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gerrit-public.fairphone.software / platform / external / webrtc / 7ebbf9007706ba51be44e065a7c14f18a6d2b009
  1. 7ebbf90 New rtc dump analyzing tool in Python by aleloi · 8 years ago
  2. 3e33bfe Fix some sign-compare warnings in webrtc/api. by kjellander · 8 years ago
  3. 839315b Use the Chromium libfuzzer template instead of rolling our own. by katrielc · 8 years ago
  4. 1a20610 Fix buffer overflow in HMAC validation of STUN messages. by katrielc · 8 years ago
  5. c853597 rtc::Buffer: Grow capacity by at least 1.5x to prevent quadratic behavior by kwiberg · 8 years ago
  6. 504f335 Roll chromium_revision 465d55d04e..6a85b3b953 (400622:400641) by buildbot · 8 years ago
  7. ac62bd4 Rewrite CreateBlackFrame in webrtcvideoengine. by nisse · 8 years ago
  8. 44bf02f Remove SdpAudioFormat's default constructor by kwiberg · 8 years ago
  9. a7d88d3 Remove audio/video distinction for probe packets. by Peter Boström · 8 years ago
  10. 02343b9 Remove dead GYP target audio_device_module_java by kjellander · 8 years ago
  11. 442e6ee Workaround java.gypi inclusion error in Chromium builds. by kjellander · 8 years ago
  12. 4c7f8ae Cleanup MIPS specific link configuration by kjellander · 8 years ago
  13. fc36a2d Roll chromium_revision a21316a36e..465d55d04e (400620:400622) by buildbot · 8 years ago
  14. ff702f5 Roll chromium_revision f66fe7e469..a21316a36e (400617:400620) by buildbot · 8 years ago
  15. 71687f3 Roll chromium_revision 5cdeb1b846..f66fe7e469 (400605:400617) by buildbot · 8 years ago
  16. a9df50a Roll chromium_revision 0962148116..5cdeb1b846 (400593:400605) by buildbot · 8 years ago
  17. ce5a874 Improve encoding time calculation for Android HW encoder. by glaznev · 8 years ago
  18. 7508f3d Roll chromium_revision 6c3ee789f0..0962148116 (400588:400593) by buildbot · 8 years ago
  19. 5023d41 GN: Update xmpp and p2p to match Chromium build by kjellander · 8 years ago
  20. a935574 Roll chromium_revision e10a42e0d1..6c3ee789f0 (400570:400588) by buildbot · 8 years ago
  21. cc5a582 Roll chromium_revision b078b9902f..e10a42e0d1 (400452:400570) by buildbot · 8 years ago
  22. 61a6946 Roll chromium_revision 3a1c71fcbd..b078b9902f (400409:400452) by buildbot · 8 years ago
  23. f03a8d4 Unpack different wav files after each INIT event of the aecdump by aluebs · 8 years ago
  24. 863a826 Use |probe_cluster_id| to cluster packets. by philipel · 8 years ago
  25. 217fb66 Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume(). by solenberg · 8 years ago
  26. 3870001 Remove some dead code from VCMJitterBuffer. by Tommi · 8 years ago
  27. 57c21f9 Remove ViEEncoder::Pause / Start by perkj · 8 years ago
  28. c13ded5 Move AudioCodingModuleImpl to anonymous namespace in audio_coding_module.cc by kwiberg · 8 years ago
  29. 434b85d Roll chromium_revision ed2c9cb4cb..3a1c71fcbd (400384:400409) by buildbot · 8 years ago
  30. ca6d5d1 Partial reland of Delete unused and almost unused frame-related methods. (patchset #1 id:1 of https://codereview.webrtc.org/2076113002/ ) by nisse · 8 years ago
  31. fd634c4 Reland of Re-enable UBsan on AGC. by minyue · 8 years ago
  32. 07ec26d Fix crash parsing malformed rtp packet by danilchap · 8 years ago
  33. 9b99499 Added a builtin audio decoder factory to the default PeerConnectionFactory constructor. by ossu · 8 years ago
  34. 62379c8 Move Camera1 specific methods to Camera1Enumerator and create CameraEnumerator interface. by sakal · 8 years ago
  35. 72e735d Revert of Delete unused and almost unused frame-related methods. (patchset #12 id:220001 of https://codereview.webrtc.org/2065733003/ ) by nisse · 8 years ago
  36. 76270de Delete unused and almost unused frame-related methods. by nisse · 8 years ago
  37. 0dbc8bf Roll chromium_revision 1a73d11e65..ed2c9cb4cb (399420:400384) by buildbot · 8 years ago
  38. e6c9e88 Android: Add Size class. by sakal · 8 years ago
  39. 50c4821 Fix missing resource file in webrtc_perf_tests.isolate by kjellander · 8 years ago
  40. 6af2e86 Refactor VideoDenoiser to work with I420Buffer, not VideoFrame. by Niels Möller · 8 years ago
  41. 6820889 Roll chromium_revision 7fa6701bc5..1a73d11e65 (398458:399420) by kjellander · 8 years ago
  42. d1523ca Fix header size check in PseudoTcp::parse(). by sergeyu · 8 years ago
  43. 8e8222d Revert of Split IncomingVideoStream into two implementations, with smoothing and without. (patchset #4 id:290001 of https://codereview.webrtc.org/2071473002/ ) by tommi · 8 years ago
  44. e03f878 Reland of Split IncomingVideoStream into two implementations, with smoothing and without. by tommi · 8 years ago
  45. 4a0f7b5 - Remove use of VoERTP_RTCP::SetLocalSSRC() for receive streams; recreate them instead. by solenberg · 8 years ago
  46. 3abb764 Avoid unnecessary HW video encoder reconfiguration by skvlad · 8 years ago
  47. 9421853 Add AudioSendStream::SetMuted() method and use it in WVoMC::MuteStream(). by solenberg · 8 years ago
  48. e565a04 Revert of Fix crash parsing malformed rtp packet (patchset #1 id:1 of https://codereview.webrtc.org/2067793003/ ) by danilchap · 8 years ago
  49. 5a45fe6 Fix crash parsing malformed rtp packet by Danil Chapovalov · 8 years ago
  50. 4867ca2 Revert of -enable UBsan on AGC. (patchset #1 id:1 of https://codereview.webrtc.org/2063643003/ ) by pbos · 8 years ago
  51. 30a3a75 Fix buffer overflow parsing malformed rtp packet by Danil Chapovalov · 8 years ago
  52. 1642620 Performance fix for H264 RBSP parsing. by Erik Språng · 8 years ago
  53. fc3a8ee Delete unused code. by Niels Möller · 8 years ago
  54. 2d014be Resolves issue with bad audio using BT headsets on iOS. by henrika · 8 years ago
  55. 5a9e7e0 Fix a few error prone lines on VideoCapturerAndroid. by Sami Kalliomaki · 8 years ago
  56. b00dc38 Delete GetExecutablePath and related unused code. by Niels Möller · 8 years ago
  57. 342f740 NetEq: Ask AudioDecoder for sample rate instead of passing it as an argument by kwiberg · 8 years ago
  58. 347d351 AudioDecoder: Remove the default implementation of SampleRateHz by kwiberg · 8 years ago
  59. 371b43b Changes synchronization offset perfomance tracking by Danil Chapovalov · 8 years ago
  60. 4f0dfbd Change initial DTLS retransmission timer from 1 second to 50ms. by Taylor Brandstetter · 8 years ago
  61. 947c02d Disable WebRtcVideoChannel2BaseTest.AddRemoveCapturer because it is flaky by Alejandro Luebs · 8 years ago
  62. 4c17abe Add DesktopCapturer::Result::MAX_VALUE by Sergey Ulanov · 8 years ago
  63. 14461d4 Fixing flaky test: WebRtcSessionTest.TestPacketOptionsAndOnPacketSent by deadbeef · 8 years ago
  64. a6219cc FileWrapper[Impl] modifications and actually remove the "Impl" class. by tommi · 8 years ago
  65. 74290b9 New rtc dump analyzing tool in Python by kjellander@webrtc.org · 8 years ago
  66. ceb9d0c Audio decoder factory test: Ensure that g722's sample rate is 16 kHz, not 8 kHz by kwiberg · 8 years ago
  67. 6808419 iSAC decoder: Remove obsolete TODO by kwiberg · 8 years ago
  68. edaa849 WebRtcVoiceCodecs: Eliminate some useless copying by kwiberg · 8 years ago
  69. 111744e Added backwards compatible version of WebRtcMediaEngineFactory::Create. by ossu · 8 years ago
  70. 71ee44c This cl: by perkj · 8 years ago
  71. 786f481 New misc scripts, header_usage.sh and author_line_count.sh. by nisse · 8 years ago
  72. 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 · 8 years ago
  73. 17c3cdd Revert of Split IncomingVideoStream into two implementations, with smoothing and without. (patchset #23 id:430001 of https://codereview.webrtc.org/2035173002/ ) by tommi · 8 years ago
  74. 37ad337 Remove EncodedFrameCallbackAdapter. by sergeyu · 8 years ago
  75. 204177f Add RTCEventLog API to ObjC. by tkchin · 8 years ago
  76. e110411 Attempt to figure out what the issue is on the Win10 FYI build bot in content_browsertests. by tommi · 8 years ago
  77. 2cc8baa Adjust the amount of VP8 encoder threads for Android builds. by Alex Glaznev · 8 years ago
  78. 4deba9a Add SigslotTester0 for testing signals without argument. by honghaiz · 8 years ago
  79. 8189b02 Configure VoE NACK through AudioReceiveStream::Config, for receive streams. Also minor refactoring of WVoE unit test. by solenberg · 8 years ago
  80. 184a3fd Forward the SignalFirstPacketReceived to RtpReceiver. by zhihuang · 8 years ago
  81. 9a38cab Voice Engine: Remove RED support by kwiberg · 8 years ago
  82. 5aaa9fa Remove thread_checker in playout_delay_oracle by isheriff · 8 years ago
  83. 971cab0 Configure VoE NACK through AudioSendStream::Config, for send streams. by solenberg · 8 years ago
  84. 05b9803 Removed unused GetOutputVolume() and SetOutputVolume() from MediaEngineInterface. by solenberg · 8 years ago
  85. 8b06ec0 Change RTC_CHECK to RTC_CHECK_EQ for improved printout of GetLastError. by tommi · 8 years ago
  86. 6806136 Remove RED support from WebRtcVoiceEngine/MediaChannel by kwiberg · 8 years ago
  87. b1963b4 Reland of Re-enable UBsan on AGC. by minyue · 8 years ago
  88. dedfd28 Support for two audio codec lists down into WebRtcVoiceEngine. by ossu · 8 years ago
  89. 79ede03 Refactor VideoCapturerAndroid tests in WebRTC. by sakal · 8 years ago
  90. 1c7eef6 Split IncomingVideoStream into two implementations, with smoothing and without. by tommi · 8 years ago
  91. e355069 Disable SctpDataMediaChannelTest.ReusesAStream. by Peter Boström · 8 years ago
  92. 0208322 GN: Add video_engine_tests by Peter Boström · 8 years ago
  93. 075af92 Initial asymmetric codec support in MediaSessionDescription by ossu · 8 years ago
  94. 87abc28 Add kwiberg@webrtc.org as root owner. by solenberg · 8 years ago
  95. 8660024 Remove webrtc_all target by kjellander · 8 years ago
  96. 7336225 Delete left-over files. by nisse · 8 years ago
  97. 1fc4810 Always on statistics for AndroidMediaEncoder. by sakal · 8 years ago
  98. 81d99b3 A missing path separator caused aecdump recordings by peah · 8 years ago
  99. 54f5a26 Report errors creating peer connection in AppRTC Demo Android. by sakal · 8 years ago
  100. e9fc75e Fixing SCTP verbose packet logging. by deadbeef · 8 years ago