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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 5bea712f9645f3e66be25699572e73b0e9590f90
  1. 5bea712 Two more sleep calls converted to use SleepMs(). by hta@webrtc.org · 11 years ago
  2. ebc0331 TSan v2 reports data races in WebRTCAudioDeviceTest.FullDuplexAudioWithAGC by henrika@webrtc.org · 11 years ago
  3. 22f789f Remove UDP transport API from VoE by pwestin@webrtc.org · 11 years ago
  4. 25dda04 Fixes memory leak in AudioLevel class reported by memory try bots. by henrika@webrtc.org · 11 years ago
  5. 63ef6e2 Fixes data race in WebRTCAudioDeviceTest.StartRecording reported by ThreadSanitizer by henrika@webrtc.org · 11 years ago
  6. e561f8c Remove UDP transport API from ViE by pwestin@webrtc.org · 11 years ago
  7. 45d75a4 Webrtc_Word32 => int32_t in video_coding/main/ by pbos@webrtc.org · 11 years ago
  8. 1562c72 Revert of r3747. by henrike@webrtc.org · 11 years ago
  9. d393127 Two more sleep calls converted to use SleepMs(). by hta@webrtc.org · 11 years ago
  10. d042a17 Fixes data race in WebRTCAudioDeviceTest.Construct reported by ThreadSanitizer by henrika@webrtc.org · 11 years ago
  11. d8322b9 Fix opus bitrate truncated to 16-bit int. This prevented setting bitrates higher by justinlin@chromium.org · 11 years ago
  12. 435b50c For VGA (640x360), currently 1 thread is used. This change increases it to 2 threads. For HD, 4 threads are enabled. by fbarchard@google.com · 11 years ago
  13. 2379013 Updated Webrtc version to 3.28 by elham@webrtc.org · 11 years ago
  14. bbf5086 Fix for issue: https://code.google.com/p/webrtc/issues/detail?id=1549 by marpan@webrtc.org · 11 years ago
  15. bcce6df Fixes build break in previous cl (https://code.google.com/p/webrtc/source/detail?r=3739) found by Android bots. by henrike@webrtc.org · 11 years ago
  16. 18881d5 Removed CPU APIs from VoEHardware. Code is now only used by test applications. by henrike@webrtc.org · 11 years ago
  17. 1ca9d42 Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE SRTP APIs. Test stubs are left in place as we still have the (De)RegisterExternalEncryption() APIs, although they are currently untested. by solenberg@webrtc.org · 11 years ago
  18. e148532 Revert 3736 "Removed CPU APIs from VoEHardware. Code is now only..." by wu@webrtc.org · 11 years ago
  19. 90edf85 Removed CPU APIs from VoEHardware. Code is now only used by test applications. by henrike@webrtc.org · 11 years ago
  20. fece2f5 Fix broken audio. by leozwang@webrtc.org · 11 years ago
  21. 11552e9 G722-stereo has been missing when creating AudioDecoder. by turaj@webrtc.org · 11 years ago
  22. 3e00311 NetEq4 fails if the first packets inserted in are out-of-band DTMFs. by turaj@webrtc.org · 11 years ago
  23. c3ab830 Fix flakiness in network up/down event tests when running under memcheck. by stefan@webrtc.org · 11 years ago
  24. 09e8463 WebRTCDemo: remove unnecessary stop & start during orientation change which isn't necessary since API v14. by fischman@webrtc.org · 11 years ago
  25. e3eea1b Add interface to signal a network down event. by stefan@webrtc.org · 11 years ago
  26. fb6a7c4 Split condition_variable_win.cc into native (for Vista and newer OS versions) and generic implementation (based on events). by henrike@webrtc.org · 11 years ago
  27. 41419d9 Remove VoE's default call in Trace::SetLevelFilter. by andrew@webrtc.org · 11 years ago
  28. eefab4e Fix potential buffer overrun when checking if a packet is RTCP. Also makes validation slightly more robust. by solenberg@webrtc.org · 11 years ago
  29. 6fc92b4 Alphabetize include order in fake_voe_external_media.h. by andrew@webrtc.org · 11 years ago
  30. 6666b90 Restart Android capture after orientation change. Also prevent an NPE on exit. by fischman@webrtc.org · 11 years ago
  31. 58a5924 Add some VoE and AudioProcessing mocks. by andrew@webrtc.org · 11 years ago
  32. f658278 Refactor unittest trace printouts to a separate class. by andrew@webrtc.org · 11 years ago
  33. 8cfba7e Enable the below APIs for iOS. by sjlee@webrtc.org · 11 years ago
  34. 60c8100 Introduced pause and resume to the pacer by pwestin@webrtc.org · 11 years ago
  35. e760243 Updated WebRTC version to 3.27 by elham@webrtc.org · 11 years ago
  36. d3eb512 Bugfix for extended RTP/RTCP test by pwestin@webrtc.org · 11 years ago
  37. 9c3b7bd Move the VIE tests to use external transport instead of the built in udp transport by pwestin@webrtc.org · 11 years ago
  38. 680fbc5 Add trace printouts to all unit tests. by andrew@webrtc.org · 11 years ago
  39. 90fa4a1 Add min and target bitrate to VideoCodec. by marpan@webrtc.org · 11 years ago
  40. 9c0b169 Move the VoE tests to use external transport instead of the built in udp transport by pwestin@webrtc.org · 11 years ago
  41. ce2d125 Creating a copy of Udp transport under webrtc/test by pwestin@webrtc.org · 11 years ago
  42. ed6b4c8 Cleanup nanosleep -> SleepMs Remove some leftover stuff by hta@webrtc.org · 11 years ago
  43. e3abb18 WebRtc_Word -> stdint in audio_coding/g711/ by pbos@webrtc.org · 11 years ago
  44. 48ec040 Remove incorrect asserts. by stefan@webrtc.org · 11 years ago
  45. 326becd WebRtc_Word -> stdint in audio_coding/cng/ by pbos@webrtc.org · 11 years ago
  46. c226567 Fix -Wstring-conversion warnings. by wu@webrtc.org · 11 years ago
  47. f99f63f Thread safety issue fix in incoming_video_stream.cc. See issue 1465. by vikasmarwaha@webrtc.org · 11 years ago
  48. d579a2a Account for header inside I420Encoder::InitEncode. by pbos@webrtc.org · 11 years ago
  49. 06d1e8f Follow-up fix for r3681. by stefan@webrtc.org · 11 years ago
  50. 6f1f826 Fixed initialization of SPL in echo_control_mobile. by kma@webrtc.org · 11 years ago
  51. aef22a7 Android: rename android_build_type gyp variable. by wjia@webrtc.org · 11 years ago
  52. 035c96a Updated WebRTC version number to 3.26 by elham@webrtc.org · 11 years ago
  53. ebdc04d Fix framerate sent to account for actually sent frames. by stefan@webrtc.org · 11 years ago
  54. 3be5a98 Change VCM interface to take target bitrate in bits per second. by stefan@webrtc.org · 11 years ago
  55. a2e9124 Generic video-codec support. by pbos@webrtc.org · 11 years ago
  56. 857be46 Revert the deletion of test_api_nack.cc in r3674. by stefan@webrtc.org · 11 years ago
  57. 1f71c06 Truncated delay quality to avoid negative return values by bjornv@webrtc.org · 11 years ago
  58. 072c9b6 Adding RTX on source by mikhal@webrtc.org · 11 years ago
  59. feaa409 Adding Opus frame length test by tina.legrand@webrtc.org · 11 years ago
  60. 897e86f Fixed a crash issue in NSX module. by kma@webrtc.org · 11 years ago
  61. 9a7b9f7 Revert r3667 and r3665 Review URL: https://webrtc-codereview.appspot.com/1199004 by pwestin@webrtc.org · 11 years ago
  62. a891566 Added destructors for tests to control destruct order by pwestin@webrtc.org · 11 years ago
  63. 25023aa Increasing size of nack list in buffered mode. by mikhal@webrtc.org · 11 years ago
  64. 66ccc6e Removed the engine API:s related to transport such as SetSendDestination, the functionality is now provided via the test frame work. by pwestin@webrtc.org · 11 years ago
  65. 9fe8f22 Refactor webrtc specific Event implementation to an EventFactory. by stefan@webrtc.org · 11 years ago
  66. d4caede Remove DTMF detection. Talk team has been in the loop and there is no need for by turaj@webrtc.org · 11 years ago
  67. b661eae Tool found: pass by value when pass by reference is better in system wrapper unit test. by henrike@webrtc.org · 11 years ago
  68. a284f1d Change intrinsic code in isac fix to let it pass chrome clang compiler. by kma@webrtc.org · 11 years ago
  69. 2c62fd9 Fixes issue detected by tool. by henrike@webrtc.org · 11 years ago
  70. 020e6ad Removed redundant VP8 width/height and made sure the generic width/height is set. by stefan@webrtc.org · 11 years ago
  71. 9b78141 Revert "Internal clean up: removing unused include line." by dwkang@webrtc.org · 11 years ago
  72. 1a8d06e Internal clean up: removing unused include line. by dwkang@webrtc.org · 11 years ago
  73. 2d2bfb0 Fixed issue 1497 in iSAC fixed point. by kma@webrtc.org · 11 years ago
  74. 38a5679 Fix frame_editing_unittest reference file handling. by kjellander@webrtc.org · 11 years ago
  75. c51b060 Optimized EstCodeLpcCoef() for iSAC with intrinsics in Android-Neon platform. by kma@webrtc.org · 11 years ago
  76. 2a3949f Lazy capture_device_info acquisition. by pbos@webrtc.org · 11 years ago
  77. 46672bb Refactor barcode decoder to use Zxing's C++ version by kjellander@webrtc.org · 11 years ago
  78. c96125a Splitting out video_coding_test executable again. by kjellander@webrtc.org · 11 years ago
  79. ad807de Fixed an assembly code error in AECM for ARMv7. by kma@webrtc.org · 11 years ago
  80. 2dbb66b Disable frame dropper for screenshare mode. by stefan@webrtc.org · 11 years ago
  81. 2a070a5 Move video_coding OWNERS to video_coding/. by stefan@webrtc.org · 11 years ago
  82. 2733e12 Fixed a ton of Python lint errors, enabled python lint checking. by phoglund@webrtc.org · 11 years ago
  83. 4621446 Fix debug file buffer bug introduced in r3574. by andrew@webrtc.org · 11 years ago
  84. ace0823 Enabling bufffering mode with no sync module or VoE by mikhal@webrtc.org · 11 years ago
  85. 66b0c5d Remove the error return on SetAGC failure introduced by r3605. by andrew@webrtc.org · 11 years ago
  86. ad3fd52 1. Updated test pages to include Chrome Frame meta tag by elham@webrtc.org · 11 years ago
  87. 333987b Adds new AEC API to audio_processing. by bjornv@webrtc.org · 11 years ago
  88. 08b9b59 Fix for build error on android introduced with r3609. by stefan@webrtc.org · 11 years ago
  89. 2654c43 Split the NACK list into multiple RTCPs if it's too big. by stefan@webrtc.org · 11 years ago
  90. df1cfd1 None of the clients of VoE use SetNetEQBGNMode(), furthermore, NetEq 4 does not provide an API to change the mode of the background noise. by turaj@webrtc.org · 11 years ago
  91. 6316d17 Expose the capture-side AudioProcessing object and allow it to be injected. by andrew@webrtc.org · 11 years ago
  92. 2c1f9d4 AEC Refactoring: Removes lint warning by bjornv@webrtc.org · 11 years ago
  93. 87d8f2d Updated version number to 3.25 by elham@webrtc.org · 11 years ago
  94. eeaacdb Refactor NACK list creation to build the NACK list as packets arrive. by stefan@webrtc.org · 11 years ago
  95. 552f230 compile fix for get_nprocs() with uClibc by phoglund@webrtc.org · 11 years ago
  96. 89cc166 Fixed coverity defects (CID 14657 and 14656). by phoglund@webrtc.org · 11 years ago
  97. 4aa2314 VideoCaptureAndroid can now capture just buffers without also rendering to a SurfaceView. by fischman@webrtc.org · 11 years ago
  98. 7d2689d Don't upsample the capture signal early. by andrew@webrtc.org · 11 years ago
  99. 3da576e Update integration tests for idempotent RTP header settings. by bemasc@google.com · 11 years ago
  100. 13a186f Refactored inline assembly code in complex_fft.c, by combining the individual __asm lines into a single block, to avoid potential register usage problems when building with different tools. by kma@webrtc.org · 11 years ago