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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 0e098e0b0340e3bb08c12cbc98c741d179cb2a63
  1. 0725df6 Extends max sample rate from 96kHz to 192kHz on the input side. by henrika@webrtc.org · 10 years ago
  2. a0acb1f sink_filter_ds.cc: add lock to Receive procedure to Pause(). by braveyao@webrtc.org · 10 years ago
  3. 5ae01bf Make ACM2 the default in voe_cmd_test. by andrew@webrtc.org · 10 years ago
  4. 15f109e Added simulations of capacity variations and wifi recordings. by stefan@webrtc.org · 10 years ago
  5. 53b062b Roll chromium_revision 255773:260462 by kjellander@webrtc.org · 10 years ago
  6. 7a8dee4 Fix ARM64 detection. by andrew@webrtc.org · 10 years ago
  7. 8f5ab19 VoiceEngine(iOS & Android): removed NOT_SUPPORTED by fischman@webrtc.org · 10 years ago
  8. 24532e0 Add tests for the RBE RemoveStream() API. by solenberg@webrtc.org · 10 years ago
  9. 7c3f468 VoE Channel: Don't register codecs when stopping receiver by henrik.lundin@webrtc.org · 10 years ago
  10. 9136607 Restore support for code coverage in WebRTC by kjellander@webrtc.org · 10 years ago
  11. ad239fe Add arm64 to typedefs.h by andrew@webrtc.org · 10 years ago
  12. 4c6d59a Allow loopback tests to do TURN when served from webrtc.googlecode.com. by andresp@webrtc.org · 10 years ago
  13. 66f5371 Add svn mime-type properties to loopback_test files so they can be served from: by andresp@webrtc.org · 10 years ago
  14. bae92ab Don't disable experimental AGC in audioproc. by andrew@webrtc.org · 10 years ago
  15. 6c57efd Re-submit: rev5775 by andresp@webrtc.org · 10 years ago
  16. 0ab635c Makes ScreenCapturerMac exclude the window specified in DesktopCapturer::SetExcludedWindow. by jiayl@webrtc.org · 10 years ago
  17. 37f807f Add API to allow deducting bitrate from incoming estimates before the capacity is distributed among outgoing video streams. For example, this can be used to reserve space for audio streams. by solenberg@webrtc.org · 10 years ago
  18. 1e05528 Protect write of send_target_bitrate. by andresp@webrtc.org · 10 years ago
  19. 0027f0a Make RTPHeaderParser skip over unknown RTP header extensions rather than bail out. by solenberg@webrtc.org · 10 years ago
  20. 765ea72 Revert 5775 "Modify bitrate controller to update bitrate based o..." by andrew@webrtc.org · 10 years ago
  21. f50914a Removing VideoCodecDerived and moving methods inside VideoCodec. by mallinath@webrtc.org · 10 years ago
  22. 0ac0bca Updated WebRTC version to 3.51 by elham@webrtc.org · 10 years ago
  23. a090cc7 iOS video_capture: move @private vars to impl. by fischman@webrtc.org · 10 years ago
  24. 09fb237 Fix race condition in RTPSEnder. by sprang@webrtc.org · 10 years ago
  25. 0b11715 Change sprintf format string from %zu to %i by henrik.lundin@webrtc.org · 10 years ago
  26. 539bbde Modify bitrate controller to update bitrate based on process call and not by andresp@webrtc.org · 10 years ago
  27. 1f49208 Adding API for setting bandwidth estimation configurations. by stefan@webrtc.org · 10 years ago
  28. 892cd1f iOS video_capture: start camera in the background. by fischman@webrtc.org · 10 years ago
  29. 1dd9fb5 iOS VideoEngine: move video_{capture,render} to ARC. by fischman@webrtc.org · 10 years ago
  30. 1a19092 Add configuration for ability to use the encode usage measure for triggering overuse/underuse. by asapersson@webrtc.org · 10 years ago
  31. 50ac4d6 Implement ViE forwarding to RBE of packets for BWE coming in through the ViENetwork::ReceivedBWEPacket API. by solenberg@webrtc.org · 10 years ago
  32. 85101db Have changes to REMB trigger RTCP to be sent immediately. by stefan@webrtc.org · 10 years ago
  33. bb1d4c7 DelayEstimator: Updates delay_quality and adds soft reset. by bjornv@webrtc.org · 10 years ago
  34. b45cf1e Run Opus with lower complexity setting on Android, iOS and/or ARM by tina.legrand@webrtc.org · 10 years ago
  35. 5ca38d1 Add targetBitrate to VideoCodec struct. by pbos@webrtc.org · 10 years ago
  36. 825acb1 Disabled some of the remote bitrate estimator baseline tests. by stefan@webrtc.org · 10 years ago
  37. 5f804f8 VoE changes to allow forwarding of packets from VoE to ViE BWE. by solenberg@webrtc.org · 10 years ago
  38. 61e9201 Add fir_filter to common_audio by aluebs@webrtc.org · 10 years ago
  39. 9b2b8ec Add AIMD option to BWE API. by stefan@webrtc.org · 10 years ago
  40. 95b5fde ACM2/NetEq4 did not decode Opus in stereo by tina.legrand@webrtc.org · 10 years ago
  41. 209791d Refactor in BitrateController module. by andresp@webrtc.org · 10 years ago
  42. 61e72f0 Fixing crash in video_render_tests in release mode. by henrikg@webrtc.org · 10 years ago
  43. 23e07d8 Remove locks in SendSideBandwidthEstimation since those are only accessed while owning locks in by andresp@webrtc.org · 10 years ago
  44. 97d92ed Adding FEC support in NetEq 4. by minyue@webrtc.org · 10 years ago
  45. d327be4 Fix "unreachable code" warnings (MSVC warning 4702) in webrtc. by pbos@webrtc.org · 10 years ago
  46. 40fee00 Adding operator== and != methods for CodecInst and VideoCodec structures. by mallinath@webrtc.org · 10 years ago
  47. bcee6b7 Use codec width/height as the encoded_image width/height. by wu@webrtc.org · 10 years ago
  48. 88fa18b Changing the buffer size (slots) to 1.5 seconds @ 30 ms packets by henrik.lundin@webrtc.org · 10 years ago
  49. 27bd3be Add ability to configure cpu overuse options via an API. by asapersson@webrtc.org · 10 years ago
  50. 55f4fe8 Prevent playout delay wrap-around in VoiceEngine by henrik.lundin@webrtc.org · 10 years ago
  51. 4d9df07 Removes error printout in voe_cmd_test which was caused by attempts to transmit RTCP packets even if a transport object was not registered. by henrika@webrtc.org · 10 years ago
  52. a183edc Extend perf tests to perform rampup on single stream. by andresp@webrtc.org · 10 years ago
  53. b903292 Adjust the captured window rect when the window is maximized. by jiayl@webrtc.org · 10 years ago
  54. 5e18933 Properly account for retransmitted packets when not using the pacer. by stefan@webrtc.org · 10 years ago
  55. f9d5709 Fixes RTX related bugs. by stefan@webrtc.org · 10 years ago
  56. 292e7f6 Disabling SendsSetSimulcastSsrcs. by pbos@webrtc.org · 10 years ago
  57. ce06c77 Revert "Changing the buffer size (in packets) to 1.5 seconds @ 30 ms packets" by henrik.lundin@webrtc.org · 10 years ago
  58. e8db41f Changing the buffer size (in packets) to 1.5 seconds @ 30 ms packets by henrik.lundin@webrtc.org · 10 years ago
  59. 16c3dcc Disable flaky CanSwitchToUseAllSsrcs. by pbos@webrtc.org · 10 years ago
  60. bef6e62 Simplify pacer interface. by pbos@webrtc.org · 10 years ago
  61. f39df52 Remove internal codecs from VideoSendStream. by pbos@webrtc.org · 10 years ago
  62. 3a70e6e Fix a deadlock in ViEEncoder::DeliverFrame. by wuchengli@chromium.org · 10 years ago
  63. 3c85e1e Adds a method to WindowCapturer to bring a window to the front. by jiayl@webrtc.org · 10 years ago
  64. d8e33dc Adding thread annotations to NetEq4 by henrik.lundin@webrtc.org · 10 years ago
  65. e85416a Add #include <cstdlib> for std::abs. by pbos@webrtc.org · 10 years ago
  66. 48bbc5a Resolves TSan v2 warnings in voe_auto_test. by henrika@webrtc.org · 10 years ago
  67. ebae8bb Re-comitting r5711: "Fixing a flaky test in video_engine_tests" by henrik.lundin@webrtc.org · 10 years ago
  68. 8d3c410 Revert 5711 "Fixing a flaky test in video_engine_tests" by turaj@webrtc.org · 10 years ago
  69. f9a6ab0 Fixing a flaky test in video_engine_tests by henrik.lundin@webrtc.org · 10 years ago
  70. 48191a6 Small refactor on send_side_bandwidth_estimation. by andresp@webrtc.org · 10 years ago
  71. ca626eb Refactor rampup tests: by andresp@webrtc.org · 10 years ago
  72. 3bf1f38 Tool to establish a loopback call via apprtc turn server. by andresp@webrtc.org · 10 years ago
  73. 81fd3e7 References to includes in third_party should be relative, not absolute. by sprang@webrtc.org · 10 years ago
  74. 340b16e Add support for YUV4MPEG file reading to tools files. (Minor fix). by mcasas@webrtc.org · 10 years ago
  75. bc5c7bc Add support for YUV4MPEG file reading to tools files. by mcasas@webrtc.org · 10 years ago
  76. b6fb76a Fix a bug where network freeze during CNG causes delay by henrik.lundin@webrtc.org · 10 years ago
  77. 78c101a Remove legacy weirdness in Merge::Downsample by henrik.lundin@webrtc.org · 10 years ago
  78. 3c00b1c Stopping network threads before tearing down test by henrik.lundin@webrtc.org · 10 years ago
  79. 64e7141 Race condition in RTPSender by sprang@webrtc.org · 10 years ago
  80. 7f78ae5 Add max delay to trace based filters and enhances drop tail queues with delay statistics. by stefan@webrtc.org · 10 years ago
  81. 15cf717 Re-landing "Routing SuspendChange to VideoSendStream::Stats" by henrik.lundin@webrtc.org · 10 years ago
  82. 9420a1f Implement minimum transmit bitrate. by pbos@webrtc.org · 10 years ago
  83. 41da329 Enable all RampUpTest.UpDownUp* tests by henrik.lundin@webrtc.org · 10 years ago
  84. c53e587 Replace labs with std::abs. by pbos@webrtc.org · 10 years ago
  85. 34bf4d6 Disable all protobuf dependent targets when enable_protobuf=0. by andrew@webrtc.org · 10 years ago
  86. 1507b50 Enable VS2013 for Windows compilation by default. by kjellander@webrtc.org · 10 years ago
  87. af634a2 Remove platform-specific code from new-API tests. by pbos@webrtc.org · 10 years ago
  88. 10c488f Implement a test for an old corner-case in NetEq by henrik.lundin@webrtc.org · 10 years ago
  89. fb9e586 Developing NetEqImpl unit tests by henrik.lundin@webrtc.org · 10 years ago
  90. 0a1a92b Disable TestOpusNewACM on Android. by andrew@webrtc.org · 10 years ago
  91. f951dfc Revert "Routing SuspendChange to VideoSendStream::Stats" by henrik.lundin@webrtc.org · 10 years ago
  92. 17757d1 Reorder includes in audio_processing_impl_unittest. by andrew@webrtc.org · 10 years ago
  93. 5ddb6fe Voice Engine GetRemoteCSRCs should return the CSRCs from rtp_receiver_ instead of _rtpRtcpModule now. by braveyao@webrtc.org · 10 years ago
  94. 697cd78 Routing SuspendChange to VideoSendStream::Stats by henrik.lundin@webrtc.org · 10 years ago
  95. 253e312 Classes and tests for audio an classifier. The class can be used to classify whether a frame of audio contains speech or music. The classifier uses the music/speech classifier in Opus. by jan.skoglund@webrtc.org · 10 years ago
  96. 39dd100 Add tests and modify tools for new float deinterleaved interface. by andrew@webrtc.org · 10 years ago
  97. e2af272 Add a float interface to PushSincResampler. by andrew@webrtc.org · 10 years ago
  98. bcec718 iOS video_render: omit no-op setNeedsDisplay by fischman@webrtc.org · 10 years ago
  99. dd5c99c AppRTCDemo(iOS): video support; part 1 of 2: webrtc/. by fischman@webrtc.org · 10 years ago
  100. f422ce1 Adding a link to issue by henrik.lundin@webrtc.org · 10 years ago