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gerrit-public.fairphone.software / platform / external / webrtc / 9a8abcbea7c89df18aa6e6bce7a462cf50562bd3
  1. 6fa69c9 Relaxed unnecessarily stringent thread checking in WebRtcAudioSendStream::OnData(). by solenberg · 8 years ago
  2. cbae0b4 Use I420Buffer rather than VideoFrameBuffer when writing pixels. by nisse · 8 years ago
  3. bc18fc0 Change onCameraOpening to take camera name as a parameter instead of camera id. by sakal · 8 years ago
  4. 9e2be5f webrtc/modules/audio_processing: Use RTC_DCHECK() instead of assert() by kwiberg · 8 years ago
  5. 3a7f35b GN: Declare resources for targets. by ehmaldonado · 8 years ago
  6. 52a5703 Enable BWE logging to command line when rtc_enable_bwe_test_logging is set to true by gaetano.carlucci · 8 years ago
  7. b471d1c Android EglBase: Include EGL error code in exceptions by magjed · 8 years ago
  8. 194f40a Refactor QualityScaler and MovingAverage by kthelgason · 8 years ago
  9. a075848 New method TimestampAligner::TranslateTimestamp by nisse · 8 years ago
  10. f8a4ecc Remove dependency of audio_device on metrics_default. by maxmorin · 8 years ago
  11. 17366bc Remove handling unused rtcp packets. by danilchap · 8 years ago
  12. cdf37a9 Delete Timing class, timing.h, and update all users. by nisse · 8 years ago
  13. d29e3ea Added build flag around the Intelligibility enhancer performance test code by peah · 8 years ago
  14. caa9cb2 Adding basic implementation of AudioNetworkAdaptor. by minyue · 8 years ago
  15. 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
  16. d59d3bb Replace a DCHECK with static_assert by kwiberg · 8 years ago
  17. ba56b6c Moved webrtc/test/channel_transport/ into webrtc/voice_engine/test/ by solenberg · 8 years ago
  18. bb723e5 Fixed video_loopback target. by charujain · 8 years ago
  19. 2b2779f Make CopyOnWriteBuffer keep capacity for SetData and Clear functions too. by Danil Chapovalov · 8 years ago
  20. 9708884 Update rtcp receiver fuzzer to use generic function by Danil Chapovalov · 8 years ago
  21. 6631e8a Minor fixes in FEC and RtpSender{,Video} by brandtr · 8 years ago
  22. 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
  23. 22487b2 webrtc/base: Use RTC_DCHECK() instead of assert() by kwiberg · 8 years ago
  24. ade2a03 Moved webrtc/test/channel_transport/ into webrtc/voice_engine/test/ by solenberg · 8 years ago
  25. 88ac853 The current scheme for setting parameters and specifying the by peah · 8 years ago
  26. b2540bb Probing: Add support for exponential startup probing by Irfan Sheriff · 8 years ago
  27. a421ddd The buffering of the farend signal is refactored in this CL. by peah · 8 years ago
  28. b3f7876 Add printStackTrace method to CameraCapturer. by sakal · 8 years ago
  29. 78ce619 Extract simulcast rate allocation outside of video encoder. by Erik Språng · 8 years ago
  30. 7b11c65 MB: Move iOS GYP bots to use limited support config by kjellander · 8 years ago
  31. 8e56521 The output signal of the AEC needs to be buffered as the by peah · 8 years ago
  32. a64a2fb Fix oversized rtp extension parsing. by Danil Chapovalov · 8 years ago
  33. 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
  34. faf708e Make rtcp parsing implementation private in RtcpReceiver: by Danil Chapovalov · 8 years ago
  35. 1a0533d Add statistics for the time it takes to start and stop the camera on Camera2. by sakal · 8 years ago
  36. 6ffb67d Add periodic logging of number of captured and dropped frames in VideoCaptureInput. Logged every minute. by asapersson · 8 years ago
  37. 11d5766 GN: Revert to default compiler optimizations for Win Release. by kjellander · 8 years ago
  38. 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
  39. 5df5434 Fix a type mistake by honghaiz · 8 years ago
  40. 2ace3f9 The audio processing module (APM) relies on two for by peah · 8 years ago
  41. 1d02d3e Remove RTC_LOGGED_* macro. by asapersson · 8 years ago
  42. d5fff50 Removing assert error when we fail to create a connection for a ping from an unknown address. by Honghai Zhang · 8 years ago
  43. ed0b0db Revert "Optimize Android NV12 capture" by jackychen · 8 years ago
  44. c8bbe3f The current scheme for setting parameters and specifying the behavior by peah · 8 years ago
  45. e753641 Adding ability to simulate EWOULDBLOCK/SignalReadyToSend. by Taylor Brandstetter · 8 years ago
  46. fc433e6 Don't use VoE legacy APIs in force_mic_volume_max tool. by solenberg · 8 years ago
  47. fac0ff0 Change SimulcastEncoderAdapter to allow a 0 for SetRates. by noahric · 8 years ago
  48. 36d38cb Optimize Android NV12 capture by magjed · 8 years ago
  49. 291cd8f CopyOnWriteBuffer::SetSize to smaller size memcpy less. by Danil Chapovalov · 8 years ago
  50. 96f2c4d Remove unused audio_e2e_harness.cc and infrastructure. by solenberg · 8 years ago
  51. 467bc84 Revert webrtc/build/mb_config.pyl accidental change by Henrik Kjellander · 8 years ago
  52. a41c13e OWNERS: Make everyone able to change *.gn,*.gni files. by Henrik Kjellander · 8 years ago
  53. 2b1b7a8 iSAC fix: Ignore overflow in signed left shift by kwiberg · 8 years ago
  54. 53cec04 GN: Move audio_coding to public_deps in voice engine by ehmaldonado · 8 years ago
  55. f06f35a Adds logging of native audio levels and UMA stats to track issues. by henrika · 8 years ago
  56. 99f8e08 Add a chart for packet loss on incoming streams. by Stefan Holmer · 8 years ago
  57. 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
  58. 2b11fd2 rtc::Optional: Tell sanitizers that unset values aren't OK to access by kwiberg · 8 years ago
  59. 463d301 Added ClearTo(seq_num) to RtpFrameReferenceFinder. by philipel · 8 years ago
  60. d547224 Reland of move all reference to carbon api (patchset #1 id:1 of https://codereview.webrtc.org/2317343003/ ) by kthelgason · 8 years ago
  61. 27c7b8f VadCore: Allow signed multiplication overflow that we don't know how to fix by kwiberg · 8 years ago
  62. 3fa3517 Filter objc headers in cpplint presubmit check by Kári Tristan Helgason · 8 years ago
  63. 9c8c586 MB: Disable more parts of the GYP build. by kjellander · 8 years ago
  64. 499dcb1 Remove references to .isolate files that are no longer needed. by kjellander · 8 years ago
  65. bd3dda6 Renamed RTCStatsReport to RTCLegacyStatsReport in objc files. by hbos · 8 years ago
  66. 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
  67. 961168a Add sakal as an OWNER to some Android files. by sakal · 8 years ago
  68. 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
  69. 2a5f371 Make UMA stats creation available in the Java interface. by sakal · 8 years ago
  70. 9365338 Only expose gflags target in non-Chromium and non-fuzzer builds. by kjellander · 8 years ago
  71. aa85cac Add magjed@ as owner of webrtc/common_video by magjed · 8 years ago
  72. 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
  73. 5865f48 Revert of Separating video settings in VideoQualityTest. (patchset #2 id:20001 of https://codereview.webrtc.org/2312613003/ ) by kjellander · 8 years ago
  74. 906f403 This CL refactors the buffering of the incoming near-end signal inside by peah · 8 years ago
  75. 0e62f2b Change owner of webrtc/test/channel_transport to solenberg@. by henrikg · 8 years ago
  76. f07fb00 Separating video settings in VideoQualityTest. by minyue · 8 years ago
  77. 3115b06 Add a DEPS gclient hook to prune corrupt mockito remote. by ehmaldonado · 8 years ago
  78. 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
  79. 0f8ea0d Avoids crash in WebRtcAudioTrack.initPlayout (part II) by henrika · 8 years ago
  80. ae9f2bd Don't use VoE legacy APIs in force_mic_volume_max tool. by solenberg · 8 years ago
  81. 49fbbe0 Force a Chromium sync on all bots. by Henrik Kjellander · 8 years ago
  82. 4e0581f Revert of move all reference to carbon api (patchset #2 id:300001 of https://codereview.webrtc.org/2321493002/ ) by magjed · 8 years ago
  83. 7e4b604 Android ThreadUtils: Propagate exceptions in invoke functions by magjed · 8 years ago
  84. 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
  85. 17e3fa1 Removed sync packet support from NetEq. by ossu · 8 years ago
  86. 2c993ce Avoids crash in WebRtcAudioTrack.initPlayout by henrika · 8 years ago
  87. 5b356f4 FilePlayer: Remove backwards compatibility stuff that we no longer need by kwiberg · 8 years ago
  88. acf9f47 GN Templates: Introduce rtc_shared_library by ehmaldonado · 8 years ago
  89. 76cd281 MB: Move Linux 32 bots from the WebRTC FYI to the main waterfall. by ehmaldonado · 8 years ago
  90. a90879b Reland of move all reference to carbon api (patchset #1 id:1 of https://codereview.webrtc.org/2316563002/ ) by kthelgason · 8 years ago
  91. 71eb61c Reland of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #1 id:1 of https://codereview.webrtc.org/2300323002/ ) by magjed · 8 years ago
  92. 4e869e9 A more useful gyp_flag_compare script by ehmaldonado · 8 years ago
  93. 243c0e8 Fixing NetEqReplacementInput for reordered and missing packets by henrik.lundin · 8 years ago
  94. ac398f2 Python event log analyzer tool: fix of indexing issue. by aleloi · 8 years ago
  95. a4c2106 This CL contains the following small changes: by aleloi · 8 years ago
  96. 250fd97 Use RateCounter for received bitrate stats: by asapersson · 8 years ago
  97. 14f1250 Do not report bucket delay for stats when pacer is paused (zero returned). by asapersson · 8 years ago
  98. a264ecc Copy RTCAudioSource.h and RTCMediaSource.h with other public header files when building the WebRTC framework for iOS / Mac by VladimirTechMan · 8 years ago
  99. 14b9d79 If encoding is inactive, don't start sending when stream is reconfigured. by Taylor Brandstetter · 8 years ago
  100. 7610f85 Adding AudioNetworkAdaptor interfaces. by minyue · 8 years ago