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gerrit.openfyde.cn / webrtc.googlesource.com / src / fdde8b3fb7d8e5b7509a5f81c60af6bc09d1e994
  1. fdde8b3 Add references to src/ copies for LICENSE etc. by andrew@webrtc.org · 14 years ago
  2. cb18121 Add an unpacker tool for audioproc debug files. by andrew@webrtc.org · 14 years ago
  3. fc9bcef Data alignment fix for SSIM. by frkoenig@google.com · 14 years ago
  4. 78c767f Rewrote codec test to use fake camera. by phoglund@webrtc.org · 14 years ago
  5. d855c1a Reverts r807 and fixes the real issue in the VCM. by stefan@webrtc.org · 14 years ago
  6. bdb55c8 This CL is an attempt to remove a crash we can see when closing down VoiceEgine. by henrika@webrtc.org · 14 years ago
  7. a6c2335 Solves crash in ADM API unit test for Core Audio on Windows by henrika@webrtc.org · 14 years ago
  8. 5423bc2 Adds correct absolute paths to all input files in ADM functional unit tests. by henrika@webrtc.org · 14 years ago
  9. 5b5c31d Update fixed point audio processing output. by andrew@webrtc.org · 14 years ago
  10. ca325ec Corrected a linux build error introduced in issue 246005. by kma@webrtc.org · 14 years ago
  11. f0cd394 Put fwrite calls under corresponding macros since they shouldn't show up in release build. by wjia@webrtc.org · 14 years ago
  12. f31826e adding a wait on the decode thread when no frames are available by mikhal@webrtc.org · 14 years ago
  13. a412924 VP8:Setting number of cores based on image size by mikhal@webrtc.org · 14 years ago
  14. 913644b For commiting changes in CL 277002, due to file structure changes introduced during by kma@webrtc.org · 14 years ago
  15. 0d0037c Return cached data instead of sleeping in CpuWrapperMac (shaves 2s off WebrtcMediaEngine creation time on Mac). by henrike@webrtc.org · 14 years ago
  16. 0a9c318 The fread result is no longer ignored. by phoglund@webrtc.org · 14 years ago
  17. 537096a Remove unnecessary objective-c compiler flags. by andrew@webrtc.org · 14 years ago
  18. c63f788 Added fake camera, rewrote one test to use it. by phoglund@webrtc.org · 14 years ago
  19. bf478fa Ensures that ADM unit tests builds on all platforms. by henrika@webrtc.org · 14 years ago
  20. 58b4f1c Fixes broken build in peerconnection unit tests by mallinath@webrtc.org · 14 years ago
  21. 4e4c939 Upgrade libvpx to e529a825 by mikhal@webrtc.org · 14 years ago
  22. f1a605c Update DEPS to support Mac clang build. by andrew@webrtc.org · 14 years ago
  23. 5eb64f0 Fix BitrateSent() API when having a default RTP module. by stefan@webrtc.org · 14 years ago
  24. 158f496 Fixes a rate control bug in the VP8 wrapper. by stefan@webrtc.org · 14 years ago
  25. aa32319 Implement unittest for proxies of MediaStreamTrackInterface and MediaStreamInterface. by perkj@webrtc.org · 14 years ago
  26. ca8b3a3 kind() method in track interface is changed to std::string to keep uniformity with other get methods by mallinath@webrtc.org · 14 years ago
  27. 96ba190 ref_count.h file name changed to refcount.h to keep as other ( most ) files are named in libjingle. by mallinath@webrtc.org · 14 years ago
  28. ead87b5 Fix potential issue where frame buffers might be freed while being decoded. by stefan@webrtc.org · 14 years ago
  29. 2b0f094 Avoid reallocating the decodedImage for every decoded frame. by stefan@webrtc.org · 14 years ago
  30. ee3dfa6 by mikhal@webrtc.org · 14 years ago
  31. 1af915d video_coding: vp8: Updating error propagation threshold by mikhal@webrtc.org · 14 years ago
  32. 11330b0 Added myself to rtp module watch by pwestin@webrtc.org · 14 years ago
  33. d75889e Change of Android makefiles to build latest video coding code. by kma@webrtc.org · 14 years ago
  34. 7cf8937 by henrika@webrtc.org · 14 years ago
  35. cedbb03 [Issue 101] Solves memory leak on Windows by henrika@webrtc.org · 14 years ago
  36. 2ebc9ce Fix broken PeerConnection Dev build. by perkj@webrtc.org · 14 years ago
  37. c4d1983 Changes in rtp_format_vp8_unittest to match the changes in CL 774. by stefan@webrtc.org · 14 years ago
  38. f553ec7 Notifier and RefCount interface and implementation class name changed according to the naming convention. by mallinath@webrtc.org · 14 years ago
  39. ae499a2 Set correct codec info before sending frame to VCM. by mflodman@webrtc.org · 14 years ago
  40. 81f25f9 Fixing build errors on Windows platform. Minor changes... by kjellander@webrtc.org · 14 years ago
  41. f3f2f6a * Add include_internal_video_capture and include_internal_video_render to include/exclude the internal VCM and VRM. by wu@webrtc.org · 14 years ago
  42. 509c9c5 operator + is evaluated before ?: by henrike@webrtc.org · 14 years ago
  43. 4df8c9a by henrike@webrtc.org · 14 years ago
  44. 7ecdf58 Enable chromium_code:1 in the Chrome build. by andrew@webrtc.org · 14 years ago
  45. ffd28f9 Request key frames to battle error propagation. by stefan@webrtc.org · 14 years ago
  46. d0752c3 video_coding: Update to hybrid mode: Set FEC values for zero below a threshold. by mikhal@webrtc.org · 14 years ago
  47. c693bac Only start ViEPerformanceMonitor when needed. by mflodman@webrtc.org · 14 years ago
  48. b5475d0 vie_auto_test will now obey the Mac .mm rules for files including objective-c code. by phoglund@webrtc.org · 14 years ago
  49. 4c63676 Updated the AEC delay logging to output values in ms. PB output updated. by bjornv@webrtc.org · 14 years ago
  50. cc412c1 Remove second instance of ViE PerformanceMonitor. by mflodman@webrtc.org · 14 years ago
  51. ce8813d Using id instead of name when setting Mac/QTKit capture device. by mflodman@webrtc.org · 14 years ago
  52. 4d5d5c1 Reorganize the audio_processing source. by andrew@webrtc.org · 14 years ago
  53. 5d3bdf7 Fix clang warnings in ViE autotest. by andrew@webrtc.org · 14 years ago
  54. 8fd93d4 Move DeliverCapturedFrame from private to protected. by wu@webrtc.org · 14 years ago
  55. 1305a1d Fix rendering in new PeerConnection API. by perkj@webrtc.org · 14 years ago
  56. 52eddf7 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 · 14 years ago
  57. 5b15cfc Fix BWE unit test build issue by stefan@webrtc.org · 14 years ago
  58. 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 · 14 years ago
  59. 5dedd0e Handling of white-space in DataLog::Combine by henrik.lundin@webrtc.org · 14 years ago
  60. 929789b vie_auto_test - moved custom call specific functions to be static, added video protect method to custom call by amyfong@webrtc.org · 14 years ago
  61. 76aea65 When _audioConfigured, should not try to use the _video. by wu@webrtc.org · 14 years ago
  62. 0d55c8f Adding peerconnection_unittest. by henrike@webrtc.org · 14 years ago
  63. 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 · 14 years ago
  64. 63257d4 Implement proxy for both audio and video tracks. by perkj@webrtc.org · 14 years ago
  65. 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 · 14 years ago
  66. f10ea31 Add IncomingFrameI420 to ViEExternalCapture interface to take captured video frame buffer as 3 planes. by wu@webrtc.org · 14 years ago
  67. 14aaaf1 Some re-organization of the fec-uep code: updated protection modes, comments, and some variable/function re-naming. by marpan@webrtc.org · 14 years ago
  68. 55c39f0 Add mallinath@webrtc.org and wu@webrtc.org as the capture owner for US office. by wu@webrtc.org · 14 years ago
  69. 58691eb Remove the DestroyDeviceInfo for mac video capture. (This is missed in r731.) by wu@webrtc.org · 14 years ago
  70. d0bdab0 Adding API to get sent total bitrate, FEC bitrate and NACK bitrate. by stefan@webrtc.org · 14 years ago
  71. e698eb7 Make the sanity check test a little more robust, and add a README file. by hta@webrtc.org · 14 years ago
  72. 26c0416 Added more tests, fixed a bug and refactored. by phoglund@webrtc.org · 14 years ago
  73. 2111d3b Removed the vad_const files and added the constants to the files where they are by bjornv@webrtc.org · 14 years ago
  74. 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 · 14 years ago
  75. c01c358 session/phone/channel.cc updates after new push of libjingle revision. by mallinath@webrtc.org · 14 years ago
  76. 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 · 14 years ago
  77. 03a8699 Fixes for build errors introduced most likely earlier today. by henrike@webrtc.org · 14 years ago
  78. 0c37811 Define NO_SOUND_SYSTEM for chromium build. by wu@webrtc.org · 14 years ago
  79. ebc405d Remove the fakeportallocator from the libjingle.gyp. by wu@webrtc.org · 14 years ago
  80. 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 · 14 years ago
  81. 5a3e20f Removed unused variables (build error) for test_fec. by marpan@webrtc.org · 14 years ago
  82. 6c2d710 * Update to use the new libjingle release. by wu@webrtc.org · 14 years ago
  83. 1da1ce0 First implementation of simulcast, adds VP8 simulcast to video engine. by pwestin@webrtc.org · 14 years ago
  84. 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 · 14 years ago
  85. 7951e81 Simple utility method for finding the project root dir (to be used by tests loading resource files) by kjellander@webrtc.org · 14 years ago
  86. 6a34d58 Implement MediaStreamProxy. by perkj@webrtc.org · 14 years ago
  87. 4c059d8 Add metric for number of packets discarded by JB due to not being decodable by stefan@webrtc.org · 14 years ago
  88. 77d7d54 Replace the DestroyDeviceInfo with a virtual destructor. by wu@webrtc.org · 14 years ago
  89. 38e400a Adding native client test page to test loopback. by perkj@webrtc.org · 14 years ago
  90. e5542a0 Add file record and play functions to voe_cmd_test, fix Play local file (path was incorrect) by amyfong@webrtc.org · 14 years ago
  91. 6330cf2 Fixed ViE AutoTest trace file names to be consistent by amyfong@webrtc.org · 14 years ago
  92. ea89922 Add VideoCaptureFactory so that we don't need to expose VideoCaptureImpl. by wu@webrtc.org · 14 years ago
  93. 199f4de Rename all .cc files which include Objective-C headers to .mm. by andrew@webrtc.org · 14 years ago
  94. a0258de Fixes test build errors (warnings treated as errors) in system_wrappers. by henrike@webrtc.org · 14 years ago
  95. 26c9ff9 Add dummy implementation of DataLog::Combine method by henrik.lundin@webrtc.org · 14 years ago
  96. 791eec7 Add API to get the number of packets discarded by the video jitter buffer due to being too late. by stefan@webrtc.org · 14 years ago
  97. 06887ae Fixes two bugs when decoding with packet losses. by stefan@webrtc.org · 14 years ago
  98. ed081a9 Print info about the local and remote resolution in the Windows client. by tommi@webrtc.org · 14 years ago
  99. 73ba416 Fix OnClose(socket, NO_ERROR) compile error on Linux. by perkj@webrtc.org · 14 years ago
  100. 1843664 DataLog: Changing from common_types to typedefs by henrik.lundin@webrtc.org · 14 years ago