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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 7d6e2a00356b107f745ff84b738a6f0a94b9b9a2
  1. 7d6e2a0 Remove TEXT(x) for BUILDINFO macros. by pbos@webrtc.org · 11 years ago
  2. 1dba621 Added a config class to ease passing a set of options across webrtc. by andresp@webrtc.org · 11 years ago
  3. e56cf2c Add svn:eol-style back which is lost in r3993 mistakenly. by braveyao@webrtc.org · 11 years ago
  4. 22aedca Revert 3977 BUG=webrtc:1749 by tnakamura@webrtc.org · 11 years ago
  5. 4de065d Reverting r3978 by elham@webrtc.org · 11 years ago
  6. 15c0af4 This is the first step to convert building the Android WebRTC demo to a proper GYP target, android ndk toolchains is being used to build the jni cpp files instead of using ndk-build. by fischman@webrtc.org · 11 years ago
  7. 238aa38 Use 2 threads for HD, or 1 for VGA or less. by fbarchard@google.com · 11 years ago
  8. f22bfed Since the layout of the Android WebRTC demo application is fixed, if we start the demo application in portrait postion, the activity will be destroyed and then created again, force the demo application to start in landscape position to avoid activity re-creation. by fischman@webrtc.org · 11 years ago
  9. af89cc3 WebRTCDemo Android doesn't hangle activity recreation correctly. by braveyao@webrtc.org · 11 years ago
  10. 41b55fa Drop Virtual webcam check script as moved into buildbot scripts. by kjellander@webrtc.org · 11 years ago
  11. 4afe86b Add fischman into OWNERS of WebRTCDemo Android. by braveyao@webrtc.org · 11 years ago
  12. 7ab7268 Fix compile errors in ViE with latest clang. by andrew@webrtc.org · 11 years ago
  13. 5dea86a Update SincResampler with the latest Chromium code. by andrew@webrtc.org · 11 years ago
  14. 90f05ed Clean creation of VideoEngine: by andresp@webrtc.org · 11 years ago
  15. a149ea3 Formatted dtmf_queue. by phoglund@webrtc.org · 11 years ago
  16. 4be3afb Add script to ensure virtual webcam is running. by kjellander@webrtc.org · 11 years ago
  17. d7ebd68 Disable clang C++11 warnings to permit OVERRIDE keyword. by pbos@webrtc.org · 11 years ago
  18. 9ccfe46 Fix VCMProcessTimer::TimeUntilProcess() unsigned-integer underflow problem. by stefan@webrtc.org · 11 years ago
  19. acdfffb Enable protobuf use in Chromium. by andrew@webrtc.org · 11 years ago
  20. bfa5ee2 Update protoc.gypi to match Chromium's latest. by andrew@webrtc.org · 11 years ago
  21. 28832e1 Refactoring for typing detection by niklas.enbom@webrtc.org · 11 years ago
  22. 06ad384 Trigger a PLI if the duration of non-decodable frames exceeds a threshold. by stefan@webrtc.org · 11 years ago
  23. 38fb7b0 VCM/Receiver: Return null when can't extract frame. by mikhal@webrtc.org · 11 years ago
  24. 957f938 Relanding 3962: VCM/JB: Porting jitter_buffer_test to gtest by mikhal@webrtc.org · 11 years ago
  25. 933f885 Relanding r3952: VCM: Updating receiver logic BUG=r1734 R=stefan@webrtc.org by mikhal@webrtc.org · 11 years ago
  26. 2fc86ee VCM/JB: Break and skip to key if possible by mikhal@webrtc.org · 11 years ago
  27. 98b2011 Fix clang errors in non-GYP_DEFINES=clang=1 build by pbos@webrtc.org · 11 years ago
  28. 7bdebfd Fix jitter buffer unittest. by stefan@webrtc.org · 11 years ago
  29. 73631c9 Correctly add packets to nack list when sequence number wraps. by stefan@webrtc.org · 11 years ago
  30. 49e9c6c Fix crash in pacer. by pwestin@webrtc.org · 11 years ago
  31. 20eb558 Revert r3952 "VCM: Updating receiver logic" by stefan@webrtc.org · 11 years ago
  32. a9fc587 Revert r3956 "VCM/JB: Porting jitter_buffer_test to gtest." by stefan@webrtc.org · 11 years ago
  33. 000fca3 Landing 1399004, Minor clean up on the un-used _measureDelay code by xians@webrtc.org · 11 years ago
  34. 02ae32e Add an option to override the TestToStderr trace printout time. by andrew@webrtc.org · 11 years ago
  35. a0975ed Consolidate all third party licenses in LICENSE_THIRD_PARTY. by andrew@webrtc.org · 11 years ago
  36. b6c7447 Updated WebRTC version number to 3.30 by elham@webrtc.org · 11 years ago
  37. 9215af6 VCM/JB: Porting jitter_buffer_test to gtest. by mikhal@webrtc.org · 11 years ago
  38. 9e4ad86 Remove 44.1 kHz workaround from AudioDevice on PulseAudio. by andrew@webrtc.org · 11 years ago
  39. 8cfa495 Remove 44.1 kHz workaround from AudioDevice on WASAPI. by andrew@webrtc.org · 11 years ago
  40. 27f61e2 Fix off-by-one buffer overflow in WebRtcNetEQ_PacketBufferInsert(). by sergeyu@chromium.org · 11 years ago
  41. b842af8 VCM: Updating receiver logic by mikhal@webrtc.org · 11 years ago
  42. 20f81fe Correct and update dir name by leozwang@webrtc.org · 11 years ago
  43. 52b2ee5 Revert 3934 "Revert 3933 "Remove traces of deprecated WebRtc_Wor..." by pbos@webrtc.org · 11 years ago
  44. 3a3a787 Get rid of some unnecessary copying when sending REMBs. by solenberg@webrtc.org · 11 years ago
  45. b31332e Formatting ACM tests by tina.legrand@webrtc.org · 11 years ago
  46. 49d151e Fix when SetMinimumPlayoutDelay is configured to 0 by pwestin@webrtc.org · 11 years ago
  47. 677ca88 Removing bad code resulting in flaky test. by pwestin@webrtc.org · 11 years ago
  48. 36bdba4 Adding trace and changing pacing constants by pwestin@webrtc.org · 11 years ago
  49. 24de1a0 Update third party license file by niklas.enbom@webrtc.org · 11 years ago
  50. 422a124 Bugfix custom call stop. by pwestin@webrtc.org · 11 years ago
  51. c0fc487 Allow voe_cmd_test to select Opus mono (now the default). by andrew@webrtc.org · 11 years ago
  52. ad9cee8 Relax VoE's max packet length threshold. by andrew@webrtc.org · 11 years ago
  53. 9e0d9a6 Disabled flaky test. by phoglund@webrtc.org · 11 years ago
  54. 2d6f0df Revert 3933 "Remove traces of deprecated WebRtc_Word types." by pbos@webrtc.org · 11 years ago
  55. e422d12 Remove traces of deprecated WebRtc_Word types. by pbos@webrtc.org · 11 years ago
  56. be6ad88 WebRTCDemo Android app to route audio to headphone when it's plugged in. by braveyao@webrtc.org · 11 years ago
  57. 4a68e95 Replace Resampler with PushResampler in transmit_mixer. by andrew@webrtc.org · 11 years ago
  58. 166153e Consolidate common_audio into a single target. by andrew@webrtc.org · 11 years ago
  59. e9c34ba Add AEC suppression level option to audioproc. by andrew@webrtc.org · 11 years ago
  60. 22b72cb Move WEBRTC_THREAD_RR and WEBRTC_CLOCK_TYPE_REALTIME to system_wrappers.gypi . by sergeyu@chromium.org · 11 years ago
  61. 9137e98 Fixes two bugs in receive statistics. by stefan@webrtc.org · 11 years ago
  62. 4d4ba47 Fixing AV sync. Increased 2 const to allow for a bigger difference in AV sync. by pwestin@webrtc.org · 11 years ago
  63. 2725aa0 VCM: Setting buffering delay in timing by mikhal@webrtc.org · 11 years ago
  64. 6ed1c8c Adding buffered mode to loopback test by mikhal@webrtc.org · 11 years ago
  65. 5f09bcc Apply Chromium C++ style to RemoteRateControl. by solenberg@webrtc.org · 11 years ago
  66. 85a1dab Add DesktopCapturer interface for desktop capturers. by sergeyu@chromium.org · 11 years ago
  67. d60f7a9 Don't reset the last je value and mode by mikhal@webrtc.org · 11 years ago
  68. b6fadb1 Add a wrapper around PushSincResampler and the old Resampler. by andrew@webrtc.org · 11 years ago
  69. c12e655 Fix two issues where we might end up busy looping in decoder_render mode. by stefan@webrtc.org · 11 years ago
  70. b8f1cf3 Enable Nack pacing. by pwestin@webrtc.org · 11 years ago
  71. 1a58dd7 Removing vie file related code from vie_custom_call by mikhal@webrtc.org · 11 years ago
  72. b13f394 Fixed remaining nits from Stefan by pwestin@webrtc.org · 11 years ago
  73. dea537b Add a push-based wrapper around SincResampler. by andrew@webrtc.org · 11 years ago
  74. 3e20f91 Add comfort noise disabling and routing mode selection to audioproc. by andrew@webrtc.org · 11 years ago
  75. 05c25a7 Removing another instance of file api by mikhal@webrtc.org · 11 years ago
  76. 3816c52 Fix the encoder pause logic. BUG=1691 by pwestin@webrtc.org · 11 years ago
  77. 40bd744 VCM: Adding API for the size(duration) of the jitter buffer. by mikhal@webrtc.org · 11 years ago
  78. b47b7e0 VCM/JB: Using last decoded state for waiting for key by mikhal@webrtc.org · 11 years ago
  79. 1886a04 VCM/JB: FrameForDecoding->IncompleteFrameForDecoding by mikhal@webrtc.org · 11 years ago
  80. 4eac481 Disabling avi file interface by mikhal@webrtc.org · 11 years ago
  81. e4ae7a2 Avoid adding duplicates in pacer lists. by pwestin@webrtc.org · 11 years ago
  82. 2dfffc3 Make sure timestamps are monotonically increasing. by stefan@webrtc.org · 11 years ago
  83. 8f97f02 Revert 3892 "VCM/JB: Using last decoded state for waiting for key" by andrew@webrtc.org · 11 years ago
  84. f292306 Adding extra options to interact with external encoder/decoder. by andresp@webrtc.org · 11 years ago
  85. a7be357 VCM/JB: Using last decoded state for waiting for key by mikhal@webrtc.org · 11 years ago
  86. d430f32 Revert "Add a default RTT to CallStats and use different values for buffered/real-time mode." by stefan@webrtc.org · 11 years ago
  87. 292ed1d Buf fix for r3883. by turaj@webrtc.org · 11 years ago
  88. 2788107 Add a default RTT to CallStats and use different values for buffered/real-time mode. by stefan@webrtc.org · 11 years ago
  89. 9ded0b1 VP8: Avoid copying the codec struct on Reset(). by pbos@webrtc.org · 11 years ago
  90. 0dae366 BUG=1351 by mflodman@webrtc.org · 11 years ago
  91. 4123abf VCM/JB: Skip to the next complete key frame by mikhal@webrtc.org · 11 years ago
  92. f13f1fc Updated the sync module with a slow moving filter by pwestin@webrtc.org · 11 years ago
  93. 92aa25b Improve AV-sync when initial delay is set and NetEq has long buffer. by turaj@webrtc.org · 11 years ago
  94. 6cb19e1 emove desktop_capture.gypi from modules.gyp by kjellander@webrtc.org · 11 years ago
  95. c11933f Removed unused variable. by mflodman@webrtc.org · 11 years ago
  96. bea854a Fixing Coverity issues. by mflodman@webrtc.org · 11 years ago
  97. a788a4d Update iOS build script to run on bots. by kjellander@webrtc.org · 11 years ago
  98. e07ec09 Revert 3876 by mikhal@webrtc.org · 11 years ago
  99. c2c65ba VCM/Receiver: Only update render time when decoding by mikhal@webrtc.org · 11 years ago
  100. b6e175d Ensure build_demo.py run subprocesses with bash shell. by kjellander@webrtc.org · 11 years ago