>
> I was thinking of taking freeverb and removing the comb
> filters and just using the allpass filters with 100% feedback thinking
> it would make it a "cleaner" sustain.
>
I think that's worth a try. Using comb filters would be problematic in
your case because the harmonics of the guitar
Wow, thanks all for the replies!
If I used a reverb would it end up giving the sustained voice too much
character? I was thinking of taking freeverb and removing the comb
filters and just using the allpass filters with 100% feedback thinking
it would make it a "cleaner" sustain. I'm leaning this
Simple OLA will produce warbles. I recommend a phase vocoder.
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Essentially, what you want is a "sustain" effect?
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Am 16.09.2016 um 19:30 schrieb Spencer Jackson:
On Fri, Sep 16, 2016 at 11:24 AM, gm wrote:
Did you consider a reverb or an FFT time stretch algorithm?
I haven't looked into an FFT algorithm. I'll have to read up on that,
but what do you mean with reverb? Would you
Probably shouldn't reveal too much of the detail, but it likely comes as
no surprise that the tradeoff between time and frequency resolution is
critical in systems that have limited CPU horsepower. The FFTs in this
case do run in the audio processing thread which is synchronous to the
buffer
Yup, I was asking because I found that on the M4 it is difficult to do proper
overlap and add while keeping reasonably small block sizes for FFTs of useful
length, so I was just curious whether you had to implement threads or just went
for block sizes synchronous with the FFT step.
If I
In the time domain I'd try to mask the whole sample e.g. with a triangle
window. Then repeat it half overlapped.
(Or was this already one of your "various forms of crossfading"?)
Maybe even add up the signal with it's reverse signal to have constant
power before overlapped playback.
But that's
Nice that it runs on the M4F, what FFT size, overlap and audio processing block
size are you using? Are you running the FFT in a separate thread?
Giulio
From: Eric Brombaugh
To: music-dsp@music.columbia.edu
Sent: Friday, 16 September 2016, 19:12
Subject: Re:
On Fri, Sep 16, 2016 at 1:30 PM, Spencer Jackson
wrote:
> On Fri, Sep 16, 2016 at 11:24 AM, gm wrote:
> > Did you consider a reverb or an FFT time stretch algorithm?
> >
>
> I haven't looked into an FFT algorithm. I'll have to read up on that,
> but
I coded up the spectral freeze core of the Audio Damage "Spectre" module
in a similar way:
http://www.audiodamage.com/hardware/product.php?pid=ADM15
It's a basic phase vocoder with forward and inverse FFTs but we added
some fun little tweaks to shift, stretch and randomize the spectrum. It
Hi,I actually implemented this a few years back using an FFT algorithm, I can
dig out the code if you need it (it was a VST written using Juce and fftw, but
there was no threading on the FFT if I remember correctly, so it is flawed as
it is and it requires running with large blocksizes.I doubt
Regarding reverbs, one classic element is the Schroeder allpass:
https://ccrma.stanford.edu/~jos/pasp/Schroeder_Allpass_Sections.html You
can use it to smear a signal without any constructive or destructive
interference. Generally speaking this lack of "color" is a disadvantage,
but for your
Did you consider a reverb or an FFT time stretch algorithm?
Am 16.09.2016 um 17:48 schrieb Spencer Jackson:
Hi all:
First post on the list. Quite some time ago I set out to create a lv2
plugin re-creation of the electroharmonix freeze guitar effect. The
idea is that when you click the button
On Fri, Sep 16, 2016 at 11:24 AM, gm wrote:
> Did you consider a reverb or an FFT time stretch algorithm?
>
I haven't looked into an FFT algorithm. I'll have to read up on that,
but what do you mean with reverb? Would you feed the loop into a
reverb or apply some
Hi all:
First post on the list. Quite some time ago I set out to create a lv2
plugin re-creation of the electroharmonix freeze guitar effect. The
idea is that when you click the button it takes a short sample and
loops it for a drone like effect, sort of a granular synthesis
sustainer thing.
Hello Andre,
If you are interested in source code, here is a multi-wavetable
implementation of a band-limited sawtooth oscillator. It windows the
partials to avoid the Gibbs Effect.
https://github.com/philburk/jsyn/blob/master/src/com/jsyn/unitgen/SawtoothOscillatorBL.java
You can combine two
> >
> > From what I understand it is impossible to get rid of aliased frequencies
> > after sampling.
> >
>
> Once a frequency component is "aliased", it loses the original identity of
> that frequency takes on the identity of the alias. Then it's just a
> frequency component. The reason we
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