 7 months ago
- be4b792 Update WebRTC code version (2021-09-27T04:05:06). by webrtc-version-updater · 3 years, 7 months ago
- adf6a9a Roll chromium_revision bd339cd24d..efa07aa310 (925034:925137) by chromium-webrtc-autoroll · 3 years, 7 months ago
- 96168ab Update WebRTC code version (2021-09-26T04:03:45). by webrtc-version-updater · 3 years, 7 months ago
- b6a73b6 Roll chromium_revision e0a13a7aa6..bd339cd24d (924934:925034) by chromium-webrtc-autoroll · 3 years, 7 months ago
- e86431b Update WebRTC code version (2021-09-25T04:04:40). by webrtc-version-updater · 3 years, 7 months ago
- 96f2069 Roll chromium_revision 716ac22917..e0a13a7aa6 (924783:924934) by chromium-webrtc-autoroll · 3 years, 7 months ago
- 9f0b333 Test LKGR finder. by Mirko Bonadei · 3 years, 7 months ago
- 889ffa4 Roll chromium_revision ca31e240dc..716ac22917 (924631:924783) by chromium-webrtc-autoroll · 3 years, 7 months ago
- f95f534 Delete deprecated kUri constants for rtp header extensions by Danil Chapovalov · 3 years, 7 months ago
- 2938694 AudioProcessing: Add use_predicted_step in GainController1 comparison and string conversion by Hanna Silen · 3 years, 7 months ago
- 8970b49 AgcManagerDirect: Add histograms for clipping prediction precision and recall by Hanna Silen · 3 years, 7 months ago
- 4275448 Use new CopyOnWriteBuffer ctor/append function in H264PacketBuffer. by philipel · 3 years, 7 months ago
- fdea8e6 Roll chromium_revision 05ebbf1d76..ca31e240dc (924489:924631) by chromium-webrtc-autoroll · 3 years, 7 months ago
- ec56775 Roll chromium_revision e1ce489c98..05ebbf1d76 (924380:924489) by chromium-webrtc-autoroll · 3 years, 7 months ago