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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / ee03b3b6ab79f68c9c7ee9d4e4de353b0d1a5192
  1. ee03b3b Disable libjingle_peerconnection_java_unittest by kjellander@webrtc.org · 10 years ago
  2. 69a0eca Removed unused mock methods in audio_processing by bjornv@webrtc.org · 10 years ago
  3. 65bf249 Add RTCP packet class. Adds packet types: sr, rr, bye, fir. by asapersson@webrtc.org · 10 years ago
  4. f59fce9 MIPS optimizations for AEC audio processing module by andrew@webrtc.org · 10 years ago
  5. 23c8d6b Updated WebRTC version to 3.50 TBR= wu@webrtc.org by elham@webrtc.org · 10 years ago
  6. f8722d5 Add an AlignedFreeDeleter and remove scoped_ptr_malloc. by andrew@webrtc.org · 10 years ago
  7. 27c1536 Minor improvement in RoundToInt16 implementation. by turaj@webrtc.org · 10 years ago
  8. 072bab2 Modified overuse detection thresholds. by asapersson@webrtc.org · 10 years ago
  9. 04e6137 Removing a variable that was never read by henrik.lundin@webrtc.org · 10 years ago
  10. a8498d9 ifdef the alsa code based on macro USE_X11 by fbarchard@google.com · 10 years ago
  11. cb06a6b Fix the break caused by r5579. by turaj@webrtc.org · 10 years ago
  12. 554bd44 Resolves memcheck issue in AudioCodingModuleTest. The issue is coditional jumnp based on uninitialized variable. by turaj@webrtc.org · 10 years ago
  13. 4b2ef8b Make WindowCapturerLinux handling window resize events. by jiayl@webrtc.org · 10 years ago
  14. 1d68dc1 Added architecture for compiling under chrome NaCl. by andresp@webrtc.org · 10 years ago
  15. 0657a86 This CL separate all ACM tests with new and old implementation of ACM and NetEq. The reason is to debug an issue with failure on Android try bots. We need to see if the error only occurs with the new ACM/NetEq, or if it is a flakiness that affects both. by tina.legrand@webrtc.org · 10 years ago
  16. 2fa9f7e Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR). by asapersson@webrtc.org · 10 years ago
  17. fa28e37 Removes VoERTP_RTCP::InsertExtraRTPPacket. by henrika@webrtc.org · 10 years ago
  18. 4e266ff Fix DesktopAndCursorComposer not to crash by sergeyu@chromium.org · 10 years ago
  19. 8513671 Move the volume quantization workaround from VoE to AGC. by andrew@webrtc.org · 10 years ago
  20. c8529ab Remove obsolete voe_unit_test. by solenberg@webrtc.org · 10 years ago
  21. 387b944 Don't print a warning if RTPPacketHistory::SetStorePacketStatus is called by mflodman@webrtc.org · 10 years ago
  22. aeb2e9e Remove unnecessary warnings. by turaj@webrtc.org · 10 years ago
  23. ae50521 Remove external encryption API for VoE. by solenberg@webrtc.org · 10 years ago
  24. 3f3e951 Incorrect overhead calculation when using FEC + RTP extension headers. by sprang@webrtc.org · 10 years ago
  25. 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
  26. e3da97c Misc small nits in NetEq by henrik.lundin@webrtc.org · 10 years ago
  27. 6f8b051 AudioProcessing is not a Module. by andrew@webrtc.org · 10 years ago
  28. cd15790 Refactoring common_audio/signal_processing: Removed two macros used by isac only. by bjornv@webrtc.org · 10 years ago
  29. 46b22d8 Adding a critical section missing in r5543. by stefan@webrtc.org · 10 years ago
  30. 0f5010d Initialize output_will_be_muted_. by andrew@webrtc.org · 10 years ago
  31. 8e98655 Increase overuse and normal use thresholds for Mac. by asapersson@webrtc.org · 10 years ago
  32. 8cb4c8d Fixes a race when writing to send_padding_. by stefan@webrtc.org · 10 years ago
  33. 5d5e87d Small refactoring of NetEq unittest for CNG with clock drift by henrik.lundin@webrtc.org · 10 years ago
  34. f4f1d1a Add a method to inform AudioProcessing that its output will be muted. by andrew@webrtc.org · 10 years ago
  35. 96b5dfa Change the type of propagation delta from int64 to int. by jiayl@webrtc.org · 10 years ago
  36. 9e3cb7b Initialize key_pressed_. by andrew@webrtc.org · 10 years ago
  37. 6ec403d Add a keypress field to the audioproc debug proto. by andrew@webrtc.org · 10 years ago
  38. 6cfc58d Set pacing bitrates in SetEncoder. by pbos@webrtc.org · 10 years ago
  39. 0fd5775 Remove unused and not working voe_extended_test. by solenberg@webrtc.org · 10 years ago
  40. 48a5cdb Reduce mixing threshold in test to avoid flakiness. by andrew@webrtc.org · 10 years ago
  41. 247df83 Add an interface for accepting keypress signals to AudioProcessing. by andrew@webrtc.org · 10 years ago
  42. 4112a51 Rename merged webrtc lib to libwebrtc_merged.a. by andrew@webrtc.org · 10 years ago
  43. e2d2804 Remove "Too long processing time of Incoming frame" logspam. by fischman@webrtc.org · 10 years ago
  44. ff986f4 Add boundary checking to supress gcc 4.8.3 warning. by turaj@webrtc.org · 10 years ago
  45. ddbd31e Remove ViE external encryption API. by solenberg@webrtc.org · 10 years ago
  46. e08d28e Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file. by michaelbai@google.com · 10 years ago
  47. dd1d6ce Restore mixing integration tests. by andrew@webrtc.org · 10 years ago
  48. 89a0796 Revert "Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file." by michaelbai@google.com · 10 years ago
  49. a68379b Add stats of incoming frame delays for debugging bandwidth estimation. by jiayl@webrtc.org · 10 years ago
  50. bac08b3 Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file. by michaelbai@google.com · 10 years ago
  51. 85835a0 Add experiment: SkipEncodingUnusedStreams by sprang@webrtc.org · 10 years ago
  52. c0b1926 Roll chromium_revision 245382:249215 by kjellander@webrtc.org · 10 years ago
  53. 992076c Fix WindowCapturerWin to unselect bitmap before destroying DC. by sergeyu@chromium.org · 10 years ago
  54. f2c28a0 Make VideoReceiveStream::GetStats() const. by sprang@webrtc.org · 10 years ago
  55. fa7c4c4 Wire up statistics in video receive stream of new API by sprang@webrtc.org · 10 years ago
  56. 8a431ef Plot the capacity of a trace-based delivery filter. by stefan@webrtc.org · 10 years ago
  57. 74ffc7b Use system's cpu_features library by michaelbai@google.com · 10 years ago
  58. 94c5692 Add delay and send/receive throughput plots to BWE simulation. by stefan@webrtc.org · 10 years ago
  59. 618154f Implementing replacement audio support in neteq_rtpplay by henrik.lundin@webrtc.org · 10 years ago
  60. 395e1b4 Fixing a bug in DummyRTPpacket by henrik.lundin@webrtc.org · 10 years ago
  61. 680d3ca Update AudioProcessing::Create docs. by andrew@webrtc.org · 10 years ago
  62. 1ca2c1f Fix a cursor capturing issue on Windows. by jiayl@webrtc.org · 10 years ago
  63. 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
  64. 1eba384 Revert "Disable the test: DtmfTest.ManualSuccessfullySendsIn/OutOfBandTelephoneEvents" by pbos@webrtc.org · 10 years ago
  65. 4f41016 Fix locking in LoopBackTransport::StorePacket. by pbos@webrtc.org · 10 years ago
  66. 3634228 Trivial rename of non-compile time consts. by andrew@webrtc.org · 10 years ago
  67. 0a7d406 Disable the test: DtmfTest.ManualSuccessfullySendsIn/OutOfBandTelephoneEvents by marpan@webrtc.org · 10 years ago
  68. cc8de94 Wire up feedback to VideoSender. by stefan@webrtc.org · 10 years ago
  69. 54a9a32 Re-enabling audio processing tests by aluebs@webrtc.org · 10 years ago
  70. 910910a Moved the new OnData interface to AudioTranport, and expose the AudioTransport pointer via voe_base by xians@webrtc.org · 10 years ago
  71. 2b38fc1 Implement single monitor capture on Mac. by jiayl@webrtc.org · 10 years ago
  72. 622a139 Fixing test name for NetEqPerformanceTest by henrik.lundin@webrtc.org · 10 years ago
  73. 4b1817f Add configuration for cpu overuse detection to video send stream. by asapersson@webrtc.org · 10 years ago
  74. aaac959 Add gyp_webrtc script to generate projects. by kjellander@webrtc.org · 10 years ago
  75. 098ffb2 Add BWE tools for parsing RTP files. by stefan@webrtc.org · 10 years ago
  76. 28429ea Fix the mouse cursor offset issue on Mac. by jiayl@webrtc.org · 10 years ago
  77. 25bec2a Move out typing detection to its own class. by henrikg@webrtc.org · 10 years ago
  78. c4fa5fa Moves the display reconfiguration callback into a separate class, by jiayl@webrtc.org · 10 years ago
  79. 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
  80. fdb30d1 Fix race when deleting video receive streams in Call. by solenberg@webrtc.org · 11 years ago
  81. 50afcf1 Fix deadlock in video_receiver.cc. by stefan@webrtc.org · 11 years ago
  82. 49e9e15 Connect webrtc::Config to WrappingBitrateEstimator by henrik.lundin@webrtc.org · 11 years ago
  83. 9d5a547 Add Config struct for experimental AGC. by andrew@webrtc.org · 11 years ago
  84. a1e140d Revert 5444 "Revert 5421 "Fix deadlock on register/unregister ob..." by mallinath@webrtc.org · 11 years ago
  85. 78fae4b Add clean test to NetEq perf test by henrik.lundin@webrtc.org · 11 years ago
  86. 76d028d VideoCaptureAndroid: stop preview in opposite order of starting. by fischman@webrtc.org · 11 years ago
  87. c091c50 Revert 5421 "Fix deadlock on register/unregister observer while ..." by mallinath@webrtc.org · 11 years ago
  88. 2ac9c51 Avoid potential dead lock in StreamStatisticianImpl by sprang@webrtc.org · 11 years ago
  89. 5a2228b Race condition in RTPSender::UpdateRtpStats by sprang@webrtc.org · 11 years ago
  90. 48ac0da Drop early packets when not sending in TransportAdapter. by sprang@webrtc.org · 11 years ago
  91. 4c9a4b4 Fix bug introduced during replace of list wrapper with std equivalents in r5378. by andresp@webrtc.org · 11 years ago
  92. 0b86761 Race in StreamStatisticianImpl::GetStatistics vs. ::IncomingPacket by sprang@webrtc.org · 11 years ago
  93. 778e73f Fix "field '_testNo' is uninitialized" warnings. by pbos@webrtc.org · 11 years ago
  94. ffd4269 Always initialize Trace in Call TraceDispatcher. by pbos@webrtc.org · 11 years ago
  95. 6a6e3eb Add a Config parameter to AudioProcessing::Create(). by andrew@webrtc.org · 11 years ago
  96. db9ad63 Android, WebRTCDemo: fixes crash issue when pressing switch camera button on devices with only one camera. by henrike@webrtc.org · 11 years ago
  97. d476500 Remove loopback setup in RtpRtcpImplTest. Changed to use two separate rtp/rtcp modules. by asapersson@webrtc.org · 11 years ago
  98. 6726cce Add new API (webrtc.gyp:webrtc) to merge_libs.gyp. by pbos@webrtc.org · 11 years ago
  99. 21b46dd Add trace-based delivery filter to BWE test framework. by stefan@webrtc.org · 11 years ago
  100. c766775 Wire up RTX in VideoReceiveStream. by pbos@webrtc.org · 11 years ago