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gerrit-public.fairphone.software / platform / external / webrtc / 6d26ef76ea778d65e2f37ce5179b884cbed0a881 / src
  1. 6d26ef7 Refactored ViESender. by mflodman@webrtc.org · 13 years ago
  2. d492f72 Added empty unit tests to get code coverage measured. by kjellander@webrtc.org · 13 years ago
  3. 55d81ea ViE Custom Call observer now using pointers, fixed protection method and miscellaneous TODO cleanup by amyfong@webrtc.org · 13 years ago
  4. ba028a3 Fix sample rate printout in process_test. by andrew@webrtc.org · 13 years ago
  5. f3d10d3 Fixed release compilation error-warnings. by phoglund@webrtc.org · 13 years ago
  6. c4c56ed Rewrote vie_auto_test to use googletest macros. by phoglund@webrtc.org · 13 years ago
  7. 48b68c0 Added support for 96 kHz sampling frequency. by bjornv@webrtc.org · 13 years ago
  8. 4257790 NetEQ-related bug in ACM by henrik.lundin@webrtc.org · 13 years ago
  9. 543c3ea Fixing Release compilation errors by kjellander@webrtc.org · 13 years ago
  10. 89ab652 Cleaning up NetEQ statistics by henrik.lundin@webrtc.org · 13 years ago
  11. df10de4 Removing statistics API from NetEQ by henrik.lundin@webrtc.org · 13 years ago
  12. 7d3e949 This CL is to support certain audio devices which don't offer volume control. Try to be more compatible to those rare cases. by braveyao@webrtc.org · 13 years ago
  13. 2b838b4 video_coding: updating the session info unit test following recent changes by mikhal@webrtc.org · 13 years ago
  14. 425b377 video_coding: Updating internal_defines to resolve latest build error. Refers to JB flush update. by mikhal@webrtc.org · 13 years ago
  15. f13388f video_coding: Requesting a key frame after a JB flush by mikhal@webrtc.org · 13 years ago
  16. 6b9a7f8 video_coding: Allowing for a decodable state independent of selective nacking by mikhal@webrtc.org · 13 years ago
  17. 828af1b Add lookahead to the delay estimator. by andrew@webrtc.org · 13 years ago
  18. 5a52939 Make DMO init safe when not supported. by andrew@webrtc.org · 13 years ago
  19. dfe89e3 Move ViE main/test/AutoTest to test/auto_test. by mflodman@webrtc.org · 13 years ago
  20. 8594f76 Add a gyp variable for AEC debug dumps. by andrew@webrtc.org · 13 years ago
  21. a249f35 Correct several makefile errors for Android build. by kma@webrtc.org · 13 years ago
  22. 6830bdd Fix xcode build. by mflodman@webrtc.org · 13 years ago
  23. 94ea32e Move video_engine/source* to video_engine/. No code changes except paths in gyp-files. by mflodman@webrtc.org · 13 years ago
  24. 274c2ef Adding empty test method required to get code coverage by kjellander@webrtc.org · 13 years ago
  25. 3caa327 VP8 wrapper: Turn on some mild amount of deblocking in post-processing. by marpan@webrtc.org · 13 years ago
  26. ce9d89d Fixes linux build error introduced in r980. by henrike@webrtc.org · 13 years ago
  27. ad98a3e Fixes TEST crash triggered by webrtc-codereview.appspot.com/268014. by henrike@webrtc.org · 13 years ago
  28. 31d3070 Addressed review comments from http://webrtc-codereview.appspot.com/256004/ by henrike@webrtc.org · 13 years ago
  29. ced1186 Changed keyword __restrict__ to __restrict. by kma@webrtc.org · 13 years ago
  30. 3798ecb Made CPU initialization on Windows lazy to prevent long startup time. by henrike@webrtc.org · 13 years ago
  31. 543611a Reverting r972 due to compilation error on Windows Release build. by kjellander@webrtc.org · 13 years ago
  32. 2f047cc Removed unnecessary variable to avoid compiler error on Win. by bjornv@webrtc.org · 13 years ago
  33. ba74924 Remove use of exceptions in NetEQ test code by henrik.lundin@webrtc.org · 13 years ago
  34. 6a9835d Delay estimator structural changes. by bjornv@webrtc.org · 13 years ago
  35. fa9b016 Optimized WebRtcIsacfix_AutocorrFix() function for iSAC fix. by kma@webrtc.org · 13 years ago
  36. f556b9d This modification is supposed to fix the webrtc issue 144/145. With this fix, people could set/get mic volume before StartSend(). by braveyao@webrtc.org · 13 years ago
  37. 917fa6b ViE Custom Call added SetImageScaleStatus toggle option and other changes. by amyfong@webrtc.org · 13 years ago
  38. cd7b57e Fixing release compilation error by kjellander@webrtc.org · 13 years ago
  39. 3f1cb8e Restructuring and adding dummy unit test target. by kjellander@webrtc.org · 13 years ago
  40. cc2ecb3 Restructuring and adding dummy unit test target. by kjellander@webrtc.org · 13 years ago
  41. b72268e Restructuring and adding dummy unit test target. by kjellander@webrtc.org · 13 years ago
  42. 64a897a Restructuring and adding dummy unit test target. by kjellander@webrtc.org · 13 years ago
  43. 8f89f09 Note: this patch may seem intimidating but it mostly moves code around and renames things. There are quite few actual changes. by phoglund@webrtc.org · 13 years ago
  44. c05b56a Fixing compilation error by kjellander@webrtc.org · 13 years ago
  45. 0403ef4 Restructuring and adding unit test targets on project level instead of in common_audio. by kjellander@webrtc.org · 13 years ago
  46. 337dc68 Included modules in webrtc.gyp and fixed build errors. by phoglund@webrtc.org · 13 years ago
  47. af26f64 Inband DTMF stereo support by niklas.enbom@webrtc.org · 13 years ago
  48. e33a102 Resubmitting http://webrtc-codereview.appspot.com/269007/ by niklas.enbom@webrtc.org · 13 years ago
  49. fcf33eb Limit number of send-side BWE increases to one per second. by stefan@webrtc.org · 13 years ago
  50. 81d4499 Microphone volume on Mac not being printed properly due by punyabrata@webrtc.org · 13 years ago
  51. 755b04a Add RMS computation for the RTP level indicator. by andrew@webrtc.org · 13 years ago
  52. 6a85b17 Potential fix for crash after Mac sleep. by andrew@webrtc.org · 13 years ago
  53. 85596d5 Setting completeFrame to true for all created encoded images. by kjellander@webrtc.org · 13 years ago
  54. cde1e7f Use a TraceNoop instance when tracing disabled (to be used in Chromium). by tommi@webrtc.org · 13 years ago
  55. bc91d5a NetEQ tests by henrik.lundin@webrtc.org · 13 years ago
  56. a02ef1a Fix broken tree. by mflodman@webrtc.org · 13 years ago
  57. 1f69c03 Added size sanity check for copying app specific RTCP data. by mflodman@webrtc.org · 13 years ago
  58. 33df533 Change luminance of all pixels by a specified value. by henrik.lundin@webrtc.org · 13 years ago
  59. 7de0765 Disables a flaky metric test. by stefan@webrtc.org · 13 years ago
  60. ded85f1 Enable WEBRTC_NO_TRACE for Chromium builds. by tommi@webrtc.org · 13 years ago
  61. 0db7dc6 Add file-playing channels to voe_cmd_test. by andrew@webrtc.org · 13 years ago
  62. cd82438 Unpack the full set of audioproc data. by andrew@webrtc.org · 13 years ago
  63. d71d480 Fixed a build error of audio conference mixer in Android. by kma@webrtc.org · 13 years ago
  64. b351d6a Reverting rev 929 due to failing assert on Linux. by stefan@webrtc.org · 13 years ago
  65. fd3a0ef RTP bw estimate fix. by mflodman@webrtc.org · 13 years ago
  66. 1144ba2 Base and codec tests now run verify output and render to file instead of to screen. by phoglund@webrtc.org · 13 years ago
  67. 50b3cbe First pass. You can now enable a stereo codec and send and receive. This does not include more advances use cases (DTMF etc), but I'd rather keep the CLs manageable. by niklas.enbom@webrtc.org · 13 years ago
  68. b61c410 Fixed a couple of Android makefiles to let voe and vie build properly. by kma@webrtc.org · 13 years ago
  69. 13318ef (1) Corrected the makefile for testing iLBC in Android, and changed the location of the test makefile to make it consistent with audio_processing. by kma@webrtc.org · 13 years ago
  70. 7a4eb28 Calculate the available bandwidth before sending a TMMBR by mflodman@webrtc.org · 13 years ago
  71. 637a59e jitter buffer update: waiting for key frame when Nack is enabled and continuity cannot be determined. by mflodman@webrtc.org · 13 years ago
  72. 855a77c Audio Coding Module: Fixing a bug that prevented the encoder from being re-initialized when changing codec from mono to stereo. by tina.legrand@webrtc.org · 13 years ago
  73. c4f129f Improve the mixing saturation protection scheme. by andrew@webrtc.org · 13 years ago
  74. d30b688 Remove TraceScan executable. by andrew@webrtc.org · 13 years ago
  75. 4b13fc9 Add delay modification to process_test. by andrew@webrtc.org · 13 years ago
  76. 2f32b5c Fixes an issue where file playing could happen at a lower sampling frequency than the file. by henrike@webrtc.org · 13 years ago
  77. eb4ef17 Removing vplib include and VideoInterpolator when not needed by mikhal@webrtc.org · 13 years ago
  78. 488ed92 Removing exceptions since not used by kjellander@webrtc.org · 13 years ago
  79. c3a4dcd Removing exceptions since not used by kjellander@webrtc.org · 13 years ago
  80. ad79d6f Removing exceptions since not used by kjellander@webrtc.org · 13 years ago
  81. 03a9eb1 RTP module: Make sure payloadName is null terminated. by mflodman@webrtc.org · 13 years ago
  82. f3c1b87 my eyes started bleeding when I saw this... by niklas.enbom@webrtc.org · 13 years ago
  83. 9dcab8f Restoring Android.mk by kjellander@webrtc.org · 13 years ago
  84. 4cd841e Fix win compile error for interpolator_test by niklas.enbom@webrtc.org · 13 years ago
  85. cff98ca Made it possible to run the voe_auto_test standard test in GTest behind a flag. The purpose is to run the whole test without any manual intervention since we want to run the test on a build bot in automated mode. by phoglund@webrtc.org · 13 years ago
  86. c58ef08 Removes system CPU measurement for Chrome build. by henrikg@webrtc.org · 13 years ago
  87. f15fbc3 Change in RTP module SendVP8 by henrik.lundin@webrtc.org · 13 years ago
  88. 9b81351 Changes for building audio coding in anroid. Only makefiles are touched. by kma@webrtc.org · 13 years ago
  89. 26d3667 Fix for broken test after r897 by henrike@webrtc.org · 13 years ago
  90. e2a34f8 Removes the API for setting RX VAD since the RX vad should always be on anyways. by henrike@webrtc.org · 13 years ago
  91. 5ae9f5e Adding logs in RTPSender::ReSendToNetwork. by mflodman@webrtc.org · 13 years ago
  92. bf48384 Restructuring and removing neteq_tests.gypi according to project structure discussed with Andrew. We want to flatten out the hierarchy and minimize the number of GYP files. by kjellander@webrtc.org · 13 years ago
  93. 36e1ad9 Restructuring and removing ilbc_test.gypi. by kjellander@webrtc.org · 13 years ago
  94. b353d21 ...and now fix the Debug build. by andrew@webrtc.org · 13 years ago
  95. 369766e Fix Release mode errors in common_video tests. by andrew@webrtc.org · 13 years ago
  96. a5c4c1f Fix for WebRTC issue 64, removed the screenupdate thread and events from start render as they are already created in the ctor. by vikasmarwaha@webrtc.org · 13 years ago
  97. 040cb71 Fix windows compilation errors and warning for test_fec. Disabled VERBOSE_OUTPUT. by marpan@webrtc.org · 13 years ago
  98. 731e9ae Fixes ACM API test to build on 32-bits machines. by tina.legrand@webrtc.org · 13 years ago
  99. 20a370e Changing the namespace of TestSuite to webrtc::test. by kjellander@webrtc.org · 13 years ago
  100. 1a8d08a Changing usage of gtest_main target, to use test_support_main instead. by kjellander@webrtc.org · 13 years ago