1. 5835adf Reorganize gyp for Android by leozwang@webrtc.org · 12 years ago
  2. 3263a7a Setting correct stride for VP8 encoder by mikhal@webrtc.org · 12 years ago
  3. 32b3f40 Adding an aligned stride test to LibYuv by mikhal@webrtc.org · 12 years ago
  4. 8187877 Reland 3135 - Previous failure was bot flakiness. ***** by tommi@webrtc.org · 12 years ago
  5. 951b6c4 Revert 3135 - This broke the Mac bots somehow. Here's the error: by tommi@webrtc.org · 12 years ago
  6. 704eb8f Restructure the video_capture code a bit to make room for a Media Foundation class implementation. by tommi@webrtc.org · 12 years ago
  7. 655d8f5 Add a kTraceTerseInfo level for non-verbose logging. by andrew@webrtc.org · 12 years ago
  8. 2009f6b Add Chromium's perf_test to testsupport. by andrew@webrtc.org · 12 years ago
  9. 0f34fd7 Updating Memory allocation for rotation and related tests. by mikhal@webrtc.org · 12 years ago
  10. 467dfe0 Fix possible race condition and access into an empty list. by stefan@webrtc.org · 12 years ago
  11. 4100b04 Move SSRC list to RemoteBitrateEstimator. by stefan@webrtc.org · 12 years ago
  12. 5ac387c Allow NetEQ to use real packet durations. by tina.legrand@webrtc.org · 12 years ago
  13. 3662aa3 Revert 3123 - Roll to libyuv r496 for Android x86 fix by avoiding stdint.h by kjellander@webrtc.org · 12 years ago
  14. 815653b Use cpu_features library from ndk when built with chromium. by wjia@webrtc.org · 12 years ago
  15. 5c38d90 Define enable_android_opensl when built with chromium. by wjia@webrtc.org · 12 years ago
  16. de727ab Fixes http://code.google.com/p/webrtc/issues/detail?id=941 by henrike@webrtc.org · 12 years ago
  17. 55cd78c Porting ARM optimization from Android to ios. by kma@webrtc.org · 12 years ago
  18. 2ec58dc Roll to libyuv r496 for Android x86 fix by avoiding stdint.h by fbarchard@google.com · 12 years ago
  19. c79505f Add warning comment Review URL: https://webrtc-codereview.appspot.com/933012 by niklas.enbom@webrtc.org · 12 years ago
  20. 7381496 Add a variable for the libvpx path. by andrew@webrtc.org · 12 years ago
  21. d0acdf6 Fix ordered comparison warnings in the RTPtimeshift unit test by tina.legrand@webrtc.org · 12 years ago
  22. b2f474e Adding ViE CallStats to keep track of call statistics. As a start, only rtt is handled. by mflodman@webrtc.org · 12 years ago
  23. c289f9f Replaced remb unittest sleep with fake clock. by mflodman@webrtc.org · 12 years ago
  24. 21b426d Revert 3111 (revert of a revert). by tommi@webrtc.org · 12 years ago
  25. 12773ea Minor cleanup of the videocapture code. No "real" code change :) by tommi@webrtc.org · 12 years ago
  26. d14ff54 Removed unnecessary lines in one of the tests and changed one parameter. by marpan@webrtc.org · 12 years ago
  27. 01c4b98 Revert 3105 - Don't crash the unit test host when tests fail. by mikhal@webrtc.org · 12 years ago
  28. 37ff69d Point codereview.settings at the newly-created webrtc-reviews google group. by fischman@webrtc.org · 12 years ago
  29. d72b3d6 Fix cpplint errors in audio_processing.h by andrew@webrtc.org · 12 years ago
  30. f3adba4 Add Android include path so that header files can follow google style by leozwang@webrtc.org · 12 years ago
  31. 0f4185f Making valgrind wrapper script work with test arguments by kjellander@webrtc.org · 12 years ago
  32. 5ea70e6 Don't crash the unit test host when tests fail. by tommi@webrtc.org · 12 years ago
  33. 2d0b81b Fix sorting issues in video_capture.gypi. No code change. by tommi@webrtc.org · 12 years ago
  34. 53a8be2 Wraparound distortion in Opus by tina.legrand@webrtc.org · 12 years ago
  35. 23ec30b Clean up TraceCallback::Print. by andrew@webrtc.org · 12 years ago
  36. 8123ed7 libyuv roll to r481 for optimization of neon yuv to/from rgb by fbarchard@google.com · 12 years ago
  37. aea2510 Fix generate_asm_header. by wjia@webrtc.org · 12 years ago
  38. 50419b0 Add libjingle-style stream-style logging. by andrew@webrtc.org · 12 years ago
  39. 1786436 Pure Neon assembly coding for WebRtcIsacfix_AutocorrNeon() in iSAC-Fix. by kma@webrtc.org · 12 years ago
  40. 9e9cc72 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
  41. 71fd288 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
  42. a36d75a Reformatted condition_variable* in system_wrappers. by phoglund@webrtc.org · 12 years ago
  43. 12b828a Fixed test memory leak + disabled base test. by phoglund@webrtc.org · 12 years ago
  44. 56a0076 Add myself to the all_webrtc watchlist. by andrew@webrtc.org · 12 years ago
  45. cb7561c Adding myself to webrtc watchlist. by fischman@webrtc.org · 12 years ago
  46. 6b9543b Add libpaced_sender to Android makefile by leozwang@webrtc.org · 12 years ago
  47. d5fbdc8 Increase number of channels that can be supported on Android by leozwang@webrtc.org · 12 years ago
  48. 571a1c0 Enable paced sender. Review URL: https://webrtc-codereview.appspot.com/965016 by pwestin@webrtc.org · 12 years ago
  49. 42aa10e Clarifies the bandwidth estimation interfaces. by stefan@webrtc.org · 12 years ago
  50. 7577ddf Refactoring acm_generic_codec by tina.legrand@webrtc.org · 12 years ago
  51. 1726661 Update parsed non ref frame info. by asapersson@webrtc.org · 12 years ago
  52. c734669 Re-create libvpx configuration files for Android by leozwang@webrtc.org · 12 years ago
  53. 8d18526 Fixes an incorrect if statement in vie_sync_module.cc. by stefan@webrtc.org · 12 years ago
  54. 6e47007 mac: Fix a port leak in threading code. by thakis@chromium.org · 12 years ago
  55. b900721 Fix OpenGL rendering of WebRTCDemo by accounting for stride != width. by fischman@webrtc.org · 12 years ago
  56. af49062 Revert 3071 - i420:verify image length by henrik.lundin@webrtc.org · 12 years ago
  57. cb76c3c Unbreak ninja/android build of webrtc. by fischman@webrtc.org · 12 years ago
  58. 6590ec3 Teach webrtc/codereview.settings how to point at svn rev's so rietveld issues get a useful URL. by fischman@webrtc.org · 12 years ago
  59. f4b2617 Re-initialize enough state on "Stop Call" to be able to stop/start multiple calls in succession. by fischman@webrtc.org · 12 years ago
  60. b518017 Adding pacing module, will replace the transmission_bucket in the RTP module. by pwestin@webrtc.org · 12 years ago
  61. f875fd2 i420:verify image length by mikhal@webrtc.org · 12 years ago
  62. 701567a Capture module: Fixing size computation for u and v planes TEST=trybots by mikhal@webrtc.org · 12 years ago
  63. 06d72d8 Add Android OWNER files by leozwang@webrtc.org · 12 years ago
  64. e8ef807 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
  65. 9cb9fc1 Reformatted atomic32 files. by phoglund@webrtc.org · 12 years ago
  66. fa65c85 Optimized function AllpassFilter2FixDec16() in isac fix for Android Neon platforms. by kma@webrtc.org · 12 years ago
  67. b952a90 Remove an unused Shutdown method from the ThreadWrapper interface. by tommi@webrtc.org · 12 years ago
  68. 1401285 Can now fully control custom calls from the command line. by phoglund@webrtc.org · 12 years ago
  69. 2a749d3 Verify output frame timestamp in VideoProcessingModuleTest.Resampler. by wu@webrtc.org · 12 years ago
  70. 206532e Fix a bug in spatial_resampler where we should set the timestamp after Scale. by wu@webrtc.org · 12 years ago
  71. 1245402 Fixed and enabled ARM assembly code in AECM and NS. by kma@webrtc.org · 12 years ago
  72. 31eae47 Implemented a build system that generates offset header files for ARM assembly files, in Android. by kma@webrtc.org · 12 years ago
  73. 055663b Updating vp8 tests by mikhal@webrtc.org · 12 years ago
  74. c862f49 Move capture level computation after all processing. by andrew@webrtc.org · 12 years ago
  75. 7096fc0 Break out unittest helpers for remote_bitrate_estimator. by stefan@webrtc.org · 12 years ago
  76. ac993fe Adding codecType to OnIncomingCapturedEncodedFrame partially reverting r3013. by mikhal@webrtc.org · 12 years ago
  77. c66e8b3 pre-factor cleanup pre-work. by pwestin@webrtc.org · 12 years ago
  78. 4cebe6c Made TickTime immutable, rewrote tick utils to be fakeable. by phoglund@webrtc.org · 12 years ago
  79. 6e9890d Removed ViEBaseObserver. by mflodman@webrtc.org · 12 years ago
  80. 8d0cef3 Updating opus in .gitignore by kjellander@webrtc.org · 12 years ago
  81. 0ad3c1a Adding Opus stereo support to WebRTC by tina.legrand@webrtc.org · 12 years ago
  82. 6dddfe9 Fix for webrtc issue 1052 on windows with vie_auto_test. by vikasmarwaha@webrtc.org · 12 years ago
  83. ddcc942 Check the channels in receive-side processing frames. by andrew@webrtc.org · 12 years ago
  84. e5b49a0 Update timestamp offset for re-transmitted packets. by asapersson@webrtc.org · 12 years ago
  85. f17d7d1 Using proper GYP references for Strmiids.lib on Windows by kjellander@webrtc.org · 12 years ago
  86. f7fa627 Reformating files in audio coding module. by tina.legrand@webrtc.org · 12 years ago
  87. a56d759 Removing use of raw buffers for I420PSNR and I420SSIM functions by kjellander@webrtc.org · 12 years ago
  88. 5f9970f Refactor OpenSL audio driver by leozwang@webrtc.org · 12 years ago
  89. 737ed3b 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
  90. 1be46fc Change src/webrtc in WATCHLIST. by mflodman@webrtc.org · 12 years ago
  91. f3ffcce Adding default trybot names to PRESUBMIT.py. by kjellander@webrtc.org · 12 years ago
  92. ef62929 Landing http://review.webrtc.org/914006/ by niklas.enbom@webrtc.org · 12 years ago
  93. 1a2a6dd Fixes a bitrate mismatch between sender and receiver. by stefan@webrtc.org · 12 years ago
  94. e034f21 Remove video_capture/test/android by andrew@webrtc.org · 12 years ago
  95. 9841d92 Reorganize modules/video_render. by andrew@webrtc.org · 12 years ago
  96. 3c01316 Fix Android build after video_capture reorg. by andrew@webrtc.org · 12 years ago
  97. 94caca7 Reorganize modules/video_capture. by andrew@webrtc.org · 12 years ago
  98. 5ff091f 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. 91a0340 Adding stride alignment by mikhal@webrtc.org · 12 years ago
  100. 6ab92ed Check if opus exists when build test app on Android by leozwang@webrtc.org · 12 years ago