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  1. 79348ea Merge WebView M40 build 1832189 into AOSP. by Torne (Richard Coles) · 9 years ago main master android-m-preview-1 android-m-preview-2
  2. c8b2e2b Update mac makefiles. by Torne (Richard Coles) · 9 years ago master-soong android-m-preview
  3. 3b77881 MIPS: Update Chromium WebView makefiles (webrtc). by Paul Lind · 9 years ago
  4. d31cda8 Set default 'mips_arch_variant%' to 'r6' by Gordana Cmiljanovic · 9 years ago
  5. bdd5979 Temporarily disable -Werror in Chromium. by Torne (Richard Coles) · 9 years ago
  6. 3bf0f33 Merge from Chromium at DEPS revision 40.0.2214.27 by Ben Murdoch · 9 years ago ub-webview-m40-release webview-m40_r1 webview-m40_r2 webview-m40_r3 webview-m40_r4
  7. 1dc4c26 Update makefiles after merge of Chromium at 40.0.2214.27 by Ben Murdoch · 9 years ago
  8. 0fbb12c Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 71616dbb8cdd3ea13e0a964d18456ca3fe002dab by Ben Murdoch · 9 years ago
  9. 71616db Merge r7729 into M40 branch. by kjellander@webrtc.org · 10 years ago
  10. 29e9d25 Create a 40 branch from trunk@7660 by tnakamura@webrtc.org · 10 years ago
  11. 5d04ee7 Merge from Chromium at DEPS revision 03655fd3f6d7 by Torne (Richard Coles) · 10 years ago
  12. 361320e Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 60ab669c4c545b328b5c8b0453eb2cdecf851651 by Torne (Richard Coles) · 10 years ago
  13. 60ab669 Remove partially defined WebRtcRTPHeader from Parse(). by pbos@webrtc.org · 10 years ago
  14. 8645a5a Use uint16s for port numbers in webrtc/p2p/base. by pkasting@chromium.org · 10 years ago
  15. a031c17 Fix WebRTC Win64 + BoringSSL build. by henrike@webrtc.org · 10 years ago
  16. eb46bb8 Volume buttons in AppRTCDemo should affect output audio volume (part II). by henrika@webrtc.org · 10 years ago
  17. 77155b0 Merge from Chromium at DEPS revision db3f05efe0f9 by Torne (Richard Coles) · 10 years ago
  18. 98dd0b8 Log formatting fix for VideoEncoderConfig. by pbos@webrtc.org · 10 years ago
  19. 01a30f2 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at b831a9e3d5f9f0563d249b726cffa8a070e58aee by Android Chromium Automerger · 10 years ago
  20. 3ebce78 Remove the state_ member from AudioDecoder by kwiberg@webrtc.org · 10 years ago
  21. b831a9e Adjust parameter in vp9 rate control test. by marpan@webrtc.org · 10 years ago
  22. 10d91f4 Increase speed setting for VP9 (from 5 to 6) and re-enable end_to_end test. by marpan@webrtc.org · 10 years ago
  23. b44eb8f Update makefiles after merge of Chromium at 5a645aa13b82 by Android Chromium Automerger · 10 years ago
  24. 941173a Remove uses of build date/time. by pbos@webrtc.org · 10 years ago
  25. 5232267 Wire up bandwidth stats to the new API and webrtcvideoengine2. by stefan@webrtc.org · 10 years ago
  26. 7f032d2 Update makefiles after merge of Chromium at 2d0da5605d75 by Android Chromium Automerger · 10 years ago
  27. 048258f Restore old behavior for Android in fileutils.cc by kjellander@webrtc.org · 10 years ago
  28. 20b2dc6 Fix android_clang build. by glaznev@webrtc.org · 10 years ago
  29. d1f71cc Revert 7623 "Remove the state_ member from AudioDecoder" by niklas.enbom@webrtc.org · 10 years ago
  30. 1145210 Revert 7625 "Don't use DCHECK when you need the side effects..." by niklas.enbom@webrtc.org · 10 years ago
  31. a91b08b Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 9a8c28f10f329c5ce91e77057933e60224000627 by Android Chromium Automerger · 10 years ago
  32. ff6cd0b Don't use DCHECK when you need the side effects... by kwiberg@webrtc.org · 10 years ago
  33. 5721e27 Remove the state_ member from AudioDecoder by kwiberg@webrtc.org · 10 years ago
  34. 385606f Add support for VP9 in webrtc::Call and video_loopback. by stefan@webrtc.org · 10 years ago
  35. c4cd81a Reduce to 2 probes when probing for initial bandwidth. by stefan@webrtc.org · 10 years ago
  36. 4706c96 Add UMA for measuring the diff between the BWE at 2 seconds compared to the BWE at 20 seconds when the BWE should have converged. by stefan@webrtc.org · 10 years ago
  37. 8f2a7fa Update makefiles after merge of Chromium at a99b7ad25d02 by Android Chromium Automerger · 10 years ago
  38. 9a8c28f Reworked paced sender queue by sprang@webrtc.org · 10 years ago
  39. 83c1dcb Update makefiles after merge of Chromium at 30ec995cdb2d by Android Chromium Automerger · 10 years ago
  40. 6651f18 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at a1fd19c12e4efdf4b8a5f92323070443d50dc34e by Android Chromium Automerger · 10 years ago
  41. 8d28158 Adds support for finch experiments to video_loopback. by stefan@webrtc.org · 10 years ago
  42. 95e0f61 Fix problem with late packets in NetEq by henrik.lundin@webrtc.org · 10 years ago
  43. e765eef Delete VideoReceiveStream channels in destructor. by pbos@webrtc.org · 10 years ago
  44. ab22837 Remove the useless dummy state parameter to WebRtcPcm16b_DecodeW16 by kwiberg@webrtc.org · 10 years ago
  45. 14f28eb Remove the useless dummy state parameter to WebRtcG711_* by kwiberg@webrtc.org · 10 years ago
  46. a1fd19c Remove the codec_type_ member from AudioDecoder by kwiberg@webrtc.org · 10 years ago
  47. 343d3cf Enables AIMD control by default. by stefan@webrtc.org · 10 years ago
  48. dba94e1 Improving error message from neteq_rtpplay by henrik.lundin@webrtc.org · 10 years ago
  49. deb9e49 Add format members to AudioConverter for DCHECKing. by andrew@webrtc.org · 10 years ago
  50. afcd610 Update rate control parameter in vp9 test. by marpan@webrtc.org · 10 years ago
  51. dd298d8 Roll chromium_revision: 28d1981..d3db2ff by marpan@webrtc.org · 10 years ago
  52. 1c755a6 Restore the void return type on WriteWavHeader. by andrew@webrtc.org · 10 years ago
  53. 7e2ad87 replace inline assembly WebRtcNsx_AnalysisUpdate by intrinsics. by andrew@webrtc.org · 10 years ago
  54. 63c5ce8 Add Opus support to neteq_rtpplay by henrik.lundin@webrtc.org · 10 years ago
  55. 3894237 Add UMA metrics for the initial (after two seconds) packet loss, round-trip time and bandwidth estimate of a WebRTC call. by stefan@webrtc.org · 10 years ago
  56. ed94bc8 Add stats for video: by asapersson@webrtc.org · 10 years ago
  57. 9b1042c Add more sanity checks to workaround the unidentified problem that CaptureThread is still running while related resouces are destroyed already. by braveyao@webrtc.org · 10 years ago
  58. 5f53af3 Adjust/increase rate control thresold for a vp9 test. by marpan@webrtc.org · 10 years ago
  59. 6637388 Add VP9 codec to VCM and vie_auto_test. by marpan@webrtc.org · 10 years ago
  60. 0eb4066 Update Android projects to API level 21. by kjellander@webrtc.org · 10 years ago
  61. 99f0de3 replace inline assembly WebRtcNsx_SynthesisUpdateNeon by intrinsics. by andrew@webrtc.org · 10 years ago
  62. 46c5634 Add a WavReader counterpart to WavWriter. by andrew@webrtc.org · 10 years ago
  63. 2dc72e3 Update makefiles after merge of Chromium at a41c404b1c7f by Android Chromium Automerger · 10 years ago
  64. 1d09eed Update makefiles after merge of Chromium at b210e2d62956 by Android Chromium Automerger · 10 years ago
  65. 796056b Update all .isolate files for the new format. by kjellander@webrtc.org · 10 years ago
  66. 4ed1b70 Update Android projects to API level 20. by kjellander@webrtc.org · 10 years ago
  67. cf59c6e Fix N7 camera aspect ratio. by glaznev@webrtc.org · 10 years ago
  68. f8586cc Build fix for MIPS32R6. by andrew@webrtc.org · 10 years ago
  69. 6aaaf9f Fix a name collision with Android libc++ by andrew@webrtc.org · 10 years ago
  70. 08b354d Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 1a02faa335e7d8076b5cf8dd9a584e72669b0c8e by Android Chromium Automerger · 10 years ago
  71. ddb84aa Implement conference-mode temporal-layer screencast. by pbos@webrtc.org · 10 years ago
  72. fbdea25 Configure A/V sync in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
  73. 4ef173b Simplify bwe tests. by stefan@webrtc.org · 10 years ago
  74. a1f6cf7 Revert "Revert part of r7561, "Refactor audio conversion functions."" by andrew@webrtc.org · 10 years ago
  75. 8a3acf6 arm64 iOS build. by tkchin@webrtc.org · 10 years ago
  76. f37dc42 Add 15 fps support for Android devices with missing 15 fps camera mode. by glaznev@webrtc.org · 10 years ago
  77. d7062cc Creating a C++ wrapper class for VAD by henrik.lundin@webrtc.org · 10 years ago
  78. 78f89f1 Revert part of r7561, "Refactor audio conversion functions." by kwiberg@webrtc.org · 10 years ago
  79. bce1329 Refactor audio conversion functions. by andrew@webrtc.org · 10 years ago
  80. 7b5a896 Use external VideoDecoders in VideoReceiveStream. by pbos@webrtc.org · 10 years ago
  81. 2ba45ee Add stats for duplicate sent and received NACK requests. by asapersson@webrtc.org · 10 years ago
  82. 2d05389 common_audio: Removed macro WEBRTC_SPL_RSHIFT_W32 by bjornv@webrtc.org · 10 years ago
  83. 3c47df6 Remove unused code in overuse detector. by asapersson@webrtc.org · 10 years ago
  84. a999336 AudioEncoder: num_10ms_frames_per_packet -> Num10MsFramesInNextPacket by kwiberg@webrtc.org · 10 years ago
  85. ff8f833 Enable G.722 for Chromium builds by henrik.lundin@webrtc.org · 10 years ago
  86. dcfa54a Make an AudioEncoder subclass for Opus by kwiberg@webrtc.org · 10 years ago
  87. 06d5119 Make NSinst_t* const and rename to self in ns_core by aluebs@webrtc.org · 10 years ago
  88. e77df57 Update makefiles after merge of Chromium at f92f0738e9e0 by Android Chromium Automerger · 10 years ago
  89. 8239efe Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 4f2aa0829e4e69972202efb7de2f53cc8858e2c9 by Android Chromium Automerger · 10 years ago
  90. 1a02faa move xmpp and p2p to webrtc by henrike@webrtc.org · 10 years ago
  91. b136cd2 Make local functions static and dropWebRtcNs_ in ns_core by aluebs@webrtc.org · 10 years ago
  92. fc14046 Make all comments whole sentences in ns_core by aluebs@webrtc.org · 10 years ago
  93. 4f2aa08 scoped_ptr.h: Renames function and change namespace scope to fix conflicts with Chromium not detected by the FYI bots. by henrike@webrtc.org · 10 years ago
  94. bc10410 audio_coding/codecs/isac/fix: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>" by bjornv@webrtc.org · 10 years ago
  95. 88b4afa common_audio: Removed trivial macro WEBRTC_SPL_UMUL_16_16 by bjornv@webrtc.org · 10 years ago
  96. e2cf507 Use neteq_unittest_tools in audio_decoder_unittests by henrik.lundin@webrtc.org · 10 years ago
  97. d18fd94 Fix double backslashes in incoming_video_stream.cc by perkj@webrtc.org · 10 years ago
  98. 3aa7f6d Update makefiles after merge of Chromium at 82ca3b654cda by Android Chromium Automerger · 10 years ago
  99. af3d97a Add a simple AudioConverter class. by andrew@webrtc.org · 10 years ago
  100. 2e49acd Only configure the SSL library in one place. by henrike@webrtc.org · 10 years ago