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  1. c4ea5ae Avoid log spamming when the dependency descriptor fail to parse. by philipel · 1 year, 3 months ago
  2. 4893638 mark RTPHeader struct RTC_EXPORT by Philipp Hancke · 1 year, 3 months ago
  3. 7d79740 Roll chromium_revision 2fb5e2fb16..92aea4500e (1094884:1095006) by chromium-webrtc-autoroll · 1 year, 3 months ago
  4. eeb2235 Retune AGC2 input volume controller speech ratio threshold by Alessio Bazzica · 1 year, 3 months ago
  5. 46053e4 Handle the case of missing certificates. by Henrik Boström · 1 year, 3 months ago
  6. 124d7c3 [Stats] Handle the case of missing certificates. by Henrik Boström · 1 year, 3 months ago
  7. bf27f35 Roll chromium_revision c97ba0b8b1..2fb5e2fb16 (1094776:1094884) by chromium-webrtc-autoroll · 1 year, 3 months ago
  8. 89870ff Reland "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" by Per Kjellander · 1 year, 3 months ago
  9. 8e1d613 Update WebRTC code version (2023-01-20T04:01:58). by webrtc-version-updater · 1 year, 3 months ago
  10. 5f59ded Roll chromium_revision 09356cf622..c97ba0b8b1 (1094626:1094776) by chromium-webrtc-autoroll · 1 year, 3 months ago
  11. 74f108a Add Fuchsia support to memory_usage. by Sarah Pham · 1 year, 3 months ago
  12. 541024a Roll chromium_revision e551fb7716..09356cf622 (1093833:1094626) by chromium-webrtc-autoroll · 1 year, 3 months ago
  13. 41010f9 Log resolved & unresolved turn server address as sensitive string. by Yury Yarashevich · 1 year, 3 months ago
  14. 4cb2ac0 User Timestamp and TimeDelta instead of raw ints in RtpSenderEgress by Danil Chapovalov · 1 year, 3 months ago
  15. bd1e5d5 Reland "Ensure RTCRtpSenders are always created with one encoding" by Florent Castelli · 1 year, 3 months ago
  16. 828de80 Populate RTCInboundRtpStreamStats::playoutId when appropriate by Fredrik Hernqvist · 1 year, 3 months ago
  17. 4abca66 Ensure FakeNetwork propages arrival_time by Per K · 1 year, 3 months ago
  18. 9f9671f Revert "Reland "Ensure RTCRtpSenders are always created with one encoding"" by Evan Shrubsole · 1 year, 3 months ago
  19. 3e61f88 Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" by Per Kjellander · 1 year, 3 months ago
  20. fc5d627 Reland "Ensure RTCRtpSenders are always created with one encoding" by Florent Castelli · 1 year, 3 months ago
  21. 949e356 Update WebRTC code version (2023-01-19T04:04:31). by webrtc-version-updater · 1 year, 3 months ago
  22. abb6416 mouse_cursor_monitor: Annotate a method with RTC_EXPORT by Salman · 1 year, 3 months ago
  23. a0bc404 Remove WebRTC-Dav1dDecoder kill switch. by philipel · 1 year, 3 months ago
  24. 299c02e Roll chromium_revision 5c3b57c4c8..e551fb7716 (1093721:1093833) by chromium-webrtc-autoroll · 1 year, 3 months ago
  25. 7391899 Add Fuchsia perf output and fix upload by Christoffer Jansson · 1 year, 3 months ago
  26. 9ece54f Delete unnecssary AudioReceiveStreamInterface::GetRtpExtensions by Per K · 1 year, 3 months ago
  27. 444741e replace use of iterators with for loops or auto by Philipp Hancke · 1 year, 3 months ago
  28. 3b96f2c Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp by Per K · 1 year, 3 months ago
  29. bdf30c8 Ensure VideoRtpReplayer use new PacketReceiver::DeliverRtp packet. by Per K · 1 year, 3 months ago
  30. e6b3f48 Reland "Move leb128 helper functions into own build target" by Danil Chapovalov · 1 year, 3 months ago
  31. 4885de4 Remove test workaround to catch scenario when packet is resent before sent by Danil Chapovalov · 1 year, 4 months ago
  32. 4ccb616 Old iOS sim bots clean up. by Jeremy Leconte · 1 year, 3 months ago
  33. 44e5d5a Revert "Ensure RTCRtpSenders are always created with one encoding" by Evan Shrubsole · 1 year, 3 months ago
  34. 2c0376e Run iOS sim bots on versions 14, 15 and 16. by Jeremy Leconte · 1 year, 3 months ago
  35. e351560 Disable RTCCameraVideoCapturerTestsWithMockedCaptureSession. by Jeremy Leconte · 1 year, 3 months ago
  36. 25ffa80 Roll chromium_revision 8496a82e1d..5c3b57c4c8 (1093621:1093721) by chromium-webrtc-autoroll · 1 year, 3 months ago
  37. d05daeb Roll chromium_revision 2762031951..8496a82e1d (1093486:1093621) by chromium-webrtc-autoroll · 1 year, 3 months ago
  38. 897f15d Roll chromium_revision f1d4d8b74c..2762031951 (1093304:1093486) by chromium-webrtc-autoroll · 1 year, 3 months ago
  39. b802369 Ensure RTCRtpSenders are always created with one encoding by Florent Castelli · 1 year, 3 months ago
  40. 478f3b7 Avoid waking up encoder thread when audio send stream is stopped. by Jakob Ivarsson · 1 year, 3 months ago
  41. 9a8aa20 Roll chromium_revision b4664ef98d..f1d4d8b74c (1093178:1093304) by chromium-webrtc-autoroll · 1 year, 3 months ago
  42. 1389c4b Add 444 10 bits support for H264 and VP9 by Sergio Garcia Murillo · 1 year, 4 months ago
  43. 94b0559 Only fill send/recv stats if there are send/receive streams by Philipp Hancke · 1 year, 3 months ago
  44. 9b23900 Update WebRTC code version (2023-01-17T04:11:09). by webrtc-version-updater · 1 year, 3 months ago
  45. 6d19e6f Roll chromium_revision de6dd5a1b5..b4664ef98d (1093073:1093178) by chromium-webrtc-autoroll · 1 year, 3 months ago
  46. f1fa6ac7 Implement time functions for Fuchsia. by Sarah Pham · 1 year, 3 months ago
  47. 40b5bd7 APM: fix TS initialization bugs with `WebRTC-Audio-GainController2` by Alessio Bazzica · 1 year, 3 months ago
  48. f7a46e5 Roll chromium_revision 09a5d11a47..de6dd5a1b5 (1092898:1093073) by chromium-webrtc-autoroll · 1 year, 3 months ago
  49. a6b9924 Remove all usage of //rtc_base target by Florent Castelli · 1 year, 3 months ago
  50. efbe753 Add RTCAudioPlayoutStats to GetStats(). by Fredrik Hernqvist · 1 year, 3 months ago
  51. f7e4071 Only generate codec stats for the video send/recv codec in use by Philipp Hancke · 1 year, 3 months ago
  52. 3dd73ae Surface the SetMetadata() method so that Chromium can use it. by Henrik Boström · 1 year, 3 months ago
  53. 6afa92a Tooling to process RtcEventNetEqSetMinimumDelay by Lionel Koenig · 1 year, 3 months ago
  54. 55ac75f Make terelius owner of rtc_event to text by Lionel Koenig · 1 year, 3 months ago
  55. 5cca086 Update WebRTC doc related to webrtc.org accounts. by Jeremy Leconte · 1 year, 3 months ago
  56. 2d14479 Update WebRTC code version (2023-01-16T04:07:02). by webrtc-version-updater · 1 year, 3 months ago
  57. 332bc4b Roll chromium_revision ad7c2cc677..09a5d11a47 (1092797:1092898) by chromium-webrtc-autoroll · 1 year, 3 months ago
  58. c01410e Roll chromium_revision 2650ba360c..ad7c2cc677 (1092690:1092797) by chromium-webrtc-autoroll · 1 year, 3 months ago
  59. a1eb4fe Roll chromium_revision 05284a0a51..2650ba360c (1092572:1092690) by chromium-webrtc-autoroll · 1 year, 3 months ago
  60. 37d4d84 Roll chromium_revision 2b544cbc99..05284a0a51 (1092425:1092572) by chromium-webrtc-autoroll · 1 year, 3 months ago
  61. 612872b2 Add RtcEvent to store when MinimumSetDelay is set on NetEq by Lionel Koenig · 1 year, 4 months ago
  62. 8da0f3a Roll chromium_revision f6c2626500..2b544cbc99 (1092306:1092425) by chromium-webrtc-autoroll · 1 year, 3 months ago
  63. 6cf46b9 Add RTPVideoHeader::SetFromMetadata() and FromMetadata(). by Henrik Boström · 1 year, 3 months ago
  64. dc39aeb Add GetRTPVideoHeaderCodecSpecifics() to metadata. by Henrik Boström · 1 year, 3 months ago
  65. bb25641 [PCLF] Add an API to add extra audio/video RTP header extensions by Artem Titov · 1 year, 3 months ago
  66. 6a9af57 Revert "Move leb128 helper functions into own build target" by Danil Chapovalov · 1 year, 3 months ago
  67. 7aa145d Remove 'iOS Debug (simulator)' from LKGR. by Jeremy Leconte · 1 year, 3 months ago
  68. 3fab086 Use RtcEventLog::EncodingType::NewFormat in VideoQualityTest. by philipel · 1 year, 4 months ago
  69. 04e9318 Roll chromium_revision 76506aa56a..f6c2626500 (1092201:1092306) by chromium-webrtc-autoroll · 1 year, 3 months ago
  70. 079e93d Add callback for raw frames for video capture by Michael Olbrich · 1 year, 5 months ago
  71. 9795589 Add "iOS Debug (simulator)" CI bot. by Jeremy Leconte · 1 year, 3 months ago
  72. e7e53fe Roll chromium_revision aff3b5b1f9..76506aa56a (1092063:1092201) by chromium-webrtc-autoroll · 1 year, 3 months ago
  73. f04a345 Roll chromium_revision 8ff1e4f84e..aff3b5b1f9 (1091923:1092063) by chromium-webrtc-autoroll · 1 year, 3 months ago
  74. c19cc29 Roll chromium_revision 00abcc8f11..8ff1e4f84e (1091815:1091923) by chromium-webrtc-autoroll · 1 year, 3 months ago
  75. b3046c2 Use PacketReceiver::DeliverRtpPaket in scenario tests by Per K · 1 year, 3 months ago
  76. 5b7896b Roll chromium_revision b924865c52..00abcc8f11 (1091696:1091815) by chromium-webrtc-autoroll · 1 year, 3 months ago
  77. b613d62 [Unwrap] Delete webrtc::Unwrapper by Evan Shrubsole · 1 year, 3 months ago
  78. d8b6b06e [Unwrap] Delete rtc::TimestampWrapAroundHandler by Evan Shrubsole · 1 year, 3 months ago
  79. 83fd843 Add a ios_x64_dbg_simulator try bot. by Jeremy Leconte · 1 year, 3 months ago
  80. 222c052 [Unwrap] Delete SequenceNumbersConformanceTest by Evan Shrubsole · 1 year, 3 months ago
  81. 4387ad6 [Unwrap] Migrate dcsctp sequence numbers to SeqNumUnwrapper by Evan Shrubsole · 1 year, 4 months ago
  82. 17f783e Skip trimming packet arrival history at the beginning by Danil Chapovalov · 1 year, 4 months ago
  83. 7787429 In remb parser discard bitrate larger than max int64_t by Danil Chapovalov · 1 year, 3 months ago
  84. 43d4eee [Unwrap] Migrate rtp_rtcp_tests to RtpSequenceNumberUnwrapper by Evan Shrubsole · 1 year, 3 months ago
  85. 9337ac8 [Unwrap] Migrate RtcEventLog parser to use RtpSequenceNumberUnwrapper by Evan Shrubsole · 1 year, 4 months ago
  86. d22dc86 Roll chromium_revision 40afafa78c..b924865c52 (1091586:1091696) by chromium-webrtc-autoroll · 1 year, 3 months ago
  87. 0bef97c Update WebRTC code version (2023-01-12T04:02:41). by webrtc-version-updater · 1 year, 3 months ago
  88. 402a440 Roll chromium_revision 709ec8ac30..40afafa78c (1091458:1091586) by chromium-webrtc-autoroll · 1 year, 3 months ago
  89. b406577 Roll chromium_revision 0c76207a3f..709ec8ac30 (1090534:1091458) by chromium-webrtc-autoroll · 1 year, 3 months ago
  90. e4c49e3 [Unwrap] Migrate RtpToNtpEstimator to use RtpTimestampUnwrapper by Evan Shrubsole · 1 year, 3 months ago
  91. e6b4cbe Add SVC fallback. by Åsa Persson · 1 year, 3 months ago
  92. d100a58 Add dimensions to video settings in objc sdk camera backend. by Andreas Pehrson · 1 year, 4 months ago
  93. b081042 Remove dimension check in SimulcastUtility::ValidSimulcastParameters by anurag · 1 year, 4 months ago
  94. 8c347eb [Unwrap] Migrate TransportFeedbackDemuxer to use RtpSequenceNumberUnwrapper by Evan Shrubsole · 1 year, 4 months ago
  95. 57e5562 [Unwrap] Use RtpTimestampUnwrapper in audio/channel_receive by Evan Shrubsole · 1 year, 4 months ago
  96. fa962ff Move leb128 helper functions into own build target by Danil Chapovalov · 1 year, 4 months ago
  97. a5ba586 Update visibility on rtc_base:log_sinks target by Florent Castelli · 1 year, 3 months ago
  98. 7b4c8ad Reland "[Unwrap] Migrate ReceiveStatisticsImpl to use RtpSequenceNumberUnwrapper" by Evan Shrubsole · 1 year, 4 months ago
  99. e137c45 stats: deprecate timestamp_us constructor and method by Philipp Hancke · 1 year, 4 months ago
  100. 128afb1 Only build fuchsia_perf_tests on fuchsia os. by Jeremy Leconte · 1 year, 3 months ago