Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / 4d905f88c65ba6691de105b6aa0978e246a7409e
  1. 4d905f8 Fix clang warnings in rtp. by andrew@webrtc.org · 13 years ago
  2. f1f93d8 Remove warning settings more stringent than Chromium's common.gypi. by andrew@webrtc.org · 13 years ago
  3. a80d026 Fix clang warnings in voice engine. by andrew@webrtc.org · 13 years ago
  4. bbd8908 Fix clang warnings in video coding. by andrew@webrtc.org · 13 years ago
  5. 49e58da Fix release mode "unused variable" warnings in peerconnection. by andrew@webrtc.org · 13 years ago
  6. 20f7428 Temporarily switch to Chrome's hosted libvpx copy. by andrew@webrtc.org · 13 years ago
  7. 87c546e Remove peerconnectionimpl_callbacks.h from libjingle.gyp. by tommi@webrtc.org · 13 years ago
  8. fac55d5 I've added two watchlist definitions (NetEQ and video codecs), and added myself to be notified when something changes. by henrik.lundin@webrtc.org · 13 years ago
  9. c6e54a9 Update to the peerconnection sample app. by tommi@webrtc.org · 13 years ago
  10. 84519ec Fixing some inconsistencies in WebRTC audio coding module. I've added setup information for all codecs which are not part of WebRTC, but possible to hook in. by tina.legrand@webrtc.org · 13 years ago
  11. d9e11b4 by zakkhoyt@google.com · 13 years ago
  12. 777ef59 Fix clang warnings in video engine. by andrew@webrtc.org · 13 years ago
  13. 243db12 media_opt_util: Fixed an assert and some code cleanup for AvgRecoveryFEC function. by marpan@google.com · 13 years ago
  14. b15bfd3 * Add the time_stamp as one parameter to the ViE ExternalRenderer interface. by wu@webrtc.org · 13 years ago
  15. ebb2744 To fix warning for unused variable. And fix some warning in test. by turajs@google.com · 13 years ago
  16. eaf3185 Take care of unused variable. by turajs@google.com · 13 years ago
  17. 9562a36 Last fixes to build with gcc 4.6. by andrew@webrtc.org · 13 years ago
  18. cdefd42 Adding code review watchlist to automatically CC e-mail addresses when new CLs are created. by mflodman@webrtc.org · 13 years ago
  19. 830099e Add a gyp flag to disable video functionality from dependencies shared by voice and video engine. by andrew@webrtc.org · 13 years ago
  20. e9f0e2e Moved _rtpReceiver to protected by pwestin@webrtc.org · 13 years ago
  21. c7d5f62 Fix build errors on Windows. by tommi@webrtc.org · 13 years ago
  22. 74c640a fix build break Review URL: http://webrtc-codereview.appspot.com/132004 by turajs@google.com · 13 years ago
  23. 7796c02 Wrap encode, decode, PLC NB functions in #define to avoid warnings. by turajs@google.com · 13 years ago
  24. 8ecd0e8 Remove Clang warning for PCM16B. by turajs@google.com · 13 years ago
  25. f990eb3 Hi, by mallinath@webrtc.org · 13 years ago
  26. eba8c32 Resolving a race condition issue related to using shared devices by punyabrata@google.com · 13 years ago
  27. 8811e5a Switch to a smoother stretch algorithm on Windows and delete buffers from previous conversations on linux when switching back to peer list. by tommi@webrtc.org · 13 years ago
  28. 3266d8d have the voe_cmd_test compiled with external transport enabled. by xians@google.com · 13 years ago
  29. e74a9ea AudioDeviceUtility::WaitForKey() pulls two characters if the first one is a newline, but discards the final value. by xians@google.com · 13 years ago
  30. 3fcabbe Modified include path after after moving files to webrtc_dev. by perkj@google.com · 13 years ago
  31. 932096c Porting gtalk alsa impl from depot to webrtc by xians@google.com · 13 years ago
  32. 46171cf video coding tests: Adding a Normal distribution to simulate packet arrival times by mikhal@webrtc.org · 13 years ago
  33. 8571af7 Updating to new VP8 rtp format by henrik.lundin@webrtc.org · 13 years ago
  34. 0973408 Fixes build issue in http://code.google.com/p/webrtc/issues/detail?id=56. by hellner@google.com · 13 years ago
  35. 81fd2bf New ACM codec database, created at compile time. by tina.legrand@webrtc.org · 13 years ago
  36. af931bd Update of iLBC reference files for version 1.1.1, new SQRT. by tina.legrand@webrtc.org · 13 years ago
  37. a41b4ce Changing iLBC to use the new improved SQRT, WebRtcSpl_SqrtFloor(). by tina.legrand@webrtc.org · 13 years ago
  38. c9cff24 Adding classes to be used for logging data within the engines and the by stefan@webrtc.org · 13 years ago
  39. 4094c49 Temporarily use digital AGC in WebRTC since Chromium can't support analog AGC. by perkj@google.com · 13 years ago
  40. c9b75e0 removing the warnings from the voe tests. by xians@google.com · 13 years ago
  41. 2aa5d50 Issue reported in WebRTC. A variable is defined and set, but never used. by tina.legrand@webrtc.org · 13 years ago
  42. 36450af Removing unsupported codecs from ptypes file by henrik.lundin@webrtc.org · 13 years ago
  43. 92bace1 Hi, by mallinath@webrtc.org · 13 years ago
  44. bd4494c Remove the divide-by-2 when mixing. by andrew@webrtc.org · 13 years ago
  45. b7ac56d video coding tests: updating quality tests following r466 by mikhal@webrtc.org · 13 years ago
  46. d24a97f video coding test: deleting unused file(resampler_test.cc) by mikhal@webrtc.org · 13 years ago
  47. 2c3b1fb video_coding tests: removing unused functionality from test_util by mikhal@webrtc.org · 13 years ago
  48. a057a95 video_coding: Updating protection logic in media optimization utility: by mikhal@webrtc.org · 13 years ago
  49. 552f173 video_coding: Moving video metrics computation to a designated file. by mikhal@webrtc.org · 13 years ago
  50. e46d69f Fix gcc 4.6 set but unused warnings in AEC. by andrew@webrtc.org · 13 years ago
  51. b62c776 moving all new version related files to webrtc_dev and removed from webrtc. by mallinath@webrtc.org · 13 years ago
  52. ffbe7a7 Cast away the unused state argument value to silence gcc 4.6 warnings. by andrew@webrtc.org · 13 years ago
  53. 7f2bbbb To remove all calls involving scratch-memory by turajs@google.com · 13 years ago
  54. ac55f7b by turajs@google.com · 13 years ago
  55. 7659b36 revert the file path in the voe_auto_test by xians@google.com · 13 years ago
  56. 350d091 Send the hangup message when asked to disconnect from a peer. by tommi@webrtc.org · 13 years ago
  57. c57f9c3 by xians@webrtc.org · 13 years ago
  58. 4fcb0ca Removing warning in video capture module for linux and auto test. by mflodman@webrtc.org · 13 years ago
  59. b55c988 Updated peerconnection_unittest slightly. Also added it to the build. by hellner@google.com · 13 years ago
  60. 23a8065 Fixed broken build due to r453. by hellner@google.com · 13 years ago
  61. b2801f3 Added the remaining test cases for the webrtcsession unittest also some minor refactoring. by hellner@google.com · 13 years ago
  62. 59af6f1 Porting Mac keypress detection from GIPS repository. by zakkhoyt@google.com · 13 years ago
  63. ba9bd69 video_coding_tests: Fix build error by mikhal@webrtc.org · 13 years ago
  64. aed0348 Roll gyp 985:1012 by andrew@webrtc.org · 13 years ago
  65. 40373cc Bugfix in unittest and some minor refactoring. by hellner@google.com · 13 years ago
  66. eb9572e Add the new peerconnection factory to the scons file. by wu@webrtc.org · 13 years ago
  67. e129ae9 by niklas.enbom@webrtc.org · 13 years ago
  68. 3227ed5 Fixed potential memory leak in unit test and removed an unnecessary copy. by hellner@google.com · 13 years ago
  69. 102b227 First version of the peerconnection client application for Linux. by tommi@webrtc.org · 13 years ago
  70. 137ece4 * Make GetReadyState accessible via the PeerConnection interface. by tommi@webrtc.org · 13 years ago
  71. 44d356d Fix unused variable warning in spatial_resampler.cc by stefan@webrtc.org · 13 years ago
  72. 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
  73. 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
  74. 5cc9c68 Fixing a warning discovered while compiling with clang. by turajs@google.com · 13 years ago
  75. 057efc8 Removed unused variables and unnecessary assert: causing build error in vpm_test. by marpan@google.com · 13 years ago
  76. 4f39000 Fix warnings on Ubuntu 11.04 (gcc 4.5) by andrew@webrtc.org · 13 years ago
  77. 37fd004 Remove the X11 headers we don't need. by wu@webrtc.org · 13 years ago
  78. cf36b2a Match new[] / delete [] by frkoenig@google.com · 13 years ago
  79. accd686 Implementation of media streams. Work in progress. by perkj@google.com · 13 years ago
  80. 49cbc51 Fix unused variable warning in video_coding. by stefan@webrtc.org · 13 years ago
  81. 7f593c1 Fix gcc 4.6 unused variable warnings in audio_processing. by andrew@webrtc.org · 13 years ago
  82. 6724cf8 VP8: Adding a flag to indicate the libvpx version. When in Cayuga, additional API's will be used. by mikhal@webrtc.org · 13 years ago
  83. 9788e18 * Add PeerConnectionProxy to forward all the API calls to signaling thread. by wu@webrtc.org · 13 years ago
  84. 4482b04 revert r430 to keep webrtc always ready to roll in chromium. by wjia@google.com · 13 years ago
  85. f9f1deb Get ready for libvpx Cayuga (v0.9.7-p1). by wjia@google.com · 13 years ago
  86. 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
  87. a386fc0 Fixes build warnings due to unused variables. by hellner@google.com · 13 years ago
  88. 9aa9996 Different solution than the one suggested in http://code.google.com/p/webrtc/issues/detail?id=56 however, should solve the same problem. by hellner@google.com · 13 years ago
  89. 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
  90. ae53bf8 The variable ‘dummy’ set but not used being treated as errors in Fedora. by wu@webrtc.org · 13 years ago
  91. ceb148c Fix compile warnings in Release configuration. by andrew@webrtc.org · 13 years ago
  92. 06ad81f video_coding: changing the UpdateMethod function (protection settings). by mikhal@webrtc.org · 13 years ago
  93. 12f1fc4 Fix initialization defect in constructor webrtc::ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(WebRtc_Word32, bool) initialization list. by perkj@google.com · 13 years ago
  94. a3fc1aa Fix Issue 59. Fix a constructor cast warning in video_X11_channel.cc. by perkj@google.com · 13 years ago
  95. a070adb Moved member RTPSender from private to protected. by pwestin@webrtc.org · 13 years ago
  96. 9d64705 by xians@webrtc.org · 13 years ago
  97. 5895ea1 Fixes volume problem controls, happening with some Logitech headsets. Originally submitted as gips p4 depot CL 38122. by punyabrata@webrtc.org · 13 years ago
  98. 9695e75 Resolve a crash related to pulseAudio where we need to check if by punyabrata@google.com · 13 years ago
  99. 288c869 Optimization of 'cftmdl': by cduvivier@google.com · 13 years ago
  100. 0e16b78 VP8: Removing VP8Latest flag following the update to the Cayuga release by mikhal@webrtc.org · 13 years ago