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gerrit-public.fairphone.software / platform / external / webrtc / 7bd242e53dacc22eaddbe27a74bdd30d672e57db
  1. 7bd242e Enabling screensharing tests for Android by ivica · 9 years ago
  2. 9359b5b Disabling AudioDeviceTest.StartStopPlayout on Android. by Henrik Kjellander · 9 years ago
  3. d89f82a Roll chromium_revision c511263..4bf3678 (352322:352512) by kjellander · 9 years ago
  4. 09f1350 Add option to reset Android video renderer first frame flag. by Alex Glaznev · 9 years ago
  5. 6caafbe Convert uint16_t to int for WebRTC cipher/crypto suite. by Guo-wei Shieh · 9 years ago
  6. 1b33da1 SurfaceTextureHelper fixes by perkj · 9 years ago
  7. 4185032 Add ThreadChecker class to ThreadUtils by perkj · 9 years ago
  8. d2838a7 Roll chromium_revision 07b4a8e..c511263 (352281:352322) by Henrik Kjellander · 9 years ago
  9. e0bce24 VideoCapturerAndroid: Add custom nativeCreateVideoCapturer() by perkj · 9 years ago
  10. 723dff1 Poll stats more often to get more stable stats in ramp-up tests. by Stefan Holmer · 9 years ago
  11. 4cd053f Only catch UnsatisfiedLinkError in Logging.java. by jiayl · 9 years ago
  12. f3a7c9d In rampup tests, set start time when starting poller thread. by Erik Språng · 9 years ago
  13. 95cd8ea Enable HW NS for N6 to fix HW AEC issue by henrika · 9 years ago
  14. dec5ebf Move sent key frame stats to send_statistics_proxy class. by asapersson · 9 years ago
  15. 42b6c63 autoroller: Allow to specify Rietveld e-mail. by Henrik Kjellander · 9 years ago
  16. 990d57d Fix file order in base.gyp. by henrikg · 9 years ago
  17. ba0f0a5 Disable flaky WebRtcVideoChannel2BaseTest.* on DrMemory/memcheck. by Peter Boström · 9 years ago
  18. 4bd8d09 Roll chromium_revision ca4c339..07b4a8e (352257:352281) by kjellander · 9 years ago
  19. 96a70f0 Exclude WebRtcVideoChannel2BaseTest.AddRemoveCapturerMultipleSources on Dr Memory. by Henrik Kjellander · 9 years ago
  20. b5fd46e Exclude WebRtcVideoChannel2BaseTest.AddRemoveCapturerMultipleSources on Memcheck by Henrik Kjellander · 9 years ago
  21. 42b4faa Fix a build issue when use external OpenSSL. by Guo-wei Shieh · 9 years ago
  22. 6df1ef6 Roll chromium_revision 4ce3c08..ca4c339 (352000:352257) by Henrik Kjellander · 9 years ago
  23. bc0938e Android VideoRendererGui: Make deep copy of incoming texture frames by Magnus Jedvert · 9 years ago
  24. 44bf6f5 Android MediaCodecVideoDecoder: Split DecoderOutputBufferInfo into DecodedByteBuffer and DecodedTextureBuffer by magjed · 9 years ago
  25. 13b96ba Adding APM configuration in AEC dump. by Minyue · 9 years ago
  26. 371dc7e WebRtc Win Desktop capture: ignore Win8+ Modern Apps' windows. by gyzhou · 9 years ago
  27. 913e645 Loopback and audio only mode. by haysc · 9 years ago
  28. f9c23ca Exclude WebRtcVideoChannel2BaseTest.GetStats on linux memcheck by Marco · 9 years ago
  29. 9dff0ba Fix MSVS project files generation. by henrikg · 9 years ago
  30. 067fb65 Roll chromium_revision 7fddcec..4ce3c08 (351973:352000) by kjellander · 9 years ago
  31. a050e98 Avoid race in RampUpTest by sprang · 9 years ago
  32. 7e31937 Android MediaCodecVideoDecoder: Cleanup to prepare for texture liftime management by Magnus Jedvert · 9 years ago
  33. 6781ea4 jni/native_handle_impl.h: Move implementation into .cc file by Magnus Jedvert · 9 years ago
  34. 417fec2 autoroller: Add CQ_EXTRA_TRYBOTS, CQ feature and --skip-cq flag. by Henrik Kjellander · 9 years ago
  35. 401025d Roll chromium_revision 354cc7d..7fddcec (351828:351973) by Henrik Kjellander · 9 years ago
  36. 1d8a506 Add a PacketOptions struct to webrtc::Transport. by stefan · 9 years ago
  37. da903ea Unify newapi::RtcpMode and RTCPMethod. by pbos · 9 years ago
  38. c8ba105 Roll chromium_revision 681f0cd..354cc7d (351698:351828) by kjellander · 9 years ago
  39. a9c584d autoroller: Always roll and improve description by Henrik Kjellander · 9 years ago
  40. 6c2ba7d autoroller: Add TBR= field and always update the checkout by Henrik Kjellander · 9 years ago
  41. 18b042f autoroller: Use HEAD instead of LKGR. by Henrik Kjellander · 9 years ago
  42. 5aaa9b4 Removed unused API functions in AudioProcessing and AudioProcessingModule by peah · 9 years ago
  43. 5629a1d Fix flaky test TestSrtpError, introduced in https://codereview.webrtc.org/1362913004. by solenberg · 9 years ago
  44. cf18b34 Align new VoE API with design. by solenberg · 9 years ago
  45. 8c471e7 Objective-C++ style guide changes for iOS ADM by henrika · 9 years ago
  46. fb30c1b Update VP8 settings to avoid spending bitrate on static areas. PERF NOTE by sprang · 9 years ago
  47. 5b14b42 Remove unused SignalMediaError and infrastructure. by solenberg · 9 years ago
  48. 49f9cdb Fix bug where rtcp::TransportFeedback may generate incorrect messages. by sprang · 9 years ago
  49. b09b660 Remove cricket::VideoFrame::Set/GetElapsedTime() by magjed · 9 years ago
  50. dfc8f4f Change 'mute' parameter of MediaChannel::SetAudioSend()/SetVideoSend() to 'enable'. by solenberg · 9 years ago
  51. edba998 Roll chromium_revision 8cf53d6..681f0cd (351112:351698) by kjellander · 9 years ago
  52. 0ecf1b2 Android focus problem on rear camera. by dchakarov.broadsoft · 9 years ago
  53. 98ab3a4 Don't link with audio codecs that we don't use by kwiberg · 9 years ago
  54. 456696a Reland Change WebRTC SslCipher to be exposed as number only by Guo-wei Shieh · 9 years ago
  55. 27dc29b Revert of Change WebRTC SslCipher to be exposed as number only. (patchset #20 id:750001 of https://codereview.webrtc.org/1337673002/ ) by guoweis · 9 years ago
  56. 4fe3c9a Change WebRTC SslCipher to be exposed as number only. by guoweis · 9 years ago
  57. 2f8a4ca Add OWNERS for ObjC dirs. by tkchin · 9 years ago
  58. d0b3143 Do not time out a port if its role switched from controlled to controlling. Also fix some comments. by honghaiz · 9 years ago
  59. 898d21c WebRTC might leak srflx ip address when multiple_routes disabled and IceTransportType is relay. by Guo-wei Shieh · 9 years ago
  60. c4d3a5d Thinning out the Transport class. by Taylor Brandstetter · 9 years ago
  61. 2b342bf Delete a connection only if it has timed out on writing and not receiving for 10 seconds. by Honghai Zhang · 9 years ago
  62. 27551c9 Android RendererCommon: Refactor getSamplingMatrix() by Magnus Jedvert · 9 years ago
  63. 4a8e9c5 Remove overrides folder. by henrikg · 9 years ago
  64. bbda54e Android MediaDecoder: Use frame pool to avoid allocations for non-surface decoding by Magnus Jedvert · 9 years ago
  65. ee2bf41 Update build files to use webrtc_overrides in Chromium instead of overrides. by henrikg · 9 years ago
  66. 6ba8e4a ACM: Remove a few local enums that were no longer used by Henrik Lundin · 9 years ago
  67. d094c04 Remove AgcManager. by Alejandro Luebs · 9 years ago
  68. a67696b Reland of Adding PeerConnectionInterface::SetConfiguration method. (patchset #1 id:1 of https://codereview.webrtc.org/1361263002/ ) by deadbeef · 9 years ago
  69. 38778b0 Add unit test for nack bandwidth constraint. by sprang · 9 years ago
  70. 98db68f If gather_continually is set to true, keep the last port allocator session running while stopping all existing process of getting ports (when p2ptransportchannel first becomes writable). by honghaiz · 9 years ago
  71. 24b52f8 Android GlRectDrawer: Add test for OES texture rendering by Magnus Jedvert · 9 years ago
  72. 1d640e5 JavaVideoRendererWrapper: Use jlongFromPointer() to convert frame pointer to jlong by Magnus Jedvert · 9 years ago
  73. 63b3454 Simplify handling of options in WebRtcVoiceMediaEngine. by solenberg · 9 years ago
  74. 86fd9ed Set RtcpSender transport at construction. by sprang · 9 years ago
  75. 38502a7 Remove isolate_deps_dir from .gitignore by Henrik Kjellander · 9 years ago
  76. 092508a Fix bug in ramp-up tests stats where rtx was accounted for in the media ssrc. by stefan · 9 years ago
  77. b5815c8 Revert of Android VideoCapturer: Send ByteBuffer instead of byte[] (patchset #1 id:1 of https://codereview.webrtc.org/1372813002/ ) by magjed · 9 years ago
  78. fb9e763 Remove last use of ACMAMRPackingFormat by henrik.lundin · 9 years ago
  79. d6024e3 Roll chromium_revision 310ea93..8cf53d6 (349094:351112) by kjellander · 9 years ago
  80. 70ab1a1 Exposing RtpSenders and RtpReceivers from PeerConnection. by deadbeef · 9 years ago
  81. 8e9cb09 Android: Add unittests for SurfaceTextureHelper by Magnus Jedvert · 9 years ago
  82. 4fa648b Adding 20-second timeout to Java and Objective-C tests. by deadbeef · 9 years ago
  83. 8108764 Analyze support in gyp_webrtc by Henrik Kjellander · 9 years ago
  84. 2d56668 Unify Transport and newapi::Transport interfaces. by pbos · 9 years ago
  85. 8387c5f Remove AMR format parameter from AudioCoder in utility by henrik.lundin · 9 years ago
  86. 1968d3f Simplify VCMTimestampMap. by pbos · 9 years ago
  87. 8c404fa When doing DisableEquivalentPhases, exclude those AllocationSequences by honghaiz · 9 years ago
  88. 1f429e3 Passing the new policy from PeerConnection RTCConfiguration to by honghaiz · 9 years ago
  89. cb3649b Android VideoCapturer: Send ByteBuffer instead of byte[] by Magnus Jedvert · 9 years ago
  90. 4b808ee ACM: Remove unused and deprecated types by Henrik Lundin · 9 years ago
  91. 1bd0e03 ACM: Removing runtime APIs related to playout mode by henrik.lundin · 9 years ago
  92. d417523 Minor fix for debug logging on Android by henrika · 9 years ago
  93. 4fbd145 Fix suspend below min bitrate in new API by making it possible to set min bitrate at the receive-side. by stefan · 9 years ago
  94. d2413e5 Fix the C++ SurfaceTextureHolder by Per · 9 years ago
  95. 1ab271c Android SurfaceTextureHelper: Don't wait for pending frames in disconnect() by Magnus Jedvert · 9 years ago
  96. 3e9eb4b Add C++ SurfaceTextureHandler by Per · 9 years ago
  97. 82d6f2a ACM: Remove ACMVQMonCallback object by Henrik Lundin · 9 years ago
  98. 69984f0 Fixes logging levels in WebRtcAudioXXX.java classes by henrika · 9 years ago
  99. d6d27e7 Update isolate.gypi to support Swarming + move .isolate files by Henrik Kjellander · 9 years ago
  100. c97be6a Disable TestUdpReadyToSendIPv4 under MSan. by deadbeef · 9 years ago