1. 9c3b7bd Move the VIE tests to use external transport instead of the built in udp transport by pwestin@webrtc.org · 11 years ago
  2. 680fbc5 Add trace printouts to all unit tests. by andrew@webrtc.org · 11 years ago
  3. 90fa4a1 Add min and target bitrate to VideoCodec. by marpan@webrtc.org · 11 years ago
  4. 9c0b169 Move the VoE tests to use external transport instead of the built in udp transport by pwestin@webrtc.org · 11 years ago
  5. ce2d125 Creating a copy of Udp transport under webrtc/test by pwestin@webrtc.org · 11 years ago
  6. ed6b4c8 Cleanup nanosleep -> SleepMs Remove some leftover stuff by hta@webrtc.org · 11 years ago
  7. e3abb18 WebRtc_Word -> stdint in audio_coding/g711/ by pbos@webrtc.org · 11 years ago
  8. 48ec040 Remove incorrect asserts. by stefan@webrtc.org · 11 years ago
  9. 326becd WebRtc_Word -> stdint in audio_coding/cng/ by pbos@webrtc.org · 11 years ago
  10. c226567 Fix -Wstring-conversion warnings. by wu@webrtc.org · 11 years ago
  11. f99f63f Thread safety issue fix in incoming_video_stream.cc. See issue 1465. by vikasmarwaha@webrtc.org · 11 years ago
  12. d579a2a Account for header inside I420Encoder::InitEncode. by pbos@webrtc.org · 11 years ago
  13. 06d1e8f Follow-up fix for r3681. by stefan@webrtc.org · 11 years ago
  14. 6f1f826 Fixed initialization of SPL in echo_control_mobile. by kma@webrtc.org · 11 years ago
  15. aef22a7 Android: rename android_build_type gyp variable. by wjia@webrtc.org · 11 years ago
  16. 035c96a Updated WebRTC version number to 3.26 by elham@webrtc.org · 11 years ago
  17. ebdc04d Fix framerate sent to account for actually sent frames. by stefan@webrtc.org · 11 years ago
  18. 3be5a98 Change VCM interface to take target bitrate in bits per second. by stefan@webrtc.org · 11 years ago
  19. a2e9124 Generic video-codec support. by pbos@webrtc.org · 11 years ago
  20. 857be46 Revert the deletion of test_api_nack.cc in r3674. by stefan@webrtc.org · 11 years ago
  21. 1f71c06 Truncated delay quality to avoid negative return values by bjornv@webrtc.org · 11 years ago
  22. 072c9b6 Adding RTX on source by mikhal@webrtc.org · 11 years ago
  23. feaa409 Adding Opus frame length test by tina.legrand@webrtc.org · 11 years ago
  24. 897e86f Fixed a crash issue in NSX module. by kma@webrtc.org · 11 years ago
  25. 9a7b9f7 Revert r3667 and r3665 Review URL: https://webrtc-codereview.appspot.com/1199004 by pwestin@webrtc.org · 11 years ago
  26. a891566 Added destructors for tests to control destruct order by pwestin@webrtc.org · 11 years ago
  27. 25023aa Increasing size of nack list in buffered mode. by mikhal@webrtc.org · 11 years ago
  28. 66ccc6e Removed the engine API:s related to transport such as SetSendDestination, the functionality is now provided via the test frame work. by pwestin@webrtc.org · 11 years ago
  29. 9fe8f22 Refactor webrtc specific Event implementation to an EventFactory. by stefan@webrtc.org · 11 years ago
  30. d4caede Remove DTMF detection. Talk team has been in the loop and there is no need for by turaj@webrtc.org · 11 years ago
  31. b661eae Tool found: pass by value when pass by reference is better in system wrapper unit test. by henrike@webrtc.org · 11 years ago
  32. a284f1d Change intrinsic code in isac fix to let it pass chrome clang compiler. by kma@webrtc.org · 11 years ago
  33. 2c62fd9 Fixes issue detected by tool. by henrike@webrtc.org · 11 years ago
  34. 020e6ad Removed redundant VP8 width/height and made sure the generic width/height is set. by stefan@webrtc.org · 11 years ago
  35. 9b78141 Revert "Internal clean up: removing unused include line." by dwkang@webrtc.org · 11 years ago
  36. 1a8d06e Internal clean up: removing unused include line. by dwkang@webrtc.org · 11 years ago
  37. 2d2bfb0 Fixed issue 1497 in iSAC fixed point. by kma@webrtc.org · 11 years ago
  38. 38a5679 Fix frame_editing_unittest reference file handling. by kjellander@webrtc.org · 11 years ago
  39. c51b060 Optimized EstCodeLpcCoef() for iSAC with intrinsics in Android-Neon platform. by kma@webrtc.org · 11 years ago
  40. 2a3949f Lazy capture_device_info acquisition. by pbos@webrtc.org · 11 years ago
  41. 46672bb Refactor barcode decoder to use Zxing's C++ version by kjellander@webrtc.org · 11 years ago
  42. c96125a Splitting out video_coding_test executable again. by kjellander@webrtc.org · 11 years ago
  43. ad807de Fixed an assembly code error in AECM for ARMv7. by kma@webrtc.org · 11 years ago
  44. 2dbb66b Disable frame dropper for screenshare mode. by stefan@webrtc.org · 11 years ago
  45. 2a070a5 Move video_coding OWNERS to video_coding/. by stefan@webrtc.org · 11 years ago
  46. 2733e12 Fixed a ton of Python lint errors, enabled python lint checking. by phoglund@webrtc.org · 11 years ago
  47. 4621446 Fix debug file buffer bug introduced in r3574. by andrew@webrtc.org · 11 years ago
  48. ace0823 Enabling bufffering mode with no sync module or VoE by mikhal@webrtc.org · 11 years ago
  49. 66b0c5d Remove the error return on SetAGC failure introduced by r3605. by andrew@webrtc.org · 11 years ago
  50. ad3fd52 1. Updated test pages to include Chrome Frame meta tag by elham@webrtc.org · 11 years ago
  51. 333987b Adds new AEC API to audio_processing. by bjornv@webrtc.org · 11 years ago
  52. 08b9b59 Fix for build error on android introduced with r3609. by stefan@webrtc.org · 11 years ago
  53. 2654c43 Split the NACK list into multiple RTCPs if it's too big. by stefan@webrtc.org · 11 years ago
  54. df1cfd1 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 · 11 years ago
  55. 6316d17 Expose the capture-side AudioProcessing object and allow it to be injected. by andrew@webrtc.org · 11 years ago
  56. 2c1f9d4 AEC Refactoring: Removes lint warning by bjornv@webrtc.org · 11 years ago
  57. 87d8f2d Updated version number to 3.25 by elham@webrtc.org · 11 years ago
  58. eeaacdb Refactor NACK list creation to build the NACK list as packets arrive. by stefan@webrtc.org · 11 years ago
  59. 552f230 compile fix for get_nprocs() with uClibc by phoglund@webrtc.org · 11 years ago
  60. 89cc166 Fixed coverity defects (CID 14657 and 14656). by phoglund@webrtc.org · 11 years ago
  61. 4aa2314 VideoCaptureAndroid can now capture just buffers without also rendering to a SurfaceView. by fischman@webrtc.org · 11 years ago
  62. 7d2689d Don't upsample the capture signal early. by andrew@webrtc.org · 11 years ago
  63. 3da576e Update integration tests for idempotent RTP header settings. by bemasc@google.com · 11 years ago
  64. 13a186f 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 · 11 years ago
  65. ad179ce Properly error check calls to AudioProcessing. by andrew@webrtc.org · 11 years ago
  66. 8e4340d Enable External MediaProcessing on Mobile by leozwang@webrtc.org · 11 years ago
  67. 8648aad Make RtpHeaderExtensionMap::Register and ::Deregister idempotent. by bemasc@google.com · 11 years ago
  68. bb2973a Return an error when greater than 16 kHz is used with AECM. by andrew@webrtc.org · 11 years ago
  69. 1dcba31 Destroy VCM and VPM instead of delete. by mflodman@webrtc.org · 11 years ago
  70. 22fc115 Limit ARM instruction "strheq" to Apple's clang compiler only. by kma@webrtc.org · 11 years ago
  71. fef10a3 Turn off error concealment in videoprocessor_integration tests. by marpan@webrtc.org · 11 years ago
  72. 51d5c6d Add supporting to V4L2_PIX_FMT_JPEG since it works same as MJPEG. by braveyao@webrtc.org · 11 years ago
  73. eeb8b8f Rewrite the jitter buffer statistics test and put make it robust under valgrind. by stefan@webrtc.org · 11 years ago
  74. 12509cf AEC Refactoring: by bjornv@webrtc.org · 11 years ago
  75. 6f93416 Fix to send a full NACK list at least roughly once every 1.5 x RTT. by stefan@webrtc.org · 11 years ago
  76. 9ea9696 Fixed a bug in WebRtcNsx_PrepareSpectrumNeon() for NS in ARM Neon platform. by kma@webrtc.org · 11 years ago
  77. 24e40df Refactor WebRtc_CreateBuffer to return the instance. by andrew@webrtc.org · 11 years ago
  78. 90aa2fe Use ninja with merge_libs.py on Windows and clean up. by andrew@webrtc.org · 11 years ago
  79. c60d33b Force a memcpy directly from the AEC ring buffer. by andrew@webrtc.org · 11 years ago
  80. 86e2e1f Remove unneeded libvpx path from vp8 include_dirs. by andrew@webrtc.org · 11 years ago
  81. 101eb2c Refactor ring_buffer interface, add a feature and a test. by andrew@webrtc.org · 11 years ago
  82. 4211e6a New attempt at fixing hard-coded libvpx source. by phoglund@webrtc.org · 11 years ago
  83. bf03dd4 Revert "Fixing hard-coded libvpx source path." by phoglund@webrtc.org · 11 years ago
  84. 7f53b4c Fixing hard-coded libvpx source path. by phoglund@webrtc.org · 11 years ago
  85. 933af52 Ported assembly coding in APM from Android to iOS. by kma@webrtc.org · 11 years ago
  86. ca65c51 Handle multiple calls to set initial delay by mikhal@webrtc.org · 11 years ago
  87. c1f0f68 Remove WEBRTC_TRACE completely when tracing is disabled. by wjia@webrtc.org · 11 years ago
  88. fa9a633 Minor bug fix in maxFPS parameter declaration. by vikasmarwaha@webrtc.org · 11 years ago
  89. c1c5aad Fix for WebRTC Issue 1384. Some cameras return 0 fps for all capabilities which causes divide-by-zero. by vikasmarwaha@webrtc.org · 11 years ago
  90. 85e32df MIPS optimizations for Signal Processing Library patch01 by andrew@webrtc.org · 11 years ago
  91. ed1c7f4 AEC refactoring: Moved typedefs to _internal.h by bjornv@webrtc.org · 11 years ago
  92. 4f33453 Changing non-const reference arguments to pointers, ACM by tina.legrand@webrtc.org · 11 years ago
  93. db1733f Misc cleanups to webrtc/android code: by fischman@webrtc.org · 11 years ago
  94. c01c6c3 Refactoring AEC: AecCore struct made private by bjornv@webrtc.org · 11 years ago
  95. 325931a Refactor AEC: PowerLevel by bjornv@webrtc.org · 11 years ago
  96. 798195e Added a pointer getter to the system_delay variable. by bjornv@webrtc.org · 11 years ago
  97. 191efa0 Refactoring AEC: Added a SetConfigCore function by bjornv@webrtc.org · 11 years ago
  98. e27e49b Moved out buffer handling to ProcessFrame() by bjornv@webrtc.org · 11 years ago
  99. 5efb2ee 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 · 11 years ago
  100. 213217c Stop and restart fix. by mflodman@webrtc.org · 11 years ago