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webrtc / src / webrtc / a2c26549012b8024442fe0920d0325aa853d7890
  1. a2c2654 Extends the ScreenCapturer interface for individual display screen cast. by jiayl@webrtc.org · 11 years ago
  2. fba4f1c Roll Chromium 238260 -> 243863 by wjia@webrtc.org · 11 years ago
  3. 84350a9 Remove empty VideoCodecGeneric struct. by pbos@webrtc.org · 11 years ago
  4. 37fb66d Changing to using factory methods for some classes in NetEq by henrik.lundin@webrtc.org · 11 years ago
  5. b3ff385 Temporarily disabling some more audio processing tests. by aluebs@webrtc.org · 11 years ago
  6. 59fdf2d Fix MouseCursorMonitorMac to return correct hotspot position. by sergeyu@chromium.org · 11 years ago
  7. 4b5d36e Removes the remaining uses of the list wrapper class and the list wrapper class. by henrike@webrtc.org · 11 years ago
  8. 8c03c4c WebRTCDemo: fix out-of-bounds array read. by fischman@webrtc.org · 11 years ago
  9. eed1f11 Updated Webrtc version to 3.49 by elham@webrtc.org · 11 years ago
  10. 083049f Removes usage of ListWrapper from several files. by henrike@webrtc.org · 11 years ago
  11. ad584b6 Revert "Activate ACM test for Android in modules_tests." (rev5364). by andresp@webrtc.org · 11 years ago
  12. 572cc28 Temporarily disabling audio processing tests. by aluebs@webrtc.org · 11 years ago
  13. 2904b71 Increasing simulation time for NetEqPerformanceTest by henrik.lundin@webrtc.org · 11 years ago
  14. b7c1e03 Enables robust delay validation in AEC delay logging. by bjornv@webrtc.org · 11 years ago
  15. 22470b5 Minor voice engine improvements around AGC. by andrew@webrtc.org · 11 years ago
  16. f3a2ef3 Android: Fixes crash when exiting WebRTCDemo. by henrike@webrtc.org · 11 years ago
  17. 3ab10f9 Activate ACM test for Android in modules_tests. by turaj@webrtc.org · 11 years ago
  18. 75e7da3 Permitting double start/stopping of streams. by pbos@webrtc.org · 11 years ago
  19. 5b1467d Adding NetEq performance test to webrtc_perf_tests by henrik.lundin@webrtc.org · 11 years ago
  20. 8d4f9ca Delay Estimator: Adds unittests for robust validation. by bjornv@webrtc.org · 11 years ago
  21. 201049c Fixing lint errors in NetEq4 by henrik.lundin@webrtc.org · 11 years ago
  22. 8b6867b Make code simpler on VCMEncodedCallback. by andresp@webrtc.org · 11 years ago
  23. 88ece35 Isolate register post encode callback in video coding module to simplify code and critical sections. by andresp@webrtc.org · 11 years ago
  24. 093b960 Isolate debug recording from video sender into a thread safe small class. by andresp@webrtc.org · 11 years ago
  25. 0b9d7ce Add another test case for AST/TOF switching. by solenberg@webrtc.org · 11 years ago
  26. 9b125e1 Delay Estimator: Converts a constant into a configurable parameter. by bjornv@webrtc.org · 11 years ago
  27. f3b2148 Init to 16 kHz in the fixed-point profile. by andrew@webrtc.org · 11 years ago
  28. 48b9892 Ensure capture_levels_ is sized correctly at init time. by andrew@webrtc.org · 11 years ago
  29. 2c358e2 Now printing less output from compare_videos.py. by phoglund@webrtc.org · 11 years ago
  30. e95dc25 Remove the requirement to call set_sample_rate_hz and friends. by andrew@webrtc.org · 11 years ago
  31. b3b6049 Remove outdated DestroyVideoSendStream comment. by pbos@webrtc.org · 11 years ago
  32. 49812e6 Wire up statistics in video send stream of new video engine api by sprang@webrtc.org · 11 years ago
  33. c902d88 Delay Estimator: robust_validation should be stored over a reset by bjornv@webrtc.org · 11 years ago
  34. e66b5bc Add include guards to forward_error_correction_internal.h by braveyao@webrtc.org · 11 years ago
  35. 1a6b274 Fix the include guard in transmit_mixer.h by braveyao@webrtc.org · 11 years ago
  36. 5d7992f Fix the include guard in transmit_mixer.h by braveyao@webrtc.org · 11 years ago
  37. 29975da Android build: make it quiet on success and not overly noisy on failure. by fischman@webrtc.org · 11 years ago
  38. fba1476 Fix the android clang bot for compiling with thread annotations. by andresp@webrtc.org · 11 years ago
  39. c9faf10 Add thread_annotations for clang targets. by andresp@webrtc.org · 11 years ago
  40. a9a7327 If the configured start bitrate is higher than the configures max by mflodman@webrtc.org · 11 years ago
  41. 9662535 Race condition in ViECapturer::RegisterObserver by sprang@webrtc.org · 11 years ago
  42. 91cebfc Update WebRTC to version 3.48 by tnakamura@webrtc.org · 11 years ago
  43. 4f1f5fa Add callbacks for receive channel RTCP statistics. by sprang@webrtc.org · 11 years ago
  44. 9edcdb0 Refactoring MediaOptimization so it can easily be turned into a thread-safe class. by andresp@webrtc.org · 11 years ago
  45. aacdb9f Integrate fake_network_pipe into direct_transport. by stefan@webrtc.org · 11 years ago
  46. a07c56f Remove metrics_unittests by kjellander@webrtc.org · 11 years ago
  47. 5596ac6 Remove media_file from VideoEngine dependencies. by pbos@webrtc.org · 11 years ago
  48. 5e0cbcf cpplint cleaning new API and its implementation files. by mflodman@webrtc.org · 11 years ago
  49. 6c172c5 Reduced execution time for CallTest::ReceivesPliAndRecovers, by dropping only one packet and made it predictable by removing rand(). by mflodman@webrtc.org · 11 years ago
  50. 5e252ac Speeding up CallTest.ReceivesAndRetransmitsNack and removed the random packet loss. by mflodman@webrtc.org · 11 years ago
  51. eb9ce11 Make MouseCursor mutable by sergeyu@chromium.org · 11 years ago
  52. 8d14e06 audio_processing_unittest: unbreak clang compilation. by fischman@webrtc.org · 11 years ago
  53. d138166 JNI Audio: remove dead members. by fischman@webrtc.org · 11 years ago
  54. 432e574 Revert "Make MouseCursor mutable" by sergeyu@chromium.org · 11 years ago
  55. 865be14 Make MouseCursor mutable by sergeyu@chromium.org · 11 years ago
  56. 5d13922 Stop transport in test SuspendBelowMinBitrate. by pbos@webrtc.org · 11 years ago
  57. 202d38d Added method for getting default module state and protect agains a by mflodman@webrtc.org · 11 years ago
  58. d60137f Modify video_render/ to allow a single old frame. by pbos@webrtc.org · 11 years ago
  59. 471354f Delete capturers after destroying streams in test. by pbos@webrtc.org · 11 years ago
  60. 5041831 Revert 5285 "Revert 5228 "Use the RTT from RtcpRttStats class if..." by asapersson@webrtc.org · 11 years ago
  61. 0443f6c Simplification of histogram normalization in delay estimator. by bjornv@webrtc.org · 11 years ago
  62. 46f7288 Revert r5294 to re-roll r5293. by pbos@webrtc.org · 11 years ago
  63. 1b3b8cb Adds robust validation functionality to the delay estimator by bjornv@webrtc.org · 11 years ago
  64. 1eb1008 Incorrect iterator++ in ModuleRtpRtcpImpl::RegisterVideoBitrateObserver by sprang@webrtc.org · 11 years ago
  65. c5a5713 Revert 5293 "Auto instantiate RBE depending on whether AST or TO..." by turaj@webrtc.org · 11 years ago
  66. 3bbc91e Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. by solenberg@webrtc.org · 11 years ago
  67. 7d7e63d Disabled tests on Android. The issue 2723 is filed to investigate the reason for tests failing. by turaj@webrtc.org · 11 years ago
  68. afceaca Measure pacer queue size based on when packets are inserted rather than captured. by stefan@webrtc.org · 11 years ago
  69. f1b92fd Fix jitter buffer delay estimate. by turaj@webrtc.org · 11 years ago
  70. 79d6daf Update talk to 58174641 together with http://review.webrtc.org/4319005/. by wu@webrtc.org · 11 years ago
  71. 7ff4089 Adding REMB to receive stream configuration, the send side will always by mflodman@webrtc.org · 11 years ago
  72. 1465cef Revert 5228 "Use the RTT from RtcpRttStats class if provided whe..." by asapersson@webrtc.org · 11 years ago
  73. f64791e Merge metrics_unittests into video_engine_tests. by pbos@webrtc.org · 11 years ago
  74. f94ccd4 Move realtime tests to webrtc_perf_tests. by pbos@webrtc.org · 11 years ago
  75. b70db6d Callback for send bitrate estimates - new roll by sprang@webrtc.org · 11 years ago
  76. dadfc9e Make sure channels in the same call are in the same channel group. by mflodman@webrtc.org · 11 years ago
  77. c49a3fa Making RemoteRateControl::min_configured_bit_rate_ configurable by henrik.lundin@webrtc.org · 11 years ago
  78. efeb8ce Update talk to 58127566 together with by wu@webrtc.org · 11 years ago
  79. da4d59e ACM 2 compatibility with ACM 1. by turaj@webrtc.org · 11 years ago
  80. 12553ad Revert 5274 "Update talk to 58113193 together with https://webrt..." by wu@webrtc.org · 11 years ago
  81. 984bee2 Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. by wu@webrtc.org · 11 years ago
  82. a48c91d Complete rewrite of demo application. by henrike@webrtc.org · 11 years ago
  83. 4fcb2f5 Remove overloaded CpuOveruseMeasure function. by asapersson@webrtc.org · 11 years ago
  84. 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
  85. 6bf67db Fix common_video_unittests in apk_tests.gyp. by pbos@webrtc.org · 11 years ago
  86. c33d37c Add SwapFrame() to VideoSendStreamInput. by pbos@webrtc.org · 11 years ago
  87. ffea4ce Revert 5259 "Callback for send bitrate estimates" by sprang@webrtc.org · 11 years ago
  88. ca63ad9 Roll chromium_revision 232627:238260 by kjellander@webrtc.org · 11 years ago
  89. 1430bc3 Callback for send bitrate estimates by sprang@webrtc.org · 11 years ago
  90. b46e68d Removed unnecessary Pulse init from VoE startup. by fischman@webrtc.org · 11 years ago
  91. aa9e768 Correctly define OVERRIDE when building with g++ 4.7 and C++11 support by andrew@webrtc.org · 11 years ago
  92. b589c65 Improve VideoSendStreamTest::MaxPacketSize by sprang@webrtc.org · 11 years ago
  93. 2cafda4 Change uses of the obsolete armv7 setting to arm_version==7. by kjellander@webrtc.org · 11 years ago
  94. 5424c16 Fix compilation errors on Fedora 20. by fischman@webrtc.org · 11 years ago
  95. 241103f Ensure WEBRTC_MODULE_UTILITY_VIDEO is undefined for enable_video==0. by andrew@webrtc.org · 11 years ago
  96. 935c8c7 Add shape in DesktopFrame. by sergeyu@chromium.org · 11 years ago
  97. 8beba83 Add new method to MockAudioProcessing. by andrew@webrtc.org · 11 years ago
  98. e388f9e Revert 5229 "Make VideoSendStreamTest::MaxPacketSize test a whol..." by andrew@webrtc.org · 11 years ago
  99. 7b72264 Allow opening an AEC dump from an existing file handle. by henrikg@webrtc.org · 11 years ago
  100. 4383539 Stop video capturers in multi-stream test. by pbos@webrtc.org · 11 years ago