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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / b7e5b2741f9cbd68e7be54e3445d28d266477c92
  1. 3021162 Fix "field '_testNo' is uninitialized" warnings. by pbos@webrtc.org · 11 years ago
  2. a370f24 Always initialize Trace in Call TraceDispatcher. by pbos@webrtc.org · 11 years ago
  3. 3fbe666 Add a Config parameter to AudioProcessing::Create(). by andrew@webrtc.org · 11 years ago
  4. b2c6a45 Android, WebRTCDemo: fixes crash issue when pressing switch camera button on devices with only one camera. by henrike@webrtc.org · 11 years ago
  5. 01d06c8 Remove loopback setup in RtpRtcpImplTest. Changed to use two separate rtp/rtcp modules. by asapersson@webrtc.org · 11 years ago
  6. ed592c7 Add new API (webrtc.gyp:webrtc) to merge_libs.gyp. by pbos@webrtc.org · 11 years ago
  7. ef6a602 Add trace-based delivery filter to BWE test framework. by stefan@webrtc.org · 11 years ago
  8. c71929d Wire up RTX in VideoReceiveStream. by pbos@webrtc.org · 11 years ago
  9. 630939f Fix deadlock on register/unregister observer while there is a an going callback. by andresp@webrtc.org · 11 years ago
  10. d242006 Fix array declarations in aec_rdft.h. by andrew@webrtc.org · 11 years ago
  11. e822c84 Set NACKed packet to -1 in TestNackRetransmission. by pbos@webrtc.org · 11 years ago
  12. 7d99cd4 Add callbacks for receive channel RTP statistics by sprang@webrtc.org · 11 years ago
  13. 888b839 Android, fixes crash on devices with only front cameras. by henrike@webrtc.org · 11 years ago
  14. 0584274 Output logs to stderr from voe_cmd_test by default. by andrew@webrtc.org · 11 years ago
  15. 83dd51d Android example apps: fixes issue where useful failure information was suppressed. by henrike@webrtc.org · 11 years ago
  16. 83a37c8 Potential dead lock in receive statistics by sprang@webrtc.org · 11 years ago
  17. 57e8ba9 Fix for libtalkmobile build error bug=b/12549061 by elham@webrtc.org · 11 years ago
  18. 7238373 Removes script for generating supplement.gypi also adds git ignore for tools/gn. by henrike@webrtc.org · 11 years ago
  19. 5b08052 Set up receiver RTX config using a std::map. by pbos@webrtc.org · 11 years ago
  20. b4263e0 Add configuration and test for extended RTCP reference time reports to new video api. by asapersson@webrtc.org · 11 years ago
  21. 4be4e8a Android, OpenSlDemo: moved to webrtc/examples/android/opensl_loopback by henrike@webrtc.org · 11 years ago
  22. 28bd30a Implement screen enumeration and individual screen capturing for Windows. by jiayl@webrtc.org · 11 years ago
  23. aea6053 Android, OpenSlDemo: fixes issue where app would crash as soon as the application is started. by henrike@webrtc.org · 11 years ago
  24. 89b5ae8 Android, WebRTCDemo: fix issue where changing remote IP was not working properly. by henrike@webrtc.org · 11 years ago
  25. 0902491 Add full path to headers by aluebs@webrtc.org · 11 years ago
  26. a7c7e2c Adds back set_sample_rate_hz() when Init is called in recordings. by bjornv@webrtc.org · 11 years ago
  27. 556423f MIPS optimizations for NS audio processing module by andrew@webrtc.org · 11 years ago
  28. 4c20797 Fix crash in MouseCursor::CopyOf() by sergeyu@chromium.org · 11 years ago
  29. 96961e5 Exclude protoc objects from merge_libs.py. by andrew@webrtc.org · 11 years ago
  30. 390a5ae Update needed to MockScreenCapturer after new methods addition to webrtc::ScreenCapturer. by mallinath@webrtc.org · 11 years ago
  31. a2c2654 Extends the ScreenCapturer interface for individual display screen cast. by jiayl@webrtc.org · 11 years ago
  32. fba4f1c Roll Chromium 238260 -> 243863 by wjia@webrtc.org · 11 years ago
  33. 84350a9 Remove empty VideoCodecGeneric struct. by pbos@webrtc.org · 11 years ago
  34. 37fb66d Changing to using factory methods for some classes in NetEq by henrik.lundin@webrtc.org · 11 years ago
  35. b3ff385 Temporarily disabling some more audio processing tests. by aluebs@webrtc.org · 11 years ago
  36. 59fdf2d Fix MouseCursorMonitorMac to return correct hotspot position. by sergeyu@chromium.org · 11 years ago
  37. 4b5d36e Removes the remaining uses of the list wrapper class and the list wrapper class. by henrike@webrtc.org · 11 years ago
  38. 8c03c4c WebRTCDemo: fix out-of-bounds array read. by fischman@webrtc.org · 11 years ago
  39. eed1f11 Updated Webrtc version to 3.49 by elham@webrtc.org · 11 years ago
  40. 083049f Removes usage of ListWrapper from several files. by henrike@webrtc.org · 11 years ago
  41. ad584b6 Revert "Activate ACM test for Android in modules_tests." (rev5364). by andresp@webrtc.org · 11 years ago
  42. 572cc28 Temporarily disabling audio processing tests. by aluebs@webrtc.org · 11 years ago
  43. 2904b71 Increasing simulation time for NetEqPerformanceTest by henrik.lundin@webrtc.org · 11 years ago
  44. b7c1e03 Enables robust delay validation in AEC delay logging. by bjornv@webrtc.org · 11 years ago
  45. 22470b5 Minor voice engine improvements around AGC. by andrew@webrtc.org · 11 years ago
  46. f3a2ef3 Android: Fixes crash when exiting WebRTCDemo. by henrike@webrtc.org · 11 years ago
  47. 3ab10f9 Activate ACM test for Android in modules_tests. by turaj@webrtc.org · 11 years ago
  48. 75e7da3 Permitting double start/stopping of streams. by pbos@webrtc.org · 11 years ago
  49. 5b1467d Adding NetEq performance test to webrtc_perf_tests by henrik.lundin@webrtc.org · 11 years ago
  50. 8d4f9ca Delay Estimator: Adds unittests for robust validation. by bjornv@webrtc.org · 11 years ago
  51. 201049c Fixing lint errors in NetEq4 by henrik.lundin@webrtc.org · 11 years ago
  52. 8b6867b Make code simpler on VCMEncodedCallback. by andresp@webrtc.org · 11 years ago
  53. 88ece35 Isolate register post encode callback in video coding module to simplify code and critical sections. by andresp@webrtc.org · 11 years ago
  54. 093b960 Isolate debug recording from video sender into a thread safe small class. by andresp@webrtc.org · 11 years ago
  55. 0b9d7ce Add another test case for AST/TOF switching. by solenberg@webrtc.org · 11 years ago
  56. 9b125e1 Delay Estimator: Converts a constant into a configurable parameter. by bjornv@webrtc.org · 11 years ago
  57. f3b2148 Init to 16 kHz in the fixed-point profile. by andrew@webrtc.org · 11 years ago
  58. 48b9892 Ensure capture_levels_ is sized correctly at init time. by andrew@webrtc.org · 11 years ago
  59. 2c358e2 Now printing less output from compare_videos.py. by phoglund@webrtc.org · 11 years ago
  60. e95dc25 Remove the requirement to call set_sample_rate_hz and friends. by andrew@webrtc.org · 11 years ago
  61. b3b6049 Remove outdated DestroyVideoSendStream comment. by pbos@webrtc.org · 11 years ago
  62. 49812e6 Wire up statistics in video send stream of new video engine api by sprang@webrtc.org · 11 years ago
  63. c902d88 Delay Estimator: robust_validation should be stored over a reset by bjornv@webrtc.org · 11 years ago
  64. e66b5bc Add include guards to forward_error_correction_internal.h by braveyao@webrtc.org · 11 years ago
  65. 1a6b274 Fix the include guard in transmit_mixer.h by braveyao@webrtc.org · 11 years ago
  66. 5d7992f Fix the include guard in transmit_mixer.h by braveyao@webrtc.org · 11 years ago
  67. 29975da Android build: make it quiet on success and not overly noisy on failure. by fischman@webrtc.org · 11 years ago
  68. fba1476 Fix the android clang bot for compiling with thread annotations. by andresp@webrtc.org · 11 years ago
  69. c9faf10 Add thread_annotations for clang targets. by andresp@webrtc.org · 11 years ago
  70. a9a7327 If the configured start bitrate is higher than the configures max by mflodman@webrtc.org · 11 years ago
  71. 9662535 Race condition in ViECapturer::RegisterObserver by sprang@webrtc.org · 11 years ago
  72. 91cebfc Update WebRTC to version 3.48 by tnakamura@webrtc.org · 11 years ago
  73. 4f1f5fa Add callbacks for receive channel RTCP statistics. by sprang@webrtc.org · 11 years ago
  74. 9edcdb0 Refactoring MediaOptimization so it can easily be turned into a thread-safe class. by andresp@webrtc.org · 11 years ago
  75. aacdb9f Integrate fake_network_pipe into direct_transport. by stefan@webrtc.org · 11 years ago
  76. a07c56f Remove metrics_unittests by kjellander@webrtc.org · 11 years ago
  77. 5596ac6 Remove media_file from VideoEngine dependencies. by pbos@webrtc.org · 11 years ago
  78. 5e0cbcf cpplint cleaning new API and its implementation files. by mflodman@webrtc.org · 11 years ago
  79. 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
  80. 5e252ac Speeding up CallTest.ReceivesAndRetransmitsNack and removed the random packet loss. by mflodman@webrtc.org · 11 years ago
  81. eb9ce11 Make MouseCursor mutable by sergeyu@chromium.org · 11 years ago
  82. 8d14e06 audio_processing_unittest: unbreak clang compilation. by fischman@webrtc.org · 11 years ago
  83. d138166 JNI Audio: remove dead members. by fischman@webrtc.org · 11 years ago
  84. 432e574 Revert "Make MouseCursor mutable" by sergeyu@chromium.org · 11 years ago
  85. 865be14 Make MouseCursor mutable by sergeyu@chromium.org · 11 years ago
  86. 5d13922 Stop transport in test SuspendBelowMinBitrate. by pbos@webrtc.org · 11 years ago
  87. 202d38d Added method for getting default module state and protect agains a by mflodman@webrtc.org · 11 years ago
  88. d60137f Modify video_render/ to allow a single old frame. by pbos@webrtc.org · 11 years ago
  89. 471354f Delete capturers after destroying streams in test. by pbos@webrtc.org · 11 years ago
  90. 5041831 Revert 5285 "Revert 5228 "Use the RTT from RtcpRttStats class if..." by asapersson@webrtc.org · 11 years ago
  91. 0443f6c Simplification of histogram normalization in delay estimator. by bjornv@webrtc.org · 11 years ago
  92. 46f7288 Revert r5294 to re-roll r5293. by pbos@webrtc.org · 11 years ago
  93. 1b3b8cb Adds robust validation functionality to the delay estimator by bjornv@webrtc.org · 11 years ago
  94. 1eb1008 Incorrect iterator++ in ModuleRtpRtcpImpl::RegisterVideoBitrateObserver by sprang@webrtc.org · 11 years ago
  95. c5a5713 Revert 5293 "Auto instantiate RBE depending on whether AST or TO..." by turaj@webrtc.org · 11 years ago
  96. 3bbc91e Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. by solenberg@webrtc.org · 11 years ago
  97. 7d7e63d Disabled tests on Android. The issue 2723 is filed to investigate the reason for tests failing. by turaj@webrtc.org · 11 years ago
  98. afceaca Measure pacer queue size based on when packets are inserted rather than captured. by stefan@webrtc.org · 11 years ago
  99. f1b92fd Fix jitter buffer delay estimate. by turaj@webrtc.org · 11 years ago
  100. 79d6daf Update talk to 58174641 together with http://review.webrtc.org/4319005/. by wu@webrtc.org · 11 years ago