1. 96edd56 Sorted headers under rtp_rtcp/. by pbos@webrtc.org · 11 years ago
  2. 69215d8 Include files from webrtc/.. paths in video_engine/. by pbos@webrtc.org · 11 years ago
  3. adf23a5 Direct3D renderer for new VideoEngine API tests. by pbos@webrtc.org · 11 years ago
  4. 717d147 Support sending multiple report blocks and keeping track of statistics on several SSRCs. by stefan@webrtc.org · 11 years ago
  5. 9de89a6 Fix uninitialized value warning in rtp_payload_registry and make sure we return an error if the payload type isn't registered. by stefan@webrtc.org · 11 years ago
  6. 452d853 Fix three uninitialized members in rtp_receiver_impl.cc. by stefan@webrtc.org · 11 years ago
  7. 08933a5 Initialize payload-type frequency in channel.cc. by pbos@webrtc.org · 11 years ago
  8. 28e2075 Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk by henrike@webrtc.org · 11 years ago
  9. 6aa6229 Update version number to 3.35 by tnakamura@webrtc.org · 11 years ago
  10. c79b929 Update version number to 3.34 by tnakamura@webrtc.org · 11 years ago
  11. fc496d9 Add root_path_android.cc to webrtc/test/Android.mk. by pbos@webrtc.org · 11 years ago
  12. f3f1358 Fixed implicit-int-conversion bugs. by pbos@webrtc.org · 11 years ago
  13. cab716c Fix a circular dependency by removing an unnecessary dependency, add a missing include_tests check and missing lib references for android. by stefan@webrtc.org · 11 years ago
  14. f56d612 Create gyp target for bwe components. by stefan@webrtc.org · 11 years ago
  15. af8d5af Initial port of FullStackTest to new VideoEngine API. by pbos@webrtc.org · 11 years ago
  16. 5fc4d34 Arguments need to be separated when implementing gyp-actions. by henrike@webrtc.org · 11 years ago
  17. 1a7b9b9 Cleanup WebRTC tracing by hclam@chromium.org · 11 years ago
  18. e80a934 Added modules_unittests.isolate for ndk-apk builds. by henrike@webrtc.org · 11 years ago
  19. a950300b Disables unit tests that don't work on Android for Android. by henrike@webrtc.org · 11 years ago
  20. a2073af Fixes build breakage when building WebRTC in Chromium and having include_tests=1. by henrike@webrtc.org · 11 years ago
  21. bd3eee3 Fixes broken gyp-condition. by henrike@webrtc.org · 11 years ago
  22. 34773d9 Unreverts revert: Makes it possible to find files used by some unit tests when running them as Chrome native tests. by henrike@webrtc.org · 11 years ago
  23. 1932fe1 Use scoped_ptr<> for loopback.cc by pbos@webrtc.org · 11 years ago
  24. 66b2e5c Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the by stefan@webrtc.org · 11 years ago
  25. d4d9480 Added gum4.html, a multiple camera opening demo, each opening with a different resolution and/or frame rate. by mcasas@webrtc.org · 11 years ago
  26. db7d82f Revert 4298 "Makes it possible to find files used by some unit t..." by pbos@webrtc.org · 11 years ago
  27. caf2fcc Makes it possible to find files used by some unit tests when running them as Chrome native tests. by henrike@webrtc.org · 11 years ago
  28. 21beaf9 Adding Stefan as VideoEngine owner, removing Per. by mflodman@webrtc.org · 11 years ago
  29. 0b8636a In AudioDeviceWindowsCore::_EnumerateEndpointDevicesAll(), continue enumerating if one individual device failed. by braveyao@webrtc.org · 11 years ago
  30. 1303af3 Makes it possible to build ndk-apks of native unit tests if the workspace is inside a chromium checkout. by henrike@webrtc.org · 11 years ago
  31. d900e8b Proper spacing for end-of-namespace comments. by pbos@webrtc.org · 11 years ago
  32. 45426ea In call to Opus decoder: frame length too large by tina.legrand@webrtc.org · 11 years ago
  33. f6f033f Possible divide by 0 in ACM. by tina.legrand@webrtc.org · 11 years ago
  34. b1698ab Error in update of read index in ACM by tina.legrand@webrtc.org · 11 years ago
  35. ecd3c80 Add Magnus to root owners. by tommi@webrtc.org · 11 years ago
  36. c66aaaf Rename unit_test.{cc,h} under module_unittest. by pbos@webrtc.org · 11 years ago
  37. 510dfad Update myself in webrtc watchlist by yujie.mao@webrtc.org · 11 years ago
  38. 65a1f2c Remove log of undefined input values in GetCodec. by pbos@webrtc.org · 11 years ago
  39. 504af45 Diff NTP and internal once in VideoCaptureImpl. by pbos@webrtc.org · 11 years ago
  40. 546c91d Build all java files into jar for each module on Android by fischman@webrtc.org · 11 years ago
  41. d4803ce WebRTCViEDemo: Use global reference when passing variables across different threads by yujie.mao@webrtc.org · 11 years ago
  42. 90cc3b9 Android opengles renderer: add thread sync to swap frame and draw native. by braveyao@webrtc.org · 11 years ago
  43. 5616aba Suppress excessive logging in video_coding by hclam@chromium.org · 11 years ago
  44. 2a7fd53 Moves tools/update.py to trunk/webrtc/tools and updates it so that it no longer pulls any information from the DEPS file. by henrike@webrtc.org · 11 years ago
  45. 83cebb2 Removes unused main function that is poluting the build. by henrike@webrtc.org · 11 years ago
  46. 0021632 Re-add WebRTCDemo dependencies as dependencies (not just inputs) because they also need to be built for this target! by fischman@webrtc.org · 11 years ago
  47. 1d4a2d5 Move TickTime::QueryOsForTicks out-of-line by fischman@webrtc.org · 11 years ago
  48. 4cf1a8a Removes kStateFree and kStateDecoding, added a free_frames_ list which simplifies finding a free frame. by stefan@webrtc.org · 11 years ago
  49. 7bcc7e3 Fixed bad parameter passing in compare_videos.py by phoglund@webrtc.org · 11 years ago
  50. 2de80dd Fix unnamed-type-template-args warnings on clang. by pbos@webrtc.org · 11 years ago
  51. 3145a64 Correctly rebuild WebRTCDemo-debug.apk after modules/ source file changes. by fischman@webrtc.org · 11 years ago
  52. e6168f5 Adding a first simple version of overuse detection, but not hooked up. by mflodman@webrtc.org · 11 years ago
  53. 1c986e7 Removed ViE file API. by mflodman@webrtc.org · 11 years ago
  54. a5fd2f1 Do basic parsing of RTCP headers in PcapFileReader to enable log filtering. by solenberg@webrtc.org · 11 years ago
  55. 892d750 Add *.DS_Store to .gitignore so that ._.DS_Store is ignored too. by solenberg@webrtc.org · 11 years ago
  56. 91811e2 Remove unused multi stream bandwidth estimator. by solenberg@webrtc.org · 11 years ago
  57. a4c5abb Make sure padding packets are sent. by stefan@webrtc.org · 11 years ago
  58. bb25256 Added changes in apprtc demo to ignore turn address through query string for FF. Also made sure that the iceServers array doesnot include transport parameter in turn url for FF. Finally removed turn: from the turn_url when creating iceservers for pre-M28 chrome. by vikasmarwaha@webrtc.org · 11 years ago
  59. 3348ae2 mac: Mark kCGLPFAFullScreen as allowed in a 10.6-only function. by sergeyu@chromium.org · 11 years ago
  60. bb4f225 Roll libvpx to 207593. -pick up libvpx roll to c259af4f. by marpan@webrtc.org · 11 years ago
  61. 6eb53f7 Fix memory bot failure by hclam@chromium.org · 11 years ago
  62. 2e402ce Enqueue packet in pacer if sending fails by hclam@chromium.org · 11 years ago
  63. 9ca7360 VCM: removing max jitter estimate by mikhal@webrtc.org · 11 years ago
  64. 0851df8 Remove unneeded *_NOT_SUPPORTED from VoEAudioProcessing. by andrew@webrtc.org · 11 years ago
  65. 8ccb9f9 Fixes some pacer/padding issues found while testing. by stefan@webrtc.org · 11 years ago
  66. 2d7617a Add dummy Android test APK to be used for buildbot automation testing. by kjellander@webrtc.org · 11 years ago
  67. d7148c8 Use 3 threads for higher than 720p resolutions by fbarchard@google.com · 11 years ago
  68. 30fb7b8 Add a log message to see video delay break down by hclam@chromium.org · 11 years ago
  69. 6cfe178 Chromium Android tools for test execution. by kjellander@webrtc.org · 11 years ago
  70. a20eb91 Make ScreenCapturerMac work in versions of OSX before Lion. by sergeyu@chromium.org · 11 years ago
  71. 9e18279 Enable ScreenCapturer unittests by sergeyu@chromium.org · 11 years ago
  72. a590b41 Use intptr_t to represent window IDs on all platforms. by sergeyu@chromium.org · 11 years ago
  73. 508a84b Wire up pacer-based padding. by stefan@webrtc.org · 11 years ago
  74. 50fb4af Revert r4145 "Revert 4127 "Switch frame list implementation to std::map."" by stefan@webrtc.org · 11 years ago
  75. c8b29a2 Revert r4146 "Revert 4104 "Refactor jitter buffer to use separate lists for de..."" by stefan@webrtc.org · 11 years ago
  76. 7262ad1 Fix AV sync issue by hclam@chromium.org · 11 years ago
  77. 9b23ecb Log current and target AV delay in ViESyncModule by hclam@chromium.org · 11 years ago
  78. 63e9888 Merge more tests into modules_{unit,integration}tests. by kjellander@webrtc.org · 11 years ago
  79. f27389c WebRTCDemo: ensures that using front and back camera work as expected. by henrike@webrtc.org · 11 years ago
  80. d4ed1a3 Fixes linker issue with no op trace. by henrike@webrtc.org · 11 years ago
  81. a193339 Apprtc CSS: Add flip to local view of FireFox and remove warning of Canary by braveyao@webrtc.org · 11 years ago
  82. fee739c Risk of division by zero. by turaj@webrtc.org · 11 years ago
  83. dd97ef4 Revert 4211 "Build all java files into jar for each module on An..." by fischman@webrtc.org · 11 years ago
  84. 20a993f Disable ViEExtendedIntegrationTest.RunsCodecTestWithoutErrors test. by kjellander@webrtc.org · 11 years ago
  85. 935d705 Disable ViEExtendedIntegrationTest.RunsCodecTestWithoutErrors test. by kjellander@webrtc.org · 11 years ago
  86. 04996cd Fix breakage due to test_fec conversion to gtest. by kjellander@webrtc.org · 11 years ago
  87. 22bbbdf Convert test_fec to gtest by kjellander@webrtc.org · 11 years ago
  88. 7124dd8 Disable ViEStandardIntegrationTest.RunsRtpRtcpTestWithoutErrors test. by kjellander@webrtc.org · 11 years ago
  89. 18275a8 Update bots to make LKGR progress. by kjellander@webrtc.org · 11 years ago
  90. b097670 G722_1/G722_1C codecs won't instantiate by tina.legrand@webrtc.org · 11 years ago
  91. 2ef9513 libyuv r723 with convert util -attenuate feature used to fix transparent pixels used by Effects. By attenuating and then unattenuating, any transparent pixels will have RGB value of black, which will filter correctly when bilinear resized. by fbarchard@google.com · 11 years ago
  92. 6c35e0b Reorganize test targets in WebRTC by kjellander@webrtc.org · 11 years ago
  93. 6d6d95e Add support for test disable files in webrtc_tests.py by kjellander@webrtc.org · 11 years ago
  94. 1374965 Build all java files into jar for each module on Android by fischman@webrtc.org · 11 years ago
  95. 4af0878 Allow the screen capturer to capture oversized cursors and cursors without alpha channel (Windows). by alexeypa@chromium.org · 11 years ago
  96. 5e03f8a Landing binary cursor image files to be used in a follow up CL. by alexeypa@chromium.org · 11 years ago
  97. dfa1c4a libyuv r722 for OWNERS file for chromium, white space fix for lint, unittests on scale use randomize to reduce overhead, and neon change from vld1.u8 to vld1.8 for better compiler portability. by fbarchard@google.com · 11 years ago
  98. fe6b571 AppRTCDemo: delete hosted android_channel.html now that it's no longer necessary. by fischman@webrtc.org · 11 years ago
  99. 5137b97 Updated WebRTC version to 3.33 by elham@webrtc.org · 11 years ago
  100. 509754c Making no NACK mode work again in VideoEngine. by mflodman@webrtc.org · 11 years ago