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  1. 9057260 Roll chromium_revision bf03673fd1..241ac98bfc (626985:627089) by chromium-webrtc-autoroll · 5 years ago
  2. 5118bbc Add ability to set max probing bitrate via GoogCcNetworkController by Erik Språng · 5 years ago
  3. d3be017 Remove unused PacketLossEstimator class by Zach Stein · 5 years ago
  4. 8c8feb9 Moves packet overhead from network nodes to simulation. by Sebastian Jansson · 5 years ago
  5. c1a0bcb Implement the encoding RtpParameter scaleResolutionDownBy by Florent Castelli · 5 years ago
  6. 411b49b Break FrameConfig out of Vp8TemporalLayers by Elad Alon · 5 years ago
  7. 31a739e Roll chromium_revision 531da0eda2..bf03673fd1 (626885:626985) by chromium-webrtc-autoroll · 5 years ago
  8. b4977de Receive-side ready for multiple channels. by Alex Loiko · 5 years ago
  9. 7a3e43a Reland of Opus multistream. by Alex Loiko · 5 years ago
  10. e5ccf5f APM: adding a missing header when dumping files in APM by Jesús de Vicente Peña · 5 years ago
  11. 68d6d44 AEC3: Remove remaining kill-switches by Gustaf Ullberg · 5 years ago
  12. 649a4c2 [clang-tidy] Apply performance-inefficient-vector-operation fixes. by Mirko Bonadei · 5 years ago
  13. 949f0fd Move FrameCountObserver from RTPSender to RtpVideoSender by Niels Möller · 5 years ago
  14. 3e8b7e9 mb: remove 'type': 'gn' because it's the default and doesn't mean anything by Oleh Prypin · 5 years ago
  15. e008248 Only instantiate TemporalLayersChecker in debug builds by Elad Alon · 5 years ago
  16. f5b216a Pass explicit frame dependency information to RtpPayloadParams by Elad Alon · 5 years ago
  17. 7248b40 Added VcmCapturer::Create loop to allow nonzero device index. by Johnny Lee · 5 years ago
  18. f7f227c Roll chromium_revision ed7fd9b77f..531da0eda2 (626752:626885) by chromium-webrtc-autoroll · 5 years ago
  19. 3d02384 Fix inverted DCHECK conditional by Steve Anton · 5 years ago
  20. 2c9ebef Use Abseil container algorithms in media/ by Steve Anton · 5 years ago
  21. 64b626b Use Abseil container algorithms in pc/ by Steve Anton · 5 years ago
  22. b7446ed Removing receive RIDs and Simulcast Layers. by Amit Hilbuch · 5 years ago
  23. 9bcf80a Roll chromium_revision fa9574f1d1..ed7fd9b77f (626644:626752) by chromium-webrtc-autoroll · 5 years ago
  24. 733e087 Ignore duplicated incoming RTCP packets in RTC event log parser. by Bjorn Terelius · 5 years ago
  25. a75f618 Roll chromium_revision 0a788fbaed..fa9574f1d1 (626455:626644) by chromium-webrtc-autoroll · 5 years ago
  26. bcd39d4 Creating Simulcast offer and answer in Peer Connection. by Amit Hilbuch · 5 years ago
  27. e76ca61 Allow use of functions in absl/algorithms by Steve Anton · 5 years ago
  28. 48c5493 Add 'UpdateAllocationLimits' in media transport. by Piotr (Peter) Slatala · 5 years ago
  29. 435ea0a Add is_fec property to RtpPacketToSend by Niels Möller · 5 years ago
  30. a3ed451 Add static factory method from FrameGenerator for FrameGeneratorCapturer. by Artem Titov · 5 years ago
  31. 37ec55e [clang-tidy] Apply performance-faster-string-find fixes. by Mirko Bonadei · 5 years ago
  32. 190713c Remove +api from internal DEPS files. by Mirko Bonadei · 5 years ago
  33. 7d61352 Remove unused defines and methods in internal_defines.h by Åsa Persson · 5 years ago
  34. 739baf0 [clang-tidy] Apply performance-for-range-copy fixes. by Mirko Bonadei · 5 years ago
  35. 2d65fff Roll chromium_revision 53292b65a5..0a788fbaed (626349:626455) by chromium-webrtc-autoroll · 5 years ago
  36. 8270904 Roll chromium_revision 334d413a77..53292b65a5 (626249:626349) by chromium-webrtc-autoroll · 5 years ago
  37. f380284 (7) Rename files to snake_case: remove forwarding headers by Steve Anton · 5 years ago
  38. 55b91b9 Only create no-op DTLS if media transport is used for both media and data by Piotr (Peter) Slatala · 5 years ago
  39. 9058e07 Roll chromium_revision 3343618014..334d413a77 (626126:626249) by chromium-webrtc-autoroll · 5 years ago
  40. d970807 Remove rtc_base/scoped_ref_ptr.h. by Mirko Bonadei · 5 years ago
  41. 9444f3a Roll chromium_revision 6a5b2b19b1..3343618014 (626014:626126) by chromium-webrtc-autoroll · 5 years ago
  42. d3a5aaa Check "rtc_include_internal_audio_device" before creating unittest for audio_device_ios_objc by Jiawei Ou · 5 years ago
  43. 63a176b Do not modify media transport config when falling back to RTP by Piotr (Peter) Slatala · 5 years ago
  44. 18f65dc Don't attempt to unwrap RTP timestamps for RTX stream. by Bjorn Terelius · 5 years ago
  45. 44b31d6 Delete leftover method MaxConfiguredBitrateVideo and member remote_ssrc_ by Niels Möller · 5 years ago
  46. 0ef117e Improving robustness of stable bandwidth estimate test. by Sebastian Jansson · 5 years ago
  47. bebca61 Delete unused method SetSelectiveRetransmissions by Niels Möller · 5 years ago
  48. 728b5a4 Fix initialization to prevent SIGSEGV by Artem Titov · 5 years ago
  49. b2d7141 Revert "Always use real VideoStreamsFactory in full stack tests" by Ilya Nikolaevskiy · 5 years ago
  50. da37473 Make webrtc::ParseCandidate() public. by Guido Urdaneta · 5 years ago
  51. 99ec6f3 AEC3: Remove unused kill-switches from AdjustConfig by Gustaf Ullberg · 5 years ago
  52. a9316c9 frame_analyzer: exit with status 1 when video files fail to open by Oleh Prypin · 5 years ago
  53. a8f9e25 Make sure lost packets are removed from FakeNetworkPipe. by Johannes Kron · 5 years ago
  54. fe490d8 Roll chromium_revision b483b4fce1..6a5b2b19b1 (625914:626014) by chromium-webrtc-autoroll · 5 years ago
  55. e47433f AEC3: Remove legacy render buffering by Gustaf Ullberg · 5 years ago
  56. 8a40edd Delete constant RTP_PAYLOAD_NAME_SIZE by Niels Möller · 5 years ago
  57. 76cf320 Roll chromium_revision eedb2069ef..b483b4fce1 (625788:625914) by chromium-webrtc-autoroll · 5 years ago
  58. b8c81c3 Roll chromium_revision 3432970f4e..eedb2069ef (625619:625788) by chromium-webrtc-autoroll · 5 years ago
  59. f50c6c2 Introduce VideoQualityAnalyzerInjectionHelper. by Artem Titov · 5 years ago
  60. 3ea55d5 Reland "Delete RtpUtility::Payload, and refactor RTPSender to not use it" by Niels Möller · 5 years ago
  61. 5affbf2 Turn off automatic quality scaling for simulcast in video_loopback. by philipel · 5 years ago
  62. 3770b99 Allow repeated feedback packets in log parser. by Sebastian Jansson · 5 years ago
  63. 84ca69a Add RTC event logging of LossNotification RTCP messages by Elad Alon · 5 years ago
  64. e2fffd7 Roll chromium_revision 1aa6cb924c..3432970f4e (625210:625619) by chromium-webrtc-autoroll · 5 years ago
  65. 68d5860 Override default manifest from Chromium in WebRTC. by Sami Kalliomäki · 5 years ago
  66. a67a9d9 Handle zero number of spatial layers at calculation of VP9 SVC padding. by Sergey Silkin · 5 years ago
  67. f8e7ccb Create new RTCP feedback message - LossIndication by Elad Alon · 5 years ago
  68. 2d05050 Revert "Roll chromium_revision 1aa6cb924c..faaba5b0a8 (625210:625596)" by Oleh Prypin · 5 years ago
  69. 81d4bf7 Revert "Delete RtpUtility::Payload, and refactor RTPSender to not use it" by Artem Titov · 5 years ago
  70. 1e27fec Negate flag name for prerender smoothing and update comments. by Rasmus Brandt · 5 years ago
  71. 2fd09a4 Remove deprecated code from audio device. by Mirko Bonadei · 5 years ago
  72. 88ca008 Roll chromium_revision 1aa6cb924c..faaba5b0a8 (625210:625596) by chromium-webrtc-autoroll · 5 years ago
  73. fc2175d Introduce QualityAnalyzingVideoEncoder and QualityAnalyzingVideoDecoder. by Artem Titov · 5 years ago
  74. 171df93 Delete RtpUtility::Payload, and refactor RTPSender to not use it by Niels Möller · 5 years ago
  75. 2820d17 Roll chromium_revision 1cac36a781..1aa6cb924c (624101:625210) by chromium-webrtc-autoroll · 5 years ago
  76. dbb49df Moving UniqueIdGenerator into rtc_base. by Amit Hilbuch · 5 years ago
  77. 6fde78c Prevent mac_framework_bundle configs from getting reset by Thomas Anderson · 5 years ago
  78. ce7032b Fixing snake_case files that were renamed in PRESUBMIT.py by Amit Hilbuch · 5 years ago
  79. 6a32de4 Fix potential race in CallTest. by Erik Språng · 5 years ago
  80. 2c58ba1 Move simulcast hysteresis factor parsing to RateControlSettings by Erik Språng · 5 years ago
  81. 83d5e86 Use EncoderSimulcastProxy for all codecs by Florent Castelli · 5 years ago
  82. b599787 Make UlpfecReceiverImpl use rtc::TimeMillis, not Clock::GetRealTimeClock by Niels Möller · 5 years ago
  83. 4b4266f Move parsing of trusted rate controller to RateControlSettings by Erik Språng · 5 years ago
  84. 470a5ea Introduces common AudioAllocationSettings class. by Sebastian Jansson · 5 years ago
  85. 33b716f Publish task queue test suite. by Danil Chapovalov · 5 years ago
  86. b0397d6 Always send abs-send-time when negotiated and do not filter it out. by Konrad Hofbauer · 5 years ago
  87. ae6e0b2 [CodeHealth] Fix use after std::move instances. by Yves Gerey · 5 years ago
  88. e54287a Correctly specify Mac version as 10.13 for iOS simulator tests by Oleh Prypin · 5 years ago
  89. df919fb Don't pretend we've received an end-of-candidates indication. by Jonas Olsson · 5 years ago
  90. 28522dc Rename new build targets to follow the recent large file rename by Karl Wiberg · 5 years ago
  91. 79f0d4d Enables feature to account for unacknowledged data. by Sebastian Jansson · 5 years ago
  92. 7e4341d Reland "Reland "Adds richer packet and ice processing to ParsedRtcEventLog."" by Sebastian Jansson · 5 years ago
  93. 41dd0bc Fix typo in rtc_base/thread_checker.h. by Mirko Bonadei · 5 years ago
  94. 067dc86 Make SetFirstSubFrameInFrame and SetLastSubFrameInFrame protected by Elad Alon · 5 years ago
  95. 3fdf90d PSFB without REMB magic word is not an error by Elad Alon · 5 years ago
  96. 18cf238 Always use real VideoStreamsFactory in full stack tests by Ilya Nikolaevskiy · 5 years ago
  97. d47d3eb Report rendered pixels statistic in full stack tests by Ilya Nikolaevskiy · 5 years ago
  98. 0500b52 Reduce webrtc_perf_tests duration on buildbots by Ilya Nikolaevskiy · 5 years ago
  99. 23213d9 Refactor FileRotatingStream to use FileWrapper rather than FileStream by Niels Möller · 5 years ago
  100. efd7034 Include video_bitrate_allocator.h, now that's in api/ by Niels Möller · 5 years ago