- c84f661 Stop using Googletest legacy APIs. by Mirko Bonadei · 6 years ago
- 042bb00 Fix RTP transport accepting invalid RTCP headers. by Piotr (Peter) Slatala · 6 years ago
- c1a0bcb Implement the encoding RtpParameter scaleResolutionDownBy by Florent Castelli · 6 years ago
- 2c9ebef Use Abseil container algorithms in media/ by Steve Anton · 6 years ago
- bcd39d4 Creating Simulcast offer and answer in Peer Connection. by Amit Hilbuch · 6 years ago
- f380284 (7) Rename files to snake_case: remove forwarding headers by Steve Anton · 6 years ago
- d970807 Remove rtc_base/scoped_ref_ptr.h. by Mirko Bonadei · 6 years ago
- 1e27fec Negate flag name for prerender smoothing and update comments. by Rasmus Brandt · 6 years ago
- 805a27e Reland "Refactor WebRtcVideoEngine tests to not use cricket::VideoCapturer, part 2." by Niels Möller · 6 years ago
- 0acffb5 Expose `jitterBufferEmittedCount` in addition to the existing `jitterBufferDelay` for `getStats()`. by Chen Xing · 6 years ago
- aec15aa (5) Rename files to snake_case: install forwarding headers by Steve Anton · 6 years ago
- 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
- 1c05765 (3) Rename files to snake_case: move the files by Steve Anton · 6 years ago
- 1ebfb6a Introduce VideoFrame::id to keep track of frames inside application. by Artem Titov · 6 years ago
- fd87da7 Delete WebRtcVideoCapturer and related classes. by Niels Möller · 6 years ago
- 3793bb4 Refactor TestVideoCapturer to support multiple sinks. by Niels Möller · 6 years ago
- 41f3a43 Remove CodecInst pt.3 by Fredrik Solenberg · 6 years ago
- cca13f6 Remove unused cryptoparams.h header by Steve Anton · 6 years ago
- e1301a8 Revert "Implement read-only codecPayloadType in RtpParameters" by Henrik Grunell · 6 years ago
- 806e06d Implement read-only codecPayloadType in RtpParameters by Florent Castelli · 6 years ago
- c57d573 RID parsing for Simulcast support. by Amit Hilbuch · 6 years ago
- 514f084 New statistic added to VideoReceiveStream to determine latency to first decode. by Benjamin Wright · 6 years ago
- 3e70781 [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. by Yves Gerey · 6 years ago
- 352ce5c Expose delayed packet outage as a cumulative metric of samples in the new getStats API. by Jakob Ivarsson · 6 years ago
- 8af8896 Expose jitter buffer flushes metric in new getStats api. by Ruslan Burakov · 6 years ago
- 5f2ffee Clean up deprecated APM stats by Sam Zackrisson · 6 years ago
- 2222a80 Delete unneeded includes of common_types.h and gn deps on webrtc_common. by Niels Möller · 6 years ago
- 38332cd Add RTCP and simulcast support for RTCRtpReceiver::getParameters() by Florent Castelli · 6 years ago
- 6eb8a16 Exposing audio and video engines directly. by Sebastian Jansson · 6 years ago
- fa0aa39 Removes templating from CompositeMediaEngine. by Sebastian Jansson · 6 years ago
- 84848f2 Adds interfaces for audio and video engines. by Sebastian Jansson · 6 years ago
- dd9390c Prevent channels being set on stopped transceiver. by Amit Hilbuch · 6 years ago
- 5571812 Adding rtcp report interval into RTCConfiguration. by Jiawei Ou · 6 years ago
- 175aa2e Implement data channels over media transport. by Bjorn Mellem · 6 years ago
- e693381 Delete struct rtc::PacketTime. by Niels Möller · 6 years ago
- 15ca5a9 Add implicit conversion between rtc:PacketTime and int64_t. by Niels Möller · 6 years ago
- 9190b82 Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap by Johannes Kron · 6 years ago
- 06aa209 Add support to adapt video without preserving aspect ratio by Magnus Jedvert · 6 years ago
- 039743e Reland "Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase" by Niels Möller · 6 years ago
- 6e8e299 Revert "Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase" by Oleh Prypin · 6 years ago
- 7e6b528 Removes FakeBaseEngine. by Sebastian Jansson · 6 years ago
- 93922dc Fix flaky unit test in rtc_unittests by Johannes Kron · 6 years ago
- 80cd25b Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase by Niels Möller · 6 years ago
- 648d28a Media engine and channel support for per-channel dscp values, specified by RtpParameter by Tim Haloun · 6 years ago
- 3c7d599 Replace _stricmp with absl::EqualsIgnoreCase by Niels Möller · 6 years ago
- 1ddc5b6 Export symbols needed by the Chromium component build (part 5). by Mirko Bonadei · 6 years ago
- cb06cac Moves fake media engine implementation to cc file. by Sebastian Jansson · 6 years ago
- 7dc9774 Delete unused code from media/base/testutils.{cc,h} by Niels Möller · 6 years ago
- d65d179 Export symbols needed by the Chromium component build (part 4). by Mirko Bonadei · 6 years ago
- 98a462c Reland "Reland "Propagate media transport to media channel."" by Anton Sukhanov · 6 years ago
- bfb444c Adds new CryptoOption crypto_options.frame.require_frame_encryption. by Benjamin Wright · 6 years ago
- 9accc9f Revert "Reland "Propagate media transport to media channel."" by Oleh Prypin · 6 years ago
- da65ed2 Reland "Propagate media transport to media channel." by Anton Sukhanov · 6 years ago
- 276827c Export symbols needed by the Chromium component build (part 3). by Mirko Bonadei · 6 years ago
- 37cf245 Revert "Propagate media transport to media channel." by Oleh Prypin · 6 years ago
- 8c16f74 Propagate media transport to media channel. by Anton Sukhanov · 6 years ago
- 3d25530 Reland "Export symbols needed by the Chromium component build (part 1)." by Mirko Bonadei · 6 years ago
- 5526e45 vp9: change x-google-profile-id to profile-id by Philipp Hancke · 6 years ago
- 16fe3f2 Revert "Export symbols needed by the Chromium component build (part 1)." by Mirko Bonadei · 6 years ago
- 99eea42 Reland "Reland "Export symbols needed by the Chromium component build (part 1)."" by Mirko Bonadei · 6 years ago
- 84583f6 Enable End-to-End Encrypted Audio Payloads. by Benjamin Wright · 6 years ago
- b49520b Revert "Reland "Export symbols needed by the Chromium component build (part 1)."" by Mirko Bonadei · 6 years ago
- 588f464 Reland "Export symbols needed by the Chromium component build (part 1)." by Mirko Bonadei · 6 years ago
- 2ea9af2 Revert "Export symbols needed by the Chromium component build (part 1)." by Mirko Bonadei · 6 years ago
- 9e24dcf Export symbols needed by the Chromium component build (part 1). by Mirko Bonadei · 6 years ago
- 6ca9836 Prepare for per-media DSCP values. Push dscp for stun packets to the port layer where they are created. by Tim Haloun · 6 years ago
- 23eba22 Add support for RtpEncodingParameters num_temporal_layers. by Åsa Persson · 6 years ago
- 892acf0 Add support for send_encodings parameters in addTransceiver by Florent Castelli · 6 years ago
- 49ac595 Add GetSources to VideoRtpReceiver by Jonas Oreland · 6 years ago
- 965e794 Add sanity checks to UpdateDelayStatistics and patch unit tests. by Johannes Kron · 6 years ago
- 84df1c7 Make fewer copies when using StringBuilder. by Jonas Olsson · 6 years ago
- bfd412e Adds integration of the FrameEncryptor/FrameDecryptor into the MediaChannel. by Benjamin Wright · 6 years ago
- 2e4419e Add option to only request a frame interval change via OnOutputFormatRequest. by Åsa Persson · 6 years ago
- 366a50c Remove simple stringstream usages. by Jonas Olsson · 6 years ago
- 3288168 Enable video adaptation for all screenshare content by Ilya Nikolaevskiy · 6 years ago
- 3df1d5d Revert removal of simulcast screenshare experimental code (killswitch checks) by Ilya Nikolaevskiy · 6 years ago
- f5f5373 Delete unused member MediaSenderInfo::packets_cached. by Niels Möller · 6 years ago
- 17aff35 Enable clang::find_bad_constructs for sdk/ (part 1). by Mirko Bonadei · 7 years ago
- a3df0f2 Remove simulcast screenshare experimental code by Ilya Nikolaevskiy · 7 years ago
- e41c433 Move sigslot to proper third_party directory by Artem Titov · 7 years ago
- 79eb4dd Enabling clang::find_bad_constructs for libjingle_peerconnection_api. by Mirko Bonadei · 7 years ago
- 89b2963 Reland "Enable simulcast screenshare by default" by Ilya Nikolaevskiy · 7 years ago
- ca536d4 Revert "Enable simulcast screenshare by default" by Ilya Nikolaevskiy · 7 years ago
- d43c692 Enable simulcast screenshare by default by Ilya Nikolaevskiy · 7 years ago
- 065a52a Reland "Remove rtc::Optional alias and api:optional target" by Danil Chapovalov · 7 years ago
- b661c65 Revert "Remove rtc::Optional alias and api:optional target" by Ilya Nikolaevskiy · 7 years ago
- 6f5b0f9 Remove rtc::Optional alias and api:optional target by Danil Chapovalov · 7 years ago
- 918f50c Use absl::make_unique and absl::WrapUnique directly by Karl Wiberg · 7 years ago
- 98badbc Add VP9 profile negotiation to SDP by Emircan Uysaler · 7 years ago
- 665174f Reformat the WebRTC code base by Yves Gerey · 7 years ago
- 00c7183 Replace rtc::Optional with absl::optional in media, ortc, p2p by Danil Chapovalov · 7 years ago
- fc9dcb6 Remove wire-up for cancelled experement on VAAPI VP8 encoding by Ilya Nikolaevskiy · 7 years ago
- f7d7e90 Replace std:remove on vector::erase in streamparams_unittest.cc by Artem Titov · 7 years ago
- 97e0488 Delete unused stats for preferred_bitrate. by Niels Möller · 7 years ago
- f859e55 Removing warning suppression flags from media. by Mirko Bonadei · 7 years ago
- 97b4ee5 Wire up VAAPI VP8 experimental support in WebRTC. by Ilya Nikolaevskiy · 7 years ago
- dacec71 Add Rtcp parameters for PeerConnection senders by Florent Castelli · 7 years ago
- 0327c2d Move VideoStreamEncoderInterface to api/. by Niels Möller · 7 years ago
- 65ec0fc Delete unneeded includes of basictypes.h. by Niels Möller · 7 years ago
- 2d2c888 Returns RTCError for setting unimplemented RtpParameters. by Seth Hampson · 7 years ago