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  1. 9cfdb20 Control PeerConnectionFactory's default min/starting/max bitrates from experiment by Elad Alon · 4 years, 9 months ago
  2. d781965 Delete StreamDataCountersCallback from ReceiveStatistics by Niels Möller · 4 years, 9 months ago
  3. 01525f9 Delete method StreamStatistician::GetDataCounters by Niels Möller · 4 years, 9 months ago
  4. 34aee67 Roll chromium_revision 514a543362..3ae2445b34 (686198:686310) by chromium-webrtc-autoroll · 4 years, 9 months ago
  5. 43faee0 Implement JNI and objc implementation for Ice Candidate Pair Change event surfacing by Alex Drake · 4 years, 9 months ago
  6. 519fc44 Roll chromium_revision 01bf391305..514a543362 (686061:686198) by chromium-webrtc-autoroll · 4 years, 9 months ago
  7. 9809cad Roll chromium_revision f0fd984a31..01bf391305 (685691:686061) by chromium-webrtc-autoroll · 4 years, 9 months ago
  8. 82d75a6 Use unit types in RoundRobingPacketQueue and PacedSender by Erik Språng · 4 years, 9 months ago
  9. 4d207a3 Add frames_in_flight metric to catch not delivered frames by Artem Titov · 4 years, 9 months ago
  10. 110a4de Roll chromium_revision 8f0166a59b..f0fd984a31 (685582:685691) by Yves Gerey · 4 years, 9 months ago
  11. 40dc98a Print stack trace when a test crash by Danil Chapovalov · 4 years, 9 months ago
  12. eea605d Make fake network degradation work also for sent audio by Erik Språng · 4 years, 9 months ago
  13. 58b496b Let StreamStatistician::GetReceiveStreamDataCounters return counters by value by Niels Möller · 4 years, 9 months ago
  14. 412282a [tsan] Guard audio_device_pulse_linux members from concurrent access. by Yves Gerey · 4 years, 9 months ago
  15. 1691e88 Remove unused fallback method in PacedSender by Erik Språng · 4 years, 9 months ago
  16. dc5ed5c Delete NACK-related methods from AudioCodingModule by Niels Möller · 4 years, 9 months ago
  17. b75d14c audioproc_f: input AEC dump as string, output audio to vector by Sonia-Florina Horchidan · 4 years, 9 months ago
  18. 81df62b Add field trial to introduce extra delay after target level calculation. by Jakob Ivarsson · 4 years, 9 months ago
  19. 1544915 Avoid capturing extraneous windows in CroppingWindowCapturerWin by Bryan Ferguson · 4 years, 9 months ago
  20. e427996 Roll chromium_revision 87ee38fb42..8f0166a59b (685466:685582) by chromium-webrtc-autoroll · 4 years, 9 months ago
  21. 6b2cec1 Use recommended min bitrate limit provided by encoder. by Sergey Silkin · 4 years, 9 months ago
  22. 48b48e5 Enable thread check in Call::GetStats(). by Tommi · 4 years, 9 months ago
  23. e4ba4ee Delete placeholder file rtc_base/function_view.h by Niels Möller · 4 years, 9 months ago
  24. a52e9bd Use StreamStatistician::BitrateReceived to produce total_bitrate_bps for GetStats. by Niels Möller · 4 years, 9 months ago
  25. 6685b32 Delete rtc_base/gunit_prod.h by Niels Möller · 4 years, 9 months ago
  26. e4b4de6 Add missing AppKit dependency by Niels Möller · 4 years, 9 months ago
  27. 273e263 Delete old placeholder file android_network_monitor_jni.h by Niels Möller · 4 years, 9 months ago
  28. b90d38a Delete unused Opus-specific methods of AudioCodingModule by Niels Möller · 4 years, 9 months ago
  29. 45fd69d Roll chromium_revision 6fb8f3c614..87ee38fb42 (685365:685466) by chromium-webrtc-autoroll · 4 years, 9 months ago
  30. 5297cf3 Delete unused class MockTargetTransferRateObserver by Niels Möller · 4 years, 9 months ago
  31. 5e4af85 Roll chromium_revision 9230e75a8c..6fb8f3c614 (685264:685365) by chromium-webrtc-autoroll · 4 years, 9 months ago
  32. 287bff3 Roll chromium_revision 498f5876be..9230e75a8c (685149:685264) by chromium-webrtc-autoroll · 4 years, 9 months ago
  33. 55251c3 Adds struct parameters parser/encoder. by Sebastian Jansson · 4 years, 9 months ago
  34. 940c2b5 AEC3: Reduce level of log messages by Gustaf Ullberg · 4 years, 9 months ago
  35. b6b7d1f Roll chromium_revision 5744654b26..498f5876be (685023:685149) by chromium-webrtc-autoroll · 4 years, 9 months ago
  36. 78a7138 Remove MediaTransport from Call. by Tommi · 4 years, 9 months ago
  37. 44327c3 Update test::CreateVideoStreams to use configured scale_resolution_down_by if set. by Åsa Persson · 4 years, 9 months ago
  38. 383adc0 Delete shim of PRId64 et al. on Windows by Oleh Prypin · 4 years, 9 months ago
  39. 0d210ee Change return type of of ReceiveStatistics::Create to unique_ptr. by Niels Möller · 4 years, 9 months ago
  40. c2fe547 Remove unused fallbacks in PacedSender by Erik Språng · 4 years, 9 months ago
  41. eac47f7 Removing unused fallback variant for the reverb computation by Per Åhgren · 4 years, 9 months ago
  42. 891d393 Call Call::GetStats() from the correct thread in ProbingEndToEndTest. by Tommi · 4 years, 9 months ago
  43. aaaf804 Call Call::GetStats() from the correct thread in VideoSendStreamTest. by Tommi · 4 years, 9 months ago
  44. efffd0a Roll chromium_revision 3d0c04364f..5744654b26 (684897:685023) by chromium-webrtc-autoroll · 4 years, 9 months ago
  45. 307448f Roll chromium_revision 006302cd2e..3d0c04364f (684781:684897) by chromium-webrtc-autoroll · 4 years, 9 months ago
  46. 5b5d97c Reland of "Reporting of decoding_codec_plc events"" by Alex Narest · 4 years, 9 months ago
  47. 2d2bbb1 Filter out duplicate receive codecs in the media engine by Steve Anton · 4 years, 9 months ago
  48. 3cc2f70 Roll chromium_revision 192da69226..006302cd2e (684664:684781) by chromium-webrtc-autoroll · 4 years, 9 months ago
  49. b168678 Add RTC_ prefix to non-standard format specifier macro "PRIdNS" by Oleh Prypin · 4 years, 9 months ago
  50. 12ebfa6 Delete RtcpStatisticsCallback from ReceiveStatistics by Niels Möller · 4 years, 9 months ago
  51. b668542 Delete unused format specifier macros for NSInteger and NSUInteger by Oleh Prypin · 4 years, 9 months ago
  52. 83bbe91 Delete deprecated rtc_event_log header by Danil Chapovalov · 4 years, 9 months ago
  53. e08648d Add `AbsoluteCaptureTime` to `RtpPacketInfo`. by Chen Xing · 4 years, 9 months ago
  54. f40a340 Remove deprecated code related to AEC2 by Per Åhgren · 5 years ago
  55. 75caef7 Delete unused ACM members isac_decoder_16k_ and isac_decoder_32k_ by Niels Möller · 4 years, 9 months ago
  56. d2845f8 Removes unused AudioAllocationSettings from voice engine. by Sebastian Jansson · 4 years, 9 months ago
  57. d23f67e Call Call::GetStats() from the correct thread in StatsEndToEndTest. by Tommi · 4 years, 9 months ago
  58. c24a5b1 Fix CallPerfTests to call Call::GetStats() from the right thread. by Tommi · 4 years, 9 months ago
  59. c653172 Delete obsolete method AudioCodingModule::SetBitRate by Niels Möller · 4 years, 9 months ago
  60. e71edc5 Roll chromium_revision 838e9d2793..192da69226 (684401:684664) by chromium-webrtc-autoroll · 4 years, 9 months ago
  61. 1e49ab2 Migrate part of Vp9 SVC tests on PC framework. Add temporal layers support. by Artem Titov · 4 years, 9 months ago
  62. 8dcaed9 Split VideoFrameWriter into yuv and y4m writers by Artem Titov · 4 years, 9 months ago
  63. 9d62a56 Roll chromium_revision 9d357a520c..838e9d2793 (684300:684401) by chromium-webrtc-autoroll · 4 years, 9 months ago
  64. 00c7ecf Surface CandidatePairChange event by Alex Drake · 4 years, 9 months ago
  65. 63c38e2 Fix for incorrect transport sequence number config for audio in scenario tests. by Sebastian Jansson · 4 years, 9 months ago
  66. 7cbee84 Reland "Adds PeerConnection scenario test framework." by Sebastian Jansson · 4 years, 9 months ago
  67. c648819 DegradedCall: fake network using TaskQueue instead of ProcessThread by Erik Språng · 4 years, 9 months ago
  68. bb1f245 Disable RunPythonTests on rtc_tools. by Mirko Bonadei · 4 years, 9 months ago
  69. 61b1590 Roll chromium_revision 8776a3887d..9d357a520c (684182:684300) by chromium-webrtc-autoroll · 4 years, 9 months ago
  70. 2e6c294 Refactor test_peer.cc to reduce amount of arguments passing around by Artem Titov · 4 years, 9 months ago
  71. e6b7b66 Fix CallClient so that it calls Call::GetStats() on the right thread. by Tommi · 4 years, 9 months ago
  72. a22cab8 Calling DebugBreak() on Windows during fatal checks instead of relying on abort(). by Tommi · 4 years, 9 months ago
  73. 7ba3b81 Delete class PlatformFile. by Niels Möller · 4 years, 9 months ago
  74. 10da4a0 Fix RtpFrameReferenceFinderFuzzer to not generate invalid input by Johannes Kron · 4 years, 9 months ago
  75. c89468a Fix CallStatsUnittests to update the RTT on the process thread (as in production). by Tommi · 4 years, 9 months ago
  76. 4d7c405 Split out RtcpCnameCallback from RtcpStatisticsCallback by Niels Möller · 4 years, 9 months ago
  77. ed44f54 In ChannelReceive, use AcmReceiver directly, not AudioCodingModule by Niels Möller · 4 years, 9 months ago
  78. e80885a Call Call::GetStats() from the correct thread in our bandwidth tests. by Tommi · 4 years, 9 months ago
  79. 5e005f4 Fix RampUp tests to call Call::GetStats() from the right thread - and remove the need for a dedicated polling thread. by Tommi · 4 years, 9 months ago
  80. bdc9096 Roll chromium_revision 2c4c2e2ea6..8776a3887d (684065:684182) by chromium-webrtc-autoroll · 4 years, 9 months ago
  81. 074f0d2 Roll chromium_revision 7c6275bdfa..2c4c2e2ea6 (683711:684065) by chromium-webrtc-autoroll · 4 years, 9 months ago
  82. 9b1700c Enable field trial LegacySimulcastLayerLimit by default by Florent Castelli · 4 years, 9 months ago
  83. 45231be AEC3: Removing unused code in the echo subtractor by Per Åhgren · 5 years ago
  84. cdbaeeb Aec3:Remove unused legacy code by Per Åhgren · 5 years ago
  85. d7ee76c Wire up field trials for some experimental screenshare settings by Erik Språng · 4 years, 9 months ago
  86. 8d41058 Remove unused rtc_tools/video_analysis.py. by Mirko Bonadei · 4 years, 9 months ago
  87. b56cca3 Remove the old `ContributingSources` class. by Chen Xing · 4 years, 9 months ago
  88. 3d351c6 Revert "Adds PeerConnection scenario test framework." by Sebastian Jansson · 4 years, 9 months ago
  89. ad5c4ac Adds PeerConnection scenario test framework. by Sebastian Jansson · 4 years, 9 months ago
  90. 139f4dc QualityScaler: Add option to try fast adapt down at start up based on initial bw estimates. by Åsa Persson · 4 years, 9 months ago
  91. fedd625 Change 2g network pc audio test to more realistic network by Artem Titov · 4 years, 9 months ago
  92. d75b3c4 Roll chromium_revision 2ab7c1917b..7c6275bdfa (683574:683711) by chromium-webrtc-autoroll · 4 years, 9 months ago
  93. a285909 Revert "Adding new top-level directory crypto/" by Mirko Bonadei · 4 years, 9 months ago
  94. df7c5f1 Roll chromium_revision 01452febf2..2ab7c1917b (683465:683574) by chromium-webrtc-autoroll · 4 years, 9 months ago
  95. 8bbdb5b Update VideoBitrateAllocator allocate to take a struct with more fields by Florent Castelli · 4 years, 9 months ago
  96. 9a9f18a Get WebRTC.Video.ReceivedPacketsLostInPercent from ReceiveStatistics by Niels Möller · 4 years, 9 months ago
  97. 054e3bb Reland "Replace the implementation of `GetContributingSources()` on the audio side." by Chen Xing · 4 years, 9 months ago
  98. 59bbd65 Add ToString method for AudioProcessing::Config by Artem Titov · 4 years, 9 months ago
  99. 6563934 Revert "Sanitize the codec list before sending it to the media engine" by Artem Titov · 4 years, 9 months ago
  100. 5e155a6 ReportBlockStatsTest: Remove usage of RTCPReportBlock (no longer used). by Åsa Persson · 4 years, 9 months ago