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

[Holger Marzen]

On Mon, 24 Sep 2018, David Kastrup wrote:

> Holger Marzen <holger at marzen.de> writes:
> 
> > 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
> here.

That was my first approach but I couldn't find an affordable stereo EQ.
Most affordable devices are 2x15 or 2x31 and not 15 or 31 band stereo.
I don't want to set the channels seperately.

> > 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?

The output of the first adapter is used for
- direct monitoring
- headphone monitoring and mixing/mastering, but without EQ

> > 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.

Or you can simply wait til the "right" moment und use the original
values. But that would not use more CPU depending on the quality
setting. So there seems to bee an interpolation happening.

It looks as if I have to do a looptest (play through the 2nd interface
and record with the 1st) to get an impression how's the signal distorted
at the different quality settings.