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gerrit-public.fairphone.software / platform / external / webrtc / 280ad1514e44bf6717e5871526dd4632f759eb3d
  1. 280ad15 Update test code to use I420Buffer when writing pixel data. by Niels Möller · 9 years ago
  2. e9a3c7f Wrap ScreenCapturer with ScreenCapturerDifferWrapper by zijiehe · 9 years ago
  3. 2dc4cde Correcting the enabling of the level controller in the audio processing simulator by peah · 9 years ago
  4. 74e1a4f PeerConnection[Interface]::GetStats(RTCStatsCollectorCallback*) added. by hbos · 9 years ago
  5. fe7d091 Fixing a couple cases that cause ProcessAllMessageQueues to hang. by Taylor Brandstetter · 9 years ago
  6. 9ecb085 Adding logs to track potential cause of not starting port allocation. by Honghai Zhang · 9 years ago
  7. fb2c1d0 Add voe_cmd_test to voice_engine/BUILD.gn (and remove it from voice_engine.gyp, together with the channel_transport gyp target) by solenberg · 9 years ago
  8. 232c56b Add logging available fps ranges to Camera2Session. by sakal · 9 years ago
  9. bb716da Fix android_junit_tests and add a GN target for them. by sakal · 9 years ago
  10. 530b3f5 Merge RtcpReceiver::Handle<Packet>Item functions into Handle<Packet> by Danil Chapovalov · 9 years ago
  11. 9fdbda6 Revert of Replace interface VideoCapturerInput with VideoSinkInterface. (patchset #13 id:280001 of https://codereview.webrtc.org/2257413002/ ) by perkj · 9 years ago
  12. 95a226f Replace VideoCapturerInput with VideoSinkInterface. by perkj · 9 years ago
  13. 91511f1 Split RtcpReceiver::HandleSenderReceiverReport into two functions by Danil Chapovalov · 9 years ago
  14. edebf45 Use I420Buffer rather than VideoFrameBuffer when writing pixels. by nisse · 9 years ago
  15. 8faf9e0 Removed the const char* (StaticString) type from RTCStatsMember. by hbos · 9 years ago
  16. 4ed5b9f Android SurfaceViewRenderer: Create EGL context on render thread by Magnus Jedvert · 9 years ago
  17. ec62374 Reland of Add arraysize to rtc_base_approved. Remove dependency of audio_device on rtc_base. (patchset #1 id:1 of https://codereview.webrtc.org/2340253003/ ) by maxmorin · 9 years ago
  18. 0a6e0dc Disable all screen-capturer tests by stefan · 9 years ago
  19. 17f008b GYP: Remove targets inside include_tests==1 that are converted to GN. by kjellander · 9 years ago
  20. 8b28b80 Assume ProjectRootPath() equals ../.. in Desktop by ehmaldonado · 9 years ago
  21. d268d6f Stash non layer-sync frames if we have not yet received an earlier frame for this layer. by philipel · 9 years ago
  22. ebc34e7 [GN] Add rtc_sdk_framework_objc target to GN by kthelgason · 9 years ago
  23. 11ace15 The VoE functionality to apply receive-side processing to VoE channels is unused. I'm removing it so we can avoid instantiating a full APM per channel (and thus also for webrtc::AudioSendStream and webrtc::AudioReceiveStream), and then never use it. by solenberg · 9 years ago
  24. 70d0124 MB: Change Android Clang to build shared instead of static. by kjellander · 9 years ago
  25. 5a20ed3 Fix undefined reference to log2 on android by Kári Tristan Helgason · 9 years ago
  26. 89fb920 Revert of Add arraysize to rtc_base_approved. Remove dependency of audio_device on rtc_base. (patchset #1 id:1 of https://codereview.webrtc.org/2346763002/ ) by maxmorin · 9 years ago
  27. 100c9d0 Add arraysize to rtc_base_approved. Remove dependency of audio_device on rtc_base. by maxmorin · 9 years ago
  28. 705ecc5 GN: Change group deps to public_deps. by kjellander · 9 years ago
  29. c26f77f Remove a couple of unnecessary dependencies on gflags by henrik.lundin · 9 years ago
  30. f807a52 iSAC: Remove unnecessary WEBRTC_LINUX define by kjellander · 9 years ago
  31. 0d14c6a Add method cricket::VideoCapturer::NeedsDenoising, use in VideoCapturerTrackSource. by nisse · 9 years ago
  32. 8295c93 [WebRTC] Add TwoCapturers test and TwoMagnifierCapturers test by zijiehe · 9 years ago
  33. 3626d7e Move CopyOnWriteBuffer functions definitions from .h to .cc by Danil Chapovalov · 9 years ago
  34. 2e164c6 Adding ChannelController to audio network adaptor. by minyue · 9 years ago
  35. fdafab8 Fix issues with rtc_stats_unittests tests so that they can run on bots. by hbos · 9 years ago
  36. 6fa69c9 Relaxed unnecessarily stringent thread checking in WebRtcAudioSendStream::OnData(). by solenberg · 9 years ago
  37. cbae0b4 Use I420Buffer rather than VideoFrameBuffer when writing pixels. by nisse · 9 years ago
  38. bc18fc0 Change onCameraOpening to take camera name as a parameter instead of camera id. by sakal · 9 years ago
  39. 9e2be5f webrtc/modules/audio_processing: Use RTC_DCHECK() instead of assert() by kwiberg · 9 years ago
  40. 3a7f35b GN: Declare resources for targets. by ehmaldonado · 9 years ago
  41. 52a5703 Enable BWE logging to command line when rtc_enable_bwe_test_logging is set to true by gaetano.carlucci · 9 years ago
  42. b471d1c Android EglBase: Include EGL error code in exceptions by magjed · 9 years ago
  43. 194f40a Refactor QualityScaler and MovingAverage by kthelgason · 9 years ago
  44. a075848 New method TimestampAligner::TranslateTimestamp by nisse · 9 years ago
  45. f8a4ecc Remove dependency of audio_device on metrics_default. by maxmorin · 9 years ago
  46. 17366bc Remove handling unused rtcp packets. by danilchap · 9 years ago
  47. cdf37a9 Delete Timing class, timing.h, and update all users. by nisse · 9 years ago
  48. d29e3ea Added build flag around the Intelligibility enhancer performance test code by peah · 9 years ago
  49. caa9cb2 Adding basic implementation of AudioNetworkAdaptor. by minyue · 9 years ago
  50. dd12892 Reland of Make rtcp parsing implementation private in RtcpReceiver (patchset #1 id:1 of https://codereview.webrtc.org/2332673003/ ) by danilchap · 9 years ago
  51. d59d3bb Replace a DCHECK with static_assert by kwiberg · 9 years ago
  52. ba56b6c Moved webrtc/test/channel_transport/ into webrtc/voice_engine/test/ by solenberg · 9 years ago
  53. bb723e5 Fixed video_loopback target. by charujain · 9 years ago
  54. 2b2779f Make CopyOnWriteBuffer keep capacity for SetData and Clear functions too. by Danil Chapovalov · 9 years ago
  55. 9708884 Update rtcp receiver fuzzer to use generic function by Danil Chapovalov · 9 years ago
  56. 6631e8a Minor fixes in FEC and RtpSender{,Video} by brandtr · 9 years ago
  57. 07d9e54 Revert of Moved webrtc/test/channel_transport/ into webrtc/voice_engine/test/ (patchset #7 id:120001 of https://codereview.webrtc.org/2319583005/ ) by solenberg · 9 years ago
  58. 22487b2 webrtc/base: Use RTC_DCHECK() instead of assert() by kwiberg · 9 years ago
  59. ade2a03 Moved webrtc/test/channel_transport/ into webrtc/voice_engine/test/ by solenberg · 9 years ago
  60. 88ac853 The current scheme for setting parameters and specifying the by peah · 9 years ago
  61. b2540bb Probing: Add support for exponential startup probing by Irfan Sheriff · 9 years ago
  62. a421ddd The buffering of the farend signal is refactored in this CL. by peah · 9 years ago
  63. b3f7876 Add printStackTrace method to CameraCapturer. by sakal · 9 years ago
  64. 78ce619 Extract simulcast rate allocation outside of video encoder. by Erik Språng · 9 years ago
  65. 7b11c65 MB: Move iOS GYP bots to use limited support config by kjellander · 9 years ago
  66. 8e56521 The output signal of the AEC needs to be buffered as the by peah · 9 years ago
  67. a64a2fb Fix oversized rtp extension parsing. by Danil Chapovalov · 9 years ago
  68. 180e452 Revert of Make rtcp parsing implementation private in RtcpReceiver (patchset #1 id:1 of https://codereview.webrtc.org/2320603002/ ) by danilchap · 9 years ago
  69. faf708e Make rtcp parsing implementation private in RtcpReceiver: by Danil Chapovalov · 9 years ago
  70. 1a0533d Add statistics for the time it takes to start and stop the camera on Camera2. by sakal · 9 years ago
  71. 6ffb67d Add periodic logging of number of captured and dropped frames in VideoCaptureInput. Logged every minute. by asapersson · 9 years ago
  72. 11d5766 GN: Revert to default compiler optimizations for Win Release. by kjellander · 9 years ago
  73. 10f606d Revert of Introduced new scheme for controlling the functionality inside the audio processing module (patchset #12 id:260001 of https://codereview.webrtc.org/2292863002/ ) by kjellander · 9 years ago
  74. 5df5434 Fix a type mistake by honghaiz · 9 years ago
  75. 2ace3f9 The audio processing module (APM) relies on two for by peah · 9 years ago
  76. 1d02d3e Remove RTC_LOGGED_* macro. by asapersson · 9 years ago
  77. d5fff50 Removing assert error when we fail to create a connection for a ping from an unknown address. by Honghai Zhang · 9 years ago
  78. ed0b0db Revert "Optimize Android NV12 capture" by jackychen · 9 years ago
  79. c8bbe3f The current scheme for setting parameters and specifying the behavior by peah · 9 years ago
  80. e753641 Adding ability to simulate EWOULDBLOCK/SignalReadyToSend. by Taylor Brandstetter · 9 years ago
  81. fc433e6 Don't use VoE legacy APIs in force_mic_volume_max tool. by solenberg · 9 years ago
  82. fac0ff0 Change SimulcastEncoderAdapter to allow a 0 for SetRates. by noahric · 9 years ago
  83. 36d38cb Optimize Android NV12 capture by magjed · 9 years ago
  84. 291cd8f CopyOnWriteBuffer::SetSize to smaller size memcpy less. by Danil Chapovalov · 9 years ago
  85. 96f2c4d Remove unused audio_e2e_harness.cc and infrastructure. by solenberg · 9 years ago
  86. 467bc84 Revert webrtc/build/mb_config.pyl accidental change by Henrik Kjellander · 9 years ago
  87. a41c13e OWNERS: Make everyone able to change *.gn,*.gni files. by Henrik Kjellander · 9 years ago
  88. 2b1b7a8 iSAC fix: Ignore overflow in signed left shift by kwiberg · 9 years ago
  89. 53cec04 GN: Move audio_coding to public_deps in voice engine by ehmaldonado · 9 years ago
  90. f06f35a Adds logging of native audio levels and UMA stats to track issues. by henrika · 9 years ago
  91. 99f8e08 Add a chart for packet loss on incoming streams. by Stefan Holmer · 9 years ago
  92. 073378e Avoids crash at device switch on iOS by ensuring that audio parameters can be updated on the fly driven by e.g. switching audio device. by henrika · 9 years ago
  93. 2b11fd2 rtc::Optional: Tell sanitizers that unset values aren't OK to access by kwiberg · 9 years ago
  94. 463d301 Added ClearTo(seq_num) to RtpFrameReferenceFinder. by philipel · 9 years ago
  95. d547224 Reland of move all reference to carbon api (patchset #1 id:1 of https://codereview.webrtc.org/2317343003/ ) by kthelgason · 9 years ago
  96. 27c7b8f VadCore: Allow signed multiplication overflow that we don't know how to fix by kwiberg · 9 years ago
  97. 3fa3517 Filter objc headers in cpplint presubmit check by Kári Tristan Helgason · 9 years ago
  98. 9c8c586 MB: Disable more parts of the GYP build. by kjellander · 9 years ago
  99. 499dcb1 Remove references to .isolate files that are no longer needed. by kjellander · 9 years ago
  100. bd3dda6 Renamed RTCStatsReport to RTCLegacyStatsReport in objc files. by hbos · 9 years ago