1. 64b626b Use Abseil container algorithms in pc/ by Steve Anton · 6 years ago
  2. b7446ed Removing receive RIDs and Simulcast Layers. by Amit Hilbuch · 6 years ago
  3. 9bcf80a Roll chromium_revision fa9574f1d1..ed7fd9b77f (626644:626752) by chromium-webrtc-autoroll · 6 years ago
  4. 733e087 Ignore duplicated incoming RTCP packets in RTC event log parser. by Bjorn Terelius · 6 years ago
  5. a75f618 Roll chromium_revision 0a788fbaed..fa9574f1d1 (626455:626644) by chromium-webrtc-autoroll · 6 years ago
  6. bcd39d4 Creating Simulcast offer and answer in Peer Connection. by Amit Hilbuch · 6 years ago
  7. e76ca61 Allow use of functions in absl/algorithms by Steve Anton · 6 years ago
  8. 48c5493 Add 'UpdateAllocationLimits' in media transport. by Piotr (Peter) Slatala · 6 years ago
  9. 435ea0a Add is_fec property to RtpPacketToSend by Niels Möller · 6 years ago
  10. a3ed451 Add static factory method from FrameGenerator for FrameGeneratorCapturer. by Artem Titov · 6 years ago
  11. 37ec55e [clang-tidy] Apply performance-faster-string-find fixes. by Mirko Bonadei · 6 years ago
  12. 190713c Remove +api from internal DEPS files. by Mirko Bonadei · 6 years ago
  13. 7d61352 Remove unused defines and methods in internal_defines.h by Åsa Persson · 6 years ago
  14. 739baf0 [clang-tidy] Apply performance-for-range-copy fixes. by Mirko Bonadei · 6 years ago
  15. 2d65fff Roll chromium_revision 53292b65a5..0a788fbaed (626349:626455) by chromium-webrtc-autoroll · 6 years ago
  16. 8270904 Roll chromium_revision 334d413a77..53292b65a5 (626249:626349) by chromium-webrtc-autoroll · 6 years ago
  17. f380284 (7) Rename files to snake_case: remove forwarding headers by Steve Anton · 6 years ago
  18. 55b91b9 Only create no-op DTLS if media transport is used for both media and data by Piotr (Peter) Slatala · 6 years ago
  19. 9058e07 Roll chromium_revision 3343618014..334d413a77 (626126:626249) by chromium-webrtc-autoroll · 6 years ago
  20. d970807 Remove rtc_base/scoped_ref_ptr.h. by Mirko Bonadei · 6 years ago
  21. 9444f3a Roll chromium_revision 6a5b2b19b1..3343618014 (626014:626126) by chromium-webrtc-autoroll · 6 years ago
  22. d3a5aaa Check "rtc_include_internal_audio_device" before creating unittest for audio_device_ios_objc by Jiawei Ou · 6 years ago
  23. 63a176b Do not modify media transport config when falling back to RTP by Piotr (Peter) Slatala · 6 years ago
  24. 18f65dc Don't attempt to unwrap RTP timestamps for RTX stream. by Bjorn Terelius · 6 years ago
  25. 44b31d6 Delete leftover method MaxConfiguredBitrateVideo and member remote_ssrc_ by Niels Möller · 6 years ago
  26. 0ef117e Improving robustness of stable bandwidth estimate test. by Sebastian Jansson · 6 years ago
  27. bebca61 Delete unused method SetSelectiveRetransmissions by Niels Möller · 6 years ago
  28. 728b5a4 Fix initialization to prevent SIGSEGV by Artem Titov · 6 years ago
  29. b2d7141 Revert "Always use real VideoStreamsFactory in full stack tests" by Ilya Nikolaevskiy · 6 years ago
  30. da37473 Make webrtc::ParseCandidate() public. by Guido Urdaneta · 6 years ago
  31. 99ec6f3 AEC3: Remove unused kill-switches from AdjustConfig by Gustaf Ullberg · 6 years ago
  32. a9316c9 frame_analyzer: exit with status 1 when video files fail to open by Oleh Prypin · 6 years ago
  33. a8f9e25 Make sure lost packets are removed from FakeNetworkPipe. by Johannes Kron · 6 years ago
  34. fe490d8 Roll chromium_revision b483b4fce1..6a5b2b19b1 (625914:626014) by chromium-webrtc-autoroll · 6 years ago
  35. e47433f AEC3: Remove legacy render buffering by Gustaf Ullberg · 6 years ago
  36. 8a40edd Delete constant RTP_PAYLOAD_NAME_SIZE by Niels Möller · 6 years ago
  37. 76cf320 Roll chromium_revision eedb2069ef..b483b4fce1 (625788:625914) by chromium-webrtc-autoroll · 6 years ago
  38. b8c81c3 Roll chromium_revision 3432970f4e..eedb2069ef (625619:625788) by chromium-webrtc-autoroll · 6 years ago
  39. f50c6c2 Introduce VideoQualityAnalyzerInjectionHelper. by Artem Titov · 6 years ago
  40. 3ea55d5 Reland "Delete RtpUtility::Payload, and refactor RTPSender to not use it" by Niels Möller · 6 years ago
  41. 5affbf2 Turn off automatic quality scaling for simulcast in video_loopback. by philipel · 6 years ago
  42. 3770b99 Allow repeated feedback packets in log parser. by Sebastian Jansson · 6 years ago
  43. 84ca69a Add RTC event logging of LossNotification RTCP messages by Elad Alon · 6 years ago
  44. e2fffd7 Roll chromium_revision 1aa6cb924c..3432970f4e (625210:625619) by chromium-webrtc-autoroll · 6 years ago
  45. 68d5860 Override default manifest from Chromium in WebRTC. by Sami Kalliomäki · 6 years ago
  46. a67a9d9 Handle zero number of spatial layers at calculation of VP9 SVC padding. by Sergey Silkin · 6 years ago
  47. f8e7ccb Create new RTCP feedback message - LossIndication by Elad Alon · 6 years ago
  48. 2d05050 Revert "Roll chromium_revision 1aa6cb924c..faaba5b0a8 (625210:625596)" by Oleh Prypin · 6 years ago
  49. 81d4bf7 Revert "Delete RtpUtility::Payload, and refactor RTPSender to not use it" by Artem Titov · 6 years ago
  50. 1e27fec Negate flag name for prerender smoothing and update comments. by Rasmus Brandt · 6 years ago
  51. 2fd09a4 Remove deprecated code from audio device. by Mirko Bonadei · 6 years ago
  52. 88ca008 Roll chromium_revision 1aa6cb924c..faaba5b0a8 (625210:625596) by chromium-webrtc-autoroll · 6 years ago
  53. fc2175d Introduce QualityAnalyzingVideoEncoder and QualityAnalyzingVideoDecoder. by Artem Titov · 6 years ago
  54. 171df93 Delete RtpUtility::Payload, and refactor RTPSender to not use it by Niels Möller · 6 years ago
  55. 2820d17 Roll chromium_revision 1cac36a781..1aa6cb924c (624101:625210) by chromium-webrtc-autoroll · 6 years ago
  56. dbb49df Moving UniqueIdGenerator into rtc_base. by Amit Hilbuch · 6 years ago
  57. 6fde78c Prevent mac_framework_bundle configs from getting reset by Thomas Anderson · 6 years ago
  58. ce7032b Fixing snake_case files that were renamed in PRESUBMIT.py by Amit Hilbuch · 6 years ago
  59. 6a32de4 Fix potential race in CallTest. by Erik Språng · 6 years ago
  60. 2c58ba1 Move simulcast hysteresis factor parsing to RateControlSettings by Erik Språng · 6 years ago
  61. 83d5e86 Use EncoderSimulcastProxy for all codecs by Florent Castelli · 6 years ago
  62. b599787 Make UlpfecReceiverImpl use rtc::TimeMillis, not Clock::GetRealTimeClock by Niels Möller · 6 years ago
  63. 4b4266f Move parsing of trusted rate controller to RateControlSettings by Erik Språng · 6 years ago
  64. 470a5ea Introduces common AudioAllocationSettings class. by Sebastian Jansson · 6 years ago
  65. 33b716f Publish task queue test suite. by Danil Chapovalov · 6 years ago
  66. b0397d6 Always send abs-send-time when negotiated and do not filter it out. by Konrad Hofbauer · 6 years ago
  67. ae6e0b2 [CodeHealth] Fix use after std::move instances. by Yves Gerey · 6 years ago
  68. e54287a Correctly specify Mac version as 10.13 for iOS simulator tests by Oleh Prypin · 6 years ago
  69. df919fb Don't pretend we've received an end-of-candidates indication. by Jonas Olsson · 6 years ago
  70. 28522dc Rename new build targets to follow the recent large file rename by Karl Wiberg · 6 years ago
  71. 79f0d4d Enables feature to account for unacknowledged data. by Sebastian Jansson · 6 years ago
  72. 7e4341d Reland "Reland "Adds richer packet and ice processing to ParsedRtcEventLog."" by Sebastian Jansson · 6 years ago
  73. 41dd0bc Fix typo in rtc_base/thread_checker.h. by Mirko Bonadei · 6 years ago
  74. 067dc86 Make SetFirstSubFrameInFrame and SetLastSubFrameInFrame protected by Elad Alon · 6 years ago
  75. 3fdf90d PSFB without REMB magic word is not an error by Elad Alon · 6 years ago
  76. 18cf238 Always use real VideoStreamsFactory in full stack tests by Ilya Nikolaevskiy · 6 years ago
  77. d47d3eb Report rendered pixels statistic in full stack tests by Ilya Nikolaevskiy · 6 years ago
  78. 0500b52 Reduce webrtc_perf_tests duration on buildbots by Ilya Nikolaevskiy · 6 years ago
  79. 23213d9 Refactor FileRotatingStream to use FileWrapper rather than FileStream by Niels Möller · 6 years ago
  80. efd7034 Include video_bitrate_allocator.h, now that's in api/ by Niels Möller · 6 years ago
  81. cd76eab Parsing of pacing factor and alr probing in RateControlSettings by Erik Språng · 6 years ago
  82. 44f0f87 Remove NetworkManager::{set_ipv6_enabled,ipv6_enabled}. by Mirko Bonadei · 6 years ago
  83. 71b5a7d Revert "Reland "Adds richer packet and ice processing to ParsedRtcEventLog."" by Sebastian Jansson · 6 years ago
  84. 6fc6a0c Reland "Adds richer packet and ice processing to ParsedRtcEventLog." by Sebastian Jansson · 6 years ago
  85. 5be3bbd Clarify and unify network delay plot annotations. by Konrad Hofbauer · 6 years ago
  86. fe3dfee Add Visual Studio Code project folder to gitignore file. by Konrad Hofbauer · 6 years ago
  87. 4007360 Remove dimensions subarray from internal iOS bots config by Artem Titarenko · 6 years ago
  88. 7121564 Move congestion window field trial parsing to new class. by Erik Språng · 6 years ago
  89. b4c6d1e Connect global task queue factory and rtc::TaskQueue by Danil Chapovalov · 6 years ago
  90. 443760d Android: Add option to print native stack traces in PeerConnectionFactory API by Magnus Jedvert · 6 years ago
  91. 1b761ca Remove simulcast constraints in SimulcastEncoderAdapter by Florent Castelli · 6 years ago
  92. e6a4793 AEC3: avoiding a warning in the reverberation decay estimator. by Jesús de Vicente Peña · 6 years ago
  93. 7510e4a Reland "Android: Add helper methods for printing native stack traces" by Magnus Jedvert · 6 years ago
  94. dfc7d63 Deprecate FirstSubFrameInFrame() and LastSubFrameInFrame() by Elad Alon · 6 years ago
  95. 05acd2b Removes clock from TransportFeedbackAdapter. by Sebastian Jansson · 6 years ago
  96. 805a27e Reland "Refactor WebRtcVideoEngine tests to not use cricket::VideoCapturer, part 2." by Niels Möller · 6 years ago
  97. 5a6ae02 Reland "Trim down FileWrapper class to be merely a wrapper owning a FILE*" by Niels Möller · 6 years ago
  98. 700615f Revert "Android: Add helper methods for printing native stack traces" by Magnus Jedvert · 6 years ago
  99. d7bc097 Change webrtc-internal iOS pool for try and ci bots by Artem Titarenko · 6 years ago
  100. 36e0f57 Replace the usage of RTC_HISTOGRAM_COMMON_BLOCK with by Qingsi Wang · 6 years ago