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  • 630c40d716 Update RtpSenderVideo::SendVideo/SendEncodedImage to take Timestamp/TimeDelta types Danil Chapovalov 2023-07-17 16:42:45 +02:00
  • 2e48e4b112 Update WebRTC code version (2023-07-20T04:13:41). webrtc-version-updater 2023-07-19 21:13:43 -07:00
  • e9e03a9160 Fix inaccurate contentType in RTCInbound/OutboundRtpStreamStats Joachim Reiersen 2023-07-19 12:47:29 -07:00
  • 2206b63af0 Prevent SDP munging of duplicate SSRCs Philipp Hancke 2023-07-18 07:23:02 +02:00
  • d3cb2f8b95 Roll chromium_revision 21b76e39ae..58a3c40eba (1172261:1172400) chromium-webrtc-autoroll 2023-07-19 09:02:33 -07:00
  • dbb89430ef Roll chromium_revision 468e35f19c..21b76e39ae (1170159:1172261) chromium-webrtc-autoroll 2023-07-19 03:02:01 -07:00
  • d3685676bf Support Reclient developer builds Junji Watanabe 2023-07-19 17:34:08 +09:00
  • b6c0ddc48d Update Fuchsia API version Bjorn Terelius 2023-07-19 10:07:02 +02:00
  • ab9458408d Update WebRTC code version (2023-07-19T04:01:48). webrtc-version-updater 2023-07-18 21:01:49 -07:00
  • 2a19c68d7b Update WebRTC code version (2023-07-18T04:10:57). webrtc-version-updater 2023-07-17 21:10:58 -07:00
  • e66a85c278 kDummyAudio now also creates Dummy ADM on Android henrika 2023-07-17 16:08:51 +02:00
  • 50d62c1adf Roll chromium_revision 9640e11e3c..468e35f19c (1165649:1170159) Björn Terelius 2023-07-17 15:41:49 +02:00
  • e2dff95ac6 Revert "Clean up WebRTC-FilterAbsSendTimeExtension field trial" Mirko Bonadei 2023-07-17 10:25:44 +02:00
  • fd5bdca28c Revert "Disable VideoCaptureTest due to flakyness" Björn Terelius 2023-07-17 08:27:15 +00:00
  • 726992d7a4 Revert "[fuchsia] remove Scenic and GFX dependencies in DesktopCapturer" Mirko Bonadei 2023-07-17 06:47:51 +00:00
  • 4272c47782 Update WebRTC code version (2023-07-17T04:01:57). webrtc-version-updater 2023-07-16 21:01:58 -07:00
  • c9302855d8 Update WebRTC code version (2023-07-16T04:02:13). webrtc-version-updater 2023-07-15 21:02:14 -07:00
  • a4592cec2c Update WebRTC code version (2023-07-15T04:12:34). webrtc-version-updater 2023-07-14 21:12:36 -07:00
  • fe5be2eb4f [fuchsia] remove Scenic and GFX dependencies in DesktopCapturer Caroline Liu 2023-07-14 17:28:22 +00:00
  • c0ed83eac2 Bail out early if the RTP send module for a SSRC was not found Philipp Hancke 2023-07-14 10:47:11 +02:00
  • 94abc09246 Update WebRTC code version (2023-07-14T04:07:27). webrtc-version-updater 2023-07-13 21:07:28 -07:00
  • aa8faa6423 Update RtcpReceiver to use Timesetamp/TimeDelta types instead of raw ints Danil Chapovalov 2023-07-13 11:00:28 +02:00
  • 775470214a Removing enum used for removed UMA metrics WebRTC.PeerConnection.Simulcast.ApplyLocalDescription and WebRTC.PeerConnection.Simulcast.ApplyRemoteDescription Angelo Tadres 2023-07-12 18:33:29 +00:00
  • cbfbb591cf Update WebRTC code version (2023-07-13T04:02:11). webrtc-version-updater 2023-07-12 21:02:12 -07:00
  • 9e5defcf74 Implement Y4mFrameGenerator::ChangeResolution() Jianhui Dai 2023-07-12 04:58:09 +08:00
  • 0e9556a90c Desktop capture: introduce capturer requesting both screen and windows Jan Grulich 2023-07-12 13:15:40 +02:00
  • 73d51f8e84 [ssci] Added Shipped field to READMEs Anne Redulla 2023-07-12 15:32:29 +10:00
  • 7b07ab93e8 Use auto to avoid implicit conversion. Taylor Brandstetter 2023-07-11 15:49:25 -07:00
  • bf6e60e5ce Issue 1447193: Remove expired histograms WebRTC.PeerConnection.Simulcast.ApplyLocalDescription, WebRTC.PeerConnection.Simulcast.ApplyRemoteDescription, WebRTC.PeerConnection.Simulcast.Disabled Angelo Tadres 2023-07-11 18:21:12 +00:00
  • 99b984a1c5 Update WebRTC code version (2023-07-12T04:12:34). webrtc-version-updater 2023-07-11 21:12:36 -07:00
  • 32a8169a65 Use common VideoFrameTypeToString helper Jianhui Dai 2023-06-30 09:57:02 +08:00
  • 4200233adc Add exception callbacks to EglThread Linus Nilsson 2023-07-11 14:12:58 +02:00
  • 6a4f409241 pacing_controller: add millisecond units to warning message Alfred E. Heggestad 2023-07-11 16:17:50 +02:00
  • ebf71114a3 Clean up WebRTC-FilterAbsSendTimeExtension field trial Philipp Hancke 2023-07-04 09:40:16 +02:00
  • 9d677f4cdc Set surrogate receive times for transformed sender frames Tony Herre 2023-07-11 12:41:06 +00:00
  • dfe026ce08 Log frame NTP timestamp in VideoEncoder::AugmentEncodedImage Philipp Hancke 2023-07-10 12:34:18 +02:00
  • 32457632a6 Update WebRTC code version (2023-07-11T04:11:40). webrtc-version-updater 2023-07-10 21:11:41 -07:00
  • 8fcc6df79d PipeWire capturer: increase buffer size to avoid buffer overflow Jan Grulich 2023-07-10 10:07:38 +02:00
  • a6c4f12fad Set noparent in api/stats/OWNERS. Henrik Boström 2023-07-10 14:04:25 +02:00
  • 4e5b89f77b Create the EGLContext on the thread it will be used on. Linus Nilsson 2023-07-10 11:24:21 +02:00
  • f186e32d9e Update WebRTC code version (2023-07-10T04:05:08). webrtc-version-updater 2023-07-09 21:05:10 -07:00
  • df81547595 Update WebRTC code version (2023-07-09T04:12:01). webrtc-version-updater 2023-07-08 21:12:02 -07:00
  • 1043047243 Update WebRTC code version (2023-07-08T04:03:39). webrtc-version-updater 2023-07-07 21:03:40 -07:00
  • 599367595d Allow StartRecording if capturer is null in test ADM Artem Titov 2023-07-07 15:14:54 +02:00
  • 2feee9da60 Update WebRTC code version (2023-07-07T04:05:37). webrtc-version-updater 2023-07-06 21:05:38 -07:00
  • e9ae738c7c rtcp_sender: uppercase protocol names (RTCP) in log messages Alfred E. Heggestad 2023-07-06 13:08:14 +02:00
  • 3f10b4917e Log SSRC for video decode errors Philipp Hancke 2023-07-04 13:31:28 +02:00
  • 26f72901c6 Mark rtc::CopyOnWriteBuffer move constructor noexcept Danil Chapovalov 2023-07-06 11:06:36 +02:00
  • cc6042d876 Let IvfVideoFrameGenerator support AV1 codec Jianhui Dai 2023-07-04 11:09:49 +08:00
  • 51712a3196 Update WebRTC code version (2023-07-05T04:11:14). webrtc-version-updater 2023-07-04 21:11:15 -07:00
  • a8b955def2 Roll chromium_revision b9b9f99812..9640e11e3c (1165535:1165649) chromium-webrtc-autoroll 2023-07-04 09:01:46 -07:00
  • 1a8c1aedbc Add raw file audio capturer/renderer for test ADM Artem Titov 2023-07-04 00:57:05 +02:00
  • 804ff860e6 Roll chromium_revision ffa2726099..b9b9f99812 (1165420:1165535) chromium-webrtc-autoroll 2023-07-04 03:01:42 -07:00
  • 5f4a7e004e Log last received RTP timestamp when requesting a PLI due to timeout Philipp Hancke 2023-07-03 11:18:37 +02:00
  • c2919fe1e2 Ugrade xcode version from 13 to 14. Jeremy Leconte 2023-07-04 10:46:32 +02:00
  • 758f26852d Fix downstream review comments for C++20 Li-Yu Yu 2023-07-04 15:35:23 +08:00
  • d4bd50b018 Update WebRTC code version (2023-07-04T04:12:28). webrtc-version-updater 2023-07-03 21:12:28 -07:00
  • 0725072604 Roll chromium_revision d13ed2a18f..ffa2726099 (1165315:1165420) chromium-webrtc-autoroll 2023-07-03 17:01:58 -07:00
  • aad4ae54d4 Roll chromium_revision 4e064eab42..d13ed2a18f (1164089:1165315) chromium-webrtc-autoroll 2023-07-03 07:02:29 -07:00
  • 283c3df2ac Fix TCPChannelClientTest.java when running on multiple shards. Jeremy Leconte 2023-07-03 14:36:41 +02:00
  • e21745a78b Video Capture PipeWire: initialize pw_stream raw pointer member Jan Grulich 2023-07-03 14:15:33 +02:00
  • 7a35f108c7 Remove dead code in offer/answer handling Philipp Hancke 2023-07-01 09:47:34 +02:00
  • d51dada719 Update WebRTC code version (2023-07-01T04:12:31). webrtc-version-updater 2023-06-30 21:12:33 -07:00
  • 2cf8eb9f78 Reland "Migrate TestAudioDeviceModule on AudioDeviceModuleImpl" Artem Titov 2023-06-30 15:26:09 +02:00
  • 6fc700ec3d Rland "Revert "Reland "Reland "Delete old Android ADM."""" Yaowen Guo 2023-04-28 14:10:37 +08:00
  • becfe2e571 Make opengl targets ios-only. Kári Tristan Helgason 2023-06-30 13:41:30 +02:00
  • 415e30fdbb Extract some test code out from audio_device_impl into own targets Artem Titov 2023-06-29 16:58:57 +02:00
  • 9948623027 Use AVCaptureDeviceDiscoverySession on all platforms Kári Tristan Helgason 2023-06-30 10:40:40 +02:00
  • 079b36bd4f Update WebRTC code version (2023-06-30T04:03:17). webrtc-version-updater 2023-06-29 21:03:18 -07:00
  • fad6617506 Roll chromium_revision c94754fca5..4e064eab42 (1163989:1164089) chromium-webrtc-autoroll 2023-06-29 07:02:14 -07:00
  • ba50cb322c Reland "Delete deprecated NSGLVideoView." Kári Tristan Helgason 2023-06-28 14:32:04 +02:00
  • f92cc6d7b4 Reland: FrameGeneratorCapturer: don't generate video before Start is called Artem Titov 2023-06-29 14:56:59 +02:00
  • b4062e5611 Add a setter for RTPTimestamp on TransformableFrameInterface Tony Herre 2023-06-29 09:02:22 +00:00
  • cabd77a5c7 Remove flexfec-03 killswitch guarding receiving FlexFEC Philipp Hancke 2023-06-27 15:05:00 +02:00
  • d797cb6ca7 Remove all split channels related code Florent Castelli 2023-06-27 20:07:09 +00:00
  • 9c62b32d28 Roll chromium_revision 8e1bd939a3..c94754fca5 (1163873:1163989) chromium-webrtc-autoroll 2023-06-29 01:02:05 -07:00
  • 9e2567cfa1 Update WebRTC code version (2023-06-29T04:02:14). webrtc-version-updater 2023-06-28 21:02:15 -07:00
  • 396d8c8a44 Roll chromium_revision 665afad9ea..8e1bd939a3 (1163663:1163873) chromium-webrtc-autoroll 2023-06-28 17:17:56 -07:00
  • 2d7ccb4149 Revert "FrameGeneratorCapturer: don't generate video before Start is called" Mirko Bonadei 2023-06-28 18:24:10 +00:00
  • 7e4b2a5265 Roll chromium_revision 2ce2f71c6c..665afad9ea (1163494:1163663) chromium-webrtc-autoroll 2023-06-28 11:03:08 -07:00
  • ea668e36a9 Revert "Delete deprecated NSGLVideoView." Mirko Bonadei 2023-06-28 14:44:37 +00:00
  • 7a24f2a7eb Call generate_buildbot_json.py to update json files. Jeremy Leconte 2023-06-28 15:32:04 +02:00
  • 54d7547faf Delete deprecated NSGLVideoView. Kári Tristan Helgason 2023-06-28 14:32:04 +02:00
  • 533a97be2b Roll chromium_revision eb64addd52..2ce2f71c6c (1162930:1163494) chromium-webrtc-autoroll 2023-06-28 05:01:13 -07:00
  • 1f31c201cd Split fake media channel classes Florent Castelli 2023-06-26 00:26:07 +00:00
  • 96293f0876 Remove usage of CreateMediaChannel in webrtc_voice_engine_unittest Florent Castelli 2023-06-27 16:09:12 +00:00
  • ceabb9e8e5 Update WebRTC code version (2023-06-28T04:03:42). webrtc-version-updater 2023-06-27 21:03:43 -07:00
  • a2e945e042 [rtc_tools/video_encoder] Output ivf for all SVC decode targets Jianhui Dai 2023-06-19 09:36:42 +08:00
  • be90237a0a rtp_rtcp/source: fix some minor typos Alfred E. Heggestad 2023-06-23 10:22:25 +02:00
  • c26d96ac56 Roll chromium_revision 24cee86ed6..eb64addd52 (1162819:1162930) chromium-webrtc-autoroll 2023-06-27 07:02:02 -07:00
  • 3adaeefbc6 Fix TimeUTCMicros resolution on Windows Philipp Hancke 2023-06-27 13:08:44 +02:00
  • 58ee9dff08 Deprecate encoded audio frame GetHeader Tony Herre 2023-06-26 15:30:06 +00:00
  • 0606eafb9f Make WebRTC-EventLogNewFormat default. Lionel Koenig 2023-06-27 08:46:19 +00:00
  • 299cdc9057 Revert mid check in SdpOfferAnswerHandler::CreateDataChannel. Tommi 2023-06-26 21:11:35 +02:00
  • 365a5717ae Use absl::optional instead of std::optional Emil Lundmark 2023-06-27 11:35:22 +02:00
  • e4e33b8ee3 Add video_encoder to default build Philipp Hancke 2023-06-14 16:55:54 +02:00
  • 69998be21c Roll chromium_revision 4492fcc3f0..24cee86ed6 (1162492:1162819) chromium-webrtc-autoroll 2023-06-27 01:01:53 -07:00
  • 9a3ab3dcca Add a method to log AnalyzingVideoSink metrics. Jeremy Leconte 2023-06-23 16:58:02 +02:00
  • 34589929fe Add audio energy metric. Jeremy Leconte 2023-06-27 09:49:13 +02:00
  • 38aa4ef5f4 Update WebRTC code version (2023-06-27T04:03:35). webrtc-version-updater 2023-06-26 21:03:38 -07:00