Also, make sure that the Realtek chip does native 44.1...<br>I feed my MOTU card from the SPDIF output of a Creative Audigy* and I had to change my workflow to 48k because the Audigy turned out to "fake" 44.1 by constant upsampling/downsampling around its fixed-48k DSP.<br>
<br>* because the FireWire connection is so fragile and the damn MOTU freezes all the time<br><br>Andras<br><br><br><div class="gmail_quote">On Fri, Nov 11, 2011 at 04:38, Ingo <span dir="ltr"><<a href="mailto:ingo@miamiwave.com">ingo@miamiwave.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">I think if you are using a mixed patch with osc and samples only the samples<br>
would play correctly and any osc~ might get detuned. Not sure?<br>
<br>
But anyway, as far as I know you can only up or downsample by the power of<br>
two.<br>
<br>
I did try to use a frequency multiplier of 0.91875 and it was playing in<br>
tune but that still gave me a samplerate of 48k at the SPDIF out instead of<br>
the 44.1k which was what I wanted.<br>
<br>
Anyway, I suspect that some instabilities as well as certain losses of sound<br>
quality may happen if the sample rate of Pd conflicts with the sample rate<br>
of the sound card. This is because of the resampling that the sound card<br>
needs to do. So getting Pd's sample rate matched up with the rate of the<br>
hardware might have advantages even if you are using only analogue outs.<br>
<br>
Ingo<br>
<br>
<br>
2011/11/10 Roman Haefeli <<a href="mailto:reduzent@gmail.com">reduzent@gmail.com</a>><br>
On Thu, 2011-11-10 at 17:14 +0100, tim vets wrote:<br>
> ..<br>
> Lastly: I wonder if there isn't a way to downsample some subpatches to<br>
> playback the 44.1kHz soundfiles in a 48kHz environment?<br>
<br>
Why would you want to run an [osc~ 440] at a different samplerate, when<br>
it plays a 440Hz anyway?<br>
<br>
Regarding audio samples, you can use [tabread4~] fed by a [line~]<br>
instead of [tabplay~] for up- or downsampling.<br>
<br>
<br>
it was just a thought, I can imagine if you would have based some<br>
sophisticated sample playback on a whole bunch of tabplay~'s or readsf~'s,<br>
that maybe you wouldn't want or have time to change all that...<br>
I checked [switch~] again and indeed you can enter 0.5 to downsample by<br>
factor 2, does that mean you could enter 0.91875000000000007 to downsample<br>
from 48 to 44.1?<br>
Roman<br>
<br>
<br>
<br>
<br>
</blockquote></div><br>