Log - 95cb56bd895dc83b8588e2b0090d417ecd9efa97 - platform/external/webrtc

184ea66 Reland "Reland "Reland "Distinguish between send and receive codecs""" by Johannes Kron · 5 years ago
99d6d81 Adding absolute capture timestamp to AudioTrackSinkInterface. by Minyue Li · 5 years ago
a104ceb Revert "Reland "Reland "Distinguish between send and receive codecs""" by Johannes Kron · 5 years ago
9bac68c Reland "Reland "Distinguish between send and receive codecs"" by Johannes Kron · 5 years ago
760fd52 Replace MockAudioDeviceModule mock refcounting with real refcounting by Steve Anton · 5 years ago
00a3087 Revert "Reland "Distinguish between send and receive codecs"" by Johannes Kron · 5 years ago
897776e Pass SDP video parameters to encoder. by Sergey Silkin · 5 years ago
133bf2b Reland "Distinguish between send and receive codecs" by Johannes Kron · 5 years ago
1acdc74 Split up EncoderStreamFactory::CreateEncoderStreams in two. by Rasmus Brandt · 5 years ago
43bfe0b Enforce VideoEncoderConfig.num_temporal_layers >= 1. by Rasmus Brandt · 5 years ago
ccbe95f Reformat GN files. by Mirko Bonadei · 5 years ago
e57b266 Revert "Distinguish between send and receive codecs" by Steve Anton · 5 years ago
c0f25cf Distinguish between send and receive codecs by Johannes Kron · 5 years ago
b2b2031 Concatenate string literals at compile time. by Jonas Olsson · 5 years ago
1546f99 Fixed timeout overflow in sctp reliability test. by Yura Yaroshevich · 5 years ago
fae6400 Add saza@ and peah@ to OWNERS of some audio files by Sam Zackrisson · 5 years ago
f5ecb5f Revert "Reland "Reland "Reland "Distinguish between send and receive video codecs"""" by Mirko Bonadei · 5 years 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 · 5 years ago
27064ad SimulcastEncoderAdapter: In passthrough mode set correct lenght for frame_types parameter by Ilya Nikolaevskiy · 5 years ago
4db28b5 Cleanup: Removes redundant includes on message_queue.h by Sebastian Jansson · 5 years ago
873610c Fix updating degradation preference in SetRtpParameters. by Mirta Dvornicic · 5 years ago
b5159fe Revert "Reland "Reland "Distinguish between send and receive video codecs""" by Olga Sharonova · 5 years 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 · 5 years ago
5cad55b Signal requested resolution alignment requirements from sinks to sources. by Rasmus Brandt · 5 years 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