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webrtc / src / 11330b003ebe9ef7e6d3bd2bd0c79b14bdb3fa83
  1. 11330b0 Added myself to rtp module watch by pwestin@webrtc.org · 13 years ago
  2. d75889e Change of Android makefiles to build latest video coding code. by kma@webrtc.org · 13 years ago
  3. 7cf8937 by henrika@webrtc.org · 13 years ago
  4. cedbb03 [Issue 101] Solves memory leak on Windows by henrika@webrtc.org · 13 years ago
  5. 2ebc9ce Fix broken PeerConnection Dev build. by perkj@webrtc.org · 13 years ago
  6. c4d1983 Changes in rtp_format_vp8_unittest to match the changes in CL 774. by stefan@webrtc.org · 13 years ago
  7. f553ec7 Notifier and RefCount interface and implementation class name changed according to the naming convention. by mallinath@webrtc.org · 13 years ago
  8. ae499a2 Set correct codec info before sending frame to VCM. by mflodman@webrtc.org · 13 years ago
  9. 81f25f9 Fixing build errors on Windows platform. Minor changes... by kjellander@webrtc.org · 13 years ago
  10. f3f2f6a * Add include_internal_video_capture and include_internal_video_render to include/exclude the internal VCM and VRM. by wu@webrtc.org · 13 years ago
  11. 509c9c5 operator + is evaluated before ?: by henrike@webrtc.org · 13 years ago
  12. 4df8c9a by henrike@webrtc.org · 13 years ago
  13. 7ecdf58 Enable chromium_code:1 in the Chrome build. by andrew@webrtc.org · 13 years ago
  14. ffd28f9 Request key frames to battle error propagation. by stefan@webrtc.org · 13 years ago
  15. d0752c3 video_coding: Update to hybrid mode: Set FEC values for zero below a threshold. by mikhal@webrtc.org · 13 years ago
  16. c693bac Only start ViEPerformanceMonitor when needed. by mflodman@webrtc.org · 13 years ago
  17. b5475d0 vie_auto_test will now obey the Mac .mm rules for files including objective-c code. by phoglund@webrtc.org · 13 years ago
  18. 4c63676 Updated the AEC delay logging to output values in ms. PB output updated. by bjornv@webrtc.org · 13 years ago
  19. cc412c1 Remove second instance of ViE PerformanceMonitor. by mflodman@webrtc.org · 13 years ago
  20. ce8813d Using id instead of name when setting Mac/QTKit capture device. by mflodman@webrtc.org · 13 years ago
  21. 4d5d5c1 Reorganize the audio_processing source. by andrew@webrtc.org · 13 years ago
  22. 5d3bdf7 Fix clang warnings in ViE autotest. by andrew@webrtc.org · 13 years ago
  23. 8fd93d4 Move DeliverCapturedFrame from private to protected. by wu@webrtc.org · 13 years ago
  24. 1305a1d Fix rendering in new PeerConnection API. by perkj@webrtc.org · 13 years ago
  25. 52eddf73 Made Tina, Andrew and Jan as OWNERS to entire common_audio and removed the sub-OWNERS files. Let me know if that's fine. by bjornv@webrtc.org · 13 years ago
  26. 5b15cfc Fix BWE unit test build issue by stefan@webrtc.org · 13 years ago
  27. 61f07c3 I have made a small fix so it will execute properly from the default working directory location (trunk), finding its resource files. by kjellander@webrtc.org · 13 years ago
  28. 5dedd0e Handling of white-space in DataLog::Combine by henrik.lundin@webrtc.org · 13 years ago
  29. 929789b vie_auto_test - moved custom call specific functions to be static, added video protect method to custom call by amyfong@webrtc.org · 13 years ago
  30. 76aea65 When _audioConfigured, should not try to use the _video. by wu@webrtc.org · 13 years ago
  31. 0d55c8f Adding peerconnection_unittest. by henrike@webrtc.org · 13 years ago
  32. 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
  33. 63257d4 Implement proxy for both audio and video tracks. by perkj@webrtc.org · 13 years ago
  34. 3765bd2 Added AEC delay logging metrics to VoE. Echo metrics and delay logging metrics are enabled simultaneously through the SetEcMetricsStatus(). Updated standard and extended VoE tests. by bjornv@webrtc.org · 13 years ago
  35. f10ea31 Add IncomingFrameI420 to ViEExternalCapture interface to take captured video frame buffer as 3 planes. by wu@webrtc.org · 13 years ago
  36. 14aaaf1 Some re-organization of the fec-uep code: updated protection modes, comments, and some variable/function re-naming. by marpan@webrtc.org · 13 years ago
  37. 55c39f0 Add mallinath@webrtc.org and wu@webrtc.org as the capture owner for US office. by wu@webrtc.org · 13 years ago
  38. 58691eb Remove the DestroyDeviceInfo for mac video capture. (This is missed in r731.) by wu@webrtc.org · 13 years ago
  39. d0bdab0 Adding API to get sent total bitrate, FEC bitrate and NACK bitrate. by stefan@webrtc.org · 13 years ago
  40. e698eb7 Make the sanity check test a little more robust, and add a README file. by hta@webrtc.org · 13 years ago
  41. 26c0416 Added more tests, fixed a bug and refactored. by phoglund@webrtc.org · 13 years ago
  42. 2111d3b Removed the vad_const files and added the constants to the files where they are by bjornv@webrtc.org · 13 years ago
  43. a59d80d Updated fixed point output data file after changes in nsx. Verified bitexactness before that CL and the CLs afterwards towards the new file. by bjornv@webrtc.org · 13 years ago
  44. c01c358 session/phone/channel.cc updates after new push of libjingle revision. by mallinath@webrtc.org · 13 years ago
  45. 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
  46. 03a8699 Fixes for build errors introduced most likely earlier today. by henrike@webrtc.org · 13 years ago
  47. 0c37811 Define NO_SOUND_SYSTEM for chromium build. by wu@webrtc.org · 13 years ago
  48. ebc405d Remove the fakeportallocator from the libjingle.gyp. by wu@webrtc.org · 13 years ago
  49. 4ee906d When WEBRTC_VIDEO_ENGINE_FILE_API is not defined, disable the code in vie_file_impl.cc and vie_file_image.cc so that we can remove the libjpeg dependency. Also disable the auto test for the vie file api. by wu@webrtc.org · 13 years ago
  50. 5a3e20f Removed unused variables (build error) for test_fec. by marpan@webrtc.org · 13 years ago
  51. 6c2d710 * Update to use the new libjingle release. by wu@webrtc.org · 13 years ago
  52. 1da1ce0 First implementation of simulcast, adds VP8 simulcast to video engine. by pwestin@webrtc.org · 13 years ago
  53. 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
  54. 7951e81 Simple utility method for finding the project root dir (to be used by tests loading resource files) by kjellander@webrtc.org · 13 years ago
  55. 6a34d58 Implement MediaStreamProxy. by perkj@webrtc.org · 13 years ago
  56. 4c059d8 Add metric for number of packets discarded by JB due to not being decodable by stefan@webrtc.org · 13 years ago
  57. 77d7d54 Replace the DestroyDeviceInfo with a virtual destructor. by wu@webrtc.org · 13 years ago
  58. 38e400a Adding native client test page to test loopback. by perkj@webrtc.org · 13 years ago
  59. e5542a0 Add file record and play functions to voe_cmd_test, fix Play local file (path was incorrect) by amyfong@webrtc.org · 13 years ago
  60. 6330cf2 Fixed ViE AutoTest trace file names to be consistent by amyfong@webrtc.org · 13 years ago
  61. ea89922 Add VideoCaptureFactory so that we don't need to expose VideoCaptureImpl. by wu@webrtc.org · 13 years ago
  62. 199f4de Rename all .cc files which include Objective-C headers to .mm. by andrew@webrtc.org · 13 years ago
  63. a0258de Fixes test build errors (warnings treated as errors) in system_wrappers. by henrike@webrtc.org · 13 years ago
  64. 26c9ff9 Add dummy implementation of DataLog::Combine method by henrik.lundin@webrtc.org · 13 years ago
  65. 791eec7 Add API to get the number of packets discarded by the video jitter buffer due to being too late. by stefan@webrtc.org · 13 years ago
  66. 06887ae Fixes two bugs when decoding with packet losses. by stefan@webrtc.org · 13 years ago
  67. ed081a9 Print info about the local and remote resolution in the Windows client. by tommi@webrtc.org · 13 years ago
  68. 73ba416 Fix OnClose(socket, NO_ERROR) compile error on Linux. by perkj@webrtc.org · 13 years ago
  69. 1843664 DataLog: Changing from common_types to typedefs by henrik.lundin@webrtc.org · 13 years ago
  70. f7b36a4 Fix bug in the server where a wait request was incorrectly handled. by tommi@webrtc.org · 13 years ago
  71. c0b2250 Fix the Windows build. Review URL: http://webrtc-codereview.appspot.com/213004 by tommi@webrtc.org · 13 years ago
  72. 5a695d6 Fix bug in the client that caused signaling messages to be dropped. by tommi@webrtc.org · 13 years ago
  73. d855bd4 C wrapper for DataLog class by henrik.lundin@webrtc.org · 13 years ago
  74. 6364d12 Fix a couple of build warnings. by tommi@webrtc.org · 13 years ago
  75. e95458c Started rewriting video_engine tests to use GUnit. by phoglund@webrtc.org · 13 years ago
  76. c8c4deb Fix Windows build. %zu isn't supported in the crt implementation by tommi@webrtc.org · 13 years ago
  77. 5a945ec A little upgrade to the HTML test page: by tommi@webrtc.org · 13 years ago
  78. 25e0b8e Python output flag and keyframe interval flags. by kjellander@webrtc.org · 13 years ago
  79. a31b254 Python output flag and keyframe interval flags. by kjellander@webrtc.org · 13 years ago
  80. 80dd19b vplib tests: Removing old and unused file and directories. by mikhal@webrtc.org · 13 years ago
  81. f6ab63c Update PeerConnection_client to open a video capture device. by perkj@webrtc.org · 13 years ago
  82. bf54ef9 Removed code under a non-existing define. by henrike@webrtc.org · 13 years ago
  83. 1a2933c Fixes a Valgrind warning triggering when the number of pending messages hit the limit. by henrike@webrtc.org · 13 years ago
  84. 2915f6f Use proper printf size_t specifier to fix Linux 32-bit build. by andrew@webrtc.org · 13 years ago
  85. b2d4921 Remove trailing whitespace in AudioDevice. by andrew@webrtc.org · 13 years ago
  86. d6132f5 by mikhal@webrtc.org · 13 years ago
  87. 3a6d4f4 Fix setting VideoCaptureModule and VideoRenderer for local and remote streams. by perkj@webrtc.org · 13 years ago
  88. 35a1756 First version of video quality measurement program and test framework. by kjellander@webrtc.org · 13 years ago
  89. 3ce62fc Move merge_libs targets to their own gyp. by andrew@webrtc.org · 13 years ago
  90. af57de00 Some code style changes in audio_processing/ns/main/source/ by Astyle, by kma@webrtc.org · 13 years ago
  91. fa41d80 Fixes session state transition and registering observer. by mallinath@webrtc.org · 13 years ago
  92. 01ca01f Adding neteq_tests to modules tests by henrik.lundin@webrtc.org · 13 years ago
  93. 29787c7 Changes to WebRtcSession after Provider(s) interface addition. by mallinath@webrtc.org · 13 years ago
  94. bbc1f10 Changed modules/audio_processing/utility/Android.mk, to correct a build error in by kma@webrtc.org · 13 years ago
  95. 487e401 Moving creation of sessiondescriptions to webrtcsession. by perkj@webrtc.org · 13 years ago
  96. bf39ff4 Some general optimization in NS. by kma@webrtc.org · 13 years ago
  97. a58224f Introduced a SPL inline function (multiple-accumulate), for preformance in ARMv7. by kma@webrtc.org · 13 years ago
  98. cb4ab65 Moved creation of objects to the signaling thread. by perkj@webrtc.org · 13 years ago
  99. bafca10 Temp hook in WebRtcSession to VideoChannel. by mallinath@webrtc.org · 13 years ago
  100. 4b6f747 Fixes a newly introduced bug in the jitter buffer where buffer reallocation by stefan@webrtc.org · 13 years ago