*C++ developer of interactive audio applications using the JUCE framework
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*Fixed-term contract of 6 months starting from May 2019, possibility of
transformation into a permanent position*
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Pompidou
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
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~?
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
Hi,
I just found out that on Linux
pd -jack
with jack running at 96kHz
will display a samplerate of 44100Hz in its audio settings while
[samplerate~] reports the actual 96kHz. I understand that Pd just
accepts jack's samplerate but it would be friendly in terms of user
interface
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
