Commit Graph

  • 885d538cdd ModuleRtpRtcpImpl2: remove RTCP send polling. Markus Handell 2021-06-21 18:57:36 +02:00
  • 20862093e5 Update WebRTC code version (2021-06-22T04:05:30). webrtc-version-updater 2021-06-21 21:05:34 -07:00
  • 049ed447b0 ModuleRtpRtcpImpl2: update test code. Markus Handell 2021-06-18 13:45:50 +02:00
  • fb7fd24b3d Removing RTC_SUPPORTS_METAL compilation flag. This flag is a holdover from before either macOS or the iOS Simulator supported Metal rendering. Jake Bromberg 2021-06-14 13:44:25 -07:00
  • c6b9ac782a RTCPSender: migrate to Timestamp. Markus Handell 2021-06-18 13:44:51 +02:00
  • e2ab77ba57 Reland "Port: migrate to TaskQueue." Markus Handell 2021-06-21 18:20:55 +00:00
  • a27cfbffdf Fix echo return loss stats and add to RTCAudioSourceStats. Taylor Brandstetter 2021-06-20 14:59:48 -07:00
  • 2e3edc1da9 RTCPSender: migrate to own configuration struct. Markus Handell 2021-06-18 13:44:13 +02:00
  • f906ec40d4 Handle null return from ToI420 in encoders Evan Shrubsole 2021-06-21 13:47:44 +02:00
  • 76a35d9ce2 Delete legacy RtpHeaderParser wrapper Danil Chapovalov 2021-06-08 12:30:46 +02:00
  • 257f81b98e Update VirtualSocketServer locking to match documentation. Niels Möller 2021-06-17 16:58:59 +02:00
  • a4aabb9213 Revert "Port: migrate to TaskQueue." Markus Handell 2021-06-21 07:02:43 +00:00
  • 06540166ca Port: migrate to TaskQueue. Markus Handell 2021-06-07 01:29:51 +02:00
  • 3f6efdc9a0 Update WebRTC code version (2021-06-21T04:05:45). webrtc-version-updater 2021-06-20 21:05:49 -07:00
  • ae278d47da openssl_adapter: document SSL_CTX_set_verify_depth behaviour Philipp Hancke 2021-06-16 10:26:56 +02:00
  • fbe995874f Update WebRTC code version (2021-06-20T04:03:02). webrtc-version-updater 2021-06-19 21:03:04 -07:00
  • 4cacaf7c8f Update WebRTC code version (2021-06-19T04:03:03). webrtc-version-updater 2021-06-18 21:03:05 -07:00
  • 7c719b0db1 Fixes off-by-one error in video capture module Johannes Kron 2021-06-17 10:37:22 +02:00
  • bad0ab0858 Delete unused class MockDelayable Niels Möller 2021-06-18 15:11:14 +02:00
  • c6d76489e3 Add jakobi to modules/audio_coding OWNERS Ivo Creusen 2021-06-18 11:14:54 +00:00
  • 6a11c844fd dcsctp: Add DcSctpSocketFactory Florent Castelli 2021-06-01 18:39:49 +02:00
  • c20f1563b6 dcsctp: Don't sent more packets before COOKIE ACK Victor Boivie 2021-06-16 12:52:42 +02:00
  • 95c30413db Update WebRTC code version (2021-06-18T04:03:27). webrtc-version-updater 2021-06-17 21:03:29 -07:00
  • 42dacda82c AGC analog clipping predictor: integrate evaluator Alessio Bazzica 2021-06-17 17:18:46 +02:00
  • 7d5418233d Avoid assembling complicated but unused video rtp header extensions Danil Chapovalov 2021-06-17 15:27:12 +02:00
  • afb28116d3 Catch possible RuntimeException from getCameraCharacteristics Xavier Lepaul 2021-06-16 09:43:06 +02:00
  • 11b92cfa65 Refactoring: Move groups-by-mid into Bundle manager Harald Alvestrand 2021-06-17 14:03:09 +00:00
  • de22ab2850 Apply IWYU to jsep_transport_controller/collection Harald Alvestrand 2021-06-17 10:45:33 +00:00
  • d354ced5ac Mark VideoSendTiming flags as invalid by default. philipel 2021-06-17 11:34:35 +02:00
  • ada810aab2 Reland "Deprecate microsecond timestamps in RTC event log." Björn Terelius 2021-05-24 10:19:55 +02:00
  • 1bb36d2c77 Change YuvConverter.convert to catch GLExceptions and return null. Fabian Bergmark 2021-06-17 11:53:24 +02:00
  • ac82bd386a Add timestamp to log message in generic_decoder.cc Johannes Kron 2021-06-17 10:50:56 +02:00
  • 41c700dbd1 Remove unnused build configs for M1 builder Christoffer Jansson 2021-06-16 12:05:52 +02:00
  • 82f21fdb99 Make WebRtcAudioReceiveStream::stream_ const. Tommi 2021-06-17 08:55:35 +02:00
  • b4100ad06a Avoid using legacy rtp parser in neteq test::Packet Danil Chapovalov 2021-06-16 14:23:22 +02:00
  • 35b21ba8d4 In RtcpTransceiver avoid extra PostTask during construction Danil Chapovalov 2021-06-16 17:43:00 +02:00
  • f9d5e55a31 Revert "Avoid video stream allocation on configuration change after timeout." Jakob Ivarsson 2021-06-16 16:03:16 +00:00
  • a3796c8090 Revert the send-side bwe behavior to slow ramp-up on lifted REMB cap. Christoffer Rodbro 2021-06-16 11:07:06 +02:00
  • ce3b3ba3cd Update WebRTC code version (2021-06-17T04:05:50). webrtc-version-updater 2021-06-16 21:05:52 -07:00
  • 4b629521b9 Roll chromium_revision 6ade74989a..6f7025c98c (893176:893293) chromium-webrtc-autoroll 2021-06-16 21:02:35 -07:00
  • e0c73656fa Roll chromium_revision 19c2bebe7d..6ade74989a (893060:893176) chromium-webrtc-autoroll 2021-06-16 15:01:54 -07:00
  • a2a073bc0b Reformat pc/g3doc/rtp.md Artem Titov 2021-06-16 16:36:48 +02:00
  • 55107c8507 Update the sync_group id without recreating audio receive streams. Tommi 2021-06-16 16:31:18 +02:00
  • 25029c4135 Roll chromium_revision b452ca696d..19c2bebe7d (892948:893060) chromium-webrtc-autoroll 2021-06-16 11:02:06 -07:00
  • 355c47309d Fix VideoRtpDepacketizerVp{8,9} copy assignment signature. philipel 2021-06-16 16:50:38 +02:00
  • 5b9d0c70c2 AGC1 add clipping predictor evaluator Alessio Bazzica 2021-06-16 14:55:40 +02:00
  • 808f49428f LOG DTLS (failed) handshake retransmission Jonas Oreland 2021-06-16 13:42:20 +02:00
  • d579e6bc7b dcsctp: Do explicit bounds checking in bounded IO Victor Boivie 2021-06-09 09:24:26 +02:00
  • 72b79986a5 Remove the createDecoder(String) overload Xavier Lepaul 2021-06-16 13:08:42 +02:00
  • 130e031010 Roll chromium_revision 570a173256..b452ca696d (892156:892948) chromium-webrtc-autoroll 2021-06-16 03:02:30 -07:00
  • 98ff0280ce AGC analog ClippingPredictor refactoring 2/2 Alessio Bazzica 2021-06-16 11:24:31 +02:00
  • 08be9baaa3 Don't recreate the audio receive stream when updating the local_ssrc. Tommi 2021-06-15 23:01:57 +02:00
  • bc03259de7 Define generate_location_tags gn arg Björn Terelius 2021-06-16 10:47:17 +02:00
  • 6a0a55907b Reland "Correctly handle retransmissions/padding in early loss detection." Erik Språng 2021-06-15 19:04:24 +02:00
  • c03d6e965e Support Java_Buffer_toI420 returning null Fabian Bergmark 2021-06-15 15:46:56 +02:00
  • cd430c84d9 Update WebRTC code version (2021-06-16T04:05:58). webrtc-version-updater 2021-06-15 21:05:59 -07:00
  • d6957c2eed Revert "Correctly handle retransmissions/padding in early loss detection." Erik Språng 2021-06-15 15:58:40 +00:00
  • e9ae4729e0 Correctly handle retransmissions/padding in early loss detection. Erik Språng 2021-06-14 15:29:00 +02:00
  • e3ceb88c72 Sanitize hostname literals when mDNS obfuscation is on. Harald Alvestrand 2021-06-15 15:02:41 +02:00
  • be53049555 Reland "Avoid sending empty receiver reports with RtcpTransceiver" Danil Chapovalov 2021-06-15 11:47:16 +00:00
  • 7a2db8acbe Modify Bundle logic to not add & destroy extra transport at add-track Harald Alvestrand 2021-06-14 15:41:30 +00:00
  • e4eb8af08f libstdc++: fix ostream operator<< usage in JsepTransportCollection Stephan Hartmann 2021-06-14 12:22:04 +00:00
  • 07bf5b5d78 Update WebRTC code version (2021-06-15T04:04:38). webrtc-version-updater 2021-06-14 21:04:41 -07:00
  • 3008bcd588 Don't recreate audio receive streams on header extension update. Tommi 2021-06-14 20:00:04 +02:00
  • 6bbe1a47e1 Roll chromium_revision e9261a56ad..570a173256 (892013:892156) chromium-webrtc-autoroll 2021-06-14 11:02:57 -07:00
  • d350006b70 Add rtp_config() accessor to ReceiveStream. Tommi 2021-06-14 19:39:45 +02:00
  • 48420fa947 Revert "Avoid sending empty receiver reports with RtcpTransceiver" Björn Terelius 2021-06-14 17:28:50 +00:00
  • 1c1f540487 Factor out common receive stream methods to a common interface. Tommi 2021-06-14 10:54:20 +02:00
  • e097282f11 Avoid recreating the audio stream when a frame decryptor is set. Tommi 2021-06-14 08:11:10 +02:00
  • e5f1a3992e Avoid sending empty receiver reports with RtcpTransceiver Danil Chapovalov 2021-06-14 16:56:02 +02:00
  • 8b6929081e Fix VideoStreamEncoder QP tests to not use SetHasInternalSource Niels Möller 2021-06-14 12:04:57 +02:00
  • b237a87a25 AGC analog ClippingPredictor refactoring 1/2 Alessio Bazzica 2021-06-11 12:37:54 +02:00
  • 1ff491b0bb Roll chromium_revision 8907aace7e..e9261a56ad (891631:892013) chromium-webrtc-autoroll 2021-06-14 03:02:11 -07:00
  • 74cc9eaca3 Don't register invalid encode complete callbacks. Peter Hanspers 2021-06-11 15:52:57 +02:00
  • 10814873c5 Avoid video stream allocation on configuration change after timeout. Jakob Ivarsson 2021-06-10 14:45:21 +02:00
  • 13dac0c268 Update WebRTC code version (2021-06-13T04:02:18). webrtc-version-updater 2021-06-12 21:02:20 -07:00
  • 0f9a8e33c0 Make stopping of the RepeatingTask safer Danil Chapovalov 2021-06-11 18:39:17 +02:00
  • 4bb81aca75 Make JsepTransportCollection self-managing for transports Harald Alvestrand 2021-06-11 13:04:59 +00:00
  • 63c96ce3c7 Roll chromium_revision d2f297f391..8907aace7e (890623:891631) chromium-webrtc-autoroll 2021-06-11 07:02:59 -07:00
  • 8d3396dabe In vp9 encoder fuzzer reduce information stored for older frames Danil Chapovalov 2021-06-11 15:43:20 +02:00
  • a63d152423 AEC3: Unbounded echo spectrum for dominant nearend detection. Gustaf Ullberg 2021-06-11 14:02:53 +02:00
  • 1b4807ff65 count webrtc pranswer usage Philipp Hancke 2021-06-11 11:49:54 +02:00
  • b22abbc11d Add kron as owner of api/uma_metrics.h Johannes Kron 2021-06-11 12:40:31 +02:00
  • ef4edafe8f Remove DEPS that was removed from chromium Andrey Logvin 2021-06-11 11:03:00 +00:00
  • ec6b655ba6 Break out pc/session_description build target (part 2) Harald Alvestrand 2021-06-10 06:51:48 +00:00
  • f5f7e8e806 Ensure that fps adaptation count can go back to zero when framerate is unrestricted. Åsa Persson 2021-06-09 22:55:38 +02:00
  • c63ae48871 Prepare for breakout of session_description.{h,cc} Harald Alvestrand 2021-06-10 10:48:31 +00:00
  • 62ec0f6130 Add small cooldown to unsignalled ssrc stream creation. Henrik Boström 2021-06-09 11:10:15 +02:00
  • ba7da8b9d2 Relax expectation in OveruseFrameDetectorTest2.ConvergesSlowly Niels Möller 2021-06-09 16:52:03 +02:00
  • 9dea393f17 Move MID/JsepTransport mappings into a new manager object. Harald Alvestrand 2021-06-10 06:03:06 +00:00
  • 64e3a36668 Update WebRTC code version (2021-06-10T04:05:52). webrtc-version-updater 2021-06-09 21:05:55 -07:00
  • 5e65dd5f25 Add MB configs for M1 bots Christoffer Jansson 2021-06-09 18:59:15 +02:00
  • 3cc68ec32e Report stats from ChannelReceive::GetAudioFrameWithInfo at 1Hz. Tommi 2021-06-09 19:30:41 +02:00
  • e2e046452a Remove a couple of locks from ChannelReceive and add thread checks. Tommi 2021-06-09 16:11:11 +02:00
  • b56a63e470 dcsctp: Prevent overflow of missing parameters Victor Boivie 2021-06-09 08:55:01 +02:00
  • 6eda26c550 Reland "Remove AudioReceiveStream::Reconfigure() method." Tommi 2021-06-09 13:46:28 +02:00
  • c0a95863bd Break out pc/session_description in its own build target (part 1) Harald Alvestrand 2021-06-09 09:33:56 +00:00
  • 8a18e5b3c9 Revert "Remove AudioReceiveStream::Reconfigure() method." Andrey Logvin 2021-06-09 10:15:47 +00:00
  • e2561e17e2 Remove AudioReceiveStream::Reconfigure() method. Tommi 2021-06-08 16:55:47 +02:00
  • 4ea80f35f1 Disable PT based demuxing if MID header extension is present. Henrik Boström 2021-06-09 10:29:50 +02:00