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.

1 files changed, 3 insertions, 2 deletions

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[] = {