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gerrit-public.fairphone.software / platform / external / webrtc / 5e5b32706aa1ac3790feac97e7dc2557845a2a71 / talk
  1. 59062d5 Rename SendAndReceiveH264SvcQqvga to VP8 instead. by pbos@webrtc.org · 10 years ago
  2. 8af1104 Avoid reading past end of string in GetLine. by decurtis@webrtc.org · 10 years ago
  3. bab7995 Convert FileMediaEngineTest to use more expects. by pbos@webrtc.org · 10 years ago
  4. 07c83a1 Disable WebRtcVideoMediaChannelSimulcastTest.SimulcastSend tests on Win (take 2) by kjellander@webrtc.org · 10 years ago
  5. 4e5115a RTCPeerConnectionFactory: Explicitly create new worker and signaling threads. by tkchin@webrtc.org · 10 years ago
  6. f6a9714 Remove peer connection and signaling calls from UI thread. by glaznev@webrtc.org · 10 years ago
  7. d95435c Disable WebRtcVideoMediaChannelSimulcastTest.SimulcastSend tests on Win by kjellander@webrtc.org · 10 years ago
  8. cbe7ca8 Roll chromium_revision 8e72e1d..271c6cc (307131:309333) by kjellander@webrtc.org · 10 years ago
  9. 3a63a3c iOS AppRTC: First unit test. by tkchin@webrtc.org · 10 years ago
  10. c37e72e Make setting identical RTP extensions a no-op. by pbos@webrtc.org · 10 years ago
  11. 433006a Fixed style issues from lint and got rid of unused fields. by wzh@webrtc.org · 10 years ago
  12. 8390c27 Add two unit tests for Android AppRTCDemo. by glaznev@webrtc.org · 10 years ago
  13. 896888b Remove min bitrate from simulcast streams. by pbos@webrtc.org · 10 years ago
  14. 9eacb8c Make P2PTestConductor use VirtualSocketServer. by pbos@webrtc.org · 10 years ago
  15. c62749f Parallelize MediaRecorder unittests. by pbos@webrtc.org · 10 years ago
  16. 27f5317 Use the prod GAE server in AppRTCDemo for iOS. by jiayl@webrtc.org · 10 years ago
  17. 5eb71eb Fix style issues from lint. by jiayl@webrtc.org · 10 years ago
  18. b2bda67 Removing old channel code from a few more places. by glaznev@webrtc.org · 10 years ago
  19. c5fd66d Accept incoming pings before remote answer is set to reduce connection latency. by jiayl@webrtc.org · 10 years ago
  20. b024da3 Add support for audio device selection in AppRTCDemo. by henrika@webrtc.org · 10 years ago
  21. 5ad4178 Move the Jingle-specific network code into webrtc/libjingle. by pthatcher@webrtc.org · 10 years ago
  22. 46d4d29 Add field trial for screenshare bitrates when using temporal layers. by sprang@webrtc.org · 10 years ago
  23. 086c8d5 Use a temporary buffer to scale a screencast in OnFrameCaptured by braveyao@webrtc.org · 10 years ago
  24. 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
  25. 7ce4a58 Add initWithCoder to RTCEAGLVideoView. by tkchin@webrtc.org · 10 years ago
  26. a6f7ba6 Add a AppRTCDemo setting to change the GAE server. by jiayl@webrtc.org · 10 years ago
  27. 742386a Enable payload-based padding by default and remove the API. by stefan@webrtc.org · 10 years ago
  28. 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
  29. 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
  30. 16a05dd Clean up the Channel code in AppRTCDemo and use GAE prod server for new signaling mode. by jiayl@webrtc.org · 10 years ago
  31. 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
  32. ce4e9a3 Refactor some receive-side stats. by pbos@webrtc.org · 10 years ago
  33. a9cf079 Rename external_hmac_ctx_t to ExternalHmacContext. by pbos@webrtc.org · 10 years ago
  34. 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
  35. 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
  36. 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
  37. 209df9b Change MockStatsObserver to grab values inside of OnComplete. by tommi@webrtc.org · 10 years ago
  38. e728ee0 Remove or rename typedefs with _t prefixes. by pbos@webrtc.org · 10 years ago
  39. 950c518 Add adapter_type into Candidate object. by guoweis@webrtc.org · 10 years ago
  40. 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
  41. 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
  42. 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
  43. 55360ae Revert "Add adapter_type into Candidate object." by guoweis@webrtc.org · 10 years ago
  44. aaf02cc Add adapter_type into Candidate object. by guoweis@webrtc.org · 10 years ago
  45. 0b1534c Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess. by pkasting@chromium.org · 10 years ago
  46. e2e199b Clean up StatsObserver's OnComplete methods (address TODOs). by tommi@webrtc.org · 10 years ago
  47. 032b802 (Auto)update libjingle 82121498-> 82126219 by buildbot@webrtc.org · 10 years ago
  48. dd0601f Remove unneeded ctor and add a more practical one by tommi@webrtc.org · 10 years ago
  49. 69bc5a3 Add thread asserts to StatsCollector. by tommi@webrtc.org · 10 years ago
  50. fb108b5 Revert r7885. by pbos@webrtc.org · 10 years ago
  51. 18a3896 Revert r7886:7887. by pbos@webrtc.org · 10 years ago
  52. e575e9c Move WebRtcVideoRenderFrame from webrtcvideoengine2.cc to webrtcvideoframe.h by magjed@webrtc.org · 10 years ago
  53. dee76f3 Move the obvious/easy Jingle-specific code into webrtc/libjingle. by pthatcher@webrtc.org · 10 years ago
  54. 8c9d79a Add adapter_type into Candidate object. by guoweis@webrtc.org · 10 years ago
  55. c57310b Switch kStatsValueName* constants to be enums instead of char*. by tommi@webrtc.org · 10 years ago
  56. 40b276e Cleanup little things found when refactoring. by pthatcher@webrtc.org · 10 years ago
  57. 2b19f06 Wire up RTT statistics to webrtc::Call. by pbos@webrtc.org · 10 years ago
  58. 1351895 Remove old_factory from WebRtcVideoEngine. by pbos@webrtc.org · 10 years ago
  59. 128faba Revert "Revert 7826 "Change Android PeerConnectionUnittest to build usin..."" by perkj@webrtc.org · 10 years ago
  60. a853077 (Auto)update libjingle 81702493-> 81755413 by buildbot@webrtc.org · 10 years ago
  61. aa2c342 Add back a constructor to fix FYI build. by tommi@webrtc.org · 10 years ago
  62. 87776a8 iAppRTCDemo: WebSocket based signaling. by tkchin@webrtc.org · 10 years ago
  63. 0babb4a Fix a comment. by pthatcher@webrtc.org · 10 years ago
  64. c9d155f Move implementation of types in statstypes. to its cc file. by tommi@webrtc.org · 10 years ago
  65. 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
  66. 5c3ee4b Add empty implementation file that will hold statstypes.h implementation. by tommi@webrtc.org · 10 years ago
  67. eef8538 Fix AppRTCDemo closing error for KK and JB Android devices. by glaznev@webrtc.org · 10 years ago
  68. 3b3c406 Revert 7826 "Change Android PeerConnectionUnittest to build usin..." by andrew@webrtc.org · 10 years ago
  69. ed7824b Change Android PeerConnectionUnittest to build using Chrome macros. by perkj@webrtc.org · 10 years ago
  70. e2a9261 Improve AppRTCDemo connection speed by sending all by glaznev@webrtc.org · 10 years ago
  71. bd8cc0b Add codereview.settings to the /talk subdirectory by kjellander@webrtc.org · 10 years ago
  72. 599e299 cricket::VideoFrame int64 to int64_t. by kjellander@webrtc.org · 10 years ago
  73. 9b5467e Fix assertion failure when closing data channel, and add a unit test. by bemasc@webrtc.org · 10 years ago
  74. 4b407aa Update AppRTCDemo README with information on 3-dot-apprtc server by glaznev@webrtc.org · 10 years ago
  75. 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
  76. 369746b Support new WebSocket signaling format. by glaznev@webrtc.org · 10 years ago
  77. 0fb6ad2 Check if cpu_monitor_ exists before Stop(). by pbos@webrtc.org · 10 years ago
  78. d8aed6b Verify that cpu_monitor exists before calling Stop(). by asapersson@webrtc.org · 10 years ago
  79. eb09542 Don't reset sequence number for a stream on deactivate/reactivate. by pthatcher@webrtc.org · 10 years ago
  80. d019551 Change minimum video encoder initialization resolution to by glaznev@webrtc.org · 10 years ago
  81. beee9ce Change back so that Android ApprtcDemo only use one MediaStream containing both audio and video. by perkj@webrtc.org · 10 years ago
  82. 146e0fd Fix the build by putting in a typecast to avoid a comparison between by pthatcher@webrtc.org · 10 years ago
  83. dea5173 Add start bitrate and vp8 hw acceleration option to Android AppRTCDemo. by glaznev@webrtc.org · 10 years ago
  84. 32ec0dd (Auto)update libjingle 81063831-> 81073932 by buildbot@webrtc.org · 10 years ago
  85. 273a414 Report encoded frame size in VideoSendStream. by pbos@webrtc.org · 10 years ago
  86. 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
  87. 3e9ad26 Refactor iOS AppRTC parsing code. by tkchin@webrtc.org · 10 years ago
  88. a71bb60 Revert 7750 "Don't reset sequence number for a stream on deactiv..." by sprang@webrtc.org · 10 years ago
  89. 31f7a0e Don't reset sequence number for a stream on deactivate/reactivate. by sprang@webrtc.org · 10 years ago
  90. 2faf7ee Revert "Revert "This adds an Android apk for running tests on the Java layer of PeerConnection."" by perkj@webrtc.org · 10 years ago
  91. 58edb83 Add video encoder fps and bitrate statistics to Android AppRTCDemo UI. by glaznev@webrtc.org · 10 years ago
  92. 0087318 Implement settable min/start/max bitrates in Call. by pbos@webrtc.org · 10 years ago
  93. dab5d92 Use mirror image for Android AppRTCDemo local preview. by glaznev@webrtc.org · 10 years ago
  94. 8562f23 OWNERS: Remove tomasl@ and mallinath@ by kjellander@webrtc.org · 10 years ago
  95. 308e7ff Revert "This adds an Android apk for running tests on the Java layer of PeerConnection." by kjellander@webrtc.org · 10 years ago
  96. 2751f2a This adds an Android apk for running tests on the Java layer of PeerConnection. by perkj@webrtc.org · 10 years ago
  97. 88d14f4 Remove expensive and unnecessary memory alloc for sending black frames on video by thorcarpenter@google.com · 10 years ago
  98. bdcf38c cricket::VideoFrame: Refactor ConvertToRgbBuffer into base class by magjed@webrtc.org · 10 years ago
  99. 4591fbd Use size_t more consistently for packet/payload lengths. by pkasting@chromium.org · 10 years ago
  100. edc6e57 Support loopback mode and command line execution by glaznev@webrtc.org · 10 years ago