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  • 33e75bb874 Roll chromium_revision 415567486a..d2f297f391 (890510:890623) chromium-webrtc-autoroll 2021-06-08 23:01:30 -07:00
  • 80d6669cf4 Roll chromium_revision 94153b0fd3..415567486a (890203:890510) chromium-webrtc-autoroll 2021-06-08 17:02:06 -07:00
  • 58126f92bf Update the only 3 remaining kFilterBilinear to kFilterBox. Henrik Boström 2021-06-07 12:14:09 +02:00
  • d9a135b088 Roll chromium_revision a72439959b..94153b0fd3 (889958:890203) chromium-webrtc-autoroll 2021-06-08 05:01:25 -07:00
  • 2aa24f1942 Move group-modifying functions into BundleManager Harald Alvestrand 2021-06-08 09:51:19 +00:00
  • 9f9bf38805 Start refactoring bundle behavior into BundleManager Harald Alvestrand 2021-06-08 04:12:37 +00:00
  • 2e1546887b Avoid generating a random id for candidate stats. Tommi 2021-06-07 21:16:38 +02:00
  • 0fb949702b Update WebRTC code version (2021-06-08T04:04:02). webrtc-version-updater 2021-06-07 21:04:05 -07:00
  • c6f2fef092 Roll chromium_revision 98040cdbe1..a72439959b (889744:889958) chromium-webrtc-autoroll 2021-06-07 15:02:11 -07:00
  • da8a45fdaa AllocationSequence: migrate from rtc::Message to TaskQueue. Markus Handell 2021-06-07 13:41:15 +02:00
  • dedcdfeba3 AllocationSequence: switch signal to callback. Markus Handell 2021-06-07 13:39:42 +02:00
  • fd89fc75cc BasicPortAllocatorSession: migrate to TaskQueue. Markus Handell 2021-06-07 13:37:24 +02:00
  • 637a9eebd7 Roll chromium_revision 8d359ae542..98040cdbe1 (889625:889744) chromium-webrtc-autoroll 2021-06-07 07:01:35 -07:00
  • 518669d6d4 Add more trace events to interesting places. Markus Handell 2021-06-07 13:30:46 +02:00
  • a94a4cc197 Roll chromium_revision 2951ce9ba1..8d359ae542 (889518:889625) chromium-webrtc-autoroll 2021-06-06 15:01:45 -07:00
  • 5032f54dd3 Update WebRTC code version (2021-06-06T04:02:33). webrtc-version-updater 2021-06-05 21:02:35 -07:00
  • b60ebfeac9 Roll chromium_revision f54fe52cfb..2951ce9ba1 (889417:889518) chromium-webrtc-autoroll 2021-06-04 17:01:44 -07:00
  • b6d50e3a17 Handle encoder_ == nullptr in VideoStreamEncoder::EncodeVideoFrame. Tommi 2021-06-04 18:39:10 +02:00
  • f277300ce5 Roll chromium_revision 44fa1f9723..f54fe52cfb (889277:889417) chromium-webrtc-autoroll 2021-06-04 13:01:40 -07:00
  • 1a778a24ba Avoid using legacy rtp header parser in the rtp_to_text tool Danil Chapovalov 2021-06-03 13:27:35 +02:00
  • 1b63db956f Move AV1X-AV1 mapping to VideoCodecTypeMime Sergey Silkin 2021-06-04 12:13:37 +02:00
  • e34380a160 Roll chromium_revision b54a8c30e7..44fa1f9723 (889150:889277) chromium-webrtc-autoroll 2021-06-04 07:02:05 -07:00
  • a334dc68f3 Make VideoSendStream::UpdateActiveSimulcastLayers not block. Tommi 2021-06-04 12:50:01 +02:00
  • d25af8ceac doc: document rtp payload type mapping behaviour Philipp Hancke 2021-06-04 06:58:53 +02:00
  • bc8e17583a Roll chromium_revision a11573a242..b54a8c30e7 (888935:889150) chromium-webrtc-autoroll 2021-06-03 21:01:47 -07:00
  • ffbfba979f Added PeerConnectionObserverJni::OnRemoveTrack() Jesús Leganés-Combarro 'piranna 2021-05-25 10:35:10 +02:00
  • 1050fbca91 Remove synchronization from VideoSendStream construction. Tommi 2021-06-03 17:58:28 +02:00
  • c2b4d9be15 Roll chromium_revision fb5254ac9f..a11573a242 (888818:888935) chromium-webrtc-autoroll 2021-06-03 11:03:59 -07:00
  • 52c7fd6be5 Modernize style in RemoteBitrateEstimatorAbsSendTime implementation Danil Chapovalov 2021-06-03 15:52:27 +02:00
  • 43eb4f5886 Roll chromium_revision fee5f397ef..fb5254ac9f (888712:888818) chromium-webrtc-autoroll 2021-06-03 05:02:29 -07:00
  • 096014345f Add a function to check if the packet in a PacketResult has been received. Fanny Linderborg 2021-06-02 14:16:10 +00:00
  • 47f5f8c160 Reduce usage of RtpHeaderParser::CreateForTest in favor of RtpPacket Danil Chapovalov 2021-06-03 12:27:12 +02:00
  • 943e2e6a57 Revert "Fix incorrect SSRC in RtpPacketSendInfo for RTX packets." Andrey Logvin 2021-06-03 10:01:03 +00:00
  • fa3ce637fa Simplify VideoSendStreamImpl constructor. Tommi 2021-06-03 12:06:02 +02:00
  • 84f4ca62f0 Remove workaround for broken nearby build in Chrome Evan Shrubsole 2021-06-03 10:33:56 +02:00
  • e902f28d2a Make VideoSendStreamImpl::configured_pacing_factor_ const Tommi 2021-06-03 12:02:02 +02:00
  • 28e9653b1f Remove dependency on RtpVideoSenderInterface from EncoderRtcpFeedback. Tommi 2021-06-03 11:52:15 +02:00
  • 1d2b22e193 Use pixels from single active stream if set for balanced degradation settings. Åsa Persson 2021-06-03 09:28:47 +02:00
  • 2a25a96973 Disable flacky tests on mac bots Ilya Nikolaevskiy 2021-06-03 10:58:44 +02:00
  • f3ff3c5b77 Reinstate killswitch for WebRTC-Bwe-ReceiverLimitCapsOnly. Christoffer Rodbro 2021-06-02 13:27:36 +02:00
  • ab229b0706 Add documentation for RTC event log Björn Terelius 2021-06-03 10:19:17 +02:00
  • 31b564959c Update comment for RtpVideoStreamReceiver2::RequestPacketRetransmit. Tommi 2021-06-01 09:50:26 +02:00
  • e93dc74483 Roll chromium_revision c165693ba5..fee5f397ef (888523:888712) chromium-webrtc-autoroll 2021-06-02 19:02:20 -07:00
  • a004715d13 Integrate ClippingPredictor into AudioProcessingImpl and AgcManagerDirect Hanna Silen 2021-06-03 03:29:38 +02:00
  • 4b3a06139b Add ClippingPredictor implementation Hanna Silen 2021-06-02 23:03:24 +02:00
  • 565ad610fa Roll chromium_revision 936a99501f..c165693ba5 (888404:888523) chromium-webrtc-autoroll 2021-06-02 11:02:58 -07:00
  • a43953a518 Add ClippingPredictor config in AudioProcessing config Hanna Silen 2021-06-02 17:13:24 +02:00
  • cbdbb8c166 Add ability to adjust the suppressor smoothing in AEC3 Per Åhgren 2021-05-07 23:17:28 +00:00
  • bd933ee29a SdpOfferAnswerHandler: Significantly reduce audio impairment. Markus Handell 2021-06-02 16:17:35 +02:00
  • 7444b19833 Add integration test for active stream toggling. Erik Språng 2021-06-02 14:02:13 +02:00
  • 44107890bb Roll chromium_revision e53f664c6c..936a99501f (888291:888404) chromium-webrtc-autoroll 2021-06-02 05:03:25 -07:00
  • fccb052ee3 Add event traces to interesting places in WebRTC. Markus Handell 2021-06-02 13:28:10 +02:00
  • 486b0401c5 Make VP8 DefaultTemporalLayers always report TL count even with no rate. Erik Språng 2021-06-02 11:07:50 +02:00
  • 1c7ff0d001 dcsctp: Stay in stream if not producing fragment Victor Boivie 2021-06-02 09:34:38 +02:00
  • 5981bf2eb6 Add resolution alignment properties to RTCVideoEncoder protocol. Peter Hanspers 2021-06-01 07:31:00 +02:00
  • aaa835c7b6 Update WebRTC code version (2021-06-02T04:02:07). webrtc-version-updater 2021-06-01 21:02:09 -07:00
  • f32b400505 Roll chromium_revision 45bbaf2c3c..e53f664c6c (888151:888291) chromium-webrtc-autoroll 2021-06-01 21:03:43 -07:00
  • 62678f5c7b Roll chromium_revision 10da87c3f6..45bbaf2c3c (888035:888151) chromium-webrtc-autoroll 2021-06-01 15:02:03 -07:00
  • bf952fa207 Roll chromium_revision ed24ed8d5d..10da87c3f6 (887902:888035) chromium-webrtc-autoroll 2021-06-01 11:02:36 -07:00
  • 78c73477c7 Add DesktopCaptureOption enumerate_current_process_windows to avoid hang Austin Orion 2021-06-01 09:49:15 -07:00
  • 803fdc4106 dcsctp: Stay within stream while producing from it Victor Boivie 2021-06-01 17:06:35 +02:00
  • f865444877 Make AV1 respect spatial layer active flag. Erik Språng 2021-06-01 16:52:24 +02:00
  • d23628dfb6 Remove RecordingState::keyframe_needed. Tommi 2021-06-01 09:49:03 +02:00
  • d99430410b Call: introduce SendStats. Markus Handell 2021-05-31 22:52:02 +02:00
  • e9fa954068 Roll chromium_revision 03cca1960d..ed24ed8d5d (887795:887902) chromium-webrtc-autoroll 2021-06-01 03:01:56 -07:00
  • 75941453c5 Fix incorrect fps_allocation printed by EncoderInfo::ToString() Erik Språng 2021-05-31 17:08:02 +02:00
  • 5cb983b353 Add basic synchronization function info to g3doc Harald Alvestrand 2021-06-01 07:46:54 +00:00
  • 3907e7b160 AudioSendStream: s/worker_queue_/rtp_transport_queue_/g Markus Handell 2021-06-01 09:07:20 +02:00
  • 58b8d297f5 fall back to payload types from lower range after exhausting [96,127] Philipp Hancke 2021-05-28 14:31:18 +02:00
  • 504fc192d0 Add ability to pass factory for RtpTransportControllerSend to PeerConnectionFactoryDependencies. Vojin Ilic 2021-05-31 14:02:28 +02:00
  • 40f1a06ca7 Update WebRTC code version (2021-06-01T04:04:24). webrtc-version-updater 2021-05-31 21:04:29 -07:00
  • 7d23535108 Populate qualityLimitationDurations stats for outbound RTP streams Byoungchan Lee 2021-05-28 21:32:04 +09:00
  • 7b4fd5ca59 dcsctp: Determine chunks to be retransmitted fast Victor Boivie 2021-05-22 22:56:07 +02:00
  • 82aa094a97 Fix incorrect SSRC in RtpPacketSendInfo for RTX packets. memetao 2021-05-31 21:16:52 +08:00
  • c48a49cd0d dcsctp: Find out quickly if to send FORWARD-TSN Victor Boivie 2021-05-22 23:05:19 +02:00
  • cc84c980c2 Fix typo in a URL in the comment Byoungchan Lee 2021-05-29 07:49:58 +09:00
  • 376cf07ea2 Add packet_sequence_checker_ to RtpVideoStreamReceiver2. Tommi 2021-05-31 19:14:38 +02:00
  • b6b7d80ea4 Roll chromium_revision fbd3997e87..03cca1960d (887673:887795) chromium-webrtc-autoroll 2021-05-31 11:03:27 -07:00
  • 90738ddb4e Split VideoReceiveStream2 init into worker / network steps. Tommi 2021-05-31 17:36:47 +02:00
  • 27d2be3583 dcsctp: Optimize SACK generation Victor Boivie 2021-05-29 21:20:12 +02:00
  • 0377bab21b Split FlexfecReceiveStreamImpl init into worker / network steps. Tommi 2021-05-31 14:26:05 +02:00
  • 261eec5456 dcsctp: Allow more outstanding fragments Victor Boivie 2021-05-29 21:27:23 +02:00
  • 8267724a85 dcsctp: Announce send buffer watermark as a_rwnd Victor Boivie 2021-05-29 22:14:34 +02:00
  • ea72ee6350 Add ClippingPredictorLevelBuffer circular buffer. Hanna Silen 2021-05-31 14:08:04 +02:00
  • 4f26a3c7e8 red: assign payload type 63 to audio/RED for opus Philipp Hancke 2021-05-27 13:47:47 +02:00
  • 5d4c3c5128 dcsctp: Add more unit tests for DataTracker Victor Boivie 2021-05-28 17:21:39 +02:00
  • 5429d71022 dcsctp: Allow heartbeats to be disabled Victor Boivie 2021-05-19 22:49:28 +02:00
  • 02df2eb1de Split AudioStream initialization into worker / network steps. Tommi 2021-05-31 12:57:53 +02:00
  • 6ad542cf11 Remove temporary using webrtc::OnCompleteFrameCallback statement. philipel 2021-05-27 14:33:23 +02:00
  • 948e40cfdf Add thread guards and constness to Call members. Tommi 2021-05-31 12:39:57 +02:00
  • cae1f1d47b Move PostTask for DeliverRtcp from PeerConnection to Call. Tommi 2021-05-31 10:51:09 +02:00
  • acd16af8c6 AudioReceiveStream: Clean up ConfigureStream. Markus Handell 2021-05-29 13:21:28 +02:00
  • c81afe365d Call: prepare receive stats for thread switch. Markus Handell 2021-05-31 09:02:01 +02:00
  • 22fead3041 Roll chromium_revision 6f6904aacc..fbd3997e87 (887571:887673) chromium-webrtc-autoroll 2021-05-31 01:02:02 -07:00
  • 5be2aa1ac3 Make generate_license.py compatible with Python 3. Byoungchan Lee 2021-05-29 13:50:31 +09:00
  • d3166afe01 Update WebRTC code version (2021-05-31T04:03:11). webrtc-version-updater 2021-05-30 21:03:14 -07:00
  • d280eaf195 Update WebRTC code version (2021-05-30T04:02:15). webrtc-version-updater 2021-05-29 21:02:17 -07:00
  • 6c94d58e34 Roll chromium_revision bbca8ebcc5..6f6904aacc (887470:887571) chromium-webrtc-autoroll 2021-05-29 11:03:22 -07:00
  • cbb4421eac Remove DeliverPacketAsync. Tommi 2021-05-28 15:26:59 +02:00
  • 785725129a Update WebRTC code version (2021-05-29T04:03:30). webrtc-version-updater 2021-05-28 21:03:31 -07:00