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webrtc / src / e6ab21b9cab43a3e6651bbe512f2c3495a43e547 / webrtc / modules
  1. e6ab21b Fixing a bug in DummyRTPpacket by henrik.lundin@webrtc.org · 11 years ago
  2. 5474491 Update AudioProcessing::Create docs. by andrew@webrtc.org · 11 years ago
  3. 20a60ea Fix a cursor capturing issue on Windows. by jiayl@webrtc.org · 11 years ago
  4. 0e5a2b5 Handle the invalid case of setting multiple stream_bitrates if there is only a single send stream registered. by stefan@webrtc.org · 11 years ago
  5. f6a638e Trivial rename of non-compile time consts. by andrew@webrtc.org · 11 years ago
  6. 422fdbf Wire up feedback to VideoSender. by stefan@webrtc.org · 11 years ago
  7. c9ee412 Re-enabling audio processing tests by aluebs@webrtc.org · 11 years ago
  8. c1e2803 Moved the new OnData interface to AudioTranport, and expose the AudioTransport pointer via voe_base by xians@webrtc.org · 11 years ago
  9. 1af5ea0 Implement single monitor capture on Mac. by jiayl@webrtc.org · 11 years ago
  10. 83aee8f Fixing test name for NetEqPerformanceTest by henrik.lundin@webrtc.org · 11 years ago
  11. 1dd9b4d Add BWE tools for parsing RTP files. by stefan@webrtc.org · 11 years ago
  12. bda5fa7 Fix the mouse cursor offset issue on Mac. by jiayl@webrtc.org · 11 years ago
  13. c693704 Move out typing detection to its own class. by henrikg@webrtc.org · 11 years ago
  14. cf1b51b Moves the display reconfiguration callback into a separate class, by jiayl@webrtc.org · 11 years ago
  15. f7c6e74 Fix deadlock in video_receiver.cc. by stefan@webrtc.org · 11 years ago
  16. c7c7a53 Add Config struct for experimental AGC. by andrew@webrtc.org · 11 years ago
  17. 7433a08 Revert 5444 "Revert 5421 "Fix deadlock on register/unregister ob..." by mallinath@webrtc.org · 11 years ago
  18. 84eb0e9 Add clean test to NetEq perf test by henrik.lundin@webrtc.org · 11 years ago
  19. 932b019 VideoCaptureAndroid: stop preview in opposite order of starting. by fischman@webrtc.org · 11 years ago
  20. 18586d3 Revert 5421 "Fix deadlock on register/unregister observer while ..." by mallinath@webrtc.org · 11 years ago
  21. a45cac0 Avoid potential dead lock in StreamStatisticianImpl by sprang@webrtc.org · 11 years ago
  22. 5314e85 Race condition in RTPSender::UpdateRtpStats by sprang@webrtc.org · 11 years ago
  23. 2397a17 Fix bug introduced during replace of list wrapper with std equivalents in r5378. by andresp@webrtc.org · 11 years ago
  24. c00adbe Race in StreamStatisticianImpl::GetStatistics vs. ::IncomingPacket by sprang@webrtc.org · 11 years ago
  25. 99eab02 Fix "field '_testNo' is uninitialized" warnings. by pbos@webrtc.org · 11 years ago
  26. e84978f Add a Config parameter to AudioProcessing::Create(). by andrew@webrtc.org · 11 years ago
  27. 871d949 Remove loopback setup in RtpRtcpImplTest. Changed to use two separate rtp/rtcp modules. by asapersson@webrtc.org · 11 years ago
  28. 99a8c7e Add trace-based delivery filter to BWE test framework. by stefan@webrtc.org · 11 years ago
  29. 8d375c9 Fix deadlock on register/unregister observer while there is a an going callback. by andresp@webrtc.org · 11 years ago
  30. 754de52 Fix array declarations in aec_rdft.h. by andrew@webrtc.org · 11 years ago
  31. 0e93257 Add callbacks for receive channel RTP statistics by sprang@webrtc.org · 11 years ago
  32. 7dba27c Potential dead lock in receive statistics by sprang@webrtc.org · 11 years ago
  33. 32c3247 Fix for libtalkmobile build error bug=b/12549061 by elham@webrtc.org · 11 years ago
  34. efaeda0 Add configuration and test for extended RTCP reference time reports to new video api. by asapersson@webrtc.org · 11 years ago
  35. 32c26eb Android, OpenSlDemo: moved to webrtc/examples/android/opensl_loopback by henrike@webrtc.org · 11 years ago
  36. 4985927 Implement screen enumeration and individual screen capturing for Windows. by jiayl@webrtc.org · 11 years ago
  37. ead202b Android, OpenSlDemo: fixes issue where app would crash as soon as the application is started. by henrike@webrtc.org · 11 years ago
  38. 4ffd9c7 Add full path to headers by aluebs@webrtc.org · 11 years ago
  39. 6a94734 Adds back set_sample_rate_hz() when Init is called in recordings. by bjornv@webrtc.org · 11 years ago
  40. ea9392d MIPS optimizations for NS audio processing module by andrew@webrtc.org · 11 years ago
  41. fb4e256 Fix crash in MouseCursor::CopyOf() by sergeyu@chromium.org · 11 years ago
  42. 7a2ca7c Update needed to MockScreenCapturer after new methods addition to webrtc::ScreenCapturer. by mallinath@webrtc.org · 11 years ago
  43. 017b619 Extends the ScreenCapturer interface for individual display screen cast. by jiayl@webrtc.org · 11 years ago
  44. 39fcfd7 Remove empty VideoCodecGeneric struct. by pbos@webrtc.org · 11 years ago
  45. d9faa46 Changing to using factory methods for some classes in NetEq by henrik.lundin@webrtc.org · 11 years ago
  46. eb31b45 Fix MouseCursorMonitorMac to return correct hotspot position. by sergeyu@chromium.org · 11 years ago
  47. 3907c2e Removes the remaining uses of the list wrapper class and the list wrapper class. by henrike@webrtc.org · 11 years ago
  48. 79cf3ac Removes usage of ListWrapper from several files. by henrike@webrtc.org · 11 years ago
  49. d0b436a Revert "Activate ACM test for Android in modules_tests." (rev5364). by andresp@webrtc.org · 11 years ago
  50. 8bc4fcf Temporarily disabling audio processing tests. by aluebs@webrtc.org · 11 years ago
  51. 2c03bf1 Increasing simulation time for NetEqPerformanceTest by henrik.lundin@webrtc.org · 11 years ago
  52. bbd47fc Enables robust delay validation in AEC delay logging. by bjornv@webrtc.org · 11 years ago
  53. 573a1b4 Android: Fixes crash when exiting WebRTCDemo. by henrike@webrtc.org · 11 years ago
  54. 7cc64b3 Activate ACM test for Android in modules_tests. by turaj@webrtc.org · 11 years ago
  55. a366e81 Adding NetEq performance test to webrtc_perf_tests by henrik.lundin@webrtc.org · 11 years ago
  56. fa8d534 Delay Estimator: Adds unittests for robust validation. by bjornv@webrtc.org · 11 years ago
  57. e7ce437 Fixing lint errors in NetEq4 by henrik.lundin@webrtc.org · 11 years ago
  58. c5aeb2a Make code simpler on VCMEncodedCallback. by andresp@webrtc.org · 11 years ago
  59. 1df9dc3 Isolate register post encode callback in video coding module to simplify code and critical sections. by andresp@webrtc.org · 11 years ago
  60. b08a12d Isolate debug recording from video sender into a thread safe small class. by andresp@webrtc.org · 11 years ago
  61. bccd53d Delay Estimator: Converts a constant into a configurable parameter. by bjornv@webrtc.org · 11 years ago
  62. d335094 Init to 16 kHz in the fixed-point profile. by andrew@webrtc.org · 11 years ago
  63. b6541ca Ensure capture_levels_ is sized correctly at init time. by andrew@webrtc.org · 11 years ago
  64. 60730cf Remove the requirement to call set_sample_rate_hz and friends. by andrew@webrtc.org · 11 years ago
  65. a89d17d Delay Estimator: robust_validation should be stored over a reset by bjornv@webrtc.org · 11 years ago
  66. 2fb72cf Add include guards to forward_error_correction_internal.h by braveyao@webrtc.org · 11 years ago
  67. 000dde9 Android build: make it quiet on success and not overly noisy on failure. by fischman@webrtc.org · 11 years ago
  68. f6acf98 Fix the android clang bot for compiling with thread annotations. by andresp@webrtc.org · 11 years ago
  69. 7fb75ec Add thread_annotations for clang targets. by andresp@webrtc.org · 11 years ago
  70. 54ae4ff Add callbacks for receive channel RTCP statistics. by sprang@webrtc.org · 11 years ago
  71. e682aa5 Refactoring MediaOptimization so it can easily be turned into a thread-safe class. by andresp@webrtc.org · 11 years ago
  72. 8ae7256 Make MouseCursor mutable by sergeyu@chromium.org · 11 years ago
  73. f8be8df audio_processing_unittest: unbreak clang compilation. by fischman@webrtc.org · 11 years ago
  74. 179908c JNI Audio: remove dead members. by fischman@webrtc.org · 11 years ago
  75. e4c9272 Revert "Make MouseCursor mutable" by sergeyu@chromium.org · 11 years ago
  76. 8fd1d26 Make MouseCursor mutable by sergeyu@chromium.org · 11 years ago
  77. e6b871b Added method for getting default module state and protect agains a by mflodman@webrtc.org · 11 years ago
  78. eb7b7bc Modify video_render/ to allow a single old frame. by pbos@webrtc.org · 11 years ago
  79. e7b1e11 Revert 5285 "Revert 5228 "Use the RTT from RtcpRttStats class if..." by asapersson@webrtc.org · 11 years ago
  80. 1e7d612 Simplification of histogram normalization in delay estimator. by bjornv@webrtc.org · 11 years ago
  81. 5ab7567 Revert r5294 to re-roll r5293. by pbos@webrtc.org · 11 years ago
  82. 5c64508 Adds robust validation functionality to the delay estimator by bjornv@webrtc.org · 11 years ago
  83. 87ad57b Incorrect iterator++ in ModuleRtpRtcpImpl::RegisterVideoBitrateObserver by sprang@webrtc.org · 11 years ago
  84. 41e2615 Revert 5293 "Auto instantiate RBE depending on whether AST or TO..." by turaj@webrtc.org · 11 years ago
  85. 341e914 Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. by solenberg@webrtc.org · 11 years ago
  86. dd393e7 Measure pacer queue size based on when packets are inserted rather than captured. by stefan@webrtc.org · 11 years ago
  87. 24301a6 Update talk to 58174641 together with http://review.webrtc.org/4319005/. by wu@webrtc.org · 11 years ago
  88. 86bb56a Revert 5228 "Use the RTT from RtcpRttStats class if provided whe..." by asapersson@webrtc.org · 11 years ago
  89. 6811b6e Callback for send bitrate estimates - new roll by sprang@webrtc.org · 11 years ago
  90. e9abd59 Making RemoteRateControl::min_configured_bit_rate_ configurable by henrik.lundin@webrtc.org · 11 years ago
  91. a92baea ACM 2 compatibility with ACM 1. by turaj@webrtc.org · 11 years ago
  92. 9ee75e9 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
  93. 724947b Add SwapFrame() to VideoSendStreamInput. by pbos@webrtc.org · 11 years ago
  94. 096e8d9 Revert 5259 "Callback for send bitrate estimates" by sprang@webrtc.org · 11 years ago
  95. f9bdbe3 Roll chromium_revision 232627:238260 by kjellander@webrtc.org · 11 years ago
  96. 2656cf9 Callback for send bitrate estimates by sprang@webrtc.org · 11 years ago
  97. 7ae8495 Removed unnecessary Pulse init from VoE startup. by fischman@webrtc.org · 11 years ago
  98. 917306d Change uses of the obsolete armv7 setting to arm_version==7. by kjellander@webrtc.org · 11 years ago
  99. eb7def2 Fix compilation errors on Fedora 20. by fischman@webrtc.org · 11 years ago
  100. de7c9e8 Ensure WEBRTC_MODULE_UTILITY_VIDEO is undefined for enable_video==0. by andrew@webrtc.org · 11 years ago