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gerrit-public.fairphone.software / platform / external / webrtc / f08fb29aa5da230ba32654a10a1317daeb73ffa8
  1. f08fb29 Disable flaky test (WebRtcVideoChannel2Base.AddRemoveCapturer) on Dr. Memory. by Åsa Persson · 9 years ago
  2. c06a716 Android: Add new renderer SurfaceViewRenderer by Magnus Jedvert · 9 years ago
  3. df1a171 Remove unused event in video_capture_input.cc. by mflodman · 9 years ago
  4. d12140a Revert change which removes GICE. by guoweis · 9 years ago
  5. fab882b Remove obsolete typingmonitor.cc/.h files. by solenberg · 9 years ago
  6. 4ed3658 Avoids crashes in Java-based InitRecording(). by henrika · 9 years ago
  7. 39720f2 ACM CodecOwner: Test that we reset speech encoder when enabling CNG or RED by kwiberg · 9 years ago
  8. 9b66d3b MockAudioEncoder: Use a dedicated marker method for test expectations by kwiberg · 9 years ago
  9. 1dd98f3 - Rename VoiceChannel::MuteStream() -> SetAudioSend() (incl. media channel) by solenberg · 9 years ago
  10. 9a78d22 Revert of Consolidate constructormagic macros with Chromium version and remove Chromium override. (patchset #4 id:60001 of https://codereview.webrtc.org/1316363005/ ) by tommi · 9 years ago
  11. 0de8ff4 Consolidate constructormagic macros with Chromium version and remove Chromium override. by henrikg · 9 years ago
  12. 11e4985 GN: Fix iOS build. by Henrik Kjellander · 9 years ago
  13. c2db810 Remove VideoRendererInterface::CanApplyRotation() by Magnus Jedvert · 9 years ago
  14. f6901b0 Remove NullVideoFrame by Magnus Jedvert · 9 years ago
  15. 8ce0bd5 Android video rendering: Fix texture matrix multiplication order by Magnus Jedvert · 9 years ago
  16. 942a699 AudioEncoderOpusTest.PacketLossRateOptimized: Fix bug and make prettier by kwiberg · 9 years ago
  17. 2feafdb Enable automatic resizing for RTX-enabled senders. by Peter Boström · 9 years ago
  18. 77d22fa Merge two files with AudioEncoderOpus tests by kwiberg · 9 years ago
  19. 529528c Android video rendering: Apply SurfaceTexture.getTransformationMatrix() by Magnus Jedvert · 9 years ago
  20. 66f4339 Remove [Voice|Video]MediaChannel::GetOptions(). by solenberg · 9 years ago
  21. c99ebc1 Remove AudioEncoder methods SetMaxBitrate and SetMaxPayloadSize by kwiberg · 9 years ago
  22. d944067 Disable flaky test (WebRtcVideoChannel2Base.GetStatsMultipleSendStreams) on Dr. Memory. by Åsa Persson · 9 years ago
  23. b04965c Hooked up RtcEventLog. It lives in Voice Engine and pointers are propagated to ACM and Call. by ivoc · 9 years ago
  24. 3f5f1c2 Change return type of AudioEncoder::SetMaxPlaybackRate to void by kwiberg · 9 years ago
  25. e9e7896 Turn webrtc::Vad into a pure virtual interface by kwiberg · 9 years ago
  26. 233bd87 Add RemoteEstimatorProxy for capturing receive times by sprang · 9 years ago
  27. 66c42df Alphabetize common_audio/OWNERS. by Andrew MacDonald · 9 years ago
  28. 7764973 Add magjed@ as owner for talk/app/webrtc/androidtests/ and talk/app/webrtc/java/jni/ by Magnus Jedvert · 9 years ago
  29. 76b3147 Disable flaky WebRtcVideoChannel2Base, EndToEndTest tests on Dr. Memory. by Åsa Persson · 9 years ago
  30. 12cfc9b Fold AudioEncoderMutable into AudioEncoder by kwiberg · 9 years ago
  31. cd3c475 Updating common_audio/OWNERS by henrik.lundin · 9 years ago
  32. 68786d2 Wire up PacketTime to ReceiveStreams. by stefan · 9 years ago
  33. e526974 Make LoadObserver settable per video send stream. Gives client flexibility and makes the implementation slightly simpler. See discussion in: https://codereview.webrtc.org/1269863005/ by solenberg · 9 years ago
  34. a9839dd Use of override keyword to fix chromium trybot by Henrik Boström · 9 years ago
  35. 04ada47 Add third_party/lss and third_party/proguard to .gitignore. by henrikg · 9 years ago
  36. f325d21 Disable VideoSendStreamTest.VP9FlexMode. by philipel · 9 years ago
  37. c3aa12d Add utility class for unwrapping 16 bit sequence numbers by sprang · 9 years ago
  38. caa5f4b Update to the neteq_rtpplay utility to support RtcEventLog input files. by ivoc · 9 years ago
  39. f3ecdb9 Replacing SSLIdentity* with scoped_refptr<RTCCertificate> in TransportChannel layer. by Henrik Boström · 9 years ago
  40. 8006f07 Remove unused TypingMonitor class. by solenberg · 9 years ago
  41. 7f6a6fc Enabling spatial layers in VP9Impl. Filter layers in the loopback test. by ivica · 9 years ago
  42. e313e02 Remove unnecessary fields from VoE SharedData. by solenberg · 9 years ago
  43. 746210f Remove unused overuse detection metric (capture jitter). by Åsa Persson · 9 years ago
  44. 3dfe5d3 Remove arraysize.h gcc hack and Chromium override. by henrikg · 9 years ago
  45. e9ad18b Remove obsolete soundclip.cc/.h files. by solenberg · 9 years ago
  46. 1c7d48d Let max default bitrate depend on resolution when configuring one video stream (was previously always 2Mbps). by Åsa Persson · 9 years ago
  47. 6322467 PRESUBMIT: Exclude some files from 80-character limit check. by Henrik Kjellander · 9 years ago
  48. 81db11a copy-red: Fill an rtc::Buffer with bytes the easy way by kwiberg · 9 years ago
  49. 86d907c Refactor the AudioDevice for iOS and improve the performance and stability by henrika · 9 years ago
  50. 05cfcd3 Full stack graphs by ivica · 9 years ago
  51. 110443c Fix for frame resolution in encoded frame callback. by Asa Persson · 9 years ago
  52. 7b38f69 Add placeholder files for talk/app/webrtc/mediacontroller.cc/.h to be able to update Chrome's libjingle.gyp before the MediaController implementation CL is submitted. by solenberg · 9 years ago
  53. c0c7d2e GN: Fix invalid configuration for Android GCC build. by Henrik Kjellander · 9 years ago
  54. bb741b3 Remove GetOutputScaling from VoiceMediaChannel. by solenberg · 9 years ago
  55. 0ab8ca8 Remove x11 from libjingle_media by phoglund · 9 years ago
  56. 88703d7 Disable base/logging.h stderr logs by default for webrtc/ tests. by andrew · 9 years ago
  57. 9eb1365 Revert of purge nss files and dependencies (patchset #1 id:1 of https://codereview.webrtc.org/1313233005/ ) by deadbeef · 9 years ago
  58. fd4df46 Fix build when using Xcode 7 which contains .tbd files instead of .dylib by hjon · 9 years ago
  59. d5ae6ae Fix ScreenCapturerWinGdi to handle DesktopFrameWin::Create() errors. by sergeyu · 9 years ago
  60. 3cc834a Add more IceCandidatePairType for host-host CandidatePair by Guo-wei Shieh · 9 years ago
  61. 250bdc7 Exclude VideoSendStreamTest.VP9FlexMode on linux_memcheck. by Marco · 9 years ago
  62. 5647a2c purge nss files and dependencies by torbjorng · 9 years ago
  63. e7a0de7 CameraEnumerationAndroid: Add getSupportedFormats() implementation using android.hardware.camera2 by Magnus Jedvert · 9 years ago
  64. 242d638 VP9 codec controls for screensharing by ivica · 9 years ago
  65. 318673c Update SendTimeHistory to store complete PacketInfo, not just send time by sprang · 9 years ago
  66. c8a1ccc Fixed base time in TransportFeedback message writing. by sprang · 9 years ago
  67. d415629 Remove AsyncHttpRequest, AutoPortAllocator, ConnectivityChecker, and HttpPortAllocator. by Peter Thatcher · 9 years ago
  68. 2f9fd5d Changed LogRtpHeader to read the header length from the packet instead of requiring an extra argument. by terelius · 9 years ago
  69. b6b0b92 Rate limit the low bandwidth / min bitrate warning to once every 10 seconds. by stefan · 9 years ago
  70. be9b7b6 Make sure ByteReader and ByteWriter classes (and their specializations) don't perform operations that have implementation-specific or undefined behavior. by sprang · 9 years ago
  71. 47d78cc Pass the encoder's internal source property through to video_sender to request a keyframe from the external encoder by sophiechang · 9 years ago
  72. dfbe679 Cleanup: Remove duplicated functions by Guo-wei Shieh · 9 years ago
  73. 9743d07 Reland "Adding unittests to AudioConferenceMixer." by minyue · 9 years ago
  74. 6ee69aa Add scrolling screenshare test to full_stack perf tests. by Erik Språng · 9 years ago
  75. 658910c Revert "Speculative revert of "- Move test cases for more natural ordering."" by stefan · 9 years ago
  76. 7fabd46 Don't set V bit in flexible mode by philipel · 9 years ago
  77. 7afc12f VideoRendererGui: Move to async rendering and remove no longer needed code by Magnus Jedvert · 9 years ago
  78. 4df08ff GN: Fix compilation with NaCl toolchain by sergeyu · 9 years ago
  79. 1a591dd Android GlUtil: Add helper functions generateTexture/deleteTexture by Magnus Jedvert · 9 years ago
  80. 6aae757 On FATAL, log which unsupported encoder the caller wanted us to create by kwiberg · 9 years ago
  81. ed4224f Android GlRectDrawer: Add fragment shader for RGB(A) textures by Magnus Jedvert · 9 years ago
  82. d4563f4 Revert of Excluding two troublesome trybots from the CQ config. (patchset #1 id:1 of https://codereview.webrtc.org/1310953006/ ) by kjellander · 9 years ago
  83. 71cfe69 For TestResolverShutdown, use address that can't be resolved. by deadbeef · 9 years ago
  84. e63d2a1 Add JNI/java wrapper for the file rotating logging class. by Jiayang Liu · 9 years ago
  85. abd0d1a Handle all RTCICEConnectionState values in ARDVideoCallViewController by hjon · 9 years ago
  86. 4d2f4d1 - Make shared EGL context used for HW video decoding member by Alex Glaznev · 9 years ago
  87. c36d4df Use committer list from chrome-infra-auth by Sergiy Byelozyorov · 9 years ago
  88. 098c1de Fixes for PNaCl build of remoting client plugin with GN. by Sergey Ulanov · 9 years ago
  89. 97579a4 Add option to enable ECDSA key for Java API. by glaznev · 9 years ago
  90. eebc099 Add magjed@ as owner for talk/app/webrtc/java/android/org/webrtc/ by magjed · 9 years ago
  91. 194ccea Do not use HW H.264 encoder on Nexus 7. by Alex Glaznev · 9 years ago
  92. 4edc39c Set the IceConnectionReceivingTimeout as a RTCConfiguration parameter. by honghaiz · 9 years ago
  93. 0f9af01 Added send stream test case for VP9 header. by philipel · 9 years ago
  94. fa7cb8e Excluding two troublesome trybots from the CQ config. by Henrik Kjellander · 9 years ago
  95. 02d283a Speculative revert of "- Move test cases for more natural ordering." by Stefan Holmer · 9 years ago
  96. 05f71fc NetEq: Fixing a corner case with depleted sync buffer by Henrik Lundin · 9 years ago
  97. 521875a Use RtcpPacket to send APP in RtcpSender by Erik Språng · 9 years ago
  98. e551f12 Revert "Adding unittests to AudioConferenceMixer." by minyuel · 9 years ago
  99. 22c2729 Adding unittests to AudioConferenceMixer. by minyuel · 9 years ago
  100. b7306ae Revert "Avoiding size_t in MIPS." by Niklas Enbom · 9 years ago