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gerrit-public.fairphone.software / platform / external / webrtc / 0d8d01001745561580fa39ffabbcdb1aba8443c3
  1. 0d8d010 Handle multiple calls to set initial delay by mikhal@webrtc.org · 11 years ago
  2. 7bf7326 Remove WEBRTC_TRACE completely when tracing is disabled. by wjia@webrtc.org · 11 years ago
  3. 10987a8 Minor bug fix in maxFPS parameter declaration. by vikasmarwaha@webrtc.org · 11 years ago
  4. bf3a9b3 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
  5. 5140e24 MIPS optimizations for Signal Processing Library patch01 by andrew@webrtc.org · 11 years ago
  6. 60f8313 AEC refactoring: Moved typedefs to _internal.h by bjornv@webrtc.org · 11 years ago
  7. 7a7a008 Changing non-const reference arguments to pointers, ACM by tina.legrand@webrtc.org · 11 years ago
  8. f61e02c Misc cleanups to webrtc/android code: by fischman@webrtc.org · 11 years ago
  9. 56a9ec3 Refactoring AEC: AecCore struct made private by bjornv@webrtc.org · 11 years ago
  10. 71e91f3 Refactor AEC: PowerLevel by bjornv@webrtc.org · 11 years ago
  11. 4d1cfae Added a pointer getter to the system_delay variable. by bjornv@webrtc.org · 11 years ago
  12. 47b274d Refactoring AEC: Added a SetConfigCore function by bjornv@webrtc.org · 11 years ago
  13. 716fd90 Moved out buffer handling to ProcessFrame() by bjornv@webrtc.org · 11 years ago
  14. 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 · 11 years ago
  15. 59b2d5f Stop and restart fix. by mflodman@webrtc.org · 11 years ago
  16. eb7ebf2 Revert 3543 by tina.legrand@webrtc.org · 11 years ago
  17. 374aa49 Changing non-const reference arguments to pointers, ACM by tina.legrand@webrtc.org · 11 years ago
  18. 0b6293a Fixed typo in vie_autotest_loopback.cc. by pbos@webrtc.org · 11 years ago
  19. 83663ef Replace gtest_prod.h include with our own FRIEND_TEST macro. by andrew@webrtc.org · 11 years ago
  20. aea96d3 Rename webrtc::StatsObserver to webrtc::CallStatsObserver by fischman@webrtc.org · 11 years ago
  21. 0a480cb Added getter for far_time_buf in AEC. Only used in AEC debug dump. by bjornv@webrtc.org · 11 years ago
  22. 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 · 11 years ago
  23. cea70f4 * Name change * Removed WebRtcAec_ function name prepending on private function. by bjornv@webrtc.org · 11 years ago
  24. 95b48c3 Update to codec unit test: by marpan@webrtc.org · 11 years ago
  25. 77fced3 fixing nack list size calculation by mikhal@webrtc.org · 11 years ago
  26. 10741b3 Updated version number to 3.24 by elham@webrtc.org · 11 years ago
  27. 1682f71 Updating watchlist by mikhal@webrtc.org · 11 years ago
  28. 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 · 11 years ago
  29. 0460c72 Remove the dependency on dxguid.lib. by tommi@webrtc.org · 11 years ago
  30. d2c3bed Move directx_sdk_path definition variable into the video_render_module gyp file. by tommi@webrtc.org · 11 years ago
  31. eb91792 Refactoring temporal layers implementation and adding VideoCodecMode for easier control of codec settings. by stefan@webrtc.org · 11 years ago
  32. 3897255 Add VoE interface to VieRTP test by mikhal@webrtc.org · 11 years ago
  33. e3d6ffe Increase threshold in codec unit test. by marpan@webrtc.org · 11 years ago
  34. ef9f76a Adding a receive side API for buffering mode. by mikhal@webrtc.org · 11 years ago
  35. 47fe573 Bug fix for webrtc issue 1391. Typo in sin_length for socket address. by vikasmarwaha@webrtc.org · 11 years ago
  36. b4cd342 This refactoring CL contains an API to get low level echo metrics stats. by bjornv@webrtc.org · 11 years ago
  37. 21a2fc9 This Cl includes by bjornv@webrtc.org · 11 years ago
  38. 325f625 Moved the actual calculations to aec_core to avoid passing up low level members. by bjornv@webrtc.org · 11 years ago
  39. 0989fb7 Make VoiceEngineImpl inherit from VoiceEngine. by tommi@webrtc.org · 11 years ago
  40. 1723857 Removed astyle from webrtc_reformat since clang-format-chrome.py handles that now. by phoglund@webrtc.org · 11 years ago
  41. 076fc12 Modify SincResampler to build in webrtc. by andrew@webrtc.org · 11 years ago
  42. 6f6acd9 Duplicated sampling frequency multiplier to aecpc_t struct to avoid a getter. by bjornv@webrtc.org · 11 years ago
  43. 4013ac4 Roll Chromium revision 176094:182149 by kjellander@webrtc.org · 11 years ago
  44. 7267ffd Moved debug file handling to aec_core from echo_cancellation.c. This removes dependency on low level member variables. by bjornv@webrtc.org · 11 years ago
  45. 3e10249 Added delay estimation test to audio processing unit tests. by bjornv@webrtc.org · 11 years ago
  46. e580be9 Add regression monitoring for audioproc and iSAC fixed-point tests. by kjellander@webrtc.org · 11 years ago
  47. 07b667d Remove MultiStreamMode from test. by stefan@webrtc.org · 11 years ago
  48. 294e5b0 Reset ssrc when calling SetSendCodec. by mflodman@webrtc.org · 11 years ago
  49. a092cbf Fixing lint warnings from previous commit by tina.legrand@webrtc.org · 11 years ago
  50. 45eab19 Import stringize_macros from Chromium. by andrew@webrtc.org · 11 years ago
  51. a8ef811 Import SincResampler from Chromium. by andrew@webrtc.org · 11 years ago
  52. 9c4e662 Fix Windows x64 errors in video_codecs_test_framework by kjellander@webrtc.org · 11 years ago
  53. 6388c3e Implement initial delay. This CL allows clients of VoE to set an initial delay. Playout of audio is delayed and the extra playout delay is maintained during the call. While packets are buffered (in NetEq) to acheive the desired delay. ACM will playout silence (zeros). Initial delay has to be set before any packet is pushed into ACM. by turaj@webrtc.org · 11 years ago
  54. e6e344a Sync libvpx and its gyp wrapper from Chromium. by andrew@webrtc.org · 11 years ago
  55. 0ee57c2 Increase maximum resolution to 4k x 3k. by fbarchard@google.com · 11 years ago
  56. 57a0049 VCM: Removing frame drop enable from Reset call BUG = 1387 by mikhal@webrtc.org · 11 years ago
  57. 18a21a0 Android NDK build tools by kjellander@webrtc.org · 11 years ago
  58. 00ab7cf Fix perf output for audioproc and iSAC fixed-point tests by kjellander@webrtc.org · 11 years ago
  59. 0cb48a0 Set SingleStream BWE in unittests. by stefan@webrtc.org · 11 years ago
  60. 63066f7 Set qpMax to 56 in for all VP8 tests. Fixes buildbot breakage. by stefan@webrtc.org · 11 years ago
  61. 3d305c6 Updates to send side streaming mode: by mikhal@webrtc.org · 11 years ago
  62. 7948147 Update version number to 3.23 by tnakamura@webrtc.org · 11 years ago
  63. 7c85074 Adding third_party/directx and winsdk_samples to svn:ignore by kjellander@webrtc.org · 11 years ago
  64. 687efe3 Adding third_party/opus to svn:ignore by kjellander@webrtc.org · 11 years ago
  65. b64732a Fix Win64 build breakage by henrikg@webrtc.org · 11 years ago
  66. 147c73e Made it possible to render custom call output to file. by phoglund@webrtc.org · 11 years ago
  67. d83b9fd Fixed a bug in iSAC transform functions on ARM-Neon platform. Performance unchanged. by kma@webrtc.org · 11 years ago
  68. 4fd5527 Don't report an error for GetEstimatedReceiveBandwidth if there is no valid by mflodman@webrtc.org · 11 years ago
  69. fe3d606 Enable indefinitely running vie_auto_test option by kjellander@webrtc.org · 11 years ago
  70. 1e7ed7a Use LOG_F interface for unsupported functions. by andrew@webrtc.org · 11 years ago
  71. 959da8d Added labels in transform_neon.S in iSAC-fix, so the tables be shared with other files in iOS build. Also, moved several code lines in the same file, in case register values cannot be preserved after a function call which could cause a crash in some platforms (e.g. iOS etc.). by kma@webrtc.org · 11 years ago
  72. a7303bd Lint-cleaned video and audio receivers. by phoglund@webrtc.org · 11 years ago
  73. c4e45f6 Updated version number to 3.22 by elham@webrtc.org · 11 years ago
  74. 23e3559 Updating Perf numbers for Win Large Test. by tina.legrand@webrtc.org · 11 years ago
  75. 244251a Moved almost all payload-related stuff to the payload registry. by phoglund@webrtc.org · 11 years ago
  76. fa53d87 Fixing/disabling Windows x64 warnings by kjellander@webrtc.org · 11 years ago
  77. 254d85a Exchange TRY by enumerating image formats in Linux video capture by braveyao@webrtc.org · 11 years ago
  78. 6ed8ebc Fix MaxChannels test; 32 -> 100. by andrew@webrtc.org · 11 years ago
  79. 4a6f62d Remove (in practice) the voice engine channel limit. by andrew@webrtc.org · 11 years ago
  80. dbe97d2 Adding a send side API for streaming by mikhal@webrtc.org · 11 years ago
  81. becf9c8 Fix mismatch between different NACK list lengths and packet buffers. by stefan@webrtc.org · 11 years ago
  82. b586507 Break out RemoteBitrateEstimator from RtpRtcp module and make RemoteBitrateEstimator::Process trigger new REMB messages. by stefan@webrtc.org · 11 years ago
  83. 46d90dc Adding three frame sizes to Opus by tina.legrand@webrtc.org · 11 years ago
  84. d087789 Adjusted net_50_5_plr_5 on Linux, removed all gilbert_elliot metrics (too flaky), added mac expectations. by phoglund@webrtc.org · 11 years ago
  85. aaad613 Implementing stereo support for G.722 by henrik.lundin@webrtc.org · 11 years ago
  86. 7050f96 Set frame length for frame converting in external renderer by braveyao@webrtc.org · 11 years ago
  87. ac46c6d Replaced relative path to reference from <(webrtc_root). by bjornv@webrtc.org · 11 years ago
  88. 9d532fd Fix propagating RED paylaod-type to ACM. by turaj@webrtc.org · 11 years ago
  89. 763faea Removing a codec from NetEq database has a bug. |funcDurationEst| is not updated. by turaj@webrtc.org · 11 years ago
  90. c0ada86 fix for issue 281. by turaj@webrtc.org · 11 years ago
  91. 8c8ad85 fix issue 1322, accept -1 as default payload-type for redundant coding (FEC). by turaj@webrtc.org · 11 years ago
  92. 119c67d Adding a max jitter filter to the JB estimate - allowing two modes, one will return the last estimate (current setting), and another will return the max value seen, and allow setting an initial value. by mikhal@webrtc.org · 11 years ago
  93. e07c661 VP8: Making key frame interval a tunnable parameter by mikhal@webrtc.org · 11 years ago
  94. 6e3968f Fix NetEq4 unit tests for VS2012 by henrik.lundin@webrtc.org · 11 years ago
  95. 73deaad Removing a hack for CNG by henrik.lundin@webrtc.org · 11 years ago
  96. 96a08ce Fixed stale regression values and calibrated some vie_auto_test values. by phoglund@webrtc.org · 11 years ago
  97. ac59dba Adding iSAC-fb support by henrik.lundin@webrtc.org · 11 years ago
  98. 3d13d9f Fix audio_e2e_test command line arguments by kjellander@webrtc.org · 11 years ago
  99. 73a702c This is a change in the iOS audio device to use VoiceProcessingIO API instead of RemoteIO. This way we don't need to use WebRTC EC and NS because it happens on the device hardware. by andrew@webrtc.org · 11 years ago
  100. 7ded92b Re-committing r3428 by bjornv@webrtc.org · 11 years ago