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webrtc / src / a6219cc3ef08dd9b2981b065b6f051d7de0866f8
  1. a6219cc FileWrapper[Impl] modifications and actually remove the "Impl" class. by tommi · 9 years ago
  2. 74290b9 New rtc dump analyzing tool in Python by kjellander@webrtc.org · 9 years ago
  3. ceb9d0c Audio decoder factory test: Ensure that g722's sample rate is 16 kHz, not 8 kHz by kwiberg · 9 years ago
  4. 6808419 iSAC decoder: Remove obsolete TODO by kwiberg · 9 years ago
  5. edaa849 WebRtcVoiceCodecs: Eliminate some useless copying by kwiberg · 9 years ago
  6. 111744e Added backwards compatible version of WebRtcMediaEngineFactory::Create. by ossu · 9 years ago
  7. 71ee44c This cl: by perkj · 9 years ago
  8. 786f481 New misc scripts, header_usage.sh and author_line_count.sh. by nisse · 9 years ago
  9. 42883f8 Revert of Attempt to figure out what the issue is on the Win10 FYI build bot in content_browsertests. (patchset #1 id:1 of https://codereview.webrtc.org/2063313003/ ) by tommi · 9 years ago
  10. 17c3cdd Revert of Split IncomingVideoStream into two implementations, with smoothing and without. (patchset #23 id:430001 of https://codereview.webrtc.org/2035173002/ ) by tommi · 9 years ago
  11. 37ad337 Remove EncodedFrameCallbackAdapter. by sergeyu · 9 years ago
  12. 204177f Add RTCEventLog API to ObjC. by tkchin · 9 years ago
  13. e110411 Attempt to figure out what the issue is on the Win10 FYI build bot in content_browsertests. by tommi · 9 years ago
  14. 2cc8baa Adjust the amount of VP8 encoder threads for Android builds. by Alex Glaznev · 9 years ago
  15. 4deba9a Add SigslotTester0 for testing signals without argument. by honghaiz · 9 years ago
  16. 8189b02 Configure VoE NACK through AudioReceiveStream::Config, for receive streams. Also minor refactoring of WVoE unit test. by solenberg · 9 years ago
  17. 184a3fd Forward the SignalFirstPacketReceived to RtpReceiver. by zhihuang · 9 years ago
  18. 9a38cab Voice Engine: Remove RED support by kwiberg · 9 years ago
  19. 5aaa9fa Remove thread_checker in playout_delay_oracle by isheriff · 9 years ago
  20. 971cab0 Configure VoE NACK through AudioSendStream::Config, for send streams. by solenberg · 9 years ago
  21. 05b9803 Removed unused GetOutputVolume() and SetOutputVolume() from MediaEngineInterface. by solenberg · 9 years ago
  22. 8b06ec0 Change RTC_CHECK to RTC_CHECK_EQ for improved printout of GetLastError. by tommi · 9 years ago
  23. 6806136 Remove RED support from WebRtcVoiceEngine/MediaChannel by kwiberg · 9 years ago
  24. b1963b4 Reland of Re-enable UBsan on AGC. by minyue · 9 years ago
  25. dedfd28 Support for two audio codec lists down into WebRtcVoiceEngine. by ossu · 9 years ago
  26. 79ede03 Refactor VideoCapturerAndroid tests in WebRTC. by sakal · 9 years ago
  27. 1c7eef6 Split IncomingVideoStream into two implementations, with smoothing and without. by tommi · 9 years ago
  28. e355069 Disable SctpDataMediaChannelTest.ReusesAStream. by Peter Boström · 9 years ago
  29. 0208322 GN: Add video_engine_tests by Peter Boström · 9 years ago
  30. 075af92 Initial asymmetric codec support in MediaSessionDescription by ossu · 9 years ago
  31. 87abc28 Add kwiberg@webrtc.org as root owner. by solenberg · 9 years ago
  32. 8660024 Remove webrtc_all target by kjellander · 9 years ago
  33. 7336225 Delete left-over files. by nisse · 9 years ago
  34. 1fc4810 Always on statistics for AndroidMediaEncoder. by sakal · 9 years ago
  35. 81d99b3 A missing path separator caused aecdump recordings by peah · 9 years ago
  36. 54f5a26 Report errors creating peer connection in AppRTC Demo Android. by sakal · 9 years ago
  37. e9fc75e Fixing SCTP verbose packet logging. by deadbeef · 9 years ago
  38. dfe6937 Revert of Roll chromium_revision 7fa6701bc5..1a73d11e65 (398458:399420) (patchset #2 id:20001 of https://codereview.webrtc.org/2061723002/ ) by kjellander · 9 years ago
  39. c49cf13 Roll chromium_revision 7fa6701bc5..1a73d11e65 (398458:399420) by kjellander · 9 years ago
  40. fd5b4e9 GN: Add peerconnection_unittests by kjellander · 9 years ago
  41. 880ffeb Optimize the repeated calls to AudioEffect.queryEffects() on Android by skvlad · 9 years ago
  42. 6379793 Removing obsolete method from channel.h. by deadbeef · 9 years ago
  43. abfdb53 Fixed partially out of screen window capture in unix by gyzhou · 9 years ago
  44. 29b1a8d Moved creation of AudioDecoderFactory to inside PeerConnectionFactory. by ossu · 9 years ago
  45. 781e0c0 GN: Fix 32-bit Mac library error by kjellander · 9 years ago
  46. 718a763 Refactor scaling. by Niels Möller · 9 years ago
  47. be99ab9 Remove unnecessary redefinition of PacketLists in rtp_fec_unittest. by Rasmus Brandt · 9 years ago
  48. fb11424 GN: Add modules_unittests by kjellander · 9 years ago
  49. 142f8c5 GN: Add rtc_pc_unittests by kjellander · 9 years ago
  50. 82a9449 GN: Add rtc_media_unittests by kjellander · 9 years ago
  51. 979c268 Do not reconnect the network change signal each time the network manager is started by skvlad · 9 years ago
  52. 5d97a9a Adding more detail to MessageQueue::Dispatch logging. by Taylor Brandstetter · 9 years ago
  53. 51e6030 Update RateStatistics to handle too-little-data case. by Erik Språng · 9 years ago
  54. 733b547 Movable support for VideoReceiveStream::Config and avoid copies. by Tommi · 9 years ago
  55. bd3380f Make VideoReceiveStream not inherit from I420FrameCallback. by Tommi · 9 years ago
  56. bdce06e Delete unused YuvFrameGenerator class. by nisse · 9 years ago
  57. 602844a Delete some unused header files. by nisse · 9 years ago
  58. 81ca735 Remove new fuzzers until their GN targets work properly in Chromium. by katrielc · 9 years ago
  59. 94cee31 GN: Enable api,media,pc and p2p for the 'webrtc' target. by kjellander · 9 years ago
  60. 2b9423f Revert of Re-enable UBsan on AGC. (patchset #8 id:300001 of https://codereview.webrtc.org/2003623003/ ) by Åsa Persson · 9 years ago
  61. 555cfe9 Use relative paths for api/p2p fuzzers. by Peter Boström · 9 years ago
  62. dd70547 PRESUBMIT: Split NATIVE_API_DIRS into two lists. by kjellander · 9 years ago
  63. e4bda243 Before validating a STUN packet, check it's big enough for a header. by katrielc · 9 years ago
  64. 101f250 Implementing auto pausing of video streams. by mflodman · 9 years ago
  65. 2c1bec3 Add suppressions for memcheck errors. by asapersson · 9 years ago
  66. 8e85b3c Moves macros ACCESS_ON/RUN_ON from thread_annotations to thread_checker by danilchap · 9 years ago
  67. f9da44d RTCPeerConnectionInterface.mm createNativeConfiguration and other clean-up. by hbos · 9 years ago
  68. d4070c6 GN: Fix Chromium breakage for remote_bitrate_estimator by Henrik Kjellander · 9 years ago
  69. 5c1d043 Fix GYP/GN for webrtc/modules/remote_bitrate_estimator by kjellander · 9 years ago
  70. da75f7c Disable flaky test (WebRtcSessionTest.TestPacketOptionsAndOnPacketSent) on Dr Memory. by asapersson · 9 years ago
  71. d9f3d56 Use a video renderer instead of a capture observer in VideoCapturerAndroidTest. by sakal · 9 years ago
  72. 1503df6 Add suppressions for memcheck errors. by asapersson · 9 years ago
  73. efec590 Reland of New method I420Buffer::SetToBlack. (patchset #1 id:1 of https://codereview.webrtc.org/2049023002/ ) by nisse · 9 years ago
  74. e1cac64 Disable all BasicPortAllocatorTests on Dr Memory (flaky). by Åsa Persson · 9 years ago
  75. 40f5400 Start integrating StatsCounter class. by asapersson · 9 years ago
  76. fc22e03 Add AVFoundation video capture support to Mac objc SDK (based on iOS) by adam.fedor · 9 years ago
  77. f2a1c89 Add r-value constructor for RefCountedObject. by sergeyu · 9 years ago
  78. d5f41ce Use the new versions of OnAddStream/OnRemoveStream in objc binding. by deadbeef · 9 years ago
  79. 73fbcf9 Don't re-determine ICE role on an ICE restart. by deadbeef · 9 years ago
  80. 3cd9a30 Allow 100 char lines for ObjC files. by tkchin · 9 years ago
  81. 1c76bf1 Hide *.xcworkspace files by adam.fedor · 9 years ago
  82. bde418d Renamed video_coding/packet_buffer_unittest.cc. by philipel · 9 years ago
  83. 2019afd Replaced ACCESS_ON alias with GUARDED_BY macros by danilchap · 9 years ago
  84. e8f8f60 Only update Intelligibility Enhancer gains every 10 chunks by aluebs · 9 years ago
  85. b643939 Disable flaky TurnPortTests on Memcheck. by Åsa Persson · 9 years ago
  86. bea8959 Hibernate the thread if there are no events in the queue. Wake it up when an event is added to the queue. by terelius · 9 years ago
  87. 9195186 NetEq: Rename Nack to NackTracker to avoid name collisions in GN by henrik.lundin · 9 years ago
  88. bbe4233 Change name of files and class in agc/histogram* in order to avoid issue file-name clash in build files by peah · 9 years ago
  89. 86f7afd Android: Fix texture leak. by Niels Möller · 9 years ago
  90. a107402 Fix UBSan errors (signed integer overflow) by kwiberg · 9 years ago
  91. 271d740 Revert of New static method I420Buffer::SetToBlack. (patchset #4 id:60001 of https://codereview.webrtc.org/2029273004/ ) by nisse · 9 years ago
  92. 0ab07d6 Add ObjC API for getting native histograms. by asapersson · 9 years ago
  93. 663f9e2 New static method I420Buffer::SetToBlack. by nisse · 9 years ago
  94. 52f56d4 Roll chromium_revision 086802955f..7fa6701bc5 (396351:398458) by kjellander · 9 years ago
  95. 2a3892a GN: Add common_audio_unittests and common_video_unittests by kjellander · 9 years ago
  96. 3bcedd3 GN: Add SDK tests to rtc_unittests. by kjellander · 9 years ago
  97. 6b4b5f3 Add sender controlled playout delay limits by isheriff · 9 years ago
  98. 5d91028 Use std::unique_ptr<> to pass frame ownership in DesktopCapturer impls. by sergeyu · 9 years ago
  99. 6ebdf6b Fix issue with parsing of incorrect (empty) Stap-A H264 NAL units. by Erik Språng · 9 years ago
  100. a601f5c Separating internal and external methods of RtpSender/RtpReceiver. by deadbeef · 9 years ago