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• dff6e25f9e Update WebRTC code version (2023-05-26T04:05:22). webrtc-version-updater 2023-05-25 21:05:24 -07:00
• 6e23fa52bf Cleanup WebRTC-PayloadTypes-Lower-Dynamic-Range trial Philipp Hancke 2023-05-22 18:27:45 +02:00
• 56d126074e PipeWire video capture: split portal and PipeWire implementations Jan Grulich 2023-05-25 18:09:13 +02:00
• 2264e7aacc Fixes distortion in WGC screen capture path henrika 2023-05-25 16:38:52 +02:00
• 40a0fa95c8 Add new padding mode to RtpPacketHistory Per K 2023-05-25 14:13:19 +02:00
• 4206d31ae1 [Analysis] Add new thresholds config schema Beining Chen 2023-05-25 12:09:12 +00:00
• cfd4cd0703 Introduce AddDefaultRecvStreamForTesting to VideoReceiveChannel API Harald Alvestrand 2023-05-25 08:41:53 +00:00
• 5c35d08703 Replace "RTRR" with "RRTR" Philipp Hancke 2023-05-25 08:23:15 +02:00
• 1ecac13149 Roll chromium_revision be3e47cd99..aae661725b (1148555:1148994) chromium-webrtc-autoroll 2023-05-25 03:04:48 -07:00
• f4d0a493b4 Adopt EglThread in EglRenderer Linus Nilsson 2023-05-25 10:43:44 +02:00
• 1cc41ea84f Remove unused Win32Window class Philipp Hancke 2023-05-24 18:49:10 +02:00
• 0f1376529a Delete RTC[NonStandard/Restricted]StatsMember. Henrik Boström 2023-05-24 17:57:12 +02:00
• 621cb2943d Fix video version of RTCInboundRtpStreamStats.jitterBufferDelay to obey spec. Rasmus Brandt 2023-05-24 13:29:14 +02:00
• 8ac66a2ba8 Update WebRTC code version (2023-05-25T04:12:48). webrtc-version-updater 2023-05-24 21:12:49 -07:00
• ec863a213f Roll chromium_revision 8f46ad499d..be3e47cd99 (1148441:1148555) chromium-webrtc-autoroll 2023-05-24 09:02:12 -07:00
• 979b0479a1 Revert "Temporarily add dummy trackId to unblock roll." Henrik Boström 2023-05-24 12:59:07 +00:00
• aa1ad7d6a4 In RtcpTransciever refactor outgoing transport interface Danil Chapovalov 2023-05-24 15:49:23 +02:00
• f4d762e776 docs: explain release note process Philipp Hancke 2023-05-24 10:26:36 +02:00
• 2057d71775 [Stats] Delete unused NonStandardGroupId. Henrik Boström 2023-05-24 11:18:56 +02:00
• 61bacd1ddd Enable WebRTC-SplitMediaChannel by default Harald Alvestrand 2023-05-15 07:12:40 +00:00
• 3df4178d59 Temporarily add dummy trackId to unblock roll. Henrik Boström 2023-05-24 14:51:02 +02:00
• 4e231eedbd Delete deprecated 'track' and 'stream' metrics from WebRTC. Henrik Boström 2023-05-24 11:24:13 +02:00
• 54c37a5500 Roll chromium_revision 9378e3160b..8f46ad499d (1148314:1148441) chromium-webrtc-autoroll 2023-05-24 03:16:20 -07:00
• 18898d7db3 Update WebRTC code version (2023-05-24T04:17:39). webrtc-version-updater 2023-05-23 21:17:40 -07:00
• 1920430ebb Roll chromium_revision 3d8a0c6a22..9378e3160b (1148203:1148314) chromium-webrtc-autoroll 2023-05-23 19:03:21 -07:00
• 531383c653 Roll chromium_revision bbf4ff3290..3d8a0c6a22 (1148035:1148203) chromium-webrtc-autoroll 2023-05-23 15:02:31 -07:00
• 4daa4e6b2f Roll chromium_revision c025c4ac7b..bbf4ff3290 (1147854:1148035) chromium-webrtc-autoroll 2023-05-23 11:02:48 -07:00
• a7d10811cd Revert "pipewire capturer: Reduce the amount of copying" Alexander Cooper 2023-05-23 16:14:46 +00:00
• 2eacbbc03a Roll chromium_revision 692840e030..c025c4ac7b (1147747:1147854) chromium-webrtc-autoroll 2023-05-23 07:20:47 -07:00
• 3369729100 Add EglThread class wrapping EglConnection and handler. Linus Nilsson 2023-05-17 22:07:42 +02:00
• ff35a37a8b Unit tests for MediaChannel creation API Harald Alvestrand 2023-05-23 11:25:30 +00:00
• 98f47a300b Delete redundant member StreamDataCounters::last_packet_received_time Danil Chapovalov 2023-05-22 18:30:28 +02:00
• 0328190ab3 Add video_codec_perf_tests to desktop and android perf test suites Sergey Silkin 2023-05-19 11:10:15 +02:00
• e3441ec456 Roll chromium_revision fb6508249a..692840e030 (1147609:1147747) chromium-webrtc-autoroll 2023-05-23 03:03:00 -07:00
• aa6d4fa622 Adds WebRTC-DisableRtxRateLimiter for enable/disable RTX rate limiter. Ying Wang 2023-05-17 20:59:49 +08:00
• f67d1fd42c OveruseFrameDetector: complete removal of mac rules kill switch. Markus Handell 2023-05-22 17:16:21 +02:00
• 06130548fc Remove unused histograms. Markus Handell 2023-05-22 20:16:44 +02:00
• 434deda6fa Cleanup RtcpReceiver from using RtcpBandwidthObser callback interface Danil Chapovalov 2023-05-22 12:56:56 +02:00
• 4858a0d9d8 Add test for split-mode SSRC callback Harald Alvestrand 2023-05-22 21:34:24 +00:00
• 85632b8191 Update WebRTC code version (2023-05-23T04:03:48). webrtc-version-updater 2023-05-22 21:03:49 -07:00
• a8f55c7c03 Roll chromium_revision 634d3c7e62..fb6508249a (1147498:1147609) chromium-webrtc-autoroll 2023-05-22 19:02:36 -07:00
• c941cdd540 Roll chromium_revision 15e9b8d197..634d3c7e62 (1147348:1147498) chromium-webrtc-autoroll 2023-05-22 15:01:38 -07:00
• 3fea51aa20 Roll chromium_revision 0eaeb41fa6..15e9b8d197 (1147199:1147348) chromium-webrtc-autoroll 2023-05-22 11:21:29 -07:00
• 13897e67c8 Change SSRC-passing for MediaChannel from external to callback Harald Alvestrand 2023-05-22 12:49:05 +00:00
• 5dc42050e1 Roll chromium_revision 65192f0ef9..0eaeb41fa6 (1147078:1147199) chromium-webrtc-autoroll 2023-05-22 07:01:47 -07:00
• 1d3452f31b RequestedResolution - Bug fix Jonas Oreland 2023-05-12 16:36:41 +02:00
• b7a688c68e Delete WebRTC.Video.BadCall.* histograms. Rasmus Brandt 2023-05-22 12:21:09 +02:00
• b4015689b8 Initial copy of flexfec_header_reader_writer. Yosef Twaik 2023-05-04 18:27:43 +03:00
• 718601a1f8 Cleanup RtcpReceiver from passing TransportFeedback via older interface Danil Chapovalov 2023-05-19 15:10:55 +02:00
• 15feded162 Increase maximum RTP padding length to 255 bytes Philipp Hancke 2023-05-18 11:03:21 +02:00
• 194f657b51 Roll chromium_revision 20c92b363d..65192f0ef9 (1146960:1147078) chromium-webrtc-autoroll 2023-05-22 01:02:48 -07:00
• 0c85f733c9 For AV1, disable error resilience on upper temporal layers Danil Chapovalov 2023-05-19 17:28:31 +02:00
• 3fb338a4d5 Update WebRTC code version (2023-05-22T04:03:29). webrtc-version-updater 2023-05-21 21:03:30 -07:00
• 0483755f4e Roll chromium_revision f5493a4850..20c92b363d (1146841:1146960) chromium-webrtc-autoroll 2023-05-21 11:13:46 -07:00
• b1b2c534be Update WebRTC code version (2023-05-21T04:02:48). webrtc-version-updater 2023-05-20 21:02:50 -07:00
• c9f0b20c53 Update WebRTC code version (2023-05-20T04:11:18). webrtc-version-updater 2023-05-19 21:11:19 -07:00
• 0f40079726 Roll chromium_revision e7ad7ca1d5..f5493a4850 (1146741:1146841) chromium-webrtc-autoroll 2023-05-19 21:02:08 -07:00
• 447fc3f555 Roll chromium_revision f5f7594337..e7ad7ca1d5 (1146453:1146741) chromium-webrtc-autoroll 2023-05-19 15:08:04 -07:00
• 9a7ca64da3 Roll chromium_revision d4f384285a..f5f7594337 (1145480:1146453) chromium-webrtc-autoroll 2023-05-19 05:04:19 -07:00
• cc1ee35a69 Reland "Avoid recreating VirtualDisplay on format changes." Linus Nilsson 2023-05-19 14:19:23 +02:00
• 328c514757 Reduce precision of RTT in RtrpTransportControllerSend Danil Chapovalov 2023-05-19 11:45:46 +02:00
• ff75eae064 Update WebRTC code version (2023-05-18T04:12:20). webrtc-version-updater 2023-05-17 21:12:21 -07:00
• f6a0680494 Roll chromium_revision d59cc17cf9..d4f384285a (1145311:1145480) chromium-webrtc-autoroll 2023-05-17 11:01:44 -07:00
• 3e39254b67 Pass rtcp message to RtpTransportController through newer interface Danil Chapovalov 2023-05-17 13:25:39 +02:00
• a0b1144745 Roll chromium_revision 2b0829702f..d59cc17cf9 (1145193:1145311) chromium-webrtc-autoroll 2023-05-17 07:06:58 -07:00
• 510890ba19 Revert "Avoid recreating VirtualDisplay on format changes." Mirko Bonadei 2023-05-17 11:15:37 +00:00
• fcd1dfad1f Avoid recreating VirtualDisplay on format changes. Linus Nilsson 2023-05-16 21:20:50 +02:00
• 4d0468eebc Roll chromium_revision 8f747c9bf2..2b0829702f (1144817:1145193) chromium-webrtc-autoroll 2023-05-17 01:22:40 -07:00
• cb1b73ab0c Update WebRTC code version (2023-05-17T04:12:05). webrtc-version-updater 2023-05-16 21:12:06 -07:00
• a2cf8ee854 Simplify handling rtcp messages in audio send channel Danil Chapovalov 2023-05-16 13:26:33 +02:00
• 9a438742fe Roll chromium_revision 91c345cf4e..8f747c9bf2 (1144710:1144817) chromium-webrtc-autoroll 2023-05-16 11:03:20 -07:00
• c37dec2882 Set use_cxx to true. Mirko Bonadei 2023-05-15 16:42:42 +00:00
• ca66eefe6d Roll chromium_revision 1e36b7ebe0..91c345cf4e (1144620:1144710) chromium-webrtc-autoroll 2023-05-16 07:22:17 -07:00
• 99869adeb5 Roll chromium_revision 30ae698dcc..1e36b7ebe0 (1144471:1144620) chromium-webrtc-autoroll 2023-05-16 03:02:17 -07:00
• 69bc3e185f Trigger bots Mirko Bonadei 2023-05-16 08:22:12 +00:00
• f3de65aebe Change ReceivedFecPacket to have list of ssrcs, seq nums and masks. Yosef Twaik 2023-05-04 18:19:54 +03:00
• 3a4cfdf344 Update WebRTC code version (2023-05-16T04:02:28). webrtc-version-updater 2023-05-15 21:02:29 -07:00
• aaa3b8fe0f Roll chromium_revision 5fb222694e..30ae698dcc (1144315:1144471) chromium-webrtc-autoroll 2023-05-15 17:05:20 -07:00
• 8cef7c9aac Roll chromium_revision 23a25491e5..5fb222694e (1144139:1144315) chromium-webrtc-autoroll 2023-05-15 13:04:52 -07:00
• 784c339f34 Expose setCodecPreferences/getCapabilities for android David Liu 2023-05-15 11:33:38 +00:00
• 54519aa378 Roll chromium_revision 1f0a8499f0..23a25491e5 (1144024:1144139) chromium-webrtc-autoroll 2023-05-15 09:03:06 -07:00
• ad363ea310 Remove spam log from IvfFileWriter. Mirko Bonadei 2023-05-15 13:42:11 +00:00
• 44ebe2abc9 Refactor HasDataChannels Tommi 2023-05-15 15:14:10 +02:00
• 3ca0aa5e6f Set use_cxx to false. Mirko Bonadei 2023-05-15 11:58:52 +00:00
• 84f6470821 Roll chromium_revision bd7ba3acbb..1f0a8499f0 (1143522:1144024) chromium-webrtc-autoroll 2023-05-15 05:01:55 -07:00
• 487c943a41 Guard send_codec variable against receive channel access Harald Alvestrand 2023-05-15 06:32:27 +00:00
• 79249155c3 Stop decoding video for m-lines which are sendonly or inactive Philipp Hancke 2023-05-11 18:13:34 +02:00
• 121f1e7a01 In RtpTransportController reduce information stored about rtcp report blocks Danil Chapovalov 2023-05-15 09:28:02 +02:00
• eb97ac5f29 Update WebRTC code version (2023-05-13T04:12:15). webrtc-version-updater 2023-05-12 21:12:17 -07:00
• 5998ddfad8 Roll chromium_revision 5cdde4d6df..bd7ba3acbb (1143390:1143522) chromium-webrtc-autoroll 2023-05-12 15:03:33 -07:00
• 7b0d7f48fb PipeWire capturer: fix fcntl call when duplicating a file descriptor Jan Grulich 2023-05-12 20:59:06 +02:00
• 016e57c68c Roll chromium_revision e89d67b6a3..5cdde4d6df (1143194:1143390) chromium-webrtc-autoroll 2023-05-12 11:03:13 -07:00
• 32dae4b844 sdp: accept bundle-only media section without rtcp-mux Philipp Hancke 2023-05-12 06:53:55 +02:00
• afdc00f1f6 Roll chromium_revision f273051fd0..e89d67b6a3 (1142666:1143194) chromium-webrtc-autoroll 2023-05-12 05:01:13 -07:00
• 63551c6f0c Initialize RTP modes from callback Harald Alvestrand 2023-05-12 08:24:02 +00:00
• a09331a603 Don't write TransmissionOffset when capture time is not set Andreas Pehrson 2023-05-12 09:36:34 +02:00
• 0280ceb1b6 Delete unused member in VideoSendStream Danil Chapovalov 2023-05-11 18:26:15 +02:00
• 71f80c0bd6 Replace RtcpReceiver::RTT function with RtcpReceiver::AverageRtt with cleaner interface Danil Chapovalov 2023-05-11 10:35:37 +02:00
• b073283b4a Roll chromium_revision c8d546c7b8..f273051fd0 (1142541:1142666) chromium-webrtc-autoroll 2023-05-11 09:17:50 -07:00
• ad7792b1a5 rtp_rtcp: fix small typo in enum class RtpPacketMediaType Alfred E. Heggestad 2023-05-11 11:30:43 +02:00