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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 708ff4d93770d863cdea26fd496ae71414eb8a53
  1. 708ff4d Resampler modifications in preparation for arbitrary audioproc rates. by andrew@webrtc.org · 10 years ago
  2. 74658f6 Fix multi-monitor support in the screen capturer for Mac. by sergeyu@chromium.org · 10 years ago
  3. 998d063 Revert r5937 "Fix multi-monitor support in the screen capturer for Mac." by sergeyu@chromium.org · 10 years ago
  4. 6958c0f Add Chromium's ScopedVector. by andrew@webrtc.org · 10 years ago
  5. ceef02b Fix multi-monitor support in the screen capturer for Mac. by sergeyu@chromium.org · 10 years ago
  6. d084db9 Fix iSAC/48000 issue with ACM2. by turaj@webrtc.org · 10 years ago
  7. dcc0dbe WebRtcAecm_Process: Reduce code duplication by kwiberg@webrtc.org · 10 years ago
  8. 732b067 StereoToMono: Remove useless call to WebRtcSpl_SatW32ToW16 by kwiberg@webrtc.org · 10 years ago
  9. 692224a Removes parts of the VoEBase sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago
  10. 66ccaff Removes VoECodec sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago
  11. 1b5b5c5 Revert "Make VoiceEngine choose ACM2 by default" by henrik.lundin@webrtc.org · 10 years ago
  12. c68f88e Removing AudioCoding duplicate tests by henrik.lundin@webrtc.org · 10 years ago
  13. bee17f6 Make VoiceEngine choose ACM2 by default by henrik.lundin@webrtc.org · 10 years ago
  14. 6cae2c8 Fix crashes due to dangling external decoder pointer. by fischman@webrtc.org · 10 years ago
  15. d8d9918 Set include_internal_video_capture=1 for video_capture_tests by kjellander@webrtc.org · 10 years ago
  16. 1a07e42 Re-enable AGC tests: by aluebs@webrtc.org · 10 years ago
  17. 1ed7008 Remove use of tmpnam. by kjellander@webrtc.org · 10 years ago
  18. feb904c Replace flooding logs in rtp_sender.cc with a comment. by andrew@webrtc.org · 10 years ago
  19. 8fb9156 iOS: baby steps to being able to include_tests=1 by fischman@webrtc.org · 10 years ago
  20. bbea098 Moved voe_neteq_stats_unittest to audio_coding_module_unittest by henrik.lundin@webrtc.org · 10 years ago
  21. 9d10769 Propagate capture ntp timestamp from rtp to renderer. by wu@webrtc.org · 10 years ago
  22. d581e2c Check if a header extension is registered before updating it and fail silently if it's not. by stefan@webrtc.org · 10 years ago
  23. 336f24b Make WebRTC Android examples build without sourcing envsetup.sh by kjellander@webrtc.org · 10 years ago
  24. 98f8320 Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition. by fischman@webrtc.org · 10 years ago
  25. 4087215 Adding a config struct to NetEq by henrik.lundin@webrtc.org · 10 years ago
  26. e7d9de3 New Packet and PacketSource classes for NetEq tests by henrik.lundin@webrtc.org · 10 years ago
  27. a48d53b Fix gyp for video_capture/ensure_initialized.cc. by primiano@chromium.org · 10 years ago
  28. 4ff0eda Removes VoECallReport sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago
  29. 91d88e1 Added a new OnMoreData() interface which will not feed the playout data to APM. by xians@webrtc.org · 10 years ago
  30. 26103b9 Fix the captured screen rect conversion. by jiayl@webrtc.org · 10 years ago
  31. c1caa69 NetEq changes. by turaj@webrtc.org · 10 years ago
  32. 8edccce Cleaned up logging in video_coding. by stefan@webrtc.org · 10 years ago
  33. f16b605 Convert WEBRTC_TRACE to LOG in utility. by asapersson@webrtc.org · 10 years ago
  34. 55bc281 Disable UsesTraceCallback by pbos@webrtc.org · 10 years ago
  35. 5b3c956 Fix loopback test for case where no constraint is given. by andresp@webrtc.org · 10 years ago
  36. 9402619 Remove usage of webrtc trace in video processing modules. by asapersson@webrtc.org · 10 years ago
  37. ccef356 Add ability to control peer connection constraints for the loopback test. by andresp@webrtc.org · 10 years ago
  38. 01f4fb6 Remove self-assignment hacks that were added to avoid unused variable warnings. by fischman@webrtc.org · 10 years ago
  39. b8f935f Move a chatty creation log in neteq to LS_VERBOSE. by andrew@webrtc.org · 10 years ago
  40. ca9bff6 Make Android-APK compile in release again. by solenberg@webrtc.org · 10 years ago
  41. 2e4c621 (landing) Exclude VoiceEngine::SetAndroidObjects in WebRTC chrome builds by henrika@webrtc.org · 10 years ago
  42. f03e467 Unbreak android APK buildbots by emptying the video_capture_tests_apk target. by fischman@webrtc.org · 10 years ago
  43. b515322 VideoCaptureAndroid: support multiple frame-rates per resolution. by fischman@webrtc.org · 10 years ago
  44. d6e5cf9 Fix DesktopSize::is_empty() for the case when only width or only height is 0. by sergeyu@chromium.org · 10 years ago
  45. 01f4592 Move output_mixer_unittest.cc to utility_unittest.cc. by andrew@webrtc.org · 10 years ago
  46. bdeb1d8 VideoCaptureAndroid: stop referencing ViERenderer by fischman@webrtc.org · 10 years ago
  47. 24224fc video_capture(iOS): move stopCapture to background thread by fischman@webrtc.org · 10 years ago
  48. ea15f8d Implement FEC support in VideoReceiveStream. by pbos@webrtc.org · 10 years ago
  49. 9968131 Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG. by andresp@webrtc.org · 10 years ago
  50. cef07f9 New NetEq test to verify correct timestamp propagation by henrik.lundin@webrtc.org · 10 years ago
  51. 5260350 Removed the disabling of include_tests from r2729. by henrike@webrtc.org · 10 years ago
  52. a32583c Updated WebRTC version to 3.52 TBR=wu@webrtc.org by elham@webrtc.org · 10 years ago
  53. 8d93b11 Clean up traces and logs in RemoteBitrateEstimator. by stefan@webrtc.org · 10 years ago
  54. 022615b Log Fixit for parts of video_engine folder. by mflodman@webrtc.org · 10 years ago
  55. a6948e2 Fix logging calls in bitrate_controller module. by andresp@webrtc.org · 10 years ago
  56. 168a51c Remove WEBRTC_TRACE use in common_video/ by pbos@webrtc.org · 10 years ago
  57. a538def Fix a crash in WindowCapturereMac when capture() fails. by jiayl@webrtc.org · 10 years ago
  58. 4d17e20 Fix the library path for android 64-bit build by michaelbai@google.com · 10 years ago
  59. f7c73b5 Consolidate audio conversion from Channel and TransmitMixer. by andrew@webrtc.org · 10 years ago
  60. b5a182a Delay Estimator: Minor refactoring and added a setter function. by bjornv@webrtc.org · 10 years ago
  61. a38c76b Rename RTPanalyze to rtp_analyze and remove old version by henrik.lundin@webrtc.org · 10 years ago
  62. cceb392 Remove AudioDevice::{Microphone,Speaker}IsAvailable. by andrew@webrtc.org · 10 years ago
  63. ffcd844 This is to get rid of a bug relating to the return of NULL in calling GetDecoder when there are DTMF packets. by minyue@webrtc.org · 10 years ago
  64. df9fa2b Add format specification to output file names by henrik.lundin@webrtc.org · 10 years ago
  65. 120c725 Extends max sample rate from 96kHz to 192kHz on the input side. by henrika@webrtc.org · 10 years ago
  66. 53da760 sink_filter_ds.cc: add lock to Receive procedure to Pause(). by braveyao@webrtc.org · 10 years ago
  67. 56aeb0e Make ACM2 the default in voe_cmd_test. by andrew@webrtc.org · 10 years ago
  68. e823012 Added simulations of capacity variations and wifi recordings. by stefan@webrtc.org · 10 years ago
  69. 38d4ad7 Roll chromium_revision 255773:260462 by kjellander@webrtc.org · 10 years ago
  70. 7830689 Fix ARM64 detection. by andrew@webrtc.org · 10 years ago
  71. 172f42a VoiceEngine(iOS & Android): removed NOT_SUPPORTED by fischman@webrtc.org · 10 years ago
  72. 3f2f440 Add tests for the RBE RemoveStream() API. by solenberg@webrtc.org · 10 years ago
  73. c9dca91 VoE Channel: Don't register codecs when stopping receiver by henrik.lundin@webrtc.org · 10 years ago
  74. 4910a7f Restore support for code coverage in WebRTC by kjellander@webrtc.org · 10 years ago
  75. d30a9ea Add arm64 to typedefs.h by andrew@webrtc.org · 10 years ago
  76. 1fb05fc Allow loopback tests to do TURN when served from webrtc.googlecode.com. by andresp@webrtc.org · 10 years ago
  77. e438054 Add svn mime-type properties to loopback_test files so they can be served from: by andresp@webrtc.org · 10 years ago
  78. 5f8dfa0 Don't disable experimental AGC in audioproc. by andrew@webrtc.org · 10 years ago
  79. d7aa228 Re-submit: rev5775 by andresp@webrtc.org · 10 years ago
  80. ba75592 Makes ScreenCapturerMac exclude the window specified in DesktopCapturer::SetExcludedWindow. by jiayl@webrtc.org · 10 years ago
  81. 2f0c5f7 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
  82. 9511bbd Protect write of send_target_bitrate. by andresp@webrtc.org · 10 years ago
  83. 2bf87a2 Make RTPHeaderParser skip over unknown RTP header extensions rather than bail out. by solenberg@webrtc.org · 10 years ago
  84. ca28c29 Revert 5775 "Modify bitrate controller to update bitrate based o..." by andrew@webrtc.org · 10 years ago
  85. 2a0cbfc Removing VideoCodecDerived and moving methods inside VideoCodec. by mallinath@webrtc.org · 10 years ago
  86. 60ae794 Updated WebRTC version to 3.51 by elham@webrtc.org · 10 years ago
  87. e965143 iOS video_capture: move @private vars to impl. by fischman@webrtc.org · 10 years ago
  88. a4f259b Fix race condition in RTPSEnder. by sprang@webrtc.org · 10 years ago
  89. 9deb87b Change sprintf format string from %zu to %i by henrik.lundin@webrtc.org · 10 years ago
  90. a0320c2 Modify bitrate controller to update bitrate based on process call and not by andresp@webrtc.org · 10 years ago
  91. 5d8c954 Adding API for setting bandwidth estimation configurations. by stefan@webrtc.org · 10 years ago
  92. 7fd6ac1 iOS video_capture: start camera in the background. by fischman@webrtc.org · 10 years ago
  93. 25bbc98 iOS VideoEngine: move video_{capture,render} to ARC. by fischman@webrtc.org · 10 years ago
  94. 154951d Add configuration for ability to use the encode usage measure for triggering overuse/underuse. by asapersson@webrtc.org · 10 years ago
  95. 2d3624c Implement ViE forwarding to RBE of packets for BWE coming in through the ViENetwork::ReceivedBWEPacket API. by solenberg@webrtc.org · 10 years ago
  96. 5374dab Have changes to REMB trigger RTCP to be sent immediately. by stefan@webrtc.org · 10 years ago
  97. 1df7a5a DelayEstimator: Updates delay_quality and adds soft reset. by bjornv@webrtc.org · 10 years ago
  98. f70d0b9 Run Opus with lower complexity setting on Android, iOS and/or ARM by tina.legrand@webrtc.org · 10 years ago
  99. 3d6910c Add targetBitrate to VideoCodec struct. by pbos@webrtc.org · 10 years ago
  100. deac6f5 Disabled some of the remote bitrate estimator baseline tests. by stefan@webrtc.org · 10 years ago