1. 340f0cd Add *.host.mk to .gitignore. by andrew@webrtc.org · 12 years ago
  2. 54da26d Move include_tests to a higher variable nesting. by andrew@webrtc.org · 12 years ago
  3. b20916e Change libaries path because of recent file structure changes by leozwang@webrtc.org · 12 years ago
  4. 490fdc5 Fix Android after iSAC -> isac move. by andrew@webrtc.org · 12 years ago
  5. 6a16e74 Move iSAC -> isac. by andrew@webrtc.org · 12 years ago
  6. cb53410 Make some dependencies more flexible. by andrew@webrtc.org · 12 years ago
  7. ea5d0e5 Don't target top-level test gyps if include_tests==0 by andrew@webrtc.org · 12 years ago
  8. 48cf686 Removed v4l2_file_player code, which is checked into the signal repo. by phoglund@webrtc.org · 12 years ago
  9. 7f3d4c6 Normalize x64 and ia32 sse2 behavior in libvpx. by andrew@webrtc.org · 12 years ago
  10. ccb7cc6 Added unittest for iSAC-Fix. by kma@webrtc.org · 12 years ago
  11. c2fde80 Fix VideoCaptureModule and unit test valgrind errors on linux. by perkj@webrtc.org · 12 years ago
  12. d7b17e4 Enable denoising by default. by stefan@webrtc.org · 12 years ago
  13. 54e22eb Made it possible to run video_capture tests on mac. by phoglund@webrtc.org · 12 years ago
  14. 743e5cf remove flaky test case in FileBeforeStreamingTest by braveyao@webrtc.org · 12 years ago
  15. da236df Added more unit tests for min-max operations in signal processing module. by kma@webrtc.org · 12 years ago
  16. 5d6be54 Make sure the video capture delay is set to an initial value on Mac. by perkj@webrtc.org · 12 years ago
  17. 9a3f83f Use relative paths in DEPS. by andrew@webrtc.org · 12 years ago
  18. f388fcc Added dynamic Neon detect in isac-fix for Android NDK build, and thus fixed a build error in the last version. by kma@webrtc.org · 12 years ago
  19. 7155098 Added run time detection of Neon architecture in iSAC-fix. by kma@webrtc.org · 12 years ago
  20. 8fe5f32 Refactor three signal processing library files. WebRTC issue 545 is solved by the way. by kma@webrtc.org · 12 years ago
  21. 1e1a250 Wrong RTP module used when calling RegisterReceiveRtpHeaderExtension in ViE channel. by mflodman@webrtc.org · 12 years ago
  22. 2d4c4ae Optimization of function CalculateResidualEnergy() for iSAC-fix in ARM Neon platforms. by kma@webrtc.org · 12 years ago
  23. 07ebdb9 Handle 96 kHz when downmixing the capture path. by andrew@webrtc.org · 12 years ago
  24. c0348fb bump version to 3.9.0 Review URL: https://webrtc-codereview.appspot.com/708007 by elham@webrtc.org · 12 years ago
  25. 4889120 Fix integer divisin truncation error. by marpan@webrtc.org · 12 years ago
  26. 10a3152 Disabled FileBeforeStreamingTest.TestStartPlayingFileLocallyWithStartPlayout. by mflodman@webrtc.org · 12 years ago
  27. 1526436 Removing RawImage. Last cl in the series. by mikhal@webrtc.org · 12 years ago
  28. 8274bf2 Added suppression for issue 716. by phoglund@webrtc.org · 12 years ago
  29. 686a731 Fix error when receiving an already sent timestamp from VoE. by andrew@webrtc.org · 12 years ago
  30. 792e974 Refactor the public interfaces to use the full path in include. by wu@webrtc.org · 12 years ago
  31. 42033b4 This change will allow us to set proper frame rate for the camera on Linux. Earlier we were setting based on the resolution irrespective of input frame rate. by mallinath@webrtc.org · 12 years ago
  32. d7a71d0 Prepare to roll Chromium to 149181. by andrew@webrtc.org · 12 years ago
  33. bf85391 Fix issue introduced in r2540 by mikhal@webrtc.org · 12 years ago
  34. 4ce52bb Use unix line endings in DEPS. by andrew@webrtc.org · 12 years ago
  35. 4147562 Fixing error introduced in r2540. by mikhal@webrtc.org · 12 years ago
  36. a2031d5 Replacing RawImage with VideoFrame in video_coding and related engine code. by mikhal@webrtc.org · 12 years ago
  37. 5fe91a8 Add license header to no_op.cc. by andrew@webrtc.org · 12 years ago
  38. 8639fd9 Use correct rtp header size for FEC packets. by marpan@webrtc.org · 12 years ago
  39. d1f3b1a Reorganize the vp8 directory. by andrew@webrtc.org · 12 years ago
  40. 6f8db36 Reorganize voice_engine/. by andrew@webrtc.org · 12 years ago
  41. c1354bd Make handling of libyuv more flexible. by andrew@webrtc.org · 12 years ago
  42. f5f69c7 Resource Preprocessor Definitions which contain spaces are handled incorrectly in Visual Studio 2010 by braveyao@webrtc.org · 12 years ago
  43. 7cbb5a0 JPEG: Replacing RawImage with VideoFrame. by mikhal@webrtc.org · 12 years ago
  44. 8d95a70 Change libvp8 library patch in makefile by leozwang@webrtc.org · 12 years ago
  45. bc934cc Temporarily disable version.py. by andrew@webrtc.org · 12 years ago
  46. ad69ca7 webrtc crashes with virtual cameras on Windows. by braveyao@webrtc.org · 12 years ago
  47. f5a91fd Make some build settings more flexible. by andrew@webrtc.org · 12 years ago
  48. a9da4c5 Landing for thakis. Original review here: by tommi@webrtc.org · 12 years ago
  49. 8495915 Make loopback mode work properly by leozwang@webrtc.org · 12 years ago
  50. d41f59a Fix Mac-gcc warnings. by andrew@webrtc.org · 12 years ago
  51. 837bc7b ilbc: Make the decode input array const by turaj@webrtc.org · 12 years ago
  52. 73db8db video conversion functions: switching from designated functions to a general one. by mikhal@webrtc.org · 12 years ago
  53. 7760963 Make webrtc compile on android in chromium by leozwang@webrtc.org · 12 years ago
  54. 6c08f26 Terminate version string by leozwang@webrtc.org · 12 years ago
  55. 71707aa Add the FEC mask type to FecProtectionParams and set the mask type in the VCM. by marpan@webrtc.org · 12 years ago
  56. d96dcef vpm: Updating module to use CalcBufferSize by mikhal@webrtc.org · 12 years ago
  57. 08329f4 Added API to port internal speech probability in NS. by bjornv@webrtc.org · 12 years ago
  58. 6182db1 vp8: Updating wrapper to use CalcBufferSize (includes odd size support). by mikhal@webrtc.org · 12 years ago
  59. 538f0ab I420: Updating computation of buffer size to use calcBufferSize (odd size support). by mikhal@webrtc.org · 12 years ago
  60. 262bded Remove files that are not needed from direct_show_base_classes.gyp by wu@webrtc.org · 12 years ago
  61. 13c09bc . by wu@webrtc.org · 12 years ago
  62. ff2f861 Corrected one error for Android build. by kma@webrtc.org · 12 years ago
  63. b95e9ca video_coding: Refatoring I420 wrapper. No functional updates. by mikhal@webrtc.org · 12 years ago
  64. 0bb817d 1. Adding odd size support to LibYuv wrapper. by mikhal@webrtc.org · 12 years ago
  65. 475c266 Re-enable WEBRTC_SVNREVISION script by leozwang@webrtc.org · 12 years ago
  66. adf8ddf Assembly coding for pitch filter in iSAC for ARMv6. by kma@webrtc.org · 12 years ago
  67. e2c16a8 Optimized a filter bank function in iSAC/fix for ARM. by kma@webrtc.org · 12 years ago
  68. cf9855d Update build.xml and api level by leozwang@webrtc.org · 12 years ago
  69. d2f7100 correct one build error in linux. by kma@webrtc.org · 12 years ago
  70. 72f8a6d Optimized PCorr2Q32() in iSAC with intrinsics in ARM Neon platform. by kma@webrtc.org · 12 years ago
  71. 1bc6d32 Only status from interesting bots are reported to the Dashboard by kjellander@webrtc.org · 12 years ago
  72. e9eb235 Remove the useless dummy audio device impl which creates threads and high res timers on windows. by xians@webrtc.org · 12 years ago
  73. 2eefb22 Improved fuzzer. It will now throw in additional refreshes, which is known to mess with lifetime assumptions. by phoglund@webrtc.org · 12 years ago
  74. 01ad758 ilbc: Mark untouched input arrays as const by turaj@webrtc.org · 12 years ago
  75. ddfdfed Pass capture time (wallclock) to the RTP sender to compute transmission offset by stefan@webrtc.org · 12 years ago
  76. 1853005 Change clock to be 64 bits in RTP module by pwestin@webrtc.org · 12 years ago
  77. 7b61049 Land: https://webrtc-codereview.appspot.com/678010/ by tommi@webrtc.org · 12 years ago
  78. fb933bd Landing: https://webrtc-codereview.appspot.com/680005/ by tommi@webrtc.org · 12 years ago
  79. e85c77b Bump WebRTC version to 3.8.1 by vikasmarwaha@webrtc.org · 12 years ago
  80. cf21b9b Fix ChromeOS build by removing an unused variable. by tommi@webrtc.org · 12 years ago
  81. ef8ca6a Wrote ClusterFuzz test for WebRTC GetUserMedia. by phoglund@webrtc.org · 12 years ago
  82. b358bd8 A command-line tool based on libyuv to convert a set of RGBA files to a YUV video. by vspasova@webrtc.org · 12 years ago
  83. 2a45209 Adde video_engine to cpplint check. by mflodman@webrtc.org · 12 years ago
  84. c5b392e Updates t resolution adaptation (cama): by marpan@webrtc.org · 12 years ago
  85. ea5b8b5 Trival changes in gui layout based on feedback by leozwang@webrtc.org · 12 years ago
  86. fb59442 Change file path to make it work on android by leozwang@webrtc.org · 12 years ago
  87. 8d59e70 In this CL four pitch-filters are integrated into a single function. I have kept the interfaces unchanged so there was no need to modify any other file. A test is uploaded to show how this CL is tested. The test engages all the functions affected by this CL and compares their output with the version of iSAC before this change. This CL is bit-exact. Furthermore, I ran iSAC release test and diff results with previous version. The test file will not be commited, as running it requires a hack in old iSAC to. Hence you don't need to code-review it. by turaj@webrtc.org · 12 years ago
  88. e06ca3c Removed nolint for include guards. by mflodman@webrtc.org · 12 years ago
  89. ab2610f Removed the last lint warnings in video_engine. by mflodman@webrtc.org · 12 years ago
  90. efe20b3 Only add Mac compiler warning for clang, not gcc. by mflodman@webrtc.org · 12 years ago
  91. 1115fdb Remove tab from DEPS. by andrew@webrtc.org · 12 years ago
  92. a5fcf7a Fixes broken Chromium build. by henrike@webrtc.org · 12 years ago
  93. 44361ab Moving from Chromium cygwin dependency to our own by kjellander@webrtc.org · 12 years ago
  94. c802e0e Changed max codec resolution. by mflodman@webrtc.org · 12 years ago
  95. d2e6779 Fix for negative transmission time offset. by asapersson@webrtc.org · 12 years ago
  96. 5f28498 First step in refactoring audio/video synchronization. Adds unittests. by stefan@webrtc.org · 12 years ago
  97. cee447a cpplint passes for vie_performance_monitor, vie_manager_base, vie_impl, vie_renderer, vie_defines and vie_render_manager. by mflodman@webrtc.org · 12 years ago
  98. 100463e Added initial nack configuration for rtp module. by asapersson@webrtc.org · 12 years ago
  99. 1b1cd78 Made cpplint pass for vie_remb, vie_ref_count, vie_sender and vie_receiver. by mflodman@webrtc.org · 12 years ago
  100. e22beab [MIPS] Adding support for MIPS architecture for WebRTC. by andrew@webrtc.org · 12 years ago