From b9e192b78a384ff13d87c606502373725042509c Mon Sep 17 00:00:00 2001 From: Chris Robinson Date: Thu, 5 Nov 2015 09:42:08 -0800 Subject: Implement a band-limited sinc resampler This is essentially a 12-point sinc resampler, unless it's resampling to a rate higher than the output, at which point it will vary between 12 and 24 points and do anti-aliasing to avoid/reduce frequencies going over nyquist. Code provided by Christopher Fitzgerald. --- utils/alsoft-config/mainwindow.cpp | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'utils/alsoft-config/mainwindow.cpp') diff --git a/utils/alsoft-config/mainwindow.cpp b/utils/alsoft-config/mainwindow.cpp index 75703f03..01f59e4b 100644 --- a/utils/alsoft-config/mainwindow.cpp +++ b/utils/alsoft-config/mainwindow.cpp @@ -61,7 +61,7 @@ static const struct { }; static const struct { - const char name[48]; + const char name[64]; const char value[16]; } speakerModeList[] = { { "Autodetect", "" }, @@ -87,10 +87,11 @@ static const struct { { "", "" } }, resamplerList[] = { { "Default", "" }, - { "Point (low quality, fast)", "point" }, + { "Point (low quality, very fast)", "point" }, { "Linear (basic quality, fast)", "linear" }, { "4-Point Sinc (good quality)", "sinc4" }, { "8-Point Sinc (high quality, slow)", "sinc8" }, + { "Band-limited Sinc (very high quality, very slow)", "bsinc" }, { "", "" } }, stereoModeList[] = { -- cgit v1.2.3