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gerrit-public.fairphone.software / platform / external / webrtc / 72e3a89b5207519e0df3069f93942b854364ee7a
  1. 72e3a89 Created a wrapper class for condition_variable that lets me write (hopefully) reliable tests for some of its properties. by hta@webrtc.org · 13 years ago
  2. b38fca1 VAD Refactoring: API change of return value from int16_t to int. by bjornv@webrtc.org · 13 years ago
  3. f477aac Removed gflags header from vie_auto_test. by vspasova@webrtc.org · 13 years ago
  4. dfa6b69 Refine the error handling made in rev2373 by braveyao@webrtc.org · 13 years ago
  5. 67f256f Use 32 as the alignment if possible in VP8 wrapper. by wu@webrtc.org · 13 years ago
  6. df596ae VAD Refactoring of GMM test section by bjornv@webrtc.org · 13 years ago
  7. 50d5ca5 Refactoring of TestAllCodecs by tina.legrand@webrtc.org · 13 years ago
  8. db2f6cf Added missing define guard to sleep.h by hta@webrtc.org · 13 years ago
  9. 86a6aac Unittest utilities - starting out with an encapsulated trace-to-screen. by hta@webrtc.org · 13 years ago
  10. e3a0712 Deregister RTP module before deleting it. by mflodman@webrtc.org · 13 years ago
  11. 41adcdb An OS-independent sleep function, and one usage thereof. by hta@webrtc.org · 13 years ago
  12. 3719800 GetRecPayloadType now logs a warning instead of and error when the user asks for the payload type while no packets have been received. by henrika@webrtc.org · 13 years ago
  13. 1905415 Correct gypi files to match the actual filenames. by stefan@webrtc.org · 13 years ago
  14. d63d06a bump version to 3.8 Review URL: https://webrtc-codereview.appspot.com/657004 by niklas.enbom@webrtc.org · 13 years ago
  15. 4de777b Refine the error processing of StopRecordingMicrophone. by braveyao@webrtc.org · 13 years ago
  16. bdf7ee5 This simple change should adress issue 471. by turaj@webrtc.org · 13 years ago
  17. 352d09a Updates to videoprocessor_integration test: by marpan@webrtc.org · 13 years ago
  18. f088448 Libyuv Scalerunittest: Added PSNR check to some tests in scaler unittest: by marpan@webrtc.org · 13 years ago
  19. 139c467 Fixed a/v sync issue. by mflodman@webrtc.org · 13 years ago
  20. 46d83fa Use digital mode on mobile by leozwang@webrtc.org · 13 years ago
  21. c35f1d2 FEC: Fix to coverity issue 14448: unintended sign extension. by marpan@webrtc.org · 13 years ago
  22. f0d4696 Add support for SSE intrinsics on gcc in libvpx. by stefan@webrtc.org · 13 years ago
  23. d418514 Bumped version number to 3.7. by mflodman@webrtc.org · 13 years ago
  24. b1c3276 VAD Refactoring: WebRtcVad_Process() by bjornv@webrtc.org · 13 years ago
  25. 5f9f1db This change make PulseAudio only start for the tests on the LinuxLargeTests bot. by kjellander@webrtc.org · 13 years ago
  26. 5e7ca60 Use new fileutil functions for trace in ACM by tina.legrand@webrtc.org · 13 years ago
  27. 1c28473 Fix master's "Start PulseAudio" step. by andrew@webrtc.org · 13 years ago
  28. 0594916 Add audio_e2e_test to LinuxLargeTests. by andrew@webrtc.org · 13 years ago
  29. 9f6577b Restore default source in e2e test. by andrew@webrtc.org · 13 years ago
  30. 6724c42 Add VoiceEngine apm settings to test application by leozwang@webrtc.org · 13 years ago
  31. be58164 Add a variable for the libjpeg include directory. by andrew@webrtc.org · 13 years ago
  32. f08f52f Fixing issues with slaves.cfg on Windows. by kjellander@webrtc.org · 13 years ago
  33. eec739f VAD Refactoring: Changed pointer structure in WebRtcVad_FindMinimum(). by bjornv@webrtc.org · 13 years ago
  34. 78a3110 Disable multi_res_encoding in libvpx. by marpan@webrtc.org · 13 years ago
  35. fa7138f Change CriticalSectionScoped to use pointer constructor by tina.legrand@webrtc.org · 13 years ago
  36. 276dc18 Add libremote_bitrate_estimator to android makefile by leozwang@webrtc.org · 13 years ago
  37. f85b35a Refactored Neon code for AECM module, by using pure assembly code. by kma@webrtc.org · 13 years ago
  38. 38506ef Disable cpu detection on arm-neon by leozwang@webrtc.org · 13 years ago
  39. d81ab13 abs() was used instead of fabsf(), which returns int and not float and therefore truncated the return value. by stefan@webrtc.org · 13 years ago
  40. f7d0c77 Added the bitrate estimator test to the trybots. by phoglund@webrtc.org · 13 years ago
  41. 90af7f8 Changing Celt to run on 20 msec frames by tina.legrand@webrtc.org · 13 years ago
  42. d2956d8 Renamed test_bwe. by phoglund@webrtc.org · 13 years ago
  43. 9354cc9 Refactoring the receive-side bandwidth estimation into its own module. by stefan@webrtc.org · 13 years ago
  44. f4c6aa2 Improve the reliablity of the audio e2e test. by andrew@webrtc.org · 13 years ago
  45. b0bcf13 Trival fix to relative paths of audio files in voe_ui_win_test by braveyao@webrtc.org · 13 years ago
  46. 5f97232 Removing a TODO in the FEC: renaming the exisiting packets mask to indicate random mode, by marpan@webrtc.org · 13 years ago
  47. cac603f Fix for the alignment problems/mismatch in ViECapture and VP8Encoder. by wu@webrtc.org · 13 years ago
  48. f4c2de9 Added some tests to videoprocessor_integrationtest, for testing: by marpan@webrtc.org · 13 years ago
  49. 8866bb1 FEC: Added another set of packet masks for the FEC. by marpan@webrtc.org · 13 years ago
  50. 20e13ed New attempt to revert r2362, since drover failed. by bjornv@webrtc.org · 13 years ago
  51. cb89c6f Revert 2363 - Refactoring the receive-side bandwidth estimation into its own module. by bjornv@webrtc.org · 13 years ago
  52. df37398 Renamed test_bwe. by phoglund@webrtc.org · 13 years ago
  53. f728814 Refactoring the receive-side bandwidth estimation into its own module. by stefan@webrtc.org · 13 years ago
  54. d2acea6 Minor style changes by bjornv@webrtc.org · 13 years ago
  55. 3007b26 Roll Chromium 134666:140240. by andrew@webrtc.org · 13 years ago
  56. da7fdf4 Fix to scaler in libyuv for odd size frames. by marpan@webrtc.org · 13 years ago
  57. ba108ae This CL contains some refactoring. Spectrum coding is main place that is affected. Therefore, I have bit-exactness test, test_spectrum_ by turaj@webrtc.org · 13 years ago
  58. 2cc5509 Fix syntax error in jpeg.gypi. by andrew@webrtc.org · 13 years ago
  59. ad6083f Added condition for which jpeg lib to use. by mflodman@webrtc.org · 13 years ago
  60. 77fd39a ACM PCM16B, fixing a copy-and-paste error. by tina.legrand@webrtc.org · 13 years ago
  61. e6f235c Attempt to fix broken encoding. by phoglund@webrtc.org · 13 years ago
  62. 9cf4d72 by niklas.enbom@webrtc.org · 13 years ago
  63. 82bf033 by niklas.enbom@webrtc.org · 13 years ago
  64. 265e38c Fixing test gypi for bit rate controller by niklas.enbom@webrtc.org · 13 years ago
  65. f1d6e0a Removed the obsolete sanity check and added new test HTML files. by phoglund@webrtc.org · 13 years ago
  66. ab12990 In the past we support calling StartPlayingFileLocally() before StartPlayout(), then when playout is started, the file would be played out immediately. by braveyao@webrtc.org · 13 years ago
  67. 899baa8 Temporarily disable first partition packet counting to avoid a bug in ProducerFec which doesn't properly handle important packets. by marpan@webrtc.org · 13 years ago
  68. 354b0ed Check return result of fwrite [Audio Module] by leozwang@webrtc.org · 13 years ago
  69. c3b2683 Refactored the pitch filter function in iSAC-fix. One important purpose is to prepare the function for assembly optimization in ARM platforms. by kma@webrtc.org · 13 years ago
  70. 5b4f36d ACM: Too short char vector by tina.legrand@webrtc.org · 13 years ago
  71. 343301f Fixing release compilation on Linux and Mac trybots by kjellander@webrtc.org · 13 years ago
  72. c03df17 Enabling audio_coding_module_test on trybots by kjellander@webrtc.org · 13 years ago
  73. 4517585 Adding separate payload types for stereo modes by tina.legrand@webrtc.org · 13 years ago
  74. c2722a0 Fixed compiler warning Review URL: https://webrtc-codereview.appspot.com/624005 by pwestin@webrtc.org · 13 years ago
  75. 29c5a23 Renamed to Network Emulator and improved error handling. by kjellander@webrtc.org · 13 years ago
  76. f5d934d Upgrade libvpx to cab6ac16 (v. 1.1.1 pre-point-release). by stefan@webrtc.org · 13 years ago
  77. 7d8c567 Ignore return value of fwrites. by andrew@webrtc.org · 13 years ago
  78. 595749f Network simulation script based on Dummynet. by kjellander@webrtc.org · 13 years ago
  79. ad0f05b Remove empty directories. by andrew@webrtc.org · 13 years ago
  80. 2e84c11 Updating bitrate controller tests to test naming conventions. by kjellander@webrtc.org · 13 years ago
  81. baaf243 Extracted a method for sending padded data. by phoglund@webrtc.org · 13 years ago
  82. bb24b14 Libvpx waterfall additional changes. The CL http://review.webrtc.org/595005/ was not complete since it didn't put the libvpx waterfall at its own port. by kjellander@webrtc.org · 13 years ago
  83. 7d3b07a Update to chromium r139469. by wu@webrtc.org · 13 years ago
  84. 36ccce4 Remove documentation folders. by andrew@webrtc.org · 13 years ago
  85. 16fcb24 Disable flaky VolumeTests only on Linux. by andrew@webrtc.org · 13 years ago
  86. e7e64e3 Fix compilation errors on ChromeOS by leozwang@webrtc.org · 13 years ago
  87. 0cb79cc Fixing gyp bug in https://webrtc-codereview.appspot.com/599006 by niklas.enbom@webrtc.org · 13 years ago
  88. dc257b5 Add option to configure error concealment and disable by default. by stefan@webrtc.org · 13 years ago
  89. 327ada1 Refactored IncomingVideoStream and VideoRenderFrame, to get code in better shape when hunting BUG=481. by mflodman@webrtc.org · 13 years ago
  90. 9259e7b Added a step for restarting pulseaudio. by phoglund@webrtc.org · 13 years ago
  91. 281b798 Resolved Coverity warnings. by bjornv@webrtc.org · 13 years ago
  92. b5ea03a Add print out stats summary to integrationtest.cc by leozwang@webrtc.org · 13 years ago
  93. 459955f Move audio_frame_operations to the utility module. by andrew@webrtc.org · 13 years ago
  94. aafa49b Disable flaky VolumeTest.DefaultSpeakerVolumeIsAtMost255. by andrew@webrtc.org · 13 years ago
  95. 5f23d64 Set the stream delay to zero if too low. by andrew@webrtc.org · 13 years ago
  96. 2fc6e38 Check return value of fwrite. [Video Module] by leozwang@webrtc.org · 13 years ago
  97. 8a7a019 Syncing tests on try master with build master. by kjellander@webrtc.org · 13 years ago
  98. 1eef9c1 Bitrate bugfixes Review URL: https://webrtc-codereview.appspot.com/609005 by pwestin@webrtc.org · 13 years ago
  99. 5abab0b Revert 2311 - Disable error concealment. by stefan@webrtc.org · 13 years ago
  100. 3348b29 Disable error concealment. by stefan@webrtc.org · 13 years ago