Re: Digital bit perfect ouptut with ALSA

[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]

 



Sergei Steshenko wrote:
> > The argument about 192 kHz is true, but why not run the DAC at the
> > 'normal' 96 kHz then? I would prefer a 48 -> 96 conversion over a 48 ->
> > 110 conversion!.
> > And how about interference: won't a 96 -> 110 conversion give any
> > interference at 14 kHz? Right in the audible band!
> > 
> > Matthijs
> > 
> 
> I think these guys running DAC at 110kHz simply do not understand what they
> are doing.
> 
> Upsampling makes sense for better analog reproduction, but when playing at
> CD sample rate (44.1kHz) they should have upsampled 2x to 88.2kHz - such
> upsampling does not introduce IMD.

Even 2x upsampling introduces distortion.  You need a sinc filter to
produce undistorted output even for simple integer ratios, and that's
physically unrealizable.  Polyphase sampling of 44.1 -> 110 kHz will
tend to introduce more distortion than a simple integer ratio, if
other parameters are constant (such as the filter base size), but with
a well-designed polyphase, it is possible that this energy is quite
low and/or concentrated in the band removed by the post-DAC analague
output filter, and that the higher sample rate allows the analogue
output filter to introduce less frequency ripple in the passed
sub-Nyquist band, making the overall balance for minimum distortion
favour the higher sample rate and polyphase upsampling.

>From their text, it sounds like that's the reason the DAC1 people say
they are doing what they do.

The other comment somebody made about "overclocking" and whether the
electronics degrades quality with that.  The DAC1 text implies they
looked at that, and chose a slightly higher sample rate which improves
the effect of the analogue output filter (perhaps more than
compensating for the polyphase effects), but not too high because of
the DAC itself not performing well above that.

I don't know if they did a good job, but it sounds plausible to me
that they could have done and their circuit may introduce less
analogue distortion overall than a 2x upsampler.  I.e. better.

On the other hand, if you care only about particular kinds of
distortion (such as polyphase artifacts) and don't care about others
(such as frequency ripple and roll-off near Nyquist), you wouldn't
want it.

But then there's people who say that all analogue output above 20kHz
or so is pointless anyway. :-)  (I have no opinion either way on this.
On the one hand, they could be right.  The upper auditory limit is
well known.  On the other hand, there is a possibility that IMD occurs
in the ear due to physical non-linearity, making it able to extract
information from higher frequencies in a rich sound field, even though
those frequencies cannot be heard by themselves).

-- Jamie

------------------------------------------------------------------------------
_______________________________________________
Alsa-user mailing list
Alsa-user@xxxxxxxxxxxxxxxxxxxxx
https://lists.sourceforge.net/lists/listinfo/alsa-user

[Index of Archives]     [ALSA Devel]     [Linux Audio Users]     [Fedora Users]     [Fedora Desktop]     [Fedora SELinux]     [Big List of Linux Books]     [Yosemite News]     [Yosemite Photos]     [KDE Users]     [Fedora Tools]

  Powered by Linux