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android / platform / external / chromium_org / third_party / libjingle / source / talk / HEAD
  1. 2eed1ff Merge from Chromium at DEPS revision 03655fd3f6d7 by Torne (Richard Coles) · 10 years ago main master master-soong ub-webview-m40-release android-m-preview android-m-preview-1 android-m-preview-2 webview-m40_r1 webview-m40_r2 webview-m40_r3 webview-m40_r4
  2. c8429e6 Merge third_party/libjingle/source/talk from https://chromium.googlesource.com/external/webrtc/trunk/talk.git at f97800413f157c911aefbf5be167dbd4806a2323 by Torne (Richard Coles) · 10 years ago
  3. f978004 Reapply "Advertise G722 as 8 kHz rather than 16 kHz"" by henrik.lundin@webrtc.org · 10 years ago
  4. b3da49e Change dummy address to use 0.0.0.0 instead of :: by perkj@webrtc.org · 10 years ago
  5. efe9322 Prevent a lot of VideoSendStream reconfigures. by pbos@webrtc.org · 10 years ago
  6. 2cc4257 Refactor webrtcvideoengines to have the default list of supported codecs being generated in runtime. by andresp@webrtc.org · 10 years ago
  7. b87c47f Reland Volume buttons in AppRTCDemo should affect output audio volume (part I). by henrika@webrtc.org · 10 years ago
  8. f1e9d8b Revert "Advertise G722 as 8 kHz rather than 16 kHz" by henrik.lundin@webrtc.org · 10 years ago
  9. eb0e231 (Auto)update libjingle 79326895-> 79329222 by buildbot@webrtc.org · 10 years ago
  10. d0a8777 Volume buttons in AppRTCDemo should affect output audio volume. by henrika@webrtc.org · 10 years ago
  11. 99766a1 Remove deprecated PeerConnection APIs. by perkj@webrtc.org · 10 years ago
  12. 59ad3d2 Removing unused method GetDefaultVideoEncoderConfig. by andresp@webrtc.org · 10 years ago
  13. ce708af Merge from Chromium at DEPS revision db3f05efe0f9 by Torne (Richard Coles) · 10 years ago
  14. 4a7f102 Merge third_party/libjingle/source/talk from https://chromium.googlesource.com/external/webrtc/trunk/talk.git at 7d974c11e23898cd59838c79751b96c45b09ec4b by Android Chromium Automerger · 10 years ago
  15. b763402 (Auto)update libjingle 79285346-> 79320771 by buildbot@webrtc.org · 10 years ago
  16. 44fec83 AppRTCDemoActivity: Add a config CheckBox for enabling/disabling CPU overuse adaptation. by mcasas@webrtc.org · 10 years ago
  17. 05e3f53 Advertise G722 as 8 kHz rather than 16 kHz by henrik.lundin@webrtc.org · 10 years ago
  18. 7d974c1 This fixes a small memory leak (found using Xcode/Instruments on iOS) in by tkchin@webrtc.org · 10 years ago
  19. 16cad08 Wire up bandwidth stats to the new API and webrtcvideoengine2. by stefan@webrtc.org · 10 years ago
  20. 31e29d7 (Auto)update libjingle 79205306-> 79244016 by buildbot@webrtc.org · 10 years ago
  21. 007bff3 (Auto)update libjingle 79200114-> 79205306 by buildbot@webrtc.org · 10 years ago
  22. c55e14e Merge third_party/libjingle/source/talk from https://chromium.googlesource.com/external/webrtc/trunk/talk.git at 19c8d5c35a3e7a7341124f3865a3a117985e7c08 by Android Chromium Automerger · 10 years ago
  23. e889fa5 Cleanup RTCVideoRenderer interface. by tkchin@webrtc.org · 10 years ago
  24. 1fc1bd0 (Auto)update libjingle 79169148-> 79192489 by buildbot@webrtc.org · 10 years ago
  25. 562748d AppRTCDemoActivity: use differnet Themes for different API levels by mcasas@webrtc.org · 10 years ago
  26. 19c8d5c Remove protected files from talk/PRESUBMIT.py. by pbos@webrtc.org · 10 years ago
  27. e540565 Falling back on single-stream on multiple SSRC. by pbos@webrtc.org · 10 years ago
  28. 7a2505e Merge third_party/libjingle/source/talk from https://chromium.googlesource.com/external/webrtc/trunk/talk.git at 4fb0d5af90819ec9666b847f6295b933f58c8301 by Android Chromium Automerger · 10 years ago
  29. 4fb0d5a ReAdd PeerConnectionInterface::AddStream to fix Chrome build. by perkj@webrtc.org · 10 years ago
  30. 2efcfc4 Change the PeerConnection proxy templates to use blocking method calls instead of using Thread::Send. by tommi@webrtc.org · 10 years ago
  31. 42536c6 Prepare for removal of PeerConnectionObserver::OnError. by perkj@webrtc.org · 10 years ago
  32. b428c1d (Auto)update libjingle 79104430-> 79104922 by buildbot@webrtc.org · 10 years ago
  33. 561e751 Android AppRTCDemo improvements: by glaznev@webrtc.org · 10 years ago
  34. 377ebd5 Implement external decoder support in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
  35. 9d70622 Disable PeerConnectionEndToEndTest.CreateDataChannelAfterNegotiate under MSan by henrik.lundin@webrtc.org · 10 years ago
  36. 11643cc Update Android projects to API level 21. by kjellander@webrtc.org · 10 years ago
  37. 3434a47 Change default JVM location to /usr/lib/jvm/java-7-openjdk-amd64 by kjellander@webrtc.org · 10 years ago
  38. 8a771f6 Update all .isolate files for the new format. by kjellander@webrtc.org · 10 years ago
  39. 74b9ec2 Update Android projects to API level 20. by kjellander@webrtc.org · 10 years ago
  40. f4b3bcc Merge third_party/libjingle/source/talk from https://chromium.googlesource.com/external/webrtc/trunk/talk.git at 5910fdfb2ca4b12096bdc5c64aee0afc7f93d426 by Android Chromium Automerger · 10 years ago
  41. 990afb6 Implement conference-mode temporal-layer screencast. by pbos@webrtc.org · 10 years ago
  42. a1feeae Configure A/V sync in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
  43. 36330f3 Adapting bitrate according to maxplaybackrate for Opus. by minyue@webrtc.org · 10 years ago
  44. 723f605 arm64 iOS build. by tkchin@webrtc.org · 10 years ago
  45. b4c42ca Improve the logging when a TCP connection is deleted. by jiayl@webrtc.org · 10 years ago
  46. abe48e1 Cleaning up r7562-7567. by minyue@webrtc.org · 10 years ago
  47. 1975557 (Auto)update libjingle 78822708-> 78823675 by buildbot@webrtc.org · 10 years ago
  48. 0042172 Revert 7563 "before rebase" due to wrong submission by minyue@webrtc.org · 10 years ago
  49. 2f5b8b6 Revert 7564 "to submit" due to wrong submission by minyue@webrtc.org · 10 years ago
  50. d19bf75 to submit by minyue@webrtc.org · 10 years ago
  51. 0defac2 before rebase by minyue@webrtc.org · 10 years ago
  52. 395822f adding default rates by minyue@webrtc.org · 10 years ago
  53. 5cc97a5 Use external VideoDecoders in VideoReceiveStream. by pbos@webrtc.org · 10 years ago
  54. ecffe16 (Auto)update libjingle 78738075-> 78738103 by buildbot@webrtc.org · 10 years ago
  55. 271ce95 ApprtDemo Android: Switch between front and back camera. by perkj@webrtc.org · 10 years ago
  56. 4bc2320 Renaming bandwidth to bitrate in webrtcvoiceengine. by minyue@webrtc.org · 10 years ago
  57. 9c34aa3 Merge third_party/libjingle/source/talk from https://chromium.googlesource.com/external/webrtc/trunk/talk.git at c103da3cd4f12eb017c74963eeaf2204cb9ed5eb by Android Chromium Automerger · 10 years ago
  58. 5910fdf move xmpp and p2p to webrtc by henrike@webrtc.org · 10 years ago
  59. c103da3 (Auto)update libjingle 78642371-> 78680406 by buildbot@webrtc.org · 10 years ago
  60. a62c37c (Auto)update libjingle 78616359-> 78642371 by buildbot@webrtc.org · 10 years ago
  61. f5dbd44 Check if a datachannel in the current local description is an sctp channel before assuming rtp. by tommi@webrtc.org · 10 years ago
  62. b09dc2b Adding setting screen to AppRTCDemo. by glaznev@webrtc.org · 10 years ago
  63. 10f2fab (Auto)update libjingle 78583324-> 78583691 by buildbot@webrtc.org · 10 years ago
  64. 1d3853c Fix the SrtpFilter crash caused by two local offers. by pthatcher@webrtc.org · 10 years ago
  65. b50a456 Merge from Chromium at DEPS revision 614f7b807940 by Torne (Richard Coles) · 10 years ago
  66. 6a9c800 Implement screencast settings for WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
  67. bdfc44c Use flags set by the port allocator. by braveyao@webrtc.org · 10 years ago
  68. 11a355d (Auto)update libjingle 78430441-> 78445452 by buildbot@webrtc.org · 10 years ago
  69. c1577e9 (Auto)update libjingle 78427027-> 78430441 by buildbot@webrtc.org · 10 years ago
  70. 6ae496e Add HD support to Android if we detect a hardware video encoder that can be used. This Change the internal class MediaCodecVideoEncoder to have a one public method for checking if the platform is supported. It also adds &hd=true to the reqest url a hardware encoder is detected. by perkj@webrtc.org · 10 years ago
  71. f9122b7 patch from issue 25469004 by pthatcher@webrtc.org · 10 years ago
  72. 7b2755f (Auto)update libjingle 78381351-> 78389679 by buildbot@webrtc.org · 10 years ago
  73. 3140efa (Auto)update libjingle 78344087-> 78381351 by buildbot@webrtc.org · 10 years ago
  74. 190c56c Add macros and APIs for webrtc histograms. by asapersson@webrtc.org · 10 years ago
  75. 98ff6c0 (Auto)update libjingle 78296920-> 78342456 by buildbot@webrtc.org · 10 years ago
  76. 7fca7cd (Auto)update libjingle 78273470-> 78296920 by buildbot@webrtc.org · 10 years ago
  77. a8314ba Merging Henrik's and Peter's changes for AppRTCDemo by glaznev@webrtc.org · 10 years ago
  78. bf7ac93 (Auto)update libjingle 78262388-> 78262615 by buildbot@webrtc.org · 10 years ago
  79. 81275c6 Remove some disabled tests in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
  80. ee11aad (Auto)update libjingle 78193292-> 78199328 by buildbot@webrtc.org · 10 years ago
  81. d6fe0ff Fix local address leakage when IceTransportsType is relay by guoweis@webrtc.org · 10 years ago
  82. 8135c2a (Auto)update libjingle 78106439-> 78193292 by buildbot@webrtc.org · 10 years ago
  83. fec5bde Merge third_party/libjingle/source/talk from https://chromium.googlesource.com/external/webrtc/trunk/talk.git at 2ce4efe64e83181207479911164695a41302276a by Android Chromium Automerger · 10 years ago
  84. b51b408 Avoid using EGLContext class for Android 4.1 and below. by glaznev@webrtc.org · 10 years ago
  85. 010c874 Set up start bitrate in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
  86. 2ce4efe Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p." by henrike@webrtc.org · 10 years ago
  87. 36d12d8 (Auto)update libjingle 77953038-> 77970462 by buildbot@webrtc.org · 10 years ago
  88. 1ad2ce0 Cleaning up Android AppRTCDemo. by glaznev@webrtc.org · 10 years ago
  89. 40aac8c Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p. by henrike@webrtc.org · 10 years ago
  90. 39523be Merge third_party/libjingle/source/talk from https://chromium.googlesource.com/external/webrtc/trunk/talk.git at 70d23c7ec5c61154de089cd01831c3ae074f95b5 by Android Chromium Automerger · 10 years ago
  91. a1fed0a (Auto)update libjingle 77701902-> 77709729 by buildbot@webrtc.org · 10 years ago
  92. 8f580fc (Auto)update libjingle 77689511-> 77696841 by buildbot@webrtc.org · 10 years ago
  93. 7f06727 Remove unused (no-op) VideoOptions. by pbos@webrtc.org · 10 years ago
  94. 70d23c7 libjingle: use _stricmp instead of deprecated stricmp. by henrike@webrtc.org · 10 years ago
  95. bfddf7a Wire up external encoders. by pbos@webrtc.org · 10 years ago
  96. 6ca136a (Auto)update libjingle 77554188-> 77629208 by buildbot@webrtc.org · 10 years ago
  97. aeaf150 Removes xmllite from talk/xmllite since webrtc/xmllite is used instead. by henrike@webrtc.org · 10 years ago
  98. 6b38543 (Auto)update libjingle 77414393-> 77554188 by buildbot@webrtc.org · 10 years ago
  99. 9782bca Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE. by xians@webrtc.org · 10 years ago
  100. 54490d0 Change setting VP8 codec specific info values by HW VP8 encoder by glaznev@webrtc.org · 10 years ago