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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 73ebe67b3e6f0a65efed02efd4eee4dfb1b7729e
  1. 73ebe67 Revert "With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps." by stefan@webrtc.org · 11 years ago
  2. 67879bc WebRtc_Word32 -> int32_t in video_engine/ by pbos@webrtc.org · 11 years ago
  3. c57ef84 WebRtc_Word32 -> int32_t in video_processing/ by pbos@webrtc.org · 11 years ago
  4. 65deb26 With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps. by stefan@webrtc.org · 11 years ago
  5. a97bf1c WebRtc_Word32 -> int32_t in common_video. by pbos@webrtc.org · 11 years ago
  6. f85a509 WebRtc_Word32 -> int32_t in utility/ by pbos@webrtc.org · 11 years ago
  7. 283c29a WebRtc_Word32 -> int32_t in media_file/ by pbos@webrtc.org · 11 years ago
  8. 5d9a1bc Fixing the flakiness of ThreadWakesTwice. by hta@webrtc.org · 11 years ago
  9. 91cab71 WebRtc_Word32 -> int32_t in test/ by pbos@webrtc.org · 11 years ago
  10. 64a144f WebRtc_Word32 -> int32_t in audio_device/ by pbos@webrtc.org · 11 years ago
  11. 54f03bc WebRtc_Word32 -> int32_t in voice_engine/ by pbos@webrtc.org · 11 years ago
  12. c0231af WebRtc_Word32 -> int32_t in system_wrappers by pbos@webrtc.org · 11 years ago
  13. 208a648 Always set render delay in ViEChannel::RegisterExternalDecoder. by pbos@webrtc.org · 11 years ago
  14. fbda0fc WebRtc_Word32 => int32_t etc. in audio_coding/ by pbos@webrtc.org · 11 years ago
  15. 8ec8955 Remove the old unused udp_transport by pwestin@webrtc.org · 11 years ago
  16. 7deebae Reduce execution time of rate control test. by marpan@webrtc.org · 11 years ago
  17. 715275c Fixed a bug in isac-fix's entropy coding function: out of bounds acces to array. by kma@webrtc.org · 11 years ago
  18. 48c4b75 WebRtc_Word32 => int32_t in video_coding/ by pbos@webrtc.org · 11 years ago
  19. b57da65 WebRtc_Word32 => int32_t for rtp_rtcp/ by pbos@webrtc.org · 11 years ago
  20. a9f28d5 Clean packets on the network when closing + made loopback test actually run again. by mflodman@webrtc.org · 11 years ago
  21. 63a1ebd WebRtc_Word32 => int32_t remote_bitrate_estimator/ by pbos@webrtc.org · 11 years ago
  22. 14e22dd Fix a crash issue on WinXP where LoadLibrary(TEXT("Kernel32.dll")) may fail. by wu@webrtc.org · 11 years ago
  23. a2576cf In streaming mode it is preferable to fade to silence when sender stops sending, or long period of packet loss. by turaj@webrtc.org · 11 years ago
  24. 1d25eac Resolves TSan v2 reports data races in voe_auto_test. by henrika@webrtc.org · 11 years ago
  25. 004f462 Add GYP target for WebRTC Video demo for Android. by kjellander@webrtc.org · 11 years ago
  26. 2ed1cd9 Permit arbitrary payload names for kVideoCodecGeneric. by pbos@webrtc.org · 11 years ago
  27. ef91cbf Remove WEBRTC_*_ENGINE_NETWORK_API use by pwestin@webrtc.org · 11 years ago
  28. dded206 Adds event traces and counters for WebRTC receive side. by edjee@google.com · 11 years ago
  29. c39749a Fix no received audio in tests. by pwestin@webrtc.org · 11 years ago
  30. 84423e9 Disabling MixingTests due to race conditions. by henrika@webrtc.org · 11 years ago
  31. c9f8871 Two more sleep calls converted to use SleepMs(). by hta@webrtc.org · 11 years ago
  32. 45ce6a8 TSan v2 reports data races in WebRTCAudioDeviceTest.FullDuplexAudioWithAGC by henrika@webrtc.org · 11 years ago
  33. e493218 Remove UDP transport API from VoE by pwestin@webrtc.org · 11 years ago
  34. 9e8a401 Fixes memory leak in AudioLevel class reported by memory try bots. by henrika@webrtc.org · 11 years ago
  35. c4efe71 Fixes data race in WebRTCAudioDeviceTest.StartRecording reported by ThreadSanitizer by henrika@webrtc.org · 11 years ago
  36. 065b64d Remove UDP transport API from ViE by pwestin@webrtc.org · 11 years ago
  37. dba5f45 Webrtc_Word32 => int32_t in video_coding/main/ by pbos@webrtc.org · 11 years ago
  38. 7873061 Revert of r3747. by henrike@webrtc.org · 11 years ago
  39. f535877 Two more sleep calls converted to use SleepMs(). by hta@webrtc.org · 11 years ago
  40. f46e1fa Fixes data race in WebRTCAudioDeviceTest.Construct reported by ThreadSanitizer by henrika@webrtc.org · 11 years ago
  41. b514117 Fix opus bitrate truncated to 16-bit int. This prevented setting bitrates higher by justinlin@chromium.org · 11 years ago
  42. a4b78ff 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
  43. f3ac3ba Updated Webrtc version to 3.28 by elham@webrtc.org · 11 years ago
  44. 55742e5 Fix for issue: https://code.google.com/p/webrtc/issues/detail?id=1549 by marpan@webrtc.org · 11 years ago
  45. 46144bb 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
  46. 3b6f728 Removed CPU APIs from VoEHardware. Code is now only used by test applications. by henrike@webrtc.org · 11 years ago
  47. fa2dd22 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
  48. 2ffc8bf Revert 3736 "Removed CPU APIs from VoEHardware. Code is now only..." by wu@webrtc.org · 11 years ago
  49. 365ca40 Removed CPU APIs from VoEHardware. Code is now only used by test applications. by henrike@webrtc.org · 11 years ago
  50. 0c0795e Fix broken audio. by leozwang@webrtc.org · 11 years ago
  51. a0bba27 G722-stereo has been missing when creating AudioDecoder. by turaj@webrtc.org · 11 years ago
  52. 88a7940 NetEq4 fails if the first packets inserted in are out-of-band DTMFs. by turaj@webrtc.org · 11 years ago
  53. 88f12ab Fix flakiness in network up/down event tests when running under memcheck. by stefan@webrtc.org · 11 years ago
  54. 6fc5215 WebRTCDemo: remove unnecessary stop & start during orientation change which isn't necessary since API v14. by fischman@webrtc.org · 11 years ago
  55. dca71b2 Add interface to signal a network down event. by stefan@webrtc.org · 11 years ago
  56. 6e34ceb 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
  57. f386e2b Remove VoE's default call in Trace::SetLevelFilter. by andrew@webrtc.org · 11 years ago
  58. aa0fcd7 Fix potential buffer overrun when checking if a packet is RTCP. Also makes validation slightly more robust. by solenberg@webrtc.org · 11 years ago
  59. 31b4448 Alphabetize include order in fake_voe_external_media.h. by andrew@webrtc.org · 11 years ago
  60. a7a643e Restart Android capture after orientation change. Also prevent an NPE on exit. by fischman@webrtc.org · 11 years ago
  61. 13f66d1 Add some VoE and AudioProcessing mocks. by andrew@webrtc.org · 11 years ago
  62. 8826e34 Refactor unittest trace printouts to a separate class. by andrew@webrtc.org · 11 years ago
  63. 0c1f10b Enable the below APIs for iOS. by sjlee@webrtc.org · 11 years ago
  64. 9522792 Introduced pause and resume to the pacer by pwestin@webrtc.org · 11 years ago
  65. f49577f Updated WebRTC version to 3.27 by elham@webrtc.org · 11 years ago
  66. 7fd368f Bugfix for extended RTP/RTCP test by pwestin@webrtc.org · 11 years ago
  67. c075e25 Move the VIE tests to use external transport instead of the built in udp transport by pwestin@webrtc.org · 11 years ago
  68. 4c27c03 Add trace printouts to all unit tests. by andrew@webrtc.org · 11 years ago
  69. e1198e6 Add min and target bitrate to VideoCodec. by marpan@webrtc.org · 11 years ago
  70. aa922de Move the VoE tests to use external transport instead of the built in udp transport by pwestin@webrtc.org · 11 years ago
  71. af6aa7b Creating a copy of Udp transport under webrtc/test by pwestin@webrtc.org · 11 years ago
  72. 495c563 Cleanup nanosleep -> SleepMs Remove some leftover stuff by hta@webrtc.org · 11 years ago
  73. 4a48fd6 WebRtc_Word -> stdint in audio_coding/g711/ by pbos@webrtc.org · 11 years ago
  74. a2df078 Remove incorrect asserts. by stefan@webrtc.org · 11 years ago
  75. e49f252 WebRtc_Word -> stdint in audio_coding/cng/ by pbos@webrtc.org · 11 years ago
  76. 0f2782f Fix -Wstring-conversion warnings. by wu@webrtc.org · 11 years ago
  77. 5815b7c Thread safety issue fix in incoming_video_stream.cc. See issue 1465. by vikasmarwaha@webrtc.org · 11 years ago
  78. 757bf0f Account for header inside I420Encoder::InitEncode. by pbos@webrtc.org · 11 years ago
  79. 6313692 Follow-up fix for r3681. by stefan@webrtc.org · 11 years ago
  80. 46618db Fixed initialization of SPL in echo_control_mobile. by kma@webrtc.org · 11 years ago
  81. 9cd73ed Android: rename android_build_type gyp variable. by wjia@webrtc.org · 11 years ago
  82. ffe2ec6 Updated WebRTC version number to 3.26 by elham@webrtc.org · 11 years ago
  83. 692c69d Fix framerate sent to account for actually sent frames. by stefan@webrtc.org · 11 years ago
  84. 72e204a Change VCM interface to take target bitrate in bits per second. by stefan@webrtc.org · 11 years ago
  85. e3339fc Generic video-codec support. by pbos@webrtc.org · 11 years ago
  86. 0b5c7f1 Revert the deletion of test_api_nack.cc in r3674. by stefan@webrtc.org · 11 years ago
  87. 87ef38e Truncated delay quality to avoid negative return values by bjornv@webrtc.org · 11 years ago
  88. 946d240 Adding RTX on source by mikhal@webrtc.org · 11 years ago
  89. cf9ab12 Adding Opus frame length test by tina.legrand@webrtc.org · 11 years ago
  90. aecc559 Fixed a crash issue in NSX module. by kma@webrtc.org · 11 years ago
  91. 912b7f7 Revert r3667 and r3665 Review URL: https://webrtc-codereview.appspot.com/1199004 by pwestin@webrtc.org · 11 years ago
  92. 40749c1 Added destructors for tests to control destruct order by pwestin@webrtc.org · 11 years ago
  93. b793abe Increasing size of nack list in buffered mode. by mikhal@webrtc.org · 11 years ago
  94. 2daec4c 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
  95. 2637d61 Refactor webrtc specific Event implementation to an EventFactory. by stefan@webrtc.org · 11 years ago
  96. 15a03fd Remove DTMF detection. Talk team has been in the loop and there is no need for by turaj@webrtc.org · 11 years ago
  97. df00856 Tool found: pass by value when pass by reference is better in system wrapper unit test. by henrike@webrtc.org · 11 years ago
  98. 0c46af6 Change intrinsic code in isac fix to let it pass chrome clang compiler. by kma@webrtc.org · 11 years ago
  99. 160b327 Fixes issue detected by tool. by henrike@webrtc.org · 11 years ago
  100. f0f1dc2 Removed redundant VP8 width/height and made sure the generic width/height is set. by stefan@webrtc.org · 11 years ago