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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 0d03514a1a95cd83126ba35102c4277555603577 / modules
  1. 0d03514 Revert "Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file." by michaelbai@google.com · 10 years ago
  2. d1e7fac Add stats of incoming frame delays for debugging bandwidth estimation. by jiayl@webrtc.org · 10 years ago
  3. 0e53607 Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file. by michaelbai@google.com · 10 years ago
  4. 9add2bb Fix WindowCapturerWin to unselect bitmap before destroying DC. by sergeyu@chromium.org · 10 years ago
  5. de93ce0 Plot the capacity of a trace-based delivery filter. by stefan@webrtc.org · 10 years ago
  6. df1f99f Add delay and send/receive throughput plots to BWE simulation. by stefan@webrtc.org · 10 years ago
  7. aa42c87 Implementing replacement audio support in neteq_rtpplay by henrik.lundin@webrtc.org · 10 years ago
  8. 7993c94 Fixing a bug in DummyRTPpacket by henrik.lundin@webrtc.org · 10 years ago
  9. 03e6a88 Update AudioProcessing::Create docs. by andrew@webrtc.org · 10 years ago
  10. 4845e6e Fix a cursor capturing issue on Windows. by jiayl@webrtc.org · 10 years ago
  11. abe3b85 Handle the invalid case of setting multiple stream_bitrates if there is only a single send stream registered. by stefan@webrtc.org · 10 years ago
  12. 7387e23 Trivial rename of non-compile time consts. by andrew@webrtc.org · 10 years ago
  13. 76dcaae Wire up feedback to VideoSender. by stefan@webrtc.org · 10 years ago
  14. 9c06023 Re-enabling audio processing tests by aluebs@webrtc.org · 10 years ago
  15. 87c8b86 Moved the new OnData interface to AudioTranport, and expose the AudioTransport pointer via voe_base by xians@webrtc.org · 10 years ago
  16. 1d23d5e Implement single monitor capture on Mac. by jiayl@webrtc.org · 10 years ago
  17. f65e883 Fixing test name for NetEqPerformanceTest by henrik.lundin@webrtc.org · 11 years ago
  18. e1c9902 Add BWE tools for parsing RTP files. by stefan@webrtc.org · 11 years ago
  19. 5e9b730 Fix the mouse cursor offset issue on Mac. by jiayl@webrtc.org · 11 years ago
  20. c13a537 Move out typing detection to its own class. by henrikg@webrtc.org · 11 years ago
  21. 54ad929 Moves the display reconfiguration callback into a separate class, by jiayl@webrtc.org · 11 years ago
  22. 4a3208e Fix deadlock in video_receiver.cc. by stefan@webrtc.org · 11 years ago
  23. 4d088c6 Add Config struct for experimental AGC. by andrew@webrtc.org · 11 years ago
  24. 7dc7514 Revert 5444 "Revert 5421 "Fix deadlock on register/unregister ob..." by mallinath@webrtc.org · 11 years ago
  25. a6b9a3f Add clean test to NetEq perf test by henrik.lundin@webrtc.org · 11 years ago
  26. cfbde68 VideoCaptureAndroid: stop preview in opposite order of starting. by fischman@webrtc.org · 11 years ago
  27. 9afb6cf Revert 5421 "Fix deadlock on register/unregister observer while ..." by mallinath@webrtc.org · 11 years ago
  28. 4406e5e Avoid potential dead lock in StreamStatisticianImpl by sprang@webrtc.org · 11 years ago
  29. 312a049 Race condition in RTPSender::UpdateRtpStats by sprang@webrtc.org · 11 years ago
  30. be35c57 Fix bug introduced during replace of list wrapper with std equivalents in r5378. by andresp@webrtc.org · 11 years ago
  31. 50d6975 Race in StreamStatisticianImpl::GetStatistics vs. ::IncomingPacket by sprang@webrtc.org · 11 years ago
  32. 3021162 Fix "field '_testNo' is uninitialized" warnings. by pbos@webrtc.org · 11 years ago
  33. 3fbe666 Add a Config parameter to AudioProcessing::Create(). by andrew@webrtc.org · 11 years ago
  34. 01d06c8 Remove loopback setup in RtpRtcpImplTest. Changed to use two separate rtp/rtcp modules. by asapersson@webrtc.org · 11 years ago
  35. ef6a602 Add trace-based delivery filter to BWE test framework. by stefan@webrtc.org · 11 years ago
  36. 630939f Fix deadlock on register/unregister observer while there is a an going callback. by andresp@webrtc.org · 11 years ago
  37. d242006 Fix array declarations in aec_rdft.h. by andrew@webrtc.org · 11 years ago
  38. 7d99cd4 Add callbacks for receive channel RTP statistics by sprang@webrtc.org · 11 years ago
  39. 83a37c8 Potential dead lock in receive statistics by sprang@webrtc.org · 11 years ago
  40. 57e8ba9 Fix for libtalkmobile build error bug=b/12549061 by elham@webrtc.org · 11 years ago
  41. b4263e0 Add configuration and test for extended RTCP reference time reports to new video api. by asapersson@webrtc.org · 11 years ago
  42. 4be4e8a Android, OpenSlDemo: moved to webrtc/examples/android/opensl_loopback by henrike@webrtc.org · 11 years ago
  43. 28bd30a Implement screen enumeration and individual screen capturing for Windows. by jiayl@webrtc.org · 11 years ago
  44. aea6053 Android, OpenSlDemo: fixes issue where app would crash as soon as the application is started. by henrike@webrtc.org · 11 years ago
  45. 0902491 Add full path to headers by aluebs@webrtc.org · 11 years ago
  46. a7c7e2c Adds back set_sample_rate_hz() when Init is called in recordings. by bjornv@webrtc.org · 11 years ago
  47. 556423f MIPS optimizations for NS audio processing module by andrew@webrtc.org · 11 years ago
  48. 4c20797 Fix crash in MouseCursor::CopyOf() by sergeyu@chromium.org · 11 years ago
  49. 390a5ae Update needed to MockScreenCapturer after new methods addition to webrtc::ScreenCapturer. by mallinath@webrtc.org · 11 years ago
  50. a2c2654 Extends the ScreenCapturer interface for individual display screen cast. by jiayl@webrtc.org · 11 years ago
  51. 84350a9 Remove empty VideoCodecGeneric struct. by pbos@webrtc.org · 11 years ago
  52. 37fb66d Changing to using factory methods for some classes in NetEq by henrik.lundin@webrtc.org · 11 years ago
  53. 59fdf2d Fix MouseCursorMonitorMac to return correct hotspot position. by sergeyu@chromium.org · 11 years ago
  54. 4b5d36e Removes the remaining uses of the list wrapper class and the list wrapper class. by henrike@webrtc.org · 11 years ago
  55. 083049f Removes usage of ListWrapper from several files. by henrike@webrtc.org · 11 years ago
  56. ad584b6 Revert "Activate ACM test for Android in modules_tests." (rev5364). by andresp@webrtc.org · 11 years ago
  57. 572cc28 Temporarily disabling audio processing tests. by aluebs@webrtc.org · 11 years ago
  58. 2904b71 Increasing simulation time for NetEqPerformanceTest by henrik.lundin@webrtc.org · 11 years ago
  59. b7c1e03 Enables robust delay validation in AEC delay logging. by bjornv@webrtc.org · 11 years ago
  60. f3a2ef3 Android: Fixes crash when exiting WebRTCDemo. by henrike@webrtc.org · 11 years ago
  61. 3ab10f9 Activate ACM test for Android in modules_tests. by turaj@webrtc.org · 11 years ago
  62. 5b1467d Adding NetEq performance test to webrtc_perf_tests by henrik.lundin@webrtc.org · 11 years ago
  63. 8d4f9ca Delay Estimator: Adds unittests for robust validation. by bjornv@webrtc.org · 11 years ago
  64. 201049c Fixing lint errors in NetEq4 by henrik.lundin@webrtc.org · 11 years ago
  65. 8b6867b Make code simpler on VCMEncodedCallback. by andresp@webrtc.org · 11 years ago
  66. 88ece35 Isolate register post encode callback in video coding module to simplify code and critical sections. by andresp@webrtc.org · 11 years ago
  67. 093b960 Isolate debug recording from video sender into a thread safe small class. by andresp@webrtc.org · 11 years ago
  68. 9b125e1 Delay Estimator: Converts a constant into a configurable parameter. by bjornv@webrtc.org · 11 years ago
  69. f3b2148 Init to 16 kHz in the fixed-point profile. by andrew@webrtc.org · 11 years ago
  70. 48b9892 Ensure capture_levels_ is sized correctly at init time. by andrew@webrtc.org · 11 years ago
  71. e95dc25 Remove the requirement to call set_sample_rate_hz and friends. by andrew@webrtc.org · 11 years ago
  72. c902d88 Delay Estimator: robust_validation should be stored over a reset by bjornv@webrtc.org · 11 years ago
  73. e66b5bc Add include guards to forward_error_correction_internal.h by braveyao@webrtc.org · 11 years ago
  74. 29975da Android build: make it quiet on success and not overly noisy on failure. by fischman@webrtc.org · 11 years ago
  75. fba1476 Fix the android clang bot for compiling with thread annotations. by andresp@webrtc.org · 11 years ago
  76. c9faf10 Add thread_annotations for clang targets. by andresp@webrtc.org · 11 years ago
  77. 4f1f5fa Add callbacks for receive channel RTCP statistics. by sprang@webrtc.org · 11 years ago
  78. 9edcdb0 Refactoring MediaOptimization so it can easily be turned into a thread-safe class. by andresp@webrtc.org · 11 years ago
  79. eb9ce11 Make MouseCursor mutable by sergeyu@chromium.org · 11 years ago
  80. 8d14e06 audio_processing_unittest: unbreak clang compilation. by fischman@webrtc.org · 11 years ago
  81. d138166 JNI Audio: remove dead members. by fischman@webrtc.org · 11 years ago
  82. 432e574 Revert "Make MouseCursor mutable" by sergeyu@chromium.org · 11 years ago
  83. 865be14 Make MouseCursor mutable by sergeyu@chromium.org · 11 years ago
  84. 202d38d Added method for getting default module state and protect agains a by mflodman@webrtc.org · 11 years ago
  85. d60137f Modify video_render/ to allow a single old frame. by pbos@webrtc.org · 11 years ago
  86. 5041831 Revert 5285 "Revert 5228 "Use the RTT from RtcpRttStats class if..." by asapersson@webrtc.org · 11 years ago
  87. 0443f6c Simplification of histogram normalization in delay estimator. by bjornv@webrtc.org · 11 years ago
  88. 46f7288 Revert r5294 to re-roll r5293. by pbos@webrtc.org · 11 years ago
  89. 1b3b8cb Adds robust validation functionality to the delay estimator by bjornv@webrtc.org · 11 years ago
  90. 1eb1008 Incorrect iterator++ in ModuleRtpRtcpImpl::RegisterVideoBitrateObserver by sprang@webrtc.org · 11 years ago
  91. c5a5713 Revert 5293 "Auto instantiate RBE depending on whether AST or TO..." by turaj@webrtc.org · 11 years ago
  92. 3bbc91e Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. by solenberg@webrtc.org · 11 years ago
  93. afceaca Measure pacer queue size based on when packets are inserted rather than captured. by stefan@webrtc.org · 11 years ago
  94. 79d6daf Update talk to 58174641 together with http://review.webrtc.org/4319005/. by wu@webrtc.org · 11 years ago
  95. 1465cef Revert 5228 "Use the RTT from RtcpRttStats class if provided whe..." by asapersson@webrtc.org · 11 years ago
  96. b70db6d Callback for send bitrate estimates - new roll by sprang@webrtc.org · 11 years ago
  97. c49a3fa Making RemoteRateControl::min_configured_bit_rate_ configurable by henrik.lundin@webrtc.org · 11 years ago
  98. da4d59e ACM 2 compatibility with ACM 1. by turaj@webrtc.org · 11 years ago
  99. 27f0841 Enables mixing and matching Java and native audio. It is used for getting best of both worlds capabilities (AEC and low latency). by henrike@webrtc.org · 11 years ago
  100. c33d37c Add SwapFrame() to VideoSendStreamInput. by pbos@webrtc.org · 11 years ago