- a104ceb Revert "Reland "Reland "Distinguish between send and receive codecs""" by Johannes Kron · 4 years, 10 months ago
- 9bac68c Reland "Reland "Distinguish between send and receive codecs"" by Johannes Kron · 4 years, 10 months ago
- 760fd52 Replace MockAudioDeviceModule mock refcounting with real refcounting by Steve Anton · 4 years, 10 months ago
- 00a3087 Revert "Reland "Distinguish between send and receive codecs"" by Johannes Kron · 4 years, 10 months ago
- 897776e Pass SDP video parameters to encoder. by Sergey Silkin · 4 years, 10 months ago
- 133bf2b Reland "Distinguish between send and receive codecs" by Johannes Kron · 4 years, 10 months ago
- 1acdc74 Split up EncoderStreamFactory::CreateEncoderStreams in two. by Rasmus Brandt · 4 years, 10 months ago
- 43bfe0b Enforce VideoEncoderConfig.num_temporal_layers >= 1. by Rasmus Brandt · 4 years, 10 months ago
- ccbe95f Reformat GN files. by Mirko Bonadei · 4 years, 10 months ago
- e57b266 Revert "Distinguish between send and receive codecs" by Steve Anton · 4 years, 10 months ago
- c0f25cf Distinguish between send and receive codecs by Johannes Kron · 4 years, 10 months ago
- b2b2031 Concatenate string literals at compile time. by Jonas Olsson · 4 years, 10 months ago
- 1546f99 Fixed timeout overflow in sctp reliability test. by Yura Yaroshevich · 4 years, 10 months ago
- fae6400 Add saza@ and peah@ to OWNERS of some audio files by Sam Zackrisson · 4 years, 10 months ago
- f5ecb5f Revert "Reland "Reland "Reland "Distinguish between send and receive video codecs"""" by Mirko Bonadei · 4 years, 10 months ago
- 9cad4dc Reland "Reland "Reland "Distinguish between send and receive video codecs""" by Johannes Kron · 5 years ago
- 0e3a3f6 Adding deadbeef to sctp/OWNERS and removing myself. by Seth Hampson · 4 years, 10 months ago
- 27064ad SimulcastEncoderAdapter: In passthrough mode set correct lenght for frame_types parameter by Ilya Nikolaevskiy · 4 years, 10 months ago
- 4db28b5 Cleanup: Removes redundant includes on message_queue.h by Sebastian Jansson · 4 years, 10 months ago
- 873610c Fix updating degradation preference in SetRtpParameters. by Mirta Dvornicic · 4 years, 10 months ago
- b5159fe Revert "Reland "Reland "Distinguish between send and receive video codecs""" by Olga Sharonova · 4 years, 11 months ago
- 4e64e60 Reland "Reland "Distinguish between send and receive video codecs"" by Johannes Kron · 5 years ago
- c8f3134 Parse max-fr and max-fs from SDP FMTP line by Johannes Kron · 4 years, 11 months ago
- 5cad55b Signal requested resolution alignment requirements from sinks to sources. by Rasmus Brandt · 4 years, 11 months ago
- 6fd58b3 Add maxFramerate support to SimulcastEncoderAdapter by Florent Castelli · 5 years ago
- f9d92ed Revert "Reland "Distinguish between send and receive video codecs"" by Ilya Nikolaevskiy · 5 years ago
- 2697ac1 Stop an SCTP connection when the DTLS transport closes. by Harald Alvestrand · 5 years ago
- 77eb338 Reland "Distinguish between send and receive video codecs" by Johannes Kron · 5 years ago
- f2d6fe6 Revert "Reland "Distinguish between send and receive video codecs"" by Johannes Kron · 5 years ago
- 26e6afe Reland "Distinguish between send and receive video codecs" by Johannes Kron · 5 years ago
- 977b265 Reduce some logging at INFO level by moving log statements by Harald Alvestrand · 5 years ago
- dcb4fcc Execute cached video encoder switching request if encoder switching is allowed after the switch request was made. by philipel · 5 years ago
- ded86c1 Remove remaining settings for using legacy AEC by Per Åhgren · 5 years ago
- f22af3c Revert "Distinguish between send and receive video codecs" by Johannes Kron · 5 years ago
- 18314bd Distinguish between send and receive video codecs by Johannes Kron · 5 years ago
- 62ea0aa Remove deprecated legacy AEC code by Per Åhgren · 5 years ago
- 33f9d2b Migrate WebRTC on FrameGeneratorInterface and remove FrameGenerator class by Artem Titov · 5 years ago
- 907dc80 Reland "Add support for RtpEncodingParameters::max_framerate" by Florent Castelli · 5 years ago
- 749f660 Enable SSRC 0 in MediaChannel methods by Saurav Das · 5 years ago
- 32565f6 WebRtcVideoEngine: Enable encoded frame sink. by Markus Handell · 5 years ago
- 41462d5 Always keep abs send time extension. by Sebastian Jansson · 5 years ago
- 934afc6 Deprecate RtpReceiver's SetParameters method by Saurav Das · 5 years ago
- 831ce5f Export more symbols to fix Chromecast component build by Ken MacKay · 5 years ago
- 269ac81 VideoReceiveStream: Enable encoded frame sink. by Markus Handell · 5 years ago
- 014dd3c Trials should always be populated in call config. by Erik Språng · 5 years ago
- 7968530 Removes caching SimulcastEncoderAdapter::GetEncoderInfo() by Erik Språng · 5 years ago
- 5cef9c3 Revert "Add support for RtpEncodingParameters::max_framerate" by Florent Castelli · 5 years ago
- 7a9a092 Delete media transport integration. by Bjorn A Mellem · 5 years ago
- 15be528 Add support for RtpEncodingParameters::max_framerate by Florent Castelli · 5 years ago
- 00376e1 Add totalInterFrameDelay to RTCInboundRTPStreamStats by Johannes Kron · 5 years ago
- e43b531 Nuke p2p/base/stun.h by Patrik Höglund · 5 years ago
- 108a2f0 Preventively fix missing braces warnings. by Mirko Bonadei · 5 years ago
- 287e464 Change VideoAdapter::OnResolutionFramerateRequest to VideoAdapter::OnSinkWants by Rasmus Brandt · 5 years ago
- 56d9452 Move stun.h to api/. by Patrik Höglund · 5 years ago
- a7a2ab4 Remove dead kDummyVideoSsrc and FPS_TO_INTERVAL from video_common.h. by Rasmus Brandt · 5 years ago
- cb459ca Remove double declaration of cricket::kH264CodecName. by Mirko Bonadei · 5 years ago
- 2b9317a Stop checking VP8BaseHeavyTl3RateAllocation field trial on every frame. by Rasmus Brandt · 5 years ago
- 9560d7d Make update_rect optional in VideoFrame by Ilya Nikolaevskiy · 5 years ago
- 6e4e688 Fixed MSAN issue with usrsctp reliability test. by Yura Yaroshevich · 5 years ago
- e114fb6 Added usrsctp reliablitiy stress test. by Yura Yaroshevich · 5 years ago
- 16cec3b Added allow_codec_switching parameter to RTCConfig. by philipel · 5 years ago
- fcf79cc Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats. by Åsa Persson · 5 years ago
- 0855e2d Delete unused members of MediaReceiverInfo and MediaSenderInfo by Niels Möller · 5 years ago
- 03fbace Remove apm_helpers, consolidate audio config in WebRtcVoiceEngine by Sam Zackrisson · 5 years ago
- 3f7e0ed Add option to make first scale factor depend on input resolution. by Åsa Persson · 5 years ago
- 86d053c Use source_sets in component builds and static_library in release builds. by Mirko Bonadei · 5 years ago
- 9429888 Delete deprecated bytes_sent/bytes_rcvd stat values by Niels Möller · 5 years ago
- 0bad15f Remove the noise_suppression() pointer to submodule interface by saza · 5 years ago
- 8038541 Update the header extensions capabilities with mid, rid and rrid by Florent Castelli · 5 years ago
- ac0a4cb Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Niels Möller · 5 years ago
- 41478c7 Remove AudioProcessing::gain_control() getter by Sam Zackrisson · 5 years ago
- 35214fc Add missing RTC_EXPORT for the component build. by Mirko Bonadei · 5 years ago
- ef0627f Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Mirko Bonadei · 5 years ago
- fbde32e Fix GetStats bytesSent/Received, wireup headerBytesSent/Received by Niels Möller · 5 years ago
- 80f53b7 Extend WebRTC-Video-MinVideoBitrate to experiment per-codec by Elad Alon · 5 years ago
- 5740f3e Clarify expectation on GlobalLock by Danil Chapovalov · 5 years ago
- ff27da5 Add/remove receive streams with SSRC 0 from media channels by Saurav Das · 5 years ago
- f4e0c29 SimulcastEncoderAdapter: support per layer fallback and single encoder proxying by Erik Språng · 5 years ago
- 9d7eb28 Don't limit simulcast layers number for screenshare based on resolution by Ilya Nikolaevskiy · 5 years ago
- 09f1195 Always pass arguments to INSTANTIATE_TEST_SUITE_P. by Mirko Bonadei · 5 years ago
- 27b0e0d Remove obsolete todo comment in simulcast.h by Åsa Persson · 5 years ago
- e942b14 New build target api:media_interface by Niels Möller · 5 years ago
- 1b83a9e Only handle each RTCP once. by Sebastian Jansson · 5 years ago
- 53227cc Remove webrtc::MinPositive from api/. by Mirko Bonadei · 5 years ago
- 738bfa7 Remove api/bitrate_constraints.h. by Mirko Bonadei · 5 years ago
- 317a1f0 Use std::make_unique instead of absl::make_unique. by Mirko Bonadei · 5 years ago
- d9cc8c0 Encoder switching based on network and/or resolution conditions. by philipel · 5 years ago
- 73ceed5 Update simulcast bitrate calculations for non-standard resolutions. by Ilya Nikolaevskiy · 5 years ago
- 7bf7a42 Delete flag VideoReceiveStream::Config::Rtp::remb by Niels Möller · 5 years ago
- eaaaf41 Introduce api/crypto/BUILD.gn. by Mirko Bonadei · 5 years ago
- 70dd165 Delete CoreAudio include from media_engine.h by Niels Möller · 5 years ago
- 65f17ca Move MediaTransportInterface out of the libjingle_peerconnection_api target by Niels Möller · 5 years ago
- fcfeefe Move rtc_error.{h,cc} to its own build target. by Mirko Bonadei · 5 years ago
- cc62b16 Add qualityLimitationResolutionChanges stat by Evan Shrubsole · 5 years ago
- 0bd2eff Reland "New build target p2p:stun_types" by Niels Möller · 5 years ago
- 91c824f Revert "New build target p2p:stun_types" by Hannes Landeholm · 5 years ago
- 66d6c3b Buffers non atomic message send with usrsctp lib. by Seth Hampson · 5 years ago
- 8c5520c Reland "Make the min video bitrate in VideoSendStream configurable." by Ying Wang · 5 years ago
- 1d2149c Revert "Make the min video bitrate in VideoSendStream configurable." by Alessio Bazzica · 5 years ago
- b2fb0b9 Make the min video bitrate in VideoSendStream configurable. by Ying Wang · 5 years ago