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gerrit-public.fairphone.software / platform / external / webrtc / e2b1501101802ad8f4b7e38cad673a19bc2ff849
  1. e2b1501 Start probes only after network is connected. by Sergey Ulanov · 8 years ago
  2. 1c062bf Fix module/desktop_capture compilation on iOS by Sergey Ulanov · 8 years ago
  3. c1dd1a5 Really disable Opus complexity tests on Android by henrik.lundin · 8 years ago
  4. d661e9c WebRTC: Replace ProjectRootPath by ResourcePath by ehmaldonado · 8 years ago
  5. 10165ab Unify VideoCodecType to/from string functionality by magjed · 8 years ago
  6. 2d60e53 H264 encoder: Include QP information in encoded images by magjed · 8 years ago
  7. e60f020 iOS AppRTCMobile: Fix SDP video codec reordering for multiple H264 profiles by magjed · 8 years ago
  8. 8271d04 This CL introduces the new functionality for setting by peah · 8 years ago
  9. 30a12fb AGC: Add a histogram for clipping adjustment by henrik.lundin · 8 years ago
  10. 24d812d DEPS: Specify WebRTC hooks and add a few dependencies by kjellander · 8 years ago
  11. ab6996d Enable QP parsing from CABAC bitstreams by kthelgason · 8 years ago
  12. 04c0722 Replace AudioConferenceMixer with AudioMixer. by aleloi · 8 years ago
  13. b426040 Add Full HD and 4K camera resolutions to AppRTCMobile Android. by sakal · 8 years ago
  14. 2df1ab4 MB: Add Win32 SyzyASan (swarming) config. by ehmaldonado · 8 years ago
  15. 17338d4 Created an AudioMixer mock in webrtc/api/test. by aleloi · 8 years ago
  16. 0eb1960 ComfortNoise: Calculate used scale factor in Q13 by ossu · 8 years ago
  17. 58f90a7 Disable Opus complexity tests on Android by henrik.lundin · 8 years ago
  18. 03d5fb1 Let MediaSession generate a FlexFEC SSRC when FlexFEC is active. by brandtr · 8 years ago
  19. 0dbb6f5 Fix the standard deviation calculation in the level controller perf tests. by ivoc · 8 years ago
  20. 820f578 RTCInboundRTPStreamStats's [fir/pli/nack]_count are collected for video. by hbos · 8 years ago
  21. 468da7c Wire up FlexFEC in VideoEngine2. by brandtr · 8 years ago
  22. d848a56 DEPS: Cleanup extra_gyp_flag and extra_gitignore.py by kjellander · 8 years ago
  23. 875862c Let Opus increase complexity for low bitrates by henrik.lundin · 8 years ago
  24. b1e6d5e Set surface view surface size to minimum of the layout size and frame size. by sakal · 8 years ago
  25. f6acc2a Move VideoDecoderSoftwareFallbackWrapper from webrtc/video_decoder.h to webrtc/media/engine/ by magjed · 8 years ago
  26. 0ce6aaf Move androidvideotracksource from api under api/android/jni. by sakal · 8 years ago
  27. f723312 Add an empty libjingle_peerconnection_metrics_default_jni target. by sakal · 8 years ago
  28. 9688e38 Add support for FEC-FR semantics in StreamParams. by brandtr · 8 years ago
  29. 96385e0 iOS: Add FlexFEC-03 field trial. by brandtr · 8 years ago
  30. fb94cd6 build_ios_libs.sh: Add command line bitcode option. by tkchin · 8 years ago
  31. 7a07f13 Fix TimeCallback used by BoringSSL. by deadbeef · 8 years ago
  32. 1b0e3aa Remove deprecated CroppingWindowCapturer::Create by zijiehe · 8 years ago
  33. 2874796 RTCStats operator== bugfix by hbos · 8 years ago
  34. f570a28 Revert of Allow custom metrics implementations on Android. (patchset #11 id:260001 of https://codereview.webrtc.org/2403463002/ ) by philipel · 8 years ago
  35. ab102f1 Update gtest-parallel and introduce gtest-parallel-wrapper. by ehmaldonado · 8 years ago
  36. de609b2 Allow custom metrics implementations on Android. by sakal · 8 years ago
  37. e718606 Make magjed@ owner of webrtc/api/android/ by magjed · 8 years ago
  38. 64d6ff7 In VoiceEngine, the settings for APM are applied in such a way that by peah · 8 years ago
  39. 40217c3 Initial rate allocation should not use fps = 0 by sprang · 8 years ago
  40. 57c1ad3 Don't declare function arguments of array type by kwiberg · 8 years ago
  41. 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
  42. 6280960 Correctly pass drawn frame size when layout aspect ratio is used in EglRenderer. by sakal · 8 years ago
  43. 96c1587 RtpPacket::payload() return rtc::ArrayView instead of raw pointer by danilchap · 8 years ago
  44. fe09560 Roll chromium_revision 5e821a778b..80ff2be807 (432715:433495) by buildbot · 8 years ago
  45. f880285 iOS: Cleanup buildbot JSON files + bump iOS version to 10.0 by kjellander · 8 years ago
  46. 3898944 Remove unused files linux.cc/.h and linuxfdwalk.c/.h. by solenberg · 8 years ago
  47. 2184155 Add more logging in ScreenCapturerIntegrationTest by zijiehe · 8 years ago
  48. ed9dccf Revert of Remove unused HttpClient class. (patchset #1 id:1 of https://codereview.webrtc.org/2511883005/ ) by honghaiz · 8 years ago
  49. 4a698f6 Remove unused HttpClient class. by solenberg · 8 years ago
  50. 01af3a3 Remove unused dbus.cc/.h and related things. by solenberg · 8 years ago
  51. 90c024f Move FirewallSocketServer to test code. by nisse · 8 years ago
  52. 00f2ee0 Changed the way we find the ProjectRootPath. by ehmaldonado · 8 years ago
  53. dedaf1c Modify audio_processing_unittest to use ResourcePath instead of ProjectRootPath. by ehmaldonado · 8 years ago
  54. bbc747c Delete WindowPicker class and subclasses. by nisse · 8 years ago
  55. 76b3049 Changed the interface AudioMixer::RemoveSource to have a void return type. by aleloi · 8 years ago
  56. a28780e Introduce ArrayView::subview function to return portion of the original view by danilchap · 8 years ago
  57. 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
  58. d7ac0a9 Revert of Move smoothing filter to common audio. (patchset #3 id:60001 of https://codereview.webrtc.org/2484153002/ ) by magjed · 8 years ago
  59. a82395b Move smoothing filter to common audio. by michaelt · 8 years ago
  60. 610c454 Add Datachannel support to Android AppRTCMobile by hekra01 · 8 years ago
  61. 1acfbd2 Expose RtpCodecParameters to VoiceMediaInfo stats. by hbos · 8 years ago
  62. 7b9feee Fix PayloadRouter::OnEncodedImage() to handle errors properly. by sergeyu · 8 years ago
  63. 81c3a03 Added a callback function OnAddTrack to PeerConnectionObserver by zhihuang · 8 years ago
  64. 5b93db2 iOS: Add AudioSendSideBwe field trial. by tkchin · 8 years ago
  65. 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
  66. 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
  67. de49803 MB: Add new perf desktop bots and remove DCHECK from Android perf by kjellander · 8 years ago
  68. aae7e7c Split out target rtc_media_base from rtc_media by magjed · 8 years ago
  69. 765edc3 Update the alpha value in the echo detector. by ivoc · 8 years ago
  70. 42043b9 Stop using hardcoded payload types for video codecs by Magnus Jedvert · 8 years ago
  71. 10111bc Passed AudioMixer to AudioState::Config. by aleloi · 8 years ago
  72. dd31071 Added an empty AudioTransportProxy to AudioState. by aleloi · 8 years ago
  73. d4adce4 Remove Absolute Send Time from list of supported header extensions for audio streams. by solenberg · 8 years ago
  74. fbb374d Add a reliability term to the echo detector. by ivoc · 8 years ago
  75. d51c4dc Delete unused files httprequest.h and httprequest.cc. by nisse · 8 years ago
  76. ffbbcac Support multiple timestamp rates for sending DTMF. by solenberg · 8 years ago
  77. 2779bab Support receiving DTMF for multiple RTP clock rates. by solenberg · 8 years ago
  78. fbfb536 Explicitly enable RED over RTX in rampup tests. by brandtr · 8 years ago
  79. afaef8b Add a new overuse estimator for the delay based BWE behind experiment. by terelius · 8 years ago
  80. b7e7b49 Use NtpTime in RtcpMeasurement instead of uint sec/uint frac. by asapersson · 8 years ago
  81. 4da3044 Add overhead per packet observer to the rtp_sender. by michaelt · 8 years ago
  82. 4a4b3cf Add interval estimator to remote bitrate estimator. by michaelt · 8 years ago
  83. 377b60c Only enable residual echo detector when needed in level controller perf tests. by ivoc · 8 years ago
  84. 0bff12a Renamed -red to -ed and -red_graph to -ed_graph in audioproc_f. by ivoc · 8 years ago
  85. 9af2b60 Propagate bitrate setting to RTCRtpSender. by denicija · 8 years ago
  86. a62f582 Integrate FlexFEC in video_loopback. by brandtr · 8 years ago
  87. dd369c6 Reduce full stack test time to 45 secs and add H264 and FlexFEC. by brandtr · 8 years ago
  88. 527d347 Reland of Declare VideoCodec.codec_specific_info private (patchset #1 id:1 of https://codereview.webrtc.org/2491613005/ ) by hta · 8 years ago
  89. 05f845d Replace c-style cast and constrain value in VCMFecMethod::ProtectionFactor. by brandtr · 8 years ago
  90. 39f9729 Add VideoSendStreamTest for FlexFEC. by brandtr · 8 years ago
  91. 1293aca Configure FlexFEC in VideoQualityTest. by brandtr · 8 years ago
  92. 1e3dfbf Add FlexFEC end-to-end test. by brandtr · 8 years ago
  93. f132167 Roll chromium_revision 3048cc9bc0..5e821a778b (432221:432715) by buildbot · 8 years ago
  94. 46c7389 Adding GetConfiguration to PeerConnection. by deadbeef · 8 years ago
  95. aee0b5d Fixed a bug where only the tests in the first shard were run. by ehmaldonado · 8 years ago
  96. 0182f85 More reliable ALR detection by Sergey Ulanov · 8 years ago
  97. 3a1c40a MB: Remove configuration for unexisting bots. by ehmaldonado · 8 years ago
  98. b4af3d6 Remove all references to GYP by Henrik Kjellander · 8 years ago
  99. 67fcad8 Relax the PostDelayed expectations a little more to address flakiness. by tommi · 8 years ago
  100. 08127a9 Reland #2 of Issue 2434073003: Extract bitrate allocation ... by Erik Språng · 8 years ago