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gerrit-public.fairphone.software / platform / external / webrtc / 3c3aef44deb4cca21ece8532e8dfae1b92c7989a
  1. 3c3aef4 Revert of Reland "Move smoothing filter to common audio". (patchset #5 id:100001 of https://codereview.webrtc.org/2520003005/ ) by minyue · 8 years ago
  2. 223641f Reland "Move smoothing filter to common audio". by minyue · 8 years ago
  3. b365b80 Revert of Modify the paths of the resource files to point to chromium/src/tools/... (patchset #1 id:1 of https://codereview.webrtc.org/2528893002/ ) by ehmaldonado · 8 years ago
  4. d8ae20b Modify the paths of the resource files to point to chromium/src/tools/... by ehmaldonado · 8 years ago
  5. 3cfb3ef Added a perf test for the residual echo detector. by ivoc · 8 years ago
  6. 37a2111 Increase the threshold for RunPlayoutAndRecordingInFullDuplex. Again. by ehmaldonado · 8 years ago
  7. 3edc7f0 AGC: Add a histogram for new level by henrik.lundin · 8 years ago
  8. c42d376 DataChannelInterface: Remove default implementation of methods. by hbos · 8 years ago
  9. 464d50f Set rtc_use_memcheck=true for the FYI bot. by ehmaldonado · 8 years ago
  10. ed8c8ed Add rtc_use_memcheck flag, update MB and GN to handle it, and add gni files listing the runtime deps by ehmaldonado · 8 years ago
  11. d44d0ba For VPN network, use the underlying network type as its type. by honghaiz · 8 years ago
  12. 4dfb8ce Make the default value of rtcp-mux policy to required. by zhihuang · 8 years ago
  13. e02407a Add myself to WATCHLIST for api/. by solenberg · 8 years ago
  14. 42eee12 RTCPeerConnectionStats: Removed fixed TODO comments. by hbos · 8 years ago
  15. 08be780 Reland of Allow custom metrics implementations on Android. (patchset #1 id:1 of https://codereview.webrtc.org/2516403002/ ) by sakal · 8 years ago
  16. 817208b Re-enables AudioDeviceTest.StartStopPlayout on Android by henrika · 8 years ago
  17. 8b64628 Add fps reduction API to SurfaceViewRenderer. by sakal · 8 years ago
  18. 4fe3b8d Add framelistener functionality to SurfaceViewRenderer. by sakal · 8 years ago
  19. 1c82884 Remove binding framebuffer from GlTextureFrameBuffer.setSize. by sakal · 8 years ago
  20. 8e321c8 CQ: Disable android_more_configs trybot by Henrik Kjellander · 8 years ago
  21. 0c5a154 Try to deflake VideoSendStream tests with FlexFEC. by brandtr · 8 years ago
  22. 0adb828 RTCCodecStats[1] added. by hbos · 8 years ago
  23. 71caaca Split avfoundationcapturer classes in separate files. by denicija · 8 years ago
  24. 90ea736 Add DesktopFrame rotation functions by zijiehe · 8 years ago
  25. e2b1501 Start probes only after network is connected. by Sergey Ulanov · 8 years ago
  26. 1c062bf Fix module/desktop_capture compilation on iOS by Sergey Ulanov · 8 years ago
  27. c1dd1a5 Really disable Opus complexity tests on Android by henrik.lundin · 8 years ago
  28. d661e9c WebRTC: Replace ProjectRootPath by ResourcePath by ehmaldonado · 8 years ago
  29. 10165ab Unify VideoCodecType to/from string functionality by magjed · 8 years ago
  30. 2d60e53 H264 encoder: Include QP information in encoded images by magjed · 8 years ago
  31. e60f020 iOS AppRTCMobile: Fix SDP video codec reordering for multiple H264 profiles by magjed · 8 years ago
  32. 8271d04 This CL introduces the new functionality for setting by peah · 8 years ago
  33. 30a12fb AGC: Add a histogram for clipping adjustment by henrik.lundin · 8 years ago
  34. 24d812d DEPS: Specify WebRTC hooks and add a few dependencies by kjellander · 8 years ago
  35. ab6996d Enable QP parsing from CABAC bitstreams by kthelgason · 8 years ago
  36. 04c0722 Replace AudioConferenceMixer with AudioMixer. by aleloi · 8 years ago
  37. b426040 Add Full HD and 4K camera resolutions to AppRTCMobile Android. by sakal · 8 years ago
  38. 2df1ab4 MB: Add Win32 SyzyASan (swarming) config. by ehmaldonado · 8 years ago
  39. 17338d4 Created an AudioMixer mock in webrtc/api/test. by aleloi · 8 years ago
  40. 0eb1960 ComfortNoise: Calculate used scale factor in Q13 by ossu · 8 years ago
  41. 58f90a7 Disable Opus complexity tests on Android by henrik.lundin · 8 years ago
  42. 03d5fb1 Let MediaSession generate a FlexFEC SSRC when FlexFEC is active. by brandtr · 8 years ago
  43. 0dbb6f5 Fix the standard deviation calculation in the level controller perf tests. by ivoc · 8 years ago
  44. 820f578 RTCInboundRTPStreamStats's [fir/pli/nack]_count are collected for video. by hbos · 8 years ago
  45. 468da7c Wire up FlexFEC in VideoEngine2. by brandtr · 8 years ago
  46. d848a56 DEPS: Cleanup extra_gyp_flag and extra_gitignore.py by kjellander · 8 years ago
  47. 875862c Let Opus increase complexity for low bitrates by henrik.lundin · 8 years ago
  48. b1e6d5e Set surface view surface size to minimum of the layout size and frame size. by sakal · 8 years ago
  49. f6acc2a Move VideoDecoderSoftwareFallbackWrapper from webrtc/video_decoder.h to webrtc/media/engine/ by magjed · 8 years ago
  50. 0ce6aaf Move androidvideotracksource from api under api/android/jni. by sakal · 8 years ago
  51. f723312 Add an empty libjingle_peerconnection_metrics_default_jni target. by sakal · 8 years ago
  52. 9688e38 Add support for FEC-FR semantics in StreamParams. by brandtr · 8 years ago
  53. 96385e0 iOS: Add FlexFEC-03 field trial. by brandtr · 8 years ago
  54. fb94cd6 build_ios_libs.sh: Add command line bitcode option. by tkchin · 8 years ago
  55. 7a07f13 Fix TimeCallback used by BoringSSL. by deadbeef · 8 years ago
  56. 1b0e3aa Remove deprecated CroppingWindowCapturer::Create by zijiehe · 8 years ago
  57. 2874796 RTCStats operator== bugfix by hbos · 8 years ago
  58. f570a28 Revert of Allow custom metrics implementations on Android. (patchset #11 id:260001 of https://codereview.webrtc.org/2403463002/ ) by philipel · 8 years ago
  59. ab102f1 Update gtest-parallel and introduce gtest-parallel-wrapper. by ehmaldonado · 8 years ago
  60. de609b2 Allow custom metrics implementations on Android. by sakal · 8 years ago
  61. e718606 Make magjed@ owner of webrtc/api/android/ by magjed · 8 years ago
  62. 64d6ff7 In VoiceEngine, the settings for APM are applied in such a way that by peah · 8 years ago
  63. 40217c3 Initial rate allocation should not use fps = 0 by sprang · 8 years ago
  64. 57c1ad3 Don't declare function arguments of array type by kwiberg · 8 years ago
  65. cc7bf88 Revert of Roll chromium_revision 5e821a778b..80ff2be807 (432715:433495) (patchset #1 id:1 of https://codereview.webrtc.org/2517933002/ ) by kjellander · 8 years ago
  66. 6280960 Correctly pass drawn frame size when layout aspect ratio is used in EglRenderer. by sakal · 8 years ago
  67. 96c1587 RtpPacket::payload() return rtc::ArrayView instead of raw pointer by danilchap · 8 years ago
  68. fe09560 Roll chromium_revision 5e821a778b..80ff2be807 (432715:433495) by buildbot · 8 years ago
  69. f880285 iOS: Cleanup buildbot JSON files + bump iOS version to 10.0 by kjellander · 8 years ago
  70. 3898944 Remove unused files linux.cc/.h and linuxfdwalk.c/.h. by solenberg · 8 years ago
  71. 2184155 Add more logging in ScreenCapturerIntegrationTest by zijiehe · 8 years ago
  72. ed9dccf Revert of Remove unused HttpClient class. (patchset #1 id:1 of https://codereview.webrtc.org/2511883005/ ) by honghaiz · 8 years ago
  73. 4a698f6 Remove unused HttpClient class. by solenberg · 8 years ago
  74. 01af3a3 Remove unused dbus.cc/.h and related things. by solenberg · 8 years ago
  75. 90c024f Move FirewallSocketServer to test code. by nisse · 8 years ago
  76. 00f2ee0 Changed the way we find the ProjectRootPath. by ehmaldonado · 8 years ago
  77. dedaf1c Modify audio_processing_unittest to use ResourcePath instead of ProjectRootPath. by ehmaldonado · 8 years ago
  78. bbc747c Delete WindowPicker class and subclasses. by nisse · 8 years ago
  79. 76b3049 Changed the interface AudioMixer::RemoveSource to have a void return type. by aleloi · 8 years ago
  80. a28780e Introduce ArrayView::subview function to return portion of the original view by danilchap · 8 years ago
  81. 509e4fe Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) by magjed · 8 years ago
  82. d7ac0a9 Revert of Move smoothing filter to common audio. (patchset #3 id:60001 of https://codereview.webrtc.org/2484153002/ ) by magjed · 8 years ago
  83. a82395b Move smoothing filter to common audio. by michaelt · 8 years ago
  84. 610c454 Add Datachannel support to Android AppRTCMobile by hekra01 · 8 years ago
  85. 1acfbd2 Expose RtpCodecParameters to VoiceMediaInfo stats. by hbos · 8 years ago
  86. 7b9feee Fix PayloadRouter::OnEncodedImage() to handle errors properly. by sergeyu · 8 years ago
  87. 81c3a03 Added a callback function OnAddTrack to PeerConnectionObserver by zhihuang · 8 years ago
  88. 5b93db2 iOS: Add AudioSendSideBwe field trial. by tkchin · 8 years ago
  89. eacbaea Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) by magjed · 8 years ago
  90. 0d0d753 Revert of Split out target rtc_media_base from rtc_media (patchset #3 id:40001 of https://codereview.webrtc.org/2471573003/ ) by magjed · 8 years ago
  91. de49803 MB: Add new perf desktop bots and remove DCHECK from Android perf by kjellander · 8 years ago
  92. aae7e7c Split out target rtc_media_base from rtc_media by magjed · 8 years ago
  93. 765edc3 Update the alpha value in the echo detector. by ivoc · 8 years ago
  94. 42043b9 Stop using hardcoded payload types for video codecs by Magnus Jedvert · 8 years ago
  95. 10111bc Passed AudioMixer to AudioState::Config. by aleloi · 8 years ago
  96. dd31071 Added an empty AudioTransportProxy to AudioState. by aleloi · 8 years ago
  97. d4adce4 Remove Absolute Send Time from list of supported header extensions for audio streams. by solenberg · 8 years ago
  98. fbb374d Add a reliability term to the echo detector. by ivoc · 8 years ago
  99. d51c4dc Delete unused files httprequest.h and httprequest.cc. by nisse · 8 years ago
  100. ffbbcac Support multiple timestamp rates for sending DTMF. by solenberg · 8 years ago