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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 53c1d3c561c92948cada27409bab1c5b9747ae8a
  1. 36208c2 Properly account for retransmitted packets when not using the pacer. by stefan@webrtc.org · 10 years ago
  2. fbd6f47 Fixes RTX related bugs. by stefan@webrtc.org · 10 years ago
  3. 8abfc7f Disabling SendsSetSimulcastSsrcs. by pbos@webrtc.org · 10 years ago
  4. 14e5fbb Revert "Changing the buffer size (in packets) to 1.5 seconds @ 30 ms packets" by henrik.lundin@webrtc.org · 10 years ago
  5. 256de0f Changing the buffer size (in packets) to 1.5 seconds @ 30 ms packets by henrik.lundin@webrtc.org · 10 years ago
  6. 45ae9e4 Disable flaky CanSwitchToUseAllSsrcs. by pbos@webrtc.org · 10 years ago
  7. 1d61e3a Simplify pacer interface. by pbos@webrtc.org · 10 years ago
  8. e2a7a77 Remove internal codecs from VideoSendStream. by pbos@webrtc.org · 10 years ago
  9. 76cd2f7 Fix a deadlock in ViEEncoder::DeliverFrame. by wuchengli@chromium.org · 10 years ago
  10. dd6cf04 Adds a method to WindowCapturer to bring a window to the front. by jiayl@webrtc.org · 10 years ago
  11. c4c3021 Adding thread annotations to NetEq4 by henrik.lundin@webrtc.org · 10 years ago
  12. fee2074 Add #include <cstdlib> for std::abs. by pbos@webrtc.org · 10 years ago
  13. df08c5d Resolves TSan v2 warnings in voe_auto_test. by henrika@webrtc.org · 10 years ago
  14. 0c0c604 Re-comitting r5711: "Fixing a flaky test in video_engine_tests" by henrik.lundin@webrtc.org · 10 years ago
  15. a50f107 Revert 5711 "Fixing a flaky test in video_engine_tests" by turaj@webrtc.org · 10 years ago
  16. 323af23 Fixing a flaky test in video_engine_tests by henrik.lundin@webrtc.org · 10 years ago
  17. 608b331 Small refactor on send_side_bandwidth_estimation. by andresp@webrtc.org · 10 years ago
  18. c6f6696 Refactor rampup tests: by andresp@webrtc.org · 10 years ago
  19. d073362 Tool to establish a loopback call via apprtc turn server. by andresp@webrtc.org · 10 years ago
  20. dc7e89c References to includes in third_party should be relative, not absolute. by sprang@webrtc.org · 10 years ago
  21. 8b04780 Add support for YUV4MPEG file reading to tools files. (Minor fix). by mcasas@webrtc.org · 10 years ago
  22. 280ab2a Add support for YUV4MPEG file reading to tools files. by mcasas@webrtc.org · 10 years ago
  23. fcc76aa Fix a bug where network freeze during CNG causes delay by henrik.lundin@webrtc.org · 10 years ago
  24. 56bf2cd Remove legacy weirdness in Merge::Downsample by henrik.lundin@webrtc.org · 10 years ago
  25. 96616cb Stopping network threads before tearing down test by henrik.lundin@webrtc.org · 11 years ago
  26. 66cff19 Race condition in RTPSender by sprang@webrtc.org · 11 years ago
  27. b67e9b7 Add max delay to trace based filters and enhances drop tail queues with delay statistics. by stefan@webrtc.org · 11 years ago
  28. 9376c69 Re-landing "Routing SuspendChange to VideoSendStream::Stats" by henrik.lundin@webrtc.org · 11 years ago
  29. 3f83f9c Implement minimum transmit bitrate. by pbos@webrtc.org · 11 years ago
  30. 691c5b2 Enable all RampUpTest.UpDownUp* tests by henrik.lundin@webrtc.org · 11 years ago
  31. 2a25b6c Replace labs with std::abs. by pbos@webrtc.org · 11 years ago
  32. 2b43f8b Disable all protobuf dependent targets when enable_protobuf=0. by andrew@webrtc.org · 11 years ago
  33. 18e5838 Enable VS2013 for Windows compilation by default. by kjellander@webrtc.org · 11 years ago
  34. ee86b90 Remove platform-specific code from new-API tests. by pbos@webrtc.org · 11 years ago
  35. 758d0b8 Implement a test for an old corner-case in NetEq by henrik.lundin@webrtc.org · 11 years ago
  36. 174152b Developing NetEqImpl unit tests by henrik.lundin@webrtc.org · 11 years ago
  37. e462911 Disable TestOpusNewACM on Android. by andrew@webrtc.org · 11 years ago
  38. 9759355 Revert "Routing SuspendChange to VideoSendStream::Stats" by henrik.lundin@webrtc.org · 11 years ago
  39. dce00af Reorder includes in audio_processing_impl_unittest. by andrew@webrtc.org · 11 years ago
  40. 3f7753c Voice Engine GetRemoteCSRCs should return the CSRCs from rtp_receiver_ instead of _rtpRtcpModule now. by braveyao@webrtc.org · 11 years ago
  41. 2ae3c62 Routing SuspendChange to VideoSendStream::Stats by henrik.lundin@webrtc.org · 11 years ago
  42. f83bc05 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
  43. b5b49c0 Add tests and modify tools for new float deinterleaved interface. by andrew@webrtc.org · 11 years ago
  44. 6a5ccc5 Add a float interface to PushSincResampler. by andrew@webrtc.org · 11 years ago
  45. 23531a3 iOS video_render: omit no-op setNeedsDisplay by fischman@webrtc.org · 11 years ago
  46. b0de73f AppRTCDemo(iOS): video support; part 1 of 2: webrtc/. by fischman@webrtc.org · 11 years ago
  47. 5b67882 Adding a link to issue by henrik.lundin@webrtc.org · 11 years ago
  48. da666ce Roll chromium_revision 249215:255773 by kjellander@webrtc.org · 11 years ago
  49. 4e9a04b Fix build breakage introduce with r5665. by stefan@webrtc.org · 11 years ago
  50. dc07590 Add option to bwe_rtp_to_text to output arrival times only in nanoseconds. by stefan@webrtc.org · 11 years ago
  51. bbbe9b8 Avoid crash in ViEEncoder::DeRegisterExternalEncoder(). by fischman@webrtc.org · 11 years ago
  52. 9201f0d Moves WEBRTC_POSIX define from header file to gyp-settings. by henrike@webrtc.org · 11 years ago
  53. bcf0a10 Remove std:: prefixes from C functions in webrtc/. by pbos@webrtc.org · 11 years ago
  54. 52db76a adding FEC support to WebRTC Opus wrapper and tests. by minyue@webrtc.org · 11 years ago
  55. 1f2820b This CL is to add Opus complexity knob and to test it. by minyue@webrtc.org · 11 years ago
  56. 9a82322 Help to land 7969005 on behalf of solenberg. The review and try is done in 7969005. by wu@webrtc.org · 11 years ago
  57. 7a56079 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
  58. 70e2ce9 NetEq4: Changing the behavior of playout_timestamp_ update by henrik.lundin@webrtc.org · 11 years ago
  59. 379c349 Potential deadlock in VideoSendStreamTest::ProducesStats by sprang@webrtc.org · 11 years ago
  60. 0435a83 Use DISABLE_ instead of commenting out tests by henrik.lundin@webrtc.org · 11 years ago
  61. c766098 Adding a new ramp-up-down-up test by henrik.lundin@webrtc.org · 11 years ago
  62. bc55f91 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
  63. 80eeab5 Switch to correct interpretation of int and float input data in audio_processing_unittest by bjornv@webrtc.org · 11 years ago
  64. 5350e31 Add a deinterleaved float interface to AudioProcessing. by andrew@webrtc.org · 11 years ago
  65. 78f1f4a VideoCaptureAndroid: don't deliver frames after stopCapture(). by fischman@webrtc.org · 11 years ago
  66. dbe9516 Including algorithm header to avoid VS2013 breakage by henrik.lundin@webrtc.org · 11 years ago
  67. 6a9c344 Fix compilation errors under clang 3.5. by pbos@webrtc.org · 11 years ago
  68. f9747a8 Re-enable libjingle_peerconnection_java_unittest since bug 2952 is fixed. by fischman@webrtc.org · 11 years ago
  69. 55a2a27 Adds APIs for reporting pacer queuing delay. by jiayl@webrtc.org · 11 years ago
  70. a1a6001 Remove ProcessingComponent's dependence on AudioProcessingImpl. by andrew@webrtc.org · 11 years ago
  71. 84641bf Call PrintWindow for the first time of capturing to capture the window frames correctly. by jiayl@webrtc.org · 11 years ago
  72. e9d42e6 Clean up CPU detection defines in SincResampler a little. by andrew@webrtc.org · 11 years ago
  73. 935694d Invalidate the whole screen when the frame size is changed. by jiayl@webrtc.org · 11 years ago
  74. b7306ea Use scoped_ptr<T[]> in SincResampler to avoid .get()[] weirdness. by andrew@webrtc.org · 11 years ago
  75. 9900e37 Add SetConfig method to FakeNetworkPipe and to DirectTransport by henrik.lundin@webrtc.org · 11 years ago
  76. fa35a17 Add experimental noise suppression flag to audioproc test by aluebs@webrtc.org · 11 years ago
  77. 1f9bc4f Missing include in experiments.h by sprang@webrtc.org · 11 years ago
  78. dea824b Split the implementation of VP8Encoder|Decoder::Create into a seperated file by wu@webrtc.org · 11 years ago
  79. 9aef225 Fix to get total number of sent and received rtcp packets. by asapersson@webrtc.org · 11 years ago
  80. 7efba48 AviRecorder is missing a critical section. by braveyao@webrtc.org · 11 years ago
  81. b1a7102 Disable libjingle_peerconnection_java_unittest by kjellander@webrtc.org · 11 years ago
  82. 5cf6d75 Removed unused mock methods in audio_processing by bjornv@webrtc.org · 11 years ago
  83. 663ba07 Add RTCP packet class. Adds packet types: sr, rr, bye, fir. by asapersson@webrtc.org · 11 years ago
  84. af83c70 MIPS optimizations for AEC audio processing module by andrew@webrtc.org · 11 years ago
  85. fd1b455 Updated WebRTC version to 3.50 TBR= wu@webrtc.org by elham@webrtc.org · 11 years ago
  86. 1d2f5dc Add an AlignedFreeDeleter and remove scoped_ptr_malloc. by andrew@webrtc.org · 11 years ago
  87. 3528ad9 Minor improvement in RoundToInt16 implementation. by turaj@webrtc.org · 11 years ago
  88. 0b95677 Modified overuse detection thresholds. by asapersson@webrtc.org · 11 years ago
  89. 64007c6 Removing a variable that was never read by henrik.lundin@webrtc.org · 11 years ago
  90. 46b817d ifdef the alsa code based on macro USE_X11 by fbarchard@google.com · 11 years ago
  91. 6ad768a Fix the break caused by r5579. by turaj@webrtc.org · 11 years ago
  92. bc43ec2 Resolves memcheck issue in AudioCodingModuleTest. The issue is coditional jumnp based on uninitialized variable. by turaj@webrtc.org · 11 years ago
  93. ab23a2d Make WindowCapturerLinux handling window resize events. by jiayl@webrtc.org · 11 years ago
  94. 1af2c14 Added architecture for compiling under chrome NaCl. by andresp@webrtc.org · 11 years ago
  95. 891f05e This CL separate all ACM tests with new and old implementation of ACM and NetEq. The reason is to debug an issue with failure on Android try bots. We need to see if the error only occurs with the new ACM/NetEq, or if it is a flakiness that affects both. by tina.legrand@webrtc.org · 11 years ago
  96. 4a15560 Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR). by asapersson@webrtc.org · 11 years ago
  97. 4845ee0 Removes VoERTP_RTCP::InsertExtraRTPPacket. by henrika@webrtc.org · 11 years ago
  98. d4ce7ff Fix DesktopAndCursorComposer not to crash by sergeyu@chromium.org · 11 years ago
  99. ad065d0 Move the volume quantization workaround from VoE to AGC. by andrew@webrtc.org · 11 years ago
  100. f6c4fc3 Remove obsolete voe_unit_test. by solenberg@webrtc.org · 11 years ago