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gerrit-public.fairphone.software / platform / external / webrtc / b8996ddac0b17dae7c788e0ea1f4cf3df6bfdcb1 / modules / audio_processing / audio_buffer.h
  1. e35b32c AGC: Removing unnneccessary copying and changing to using const by Per Åhgren · 5 years ago
  2. 422b9e0 Run fullband processing at output rate on ARM by Gustaf Ullberg · 5 years ago
  3. 3e8bf28 Increase the maximum supported sample rate to 384000 Hz and add tests by Per Åhgren · 5 years ago
  4. 0aefbf0 Use the AEC3 high-pass filter for the whole APM by Per Åhgren · 5 years ago
  5. d47941e Reland "Simplification and refactoring of the AudioBuffer code" by Per Åhgren · 5 years ago
  6. f254e9e Revert "Simplification and refactoring of the AudioBuffer code" by Steve Anton · 5 years ago
  7. 81c0cf2 Simplification and refactoring of the AudioBuffer code by Per Åhgren · 5 years ago
  8. 928146f Removing all external access to the integer sample data in AudioBuffer by Per Åhgren · 5 years ago
  9. a135127 Remove all AudioBuffer code that is not related to storing audio data by Per Åhgren · 5 years ago
  10. a4d8737 Format almost everything. by Jonas Olsson · 5 years ago
  11. 200feba Make AEC3 the default desktop AEC option in WebRTC by Per Åhgren · 6 years ago
  12. 988cc08 [Cleanup] Add missing #include. Remove useless ones. by Yves Gerey · 6 years ago
  13. a12c42a Delete root header file typedef.h. by Niels Möller · 6 years ago
  14. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  15. bbf21a3 Remove dependencies on modules:module_api from AudioProcessing. by Fredrik Solenberg · 7 years ago
  16. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 7 years ago
  17. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  18. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/modules/audio_processing/audio_buffer.h]
  19. c7daea8 Make AudioBuffer::InterleaveTo const by kthelgason · 8 years ago
  20. a181c9a Keep track of the user-facing number of channels in a ChannelBuffer by Alejandro Luebs · 8 years ago
  21. 4a206a9 Remove webrtc::ScopedVector by kwiberg · 9 years ago
  22. 88788ad Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/ by kwiberg · 9 years ago
  23. 6955870 Convert channel counts to size_t. by Peter Kasting · 9 years ago
  24. ff761fb modules: more interface -> include renames by Henrik Kjellander · 9 years ago
  25. 98f5351 system_wrappers: rename interface -> include by Henrik Kjellander · 9 years ago
  26. dce40cf Update a ton of audio code to use size_t more correctly and in general reduce by Peter Kasting · 9 years ago
  27. 60d9b33 Integrate Intelligibility with APM by ekmeyerson · 9 years ago
  28. 86c6d33 Allow more than 2 input channels in AudioProcessing. by Michael Graczyk · 9 years ago
  29. 64e753c Revert of Allow more than 2 input channels in AudioProcessing. (patchset #13 id:240001 of https://codereview.webrtc.org/1226093007/) by magjed · 9 years ago
  30. c204754 Allow more than 2 input channels in AudioProcessing. by Michael Graczyk · 9 years ago
  31. 05c7605 Add resampling support in AudioBuffer::DeinterleaveFrom by Alejandro Luebs · 9 years ago
  32. 5a92aa8 Add 3-band filter-bank implementation by Alejandro Luebs · 10 years ago
  33. 3aca0b0 Add 48kHz support to Beamformer by aluebs@webrtc.org · 10 years ago
  34. 00b8f6b Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away by kwiberg@webrtc.org · 10 years ago
  35. d35a5c3 Make ChannelBuffer aware of frequency bands by aluebs@webrtc.org · 10 years ago
  36. 035e912 Move channel_buffer.{h,cc} to common_audio. by kjellander@webrtc.org · 10 years ago
  37. 27d106b Move the downmixing out of AudioBuffer by aluebs@webrtc.org · 10 years ago
  38. c5ebbd9 Support 48kHz in Noise Suppression by aluebs@webrtc.org · 10 years ago
  39. a7384a1 Simplify audio_buffer APIs by aluebs@webrtc.org · 10 years ago
  40. 8789376 Move ChannelBuffer class to channel_buffer file by aluebs@webrtc.org · 10 years ago
  41. 79b9eba Implement 3 band splitting filter bank by upsampling and splitting twice into 2 bands by aluebs@webrtc.org · 10 years ago
  42. 087da13 Add empty 3 band splitting filter API by aluebs@webrtc.org · 10 years ago
  43. be05c74 Wrap the splitting filter in its own class by aluebs@webrtc.org · 10 years ago
  44. bfacaab Add accessors for array of channel pointers in AudioBuffer. They are by claguna@google.com · 10 years ago
  45. e364ac9 AudioBuffer: Optimize const accesses to arrays that autoconvert int16<->float by kwiberg@webrtc.org · 10 years ago
  46. 2b6bc8d AudioBuffer: Eliminate the SplitChannelBuffer class by kwiberg@webrtc.org · 10 years ago
  47. 2561d52 Simplify AudioBuffer::mixed_low_pass_data API by aluebs@webrtc.org · 10 years ago
  48. fb2e7c2 Document that channels are stored contiguously in AudioBuffer by aluebs@webrtc.org · 10 years ago
  49. 38214d5 EchoCancellationImpl::ProcessRenderAudio: Use float samples directly by kwiberg@webrtc.org · 10 years ago
  50. c0035a6 Remove an optimization that's no longer worth the extra complexity it causes by kwiberg@webrtc.org · 10 years ago
  51. d5da250 Revert "Revert "Audio processing: Feed each processing step its choice by mflodman@webrtc.org · 10 years ago
  52. 21299d4 Remove the use of AudioFrame::energy_ from AudioProcessing and VoE. by andrew@webrtc.org · 10 years ago
  53. b1a66d1 Revert "Audio processing: Feed each processing step its choice of int or float data" by mflodman@webrtc.org · 10 years ago
  54. 934a265 Audio processing: Feed each processing step its choice of int or float data by kwiberg@webrtc.org · 10 years ago
  55. 4cc7636 AudioBuffer: Eliminate data_was_mixed_, and document what's left of data_ by kwiberg@webrtc.org · 11 years ago
  56. 65f9338 Fix constness of AudioBuffer accessors. by andrew@webrtc.org · 11 years ago
  57. 103657b Add keyboard channel support to AudioBuffer. by andrew@webrtc.org · 11 years ago
  58. ddbb8a2 Support arbitrary input/output rates and downmixing in AudioProcessing. by andrew@webrtc.org · 11 years ago
  59. 17e4064 Add a deinterleaved float interface to AudioProcessing. by andrew@webrtc.org · 11 years ago
  60. 7fad4b8 Include files from webrtc/.. paths in audio_processing/ by pbos@webrtc.org · 11 years ago
  61. 14b43be Move src/ -> webrtc/ by andrew@webrtc.org · 12 years ago[Renamed from src/modules/audio_processing/audio_buffer.h]
  62. 755b04a Add RMS computation for the RTP level indicator. by andrew@webrtc.org · 13 years ago
  63. 4d5d5c1 Reorganize the audio_processing source. by andrew@webrtc.org · 13 years ago[Renamed from src/modules/audio_processing/main/source/audio_buffer.h]
  64. ed083d4 Modify the _vadActivity member of the AudioFrame passed to AudioProcessing. by andrew@webrtc.org · 13 years ago
  65. 470e71d by niklase@google.com · 13 years ago