Wow!! That one line did it thanks so much.
On Wed, Mar 27, 2019 at 10:09 PM Charles Z Henry wrote:
> fexpr~ ($y1+1)*($y1<2);
> $x1*($y1==0)+$x2*($y1==1)+$x3*($y1==2)
>
> might have a *lot* of high frequency content.
>
> you could replace the counter in the above fexpr~ with a slower
> counter
fexpr~ ($y1+1)*($y1<2);
$x1*($y1==0)+$x2*($y1==1)+$x3*($y1==2)
might have a *lot* of high frequency content.
you could replace the counter in the above fexpr~ with a slower
counter and it would have a coarser effect. That would reduce the
pitch of the noise. If the counter got slow enough, it w
there's another option you're missing which doesn't involve storage
When you do your interleaving in a literal way, think of it as the sum
of 3 signals:
A,0,0,B,0,0,C,0,0,...
0,1,0,0,2,0,0,3,0,...
0,0,!,0,0,@,0,0,#,...
Those signals are versions of the original, except pitched down by
factor of 3
yeah, or “sample” the input signals at samplerate/3 … which is what i guess
would be the way to go.
or just only use every 3rd sample from each stream to generate the output.
> On 26 Mar 2019, at 12:59, IOhannes m zmölnig wrote:
>
> On 3/26/19 11:11 AM, RT wrote:
>> In that case
>
> the probl
On 3/26/19 11:11 AM, RT wrote:
> In that case
the problem being, that "that case" is *the* case for a realtime system
(like Pd).
> I would think being able to overwrite the current / played 3
> seconds of memory with the next 3 seconds of memory would be needed
> tabwrite~ tabread~ tabwrite~?
i'
In that case I would think being able to overwrite the current / played 3
seconds of memory with the next 3 seconds of memory would be needed
tabwrite~ tabread~ tabwrite~?
On Tue, Mar 26, 2019 at 4:45 AM IOhannes m zmoelnig wrote:
> On 26.03.19 01:08, RT wrote:
> > I expect some-type of delay be
On 26.03.19 01:08, RT wrote:
> I expect some-type of delay because of processing but each of the 3 signals
i guess peter's question was more along the lines:
assume your soundcard is set to a sample rate of 44.1kHz.
therefore, each of your three signals will create 44100 samples per second.
if you
I expect some-type of delay because of processing but each of the 3 signals
will be generated using OSC~ (See patch or included image with original
message showing possible logic). I guess you could look at it as taking
in audio from 3 microphones and processing the sound on each mic then
outputt
* RT [2019-03-25 18:57]:
> I have a Pd patch with 3 separate signals and I want to sequentially weave
> the signals together so instead of having three separate signals I create 1
> large signal that is created by weaving 3 signals together.
>
> Example: (note: the commas are just used as separat
What comes to my mind is to write the three signals on three new arrays
which are three times the size of the original ones. These arrays would
have each sample from the original signal zero-padded by two samples. That
can be done by reading the original signals with tabread~ at a rate which
is 1/3
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