[Jack-Devel] audioadapter: quality setting important when not changing sampling frequency?

David Kastrup dak at gnu.org
Mon Sep 24 13:51:12 CEST 2018

Holger Marzen <holger at marzen.de> writes:

> Hi all,
> since I only need one microphone when recording I am happy with my
> Focusrite Scarlett 2i4. I use the 2i4 for recording and direct
> monitoring with the line outputs feeding a headphones amp. I use the
> builtin headphones amp as well, both when recording and for mixing.
> So far, so good.
> Then I wanted 2 additional outputs to feed my active speakers (Neumann
> KH120A plus KH805 sub). Because of my room's damping of the high
> frequencies (at least compared to my headphones AKG K812) I need an EQ
> before them. So I use a Scarlett 2i2 for this purpose, with CALF
> Jackhost and a Calf EQ feeding this outputs.

This may sound defaitist but I'd lean towards just using an analog EQ

> Now I have 2 audio interfaces that can't be synchronized with a
> wordclock cable. So I have to drive the 2i2 for the speakers in an async
> way. Since I use this output only for mixing/mastering I can live with
> some additional latency.
> I tried several ways to use the 2nd interface:
> - alsa_out
> - zita_j2a
> - audioadapter (jack_load -i "-dhw:2 -q0 -n2 -p128 -r48000" 2i2 audioadapter)
> The audioadapter is the most robust solution and doesn't crash when I
> resize jackd's bufsize on the fly, so I'd like to stay with it.

Wait, I am missing something here.  "since I use this output only for
mixing/mastering".  How does your first audio adapter even come into
play here?

> As I played around with the quality setting (-q0 ... -q4) I noticed an
> enormous CPU load with -q4 (sinc) but couldn't hear any differences in
> audio quality. -q0 sounds as good as -q4.
> Now my question:
> When the audioadapter has to resample to the same frequency (from 48000
> to 48000), do higher -q settings improve the audio quality? As I
> understand there is no interpolation needed at all so -q0 would be
> enough to have a perfect audio quality.
> Any comments/ideas to this?

Interpolation will be needed when the clocks drift.  Basically you need
to delay by fractional sample amounts and the kind of interpolation used
for that makes a difference with regard to the precision particularly at
comparatively high frequencies.  Simple forms of interpolation will
cause fluctuations in the high frequency response.  With linear
interpolation, the frequency response will fluctuate between accurate
and a cosine bell shape.  That would be rather extreme in the high
range, so you want better than that.  The quality of interpolation will
be mostly audible in the high range, though, as long as you _do_
interpolate rather than skipping or duplicating samples (which would
result in clicks across the spectrum).

David Kastrup

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