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gerrit-public.fairphone.software / platform / external / webrtc / 2f44673d665899ca788ae44247a9a7f4764f5e2b
  1. 2f44673 WebRtc_Word32 => int32_t for rtp_rtcp/ by pbos@webrtc.org · 11 years ago
  2. 367804c Clean packets on the network when closing + made loopback test actually run again. by mflodman@webrtc.org · 11 years ago
  3. ff7e130 WebRtc_Word32 => int32_t remote_bitrate_estimator/ by pbos@webrtc.org · 11 years ago
  4. 37bf584 Show stats from both sides by hta@webrtc.org · 11 years ago
  5. 222e994 Migrating Apprtc to use new TURN service which supports time-limited TURN credentials. by vikasmarwaha@webrtc.org · 11 years ago
  6. 123b618 Fix a crash issue on WinXP where LoadLibrary(TEXT("Kernel32.dll")) may fail. by wu@webrtc.org · 11 years ago
  7. 2e6b7e9 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
  8. 19da719 Resolves TSan v2 reports data races in voe_auto_test. by henrika@webrtc.org · 11 years ago
  9. 10eb920 Add GYP target for WebRTC Video demo for Android. by kjellander@webrtc.org · 11 years ago
  10. b5bf54c Permit arbitrary payload names for kVideoCodecGeneric. by pbos@webrtc.org · 11 years ago
  11. b9e402d Remove WEBRTC_*_ENGINE_NETWORK_API use by pwestin@webrtc.org · 11 years ago
  12. 79b0289 Adds event traces and counters for WebRTC receive side. by edjee@google.com · 11 years ago
  13. 835dbf4 Fix no received audio in tests. by pwestin@webrtc.org · 11 years ago
  14. aa527bb Disabling MixingTests due to race conditions. by henrika@webrtc.org · 11 years ago
  15. fcb7c38 Two more sleep calls converted to use SleepMs(). by hta@webrtc.org · 11 years ago
  16. bb8ada6 TSan v2 reports data races in WebRTCAudioDeviceTest.FullDuplexAudioWithAGC by henrika@webrtc.org · 11 years ago
  17. 0c45957 Remove UDP transport API from VoE by pwestin@webrtc.org · 11 years ago
  18. 0746ce1 Fixes memory leak in AudioLevel class reported by memory try bots. by henrika@webrtc.org · 11 years ago
  19. d108a46 Fixes data race in WebRTCAudioDeviceTest.StartRecording reported by ThreadSanitizer by henrika@webrtc.org · 11 years ago
  20. 82dcc9f Remove UDP transport API from ViE by pwestin@webrtc.org · 11 years ago
  21. 7b859cc Webrtc_Word32 => int32_t in video_coding/main/ by pbos@webrtc.org · 11 years ago
  22. cfc07c9 Revert of r3747. by henrike@webrtc.org · 11 years ago
  23. 95d8873 Two more sleep calls converted to use SleepMs(). by hta@webrtc.org · 11 years ago
  24. 4ff956f Fixes data race in WebRTCAudioDeviceTest.Construct reported by ThreadSanitizer by henrika@webrtc.org · 11 years ago
  25. 46e626d Fix gflags compile error on x86 Android by kjellander@webrtc.org · 11 years ago
  26. f81fad6 Fix opus bitrate truncated to 16-bit int. This prevented setting bitrates higher by justinlin@chromium.org · 11 years ago
  27. 747c4cc 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
  28. 65243bd Updated Webrtc version to 3.28 by elham@webrtc.org · 11 years ago
  29. 7f6b7cb Revert r3743. by marpan@webrtc.org · 11 years ago
  30. e882a47 Roll libvpx to 191157. -Pick up the libvpx roll to 8015a9ae. by marpan@webrtc.org · 11 years ago
  31. 29f34b8 Fix for issue: https://code.google.com/p/webrtc/issues/detail?id=1549 by marpan@webrtc.org · 11 years ago
  32. 626c663 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
  33. 93bea51 Removed CPU APIs from VoEHardware. Code is now only used by test applications. by henrike@webrtc.org · 11 years ago
  34. a442d4d 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
  35. 80fccc2 Revert 3736 "Removed CPU APIs from VoEHardware. Code is now only..." by wu@webrtc.org · 11 years ago
  36. 4c138e8 Removed CPU APIs from VoEHardware. Code is now only used by test applications. by henrike@webrtc.org · 11 years ago
  37. 458194b Fix broken audio. by leozwang@webrtc.org · 11 years ago
  38. 4b1cd5c G722-stereo has been missing when creating AudioDecoder. by turaj@webrtc.org · 11 years ago
  39. 4d06db5 NetEq4 fails if the first packets inserted in are out-of-band DTMFs. by turaj@webrtc.org · 11 years ago
  40. e1a7193 Fix flakiness in network up/down event tests when running under memcheck. by stefan@webrtc.org · 11 years ago
  41. add50b9 WebRTCDemo: remove unnecessary stop & start during orientation change which isn't necessary since API v14. by fischman@webrtc.org · 11 years ago
  42. bfacda6 Add interface to signal a network down event. by stefan@webrtc.org · 11 years ago
  43. 686001d 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
  44. 1b31c78 Remove VoE's default call in Trace::SetLevelFilter. by andrew@webrtc.org · 11 years ago
  45. d8a6e72 Fix potential buffer overrun when checking if a packet is RTCP. Also makes validation slightly more robust. by solenberg@webrtc.org · 11 years ago
  46. 0633ccc Alphabetize include order in fake_voe_external_media.h. by andrew@webrtc.org · 11 years ago
  47. 0e3077a Restart Android capture after orientation change. Also prevent an NPE on exit. by fischman@webrtc.org · 11 years ago
  48. c83a00a Add some VoE and AudioProcessing mocks. by andrew@webrtc.org · 11 years ago
  49. b87cc85 Refactor unittest trace printouts to a separate class. by andrew@webrtc.org · 11 years ago
  50. b4c441a Enable the below APIs for iOS. by sjlee@webrtc.org · 11 years ago
  51. 7b48ced libyuv r618 roll. Includes new psnr tool for evaluating codec quality. by fbarchard@google.com · 11 years ago
  52. db41856 Introduced pause and resume to the pacer by pwestin@webrtc.org · 11 years ago
  53. 14c9909 Updated WebRTC version to 3.27 by elham@webrtc.org · 11 years ago
  54. a078d5c Bugfix for extended RTP/RTCP test by pwestin@webrtc.org · 11 years ago
  55. 26e35e1 Move the VIE tests to use external transport instead of the built in udp transport by pwestin@webrtc.org · 11 years ago
  56. c1ffd33 Add trace printouts to all unit tests. by andrew@webrtc.org · 11 years ago
  57. 94bc4cf Add min and target bitrate to VideoCodec. by marpan@webrtc.org · 11 years ago
  58. e308239 Move the VoE tests to use external transport instead of the built in udp transport by pwestin@webrtc.org · 11 years ago
  59. e86f43b Roll Opus 1.0.2 by tina.legrand@webrtc.org · 11 years ago
  60. 3ed599a Bandwidth stats display in constraints-and-stats. by hta@webrtc.org · 11 years ago
  61. 999e900 Creating a copy of Udp transport under webrtc/test by pwestin@webrtc.org · 11 years ago
  62. 2cec0b1 Cleanup nanosleep -> SleepMs Remove some leftover stuff by hta@webrtc.org · 11 years ago
  63. ae4e2b3 WebRtc_Word -> stdint in audio_coding/g711/ by pbos@webrtc.org · 11 years ago
  64. 836af79 Remove incorrect asserts. by stefan@webrtc.org · 11 years ago
  65. 01b507a WebRtc_Word -> stdint in audio_coding/cng/ by pbos@webrtc.org · 11 years ago
  66. af33b62 Fix -Wstring-conversion warnings. by wu@webrtc.org · 11 years ago
  67. 455370d Thread safety issue fix in incoming_video_stream.cc. See issue 1465. by vikasmarwaha@webrtc.org · 11 years ago
  68. f354e1f Add audio/video only option in apprtc by braveyao@webrtc.org · 11 years ago
  69. ebf49da Url option to change the resolution. by vikasmarwaha@webrtc.org · 11 years ago
  70. 8685090 Account for header inside I420Encoder::InitEncode. by pbos@webrtc.org · 11 years ago
  71. 3d0b0d6 Follow-up fix for r3681. by stefan@webrtc.org · 11 years ago
  72. ecfd328 Changed stats reporting to not use local/remote by hta@webrtc.org · 11 years ago
  73. 31829a7 Fixed initialization of SPL in echo_control_mobile. by kma@webrtc.org · 11 years ago
  74. 95a8ddd Android: rename android_build_type gyp variable. by wjia@webrtc.org · 11 years ago
  75. f1ea0df Updated WebRTC version number to 3.26 by elham@webrtc.org · 11 years ago
  76. f4944d4 Fix framerate sent to account for actually sent frames. by stefan@webrtc.org · 11 years ago
  77. abc9d5b Change VCM interface to take target bitrate in bits per second. by stefan@webrtc.org · 11 years ago
  78. 8911ce4 Generic video-codec support. by pbos@webrtc.org · 11 years ago
  79. 3cb42b1 Remove GCC 4.6 bot from LKGR parsing. by kjellander@webrtc.org · 11 years ago
  80. 71335ce Have git ignore ".swp" files. by pbos@webrtc.org · 11 years ago
  81. 4121146 Revert the deletion of test_api_nack.cc in r3674. by stefan@webrtc.org · 11 years ago
  82. 04ecd49 Truncated delay quality to avoid negative return values by bjornv@webrtc.org · 11 years ago
  83. bda7f30 Adding RTX on source by mikhal@webrtc.org · 11 years ago
  84. 73222cf Adding Opus frame length test by tina.legrand@webrtc.org · 11 years ago
  85. d613c20 Adding new directories and watchers to the WATCHLISTS. by stefan@webrtc.org · 11 years ago
  86. eddc5a6 Updated local-audio-rendering.html to remove unmute. by vikasmarwaha@webrtc.org · 11 years ago
  87. 33f22d0 Fixed a crash issue in NSX module. by kma@webrtc.org · 11 years ago
  88. 684f057 Revert r3667 and r3665 Review URL: https://webrtc-codereview.appspot.com/1199004 by pwestin@webrtc.org · 11 years ago
  89. 2dc0367 Added destructors for tests to control destruct order by pwestin@webrtc.org · 11 years ago
  90. 15960c2 Increasing size of nack list in buffered mode. by mikhal@webrtc.org · 11 years ago
  91. 361bac7 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
  92. 2baf5f5 Refactor webrtc specific Event implementation to an EventFactory. by stefan@webrtc.org · 11 years ago
  93. b7edd06 Remove DTMF detection. Talk team has been in the loop and there is no need for by turaj@webrtc.org · 11 years ago
  94. 728b7ea Tool found: pass by value when pass by reference is better in system wrapper unit test. by henrike@webrtc.org · 11 years ago
  95. d6cd64a Change intrinsic code in isac fix to let it pass chrome clang compiler. by kma@webrtc.org · 11 years ago
  96. 23875c1 Fixes issue detected by tool. by henrike@webrtc.org · 11 years ago
  97. 6ddb907 Corrected dashboard script error. by phoglund@webrtc.org · 11 years ago
  98. 03e3117 Removed redundant VP8 width/height and made sure the generic width/height is set. by stefan@webrtc.org · 11 years ago
  99. 7473f89 Revert "Internal clean up: removing unused include line." by dwkang@webrtc.org · 11 years ago
  100. 25316b2 Internal clean up: removing unused include line. by dwkang@webrtc.org · 11 years ago