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gerrit-public.fairphone.software / platform / external / webrtc / 0df3eb03c9a6a8299d7e18c8c314ca58c2f0681e
  1. 0df3eb0 provide RSA2048 as per RFC by torbjorng · 9 years ago
  2. f839dcc Add stats for rendered pixels (sqrt(w*h)) per second: by asapersson · 9 years ago
  3. e0e88cf Roll chromium_revision 22ce537..2a2c52e (352889:353013) by kjellander · 9 years ago
  4. e78e2c7 Using different sequence numbers for different SSRCs by ivica · 9 years ago
  5. fddf6e5 Use WebRTC logging in MediaCodec JNI code. by Alex Glaznev · 9 years ago
  6. 21622a1 Add option to print peer connection factory Java stack traces. by Alex Glaznev · 9 years ago
  7. 91b348c Android MediaCodecVideoDecoder: Manage lifetime of texture frames by Magnus Jedvert · 9 years ago
  8. 87962a9 Roll chromium_revision df4218d..22ce537 (352811:352889) by kjellander · 9 years ago
  9. c7199c2 Read the number of TLs for VP9 too + cleanup by ivica · 9 years ago
  10. 7854328 Fix minor GYP error in webrtc/tools/internal_tools.gyp by Henrik Kjellander · 9 years ago
  11. 172f009 Get rid of SCHANNEL code. by torbjorng · 9 years ago
  12. 0a6380f Roll chromium_revision d47c242..df4218d (352743:352811) by kjellander · 9 years ago
  13. 70a5e0e Remove (u)int typedefs from basictypes.h. by Peter Boström · 9 years ago
  14. 0c4e06b Use suffixed {uint,int}{8,16,32,64}_t types. by Peter Boström · 9 years ago
  15. 8d15bd6 Reland of Collecting encode_time_ms for each frame (patchset #1 id:1 of https://codereview.webrtc.org/1383283005/ ) by ivica · 9 years ago
  16. d97ec30 Remove default receive channel from WVoE; baby step 0. by solenberg · 9 years ago
  17. 67bcb60 GN: Port frame_analyzer and rgba_to_i420_converter targets by Henrik Kjellander · 9 years ago
  18. 7f315f5 Roll chromium_revision 6ebca7d..d47c242 (352640:352743) by kjellander · 9 years ago
  19. a38e31a Update lower-level codereview.settings files. by Henrik Kjellander · 9 years ago
  20. a10492f Fix VS 2015 warning by adding an additional cast by brucedawson · 9 years ago
  21. 4139c0f Java binding for RtpSender/RtpReceiver. by deadbeef · 9 years ago
  22. 1095069 Revert "Transport sequence number should be set also for retransmissions." by Alejandro Luebs · 9 years ago
  23. a295320 Roll chromium_revision 2a1b03e..6ebca7d (352576:352640) by kjellander · 9 years ago
  24. 0a6c4ca Catching more errors when parsing ICE server URLs. by deadbeef · 9 years ago
  25. 8104479 Revert of Collecting encode_time_ms for each frame (patchset #13 id:220001 of https://codereview.webrtc.org/1374233002/ ) by kjellander · 9 years ago
  26. 092b133 Collecting encode_time_ms for each frame. by ivica · 9 years ago
  27. af4ced9 Transport sequence number should be set also for retransmissions. by sprang · 9 years ago
  28. 86fa298 Roll chromium_revision 4bf3678..2a1b03e (352512:352576) by kjellander · 9 years ago
  29. 5d0379d Remove kSkipFrame usage. by Peter Boström · 9 years ago
  30. 13c433c Add delay metric (includes network delay (rtt/2) + jitter delay + decode time + render delay): by asapersson · 9 years ago
  31. 7bd242e Enabling screensharing tests for Android by ivica · 9 years ago
  32. 9359b5b Disabling AudioDeviceTest.StartStopPlayout on Android. by Henrik Kjellander · 9 years ago
  33. d89f82a Roll chromium_revision c511263..4bf3678 (352322:352512) by kjellander · 9 years ago
  34. 09f1350 Add option to reset Android video renderer first frame flag. by Alex Glaznev · 9 years ago
  35. 6caafbe Convert uint16_t to int for WebRTC cipher/crypto suite. by Guo-wei Shieh · 9 years ago
  36. 1b33da1 SurfaceTextureHelper fixes by perkj · 9 years ago
  37. 4185032 Add ThreadChecker class to ThreadUtils by perkj · 9 years ago
  38. d2838a7 Roll chromium_revision 07b4a8e..c511263 (352281:352322) by Henrik Kjellander · 9 years ago
  39. e0bce24 VideoCapturerAndroid: Add custom nativeCreateVideoCapturer() by perkj · 9 years ago
  40. 723dff1 Poll stats more often to get more stable stats in ramp-up tests. by Stefan Holmer · 9 years ago
  41. 4cd053f Only catch UnsatisfiedLinkError in Logging.java. by jiayl · 9 years ago
  42. f3a7c9d In rampup tests, set start time when starting poller thread. by Erik Språng · 9 years ago
  43. 95cd8ea Enable HW NS for N6 to fix HW AEC issue by henrika · 9 years ago
  44. dec5ebf Move sent key frame stats to send_statistics_proxy class. by asapersson · 9 years ago
  45. 42b6c63 autoroller: Allow to specify Rietveld e-mail. by Henrik Kjellander · 9 years ago
  46. 990d57d Fix file order in base.gyp. by henrikg · 9 years ago
  47. ba0f0a5 Disable flaky WebRtcVideoChannel2BaseTest.* on DrMemory/memcheck. by Peter Boström · 9 years ago
  48. 4bd8d09 Roll chromium_revision ca4c339..07b4a8e (352257:352281) by kjellander · 9 years ago
  49. 96a70f0 Exclude WebRtcVideoChannel2BaseTest.AddRemoveCapturerMultipleSources on Dr Memory. by Henrik Kjellander · 9 years ago
  50. b5fd46e Exclude WebRtcVideoChannel2BaseTest.AddRemoveCapturerMultipleSources on Memcheck by Henrik Kjellander · 9 years ago
  51. 42b4faa Fix a build issue when use external OpenSSL. by Guo-wei Shieh · 9 years ago
  52. 6df1ef6 Roll chromium_revision 4ce3c08..ca4c339 (352000:352257) by Henrik Kjellander · 9 years ago
  53. bc0938e Android VideoRendererGui: Make deep copy of incoming texture frames by Magnus Jedvert · 9 years ago
  54. 44bf6f5 Android MediaCodecVideoDecoder: Split DecoderOutputBufferInfo into DecodedByteBuffer and DecodedTextureBuffer by magjed · 9 years ago
  55. 13b96ba Adding APM configuration in AEC dump. by Minyue · 9 years ago
  56. 371dc7e WebRtc Win Desktop capture: ignore Win8+ Modern Apps' windows. by gyzhou · 9 years ago
  57. 913e645 Loopback and audio only mode. by haysc · 9 years ago
  58. f9c23ca Exclude WebRtcVideoChannel2BaseTest.GetStats on linux memcheck by Marco · 9 years ago
  59. 9dff0ba Fix MSVS project files generation. by henrikg · 9 years ago
  60. 067fb65 Roll chromium_revision 7fddcec..4ce3c08 (351973:352000) by kjellander · 9 years ago
  61. a050e98 Avoid race in RampUpTest by sprang · 9 years ago
  62. 7e31937 Android MediaCodecVideoDecoder: Cleanup to prepare for texture liftime management by Magnus Jedvert · 9 years ago
  63. 6781ea4 jni/native_handle_impl.h: Move implementation into .cc file by Magnus Jedvert · 9 years ago
  64. 417fec2 autoroller: Add CQ_EXTRA_TRYBOTS, CQ feature and --skip-cq flag. by Henrik Kjellander · 9 years ago
  65. 401025d Roll chromium_revision 354cc7d..7fddcec (351828:351973) by Henrik Kjellander · 9 years ago
  66. 1d8a506 Add a PacketOptions struct to webrtc::Transport. by stefan · 9 years ago
  67. da903ea Unify newapi::RtcpMode and RTCPMethod. by pbos · 9 years ago
  68. c8ba105 Roll chromium_revision 681f0cd..354cc7d (351698:351828) by kjellander · 9 years ago
  69. a9c584d autoroller: Always roll and improve description by Henrik Kjellander · 9 years ago
  70. 6c2ba7d autoroller: Add TBR= field and always update the checkout by Henrik Kjellander · 9 years ago
  71. 18b042f autoroller: Use HEAD instead of LKGR. by Henrik Kjellander · 9 years ago
  72. 5aaa9b4 Removed unused API functions in AudioProcessing and AudioProcessingModule by peah · 9 years ago
  73. 5629a1d Fix flaky test TestSrtpError, introduced in https://codereview.webrtc.org/1362913004. by solenberg · 9 years ago
  74. cf18b34 Align new VoE API with design. by solenberg · 9 years ago
  75. 8c471e7 Objective-C++ style guide changes for iOS ADM by henrika · 9 years ago
  76. fb30c1b Update VP8 settings to avoid spending bitrate on static areas. PERF NOTE by sprang · 9 years ago
  77. 5b14b42 Remove unused SignalMediaError and infrastructure. by solenberg · 9 years ago
  78. 49f9cdb Fix bug where rtcp::TransportFeedback may generate incorrect messages. by sprang · 9 years ago
  79. b09b660 Remove cricket::VideoFrame::Set/GetElapsedTime() by magjed · 9 years ago
  80. dfc8f4f Change 'mute' parameter of MediaChannel::SetAudioSend()/SetVideoSend() to 'enable'. by solenberg · 9 years ago
  81. edba998 Roll chromium_revision 8cf53d6..681f0cd (351112:351698) by kjellander · 9 years ago
  82. 0ecf1b2 Android focus problem on rear camera. by dchakarov.broadsoft · 9 years ago
  83. 98ab3a4 Don't link with audio codecs that we don't use by kwiberg · 9 years ago
  84. 456696a Reland Change WebRTC SslCipher to be exposed as number only by Guo-wei Shieh · 9 years ago
  85. 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
  86. 4fe3c9a Change WebRTC SslCipher to be exposed as number only. by guoweis · 9 years ago
  87. 2f8a4ca Add OWNERS for ObjC dirs. by tkchin · 9 years ago
  88. d0b3143 Do not time out a port if its role switched from controlled to controlling. Also fix some comments. by honghaiz · 9 years ago
  89. 898d21c WebRTC might leak srflx ip address when multiple_routes disabled and IceTransportType is relay. by Guo-wei Shieh · 9 years ago
  90. c4d3a5d Thinning out the Transport class. by Taylor Brandstetter · 9 years ago
  91. 2b342bf Delete a connection only if it has timed out on writing and not receiving for 10 seconds. by Honghai Zhang · 9 years ago
  92. 27551c9 Android RendererCommon: Refactor getSamplingMatrix() by Magnus Jedvert · 9 years ago
  93. 4a8e9c5 Remove overrides folder. by henrikg · 9 years ago
  94. bbda54e Android MediaDecoder: Use frame pool to avoid allocations for non-surface decoding by Magnus Jedvert · 9 years ago
  95. ee2bf41 Update build files to use webrtc_overrides in Chromium instead of overrides. by henrikg · 9 years ago
  96. 6ba8e4a ACM: Remove a few local enums that were no longer used by Henrik Lundin · 9 years ago
  97. d094c04 Remove AgcManager. by Alejandro Luebs · 9 years ago
  98. a67696b Reland of Adding PeerConnectionInterface::SetConfiguration method. (patchset #1 id:1 of https://codereview.webrtc.org/1361263002/ ) by deadbeef · 9 years ago
  99. 38778b0 Add unit test for nack bandwidth constraint. by sprang · 9 years ago
  100. 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