- e99f687 Move WebRTC to non deprecated jsoncpp APIs. by Mirko Bonadei · 2 years, 3 months ago main
- 06b8f7e Move supported_platforms.md under g3doc/. by Mirko Bonadei · 2 years, 3 months ago
- e6ab520 First draft of WebRTC supported platforms/compilers documentation. by Mirko Bonadei · 2 years, 3 months ago
- 7208457 Same length for all ARM64 platforms Update more audio checksums for M1 by Christoffer Jansson · 2 years, 3 months ago lkgr
- 74543b7 PlatformThreadTest: fix flake. by Markus Handell · 2 years, 3 months ago
- 2b3a10e Add MAC arm64 platform and update checksums for acm unittest by Christoffer Jansson · 2 years, 3 months ago
- 5e82c75 Remove TODOs to remove SetAudioPlayback and SetAudioRecording by Harald Alvestrand · 2 years, 3 months ago
- 482b7c0 Fix -Wimplicit-int-float-conversions. by Peter Kasting · 2 years, 3 months ago
- 64851c0 Reland: Fix echo return loss stats and add to RTCAudioSourceStats. by Taylor Brandstetter · 2 years, 3 months ago
- a987429 Update WebRTC code version (2021-06-25T04:04:09). by webrtc-version-updater · 2 years, 3 months ago
- c830bd6 Remove ssl_certificate() accessor. by Harald Alvestrand · 2 years, 3 months ago
- e9a74c9 Public RtpVideoFrameAssembler by philipel · 2 years, 3 months ago
- 4e51334 AV1 OBU test helper. by philipel · 2 years, 3 months ago
- 28e582d Removing RTC_SUPPORTS_METAL compilation flag. This flag is a holdover from before either macOS or the iOS Simulator supported Metal rendering. by Jake Bromberg · 2 years, 3 months ago
- f2ed401 Fix unscaled timestamps passed to nack_tracker by Jared Siskin · 2 years, 3 months ago
- 9233af3 Update dependencies on deprecated target rtc_base:critical_section by Niels Möller · 2 years, 3 months ago
- 0742f52 Triggering build after flaky builders (asan). by Tommi · 2 years, 3 months ago
- eb61b7f ModuleRtcRtcpImpl2: remove Module inheritance. by Markus Handell · 2 years, 3 months ago
- 6e65f6a Deprecating AbsoluteCaptureTimeReceiver by Minyue Li · 2 years, 3 months ago
- 3f7b717 RTCPSender: remove compatibility ctor & method. by Markus Handell · 2 years, 3 months ago
- 49cb459 TaskQueueStdlib: initialize the thread last. by Markus Handell · 2 years, 3 months ago
- 0fe60bd Add RecursiveCriticalSection to the don't-use list of primitives by Harald Alvestrand · 2 years, 3 months ago
- c413c55 Replace use of RecursiveCriticalSection in VirtualSocketServer by Niels Möller · 2 years, 3 months ago
- fe6580f Revert "Fix echo return loss stats and add to RTCAudioSourceStats." by Evan Shrubsole · 2 years, 3 months ago
- 9e2b315 Minor code cleanup of WebRtcVideoReceiveStream. by Tommi · 2 years, 3 months ago
- 885d538 ModuleRtpRtcpImpl2: remove RTCP send polling. by Markus Handell · 2 years, 3 months ago
- 2086209 Update WebRTC code version (2021-06-22T04:05:30). by webrtc-version-updater · 2 years, 3 months ago
- 049ed44 ModuleRtpRtcpImpl2: update test code. by Markus Handell · 2 years, 3 months ago
- fb7fd24 Removing RTC_SUPPORTS_METAL compilation flag. This flag is a holdover from before either macOS or the iOS Simulator supported Metal rendering. by Jake Bromberg · 2 years, 3 months ago
- c6b9ac7 RTCPSender: migrate to Timestamp. by Markus Handell · 2 years, 3 months ago
- e2ab77b Reland "Port: migrate to TaskQueue." by Markus Handell · 2 years, 3 months ago
- a27cfbf Fix echo return loss stats and add to RTCAudioSourceStats. by Taylor Brandstetter · 2 years, 3 months ago
- 2e3edc1 RTCPSender: migrate to own configuration struct. by Markus Handell · 2 years, 3 months ago
- f906ec4 Handle null return from ToI420 in encoders by Evan Shrubsole · 2 years, 3 months ago
- 76a35d9 Delete legacy RtpHeaderParser wrapper by Danil Chapovalov · 2 years, 3 months ago
- 257f81b Update VirtualSocketServer locking to match documentation. by Niels Möller · 2 years, 3 months ago
- a4aabb9 Revert "Port: migrate to TaskQueue." by Markus Handell · 2 years, 3 months ago
- 0654016 Port: migrate to TaskQueue. by Markus Handell · 2 years, 3 months ago
- 3f6efdc Update WebRTC code version (2021-06-21T04:05:45). by webrtc-version-updater · 2 years, 3 months ago
- ae278d4 openssl_adapter: document SSL_CTX_set_verify_depth behaviour by Philipp Hancke · 2 years, 3 months ago
- fbe9958 Update WebRTC code version (2021-06-20T04:03:02). by webrtc-version-updater · 2 years, 3 months ago
- 4cacaf7 Update WebRTC code version (2021-06-19T04:03:03). by webrtc-version-updater · 2 years, 3 months ago
- 7c719b0 Fixes off-by-one error in video capture module by Johannes Kron · 2 years, 3 months ago
- bad0ab0 Delete unused class MockDelayable by Niels Möller · 2 years, 3 months ago
- c6d7648 Add jakobi to modules/audio_coding OWNERS by Ivo Creusen · 2 years, 3 months ago
- 6a11c84 dcsctp: Add DcSctpSocketFactory by Florent Castelli · 2 years, 3 months ago
- c20f156 dcsctp: Don't sent more packets before COOKIE ACK by Victor Boivie · 2 years, 3 months ago
- 95c3041 Update WebRTC code version (2021-06-18T04:03:27). by webrtc-version-updater · 2 years, 3 months ago
- 42dacda AGC analog clipping predictor: integrate evaluator by Alessio Bazzica · 2 years, 3 months ago
- 7d54182 Avoid assembling complicated but unused video rtp header extensions by Danil Chapovalov · 2 years, 3 months ago
- afb2811 Catch possible `RuntimeException` from `getCameraCharacteristics` by Xavier Lepaul · 2 years, 3 months ago
- 11b92cf Refactoring: Move groups-by-mid into Bundle manager by Harald Alvestrand · 2 years, 3 months ago
- de22ab2 Apply IWYU to jsep_transport_controller/collection by Harald Alvestrand · 2 years, 3 months ago
- d354ced Mark VideoSendTiming flags as invalid by default. by philipel · 2 years, 3 months ago
- ada810a Reland "Deprecate microsecond timestamps in RTC event log." by Björn Terelius · 2 years, 3 months ago
- 1bb36d2 Change YuvConverter.convert to catch GLExceptions and return null. by Fabian Bergmark · 2 years, 3 months ago
- ac82bd3 Add timestamp to log message in generic_decoder.cc by Johannes Kron · 2 years, 3 months ago
- 41c700d Remove unnused build configs for M1 builder by Christoffer Jansson · 2 years, 3 months ago
- 82f21fd Make WebRtcAudioReceiveStream::stream_ const. by Tommi · 2 years, 3 months ago
- b4100ad Avoid using legacy rtp parser in neteq test::Packet by Danil Chapovalov · 2 years, 3 months ago
- 35b21ba In RtcpTransceiver avoid extra PostTask during construction by Danil Chapovalov · 2 years, 3 months ago
- f9d5e55 Revert "Avoid video stream allocation on configuration change after timeout." by Jakob Ivarsson · 2 years, 3 months ago
- a3796c8 Revert the send-side bwe behavior to slow ramp-up on lifted REMB cap. by Christoffer Rodbro · 2 years, 3 months ago
- ce3b3ba Update WebRTC code version (2021-06-17T04:05:50). by webrtc-version-updater · 2 years, 3 months ago
- 4b62952 Roll chromium_revision 6ade74989a..6f7025c98c (893176:893293) by chromium-webrtc-autoroll · 2 years, 3 months ago
- e0c7365 Roll chromium_revision 19c2bebe7d..6ade74989a (893060:893176) by chromium-webrtc-autoroll · 2 years, 3 months ago
- a2a073b Reformat pc/g3doc/rtp.md by Artem Titov · 2 years, 3 months ago
- 55107c8 Update the sync_group id without recreating audio receive streams. by Tommi · 2 years, 3 months ago
- 25029c4 Roll chromium_revision b452ca696d..19c2bebe7d (892948:893060) by chromium-webrtc-autoroll · 2 years, 3 months ago
- 355c473 Fix VideoRtpDepacketizerVp{8,9} copy assignment signature. by philipel · 2 years, 3 months ago
- 5b9d0c7 AGC1 add clipping predictor evaluator by Alessio Bazzica · 2 years, 3 months ago
- 808f494 LOG DTLS (failed) handshake retransmission by Jonas Oreland · 2 years, 3 months ago
- d579e6b dcsctp: Do explicit bounds checking in bounded IO by Victor Boivie · 2 years, 3 months ago
- 72b7998 Remove the `createDecoder(String)` overload by Xavier Lepaul · 2 years, 3 months ago
- 130e031 Roll chromium_revision 570a173256..b452ca696d (892156:892948) by chromium-webrtc-autoroll · 2 years, 3 months ago
- 98ff028 AGC analog ClippingPredictor refactoring 2/2 by Alessio Bazzica · 2 years, 3 months ago
- 08be9ba Don't recreate the audio receive stream when updating the local_ssrc. by Tommi · 2 years, 3 months ago
- bc03259 Define generate_location_tags gn arg by Björn Terelius · 2 years, 3 months ago
- 6a0a559 Reland "Correctly handle retransmissions/padding in early loss detection." by Erik Språng · 2 years, 3 months ago
- c03d6e9 Support Java_Buffer_toI420 returning null by Fabian Bergmark · 2 years, 3 months ago
- cd430c8 Update WebRTC code version (2021-06-16T04:05:58). by webrtc-version-updater · 2 years, 3 months ago
- d6957c2 Revert "Correctly handle retransmissions/padding in early loss detection." by Erik Språng · 2 years, 3 months ago
- e9ae472 Correctly handle retransmissions/padding in early loss detection. by Erik Språng · 2 years, 3 months ago
- e3ceb88 Sanitize hostname literals when mDNS obfuscation is on. by Harald Alvestrand · 2 years, 3 months ago
- be53049 Reland "Avoid sending empty receiver reports with RtcpTransceiver" by Danil Chapovalov · 2 years, 3 months ago
- 7a2db8a Modify Bundle logic to not add & destroy extra transport at add-track by Harald Alvestrand · 2 years, 3 months ago
- e4eb8af libstdc++: fix ostream operator<< usage in JsepTransportCollection by Stephan Hartmann · 2 years, 3 months ago
- 07bf5b5 Update WebRTC code version (2021-06-15T04:04:38). by webrtc-version-updater · 2 years, 3 months ago
- 3008bcd Don't recreate audio receive streams on header extension update. by Tommi · 2 years, 3 months ago
- 6bbe1a4 Roll chromium_revision e9261a56ad..570a173256 (892013:892156) by chromium-webrtc-autoroll · 2 years, 3 months ago
- d350006 Add rtp_config() accessor to ReceiveStream. by Tommi · 2 years, 3 months ago
- 48420fa Revert "Avoid sending empty receiver reports with RtcpTransceiver" by Björn Terelius · 2 years, 3 months ago
- 1c1f540 Factor out common receive stream methods to a common interface. by Tommi · 2 years, 3 months ago
- e097282 Avoid recreating the audio stream when a frame decryptor is set. by Tommi · 2 years, 3 months ago
- e5f1a39 Avoid sending empty receiver reports with RtcpTransceiver by Danil Chapovalov · 2 years, 3 months ago
- 8b69290 Fix VideoStreamEncoder QP tests to not use SetHasInternalSource by Niels Möller · 2 years, 3 months ago
- b237a87 AGC analog ClippingPredictor refactoring 1/2 by Alessio Bazzica · 2 years, 3 months ago
- 1ff491b Roll chromium_revision 8907aace7e..e9261a56ad (891631:892013) by chromium-webrtc-autoroll · 2 years, 3 months ago
- 74cc9ea Don't register invalid encode complete callbacks. by Peter Hanspers · 2 years, 3 months ago
- 1081487 Avoid video stream allocation on configuration change after timeout. by Jakob Ivarsson · 2 years, 3 months ago