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gerrit-public.fairphone.software / platform / external / webrtc / 44eb87e6dd3ddf9711d480521e006ce9084fdbd6
  1. 44eb87e Tool to establish a loopback call via apprtc turn server. by andresp@webrtc.org · 11 years ago
  2. 26caf0e Suppresses/disables tsan/memcheck issues due to sync of 63111035. by henrike@webrtc.org · 11 years ago
  3. 18e5911 (Auto)update libjingle 63089643-> 63111035 by henrike@webrtc.org · 11 years ago
  4. cf6f46d References to includes in third_party should be relative, not absolute. by sprang@webrtc.org · 11 years ago
  5. 4375e1a Add support for YUV4MPEG file reading to tools files. (Minor fix). by mcasas@webrtc.org · 11 years ago
  6. 6e2d012 Add support for YUV4MPEG file reading to tools files. by mcasas@webrtc.org · 11 years ago
  7. 24779fe Fix a bug where network freeze during CNG causes delay by henrik.lundin@webrtc.org · 11 years ago
  8. 367000f Remove legacy weirdness in Merge::Downsample by henrik.lundin@webrtc.org · 11 years ago
  9. f45a550 (Auto)update libjingle 63019975-> 63089643 by henrike@webrtc.org · 11 years ago
  10. 54464e6 Stopping network threads before tearing down test by henrik.lundin@webrtc.org · 11 years ago
  11. 5a320fb Race condition in RTPSender by sprang@webrtc.org · 11 years ago
  12. 4168901 Add max delay to trace based filters and enhances drop tail queues with delay statistics. by stefan@webrtc.org · 11 years ago
  13. b10363f Re-landing "Routing SuspendChange to VideoSendStream::Stats" by henrik.lundin@webrtc.org · 11 years ago
  14. 3349ae0 Implement minimum transmit bitrate. by pbos@webrtc.org · 11 years ago
  15. 6ea4f63 Enable all RampUpTest.UpDownUp* tests by henrik.lundin@webrtc.org · 11 years ago
  16. b5f3029 Replace labs with std::abs. by pbos@webrtc.org · 11 years ago
  17. 827faae Fixing incorrect memset. by mallinath@webrtc.org · 11 years ago
  18. dd5d804 Disable all protobuf dependent targets when enable_protobuf=0. by andrew@webrtc.org · 11 years ago
  19. c7bec84 (Auto)update libjingle 62948689-> 63019975 by henrike@webrtc.org · 11 years ago
  20. 9269ba1 (Git)ignore all of /net. Works around issue: gclient sync, git clean -df, gclient runhooks -> failure (regression in r4466). by henrike@webrtc.org · 11 years ago
  21. c2313fb Enable VS2013 for Windows compilation by default. by kjellander@webrtc.org · 11 years ago
  22. 95153cc Remove platform-specific code from new-API tests. by pbos@webrtc.org · 11 years ago
  23. ca8cb95 Implement a test for an old corner-case in NetEq by henrik.lundin@webrtc.org · 11 years ago
  24. 04ea232 Developing NetEqImpl unit tests by henrik.lundin@webrtc.org · 11 years ago
  25. 10bd88e (Auto)update libjingle 62871616-> 62948689 by henrike@webrtc.org · 11 years ago
  26. 21df847 Disable TestOpusNewACM on Android. by andrew@webrtc.org · 11 years ago
  27. be39470 Revert "Routing SuspendChange to VideoSendStream::Stats" by henrik.lundin@webrtc.org · 11 years ago
  28. 12acd6e Reorder includes in audio_processing_impl_unittest. by andrew@webrtc.org · 11 years ago
  29. cdefc91 Voice Engine GetRemoteCSRCs should return the CSRCs from rtp_receiver_ instead of _rtpRtcpModule now. by braveyao@webrtc.org · 11 years ago
  30. 1598b80 Routing SuspendChange to VideoSendStream::Stats by henrik.lundin@webrtc.org · 11 years ago
  31. c3d13d3 Classes and tests for audio an classifier. The class can be used to classify whether a frame of audio contains speech or music. The classifier uses the music/speech classifier in Opus. by jan.skoglund@webrtc.org · 11 years ago
  32. a8b9737 Add tests and modify tools for new float deinterleaved interface. by andrew@webrtc.org · 11 years ago
  33. 3046b84 Adding new data files for audio classifier unit testing on Android try bots by jan.skoglund@webrtc.org · 11 years ago
  34. d3d6bce (Auto)update libjingle 62865357-> 62871616 by henrike@webrtc.org · 11 years ago
  35. d32797f Add a float interface to PushSincResampler. by andrew@webrtc.org · 11 years ago
  36. bc206ea iOS video_render: omit no-op setNeedsDisplay by fischman@webrtc.org · 11 years ago
  37. f792d17 AppRTCDemo(iOS): video support; part 1 of 2: webrtc/. by fischman@webrtc.org · 11 years ago
  38. 0537634 (Auto)update libjingle 62713454-> 62865357 by henrike@webrtc.org · 11 years ago
  39. 4a47be0 Disable CallTest.ReceivesAndRetransmitsNack for TSan by kjellander@webrtc.org · 11 years ago
  40. 36b6221 Adding a link to issue by henrik.lundin@webrtc.org · 11 years ago
  41. 6b0cbcb Roll chromium_revision 249215:255773 by kjellander@webrtc.org · 11 years ago
  42. 9b5f4d8 Fix build breakage introduce with r5665. by stefan@webrtc.org · 11 years ago
  43. f9e7c9d Add option to bwe_rtp_to_text to output arrival times only in nanoseconds. by stefan@webrtc.org · 11 years ago
  44. a01daf0 RTCPeerConnectionTest(objc): deflake by ignoring ICECompleted. by fischman@webrtc.org · 11 years ago
  45. 13320ea PeerConnectionTest(objc): expect ICE Completed state post 61460797-p10 by fischman@webrtc.org · 11 years ago
  46. 7811469 Roll libvpx 251850:254609 by marpan@webrtc.org · 11 years ago
  47. 11aab0e Populate VoiceReceiverInfo::delay_estimate_ms, jitter_buffer_ms, and jitter_buffer_preferred_ms to getStats. by jiayl@webrtc.org · 11 years ago
  48. 64e0405 Avoid crash in ViEEncoder::DeRegisterExternalEncoder(). by fischman@webrtc.org · 11 years ago
  49. cc08e3f Moves WEBRTC_POSIX define from header file to gyp-settings. by henrike@webrtc.org · 11 years ago
  50. 3ecc162 Remove std:: prefixes from C functions in webrtc/. by pbos@webrtc.org · 11 years ago
  51. 371243d Remove std:: prefixes from C functions in talk/. by pbos@webrtc.org · 11 years ago
  52. 46509c8 adding FEC support to WebRTC Opus wrapper and tests. by minyue@webrtc.org · 11 years ago
  53. 0454688 This CL is to add Opus complexity knob and to test it. by minyue@webrtc.org · 11 years ago
  54. ebdb0e3 Help to land 7969005 on behalf of solenberg. The review and try is done in 7969005. by wu@webrtc.org · 11 years ago
  55. 79047f9 (Auto)update libjingle 62691533-> 62713454 by henrike@webrtc.org · 11 years ago
  56. 2d213e4 (Auto)update libjingle 62550414-> 62691533 by henrike@webrtc.org · 11 years ago
  57. f714e7f Remove abs() use in PseudoTcp::process. by pbos@webrtc.org · 11 years ago
  58. 4584697 Fixes a bug in the simulation framework where the time offset is accumulating as the packet trace is repeated, causing increasingly large gaps with no packets being transmitted. by stefan@webrtc.org · 11 years ago
  59. ed865b5 NetEq4: Changing the behavior of playout_timestamp_ update by henrik.lundin@webrtc.org · 11 years ago
  60. 60ad5fd Potential deadlock in VideoSendStreamTest::ProducesStats by sprang@webrtc.org · 11 years ago
  61. 998cb8f Use DISABLE_ instead of commenting out tests by henrik.lundin@webrtc.org · 11 years ago
  62. 845862f Adding a new ramp-up-down-up test by henrik.lundin@webrtc.org · 11 years ago
  63. a0d11da Remove upper check for number of cores in VCM, I didn't find any good reasons for checking this. by mflodman@webrtc.org · 11 years ago
  64. cf85f1c Reorganize libjingle path variables. by kjellander@webrtc.org · 11 years ago
  65. 9f4d212 adding sha1 files for audio classifier test by jan.skoglund@webrtc.org · 11 years ago
  66. 3e0b60f Switch to correct interpretation of int and float input data in audio_processing_unittest by bjornv@webrtc.org · 11 years ago
  67. 17e4064 Add a deinterleaved float interface to AudioProcessing. by andrew@webrtc.org · 11 years ago
  68. b90991d Update libjingle 62472237->62550414 by henrike@webrtc.org · 11 years ago
  69. 7bd4a27 VideoCaptureAndroid: don't deliver frames after stopCapture(). by fischman@webrtc.org · 11 years ago
  70. be50ab6 Including algorithm header to avoid VS2013 breakage by henrik.lundin@webrtc.org · 11 years ago
  71. 52e898d Add .bin and .rx files to svn:ignore in resources by kjellander@webrtc.org · 11 years ago
  72. 24dae94 Add pthatcher@webrtc.org to talk/OWNERS. by pbos@webrtc.org · 11 years ago
  73. a25a92e Add third_party dependencies to svn:ignore by kjellander@webrtc.org · 11 years ago
  74. db41b4d Remove the deprecated GetStats method from PeerConnectionInterface. by jiayl@webrtc.org · 11 years ago
  75. 80bbf4c Enable test SSLStreamAdapterTestDTLS.TestDTLSConnectWithSmallMtu since it does not fail anymore. by jiayl@webrtc.org · 11 years ago
  76. 40b3b68 Update libjingle 62364298->62472237 by henrike@webrtc.org · 11 years ago
  77. 1bbfb57 Rollback of r5629 "(Auto)update libjingle 62364298-> 62368661". by henrike@webrtc.org · 11 years ago
  78. 0117d1c Fix compilation errors under clang 3.5. by pbos@webrtc.org · 11 years ago
  79. 31413dc (Auto)update libjingle 62364298-> 62368661 by henrike@webrtc.org · 11 years ago
  80. 10adbef Exclude /out* instead of just /out from pylint checks. by fischman@webrtc.org · 11 years ago
  81. 2bd5944 Re-enable libjingle_peerconnection_java_unittest since bug 2952 is fixed. by fischman@webrtc.org · 11 years ago
  82. d3dc424 Remove posting of ICE messages from WebRTCSession in PeerConnection to signaling thread. by mallinath@webrtc.org · 11 years ago
  83. bcfc167 AppRTCDemo(android): don't send local SDP until it's set. by fischman@webrtc.org · 11 years ago
  84. b898ce9 Revert of r5622 "disable unit tests" as it should be fixed in r5623. by henrike@webrtc.org · 11 years ago
  85. b8395eb (Auto)update libjingle 62293974-> 62364298 by henrike@webrtc.org · 11 years ago
  86. eec3843 TSAN only disable of two of libjingle's tests for atomic ops as they are failing for TSAN-bot. by henrike@webrtc.org · 11 years ago
  87. 9fd8d87 Adds APIs for reporting pacer queuing delay. by jiayl@webrtc.org · 11 years ago
  88. 56e4a05 Remove ProcessingComponent's dependence on AudioProcessingImpl. by andrew@webrtc.org · 11 years ago
  89. 806768a (Auto)update libjingle 62281784-> 62293974 by henrike@webrtc.org · 11 years ago
  90. 704bf9e (Auto)update libjingle 62063505-> 62278774 by henrike@webrtc.org · 11 years ago
  91. f0fc72f Call PrintWindow for the first time of capturing to capture the window frames correctly. by jiayl@webrtc.org · 11 years ago
  92. 00073aa Clean up CPU detection defines in SincResampler a little. by andrew@webrtc.org · 11 years ago
  93. 0231e80 Invalidate the whole screen when the frame size is changed. by jiayl@webrtc.org · 11 years ago
  94. 2038920 Use scoped_ptr<T[]> in SincResampler to avoid .get()[] weirdness. by andrew@webrtc.org · 11 years ago
  95. c0e9aeb Add SetConfig method to FakeNetworkPipe and to DirectTransport by henrik.lundin@webrtc.org · 11 years ago
  96. eaadeca iOS, AppRTCDemo: Fixes exception due to JSON for ice using "urls" instead of "url", which is introduced by r5599. by braveyao@webrtc.org · 11 years ago
  97. 90173e1 Roll libvpx 248011:251850 by marpan@webrtc.org · 11 years ago
  98. bc1d224 Add experimental noise suppression flag to audioproc test by aluebs@webrtc.org · 11 years ago
  99. 050892a Missing include in experiments.h by sprang@webrtc.org · 11 years ago
  100. 7f52a6e Split the implementation of VP8Encoder|Decoder::Create into a seperated file by wu@webrtc.org · 11 years ago