1. 0077272 Roll chromium_revision 4101b15..fbc4ecf (376664:376680) by kjellander · 9 years ago
  2. da1d656 Roll chromium_revision 789f25d..4101b15 (376663:376664) by kjellander · 9 years ago
  3. b9f943d Roll chromium_revision 1120bd3..789f25d (376660:376663) by kjellander · 9 years ago
  4. a18f638 Include "sharedexclusivelock.cc" in Chromium GN build. by jbauch · 9 years ago
  5. fa830dc Roll chromium_revision 9dc1788..1120bd3 (376655:376660) by kjellander · 9 years ago
  6. bf81175 Roll chromium_revision 5618e25..9dc1788 (376642:376655) by kjellander · 9 years ago
  7. 330d3d8 Roll chromium_revision fa5d546..5618e25 (376142:376642) by kjellander · 9 years ago
  8. b9dd7c5 Remove GetTransport() from TransportChannelImpl by mikescarlett · 9 years ago
  9. 9bf5cde Update build_ios_libs.sh script to build new Objective-C API and gather header files. by hjon · 9 years ago
  10. 91fe304 vp9: Adjust parameter for a test in videoprocessor_integrationtest.cc by Marco · 9 years ago
  11. a9d0892 Add initial bitrate and frame resolution parameters to quality scaler. by Alex Glaznev · 9 years ago
  12. 0013dcc Simplify SSRC usage inside ViEEncoder. by Peter Boström · 9 years ago
  13. 7254890 Nuke SetSenderBufferingMode. by Peter Boström · 9 years ago
  14. da9ae0c Revert of CQ: Change Android trybots to not run device tests. (patchset #1 id:1 of https://codereview.webrtc.org/1715643002/ ) by kjellander · 9 years ago
  15. e2d83d6 Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt() by sprang · 9 years ago
  16. 45c44f0 Simplify EncoderStateFeedback. by Peter Boström · 9 years ago
  17. 9674d7c Revert of Prevent data race in MessageQueue. (patchset #3 id:40001 of https://codereview.webrtc.org/1675923002/ ) by jbauch · 9 years ago
  18. fc968a2 Fix sequence-number replay race for padding. by Peter Boström · 9 years ago
  19. 88788ad Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/ by kwiberg · 9 years ago
  20. df88460 Prevent data race in MessageQueue. by jbauch · 9 years ago
  21. 1e80ce4 webrtc::RtpPacket name freed for better RtpPacket by Danil Chapovalov · 9 years ago
  22. c51d694 CQ: Disable linux_baremetal pending installation fix. by kjellander@webrtc.org · 9 years ago
  23. 728012e Changed the semantics of Buffer::Clear to not alter the capacity by ossu · 9 years ago
  24. ecdeb4c CQ: Change Android trybots to not run device tests. by kjellander@webrtc.org · 9 years ago
  25. c4e3ead Blacklist "build/c++11" cpplint filter. by jbauch · 9 years ago
  26. 4458d09 Drop support for playing output through aplay in intelligibility_proc by Alejandro Luebs · 9 years ago
  27. b3fb71c Add RTCAudioSession proxy class. by Zeke Chin · 9 years ago
  28. 9ac4df1 iOS: Enable modules_unittests and common_audio_unittests by kjellander · 9 years ago
  29. 235aaa7 Fix Linux 32-bit compilation after sysroot switch. by Henrik Kjellander · 9 years ago
  30. 66a9928 Roll chromium_revision 1d144ca..fa5d546 (375480:376142) by kjellander@webrtc.org · 9 years ago
  31. 0e2e50c Always append the BYE packet type at the end by aleungbroadsoft · 9 years ago
  32. 452df1c Suppress UBSan errors in common_audio by henrik.lundin · 9 years ago
  33. f45381e VideoCapturerAndroid: Report onFirstFrameAvailable() for textures as well by Magnus Jedvert · 9 years ago
  34. 5199c74 AndroidVideoCapturer getSupportedFormats(): Change interface from JSON string to List/vector by Magnus Jedvert · 9 years ago
  35. 347c0bb Android GLShader: Check return value of glCreateShader() by magjed · 9 years ago
  36. 3ee73a5 Make RemoteBitrateEstimator::GetStats() virtual. by Stefan Holmer · 9 years ago
  37. fd22e6c Change PeerConnectionFactory.setVideoHwAccelerationOptions to be able to replace Egl context. by Per · 9 years ago
  38. 74db777 Revert of Remove GetTransport() from TransportChannelImpl (patchset #3 id:40001 of https://codereview.webrtc.org/1691673002/ ) by guidou · 9 years ago
  39. 59c634b Re-add RemoteBitrateEstimator::GetStats. by Stefan Holmer · 9 years ago
  40. 3234819 Fix and simplify the power estimation in the IntelligibilityEnhancer by Alejandro Luebs · 9 years ago
  41. ee18220 Remove GetTransport() from TransportChannelImpl by mikescarlett · 9 years ago
  42. ee75c7a Compile rtc_base_objc for Mac. by tkchin · 9 years ago
  43. e3c6c82 When doing continual gathering, remove the local ports when a corresponding network is dropped. by honghaiz · 9 years ago
  44. a08bb0d Disabled the test EndToEndTest RestartingSendStreamPreservesRtpState due to the test being flaky. by peah · 9 years ago
  45. b7f89d6 Replace scoped_ptr with unique_ptr in webrtc/voice_engine/ by kwiberg · 9 years ago
  46. dabf07f Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/vad/ by kwiberg · 9 years ago
  47. a293ef0 Apply VideoOptions per stream. by nisse · 9 years ago
  48. 789ba92 Simplify CongestionController. by Stefan Holmer · 9 years ago
  49. bad7804 Remove unused VideoSendStream TransportAdapter. by Peter Boström · 9 years ago
  50. 62eaacf Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/test/ by kwiberg · 9 years ago
  51. 28c99bc iOS: Include legacy objc API in all.gyp + fix H264 libyuv dependency by kjellander · 9 years ago
  52. 4b4dc86 Remove conference_mode flag from AudioOptions and VideoOptions. by nisse · 9 years ago
  53. 22785c7 Exclude legacy objc API tests properly. by kjellander · 9 years ago
  54. 69e59e6 [rtp_rtcp] rtc::scoped_ptr<rtcp::RawPacket> replaced with rtc::Buffer by danilchap · 9 years ago
  55. 67680c1 Ignore padding-only RTX packets in test. by Peter Boström · 9 years ago
  56. a332e2d Added boilerplate code for being able to test the upcoming AEC functionality. by peah · 9 years ago
  57. 0206000 iOS: Add resource files for tests and implement OutputPath by kjellander · 9 years ago
  58. 85d8bb0 Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/transient/ by kwiberg · 9 years ago
  59. 9d3584c Implementing unified plan encoding of msid. by deadbeef · 9 years ago
  60. 25d6a0f Adding TSan suppressions temporarily to fix some flaky unit tests. by deadbeef · 9 years ago
  61. e1a0c94 Add network cost as part of the connection ranking. by honghaiz · 9 years ago
  62. 2c38c20 Fix out-of-buffer write in iLBC by henrik.lundin · 9 years ago
  63. 44c65e9 Enable adaptive threshold experiment by default. by Stefan Holmer · 9 years ago
  64. 9d0c432 Remove video-codec max bitrate from TMMBN. by Peter Boström · 9 years ago
  65. d20327c Increase the allowed number of probe packets in test to please msan. by Stefan Holmer · 9 years ago
  66. ee31f0a Fix out-of-buffer read in iLBC by henrik.lundin · 9 years ago
  67. 62a5ccd Update bitrate only when we have incoming packet. by Stefan Holmer · 9 years ago
  68. 58cf5f1 Changed order of events when synthesizing a call. by peah · 9 years ago
  69. 0453ef8 Prevent busy-looping PacedSender on small packets. by Peter Boström · 9 years ago
  70. 1794b26 Extract ViESyncModule outside ViEChannel. by Peter Boström · 9 years ago
  71. a3dc79e Move SSLIdentity Generate() implementations from .h to .cc file. by Torbjorn Granlund · 9 years ago
  72. 71e92dc Avoid overflow in WebRtcSpl_Sqrt by henrik.lundin · 9 years ago
  73. 092c951 Roll chromium_revision aefd358..1d144ca (375443:375480) by kjellander · 9 years ago
  74. e8dc081 Implement certificate lifetime parameter as required by WebRTC RFC. by torbjorng · 9 years ago
  75. b1ae3a4 Stop decoders in VideoReceiveStream destructor. by Peter Boström · 9 years ago
  76. 461121c Replaced eglbase_jni with with holding a EglBase in PeerConnectionFactory. by perkj · 9 years ago
  77. 8259c2d Roll chromium_revision 8d1f312..aefd358 (375401:375443) by kjellander · 9 years ago
  78. 8110482 Rename gtest_exclude for rtc_pc_unittests by kjellander@webrtc.org · 9 years ago
  79. 56e6269 Rename gtest_exclude for rtc_media_unittests. by Peter Boström · 9 years ago
  80. 88c52a7 Disable VerifyHistogramStatsWithRed on DrMemory. by Peter Boström · 9 years ago
  81. 16c5a96 Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/ by kwiberg · 9 years ago
  82. 3959397 Reland of Don't send FEC for H.264 with NACK enabled. (patchset #1 id:1 of https://codereview.webrtc.org/1692783005/ ) by Peter Boström · 9 years ago
  83. cde5d6b removed five redundant avsync tests to make webrtc_perf_test faster by Danil Chapovalov · 9 years ago
  84. e829f58 Rename libjingle_p2p_unittest -> rtc_pc_unittests by kjellander@webrtc.org · 9 years ago
  85. 3747838 Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/test/ by kwiberg · 9 years ago
  86. 1d9ada7 Roll chromium_revision 99c33e8..8d1f312 (375382:375401) by kjellander · 9 years ago
  87. 2d0c332 Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/neteq/ by kwiberg · 9 years ago
  88. 04af839 Move refcount.h and scoped_ref_ptr.h to rtc_base_approved. BUG= by tommi · 9 years ago
  89. 290ab41 Roll chromium_revision ee3223b..99c33e8 (375377:375382) by kjellander · 9 years ago
  90. 91d9756 Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/codecs/ by kwiberg · 9 years ago
  91. 3f5f5aa Roll chromium_revision e1bc525..ee3223b (375361:375377) by kjellander · 9 years ago
  92. be61562 Moved the GainControlForNewAGC class to be a separate file. by peah · 9 years ago
  93. 46e2cb8 Roll chromium_revision 96c72eb..e1bc525 (375334:375361) by kjellander · 9 years ago
  94. 88ec91d Roll chromium_revision c9db86b..96c72eb (374913:375334) by kjellander · 9 years ago
  95. 88b0a22 Add VP9 to full stack tests. by asapersson · 9 years ago
  96. 29ffdc1 Revert of Don't send FEC for H.264 with NACK enabled. (patchset #5 id:80001 of https://codereview.webrtc.org/1687303002/ ) by deadbeef · 9 years ago
  97. 5e7834e Android: Make VideoCapturer an interface for all VideoCapturers to implement by Magnus Jedvert · 9 years ago
  98. e78765b Removes Nexus 5 from AEC and NS blacklists by henrika · 9 years ago
  99. 579e832 Fix race on VCM protection callback. by Peter Boström · 9 years ago
  100. b72dada Remove Reset from conditionally-compiled decoders. by Peter Boström · 9 years ago