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webrtc / src / webrtc / 29763d300b8c37fbfb4ceb6dbbbe38cfbd0b906c
  1. 29763d3 Unpack different wav files after each INIT event of the aecdump by aluebs · 9 years ago
  2. fde23c8 Use |probe_cluster_id| to cluster packets. by philipel · 9 years ago
  3. 784336a Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume(). by solenberg · 9 years ago
  4. 7e87577 Remove some dead code from VCMJitterBuffer. by Tommi · 9 years ago
  5. b3f05f4 Remove ViEEncoder::Pause / Start by perkj · 9 years ago
  6. 7ac257b Move AudioCodingModuleImpl to anonymous namespace in audio_coding_module.cc by kwiberg · 9 years ago
  7. ea3460e Partial reland of Delete unused and almost unused frame-related methods. (patchset #1 id:1 of https://codereview.webrtc.org/2076113002/ ) by nisse · 9 years ago
  8. 38bbfbb Reland of Re-enable UBsan on AGC. by minyue · 9 years ago
  9. 4cfc3f9 Fix crash parsing malformed rtp packet by danilchap · 9 years ago
  10. 0398f3b Added a builtin audio decoder factory to the default PeerConnectionFactory constructor. by ossu · 9 years ago
  11. 9ca3abd Move Camera1 specific methods to Camera1Enumerator and create CameraEnumerator interface. by sakal · 9 years ago
  12. 389d403 Revert of Delete unused and almost unused frame-related methods. (patchset #12 id:220001 of https://codereview.webrtc.org/2065733003/ ) by nisse · 9 years ago
  13. ab2e723 Delete unused and almost unused frame-related methods. by nisse · 9 years ago
  14. 03792ee Android: Add Size class. by sakal · 9 years ago
  15. cb33bd5 Fix missing resource file in webrtc_perf_tests.isolate by kjellander · 9 years ago
  16. 383f4c9 Refactor VideoDenoiser to work with I420Buffer, not VideoFrame. by Niels Möller · 9 years ago
  17. df48a57 Roll chromium_revision 7fa6701bc5..1a73d11e65 (398458:399420) by kjellander · 9 years ago
  18. 502d9f0 Fix header size check in PseudoTcp::parse(). by sergeyu · 9 years ago
  19. 896b6c5 Revert of Split IncomingVideoStream into two implementations, with smoothing and without. (patchset #4 id:290001 of https://codereview.webrtc.org/2071473002/ ) by tommi · 9 years ago
  20. c289ab1 Reland of Split IncomingVideoStream into two implementations, with smoothing and without. by tommi · 9 years ago
  21. d056573 - Remove use of VoERTP_RTCP::SetLocalSSRC() for receive streams; recreate them instead. by solenberg · 9 years ago
  22. 48be054 Avoid unnecessary HW video encoder reconfiguration by skvlad · 9 years ago
  23. d9cd888 Add AudioSendStream::SetMuted() method and use it in WVoMC::MuteStream(). by solenberg · 9 years ago
  24. e207aed Revert of Fix crash parsing malformed rtp packet (patchset #1 id:1 of https://codereview.webrtc.org/2067793003/ ) by danilchap · 9 years ago
  25. aedaa42 Fix crash parsing malformed rtp packet by Danil Chapovalov · 9 years ago
  26. 5e251e4 Revert of -enable UBsan on AGC. (patchset #1 id:1 of https://codereview.webrtc.org/2063643003/ ) by pbos · 9 years ago
  27. b021dea Fix buffer overflow parsing malformed rtp packet by Danil Chapovalov · 9 years ago
  28. 79b280a Performance fix for H264 RBSP parsing. by Erik Språng · 9 years ago
  29. a259d26 Delete unused code. by Niels Möller · 9 years ago
  30. 471c2be Resolves issue with bad audio using BT headsets on iOS. by henrika · 9 years ago
  31. 58360e3 Fix a few error prone lines on VideoCapturerAndroid. by Sami Kalliomaki · 9 years ago
  32. 6061fcc Delete GetExecutablePath and related unused code. by Niels Möller · 9 years ago
  33. 1357a53 NetEq: Ask AudioDecoder for sample rate instead of passing it as an argument by kwiberg · 9 years ago
  34. 42abf60 AudioDecoder: Remove the default implementation of SampleRateHz by kwiberg · 9 years ago
  35. 8e697aa Changes synchronization offset perfomance tracking by Danil Chapovalov · 9 years ago
  36. fa82254 Change initial DTLS retransmission timer from 1 second to 50ms. by Taylor Brandstetter · 9 years ago
  37. 19923e4 Disable WebRtcVideoChannel2BaseTest.AddRemoveCapturer because it is flaky by Alejandro Luebs · 9 years ago
  38. 719b681 Add DesktopCapturer::Result::MAX_VALUE by Sergey Ulanov · 9 years ago
  39. f728aac Fixing flaky test: WebRtcSessionTest.TestPacketOptionsAndOnPacketSent by deadbeef · 9 years ago
  40. a6d985c FileWrapper[Impl] modifications and actually remove the "Impl" class. by tommi · 9 years ago
  41. e4795d7 Audio decoder factory test: Ensure that g722's sample rate is 16 kHz, not 8 kHz by kwiberg · 9 years ago
  42. b760e80 iSAC decoder: Remove obsolete TODO by kwiberg · 9 years ago
  43. 8e3caa1 WebRtcVoiceCodecs: Eliminate some useless copying by kwiberg · 9 years ago
  44. 51c8196 Added backwards compatible version of WebRtcMediaEngineFactory::Create. by ossu · 9 years ago
  45. 42209be This cl: by perkj · 9 years ago
  46. 732c17c New misc scripts, header_usage.sh and author_line_count.sh. by nisse · 9 years ago
  47. d173adb 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
  48. 36500f5 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
  49. 28120a0 Remove EncodedFrameCallbackAdapter. by sergeyu · 9 years ago
  50. b6f1589 Add RTCEventLog API to ObjC. by tkchin · 9 years ago
  51. fcf949d Attempt to figure out what the issue is on the Win10 FYI build bot in content_browsertests. by tommi · 9 years ago
  52. 0f6395b Adjust the amount of VP8 encoder threads for Android builds. by Alex Glaznev · 9 years ago
  53. 0c58e51 Add SigslotTester0 for testing signals without argument. by honghaiz · 9 years ago
  54. ff1d51a Configure VoE NACK through AudioReceiveStream::Config, for receive streams. Also minor refactoring of WVoE unit test. by solenberg · 9 years ago
  55. a2636be Forward the SignalFirstPacketReceived to RtpReceiver. by zhihuang · 9 years ago
  56. a777bf6 Voice Engine: Remove RED support by kwiberg · 9 years ago
  57. 338a67c Remove thread_checker in playout_delay_oracle by isheriff · 9 years ago
  58. 5a37e3e Configure VoE NACK through AudioSendStream::Config, for send streams. by solenberg · 9 years ago
  59. e4800a7 Removed unused GetOutputVolume() and SetOutputVolume() from MediaEngineInterface. by solenberg · 9 years ago
  60. 360b085 Change RTC_CHECK to RTC_CHECK_EQ for improved printout of GetLastError. by tommi · 9 years ago
  61. 8f99654 Remove RED support from WebRtcVoiceEngine/MediaChannel by kwiberg · 9 years ago
  62. ec74df9 Reland of Re-enable UBsan on AGC. by minyue · 9 years ago
  63. c0c552c Support for two audio codec lists down into WebRtcVoiceEngine. by ossu · 9 years ago
  64. 1d24c65 Refactor VideoCapturerAndroid tests in WebRTC. by sakal · 9 years ago
  65. 4f834e9 Split IncomingVideoStream into two implementations, with smoothing and without. by tommi · 9 years ago
  66. ac0b785 Disable SctpDataMediaChannelTest.ReusesAStream. by Peter Boström · 9 years ago
  67. c9da01c GN: Add video_engine_tests by Peter Boström · 9 years ago
  68. d8878f5 Initial asymmetric codec support in MediaSessionDescription by ossu · 9 years ago
  69. 4fcc511 Add kwiberg@webrtc.org as root owner. by solenberg · 9 years ago
  70. 827c515 Remove webrtc_all target by kjellander · 9 years ago
  71. d7b1131 Delete left-over files. by nisse · 9 years ago
  72. c43d129 Always on statistics for AndroidMediaEncoder. by sakal · 9 years ago
  73. 4af0d53 A missing path separator caused aecdump recordings by peah · 9 years ago
  74. f3f5e56 Report errors creating peer connection in AppRTC Demo Android. by sakal · 9 years ago
  75. 999c587 Fixing SCTP verbose packet logging. by deadbeef · 9 years ago
  76. 5398307 Revert of Roll chromium_revision 7fa6701bc5..1a73d11e65 (398458:399420) (patchset #2 id:20001 of https://codereview.webrtc.org/2061723002/ ) by kjellander · 9 years ago
  77. 83aafc7 Roll chromium_revision 7fa6701bc5..1a73d11e65 (398458:399420) by kjellander · 9 years ago
  78. d7c4e64 GN: Add peerconnection_unittests by kjellander · 9 years ago
  79. 9b9c620 Optimize the repeated calls to AudioEffect.queryEffects() on Android by skvlad · 9 years ago
  80. b7f0831 Removing obsolete method from channel.h. by deadbeef · 9 years ago
  81. eabaee1 Fixed partially out of screen window capture in unix by gyzhou · 9 years ago
  82. e840777 Moved creation of AudioDecoderFactory to inside PeerConnectionFactory. by ossu · 9 years ago
  83. c2afc03 GN: Fix 32-bit Mac library error by kjellander · 9 years ago
  84. 34d0c32 Refactor scaling. by Niels Möller · 9 years ago
  85. c4dedf3 Remove unnecessary redefinition of PacketLists in rtp_fec_unittest. by Rasmus Brandt · 9 years ago
  86. becd982 GN: Add modules_unittests by kjellander · 9 years ago
  87. a996c6a GN: Add rtc_pc_unittests by kjellander · 9 years ago
  88. d9eaaa0 GN: Add rtc_media_unittests by kjellander · 9 years ago
  89. bdf80af Do not reconnect the network change signal each time the network manager is started by skvlad · 9 years ago
  90. c0bec8f Adding more detail to MessageQueue::Dispatch logging. by Taylor Brandstetter · 9 years ago
  91. 475c05f Update RateStatistics to handle too-little-data case. by Erik Språng · 9 years ago
  92. 87875ef Movable support for VideoReceiveStream::Config and avoid copies. by Tommi · 9 years ago
  93. a5ab043 Make VideoReceiveStream not inherit from I420FrameCallback. by Tommi · 9 years ago
  94. 5649111 Delete unused YuvFrameGenerator class. by nisse · 9 years ago
  95. 6ab85a4 Delete some unused header files. by nisse · 9 years ago
  96. 456c3c7 Remove new fuzzers until their GN targets work properly in Chromium. by katrielc · 9 years ago
  97. 6ec4394 GN: Enable api,media,pc and p2p for the 'webrtc' target. by kjellander · 9 years ago
  98. 8d2c27a Revert of Re-enable UBsan on AGC. (patchset #8 id:300001 of https://codereview.webrtc.org/2003623003/ ) by Åsa Persson · 9 years ago
  99. bcb3e12 Use relative paths for api/p2p fuzzers. by Peter Boström · 9 years ago
  100. a2e19aa Before validating a STUN packet, check it's big enough for a header. by katrielc · 9 years ago