Hello Mark,
On 27/02/2014 3:52 PM, Mark Garvin wrote:
Most sample banks these days seem to be in NKI format (Native
Instruments). They have the ability to map ranges of a keyboard into
different samples so the timbres don't become munchkin-ized or
Vader-ized. IOW, natural sound within each regis
>1. Re: Hosting playback module for samples (Ross Bencina)
>
> From: Ross Bencina
> Hi Mark,
>
> I'm not really sure that I understand the problem. Can you be more
> specific about the problems that you're facing?
Hi Ross,
Specific: (Forgive me if you know all of this): My code is C#, but
Hello Theo,
On 27/02/2014, Theo Verelst wrote:
> http://www.youtube.com/watch?v=5snh9UvAno4&feature=youtu.be
May I ask what kind of software do you use at 0:18? I couldn't
recognize it. What are those graph processors actually doing, and what
is their role for this example? Could you give a
Hi
Whilst among other things trying to re-remaster some of the nice music
tracks I've listen to since they came out, I've found out my
"collectors" Abba CDs in spite of being pretty well corrected to sound
ok, except for quite dreadfully compressed, sounded like someone had
been dubbing in ot
>TIIR and resampling might both be JOS, but otherwise they're not the same
thing. resampling is Julius and Gossett and TIIR is Julius and Wang.
Oh man. It was a long time ago that I looked on TIIR and I have been more
used to JOS resampling in the last years. I basically confused the idea of
TIIR
> Date: Wed, 26 Feb 2014 16:42:00 -0500
> From: r...@audioimagination.com
> To: music-dsp@music.columbia.edu
> Subject: Re: [music-dsp] R: R: R: R: R: Best way to do sine hard sync?
>
> On 2/26/14 3:08 PM, Marco Lo Monaco wrote:
> > Yes I am talking about IIR and feedback of course and I ha
On 2/26/14 3:08 PM, Marco Lo Monaco wrote:
Yes I am talking about IIR and feedback of course and I have no problem
actually in converting any analog network changing at audio rate (thus
modulation of filter parameters etc). In my case I used a simple leaky
integrator time varying at audio rate tu
Hi Mark,
I'm not really sure that I understand the problem. Can you be more
specific about the problems that you're facing?
Personally I would avoid managed code for anything real-time (ducks).
You're need to build a simple audio engine (consider PortAudio or the
ASIO SDK). And write some V
I dont think in literature exist proofs of timevarying filter at audio rate
with no artifacts: afaik there is the minimum norm class and other
techniques to understand the minimum requirements for a topology to be
changed at every N sample minimum. If I am wrong I would love to have the
details and
Hi Mark,
if you just need a simple VST2 Windows host there's enough source code
that will give you access tho VSTi. You could use the MiniHost from the
VST2 SDK (hope you've got a copy. The download/support is gone since
some weeks). And of course you can have a look at Hermann Seib's VstHost:
On 2/26/14 1:55 PM, Marco Lo Monaco wrote:
Actually I can do a filter with time varying filter at audio rate with _NO_
glithces or artifacts, stable (essentially behaving like an analog one).
you're not the only one that can do that. but, with an IIR, there are
problems that arise and must be
Actually I can do a filter with time varying filter at audio rate with _NO_
glithces or artifacts, stable (essentially behaving like an analog one).
The point is that the BLIT harmonic content changes in modulation (because
you have to cut harmonics as you are sweeping hi) and that results in spike
On 2/26/14 12:37 PM, Marco Lo Monaco wrote:
Moreover in my experience BLIT with leaky integrators fails on frequency
modulations,
i can imagine why. it's sorta like how some IIR filter topologies fail
with coefficient modulation.
this is another reason that i am a proponent of wavetable syn
Agreed 100%.
Moreover in my experience BLIT with leaky integrators fails on frequency
modulations, other approach like BLIT-SWS are more complicated but if memory
is not an issue wavetable is the choice.
BTW a smarter approach is needed for hardsync because a user request is to
change at processti
On 2/26/14 10:49 AM, Marco Lo Monaco wrote:
Yes it is. And it is true even in analog domain...if you only could have
dirac pulse realized on a circuit :)
Mathematically and in continuous time they are the same: it is the basic
starting concept of BLIT (see also Stilson paper)
Hope to have helped.
PortAudio!
http://www.portaudio.com
best,
douglas
On Wed, Feb 26, 2014 at 11:56 AM, Mark Garvin wrote:
> I realize that this is slightly off the beaten path for this group,
> but it's a problem that I've been trying to solve for a few years:
>
> I had written software for notation-based comp
On 26/02/2014, Risto Holopainen wrote:
> Now, for my part, I find soft sync much more useful. I don't know what
> attempts there have been to do soft sync in digital oscillators, if anyone
> knows I'd be interested.
Nobody agrees on whether soft sync is "knock the waveform into
reverse" (the Ales
I realize that this is slightly off the beaten path for this group,
but it's a problem that I've been trying to solve for a few years:
I had written software for notation-based composition and playback of
orchestral scores. That was done via MIDI. I was working on porting
the original C++ to C#, a
On 26/02/2014, robert bristow-johnson wrote:
> the simplest thing, if you want the slave perfectly locked to the
> master, is to derive both master and slave phase from the very same
> phase accumulator. there might be scaling and wrap around, but that
> should be easy.
I think that's how most V
I thought about that, but it's still wrong for every case but the
master and slave having a 1:1 frequency ratio. Hence the need to
multiply by the m:s ratio as well. (And of course, the slave should be
bandlimited according to its frequency; that pretty much goes without
saying. Which makes it all
robert bristow-johnson:
for a slave doing a sine, i wonder what you would expect to hear as
the master/slave frequency ratio changes. while i have heard sync
saws and sync squares, i don't think i ever heard a sync sine and it
would seem to me to go from no harmonics to lotsa harmonics prett
It is.
The DC offset defines the slope and the impulse is the reset, if sgn(DC offset)
* sgn (impulse) == -1.
Of course, a leaky integrator, like Sampo suggested, is helpful to suppress
accumulated precision errors.
Best,
Steffan
On 26 Feb 2014, at 15:41, robert bristow-johnson
wrote:
On 2/26/14 10:50 AM, Tobias Münzer wrote:
an easy fix to avoid this kind of phase jitter is to add the
fractional part of the master oscillator after the zero crossing to
the slave.
Basically: Do not reset the slave to zero, but to the fractional rest
of the master.
the simplest thing, if
Yes,
an easy fix to avoid this kind of phase jitter is to add the fractional
part of the master oscillator after the zero crossing to the slave.
Basically: Do not reset the slave to zero, but to the fractional rest of
the master.
There is still a lot of aliasing if you don't band limit the tr
Yes it is. And it is true even in analog domain...if you only could have
dirac pulse realized on a circuit :)
Mathematically and in continuous time they are the same: it is the basic
starting concept of BLIT (see also Stilson paper)
Hope to have helped...and sorry if I misunderstood your words Robe
"bandlimit the slave's phase" -> "bandlimit the slave's output". oops.
and I do know how to spell interpolation, honest.
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On 26/02/2014, Risto Holopainen wrote:
> When it comes to programming hard sync, I would use oversampling. I'm not
> saying that you should, I'm just lazy enough to do it the easy way.
You need to oversample a *lot* to chase away aliasing, though. The
Alesis Fusion - and its descendant, the M-Aud
Am 26.02.2014 15:15, schrieb robert bristow-johnson:
On 2/26/14 4:03 AM, Marco Lo Monaco wrote:
yup that was the BLIT stuff, i think, so a sawtooth is the integral if
this BandLimited Impulse Train (with a little DC added).
Ahaha, funny! Did you set sarcasm mode = on? :)))
i guess i hadn't.
On 2/26/14 9:28 AM, Sampo Syreeni wrote:
On 2014-02-26, robert bristow-johnson wrote:
On 2/26/14 4:03 AM, Marco Lo Monaco wrote:
yup that was the BLIT stuff, i think, so a sawtooth is the
integral if this BandLimited Impulse Train (with a little DC added).
Ahaha, funny! Did you set sarcasm m
On 2014-02-26, robert bristow-johnson wrote:
On 2/26/14 4:03 AM, Marco Lo Monaco wrote:
yup that was the BLIT stuff, i think, so a sawtooth is the integral if
this BandLimited Impulse Train (with a little DC added).
Ahaha, funny! Did you set sarcasm mode = on? :)))
i guess i hadn't. a ban
On 2/26/14 4:03 AM, Marco Lo Monaco wrote:
yup that was the BLIT stuff, i think, so a sawtooth is the integral if
this BandLimited Impulse Train (with a little DC added).
Ahaha, funny! Did you set sarcasm mode = on? :)))
i guess i hadn't. a bandlimited sawtooth is not the integral of a BLIT
Here's some examples made with a eurorack modular. The master oscillator is the
tiptop Z3000 and the slave is an intellijel rubicon. The sine output of the
slave oscillator is always in the right channel and the modulating output of
the master oscillator is in the left channel throughout.
1) a s
I have an almost embarrassing question on this subject. Having lead far
too sheltered a life, I have never ~knowingly~ heard analog hard sync,
only this or that digital emulation; partly as I have never owned an
analog synth. So - is there an example file anywhere with the sampled
(with suitabl
Hi thanks a lot for your answers,
25/02/2014 20:48, Ross Bencina wrote:
The approach that I am familiar with is the "corrective grains"
approach (AKA BLIT/BLEP/BLAMP etc) where you basically run a
granulator that generates grains that cancel the aliasing caused by
the phase discontinuity. The
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>> yup that was the BLIT stuff, i think, so a sawtooth is the integral if
this BandLimited Impulse Train (with a little DC added).
Ahaha, funny! Did you set sarcasm mode = on? :)))
Ciao Robert
Marco
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Hello Tobias,
following Ross advice one of the drawback that you will have to deal is the
CPU usage at high master frequencies. Placing in an overlap and add fashion
a grain is very convenient at low freqs but not so much at high. More over
you will have to deal with some sort of DC offsets on the
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