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webrtc / src / / f5d4cb19589f0b0a3736e4f653dc5a75b9df1ceb
  1. f5d4cb1 Include files from webrtc/.. paths in video_engine/ by pbos@webrtc.org · 12 years ago
  2. 9f557c1 Improve wraparound handling in the render time extrapolator. by stefan@webrtc.org · 12 years ago
  3. 14d7700 Moved command line parsing to internal tools and moved back the mic volume thingie. by phoglund@webrtc.org · 12 years ago
  4. e874a8f Enable WebRTC demo application on x86 Android by fischman@webrtc.org · 12 years ago
  5. 8630cfe Guarding certain operations, e.g. bandwidth estimation, RTCP statistics update etc., not to be run on sync RTPS. by turaj@webrtc.org · 12 years ago
  6. fe307e1 Add one unit test for NACKing a key frame by hclam@chromium.org · 12 years ago
  7. b3e5acf Cleanup traces in WebRTC by hclam@chromium.org · 12 years ago
  8. b9bb3d1 Avoid resetting encoder on identical settings. by pbos@webrtc.org · 12 years ago
  9. 890f609 Bugfix: VCM would report wrong sentBitrate by marpan@webrtc.org · 12 years ago
  10. 9919ad5 Formatted FEC stuff. by phoglund@webrtc.org · 12 years ago
  11. 5c1948d Moved force_volume_max to its own gyp file to avoid a circular dependency. by phoglund@webrtc.org · 12 years ago
  12. 61d3c55 Wrote a small portable tool for forcing the mic volume to 100%. by phoglund@webrtc.org · 12 years ago
  13. 29d5839 New VideoEngine API implementation on top of old one, first steps. by pbos@webrtc.org · 12 years ago
  14. 2038214 Log too long non-decodable duration events. by stefan@webrtc.org · 12 years ago
  15. 4dee309 Remove SetOverUseDetectorOptions and cleaned ViESharedData. by mflodman@webrtc.org · 12 years ago
  16. 7ebbea1 Add handling of the absolute send time header extension to the rtp_rtcp module. by solenberg@webrtc.org · 12 years ago
  17. 59a0667 Updated apprtc demo to interop with firefox. by vikasmarwaha@webrtc.org · 12 years ago
  18. 40298d4 Added webaudio-and-webtrc.html to the demos index.html. by vikasmarwaha@webrtc.org · 12 years ago
  19. 8c2e78b Roll chromium_revision 193311:199267 by fischman@webrtc.org · 12 years ago
  20. 6cfa390 Updating NACK RTX test by mikhal@webrtc.org · 12 years ago
  21. cb20a5b VCM/JB: Bug fix in ExtractAndSetDecode BUG=1771 R=stefan@webrtc.org by mikhal@webrtc.org · 12 years ago
  22. 5add4ad RemoteBitrateEstimatorTest::TestRateIncreaseReordering sent in arrival timestamps in non monotonically increasing order. Fixed. by solenberg@webrtc.org · 12 years ago
  23. c93b1d0 CoreAudio Win: release resources safely under certain rare circumstance in GTalkplugin by braveyao@webrtc.org · 12 years ago
  24. e2a8006 Linux support for typing detection by niklas.enbom@webrtc.org · 12 years ago
  25. 4ce83893 Address sanitizer out of bounds read in iSAC by turaj@webrtc.org · 12 years ago
  26. 6bee05a Remove const for plain data types in common_video/ by pbos@webrtc.org · 12 years ago
  27. 29b2219 Adding a factory to remote bitrate estimator and allow it to be set via config. by andresp@webrtc.org · 12 years ago
  28. 1673481 Fixes a bug where the render buffer size (and indirectly the non-continuous duration) was computed incorrectly. by stefan@webrtc.org · 12 years ago
  29. 736c6f7 Fixed more perf expectations. by phoglund@webrtc.org · 12 years ago
  30. 80c7e3b Adjusted perf expectations for mac large tests. by phoglund@webrtc.org · 12 years ago
  31. bb984f5 Removed Mac capture crash and memory leak. by mflodman@webrtc.org · 12 years ago
  32. a6ff845 Add script for comparing video quality by kjellander@webrtc.org · 12 years ago
  33. 6d07ad9 Added protoc_wrapper to blacklist, fixed tools/PRESUBMIT.py which was passing in the wrong args to CheckLongLines. by phoglund@webrtc.org · 12 years ago
  34. 527f6c6 Reformatted FEC tables. by phoglund@webrtc.org · 12 years ago
  35. 8e3b5948 Remove const for plain data types in common_audio/ by pbos@webrtc.org · 12 years ago
  36. 9213521 Remove const for plain data types in voice_engine/ by pbos@webrtc.org · 12 years ago
  37. 185bae4 Replace ExtraCodecOptions with new Config class that supports multiple settings at once. by andresp@webrtc.org · 12 years ago
  38. c9cb4ff Fix typo in log statement. witdh should be width. by fbarchard@google.com · 12 years ago
  39. 7bfb3a3 Add more tracing for key frames. by justinlin@chromium.org · 12 years ago
  40. 941fcc5 Increased the limit for KViEMaxCaptureDevices from 10 to 256. See issue 1343. by vikasmarwaha@webrtc.org · 12 years ago
  41. 1993a55 Added Stereo url paramter to apprtc demo. by vikasmarwaha@webrtc.org · 12 years ago
  42. 52b3905 Updated WebRTC version to 3.31 TBR=wu@webrtc.org by elham@webrtc.org · 12 years ago
  43. 43bf6ce Revert 4008 "Avoid resetting video encoder for similar configs." by phoglund@webrtc.org · 12 years ago
  44. c53480f Disabled flaky codec test (RunsCodecTestWithoutErrors) by phoglund@webrtc.org · 12 years ago
  45. aa4efd1 Avoid resetting video encoder for similar configs. by pbos@webrtc.org · 12 years ago
  46. 7707d06 Wiring down config from video engine until video coding and remote bitrate estimator modules instantiation. by andresp@webrtc.org · 12 years ago
  47. 7a5615b New WebAudio-WebRTC demo. by henrika@webrtc.org · 12 years ago
  48. 7ee8228 Remove TEXT(x) for BUILDINFO macros. by pbos@webrtc.org · 12 years ago
  49. 6b68c28 Added a config class to ease passing a set of options across webrtc. by andresp@webrtc.org · 12 years ago
  50. 9ecd686 Add svn:eol-style back which is lost in r3993 mistakenly. by braveyao@webrtc.org · 12 years ago
  51. a404d1d Change watchlist. by leozwang@webrtc.org · 12 years ago
  52. 7311083 Revert 3977 BUG=webrtc:1749 by tnakamura@webrtc.org · 12 years ago
  53. 05ea12f Reverting r3978 by elham@webrtc.org · 12 years ago
  54. d6ed000 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 · 12 years ago
  55. 571b336 Updating perf by mikhal@webrtc.org · 12 years ago
  56. 1e3c794 Use 2 threads for HD, or 1 for VGA or less. by fbarchard@google.com · 12 years ago
  57. 0680670 Updating perf by mikhal@webrtc.org · 12 years ago
  58. 6a36f0e 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 · 12 years ago
  59. e525309 WebRTCDemo Android doesn't hangle activity recreation correctly. by braveyao@webrtc.org · 12 years ago
  60. 219762a Drop Virtual webcam check script as moved into buildbot scripts. by kjellander@webrtc.org · 12 years ago
  61. ebdfa8d Add fischman into OWNERS of WebRTCDemo Android. by braveyao@webrtc.org · 12 years ago
  62. d72262d Fix compile errors in ViE with latest clang. by andrew@webrtc.org · 12 years ago
  63. c6a3755 Update SincResampler with the latest Chromium code. by andrew@webrtc.org · 12 years ago
  64. 4427273 Clean creation of VideoEngine: by andresp@webrtc.org · 12 years ago
  65. 6155be2 Add /tools/protoc_wrappers to .gitignore. by andrew@webrtc.org · 12 years ago
  66. aeb7d87 Tweaked webrtc_reformat. by phoglund@webrtc.org · 12 years ago
  67. 315d398 Formatted dtmf_queue. by phoglund@webrtc.org · 12 years ago
  68. 73a4d5a Add script to ensure virtual webcam is running. by kjellander@webrtc.org · 12 years ago
  69. f6d67ae Disable clang C++11 warnings to permit OVERRIDE keyword. by pbos@webrtc.org · 12 years ago
  70. d98e784 Fix VCMProcessTimer::TimeUntilProcess() unsigned-integer underflow problem. by stefan@webrtc.org · 12 years ago
  71. b55a12a Enable protobuf use in Chromium. by andrew@webrtc.org · 12 years ago
  72. e53084f Update protoc.gypi to match Chromium's latest. by andrew@webrtc.org · 12 years ago
  73. 3be565b Refactoring for typing detection by niklas.enbom@webrtc.org · 12 years ago
  74. ef14488 Trigger a PLI if the duration of non-decodable frames exceeds a threshold. by stefan@webrtc.org · 12 years ago
  75. 8f86cc8 VCM/Receiver: Return null when can't extract frame. by mikhal@webrtc.org · 12 years ago
  76. 474e915 Relanding 3962: VCM/JB: Porting jitter_buffer_test to gtest by mikhal@webrtc.org · 12 years ago
  77. 759b041 Relanding r3952: VCM: Updating receiver logic BUG=r1734 R=stefan@webrtc.org by mikhal@webrtc.org · 12 years ago
  78. 9c7685f VCM/JB: Break and skip to key if possible by mikhal@webrtc.org · 12 years ago
  79. 3004c79 Fix clang errors in non-GYP_DEFINES=clang=1 build by pbos@webrtc.org · 12 years ago
  80. d3a1959 Fix jitter buffer unittest. by stefan@webrtc.org · 12 years ago
  81. a5dee33 Correctly add packets to nack list when sequence number wraps. by stefan@webrtc.org · 12 years ago
  82. 0f29810 Fix crash in pacer. by pwestin@webrtc.org · 12 years ago
  83. 4ce19b1 Revert r3952 "VCM: Updating receiver logic" by stefan@webrtc.org · 12 years ago
  84. 2737590 Revert r3956 "VCM/JB: Porting jitter_buffer_test to gtest." by stefan@webrtc.org · 12 years ago
  85. 233c58d Landing 1399004, Minor clean up on the un-used _measureDelay code by xians@webrtc.org · 12 years ago
  86. 59aaebc Add an option to override the TestToStderr trace printout time. by andrew@webrtc.org · 12 years ago
  87. f9c289b Consolidate all third party licenses in LICENSE_THIRD_PARTY. by andrew@webrtc.org · 12 years ago
  88. df3da84 Updated WebRTC version number to 3.30 by elham@webrtc.org · 12 years ago
  89. 45f2da0 VCM/JB: Porting jitter_buffer_test to gtest. by mikhal@webrtc.org · 12 years ago
  90. a31c428 Remove 44.1 kHz workaround from AudioDevice on PulseAudio. by andrew@webrtc.org · 12 years ago
  91. 7cb766b Remove 44.1 kHz workaround from AudioDevice on WASAPI. by andrew@webrtc.org · 12 years ago
  92. bd4a2fe Fix off-by-one buffer overflow in WebRtcNetEQ_PacketBufferInsert(). by sergeyu@chromium.org · 12 years ago
  93. d3cd565 VCM: Updating receiver logic by mikhal@webrtc.org · 12 years ago
  94. d293a58e Correct and update dir name by leozwang@webrtc.org · 12 years ago
  95. 77f6b21 Revert 3934 "Revert 3933 "Remove traces of deprecated WebRtc_Wor..." by pbos@webrtc.org · 12 years ago
  96. 2580bc4 Get rid of some unnecessary copying when sending REMBs. by solenberg@webrtc.org · 12 years ago
  97. d5726a12 Formatting ACM tests by tina.legrand@webrtc.org · 12 years ago
  98. 03efc89 Fix when SetMinimumPlayoutDelay is configured to 0 by pwestin@webrtc.org · 12 years ago
  99. 42636e8 Removing bad code resulting in flaky test. by pwestin@webrtc.org · 12 years ago
  100. 52b4e88 Adding trace and changing pacing constants by pwestin@webrtc.org · 12 years ago