1. 13790be Add suppression for "uninitialized memory" error in getaddrinfo. by deadbeef · 8 years ago
  2. 8290ddf Revert of Delete SignalThread class. (patchset #20 id:380001 of https://codereview.webrtc.org/2915253002/ ) by deadbeef · 8 years ago
  3. 1095ada Revert of Injectable Obj-C video codecs (patchset #8 id:140001 of https://codereview.webrtc.org/2966023002/ ) by tkchin · 8 years ago
  4. aa41f0c API for periodically regathering ICE candidates by Steve Anton · 8 years ago
  5. f1f9889 Roll chromium_revision 2c2aec8ef0..72d8fe0dd3 (485610:485684) by buildbot · 8 years ago
  6. b8e7a7d Use correct value for CFBundleExecutable and CFBundleName. by Sylvain Defresne · 8 years ago
  7. 950c1c9 TransmitMixer: Check GetSendCodec return value. by ossu · 8 years ago
  8. 29d0840 Reland of Refactor timing frame logic to work with encoders with internal sources (patchset #1 id:1 of https://codereview.webrtc.org/2980533002/ ) by ilnik · 8 years ago
  9. 2fbd771 Roll chromium_revision dea6441d2d..2c2aec8ef0 (485283:485610) by buildbot · 8 years ago
  10. 82c5593 Let alr detector use a budged to detect underuse. by tschumim · 8 years ago
  11. 87b6ddb Suppress MissingPrefix using tools attribute by Ingemar Ådahl · 8 years ago
  12. c024740 Use relative paths in GN files. by jianjun.zhu · 8 years ago
  13. c45d6d9 Remove dependency on rtc::Thread and rtc_base from audio_mixer_unittests. by tommi · 8 years ago
  14. 14c11a4 Add adaptive notch filter to remove narrowband echo components in AEC3 by peah · 8 years ago
  15. 5e6685f Robustification of the AEC3 echo removal in the first part of the call by peah · 8 years ago
  16. c1abde7 Call should allow pass through of keep-alive packets. by sprang · 8 years ago
  17. 7fc3f15 Now uses CallStaticObjectMethodV to an variable argument list argument by henrika · 8 years ago
  18. 1e50748 Revert of Make "set_ignore_non_default_routes" actually use its argument. (patchset #1 id:1 of https://codereview.webrtc.org/2974873002/ ) by sprang · 8 years ago
  19. e5c4a81 Move RTP keep-alive config from VideoSendStream::Config to Call::Config by sprang · 8 years ago
  20. 2910357 Transparency improvements in the echo canceller 3 by peah · 8 years ago
  21. 863f03b Fix video_replay tool to respect RTX stream and fix default parameters. by ilnik · 8 years ago
  22. d8cf08f Don't call CreateDtlsTransport_n from non-network thread in WebRtcSession by deadbeef · 8 years ago
  23. 05314c3 Make "set_ignore_non_default_routes" actually use its argument. by deadbeef · 8 years ago
  24. 48956a1 Remove unused headers from remote_bitrate_estimator_single_stream.cc by eladalon · 8 years ago
  25. 0b1e2f3 Revert of Refactor timing frame logic to work with encoders with internal sources (patchset #2 id:20001 of https://codereview.webrtc.org/2968153002/ ) by ilnik · 8 years ago
  26. bc8ee33 Remove verbose logs from audio_coding_module.cc. by Noah Richards · 8 years ago
  27. a7a4535 Refactor timing frame logic to work with encoders with internal sources by ilnik · 8 years ago
  28. 5ce5a47 Roll chromium_revision b98c4727fc..dea6441d2d (485244:485283) by buildbot · 8 years ago
  29. d2702ef Fix flaky test VideoSendStreamTest.SendsKeepAlive by sprang · 8 years ago
  30. fe53355 Fix presubmit check for webrtc/base. by ehmaldonado · 8 years ago
  31. ea0e084 Add presubmit check to prevent further changes to webrtc/base. by ehmaldonado · 8 years ago
  32. 78460a7 Roll chromium_revision 8e79102feb..b98c4727fc (485221:485244) by buildbot · 8 years ago
  33. 370dd47 Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ ) by ehmaldonado · 8 years ago
  34. 9483b49 Remove remains of webrtc/base by ehmaldonado · 8 years ago
  35. ccd7328 Roll chromium_revision a28aeb7be0..8e79102feb (485062:485221) by buildbot · 8 years ago
  36. 6c9556e Prevent warnings in PacketRouterTest and PacketRouterRembTest by eladalon · 8 years ago
  37. 6cc2561 Remove webrtc::VideoEncoderFactory by magjed · 8 years ago
  38. c453b08 Adding log in OrtcFactoryIntegrationTest and fix a bug. by minyue-webrtc · 8 years ago
  39. 0b249c2 Refactor gunit for synergy to gtest. by minyue-webrtc · 8 years ago
  40. 539d104 Revert of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt (patchset #2 id:20001 of https://codereview.webrtc.org/2964863002/ ) by mbonadei · 8 years ago
  41. bffe597 Convert occurrences of deprecated WEBRTC_TRACE logging to LOG style logging in webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc. by saza · 8 years ago
  42. 5de0680 Add a check in the BlockBuffer of AEC2 to guard for buffer overflows. by saza · 8 years ago
  43. 3ffa72d Add AudioFrame::ResetWithoutMuting() to address performance regression. by Jonathan Yu · 8 years ago
  44. f1377f7 Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on the worker thread. by perkj · 8 years ago
  45. cf39dd5 Add RTC_FROM_HERE location information to two DCHECKs in ProcessThread. by tommi · 8 years ago
  46. f88f72d Roll chromium_revision 85f2caa791..a28aeb7be0 (484906:485062) by buildbot · 8 years ago
  47. 4dde3df Move SrtpSession and tests to their own files. by zstein · 8 years ago
  48. 7d0a77e Handle case where UDP packet contains multiple DTLS records. by jbauch · 8 years ago
  49. 7480da4 Trace loggging: Check for g_event_logger is not null before calling it. by jtteh · 8 years ago
  50. fb660ae Decreased the adaptation rate for the adaptive filter in the echo canceller 3 by peah · 8 years ago
  51. 991521e Roll chromium_revision 6ddb8d2861..85f2caa791 (484869:484906) by buildbot · 8 years ago
  52. de5ff8e Fix a variable naming typo by henrik.lundin · 8 years ago
  53. e67bedb External APM usage downstream dependency support cleanup by peah · 8 years ago
  54. a0349c1 Injectable Obj-C video codecs by Anders Carlsson · 8 years ago
  55. c1a58ba Roll chromium_revision a5ed187842..6ddb8d2861 (484843:484869) by buildbot · 8 years ago
  56. eaaae9e base->rtc_base: Update .c, .mm and .java files. by ehmaldonado · 8 years ago
  57. 0e9bd8e Roll chromium_revision 4b357464fd..a5ed187842 (484824:484843) by buildbot · 8 years ago
  58. f04afde Report interframe delay sum in old GetStats by ilnik · 8 years ago
  59. 5b36173 Support re-entrant calls to MessageQueueManager::Clear. by jbauch · 8 years ago
  60. 876088a Roll chromium_revision baaa9eae93..4b357464fd (484696:484824) by buildbot · 8 years ago
  61. 4a494ff desktop_capture: crop border in window_capture on Win8/10 by braveyao · 8 years ago
  62. f07e6b4 Roll chromium_revision 2e0945b687..baaa9eae93 (484611:484696) by buildbot · 8 years ago
  63. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  64. ea39dfa Roll chromium_revision c33c6bfd24..2e0945b687 (484321:484611) by buildbot · 8 years ago
  65. 9e3f1e4 Fixed a miscalculation of sent bitrate caused by mixup of time units by Sebastian Jansson · 8 years ago
  66. d66072b Moving asm code out of common_audio_c sources list by mbonadei · 8 years ago
  67. 3b03476 Remove MAIN_NIB_FILE from Info.plist because the substitution is broken by oprypin · 8 years ago
  68. a449107 Let NetEq reset the AudioFrame during muted state by henrik.lundin · 8 years ago
  69. 02569ad Update screen simulcast config by sprang · 8 years ago
  70. 168794c Implement RTP keepalive in native stack. by sprang · 8 years ago
  71. 5c0d703 Moving asm code out of isac_fix_c sources list by mbonadei · 8 years ago
  72. 05db21d Reland of move webrtc/tools (patchset #1 id:1 of https://codereview.webrtc.org/2973493002/ ) by ehmaldonado · 8 years ago
  73. 2edc684 Report timing frames info in GetStats. by ilnik · 8 years ago
  74. 5b7fc8c A few simplifications to CodecDatabase and VCMGenericDecoder. by tommi · 8 years ago
  75. 7025244 Roll chromium_revision f45f1f992e..c33c6bfd24 (484285:484321) by buildbot · 8 years ago
  76. 6aa9511 Fix null ref in NetworkMonitorAutoDetect if Connectivity Manager service is unavailable by bdodson · 8 years ago
  77. e4f63a1 Roll chromium_revision c01b31617b..f45f1f992e (484252:484285) by buildbot · 8 years ago
  78. b16a01f Revert "Reland "Adding ANA config event to debug dump."" by minyue-webrtc · 8 years ago
  79. 63d146b NetEq: Rectify the implementation of PacketBuffer::DiscardOldPackets by henrik.lundin · 8 years ago
  80. 440ea8c Roll chromium_revision 3fe2409358..c01b31617b (484231:484252) by buildbot · 8 years ago
  81. 191113a Added implementation of four functions in the BBR congestion controller: by gnish · 8 years ago
  82. bc0c4f5 Roll chromium_revision 6da2ebcead..3fe2409358 (484119:484231) by buildbot · 8 years ago
  83. fae474c Implement packet discard rate in NetEq. by minyue-webrtc · 8 years ago
  84. 889d965 Fix issue with zero rtt reports when using FlexFEC and add perf test. by stefan · 8 years ago
  85. 070efc0 Improves WebRTC.Audio.AveragePlayoutCallbacksBetweenGlitches UMA stat by henrika · 8 years ago
  86. abee2d8 Roll chromium_revision 2fe6dc66f8..6da2ebcead (484092:484119) by buildbot · 8 years ago
  87. 0fc6d87 Roll chromium_revision cf58257d56..2fe6dc66f8 (483646:484092) by buildbot · 8 years ago
  88. f720704 Added philipel@webrtc.org to webrtc/modules/remote_bitrate_estimator/OWNERS. by philipel · 8 years ago
  89. cb576c5 Fixes build issue based on usage of Android O specific API by henrika · 8 years ago
  90. c43f68e Fix do not unregister bluetooth receiver if it was not registered by Gustavo Garcia · 8 years ago
  91. cc8856c Remove unused static VideoEncoder functions by magjed · 8 years ago
  92. f612998 Override bots to use libstdc++ on Linux by oprypin · 8 years ago
  93. 8eadead Adds support for USB audio devices in AppRTCMobile on Android. by henrika · 8 years ago
  94. a9521e2 Reduce send rate to 50% if overusing before we have an acknowledged bitrate. by terelius · 8 years ago
  95. 2c3161c Changed default value for the duration of the echo in echocanceller 3 by peah · 8 years ago
  96. 0d7f04d Reland of Add received audio/video call duration metrics based on packets. by saza · 8 years ago
  97. 38fecaf Revert of Remove webrtc/tools (patchset #1 id:1 of https://codereview.webrtc.org/2970743003/ ) by ehmaldonado · 8 years ago
  98. d3588cf Improved low-level echo handling in echo canceller 3 by peah · 8 years ago
  99. ed56680 Remove webrtc/tools by ehmaldonado · 8 years ago
  100. 382f21c Revert of Add received audio and video call duration metrics based on packets. (patchset #4 id:140001 of https://codereview.webrtc.org/2957073002/ ) by saza · 8 years ago