That is certainly true with bass (electric or upright) as well. (I'm
watching this discussion with fascination!)
Phil
On 4/29/14, 12:10 PM, katja wrote:
Hi Simon,
I'd be curious to see this adaptive filtering work in practice. Could
you share a patch, once you have that working? Vocals mostly don't
exceed a 3 octave range either. Only thing is, in vocals the strongest
component is sometimes not the first harmonic but the second, when
speaking or singing the lowest notes in the range.
Katja
On Tue, Apr 29, 2014 at 7:58 PM, Simon Iten <itensi...@gmail.com> wrote:
katja,
exactly! i filter the input based on the output of the pitch detection. i used
this for quite some time with my doublebass (but with a pickup per string) and
it works perfectly. i get no octave jumps or glitches at all. the version i
shared here is planned to be used for vocals, i have to see if it works as good…
the trick is not to filter too much in order to “let through” new notes but
enough to filter out strong overtones (mainly octaves). it also helps to have
filters in parallel. and of course you cut the range before that in order to
fit your input.
on a bass string this is very easy since on a double-bass you have a 3 octave
range per string you can cut many frequencies before even starting filtering.
this is how i did it and it worked.
i adapted this system from the gr300 also. there it’s even easier. just two
filters per string. one in the lower section (0-7th fret and one from 7-22
fret) they get switched via transistors based on the output voltage of the p/v
circuit. they are 2nd order bandpass filters.
cheers, simon
On 29 Apr 2014, at 19:37, katja <katjavet...@gmail.com> wrote:
Hi Simon,
See attachment for an upsampled version. I used a 6th order lo pass
filter with cut off at 1/4 of the original sampling rate. This seems
to work with max. 8 times upsampling. Period length error is then
limited to 1/8 sample.
You mentioned adaptive filtering of a real life input signal. Are you
planning to control filter cut off frequency with the pitch detection
result? Did you already try that? I wonder how that could work at all,
because the pitch result comes only after the adaptive filter.
Katja
On Tue, Apr 29, 2014 at 3:44 PM, Simon Iten <itensi...@gmail.com> wrote:
Katja thanks for your Inputs! Will Look at the Patch tonight. Simple lowpass
Filtering? I tried to upsample with a Block object but the biquad object
stopped outputting Pulses. If you don't mind doing a Version with upsampling
that would be fantastic.
Well i just copied from the Gr300 schematic, so no credits for me :)
Am 29.04.2014 um 13:12 schrieb katja <katjavet...@gmail.com>:
Hi Simon,
So your method counts samples per (zero-crossing) cycle, is what I
learned from studying the patch. Very nice how you do this with tilde
objects. It seems possible to get equivalent result with only one
[rpole~], when using the positive pulse as trigger for [samphold~] and
with two samples delay for [rpole~]. You get the integrator's maximum
everytime. See attached patch.
Of course it still counts integer number of samples. Upsampling would
indeed improve accuracy. An upsampled signal needs filtering to remove
spectral images, did you try that?
Katja
On Tue, Apr 29, 2014 at 8:10 AM, simon <itensi...@gmail.com> wrote:
nice changes with expr~ ! but i think you missed the point of the beginning
of the patch. read in my first e-mail for an explanation of what this patch
does exactly. it is an gr300 analog guitar synthesizer clone (well one voice
of it). it is intended for real-life signals so there needs to be an
adaptive filter in the beginning (with the pitch info we get from the two
rpole~
objects) and the signal needs to be squared to get the longest possible
sustain (envelope is re added later obviously). also i think response is
faster when squared, or not?
thanks for the changes, greatly appreciated!
simon
Well i know exactly what the Patch does... I just dont know why the two
numbers before the Addition Need to be -1 And -2 :-)
Will Look at your Version asap.
Cheers
Am 29.04.2014 um 02:00 schrieb Alexandre Torres Porres <por...@gmail.com>:
I have no idea what the patch is doing either, but I was able to clean it a
lot.
many things that didn't need to be there
cheers
2014-04-28 3:52 GMT-03:00 Simon Iten <itensi...@gmail.com>:
roman, thanks for your inputs.
i tried both fexpr and expr and sticked to fexpr at some point, don’t know
why though. will change it back! (i remember reading that fexpr was more
expensive but also more precise)
to make the whole thing work with real world signals (bass guitar in my
case) you have to add an adaptive filter in the beginning of the chain
(which is very easy because you get the frequency information hehe…) this
will filter out overtones and prevent octave jumping.
thanks
simon
On 28 Apr 2014, at 08:39, Roman Haefeli <reduz...@gmail.com> wrote:
That works very well. Good job and thanks for sharing!
One minor thing jumped to my eye: Your patch uses some instances of
[fexpr~] and all of them actually don't need [fexpr~] functionality. I
experienced that [fexpr~] is quite expensive, which seems apparent
considering it is designed for feedback algorithms. I don't know if
[fexpr~] is also expensive when you use it not for feedbacks as your
patch does. Anyway, you could replace them by likely less expensive
[expr~] instances:
[fexpr~ $x1>=0] -> [expr~ $v1>=0]
Roman
On Mon, 2014-04-28 at 00:59 +0200, simon wrote:
hey miller and list,
find attached a version that works beautifully. it's a dirty hack
without upsampling but it works extremly well. don't ask me why, i have no
idea.
thanks for all the help miller, really appreciate it! and thanks for pd
in general :-)
cheers,
simon
On Apr 27, 2014, at 8:59 PM, Simon Iten wrote:
sorry this one went off-list :-)
On 27 Apr 2014, at 19:05, simon <itensi...@gmail.com> wrote:
sure,
here is the version with biquad in a subpatch with a block opject to
upsample. probably i'm doing something wrong, i just copied from the block
help-patch.
<sinetosawtoothupsample.pd>
On Apr 27, 2014, at 5:48 PM, Miller Puckette wrote:
Drat, I don't have any explanation for this... can you send me the
patch
again?
cheers
M
On Sun, Apr 27, 2014 at 05:44:22PM +0200, simon wrote:
hmm, changing change to biquad does also not work. i mean it does
as long as i don't upsample in the subpatch. as soon as i change the block
object i get square instead of pulses...
On Apr 27, 2014, at 3:48 PM, Miller Puckette wrote:
Actually I don't know where the change~ object is from - I've nver
seen t
before. I would just use biquad~ 0 0 1 -1 0 (assuming that
change~ simply
ubtracts the previous sample from teh current one as I guessed
from the patch :)
M
On Sun, Apr 27, 2014 at 03:40:01PM +0200, Simon Iten wrote:
ok tried to upsample the whole thing (after the osc~) and now
change~ does nothing anymore… it just spits out the same square wave i feed
in…clues?
On 27 Apr 2014, at 13:05, Simon Iten <itensi...@gmail.com> wrote:
crosspost! sorry about the noise. thanks for the inputs i will
try to to this. not sure if i can. otherwise i will ask back if that’s ok!
On 27 Apr 2014, at 13:03, Simon Iten <itensi...@gmail.com>
wrote:
so if i would measure at the peak of the sawtooth and would
upsample inside the pd patch, i would get higher resolution, right?
any ideas how i can measure at the peak? (using the rpole
output on both samphold inputs does not work and delaying one of them is
also not working)
which
i would highly recommend you try this method with your gk-3
equipped guitar (one for each string) since you only have to cover a two
octave range per string the error is tolerable. (you can add an offset to
make it fit)
On 27 Apr 2014, at 12:56, Miller Puckette <m...@ucsd.edu> wrote:
That is an excellent, witty way to measure pulse withs using
only tilde obects - my hat's off to you.
The methond only has limited accuracy since its measurement is
in
samples. For instance, a 1/2 cycle of a 440-hz. tone at 44.1
kHz is
only 50 samples, so there's only 2% accuracy. That's about
1/3 of a
half tone (30-ish cents) which would sound horribly out of
tune.
There's an alternative sine-to-sawtooth recipe described here:
http://msp.ucsd.edu/Publications/icmc10.pdf
This is the basis of my guitar processing patch, smeck, but
should be more
broadly useful. But it has its own limitations: the sawtooth
you get out
is wiggly if the input sn't a pure sinusoid.
There's also the possibility of simply pitch tracking with
sigmund~. Use
a maximum frequency around 6000 and a maximum of 6 partals
(default 50!)
for best results.
cheers
M
On Sun, Apr 27, 2014 at 11:27:33AM +0200, Simon Iten wrote:
dear list,
i have a strange problem with my “sinetosawtooth” patch.
it is basically a version of the pitch to voltage conversion
used in the old gr300 guitar synths from roland.
i cut out all the clutter to make it easier to look at and
understand. (cut out the adaptive filtering at the input since i use a sine
wave for this example and not a guitar string)
here is how it works (or should):
-an input signal gets amplified by a large factor and
clipped. this squares the input.
-the square wave is converted to pulses.
-the pulses from the rising of the square wave are used to
set and reset an accumulating filter (rpole~)
this results in a sawtooth wave that varies in amplitude
depending on the frequency of the input.
-a sample and hold samples the peak of the sawtooth and holds
it until the next peak occurs. this, after a conversion gives us the input
frequency. yeah!
in the example patch i used the falling edges of the
square wave to trigger the sample and hold. this samples the sawtooth
amplitude after half the rising. (this is also why i have 22050 in fexpr~
and not 44100) i could not figure out how to sample the peak of the
sawtooth, so suggestions here are very welcome.
now to the problem:
the extracted frequency does not exactly correspond to the
input frequency. it is pretty close at low frequencies but gets worse at
higher frequencies. the factor is not constant. at even higher frequencies
(around 5000 hertz) the reported frequency gets totally out of control.
i first thought this is because the samphold~ object is
inaccurate. but i then saw that the sawtooth wave from the rpole~ object has
no constant amplitude even with the input frequency not changing. so it
seems that either rpole~ or change~ is not accurate.
or the problem is that i sample in the middle of the rising
and not at the top ( as described earlier)
attached the sinetosawtooth patch. set your sound card to
44100 or change the 22050 in fexpr~ to half the sampling frequency.
i would really appreciate if somebody could have a look at
this,
thanks, simon
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