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gerrit-public.fairphone.software / platform / external / webrtc / 84d84471f5f0031436e798657784b8938a25ac0f / talk
  1. b024da3 Add support for audio device selection in AppRTCDemo. by henrika@webrtc.org · 10 years ago
  2. 5ad4178 Move the Jingle-specific network code into webrtc/libjingle. by pthatcher@webrtc.org · 10 years ago
  3. 46d4d29 Add field trial for screenshare bitrates when using temporal layers. by sprang@webrtc.org · 10 years ago
  4. 086c8d5 Use a temporary buffer to scale a screencast in OnFrameCaptured by braveyao@webrtc.org · 10 years ago
  5. 4c0544a Move Jingle-specific files from talk/session/media to webrtc/libjingle/session/media. This is part of an ongoing effort to remove Jingle-specific files from the WebRTC repository. by pthatcher@webrtc.org · 10 years ago
  6. 7ce4a58 Add initWithCoder to RTCEAGLVideoView. by tkchin@webrtc.org · 10 years ago
  7. a6f7ba6 Add a AppRTCDemo setting to change the GAE server. by jiayl@webrtc.org · 10 years ago
  8. 742386a Enable payload-based padding by default and remove the API. by stefan@webrtc.org · 10 years ago
  9. 5647877 Breakup Transports and TransportParsers and move TransportParsers into webrtc/libjingle. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository. by pthatcher@webrtc.org · 10 years ago
  10. aacc234 Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository. by pthatcher@webrtc.org · 10 years ago
  11. 16a05dd Clean up the Channel code in AppRTCDemo and use GAE prod server for new signaling mode. by jiayl@webrtc.org · 10 years ago
  12. f5847d7 Move session/tunnel to webrtc/libjingle. This is part of the ongoing effort to move Jingle-specific things out of WebRTC and into its own repository. I won't submit this until all other projects have moved off of compiling this as well. by pthatcher@webrtc.org · 10 years ago
  13. ce4e9a3 Refactor some receive-side stats. by pbos@webrtc.org · 10 years ago
  14. a9cf079 Rename external_hmac_ctx_t to ExternalHmacContext. by pbos@webrtc.org · 10 years ago
  15. 4cb3856 Revert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository." by pthatcher@webrtc.org · 10 years ago
  16. 536f999 Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository. by pthatcher@webrtc.org · 10 years ago
  17. bc03192 Move jingle examples from talk/ into webrtc/libjingle. This is part of the effor to move Jingle out of WebRTC and into its own repository. by pthatcher@webrtc.org · 10 years ago
  18. 209df9b Change MockStatsObserver to grab values inside of OnComplete. by tommi@webrtc.org · 10 years ago
  19. e728ee0 Remove or rename typedefs with _t prefixes. by pbos@webrtc.org · 10 years ago
  20. 950c518 Add adapter_type into Candidate object. by guoweis@webrtc.org · 10 years ago
  21. f050791 Revert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository." by pthatcher@webrtc.org · 10 years ago
  22. 4afb599 Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository. by pthatcher@webrtc.org · 10 years ago
  23. e2b7585 Move ViewRequest and MediaStreams to streamparams.h, and remove dependency on mediasessionclient.h and mediamessages.h. This is part of the effort to remove Jingle-specific code from WebRTC and into its own repository. by pthatcher@webrtc.org · 10 years ago
  24. 55360ae Revert "Add adapter_type into Candidate object." by guoweis@webrtc.org · 10 years ago
  25. aaf02cc Add adapter_type into Candidate object. by guoweis@webrtc.org · 10 years ago
  26. 0b1534c Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess. by pkasting@chromium.org · 10 years ago
  27. e2e199b Clean up StatsObserver's OnComplete methods (address TODOs). by tommi@webrtc.org · 10 years ago
  28. 032b802 (Auto)update libjingle 82121498-> 82126219 by buildbot@webrtc.org · 10 years ago
  29. dd0601f Remove unneeded ctor and add a more practical one by tommi@webrtc.org · 10 years ago
  30. 69bc5a3 Add thread asserts to StatsCollector. by tommi@webrtc.org · 10 years ago
  31. fb108b5 Revert r7885. by pbos@webrtc.org · 10 years ago
  32. 18a3896 Revert r7886:7887. by pbos@webrtc.org · 10 years ago
  33. e575e9c Move WebRtcVideoRenderFrame from webrtcvideoengine2.cc to webrtcvideoframe.h by magjed@webrtc.org · 10 years ago
  34. dee76f3 Move the obvious/easy Jingle-specific code into webrtc/libjingle. by pthatcher@webrtc.org · 10 years ago
  35. 8c9d79a Add adapter_type into Candidate object. by guoweis@webrtc.org · 10 years ago
  36. c57310b Switch kStatsValueName* constants to be enums instead of char*. by tommi@webrtc.org · 10 years ago
  37. 40b276e Cleanup little things found when refactoring. by pthatcher@webrtc.org · 10 years ago
  38. 2b19f06 Wire up RTT statistics to webrtc::Call. by pbos@webrtc.org · 10 years ago
  39. 1351895 Remove old_factory from WebRtcVideoEngine. by pbos@webrtc.org · 10 years ago
  40. 128faba Revert "Revert 7826 "Change Android PeerConnectionUnittest to build usin..."" by perkj@webrtc.org · 10 years ago
  41. a853077 (Auto)update libjingle 81702493-> 81755413 by buildbot@webrtc.org · 10 years ago
  42. aa2c342 Add back a constructor to fix FYI build. by tommi@webrtc.org · 10 years ago
  43. 87776a8 iAppRTCDemo: WebSocket based signaling. by tkchin@webrtc.org · 10 years ago
  44. 0babb4a Fix a comment. by pthatcher@webrtc.org · 10 years ago
  45. c9d155f Move implementation of types in statstypes. to its cc file. by tommi@webrtc.org · 10 years ago
  46. a954c07 AppRTCDemo (Android): built-in AEC should be enabled if device supports it and in combination with Java-based audio layer by henrika@webrtc.org · 10 years ago
  47. 5c3ee4b Add empty implementation file that will hold statstypes.h implementation. by tommi@webrtc.org · 10 years ago
  48. eef8538 Fix AppRTCDemo closing error for KK and JB Android devices. by glaznev@webrtc.org · 10 years ago
  49. 3b3c406 Revert 7826 "Change Android PeerConnectionUnittest to build usin..." by andrew@webrtc.org · 10 years ago
  50. ed7824b Change Android PeerConnectionUnittest to build using Chrome macros. by perkj@webrtc.org · 10 years ago
  51. e2a9261 Improve AppRTCDemo connection speed by sending all by glaznev@webrtc.org · 10 years ago
  52. bd8cc0b Add codereview.settings to the /talk subdirectory by kjellander@webrtc.org · 10 years ago
  53. 599e299 cricket::VideoFrame int64 to int64_t. by kjellander@webrtc.org · 10 years ago
  54. 9b5467e Fix assertion failure when closing data channel, and add a unit test. by bemasc@webrtc.org · 10 years ago
  55. 4b407aa Update AppRTCDemo README with information on 3-dot-apprtc server by glaznev@webrtc.org · 10 years ago
  56. 7169afd With IPv6 enabled, it's important to know whether IPv6 is really used or not. BestConnection is tracked for this purpose. Also added a test case to verify the end to end behavior. by guoweis@webrtc.org · 10 years ago
  57. 369746b Support new WebSocket signaling format. by glaznev@webrtc.org · 10 years ago
  58. 0fb6ad2 Check if cpu_monitor_ exists before Stop(). by pbos@webrtc.org · 10 years ago
  59. d8aed6b Verify that cpu_monitor exists before calling Stop(). by asapersson@webrtc.org · 10 years ago
  60. eb09542 Don't reset sequence number for a stream on deactivate/reactivate. by pthatcher@webrtc.org · 10 years ago
  61. d019551 Change minimum video encoder initialization resolution to by glaznev@webrtc.org · 10 years ago
  62. beee9ce Change back so that Android ApprtcDemo only use one MediaStream containing both audio and video. by perkj@webrtc.org · 10 years ago
  63. 146e0fd Fix the build by putting in a typecast to avoid a comparison between by pthatcher@webrtc.org · 10 years ago
  64. dea5173 Add start bitrate and vp8 hw acceleration option to Android AppRTCDemo. by glaznev@webrtc.org · 10 years ago
  65. 32ec0dd (Auto)update libjingle 81063831-> 81073932 by buildbot@webrtc.org · 10 years ago
  66. 273a414 Report encoded frame size in VideoSendStream. by pbos@webrtc.org · 10 years ago
  67. 2c13f65 Add a platform specific typedef for SOCKET in the peerconnection_server example since it's not universally 'int'. by tommi@webrtc.org · 10 years ago
  68. 3e9ad26 Refactor iOS AppRTC parsing code. by tkchin@webrtc.org · 10 years ago
  69. a71bb60 Revert 7750 "Don't reset sequence number for a stream on deactiv..." by sprang@webrtc.org · 10 years ago
  70. 31f7a0e Don't reset sequence number for a stream on deactivate/reactivate. by sprang@webrtc.org · 10 years ago
  71. 2faf7ee Revert "Revert "This adds an Android apk for running tests on the Java layer of PeerConnection."" by perkj@webrtc.org · 10 years ago
  72. 58edb83 Add video encoder fps and bitrate statistics to Android AppRTCDemo UI. by glaznev@webrtc.org · 10 years ago
  73. 0087318 Implement settable min/start/max bitrates in Call. by pbos@webrtc.org · 10 years ago
  74. dab5d92 Use mirror image for Android AppRTCDemo local preview. by glaznev@webrtc.org · 10 years ago
  75. 8562f23 OWNERS: Remove tomasl@ and mallinath@ by kjellander@webrtc.org · 10 years ago
  76. 308e7ff Revert "This adds an Android apk for running tests on the Java layer of PeerConnection." by kjellander@webrtc.org · 10 years ago
  77. 2751f2a This adds an Android apk for running tests on the Java layer of PeerConnection. by perkj@webrtc.org · 10 years ago
  78. 88d14f4 Remove expensive and unnecessary memory alloc for sending black frames on video by thorcarpenter@google.com · 10 years ago
  79. bdcf38c cricket::VideoFrame: Refactor ConvertToRgbBuffer into base class by magjed@webrtc.org · 10 years ago
  80. 4591fbd Use size_t more consistently for packet/payload lengths. by pkasting@chromium.org · 10 years ago
  81. edc6e57 Support loopback mode and command line execution by glaznev@webrtc.org · 10 years ago
  82. f58b455 cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after. by magjed@webrtc.org · 10 years ago
  83. 6f6ef72 Add DCHECK to ensure that NetEq's packet buffer is not empty by henrik.lundin@webrtc.org · 10 years ago
  84. 2176db3 AppRTCDemoActivity: Add a config CheckBox for enabling/disabling CPU overuse adaptation. (re-land) by henrika@webrtc.org · 10 years ago
  85. 930e004 Add jmi field for packets discarded due to network error by guoweis@webrtc.org · 10 years ago
  86. c72a22c Add preliminary empty file videoframefactory.cc by magjed@webrtc.org · 10 years ago
  87. 4ef22d1 Setting Opus FEC as default by minyue@webrtc.org · 10 years ago
  88. 4ec19e3 Revert 7707 "cricket::VideoAdapter: Drop frames before spending ..." by tommi@webrtc.org · 10 years ago
  89. 858dbbc cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after. by magjed@webrtc.org · 10 years ago
  90. 6a782c2 Revert 7693 "Add jmi field for packets discarded due to network error" breaks chromium's webrtc_cases. by henrike@webrtc.org · 10 years ago
  91. a73d746 Revert 7702 "cricket::VideoAdapter: Drop frames before spending ..." by magjed@webrtc.org · 10 years ago
  92. bbd8cad cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after. by magjed@webrtc.org · 10 years ago
  93. ece3890 Report total bitrate for all streams in GetStats. by pbos@webrtc.org · 10 years ago
  94. 35c1ace Revert 7698 "WebRtcVideoMediaChannel::SetSendParams: Don't cap r..." by magjed@webrtc.org · 10 years ago
  95. a1f5b96 Remove unnecessary copying of libjingle resource files. by kjellander@webrtc.org · 10 years ago
  96. 52da44b WebRtcVideoMediaChannel::SetSendParams: Don't cap resolution by magjed@webrtc.org · 10 years ago
  97. 312614a Add jmi field for packets discarded due to network error by guoweis@webrtc.org · 10 years ago
  98. 6ca6190 Fix a SCTP message reordering issue in datachannel.cc. by jiayl@webrtc.org · 10 years ago
  99. 8038d42 Follow-up fixes for G722 by henrik.lundin@webrtc.org · 10 years ago
  100. c492231 Removes talk/xmllite, talk/xmpp and talk/p2p as they are no longer used by gyp/gn builds. by henrike@webrtc.org · 10 years ago