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gerrit-public.fairphone.software / platform / external / webrtc / 70d01242f8898aeb9dddfbbeaf3f22421479fe7c
  1. 70d0124 MB: Change Android Clang to build shared instead of static. by kjellander · 8 years ago
  2. 5a20ed3 Fix undefined reference to log2 on android by Kári Tristan Helgason · 8 years ago
  3. 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 · 8 years ago
  4. 100c9d0 Add arraysize to rtc_base_approved. Remove dependency of audio_device on rtc_base. by maxmorin · 8 years ago
  5. 705ecc5 GN: Change group deps to public_deps. by kjellander · 8 years ago
  6. c26f77f Remove a couple of unnecessary dependencies on gflags by henrik.lundin · 8 years ago
  7. f807a52 iSAC: Remove unnecessary WEBRTC_LINUX define by kjellander · 8 years ago
  8. 0d14c6a Add method cricket::VideoCapturer::NeedsDenoising, use in VideoCapturerTrackSource. by nisse · 8 years ago
  9. 8295c93 [WebRTC] Add TwoCapturers test and TwoMagnifierCapturers test by zijiehe · 8 years ago
  10. 3626d7e Move CopyOnWriteBuffer functions definitions from .h to .cc by Danil Chapovalov · 8 years ago
  11. 2e164c6 Adding ChannelController to audio network adaptor. by minyue · 8 years ago
  12. fdafab8 Fix issues with rtc_stats_unittests tests so that they can run on bots. by hbos · 8 years ago
  13. 6fa69c9 Relaxed unnecessarily stringent thread checking in WebRtcAudioSendStream::OnData(). by solenberg · 8 years ago
  14. cbae0b4 Use I420Buffer rather than VideoFrameBuffer when writing pixels. by nisse · 8 years ago
  15. bc18fc0 Change onCameraOpening to take camera name as a parameter instead of camera id. by sakal · 8 years ago
  16. 9e2be5f webrtc/modules/audio_processing: Use RTC_DCHECK() instead of assert() by kwiberg · 8 years ago
  17. 3a7f35b GN: Declare resources for targets. by ehmaldonado · 8 years ago
  18. 52a5703 Enable BWE logging to command line when rtc_enable_bwe_test_logging is set to true by gaetano.carlucci · 8 years ago
  19. b471d1c Android EglBase: Include EGL error code in exceptions by magjed · 8 years ago
  20. 194f40a Refactor QualityScaler and MovingAverage by kthelgason · 8 years ago
  21. a075848 New method TimestampAligner::TranslateTimestamp by nisse · 8 years ago
  22. f8a4ecc Remove dependency of audio_device on metrics_default. by maxmorin · 8 years ago
  23. 17366bc Remove handling unused rtcp packets. by danilchap · 8 years ago
  24. cdf37a9 Delete Timing class, timing.h, and update all users. by nisse · 8 years ago
  25. d29e3ea Added build flag around the Intelligibility enhancer performance test code by peah · 8 years ago
  26. caa9cb2 Adding basic implementation of AudioNetworkAdaptor. by minyue · 8 years ago
  27. dd12892 Reland of Make rtcp parsing implementation private in RtcpReceiver (patchset #1 id:1 of https://codereview.webrtc.org/2332673003/ ) by danilchap · 8 years ago
  28. d59d3bb Replace a DCHECK with static_assert by kwiberg · 8 years ago
  29. ba56b6c Moved webrtc/test/channel_transport/ into webrtc/voice_engine/test/ by solenberg · 8 years ago
  30. bb723e5 Fixed video_loopback target. by charujain · 8 years ago
  31. 2b2779f Make CopyOnWriteBuffer keep capacity for SetData and Clear functions too. by Danil Chapovalov · 8 years ago
  32. 9708884 Update rtcp receiver fuzzer to use generic function by Danil Chapovalov · 8 years ago
  33. 6631e8a Minor fixes in FEC and RtpSender{,Video} by brandtr · 8 years ago
  34. 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 · 8 years ago
  35. 22487b2 webrtc/base: Use RTC_DCHECK() instead of assert() by kwiberg · 8 years ago
  36. ade2a03 Moved webrtc/test/channel_transport/ into webrtc/voice_engine/test/ by solenberg · 8 years ago
  37. 88ac853 The current scheme for setting parameters and specifying the by peah · 8 years ago
  38. b2540bb Probing: Add support for exponential startup probing by Irfan Sheriff · 8 years ago
  39. a421ddd The buffering of the farend signal is refactored in this CL. by peah · 8 years ago
  40. b3f7876 Add printStackTrace method to CameraCapturer. by sakal · 8 years ago
  41. 78ce619 Extract simulcast rate allocation outside of video encoder. by Erik Språng · 8 years ago
  42. 7b11c65 MB: Move iOS GYP bots to use limited support config by kjellander · 8 years ago
  43. 8e56521 The output signal of the AEC needs to be buffered as the by peah · 8 years ago
  44. a64a2fb Fix oversized rtp extension parsing. by Danil Chapovalov · 8 years ago
  45. 180e452 Revert of Make rtcp parsing implementation private in RtcpReceiver (patchset #1 id:1 of https://codereview.webrtc.org/2320603002/ ) by danilchap · 8 years ago
  46. faf708e Make rtcp parsing implementation private in RtcpReceiver: by Danil Chapovalov · 8 years ago
  47. 1a0533d Add statistics for the time it takes to start and stop the camera on Camera2. by sakal · 8 years ago
  48. 6ffb67d Add periodic logging of number of captured and dropped frames in VideoCaptureInput. Logged every minute. by asapersson · 8 years ago
  49. 11d5766 GN: Revert to default compiler optimizations for Win Release. by kjellander · 8 years ago
  50. 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 · 8 years ago
  51. 5df5434 Fix a type mistake by honghaiz · 8 years ago
  52. 2ace3f9 The audio processing module (APM) relies on two for by peah · 8 years ago
  53. 1d02d3e Remove RTC_LOGGED_* macro. by asapersson · 8 years ago
  54. d5fff50 Removing assert error when we fail to create a connection for a ping from an unknown address. by Honghai Zhang · 8 years ago
  55. ed0b0db Revert "Optimize Android NV12 capture" by jackychen · 8 years ago
  56. c8bbe3f The current scheme for setting parameters and specifying the behavior by peah · 8 years ago
  57. e753641 Adding ability to simulate EWOULDBLOCK/SignalReadyToSend. by Taylor Brandstetter · 8 years ago
  58. fc433e6 Don't use VoE legacy APIs in force_mic_volume_max tool. by solenberg · 8 years ago
  59. fac0ff0 Change SimulcastEncoderAdapter to allow a 0 for SetRates. by noahric · 8 years ago
  60. 36d38cb Optimize Android NV12 capture by magjed · 8 years ago
  61. 291cd8f CopyOnWriteBuffer::SetSize to smaller size memcpy less. by Danil Chapovalov · 8 years ago
  62. 96f2c4d Remove unused audio_e2e_harness.cc and infrastructure. by solenberg · 8 years ago
  63. 467bc84 Revert webrtc/build/mb_config.pyl accidental change by Henrik Kjellander · 8 years ago
  64. a41c13e OWNERS: Make everyone able to change *.gn,*.gni files. by Henrik Kjellander · 8 years ago
  65. 2b1b7a8 iSAC fix: Ignore overflow in signed left shift by kwiberg · 8 years ago
  66. 53cec04 GN: Move audio_coding to public_deps in voice engine by ehmaldonado · 8 years ago
  67. f06f35a Adds logging of native audio levels and UMA stats to track issues. by henrika · 8 years ago
  68. 99f8e08 Add a chart for packet loss on incoming streams. by Stefan Holmer · 8 years ago
  69. 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 · 8 years ago
  70. 2b11fd2 rtc::Optional: Tell sanitizers that unset values aren't OK to access by kwiberg · 8 years ago
  71. 463d301 Added ClearTo(seq_num) to RtpFrameReferenceFinder. by philipel · 8 years ago
  72. d547224 Reland of move all reference to carbon api (patchset #1 id:1 of https://codereview.webrtc.org/2317343003/ ) by kthelgason · 8 years ago
  73. 27c7b8f VadCore: Allow signed multiplication overflow that we don't know how to fix by kwiberg · 8 years ago
  74. 3fa3517 Filter objc headers in cpplint presubmit check by Kári Tristan Helgason · 8 years ago
  75. 9c8c586 MB: Disable more parts of the GYP build. by kjellander · 8 years ago
  76. 499dcb1 Remove references to .isolate files that are no longer needed. by kjellander · 8 years ago
  77. bd3dda6 Renamed RTCStatsReport to RTCLegacyStatsReport in objc files. by hbos · 8 years ago
  78. b0afd97 Revert of Only expose gflags target in non-Chromium and non-fuzzer builds. (patchset #1 id:40001 of https://codereview.webrtc.org/2321963002/ ) by kjellander · 8 years ago
  79. 961168a Add sakal as an OWNER to some Android files. by sakal · 8 years ago
  80. ce2e136 Update AvgCounter to have the ability to include last period metric for subsequent intervals without samples (e.g. for non-periodic updated stats). by asapersson · 8 years ago
  81. 2a5f371 Make UMA stats creation available in the Java interface. by sakal · 8 years ago
  82. 9365338 Only expose gflags target in non-Chromium and non-fuzzer builds. by kjellander · 8 years ago
  83. aa85cac Add magjed@ as owner of webrtc/common_video by magjed · 8 years ago
  84. 432950c Revert of Add a DEPS gclient hook to prune corrupt mockito remote. (patchset #1 id:1 of https://codereview.webrtc.org/2326523002/ ) by kjellander · 8 years ago
  85. 5865f48 Revert of Separating video settings in VideoQualityTest. (patchset #2 id:20001 of https://codereview.webrtc.org/2312613003/ ) by kjellander · 8 years ago
  86. 906f403 This CL refactors the buffering of the incoming near-end signal inside by peah · 8 years ago
  87. 0e62f2b Change owner of webrtc/test/channel_transport to solenberg@. by henrikg · 8 years ago
  88. f07fb00 Separating video settings in VideoQualityTest. by minyue · 8 years ago
  89. 3115b06 Add a DEPS gclient hook to prune corrupt mockito remote. by ehmaldonado · 8 years ago
  90. 13eef785 Revert of Don't use VoE legacy APIs in force_mic_volume_max tool. (patchset #5 id:80001 of https://codereview.webrtc.org/2268183007/ ) by solenberg · 8 years ago
  91. 0f8ea0d Avoids crash in WebRtcAudioTrack.initPlayout (part II) by henrika · 8 years ago
  92. ae9f2bd Don't use VoE legacy APIs in force_mic_volume_max tool. by solenberg · 8 years ago
  93. 49fbbe0 Force a Chromium sync on all bots. by Henrik Kjellander · 8 years ago
  94. 4e0581f Revert of move all reference to carbon api (patchset #2 id:300001 of https://codereview.webrtc.org/2321493002/ ) by magjed · 8 years ago
  95. 7e4b604 Android ThreadUtils: Propagate exceptions in invoke functions by magjed · 8 years ago
  96. 22c8d5a Revert of Setting up an RTP input fuzzer for NetEq (patchset #2 id:20001 of https://codereview.webrtc.org/2315633002/ ) by henrik.lundin · 8 years ago
  97. 17e3fa1 Removed sync packet support from NetEq. by ossu · 8 years ago
  98. 2c993ce Avoids crash in WebRtcAudioTrack.initPlayout by henrika · 8 years ago
  99. 5b356f4 FilePlayer: Remove backwards compatibility stuff that we no longer need by kwiberg · 8 years ago
  100. acf9f47 GN Templates: Introduce rtc_shared_library by ehmaldonado · 8 years ago