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gerrit-public.fairphone.software / platform / external / webrtc / aee87f0c875994effe9003004279485a2b716a68 / third_party_mods
  1. caef503 Removing PeerConnection sample client and libjingle from webrtc. by wu@webrtc.org · 13 years ago
  2. 59f16ec Introduced ARM version of WebRtcSpl_SqrtFloor(). Function cycles reduced by ~ 30% in a real time VOE test in an android device (Nexus-S, ARMv7a). by kma@webrtc.org · 13 years ago
  3. df55166 Adding libsrtp in libjingle.gyp and changing DEPS to retrieve libsrtp code from chromium repository. by mallinath@webrtc.org · 13 years ago
  4. 2a61e15 PortAllocator is now passed to PeerConnection instead of PeerConnectionFactory in new libjingle release. by mallinath@webrtc.org · 13 years ago
  5. 9f9af7d Remove Peerconnection Dev branch. by perkj@webrtc.org · 13 years ago
  6. 36a992b Merge streamparams and mediasession from libjingle and made necessary changes in peerconnection. by perkj@webrtc.org · 13 years ago
  7. 8885d22 by henrike@webrtc.org · 13 years ago
  8. 4d8c818 The implementation before this change list keeps the ownership of memory that is used by peer connection instances in the peer connection manager. This means that if the peer connection manager is deleted before all peer connections it has created, these peer connections will be pointing to invalid memory. by henrike@webrtc.org · 13 years ago
  9. 35a12cd Fix comment. by perkj@webrtc.org · 13 years ago
  10. 8129752 Add refcount and scoped_refptr. by perkj@webrtc.org · 13 years ago
  11. 94cfde7 Removed scoped_refptr from libjingle.gyp by perkj@webrtc.org · 13 years ago
  12. 7e08613 Move refcount and scoped_refptr to merge with libjingle. Deleted scoped_refptr_msg.h. by perkj@webrtc.org · 13 years ago
  13. 58b4f1c Fixes broken build in peerconnection unit tests by mallinath@webrtc.org · 13 years ago
  14. aa32319 Implement unittest for proxies of MediaStreamTrackInterface and MediaStreamInterface. by perkj@webrtc.org · 13 years ago
  15. ca8b3a3 kind() method in track interface is changed to std::string to keep uniformity with other get methods by mallinath@webrtc.org · 13 years ago
  16. 96ba190 ref_count.h file name changed to refcount.h to keep as other ( most ) files are named in libjingle. by mallinath@webrtc.org · 13 years ago
  17. 2ebc9ce Fix broken PeerConnection Dev build. by perkj@webrtc.org · 13 years ago
  18. f553ec7 Notifier and RefCount interface and implementation class name changed according to the naming convention. by mallinath@webrtc.org · 13 years ago
  19. 1305a1d Fix rendering in new PeerConnection API. by perkj@webrtc.org · 13 years ago
  20. 0d55c8f Adding peerconnection_unittest. by henrike@webrtc.org · 13 years ago
  21. 5cb3064 The change will separate the media tracks based on media type. MediaStreamInterface currently will have list for audio and video. This way we don't need to check for the track type before converting to respective mediatrack. by mallinath@webrtc.org · 13 years ago
  22. 63257d4 Implement proxy for both audio and video tracks. by perkj@webrtc.org · 13 years ago
  23. c01c358 session/phone/channel.cc updates after new push of libjingle revision. by mallinath@webrtc.org · 13 years ago
  24. ebc0a00 One of Justin comment was to have XXXXInterface and XXXX, rather than XXXX and XXXXImpl. So here are the changes, i don't like to call some the classes as interfaces like MediaStreamTrackListInterface, but they fit the criteria to be called as interface. by mallinath@webrtc.org · 13 years ago
  25. 03a8699 Fixes for build errors introduced most likely earlier today. by henrike@webrtc.org · 13 years ago
  26. 0c37811 Define NO_SOUND_SYSTEM for chromium build. by wu@webrtc.org · 13 years ago
  27. ebc405d Remove the fakeportallocator from the libjingle.gyp. by wu@webrtc.org · 13 years ago
  28. 6c2d710 * Update to use the new libjingle release. by wu@webrtc.org · 13 years ago
  29. 103f33b Changes after comments received from Justin and Harald. Few comments are not implemented like moving track implementation to base<> and then have child classes based on the type of track. by mallinath@webrtc.org · 13 years ago
  30. 6a34d58 Implement MediaStreamProxy. by perkj@webrtc.org · 13 years ago
  31. 77d7d54 Replace the DestroyDeviceInfo with a virtual destructor. by wu@webrtc.org · 13 years ago
  32. 38e400a Adding native client test page to test loopback. by perkj@webrtc.org · 13 years ago
  33. ea89922 Add VideoCaptureFactory so that we don't need to expose VideoCaptureImpl. by wu@webrtc.org · 13 years ago
  34. 73ba416 Fix OnClose(socket, NO_ERROR) compile error on Linux. by perkj@webrtc.org · 13 years ago
  35. f6ab63c Update PeerConnection_client to open a video capture device. by perkj@webrtc.org · 13 years ago
  36. 3a6d4f4 Fix setting VideoCaptureModule and VideoRenderer for local and remote streams. by perkj@webrtc.org · 13 years ago
  37. fa41d80 Fixes session state transition and registering observer. by mallinath@webrtc.org · 13 years ago
  38. 29787c7 Changes to WebRtcSession after Provider(s) interface addition. by mallinath@webrtc.org · 13 years ago
  39. 487e401 Moving creation of sessiondescriptions to webrtcsession. by perkj@webrtc.org · 13 years ago
  40. cb4ab65 Moved creation of objects to the signaling thread. by perkj@webrtc.org · 13 years ago
  41. bafca10 Temp hook in WebRtcSession to VideoChannel. by mallinath@webrtc.org · 13 years ago
  42. 1b6ff7a Connecting PeerConnectionImpl with WebrtcSession and MediaStreamHandlers. by perkj@webrtc.org · 13 years ago
  43. 666f56b MediaStreamHandler implements eventhandlers for streams and tracks. by perkj@webrtc.org · 13 years ago
  44. 236fcaa Interface changes after we have the Serialize and Deserialize. by wu@webrtc.org · 13 years ago
  45. ed6d555 * Add the crypto serialize and deserialize. * Populate candidates test data. by wu@webrtc.org · 13 years ago
  46. ee2c391 more webrtc session changes. Transport and TransportChannel handling is complete. Need work on session state. by mallinath@webrtc.org · 13 years ago
  47. 99239d5 First compiling version of peerconnection_client_dev using the new Peerconnection API. by perkj@webrtc.org · 13 years ago
  48. c93e363 * Add Deserize for PeerConnectionMessage by wu@webrtc.org · 13 years ago
  49. e804ee1 This patch hooks up PeerConnectionImpl to PeerConnectionSignaling. by perkj@webrtc.org · 13 years ago
  50. 78083bf * Add Serialize functions to PeerConnectionMessage. by wu@webrtc.org · 13 years ago
  51. 9a1249d first cut of webrtcsession. Doesn't do much other than creating files and empty function bodies. by mallinath@webrtc.org · 13 years ago
  52. 5045f67 Add SignalUpdateSessionDescription to PeerConnectionSignaling. by perkj@webrtc.org · 13 years ago
  53. 2f56ff4 Implementation of PcSignaling. A Class to handle signaling between peerconnections. by perkj@webrtc.org · 13 years ago
  54. c389aa2 Fix the bad video issue on Window client by increasing the rtp recv buffer size. by ronghuawu@google.com · 13 years ago
  55. 679e64d Cleaning up of Peerconnection API. by perkj@webrtc.org · 13 years ago
  56. c49db5e The files included in devicemanager.h/cc still have some conflict with chromium. Let's keep the devicemanager mods for now and I will see how can we solve this next. by wu@webrtc.org · 13 years ago
  57. cb99f78 * Update to use libjingle r85. by wu@webrtc.org · 13 years ago
  58. b27f3f1 Update to use the new opensource jsoncpp and remove jsoncpp mods. by wu@webrtc.org · 13 years ago
  59. d3185fe refactor the gyp file to gypi file. by xians@google.com · 13 years ago
  60. 0cc68dc Change Video capture module to be reference counting. Also prevent the module from beeing deleted using the interface. by perkj@webrtc.org · 13 years ago
  61. c273019 linking error after tommi's changes. by mallinath@webrtc.org · 13 years ago
  62. 73f98ae Temporarily switch the numeric locale formatting to 'classic' while we process the signaling message. by tommi@webrtc.org · 13 years ago
  63. 73d6551 Adds reference counting to the ADM. by henrika@google.com · 13 years ago
  64. 2d9af90 Fix error when building Peerconnection in Chrome. by perkj@webrtc.org · 13 years ago
  65. e5ea752 New Peerconnection manager implementation. Ready for review. by perkj@google.com · 13 years ago
  66. 5a15ab9 Move the WebRtcDeviceManager and WebRtcMediaEngine to libjingle. by wu@webrtc.org · 13 years ago
  67. 87c546e Remove peerconnectionimpl_callbacks.h from libjingle.gyp. by tommi@webrtc.org · 13 years ago
  68. b15bfd3 * Add the time_stamp as one parameter to the ViE ExternalRenderer interface. by wu@webrtc.org · 13 years ago
  69. f990eb3 Hi, by mallinath@webrtc.org · 13 years ago
  70. 3fcabbe Modified include path after after moving files to webrtc_dev. by perkj@google.com · 13 years ago
  71. 4094c49 Temporarily use digital AGC in WebRTC since Chromium can't support analog AGC. by perkj@google.com · 13 years ago
  72. 92bace1 Hi, by mallinath@webrtc.org · 13 years ago
  73. b62c776 moving all new version related files to webrtc_dev and removed from webrtc. by mallinath@webrtc.org · 13 years ago
  74. b55c988 Updated peerconnection_unittest slightly. Also added it to the build. by hellner@google.com · 13 years ago
  75. b2801f3 Added the remaining test cases for the webrtcsession unittest also some minor refactoring. by hellner@google.com · 13 years ago
  76. 40373cc Bugfix in unittest and some minor refactoring. by hellner@google.com · 13 years ago
  77. eb9572e Add the new peerconnection factory to the scons file. by wu@webrtc.org · 13 years ago
  78. 3227ed5 Fixed potential memory leak in unit test and removed an unnecessary copy. by hellner@google.com · 13 years ago
  79. 137ece4 * Make GetReadyState accessible via the PeerConnection interface. by tommi@webrtc.org · 13 years ago
  80. 1cdc6b5 This CL adding a factory class which has the responsibility of creating peerconnection objects. This is very basic class doesn't do any reference count, user has the responsibility to delete the object externally. by mallinath@webrtc.org · 13 years ago
  81. d1015fe Replaced regular sleep with a talk_base::Thread::ProcessMessages(..) call so that Posts get some execution time from the main thread. by hellner@google.com · 13 years ago
  82. accd686 Implementation of media streams. Work in progress. by perkj@google.com · 13 years ago
  83. 9788e18 * Add PeerConnectionProxy to forward all the API calls to signaling thread. by wu@webrtc.org · 13 years ago
  84. dec6aa5 This CL will remove sending any signal after calling Close and RemoveStream. I am thinking to remove Close method at all, since application can directly delete the object if it wants to end the call with all active streams. Will send that change later in a different CL. by mallinath@webrtc.org · 13 years ago
  85. 87c9b74 * Use the current thread as the signaling thread and worker thread to keep the unit test simple and easier to debug. by wu@webrtc.org · 13 years ago
  86. 6f555dc by mallinath@webrtc.org · 13 years ago
  87. eb29a97 * Remove the previous renderer before set a new one. by wu@webrtc.org · 13 years ago
  88. bca7fa0 by mallinath@webrtc.org · 13 years ago
  89. 310689e by mallinath@webrtc.org · 13 years ago
  90. 765c918 Changes based on the review comments. by wu@webrtc.org · 13 years ago
  91. bfc63ae by mallinath@webrtc.org · 13 years ago
  92. 18cec47 Bug fix for OnRemoveStream. by ronghuawu@google.com · 13 years ago
  93. 467b1a9 by mallinath@webrtc.org · 13 years ago
  94. 74a49a8 Two changes: by wu@webrtc.org · 13 years ago
  95. 424e76a by mallinath@google.com · 13 years ago
  96. 8910f27 Switch to webrtc.org accounts (for those which exist). by andrew@webrtc.org · 13 years ago
  97. 492dbc2 Use the full path instead of the current directory. by ronghuawu@google.com · 13 years ago
  98. 35f5345 * Point the webrtc libjingle dependency to third_party_mods. by ronghuawu@google.com · 13 years ago
  99. a852de7 The new PeerConnection Api (under development) from p4 libjingle/...@38654. by ronghuawu@google.com · 13 years ago
  100. 907c355 Add owner file for the third_party_mods\libjingle. by ronghuawu@google.com · 13 years ago