- 1db921e Non-integer frame rate in Android HW encoder by Sergey Silkin · 2 years, 9 months ago
- efece42 Allow remote SDP offers to be "active" or "passive" by Harald Alvestrand · 2 years, 9 months ago
- 34cc986 [DVQA] Add detailed printout of when frame was dropped by Artem Titov · 2 years, 9 months ago
- c68796e Calculate frame timestamps based on target frame rate by Sergey Silkin · 2 years, 9 months ago
- 51238e6 Keep `transport_queue_safety_` alive until stopped permanently. by Tommi · 2 years, 9 months ago
- 1039392 Add a unittest to NetEq StatisticsCalculator for discarded packets counter. by Minyue Li · 2 years, 9 months ago
- 81f24c0 Add jakobi@webrtc.org to audio/OWNERS by saza · 2 years, 9 months ago
- 7a46cc5 Remove 3DES from WebRTC by David Benjamin · 2 years, 9 months ago
- 8d8bc62 Propogate already collected stats for frames in flight in DVQA to comparisons by Artem Titov · 2 years, 9 months ago
- 57d1310 A test to verify counter on discarded packet upon reordering. by Minyue Li · 2 years, 9 months ago
- cc69ea4 Fix parsing of vp9 skip level segmentation feature by Erik Språng · 2 years, 9 months ago
- 062acd9 Move frame drop functionality in VideoAdapter into a separate class. by Åsa Persson · 2 years, 9 months ago
- 0ca62e3 dcsctp: Avoid bundling FORWARD-TSN and DATA chunks by Victor Boivie · 2 years, 9 months ago
- 3bb74f3 Change VideoDecoderFactory::QueryCodecSupport to use reference_scaling by Johannes Kron · 2 years, 9 months ago
- d60b4ce Roll chromium_revision 5e754b1b6a..e35a3c7a8a (913170:913273) by chromium-webrtc-autoroll · 2 years, 9 months ago
- 37e2235 Roll chromium_revision 22b8bcea20..5e754b1b6a (913017:913170) by chromium-webrtc-autoroll · 2 years, 9 months ago
- 3ec9e03 dcsctp: Removing all references to unordered_map by Victor Boivie · 2 years, 9 months ago
- 51b96a7 Roll chromium_revision 680b7dae9d..22b8bcea20 (912910:913017) by chromium-webrtc-autoroll · 2 years, 9 months ago
- b6f19d7 Reland "Update remaining usage of VideoDecoder::InitDecode to Configure" by Danil Chapovalov · 2 years, 9 months ago
- d5a0efe Delete deprecated EncoderSimulcastProxy constructor by Danil Chapovalov · 2 years, 9 months ago
- 0347a08 Fix _hRecThread,_hPlayThread RTC_DCHECK reverse bug. by chengqian · 2 years, 9 months ago
- f135800 Roll chromium_revision 47dc8e2f50..680b7dae9d (912091:912910) by Mirko Bonadei · 2 years, 9 months ago
- 124889f Update WebRTC code version (2021-08-18T04:05:26). by webrtc-version-updater · 2 years, 9 months ago
- 9d8c3d9 Use separate queue for alive frames when self view is enabled in DVQA by Artem Titov · 2 years, 9 months ago
- e57a493 Reland "Rename vp9::FrameInfo to vp9::UncompressedHeader and add more fields." by Erik Språng · 2 years, 9 months ago
- 923d2c2 dcsctp: fixed grammar in one comment, added comment regarding the threading contract by Sergey Sukhanov · 2 years, 9 months ago
- d0b8879 Delete AsyncSocket class, merge into Socket class by Niels Möller · 2 years, 9 months ago
- 45b3e53 Improve webrtc fuzzer coverage of VP9 bitstream parser. by Erik Språng · 2 years, 9 months ago
- fb19596 Allow setting different number of temporal layers per simulcast layer. by Åsa Persson · 2 years, 9 months ago
- 29dddff usrsctp: Remove usage of usrsctp_getladdrs() by Florent Castelli · 2 years, 9 months ago
- 24e79f6 Add missing header (for unique_ptr). by Mirko Bonadei · 2 years, 9 months ago
- 1fdafae Calculate bitrate and frame rate mismatches in video codec tests by Sergey Silkin · 2 years, 9 months ago
- 773a222 red: enable opus-red by default by Philipp Hancke · 2 years, 9 months ago
- d6da4c2 Revert "Update remaining usage of VideoDecoder::InitDecode to Configure" by Mirko Bonadei · 2 years, 9 months ago
- 5bf0bb3 Enable WebRTC-UseStandardBytesStats in E2E tests by default. by Mirko Bonadei · 2 years, 9 months ago
- 14ef633 dcsctp: Don't send small packets when cwnd full by Victor Boivie · 2 years, 9 months ago
- ca0a08a Update remaining usage of VideoDecoder::InitDecode to Configure by Danil Chapovalov · 2 years, 9 months ago
- 82c3a6f Extract frames comparator out from DVQA by Artem Titov · 2 years, 9 months ago
- be9281b dcsctp: Increase cwnd by serialized chunk size by Victor Boivie · 2 years, 9 months ago
- a324950 Update WebRTC code version (2021-08-17T04:05:32). by webrtc-version-updater · 2 years, 9 months ago
- d912446 dcsctp: Refactor chunk acking by Victor Boivie · 2 years, 9 months ago
- abf6188 dcsctp: Add PacketSender by Victor Boivie · 2 years, 9 months ago
- 6b89130 Fix array_view nested namespace. by Mirko Bonadei · 2 years, 9 months ago
- ac09f0d Remove last traces of deferred sequencing. by Erik Språng · 2 years, 9 months ago
- ffce8e3 Migrate android video decoder wrapper from InitDecode to Configure by Danil Chapovalov · 2 years, 9 months ago
- 600bb8c dcsctp: Migrating to using absl::bind_front by Victor Boivie · 2 years, 9 months ago
- 8df32eb dcsctp: Add API to indicate packet send status by Victor Boivie · 2 years, 9 months ago
- 1921481 Fix some -Wunreachable-code-aggressive warnings by Nico Weber · 2 years, 9 months ago
- 9610671 Drop support for PipeWire 0.2 by Jan Grulich · 2 years, 9 months ago
- e1afe72 Roll chromium_revision 1d52d174cb..47dc8e2f50 (911982:912091) by chromium-webrtc-autoroll · 2 years, 9 months ago
- c617c33 Update WebRTC code version (2021-08-16T04:02:11). by webrtc-version-updater · 2 years, 9 months ago
- 750948b Pass dSYM when creating XCFramework only if dSYM exists by Byoungchan Lee · 2 years, 9 months ago
- ba0a306 Move check for number_of_cores parameter validitity by Danil Chapovalov · 2 years, 9 months ago
- 10ee27e Roll chromium_revision d76e910d24..1d52d174cb (911810:911982) by chromium-webrtc-autoroll · 2 years, 9 months ago
- 75f2228 Update WebRTC code version (2021-08-14T04:03:09). by webrtc-version-updater · 2 years, 9 months ago
- a7d32e3 Roll chromium_revision cce6e710fd..d76e910d24 (911687:911810) by chromium-webrtc-autoroll · 2 years, 9 months ago
- 54abf98 Remove the now unused non-deferred sequencing code path. by Erik Språng · 2 years, 9 months ago
- 355b8d2 Use VideoDecoder::Configure interface when setting up decoder by Danil Chapovalov · 2 years, 9 months ago
- b6bbdeb Allow RTP module thread checking to know PacketRouter status. by Erik Språng · 2 years, 9 months ago
- 05a9e5a Fix race in CallPerfTest.Bitrate_Kbps_PadsToMinTransmitBitrate by Niels Möller · 2 years, 9 months ago
- 95f6e8b Relax video_codec parameter for RtpVideoStreamReceiver::AddReceiveCodec by Danil Chapovalov · 2 years, 9 months ago
- b2745ba Condition frame pacing on min/max playout delay by Johannes Kron · 2 years, 9 months ago
- edfaaef Propagate socket write errors for DtlsTransport by Victor Boivie · 2 years, 9 months ago
- 55a2f77 Roll chromium_revision fcc4764793..cce6e710fd (911577:911687) by chromium-webrtc-autoroll · 2 years, 9 months ago
- 722a8a6 Tolerate more not completed measurements for CPD uploads by landrey · 2 years, 9 months ago
- dcaf1e7 Update WebRTC code version (2021-08-13T04:05:09). by webrtc-version-updater · 2 years, 9 months ago
- 1a292b7 Roll chromium_revision 8baabcff97..fcc4764793 (911441:911577) by chromium-webrtc-autoroll · 2 years, 9 months ago
- 04a9ce1 Roll chromium_revision ea375c193d..8baabcff97 (911333:911441) by chromium-webrtc-autoroll · 2 years, 9 months ago
- 0c2a9ca fix some typos by Philipp Hancke · 2 years, 9 months ago
- e0fb45c dcsctp: Add burst limiter for sent packets by Victor Boivie · 2 years, 9 months ago
- a9e7f71 Roll chromium_revision 3857505ab0..ea375c193d (911218:911333) by chromium-webrtc-autoroll · 2 years, 9 months ago
- d08930d Migrate test VideoDecoders to new VideoDecoder::Configure by Danil Chapovalov · 2 years, 9 months ago
- 40f7a5b Extract CPU measurer from DVQA by Artem Titov · 2 years, 9 months ago
- da6a9d5 Use GTEST_SKIP() instead of early return. by Mirko Bonadei · 2 years, 9 months ago
- aa918e0 Delete always-true BasicPortAllocatorSession::AddAllocatedPort argument by Niels Möller · 2 years, 9 months ago
- 69dd142 Register audio send stream in packet router on Start(). by Erik Språng · 2 years, 9 months ago
- 8b18304 Extract internal DVQA shared objects for future split by Artem Titov · 2 years, 9 months ago
- a17ec763 Extract various DVQA stats related objects into its own header by Artem Titov · 2 years, 9 months ago
- 2a6d48b Roll chromium_revision 28cd08bbe9..3857505ab0 (911167:911218) by chromium-webrtc-autoroll · 2 years, 9 months ago
- 5f1d406 Move legacy RtpRtcpModule to deferred sequencing. by Erik Språng · 2 years, 9 months ago
- 7fa3f40 Migrate software fallback wrapper to new VideoDecoder::Configure by Danil Chapovalov · 2 years, 9 months ago
- 647d326 Add tracking of video encoder/decoder used for stream in DVQA by Artem Titov · 2 years, 9 months ago
- 9ff59a3 Roll chromium_revision cf08193ade..28cd08bbe9 (911056:911167) by chromium-webrtc-autoroll · 2 years, 9 months ago
- a30819b Update WebRTC code version (2021-08-12T04:05:19). by webrtc-version-updater · 2 years, 9 months ago
- 44aa6c4 Roll chromium_revision c0a0d75300..cf08193ade (910939:911056) by chromium-webrtc-autoroll · 2 years, 9 months ago
- 8591eff Reland "Fix bug where we assume new m= sections will always be bundled." by Taylor Brandstetter · 2 years, 9 months ago
- 1c7ecef Reland "Modify Bundle logic to not add & destroy extra transport at add-track" by Taylor Brandstetter · 2 years, 9 months ago
- b7eb18d Roll chromium_revision a48436a40a..c0a0d75300 (910736:910939) by chromium-webrtc-autoroll · 2 years, 9 months ago
- 4f776ac Use make_ref_counted in AudioProcessingBuilder by Niels Möller · 2 years, 11 months ago
- 897bf9b Fix frame counters for peers added in the middle of the call by Artem Titov · 2 years, 9 months ago
- 6e2458d Add lock to guard rtp packet sequencer. by Erik Språng · 2 years, 9 months ago
- 8729d78 Delete AsyncSocketAdapter::Attach, make socket construction time const by Niels Möller · 2 years, 9 months ago
- a820cc2 Change signatures of SDP ParseFailed* functions by Niels Möller · 2 years, 9 months ago
- 9367cff red: change default redundancy level to 1 by Philipp Hancke · 2 years, 9 months ago
- 6144b84 red: fix renegotiation by Philipp Hancke · 3 years ago
- 221fa44 Roll chromium_revision 304823aaf0..a48436a40a (910626:910736) by chromium-webrtc-autoroll · 2 years, 9 months ago
- 10ed5f9 Increase sigslot internal pointer representation to 24 bytes. by Bjorn Terelius · 2 years, 9 months ago
- 53d4be2 Migrate software decoders to new VideoDecoder::Configure by Danil Chapovalov · 2 years, 9 months ago
- bf75041 Update stats_types.cc to use make_ref_counted. by Niels Möller · 2 years, 10 months ago
- 6c02c33 Add henrik.lundin as owner in audio/ by Henrik Lundin · 2 years, 9 months ago