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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 3f3e951cfc524543c58839ed380b170a0433bea0
  1. 3f3e951 Incorrect overhead calculation when using FEC + RTP extension headers. by sprang@webrtc.org · 10 years ago
  2. 15e3511 Reset estimate if no frame has been seen for a certain time (to avoid large jitter if stop sending). by asapersson@webrtc.org · 10 years ago
  3. e3da97c Misc small nits in NetEq by henrik.lundin@webrtc.org · 10 years ago
  4. 6f8b051 AudioProcessing is not a Module. by andrew@webrtc.org · 10 years ago
  5. cd15790 Refactoring common_audio/signal_processing: Removed two macros used by isac only. by bjornv@webrtc.org · 10 years ago
  6. 46b22d8 Adding a critical section missing in r5543. by stefan@webrtc.org · 10 years ago
  7. 0f5010d Initialize output_will_be_muted_. by andrew@webrtc.org · 10 years ago
  8. 8e98655 Increase overuse and normal use thresholds for Mac. by asapersson@webrtc.org · 10 years ago
  9. 8cb4c8d Fixes a race when writing to send_padding_. by stefan@webrtc.org · 10 years ago
  10. 5d5e87d Small refactoring of NetEq unittest for CNG with clock drift by henrik.lundin@webrtc.org · 10 years ago
  11. f4f1d1a Add a method to inform AudioProcessing that its output will be muted. by andrew@webrtc.org · 10 years ago
  12. 96b5dfa Change the type of propagation delta from int64 to int. by jiayl@webrtc.org · 10 years ago
  13. 9e3cb7b Initialize key_pressed_. by andrew@webrtc.org · 10 years ago
  14. 6ec403d Add a keypress field to the audioproc debug proto. by andrew@webrtc.org · 10 years ago
  15. 6cfc58d Set pacing bitrates in SetEncoder. by pbos@webrtc.org · 10 years ago
  16. 0fd5775 Remove unused and not working voe_extended_test. by solenberg@webrtc.org · 10 years ago
  17. 48a5cdb Reduce mixing threshold in test to avoid flakiness. by andrew@webrtc.org · 10 years ago
  18. 247df83 Add an interface for accepting keypress signals to AudioProcessing. by andrew@webrtc.org · 10 years ago
  19. 4112a51 Rename merged webrtc lib to libwebrtc_merged.a. by andrew@webrtc.org · 10 years ago
  20. e2d2804 Remove "Too long processing time of Incoming frame" logspam. by fischman@webrtc.org · 10 years ago
  21. ff986f4 Add boundary checking to supress gcc 4.8.3 warning. by turaj@webrtc.org · 10 years ago
  22. ddbd31e Remove ViE external encryption API. by solenberg@webrtc.org · 10 years ago
  23. e08d28e Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file. by michaelbai@google.com · 10 years ago
  24. dd1d6ce Restore mixing integration tests. by andrew@webrtc.org · 10 years ago
  25. 89a0796 Revert "Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file." by michaelbai@google.com · 10 years ago
  26. a68379b Add stats of incoming frame delays for debugging bandwidth estimation. by jiayl@webrtc.org · 10 years ago
  27. bac08b3 Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file. by michaelbai@google.com · 10 years ago
  28. 85835a0 Add experiment: SkipEncodingUnusedStreams by sprang@webrtc.org · 10 years ago
  29. c0b1926 Roll chromium_revision 245382:249215 by kjellander@webrtc.org · 10 years ago
  30. 992076c Fix WindowCapturerWin to unselect bitmap before destroying DC. by sergeyu@chromium.org · 10 years ago
  31. f2c28a0 Make VideoReceiveStream::GetStats() const. by sprang@webrtc.org · 10 years ago
  32. fa7c4c4 Wire up statistics in video receive stream of new API by sprang@webrtc.org · 10 years ago
  33. 8a431ef Plot the capacity of a trace-based delivery filter. by stefan@webrtc.org · 10 years ago
  34. 74ffc7b Use system's cpu_features library by michaelbai@google.com · 10 years ago
  35. 94c5692 Add delay and send/receive throughput plots to BWE simulation. by stefan@webrtc.org · 10 years ago
  36. 618154f Implementing replacement audio support in neteq_rtpplay by henrik.lundin@webrtc.org · 10 years ago
  37. 395e1b4 Fixing a bug in DummyRTPpacket by henrik.lundin@webrtc.org · 10 years ago
  38. 680d3ca Update AudioProcessing::Create docs. by andrew@webrtc.org · 10 years ago
  39. 1ca2c1f Fix a cursor capturing issue on Windows. by jiayl@webrtc.org · 10 years ago
  40. 55367d5 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
  41. 1eba384 Revert "Disable the test: DtmfTest.ManualSuccessfullySendsIn/OutOfBandTelephoneEvents" by pbos@webrtc.org · 10 years ago
  42. 4f41016 Fix locking in LoopBackTransport::StorePacket. by pbos@webrtc.org · 10 years ago
  43. 3634228 Trivial rename of non-compile time consts. by andrew@webrtc.org · 10 years ago
  44. 0a7d406 Disable the test: DtmfTest.ManualSuccessfullySendsIn/OutOfBandTelephoneEvents by marpan@webrtc.org · 10 years ago
  45. cc8de94 Wire up feedback to VideoSender. by stefan@webrtc.org · 10 years ago
  46. 54a9a32 Re-enabling audio processing tests by aluebs@webrtc.org · 10 years ago
  47. 910910a Moved the new OnData interface to AudioTranport, and expose the AudioTransport pointer via voe_base by xians@webrtc.org · 10 years ago
  48. 2b38fc1 Implement single monitor capture on Mac. by jiayl@webrtc.org · 10 years ago
  49. 622a139 Fixing test name for NetEqPerformanceTest by henrik.lundin@webrtc.org · 10 years ago
  50. 4b1817f Add configuration for cpu overuse detection to video send stream. by asapersson@webrtc.org · 10 years ago
  51. aaac959 Add gyp_webrtc script to generate projects. by kjellander@webrtc.org · 10 years ago
  52. 098ffb2 Add BWE tools for parsing RTP files. by stefan@webrtc.org · 10 years ago
  53. 28429ea Fix the mouse cursor offset issue on Mac. by jiayl@webrtc.org · 10 years ago
  54. 25bec2a Move out typing detection to its own class. by henrikg@webrtc.org · 10 years ago
  55. c4fa5fa Moves the display reconfiguration callback into a separate class, by jiayl@webrtc.org · 10 years ago
  56. 4f23307 Added new capture callback interface to pass the capture callback to a specific voe channel from libjingle webrtcvoiceengine.cc. by xians@webrtc.org · 10 years ago
  57. fdb30d1 Fix race when deleting video receive streams in Call. by solenberg@webrtc.org · 10 years ago
  58. 50afcf1 Fix deadlock in video_receiver.cc. by stefan@webrtc.org · 10 years ago
  59. 49e9e15 Connect webrtc::Config to WrappingBitrateEstimator by henrik.lundin@webrtc.org · 10 years ago
  60. 9d5a547 Add Config struct for experimental AGC. by andrew@webrtc.org · 10 years ago
  61. a1e140d Revert 5444 "Revert 5421 "Fix deadlock on register/unregister ob..." by mallinath@webrtc.org · 10 years ago
  62. 78fae4b Add clean test to NetEq perf test by henrik.lundin@webrtc.org · 11 years ago
  63. 76d028d VideoCaptureAndroid: stop preview in opposite order of starting. by fischman@webrtc.org · 11 years ago
  64. c091c50 Revert 5421 "Fix deadlock on register/unregister observer while ..." by mallinath@webrtc.org · 11 years ago
  65. 2ac9c51 Avoid potential dead lock in StreamStatisticianImpl by sprang@webrtc.org · 11 years ago
  66. 5a2228b Race condition in RTPSender::UpdateRtpStats by sprang@webrtc.org · 11 years ago
  67. 48ac0da Drop early packets when not sending in TransportAdapter. by sprang@webrtc.org · 11 years ago
  68. 4c9a4b4 Fix bug introduced during replace of list wrapper with std equivalents in r5378. by andresp@webrtc.org · 11 years ago
  69. 0b86761 Race in StreamStatisticianImpl::GetStatistics vs. ::IncomingPacket by sprang@webrtc.org · 11 years ago
  70. 778e73f Fix "field '_testNo' is uninitialized" warnings. by pbos@webrtc.org · 11 years ago
  71. ffd4269 Always initialize Trace in Call TraceDispatcher. by pbos@webrtc.org · 11 years ago
  72. 6a6e3eb Add a Config parameter to AudioProcessing::Create(). by andrew@webrtc.org · 11 years ago
  73. db9ad63 Android, WebRTCDemo: fixes crash issue when pressing switch camera button on devices with only one camera. by henrike@webrtc.org · 11 years ago
  74. d476500 Remove loopback setup in RtpRtcpImplTest. Changed to use two separate rtp/rtcp modules. by asapersson@webrtc.org · 11 years ago
  75. 6726cce Add new API (webrtc.gyp:webrtc) to merge_libs.gyp. by pbos@webrtc.org · 11 years ago
  76. 21b46dd Add trace-based delivery filter to BWE test framework. by stefan@webrtc.org · 11 years ago
  77. c766775 Wire up RTX in VideoReceiveStream. by pbos@webrtc.org · 11 years ago
  78. aa2c3ae Fix deadlock on register/unregister observer while there is a an going callback. by andresp@webrtc.org · 11 years ago
  79. fbbbc11 Fix array declarations in aec_rdft.h. by andrew@webrtc.org · 11 years ago
  80. bb55b6d Set NACKed packet to -1 in TestNackRetransmission. by pbos@webrtc.org · 11 years ago
  81. 224933c Add callbacks for receive channel RTP statistics by sprang@webrtc.org · 11 years ago
  82. bd389d5 Android, fixes crash on devices with only front cameras. by henrike@webrtc.org · 11 years ago
  83. ccee3c3 Output logs to stderr from voe_cmd_test by default. by andrew@webrtc.org · 11 years ago
  84. e412bb6 Android example apps: fixes issue where useful failure information was suppressed. by henrike@webrtc.org · 11 years ago
  85. 7262655 Potential dead lock in receive statistics by sprang@webrtc.org · 11 years ago
  86. c85c797 Fix for libtalkmobile build error bug=b/12549061 by elham@webrtc.org · 11 years ago
  87. e1b9880 Removes script for generating supplement.gypi also adds git ignore for tools/gn. by henrike@webrtc.org · 11 years ago
  88. 2752bb1 Set up receiver RTX config using a std::map. by pbos@webrtc.org · 11 years ago
  89. 64339f0 Add configuration and test for extended RTCP reference time reports to new video api. by asapersson@webrtc.org · 11 years ago
  90. b6a78e5 Android, OpenSlDemo: moved to webrtc/examples/android/opensl_loopback by henrike@webrtc.org · 11 years ago
  91. 5426f84 Implement screen enumeration and individual screen capturing for Windows. by jiayl@webrtc.org · 11 years ago
  92. ac4c7ea Android, OpenSlDemo: fixes issue where app would crash as soon as the application is started. by henrike@webrtc.org · 11 years ago
  93. 6b84a14 Android, WebRTCDemo: fix issue where changing remote IP was not working properly. by henrike@webrtc.org · 11 years ago
  94. 1240a3d Add full path to headers by aluebs@webrtc.org · 11 years ago
  95. d8163b6 Adds back set_sample_rate_hz() when Init is called in recordings. by bjornv@webrtc.org · 11 years ago
  96. f180890 MIPS optimizations for NS audio processing module by andrew@webrtc.org · 11 years ago
  97. 08ef082 Fix crash in MouseCursor::CopyOf() by sergeyu@chromium.org · 11 years ago
  98. 0a71d11 Exclude protoc objects from merge_libs.py. by andrew@webrtc.org · 11 years ago
  99. df9f867 Update needed to MockScreenCapturer after new methods addition to webrtc::ScreenCapturer. by mallinath@webrtc.org · 11 years ago
  100. 7f1a443 Extends the ScreenCapturer interface for individual display screen cast. by jiayl@webrtc.org · 11 years ago