1. 804d552 Add a logging_no_op.cc when enable_tracing==0. by andrew@webrtc.org · 12 years ago
  2. 856edd5 Remove operator overloading from RTPFragmentationHeader. by andrew@webrtc.org · 12 years ago
  3. e3ada29 Fixes (or at least reduces) the flakiness in the full stack test by making sure the different frame monitors are registered and deregistered in the right order. Also makes sure only local preview frames which are actually transmitted are rendered by moving the local preview rendering to an effect filter. by stefan@webrtc.org · 12 years ago
  4. 95430b6 Condition for DirectX variable on Windows by kjellander@webrtc.org · 12 years ago
  5. fca1d1d Removed codec comparison test: it didn't work and probably never will. by phoglund@webrtc.org · 12 years ago
  6. 2bf4761 Adding Direct X SDK include directory. by kjellander@webrtc.org · 12 years ago
  7. 1d50745 Remove ViE lint warnings that should have been caught at upload time. by mflodman@webrtc.org · 12 years ago
  8. eb72e23 Removed not used include. by mflodman@webrtc.org · 12 years ago
  9. e905921 Setting capture stride to width by mikhal@webrtc.org · 12 years ago
  10. 551e488 Ensure opus_demo has a targets block. by andrew@webrtc.org · 12 years ago
  11. fb0c0d9 Add winsdk_samples to provide directshow_baseclasses. by andrew@webrtc.org · 12 years ago
  12. b13c5e2 Build opus_demo by leozwang@webrtc.org · 12 years ago
  13. 4c54650 Reformatted most of the CPU stuff in system_wrappers. by phoglund@webrtc.org · 12 years ago
  14. ab9aa45 Reorganize gyp for Android by leozwang@webrtc.org · 12 years ago
  15. 05eec40 Setting correct stride for VP8 encoder by mikhal@webrtc.org · 12 years ago
  16. 052382e Adding an aligned stride test to LibYuv by mikhal@webrtc.org · 12 years ago
  17. 0f224ff Reland 3135 - Previous failure was bot flakiness. ***** by tommi@webrtc.org · 12 years ago
  18. 3b7f2ab Revert 3135 - This broke the Mac bots somehow. Here's the error: by tommi@webrtc.org · 12 years ago
  19. a882006 Restructure the video_capture code a bit to make room for a Media Foundation class implementation. by tommi@webrtc.org · 12 years ago
  20. bc687c5 Add a kTraceTerseInfo level for non-verbose logging. by andrew@webrtc.org · 12 years ago
  21. d064f58 Add Chromium's perf_test to testsupport. by andrew@webrtc.org · 12 years ago
  22. 3082003 Updating Memory allocation for rotation and related tests. by mikhal@webrtc.org · 12 years ago
  23. 8e3d40c Fix possible race condition and access into an empty list. by stefan@webrtc.org · 12 years ago
  24. 7d32491 Move SSRC list to RemoteBitrateEstimator. by stefan@webrtc.org · 12 years ago
  25. c05b561 Allow NetEQ to use real packet durations. by tina.legrand@webrtc.org · 12 years ago
  26. 0739180 Use cpu_features library from ndk when built with chromium. by wjia@webrtc.org · 12 years ago
  27. 10b747a Define enable_android_opensl when built with chromium. by wjia@webrtc.org · 12 years ago
  28. 03a161e Fixes http://code.google.com/p/webrtc/issues/detail?id=941 by henrike@webrtc.org · 12 years ago
  29. b238aca Porting ARM optimization from Android to ios. by kma@webrtc.org · 12 years ago
  30. ece4890 Add warning comment Review URL: https://webrtc-codereview.appspot.com/933012 by niklas.enbom@webrtc.org · 12 years ago
  31. 641b4aa Fix ordered comparison warnings in the RTPtimeshift unit test by tina.legrand@webrtc.org · 12 years ago
  32. e296783 Adding ViE CallStats to keep track of call statistics. As a start, only rtt is handled. by mflodman@webrtc.org · 12 years ago
  33. e6527c1 Replaced remb unittest sleep with fake clock. by mflodman@webrtc.org · 12 years ago
  34. e293cde Revert 3111 (revert of a revert). by tommi@webrtc.org · 12 years ago
  35. 5c247c4 Minor cleanup of the videocapture code. No "real" code change :) by tommi@webrtc.org · 12 years ago
  36. 85a15fd Removed unnecessary lines in one of the tests and changed one parameter. by marpan@webrtc.org · 12 years ago
  37. f5366c1 Revert 3105 - Don't crash the unit test host when tests fail. by mikhal@webrtc.org · 12 years ago
  38. 7fa848b Fix cpplint errors in audio_processing.h by andrew@webrtc.org · 12 years ago
  39. 6fb7314 Add Android include path so that header files can follow google style by leozwang@webrtc.org · 12 years ago
  40. d894331 Don't crash the unit test host when tests fail. by tommi@webrtc.org · 12 years ago
  41. c9d3cd1 Fix sorting issues in video_capture.gypi. No code change. by tommi@webrtc.org · 12 years ago
  42. a049d6e Wraparound distortion in Opus by tina.legrand@webrtc.org · 12 years ago
  43. d75680a Clean up TraceCallback::Print. by andrew@webrtc.org · 12 years ago
  44. 1b790df Fix generate_asm_header. by wjia@webrtc.org · 12 years ago
  45. d898c01 Add libjingle-style stream-style logging. by andrew@webrtc.org · 12 years ago
  46. 27fe999 Pure Neon assembly coding for WebRtcIsacfix_AutocorrNeon() in iSAC-Fix. by kma@webrtc.org · 12 years ago
  47. c0bf9f0 Relanding r3071 - updates for i420: Making sure that decoded frame is complete and buffer size is sufficient. Re-landing is possible following r3094 - which disabled a problematic test. by mikhal@webrtc.org · 12 years ago
  48. 0734656 Fixed indentation and added the description of how to supply argument with specification of a name for the ouputfile where the contentMetrics etc. are logged. by brykt@google.com · 12 years ago
  49. 87beb44 Reformatted condition_variable* in system_wrappers. by phoglund@webrtc.org · 12 years ago
  50. d697e19 Fixed test memory leak + disabled base test. by phoglund@webrtc.org · 12 years ago
  51. 7e19b40 Add libpaced_sender to Android makefile by leozwang@webrtc.org · 12 years ago
  52. 7a3faf9 Increase number of channels that can be supported on Android by leozwang@webrtc.org · 12 years ago
  53. 5e87b5f Enable paced sender. Review URL: https://webrtc-codereview.appspot.com/965016 by pwestin@webrtc.org · 12 years ago
  54. 376495a Clarifies the bandwidth estimation interfaces. by stefan@webrtc.org · 12 years ago
  55. 2bd7fc7 Refactoring acm_generic_codec by tina.legrand@webrtc.org · 12 years ago
  56. 8b48cdc Update parsed non ref frame info. by asapersson@webrtc.org · 12 years ago
  57. 64ff6c9 Fixes an incorrect if statement in vie_sync_module.cc. by stefan@webrtc.org · 12 years ago
  58. b868710 mac: Fix a port leak in threading code. by thakis@chromium.org · 12 years ago
  59. bb1c56d Fix OpenGL rendering of WebRTCDemo by accounting for stride != width. by fischman@webrtc.org · 12 years ago
  60. 155be41 Revert 3071 - i420:verify image length by henrik.lundin@webrtc.org · 12 years ago
  61. 5784346 Unbreak ninja/android build of webrtc. by fischman@webrtc.org · 12 years ago
  62. 9672571 Re-initialize enough state on "Stop Call" to be able to stop/start multiple calls in succession. by fischman@webrtc.org · 12 years ago
  63. cba9d52 Adding pacing module, will replace the transmission_bucket in the RTP module. by pwestin@webrtc.org · 12 years ago
  64. 00453c8 i420:verify image length by mikhal@webrtc.org · 12 years ago
  65. 03a09a2 Capture module: Fixing size computation for u and v planes TEST=trybots by mikhal@webrtc.org · 12 years ago
  66. d6a40b6 Add Android OWNER files by leozwang@webrtc.org · 12 years ago
  67. 2493070 Added possibility to run quality modes test. Added possibility to input arguments to the test. The test will (for each frame) log the values in contentMetrics to a txt-file. The txt-file can optionally be saved in a specific place. Fixed an issue where video_coding_test crashed if there weren't any parameter submitted to an input argument. by brykt@google.com · 12 years ago
  68. 9ca84f7 Reformatted atomic32 files. by phoglund@webrtc.org · 12 years ago
  69. d0ea5f0 Optimized function AllpassFilter2FixDec16() in isac fix for Android Neon platforms. by kma@webrtc.org · 12 years ago
  70. 09ea027 Remove an unused Shutdown method from the ThreadWrapper interface. by tommi@webrtc.org · 12 years ago
  71. 8969e93 Can now fully control custom calls from the command line. by phoglund@webrtc.org · 12 years ago
  72. 4be55f2 Verify output frame timestamp in VideoProcessingModuleTest.Resampler. by wu@webrtc.org · 12 years ago
  73. d662a92 Fix a bug in spatial_resampler where we should set the timestamp after Scale. by wu@webrtc.org · 12 years ago
  74. b9c67af Fixed and enabled ARM assembly code in AECM and NS. by kma@webrtc.org · 12 years ago
  75. 9336469 Implemented a build system that generates offset header files for ARM assembly files, in Android. by kma@webrtc.org · 12 years ago
  76. 6bbb555 Updating vp8 tests by mikhal@webrtc.org · 12 years ago
  77. e5ac24f Move capture level computation after all processing. by andrew@webrtc.org · 12 years ago
  78. 686a447 Break out unittest helpers for remote_bitrate_estimator. by stefan@webrtc.org · 12 years ago
  79. 4d2dafc Adding codecType to OnIncomingCapturedEncodedFrame partially reverting r3013. by mikhal@webrtc.org · 12 years ago
  80. f0f97ca pre-factor cleanup pre-work. by pwestin@webrtc.org · 12 years ago
  81. 1755c25 Made TickTime immutable, rewrote tick utils to be fakeable. by phoglund@webrtc.org · 12 years ago
  82. e7d1c04 Removed ViEBaseObserver. by mflodman@webrtc.org · 12 years ago
  83. 8091fb9 Adding Opus stereo support to WebRTC by tina.legrand@webrtc.org · 12 years ago
  84. 6583f63 Fix for webrtc issue 1052 on windows with vie_auto_test. by vikasmarwaha@webrtc.org · 12 years ago
  85. 0dd483f Check the channels in receive-side processing frames. by andrew@webrtc.org · 12 years ago
  86. a8054d4 Update timestamp offset for re-transmitted packets. by asapersson@webrtc.org · 12 years ago
  87. 2988fef Using proper GYP references for Strmiids.lib on Windows by kjellander@webrtc.org · 12 years ago
  88. c0cf1db Reformating files in audio coding module. by tina.legrand@webrtc.org · 12 years ago
  89. 5028b3f Removing use of raw buffers for I420PSNR and I420SSIM functions by kjellander@webrtc.org · 12 years ago
  90. b285bce Refactor OpenSL audio driver by leozwang@webrtc.org · 12 years ago
  91. 5be3165 libyuv wrapper: 1. Updating rotation settings - in case of 90 or 270 degree rotations, width and height should be updated accordingly. 2. Test clean-up. by mikhal@webrtc.org · 12 years ago
  92. ab360a7 Landing http://review.webrtc.org/914006/ by niklas.enbom@webrtc.org · 12 years ago
  93. e75e29e Fixes a bitrate mismatch between sender and receiver. by stefan@webrtc.org · 12 years ago
  94. e631741 Remove video_capture/test/android by andrew@webrtc.org · 12 years ago
  95. 07e96da Reorganize modules/video_render. by andrew@webrtc.org · 12 years ago
  96. 86bdb3c Fix Android build after video_capture reorg. by andrew@webrtc.org · 12 years ago
  97. 5e8ee6f Reorganize modules/video_capture. by andrew@webrtc.org · 12 years ago
  98. 582a607 Init capturePicture with GetCaptureDeviceSnapshot so that the SetRenderStartImage test won't depend on the previous test which may be disabled by the include_timing_dependent_tests flag. This is a fix for LinuxLargeTests. by wu@webrtc.org · 12 years ago
  99. 7012d2b Adding stride alignment by mikhal@webrtc.org · 12 years ago
  100. fac2c48 Check if opus exists when build test app on Android by leozwang@webrtc.org · 12 years ago