Google Git
Sign in
webrtc / src / HEAD
  1. f3c8615 Revert "Min mic analog level: override minimum and behavior on Mac" by Alessio Bazzica · 1 year, 10 months ago
  2. 9dcbfd8 Revert "In bitrate estimator Improve handing send time of out of order packets" by Mirko Bonadei · 1 year, 10 months ago
  3. ffd2241 Roll chromium_revision d0810c09a2..12013b696a (1024069:1024190) by chromium-webrtc-autoroll · 1 year, 10 months ago
  4. 70d6f2e Update WebRTC code version (2022-07-14T04:06:31). by webrtc-version-updater · 1 year, 10 months ago
  5. 84ef31f Roll chromium_revision ef902d1658..d0810c09a2 (1023960:1024069) by chromium-webrtc-autoroll · 1 year, 10 months ago
  6. 09196e4 Roll chromium_revision 0f2fa982fe..ef902d1658 (1023827:1023960) by chromium-webrtc-autoroll · 1 year, 10 months ago
  7. d0a6fd2 Revert "`AgcManagerDirect`: stop enforcing min mic level override with 0 level" by Alessio Bazzica · 1 year, 10 months ago
  8. 94d259c Roll chromium_revision d05b59b211..0f2fa982fe (1023693:1023827) by chromium-webrtc-autoroll · 1 year, 10 months ago
  9. b5d77a0 webrtc: Blank desktop capturer regards empty frame as a blank frame by Sunggook Chue · 1 year, 10 months ago
  10. edd8c25 Roll chromium_revision 3fcc05638a..d05b59b211 (1023689:1023693) by Björn Terelius · 1 year, 10 months ago
  11. 2b1f509 Disallow invalid arguments in RestoreEncodingLayers. by Henrik Boström · 1 year, 10 months ago
  12. 02bfcf5 Compare only SdpVideoFormat::name and SdpVideoFormat::parameters in the VideoEncoderFactoryTemplate. by philipel · 1 year, 10 months ago
  13. 046e6d1 Update WebRTC code version (2022-07-13T04:05:07). by webrtc-version-updater · 1 year, 10 months ago
  14. b7821ce Remove unnecessary overload in RtcEventLogOutput by Ali Tofigh · 1 year, 10 months ago
  15. c8f9c56 Roll chromium_revision ab0e768796..93fe9a662b (1023140:1023242) by chromium-webrtc-autoroll · 1 year, 10 months ago
  16. 078bbf1 Roll chromium_revision 0fe12b43a6..ab0e768796 (1022960:1023140) by chromium-webrtc-autoroll · 1 year, 10 months ago
  17. 9e9bc64 Update visibility for dav1d_decoder. by philipel · 1 year, 10 months ago
  18. cb56277 libstdc++: add missing atomic include for std::atomic by Stephan Hartmann · 1 year, 11 months ago
  19. 5525e631 Roll chromium_revision cf9b6945ba..0fe12b43a6 (1022166:1022960) by Björn Terelius · 1 year, 10 months ago
  20. eacb439 Update WebRTC code version (2022-07-12T04:02:34). by webrtc-version-updater · 1 year, 10 months ago
  21. c501f30 sdp: temporarily relax channel requirements for statically assigned payload types by Philipp Hancke · 1 year, 11 months ago
  22. 4058b33 Update WebRTC code version (2022-07-09T04:04:56). by webrtc-version-updater · 1 year, 11 months ago
  23. b025b14 Roll chromium_revision b5f027ec1f..cf9b6945ba (1022047:1022166) by chromium-webrtc-autoroll · 1 year, 11 months ago
  24. 52747f1 Reland "Fix overflow due to rounding in AbsoluteSendTime::To24Bits" by Danil Chapovalov · 1 year, 11 months ago
  25. 9c125c6 Migrate test/ to absl::AnyInvocable based TaskQueueBase interface by Danil Chapovalov · 1 year, 11 months ago
  26. b981394 Remove NackSender argument from RtpVideoStreamReceiver2. by philipel · 1 year, 11 months ago
  27. 6dcd43f GCC: make UnitBase::operator*(double) constexpr by Stephan Hartmann · 1 year, 11 months ago
  28. 2295ddb In bitrate estimator Improve handing send time of out of order packets by Danil Chapovalov · 1 year, 11 months ago
  29. fcfa80f Update TaskQueueGcd implementation to absl::AnyInvocable by Danil Chapovalov · 1 year, 11 months ago
  30. 9799fe0 peerconnection: move first connect metrics gathering to helper function by Philipp Hancke · 1 year, 11 months ago
  31. 9f11047 [PCLF] Fix deadlock when stats are requested during peer destruction by Artem Titov · 1 year, 11 months ago
  32. dd4884d Roll chromium_revision f01d873206..b5f027ec1f (1021933:1022047) by chromium-webrtc-autoroll · 1 year, 11 months ago
  33. c8602cd Update WebRTC code version (2022-07-08T04:06:07). by webrtc-version-updater · 1 year, 11 months ago
  34. 401158b Roll chromium_revision d8666b6883..f01d873206 (1021819:1021933) by chromium-webrtc-autoroll · 1 year, 11 months ago
  35. 9b29b87 Roll chromium_revision b78a622762..d8666b6883 (1021700:1021819) by chromium-webrtc-autoroll · 1 year, 11 months ago
  36. cbfdcc5 Roll chromium_revision 771d9b3fa0..b78a622762 (1021586:1021700) by chromium-webrtc-autoroll · 1 year, 11 months ago
  37. 28bc2ca Remove unused WebRTC-LimitPaddingSize field trial by Erik Språng · 1 year, 11 months ago
  38. eb91fe4 Remove unnecessary std::string overloads by Ali Tofigh · 1 year, 11 months ago
  39. 26910ff Make dav1d the default AV1 decoder. by philipel · 1 year, 11 months ago
  40. 7b19036 Migrate p2p/ to absl::AnyInvocable based TaskQueueBase interface by Danil Chapovalov · 1 year, 11 months ago
  41. 677c1dd Migrate rtp_rtcp to absl::AnyInvocable based TaskQueueBase interface by Danil Chapovalov · 1 year, 11 months ago
  42. c52e627 Remove WebRTC-Pacer-DynamicPaddingTarget field trial by Erik Språng · 1 year, 11 months ago
  43. b858d3f Remove unused field trial kill-switch WebRTC-LazyPacerStart. by Erik Språng · 1 year, 11 months ago
  44. a30439b Migrate pc/ to absl::AnyInvocable based TaskQueueBase interface by Danil Chapovalov · 1 year, 11 months ago
  45. b7128ed Migrate call/ to absl::AnyInvocable based TaskQueueBase interface by Danil Chapovalov · 1 year, 11 months ago
  46. 3b526d4 Roll chromium_revision 3b91a07b1a..771d9b3fa0 (1021211:1021586) by chromium-webrtc-autoroll · 1 year, 11 months ago
  47. 62c20f3 sdp: temporarily relax clockrate requirements for statically assigned payload types by Philipp Hancke · 1 year, 11 months ago
  48. 4bcf809 In rtc::Thread implement posting AnyInvocable by Danil Chapovalov · 1 year, 11 months ago
  49. 791294a Revert "Fix overflow due to rounding in AbsoluteSendTime::To24Bits" by Danil Chapovalov · 1 year, 11 months ago
  50. 3f20765 Remove unused dependencies by Byoungchan Lee · 1 year, 11 months ago
  51. b22b095 Update WebRTC code version (2022-07-07T04:02:12). by webrtc-version-updater · 1 year, 11 months ago
  52. 8179774 Reland "Wait for frames to arrive in WgcCapturer instead of returning nothing." by Austin Orion · 1 year, 11 months ago
  53. 298450e Roll chromium_revision af0b70c101..3b91a07b1a (1021083:1021211) by chromium-webrtc-autoroll · 1 year, 11 months ago
  54. a17651f Fix overflow due to rounding in AbsoluteSendTime::To24Bits by Danil Chapovalov · 1 year, 11 months ago
  55. 0be8eba Migrate pacing and video_coding to absl::AnyInvocable based TaskQueueBase interface by Danil Chapovalov · 1 year, 11 months ago
  56. f7b30e0 A few cleanup things for the port classes to clarify test code. by Tommi · 1 year, 11 months ago
  57. 95eeaa7 Migrate video/ to absl::AnyInvocable based TaskQueueBase interface by Danil Chapovalov · 1 year, 11 months ago
  58. f9f9d54 Use TimeDelta for harmonic framerate calculation in DVQA. by Björn Terelius · 1 year, 11 months ago
  59. 92159dc [PCLF] Remove references to the old location of VideoQualityAnalyzerInterface by Artem Titov · 1 year, 11 months ago
  60. 0018def test: fix flexfec test by Philipp Hancke · 1 year, 11 months ago
  61. dde7fe4 Refactor RepeatingTask to use absl::AnyInvocable functions of TaskQueue by Danil Chapovalov · 1 year, 11 months ago
  62. e76daab `AgcManagerDirect`: stop enforcing min mic level override with 0 level by Alessio Bazzica · 1 year, 11 months ago
  63. c9cad23 Min mic analog level: override minimum and behavior on Mac by Alessio Bazzica · 1 year, 11 months ago
  64. ecf88f4 Migrate net/dcsctp/ to absl::AnyInvocable based TaskQueueBase interface by Danil Chapovalov · 1 year, 11 months ago
  65. 25e268a Demote RtpStreamReceiverController AddSink/RemoveSink to private by Niels Möller · 1 year, 11 months ago
  66. ed66b77 Roll chromium_revision 40b11309e8..af0b70c101 (1020968:1021083) by chromium-webrtc-autoroll · 1 year, 11 months ago
  67. c8680c5 dcsctp: Generate lifecycle events by Victor Boivie · 2 years ago
  68. cb99ccd Update/delete old TODOs by Niels Möller · 1 year, 11 months ago
  69. 6183a0f Add default conversational speech file to the .rodata section. by Mirko Bonadei · 1 year, 11 months ago
  70. ea8eff3 Delete rtp_sender_ check in ModuleRtpRtcpImpl::SetSendingMediaStatus by Niels Möller · 1 year, 11 months ago
  71. f25a3ee Update WebRTC code version (2022-07-06T04:03:28). by webrtc-version-updater · 1 year, 11 months ago
  72. e9e4e34 Roll chromium_revision 3ba89edf17..40b11309e8 (1020846:1020968) by chromium-webrtc-autoroll · 1 year, 11 months ago
  73. a7e15a2 Introduce helper to guard an invocable with a safety flag by Danil Chapovalov · 1 year, 11 months ago
  74. 4a93da3 dcsctp: Report acked/abandoned messages when acked by Victor Boivie · 2 years ago
  75. 72b12d4 Roll chromium_revision 89179a0330..3ba89edf17 (1020743:1020846) by chromium-webrtc-autoroll · 1 year, 11 months ago
  76. b6ff84b Reland "When VP9 SVC is used, use SvcConfig to set max bitrate for the stream." by Erik Språng · 1 year, 11 months ago
  77. 7992457 Add temporary accessors for numberOfTemporalLayers by Niels Möller · 1 year, 11 months ago
  78. a1a7c63 Let PCF.GetRtpSenderCapabilities return codecs' scalabilityModes. by Byoungchan Lee · 1 year, 11 months ago
  79. e1c707c Remove unused incomplete_frame argument from JitterEstimator. by philipel · 1 year, 11 months ago
  80. 39b1b42 Use designated initializers for webrtc::SimulcastStream by Niels Möller · 1 year, 11 months ago
  81. 11fdb08 Implement RTCInboundRTPStreamStats.JitterBufferTargetDelay by Ivo Creusen · 1 year, 11 months ago
  82. 63299a3 Add absl::string_view overload for RtcEventLogOutput::Write by Björn Terelius · 1 year, 11 months ago
  83. 8feb6fd Introduce new interface for TaskQueueBase using absl::AnyInvocable by Danil Chapovalov · 1 year, 11 months ago
  84. 4f1af11 Revert "When VP9 SVC is used, use SvcConfig to set max bitrate for the stream." by Erik Språng · 1 year, 11 months ago
  85. 2ad75b3 Remove testonly from unpack_aecdump. by Mirko Bonadei · 1 year, 11 months ago
  86. 6939f63 Update old TODO comments by Niels Möller · 1 year, 11 months ago
  87. c8152fe Update/delete old TODOs by Niels Möller · 1 year, 11 months ago
  88. fb9fbdf Delete unused UlpfecReceiver::ProcessReceivedFec return value by Niels Möller · 1 year, 11 months ago
  89. 22a6253 Make PeerConnectionInterface::SetConfiguration pure virtual by Niels Möller · 1 year, 11 months ago
  90. b5b159d Update old TODO comments by Niels Möller · 1 year, 11 months ago
  91. 27b35a7 Remove KeyFrameRequestSender argument from RtpVideoStreamReceiver2. by philipel · 1 year, 11 months ago
  92. 865e45d Add default values for SimulcastStream members by Niels Möller · 1 year, 11 months ago
  93. 56257af Cleanup ReceiveSideCongestionController: remove inner wrapper helper by Danil Chapovalov · 1 year, 11 months ago
  94. 9927cb7 Roll chromium_revision efe4047c2d..89179a0330 (1020566:1020743) by chromium-webrtc-autoroll · 1 year, 11 months ago
  95. d07186c Delete useless test fixture H264SpsParserTest by Niels Möller · 1 year, 11 months ago
  96. f785989 Rename StatsCollector to LegacyStatsCollector. by Henrik Boström · 1 year, 11 months ago
  97. 58fbd1b Add manifest_merger to DEPS. by Mirko Bonadei · 1 year, 11 months ago
  98. 5023ffb DCHECK that RTCStatsCollector does not block-invoke more than twice. by Henrik Boström · 1 year, 11 months ago
  99. 0a72a28 Update WebRTC code version (2022-07-05T04:05:09). by webrtc-version-updater · 1 year, 11 months ago
  100. 3719a0c stats: use decoded framerate for inbound-rtp framesPerSecond by Philipp Hancke · 1 year, 11 months ago