Google Git
Sign in
webrtc / src / HEAD
  1. 51dbe82 setOfferedHeaderExtensions: stop any filtered extension by Philipp Hancke · 3 hours ago lkgr main master
  2. e9c3e51 Add a DEPS hook to download llvm-cov and llvm-profdata based on .gclient custom_vars. by Björn Terelius · 18 hours ago
  3. c7a0620 Add an ICE switch reason for a switch requested by an application. by Sameer Vijaykar · 20 hours ago
  4. 95d12ad Create unit test for the population of capture_start_ntp_time by Harald Alvestrand · 22 hours ago
  5. 4b0d6f9 Upgrade Linux MSan to Focal by Tom Anderson · 2 days ago
  6. b0e1cb2 Adds WebRTC.DesktopCapture.Win.DirectXCapturerResult UMA by henrika · 3 days ago
  7. fd29662 Fix typo in histogram name by Johannes Kron · 4 days ago
  8. 5e7301f Remove rid and rrid from list of extensions that can be used for audio by Philipp Hancke · 4 days ago
  9. c6ff4bc Do not transfer ownership of codecs to tester by Sergey Silkin · 4 days ago
  10. be9b576 Move video video receiver transformable frame to modules/rtc_rtcp/source by Tony Herre · 4 days ago
  11. b311f6a Add UMA histograms to track usage of fullscreen detection by Johannes Kron · 4 days ago
  12. 85abbdf RtcEventLogImpl: Add test cases by Xuanxi Leng · 4 days ago
  13. d1831cb Treat non DTLS/SCTP Protocol Based Data Channels as Unsupported Media by Dor Hen · 4 days ago
  14. f0be3be Add pipewire/portal video capture support by Michael Olbrich · 5 days ago
  15. fad9a6d Delete deprecated Create method and config from AudioCodingModule by Henrik Lundin · 5 days ago
  16. 101c6aa Remove leftover function signatures. by Fredrik Solenberg · 5 days ago
  17. 6c60f72 Refactor video codec testing stats by Sergey Silkin · 5 days ago
  18. 97d1c34 Enable rotation tests marked as expected failures by Andreas Pehrson · 5 days ago
  19. 65ab5fd Cleanup RemoteEstimatorProxy::IncomingPacket by Danil Chapovalov · 5 days ago
  20. ba846cc Add a test that shows when channel_receive fires RR by Harald Alvestrand · 6 days ago
  21. 84f7569 Break apart AudioCodingModule and AcmReceiver by Henrik Lundin · 6 days ago
  22. c5455e7 Allow RTX ssrc to be updated on receive streams by Per K · 6 days ago
  23. be03c09 Only serialize non-stopped RTP header extensions by Philipp Hancke · 6 days ago
  24. 1f206b8 Use ArrayView in the IncomingRtcpPacket function. by Harald Alvestrand · 6 days ago
  25. 16a8792 Propagate received video csrcs to encodedframe metadata by Tony Herre · 6 days ago
  26. 0507fbd [Desktop Capture] Remove disabled test. by mark a. foltz · 6 days ago
  27. cb4b0a6 Check FMA3 support before use it in SincResampler by Henrik Lundin · 7 days ago
  28. 0b7184c Add possibility to set MetricsSet metadata. by Mirko Bonadei · 7 days ago
  29. 217b384 Remove rtp header extension from config of Call audio and video receivers by Per K · 7 days ago
  30. 3541732 Add a Config struct to AcmReceiver, and a ctor using it by Henrik Lundin · 7 days ago
  31. 3274051 Add method to get FD for physical socket by Artem Titov · 7 days ago
  32. d78f8e7 Fix doc path by Artem Titov · 7 days ago
  33. e592283 Add 'metadata' field to MetricsSet proto. by Mirko Bonadei · 7 days ago
  34. a617867 Reland "Migrate WebRTC documentation to new renderer" by Artem Titov · 7 days ago
  35. 2367103 Add android-tiramisuprivacysandbox to DEPS. by Mirko Bonadei · 7 days ago
  36. 3963a95 Enforce policy that SDP munging requires special approval by Harald Alvestrand · 8 days ago
  37. 4822031 shared_screencast_stream: Set WL capturer id by Salman · 8 days ago
  38. 9214718 Roll chromium_revision 2fd12db504..e182675fbb (1098462:1098562) by chromium-webrtc-autoroll · 8 days ago
  39. dad91a6 Send periodic TransportFeedback based on extension version by Per K · 8 days ago
  40. 2ded55e Cleanup Thread::BlockingCall by Danil Chapovalov · 8 days ago
  41. 8155227 sdp: add rtcp-fb:* lines for common feedback by Philipp Hancke · 8 days ago
  42. 68564bb [infra] Clean up mb_config.pyl after reclient migration by Junji Watanabe · 8 days ago
  43. 942abaa Roll chromium_revision 7faf2e057e..2fd12db504 (1098356:1098462) by chromium-webrtc-autoroll · 8 days ago
  44. f6a1f7e Roll chromium_revision ff5a43c15d..7faf2e057e (1098253:1098356) by chromium-webrtc-autoroll · 9 days ago
  45. 863a07f Update WebRTC code version (2023-01-29T04:02:18). by webrtc-version-updater · 9 days ago
  46. dc7a14b Roll chromium_revision 5195bd9e41..ff5a43c15d (1098136:1098253) by chromium-webrtc-autoroll · 10 days ago
  47. 6ff7713 base_capturer_pipewire: Send frames via callback by Salman · 10 days ago
  48. bf216a7 Roll chromium_revision d9876846b5..5195bd9e41 (1097987:1098136) by chromium-webrtc-autoroll · 11 days ago
  49. 26340b0 desktop_capturer: Support frame rate negotiation via pipewire by Salman · 11 days ago
  50. 4c4566a Roll chromium_revision 6da6f5ebed..d9876846b5 (1097837:1097987) by chromium-webrtc-autoroll · 11 days ago
  51. 64ce699 Propagate Video CSRCs modified by an insertable streams frame transform by Tony Herre · 11 days ago
  52. db20831 Update RTP timestamp based on capture timestamp when audio send stream is resumed. by Jakob Ivarsson · 11 days ago
  53. 9d21eb3 Roll chromium_revision 1f33300013..6da6f5ebed (1097480:1097837) by chromium-webrtc-autoroll · 11 days ago
  54. a2653bc Export some more symbols for use in chromium tests. by Sameer Vijaykar · 11 days ago
  55. ca1cfd4 Add missing `src/third_party/jdk11` dependency and roll chromium into webrtc by landrey · 11 days ago
  56. 0f2ce5c Revert "Migrate WebRTC documentation to new renderer" by Artem Titov · 12 days ago
  57. b0d8a37 Ensure CallTest derived tests per default set min/max audio bitrate. by Per K · 12 days ago
  58. 6c032cb in rtcp::TransportFeedback do not memorise all described packet by Danil Chapovalov · 12 days ago
  59. dcb09ff Reset encoder when audio send stream is stopped. by Jakob Ivarsson · 12 days ago
  60. 3eceaf4 Migrate WebRTC documentation to new renderer by Artem Titov · 12 days ago
  61. 94d5f6a Add missing include by Bjorn Terelius · 12 days ago
  62. 66efab2 Measure RTCPMuxPolicy at time of connect by Philipp Hancke · 12 days ago
  63. 73e0cc8 Delete unused Audio Bwe integration test. by Per K · 12 days ago
  64. cfbb247 Update WebRTC code version (2023-01-26T04:01:54). by webrtc-version-updater · 12 days ago
  65. e15b9ff Add a basic unittest for webrtc::voe::ChannelReceive by Harald Alvestrand · 13 days ago
  66. 664cf14 Reland "Delete PacketReceiver::DeliverPacket from all implementations" by Per K · 13 days ago
  67. 7a67dce prefer absl::optional for rtx-time by Philipp Hancke · 13 days ago
  68. 7c43d24 Roll chromium_revision a484be4b74..e5191e93ab (1096680:1096792) by chromium-webrtc-autoroll · 13 days ago
  69. 5adc2b6 Correct RTCAudioPlayoutStats type and add kind field. by Fredrik Hernqvist · 13 days ago
  70. 0c1c972 Fix gtest-output and resultdb for fuchsia by Christoffer Jansson · 13 days ago
  71. 5683a12 Increase expiration timeout for Perf bots by landrey · 13 days ago
  72. 5671c64 Stop overriding extensions in rampup tests by Per K · 13 days ago
  73. 8773349 Update win10 mixins. by Jeremy Leconte · 13 days ago
  74. 92dcc2d Roll chromium_revision 98774e2693..a484be4b74 (1096567:1096680) by chromium-webrtc-autoroll · 13 days ago
  75. 10f1bf3 Remove unused enum `FrameCombiner::LimiterType` by Alessio Bazzica · 13 days ago
  76. f2a083f Revert "Delete PacketReceiver::DeliverPacket from all implementations" by Andrey Logvin · 13 days ago
  77. 07577b5 Update WebRTC code version (2023-01-25T04:11:56). by webrtc-version-updater · 13 days ago
  78. 508979b Roll chromium_revision 87e5077aae..98774e2693 (1096404:1096567) by chromium-webrtc-autoroll · 13 days ago
  79. cc1c932 Roll chromium_revision 78e2f876c5..87e5077aae (1096262:1096404) by chromium-webrtc-autoroll · 14 days ago
  80. ea36cc2 Roll chromium_revision 632253d282..78e2f876c5 (1096118:1096262) by chromium-webrtc-autoroll · 14 days ago
  81. 897ea04 Delete PacketReceiver::DeliverPacket from all implementations by Per K · 14 days ago
  82. 0540627 SVC: Add test for SVC fallback by Florent Castelli · 14 days ago
  83. 1e43ce6 Roll chromium_revision f8b9751f30..632253d282 (1095656:1096118) by chromium-webrtc-autoroll · 14 days ago
  84. e2c29c5 Use PacketReceiver::DeliverRtpPacket in RtpReplayer by Per K · 2 weeks ago
  85. 0793ee7 Remove FakePortAllocator's dependency on ScopedKeyValueConfig. by Sameer Vijaykar · 2 weeks ago
  86. ace52a8 [infra] Remove CQ shadow builders with reclient by Junji Watanabe · 2 weeks ago
  87. 2810c14 [infra] Add todo for reclient migration cleanup by Junji Watanabe · 2 weeks ago
  88. 22821de Make capture timestamp optional in ADM. by Jakob Ivarsson · 2 weeks ago
  89. 6e62729 Roll chromium_revision cdf104c2c9..f8b9751f30 (1095545:1095656) by chromium-webrtc-autoroll · 2 weeks ago
  90. 5b55b27 Version 3: Various changes on the pre-echo delay estimator: by Jesús de Vicente Peña · 2 weeks ago
  91. 438b5b4 WebRtcVideoChannel creates default stream with dummy SSRC on received RTX packet. by Per K · 2 weeks ago
  92. 05ce032 Remove the //rtc_base target by Florent Castelli · 2 weeks ago
  93. 9ad10bc Only generate codec stats for the voice send and receive codec by Philipp Hancke · 2 weeks ago
  94. 41748c9 Roll chromium_revision a3aaddaf0c..cdf104c2c9 (1095442:1095545) by chromium-webrtc-autoroll · 2 weeks ago
  95. d3b9e71 Migrate linux_libfuzzer_rel to use reclient by Junji Watanabe · 2 weeks ago
  96. d506651 Update WebRTC code version (2023-01-23T04:11:41). by webrtc-version-updater · 2 weeks ago
  97. 5e7ae14 Roll chromium_revision 0c3ca8fbc7..a3aaddaf0c (1095332:1095442) by chromium-webrtc-autoroll · 2 weeks ago
  98. c8270b1d Roll chromium_revision e0c02c1406..0c3ca8fbc7 (1095203:1095332) by chromium-webrtc-autoroll · 2 weeks ago
  99. d2c15aa Roll chromium_revision 92aea4500e..e0c02c1406 (1095006:1095203) by chromium-webrtc-autoroll · 3 weeks ago
  100. 62ba379 Add some RTC_EXPORT needed by Chromium. by Mirko Bonadei · 3 weeks ago