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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 40220194bb03637cbc5b21ef36b6abd4e20cbd75 / modules
  1. c141982 Fix mistake in rtp/rtcp/BUILD.gn introduced with r6804. by stefan@webrtc.org · 10 years ago
  2. e75b348 Add H.264 packetization. by stefan@webrtc.org · 10 years ago
  3. 216021d Use C functions in aec for MIPS by andrew@webrtc.org · 10 years ago
  4. fa50854 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 4a1b3e3a69d349b0d3e91f607f24e02d8b975688 by Android Chromium Automerger · 10 years ago
  5. 92a0d54 Integrate rtcp packet class to rtcp receiver tests. by asapersson@webrtc.org · 10 years ago
  6. 4a1b3e3 Make sure padding is sent on the first sending RTP module. by mflodman@webrtc.org · 10 years ago
  7. e291f57 The lastest commit on this file was in by minyue@webrtc.org · 10 years ago
  8. 050346b Remove remains of WEBRTC_NO_STL. by andresp@webrtc.org · 10 years ago
  9. e4834e0 MIPS optimizations for ISAC (patch #2) by andrew@webrtc.org · 10 years ago
  10. 254879d This is to re-open an earlier CL by minyue@webrtc.org · 10 years ago
  11. 08e28eb Runtime guard for iOS7 property. by tkchin@webrtc.org · 10 years ago
  12. da5452b Fix crash in AudioDeviceUtilityIOS::~AudioDeviceUtilityIOS. by tkchin@webrtc.org · 10 years ago
  13. d89fa97 This is related to an earlier CL of enabling Opus 48 kHz. by minyue@webrtc.org · 10 years ago
  14. 84649c0 AudioBuffer: Optimize const accesses to arrays that autoconvert int16<->float by kwiberg@webrtc.org · 10 years ago
  15. 8bd216f Reduce runtime of RingBufferTest by a factor of 100. by andrew@webrtc.org · 10 years ago
  16. 9ae7d44 Use _numMixedParticipants instead of audioFrameList->size() to determine if there're more than one participants. by wu@webrtc.org · 10 years ago
  17. 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
  18. c911a13 Remove unused ExperimentalNS API in AudioProcessing by aluebs@webrtc.org · 10 years ago
  19. 4309681 AudioBuffer: Eliminate the SplitChannelBuffer class by kwiberg@webrtc.org · 10 years ago
  20. eb15100 Simplify AudioBuffer::mixed_low_pass_data API by aluebs@webrtc.org · 10 years ago
  21. 7036325 AudioBuffer: Let ChannelBuffer handle bounds checking of channel parameter by kwiberg@webrtc.org · 10 years ago
  22. bde2bcb Add unit test for MediaFile WAV file writing by kwiberg@webrtc.org · 10 years ago
  23. fbdd355 Fixes up rtc so that it compiles on iOS 8 SDK. by tkchin@webrtc.org · 10 years ago
  24. bc9711f r6709 lacks a change in BUILD.gn by minyue@webrtc.org · 10 years ago
  25. 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
  26. bb77419 Compile-time guard for iOS7 specific property. by tkchin@webrtc.org · 10 years ago
  27. f3d2702 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 82383d9b14ff8e5fedf5a70229eb0ac6b512909a by Android Chromium Automerger · 10 years ago
  28. 7a4d45f Remove old padding path in RTPSender. by pbos@webrtc.org · 10 years ago
  29. fac3d8a nrsh1 is written before tmp321 is read, so needs to be earlyclobber by kwiberg@webrtc.org · 10 years ago
  30. df6904d Fix an invalid memory access due to typo in win/cursor.cc. by jiayl@webrtc.org · 10 years ago
  31. 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
  32. 8c82443 Improvements to the pacer where it lost some budget due to truncation errors. by stefan@webrtc.org · 10 years ago
  33. 2a34c4c Fix breakage introduced by r6691. by pbos@webrtc.org · 10 years ago
  34. 442dbd4 Make RTCP sender report send media bytes. by pbos@webrtc.org · 10 years ago
  35. 15097fc Remove the VPM denoiser. by pbos@webrtc.org · 10 years ago
  36. 82383d9 Fix deadlock in Android stopCapture() call. by glaznev@webrtc.org · 10 years ago
  37. ccf0fef GN: Fix include paths for WebRTC in Chromium build. by kjellander@webrtc.org · 10 years ago
  38. 5faa6d1 Fix bugs introduced by https://code.google.com/p/webrtc/source/detail?r=6667 . by tommi@webrtc.org · 10 years ago
  39. 502a271 Remove always-true expression. by tommi@webrtc.org · 10 years ago
  40. 9fbd3ec Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC. --- by tommi@webrtc.org · 10 years ago
  41. 0cb22cf Thread annotate RTCPSender. by pbos@webrtc.org · 10 years ago
  42. 55b0f2e Move pacer to fully use webrtc::Clock instead of webrtc::TickTime. by stefan@webrtc.org · 10 years ago
  43. d20c29a Document that channels are stored contiguously in AudioBuffer by aluebs@webrtc.org · 10 years ago
  44. a301f1a Remove unnecessary build message. by tommi@webrtc.org · 10 years ago
  45. 09da1a7 Remove the send-side cname getter APIs from voice and video engine. by stefan@webrtc.org · 10 years ago
  46. 477e6bc Update makefiles after merge of Chromium at 282385 by Android Chromium Automerger · 10 years ago
  47. 9aa3497 Count total bytes sent in RTPSender::Bytes(). by pbos@webrtc.org · 10 years ago
  48. 579d63c Fix data race in VCMTiming::ResetDecodeTime. by pbos@webrtc.org · 10 years ago
  49. 8c95e83 Refactor registerable callbacks for FrameCountObserver from rtp_rtcp module into vie_channel. by andresp@webrtc.org · 10 years ago
  50. f8bddb4 audio_processing: Updates aec_core_sse2.c with changes made to aec_common.h by bjornv@webrtc.org · 10 years ago
  51. 31ab61c Neon version of SubbandCoherence() by bjornv@webrtc.org · 10 years ago
  52. 71ba40d Neon version of rftbsub_128() by bjornv@webrtc.org · 10 years ago
  53. 91b4389 Revert "Remove remains of WEBRTC_NO_STL." (rev 6641). by andresp@webrtc.org · 10 years ago
  54. 61f437e Remove remains of WEBRTC_NO_STL. by andresp@webrtc.org · 10 years ago
  55. 75f7656 Create FullScreenChromeWindowDetector in DesktopConfigurationOptions::CreateDefault. by jiayl@webrtc.org · 10 years ago
  56. 10b9861 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 138adbb0bcdab60afda25a8727e5a071abc4ae36 by Android Chromium Automerger · 10 years ago
  57. 138adbb delay_estimator: Increases test coverage and makes input spectrum const by bjornv@webrtc.org · 10 years ago
  58. 678f190 Implement a work around for Chrome full-screen tab switch on Mac. by jiayl@webrtc.org · 10 years ago
  59. 62ef953 Neon version of rftfsub_128() by bjornv@webrtc.org · 10 years ago
  60. f8ec08e Refactor registerable callbacks for VideoBitrateObserver from rtp_rtcp module into vie_channel. by andresp@webrtc.org · 10 years ago
  61. 6aae61c Some refactoring inside rtp_rtcp/. by pbos@webrtc.org · 10 years ago
  62. 4109f65 Fixing compile error. by phoglund@webrtc.org · 10 years ago
  63. 6014599 Adding explicit check for using dummy file devices. by phoglund@webrtc.org · 10 years ago
  64. 2fd91bd Preserve RTP states for restarted VideoSendStreams. by pbos@webrtc.org · 10 years ago
  65. 857f934 Add initial gn build files for video_coding and video_processing. by stefan@webrtc.org · 10 years ago
  66. 0df907b Fix pacer to accept duplicate sequence numbers on different SSRCs. by pbos@webrtc.org · 10 years ago
  67. 7376819 Add missing break introduced in r6603. by stefan@webrtc.org · 10 years ago
  68. 14c7409 Fix test issues and a win compile error introduced with r6605. by stefan@webrtc.org · 10 years ago
  69. 2c02d82 Revert conversion from TickTime to int64_t in paced sender. by stefan@webrtc.org · 10 years ago
  70. 2d4a80c Add boilerplate code for H.264. by stefan@webrtc.org · 10 years ago
  71. 65afbf3 Configure RTX send status on new modules. by pbos@webrtc.org · 10 years ago
  72. c9995bc Introduces PacedVideoSender to test framework and moves the Pacer to use Clock. by stefan@webrtc.org · 10 years ago
  73. 74e26a6 Fix memcheck error in r6594. by marpan@webrtc.org · 10 years ago
  74. 0da4394 Fix for FEC decoding with sequence number wrap-around. by marpan@webrtc.org · 10 years ago
  75. 3ded580 Update makefiles after merge of Chromium at 281279 by Android Chromium Automerger · 10 years ago
  76. c7343a3 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at d13c3753199496aeddc73ec88548da73283c312f by Android Chromium Automerger · 10 years ago
  77. bf79df3 delay_estimator: Allows dynamically used history sizes by bjornv@webrtc.org · 10 years ago
  78. 4cff6c8 Make experimental NS API not purely virtual by aluebs@webrtc.org · 10 years ago
  79. 1465d80 common_audio: Removes macro WEBRTC_SPL_SHIFT_W16 by bjornv@webrtc.org · 10 years ago
  80. 38a0cee EchoCancellationImpl::ProcessRenderAudio: Use float samples directly by kwiberg@webrtc.org · 10 years ago
  81. d13c375 Implement BUILD.gn for desktop_capture. by jiayl@webrtc.org · 10 years ago
  82. 2836899 Add tkchin@ to OWNERS. by tkchin@webrtc.org · 10 years ago
  83. 34c5b23 Fix compile error introduced with r6571. by stefan@webrtc.org · 10 years ago
  84. 96583a9 Fixes a potential BWE clock mismatch bug. by stefan@webrtc.org · 10 years ago
  85. 2220b7a audio_processing/aec: Refactors NonLinearProcessing to prepare for NEON optimizations by bjornv@webrtc.org · 10 years ago
  86. 96a2a61 Neon version of cftmdl_128() by bjornv@webrtc.org · 10 years ago
  87. 8f02f89 Add ExperimentalNs support in Config by aluebs@webrtc.org · 10 years ago
  88. b9cd54f Neon version of cft1st_128() by bjornv@webrtc.org · 10 years ago
  89. 983db6a Make MediaOptimization thread-safe. by wuchengli@chromium.org · 10 years ago
  90. 958883a Receiver bit-exactness test for AudioCoding Module by henrik.lundin@webrtc.org · 10 years ago
  91. b3f0584 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 9ff0df06bd431ddbf595620f94ae515bbdcde2da by Android Chromium Automerger · 10 years ago
  92. 9ff0df0 Fixes a bug causing NACKs to be dropped excessively at the send-side. by stefan@webrtc.org · 10 years ago
  93. 9702d56 fix after r6472 in rtp_sender, comparison between signed and unsigned integer expressions. by henrike@webrtc.org · 10 years ago
  94. 841f8c8 Update makefiles after merge of Chromium at 279716 by Android Chromium Automerger · 10 years ago
  95. 4c21d3a Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at c34b9e5d5cd44c31c4f9da649b71d0d3132cf516 by Android Chromium Automerger · 10 years ago
  96. 7eec1dd Add RTCP packet types to packet builder: by asapersson@webrtc.org · 10 years ago
  97. b5272df This is to compare NetEq with various codecs under a shared packet loss pattern. by minyue@webrtc.org · 10 years ago
  98. ccbe08e Neon version of FilterFar() by bjornv@webrtc.org · 10 years ago
  99. f1a6eac Remove payload duplication in AudioDecoderTest by henrik.lundin@webrtc.org · 10 years ago
  100. c34b9e5 Removing neteq decode lock and friends by henrik.lundin@webrtc.org · 10 years ago