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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / libjingle / source / talk / refs/tags/FP2-open-16.12.0 / app / webrtc
  1. 40539b8 Fix a problem in Thread::Send. by jiayl@webrtc.org · 10 years ago
  2. 56dcc5b Change Android video renderer to maintain video aspect by glaznev@webrtc.org · 10 years ago
  3. 928c130 Switch HW video decoder to output byte buffers if video by glaznev@webrtc.org · 10 years ago
  4. b015440 Enable ipv6 by default for webrtc under a Finch experiment. by guoweis@webrtc.org · 10 years ago
  5. 1373d9f Reapply 23529005 after fixing the build break issue (Chromium:582133002) by guoweis@webrtc.org · 10 years ago
  6. 88b24bb Fix HW video decoder crash on some Android KK devices. by glaznev@webrtc.org · 10 years ago
  7. 0d6677d Fixing compilation failure in peerconnection_jni.cc with WEBRTC_CHROMIUM_BUILD. by glaznev@webrtc.org · 10 years ago
  8. 637a5ca A few fixes to avoid crash in HW codec on device orientation change. by glaznev@webrtc.org · 10 years ago
  9. 34f7659 Revert maximum video codec resolution on Android back to 720p again. by glaznev@webrtc.org · 10 years ago
  10. 131bfb7 Enable HW video decoding on Qualcomm devices. by glaznev@webrtc.org · 10 years ago
  11. 19eb91c Split video engine android initialization into each internal module initialization. by andresp@webrtc.org · 10 years ago
  12. cacae61 Java VideoRenderer class may be backed by two different native by glaznev@webrtc.org · 10 years ago
  13. 1f59bcb Revert 7184 "Enable ipv6 by default for webrtc under a Finch exp..." by kjellander@webrtc.org · 10 years ago
  14. 109ba4e Add a target for the approved subset of rtc_base. by andrew@webrtc.org · 10 years ago
  15. a846c20 HW video decoding optimization to better support HD resolution: by glaznev@webrtc.org · 10 years ago
  16. 6d3e4cf Enable ipv6 by default for webrtc under a Finch experiment. by guoweis@webrtc.org · 10 years ago
  17. 22ea492 Make BW checks > 0 in peerconnection_unittest.cc. by pbos@webrtc.org · 10 years ago
  18. 6a9dda8 Temporary revert maximum video codec resolution back to 1080p. by glaznev@webrtc.org · 10 years ago
  19. 938f588 Disabling initializeAndroidGlobals when built with WEBRTC_CHROMIUM_BUILD. by glaznev@webrtc.org · 10 years ago
  20. 1c84149 Mark all virtual overrides in the hierarchy of Module as virtual and OVERRIDE. by henrik.lundin@webrtc.org · 10 years ago
  21. ffa7ab2 Fix frame rate selection for Android camera. by glaznev@webrtc.org · 10 years ago
  22. ded08bf (Auto)update libjingle 75141932-> 75179475 by buildbot@webrtc.org · 10 years ago
  23. 966f092 Fixes two issues in how we handle OfferToReceiveX for CreateOffer: by jiayl@webrtc.org · 10 years ago
  24. 2ca5657 Relanding https://code.google.com/p/webrtc/source/detail?r=7093, after it got by mallinath@webrtc.org · 10 years ago
  25. 3b7f619 Peerconnection_jni to use webrtc/base/checks.h instead of implementing its own. by andresp@webrtc.org · 10 years ago
  26. 87dac0a Revert 7093: "Implementing ICE Transports type handling in libjingle transport." by henrike@webrtc.org · 10 years ago
  27. df1715c Fix a bot-breaking memory leak from early returning in ParseMediaDescription. by jiayl@webrtc.org · 10 years ago
  28. 5e09ab5 Revert "Fixes two issues in how we handle OfferToReceiveX for CreateOffer:" because it broke content_browsertests on Android. by jiayl@webrtc.org · 10 years ago
  29. 265eb22 Implementing ICE Transports type handling in libjingle transport. by mallinath@webrtc.org · 10 years ago
  30. 6ea7f02 Updated SCTP SDP attributes according to draft-ietf-mmusic-sctp-sdp-07 by jiayl@webrtc.org · 10 years ago
  31. 0d3bdd0 Fix an issue in MediaStreamSignaling that a remotely create DataChannel is added to the list twice. by jiayl@webrtc.org · 10 years ago
  32. 469a71a Reduce maximum video resolution for Android. by glaznev@webrtc.org · 10 years ago
  33. 1e086db Fixes two issues in how we handle OfferToReceiveX for CreateOffer: by jiayl@webrtc.org · 10 years ago
  34. 32b3ea4 Abort Negotiate() if DoCreateOffer() fails. by pbos@webrtc.org · 10 years ago
  35. fdfa168 Remove deprecated RTCVideoRenderer constructor. by tkchin@webrtc.org · 10 years ago
  36. 662c7aa When the peerconnection creates the offer with a constraint to disable the audio offering, stats will not get properly updated. by jiayl@webrtc.org · 10 years ago
  37. 65ab28e Add 60 fps video support by niklas.enbom@webrtc.org · 10 years ago
  38. ec7a06f - Make local constant non-static. - Remove spammy log line. by solenberg@webrtc.org · 10 years ago
  39. faebcce Remove test constructor in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
  40. 146a241 Move constant so it is not stripped out for TSAN bots. by kjellander@webrtc.org · 10 years ago
  41. 1d1a138 As expected, r6569 (https://code.google.com/p/webrtc/source/detail?r=6965) caused memcheck bots to complain. Adding expections for that, in line with outher peerconnection tests. by solenberg@webrtc.org · 10 years ago
  42. 5462dd9 (Auto)update libjingle 73927775-> 74032598 by buildbot@webrtc.org · 10 years ago
  43. 9460de6 (Auto)update libjingle 73794259-> 73891518 by buildbot@webrtc.org · 10 years ago
  44. e9db8f6 (Auto)update libjingle 73399579-> 73626167 by henrike@webrtc.org · 10 years ago
  45. cdfebb3 Revert 6897 (i.e. Reland 6863) - "Revert 6863 "Refactor StatsCollector and associated..." by tommi@webrtc.org · 10 years ago
  46. ce1ee0d Revert 6863 "Refactor StatsCollector and associated types." by niklas.enbom@webrtc.org · 10 years ago
  47. ab4404d Fixes failure triggered by include order re-ordering. by henrike@webrtc.org · 10 years ago
  48. cf81adf (Auto)update libjingle 73222930-> 73226398 by buildbot@webrtc.org · 10 years ago
  49. e24ed48 Fix the audio source failure due to unsupported constraints. by xians@webrtc.org · 10 years ago
  50. eea342e Removing ASSERT for tcp candidate for port 0 and 9, as Android clients by mallinath@webrtc.org · 10 years ago
  51. b20d541 Refactor StatsCollector and associated types. by tommi@webrtc.org · 10 years ago
  52. c639043 Fix a bug in parsing IceCandidate with IPV6 address. by jiayl@webrtc.org · 10 years ago
  53. 58c89b1 Encoding and Decoding of TCP candidates as defined in RFC 6544. by mallinath@webrtc.org · 10 years ago
  54. b752f54 (Auto)update libjingle 72847605-> 72850595 by buildbot@webrtc.org · 10 years ago
  55. 0a4294d Maintain the order of the m-lines in CreateOffer and CreateAnswer. by jiayl@webrtc.org · 10 years ago
  56. 4fa2e85 Adds the support of RTCOfferOptions for PeerConnectionInterface::CreateOffer. by jiayl@webrtc.org · 10 years ago
  57. 2a86ce2 (Auto)update libjingle 72097588-> 72159069 by buildbot@webrtc.org · 10 years ago
  58. 459f356 (Auto)update libjingle 72016417-> 72097588 by buildbot@webrtc.org · 10 years ago
  59. 251a1c7 Revert of 6778 "Refactor StatsCollector and associated types." Breakes FYI bots. by henrike@webrtc.org · 10 years ago
  60. f54b0eb Refactor StatsCollector and associated types. by tommi@webrtc.org · 10 years ago
  61. 4c20a21 Fix a crash in statscollector.cc caused by invoking methods on the worker thread which destroys the Transport. by jiayl@webrtc.org · 10 years ago
  62. 312862c (Auto)update libjingle 71829282-> 71834788 by buildbot@webrtc.org · 10 years ago
  63. 8e88c29 Re-revert of 6747 "Refactor StatsCollector and associated types." by henrike@webrtc.org · 10 years ago
  64. dbb1185 (Auto)update libjingle 71775619-> 71778545 by buildbot@webrtc.org · 10 years ago
  65. 4352adb Revert 6747 "Refactor StatsCollector and associated types." Breakes FYI bots. by henrike@webrtc.org · 10 years ago
  66. 48e0f37 Revert 6766 "Temporarily add a default ctor to StatsReport and make |id| non const. As soon as I've updated the chrome side, I'll revert this cl." by henrike@webrtc.org · 10 years ago
  67. 394c028 (Auto)update libjingle 71766184-> 71775619 by buildbot@webrtc.org · 10 years ago
  68. f2399c8 Temporarily add a default ctor to StatsReport and make |id| non const. by tommi@webrtc.org · 10 years ago
  69. 7db3e8d Add VP8 video decoding hw acceleration support to Java Peerconnection library. by glaznev@webrtc.org · 10 years ago
  70. b91678d (Auto)update libjingle 71575585-> 71599033 by buildbot@webrtc.org · 10 years ago
  71. 93f95be Disable GetStatsForInvalidTrack while I rewrite it. by tommi@webrtc.org · 10 years ago
  72. 2aec728 Refactor StatsCollector and associated types. by tommi@webrtc.org · 10 years ago
  73. c0acc70 Revert 6745 "Refactor StatsCollector and associated types." by tommi@webrtc.org · 10 years ago
  74. 2a0c1e3 Refactor StatsCollector and associated types. by tommi@webrtc.org · 10 years ago
  75. 04d269c Ignore empty data in DataChannel::Send to match FF's behavior. by jiayl@webrtc.org · 10 years ago
  76. 12569f9 Revert "Reland r6707 with the fix for callclient.cc." by jiayl@webrtc.org · 10 years ago
  77. 77abf9b (Auto)update libjingle 71456173-> 71456344 by buildbot@webrtc.org · 10 years ago
  78. 4417c9c Reland r6707 with the fix for callclient.cc. by jiayl@webrtc.org · 10 years ago
  79. 624cbc5 (Auto)update libjingle 71452608-> 71453580 by buildbot@webrtc.org · 10 years ago
  80. bf21bdd Creates the default track if the remote media content is send-only and there is no stream in the SDP. by jiayl@webrtc.org · 10 years ago
  81. 88853c7 Assigning a priority to TURN server list passed to PeerConnection. First entry in the TURN server list will get the highest priotity and so forth. by mallinath@webrtc.org · 10 years ago
  82. 3588cd9 fix by jiayl@webrtc.org · 10 years ago
  83. 4604619 Revert 6707 "Add support of multiple STUN servers in UDPPort." by wu@webrtc.org · 10 years ago
  84. c7db594 Make sure b lines appear before all the a lines. Per RFC 4566, the order of media description should be: by wu@webrtc.org · 10 years ago
  85. 5367a5b Add support of multiple STUN servers in UDPPort. by jiayl@webrtc.org · 10 years ago
  86. 676a3f8 Minor refactoring of StatsCollector. by tommi@webrtc.org · 10 years ago
  87. 2829d69 A step towards changing StatsReport::Value::name to an enum. by tommi@webrtc.org · 10 years ago
  88. fd1e42f Make StatsCollector depend on always having a valid session pointer. by tommi@webrtc.org · 10 years ago
  89. 9d8544c (Auto)update libjingle 71107853-> 71115715 by buildbot@webrtc.org · 10 years ago
  90. a26d7d6 (Auto)update libjingle 71099685-> 71107853 by buildbot@webrtc.org · 10 years ago
  91. b873792 Fixed the stats problem when new track is using the same ssrc as the previous track. by xians@webrtc.org · 10 years ago
  92. b7104bb Change Timing::WallTimeNow to be static. by tommi@webrtc.org · 10 years ago
  93. 1d74a2c Add tkchin@ to OWNERS. by tkchin@webrtc.org · 10 years ago
  94. 6eff6e0 Change SdpSerializeCandidate to output candidate line without the "a=" and without the leading \r\n", i.e. candidate-attribute as defined in section 15.1 of [ICE]. by wu@webrtc.org · 10 years ago
  95. d24be91 (Auto)update libjingle 69648312-> 69830415 by buildbot@webrtc.org · 10 years ago
  96. 2a1d7c0 Limits the send and receive buffer by bytes, not by packets. by jiayl@webrtc.org · 10 years ago
  97. 39c443b Re-evalutes the ICE role on ICE restart. Also unifies the logic of ICE restart. by jiayl@webrtc.org · 10 years ago
  98. 37998a1 (Auto)update libjingle 69617317-> 69623266 by buildbot@webrtc.org · 10 years ago
  99. 3443a94 (Auto)update libjingle 69588980-> 69589535 by buildbot@webrtc.org · 10 years ago
  100. 6214542 (Auto)update libjingle 69555283-> 69567902 by buildbot@webrtc.org · 10 years ago