Re: [Jack-Devel] Variable delay line problem
Thanks for the Matlab example that's the most explicit I've seen.
I guess I have the algo right (not 100% sure though) thanks to you and Chris.
Now my problem seems to be with the 'n' variable.
I understood from the Matlab example that it is simply a counter from delay buffer length to size of the audio file.
Now, I'm operating on jack buffer that are 512 samples long. So I can't simply rely on that.
How an I maintain such an increasing global var if my sound source is my computer line in for instance ?
Can I modulo it to something ?
Thank again
Le 14 avr. 2012 à 13:32, Friedrich Ewaldt <[hidden]> a écrit :
> Chris Caudle schrieb am 13.04.2012 23:29:
>> On Fri, April 13, 2012 11:52 am, Stéphane VAXELAIRE wrote:
>>> Thanks, I did try that, but when I do, the sine does not seem to evolve.
>>
>> Maybe I suggested the division in the incorrect location.
>> I tried the equivalent in Octave, and it seems to work with:
>> cos(n/Fs*2*pi*freq)
>>
>> I don't remember how operator order of precedence works in Octave, so I
>> also tried:
>> cos(2*pi*n*(freq/sample_rate))
>>
>> and that also seemed to work like expected.
>>
>> By work as expected, I just mean it generated cosine values of the
>> expected period. I did not try to actually implement a variable delay so
>> that I could make a flanger in Octave code.
>>
>
> If you are looking for a flanger implemented in Matlab (most probably works with litte or no changes also in Octave) you may take a look at e.g.
> http://www.mathworks.com/matlabcentral/fileexchange/6656-flange-effect
>
> or this nice presentation containing flanger Matlab code as well:
>
> http://www.cs.cf.ac.uk/Dave/CM0268/PDF/10_CM0268_Audio_FX.pdf
>
> or in general:
>
> http://www.google.com/search?q=flanger+octave+OR+matlab
>
>> Are you doing any prototype code in Octave, or is everything in C?
>>
>> Also, I didn't really pay attention to how you were implementing the
>> delay. I had assumed you would do an interpolated delay so you can make
>> smooth phase adjustments. If you are just grabbing samples from the delay
>> buffer, you are going to have discontinuous jumps between samples, and
>> that is going to sound really bad, just like having lots of under-runs
>> (same effect, you are just skipping over some samples and not playing
>> them).
>>
>
>
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