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webrtc / src / 2f3b75d30d1664cd9b0c2c7285c1c31974147c3d
  1. 2f3b75d Reset prev_unwrapped_timestamp_ in TimestampExtrapolator::Reset by philipel · 1 year, 1 month ago
  2. ed050bf Remove TRACE_ASYNC without matching TRACE_BEGIN in rtc_sender_video by Evan Shrubsole · 1 year, 1 month ago
  3. 4c990e2 dcsctp: Add per-stream-limit, refactor limits. by Victor Boivie · 1 year, 1 month ago
  4. 0fa9088 Deprecate VideoFrame::timestamp() and set_timestamp by Per K · 1 year, 1 month ago
  5. 2725317 Propagate Environment through SimulcastEncoderAdapter when provided by Danil Chapovalov · 1 year, 1 month ago
  6. b8abf51 Remove TRACE_ASYNC without matching TRACE_BEGIN in rtc_sender_audio by Evan Shrubsole · 1 year, 1 month ago
  7. 6417bbf Change Port::Type() to IceCandidateType by Tommi · 1 year, 1 month ago
  8. 9849bfd Remove unused TRACE_*COPY* macros by Evan Shrubsole · 1 year, 1 month ago
  9. c6e502e Update WebRTC code version (2024-03-13T04:03:28). by webrtc-version-updater · 1 year, 1 month ago
  10. 4473d75 Add TCP keep-alive options to rtc::Socket by Tim Na · 1 year, 1 month ago
  11. f3096af Propagate Environment to create VideoEncoder through java wrappers by Danil Chapovalov · 1 year, 1 month ago
  12. afaae4e Remove remaining .cc files from rtc_media_base by Harald Alvestrand · 1 year, 1 month ago
  13. 8df31c9 Propagate ECN information on posix sockets to rtc::ReceivedPacket by Per K · 1 year, 1 month ago
  14. 329f0ea Provide Environment when creating VideoEncoder in test code by Danil Chapovalov · 1 year, 1 month ago
  15. a1d8665 Allow including internal-only tryjobs via a footer by Oleh Prypin · 1 year, 1 month ago
  16. 1fc79ce Temporarily remove Linux MSan from LKGR by Björn Terelius · 1 year, 1 month ago
  17. a70274a Remove duplicated parentheses from deprecated attribute by Keiichi Enomoto · 1 year, 1 month ago
  18. cd3d29b pc: Simplify StreamId class by Victor Boivie · 1 year, 1 month ago
  19. b4913a5 Add factory functions to pass Environment to VideoEncoders by Danil Chapovalov · 1 year, 1 month ago
  20. 83d29d5 Remove GetScalabilityMode2. by Jeremy Leconte · 1 year, 1 month ago
  21. 793add9 Temporarily remove linux_msan from cq by Björn Terelius · 1 year, 1 month ago
  22. 0268a05 Update WebRTC code version (2024-03-09T04:12:29). by webrtc-version-updater · 1 year, 1 month ago
  23. 0242939 Reland "Deprecate old constructors and set_type() in Candidate and Port" by Tomas Gunnarsson · 1 year, 1 month ago
  24. 17e3580 Add AV1 encoder speed setting for screen share by Johannes Kron · 1 year, 1 month ago
  25. 9a9f6a8 Add VideoEncoderFactory::Create to pass Environment for VideoEncoder construction by Danil Chapovalov · 1 year, 1 month ago
  26. cd54fd8 sctp: Pass webrtc::Environment to DcSctpTransport by Victor Boivie · 1 year, 1 month ago
  27. 4c1c915 Update WebRTC code version (2024-03-08T04:01:32). by webrtc-version-updater · 1 year, 1 month ago
  28. 51f98ccb Prepare the removal of GetScalabilityMode2. by Jeremy Leconte · 1 year, 1 month ago
  29. b41f07b Explicitly initialize the SctpTransportState to kNew by Bjorn Terelius · 1 year, 1 month ago
  30. d055f77 Delete legacy name AudioLevel in favor of the AudioLevelExtension by Danil Chapovalov · 1 year, 1 month ago
  31. ed8390d Revert "Deprecate old constructors and set_type() in Candidate and Port" by Ilya Nikolaevskiy · 1 year, 1 month ago
  32. dd39c03 Update WebRTC code version (2024-03-07T04:13:24). by webrtc-version-updater · 1 year, 1 month ago
  33. aaa6851 Deprecate old constructors and set_type() in Candidate and Port by Tommi · 1 year, 1 month ago
  34. ac2720e Remove unnecessary RtcEventLog parameter in RtpTransportControllerSend::CreateRtpVideoSender by Danil Chapovalov · 1 year, 1 month ago
  35. 5ace071 Remove unused PacketOptions::additional_data. by philipel · 1 year, 1 month ago
  36. 36e3875 Update WebRTC code version (2024-03-06T04:06:44). by webrtc-version-updater · 1 year, 1 month ago
  37. c9bb2c6 Propagate Environment into VideoStreamEncoder by Danil Chapovalov · 1 year, 1 month ago
  38. 9f11b96 add xctest to gn args for ios sim by Christoffer Dewerin · 1 year, 1 month ago
  39. cebded9b Update WebRTC code version (2024-03-05T04:11:56). by webrtc-version-updater · 1 year, 1 month ago
  40. e166214 Roll chromium_revision d6b7dad43f..b9338390df (1267774:1267934) by chromium-webrtc-autoroll · 1 year, 2 months ago
  41. 38c1ab1 Delete CreateVideoDecoder from VideoDecoderFactory interface by Danil Chapovalov · 1 year, 2 months ago
  42. 80d0728 Roll chromium_revision 67f77562a2..d6b7dad43f (1267659:1267774) by chromium-webrtc-autoroll · 1 year, 2 months ago
  43. 206bdaf Update WebRTC code version (2024-03-04T04:13:18). by webrtc-version-updater · 1 year, 2 months ago
  44. 04d2268 Roll chromium_revision 0bfdc8c539..67f77562a2 (1267549:1267659) by chromium-webrtc-autoroll · 1 year, 2 months ago
  45. 89e62f3 Update WebRTC code version (2024-03-03T04:12:48). by webrtc-version-updater · 1 year, 2 months ago
  46. c8068f6 Roll chromium_revision 16ca06c8c8..0bfdc8c539 (1267445:1267549) by chromium-webrtc-autoroll · 1 year, 2 months ago
  47. 7759086 Update WebRTC code version (2024-03-02T04:12:36). by webrtc-version-updater · 1 year, 2 months ago
  48. 572ce27 Roll chromium_revision 6312fa2472..16ca06c8c8 (1267340:1267445) by chromium-webrtc-autoroll · 1 year, 2 months ago
  49. 59ed9e8 Roll chromium_revision 38dcc53cf5..6312fa2472 (1267235:1267340) by chromium-webrtc-autoroll · 1 year, 2 months ago
  50. ede7529 Roll chromium_revision 1e40594b88..38dcc53cf5 (1267092:1267235) by chromium-webrtc-autoroll · 1 year, 2 months ago
  51. e5ac106 Roll chromium_revision f770766245..1e40594b88 (1266950:1267092) by chromium-webrtc-autoroll · 1 year, 2 months ago
  52. 015de61 Roll chromium_revision 248b5659e1..f770766245 (1266836:1266950) by chromium-webrtc-autoroll · 1 year, 2 months ago
  53. a5cd664 Add killswitch for receive-only setCodecPreferences change by Philipp Hancke · 1 year, 2 months ago
  54. 16ac10d PipeWire camera: use length of device id instead display name by Jan Grulich · 1 year, 2 months ago
  55. 20a9029 sdp: set content to rejected if the list of common codecs is empty by Philipp Hancke · 1 year, 2 months ago
  56. e5c04cb Roll chromium_revision 19f774aa77..248b5659e1 (1266733:1266836) by chromium-webrtc-autoroll · 1 year, 2 months ago
  57. 80000bc Update WebRTC code version (2024-02-29T04:06:33). by webrtc-version-updater · 1 year, 2 months ago
  58. c935bb2 Roll chromium_revision 07799e8663..19f774aa77 (1266593:1266733) by chromium-webrtc-autoroll · 1 year, 2 months ago
  59. 3351c9c Roll chromium_revision f6623eb6d1..07799e8663 (1266346:1266593) by chromium-webrtc-autoroll · 1 year, 2 months ago
  60. 3afa1b2 Add a SimulcastStream::GetScalabilityMode2 method that returns an optional. by Jeremy Leconte · 1 year, 2 months ago
  61. b9ce3b7 Delete deprecated VP8Decoder::Create by Danil Chapovalov · 1 year, 2 months ago
  62. 95977d8 Roll chromium_revision b2ff29df8d..f6623eb6d1 (1266241:1266346) by chromium-webrtc-autoroll · 1 year, 2 months ago
  63. fb4ad29 Continue breakup of media/rtc_media_base by Harald Alvestrand · 1 year, 2 months ago
  64. dcc1534 Delete rtc::TaskQueue by Danil Chapovalov · 1 year, 2 months ago
  65. bbff58d Introduce "well-known" SdpVideoFormat codecs by Philipp Hancke · 1 year, 2 months ago
  66. 2825f0a Update WebRTC code version (2024-02-28T04:11:45). by webrtc-version-updater · 1 year, 2 months ago
  67. 55d33f1 Roll chromium_revision aa1c478fe0..b2ff29df8d (1266091:1266241) by chromium-webrtc-autoroll · 1 year, 2 months ago
  68. c27339f Roll chromium_revision 511e819ce4..aa1c478fe0 (1265924:1266091) by chromium-webrtc-autoroll · 1 year, 2 months ago
  69. c3d937b In RtpFrameReferenceFinder discard frames with too large spatial id by Danil Chapovalov · 1 year, 2 months ago
  70. 8cd50cb Delete PacketTransportInternal::SignalReadPacket by Per K · 1 year, 2 months ago
  71. a8c4727 Roll chromium_revision a44e4a5e32..511e819ce4 (1265726:1265924) by chromium-webrtc-autoroll · 1 year, 2 months ago
  72. 334e913 Video capture PipeWire: add support for DMABuf buffer type by Jan Grulich · 1 year, 2 months ago
  73. 9e0bf9b Propagate rtc::ReceivedPacket further in RtpTransport by Per K · 1 year, 2 months ago
  74. f7682f0 Delete RTCWrappedNativeVideoDecoder by Danil Chapovalov · 1 year, 2 months ago
  75. f4aadf3 Change RtpTransport and DsctTransport to receives packets through ReceivedPacketCallback by Per K · 1 year, 2 months ago
  76. 524a06b Change BuiltInNetworkBehaviorConfig.loss_percent to double by Florent Castelli · 1 year, 2 months ago
  77. e39f6fd Roll chromium_revision af00cd208a..a44e4a5e32 (1264869:1265726) by chromium-webrtc-autoroll · 1 year, 2 months ago
  78. 2514dd7 Increase WebRTC default receive buffer size to 1MB. by Erik Språng · 1 year, 2 months ago
  79. 7a00882 Disable checks for googletest for WebRTC by Christoffer Dewerin · 1 year, 2 months ago
  80. 51532fd Test handling of rejected m-lines without transport description by Philipp Hancke · 1 year, 2 months ago
  81. 058bfe3 PipeWire capturer: set capturer as failed when session is closed by Jan Grulich · 1 year, 2 months ago
  82. 8e137d0 Replace use of SignalReadPacket in DtlsTransport by Per K · 1 year, 2 months ago
  83. 09e81cc Update WebRTC code version (2024-02-27T04:11:00). by webrtc-version-updater · 1 year, 2 months ago
  84. c32a509 Export h.265 bitstream parser APIs. by Qiu Jianlin · 1 year, 2 months ago
  85. 91ebd5f Add missing absl::optional includes by Danil Chapovalov · 1 year, 2 months ago
  86. 7098d11 Revert "p2p: separate ICE tie breaker and foundation seed" by Christoffer Dewerin · 1 year, 2 months ago
  87. 1461363 Make PeerConnectionInteface methods pure virtual. by Per K · 1 year, 2 months ago
  88. 5261619 Remove rtc::TaskQueue in AudioDeviceBuffer by Danil Chapovalov · 1 year, 2 months ago
  89. 3f7566a Cleanup rtc::TaskQueue in AsyncAudioProcessing by Danil Chapovalov · 1 year, 2 months ago
  90. 97df932 Remove multiplex codec. by Markus Handell · 1 year, 2 months ago
  91. db2f52b Reland "Make setCodecPreferences only look at receive codecs" by Philipp Hancke · 1 year, 2 months ago
  92. d99499a p2p: separate ICE tie breaker and foundation seed by Philipp Hancke · 1 year, 2 months ago
  93. 7c5f9cf Add nonstandard x-google-per-layer-pli fmtp for enabling per-layer keyFrames in response to PLIs by Philipp Hancke · 1 year, 2 months ago
  94. 3fff83d Update WebRTC code version (2024-02-26T04:03:26). by webrtc-version-updater · 1 year, 2 months ago
  95. 7391ecf Add directory for ChromiumOS specific tools by Li-Yu Yu · 1 year, 2 months ago
  96. c7a4b2a Change internal candidate type to enum by Tommi · 1 year, 2 months ago
  97. a021d99 Move implementation of PacketTransportInternal::RegisterReceivedPacketCallback by Per K · 1 year, 2 months ago
  98. d0491a3 Update WebRTC code version (2024-02-25T04:13:05). by webrtc-version-updater · 1 year, 2 months ago
  99. 4438dd3 Roll chromium_revision 1f0d2a10bd..af00cd208a (1264465:1264869) by chromium-webrtc-autoroll · 1 year, 2 months ago
  100. 3fe69c5 Update MockPeerConnectionInterface and fake with missing methods. by Per K · 1 year, 2 months ago