1. 222efdc 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
  2. 39445b0 Adding Stefan as VideoEngine owner, removing Per. by mflodman@webrtc.org · 11 years ago
  3. 531a99b In AudioDeviceWindowsCore::_EnumerateEndpointDevicesAll(), continue enumerating if one individual device failed. by braveyao@webrtc.org · 11 years ago
  4. a0ebe97 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
  5. 3b89e10 Proper spacing for end-of-namespace comments. by pbos@webrtc.org · 11 years ago
  6. cbb3966 In call to Opus decoder: frame length too large by tina.legrand@webrtc.org · 11 years ago
  7. 1bd3140 Possible divide by 0 in ACM. by tina.legrand@webrtc.org · 11 years ago
  8. cbb535a Error in update of read index in ACM by tina.legrand@webrtc.org · 11 years ago
  9. 98ac1e8 Rename unit_test.{cc,h} under module_unittest. by pbos@webrtc.org · 11 years ago
  10. 438be80 Remove log of undefined input values in GetCodec. by pbos@webrtc.org · 11 years ago
  11. 5cf0fd1 Diff NTP and internal once in VideoCaptureImpl. by pbos@webrtc.org · 11 years ago
  12. 50c1aef Build all java files into jar for each module on Android by fischman@webrtc.org · 11 years ago
  13. 5e742a8 WebRTCViEDemo: Use global reference when passing variables across different threads by yujie.mao@webrtc.org · 11 years ago
  14. e63c003 Android opengles renderer: add thread sync to swap frame and draw native. by braveyao@webrtc.org · 11 years ago
  15. af60a80 Suppress excessive logging in video_coding by hclam@chromium.org · 11 years ago
  16. 8a5cb95 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
  17. c1624d5 Removes unused main function that is poluting the build. by henrike@webrtc.org · 11 years ago
  18. c7eab28 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
  19. d305e11 Move TickTime::QueryOsForTicks out-of-line by fischman@webrtc.org · 11 years ago
  20. 8148118 Removes kStateFree and kStateDecoding, added a free_frames_ list which simplifies finding a free frame. by stefan@webrtc.org · 11 years ago
  21. c10fc53 Fixed bad parameter passing in compare_videos.py by phoglund@webrtc.org · 11 years ago
  22. 3656192 Fix unnamed-type-template-args warnings on clang. by pbos@webrtc.org · 11 years ago
  23. 555f1cd Correctly rebuild WebRTCDemo-debug.apk after modules/ source file changes. by fischman@webrtc.org · 11 years ago
  24. f8f91d6 Adding a first simple version of overuse detection, but not hooked up. by mflodman@webrtc.org · 11 years ago
  25. 0291c80 Removed ViE file API. by mflodman@webrtc.org · 11 years ago
  26. 0be9202 Do basic parsing of RTCP headers in PcapFileReader to enable log filtering. by solenberg@webrtc.org · 11 years ago
  27. 20cfda6 Remove unused multi stream bandwidth estimator. by solenberg@webrtc.org · 11 years ago
  28. 8ee45da Make sure padding packets are sent. by stefan@webrtc.org · 11 years ago
  29. c8c333d mac: Mark kCGLPFAFullScreen as allowed in a 10.6-only function. by sergeyu@chromium.org · 11 years ago
  30. 0d35c78 Fix memory bot failure by hclam@chromium.org · 11 years ago
  31. 0f6f7cb Enqueue packet in pacer if sending fails by hclam@chromium.org · 11 years ago
  32. 44050b2 VCM: removing max jitter estimate by mikhal@webrtc.org · 11 years ago
  33. f47d0f8 Remove unneeded *_NOT_SUPPORTED from VoEAudioProcessing. by andrew@webrtc.org · 11 years ago
  34. 7533659 Fixes some pacer/padding issues found while testing. by stefan@webrtc.org · 11 years ago
  35. c9cb798 Use 3 threads for higher than 720p resolutions by fbarchard@google.com · 11 years ago
  36. 5b9adb0 Add a log message to see video delay break down by hclam@chromium.org · 11 years ago
  37. b82ee51 Make ScreenCapturerMac work in versions of OSX before Lion. by sergeyu@chromium.org · 11 years ago
  38. 9ca71b1 Enable ScreenCapturer unittests by sergeyu@chromium.org · 11 years ago
  39. ba458e2 Use intptr_t to represent window IDs on all platforms. by sergeyu@chromium.org · 11 years ago
  40. 69f7605 Wire up pacer-based padding. by stefan@webrtc.org · 11 years ago
  41. 67ca2b4 Revert r4145 "Revert 4127 "Switch frame list implementation to std::map."" by stefan@webrtc.org · 11 years ago
  42. 376ae3e Revert r4146 "Revert 4104 "Refactor jitter buffer to use separate lists for de..."" by stefan@webrtc.org · 11 years ago
  43. b06dd93 Fix AV sync issue by hclam@chromium.org · 11 years ago
  44. 9540e2a Log current and target AV delay in ViESyncModule by hclam@chromium.org · 11 years ago
  45. 8dbc8ab Merge more tests into modules_{unit,integration}tests. by kjellander@webrtc.org · 11 years ago
  46. 6828566 WebRTCDemo: ensures that using front and back camera work as expected. by henrike@webrtc.org · 11 years ago
  47. 19b339a Fixes linker issue with no op trace. by henrike@webrtc.org · 11 years ago
  48. 7b2c430 Risk of division by zero. by turaj@webrtc.org · 11 years ago
  49. cefb004 Revert 4211 "Build all java files into jar for each module on An..." by fischman@webrtc.org · 11 years ago
  50. 8ed5369 Disable ViEExtendedIntegrationTest.RunsCodecTestWithoutErrors test. by kjellander@webrtc.org · 11 years ago
  51. 83163e0 Disable ViEExtendedIntegrationTest.RunsCodecTestWithoutErrors test. by kjellander@webrtc.org · 11 years ago
  52. 789e98b Fix breakage due to test_fec conversion to gtest. by kjellander@webrtc.org · 11 years ago
  53. f04f54a Convert test_fec to gtest by kjellander@webrtc.org · 11 years ago
  54. f09f7b2 Disable ViEStandardIntegrationTest.RunsRtpRtcpTestWithoutErrors test. by kjellander@webrtc.org · 11 years ago
  55. 6ab2b1f G722_1/G722_1C codecs won't instantiate by tina.legrand@webrtc.org · 11 years ago
  56. 46cec2a Reorganize test targets in WebRTC by kjellander@webrtc.org · 11 years ago
  57. 0604490 Build all java files into jar for each module on Android by fischman@webrtc.org · 11 years ago
  58. e8c9ecd Allow the screen capturer to capture oversized cursors and cursors without alpha channel (Windows). by alexeypa@chromium.org · 11 years ago
  59. a463a4b Landing binary cursor image files to be used in a follow up CL. by alexeypa@chromium.org · 11 years ago
  60. f1bcae0 Updated WebRTC version to 3.33 by elham@webrtc.org · 11 years ago
  61. b4c89a4 Making no NACK mode work again in VideoEngine. by mflodman@webrtc.org · 11 years ago
  62. 63988b2 RW lock access to ssrc maps in VideoCall. by pbos@webrtc.org · 11 years ago
  63. 203d656 Add back the WEBRTC_DIRECT_TRACE flag. by solenberg@webrtc.org · 11 years ago
  64. 0a86a9f AudioDeviceAndroidOpenSLES: NULL variables might be referenced in StopPlayout() by braveyao@webrtc.org · 11 years ago
  65. 1125d89 Revert some variables to uint32_t to fix compile errors on Mac gcc. by andrew@webrtc.org · 11 years ago
  66. 64e2651 Allow audio devices with up to 64 channels on Mac. by andrew@webrtc.org · 11 years ago
  67. 0e7cd85 Fixed Rtp/Rtcp tests by pwestin@webrtc.org · 11 years ago
  68. cfe2a74 Fix relative path to .gitignore and other minor changes. by andrew@webrtc.org · 11 years ago
  69. a18c6e5 Removing functionality for inserting pre-encoded frames instead of raw by mflodman@webrtc.org · 11 years ago
  70. 029c3f4 Add script for appending entries to .gitignore. by andrew@webrtc.org · 11 years ago
  71. 9aeef32 Fix size_t to int conversion error on Win64. by andrew@webrtc.org · 11 years ago
  72. a653113 Remove fake screen capturer because it's not used anywhere. by sergeyu@chromium.org · 11 years ago
  73. 5f545ff Fix for STL vector function data not available. by pwestin@webrtc.org · 11 years ago
  74. 4aa9f1a Connect ACM with RTP module for audio NACK. by pwestin@webrtc.org · 11 years ago
  75. d0631e3 Nack for audio. by turaj@webrtc.org · 11 years ago
  76. 31fad0b Fix leaks in DesktopRegion by sergeyu@chromium.org · 11 years ago
  77. 69bab25 Implement DetectNumberOfCores on Android and make it consistent on Linux and Android by fischman@webrtc.org · 11 years ago
  78. b8171ff Wire up Nack for Voe by pwestin@webrtc.org · 11 years ago
  79. 6fb2ca3 Fix init list for VideoSendStream::Config::Rtp. by pbos@webrtc.org · 11 years ago
  80. 6f1c3ef Stats+Config moved into VideoSend/ReceiveStreams. by pbos@webrtc.org · 11 years ago
  81. 53304e8 Merge webrtc_utility_unittests into modules_unittests. by kjellander@webrtc.org · 11 years ago
  82. 266fc69 Restore relative include paths to libyuv. by andrew@webrtc.org · 11 years ago
  83. 9b82368 Issue 1847, memcopy is wrong and unnecessary, it is sufficient to store the pointer before clearing the instance, and write back the pointer. by turaj@webrtc.org · 11 years ago
  84. ae05178 resolve b9050210. Avoid pushing sync packet before any packet received. Do not turn on AV-sync if initial delay is zero. by turaj@webrtc.org · 11 years ago
  85. 6c82a7e Move screen capturers from chromium to webrtc. by sergeyu@chromium.org · 11 years ago
  86. 12bce3b Refactor padding and rtp header functionality. by stefan@webrtc.org · 11 years ago
  87. d8ecee5 Update the remote bitrate estimator before passing the packet to the RTP module. by stefan@webrtc.org · 11 years ago
  88. e54928f Remove XvRenderer. by pbos@webrtc.org · 11 years ago
  89. 751253d Fix build error introduced with r4168. by stefan@webrtc.org · 11 years ago
  90. 695ff2a Add support for padding in pacer. by stefan@webrtc.org · 11 years ago
  91. 026d1ce Include files from webrtc/.. paths in common_video/ by pbos@webrtc.org · 11 years ago
  92. cff5c03 Include files from webrtc/.. paths in tools/ by pbos@webrtc.org · 11 years ago
  93. 1cc4ed7 Disable neteq_unittests on Win x64 in code. by kjellander@webrtc.org · 11 years ago
  94. ec5caf3 Disable audio_decoder_unittests on Win x64 in code. by kjellander@webrtc.org · 11 years ago
  95. f4fc8ba Disable audio_coding_unittests on Win x64 in code. by kjellander@webrtc.org · 11 years ago
  96. 3b35ec6 Do not hold a lock when calling VCMReceiveCallback::FrameToRender. by fischman@webrtc.org · 11 years ago
  97. aec1bc8 Optimized DesktopRegion implementation. by sergeyu@chromium.org · 11 years ago
  98. ad9ee0d Removed unused class members to enable clang=1 android build. by fischman@webrtc.org · 11 years ago
  99. 0016110 Setting SSRC in vie_loopback_test by mikhal@webrtc.org · 11 years ago
  100. 915ca75 Fix error in mixing test for supported sample rates. by andrew@webrtc.org · 11 years ago