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webrtc / src / b0f4b3da055cb09813d52f417f64ce2275887fea / webrtc
  1. b0f4b3d Improving error message from neteq_rtpplay by henrik.lundin@webrtc.org · 10 years ago
  2. 5804936 Add format members to AudioConverter for DCHECKing. by andrew@webrtc.org · 10 years ago
  3. e451b75 Update rate control parameter in vp9 test. by marpan@webrtc.org · 10 years ago
  4. 4765ca5 Roll chromium_revision: 28d1981..d3db2ff by marpan@webrtc.org · 10 years ago
  5. f866b2d Restore the void return type on WriteWavHeader. by andrew@webrtc.org · 10 years ago
  6. b81e304 replace inline assembly WebRtcNsx_AnalysisUpdate by intrinsics. by andrew@webrtc.org · 10 years ago
  7. f947180 Add Opus support to neteq_rtpplay by henrik.lundin@webrtc.org · 10 years ago
  8. 548b228 Add UMA metrics for the initial (after two seconds) packet loss, round-trip time and bandwidth estimate of a WebRTC call. by stefan@webrtc.org · 10 years ago
  9. 96dc685 Add stats for video: by asapersson@webrtc.org · 10 years ago
  10. bf09976 Add more sanity checks to workaround the unidentified problem that CaptureThread is still running while related resouces are destroyed already. by braveyao@webrtc.org · 10 years ago
  11. ed45896 Adjust/increase rate control thresold for a vp9 test. by marpan@webrtc.org · 10 years ago
  12. 5b88317 Add VP9 codec to VCM and vie_auto_test. by marpan@webrtc.org · 10 years ago
  13. 5072e0f Update Android projects to API level 21. by kjellander@webrtc.org · 10 years ago
  14. 818c9f9 replace inline assembly WebRtcNsx_SynthesisUpdateNeon by intrinsics. by andrew@webrtc.org · 10 years ago
  15. a3ed713 Add a WavReader counterpart to WavWriter. by andrew@webrtc.org · 10 years ago
  16. 78c222b Update all .isolate files for the new format. by kjellander@webrtc.org · 10 years ago
  17. 8a130c1 Update Android projects to API level 20. by kjellander@webrtc.org · 10 years ago
  18. 053c6ab Fix N7 camera aspect ratio. by glaznev@webrtc.org · 10 years ago
  19. 508c916 Build fix for MIPS32R6. by andrew@webrtc.org · 10 years ago
  20. cc476aa Fix a name collision with Android libc++ by andrew@webrtc.org · 10 years ago
  21. b7ed779 Implement conference-mode temporal-layer screencast. by pbos@webrtc.org · 10 years ago
  22. 3bf3d23 Configure A/V sync in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
  23. 4abadab Simplify bwe tests. by stefan@webrtc.org · 10 years ago
  24. 8328e7c Revert "Revert part of r7561, "Refactor audio conversion functions."" by andrew@webrtc.org · 10 years ago
  25. 14146e4 arm64 iOS build. by tkchin@webrtc.org · 10 years ago
  26. d0cf68e Add 15 fps support for Android devices with missing 15 fps camera mode. by glaznev@webrtc.org · 10 years ago
  27. 8aa4d2d Creating a C++ wrapper class for VAD by henrik.lundin@webrtc.org · 10 years ago
  28. bcfb4d0 Revert part of r7561, "Refactor audio conversion functions." by kwiberg@webrtc.org · 10 years ago
  29. 4fc4add Refactor audio conversion functions. by andrew@webrtc.org · 10 years ago
  30. 776e6f2 Use external VideoDecoders in VideoReceiveStream. by pbos@webrtc.org · 10 years ago
  31. 2dd3134 Add stats for duplicate sent and received NACK requests. by asapersson@webrtc.org · 10 years ago
  32. f567095 common_audio: Removed macro WEBRTC_SPL_RSHIFT_W32 by bjornv@webrtc.org · 10 years ago
  33. 7f10513 Remove unused code in overuse detector. by asapersson@webrtc.org · 10 years ago
  34. decd930 AudioEncoder: num_10ms_frames_per_packet -> Num10MsFramesInNextPacket by kwiberg@webrtc.org · 10 years ago
  35. cfe3845 Enable G.722 for Chromium builds by henrik.lundin@webrtc.org · 10 years ago
  36. 663fdd0 Make an AudioEncoder subclass for Opus by kwiberg@webrtc.org · 10 years ago
  37. ffeaeed Make NSinst_t* const and rename to self in ns_core by aluebs@webrtc.org · 10 years ago
  38. 269fb4b move xmpp and p2p to webrtc by henrike@webrtc.org · 10 years ago
  39. 8b1b23f Make local functions static and dropWebRtcNs_ in ns_core by aluebs@webrtc.org · 10 years ago
  40. 28b5467 Make all comments whole sentences in ns_core by aluebs@webrtc.org · 10 years ago
  41. bd6bdca scoped_ptr.h: Renames function and change namespace scope to fix conflicts with Chromium not detected by the FYI bots. by henrike@webrtc.org · 10 years ago
  42. a296725 audio_coding/codecs/isac/fix: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>" by bjornv@webrtc.org · 10 years ago
  43. 67ca26e common_audio: Removed trivial macro WEBRTC_SPL_UMUL_16_16 by bjornv@webrtc.org · 10 years ago
  44. ff8a98e Use neteq_unittest_tools in audio_decoder_unittests by henrik.lundin@webrtc.org · 10 years ago
  45. 820efd5b Fix double backslashes in incoming_video_stream.cc by perkj@webrtc.org · 10 years ago
  46. aada86b Add a simple AudioConverter class. by andrew@webrtc.org · 10 years ago
  47. 33a0e2d Only configure the SSL library in one place. by henrike@webrtc.org · 10 years ago
  48. aca5803 Move (test) RtpFileReader to a lightweight target. by pbos@webrtc.org · 10 years ago
  49. b787f4c Move scoped_ptr "free" functions into the webrtc namespace. by andrew@webrtc.org · 10 years ago
  50. df42988 Upgrade our scoped_ptr copy to match Chromium's latest. by andrew@webrtc.org · 10 years ago
  51. a37f1dd Cleaning up audio_decoder_test.cc and adding ResampleInputAudioFile by henrik.lundin@webrtc.org · 10 years ago
  52. 0552356 isacfix: Refactor big-endian reading and writing by kwiberg@webrtc.org · 10 years ago
  53. 9fed099 Increase max trace message size to 1024 characters. by pbos@webrtc.org · 10 years ago
  54. c86ec3e Fix ::~LogMessage to print as a string. by pbos@webrtc.org · 10 years ago
  55. 39b1743 Adding the subtool rtcBot report visualizer by houssainy@google.com · 10 years ago
  56. ad3b5a5 Move min transmit bitrate to VideoEncoderConfig. by pbos@webrtc.org · 10 years ago
  57. 7e19a11 Break out WebRtcNs_ComputeDdUpdate function in ns_core by aluebs@webrtc.org · 10 years ago
  58. f8ea0d5 Break out WebRtcNs_UpdateNoise function in ns_core by aluebs@webrtc.org · 10 years ago
  59. 799e88a Break out FFT function in ns_core by aluebs@webrtc.org · 10 years ago
  60. 8454ad8 Break out ComputeSnr function in ns_core by aluebs@webrtc.org · 10 years ago
  61. 0d3e254 Adding three video conference bots test by houssainy@google.com · 10 years ago
  62. 0e19d0c Adding file from test.webrtc.org domain to be downloaded by houssainy@google.com · 10 years ago
  63. 580d367 Add macros and APIs for webrtc histograms. by asapersson@webrtc.org · 10 years ago
  64. 82462aa Adds support for sending first set of packets at increasingly higher bitrates to probe the link and faster ramp up to a high bitrate. by stefan@webrtc.org · 10 years ago
  65. 2192701 Using the Unused turn configuration in two way test by houssainy@google.com · 10 years ago
  66. ad553a2 Let video_loopback use internal VCM capturers. by pbos@webrtc.org · 10 years ago
  67. fce8f5d NOTE: This code review based on the running issue: by houssainy@google.com · 10 years ago
  68. 3382059 Adding Two way video and audio streaming test to RtcBot by houssainy@google.com · 10 years ago
  69. e9b7d03 HTTPS Server used instead of HTTP for loading the bots to avoid the media permission pop-up clicks every time running the test. by houssainy@google.com · 10 years ago
  70. 32452b2 Make ReconfigureVideoEncoder use current bitrate. by pbos@webrtc.org · 10 years ago
  71. 3f8f555 Disable TestVp8Impl.BaseUnitTest on MSan. by pbos@webrtc.org · 10 years ago
  72. 76960d5 For FIR packet, payload length is zero, so SendToNetwork function is failing. by stefan@webrtc.org · 10 years ago
  73. 67cf1d7 Break out WebRtcNs_Windowing function in ns_core by aluebs@webrtc.org · 10 years ago
  74. 0e70992 Break out WebRtcNs_Energy function in ns_core by aluebs@webrtc.org · 10 years ago
  75. 7634c09 Break out WebRtcNs_IFFT function in ns_core by aluebs@webrtc.org · 10 years ago
  76. 333e255 Break out WebRtcNs_UpdateBuffer function in ns_core by aluebs@webrtc.org · 10 years ago
  77. def1e97 Implement AudioEncoderPcmU/A classes and convert AudioDecoder tests by henrik.lundin@webrtc.org · 10 years ago
  78. 78ea06d audio_coding/codecs/ilbc: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> by bjornv@webrtc.org · 10 years ago
  79. 913f7b8 Fix for glitches in ACM when switching desired output sample rate by henrik.lundin@webrtc.org · 10 years ago
  80. b69ea9a common_audio: Replaced invalid operand in min_max_operations_neon.S" by bjornv@webrtc.org · 10 years ago
  81. b35b136 Make avg_{psnr,ssim}_threshold_ const. by pbos@webrtc.org · 10 years ago
  82. 2abebe7 audio_coding/codecs/isac/main: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> by bjornv@webrtc.org · 10 years ago
  83. a5ce7bb audio_coding/neteq: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> by bjornv@webrtc.org · 10 years ago
  84. 28100cb Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p." by henrike@webrtc.org · 10 years ago
  85. b1dac33 Revert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..." by henrike@webrtc.org · 10 years ago
  86. 0371a37 Moving creating TURN configration to the host machine instead of the bots - rtcBot by houssainy@google.com · 10 years ago
  87. f7030d4 Query Android device orientation on every camera frame received. by glaznev@webrtc.org · 10 years ago
  88. c221db6 Test names changed from e.g) testOneWayVideo/chrome=>chrome to testOneWayVideo/chrome-chrome. by houssainy@google.com · 10 years ago
  89. 264e66f Add encoded_timestamp to AudioEncoder base class by henrik.lundin@webrtc.org · 10 years ago
  90. 9ea6f8a New interface class AudioEncoder by henrik.lundin@webrtc.org · 10 years ago
  91. 458c2c3 Improve rtcbot to load all test files at start and allow them to registerTests by andresp@webrtc.org · 10 years ago
  92. 9aed002 Add ability to include a larger time span (in addition to encode time) for measuring the processing time of a frame. by asapersson@webrtc.org · 10 years ago
  93. d1ba6d9 Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p. by henrike@webrtc.org · 10 years ago
  94. 3e2f8ff Selecting bot_type changed to be specified in the test file by houssainy@google.com · 10 years ago
  95. e93cbd1 Fix data races in ThreadTest.ThreeThreadsInvoke. by pbos@webrtc.org · 10 years ago
  96. f87c0af audio_processing: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> by bjornv@webrtc.org · 10 years ago
  97. f02ba9b audio_processing/agc: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> by bjornv@webrtc.org · 10 years ago
  98. 8dc00d7 audio_processing/ns: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> by bjornv@webrtc.org · 10 years ago
  99. 99e561f Extend AcmSwitchingOutputFrequencyOldApi with more frequencies by henrik.lundin@webrtc.org · 10 years ago
  100. fab5439 common_audio: Removed version API from signal_processing by bjornv@webrtc.org · 10 years ago