Gitiles
Code Review Sign In
gerrit.openfyde.cn / webrtc.googlesource.com / src / 43cb716e55eab1cf1ae8afd2ac79b51a604d0fa5 / webrtc / modules / audio_processing / include / audio_processing.h
  1. b829d9f Add AudioOption for residual echo detector, and enable the echo detector by default on non-mobile platforms. by ivoc · 9 years ago
  2. 7aba029 Make use of new APM statistics interface. by ivoc · 9 years ago
  3. d0a151c Update default values for APM stats to match old behavior. by ivoc · 9 years ago
  4. 3e9a537 Original CL: https://codereview.webrtc.org/2433153003/, commit 8b8d3e4c30e8ea3846b58dfd36d1fd35a7799df4. by ivoc · 9 years ago
  5. 9f4a4a0 Add empty residual echo detector. by ivoc · 9 years ago
  6. 48dfab5 Revert of New statistics interface for APM (patchset #11 id:200001 of https://codereview.webrtc.org/2433153003/ ) by ivoc · 9 years ago
  7. 8b8d3e4 New statistics interface for APM by ivoc · 9 years ago
  8. 73a28ee The AudioProcessing class is used as an interface by peah · 9 years ago
  9. c19f312 This CL adds functionality in the level controller to by peah · 9 years ago
  10. de65ddc This CL renames variables and method and removes some one-line by peah · 9 years ago
  11. d59d3bb Replace a DCHECK with static_assert by kwiberg · 9 years ago
  12. 88ac853 The current scheme for setting parameters and specifying the by peah · 9 years ago
  13. 10f606d Revert of Introduced new scheme for controlling the functionality inside the audio processing module (patchset #12 id:260001 of https://codereview.webrtc.org/2292863002/ ) by kjellander · 9 years ago
  14. c8bbe3f The current scheme for setting parameters and specifying the behavior by peah · 9 years ago
  15. 88499ec Moving/renaming webrtc/common.h. by solenberg · 9 years ago
  16. 7d67e45 Revert of Added functionality for specifying the initial signal level to use for the gain estimation in the l… (patchset #8 id:160001 of https://codereview.webrtc.org/2254973003/ ) by peah · 9 years ago
  17. 57fec1d This CL adds functionality in the level controller to by peah · 9 years ago
  18. 5f09980 Removed inline definitions and added destructors to fix chromium-style. by aleloi · 9 years ago
  19. f4022ff Pull out the PostFilter to its own NonlinearBeamformer API by Alejandro Luebs · 9 years ago
  20. ca4cac7 New module for the adaptive level controlling functionality in the audio processing module by peah · 9 years ago
  21. a3c51ea Revert "Pull out the PostFilter to its own NonlinearBeamformer API" by Alejandro Luebs · 9 years ago
  22. b983112 Pull out the PostFilter to its own NonlinearBeamformer API by Alejandro Luebs · 9 years ago
  23. c9b0c26 Surface the IntelligibilityEnhancer on MediaConstraints by Alejandro Luebs · 9 years ago
  24. 3815655 Change aggregation window of aecDivergentFilterFraction to 1 second. by minyue · 9 years ago
  25. 0332c2d Added support in the AEC for refined filter adaptation. by peah · 9 years ago
  26. 5045337 Pulling AEC divergent filter fraction. by minyue · 9 years ago
  27. b031955 Deprecate AudioProcessing::AnalyzeReverseStream(AudioFrame) API by aluebs · 9 years ago
  28. da116c4 Use ProcessReverseStream in VoiceEngines OutputMixer by aluebs · 9 years ago
  29. 776593b Reland: Drop the 16kHz sample rate restriction on AECM and zero out higher bands by aluebs · 9 years ago
  30. dfc2870 Revert of Drop the 16kHz sample rate restriction on AECM and zero out higher bands (patchset #3 id:40001 of https://codereview.webrtc.org/1774553002/ ) by perkj · 9 years ago
  31. f687d53 Drop the 16kHz sample rate restriction on AECM and zero out higher bands by Alex Luebs · 9 years ago
  32. 6ebc4d3 Changed name for the upcoming AEC from NextGenerationAec to AEC3. by peah · 9 years ago
  33. 50e21bd This CL introduces namespaces in the aec c++ files by peah · 9 years ago
  34. a332e2d Added boilerplate code for being able to test the upcoming AEC functionality. by peah · 9 years ago
  35. fa639f0 Surface the noise estimate of the NS to be used by other components by Alejandro Luebs · 9 years ago
  36. d66b44d Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87. by ivoc · 10 years ago
  37. 688e308 Re-land: "Use an explicit identifier in Config" by aluebs · 10 years ago
  38. fca54f4 Revert of Use an explicit identifier in Config (patchset #4 id:60001 of https://codereview.webrtc.org/1538643004/ ) by tommi · 10 years ago
  39. 25249d9 Use an explicit identifier in Config by aluebs · 10 years ago
  40. 6955870 Convert channel counts to size_t. by Peter Kasting · 10 years ago
  41. b2328d1 Remove additional channel constraints when Beamforming is enabled in AudioProcessing by aluebs · 10 years ago
  42. a4df27b Revert of Reland "Added option to specify a maximum file size when recording an AEC dump." (patchset #2 id:20001 of https://codereview.webrtc.org/1541633002/ ) by ivoc · 10 years ago
  43. f4f5cb0 Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87. by ivoc · 10 years ago
  44. 36d4c54 Revert of Added option to specify a maximum file size when recording an AEC dump. (patchset #5 id:120001 of https://codereview.webrtc.org/1413483003/ ) by ivoc · 10 years ago
  45. ae2c5ad Added option to specify a maximum file size when recording an AEC dump. by ivoc · 10 years ago
  46. 66085be Bugfix that fixes the error where the audio processing module is called by peah · 10 years ago
  47. cb3f9bd Make the nonlinear beamformer steerable by Alejandro Luebs · 10 years ago
  48. 5aaa9b4 Removed unused API functions in AudioProcessing and AudioProcessingModule by peah · 10 years ago
  49. cdfe20b Fix the maximum native sample rate in AudioProcessing by Alejandro Luebs · 10 years ago
  50. dce40cf Update a ton of audio code to use size_t more correctly and in general reduce by Peter Kasting · 10 years ago
  51. 60d9b33 Integrate Intelligibility with APM by ekmeyerson · 10 years ago
  52. b3b79b6 Clean up the Config to enable 48kHz support in AudioProcessing by aluebs · 10 years ago
  53. 86c6d33 Allow more than 2 input channels in AudioProcessing. by Michael Graczyk · 10 years ago
  54. 64e753c Revert of Allow more than 2 input channels in AudioProcessing. (patchset #13 id:240001 of https://codereview.webrtc.org/1226093007/) by magjed · 10 years ago
  55. c204754 Allow more than 2 input channels in AudioProcessing. by Michael Graczyk · 10 years ago
  56. 4e7aa43 audio_processing: Adds two UMA histograms logging delay jumps in AEC by Bjorn Volcker · 10 years ago
  57. 894ad94 Fix occurrences of const typed declaration without initialization by eblima · 10 years ago
  58. 366e952 Follow-up: Remove old ReportedDelay AEC config by henrik.lundin · 10 years ago
  59. 0f133b9 Rename APM Config ReportedDelay to DelayAgnostic by henrik.lundin · 10 years ago
  60. b02af18 Follow-up: Remove old DelayCorrection AEC config by Henrik Lundin · 10 years ago
  61. 441f634 Re-land r9378 "Rename APM Config DelayCorrection to ExtendedFilter" by Henrik Lundin · 10 years ago
  62. 3fbf3f8 Revert r9378 "Rename APM Config DelayCorrection to ExtendedFilter" by Henrik Lundin · 10 years ago
  63. 5f4b7e2 Rename APM Config DelayCorrection to ExtendedFilter by Henrik Lundin · 10 years ago
  64. 4774874 Enable AudioProcessing48kHzSupport by default by Alejandro Luebs · 10 years ago
  65. fb49451 Disables mic bump-up level if not built with chromium by Bjorn Volcker · 10 years ago
  66. adc46c4 audio_processing/agc: Adds config to set minimum microphone volume at startup by Bjorn Volcker · 10 years ago
  67. dfa3605 Reparent Nonlinear beamformer under beamforming interface. by Michael Graczyk · 10 years ago
  68. 0f663de Rename Beamformer to NonlinearBeamformer. by mgraczyk@chromium.org · 10 years ago
  69. 1d88394 Add support for arbitrary array geometries in Beamformer by aluebs@webrtc.org · 10 years ago
  70. c9ce07e Add Config option to enable 48kHz support in AudioProcessing by aluebs@webrtc.org · 10 years ago
  71. b1786db audio_processing: Added a new AEC delay metric value that gives the amount of poor delays by bjornv@webrtc.org · 10 years ago
  72. d82f55d Only adapt AGC when the desired signal is present by aluebs@webrtc.org · 11 years ago
  73. 46323b3 Remove useless AudioProcessing::Create() overload. by andrew@webrtc.org · 11 years ago
  74. fb7a039 Use array geometry in Beamformer by aluebs@webrtc.org · 11 years ago
  75. ae643ce Wire up Beamformer in AudioProcessing by aluebs@webrtc.org · 11 years ago
  76. 087da13 Add empty 3 band splitting filter API by aluebs@webrtc.org · 11 years ago
  77. e46bc77 Reland 28629004: adding new AEC dump start interface for chrome. by xians@webrtc.org · 11 years ago
  78. 79a7148 Revert 7337 "Reland 28629004: adding new AEC dump start interfac..." by xians@webrtc.org · 11 years ago
  79. 7aad5e5 Revert 7338 "Fixed the android build by making the interface pur..." by xians@webrtc.org · 11 years ago
  80. 90d1979 Fixed the android build by making the interface pure virtual. by xians@webrtc.org · 11 years ago
  81. 14092e0 Reland 28629004: adding new AEC dump start interface for chrome by xians@webrtc.org · 11 years ago
  82. 8752061 Revert 7334 "adding new AEC dump start interface for chrome." by xians@webrtc.org · 11 years ago
  83. 2e417d6 adding new AEC dump start interface for chrome. by xians@webrtc.org · 11 years ago
  84. 4065988 Remove unused ExperimentalNS API in AudioProcessing by aluebs@webrtc.org · 11 years ago
  85. 224a140 Make experimental NS API not purely virtual by aluebs@webrtc.org · 11 years ago
  86. 9825afc Add ExperimentalNs support in Config by aluebs@webrtc.org · 11 years ago
  87. 3f83072 modules/audio_processing: Adds a config for reported delays by bjornv@webrtc.org · 11 years ago
  88. 382c0c2 Allow the RTP level indicator computation to work at any sample rate. by andrew@webrtc.org · 11 years ago
  89. e44a84d Only clamp to 16 kHz when AECM is enabled. by andrew@webrtc.org · 11 years ago
  90. 82a045a APM: limit native sample rate to 16kHz on mobile. by fischman@webrtc.org · 11 years ago
  91. 46b31b1 Restore sample_rate_hz() until Chromium is updated to not use it. by andrew@webrtc.org · 11 years ago
  92. ddbb8a2 Support arbitrary input/output rates and downmixing in AudioProcessing. by andrew@webrtc.org · 11 years ago
  93. 40ee3d0 Consolidate audio conversion from Channel and TransmitMixer. by andrew@webrtc.org · 11 years ago
  94. a8b9737 Add tests and modify tools for new float deinterleaved interface. by andrew@webrtc.org · 11 years ago
  95. 17e4064 Add a deinterleaved float interface to AudioProcessing. by andrew@webrtc.org · 11 years ago
  96. 56e4a05 Remove ProcessingComponent's dependence on AudioProcessingImpl. by andrew@webrtc.org · 11 years ago
  97. f92aaff AudioProcessing is not a Module. by andrew@webrtc.org · 11 years ago
  98. 17342e5 Add a method to inform AudioProcessing that its output will be muted. by andrew@webrtc.org · 11 years ago
  99. 75dd288 Add an interface for accepting keypress signals to AudioProcessing. by andrew@webrtc.org · 11 years ago
  100. 5474491 Update AudioProcessing::Create docs. by andrew@webrtc.org · 11 years ago