external/webrtc.git -

Age	Commit message (Expand)	Author
2020-07-07	Migrate media/ to webrtc::Mutex.	Markus Handell
2020-07-07	Relanding: Fix data channel message integrity violation	Taylor Brandstetter
2020-07-01	Revert "Fix data channel message integrity violation"	Alessio Bazzica
2020-07-01	Fix data channel message integrity violation	Lennart Grahl
2020-06-25	Add RtpEncodingParameters.adaptive_ptime.	Jakob Ivarsson
2020-06-17	Delete MediaTransportInterface and DatagramTransportInterface	Niels Möller
2020-06-08	Use absl_deps in order to preapre to the Abseil component build release.	Mirko Bonadei
2020-04-16	Remove WebRTC-ExperimentalScreenshareSettings.	Mirko Bonadei
2020-04-02	Add AV1 to default video encoder factory	Danil Chapovalov
2020-03-29	Reland "Distinguish between send and receive codecs"	Johannes Kron
2020-03-16	RtpTransceiverInterface: add header_extensions_to_offer()	Markus Handell
2020-03-11	Add Av1 Decoder wrapper behind a build flag	Danil Chapovalov
2020-03-05	Implement new specification for degradation preference	Florent Castelli
2020-02-28	Insert frame transformer between Encoded and Packetizer.	Marina Ciocea
2020-02-11	Use I420Buffer::ScaleFrom to clean scaling in SimulcastEncoderAdapter	Evan Shrubsole
2020-01-21	Reformat GN files.	Mirko Bonadei
2019-12-19	Parse max-fr and max-fs from SDP FMTP line	Johannes Kron
2019-12-19	Signal requested resolution alignment requirements from sinks to sources.	Rasmus Brandt
2019-12-04	WebRtcVideoEngine: Enable encoded frame sink.	Markus Handell
2019-12-03	Export more symbols to fix Chromecast component build	Ken MacKay
2019-12-03	Trials should always be populated in call config.	Erik Språng
2019-11-26	Delete media transport integration.	Bjorn A Mellem
2019-11-22	Nuke p2p/base/stun.h	Patrik Höglund
2019-11-18	Move stun.h to api/.	Patrik Höglund
2019-11-04	Stop checking VP8BaseHeavyTl3RateAllocation field trial on every frame.	Rasmus Brandt
2019-10-25	Added usrsctp reliablitiy stress test.	Yura Yaroshevich
2019-10-21	Remove apm_helpers, consolidate audio config in WebRtcVoiceEngine	Sam Zackrisson
2019-10-18	Add option to make first scale factor depend on input resolution.	Åsa Persson
2019-10-17	Use source_sets in component builds and static_library in release builds.	Mirko Bonadei
2019-10-15	Remove AudioProcessing::gain_control() getter	Sam Zackrisson
2019-10-11	Extend WebRTC-Video-MinVideoBitrate to experiment per-codec	Elad Alon
2019-10-11	Clarify expectation on GlobalLock	Danil Chapovalov
2019-10-01	SimulcastEncoderAdapter: support per layer fallback and single encoder proxying	Erik Språng
2019-09-19	New build target api:media_interface	Niels Möller
2019-09-18	Remove api/bitrate_constraints.h.	Mirko Bonadei
2019-09-17	Use std::make_unique instead of absl::make_unique.	Mirko Bonadei
2019-09-13	Introduce api/crypto/BUILD.gn.	Mirko Bonadei
2019-09-13	Move MediaTransportInterface out of the libjingle_peerconnection_api target	Niels Möller
2019-09-11	Move rtc_error.{h,cc} to its own build target.	Mirko Bonadei
2019-09-06	Reland "New build target p2p:stun_types"	Niels Möller
2019-09-06	Revert "New build target p2p:stun_types"	Hannes Landeholm
2019-09-03	Buffers non atomic message send with usrsctp lib.	Seth Hampson
2019-09-02	Split out RtpSource from libjingle_peerconnection_api	Niels Möller
2019-09-02	New build target p2p:stun_types	Niels Möller
2019-08-29	Make the RtpHeaderParserImpl available to tests and tools only.	Tommi
2019-08-29	Delete unneeded dependencies on libjingle_peerconnection_api	Niels Möller
2019-08-29	New target for api/rtp_parameters.h and api/media_types.h.	Niels Möller
2019-08-07	Delete deprecated rtc_event_log header	Danil Chapovalov
2019-08-07	Removes unused AudioAllocationSettings from voice engine.	Sebastian Jansson
2019-08-05	Wire up field trials for some experimental screenshare settings	Erik Språng