Gitiles
Code Review Sign In
gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / d8b9cd100e889aa8b4a7d8b391ebd39a1298913e
  1. d8b9cd1 Change how background noise mode in NetEq is set by henrik.lundin@webrtc.org · 10 years ago
  2. ac772a4 RTP video playback tool using Call APIs. by pbos@webrtc.org · 10 years ago
  3. 6c3f505 Fix crashing fake network pipe tests. by stefan@webrtc.org · 10 years ago
  4. b9ca3e2 Fixing two bugs in voe_cmd_test. by minyue@webrtc.org · 10 years ago
  5. 617e272 Add end-to-end H.264 packetization test. by stefan@webrtc.org · 10 years ago
  6. dcc85c0 Change the way we reference enumerators in H.264 packetization code to be standard C++ compliant. by stefan@webrtc.org · 10 years ago
  7. 284ac14 initialize packet len in NETEQTEST_DummyRTPpacket.cc and NETEQTEST_RTPpacket.cc to fix build error on vs2013 by fbarchard@google.com · 10 years ago
  8. 6c0337d Fix some code styles. by pbos@webrtc.org · 10 years ago
  9. b0512f9 Fix implicite cast from signed int to unsigned int in unittest.cc by fbarchard@google.com · 10 years ago
  10. a288b8c Fix potential crash when depacketizing VP8. by stefan@webrtc.org · 10 years ago
  11. c4a5794 Unbreaks linux.cc in Chromium. by henrike@webrtc.org · 10 years ago
  12. 96c18e0 This is a setup to solve https://code.google.com/p/webrtc/issues/detail?id=1906 by minyue@webrtc.org · 10 years ago
  13. 7425710 Fix for retransmission. Base layer packets were not retransmitted. by asapersson@webrtc.org · 10 years ago
  14. c141982 Fix mistake in rtp/rtcp/BUILD.gn introduced with r6804. by stefan@webrtc.org · 10 years ago
  15. e75b348 Add H.264 packetization. by stefan@webrtc.org · 10 years ago
  16. 4068313 Add simulation of network effects to video_loopback tool. by stefan@webrtc.org · 10 years ago
  17. 216021d Use C functions in aec for MIPS by andrew@webrtc.org · 10 years ago
  18. 92a0d54 Integrate rtcp packet class to rtcp receiver tests. by asapersson@webrtc.org · 10 years ago
  19. cffc685 merge_libs.py: fixes Windows breakage: there should be no space after "lib /OUT:". by henrike@webrtc.org · 10 years ago
  20. 3ff4222 webrtc/base: FileModifyTime -> OlderThan as that's what it was ever used as. Needed for cl/70828325. by henrike@webrtc.org · 10 years ago
  21. b957cd8 Fix compilation on windows with clang, indentation cleanups by sergeyu@chromium.org · 10 years ago
  22. 5322628 Fixes "argument list too long" problem on Linux by using the "find" command instead of re-implementing one in python. by henrike@webrtc.org · 10 years ago
  23. ca4bc68 Remove timestamp retreival warning/error. by turaj@webrtc.org · 10 years ago
  24. 648d555 Revert "Fix compilation on windows with clang, indentation cleanups" by sergeyu@chromium.org · 10 years ago
  25. 2189f34 Fix compilation on windows with clang, indentation cleanups by sergeyu@chromium.org · 10 years ago
  26. 4a1b3e3 Make sure padding is sent on the first sending RTP module. by mflodman@webrtc.org · 10 years ago
  27. a4dc1ae Fix flaky ramp-up test. by stefan@webrtc.org · 10 years ago
  28. e291f57 The lastest commit on this file was in by minyue@webrtc.org · 10 years ago
  29. 00f7b82 Remove no longer used SkipEncodingUnusedStreams. by andresp@webrtc.org · 10 years ago
  30. 050346b Remove remains of WEBRTC_NO_STL. by andresp@webrtc.org · 10 years ago
  31. e4834e0 MIPS optimizations for ISAC (patch #2) by andrew@webrtc.org · 10 years ago
  32. f0a119f Check before send/receive rtp header extensions. by pbos@webrtc.org · 10 years ago
  33. 254879d This is to re-open an earlier CL by minyue@webrtc.org · 10 years ago
  34. 08e28eb Runtime guard for iOS7 property. by tkchin@webrtc.org · 10 years ago
  35. da5452b Fix crash in AudioDeviceUtilityIOS::~AudioDeviceUtilityIOS. by tkchin@webrtc.org · 10 years ago
  36. d89fa97 This is related to an earlier CL of enabling Opus 48 kHz. by minyue@webrtc.org · 10 years ago
  37. c617654 Sleep in ThreadTest thread functions. by pbos@webrtc.org · 10 years ago
  38. 84649c0 AudioBuffer: Optimize const accesses to arrays that autoconvert int16<->float by kwiberg@webrtc.org · 10 years ago
  39. 8bd216f Reduce runtime of RingBufferTest by a factor of 100. by andrew@webrtc.org · 10 years ago
  40. 9ae7d44 Use _numMixedParticipants instead of audioFrameList->size() to determine if there're more than one participants. by wu@webrtc.org · 10 years ago
  41. f147639 Fix issue where padding is sent before media with undefined timestamps if not abs-send-time is enabled. by stefan@webrtc.org · 10 years ago
  42. c911a13 Remove unused ExperimentalNS API in AudioProcessing by aluebs@webrtc.org · 10 years ago
  43. 4309681 AudioBuffer: Eliminate the SplitChannelBuffer class by kwiberg@webrtc.org · 10 years ago
  44. eb15100 Simplify AudioBuffer::mixed_low_pass_data API by aluebs@webrtc.org · 10 years ago
  45. 7036325 AudioBuffer: Let ChannelBuffer handle bounds checking of channel parameter by kwiberg@webrtc.org · 10 years ago
  46. bde2bcb Add unit test for MediaFile WAV file writing by kwiberg@webrtc.org · 10 years ago
  47. fbdd355 Fixes up rtc so that it compiles on iOS 8 SDK. by tkchin@webrtc.org · 10 years ago
  48. bc9711f r6709 lacks a change in BUILD.gn by minyue@webrtc.org · 10 years ago
  49. 31b38da Raw packet loss rate reported by RTP_RTCP module may vary too drastically over time. This CL is to add a filter to the value in VoE before lending it to audio coding module. by minyue@webrtc.org · 10 years ago
  50. bb77419 Compile-time guard for iOS7 specific property. by tkchin@webrtc.org · 10 years ago
  51. 6111d79 Print an info log instead of return an error if an external encoder is de-registered, but no corresponding internal encoder can be registered automatically. by stefan@webrtc.org · 10 years ago
  52. 7a4d45f Remove old padding path in RTPSender. by pbos@webrtc.org · 10 years ago
  53. 9f7856a int16<->float conversions: Use size_t for array length argument, not int by kwiberg@webrtc.org · 10 years ago
  54. b5966bd Define convenient FATAL_ERROR() and FATAL_ERROR_IF() macros by kwiberg@webrtc.org · 10 years ago
  55. fac3d8a nrsh1 is written before tmp321 is read, so needs to be earlyclobber by kwiberg@webrtc.org · 10 years ago
  56. df6904d Fix an invalid memory access due to typo in win/cursor.cc. by jiayl@webrtc.org · 10 years ago
  57. a098325 After an audio interruption the audio unit no longer invokes its render callback, which results in a loss of audio. Restarting the audio unit post interruption fixes the issue. by tkchin@webrtc.org · 10 years ago
  58. acb7c52 Remove Thread::RunningForChannelManager(). by tkchin@webrtc.org · 10 years ago
  59. 8c82443 Improvements to the pacer where it lost some budget due to truncation errors. by stefan@webrtc.org · 10 years ago
  60. 2a34c4c Fix breakage introduced by r6691. by pbos@webrtc.org · 10 years ago
  61. 442dbd4 Make RTCP sender report send media bytes. by pbos@webrtc.org · 10 years ago
  62. ec8e147 Eliminate unnecessary #include by kwiberg@webrtc.org · 10 years ago
  63. 8314016 rtc::Fatal output: Print space between # and message by kwiberg@webrtc.org · 10 years ago
  64. 15097fc Remove the VPM denoiser. by pbos@webrtc.org · 10 years ago
  65. d3de227 Rebase webrtc/base with r6682 version of talk/base: by henrike@webrtc.org · 10 years ago
  66. 82383d9 Fix deadlock in Android stopCapture() call. by glaznev@webrtc.org · 10 years ago
  67. ccf0fef GN: Fix include paths for WebRTC in Chromium build. by kjellander@webrtc.org · 10 years ago
  68. 5faa6d1 Fix bugs introduced by https://code.google.com/p/webrtc/source/detail?r=6667 . by tommi@webrtc.org · 10 years ago
  69. 502a271 Remove always-true expression. by tommi@webrtc.org · 10 years ago
  70. 9fbd3ec Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC. --- by tommi@webrtc.org · 10 years ago
  71. 0cb22cf Thread annotate RTCPSender. by pbos@webrtc.org · 10 years ago
  72. 55b0f2e Move pacer to fully use webrtc::Clock instead of webrtc::TickTime. by stefan@webrtc.org · 10 years ago
  73. c928d36 Cast payload types to int for logging. by pbos@webrtc.org · 10 years ago
  74. d20c29a Document that channels are stored contiguously in AudioBuffer by aluebs@webrtc.org · 10 years ago
  75. a301f1a Remove unnecessary build message. by tommi@webrtc.org · 10 years ago
  76. 09da1a7 Remove the send-side cname getter APIs from voice and video engine. by stefan@webrtc.org · 10 years ago
  77. c4b828d Rebase webrtc/base with r6655 version of talk/base: by henrike@webrtc.org · 10 years ago
  78. 9aa3497 Count total bytes sent in RTPSender::Bytes(). by pbos@webrtc.org · 10 years ago
  79. 579d63c Fix data race in VCMTiming::ResetDecodeTime. by pbos@webrtc.org · 10 years ago
  80. 51c9def Skip encoding in fake VP8 encoder. by pbos@webrtc.org · 10 years ago
  81. 408fa71 Remove more unused tsan suppressions and fix call test passing the same decoder to multiple received streams. by andresp@webrtc.org · 10 years ago
  82. 54f889f Support VP8 encoder settings in VideoSendStream. by pbos@webrtc.org · 10 years ago
  83. 8c95e83 Refactor registerable callbacks for FrameCountObserver from rtp_rtcp module into vie_channel. by andresp@webrtc.org · 10 years ago
  84. f8bddb4 audio_processing: Updates aec_core_sse2.c with changes made to aec_common.h by bjornv@webrtc.org · 10 years ago
  85. 31ab61c Neon version of SubbandCoherence() by bjornv@webrtc.org · 10 years ago
  86. 71ba40d Neon version of rftbsub_128() by bjornv@webrtc.org · 10 years ago
  87. 91b4389 Revert "Remove remains of WEBRTC_NO_STL." (rev 6641). by andresp@webrtc.org · 10 years ago
  88. 61f437e Remove remains of WEBRTC_NO_STL. by andresp@webrtc.org · 10 years ago
  89. 75f7656 Create FullScreenChromeWindowDetector in DesktopConfigurationOptions::CreateDefault. by jiayl@webrtc.org · 10 years ago
  90. fedbe8b Thread annotations for vie_encoder.cc/.h by stefan@webrtc.org · 10 years ago
  91. 8e97a43 Remove unnecessary race suppressions copied from chromium. by andresp@webrtc.org · 10 years ago
  92. 79b66f4 Add full stack test cases with a fake network pipe. by stefan@webrtc.org · 10 years ago
  93. 138adbb delay_estimator: Increases test coverage and makes input spectrum const by bjornv@webrtc.org · 10 years ago
  94. 678f190 Implement a work around for Chrome full-screen tab switch on Mac. by jiayl@webrtc.org · 10 years ago
  95. 62ef953 Neon version of rftfsub_128() by bjornv@webrtc.org · 10 years ago
  96. f8ec08e Refactor registerable callbacks for VideoBitrateObserver from rtp_rtcp module into vie_channel. by andresp@webrtc.org · 10 years ago
  97. 49927ed Refactor ramp-up tests to have separate help files for the test classes, to make things more reusable. by stefan@webrtc.org · 10 years ago
  98. f247ac6 Change Timing::WallTimeNow to be static. by tommi@webrtc.org · 10 years ago
  99. 6aae61c Some refactoring inside rtp_rtcp/. by pbos@webrtc.org · 10 years ago
  100. 4109f65 Fixing compile error. by phoglund@webrtc.org · 10 years ago