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gerrit-public.fairphone.software / platform / external / webrtc / 7fabd46a89675da596b28bb43c8fd3c561fbe85e
  1. 7fabd46 Don't set V bit in flexible mode by philipel · 9 years ago
  2. 7afc12f VideoRendererGui: Move to async rendering and remove no longer needed code by Magnus Jedvert · 9 years ago
  3. 4df08ff GN: Fix compilation with NaCl toolchain by sergeyu · 9 years ago
  4. 1a591dd Android GlUtil: Add helper functions generateTexture/deleteTexture by Magnus Jedvert · 9 years ago
  5. 6aae757 On FATAL, log which unsupported encoder the caller wanted us to create by kwiberg · 9 years ago
  6. ed4224f Android GlRectDrawer: Add fragment shader for RGB(A) textures by Magnus Jedvert · 9 years ago
  7. 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
  8. 71cfe69 For TestResolverShutdown, use address that can't be resolved. by deadbeef · 9 years ago
  9. e63d2a1 Add JNI/java wrapper for the file rotating logging class. by Jiayang Liu · 9 years ago
  10. abd0d1a Handle all RTCICEConnectionState values in ARDVideoCallViewController by hjon · 9 years ago
  11. 4d2f4d1 - Make shared EGL context used for HW video decoding member by Alex Glaznev · 9 years ago
  12. c36d4df Use committer list from chrome-infra-auth by Sergiy Byelozyorov · 9 years ago
  13. 098c1de Fixes for PNaCl build of remoting client plugin with GN. by Sergey Ulanov · 9 years ago
  14. 97579a4 Add option to enable ECDSA key for Java API. by glaznev · 9 years ago
  15. eebc099 Add magjed@ as owner for talk/app/webrtc/java/android/org/webrtc/ by magjed · 9 years ago
  16. 194ccea Do not use HW H.264 encoder on Nexus 7. by Alex Glaznev · 9 years ago
  17. 4edc39c Set the IceConnectionReceivingTimeout as a RTCConfiguration parameter. by honghaiz · 9 years ago
  18. 0f9af01 Added send stream test case for VP9 header. by philipel · 9 years ago
  19. fa7cb8e Excluding two troublesome trybots from the CQ config. by Henrik Kjellander · 9 years ago
  20. 02d283a Speculative revert of "- Move test cases for more natural ordering." by Stefan Holmer · 9 years ago
  21. 05f71fc NetEq: Fixing a corner case with depleted sync buffer by Henrik Lundin · 9 years ago
  22. 521875a Use RtcpPacket to send APP in RtcpSender by Erik Språng · 9 years ago
  23. e551f12 Revert "Adding unittests to AudioConferenceMixer." by minyuel · 9 years ago
  24. 22c2729 Adding unittests to AudioConferenceMixer. by minyuel · 9 years ago
  25. b7306ae Revert "Avoiding size_t in MIPS." by Niklas Enbom · 9 years ago
  26. 32e2f46 Avoiding size_t in MIPS. by minyuel · 9 years ago
  27. 2c27430 Print some output in long perf tests, to keep them alive by Erik Språng · 9 years ago
  28. 0f4b373 Stylizing AudioConferenceMixer. by minyuel · 9 years ago
  29. ca28fdc Use RtcpPacket to send XR (RTRR, DLRR, VOIP) in RtcpSender by Erik Språng · 9 years ago
  30. c252dab CameraEnumerationAndroid: Make getSupportedFormats() an interface by Magnus Jedvert · 9 years ago
  31. c92c23d Roll chromium_revision f8d6ba9..a28d8d5 (337800:346100) by Patrik Höglund · 9 years ago
  32. c20a5dc - Move test cases for more natural ordering. by Fredrik Solenberg · 9 years ago
  33. 3c4ef29 NetEq: Allow negative shift in BackgroundNoise::SaveParameters by Henrik Lundin · 9 years ago
  34. 3a14bf3 Replacing SSLIdentity* with scoped_refptr<RTCCertificate> in the cricket::TransportDescriptionFactory layers. by Henrik Boström · 9 years ago
  35. a6cba3a Java VideoRenderer.Callbacks: Make renderFrame() interface asynchronous by Magnus Jedvert · 9 years ago
  36. 1380e26 Convert some more things to size_t. by Peter Kasting · 9 years ago
  37. e8386d2 Added send-thresholding and fixed text packet dumping. Also a little squelch for the over-max-MTU log spam we see in there. by Lally Singh · 9 years ago
  38. 79de90b Do not explicitly delete OpenGL shaders in VideoRendererGui. by Alex Glaznev · 9 years ago
  39. f42376c Wire up currently-received video codec to stats. by pbos · 9 years ago
  40. 6813ec8 VideoCapturerAndroid: Move to android folder and split out camera enumeration into separate file by magjed · 9 years ago
  41. 9e69abf Added logging using the raw variant of the new aec logging macros by peah · 9 years ago
  42. 4fbae2b Add send transports to individual webrtc::Call streams. by solenberg · 9 years ago
  43. 6480d03 Make jni_helpers build on arm32. by phoglund · 9 years ago
  44. 6ec1f92 AndroidVideoCapturer: Delegate framerate choice to VideoCapturerAndroid.java by Magnus Jedvert · 9 years ago
  45. 98f3cc5 NetEq: Removing two asserts by henrik.lundin · 9 years ago
  46. 1e346b2 NetEq: Minor follow-up fix in StatisticsCalculator by henrik.lundin · 9 years ago
  47. 116c84e NetEq: Fixing a bug that caused rtc::checked_cast to trigger by henrik.lundin · 9 years ago
  48. 9c3efd0 Reland: Implement NetEq's CurrentDelay function by henrik.lundin · 9 years ago
  49. a567bf3 Rename local variable to avoid shadowing by kwiberg · 9 years ago
  50. 3154568 Using 'override' keyword in dtlstransport.h. by Henrik Boström · 9 years ago
  51. 4376648 AudioDecoder: Replace Init() with Reset() by Karl Wiberg · 9 years ago
  52. 1c3dd38 Android: Fix memory leak for remote MediaStream by Magnus Jedvert · 9 years ago
  53. 7391881 Revert of Added send-thresholding and fixed text packet dumping. (patchset #4 id:160001 of https://codereview.webrtc.org/1266033005/ ) by tommi · 9 years ago
  54. fdac516 Disallow simulcast for H.264. by noahric · 9 years ago
  55. d828198 Replaces SSLIdentity* with scoped_refptr<RTCCertificate> in the cricket::Transport layer. by Henrik Boström · 9 years ago
  56. d83df50 Use RtcpPacket to send TMMBN in RtcpSender by sprang · 9 years ago
  57. c47a01d Fix AppRTCDemo crash when room is connected after PC is destroyed. by Alex Glaznev · 9 years ago
  58. 13d35f6 Add check to prevent TURN usage if the protocol is not allowed. by Guo-wei Shieh · 9 years ago
  59. 2f20fbe Fix MIPS compile. by Peter Kasting · 9 years ago
  60. 0163fb2 AudioCodingModuleImpl::Encode: Use a Buffer instead of a stack-allocated array by Karl Wiberg · 9 years ago
  61. d838d27 Added send-thresholding and fixed text packet dumping. Also a little squelch for the over-max-MTU log spam we see in there. by Lally Singh · 9 years ago
  62. 3318f98 VideoFrameBuffer: Make non-const data access explicit by Magnus Jedvert · 9 years ago
  63. 85ad62b Remove per-frame captured frame logging. by Noah Richards · 9 years ago
  64. af9fb21 - Use C++11 loops in WebRtcVoiceMediaEngine/Channel. by Fredrik Solenberg · 9 years ago
  65. c464f50 AndroidVideoCapturerJni: Fix threading issues by Magnus Jedvert · 9 years ago
  66. c464b40 Android RendererCommon: Add unittests for getTextureMatrix() by Magnus Jedvert · 9 years ago
  67. 1eb87c7 TCPConnection can never be deteted if they fail to connect. by Guo-wei Shieh · 9 years ago
  68. 9b35115 Move mock_nonlinear_beamformer to only be a header by Alejandro Luebs · 9 years ago
  69. b274547 rtc::Bind: Capture scoped_refptr reference arguments by value by Magnus Jedvert · 9 years ago
  70. f4772ee Get rid of unused types and constants in acm_common_defs.h by Karl Wiberg · 9 years ago
  71. 1bb8cf8 NetEq/ACM: Refactor how packet waiting times are calculated by Henrik Lundin · 9 years ago
  72. 7230a21 Android RendererCommon: Add unittests for getDisplaySize() by Magnus Jedvert · 9 years ago
  73. b6cac8f Get rid of the manual destructor in AudioCodingModuleImpl by Karl Wiberg · 9 years ago
  74. 87a8fbb Fixing Pylint errors for plot_dynamics.py by Ivica Kicic · 9 years ago
  75. 87713d0 RTCCertificates added to RTCConfiguration, used by WebRtcSession/-DescriptionFactory. by Henrik Boström · 9 years ago
  76. dd00f11 Remove no-op and unused methods from AudioCodingModule by Karl Wiberg · 9 years ago
  77. 7ef9d91 Android: Remove VideoRenderer.Callbacks.canApplyRotation() by Magnus Jedvert · 9 years ago
  78. bc2296d Add a base class to Wav{Reader,Writer} to access shared parameters. by Andrew MacDonald · 9 years ago
  79. 41eeff4 More iOS compile fixes. by Peter Kasting · 9 years ago
  80. deb4875 Fix typos in https://codereview.webrtc.org/1230503003/ not caught by trybots. by Peter Kasting · 9 years ago
  81. dce40cf Update a ton of audio code to use size_t more correctly and in general reduce by Peter Kasting · 9 years ago
  82. b594041 TcpPort Reconnect should inform upper layer to start sending again. by Guo-wei Shieh · 9 years ago
  83. 39b8eb3 Fix Chromium GN build problem introduced in 608c3cfe by Karl Wiberg · 9 years ago
  84. 4e14f09 Add support for external decoders in ACM by kwiberg · 9 years ago
  85. e7cdc7f No-op CL to test if video engine core problem solved. by phoglund · 9 years ago
  86. d8ee4f9 Use RtcpPacket to send BYE in RtcpSender by sprang · 9 years ago
  87. 608c3cf iSAC: Make separate AudioEncoder and AudioDecoder objects by kwiberg · 9 years ago
  88. 2159b89 Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots. by Peter Thatcher · 9 years ago
  89. 9deaa86 Fix initialization/termination of AudioDeviceTemplate by kaorimatz · 9 years ago
  90. 7612f17 Fix accidental redeclaration. by Andrew MacDonald · 9 years ago
  91. c0775c0 Fix accessing uninitialized variables when not processing a reverse stream. by Andrew MacDonald · 9 years ago
  92. ea1012b address comments from https://codereview.webrtc.org/1277263002/ by guoweis · 9 years ago
  93. 5bdafd4 Revert "Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots."" by minyuel · 9 years ago
  94. 81a3e60 Use RtcpPacket to send TMMBR in RtcpSender by sprang · 9 years ago
  95. dd4edc5 Reland of Use RtcpPacket to send REMB in RtcpSender (patchset #1 id:1 of https://codereview.webrtc.org/1300863002/ ) by sprang · 9 years ago
  96. c232096 Remove cricket::VideoProcessor and AddVideoProcessor() functionality by Magnus Jedvert · 9 years ago
  97. 9d15c66 Include webrtc/base/json.h rather than from jsoncpp directly. by phoglund · 9 years ago
  98. 22ff75a Add unit tests for more packet types in rtcp_sender_unittest. by asapersson · 9 years ago
  99. bfab5cb Fix some minor errors with the voice engine caused by the refactor CL https://codereview.webrtc.org/1229283003/. by Peter Thatcher · 9 years ago
  100. a5b273a Fixing problems with RTP extension ID conflict resolution by deadbeef · 9 years ago