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  • f53597140f In RtpSource represent time with Timestamp type instead of int64_t Danil Chapovalov 2023-08-28 16:32:25 +00:00
  • 465bc0fd87 Validate rejected m-lines less strictly Philipp Hancke 2023-08-25 12:42:36 +02:00
  • 2ea5f15a5a Update WebRTC code version (2023-08-29T04:11:51). webrtc-version-updater 2023-08-28 21:11:52 -07:00
  • f43ca8da0b Roll chromium_revision 91ca53b76a..0e815932e7 (1189192:1189306) chromium-webrtc-autoroll 2023-08-28 19:01:50 -07:00
  • e17ac9fb05 Roll chromium_revision f530c164b2..91ca53b76a (1189050:1189192) chromium-webrtc-autoroll 2023-08-28 15:02:02 -07:00
  • 841c0fbc14 Roll chromium_revision 5e8632e21c..f530c164b2 (1188919:1189050) chromium-webrtc-autoroll 2023-08-28 11:02:42 -07:00
  • a041a97f63 Reset RobustThroughputEstimator if recv timestamp jump backwards Per K 2023-08-25 16:17:25 +02:00
  • 14e5d4ce5e Support sending IncomingFrames in audio Palak Agarwal 2023-08-22 16:58:33 +02:00
  • 93316a936f Roll chromium_revision c92a3a5750..5e8632e21c (1188810:1188919) chromium-webrtc-autoroll 2023-08-28 05:01:15 -07:00
  • 2111bb5be3 Delete some unused sigslot includes Harald Alvestrand 2023-08-28 10:23:32 +00:00
  • 3924bc272c Remove sigslot include from async_udp_socket.cc Harald Alvestrand 2023-08-28 11:06:45 +00:00
  • 5f14f9e6ed Remove VCMEncodedFrame from webrtc::EncodedFrame inheritance Tony Herre 2023-08-25 14:53:44 +02:00
  • 4bf853c7e8 Update WebRTC code version (2023-08-28T04:13:13). webrtc-version-updater 2023-08-27 21:13:14 -07:00
  • 1d09484c8b Roll chromium_revision 6d201146b8..c92a3a5750 (1188708:1188810) chromium-webrtc-autoroll 2023-08-27 17:14:03 -07:00
  • efa5fe7813 Update WebRTC code version (2023-08-27T04:04:26). webrtc-version-updater 2023-08-26 21:04:26 -07:00
  • 6de97acb4e Roll chromium_revision d863b89838..6d201146b8 (1188606:1188708) chromium-webrtc-autoroll 2023-08-26 11:03:21 -07:00
  • 174ca703e1 Update WebRTC code version (2023-08-26T04:07:31). webrtc-version-updater 2023-08-25 21:07:32 -07:00
  • 7fedd0696e Roll chromium_revision 1422b89a54..d863b89838 (1188481:1188606) chromium-webrtc-autoroll 2023-08-25 17:04:06 -07:00
  • c75769786f Roll chromium_revision 08be09ec4a..1422b89a54 (1188371:1188481) chromium-webrtc-autoroll 2023-08-25 13:02:04 -07:00
  • da711e0425 Roll chromium_revision 6aaff43ef6..08be09ec4a (1188260:1188371) chromium-webrtc-autoroll 2023-08-25 09:14:11 -07:00
  • 8cfae1bb3f Cleanup usage of lookups in video SendDelayStats helper Danil Chapovalov 2023-08-25 09:40:15 +00:00
  • b971a534e9 Fix inconsistent logics to check if temporal layer is supported Wei Zhou 2023-08-23 14:09:54 -07:00
  • b5dedfc856 In AimdRateControl, add trial to use current bitrate as min upper limit Per K 2023-08-25 14:18:47 +02:00
  • 4d25a77fd3 Deprecate AsyncResolver config fields and remove internal usage. Harald Alvestrand 2023-08-25 11:07:28 +00:00
  • 92fe5c4db3 Roll chromium_revision 2c59b7b527..6aaff43ef6 (1188138:1188260) chromium-webrtc-autoroll 2023-08-25 03:03:07 -07:00
  • 45a985c71d Check use_rtx() in PeerConnectionFactory::GetRtpSenderCapabilities Joachim Reiersen 2023-08-24 10:00:01 -07:00
  • 70cea9bda8 [SctpDataChannel] Don't use PostTask for observer registration. Tommi 2023-08-25 01:12:04 +02:00
  • 0720e9ac1d Update WebRTC code version (2023-08-25T04:12:20). webrtc-version-updater 2023-08-24 21:12:21 -07:00
  • 371a82cdab Make field trial string enable_retransmit_all_layers enabled by default. Ying Wang 2023-08-24 13:57:43 +08:00
  • 68a64dd208 Roll chromium_revision 30d5b1ee39..2c59b7b527 (1188023:1188138) chromium-webrtc-autoroll 2023-08-24 19:03:35 -07:00
  • eab0a5a0cf Roll chromium_revision 996bf8ce67..30d5b1ee39 (1187868:1188023) chromium-webrtc-autoroll 2023-08-24 15:01:37 -07:00
  • 572c6efd23 Roll chromium_revision 6dae2061c5..996bf8ce67 (1187756:1187868) chromium-webrtc-autoroll 2023-08-24 09:13:43 -07:00
  • 66bf3f472c Make PendingTaskSafetyFlag compatible with component builds Tommi 2023-08-24 15:29:10 +02:00
  • 394ed51669 Roll chromium_revision 9a6d0f811c..6dae2061c5 (1187201:1187756) chromium-webrtc-autoroll 2023-08-24 05:01:34 -07:00
  • 43a5dd86c2 Implement codec selection api Florent Castelli 2023-04-12 10:45:07 +00:00
  • f45c2ceae0 Fix TSAN conflict in AsyncDnsResolver Harald Alvestrand 2023-08-24 09:19:52 +00:00
  • d36be2fc22 Revise AsyncDnsResolver unittest Harald Alvestrand 2023-08-24 08:54:28 +00:00
  • b8617d14a6 Use the AsyncDnsResolver in PeerConnection defaults Harald Alvestrand 2023-08-23 09:01:53 +00:00
  • 7084e1b6d9 In VideoPlayoutDelay delete access to integer representation of min/max values Danil Chapovalov 2023-08-23 13:16:22 +02:00
  • 093a939572 Fix includes in rtp_header_extensions.cc Danil Chapovalov 2023-08-23 13:49:57 +02:00
  • 38422ef644 Roll chromium_revision 540343ddd9..9a6d0f811c (1187093:1187201) chromium-webrtc-autoroll 2023-08-23 05:02:11 -07:00
  • 1692e54146 FrameDumpingEncoder: relax threading assumptions. Markus Handell 2023-08-23 12:44:18 +02:00
  • c08df4bdca Remove DCHECK when processing StapA in h264_sps_pps_tracker.cc Joachim Reiersen 2023-08-18 11:15:37 -07:00
  • 8d4a5f1122 Add an async DNS resolver suitable for defaulting. Harald Alvestrand 2023-08-22 13:53:16 +00:00
  • c89fdbe624 Roll chromium_revision 84c0a4dbcf..540343ddd9 (1186981:1187093) chromium-webrtc-autoroll 2023-08-23 01:01:34 -07:00
  • 1a4cf30047 Avoiding to increase an iterator when the result can be larger than their container end. Jesús de Vicente Peña 2023-08-22 16:31:13 +02:00
  • 45ccf63c0f Update WebRTC code version (2023-08-23T04:04:00). webrtc-version-updater 2023-08-22 21:04:02 -07:00
  • 4c0e50666d Roll chromium_revision a98b2dd04f..84c0a4dbcf (1186854:1186981) chromium-webrtc-autoroll 2023-08-22 19:02:04 -07:00
  • 6f0ff21b9e Roll chromium_revision 5fd3c86bd7..a98b2dd04f (1186671:1186854) chromium-webrtc-autoroll 2023-08-22 15:02:52 -07:00
  • 8284f2b4e8 Roll chromium_revision f96125ad3f..5fd3c86bd7 (1186488:1186671) chromium-webrtc-autoroll 2023-08-22 11:01:29 -07:00
  • 411639ede8 Introduce a frame dumping encoder wrapper. Markus Handell 2023-08-22 12:58:47 +02:00
  • 56c34a9ff4 Roll chromium_revision 7010c0507b..f96125ad3f (1186337:1186488) chromium-webrtc-autoroll 2023-08-22 07:01:33 -07:00
  • 179cec2be0 Reduce logging verbosity of DTLS-SRTP RTCP transport Philipp Hancke 2023-08-22 11:08:32 +02:00
  • 6b7bbe2e33 Adding new WebRTC.Video.GenericDecoder histograms henrika 2023-08-22 13:42:32 +02:00
  • a586d4eccc Fix fuzzing test issues reported by Chromium fuzzing test qwu16 2023-08-21 15:58:15 +08:00
  • 47faf32287 Add rtc_common_public_deps Arthur Sonzogni 2023-08-22 09:55:36 +00:00
  • 5f3b3db105 Cleanup flexfec03 TODOs and logs to indicate there is no intent to implement additional features there Danil Chapovalov 2023-08-21 19:32:41 +02:00
  • cc4def36b4 Roll chromium_revision 5b6526e174..7010c0507b (1186224:1186337) chromium-webrtc-autoroll 2023-08-22 01:01:18 -07:00
  • 06717773a5 Move EncodedImage::playout_delay_ to private section of the class Danil Chapovalov 2023-08-21 18:17:31 +02:00
  • 233165b239 Replace all RTPSender::SendToNetwork with EnqueuePackets Danil Chapovalov 2023-08-21 17:34:55 +02:00
  • 798708ff9d Update WebRTC code version (2023-08-22T04:11:35). webrtc-version-updater 2023-08-21 21:11:37 -07:00
  • 1ee5bfec9f Roll chromium_revision 099aa05d92..5b6526e174 (1186097:1186224) chromium-webrtc-autoroll 2023-08-21 19:02:07 -07:00
  • 994b2c9d4e Roll chromium_revision f1c4fdda5b..099aa05d92 (1185958:1186097) chromium-webrtc-autoroll 2023-08-21 15:02:25 -07:00
  • 86f9fa2384 Roll chromium_revision 933bc00fdd..f1c4fdda5b (1185843:1185958) chromium-webrtc-autoroll 2023-08-21 11:04:11 -07:00
  • 7073740dad Roll chromium_revision dd175d86ee..933bc00fdd (1185733:1185843) chromium-webrtc-autoroll 2023-08-21 07:02:13 -07:00
  • 960882047d Add mock of GetCaptureTimeIdentifier to MockTransformableVideoFrame Tony Herre 2023-08-18 13:11:23 +02:00
  • b0b03a87b7 Mark api video timing classes with RTC_EXPORT Danil Chapovalov 2023-08-21 13:52:14 +02:00
  • f263e1eb03 Support receiving cloned encoded audio frames Palak Agarwal 2023-08-18 14:09:53 +02:00
  • 2886da25da Roll chromium_revision 2e84995879..dd175d86ee (1185614:1185733) chromium-webrtc-autoroll 2023-08-21 01:01:20 -07:00
  • 1997837d16 Add stream label to test video source for better debugablity and testability Artem Titov 2023-08-16 10:09:19 +02:00
  • 952e0b75f9 Update WebRTC code version (2023-08-21T04:02:03). webrtc-version-updater 2023-08-20 21:02:05 -07:00
  • f1551b5a8e Roll chromium_revision 8486e16478..2e84995879 (1185512:1185614) chromium-webrtc-autoroll 2023-08-20 03:02:43 -07:00
  • d95b5758ef Update WebRTC code version (2023-08-20T04:02:25). webrtc-version-updater 2023-08-19 21:02:26 -07:00
  • 45b5256789 Roll chromium_revision 1a5b138d55..8486e16478 (1185348:1185512) chromium-webrtc-autoroll 2023-08-19 01:04:02 -07:00
  • 45dda0045c Update WebRTC code version (2023-08-19T04:03:39). webrtc-version-updater 2023-08-18 21:03:41 -07:00
  • df810ab2ad Roll chromium_revision a7b5beaadc..1a5b138d55 (1185128:1185348) chromium-webrtc-autoroll 2023-08-18 13:02:34 -07:00
  • ad3f1bcc1b Adopt EglThread in EglRenderer Linus Nilsson 2023-08-18 14:15:33 +02:00
  • 47f4e55612 Log video encoder InitEncode error code Philipp Hancke 2023-08-18 14:18:40 +02:00
  • 972f28335a Added HEVC parser for WebRTC qwu16 2023-08-15 17:16:54 +08:00
  • b071871aa0 Remove unused ALR experiment settings from VideoStreamEncoder. philipel 2023-08-18 15:22:25 +02:00
  • c146b5f77b Represent unset VideoPlayoutDelay with nullopt rather than special value Danil Chapovalov 2023-08-16 11:42:52 +02:00
  • 0250b69fd4 Delete unused constructor of FakeNetworkPipe Danil Chapovalov 2023-08-18 13:57:41 +02:00
  • 44e54a75a7 Roll chromium_revision 4b024ef0b5..a7b5beaadc (1185015:1185128) chromium-webrtc-autoroll 2023-08-18 05:02:01 -07:00
  • 70483a59aa Add unscaled getWidth/Height to TextureBuffer interface Magnus Jedvert 2023-08-18 13:10:36 +02:00
  • 31251fa817 Delete deprecated RTPSender::SetCsrcs Danil Chapovalov 2023-08-15 13:32:54 +02:00
  • 68dafcb6c1 Update WebRTC code version (2023-08-18T04:05:33). webrtc-version-updater 2023-08-17 21:05:33 -07:00
  • dce991342c Roll chromium_revision 80b8365885..4b024ef0b5 (1184627:1185015) chromium-webrtc-autoroll 2023-08-17 19:02:53 -07:00
  • e5af52e1f2 Make DesktopFrameCGImage::CreateFromCGImage() public Johannes Kron 2023-08-10 11:47:15 +00:00
  • a39d7643ef Roll chromium_revision 9f26c488cf..80b8365885 (1184507:1184627) chromium-webrtc-autoroll 2023-08-17 03:01:58 -07:00
  • 69e2637787 Fix playout delay logging typo. Qiu Jianlin 2023-08-17 13:03:18 +08:00
  • 4f89506e53 Update WebRTC code version (2023-08-17T04:03:59). webrtc-version-updater 2023-08-16 21:04:00 -07:00
  • 6cb8b4fbfb Roll chromium_revision c7e600fcc1..9f26c488cf (1184329:1184507) chromium-webrtc-autoroll 2023-08-16 19:02:10 -07:00
  • ee15bea056 Roll chromium_revision b88d5ed17e..c7e600fcc1 (1183080:1184329) chromium-webrtc-autoroll 2023-08-16 13:02:27 -07:00
  • 1e42d83db4 Make VCMReceiveCallback::FrameToRender pure virtual again Philipp Hancke 2023-08-15 16:13:37 +02:00
  • ff2462e54a Update WebRTC code version (2023-08-16T04:02:29). webrtc-version-updater 2023-08-15 21:02:31 -07:00
  • 392e4714e7 Remove deprecated TransformableAudioFrameInterface::getHeader() method Tony Herre 2023-08-14 16:22:50 +02:00
  • d43af9172b Remove internal overrides using old SendRtp and SendRtcp interfaces. Harald Alvestrand 2023-08-15 11:41:45 +00:00
  • 5165743926 Reland "Fix definition of keyframes decoded statistics" Philipp Hancke 2023-08-15 10:20:50 +02:00
  • 48852c40f4 Drop email recipients from lkgr finder. Mirko Bonadei 2023-08-14 15:11:17 +00:00
  • c2da2e2baf Update WebRTC code version (2023-08-15T04:12:47). webrtc-version-updater 2023-08-14 21:12:48 -07:00