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gerrit-public.fairphone.software / platform / external / webrtc / a48dd43fe185ed6a826b7ce3bfee0b2187329159 / data
  1. e28dedf Remove old data files. by Rasmus Brandt · 6 years ago
  2. d661e9c WebRTC: Replace ProjectRootPath by ResourcePath by ehmaldonado · 8 years ago
  3. dedaf1c Modify audio_processing_unittest to use ResourcePath instead of ProjectRootPath. by ehmaldonado · 8 years ago
  4. 8e56521 The output signal of the AEC needs to be buffered as the by peah · 8 years ago
  5. 2ace3f9 The audio processing module (APM) relies on two for by peah · 8 years ago
  6. 58530ed Updating APM unittests on the echo metrics. by minyue · 8 years ago
  7. 53f7ada Add OWNERS file to data/audio_processing by henrik.lundin · 9 years ago
  8. b1fc54d Corrected the delay agnostic AEC behavior during periods of silent farend signal. by peah · 9 years ago
  9. eb3603b Don't always downsample to 16kHz in the reverse stream in APM by aluebs · 9 years ago
  10. 0bf612b This CL is partially reverting the effects that by peah · 9 years ago
  11. 40cbec5 Fix the number of frames used when interleaving in AudioBuffer::InterleaveTo() by Alejandro Luebs · 9 years ago
  12. df6416a Dont always downsample to 16kHz in the reverse stream in APM by aluebs · 9 years ago
  13. 776593b Reland: Drop the 16kHz sample rate restriction on AECM and zero out higher bands by aluebs · 9 years ago
  14. 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
  15. f687d53 Drop the 16kHz sample rate restriction on AECM and zero out higher bands by Alex Luebs · 9 years ago
  16. 4774874 Enable AudioProcessing48kHzSupport by default by Alejandro Luebs · 10 years ago
  17. b1786db audio_processing: Added a new AEC delay metric value that gives the amount of poor delays by bjornv@webrtc.org · 10 years ago
  18. f17ee9c Add case to ApmTest.Process to test the extended filter mode by aluebs@webrtc.org · 10 years ago
  19. 8328e7c Revert "Revert part of r7561, "Refactor audio conversion functions."" by andrew@webrtc.org · 10 years ago
  20. bcfb4d0 Revert part of r7561, "Refactor audio conversion functions." by kwiberg@webrtc.org · 10 years ago
  21. 4fc4add Refactor audio conversion functions. by andrew@webrtc.org · 10 years ago
  22. b6af428 Adjust speech probability in NS when echo by aluebs@webrtc.org · 10 years ago
  23. 30be827 Enable render downmixing to mono in AudioProcessing. by andrew@webrtc.org · 10 years ago
  24. a0ce9fa Call NS AnalyzeCaptureAudio before AEC by aluebs@webrtc.org · 10 years ago
  25. c98ce3b modules/audio_processing: Updates output_data_fixed.pb test file by bjornv@webrtc.org · 10 years ago
  26. 12cd443 Noise suppression: Change signature to work on floats instead of ints by kwiberg@webrtc.org · 10 years ago
  27. d5da250 Revert "Revert "Audio processing: Feed each processing step its choice by mflodman@webrtc.org · 11 years ago
  28. b1a66d1 Revert "Audio processing: Feed each processing step its choice of int or float data" by mflodman@webrtc.org · 11 years ago
  29. 934a265 Audio processing: Feed each processing step its choice of int or float data by kwiberg@webrtc.org · 11 years ago
  30. 382c0c2 Allow the RTP level indicator computation to work at any sample rate. by andrew@webrtc.org · 11 years ago
  31. bbd47fc Enables robust delay validation in AEC delay logging. by bjornv@webrtc.org · 11 years ago
  32. 3994e03 ADM no longer reads PCM files from data/audio_device. Now uses the resource folder instead. by henrika@webrtc.org · 12 years ago
  33. 6f90983 Remove data files from data/audio_coding by tina.legrand@webrtc.org · 12 years ago
  34. 08329f4 Added API to port internal speech probability in NS. by bjornv@webrtc.org · 12 years ago
  35. 9dc45da Move trunk/test/data -> trunk/data by andrew@webrtc.org · 13 years ago