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webrtc / src / 38ebf98c2a7ed1871f1e32087efddce81023eb20
  1. 38ebf98 Refactor barcode decoder to use Zxing's C++ version by kjellander@webrtc.org · 12 years ago
  2. 755e19a - Checks the OS and runs the appropriate commands for Dummynet (ipfw) by jansson@webrtc.org · 12 years ago
  3. 971278a Splitting out video_coding_test executable again. by kjellander@webrtc.org · 12 years ago
  4. 3137a21 Dtmf twinkle-twinkle. by wu@webrtc.org · 12 years ago
  5. df123ed Roll libvpx 180104:186754. by andrew@webrtc.org · 12 years ago
  6. 603a7f4 Add third_party/ dependencies to svn:ignore by kjellander@webrtc.org · 12 years ago
  7. 2951a6d Fixed an assembly code error in AECM for ARMv7. by kma@webrtc.org · 12 years ago
  8. 84cd8e3 Disable frame dropper for screenshare mode. by stefan@webrtc.org · 12 years ago
  9. 7c16c3c Move video_coding OWNERS to video_coding/. by stefan@webrtc.org · 12 years ago
  10. 5d371393 Fixed a ton of Python lint errors, enabled python lint checking. by phoglund@webrtc.org · 12 years ago
  11. 52b57cc Fix debug file buffer bug introduced in r3574. by andrew@webrtc.org · 12 years ago
  12. efe4edb Enabling bufffering mode with no sync module or VoE by mikhal@webrtc.org · 12 years ago
  13. 488d4c9 Submit symlink in apprtc from Linux since it fails from Win by braveyao@webrtc.org · 12 years ago
  14. 07db4a6 Add symlink of adapter.js from apprtc to base by braveyao@webrtc.org · 12 years ago
  15. a9a1df0 Remove the error return on SetAGC failure introduced by r3605. by andrew@webrtc.org · 12 years ago
  16. 64dc671 Roll libyuv to r590 by fbarchard@google.com · 12 years ago
  17. 90eb5c8 1. Updated test pages to include Chrome Frame meta tag by elham@webrtc.org · 12 years ago
  18. 91d11b3 Adds new AEC API to audio_processing. by bjornv@webrtc.org · 12 years ago
  19. db3f427 Using adapter.js and getRemoteStreams by hta@webrtc.org · 12 years ago
  20. 1dc0aa2 Fix for build error on android introduced with r3609. by stefan@webrtc.org · 12 years ago
  21. a271070 Split the NACK list into multiple RTCPs if it's too big. by stefan@webrtc.org · 12 years ago
  22. a856db2 Moved trace function to adapter.js and removed from pc1 & multiple.html. by vikasmarwaha@webrtc.org · 12 years ago
  23. 24045c5 None of the clients of VoE use SetNetEQBGNMode(), furthermore, NetEq 4 does not provide an API to change the mode of the background noise. by turaj@webrtc.org · 12 years ago
  24. 7881b57 Updated path of adapter.js for dtmf & pc1-audio demos. by vikasmarwaha@webrtc.org · 12 years ago
  25. f0a90c3 Expose the capture-side AudioProcessing object and allow it to be injected. by andrew@webrtc.org · 12 years ago
  26. 7f95732 AEC Refactoring: Removes lint warning by bjornv@webrtc.org · 12 years ago
  27. 99f1346 Typo in index.html and updated svn propset for dtmf & pc1-audio demos. by vikasmarwaha@webrtc.org · 12 years ago
  28. b203540 Redirect webrtc-demos.appspot.com to svn site and added dtmf & pc1-audio demos. Also updated index page to include information about new demos. by vikasmarwaha@webrtc.org · 12 years ago
  29. ec6226e Updated version number to 3.25 by elham@webrtc.org · 12 years ago
  30. a64300a Refactor NACK list creation to build the NACK list as packets arrive. by stefan@webrtc.org · 12 years ago
  31. 17b867a compile fix for get_nprocs() with uClibc by phoglund@webrtc.org · 12 years ago
  32. 44f85a4 Fixed coverity defects (CID 14657 and 14656). by phoglund@webrtc.org · 12 years ago
  33. 73ec386 VideoCaptureAndroid can now capture just buffers without also rendering to a SurfaceView. by fischman@webrtc.org · 12 years ago
  34. 2412085 Don't upsample the capture signal early. by andrew@webrtc.org · 12 years ago
  35. ea38614 Update integration tests for idempotent RTP header settings. by bemasc@google.com · 12 years ago
  36. 7d6f113 Refactored inline assembly code in complex_fft.c, by combining the individual __asm lines into a single block, to avoid potential register usage problems when building with different tools. by kma@webrtc.org · 12 years ago
  37. 6be1e93 Properly error check calls to AudioProcessing. by andrew@webrtc.org · 12 years ago
  38. 9ee5a4c Enable External MediaProcessing on Mobile by leozwang@webrtc.org · 12 years ago
  39. 603ae3e Make RtpHeaderExtensionMap::Register and ::Deregister idempotent. by bemasc@google.com · 12 years ago
  40. 78693fe Return an error when greater than 16 kHz is used with AECM. by andrew@webrtc.org · 12 years ago
  41. 6648093 Destroy VCM and VPM instead of delete. by mflodman@webrtc.org · 12 years ago
  42. 527fb4d Revert "Will now run pylint on all python files if there's at least one modified python file in the checkin." by phoglund@webrtc.org · 12 years ago
  43. a0936a6 Limit ARM instruction "strheq" to Apple's clang compiler only. by kma@webrtc.org · 12 years ago
  44. 7d052c3 Turn off error concealment in videoprocessor_integration tests. by marpan@webrtc.org · 12 years ago
  45. 6b6eb44 Add supporting to V4L2_PIX_FMT_JPEG since it works same as MJPEG. by braveyao@webrtc.org · 12 years ago
  46. 9e25413 Rewrite the jitter buffer statistics test and put make it robust under valgrind. by stefan@webrtc.org · 12 years ago
  47. 98fce15 Adding webrtc-sample demos under trunk/samples. by vikasmarwaha@webrtc.org · 12 years ago
  48. 132c15d AEC Refactoring: by bjornv@webrtc.org · 12 years ago
  49. e1c4ed9 Fix to send a full NACK list at least roughly once every 1.5 x RTT. by stefan@webrtc.org · 12 years ago
  50. 83561fb Fixed a bug in WebRtcNsx_PrepareSpectrumNeon() for NS in ARM Neon platform. by kma@webrtc.org · 12 years ago
  51. 91f3255 Refactor WebRtc_CreateBuffer to return the instance. by andrew@webrtc.org · 12 years ago
  52. ff0cd88 PeerConnection client and server can now be built from libjingle trunk. So we no longer need this folder. The getting-started page has also been updated. by wu@webrtc.org · 12 years ago
  53. dd5482c Use ninja with merge_libs.py on Windows and clean up. by andrew@webrtc.org · 12 years ago
  54. 9fbd9ca Force a memcpy directly from the AEC ring buffer. by andrew@webrtc.org · 12 years ago
  55. ac1f877 Remove unneeded libvpx path from vp8 include_dirs. by andrew@webrtc.org · 12 years ago
  56. 9ae1354 Refactor ring_buffer interface, add a feature and a test. by andrew@webrtc.org · 12 years ago
  57. 8a06623 New attempt at fixing hard-coded libvpx source. by phoglund@webrtc.org · 12 years ago
  58. 9a6623b Revert "Fixing hard-coded libvpx source path." by phoglund@webrtc.org · 12 years ago
  59. 8571c90 Fixing hard-coded libvpx source path. by phoglund@webrtc.org · 12 years ago
  60. 2f9bd24 Ported assembly coding in APM from Android to iOS. by kma@webrtc.org · 12 years ago
  61. 0d8d010 Handle multiple calls to set initial delay by mikhal@webrtc.org · 12 years ago
  62. 7bf7326 Remove WEBRTC_TRACE completely when tracing is disabled. by wjia@webrtc.org · 12 years ago
  63. 10987a8 Minor bug fix in maxFPS parameter declaration. by vikasmarwaha@webrtc.org · 12 years ago
  64. bf3a9b3 Fix for WebRTC Issue 1384. Some cameras return 0 fps for all capabilities which causes divide-by-zero. by vikasmarwaha@webrtc.org · 12 years ago
  65. 5140e24 MIPS optimizations for Signal Processing Library patch01 by andrew@webrtc.org · 12 years ago
  66. 60f8313 AEC refactoring: Moved typedefs to _internal.h by bjornv@webrtc.org · 12 years ago
  67. 7a7a008 Changing non-const reference arguments to pointers, ACM by tina.legrand@webrtc.org · 12 years ago
  68. f61e02c Misc cleanups to webrtc/android code: by fischman@webrtc.org · 12 years ago
  69. 56a9ec3 Refactoring AEC: AecCore struct made private by bjornv@webrtc.org · 12 years ago
  70. 71e91f3 Refactor AEC: PowerLevel by bjornv@webrtc.org · 12 years ago
  71. 4d1cfae Added a pointer getter to the system_delay variable. by bjornv@webrtc.org · 12 years ago
  72. 47b274d Refactoring AEC: Added a SetConfigCore function by bjornv@webrtc.org · 12 years ago
  73. 716fd90 Moved out buffer handling to ProcessFrame() by bjornv@webrtc.org · 12 years ago
  74. ee7202f Removed unused get_config function. The configuration is already stored and handled in the audio processing module, so there is no need for this functionality. by bjornv@webrtc.org · 12 years ago
  75. 59b2d5f Stop and restart fix. by mflodman@webrtc.org · 12 years ago
  76. eb7ebf2 Revert 3543 by tina.legrand@webrtc.org · 12 years ago
  77. 374aa49 Changing non-const reference arguments to pointers, ACM by tina.legrand@webrtc.org · 12 years ago
  78. 0b6293a Fixed typo in vie_autotest_loopback.cc. by pbos@webrtc.org · 12 years ago
  79. 83663ef Replace gtest_prod.h include with our own FRIEND_TEST macro. by andrew@webrtc.org · 12 years ago
  80. aea96d3 Rename webrtc::StatsObserver to webrtc::CallStatsObserver by fischman@webrtc.org · 12 years ago
  81. 0a480cb Added getter for far_time_buf in AEC. Only used in AEC debug dump. by bjornv@webrtc.org · 12 years ago
  82. 5fc8292 This refactoring CL moves the nlp_mode member value from aecpc_t to aec_t, since it it never used at that level. Further, I removed two suppression variables by depending on nlp_mode directly. by bjornv@webrtc.org · 12 years ago
  83. cea70f4 * Name change * Removed WebRtcAec_ function name prepending on private function. by bjornv@webrtc.org · 12 years ago
  84. 95b48c3 Update to codec unit test: by marpan@webrtc.org · 12 years ago
  85. 77fced3 fixing nack list size calculation by mikhal@webrtc.org · 12 years ago
  86. 10741b3 Updated version number to 3.24 by elham@webrtc.org · 12 years ago
  87. 1682f71 Updating watchlist by mikhal@webrtc.org · 12 years ago
  88. ba23d11 Will now run pylint on all python files if there's at least one modified python file in the checkin. by phoglund@webrtc.org · 12 years ago
  89. 0460c72 Remove the dependency on dxguid.lib. by tommi@webrtc.org · 12 years ago
  90. d2c3bed Move directx_sdk_path definition variable into the video_render_module gyp file. by tommi@webrtc.org · 12 years ago
  91. eb91792 Refactoring temporal layers implementation and adding VideoCodecMode for easier control of codec settings. by stefan@webrtc.org · 12 years ago
  92. 3897255 Add VoE interface to VieRTP test by mikhal@webrtc.org · 12 years ago
  93. e3d6ffe Increase threshold in codec unit test. by marpan@webrtc.org · 12 years ago
  94. ef9f76a Adding a receive side API for buffering mode. by mikhal@webrtc.org · 12 years ago
  95. 47fe573 Bug fix for webrtc issue 1391. Typo in sin_length for socket address. by vikasmarwaha@webrtc.org · 12 years ago
  96. b4cd342 This refactoring CL contains an API to get low level echo metrics stats. by bjornv@webrtc.org · 12 years ago
  97. 21a2fc9 This Cl includes by bjornv@webrtc.org · 12 years ago
  98. 325f625 Moved the actual calculations to aec_core to avoid passing up low level members. by bjornv@webrtc.org · 12 years ago
  99. 0989fb7 Make VoiceEngineImpl inherit from VoiceEngine. by tommi@webrtc.org · 12 years ago
  100. 1723857 Removed astyle from webrtc_reformat since clang-format-chrome.py handles that now. by phoglund@webrtc.org · 12 years ago