From recompas at gmail.com  Fri Mar  1 18:03:46 2013
From: recompas at gmail.com (Travis Thatcher)
Date: Fri, 1 Mar 2013 11:03:46 -0600
Subject: [sdiy] Roland JX-3p vca - noise
In-Reply-To: <512FC05B.9030706@mnet-online.de>
References: <CAG0qc=3pQXGgGV7HWMRbrQrmY_rOYHnNeuMkkauaJb4-rJGi4w@mail.gmail.com>
	<1361909085.17434.YahooMailClassic@web181602.mail.ne1.yahoo.com>
	<CAG0qc=2g6KtFPfFT2p1krAmh4LE-75rhKnnLFMH0ncAm22FnEw@mail.gmail.com>
	<512DC3A2.7050608@btinternet.com>
	<CAG0qc=2pTU1qrAMnOGyjkriTyfXbvMWC1QZpyfgodHoBNubfvw@mail.gmail.com>
	<4A1461B3-218E-42D6-90C9-465141810946@electricdruid.net>
	<512F71E4.5080500@bugbrand.co.uk> <512FC05B.9030706@mnet-online.de>
Message-ID: <CAG0qc=2zgqJU20ySLv5Jf7+X4aMq_SKW2Dfpe9fqi0YLh-hbnA@mail.gmail.com>

I seemed to have fixed my issue.  I cleaned out the jacks w rubbing
alcohol and deoxit, then re-soldered them. I was still having the
cracking issue randomly and also it seemed as though the right channel
wasn't coming through properly. In the headphone output the right
channel was way lower and when cables were plugged into both channel
jacks only the left actually output signal.  I started from before the
muting transistors and traced the signal path to find that the
transistors were okay but the volume pot was actually to blame. It
seemed okay but the trace of signal pin of the voltage divider had
become disconnected from the circuit.  A quick jumper wire and all was
well and my jx had its beautiful noisy stereo panned chorus back.
Thanks for the help.  I'm not sure what it is about Roland gear of
this era, but I've seen numerous 606s and a few JX3ps where the traces
just seem to be lifting off the boards and the soldermask all crinkly.
 What are some good tips for preserving these devices for years to
come? I've seen Minimoog vco pcbs that are a good 10-12 years older
look way better than these things.

On Thu, Feb 28, 2013 at 2:38 PM, Florian Anwander
<fanwander at mnet-online.de> wrote:
> Hello
>
>
>> If anyone has a part number for the output sockets, please share.
>> Could be that I've experienced similar(ish) problems.
>> I did get a new socket that may be suitable from Rapid, but haven't
>> taken the time to check it yet..
>
> I would desolder the sockets, wash them with medical alcohol, apply Caig
> Deoxit, let the Deoxit dry and solder the sockets back. I don't like
> spraying Deoxit in any electronic device, because 90% of the spray reach
> parts, that it shouldn't touch; but using it on the separated sockets does
> wonders.
>
> Florian
>
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy



-- 
http://www.travisthatcher.com
http://www.soundcloud.com/travisthatcher
http://voiceofsaturn.blogspot.com


From metadata at gmx.com  Sat Mar  2 03:58:39 2013
From: metadata at gmx.com (neil harper)
Date: Fri, 01 Mar 2013 21:58:39 -0500
Subject: [sdiy] VCA response - continously variable between linear and
	exponential
Message-ID: <1362193119.1910.135.camel@nwwtlaptop>

hey all,
i'm looking for some advice/pointers on how to get a VCA response curve
to be continuously variable between linear and exponential. i can do
switchable between the two, but is there any trick to getting everything
inbetween?

if it helps, my VCA is lm13700 based but i don't mind starting with
something else.

-- 
//neil harper
every wave is new until it breaks



From dixon at mail.ubc.ca  Sat Mar  2 05:21:10 2013
From: dixon at mail.ubc.ca (David G Dixon)
Date: Fri, 1 Mar 2013 20:21:10 -0800
Subject: [sdiy] VCA response - continously variable between linear
	andexponential
In-Reply-To: <1362193119.1910.135.camel@nwwtlaptop>
References: <1362193119.1910.135.camel@nwwtlaptop>
Message-ID: <9BF8A6C09E98466CAE6854E363434E2A@galvanox01>

> i'm looking for some advice/pointers on how to get a VCA 
> response curve to be continuously variable between linear and 
> exponential. i can do switchable between the two, but is 
> there any trick to getting everything inbetween?
> 
> if it helps, my VCA is lm13700 based but i don't mind 
> starting with something else.

I don't know how to do it with an OTA, but it's pretty easy with a 2164 VCA,
if you start with Mike Irwin's circuit from EDN -- the last article in this
PDF: http://www.edn.com/file/18017-72502di.pdf

The thing to do is to add a separate CV inverter which feeds the amplifying
VCA's VC pin directly for the exponential response.  The CV output from this
and the CV output from the linearizing servo can be connected to the poles
of an SPDT switch, with the centre terminal going to the VC pin of the
amplifying VCA, or the switch can be replaced with a pot, with the wiper
going to the amplifying VCA's VC pin.

I'll send you the schematic (sorry, I have no website).



From cheater00 at gmail.com  Sat Mar  2 08:28:08 2013
From: cheater00 at gmail.com (cheater cheater)
Date: Sat, 2 Mar 2013 08:28:08 +0100
Subject: [sdiy] VCA response - continously variable between linear
	andexponential
In-Reply-To: <9BF8A6C09E98466CAE6854E363434E2A@galvanox01>
References: <1362193119.1910.135.camel@nwwtlaptop>
	<9BF8A6C09E98466CAE6854E363434E2A@galvanox01>
Message-ID: <CA+9GZUiRo8aH-qq_k+F02NM_p0G8Tp7N3TNj14zm9Oo8Sv+9wA@mail.gmail.com>

David, that may look like a good idea, but it's different - it's
mixing lin and exp. It's not going to give you the same range of
sounds you could get from a circuit which can vary the exponential
base.

Cheers,
D.

On Sat, Mar 2, 2013 at 5:21 AM, David G Dixon <dixon at mail.ubc.ca> wrote:
>> i'm looking for some advice/pointers on how to get a VCA
>> response curve to be continuously variable between linear and
>> exponential. i can do switchable between the two, but is
>> there any trick to getting everything inbetween?
>>
>> if it helps, my VCA is lm13700 based but i don't mind
>> starting with something else.
>
> I don't know how to do it with an OTA, but it's pretty easy with a 2164 VCA,
> if you start with Mike Irwin's circuit from EDN -- the last article in this
> PDF: http://www.edn.com/file/18017-72502di.pdf
>
> The thing to do is to add a separate CV inverter which feeds the amplifying
> VCA's VC pin directly for the exponential response.  The CV output from this
> and the CV output from the linearizing servo can be connected to the poles
> of an SPDT switch, with the centre terminal going to the VC pin of the
> amplifying VCA, or the switch can be replaced with a pot, with the wiper
> going to the amplifying VCA's VC pin.
>
> I'll send you the schematic (sorry, I have no website).
>
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy


From synth1 at airmail.net  Sat Mar  2 15:40:17 2013
From: synth1 at airmail.net (Paul Schreiber)
Date: Sat, 2 Mar 2013 08:40:17 -0600
Subject: [sdiy] New DIY supplier
Message-ID: <002c01ce1753$dbf89970$93e9cc50$@airmail.net>

Check out SynthCube:

www.synthcube.com

They have MOTM, MU, some Frac and other assorted pc boards, parts, panels
and stuff. More to be added daily.

Paul S.




From houshu at muj.biglobe.ne.jp  Sat Mar  2 17:53:37 2013
From: houshu at muj.biglobe.ne.jp (houshu at muj.biglobe.ne.jp)
Date: Sun, 03 Mar 2013 01:53:37 +0900 (JST)
Subject: [sdiy] VCA response - continously variable between linear
	and	exponential
In-Reply-To: <1362193119.1910.135.camel@nwwtlaptop>
Message-ID: <a11924c9-1f0e-4c64-94d4-ebc7d3391e3a@bvea33570>

Hello Neil and List,

I've designed continuous curve VCA.
Precise exponential can not be obtained.
But, fractional power curves were available.

Final VCA design (Built for battery operation, and worked as in simulations.).
  http://houshu.at.webry.info/201112/article_2.html

Current source designs:
  http://houshu.at.webry.info/201111/article_1.html
  http://houshu.at.webry.info/201111/article_2.html
  http://houshu.at.webry.info/201201/article_1.html

This is basically a function generator combining log and exponential, 
which may be analyzed by theory of translinear circuit.

Continuous curve control is possible, but the required
resistor range is too large to obtain linear-like response.  
So I chose 3 curves.

Here is comparison of the 3 curves and linear and exponential: 
http://userdisk.webry.biglobe.ne.jp/000/024/65/N000/000/000/132039778125213127511_Lin-Exp1_dc.PNG

If you can wait for 1 month, I'll design a current source circuit 
for LM13700, though I myself would not test it on a breadboard.

Sam HOSHUYAMA
Saitama, JAPAN

----- "neil harper" <metadata at gmx.com>
hey all,
i'm looking for some advice/pointers on how to get a VCA response curve
to be continuously variable between linear and exponential. i can do
switchable between the two, but is there any trick to getting everything
inbetween?

if it helps, my VCA is lm13700 based but i don't mind starting with
something else.

-- 
//neil harper
every wave is new until it breaks

_______________________________________________
Synth-diy mailing list
Synth-diy at dropmix.xs4all.nl
http://dropmix.xs4all.nl/mailman/listinfo/synth-diy


From davebr at earthlink.net  Sat Mar  2 21:57:52 2013
From: davebr at earthlink.net (Dave Brown)
Date: Sat, 2 Mar 2013 12:57:52 -0800
Subject: [sdiy] New DIY supplier
In-Reply-To: <002c01ce1753$dbf89970$93e9cc50$@airmail.net>
References: <002c01ce1753$dbf89970$93e9cc50$@airmail.net>
Message-ID: <002601ce1788$9bab02b0$d3010810$@earthlink.net>

The store doesn't work at all with Opera.  I had to go with IE to see more
than one product. I hate fancy web pages - just irritating.

Dave

-----Original Message-----
From: synth-diy-bounces at dropmix.xs4all.nl
[mailto:synth-diy-bounces at dropmix.xs4all.nl] On Behalf Of Paul Schreiber
Sent: Saturday, March 02, 2013 6:40 AM
To: synthdiy diy
Subject: [sdiy] New DIY supplier

Check out SynthCube:

www.synthcube.com

They have MOTM, MU, some Frac and other assorted pc boards, parts, panels
and stuff. More to be added daily.

Paul S.





From davebr at earthlink.net  Sat Mar  2 23:28:28 2013
From: davebr at earthlink.net (Dave Brown)
Date: Sat, 2 Mar 2013 14:28:28 -0800
Subject: [sdiy] New DIY supplier
In-Reply-To: <002601ce1788$9bab02b0$d3010810$@earthlink.net>
References: <002c01ce1753$dbf89970$93e9cc50$@airmail.net>
	<002601ce1788$9bab02b0$d3010810$@earthlink.net>
Message-ID: <001601ce1795$43f5a0e0$cbe0e2a0$@earthlink.net>

Actually while I am on the topic of web pages and carts, are there any
zencart experts on the list?  I need some help figuring out how to configure
a zencart for international shipping.  I was hoping to do it flat rate for
USA and rest of world.  Also how to apply attributes to an entire group of
products.   I am learning by doing and it is a bit confusing.  Contact me
off list if you would, please.

Dave

-----Original Message-----
From: synth-diy-bounces at dropmix.xs4all.nl
[mailto:synth-diy-bounces at dropmix.xs4all.nl] On Behalf Of Paul Schreiber
Sent: Saturday, March 02, 2013 6:40 AM
To: synthdiy diy
Subject: [sdiy] New DIY supplier

Check out SynthCube:

www.synthcube.com

They have MOTM, MU, some Frac and other assorted pc boards, parts, panels
and stuff. More to be added daily.

Paul S.



_______________________________________________
Synth-diy mailing list
Synth-diy at dropmix.xs4all.nl
http://dropmix.xs4all.nl/mailman/listinfo/synth-diy




From gjames at kddlab.com  Sun Mar  3 02:56:24 2013
From: gjames at kddlab.com (Greg James)
Date: Sat, 2 Mar 2013 20:56:24 -0500
Subject: [sdiy] New DIY supplier
In-Reply-To: <002c01ce1753$dbf89970$93e9cc50$@airmail.net>
References: <002c01ce1753$dbf89970$93e9cc50$@airmail.net>
Message-ID: <EF9A62A1F17F4CE9A31752136F8E0015@KDDLab05>

Some prices are reasonable and some are, well.

$29.00 for a '910 PCB?

Are these NOS/End-of-life thru hole, or are they/you still making new runs
of the old PCBs?

Greg James

-----Original Message-----
From: synth-diy-bounces at dropmix.xs4all.nl
[mailto:synth-diy-bounces at dropmix.xs4all.nl] On Behalf Of Paul Schreiber
Sent: Saturday, March 02, 2013 9:40 AM
To: synthdiy diy
Subject: [sdiy] New DIY supplier

Check out SynthCube:

www.synthcube.com

They have MOTM, MU, some Frac and other assorted pc boards, parts, panels
and stuff. More to be added daily.

Paul S.


_______________________________________________
Synth-diy mailing list
Synth-diy at dropmix.xs4all.nl
http://dropmix.xs4all.nl/mailman/listinfo/synth-diy



From rsdio at sounds.wa.com  Sun Mar  3 03:11:22 2013
From: rsdio at sounds.wa.com (rsdio at sounds.wa.com)
Date: Sat, 2 Mar 2013 18:11:22 -0800
Subject: [sdiy] VCA response - continously variable between linear
	and	exponential
In-Reply-To: <a11924c9-1f0e-4c64-94d4-ebc7d3391e3a@bvea33570>
References: <a11924c9-1f0e-4c64-94d4-ebc7d3391e3a@bvea33570>
Message-ID: <C0E60B43-8135-4A91-A73E-10D8BDE9281D@sounds.wa.com>

Why would you need continuously variable curves? Are you just talking  
about shifting away from the standard 1 V per octave, but still being  
based on the octave standard?

In my experience, a VCO either responds to CV with a linear  
relationship between voltage and frequency (simpler circuits, easier  
to get precise), or it responds to CV with a logarithmic relationship  
between voltage and frequency (allowing a pitch bend CV to apply the  
same relative pitch change regardless of octave). The latter happens  
to have a linear relationship between voltage and octave or  
semitones. I cannot think of any reason to have a response other than  
one of those two. Note that the relationship between pitch (Hz or  
period) and octave is not linear.

As I mentioned, the linear response is more basic, and because it is  
linear you can make it as precise as you want. Unfortunately, a pitch  
bend CV applied in one octave will have a different relative pitch  
offset than in any other octave, making it impossible to pitch bend a  
chord on a polyphonic keyboard (unless you specifically desire  
inharmonious results). Thus, the only reason for non-linear CV of  
pitch is to allow mixing multiple CV sources (e.g. from keyboard,  
pitch bend, octave/footage selector, etc), and in that case the only  
logical choice is the specific curve of an octave relationship. The  
only disadvantage of the latter is that it is very difficult to  
create this non-linear curve. It's easiest to use, but hardest to  
create. Thus, the only two logical choices seem to be a Hz scale or  
an octave scale, e.g. V/Hz, Hz/V, V/octave, octave/V, V/period,  
period/V. There are lots of variations, but they still basically fall  
into just two categories in terms of the exponential or linear  
aspect, and I can't see the utility of anything "between" linear and  
exponential.

I imagine that the only useful variation would be the value for "n"  
in the expression, "n V per octave." The industry has mostly settled  
on 1 V per octave, with some using 1.2 V per octave, but my point is  
that it's always something per octave unless it's totally linear.

By the way, I'm sure everyone here understands the above, and I don't  
intend to insult anyone by spelling it all out, but I really want to  
make sure there's not something I'm missing. It seems like there was  
an implication that a cubic or quadratic or higher power exponential  
curves might be desirable, but I can't see why. I can see why 2 V per  
octave or 0.5 V per octave might be incredibly useful, because it  
would allow a different range of pitch for the same power supply. But  
unless I'm using the terminology incorrectly, that's not continuously  
variable between linear and exponential.

Brian Willoughby
Sound Consulting


----- "neil harper" <metadata at gmx.com>
> hey all,
> i'm looking for some advice/pointers on how to get a VCA response  
> curve
> to be continuously variable between linear and exponential. i can do
> switchable between the two, but is there any trick to getting  
> everything
> inbetween?
>
> if it helps, my VCA is lm13700 based but i don't mind starting with
> something else.




From dixon at mail.ubc.ca  Sun Mar  3 04:36:41 2013
From: dixon at mail.ubc.ca (David G Dixon)
Date: Sat, 2 Mar 2013 19:36:41 -0800
Subject: [sdiy] VCA response - continously variable between
	linearand	exponential
In-Reply-To: <C0E60B43-8135-4A91-A73E-10D8BDE9281D@sounds.wa.com>
References: <a11924c9-1f0e-4c64-94d4-ebc7d3391e3a@bvea33570>
	<C0E60B43-8135-4A91-A73E-10D8BDE9281D@sounds.wa.com>
Message-ID: <9D5772D144144B7C8DB83140C8F9537E@galvanox01>

I think we're having an "Emily Litella moment" here:

We're talking about VCAs, not VCOs!
 

> -----Original Message-----
> From: synth-diy-bounces at dropmix.xs4all.nl 
> [mailto:synth-diy-bounces at dropmix.xs4all.nl] On Behalf Of 
> rsdio at sounds.wa.com
> Sent: Saturday, March 02, 2013 6:11 PM
> To: neil harper
> Cc: Synth-Diy
> Subject: Re: [sdiy] VCA response - continously variable 
> between linearand exponential
> 
> Why would you need continuously variable curves? Are you just 
> talking about shifting away from the standard 1 V per octave, 
> but still being based on the octave standard?
> 
> In my experience, a VCO either responds to CV with a linear 
> relationship between voltage and frequency (simpler circuits, 
> easier to get precise), or it responds to CV with a 
> logarithmic relationship between voltage and frequency 
> (allowing a pitch bend CV to apply the same relative pitch 
> change regardless of octave). The latter happens to have a 
> linear relationship between voltage and octave or semitones. 
> I cannot think of any reason to have a response other than 
> one of those two. Note that the relationship between pitch (Hz or
> period) and octave is not linear.
> 
> As I mentioned, the linear response is more basic, and 
> because it is linear you can make it as precise as you want. 
> Unfortunately, a pitch bend CV applied in one octave will 
> have a different relative pitch offset than in any other 
> octave, making it impossible to pitch bend a chord on a 
> polyphonic keyboard (unless you specifically desire 
> inharmonious results). Thus, the only reason for non-linear 
> CV of pitch is to allow mixing multiple CV sources (e.g. from 
> keyboard, pitch bend, octave/footage selector, etc), and in 
> that case the only logical choice is the specific curve of an 
> octave relationship. The only disadvantage of the latter is 
> that it is very difficult to create this non-linear curve. 
> It's easiest to use, but hardest to create. Thus, the only 
> two logical choices seem to be a Hz scale or an octave scale, 
> e.g. V/Hz, Hz/V, V/octave, octave/V, V/period, period/V. 
> There are lots of variations, but they still basically fall 
> into just two categories in terms of the exponential or 
> linear aspect, and I can't see the utility of anything 
> "between" linear and exponential.
> 
> I imagine that the only useful variation would be the value for "n"  
> in the expression, "n V per octave." The industry has mostly 
> settled on 1 V per octave, with some using 1.2 V per octave, 
> but my point is that it's always something per octave unless 
> it's totally linear.
> 
> By the way, I'm sure everyone here understands the above, and 
> I don't intend to insult anyone by spelling it all out, but I 
> really want to make sure there's not something I'm missing. 
> It seems like there was an implication that a cubic or 
> quadratic or higher power exponential curves might be 
> desirable, but I can't see why. I can see why 2 V per octave 
> or 0.5 V per octave might be incredibly useful, because it 
> would allow a different range of pitch for the same power 
> supply. But unless I'm using the terminology incorrectly, 
> that's not continuously variable between linear and exponential.
> 
> Brian Willoughby
> Sound Consulting
> 
> 
> ----- "neil harper" <metadata at gmx.com>
> > hey all,
> > i'm looking for some advice/pointers on how to get a VCA response 
> > curve to be continuously variable between linear and exponential. i 
> > can do switchable between the two, but is there any trick 
> to getting 
> > everything inbetween?
> >
> > if it helps, my VCA is lm13700 based but i don't mind starting with 
> > something else.
> 
> 
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy



From rsdio at sounds.wa.com  Sun Mar  3 05:49:17 2013
From: rsdio at sounds.wa.com (rsdio at sounds.wa.com)
Date: Sat, 2 Mar 2013 20:49:17 -0800
Subject: [sdiy] VCA response - continously variable between
	linearand	exponential
In-Reply-To: <9D5772D144144B7C8DB83140C8F9537E@galvanox01>
References: <a11924c9-1f0e-4c64-94d4-ebc7d3391e3a@bvea33570>
	<C0E60B43-8135-4A91-A73E-10D8BDE9281D@sounds.wa.com>
	<9D5772D144144B7C8DB83140C8F9537E@galvanox01>
Message-ID: <71AB17E6-1241-4C89-8094-F6FCB3D080F3@sounds.wa.com>

Ha!

I totally forgot about the Emily Litella character, or at least the  
name, but I get it now. I totally missed the boat on the VCO / VCA  
distinction. Usually I catch myself before I write an entire essay.

But in an attempt to save a little face, isn't exponential VCA  
response an attempt to cater to the human hearing system's perception  
of intensity as a logarithmic function? We hear volume in relative  
terms, not absolute. Decibels are logarithmic, and almost perfectly  
map to our perception of intensity of sound.

In other words, I again ask: Why would you want anything besides  
linear or pure exponential? Something in between doesn't seem  
terribly useful. (not that a synth designer shouldn't create  
additional parameters for variation)

Brian


On Mar 2, 2013, at 19:36, David G Dixon wrote:
>
> I think we're having an "Emily Litella moment" here:
>
> We're talking about VCAs, not VCOs!



From tom at electricdruid.net  Sun Mar  3 11:53:40 2013
From: tom at electricdruid.net (Tom Wiltshire)
Date: Sun, 3 Mar 2013 10:53:40 +0000
Subject: [sdiy] VCA response - continously variable between
	linearand	exponential
In-Reply-To: <71AB17E6-1241-4C89-8094-F6FCB3D080F3@sounds.wa.com>
References: <a11924c9-1f0e-4c64-94d4-ebc7d3391e3a@bvea33570>
	<C0E60B43-8135-4A91-A73E-10D8BDE9281D@sounds.wa.com>
	<9D5772D144144B7C8DB83140C8F9537E@galvanox01>
	<71AB17E6-1241-4C89-8094-F6FCB3D080F3@sounds.wa.com>
Message-ID: <4FD73A85-64CE-4B22-9F55-8E2C102629C3@electricdruid.net>


On 3 Mar 2013, at 04:49, rsdio at sounds.wa.com wrote:

> In other words, I again ask: Why would you want anything besides linear or pure exponential? Something in between doesn't seem terribly useful. (not that a synth designer shouldn't create additional parameters for variation)
> 
> Brian

We're talking about VCAs, so the reason is to do with amplitude envelopes. Although we might need a pure exponential response to get a smooth ramp-sounding envelope, that isn't remotely realistic. What sound has an envelope that rises smoothly and consistently?!  In reality, envelope shapes are pretty variable (some even have little wobbles in them). So whilst we might want a linear VCA to go with our expo envelope, we might want an expo VCA to go with our liner envelope, and when the envelopes get more interesting than that, the VCA response could get more interesting too. That would all provide realistic variety in the volume curves we're able to generate.

In some ways, though, I consider a linear VCA to be the ideal, and reckon all the messing around with the shapes should be done by the envelope generators. So I sort-of agree that expo and linear pretty much cover it, and even expo's a bit superfluous. If you want a bendy curve, use a bendy curve generator.

Regards,
Tom

From rsdio at sounds.wa.com  Sun Mar  3 12:15:55 2013
From: rsdio at sounds.wa.com (rsdio at sounds.wa.com)
Date: Sun, 3 Mar 2013 03:15:55 -0800
Subject: [sdiy] Roland JX-3p vca - noise
In-Reply-To: <CAG0qc=16Ggi0PSHJZpNPqQ7oMNVkzmi_VNAEEZEHM8Fy7hnuOg@mail.gmail.com>
References: <CAG0qc=3pQXGgGV7HWMRbrQrmY_rOYHnNeuMkkauaJb4-rJGi4w@mail.gmail.com>
	<1361909085.17434.YahooMailClassic@web181602.mail.ne1.yahoo.com>
	<CAG0qc=2g6KtFPfFT2p1krAmh4LE-75rhKnnLFMH0ncAm22FnEw@mail.gmail.com>
	<512DC3A2.7050608@btinternet.com>
	<CAG0qc=2pTU1qrAMnOGyjkriTyfXbvMWC1QZpyfgodHoBNubfvw@mail.gmail.com>
	<4A1461B3-218E-42D6-90C9-465141810946@electricdruid.net>
	<512F71E4.5080500@bugbrand.co.uk>
	<CAG0qc=16Ggi0PSHJZpNPqQ7oMNVkzmi_VNAEEZEHM8Fy7hnuOg@mail.gmail.com>
Message-ID: <74393D4E-2AFF-4206-9A54-747D88C22A30@sounds.wa.com>


On Feb 28, 2013, at 07:25, Travis Thatcher wrote:
> I ended up working till late last night so I didn't have a chance to
> crack open my synth (which i'll do tonight) but assuming the muting
> transistors have gone south, would it be un-wise to replace the
> 2sc2878 with a regular 2n3904? Any other drop in replacements that'd
> work if not?

Well, the 2SC2878 claims to be designed specifically for Muting and  
Switching Applications, and the first bullet item in the data sheet  
is the high emitter-base voltage of 25 V. The 2N3904 only handles a 6  
V emitter-base voltage. In addition the collector current rating for  
the 2SC2878 is 300 mA, while the 2N3904 only handles 200 mA. The  
2N3904 has an order of magnitude more bandwidth, but it's probably  
not needed. Of course, the JX-3P circuit may not exceed the ratings  
of the 2N3904, but I'd be hesitant to replace a transistor with  
something different if it doesn't at least match the ratings.

I think that you ended up solving this problem, so I won't try to  
find a transistor that seems to match the capabilities of the 2SC2878  
(which is not available at Mouser).


> Also, would it be really harmful to just bypass the muting circuit
> altogether? Assuming I kept the volume on my mixer low on power
> on/off?

I have "fixed" certain home stereo equipment by removing the muting  
circuit. It certainly doesn't harm the normal operations. I  
instructed the owner to turn off his power amplifier before turning  
off the audio device, explaining that I'd removed one muting circuit  
(since it failed in the mute operation mode).

If it were my JX-3P, I'd want to eventually fix the thing back to  
factory original, but I wouldn't feel too bad using it with the mute  
defeated while waiting on parts.

Brian Willoughby
Sound Consulting



From aym-htnys at teaser.fr  Sun Mar  3 12:23:11 2013
From: aym-htnys at teaser.fr (Andre Majorel)
Date: Sun, 3 Mar 2013 12:23:11 +0100
Subject: [sdiy] VCA response - continously variable between linearand
 exponential
In-Reply-To: <71AB17E6-1241-4C89-8094-F6FCB3D080F3@sounds.wa.com>
References: <a11924c9-1f0e-4c64-94d4-ebc7d3391e3a@bvea33570>
	<C0E60B43-8135-4A91-A73E-10D8BDE9281D@sounds.wa.com>
	<9D5772D144144B7C8DB83140C8F9537E@galvanox01>
	<71AB17E6-1241-4C89-8094-F6FCB3D080F3@sounds.wa.com>
Message-ID: <20130303112311.GA14520@aym.net2.nerim.net>

On 2013-03-02 20:49 -0800, rsdio at sounds.wa.com wrote:

> In other words, I again ask: Why would you want anything besides
> linear or pure exponential? Something in between doesn't seem
> terribly useful. (not that a synth designer shouldn't create
> additional parameters for variation)

I'm not sure purely exponential attacks and decays sound most
natural. Brass and strings swells start fast and end slow. So do
decays of piano strings, not just because higher partials decay
faster but also because there are two planes of vibration with
different decay rates (see _The coupled motions of piano
strings_ by Gabriel Weinreich, Scientific American vol. 240, no.
1, 1979-01).

VCA with weird responses might be useful in getting the desired
dynamics when driven by simple ADSR envelope generators.

-- 
Andr? Majorel http://www.teaser.fr/~amajorel/


From gjames at kddlab.com  Sun Mar  3 16:23:32 2013
From: gjames at kddlab.com (Greg James)
Date: Sun, 3 Mar 2013 10:23:32 -0500
Subject: [sdiy] New DIY supplier
In-Reply-To: <EF9A62A1F17F4CE9A31752136F8E0015@KDDLab05>
References: <002c01ce1753$dbf89970$93e9cc50$@airmail.net>
	<EF9A62A1F17F4CE9A31752136F8E0015@KDDLab05>
Message-ID: <A58E87DFBD714BFEBDDBDA5F29DDAD0F@KDDLab05>

So I see they lowered the price on the '910 PCB back to $19.00. That's much
better.

I'm still wondering if they are authorized to make new runs of the MOTM 1.0
PCBs (i.e. become a reliable future source).

Greg James

-----Original Message-----
From: synth-diy-bounces at dropmix.xs4all.nl
[mailto:synth-diy-bounces at dropmix.xs4all.nl] On Behalf Of Greg James
Sent: Saturday, March 02, 2013 8:56 PM
To: 'Paul Schreiber'; 'synthdiy diy'
Subject: Re: [sdiy] New DIY supplier

Some prices are reasonable and some are, well.

$29.00 for a '910 PCB?

Are these NOS/End-of-life thru hole, or are they/you still making new runs
of the old PCBs?

Greg James

-----Original Message-----
From: synth-diy-bounces at dropmix.xs4all.nl
[mailto:synth-diy-bounces at dropmix.xs4all.nl] On Behalf Of Paul Schreiber
Sent: Saturday, March 02, 2013 9:40 AM
To: synthdiy diy
Subject: [sdiy] New DIY supplier

Check out SynthCube:

www.synthcube.com

They have MOTM, MU, some Frac and other assorted pc boards, parts, panels
and stuff. More to be added daily.

Paul S.


_______________________________________________
Synth-diy mailing list
Synth-diy at dropmix.xs4all.nl
http://dropmix.xs4all.nl/mailman/listinfo/synth-diy

_______________________________________________
Synth-diy mailing list
Synth-diy at dropmix.xs4all.nl
http://dropmix.xs4all.nl/mailman/listinfo/synth-diy



From don at till.com  Sun Mar  3 16:23:35 2013
From: don at till.com (Donald Tillman)
Date: Sun, 3 Mar 2013 07:23:35 -0800
Subject: [sdiy] VCA response - continously variable between linear and
	exponential
In-Reply-To: <1362193119.1910.135.camel@nwwtlaptop>
References: <1362193119.1910.135.camel@nwwtlaptop>
Message-ID: <6727B884-C8D3-48CD-A4CA-5C51C6232E69@till.com>

On Mar 1, 2013, at 6:58 PM, neil harper <metadata at gmx.com> wrote:

> i'm looking for some advice/pointers on how to get a VCA response curve
> to be continuously variable between linear and exponential. i can do
> switchable between the two, but is there any trick to getting everything
> in between?
> 
> if it helps, my VCA is lm13700 based but i don't mind starting with
> something else.


The ARP2600 addressed this with separate linear and exponential control voltage inputs.  Each has an input pot, so it's a simple matter to pipe a control voltage into both inputs and adjust the contributions of each.  That should be easy to implement.

Another approach is for a pot to add some resistance in series with the emitter of the transistor doing the exponentiation.

It may be good to think about exactly what sort of transition between exponential and linear would be the most musically useful.


  -- Don

--
Don Tillman
Palo Alto, California
don at till.com
http://www.till.com






From synth1 at airmail.net  Sun Mar  3 16:30:39 2013
From: synth1 at airmail.net (Paul Schreiber)
Date: Sun, 3 Mar 2013 09:30:39 -0600
Subject: [sdiy] New DIY supplier
In-Reply-To: <A58E87DFBD714BFEBDDBDA5F29DDAD0F@KDDLab05>
References: <002c01ce1753$dbf89970$93e9cc50$@airmail.net>
	<EF9A62A1F17F4CE9A31752136F8E0015@KDDLab05>
	<A58E87DFBD714BFEBDDBDA5F29DDAD0F@KDDLab05>
Message-ID: <003601ce1824$0fb12540$2f136fc0$@airmail.net>

I'm still wondering if they are authorized to make new runs of the MOTM 1.0
PCBs (i.e. become a reliable future source).
============================================================================
================

No, BrideChamber owns those rights (the Gerber files to run existing
boards).

This is not to say that additional offerings may be forthcoming that address
both DIY people and assembled module people that are not exclusive to
BrideChamber.

The MOTM 1.0 pc boards offered by SynthCube were purchased from proper
channels so they are 'legit'.

Paul S.




From dixon at mail.ubc.ca  Sun Mar  3 20:21:48 2013
From: dixon at mail.ubc.ca (David G Dixon)
Date: Sun, 3 Mar 2013 11:21:48 -0800
Subject: [sdiy] VCA response - continously variable between
	linearand	exponential
In-Reply-To: <71AB17E6-1241-4C89-8094-F6FCB3D080F3@sounds.wa.com>
References: <a11924c9-1f0e-4c64-94d4-ebc7d3391e3a@bvea33570>
	<C0E60B43-8135-4A91-A73E-10D8BDE9281D@sounds.wa.com>
	<9D5772D144144B7C8DB83140C8F9537E@galvanox01>
	<71AB17E6-1241-4C89-8094-F6FCB3D080F3@sounds.wa.com>
Message-ID: <E62A6A2D7C5A457B9DA9F9EDBFAA61DB@galvanox01>

You make a good point, and I've built many VCAs with Expo/Linear switches.
I personally almost never use the Expo mode.  However, the exponential mode
is useful (other than when being driven by linear envelopes) for percussive
sounds or for just giving some "pop" to amplitude envelopes, but pure
exponential can be very harsh, giving a sharp spike in gain right at the end
of the envelope.  In that case, it's nice to be able to temper the
exponential response a bit with some linearity.

> Ha!
> 
> I totally forgot about the Emily Litella character, or at 
> least the name, but I get it now. I totally missed the boat 
> on the VCO / VCA distinction. Usually I catch myself before I 
> write an entire essay.
> 
> But in an attempt to save a little face, isn't exponential 
> VCA response an attempt to cater to the human hearing 
> system's perception of intensity as a logarithmic function? 
> We hear volume in relative terms, not absolute. Decibels are 
> logarithmic, and almost perfectly map to our perception of 
> intensity of sound.
> 
> In other words, I again ask: Why would you want anything 
> besides linear or pure exponential? Something in between 
> doesn't seem terribly useful. (not that a synth designer 
> shouldn't create additional parameters for variation)
> 
> Brian
> 
> 
> On Mar 2, 2013, at 19:36, David G Dixon wrote:
> >
> > I think we're having an "Emily Litella moment" here:
> >
> > We're talking about VCAs, not VCOs!
> 



From tom at electricdruid.net  Mon Mar  4 01:54:13 2013
From: tom at electricdruid.net (Tom Wiltshire)
Date: Mon, 4 Mar 2013 00:54:13 +0000
Subject: [sdiy] VCA response - continously variable between linear
	andexponential
In-Reply-To: <CA+9GZUiRo8aH-qq_k+F02NM_p0G8Tp7N3TNj14zm9Oo8Sv+9wA@mail.gmail.com>
References: <1362193119.1910.135.camel@nwwtlaptop>
	<9BF8A6C09E98466CAE6854E363434E2A@galvanox01>
	<CA+9GZUiRo8aH-qq_k+F02NM_p0G8Tp7N3TNj14zm9Oo8Sv+9wA@mail.gmail.com>
Message-ID: <8C34858A-DE63-4739-81E0-4C450F94FB6F@electricdruid.net>

Hi Damian,

On 2 Mar 2013, at 07:28, cheater cheater <cheater00 at gmail.com> wrote:

> David, that may look like a good idea, but it's different - it's
> mixing lin and exp. It's not going to give you the same range of
> sounds you could get from a circuit which can vary the exponential
> base.

Why not?

I'm pretty sure David is right, since he's not mixing between a lin and a log/expo *output* from a VCA, but rather producing a CV which is a blend of lin and log going in.

But then I thought about it some more and wondered if crossfading between the output of a lin VCA and a expo VCA isn't the same thing anyway. I haven't done the maths to prove it either way.

Interesting question though?maybe I *will* have to do the maths.

Regards,
Tom



From sdiy-mz01 at blauwurf.info  Mon Mar  4 02:13:29 2013
From: sdiy-mz01 at blauwurf.info (Michael Zacherl)
Date: Mon, 4 Mar 2013 02:13:29 +0100
Subject: [sdiy] Polivoks demos (was: plain text)
In-Reply-To: <000601ce154a$da666570$8f333050$@earthlink.net>
References: <001201ce0c83$e1df25f0$a59d71d0$@earthlink.net>
	<AD5036DE-4AAA-42E1-BB64-31FB90DF5680@blauwurf.info>
	<000a01ce0c88$f5f62d40$e1e287c0$@earthlink.net>
	<99553ACA-705F-45A6-A71D-DA47E8B402FE@blauwurf.info>
	<000601ce154a$da666570$8f333050$@earthlink.net>
Message-ID: <EDF193E0-7BE9-407E-AE52-54B462CBFD63@blauwurf.info>


On 28.2.2013, at 01:30 , Dave Brown wrote:

> I posted a short video with some sounds on my site. - Dave

great!  thanks! Michael.



> -----Original Message-----
> From: Michael Zacherl [mailto:sdiy-mz01 at blauwurf.info] 
> Sent: Saturday, February 16, 2013 1:22 PM
> To: Dave Brown
> Subject: Re: [sdiy] plain text
> 
> good!
> btw, the Polivoks looks great! Curious about sounds, I am as well!  ;-) m.
> 
> On 16.2.2013, at 22:02 , Dave Brown wrote:
> 
>> Thanks.  That will save me at least two emails for each of my postings.
>> Interesting though, Outlook still formats it as html so it must 
>> convert it on sending.
>> 
>> Dave
>> 
>> -----Original Message-----
>> From: Michael Zacherl [mailto:sdiy-mz01 at blauwurf.info]
>> Sent: Saturday, February 16, 2013 12:57 PM
>> To: Dave Brown
>> Subject: Re: [sdiy] plain text
>> 
>> 'get through' in terms of other can read it?
>> yeah, it did!  ;-)  m.
>> 
>> On 16.2.2013, at 21:26 , Dave Brown wrote:
>> 
>>> I usually forget to set my email reader to plain text before sending 
>>> a message to the group.  I found that I could associate the message 
>>> format with the address in Outlook.  If this gets through to the list 
>>> then that feature works.  It sure will be more reliable than my memory.
>>> 
>>> 
>>> 
>>> David J. Brown
>>> 
>>> Email davebr at earthlink.net
>>> 
>>> Website http://modularsynthesis.com
>>> 
>>> 
>>> 
>>> 
>>> 
>>> _______________________________________________
>>> Synth-diy mailing list
>>> Synth-diy at dropmix.xs4all.nl
>>> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>> 
>> 
>> 
>> --
>> feed your perception: http://blauwurf.at 
>> http://soundcloud.com/noiseconformist
>> 
>> 
>> 
>> 
>> 
> 
> 
> 
> --
> feed your perception: http://blauwurf.at
> http://soundcloud.com/noiseconformist
> 
> 
> 
> 
> 



--
nonconform? noiseconform: http://blauwurf.at
http://soundcloud.com/noiseconformist






From gjames at kddlab.com  Mon Mar  4 05:26:36 2013
From: gjames at kddlab.com (Greg James)
Date: Sun, 3 Mar 2013 23:26:36 -0500
Subject: [sdiy] New DIY supplier
In-Reply-To: <003601ce1824$0fb12540$2f136fc0$@airmail.net>
References: <002c01ce1753$dbf89970$93e9cc50$@airmail.net><EF9A62A1F17F4CE9A31752136F8E0015@KDDLab05><A58E87DFBD714BFEBDDBDA5F29DDAD0F@KDDLab05>
	<003601ce1824$0fb12540$2f136fc0$@airmail.net>
Message-ID: <DC281D391DBF48E796C01A2F46849A56@KDDLab05>

Thanks for the clarification.

-----Original Message-----
From: synth-diy-bounces at dropmix.xs4all.nl
[mailto:synth-diy-bounces at dropmix.xs4all.nl] On Behalf Of Paul Schreiber
Sent: Sunday, March 03, 2013 10:31 AM
To: 'synthdiy diy'
Subject: Re: [sdiy] New DIY supplier

I'm still wondering if they are authorized to make new runs of the MOTM 1.0
PCBs (i.e. become a reliable future source).
============================================================================
================

No, BrideChamber owns those rights (the Gerber files to run existing
boards).

This is not to say that additional offerings may be forthcoming that address
both DIY people and assembled module people that are not exclusive to
BrideChamber.

The MOTM 1.0 pc boards offered by SynthCube were purchased from proper
channels so they are 'legit'.

Paul S.


_______________________________________________
Synth-diy mailing list
Synth-diy at dropmix.xs4all.nl
http://dropmix.xs4all.nl/mailman/listinfo/synth-diy



From dougt55 at yahoo.com  Tue Mar  5 03:22:22 2013
From: dougt55 at yahoo.com (Doug Terrebonne)
Date: Mon, 4 Mar 2013 18:22:22 -0800 (PST)
Subject: [sdiy] OSCar transformer
Message-ID: <1362450142.81893.YahooMailRC@web181606.mail.ne1.yahoo.com>

Hi, I need to replace the power transformer in an OSCar. The original has dual 
9V secondary output s however the middle 2 taps are tied to ground so is there 
any reason I couldn't replace it with an 18V center-tapped?

PS schem on pages 5-6 here - 

http://elektron-users.com/index.php?option=com_docman&task=doc_download&gid=415&Itemid=30


Thanks,
Doug



From Stromeko at nexgo.de  Tue Mar  5 08:12:43 2013
From: Stromeko at nexgo.de (ASSI)
Date: Tue, 05 Mar 2013 08:12:43 +0100
Subject: [sdiy] OSCar transformer
In-Reply-To: <1362450142.81893.YahooMailRC@web181606.mail.ne1.yahoo.com>
References: <1362450142.81893.YahooMailRC@web181606.mail.ne1.yahoo.com>
Message-ID: <2030769.KhqhioAEtO@rainer>

On Monday 04 March 2013, 18:22:22, Doug Terrebonne wrote:
> Hi, I need to replace the power transformer in an OSCar. The original has
> dual 9V secondary output s however the middle 2 taps are tied to ground
> so is there any reason I couldn't replace it with an 18V center-tapped?

Center-tapped should be OK.  Depending on the load characteristic of the old 
one and the new one you might perhaps want to use a slightly lower output 
voltage (like 2x15V).  I'd definitely drop the voltage if the regulators 
were also replaced with low-dropout ones.  Also, the electrolytics after the 
rectifier would enjoy a longer life if they were 25V rated (modern ones 
should still be smaller than the 16V ones you have in there now).


Regards,
Achim.
-- 
+<[Q+ Matrix-12 WAVE#46+305 Neuron microQkb Andromeda XTk Blofeld]>+

Factory and User Sound Singles for Waldorf Q+, Q and microQ:
http://Synth.Stromeko.net/Downloads.html#WaldorfSounds



From cheater00 at gmail.com  Tue Mar  5 13:54:45 2013
From: cheater00 at gmail.com (cheater cheater)
Date: Tue, 5 Mar 2013 13:54:45 +0100
Subject: [sdiy] VCA response - continously variable between linear
	andexponential
In-Reply-To: <8C34858A-DE63-4739-81E0-4C450F94FB6F@electricdruid.net>
References: <1362193119.1910.135.camel@nwwtlaptop>
	<9BF8A6C09E98466CAE6854E363434E2A@galvanox01>
	<CA+9GZUiRo8aH-qq_k+F02NM_p0G8Tp7N3TNj14zm9Oo8Sv+9wA@mail.gmail.com>
	<8C34858A-DE63-4739-81E0-4C450F94FB6F@electricdruid.net>
Message-ID: <CA+9GZUjYrP-1UtAStZbX_M_gt+wbF9+MPsROEacQTXj0eEBOhA@mail.gmail.com>

Hi Tom and David,
in case of a linear VCA where you distribute gains before the expo
converter (envelope goes to voltage divider pot, which then goes into
amp env, and into exponentiator which goes into amp env):

the output of an amplifier of gain A is:
output = A*(u*env + exp(v*env))*input

which is equal to:
A*u*env*input + A*exp(v*env)*input

in case you mix after the exponentiator (envelope goes to 50% voltage
divider resistor network, one side of that goes through expo, then
this gets mixed with the other side of the voltage divider according
to a voltage divider pot):

the output is:
output = A*(u*env + v*exp(env))*input

which is equal to:
A*u*env*input + A*v*exp(env)*input

therefore, in the simplest model, it's exactly the same.



In case of an exponential VCA:

output = exp(A+(u*log(env) + (v*env))*input
which is equivalent to:
output = exp(A) * exp(u*log(env) + v*env)*input
and to:
output = exp(A) * env^u * exp(env)^v * input

which means it's like chaining a linear and then exponential VCA and
changing their respective depth of effect with the voltage divider
pot.

The equation you are looking for is:

output = (b^(A*env)-1)/(b-1) * input
where b is variable by a pot.

see graphs here: http://wolfr.am/XntYOd

They agree exactly with what Hoshuyama posted - thanks for the info, Hoshu!

Note that you can't achieve his result by just biasing the exponential
VCA. It ends up being the same thing for every bias, but with more and
more distortion as you go in one direction, and more noise if you go
in the other:

http://wolfr.am/YaofHk

Cheers,
D.

On Mon, Mar 4, 2013 at 1:54 AM, Tom Wiltshire <tom at electricdruid.net> wrote:
> Hi Damian,
>
> On 2 Mar 2013, at 07:28, cheater cheater <cheater00 at gmail.com> wrote:
>
>> David, that may look like a good idea, but it's different - it's
>> mixing lin and exp. It's not going to give you the same range of
>> sounds you could get from a circuit which can vary the exponential
>> base.
>
> Why not?
>
> I'm pretty sure David is right, since he's not mixing between a lin and a log/expo *output* from a VCA, but rather producing a CV which is a blend of lin and log going in.
>
> But then I thought about it some more and wondered if crossfading between the output of a lin VCA and a expo VCA isn't the same thing anyway. I haven't done the maths to prove it either way.
>
> Interesting question though?maybe I *will* have to do the maths.
>
> Regards,
> Tom
>


From cheater00 at gmail.com  Tue Mar  5 14:09:31 2013
From: cheater00 at gmail.com (cheater cheater)
Date: Tue, 5 Mar 2013 14:09:31 +0100
Subject: [sdiy] Polivoks filter
In-Reply-To: <000001ce17a5$d547d3a0$7fd77ae0$@earthlink.net>
References: <000001ce157f$bfbabfc0$3f303f40$@earthlink.net>
	<CA+9GZUiAux9_pPKFCqujkr1uxmjYMbkYZmYH=DZ-Ayu24TG7Xw@mail.gmail.com>
	<000001ce17a5$d547d3a0$7fd77ae0$@earthlink.net>
Message-ID: <CA+9GZUhTn5VG4+QwOrTGFv_wGfrKv+yhdT4kzo=-twq9PF4csw@mail.gmail.com>

Hi Dave, thanks a lot! Those are really nicely made. At some points
there's quite a bit of distortion especially in the LP demo. Is that
when you turn up the resonance, or is there some sort of gain
parameter you're playing with?

Cheers,
D.

On Sun, Mar 3, 2013 at 1:27 AM, Dave Brown <davebr at earthlink.net> wrote:
> I put a couple of wave files up on the site.  Nothing fancy but I go through all the controls. - Dave
>
> -----Original Message-----
> From: cheater cheater [mailto:cheater00 at gmail.com]
> Sent: Thursday, February 28, 2013 5:01 AM
> To: Dave Brown
> Cc: synth-diy
> Subject: Re: [sdiy] Polivoks filter
>
> Hi Dave,
> thanks a lot for the demo, but it's a bit difficult to appreciate the filter's sound in the setup you've used. If I could ask you, would you redo the demo, recording to line-in, and try to tune the filter an octave or two lower? Due to reverb and other acoustic properties the filter kind of sounds like any old filter :) and then it's generally difficult to figure out what a filter is doing if the input signal is pitched way up. The typical test is a sawtooth wave in the lower registers, going through a filter with a quick attack and long (0.5 -
> 1 second) decay and a lot of positive env mod.
>
> Cheers,
> D.
>
> On Thu, Feb 28, 2013 at 7:49 AM, Dave Brown <davebr at earthlink.net> wrote:
>> I received my front panel and finished building my Polivoks filter.  I updated the schematics and highlighted my few mods.  I put a 4 minute video showing some of the various modes and sounds.  I do like this filter a lot and I like having both the soft and hard resonance.  I like the soft resonance for traditional filter sound and the hard resonance for adding somewhat uncontrolled edge to the waveforms.  The tracking is pretty good although I haven't done a complete test of the range of tracking.  The beginning of the video demonstrates a sequencer driving both a VCO and the filter in self-oscillation which tracks pretty well.  Since there is no fine control the tuning is a bit tricky.
>>
>> http://modularsynthesis.com/kuzmin/polivoks/polivoks_vcf.htm
>>
>> Dave
>>
>> -----Original Message-----
>> From: synth-diy-bounces at dropmix.xs4all.nl
>> [mailto:synth-diy-bounces at dropmix.xs4all.nl] On Behalf Of Dave Brown
>> Sent: Saturday, February 16, 2013 12:25 PM
>> To: 'cheater cheater'
>> Cc: 'synth-diy'
>> Subject: Re: [sdiy] Polivoks filter
>>
>> I will.  Right now I think I will finish building it in a module before using or recording.  It will be a lot easier and I don't risk shorts with wires and cables flying everywhere.  A couple of times now the weight of all the cables has pulled my PCB to the floor ...
>>
>> Dave
>>
>> -----Original Message-----
>> From: cheater cheater [mailto:cheater00 at gmail.com]
>> Sent: Saturday, February 16, 2013 11:00 AM
>> To: Dave Brown
>> Cc: synth-diy
>> Subject: Re: [sdiy] Polivoks filter
>>
>> Hi Dave,
>> that looks great! Would you like to record some demos and post wav's somewhere?
>>
>> Thanks,
>> D.
>>
>> On Sat, Feb 16, 2013 at 7:11 PM, Dave Brown <davebr at earthlink.net> wrote:
>>> I just finished hand building a Polivoks filter using the Marc
>>> Bareille schematics with the reversing attenuator.  I ended up making
>>> a couple of minor changes to reduce the filter to near unity gain and
>>> softening the self-resonance output level so it would not clip rail to rail.
>>>
>>> I built this filter using NTE888M programmable op-amps since they are
>>> readily available.  I did some frequency plots with my scope using XY
>>> mode and was disappointed with the DC level shift I saw on the low pass output.
>>> I got some of the ??1208 Russian parts to compare and discovered the
>>> same DC shift.  I had repaired a Polivoks filter previously and the
>>> frequency plots did not show any DC level shift.  I eventually
>>> figured out the DC shift was occurring on the VCO output as I swept
>>> to high frequencies.  I tried different VCOs and repeated the
>>> measurements with a VCO that had a constant amplitude output with no
>>> DC shift over the frequency range of the measurement.  Both the
>>> NTE888M and ??1208 Russian parts looked good with no DC shift in the low pass output.
>>>
>>> I saw only two minor differences between the NTE888M parts and the
>>> ??1208 Russian parts.  The NTE888M parts have slightly higher gain
>>> and the cutoff or center frequencies was slightly higher at the same
>>> control settings and input CV.  I assume the frequency shift is due
>>> to a slightly different Iset response curve or a different Vbe bias point for the Iset control pin.
>>>
>>> I have some uA776 parts on order and will also do the same comparison
>>> (assuming the parts are genuine).
>>>
>>> Now I need to finish this filter with a front panel and play with it some.
>>> Details, photos, and scope images are on my site on my Polivoks Filter page.
>>> http://modularsynthesis.com/kuzmin/polivoks/polivoks_vcf.htm
>>>
>>> David J. Brown
>>> Email davebr at earthlink.net
>>> Website http://modularsynthesis.com
>>
>>
>>
>> _______________________________________________
>> Synth-diy mailing list
>> Synth-diy at dropmix.xs4all.nl
>> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>
>


From jacobwatters at gmail.com  Tue Mar  5 15:25:40 2013
From: jacobwatters at gmail.com (Jacob Watters)
Date: Tue, 5 Mar 2013 09:25:40 -0500
Subject: [sdiy] monotron DC power
In-Reply-To: <CAEQ8eXPChR+DUA8zWoU7Oi=PMSqDGp_FJ4Vg6powJ+oYsZHHDA@mail.gmail.com>
References: <CAEQ8eXPChR+DUA8zWoU7Oi=PMSqDGp_FJ4Vg6powJ+oYsZHHDA@mail.gmail.com>
Message-ID: <CAKcdib10p9-2FMJ-jBvq_y-tkqDo5T+G7RJdfRgXd6BB8G=0-A@mail.gmail.com>

You can power it from 5V, but connect the supply to the +V solder
point and NOT to where the batteries go.

Also, if you do this then you should not have batteries in the
Monotron at the same time.

There are a couple of links to the schematics I used near the bottom
of this page: http://www.jacobwatters.com/monotron-mod.php

Let me know if you have any other questions.  I have helped a few
other people already who contacted me directly.


-Jacob Watters


On Tue, Mar 5, 2013 at 9:00 AM, Marc Nostromo [M-.-n]
<marc.nostromo at gmail.com> wrote:
> Hey Jacob
>
> I'm toying with modding my monotron and would like to power it from
> DC. I've done quite a bit of research  and could not find if it can be
> powered directly from 5v or if I need to use a power regulator to
> bring the voltage to 3v.
>
> Could you tell me what you've done ?
>
> Thanks a lot,
> Marc.
>
> --
> http://marc-nostromo.com


From davebr at earthlink.net  Tue Mar  5 19:56:42 2013
From: davebr at earthlink.net (Dave Brown)
Date: Tue, 5 Mar 2013 10:56:42 -0800
Subject: [sdiy] Polivoks filter
In-Reply-To: <CA+9GZUhTn5VG4+QwOrTGFv_wGfrKv+yhdT4kzo=-twq9PF4csw@mail.gmail.com>
References: <000001ce157f$bfbabfc0$3f303f40$@earthlink.net>
	<CA+9GZUiAux9_pPKFCqujkr1uxmjYMbkYZmYH=DZ-Ayu24TG7Xw@mail.gmail.com>
	<000001ce17a5$d547d3a0$7fd77ae0$@earthlink.net>
	<CA+9GZUhTn5VG4+QwOrTGFv_wGfrKv+yhdT4kzo=-twq9PF4csw@mail.gmail.com>
Message-ID: <000f01ce19d3$2dc608b0$89521a10$@earthlink.net>

I'm only playing with the Polivoks filter controls.  You can hear the hard resonance break into clipping around 0:49 and again at 1:50.  I think that is what you are talking about.  The output goes from +/-10V to +/-29 volts.  I added comments about the hard resonance clipping to both the LP and BP demo.

Dave

-----Original Message-----
From: cheater cheater [mailto:cheater00 at gmail.com] 
Sent: Tuesday, March 05, 2013 5:10 AM
To: Dave Brown; synth-diy
Subject: Re: [sdiy] Polivoks filter

Hi Dave, thanks a lot! Those are really nicely made. At some points there's quite a bit of distortion especially in the LP demo. Is that when you turn up the resonance, or is there some sort of gain parameter you're playing with?

Cheers,
D.

On Sun, Mar 3, 2013 at 1:27 AM, Dave Brown <davebr at earthlink.net> wrote:
> I put a couple of wave files up on the site.  Nothing fancy but I go 
> through all the controls. - Dave
>
> -----Original Message-----
> From: cheater cheater [mailto:cheater00 at gmail.com]
> Sent: Thursday, February 28, 2013 5:01 AM
> To: Dave Brown
> Cc: synth-diy
> Subject: Re: [sdiy] Polivoks filter
>
> Hi Dave,
> thanks a lot for the demo, but it's a bit difficult to appreciate the 
> filter's sound in the setup you've used. If I could ask you, would you 
> redo the demo, recording to line-in, and try to tune the filter an 
> octave or two lower? Due to reverb and other acoustic properties the 
> filter kind of sounds like any old filter :) and then it's generally 
> difficult to figure out what a filter is doing if the input signal is 
> pitched way up. The typical test is a sawtooth wave in the lower 
> registers, going through a filter with a quick attack and long (0.5 -
> 1 second) decay and a lot of positive env mod.
>
> Cheers,
> D.
>
> On Thu, Feb 28, 2013 at 7:49 AM, Dave Brown <davebr at earthlink.net> wrote:
>> I received my front panel and finished building my Polivoks filter.  I updated the schematics and highlighted my few mods.  I put a 4 minute video showing some of the various modes and sounds.  I do like this filter a lot and I like having both the soft and hard resonance.  I like the soft resonance for traditional filter sound and the hard resonance for adding somewhat uncontrolled edge to the waveforms.  The tracking is pretty good although I haven't done a complete test of the range of tracking.  The beginning of the video demonstrates a sequencer driving both a VCO and the filter in self-oscillation which tracks pretty well.  Since there is no fine control the tuning is a bit tricky.
>>
>> http://modularsynthesis.com/kuzmin/polivoks/polivoks_vcf.htm
>>
>> Dave
>>
>> -----Original Message-----
>> From: synth-diy-bounces at dropmix.xs4all.nl
>> [mailto:synth-diy-bounces at dropmix.xs4all.nl] On Behalf Of Dave Brown
>> Sent: Saturday, February 16, 2013 12:25 PM
>> To: 'cheater cheater'
>> Cc: 'synth-diy'
>> Subject: Re: [sdiy] Polivoks filter
>>
>> I will.  Right now I think I will finish building it in a module before using or recording.  It will be a lot easier and I don't risk shorts with wires and cables flying everywhere.  A couple of times now the weight of all the cables has pulled my PCB to the floor ...
>>
>> Dave
>>
>> -----Original Message-----
>> From: cheater cheater [mailto:cheater00 at gmail.com]
>> Sent: Saturday, February 16, 2013 11:00 AM
>> To: Dave Brown
>> Cc: synth-diy
>> Subject: Re: [sdiy] Polivoks filter
>>
>> Hi Dave,
>> that looks great! Would you like to record some demos and post wav's somewhere?
>>
>> Thanks,
>> D.
>>
>> On Sat, Feb 16, 2013 at 7:11 PM, Dave Brown <davebr at earthlink.net> wrote:
>>> I just finished hand building a Polivoks filter using the Marc 
>>> Bareille schematics with the reversing attenuator.  I ended up 
>>> making a couple of minor changes to reduce the filter to near unity 
>>> gain and softening the self-resonance output level so it would not clip rail to rail.
>>>
>>> I built this filter using NTE888M programmable op-amps since they 
>>> are readily available.  I did some frequency plots with my scope 
>>> using XY mode and was disappointed with the DC level shift I saw on the low pass output.
>>> I got some of the ??1208 Russian parts to compare and discovered the 
>>> same DC shift.  I had repaired a Polivoks filter previously and the 
>>> frequency plots did not show any DC level shift.  I eventually 
>>> figured out the DC shift was occurring on the VCO output as I swept 
>>> to high frequencies.  I tried different VCOs and repeated the 
>>> measurements with a VCO that had a constant amplitude output with no 
>>> DC shift over the frequency range of the measurement.  Both the 
>>> NTE888M and ??1208 Russian parts looked good with no DC shift in the low pass output.
>>>
>>> I saw only two minor differences between the NTE888M parts and the
>>> ??1208 Russian parts.  The NTE888M parts have slightly higher gain 
>>> and the cutoff or center frequencies was slightly higher at the same 
>>> control settings and input CV.  I assume the frequency shift is due 
>>> to a slightly different Iset response curve or a different Vbe bias point for the Iset control pin.
>>>
>>> I have some uA776 parts on order and will also do the same 
>>> comparison (assuming the parts are genuine).
>>>
>>> Now I need to finish this filter with a front panel and play with it some.
>>> Details, photos, and scope images are on my site on my Polivoks Filter page.
>>> http://modularsynthesis.com/kuzmin/polivoks/polivoks_vcf.htm
>>>
>>> David J. Brown
>>> Email davebr at earthlink.net
>>> Website http://modularsynthesis.com





From xyzzy at sysabend.org  Tue Mar  5 21:30:02 2013
From: xyzzy at sysabend.org (Tom Arnold)
Date: Tue, 5 Mar 2013 12:30:02 -0800
Subject: [sdiy] New DIY supplier
In-Reply-To: <002c01ce1753$dbf89970$93e9cc50$@airmail.net>
References: <002c01ce1753$dbf89970$93e9cc50$@airmail.net>
Message-ID: <20130305203002.GH26634@moo.sysabend.org>

On Sat, Mar 02, 2013 at 08:40:17AM -0600, Paul Schreiber wrote:
> Check out SynthCube:
> 
> www.synthcube.com

Interesting site, lots of good stuff, be nice if they had a bit more
information about themselves.  I'd rate sending money to them right up there
with dealing with a Chinese supplier with no feedback on Ebay.  Maybe an
about page with useful info?  I dunno.  Its a gripe of mine.

-- 
 -----------------------------------------------------------------
 - Tom Arnold       -  Free Synth DIY webspace http://www.sdiy.org 
 - SynthGeek        -  "...is it a virus, a drug, or a religion?"
 - K6AET            -  Juanita Shrugs. "What's the difference? 
 --------------------         -- Neal Stephenson,  Snow Crash


From eidorian at aladan.net  Tue Mar  5 21:42:41 2013
From: eidorian at aladan.net (Adrian Corston)
Date: Wed, 06 Mar 2013 07:12:41 +1030
Subject: [sdiy] New DIY supplier
In-Reply-To: <20130305203002.GH26634@moo.sysabend.org>
References: <002c01ce1753$dbf89970$93e9cc50$@airmail.net>
	<20130305203002.GH26634@moo.sysabend.org>
Message-ID: <513658C1.6020006@aladan.net>

Synthcobe is relatively well known on Muffwigglers, perhaps check there 
with people you know?

Cheers,
A.


On 6/03/13 7:00 AM, Tom Arnold wrote:
> On Sat, Mar 02, 2013 at 08:40:17AM -0600, Paul Schreiber wrote:
>> Check out SynthCube:
>>
>> www.synthcube.com
> Interesting site, lots of good stuff, be nice if they had a bit more
> information about themselves.  I'd rate sending money to them right up there
> with dealing with a Chinese supplier with no feedback on Ebay.  Maybe an
> about page with useful info?  I dunno.  Its a gripe of mine.
>



From blacet at blacet.com  Tue Mar  5 22:04:55 2013
From: blacet at blacet.com (blacet at blacet.com)
Date: Tue, 5 Mar 2013 13:04:55 -0800
Subject: [sdiy] New DIY supplier
In-Reply-To: <513658C1.6020006@aladan.net>
References: <002c01ce1753$dbf89970$93e9cc50$@airmail.net>
	<20130305203002.GH26634@moo.sysabend.org>
	<513658C1.6020006@aladan.net>
Message-ID: <af049473c47e0b8132952ae80f681c26.squirrel@webmail.blacet.com>

I've had a couple dealings with him. Seemed quite professional.

One of my pet peeves with any web site is the lack of where the business
is located. You can sometimes find a physical address by digging around
but sometimes not. They don't want to loose any business due to being too
far away from the customer; effecting shipping charges/times for example.

I've even called a few and met with paranoia. In one case, they only could
say that "they had warehouses all over".


> Synthcobe is relatively well known on Muffwigglers, perhaps check there
> with people you know?
>
> Cheers,
> A.
>
>
> On 6/03/13 7:00 AM, Tom Arnold wrote:
>> On Sat, Mar 02, 2013 at 08:40:17AM -0600, Paul Schreiber wrote:
>>> Check out SynthCube:
>>>
>>> www.synthcube.com
>> Interesting site, lots of good stuff, be nice if they had a bit more
>> information about themselves.  I'd rate sending money to them right up
>> there
>> with dealing with a Chinese supplier with no feedback on Ebay.  Maybe an
>> about page with useful info?  I dunno.  Its a gripe of mine.
>>
>
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>




From dixon at mail.ubc.ca  Tue Mar  5 22:29:51 2013
From: dixon at mail.ubc.ca (David G Dixon)
Date: Tue, 5 Mar 2013 13:29:51 -0800
Subject: [sdiy] New DIY supplier
In-Reply-To: <af049473c47e0b8132952ae80f681c26.squirrel@webmail.blacet.com>
References: <002c01ce1753$dbf89970$93e9cc50$@airmail.net><20130305203002.GH26634@moo.sysabend.org><513658C1.6020006@aladan.net>
	<af049473c47e0b8132952ae80f681c26.squirrel@webmail.blacet.com>
Message-ID: <B1A1E01A4B0043469F554BBCB782410E@galvanox01>

I feel exactly the same way.  I want to know who I'm dealing with.  The
first thing I look for on a website is some information about the specific
individuals running the company.

> One of my pet peeves with any web site is the lack of where 
> the business is located. You can sometimes find a physical 
> address by digging around but sometimes not. They don't want 
> to loose any business due to being too far away from the 
> customer; effecting shipping charges/times for example.
> 
> I've even called a few and met with paranoia. In one case, 
> they only could say that "they had warehouses all over".



From xyzzy at sysabend.org  Tue Mar  5 22:32:53 2013
From: xyzzy at sysabend.org (Tom Arnold)
Date: Tue, 5 Mar 2013 13:32:53 -0800
Subject: [sdiy] New DIY supplier
In-Reply-To: <20130305134538.3a027d564b495c443912516e20da7489.fbc5df06f9.wbe@email15.secureserver.net>
References: <20130305134538.3a027d564b495c443912516e20da7489.fbc5df06f9.wbe@email15.secureserver.net>
Message-ID: <20130305213253.GI26634@moo.sysabend.org>

On Tue, Mar 05, 2013 at 01:45:41PM -0700, info at synthcube.com wrote:
>    thanks for the feedback
> 
>    We have been loading inventory and have not yet written the 'about us'
>    page yet- but its coming :)
> 
>    we're not from china, we're from boston (concord actually)
> 
>    ebay feedback has been 100% positive
> 
>    we hope to keep developing a positive reputation over time

Thats the answer I was hoping for. 

Thanks!

-- 
 --------------------------------------------------------------------
 - Tom Arnold           -  "...is it a virus, a drug, or a religion?"  
 - Sysabend Caretaker   -    Juanita Shrugs. "What's the difference?"
 ------------------------       -- Neal Stephenson,  Snow Crash 



From jays at aracnet.com  Tue Mar  5 22:37:07 2013
From: jays at aracnet.com (Jay Schwichtenberg)
Date: Tue, 5 Mar 2013 13:37:07 -0800
Subject: [sdiy] New DIY supplier
In-Reply-To: <af049473c47e0b8132952ae80f681c26.squirrel@webmail.blacet.com>
References: <002c01ce1753$dbf89970$93e9cc50$@airmail.net>	<20130305203002.GH26634@moo.sysabend.org>	<513658C1.6020006@aladan.net>
	<af049473c47e0b8132952ae80f681c26.squirrel@webmail.blacet.com>
Message-ID: <000101ce19e9$96376450$c2a62cf0$@com>

It's getting real hard to figure that out anymore. I had some photo work
done recently. I'd looked around on the internet for places that did what I
wanted and there were a number of them. I saw one that was in Washington
which was closer to me than any of the others. Checked them out and
everything seemed ok. Placed my order and about 5 days latter I got a
package from Thailand.


-----Original Message-----
From: synth-diy-bounces at dropmix.xs4all.nl
[mailto:synth-diy-bounces at dropmix.xs4all.nl] On Behalf Of blacet at blacet.com
Sent: Tuesday, March 05, 2013 1:05 PM
To: synth-diy at dropmix.xs4all.nl
Subject: Re: [sdiy] New DIY supplier

I've had a couple dealings with him. Seemed quite professional.

One of my pet peeves with any web site is the lack of where the business
is located. You can sometimes find a physical address by digging around
but sometimes not. They don't want to loose any business due to being too
far away from the customer; effecting shipping charges/times for example.

I've even called a few and met with paranoia. In one case, they only could
say that "they had warehouses all over".


> Synthcobe is relatively well known on Muffwigglers, perhaps check there
> with people you know?
>
> Cheers,
> A.
>
>
> On 6/03/13 7:00 AM, Tom Arnold wrote:
>> On Sat, Mar 02, 2013 at 08:40:17AM -0600, Paul Schreiber wrote:
>>> Check out SynthCube:
>>>
>>> www.synthcube.com
>> Interesting site, lots of good stuff, be nice if they had a bit more
>> information about themselves.  I'd rate sending money to them right up
>> there
>> with dealing with a Chinese supplier with no feedback on Ebay.  Maybe an
>> about page with useful info?  I dunno.  Its a gripe of mine.
>>
>
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>


_______________________________________________
Synth-diy mailing list
Synth-diy at dropmix.xs4all.nl
http://dropmix.xs4all.nl/mailman/listinfo/synth-diy




From cheater00 at gmail.com  Wed Mar  6 14:20:38 2013
From: cheater00 at gmail.com (cheater cheater)
Date: Wed, 6 Mar 2013 14:20:38 +0100
Subject: [sdiy] Polivoks filter
In-Reply-To: <000f01ce19d3$2dc608b0$89521a10$@earthlink.net>
References: <000001ce157f$bfbabfc0$3f303f40$@earthlink.net>
	<CA+9GZUiAux9_pPKFCqujkr1uxmjYMbkYZmYH=DZ-Ayu24TG7Xw@mail.gmail.com>
	<000001ce17a5$d547d3a0$7fd77ae0$@earthlink.net>
	<CA+9GZUhTn5VG4+QwOrTGFv_wGfrKv+yhdT4kzo=-twq9PF4csw@mail.gmail.com>
	<000f01ce19d3$2dc608b0$89521a10$@earthlink.net>
Message-ID: <CA+9GZUgvdheRqnYANVHO_Dn6oVOo-gtH87+XJbPWwD=Zc=CpOA@mail.gmail.com>

interesting. why is there "hard" and "soft" resonance? are those two
available in parallel at the same time, or are they selectable via
switch?

Cheers,
D.

On Tue, Mar 5, 2013 at 7:56 PM, Dave Brown <davebr at earthlink.net> wrote:
> I'm only playing with the Polivoks filter controls.  You can hear the hard resonance break into clipping around 0:49 and again at 1:50.  I think that is what you are talking about.  The output goes from +/-10V to +/-29 volts.  I added comments about the hard resonance clipping to both the LP and BP demo.
>
> Dave
>
> -----Original Message-----
> From: cheater cheater [mailto:cheater00 at gmail.com]
> Sent: Tuesday, March 05, 2013 5:10 AM
> To: Dave Brown; synth-diy
> Subject: Re: [sdiy] Polivoks filter
>
> Hi Dave, thanks a lot! Those are really nicely made. At some points there's quite a bit of distortion especially in the LP demo. Is that when you turn up the resonance, or is there some sort of gain parameter you're playing with?
>
> Cheers,
> D.
>
> On Sun, Mar 3, 2013 at 1:27 AM, Dave Brown <davebr at earthlink.net> wrote:
>> I put a couple of wave files up on the site.  Nothing fancy but I go
>> through all the controls. - Dave
>>
>> -----Original Message-----
>> From: cheater cheater [mailto:cheater00 at gmail.com]
>> Sent: Thursday, February 28, 2013 5:01 AM
>> To: Dave Brown
>> Cc: synth-diy
>> Subject: Re: [sdiy] Polivoks filter
>>
>> Hi Dave,
>> thanks a lot for the demo, but it's a bit difficult to appreciate the
>> filter's sound in the setup you've used. If I could ask you, would you
>> redo the demo, recording to line-in, and try to tune the filter an
>> octave or two lower? Due to reverb and other acoustic properties the
>> filter kind of sounds like any old filter :) and then it's generally
>> difficult to figure out what a filter is doing if the input signal is
>> pitched way up. The typical test is a sawtooth wave in the lower
>> registers, going through a filter with a quick attack and long (0.5 -
>> 1 second) decay and a lot of positive env mod.
>>
>> Cheers,
>> D.
>>
>> On Thu, Feb 28, 2013 at 7:49 AM, Dave Brown <davebr at earthlink.net> wrote:
>>> I received my front panel and finished building my Polivoks filter.  I updated the schematics and highlighted my few mods.  I put a 4 minute video showing some of the various modes and sounds.  I do like this filter a lot and I like having both the soft and hard resonance.  I like the soft resonance for traditional filter sound and the hard resonance for adding somewhat uncontrolled edge to the waveforms.  The tracking is pretty good although I haven't done a complete test of the range of tracking.  The beginning of the video demonstrates a sequencer driving both a VCO and the filter in self-oscillation which tracks pretty well.  Since there is no fine control the tuning is a bit tricky.
>>>
>>> http://modularsynthesis.com/kuzmin/polivoks/polivoks_vcf.htm
>>>
>>> Dave
>>>
>>> -----Original Message-----
>>> From: synth-diy-bounces at dropmix.xs4all.nl
>>> [mailto:synth-diy-bounces at dropmix.xs4all.nl] On Behalf Of Dave Brown
>>> Sent: Saturday, February 16, 2013 12:25 PM
>>> To: 'cheater cheater'
>>> Cc: 'synth-diy'
>>> Subject: Re: [sdiy] Polivoks filter
>>>
>>> I will.  Right now I think I will finish building it in a module before using or recording.  It will be a lot easier and I don't risk shorts with wires and cables flying everywhere.  A couple of times now the weight of all the cables has pulled my PCB to the floor ...
>>>
>>> Dave
>>>
>>> -----Original Message-----
>>> From: cheater cheater [mailto:cheater00 at gmail.com]
>>> Sent: Saturday, February 16, 2013 11:00 AM
>>> To: Dave Brown
>>> Cc: synth-diy
>>> Subject: Re: [sdiy] Polivoks filter
>>>
>>> Hi Dave,
>>> that looks great! Would you like to record some demos and post wav's somewhere?
>>>
>>> Thanks,
>>> D.
>>>
>>> On Sat, Feb 16, 2013 at 7:11 PM, Dave Brown <davebr at earthlink.net> wrote:
>>>> I just finished hand building a Polivoks filter using the Marc
>>>> Bareille schematics with the reversing attenuator.  I ended up
>>>> making a couple of minor changes to reduce the filter to near unity
>>>> gain and softening the self-resonance output level so it would not clip rail to rail.
>>>>
>>>> I built this filter using NTE888M programmable op-amps since they
>>>> are readily available.  I did some frequency plots with my scope
>>>> using XY mode and was disappointed with the DC level shift I saw on the low pass output.
>>>> I got some of the ??1208 Russian parts to compare and discovered the
>>>> same DC shift.  I had repaired a Polivoks filter previously and the
>>>> frequency plots did not show any DC level shift.  I eventually
>>>> figured out the DC shift was occurring on the VCO output as I swept
>>>> to high frequencies.  I tried different VCOs and repeated the
>>>> measurements with a VCO that had a constant amplitude output with no
>>>> DC shift over the frequency range of the measurement.  Both the
>>>> NTE888M and ??1208 Russian parts looked good with no DC shift in the low pass output.
>>>>
>>>> I saw only two minor differences between the NTE888M parts and the
>>>> ??1208 Russian parts.  The NTE888M parts have slightly higher gain
>>>> and the cutoff or center frequencies was slightly higher at the same
>>>> control settings and input CV.  I assume the frequency shift is due
>>>> to a slightly different Iset response curve or a different Vbe bias point for the Iset control pin.
>>>>
>>>> I have some uA776 parts on order and will also do the same
>>>> comparison (assuming the parts are genuine).
>>>>
>>>> Now I need to finish this filter with a front panel and play with it some.
>>>> Details, photos, and scope images are on my site on my Polivoks Filter page.
>>>> http://modularsynthesis.com/kuzmin/polivoks/polivoks_vcf.htm
>>>>
>>>> David J. Brown
>>>> Email davebr at earthlink.net
>>>> Website http://modularsynthesis.com
>
>
>


From tom at electricdruid.net  Wed Mar  6 15:55:58 2013
From: tom at electricdruid.net (Tom Wiltshire)
Date: Wed, 6 Mar 2013 14:55:58 +0000
Subject: [sdiy] Polivoks filter
In-Reply-To: <CA+9GZUgvdheRqnYANVHO_Dn6oVOo-gtH87+XJbPWwD=Zc=CpOA@mail.gmail.com>
References: <000001ce157f$bfbabfc0$3f303f40$@earthlink.net>
	<CA+9GZUiAux9_pPKFCqujkr1uxmjYMbkYZmYH=DZ-Ayu24TG7Xw@mail.gmail.com>
	<000001ce17a5$d547d3a0$7fd77ae0$@earthlink.net>
	<CA+9GZUhTn5VG4+QwOrTGFv_wGfrKv+yhdT4kzo=-twq9PF4csw@mail.gmail.com>
	<000f01ce19d3$2dc608b0$89521a10$@earthlink.net>
	<CA+9GZUgvdheRqnYANVHO_Dn6oVOo-gtH87+XJbPWwD=Zc=CpOA@mail.gmail.com>
Message-ID: <7BF41A72-2F14-475F-945E-CB62E9B9336F@electricdruid.net>

Read his interesting and informative page on the topic:

	http://modularsynthesis.com/kuzmin/polivoks/polivoks_vcf.htm

The bit about the resonance and adding resonance limiting is about 1/4 the way down.

T.

On 6 Mar 2013, at 13:20, cheater cheater <cheater00 at gmail.com> wrote:

> interesting. why is there "hard" and "soft" resonance? are those two
> available in parallel at the same time, or are they selectable via
> switch?
> 
> Cheers,
> D.
> 
> On Tue, Mar 5, 2013 at 7:56 PM, Dave Brown <davebr at earthlink.net> wrote:
>> I'm only playing with the Polivoks filter controls.  You can hear the hard resonance break into clipping around 0:49 and again at 1:50.  I think that is what you are talking about.  The output goes from +/-10V to +/-29 volts.  I added comments about the hard resonance clipping to both the LP and BP demo.
>> 
>> Dave
>> 
>> -----Original Message-----
>> From: cheater cheater [mailto:cheater00 at gmail.com]
>> Sent: Tuesday, March 05, 2013 5:10 AM
>> To: Dave Brown; synth-diy
>> Subject: Re: [sdiy] Polivoks filter
>> 
>> Hi Dave, thanks a lot! Those are really nicely made. At some points there's quite a bit of distortion especially in the LP demo. Is that when you turn up the resonance, or is there some sort of gain parameter you're playing with?
>> 
>> Cheers,
>> D.
>> 
>> On Sun, Mar 3, 2013 at 1:27 AM, Dave Brown <davebr at earthlink.net> wrote:
>>> I put a couple of wave files up on the site.  Nothing fancy but I go
>>> through all the controls. - Dave
>>> 
>>> -----Original Message-----
>>> From: cheater cheater [mailto:cheater00 at gmail.com]
>>> Sent: Thursday, February 28, 2013 5:01 AM
>>> To: Dave Brown
>>> Cc: synth-diy
>>> Subject: Re: [sdiy] Polivoks filter
>>> 
>>> Hi Dave,
>>> thanks a lot for the demo, but it's a bit difficult to appreciate the
>>> filter's sound in the setup you've used. If I could ask you, would you
>>> redo the demo, recording to line-in, and try to tune the filter an
>>> octave or two lower? Due to reverb and other acoustic properties the
>>> filter kind of sounds like any old filter :) and then it's generally
>>> difficult to figure out what a filter is doing if the input signal is
>>> pitched way up. The typical test is a sawtooth wave in the lower
>>> registers, going through a filter with a quick attack and long (0.5 -
>>> 1 second) decay and a lot of positive env mod.
>>> 
>>> Cheers,
>>> D.
>>> 
>>> On Thu, Feb 28, 2013 at 7:49 AM, Dave Brown <davebr at earthlink.net> wrote:
>>>> I received my front panel and finished building my Polivoks filter.  I updated the schematics and highlighted my few mods.  I put a 4 minute video showing some of the various modes and sounds.  I do like this filter a lot and I like having both the soft and hard resonance.  I like the soft resonance for traditional filter sound and the hard resonance for adding somewhat uncontrolled edge to the waveforms.  The tracking is pretty good although I haven't done a complete test of the range of tracking.  The beginning of the video demonstrates a sequencer driving both a VCO and the filter in self-oscillation which tracks pretty well.  Since there is no fine control the tuning is a bit tricky.
>>>> 
>>>> http://modularsynthesis.com/kuzmin/polivoks/polivoks_vcf.htm
>>>> 
>>>> Dave
>>>> 
>>>> -----Original Message-----
>>>> From: synth-diy-bounces at dropmix.xs4all.nl
>>>> [mailto:synth-diy-bounces at dropmix.xs4all.nl] On Behalf Of Dave Brown
>>>> Sent: Saturday, February 16, 2013 12:25 PM
>>>> To: 'cheater cheater'
>>>> Cc: 'synth-diy'
>>>> Subject: Re: [sdiy] Polivoks filter
>>>> 
>>>> I will.  Right now I think I will finish building it in a module before using or recording.  It will be a lot easier and I don't risk shorts with wires and cables flying everywhere.  A couple of times now the weight of all the cables has pulled my PCB to the floor ...
>>>> 
>>>> Dave
>>>> 
>>>> -----Original Message-----
>>>> From: cheater cheater [mailto:cheater00 at gmail.com]
>>>> Sent: Saturday, February 16, 2013 11:00 AM
>>>> To: Dave Brown
>>>> Cc: synth-diy
>>>> Subject: Re: [sdiy] Polivoks filter
>>>> 
>>>> Hi Dave,
>>>> that looks great! Would you like to record some demos and post wav's somewhere?
>>>> 
>>>> Thanks,
>>>> D.
>>>> 
>>>> On Sat, Feb 16, 2013 at 7:11 PM, Dave Brown <davebr at earthlink.net> wrote:
>>>>> I just finished hand building a Polivoks filter using the Marc
>>>>> Bareille schematics with the reversing attenuator.  I ended up
>>>>> making a couple of minor changes to reduce the filter to near unity
>>>>> gain and softening the self-resonance output level so it would not clip rail to rail.
>>>>> 
>>>>> I built this filter using NTE888M programmable op-amps since they
>>>>> are readily available.  I did some frequency plots with my scope
>>>>> using XY mode and was disappointed with the DC level shift I saw on the low pass output.
>>>>> I got some of the ??1208 Russian parts to compare and discovered the
>>>>> same DC shift.  I had repaired a Polivoks filter previously and the
>>>>> frequency plots did not show any DC level shift.  I eventually
>>>>> figured out the DC shift was occurring on the VCO output as I swept
>>>>> to high frequencies.  I tried different VCOs and repeated the
>>>>> measurements with a VCO that had a constant amplitude output with no
>>>>> DC shift over the frequency range of the measurement.  Both the
>>>>> NTE888M and ??1208 Russian parts looked good with no DC shift in the low pass output.
>>>>> 
>>>>> I saw only two minor differences between the NTE888M parts and the
>>>>> ??1208 Russian parts.  The NTE888M parts have slightly higher gain
>>>>> and the cutoff or center frequencies was slightly higher at the same
>>>>> control settings and input CV.  I assume the frequency shift is due
>>>>> to a slightly different Iset response curve or a different Vbe bias point for the Iset control pin.
>>>>> 
>>>>> I have some uA776 parts on order and will also do the same
>>>>> comparison (assuming the parts are genuine).
>>>>> 
>>>>> Now I need to finish this filter with a front panel and play with it some.
>>>>> Details, photos, and scope images are on my site on my Polivoks Filter page.
>>>>> http://modularsynthesis.com/kuzmin/polivoks/polivoks_vcf.htm
>>>>> 
>>>>> David J. Brown
>>>>> Email davebr at earthlink.net
>>>>> Website http://modularsynthesis.com
>> 
>> 
>> 
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy



From dave.leith at gmail.com  Thu Mar  7 02:32:15 2013
From: dave.leith at gmail.com (Dave Leith)
Date: Wed, 6 Mar 2013 17:32:15 -0800
Subject: [sdiy] Looking for Ca3280 buy / trade IG00151 VCA
Message-ID: <CANcF2i2_Y-=Uq=3q3a7oVLZwjFcUQ0hbCQwz4Odqxh9qT6BQhg@mail.gmail.com>

Hello SDIYers

I'm looking for a reliable source for about 10 CA3280. Sure, there's
lots on Ebay but suggestions of good sellers would be much
appreciated. I've just been burned for some counterfeit MN3205s and
Reticon SAD1024s so I'd like to avoid more of the same.

I'd appreciate any suggestions, or personal sales/trades of the IC's
if possible.
I have a bunch of NOS Yamaha IG00151 VCA if someone would like to do a trade.

Thanks
Dave Leith
Buchla clone - http://minimalist.davidleith.com/DeitzModSynth10.html


From cheater00 at gmail.com  Thu Mar  7 12:00:30 2013
From: cheater00 at gmail.com (cheater cheater)
Date: Thu, 7 Mar 2013 12:00:30 +0100
Subject: [sdiy] Polivoks filter
In-Reply-To: <7BF41A72-2F14-475F-945E-CB62E9B9336F@electricdruid.net>
References: <000001ce157f$bfbabfc0$3f303f40$@earthlink.net>
	<CA+9GZUiAux9_pPKFCqujkr1uxmjYMbkYZmYH=DZ-Ayu24TG7Xw@mail.gmail.com>
	<000001ce17a5$d547d3a0$7fd77ae0$@earthlink.net>
	<CA+9GZUhTn5VG4+QwOrTGFv_wGfrKv+yhdT4kzo=-twq9PF4csw@mail.gmail.com>
	<000f01ce19d3$2dc608b0$89521a10$@earthlink.net>
	<CA+9GZUgvdheRqnYANVHO_Dn6oVOo-gtH87+XJbPWwD=Zc=CpOA@mail.gmail.com>
	<7BF41A72-2F14-475F-945E-CB62E9B9336F@electricdruid.net>
Message-ID: <CA+9GZUhoaDjK64qVmt4-fHutTPuMbXq0boxjqodf6HP137Pwow@mail.gmail.com>

Got it! Thanks a lot for pointing out where to look, Tom

Cheers,
D.

On Wed, Mar 6, 2013 at 3:55 PM, Tom Wiltshire <tom at electricdruid.net> wrote:
> Read his interesting and informative page on the topic:
>
>         http://modularsynthesis.com/kuzmin/polivoks/polivoks_vcf.htm
>
> The bit about the resonance and adding resonance limiting is about 1/4 the way down.
>
> T.
>
> On 6 Mar 2013, at 13:20, cheater cheater <cheater00 at gmail.com> wrote:
>
>> interesting. why is there "hard" and "soft" resonance? are those two
>> available in parallel at the same time, or are they selectable via
>> switch?
>>
>> Cheers,
>> D.
>>
>> On Tue, Mar 5, 2013 at 7:56 PM, Dave Brown <davebr at earthlink.net> wrote:
>>> I'm only playing with the Polivoks filter controls.  You can hear the hard resonance break into clipping around 0:49 and again at 1:50.  I think that is what you are talking about.  The output goes from +/-10V to +/-29 volts.  I added comments about the hard resonance clipping to both the LP and BP demo.
>>>
>>> Dave
>>>
>>> -----Original Message-----
>>> From: cheater cheater [mailto:cheater00 at gmail.com]
>>> Sent: Tuesday, March 05, 2013 5:10 AM
>>> To: Dave Brown; synth-diy
>>> Subject: Re: [sdiy] Polivoks filter
>>>
>>> Hi Dave, thanks a lot! Those are really nicely made. At some points there's quite a bit of distortion especially in the LP demo. Is that when you turn up the resonance, or is there some sort of gain parameter you're playing with?
>>>
>>> Cheers,
>>> D.
>>>
>>> On Sun, Mar 3, 2013 at 1:27 AM, Dave Brown <davebr at earthlink.net> wrote:
>>>> I put a couple of wave files up on the site.  Nothing fancy but I go
>>>> through all the controls. - Dave
>>>>
>>>> -----Original Message-----
>>>> From: cheater cheater [mailto:cheater00 at gmail.com]
>>>> Sent: Thursday, February 28, 2013 5:01 AM
>>>> To: Dave Brown
>>>> Cc: synth-diy
>>>> Subject: Re: [sdiy] Polivoks filter
>>>>
>>>> Hi Dave,
>>>> thanks a lot for the demo, but it's a bit difficult to appreciate the
>>>> filter's sound in the setup you've used. If I could ask you, would you
>>>> redo the demo, recording to line-in, and try to tune the filter an
>>>> octave or two lower? Due to reverb and other acoustic properties the
>>>> filter kind of sounds like any old filter :) and then it's generally
>>>> difficult to figure out what a filter is doing if the input signal is
>>>> pitched way up. The typical test is a sawtooth wave in the lower
>>>> registers, going through a filter with a quick attack and long (0.5 -
>>>> 1 second) decay and a lot of positive env mod.
>>>>
>>>> Cheers,
>>>> D.
>>>>
>>>> On Thu, Feb 28, 2013 at 7:49 AM, Dave Brown <davebr at earthlink.net> wrote:
>>>>> I received my front panel and finished building my Polivoks filter.  I updated the schematics and highlighted my few mods.  I put a 4 minute video showing some of the various modes and sounds.  I do like this filter a lot and I like having both the soft and hard resonance.  I like the soft resonance for traditional filter sound and the hard resonance for adding somewhat uncontrolled edge to the waveforms.  The tracking is pretty good although I haven't done a complete test of the range of tracking.  The beginning of the video demonstrates a sequencer driving both a VCO and the filter in self-oscillation which tracks pretty well.  Since there is no fine control the tuning is a bit tricky.
>>>>>
>>>>> http://modularsynthesis.com/kuzmin/polivoks/polivoks_vcf.htm
>>>>>
>>>>> Dave
>>>>>
>>>>> -----Original Message-----
>>>>> From: synth-diy-bounces at dropmix.xs4all.nl
>>>>> [mailto:synth-diy-bounces at dropmix.xs4all.nl] On Behalf Of Dave Brown
>>>>> Sent: Saturday, February 16, 2013 12:25 PM
>>>>> To: 'cheater cheater'
>>>>> Cc: 'synth-diy'
>>>>> Subject: Re: [sdiy] Polivoks filter
>>>>>
>>>>> I will.  Right now I think I will finish building it in a module before using or recording.  It will be a lot easier and I don't risk shorts with wires and cables flying everywhere.  A couple of times now the weight of all the cables has pulled my PCB to the floor ...
>>>>>
>>>>> Dave
>>>>>
>>>>> -----Original Message-----
>>>>> From: cheater cheater [mailto:cheater00 at gmail.com]
>>>>> Sent: Saturday, February 16, 2013 11:00 AM
>>>>> To: Dave Brown
>>>>> Cc: synth-diy
>>>>> Subject: Re: [sdiy] Polivoks filter
>>>>>
>>>>> Hi Dave,
>>>>> that looks great! Would you like to record some demos and post wav's somewhere?
>>>>>
>>>>> Thanks,
>>>>> D.
>>>>>
>>>>> On Sat, Feb 16, 2013 at 7:11 PM, Dave Brown <davebr at earthlink.net> wrote:
>>>>>> I just finished hand building a Polivoks filter using the Marc
>>>>>> Bareille schematics with the reversing attenuator.  I ended up
>>>>>> making a couple of minor changes to reduce the filter to near unity
>>>>>> gain and softening the self-resonance output level so it would not clip rail to rail.
>>>>>>
>>>>>> I built this filter using NTE888M programmable op-amps since they
>>>>>> are readily available.  I did some frequency plots with my scope
>>>>>> using XY mode and was disappointed with the DC level shift I saw on the low pass output.
>>>>>> I got some of the ??1208 Russian parts to compare and discovered the
>>>>>> same DC shift.  I had repaired a Polivoks filter previously and the
>>>>>> frequency plots did not show any DC level shift.  I eventually
>>>>>> figured out the DC shift was occurring on the VCO output as I swept
>>>>>> to high frequencies.  I tried different VCOs and repeated the
>>>>>> measurements with a VCO that had a constant amplitude output with no
>>>>>> DC shift over the frequency range of the measurement.  Both the
>>>>>> NTE888M and ??1208 Russian parts looked good with no DC shift in the low pass output.
>>>>>>
>>>>>> I saw only two minor differences between the NTE888M parts and the
>>>>>> ??1208 Russian parts.  The NTE888M parts have slightly higher gain
>>>>>> and the cutoff or center frequencies was slightly higher at the same
>>>>>> control settings and input CV.  I assume the frequency shift is due
>>>>>> to a slightly different Iset response curve or a different Vbe bias point for the Iset control pin.
>>>>>>
>>>>>> I have some uA776 parts on order and will also do the same
>>>>>> comparison (assuming the parts are genuine).
>>>>>>
>>>>>> Now I need to finish this filter with a front panel and play with it some.
>>>>>> Details, photos, and scope images are on my site on my Polivoks Filter page.
>>>>>> http://modularsynthesis.com/kuzmin/polivoks/polivoks_vcf.htm
>>>>>>
>>>>>> David J. Brown
>>>>>> Email davebr at earthlink.net
>>>>>> Website http://modularsynthesis.com
>>>
>>>
>>>
>> _______________________________________________
>> Synth-diy mailing list
>> Synth-diy at dropmix.xs4all.nl
>> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>


From tom at electricdruid.net  Thu Mar  7 14:07:41 2013
From: tom at electricdruid.net (Tom Wiltshire)
Date: Thu, 7 Mar 2013 13:07:41 +0000
Subject: [sdiy] Best way to get a +3.3V rail from +15V?
Message-ID: <B34B10F6-7C02-4D43-947C-DE7BC42DDFEE@electricdruid.net>

Hi all,

I'm putting a little dsPIC prototype dev board together for mucking about on. I've done such things before, but I've always had a +5V supply available, so when I've needed a 3.3V rail for the dsPIC, I've stuck a regulator on the 5V supply. Which was fine, but that's not the situation this time.

If I've only got +/-15V available, what's the best way to derive a 3.3V rail?


I've used 3.3V regulators on the 15V rail before and found they got hot, which wasn't a surprise seeing how much voltage they've got to lose somewhere. But perhaps I was doing something wrong?

Thanks,
Tom




From jacobwatters at gmail.com  Thu Mar  7 14:21:26 2013
From: jacobwatters at gmail.com (Jacob Watters)
Date: Thu, 7 Mar 2013 08:21:26 -0500
Subject: [sdiy] Best way to get a +3.3V rail from +15V?
In-Reply-To: <B34B10F6-7C02-4D43-947C-DE7BC42DDFEE@electricdruid.net>
References: <B34B10F6-7C02-4D43-947C-DE7BC42DDFEE@electricdruid.net>
Message-ID: <CAKcdib3SEjbR79bUpHEMU4fgN8rEU8if-KED9c89H+py-bzfXg@mail.gmail.com>

Hello Tom.  I am currently running a dsPIC with a 3.3V regulator that
is powered from a +/-8V rail.  It might not be the ideal solution, but
maybe you can alleviate some of the heat problem by distributing it
across two regulators, stepping down first to 8V and then 3.3V.

-Jacob Watters

On Thu, Mar 7, 2013 at 8:07 AM, Tom Wiltshire <tom at electricdruid.net> wrote:
> Hi all,
>
> I'm putting a little dsPIC prototype dev board together for mucking about on. I've done such things before, but I've always had a +5V supply available, so when I've needed a 3.3V rail for the dsPIC, I've stuck a regulator on the 5V supply. Which was fine, but that's not the situation this time.
>
> If I've only got +/-15V available, what's the best way to derive a 3.3V rail?
>
>
> I've used 3.3V regulators on the 15V rail before and found they got hot, which wasn't a surprise seeing how much voltage they've got to lose somewhere. But perhaps I was doing something wrong?
>
> Thanks,
> Tom
>
>
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy


From jarno.verhoeven at ziggo.nl  Thu Mar  7 14:27:10 2013
From: jarno.verhoeven at ziggo.nl (J. Verhoeven)
Date: Thu, 07 Mar 2013 14:27:10 +0100
Subject: [sdiy] Best way to get a +3.3V rail from +15V?
In-Reply-To: <7720cd393fed5.5138958e@ziggo.nl>
References: <B34B10F6-7C02-4D43-947C-DE7BC42DDFEE@electricdruid.net>
	<7720cd393fed5.5138958e@ziggo.nl>
Message-ID: <76708c4d3a5af.5138a3be@ziggo.nl>

Hi Tom,

Something like this?
http://nl.farnell.com/xp-power/ip1203sa/dc-dc-converter-3w-4-1-i-p3-3v/dp/1859151

I just picked one, they have several types available. I recently bought some 2x15v DC/DC converters for little money on eBay, still need to try them out, those are by Traco Power and SMT.

Jarno.


On 03/07/13, Tom Wiltshire  <tom at electricdruid.net> wrote:
> Hi all,
> 
> I'm putting a little dsPIC prototype dev board together for mucking about on. I've done such things before, but I've always had a +5V supply available, so when I've needed a 3.3V rail for the dsPIC, I've stuck a regulator on the 5V supply. Which was fine, but that's not the situation this time.
> 
> If I've only got +/-15V available, what's the best way to derive a 3.3V rail?
> 
> 
> I've used 3.3V regulators on the 15V rail before and found they got hot, which wasn't a surprise seeing how much voltage they've got to lose somewhere. But perhaps I was doing something wrong?
> 
> Thanks,
> Tom
> 
> 
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
> 
> 


From osalas at electronic-sea.net  Thu Mar  7 14:36:29 2013
From: osalas at electronic-sea.net (Oscar Salas)
Date: Thu, 7 Mar 2013 15:36:29 +0200
Subject: [sdiy] Best way to get a +3.3V rail from +15V?
In-Reply-To: <B34B10F6-7C02-4D43-947C-DE7BC42DDFEE@electricdruid.net>
References: <B34B10F6-7C02-4D43-947C-DE7BC42DDFEE@electricdruid.net>
Message-ID: <d7ca3eaf169017e3d2e4329d7ec774b4.squirrel@webmail.electronic-sea.net>

How much current do you need?

> Hi all,
>
> I'm putting a little dsPIC prototype dev board together for mucking about
> on. I've done such things before, but I've always had a +5V supply
> available, so when I've needed a 3.3V rail for the dsPIC, I've stuck a
> regulator on the 5V supply. Which was fine, but that's not the situation
> this time.
>
> If I've only got +/-15V available, what's the best way to derive a 3.3V
> rail?
>
>
> I've used 3.3V regulators on the 15V rail before and found they got hot,
> which wasn't a surprise seeing how much voltage they've got to lose
> somewhere. But perhaps I was doing something wrong?
>
> Thanks,
> Tom
>
>
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>
>


electronic-sea.net



From blacet at blacet.com  Thu Mar  7 15:26:24 2013
From: blacet at blacet.com (blacet at blacet.com)
Date: Thu, 7 Mar 2013 06:26:24 -0800
Subject: [sdiy] Best way to get a +3.3V rail from +15V?
In-Reply-To: <B34B10F6-7C02-4D43-947C-DE7BC42DDFEE@electricdruid.net>
References: <B34B10F6-7C02-4D43-947C-DE7BC42DDFEE@electricdruid.net>
Message-ID: <33782418419770c68fcc40197a9140e1.squirrel@webmail.blacet.com>

You can always use a series low value resistor to drop the 15V a bit and
take some of the heat before the regulator. It depends on how much current
you need and this will effect resistor value/wattage and regulator type/
heatsink requirements.


> Hi all,
>
> I'm putting a little dsPIC prototype dev board together for mucking about
> on. I've done such things before, but I've always had a +5V supply
> available, so when I've needed a 3.3V rail for the dsPIC, I've stuck a
> regulator on the 5V supply. Which was fine, but that's not the situation
> this time.
>
> If I've only got +/-15V available, what's the best way to derive a 3.3V
> rail?
>
>
> I've used 3.3V regulators on the 15V rail before and found they got hot,
> which wasn't a surprise seeing how much voltage they've got to lose
> somewhere. But perhaps I was doing something wrong?
>
> Thanks,
> Tom
>
>
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>




From music.maker at gte.net  Thu Mar  7 16:40:37 2013
From: music.maker at gte.net (Scott Gravenhorst)
Date: Thu, 07 Mar 2013 08:40:37 -0700
Subject: [sdiy] Best way to get a +3.3V rail from +15V?
Message-ID: <201303071640.r27GebQd020199@linux7.lan>


I'm wondering about using another regulator between +15 and +3.3.  Maybe a +8v.  I would think
that the heat dissapation should be less for the +3.3 regulator.  Some heat might also be expected
on the +8v reg.  It's either that or a power resistor as previously mentioned.

-- ScottG
________________________________________________________________________
-- Scott Gravenhorst
-- FPGA MIDI Synth Info: jovianpyx.dyndns.org:8080/public/FPGA_synth/
-- FatMan Mods Etc.: jovianpyx.dyndns.org:8080/public/fatman/
-- Some Random Electronics Bits: jovianpyx.dyndns.org:8080/public/electronics/
-- When the going gets tough, the tough use the command line.



From rburnett at richieburnett.co.uk  Thu Mar  7 16:54:45 2013
From: rburnett at richieburnett.co.uk (Richie Burnett)
Date: Thu, 07 Mar 2013 15:54:45 +0000
Subject: [sdiy] Best way to get a +3.3V rail from +15V?
Message-ID: <lqgnaw1op4tm7jj6ynthyayf.1362671685159@email.android.com>

If you want the most compact & efficient solution, then use a switching regulator. (It's technically a buck converter.) There's probably something suitable in National Semiconductor's Simple Switcher range. All you need is the switching IC, a Schottky diode and a couple of resistors and Low-ESR electrolytics.

If it's just for a quick lash-up you could always just go with a linear regulator though, bolt on an appropriately sized heatsink and benefit from some winter heating! ;-)

-Richie,


Sent from my Sony Ericsson Xperia ray

Tom Wiltshire <tom at electricdruid.net> wrote:

>Hi all,
>
>I'm putting a little dsPIC prototype dev board together for mucking about on. I've done such things before, but I've always had a +5V supply available, so when I've needed a 3.3V rail for the dsPIC, I've stuck a regulator on the 5V supply. Which was fine, but that's not the situation this time.
>
>If I've only got +/-15V available, what's the best way to derive a 3.3V rail?
>
>
>I've used 3.3V regulators on the 15V rail before and found they got hot, which wasn't a surprise seeing how much voltage they've got to lose somewhere. But perhaps I was doing something wrong?
>
>Thanks,
>Tom
>
>
>_______________________________________________
>Synth-diy mailing list
>Synth-diy at dropmix.xs4all.nl
>http://dropmix.xs4all.nl/mailman/listinfo/synth-diy

From tom at electricdruid.net  Thu Mar  7 17:02:15 2013
From: tom at electricdruid.net (Tom Wiltshire)
Date: Thu, 7 Mar 2013 16:02:15 +0000
Subject: [sdiy] Best way to get a +3.3V rail from +15V?
In-Reply-To: <33782418419770c68fcc40197a9140e1.squirrel@webmail.blacet.com>
References: <B34B10F6-7C02-4D43-947C-DE7BC42DDFEE@electricdruid.net>
	<33782418419770c68fcc40197a9140e1.squirrel@webmail.blacet.com>
Message-ID: <DBD89771-9656-476B-AC72-83E27C650E94@electricdruid.net>

A similar board I did on stripboard (the new one is getting a PCB) draws 101mA on the 3.3V rail. This board will have a 74HC594 driving some LEDs too, so I need perhaps 200mA. Does that change the picture any?

Thanks,
Tom


On 7 Mar 2013, at 14:26, blacet at blacet.com wrote:

> You can always use a series low value resistor to drop the 15V a bit and
> take some of the heat before the regulator. It depends on how much current
> you need and this will effect resistor value/wattage and regulator type/
> heatsink requirements.
> 
> 
>> Hi all,
>> 
>> I'm putting a little dsPIC prototype dev board together for mucking about
>> on. I've done such things before, but I've always had a +5V supply
>> available, so when I've needed a 3.3V rail for the dsPIC, I've stuck a
>> regulator on the 5V supply. Which was fine, but that's not the situation
>> this time.
>> 
>> If I've only got +/-15V available, what's the best way to derive a 3.3V
>> rail?
>> 
>> 
>> I've used 3.3V regulators on the 15V rail before and found they got hot,
>> which wasn't a surprise seeing how much voltage they've got to lose
>> somewhere. But perhaps I was doing something wrong?
>> 
>> Thanks,
>> Tom
>> 
>> 
>> _______________________________________________
>> Synth-diy mailing list
>> Synth-diy at dropmix.xs4all.nl
>> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>> 
> 
> 
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy



From neil.johnson97 at ntlworld.com  Thu Mar  7 17:21:06 2013
From: neil.johnson97 at ntlworld.com (Neil Johnson)
Date: Thu, 7 Mar 2013 16:21:06 +0000
Subject: [sdiy] Best way to get a +3.3V rail from +15V?
In-Reply-To: <DBD89771-9656-476B-AC72-83E27C650E94@electricdruid.net>
References: <B34B10F6-7C02-4D43-947C-DE7BC42DDFEE@electricdruid.net>
	<33782418419770c68fcc40197a9140e1.squirrel@webmail.blacet.com>
	<DBD89771-9656-476B-AC72-83E27C650E94@electricdruid.net>
Message-ID: <CAEaByMRQ20pCPsAF-_rO3KXF8KL0mT1Ffxuq8_QndeOAhtePTw@mail.gmail.com>

Tom,

> A similar board I did on stripboard (the new one is getting a PCB) draws 101mA on the 3.3V rail. This board will have a 74HC594 driving some LEDs too, so I need perhaps 200mA. Does that change the picture any?

A linear regulator would dissipate about 2.4W providing that amount of
current, from +15V down to +3V3.  That's quite a lot, and unless you
use a large heatsink then you've got yourself a nice little finger
burner.

Look in Farnell or Rapid.  For example:

http://www.rapidonline.com/Electrical-Power/R-783-3-1-0-Sgl-Out-Converter-Sil-84-2572

Neil
-- 
Modules and more: http://www.cesyg.com
Homepage: http://www.njohnson.co.uk


From neil.johnson97 at ntlworld.com  Thu Mar  7 17:49:37 2013
From: neil.johnson97 at ntlworld.com (Neil Johnson)
Date: Thu, 7 Mar 2013 16:49:37 +0000
Subject: [sdiy] Best way to get a +3.3V rail from +15V?
In-Reply-To: <201303071640.r27GebQd020199@linux7.lan>
References: <201303071640.r27GebQd020199@linux7.lan>
Message-ID: <CAEaByMTBbdbjaCt3UNbLHghNPUgM92Q68NfLnP79FjtC5rdaCw@mail.gmail.com>

Scott Gravenhorst wrote:
>
> I'm wondering about using another regulator between +15 and +3.3.  Maybe a +8v.  I would think
> that the heat dissapation should be less for the +3.3 regulator.  Some heat might also be expected
> on the +8v reg.  It's either that or a power resistor as previously mentioned.

How would that help?

Single regulator:
  Vdrop x Iload = (15 - 3.3) * 0.2 = 11.7 * 0.2 = 2.34W

Dual regulator:
  Vdrop1 x Iload = (15 - 8) * 0.2 = 7 * 0.2 = 1.4W
  Vdrop2 x Iload = (8 - 3.3) * 0.2 = 4.7 * 0.2 = 0.94W

  Ptot = 1.4 + 0.94 = 2.34W

The same power is dissipated, this time shared between two heatsinks.
You'd get the same effect using a single regulator and a larger
heatsink.

Neil
-- 
Modules and more: http://www.cesyg.com
Homepage: http://www.njohnson.co.uk


From blacet at blacet.com  Thu Mar  7 18:00:20 2013
From: blacet at blacet.com (blacet at blacet.com)
Date: Thu, 7 Mar 2013 09:00:20 -0800
Subject: [sdiy] Best way to get a +3.3V rail from +15V?
In-Reply-To: <CAEaByMTBbdbjaCt3UNbLHghNPUgM92Q68NfLnP79FjtC5rdaCw@mail.gmail.com>
References: <201303071640.r27GebQd020199@linux7.lan>
	<CAEaByMTBbdbjaCt3UNbLHghNPUgM92Q68NfLnP79FjtC5rdaCw@mail.gmail.com>
Message-ID: <acf0790a78f33e041b9b43bf2e741aab.squirrel@webmail.blacet.com>

Just as an example, I've used a 33 ohm 1W resistor in line with a LM317T
(with very small heatsink) on a board needing 5V @ 150 mA from 15V. Both
the resistor and regulator get quite warm; too hot to touch for very long
but well under the max operating temp.

The switching regulators are attractive except for they all seem to be
very small surface mount.



> Scott Gravenhorst wrote:
>>
>> I'm wondering about using another regulator between +15 and +3.3.  Maybe
>> a +8v.  I would think
>> that the heat dissapation should be less for the +3.3 regulator.  Some
>> heat might also be expected
>> on the +8v reg.  It's either that or a power resistor as previously
>> mentioned.
>
> How would that help?
>
> Single regulator:
>   Vdrop x Iload = (15 - 3.3) * 0.2 = 11.7 * 0.2 = 2.34W
>
> Dual regulator:
>   Vdrop1 x Iload = (15 - 8) * 0.2 = 7 * 0.2 = 1.4W
>   Vdrop2 x Iload = (8 - 3.3) * 0.2 = 4.7 * 0.2 = 0.94W
>
>   Ptot = 1.4 + 0.94 = 2.34W
>
> The same power is dissipated, this time shared between two heatsinks.
> You'd get the same effect using a single regulator and a larger
> heatsink.
>
> Neil
> --
> Modules and more: http://www.cesyg.com
> Homepage: http://www.njohnson.co.uk
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>




From osalas at electronic-sea.net  Thu Mar  7 18:05:24 2013
From: osalas at electronic-sea.net (Oscar Salas)
Date: Thu, 7 Mar 2013 19:05:24 +0200
Subject: [sdiy] Best way to get a +3.3V rail from +15V?
In-Reply-To: <DBD89771-9656-476B-AC72-83E27C650E94@electricdruid.net>
References: <B34B10F6-7C02-4D43-947C-DE7BC42DDFEE@electricdruid.net>
	<33782418419770c68fcc40197a9140e1.squirrel@webmail.blacet.com>
	<DBD89771-9656-476B-AC72-83E27C650E94@electricdruid.net>
Message-ID: <27839577624ecdeab9dd351198e09d10.squirrel@webmail.electronic-sea.net>

This is possible to be done with a linear standard regulator like the
UA78M33 without heat sink. It has a Junction to Ambient Thermal Resistance
of 29 C/W and a maximum absolute junction temperature of 150C. Recommended
125C.

For an ambient temperature of 40C and a maximum junction temperature of
125C (125-40)/29=2.9W

The power to dissipate: (15-3.3)*0.2=~2.3W




> A similar board I did on stripboard (the new one is getting a PCB) draws
> 101mA on the 3.3V rail. This board will have a 74HC594 driving some LEDs
> too, so I need perhaps 200mA. Does that change the picture any?
>
> Thanks,
> Tom
>
>
> On 7 Mar 2013, at 14:26, blacet at blacet.com wrote:
>
>> You can always use a series low value resistor to drop the 15V a bit and
>> take some of the heat before the regulator. It depends on how much
>> current
>> you need and this will effect resistor value/wattage and regulator type/
>> heatsink requirements.
>>
>>
>>> Hi all,
>>>
>>> I'm putting a little dsPIC prototype dev board together for mucking
>>> about
>>> on. I've done such things before, but I've always had a +5V supply
>>> available, so when I've needed a 3.3V rail for the dsPIC, I've stuck a
>>> regulator on the 5V supply. Which was fine, but that's not the
>>> situation
>>> this time.
>>>
>>> If I've only got +/-15V available, what's the best way to derive a 3.3V
>>> rail?
>>>
>>>
>>> I've used 3.3V regulators on the 15V rail before and found they got
>>> hot,
>>> which wasn't a surprise seeing how much voltage they've got to lose
>>> somewhere. But perhaps I was doing something wrong?
>>>
>>> Thanks,
>>> Tom
>>>
>>>
>>> _______________________________________________
>>> Synth-diy mailing list
>>> Synth-diy at dropmix.xs4all.nl
>>> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>>>
>>
>>
>> _______________________________________________
>> Synth-diy mailing list
>> Synth-diy at dropmix.xs4all.nl
>> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>
>


electronic-sea.net



From jlewis184 at gmail.com  Thu Mar  7 18:10:12 2013
From: jlewis184 at gmail.com (Josh)
Date: Thu, 7 Mar 2013 12:10:12 -0500
Subject: [sdiy] Best way to get a +3.3V rail from +15V?
In-Reply-To: <CAEaByMTBbdbjaCt3UNbLHghNPUgM92Q68NfLnP79FjtC5rdaCw@mail.gmail.com>
References: <201303071640.r27GebQd020199@linux7.lan>
	<CAEaByMTBbdbjaCt3UNbLHghNPUgM92Q68NfLnP79FjtC5rdaCw@mail.gmail.com>
Message-ID: <CAPCse3x23Wwq08rK3jzVn8p-v4yTXkOKFvtEnidVdHuuqx+eaA@mail.gmail.com>

The same total power is dissipated when using two devices, but they
are sharing the load.  Like two skinny dudes picking up a big box
instead of one bodybuilder.  In some cases it's easier to find the
skinny dudes.

Inserting a zener diode in series before the 3V3 regulator and
dropping the voltage there is also an option.  I have a dsPIC design
similar to yours where the 3V3 rail is stepped down from 15V.  I drop
8.7V across the zener and the rest across a linear regulator.  The
1N5345BRLG zener diode can dissipate a couple watts no problem.  I'm
also only using the PCB for heatsinking the 3V3 regulator.

And this is much simpler than a switching solution.

-Josh

On Thu, Mar 7, 2013 at 11:49 AM, Neil Johnson
<neil.johnson97 at ntlworld.com> wrote:
> Scott Gravenhorst wrote:
>>
>> I'm wondering about using another regulator between +15 and +3.3.  Maybe a +8v.  I would think
>> that the heat dissapation should be less for the +3.3 regulator.  Some heat might also be expected
>> on the +8v reg.  It's either that or a power resistor as previously mentioned.
>
> How would that help?
>
> Single regulator:
>   Vdrop x Iload = (15 - 3.3) * 0.2 = 11.7 * 0.2 = 2.34W
>
> Dual regulator:
>   Vdrop1 x Iload = (15 - 8) * 0.2 = 7 * 0.2 = 1.4W
>   Vdrop2 x Iload = (8 - 3.3) * 0.2 = 4.7 * 0.2 = 0.94W
>
>   Ptot = 1.4 + 0.94 = 2.34W
>
> The same power is dissipated, this time shared between two heatsinks.
> You'd get the same effect using a single regulator and a larger
> heatsink.
>
> Neil
> --
> Modules and more: http://www.cesyg.com
> Homepage: http://www.njohnson.co.uk
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy


From neil.johnson97 at ntlworld.com  Thu Mar  7 18:11:46 2013
From: neil.johnson97 at ntlworld.com (Neil Johnson)
Date: Thu, 7 Mar 2013 17:11:46 +0000
Subject: [sdiy] Best way to get a +3.3V rail from +15V?
In-Reply-To: <acf0790a78f33e041b9b43bf2e741aab.squirrel@webmail.blacet.com>
References: <201303071640.r27GebQd020199@linux7.lan>
	<CAEaByMTBbdbjaCt3UNbLHghNPUgM92Q68NfLnP79FjtC5rdaCw@mail.gmail.com>
	<acf0790a78f33e041b9b43bf2e741aab.squirrel@webmail.blacet.com>
Message-ID: <CAEaByMQsH_KuYE5VOs13zym-Gn-ggZ4n=CfYN0+AMM0eoYY-VA@mail.gmail.com>

blacet wrote:
> Just as an example, I've used a 33 ohm 1W resistor in line with a LM317T
> (with very small heatsink) on a board needing 5V @ 150 mA from 15V. Both
> the resistor and regulator get quite warm; too hot to touch for very long
> but well under the max operating temp.
>
> The switching regulators are attractive except for they all seem to be
> very small surface mount.

Except this one:

http://www.rapidonline.com/Electrical-Power/R-783-3-1-0-Sgl-Out-Converter-Sil-84-2572

And I've use NatSemi "SimpleSwitcher" devices before.  Very easy to
use and available in through-hole parts.  The LM2575 or TL2575 is a
good one.

Neil
-- 
Modules and more: http://www.cesyg.com
Homepage: http://www.njohnson.co.uk


From osalas at electronic-sea.net  Thu Mar  7 18:21:35 2013
From: osalas at electronic-sea.net (Oscar Salas)
Date: Thu, 7 Mar 2013 19:21:35 +0200
Subject: [sdiy] Best way to get a +3.3V rail from +15V?
In-Reply-To: <27839577624ecdeab9dd351198e09d10.squirrel@webmail.electronic-sea.net>
References: <B34B10F6-7C02-4D43-947C-DE7BC42DDFEE@electricdruid.net>
	<33782418419770c68fcc40197a9140e1.squirrel@webmail.blacet.com>
	<DBD89771-9656-476B-AC72-83E27C650E94@electricdruid.net>
	<27839577624ecdeab9dd351198e09d10.squirrel@webmail.electronic-sea.net>
Message-ID: <d13088cf97f5ba8f69501a1694712e26.squirrel@webmail.electronic-sea.net>

Sorry, my previous mail was not finished.

This is possible to be done with a linear standard regulator like the
UA78M33 without heat sink. It has a Junction to Ambient Thermal Resistance
of 29 C/W and a maximum absolute junction temperature of 150C. Recommended
125C.

For an ambient temperature of 40C and a maximum junction temperature of
125C the maximum power output is 2.9W (125-40)/29=2.9W
The power to dissipate in your application: (15-3.3)*0.2=~2.3W

Also, you could attach the regulator to the PCB to improve slightly the
power dissipation.
To output more current or to operate it at lower temperature, you could
add a UA78M09 before the UA78M33.

Regards,
Oscar.




>> A similar board I did on stripboard (the new one is getting a PCB) draws
>> 101mA on the 3.3V rail. This board will have a 74HC594 driving some LEDs
>> too, so I need perhaps 200mA. Does that change the picture any?
>>
>> Thanks,
>> Tom
>>
>>
>> On 7 Mar 2013, at 14:26, blacet at blacet.com wrote:
>>
>>> You can always use a series low value resistor to drop the 15V a bit
>>> and
>>> take some of the heat before the regulator. It depends on how much
>>> current
>>> you need and this will effect resistor value/wattage and regulator
>>> type/
>>> heatsink requirements.
>>>
>>>
>>>> Hi all,
>>>>
>>>> I'm putting a little dsPIC prototype dev board together for mucking
>>>> about
>>>> on. I've done such things before, but I've always had a +5V supply
>>>> available, so when I've needed a 3.3V rail for the dsPIC, I've stuck a
>>>> regulator on the 5V supply. Which was fine, but that's not the
>>>> situation
>>>> this time.
>>>>
>>>> If I've only got +/-15V available, what's the best way to derive a
>>>> 3.3V
>>>> rail?
>>>>
>>>>
>>>> I've used 3.3V regulators on the 15V rail before and found they got
>>>> hot,
>>>> which wasn't a surprise seeing how much voltage they've got to lose
>>>> somewhere. But perhaps I was doing something wrong?
>>>>
>>>> Thanks,
>>>> Tom
>>>>
>>>>
>>>> _______________________________________________
>>>> Synth-diy mailing list
>>>> Synth-diy at dropmix.xs4all.nl
>>>> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>>>>
>>>
>>>
>>> _______________________________________________
>>> Synth-diy mailing list
>>> Synth-diy at dropmix.xs4all.nl
>>> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>>
>> _______________________________________________
>> Synth-diy mailing list
>> Synth-diy at dropmix.xs4all.nl
>> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>>
>>
>
>
> electronic-sea.net
>
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>
>


electronic-sea.net



From osalas at electronic-sea.net  Thu Mar  7 18:38:12 2013
From: osalas at electronic-sea.net (Oscar Salas)
Date: Thu, 7 Mar 2013 19:38:12 +0200
Subject: [sdiy] Best way to get a +3.3V rail from +15V?
In-Reply-To: <d13088cf97f5ba8f69501a1694712e26.squirrel@webmail.electronic-sea.net>
References: <B34B10F6-7C02-4D43-947C-DE7BC42DDFEE@electricdruid.net>
	<33782418419770c68fcc40197a9140e1.squirrel@webmail.blacet.com>
	<DBD89771-9656-476B-AC72-83E27C650E94@electricdruid.net>
	<27839577624ecdeab9dd351198e09d10.squirrel@webmail.electronic-sea.net>
	<d13088cf97f5ba8f69501a1694712e26.squirrel@webmail.electronic-sea.net>
Message-ID: <3d4e31fa21e613fdaa120801eba2aaca.squirrel@webmail.electronic-sea.net>

Looking now the LM317 datasheets from National and from Texas instruments,
National Specifies the j-a thermal resistance 50C/W for the TO-220 package
and Texas Instruments specifies 29C/W. The same than specifies for the
UA78M33.
So I would test my suggestion or use a UA78M09 before the UA78M33.

Oscar


> Sorry, my previous mail was not finished.
>
> This is possible to be done with a linear standard regulator like the
> UA78M33 without heat sink. It has a Junction to Ambient Thermal Resistance
> of 29 C/W and a maximum absolute junction temperature of 150C. Recommended
> 125C.
>
> For an ambient temperature of 40C and a maximum junction temperature of
> 125C the maximum power output is 2.9W (125-40)/29=2.9W
> The power to dissipate in your application: (15-3.3)*0.2=~2.3W
>
> Also, you could attach the regulator to the PCB to improve slightly the
> power dissipation.
> To output more current or to operate it at lower temperature, you could
> add a UA78M09 before the UA78M33.
>
> Regards,
> Oscar.
>
>
>
>
>>> A similar board I did on stripboard (the new one is getting a PCB)
>>> draws
>>> 101mA on the 3.3V rail. This board will have a 74HC594 driving some
>>> LEDs
>>> too, so I need perhaps 200mA. Does that change the picture any?
>>>
>>> Thanks,
>>> Tom
>>>
>>>
>>> On 7 Mar 2013, at 14:26, blacet at blacet.com wrote:
>>>
>>>> You can always use a series low value resistor to drop the 15V a bit
>>>> and
>>>> take some of the heat before the regulator. It depends on how much
>>>> current
>>>> you need and this will effect resistor value/wattage and regulator
>>>> type/
>>>> heatsink requirements.
>>>>
>>>>
>>>>> Hi all,
>>>>>
>>>>> I'm putting a little dsPIC prototype dev board together for mucking
>>>>> about
>>>>> on. I've done such things before, but I've always had a +5V supply
>>>>> available, so when I've needed a 3.3V rail for the dsPIC, I've stuck
>>>>> a
>>>>> regulator on the 5V supply. Which was fine, but that's not the
>>>>> situation
>>>>> this time.
>>>>>
>>>>> If I've only got +/-15V available, what's the best way to derive a
>>>>> 3.3V
>>>>> rail?
>>>>>
>>>>>
>>>>> I've used 3.3V regulators on the 15V rail before and found they got
>>>>> hot,
>>>>> which wasn't a surprise seeing how much voltage they've got to lose
>>>>> somewhere. But perhaps I was doing something wrong?
>>>>>
>>>>> Thanks,
>>>>> Tom
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Synth-diy mailing list
>>>>> Synth-diy at dropmix.xs4all.nl
>>>>> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Synth-diy mailing list
>>>> Synth-diy at dropmix.xs4all.nl
>>>> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>>>
>>> _______________________________________________
>>> Synth-diy mailing list
>>> Synth-diy at dropmix.xs4all.nl
>>> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>>>
>>>
>>
>>
>> electronic-sea.net
>>
>> _______________________________________________
>> Synth-diy mailing list
>> Synth-diy at dropmix.xs4all.nl
>> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>>
>>
>
>
> electronic-sea.net
>
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>
>


electronic-sea.net



From ol.gillet at gmail.com  Thu Mar  7 18:59:57 2013
From: ol.gillet at gmail.com (Olivier Gillet)
Date: Thu, 7 Mar 2013 18:59:57 +0100
Subject: [sdiy] Best way to get a +3.3V rail from +15V?
In-Reply-To: <3d4e31fa21e613fdaa120801eba2aaca.squirrel@webmail.electronic-sea.net>
References: <B34B10F6-7C02-4D43-947C-DE7BC42DDFEE@electricdruid.net>
	<33782418419770c68fcc40197a9140e1.squirrel@webmail.blacet.com>
	<DBD89771-9656-476B-AC72-83E27C650E94@electricdruid.net>
	<27839577624ecdeab9dd351198e09d10.squirrel@webmail.electronic-sea.net>
	<d13088cf97f5ba8f69501a1694712e26.squirrel@webmail.electronic-sea.net>
	<3d4e31fa21e613fdaa120801eba2aaca.squirrel@webmail.electronic-sea.net>
Message-ID: <CAJoLJ5U1bEmaGEx9pxOzd1oSAHvdbAu=sqxRM9m0irgzgBRkmA@mail.gmail.com>

Another ready to use DC-DC conversion module I stumbled upon:
http://fr.mouser.com/ProductDetail/Murata-Power-Solutions/OKI-78SR-33-15-W36H-C/?qs=sGAEpiMZZMtwaiKVUtQsNa9RSQZ1iZ%2fUnO1fxRs35eY%3d

Roughly the same outline as a TO-220 Vreg. I've seen this kind of
Murata 78xx-compatible DC-DC modules on the Makenoise +5V adapter, and
people have been using them as a direct replacement of a 7805 for my
DIY synth kit Ambika.

Olivier


From Stromeko at nexgo.de  Thu Mar  7 19:15:08 2013
From: Stromeko at nexgo.de (ASSI)
Date: Thu, 07 Mar 2013 19:15:08 +0100
Subject: [sdiy] Best way to get a +3.3V rail from +15V?
In-Reply-To: <3d4e31fa21e613fdaa120801eba2aaca.squirrel@webmail.electronic-sea.net>
References: <B34B10F6-7C02-4D43-947C-DE7BC42DDFEE@electricdruid.net>
	<d13088cf97f5ba8f69501a1694712e26.squirrel@webmail.electronic-sea.net>
	<3d4e31fa21e613fdaa120801eba2aaca.squirrel@webmail.electronic-sea.net>
Message-ID: <1560428.ztlexi40Qr@rainer>

On Thursday 07 March 2013, 19:38:12, Oscar Salas wrote:
> Looking now the LM317 datasheets from National and from Texas instruments,
> National Specifies the j-a thermal resistance 50C/W for the TO-220
> package and Texas Instruments specifies 29C/W. The same than specifies
> for the UA78M33.

Keep in mind these figures are for free convection and constant ambient 
temperature (i.e. infinity capacity for taking up heat).  In a (semi-)closed 
box the figures will be different, especially if the air inside does get 
warmer than 40?C (it will).  My personal rule of thumb: if something on the 
bench gets hotter than you can touch for a few seconds (about 60?C) then it 
needs cooling when put inside any enclosure.


Regards,
Achim.
-- 
+<[Q+ Matrix-12 WAVE#46+305 Neuron microQkb Andromeda XTk Blofeld]>+

SD adaptation for Waldorf rackAttack V1.04R1:
http://Synth.Stromeko.net/Downloads.html#WaldorfSDada



From blacet at blacet.com  Thu Mar  7 19:25:17 2013
From: blacet at blacet.com (blacet at blacet.com)
Date: Thu, 7 Mar 2013 10:25:17 -0800
Subject: [sdiy] Best way to get a +3.3V rail from +15V?
In-Reply-To: <CAJoLJ5U1bEmaGEx9pxOzd1oSAHvdbAu=sqxRM9m0irgzgBRkmA@mail.gmail.com>
References: <B34B10F6-7C02-4D43-947C-DE7BC42DDFEE@electricdruid.net>
	<33782418419770c68fcc40197a9140e1.squirrel@webmail.blacet.com>
	<DBD89771-9656-476B-AC72-83E27C650E94@electricdruid.net>
	<27839577624ecdeab9dd351198e09d10.squirrel@webmail.electronic-sea.net>
	<d13088cf97f5ba8f69501a1694712e26.squirrel@webmail.electronic-sea.net>
	<3d4e31fa21e613fdaa120801eba2aaca.squirrel@webmail.electronic-sea.net>
	<CAJoLJ5U1bEmaGEx9pxOzd1oSAHvdbAu=sqxRM9m0irgzgBRkmA@mail.gmail.com>
Message-ID: <59a5d8e23bd2d0ceabb96477b989f9d7.squirrel@webmail.blacet.com>

This looks good. Also in vertical mount:
http://www.mouser.com/ProductDetail/Murata-Power-Solutions/OKI-78SR-33-15-W36-C/?qs=%2fha2pyFadug1ZrXHkIky7LtoJCPWCweGXaSSyT98oJvt1QArCaucX9tJYw9tEMiA


> Another ready to use DC-DC conversion module I stumbled upon:
> http://fr.mouser.com/ProductDetail/Murata-Power-Solutions/OKI-78SR-33-15-W36H-C/?qs=sGAEpiMZZMtwaiKVUtQsNa9RSQZ1iZ%2fUnO1fxRs35eY%3d
>
> Roughly the same outline as a TO-220 Vreg. I've seen this kind of
> Murata 78xx-compatible DC-DC modules on the Makenoise +5V adapter, and
> people have been using them as a direct replacement of a 7805 for my
> DIY synth kit Ambika.
>
> Olivier
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>




From jim at cernproductions.com  Thu Mar  7 21:13:14 2013
From: jim at cernproductions.com (Jim Credland)
Date: Thu, 07 Mar 2013 20:13:14 +0000
Subject: [sdiy] Best way to get a +3.3V rail from +15V?
In-Reply-To: <201303071640.r27GebQd020199@linux7.lan>
References: <201303071640.r27GebQd020199@linux7.lan>
Message-ID: <1362687194.30508.140661201394549.735061E5@webmail.messagingengine.com>

I've been using a MAX5033 switching regulator on a tiny PCB to get a 5V
rail and then an LDO part on the module itself to get the 3.3V out.  The
MAX thing was a piece of cake and seems remarkably low-noise.  

What I'd really like to do is convert the -12V rail which is under used
in the synth though to the +5V rail.  I'm sure it's possible but even
thinking about the problem turns my head upside down.  Do I use two
parallel step up converters? 

On Thu, Mar 7, 2013, at 03:40 PM, Scott Gravenhorst wrote:
> 
> I'm wondering about using another regulator between +15 and +3.3.  Maybe
> a +8v.  I would think
> that the heat dissapation should be less for the +3.3 regulator.  Some
> heat might also be expected
> on the +8v reg.  It's either that or a power resistor as previously
> mentioned.
> 
> -- ScottG
> ________________________________________________________________________
> -- Scott Gravenhorst
> -- FPGA MIDI Synth Info: jovianpyx.dyndns.org:8080/public/FPGA_synth/
> -- FatMan Mods Etc.: jovianpyx.dyndns.org:8080/public/fatman/
> -- Some Random Electronics Bits:
> jovianpyx.dyndns.org:8080/public/electronics/
> -- When the going gets tough, the tough use the command line.
> 
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy


-- 
  Jim Credland
  jim at cernproductions.com


From timr at circuitabbey.com  Thu Mar  7 22:24:12 2013
From: timr at circuitabbey.com (Tim Ressel)
Date: Thu, 7 Mar 2013 13:24:12 -0800 (PST)
Subject: [sdiy] Best way to get a +3.3V rail from +15V?
In-Reply-To: <1362687194.30508.140661201394549.735061E5@webmail.messagingengine.com>
References: <201303071640.r27GebQd020199@linux7.lan>
	<1362687194.30508.140661201394549.735061E5@webmail.messagingengine.com>
Message-ID: <1362691452.28803.YahooMailNeo@web5816.biz.mail.ne1.yahoo.com>

That is the solution I would have gone with. the 125KHz switching speed keeps the inductor low and the noise above the audio range. The only caveat is layout. If your not careful with the ground loops you can make lots of radio noise or even make the beast misbehave. Since we are not concerned with getting a CE mark, it'll probably be fine.


Tim Ressel
Circuit Abbey
503-750-9331
timr at circuitabbey.com




>________________________________
> From: Jim Credland <jim at cernproductions.com>
>To: synth-diy at dropmix.xs4all.nl 
>Sent: Thursday, March 7, 2013 12:13 PM
>Subject: Re: [sdiy] Best way to get a +3.3V rail from +15V?
> 
>I've been using a MAX5033 switching regulator on a tiny PCB to get a 5V
>rail and then an LDO part on the module itself to get the 3.3V out.? The
>MAX thing was a piece of cake and seems remarkably low-noise.? 
>
>What I'd really like to do is convert the -12V rail which is under used
>in the synth though to the +5V rail.? I'm sure it's possible but even
>thinking about the problem turns my head upside down.? Do I use two
>parallel step up converters? 
>
>On Thu, Mar 7, 2013, at 03:40 PM, Scott Gravenhorst wrote:
>> 
>> I'm wondering about using another regulator between +15 and +3.3.? Maybe
>> a +8v.? I would think
>> that the heat dissapation should be less for the +3.3 regulator.? Some
>> heat might also be expected
>> on the +8v reg.? It's either that or a power resistor as previously
>> mentioned.
>> 
>> -- ScottG
>> ________________________________________________________________________
>> -- Scott Gravenhorst
>> -- FPGA MIDI Synth Info: jovianpyx.dyndns.org:8080/public/FPGA_synth/
>> -- FatMan Mods Etc.: jovianpyx.dyndns.org:8080/public/fatman/
>> -- Some Random Electronics Bits:
>> jovianpyx.dyndns.org:8080/public/electronics/
>> -- When the going gets tough, the tough use the command line.
>> 
>> _______________________________________________
>> Synth-diy mailing list
>> Synth-diy at dropmix.xs4all.nl
>> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>
>
>-- 
>? Jim Credland
>? jim at cernproductions.com
>_______________________________________________
>Synth-diy mailing list
>Synth-diy at dropmix.xs4all.nl
>http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>
>
>?


From spr at spridley.freeserve.co.uk  Thu Mar  7 22:51:53 2013
From: spr at spridley.freeserve.co.uk (Steve Ridley)
Date: Thu, 7 Mar 2013 22:51:53 +0100 (CET)
Subject: [sdiy] Best way to get a +3.3V rail from +15V?
In-Reply-To: <CAPCse3x23Wwq08rK3jzVn8p-v4yTXkOKFvtEnidVdHuuqx+eaA@mail.gmail.com>
Message-ID: <1008182.132971362693113426.JavaMail.www@wwinf3711>


> And this is much simpler than a switching solution.

Simpler? Try Traco TSR series or XP Power SR range devices. 

- Similar footprint to a TO220 (not smd - old folks can see them)
- Small part count - add two caps and a small inductor
- >90% efficient so barely warm - no more "ouch! (insert expletive here) moments when you touch it.
- no clumsy heatsink cluttering up your development board
- At 200mA out it'll draw less than 50mA from the 15v.


Complicated? Linear:

- Juggling heat dissipation across two devices in non-free space
- Mounting heatsinks so they don't get in the way, burn you or take up acres of space
- Deal with uneven load your +/-15v supply
etc.

Worth a try?


Steve



From loscha at gmail.com  Fri Mar  8 06:48:59 2013
From: loscha at gmail.com (Loscha)
Date: Fri, 8 Mar 2013 16:48:59 +1100
Subject: [sdiy] [ot] Ensoniq Mirage
Message-ID: <CAA25cvdKxE3C13_-U5mR1rj0Do7Fz8EZQfhkKPb+2Ppp4zUmCw@mail.gmail.com>

Hi All,

 I'm helping someone sell an Ensoniq Mirage. It looks in great
condition, but we don't have any System or Formatting disks.

Is there anyone in Melbourne who could help us out, please? The Mirage
is located in Northcote, Melbourne.

Thanks in advance!!

/Edward


From stewpye at optusnet.com.au  Fri Mar  8 07:59:49 2013
From: stewpye at optusnet.com.au (Stewart Pye)
Date: Fri, 08 Mar 2013 16:59:49 +1000
Subject: [sdiy] Best way to get a +3.3V rail from +15V?
In-Reply-To: <B34B10F6-7C02-4D43-947C-DE7BC42DDFEE@electricdruid.net>
References: <B34B10F6-7C02-4D43-947C-DE7BC42DDFEE@electricdruid.net>
Message-ID: <51398C65.3050507@optusnet.com.au>

Use something like this:
http://www.murata-ps.com/data/power/oki-78sr.pdf

We use the 5V version in several products where I work. I'm on holidays 
and can't remember where we purchase them, but they cost around AUD $2. 
We have dropped them in to PCB's designed for 7805 regulators and they 
work fine...

3.3V, 1.5 amps at 85% efficiency, no heatsink. Woot woot!

Regards,
Stewart.



On 3/7/2013 11:07 PM, Tom Wiltshire wrote:
> Hi all,
>
> I'm putting a little dsPIC prototype dev board together for mucking about on. I've done such things before, but I've always had a +5V supply available, so when I've needed a 3.3V rail for the dsPIC, I've stuck a regulator on the 5V supply. Which was fine, but that's not the situation this time.
>
> If I've only got +/-15V available, what's the best way to derive a 3.3V rail?
>
>
> I've used 3.3V regulators on the 15V rail before and found they got hot, which wasn't a surprise seeing how much voltage they've got to lose somewhere. But perhaps I was doing something wrong?
>
> Thanks,
> Tom
>
>
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>



From recompas at gmail.com  Fri Mar  8 16:32:18 2013
From: recompas at gmail.com (Travis Thatcher)
Date: Fri, 8 Mar 2013 09:32:18 -0600
Subject: [sdiy] MXR Pitch Transposer Schem/Circuit
Message-ID: <CAG0qc=3CXvBcjexWXPFqSTOoPq5WF6DAUP-WkM6UFsb9A+cpmQ@mail.gmail.com>

While not technically SDIY (I apologize) I just got a black face MXR
Pitch transposer yesterday. Its a pretty amazing sounding piece of
gear with very little info on the web about it.  I did some quick
experimenting with the cv input and a sequencer and was very pleased
with the results but I was curious if anyone else had any experience
with these, particularly any mods or schematics.

I'm assuming its an older bbd based design, though the chip which I'm
guess is the bbd has had the labeling info removed. It does great
pitch shifting and with the regen cranked it goes from weird flanger
to space sounds pretty quickly. It sounds very similar to a friend's
mxr flanger that I fixed so I'm assuming the circuit is roughly the
same. Kind of interested how it works!



-- 
http://www.travisthatcher.com
http://www.soundcloud.com/travisthatcher
http://voiceofsaturn.blogspot.com


From tom at electricdruid.net  Fri Mar  8 17:13:38 2013
From: tom at electricdruid.net (Tom Wiltshire)
Date: Fri, 8 Mar 2013 16:13:38 +0000
Subject: [sdiy] MXR Pitch Transposer Schem/Circuit
In-Reply-To: <CAG0qc=3CXvBcjexWXPFqSTOoPq5WF6DAUP-WkM6UFsb9A+cpmQ@mail.gmail.com>
References: <CAG0qc=3CXvBcjexWXPFqSTOoPq5WF6DAUP-WkM6UFsb9A+cpmQ@mail.gmail.com>
Message-ID: <EDED40BF-DD73-4FBF-AF96-C6538FEC5690@electricdruid.net>

That's a damn interesting gizmo. Does anyone have any idea how it works?

On 8 Mar 2013, at 15:32, Travis Thatcher <recompas at gmail.com> wrote:

> While not technically SDIY (I apologize) I just got a black face MXR
> Pitch transposer yesterday. Its a pretty amazing sounding piece of
> gear with very little info on the web about it.  I did some quick
> experimenting with the cv input and a sequencer and was very pleased
> with the results but I was curious if anyone else had any experience
> with these, particularly any mods or schematics.
> 
> I'm assuming its an older bbd based design, though the chip which I'm
> guess is the bbd has had the labeling info removed. It does great
> pitch shifting and with the regen cranked it goes from weird flanger
> to space sounds pretty quickly. It sounds very similar to a friend's
> mxr flanger that I fixed so I'm assuming the circuit is roughly the
> same. Kind of interested how it works!
> 
> 
> 
> -- 
> http://www.travisthatcher.com
> http://www.soundcloud.com/travisthatcher
> http://voiceofsaturn.blogspot.com
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy



From steve at bendentech.co.uk  Fri Mar  8 17:44:34 2013
From: steve at bendentech.co.uk (Steve Lenham)
Date: Fri, 08 Mar 2013 16:44:34 +0000
Subject: [sdiy] MXR Pitch Transposer Schem/Circuit
In-Reply-To: <EDED40BF-DD73-4FBF-AF96-C6538FEC5690@electricdruid.net>
References: <CAG0qc=3CXvBcjexWXPFqSTOoPq5WF6DAUP-WkM6UFsb9A+cpmQ@mail.gmail.com>
	<EDED40BF-DD73-4FBF-AF96-C6538FEC5690@electricdruid.net>
Message-ID: <513A1572.4050307@bendentech.co.uk>

Unless the blackface version is very different to the blue one I worked 
on, the MXR pitch transposer isn't a BBD design. It is a digital one, 
with an 8-bit logarithmic ADC/DAC and a small block of RAM.

Basically a circular buffer is set up in the RAM, with a read and a 
write pointer which chase each other round at different rates. Some 
fancy crossfading is done to get rid of the discontinuities when the 
pointers cross over.

For a better insight, go to www.spinsemi.com and search around for 
information on the pitch shifter algorithms for their FV-1 effects DSP. 
Spin Semi was set up by the late Keith Barr, who IIRC founded both MXR 
and Alesis, and the software algorithm seems to follow the same 
principle as the MXR hardware.

Almost as cool as the effect itself are the touch-sensitive knobs - when 
you touch any of the four preset shift-amount knobs, it automatically 
selects that preset, so you can hop instantly between different effects.

HTH,

Steve L.


On 08/03/2013 16:13, Tom Wiltshire wrote:
> That's a damn interesting gizmo. Does anyone have any idea how it works?
>
> On 8 Mar 2013, at 15:32, Travis Thatcher <recompas at gmail.com> wrote:
>
>> While not technically SDIY (I apologize) I just got a black face MXR
>> Pitch transposer yesterday. Its a pretty amazing sounding piece of
>> gear with very little info on the web about it.  I did some quick
>> experimenting with the cv input and a sequencer and was very pleased
>> with the results but I was curious if anyone else had any experience
>> with these, particularly any mods or schematics.
>>
>> I'm assuming its an older bbd based design, though the chip which I'm
>> guess is the bbd has had the labeling info removed. It does great
>> pitch shifting and with the regen cranked it goes from weird flanger
>> to space sounds pretty quickly. It sounds very similar to a friend's
>> mxr flanger that I fixed so I'm assuming the circuit is roughly the
>> same. Kind of interested how it works!
>>
>>
>>
>> --
>> http://www.travisthatcher.com
>> http://www.soundcloud.com/travisthatcher
>> http://voiceofsaturn.blogspot.com
>> _______________________________________________
>> Synth-diy mailing list
>> Synth-diy at dropmix.xs4all.nl
>> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>



From gordonjcp at gjcp.net  Fri Mar  8 21:15:44 2013
From: gordonjcp at gjcp.net (Gordon JC Pearce)
Date: Fri, 08 Mar 2013 20:15:44 +0000
Subject: [sdiy] [ot] Ensoniq Mirage
In-Reply-To: <CAA25cvdKxE3C13_-U5mR1rj0Do7Fz8EZQfhkKPb+2Ppp4zUmCw@mail.gmail.com>
References: <CAA25cvdKxE3C13_-U5mR1rj0Do7Fz8EZQfhkKPb+2Ppp4zUmCw@mail.gmail.com>
Message-ID: <513A46F0.8000303@gjcp.net>

On Fri, Mar 08, 2013 at 04:48:59PM +1100, Loscha wrote:
> Hi All,
>
>  I'm helping someone sell an Ensoniq Mirage. It looks in great
> condition, but we don't have any System or Formatting disks.
>
> Is there anyone in Melbourne who could help us out, please? The Mirage
> is located in Northcote, Melbourne.
>
> Thanks in advance!!

If you have some DD floppies (HD won't work well at all) and a PC 
running Linux with an onboard floppy controller - ideally, with an Intel 
chipset - then it's pish easy to create blank floppies for testing.

Format a blank disk like this:
$ superformat /dev/fd0 tracksize=11b mss ssize=1024 dd --zero-based

This will lay down the weirdass 5632-byte tracks (five sectors of 1024 
bytes, one of 512 bytes, 80 tracks, single-sided).  Now you need to get 
an image onto the disk.

A company called Syntaur claims to have been granted the exclusive right 
to distribute Mirage disks including the OS disks that were provided 
with the machine.  For a long time, the general consensus was that 
copying the OS 3.2 disk was okay, but not FMT-2 or MASOS because they 
weren't part of the original kit and Syntaur sell them.  So, I can't 
tell you on a public mailing list where to find images.

Theoretically you could set up the disk parameters and just use dd to 
write the image to the disk.  In practice, it's easier to use this handy 
wee utility:

http://lovesthepython.org/~gordonjcp/writedisk.c

Grab it, read the instructions at the top of the file to compile it, and 
write a disk.

A healthy Mirage will give ten or eleven clunks from the drive as it 
loads the OS, pause for a couple of seconds as it tunes the filter (the 
output is muted but with everything full up you may faintly hear a 
"pewpewpewpew" noise), and then load in the first lower and upper half 
off the disk (twelve clunks - pause - twelve clunks).

Hope this helps

-- 
Gordonjcp MM0YEQ






From mars at pingdynasty.com  Fri Mar  8 21:32:59 2013
From: mars at pingdynasty.com (Martin Klang)
Date: Fri, 8 Mar 2013 20:32:59 +0000
Subject: [sdiy] [ot] Ensoniq Mirage
In-Reply-To: <513A46F0.8000303@gjcp.net>
References: <CAA25cvdKxE3C13_-U5mR1rj0Do7Fz8EZQfhkKPb+2Ppp4zUmCw@mail.gmail.com>
	<513A46F0.8000303@gjcp.net>
Message-ID: <D6D2F726-246A-4D86-B901-BA0E51A1E903@pingdynasty.com>


I've not tried the linux/superformat approach, seems much easier than what I went through.
I made disks for my SD-1, don't know how different that is from the mirage but imagine the raw disk format is the same.

My notes-to-self are here:
http://hq.pingdynasty.com/blog/2011/04/04/ensoniq-sd-1-sequencer-disks/

oh and I doubt you need DD floppies, just tape over the notch in an HD (much easier to get hold of!) and you should be good to go.

hth,

/m

On 8 Mar 2013, at 20:15, Gordon JC Pearce wrote:

> On Fri, Mar 08, 2013 at 04:48:59PM +1100, Loscha wrote:
>> Hi All,
>> 
>> I'm helping someone sell an Ensoniq Mirage. It looks in great
>> condition, but we don't have any System or Formatting disks.
>> 
>> Is there anyone in Melbourne who could help us out, please? The Mirage
>> is located in Northcote, Melbourne.
>> 
>> Thanks in advance!!
> 
> If you have some DD floppies (HD won't work well at all) and a PC running Linux with an onboard floppy controller - ideally, with an Intel chipset - then it's pish easy to create blank floppies for testing.
> 
> Format a blank disk like this:
> $ superformat /dev/fd0 tracksize=11b mss ssize=1024 dd --zero-based
> 
> This will lay down the weirdass 5632-byte tracks (five sectors of 1024 bytes, one of 512 bytes, 80 tracks, single-sided).  Now you need to get an image onto the disk.
> 
> A company called Syntaur claims to have been granted the exclusive right to distribute Mirage disks including the OS disks that were provided with the machine.  For a long time, the general consensus was that copying the OS 3.2 disk was okay, but not FMT-2 or MASOS because they weren't part of the original kit and Syntaur sell them.  So, I can't tell you on a public mailing list where to find images.
> 
> Theoretically you could set up the disk parameters and just use dd to write the image to the disk.  In practice, it's easier to use this handy wee utility:
> 
> http://lovesthepython.org/~gordonjcp/writedisk.c
> 
> Grab it, read the instructions at the top of the file to compile it, and write a disk.
> 
> A healthy Mirage will give ten or eleven clunks from the drive as it loads the OS, pause for a couple of seconds as it tunes the filter (the output is muted but with everything full up you may faintly hear a "pewpewpewpew" noise), and then load in the first lower and upper half off the disk (twelve clunks - pause - twelve clunks).
> 
> Hope this helps
> 
> -- 
> Gordonjcp MM0YEQ
> 
> 
> 
> 
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy



From gordonjcp at gjcp.net  Fri Mar  8 22:07:32 2013
From: gordonjcp at gjcp.net (Gordon JC Pearce)
Date: Fri, 08 Mar 2013 21:07:32 +0000
Subject: [sdiy] [ot] Ensoniq Mirage
In-Reply-To: <D6D2F726-246A-4D86-B901-BA0E51A1E903@pingdynasty.com>
References: <CAA25cvdKxE3C13_-U5mR1rj0Do7Fz8EZQfhkKPb+2Ppp4zUmCw@mail.gmail.com>
	<513A46F0.8000303@gjcp.net>
	<D6D2F726-246A-4D86-B901-BA0E51A1E903@pingdynasty.com>
Message-ID: <513A5314.9050709@gjcp.net>

On 08/03/13 20:32, Martin Klang wrote:

(Yes Martin, I didn't mean Reply All but Reply to List which appears to 
have sent to you and not the list.  Reply-To munging considered batshit 
crazy...)
>
> I've not tried the linux/superformat approach, seems much easier than
> what I went through. I made disks for my SD-1, don't know how
> different that is from the mirage but imagine the raw disk format is
> the same.

I believe the SD1 uses a more conventional DSDD 10-sector 80-track 
format, like the EPS.

> My notes-to-self are here:
> http://hq.pingdynasty.com/blog/2011/04/04/ensoniq-sd-1-sequencer-disks/

Looks intertesting.  I wonder how hard it would be to extract the files 
off the SD-1 disks?

>  oh and I doubt you need DD floppies, just tape over the notch in an
> HD (much easier to get hold of!) and you should be good to go.

Nope.  HD disks will not work at all, for this.  Not even a little bit.

If you're very lucky, it'll read once or twice on the Mirage, and then 
you get to reformat the disk.

Get DD disks, and save yourself a lot of hassle and error.

-- 
Gordonjcp MM0YEQ



From modcan at sympatico.ca  Fri Mar  8 23:53:16 2013
From: modcan at sympatico.ca (BRUCE DUNCAN)
Date: Fri, 8 Mar 2013 17:53:16 -0500
Subject: [sdiy] Best way to get a +3.3V rail from +15V?
In-Reply-To: <1008182.132971362693113426.JavaMail.www@wwinf3711>
References: <CAPCse3x23Wwq08rK3jzVn8p-v4yTXkOKFvtEnidVdHuuqx+eaA@mail.gmail.com>,
	<1008182.132971362693113426.JavaMail.www@wwinf3711>
Message-ID: <BLU170-W1260AEBB88B529737F587A3BBE60@phx.gbl>



One thing I wonder about is using say 10+ small SMPS sub regulators in a modular environment is what 
the cumulative effect on main transformer based power supply on startup.
I read somewhere that SMPS look like a dead short on startup. Could this cause
over current tripping on say Power One supplies with current limiting?
Having a lot of modules in a system resulting in many input caps in parallel has often been a problem
with power supply startup and just wondering if could cause similar issues.

Bruce



----------------------------------------
> Date: Thu, 7 Mar 2013 22:51:53 +0100
> From: spr at spridley.freeserve.co.uk
> To: synth-diy at dropmix.xs4all.nl
> CC: synth-diy at dropmix.xs4all.nl
> Subject: Re: [sdiy] Best way to get a +3.3V rail from +15V?
>
>
> > And this is much simpler than a switching solution.
>
> Simpler? Try Traco TSR series or XP Power SR range devices.
>
> - Similar footprint to a TO220 (not smd - old folks can see them)
> - Small part count - add two caps and a small inductor
> - >90% efficient so barely warm - no more "ouch! (insert expletive here) moments when you touch it.
> - no clumsy heatsink cluttering up your development board
> - At 200mA out it'll draw less than 50mA from the 15v.
>
>
> Complicated? Linear:
>
> - Juggling heat dissipation across two devices in non-free space
> - Mounting heatsinks so they don't get in the way, burn you or take up acres of space
> - Deal with uneven load your +/-15v supply
> etc.
>
> Worth a try?
>
>
> Steve
>
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
 		 	   		  

From jim at cernproductions.com  Sat Mar  9 10:48:08 2013
From: jim at cernproductions.com (Jim Credland)
Date: Sat, 09 Mar 2013 09:48:08 +0000
Subject: [sdiy] Best way to get a +3.3V rail from +15V?
In-Reply-To: <1362691452.28803.YahooMailNeo@web5816.biz.mail.ne1.yahoo.com>
References: <201303071640.r27GebQd020199@linux7.lan>
	<1362687194.30508.140661201394549.735061E5@webmail.messagingengine.com>
	<1362691452.28803.YahooMailNeo@web5816.biz.mail.ne1.yahoo.com>
Message-ID: <1362822488.1430.140661202037533.0DEE314F@webmail.messagingengine.com>

I diligently followed the rules in the datasheet for keeping it to a
minimum.  Nice star grounding (looks pretty on the PCB), everything very
close together.  I'm curious to know whether I've nailed it, but no
decent idea how to measure for radio noise.  Stupidly I met an EMC
specialist from UKAS at work last week (in connection with something
entirely different) and didn't ask him! Doh!

On Thu, Mar 7, 2013, at 09:24 PM, Tim Ressel wrote:
> That is the solution I would have gone with. the 125KHz switching speed
> keeps the inductor low and the noise above the audio range. The only
> caveat is layout. If your not careful with the ground loops you can make
> lots of radio noise or even make the beast misbehave. Since we are not
> concerned with getting a CE mark, it'll probably be fine.
> 
> 
> Tim Ressel
> Circuit Abbey


From magnus at rubidium.dyndns.org  Sat Mar  9 11:13:22 2013
From: magnus at rubidium.dyndns.org (Magnus Danielson)
Date: Sat, 09 Mar 2013 11:13:22 +0100
Subject: [sdiy] Best way to get a +3.3V rail from +15V?
In-Reply-To: <B34B10F6-7C02-4D43-947C-DE7BC42DDFEE@electricdruid.net>
References: <B34B10F6-7C02-4D43-947C-DE7BC42DDFEE@electricdruid.net>
Message-ID: <513B0B42.4090400@rubidium.dyndns.org>

On 03/07/2013 02:07 PM, Tom Wiltshire wrote:
> Hi all,
>
> I'm putting a little dsPIC prototype dev board together for mucking about on. I've done such things before, but I've always had a +5V supply available, so when I've needed a 3.3V rail for the dsPIC, I've stuck a regulator on the 5V supply. Which was fine, but that's not the situation this time.
>
> If I've only got +/-15V available, what's the best way to derive a 3.3V rail?
>
>
> I've used 3.3V regulators on the 15V rail before and found they got hot, which wasn't a surprise seeing how much voltage they've got to lose somewhere. But perhaps I was doing something wrong?

You should get a small switcher for a drop-switching. That way you avoid 
having to burn (15-3.3) times the current you use. For the current that 
I assume you will consume, it will be a small switcher and a handful of 
components.

Maxim and TI springs to mind, among others.

Be sure to layout it well and keep good filtering for isolation.

Cheers,
Magnus


From tom at electricdruid.net  Sat Mar  9 16:34:50 2013
From: tom at electricdruid.net (Tom Wiltshire)
Date: Sat, 9 Mar 2013 15:34:50 +0000
Subject: [sdiy] MXR Pitch Transposer Schem/Circuit
In-Reply-To: <513A1572.4050307@bendentech.co.uk>
References: <CAG0qc=3CXvBcjexWXPFqSTOoPq5WF6DAUP-WkM6UFsb9A+cpmQ@mail.gmail.com>
	<EDED40BF-DD73-4FBF-AF96-C6538FEC5690@electricdruid.net>
	<513A1572.4050307@bendentech.co.uk>
Message-ID: <F21FB5CD-C994-47C5-A3B7-AAF727D43409@electricdruid.net>

Thanks, Steve. The Spin Semi pages are a goldmine. For anyone who hasn't visited, I recommend:

http://www.spinsemi.com/knowledge_base/effects.html

T.

On 8 Mar 2013, at 16:44, Steve Lenham <steve at bendentech.co.uk> wrote:

> Unless the blackface version is very different to the blue one I worked on, the MXR pitch transposer isn't a BBD design. It is a digital one, with an 8-bit logarithmic ADC/DAC and a small block of RAM.
> 
> Basically a circular buffer is set up in the RAM, with a read and a write pointer which chase each other round at different rates. Some fancy crossfading is done to get rid of the discontinuities when the pointers cross over.
> 
> For a better insight, go to www.spinsemi.com and search around for information on the pitch shifter algorithms for their FV-1 effects DSP. Spin Semi was set up by the late Keith Barr, who IIRC founded both MXR and Alesis, and the software algorithm seems to follow the same principle as the MXR hardware.
> 
> Almost as cool as the effect itself are the touch-sensitive knobs - when you touch any of the four preset shift-amount knobs, it automatically selects that preset, so you can hop instantly between different effects.
> 
> HTH,
> 
> Steve L.


From tim.parkhurst at gmail.com  Sat Mar  9 19:10:05 2013
From: tim.parkhurst at gmail.com (Tim Parkhurst)
Date: Sat, 9 Mar 2013 10:10:05 -0800
Subject: [sdiy] Contest to win a Tek 465
Message-ID: <CACDGQT_WLByiT61aNG67YNj_3TRaxnMw-nV16X0KDKFTZvLm2A@mail.gmail.com>

Cool contest - Register at this site, send in a picture of your messy
/ interesting workbench, do a few other things, and you're entered in
a contest to win a classic Tek 465 scope. :)

http://www.scopejunction.com/author.asp?section_id=1835&doc_id=260090&&goback=.gde_2316414_member_221102380



Tim (I'm sure we have some VERY 'interesting' workbenches out there) Servo
---
"Sire, the church of God is an anvil that has worn out many hammers."
- H.L. Hastings


From blacet at blacet.com  Sat Mar  9 19:31:16 2013
From: blacet at blacet.com (blacet at blacet.com)
Date: Sat, 9 Mar 2013 10:31:16 -0800
Subject: [sdiy] Blacet Specials
In-Reply-To: <CACDGQT_WLByiT61aNG67YNj_3TRaxnMw-nV16X0KDKFTZvLm2A@mail.gmail.com>
References: <CACDGQT_WLByiT61aNG67YNj_3TRaxnMw-nV16X0KDKFTZvLm2A@mail.gmail.com>
Message-ID: <cdcb5ee11c60d43bde0bdbc4382083c6.squirrel@webmail.blacet.com>

A couple things of possible interest to DIY and AH:

Sale on 3.5mm Patch Cable Kits: was $13, now $10. The very best quality
metal plugs and color coding heatshrink.

PSCONN2 Power Distribution Board now in kit form. Save nearly 50% off
assembled price of $29; only $15 for the kit.

On the Store page:
http://www.blacet.com/store2.html





From houshu at muj.biglobe.ne.jp  Sun Mar 10 02:49:03 2013
From: houshu at muj.biglobe.ne.jp (houshu at muj.biglobe.ne.jp)
Date: Sun, 10 Mar 2013 10:49:03 +0900 (JST)
Subject: [sdiy] VCA response - continuously variable between linear
	and	exponential
In-Reply-To: <a11924c9-1f0e-4c64-94d4-ebc7d3391e3a@bvea33570>
Message-ID: <85281c64-7e87-409a-aae0-ea2c99fa0852@bvea33570>

Hello Neil and List,

I designed variable current sources for LM13700
with bipolar power supply.

http://houshu.at.webry.info/201303/article_1.html

As usual, I tested it only on LTspice.
Quite good exponential curves were obtained thanks to 
low impedance design.  If you don't need precise linear response,
An audio-curve 2k-ohm pot seems to be good for continuous control.  
If available, reverse-log curve pot with switch would be the best
to have precise linear response.

Temperature dependency was not so bad.  The response shrunk just 10%
in horizontal axis even when the temperature changed from 5 deg.
(in C) to 35 deg.

Sam HOSHUYAMA
Saitama, JAPAN

----- Original Message -----
From: houshu at muj.biglobe.ne.jp
To: "neil harper" <metadata at gmx.com>
Cc: "Synth-Diy" <synth-diy at dropmix.xs4all.nl>, houshu at muj.biglobe.ne.jp
Sent: Sunday, March 3, 2013 01:53:37
Subject: Re: [sdiy] VCA response - continuously variable between linear and	exponential

Hello Neil and List,

I've designed continuous curve VCA.
Precise exponential can not be obtained.
But, fractional power curves were available.

Final VCA design (Built for battery operation, and worked as in simulations.).
  http://houshu.at.webry.info/201112/article_2.html

Current source designs:
  http://houshu.at.webry.info/201111/article_1.html
  http://houshu.at.webry.info/201111/article_2.html
  http://houshu.at.webry.info/201201/article_1.html

This is basically a function generator combining log and exponential, 
which may be analyzed by theory of translinear circuit.

Continuous curve control is possible, but the required
resistor range is too large to obtain linear-like response.  
So I chose 3 curves.

Here is comparison of the 3 curves and linear and exponential: 
http://userdisk.webry.biglobe.ne.jp/000/024/65/N000/000/000/132039778125213127511_Lin-Exp1_dc.PNG

If you can wait for 1 month, I'll design a current source circuit 
for LM13700, though I myself would not test it on a breadboard.

Sam HOSHUYAMA
Saitama, JAPAN

----- "neil harper" <metadata at gmx.com>
hey all,
i'm looking for some advice/pointers on how to get a VCA response curve
to be continuously variable between linear and exponential. i can do
switchable between the two, but is there any trick to getting everything
inbetween?

if it helps, my VCA is lm13700 based but i don't mind starting with
something else.

-- 
//neil harper
every wave is new until it breaks

_______________________________________________
Synth-diy mailing list
Synth-diy at dropmix.xs4all.nl
http://dropmix.xs4all.nl/mailman/listinfo/synth-diy


From metadata at gmx.com  Sun Mar 10 03:03:41 2013
From: metadata at gmx.com (neil harper)
Date: Sat, 09 Mar 2013 21:03:41 -0500
Subject: [sdiy] VCA response - continuously variable between linear and
 exponential
In-Reply-To: <85281c64-7e87-409a-aae0-ea2c99fa0852@bvea33570>
References: <85281c64-7e87-409a-aae0-ea2c99fa0852@bvea33570>
Message-ID: <1362881021.1910.263.camel@nwwtlaptop>

> As usual, I tested it only on LTspice.
> Quite good exponential curves were obtained thanks to 
> low impedance design.  If you don't need precise linear response,
> An audio-curve 2k-ohm pot seems to be good for continuous control.  
> If available, reverse-log curve pot with switch would be the best
> to have precise linear response.
> 

hi Sam, this looks great!

the 2k pot is R16 in the diagram and simulation, correct?







From houshu at muj.biglobe.ne.jp  Sun Mar 10 03:07:34 2013
From: houshu at muj.biglobe.ne.jp (houshu at muj.biglobe.ne.jp)
Date: Sun, 10 Mar 2013 11:07:34 +0900 (JST)
Subject: [sdiy] VCA response - continuously variable between linear and
	exponential
In-Reply-To: <1362881021.1910.263.camel@nwwtlaptop>
Message-ID: <a6ea2c7a-efa7-4265-b965-04dc7c068994@bvea33570>

Hi Neil,

>the 2k pot is R16 in the diagram and simulation, correct?
Exactly.

----- Original Message -----
From: "neil harper" <metadata at gmx.com>
To: houshu at muj.biglobe.ne.jp
Cc: "Synth-Diy" <synth-diy at dropmix.xs4all.nl>
Sent: Sunday, March 10, 2013 11:03:41
Subject: Re: [sdiy] VCA response - continuously variable between linear and exponential

> As usual, I tested it only on LTspice.
> Quite good exponential curves were obtained thanks to 
> low impedance design.  If you don't need precise linear response,
> An audio-curve 2k-ohm pot seems to be good for continuous control.  
> If available, reverse-log curve pot with switch would be the best
> to have precise linear response.
> 

hi Sam, this looks great!

the 2k pot is R16 in the diagram and simulation, correct?







From gordonjcp at gjcp.net  Sun Mar 10 11:00:47 2013
From: gordonjcp at gjcp.net (Gordon JC Pearce)
Date: Sun, 10 Mar 2013 10:00:47 +0000
Subject: [sdiy] MXR Pitch Transposer Schem/Circuit
In-Reply-To: <EDED40BF-DD73-4FBF-AF96-C6538FEC5690@electricdruid.net>
References: <CAG0qc=3CXvBcjexWXPFqSTOoPq5WF6DAUP-WkM6UFsb9A+cpmQ@mail.gmail.com>
	<EDED40BF-DD73-4FBF-AF96-C6538FEC5690@electricdruid.net>
Message-ID: <513C59CF.5040909@gjcp.net>

On 08/03/13 16:13, Tom Wiltshire wrote:
> That's a damn interesting gizmo. Does anyone have any idea how it works?
>

Someone on the SQ80 mailing list - where we were discussing time-based 
effects the other day, as it happens - mentioned a pitch shifter that 
modulated the delay clock with a sawtooth.  I can see how that would 
give a steady-ish pitch shift but surely it would "hiccup" as the saw reset?

-- 
Gordonjcp MM0YEQ




From martin at martinfay.com  Sun Mar 10 11:45:23 2013
From: martin at martinfay.com (Martin Fay)
Date: Sun, 10 Mar 2013 10:45:23 +0000
Subject: [sdiy] MXR Pitch Transposer Schem/Circuit
In-Reply-To: <513C59CF.5040909@gjcp.net>
References: <CAG0qc=3CXvBcjexWXPFqSTOoPq5WF6DAUP-WkM6UFsb9A+cpmQ@mail.gmail.com>
	<EDED40BF-DD73-4FBF-AF96-C6538FEC5690@electricdruid.net>
	<513C59CF.5040909@gjcp.net>
Message-ID: <0B91C209-CE83-4DEE-A504-89CFEBDC3812@martinfay.com>

Hi Gordon,

I believe the trick is to have 2 read pointers into the delay buffer  
and crossfade between those outputs to avoid reading close to the saw  
reset.

Martin

Sent from my iPhone

On 10 Mar 2013, at 10:00, Gordon JC Pearce <gordonjcp at gjcp.net> wrote:

> On 08/03/13 16:13, Tom Wiltshire wrote:
>> That's a damn interesting gizmo. Does anyone have any idea how it  
>> works?
>>
>
> Someone on the SQ80 mailing list - where we were discussing time- 
> based effects the other day, as it happens - mentioned a pitch  
> shifter that modulated the delay clock with a sawtooth.  I can see  
> how that would give a steady-ish pitch shift but surely it would  
> "hiccup" as the saw reset?
>
> -- 
> Gordonjcp MM0YEQ
>
>
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy


From tim.parkhurst at gmail.com  Sun Mar 10 20:31:35 2013
From: tim.parkhurst at gmail.com (Tim Parkhurst)
Date: Sun, 10 Mar 2013 12:31:35 -0700
Subject: [sdiy] Nice education site
Message-ID: <CACDGQT8XZSdGfd4eQpoxAZMa=uU_Y+O2O_C=YmSq4yH_HqPn9w@mail.gmail.com>

Nice site with tons of analog and op-amp design hints and tips (this
is just the non-inv summing amp page, but there are others worth
checking out):

http://masteringelectronicsdesign.com/design-a-summing-amplifier-calculator/



Tim (old dog, learning a few new tricks) Servo
---
"Sire, the church of God is an anvil that has worn out many hammers."
- H.L. Hastings


From rburnett at richieburnett.co.uk  Sun Mar 10 21:21:07 2013
From: rburnett at richieburnett.co.uk (Richie Burnett)
Date: Sun, 10 Mar 2013 20:21:07 +0000
Subject: [sdiy] MXR Pitch Transposer Schem/Circuit
Message-ID: <hv8yx3f7j9lbt1tp9h53xy53.1362943642538@email.android.com>

The two read-pointers with cross-fading to minimise the glitches is really the bare minimum for pitch shifting. It still sounds quite rough though in practice. Even though you fade the contribution of each read pointer down to zero when it crosses the write pointer theres still a problem unless a whole number of audio cycles fit perfectly into the ring buffer's memory. Essentially the waveforms from the two read pointers are rarely coherent for arbitrary audio inputs. So when you cross-fade between them you get constructive/destructive interference that causes wobbles in the amplitude and phase of the pitch shifted audio output. This sounds rough.

Any decent pro-audio pitch shifter these days also performs an auto-correlation on the incoming audio to determine any periodicity. Then each of the two read pointers are moved by whole numbers of audio periods when they would cross the write pointer and otherwise cause a glitch. If you nudge the read pointers by whole numbers of audio cycles then there should be no glitches in theory. However cross-fading is still used to minimise any glitches caused by less than perfect period detection or poor auto-correlation when pitch-shifting complex polyphonic material. At least now the waveforms from the two playback pointers are as coherent as they can be. Almost identical and in-phase. So the cross-fading now causes minimal destructive interference and amplitude wobble.

That's how pitch shifting has been done since Eventide's H949 Harmonizer in 1977. Sean Costello provides a good explanation here:

http://valhalladsp.wordpress.com/2010/05/07/pitch-shifting-the-h949-and-de-glitching/ 

-Richie,


Sent from my Sony Ericsson Xperia ray

Martin Fay <martin at martinfay.com> wrote:

>Hi Gordon,
>
>I believe the trick is to have 2 read pointers into the delay buffer  
>and crossfade between those outputs to avoid reading close to the saw  
>reset.
>
>Martin
>
>Sent from my iPhone
>
>On 10 Mar 2013, at 10:00, Gordon JC Pearce <gordonjcp at gjcp.net> wrote:
>
>> On 08/03/13 16:13, Tom Wiltshire wrote:
>>> That's a damn interesting gizmo. Does anyone have any idea how it  
>>> works?
>>>
>>
>> Someone on the SQ80 mailing list - where we were discussing time- 
>> based effects the other day, as it happens - mentioned a pitch  
>> shifter that modulated the delay clock with a sawtooth.  I can see  
>> how that would give a steady-ish pitch shift but surely it would  
>> "hiccup" as the saw reset?
>>
>> -- 
>> Gordonjcp MM0YEQ
>>
>>
>> _______________________________________________
>> Synth-diy mailing list
>> Synth-diy at dropmix.xs4all.nl
>> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>_______________________________________________
>Synth-diy mailing list
>Synth-diy at dropmix.xs4all.nl
>http://dropmix.xs4all.nl/mailman/listinfo/synth-diy

From patchell at cox.net  Sun Mar 10 23:46:19 2013
From: patchell at cox.net (Jim Patchell)
Date: Sun, 10 Mar 2013 15:46:19 -0700
Subject: [sdiy] [OT]Rachel Flowers Wailing on Kieth Emmersons Moog
Message-ID: <513D0D3B.40507@cox.net>

http://www.youtube.com/watch?v=x7qujZpMHbA

Rachel Flowers is a fellow student at the Santa Barbara Braille 
Institute.  When this video was made, she was about 18.  Her playing 
really makes you sit up and listen.  Thought most of you might enjoy this.

What I find funny is that she got to try out the new mods on Kieths Moog 
before he got to.  On the video, she also plays the piano and a Hammond C-3.

-Jim



From rsdio at sounds.wa.com  Mon Mar 11 05:01:38 2013
From: rsdio at sounds.wa.com (rsdio at sounds.wa.com)
Date: Sun, 10 Mar 2013 21:01:38 -0700
Subject: [sdiy] [ot] Ensoniq Mirage
In-Reply-To: <513A5314.9050709@gjcp.net>
References: <CAA25cvdKxE3C13_-U5mR1rj0Do7Fz8EZQfhkKPb+2Ppp4zUmCw@mail.gmail.com>
	<513A46F0.8000303@gjcp.net>
	<D6D2F726-246A-4D86-B901-BA0E51A1E903@pingdynasty.com>
	<513A5314.9050709@gjcp.net>
Message-ID: <5F7A1AC1-DE22-4920-94C1-FFD3C1AE0DBD@sounds.wa.com>


On Mar 8, 2013, at 13:07, Gordon JC Pearce wrote:
> On 08/03/13 20:32, Martin Klang wrote:
>> I've not tried the linux/superformat approach, seems much easier than
>> what I went through. I made disks for my SD-1, don't know how
>> different that is from the mirage but imagine the raw disk format is
>> the same.
>
> I believe the SD1 uses a more conventional DSDD 10-sector 80-track  
> format, like the EPS.

I wouldn't call it conventional. Ensoniq used 10 sectors, while the  
ubiquitous IBM-PC DOS used only 9 sectors. Where DOS put 720 KB on a  
DD floppy, Ensoniq managed to fit 800 KB. Not much, but it is 11%  
more. There is custom software out there to read and write Ensoniq  
floppies.

Brian Willoughby
Sound Consulting



From martin at martinfay.com  Mon Mar 11 10:55:19 2013
From: martin at martinfay.com (Martin Fay)
Date: Mon, 11 Mar 2013 09:55:19 +0000
Subject: [sdiy] MXR Pitch Transposer Schem/Circuit
References: <5E88E816-B298-47BF-9653-41CBD8883870@martinfay.com>
Message-ID: <524076CE-F656-4601-A79D-B8BB9A16823F@martinfay.com>

Forgot to copy the list...
>

> Hi Richie,
>
> Sean's blog has a whole series of posts on pitchshifting covering  
> tape-based onward. My description more or less matches the H910  
> which used triangle waves for the crossfade.
>
> I think something like this could be done with BBDs but the  
> artifacts of the modulated clock would surely be "interesting"!
>
> Martin
>
> Sent from my iPhone
>
> On 10 Mar 2013, at 20:21, Richie Burnett  
> <rburnett at richieburnett.co.uk> wrote:
>
>> The two read-pointers with cross-fading to minimise the glitches is  
>> really the bare minimum for pitch shifting. It still sounds quite  
>> rough though in practice. Even though you fade the contribution of  
>> each read pointer down to zero when it crosses the write pointer  
>> theres still a problem unless a whole number of audio cycles fit  
>> perfectly into the ring buffer's memory. Essentially the waveforms  
>> from the two read pointers are rarely coherent for arbitrary audio  
>> inputs. So when you cross-fade between them you get constructive/ 
>> destructive interference that causes wobbles in the amplitude and  
>> phase of the pitch shifted audio output. This sounds rough.
>>
>> Any decent pro-audio pitch shifter these days also performs an auto- 
>> correlation on the incoming audio to determine any periodicity.  
>> Then each of the two read pointers are moved by whole numbers of  
>> audio periods when they would cross the write pointer and otherwise  
>> cause a glitch. If you nudge the read pointers by whole numbers of  
>> audio cycles then there should be no glitches in theory. However  
>> cross-fading is still used to minimise any glitches caused by less  
>> than perfect period detection or poor auto-correlation when pitch- 
>> shifting complex polyphonic material. At least now the waveforms  
>> from the two playback pointers are as coherent as they can be.  
>> Almost identical and in-phase. So the cross-fading now causes  
>> minimal destructive interference and amplitude wobble.
>>
>> That's how pitch shifting has been done since Eventide's H949  
>> Harmonizer in 1977. Sean Costello provides a good explanation here:
>>
>> http://valhalladsp.wordpress.com/2010/05/07/pitch-shifting-the-h949-and-de-glitching/
>>
>> -Richie,
>>
>>
>> Sent from my Sony Ericsson Xperia ray
>>
>> Martin Fay <martin at martinfay.com> wrote:
>>
>>> Hi Gordon,
>>>
>>> I believe the trick is to have 2 read pointers into the delay buffer
>>> and crossfade between those outputs to avoid reading close to the  
>>> saw
>>> reset.
>>>
>>> Martin
>>>
>>> Sent from my iPhone
>>>
>>> On 10 Mar 2013, at 10:00, Gordon JC Pearce <gordonjcp at gjcp.net>  
>>> wrote:
>>>
>>>> On 08/03/13 16:13, Tom Wiltshire wrote:
>>>>> That's a damn interesting gizmo. Does anyone have any idea how it
>>>>> works?
>>>>>
>>>>
>>>> Someone on the SQ80 mailing list - where we were discussing time-
>>>> based effects the other day, as it happens - mentioned a pitch
>>>> shifter that modulated the delay clock with a sawtooth.  I can see
>>>> how that would give a steady-ish pitch shift but surely it would
>>>> "hiccup" as the saw reset?
>>>>
>>>> -- 
>>>> Gordonjcp MM0YEQ
>>>>
>>>>
>>>> _______________________________________________
>>>> Synth-diy mailing list
>>>> Synth-diy at dropmix.xs4all.nl
>>>> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>>> _______________________________________________
>>> Synth-diy mailing list
>>> Synth-diy at dropmix.xs4all.nl
>>> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy


From spr at spridley.freeserve.co.uk  Mon Mar 11 11:42:53 2013
From: spr at spridley.freeserve.co.uk (Steve Ridley)
Date: Mon, 11 Mar 2013 11:42:53 +0100 (CET)
Subject: [sdiy] MXR Pitch Transposer Schem/Circuit
In-Reply-To: <524076CE-F656-4601-A79D-B8BB9A16823F@martinfay.com>
Message-ID: <26765649.17571362998573285.JavaMail.www@wwinf3704>



>> I think something like this could be done with BBDs but the
>> artifacts of the modulated clock would surely be "interesting"!

I vaguely remember seeing a circuit (or app note) for this. It used two
BBDs and was intended for dictaphones - so they could play stuff back
slow enough to type but keep the pitch fairly constant. I guess quality
wasn't critical.



Steve



From tom at electricdruid.net  Mon Mar 11 12:09:29 2013
From: tom at electricdruid.net (Tom Wiltshire)
Date: Mon, 11 Mar 2013 11:09:29 +0000
Subject: [sdiy] MXR Pitch Transposer Schem/Circuit
In-Reply-To: <26765649.17571362998573285.JavaMail.www@wwinf3704>
References: <26765649.17571362998573285.JavaMail.www@wwinf3704>
Message-ID: <DE378427-BAF2-4065-939E-7D4F2FA2E897@electricdruid.net>


On 11 Mar 2013, at 10:42, Steve Ridley <spr at spridley.freeserve.co.uk> wrote:

> 
> 
>>> I think something like this could be done with BBDs but the
>>> artifacts of the modulated clock would surely be "interesting"!
> 
> I vaguely remember seeing a circuit (or app note) for this. It used two
> BBDs and was intended for dictaphones - so they could play stuff back
> slow enough to type but keep the pitch fairly constant. I guess quality
> wasn't critical.
> 
> Steve

That was what I'd thought of doing too. I've done some experiments recently using the new NCO peripheral on the enhanced midrange PICs to generate a BBD clock, which works very nicely. You can wrap up a VC digital LFO and a BBD clock chip all onto one 8-pin DIP. This discussion reminded me of that.

If there was more than one NCO available (big if - I don't know if there's a PIC with two NCOs), you could organise the chip to provide the ramp-modulated clocks, and probably the crossfade signals too. Turning all that into software makes the whole thing much more manageable. It'd be a PIC, maybe a DAC (for the crossfade signals? Or use PWM?) a couple of BBDs and a LM13700 (two VCAs for the crossfading). Four or maybe five ICs.

Using modern technology to get a utterly obsolete and lo-fi pitch shifting technique!! Isn't that what it's for?!

Tom



From rainer at buchty.net  Mon Mar 11 13:05:50 2013
From: rainer at buchty.net (Rainer Buchty)
Date: Mon, 11 Mar 2013 13:05:50 +0100 (CET)
Subject: [sdiy] [ot] Ensoniq Mirage
In-Reply-To: <5F7A1AC1-DE22-4920-94C1-FFD3C1AE0DBD@sounds.wa.com>
References: <CAA25cvdKxE3C13_-U5mR1rj0Do7Fz8EZQfhkKPb+2Ppp4zUmCw@mail.gmail.com>
	<513A46F0.8000303@gjcp.net>
	<D6D2F726-246A-4D86-B901-BA0E51A1E903@pingdynasty.com>
	<513A5314.9050709@gjcp.net>
	<5F7A1AC1-DE22-4920-94C1-FFD3C1AE0DBD@sounds.wa.com>
Message-ID: <alpine.DEB.2.00.1303111254530.19170@localhost>

On Sun, 10 Mar 2013, rsdio at sounds.wa.com wrote:

> I wouldn't call it conventional. Ensoniq used 10 sectors, while the 
> ubiquitous IBM-PC DOS used only 9 sectors.

Also with PCs you could've gone higher, both sector and track count. 
There were several tools around which allowed that.

Multiple sector sizes, however, is something which is not supported very 
well on PC floppy controllers (which are NEC 765 descendants by nature) 
but required more intelligent ones like the ubiquitous WD1772.

On the Atari ST there was one particularly aggressive formatter, which 
IIRC went up to track 82 or 83, squeezing out the theoretical 1MB per DD 
disk.

> Where DOS put 720 KB on a DD floppy, Ensoniq managed to fit 800 KB. 
> Not much, but it is 11% more. There is custom software out there to 
> read and write Ensoniq floppies.

Ensoniq even managed to get 880kB onto a disk (80*2*(5*1024+512)) or 
440kB per side, at least on the machines using their MSS format.

That's why the Ensoniqs are equally picky about disk quality as e.g. the 
Commodore Amiga. Disks that fail formatting on those machines often will 
happily work on PCs -- sometimes at reduced capacity, though, for 
marking bad sectors as occupied.

Rainer



From ka4hjh at gmail.com  Mon Mar 11 19:51:55 2013
From: ka4hjh at gmail.com (KA4HJH)
Date: Mon, 11 Mar 2013 14:51:55 -0400
Subject: [sdiy] Incredible bargain on craigslist
Message-ID: <E735814F-4C44-4D0C-9EED-1572B90EDDFE@gmail.com>

Warning: STRONG language

http://losangeles.craigslist.org/sfv/msg/3668865429.html


Terry Bowman, KA4HJH
"The Mac Doctor"



From recompas at gmail.com  Mon Mar 11 22:02:08 2013
From: recompas at gmail.com (Travis Thatcher)
Date: Mon, 11 Mar 2013 16:02:08 -0500
Subject: [sdiy] Troubleshooting Roland PG300
Message-ID: <CAG0qc=0oJxEaxM=79gxEX-SDGM1UaLStViC6Dz8YQ_ZCU0=Zaw@mail.gmail.com>

I'm working on a PG300 for a friend of mine. He said it was working
for a little while when he got it and then one day just stopped
sending data.  At first look all connections checked out and the 5v
regulator was working properly. I Tested the inputs to the CPU with a
scope and saw voltages changing with the sliders, however on the
output side of the cpu, pin 17 there was no change.  The crystal is
good, and power seemed fine.  I heated up the board a bit with a hot
air gun to check for intermittent connections and briefly got some
sysex activity but it then the board cooled and ceased transmitting
again.  I went over most of the solder joints around the cpu/eprom and
checked the traces with a meter. Everything seems okay and I'm a bit
stumped.  I'm planning on removing the hex inverter, IC2 to test it
tonight out of circuit and see if perhaps it was pulling down the
output from the CPU...beyond that or a broken cpu I can't really think
of anything else.  Anybody have any ideas?

-- 
http://www.travisthatcher.com
http://www.soundcloud.com/travisthatcher
http://voiceofsaturn.blogspot.com


From eric at ericframpton.com  Tue Mar 12 06:42:47 2013
From: eric at ericframpton.com (Eric Frampton)
Date: Tue, 12 Mar 2013 01:42:47 -0400
Subject: [sdiy] Oberheim DSX Test Cassette
Message-ID: <57C7594F-EF9F-4D81-8DE5-A6A5F8CDFCA0@ericframpton.com>

Hey -

Does anybody have one of these, or have a rip of it to a WAV file that they could send me? I'm trying to get a DSX up and running and want to make sure everything's dialed in right.

TIA,
Eric



From goaqihai at yahoo.co.uk  Tue Mar 12 15:06:21 2013
From: goaqihai at yahoo.co.uk (ganesha)
Date: Tue, 12 Mar 2013 14:06:21 +0000 (GMT)
Subject: [sdiy] Complete Roland System 700 service manual
Message-ID: <1363097181.74792.YahooMailNeo@web171502.mail.ir2.yahoo.com>

Hi,
I'm looking for the complete Roland System 700 service manual, including the PCB layouts. 
I can find most of the schematics, but not the complete service manual.
Can someone mail me a copy?
Thanks!


From marc.nostromo at gmail.com  Tue Mar 12 15:19:53 2013
From: marc.nostromo at gmail.com (Marc Nostromo [M-.-n])
Date: Tue, 12 Mar 2013 15:19:53 +0100
Subject: [sdiy] monotron DC power
In-Reply-To: <CAKcdib10p9-2FMJ-jBvq_y-tkqDo5T+G7RJdfRgXd6BB8G=0-A@mail.gmail.com>
References: <CAEQ8eXPChR+DUA8zWoU7Oi=PMSqDGp_FJ4Vg6powJ+oYsZHHDA@mail.gmail.com>
	<CAKcdib10p9-2FMJ-jBvq_y-tkqDo5T+G7RJdfRgXd6BB8G=0-A@mail.gmail.com>
Message-ID: <CAEQ8eXOKsukQPjMVFTq1U5QHz8axSZytnTPfXxNYb5g9uxebAA@mail.gmail.com>

Great Jacob. That will help. I think I got all of what I wanted pretty
much figured out.

I'm still a bit puzzled about the BIAS point. I understand it gives
some virtual ground between 0/5v but why has it been labeled so
clearely ? So far I seem to be able to control pitch and cut-off
without really using it and I'm wondering what I might be doing wrong.

Thanks
Marc.

2013/3/5 Jacob Watters <jacobwatters at gmail.com>:
> You can power it from 5V, but connect the supply to the +V solder
> point and NOT to where the batteries go.
>
> Also, if you do this then you should not have batteries in the
> Monotron at the same time.
>
> There are a couple of links to the schematics I used near the bottom
> of this page: http://www.jacobwatters.com/monotron-mod.php
>
> Let me know if you have any other questions.  I have helped a few
> other people already who contacted me directly.
>
>
> -Jacob Watters
>
>
> On Tue, Mar 5, 2013 at 9:00 AM, Marc Nostromo [M-.-n]
> <marc.nostromo at gmail.com> wrote:
>> Hey Jacob
>>
>> I'm toying with modding my monotron and would like to power it from
>> DC. I've done quite a bit of research  and could not find if it can be
>> powered directly from 5v or if I need to use a power regulator to
>> bring the voltage to 3v.
>>
>> Could you tell me what you've done ?
>>
>> Thanks a lot,
>> Marc.
>>
>> --
>> http://marc-nostromo.com



--
http://marc-nostromo.com


From recompas at gmail.com  Tue Mar 12 15:35:45 2013
From: recompas at gmail.com (Travis Thatcher)
Date: Tue, 12 Mar 2013 09:35:45 -0500
Subject: [sdiy] PG-300 woes pt 2
Message-ID: <CAG0qc=3D8K4uS9VM62kZRnpfLu0HQdewtb7G=3QLwaGd7BfCvg@mail.gmail.com>

It seems I've unfortunately hit a bit of a wall with this PG300.

Last night I checked the hex invertor chip out of circuit and it seems
fine. Also checked for continuity on all the traces for the cpu, eprom
and supporting components and everything checked out.  I'm wondering
if the eeprom got corrupted or perhaps the cpu itself is dead. I'm not
entirely sure how the multiplexor control works on these but I noticed
that no matter what control is moved, the address controls of the
mux's are all set to high, so taking the input from pin 4.  I'm unsure
if these control signals not changing is an indication of what the
problem could be. The output of each MUX does make it to the CPU.

There isn't anything weird that would require the PG300 to be
connected to a synth in order to properly function?  If I monitor the
output of the CPU, at pin 17 with a scope, I'm assuming I should see a
signal if it is properly working?  I was also checking, obviously,
with a midi monitor program.

Any help would be very appreciated, I'm kind of stuck!

-- 
http://www.travisthatcher.com
http://www.soundcloud.com/travisthatcher
http://voiceofsaturn.blogspot.com


From luis.marka at terra.com.br  Tue Mar 12 15:48:56 2013
From: luis.marka at terra.com.br (=?ISO-8859-1?Q?Lu=EDs_Marka?=)
Date: Tue, 12 Mar 2013 11:48:56 -0300
Subject: [sdiy] PG-300 woes pt 2
In-Reply-To: <CAG0qc=2acyhO5uw3t2wuSsH7EPmRCqwt306b_c0bABTP0m5_xQ@mail.gmail.com>
References: <djwk2usbm1qvjh2tn3rnbhuh.1363100815213@email.android.com>
	<CAG0qc=2acyhO5uw3t2wuSsH7EPmRCqwt306b_c0bABTP0m5_xQ@mail.gmail.com>
Message-ID: <513F4058.6020007@terra.com.br>

Hi Travis,
I would check if the I/O in the EPROM pins are OK, that is, the EPROM is 
receiving address data and spits out its contents.
Think the only way to find out if the program in the EPROM is sound is 
to remove the chip, read its contents and compare to the contents of a 
good EPROM.

Also, I found this post on Analogue Heaven that may help:

http://analogue-heaven.1065350.n5.nabble.com/Roland-CPU-for-PG300-Programmer-for-MKS50-analog-synth-td102850.html

On 3/12/2013 12:31 PM, Travis Thatcher wrote:
> Thanks Richie,
>
> I did check the clock pins of the cpu and measured it looked like a
> 12MHz clock on pins 30 and 31, rather I saw a nice wave-form which I
> took to be the clock. I'll look more closely at it. I'll also look at
> the reset pin.  How exactly does that reset IC work? I'm assuming that
> pin is high when it is to be held in reset and low when it should be
> in run mode?
>
> On Tue, Mar 12, 2013 at 10:18 AM, Richie Burnett
> <rburnett at richieburnett.co.uk> wrote:
>> There should be plenty of activity on the i/o lines from the micro as it reads the multiplexed pot levels and switch positions many times every second. If you're not seeing a hive of activity then I would suspect a faulty CPU clock, CPU being held in reset, or corrupt program memory.
>>
>> You said that it momentarily worked when you heated the board. This is unlikely to help a corrupted ROM, but high temperatures are known to aid the starting of oscillations in crystal oscillators that are marginal. Also check for any damp or flux residue around the crystal and it's connection to the CPU.
>>
>> Check the reset pin status to check it's not being held in reset all the time!
>>
>> If any chips are in sockets, i'd suspect poor contacts, and remove and re-insert them just in case.
>>
>> Good luck. And let us know the fault when you get it working!
>>
>> -Richie,
>>
>> Sent from my Sony Ericsson Xperia ray
>>
>> Travis Thatcher <recompas at gmail.com> wrote:
>>
>>> It seems I've unfortunately hit a bit of a wall with this PG300.
>>>
>>> Last night I checked the hex invertor chip out of circuit and it seems
>>> fine. Also checked for continuity on all the traces for the cpu, eprom
>>> and supporting components and everything checked out.  I'm wondering
>>> if the eeprom got corrupted or perhaps the cpu itself is dead. I'm not
>>> entirely sure how the multiplexor control works on these but I noticed
>>> that no matter what control is moved, the address controls of the
>>> mux's are all set to high, so taking the input from pin 4.  I'm unsure
>>> if these control signals not changing is an indication of what the
>>> problem could be. The output of each MUX does make it to the CPU.
>>>
>>> There isn't anything weird that would require the PG300 to be
>>> connected to a synth in order to properly function?  If I monitor the
>>> output of the CPU, at pin 17 with a scope, I'm assuming I should see a
>>> signal if it is properly working?  I was also checking, obviously,
>>> with a midi monitor program.
>>>
>>> Any help would be very appreciated, I'm kind of stuck!
>>>
>>> --
>>> http://www.travisthatcher.com
>>> http://www.soundcloud.com/travisthatcher
>>> http://voiceofsaturn.blogspot.com
>>> _______________________________________________
>>> Synth-diy mailing list
>>> Synth-diy at dropmix.xs4all.nl
>>> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>
>



From rburnett at richieburnett.co.uk  Tue Mar 12 16:18:57 2013
From: rburnett at richieburnett.co.uk (Richie Burnett)
Date: Tue, 12 Mar 2013 15:18:57 +0000
Subject: [sdiy] PG-300 woes pt 2
Message-ID: <djwk2usbm1qvjh2tn3rnbhuh.1363100815213@email.android.com>

There should be plenty of activity on the i/o lines from the micro as it reads the multiplexed pot levels and switch positions many times every second. If you're not seeing a hive of activity then I would suspect a faulty CPU clock, CPU being held in reset, or corrupt program memory.

You said that it momentarily worked when you heated the board. This is unlikely to help a corrupted ROM, but high temperatures are known to aid the starting of oscillations in crystal oscillators that are marginal. Also check for any damp or flux residue around the crystal and it's connection to the CPU.

Check the reset pin status to check it's not being held in reset all the time!

If any chips are in sockets, i'd suspect poor contacts, and remove and re-insert them just in case.

Good luck. And let us know the fault when you get it working!

-Richie,

Sent from my Sony Ericsson Xperia ray

Travis Thatcher <recompas at gmail.com> wrote:

>It seems I've unfortunately hit a bit of a wall with this PG300.
>
>Last night I checked the hex invertor chip out of circuit and it seems
>fine. Also checked for continuity on all the traces for the cpu, eprom
>and supporting components and everything checked out.  I'm wondering
>if the eeprom got corrupted or perhaps the cpu itself is dead. I'm not
>entirely sure how the multiplexor control works on these but I noticed
>that no matter what control is moved, the address controls of the
>mux's are all set to high, so taking the input from pin 4.  I'm unsure
>if these control signals not changing is an indication of what the
>problem could be. The output of each MUX does make it to the CPU.
>
>There isn't anything weird that would require the PG300 to be
>connected to a synth in order to properly function?  If I monitor the
>output of the CPU, at pin 17 with a scope, I'm assuming I should see a
>signal if it is properly working?  I was also checking, obviously,
>with a midi monitor program.
>
>Any help would be very appreciated, I'm kind of stuck!
>
>-- 
>http://www.travisthatcher.com
>http://www.soundcloud.com/travisthatcher
>http://voiceofsaturn.blogspot.com
>_______________________________________________
>Synth-diy mailing list
>Synth-diy at dropmix.xs4all.nl
>http://dropmix.xs4all.nl/mailman/listinfo/synth-diy

From recompas at gmail.com  Tue Mar 12 16:31:20 2013
From: recompas at gmail.com (Travis Thatcher)
Date: Tue, 12 Mar 2013 10:31:20 -0500
Subject: [sdiy] PG-300 woes pt 2
In-Reply-To: <djwk2usbm1qvjh2tn3rnbhuh.1363100815213@email.android.com>
References: <djwk2usbm1qvjh2tn3rnbhuh.1363100815213@email.android.com>
Message-ID: <CAG0qc=2acyhO5uw3t2wuSsH7EPmRCqwt306b_c0bABTP0m5_xQ@mail.gmail.com>

Thanks Richie,

I did check the clock pins of the cpu and measured it looked like a
12MHz clock on pins 30 and 31, rather I saw a nice wave-form which I
took to be the clock. I'll look more closely at it. I'll also look at
the reset pin.  How exactly does that reset IC work? I'm assuming that
pin is high when it is to be held in reset and low when it should be
in run mode?

On Tue, Mar 12, 2013 at 10:18 AM, Richie Burnett
<rburnett at richieburnett.co.uk> wrote:
> There should be plenty of activity on the i/o lines from the micro as it reads the multiplexed pot levels and switch positions many times every second. If you're not seeing a hive of activity then I would suspect a faulty CPU clock, CPU being held in reset, or corrupt program memory.
>
> You said that it momentarily worked when you heated the board. This is unlikely to help a corrupted ROM, but high temperatures are known to aid the starting of oscillations in crystal oscillators that are marginal. Also check for any damp or flux residue around the crystal and it's connection to the CPU.
>
> Check the reset pin status to check it's not being held in reset all the time!
>
> If any chips are in sockets, i'd suspect poor contacts, and remove and re-insert them just in case.
>
> Good luck. And let us know the fault when you get it working!
>
> -Richie,
>
> Sent from my Sony Ericsson Xperia ray
>
> Travis Thatcher <recompas at gmail.com> wrote:
>
>>It seems I've unfortunately hit a bit of a wall with this PG300.
>>
>>Last night I checked the hex invertor chip out of circuit and it seems
>>fine. Also checked for continuity on all the traces for the cpu, eprom
>>and supporting components and everything checked out.  I'm wondering
>>if the eeprom got corrupted or perhaps the cpu itself is dead. I'm not
>>entirely sure how the multiplexor control works on these but I noticed
>>that no matter what control is moved, the address controls of the
>>mux's are all set to high, so taking the input from pin 4.  I'm unsure
>>if these control signals not changing is an indication of what the
>>problem could be. The output of each MUX does make it to the CPU.
>>
>>There isn't anything weird that would require the PG300 to be
>>connected to a synth in order to properly function?  If I monitor the
>>output of the CPU, at pin 17 with a scope, I'm assuming I should see a
>>signal if it is properly working?  I was also checking, obviously,
>>with a midi monitor program.
>>
>>Any help would be very appreciated, I'm kind of stuck!
>>
>>--
>>http://www.travisthatcher.com
>>http://www.soundcloud.com/travisthatcher
>>http://voiceofsaturn.blogspot.com
>>_______________________________________________
>>Synth-diy mailing list
>>Synth-diy at dropmix.xs4all.nl
>>http://dropmix.xs4all.nl/mailman/listinfo/synth-diy



-- 
http://www.travisthatcher.com
http://www.soundcloud.com/travisthatcher
http://voiceofsaturn.blogspot.com


From dfevans at sekrit.eu  Wed Mar 13 10:45:45 2013
From: dfevans at sekrit.eu (David Evans)
Date: Wed, 13 Mar 2013 09:45:45 +0000
Subject: [sdiy] Buying Digilent stuff in the Uk
Message-ID: <302FE53B-6A77-473C-8124-708FDD8EDA90@sekrit.eu>

Hi, all.

I want to get a Digilent FX2 breadboard and a PmodBB breadboard PMod. Farnell carries neither, based on a quick search. Do I need to just bite the bullet and pay the ~$38 shipping that Digilent itself charges?

Ta!




From Peter.Ullrich at kapsch.net  Wed Mar 13 10:53:37 2013
From: Peter.Ullrich at kapsch.net (Ullrich Peter)
Date: Wed, 13 Mar 2013 10:53:37 +0100
Subject: [sdiy] Buying Digilent stuff in the Uk
Message-ID: <B156A86831A33949A0977840648802478B1366B733@S060VS20.kapsch.co.at>

Hi!

Trenz Electronic in Germany is also a distributor for whole Europe.

http://shop.trenz-electronic.de/catalog/product_info.php?manufacturers_id=16&products_id=511

You have to register to see the postage costs...

Ciao
Peter

-----Urspr?ngliche Nachricht-----
Von: synth-diy-bounces at dropmix.xs4all.nl [mailto:synth-diy-bounces at dropmix.xs4all.nl] Im Auftrag von David Evans
Gesendet: Mittwoch, 13. M?rz 2013 10:46
An: SDIY List
Betreff: [sdiy] Buying Digilent stuff in the Uk

Hi, all.

I want to get a Digilent FX2 breadboard and a PmodBB breadboard PMod. Farnell carries neither, based on a quick search. Do I need to just bite the bullet and pay the ~$38 shipping that Digilent itself charges?

Ta!


_______________________________________________
Synth-diy mailing list
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The information contained in this e-mail message is privileged and
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who receives this message and who is not the addressee, one of his
employees or an agent entitled to hand it over to the addressee, is
informed that he may not use, disclose or reproduce the contents
thereof, and is kindly asked to notify the sender and delete the e-mail
immediately.




From dfevans at sekrit.eu  Wed Mar 13 17:10:54 2013
From: dfevans at sekrit.eu (David Evans)
Date: Wed, 13 Mar 2013 16:10:54 +0000
Subject: [sdiy] Buying Digilent stuff in the Uk
In-Reply-To: <B156A86831A33949A0977840648802478B1366B733@S060VS20.kapsch.co.at>
References: <B156A86831A33949A0977840648802478B1366B733@S060VS20.kapsch.co.at>
Message-ID: <FEBE4636-3139-4664-9019-7AAB0088D9E0@sekrit.eu>

On 13 Mar 2013, at 09:53, Ullrich Peter wrote:

> Hi!
> 
> Trenz Electronic in Germany is also a distributor for whole Europe.
> 
> http://shop.trenz-electronic.de/catalog/product_info.php?manufacturers_id=16&products_id=511
> 

Fantastic. Thanks!




From subjectivity at hotmail.com  Thu Mar 14 16:04:42 2013
From: subjectivity at hotmail.com (dan snazelle)
Date: Thu, 14 Mar 2013 11:04:42 -0400
Subject: [sdiy] is raspberry pi the future?
Message-ID: <BLU0-SMTP230D6940684AB514B9ADF4BC8EC0@phx.gbl>

Has anyone here gotten one of these yet to do audio work? it seems like such a powerful platform with so much ram!! 



512mb instead of the 512k often found on ucontrollers.






From tom at electricdruid.net  Thu Mar 14 16:42:03 2013
From: tom at electricdruid.net (Tom Wiltshire)
Date: Thu, 14 Mar 2013 15:42:03 +0000
Subject: [sdiy] is raspberry pi the future?
In-Reply-To: <BLU0-SMTP230D6940684AB514B9ADF4BC8EC0@phx.gbl>
References: <BLU0-SMTP230D6940684AB514B9ADF4BC8EC0@phx.gbl>
Message-ID: <A61FF618-7614-4D4D-8E4A-32042FDB5815@electricdruid.net>


On 14 Mar 2013, at 15:04, dan snazelle <subjectivity at hotmail.com> wrote:

> Has anyone here gotten one of these yet to do audio work? it seems like such a powerful platform with so much ram!! 
> 
> 
> 
> 512mb instead of the 512k often found on controllers.

I used to *dream* of having 512K! We only had 16K on the dsPIC, and we thought we were lucky?

To actually answer the question, I've got a Raspberry Pi but I haven't done anything audio with it yet. Is there anything online about audio on Pi yet?

T.



From Peter.Ullrich at kapsch.net  Thu Mar 14 16:46:02 2013
From: Peter.Ullrich at kapsch.net (Ullrich Peter)
Date: Thu, 14 Mar 2013 16:46:02 +0100
Subject: [sdiy] is raspberry pi the future?
Message-ID: <B156A86831A33949A0977840648802478B1623E0E7@S060VS20.kapsch.co.at>

Some hints on audio:

http://dbader.org/blog/crackle-free-audio-on-the-raspberry-pi-with-mpd-and-pulseaudio


-----Urspr?ngliche Nachricht-----
Von: synth-diy-bounces at dropmix.xs4all.nl [mailto:synth-diy-bounces at dropmix.xs4all.nl] Im Auftrag von Tom Wiltshire
Gesendet: Donnerstag, 14. M?rz 2013 16:42
An: dan snazelle
Cc: *SYNTH DIY
Betreff: Re: [sdiy] is raspberry pi the future?


On 14 Mar 2013, at 15:04, dan snazelle <subjectivity at hotmail.com> wrote:

> Has anyone here gotten one of these yet to do audio work? it seems like such a powerful platform with so much ram!! 
> 
> 
> 
> 512mb instead of the 512k often found on controllers.

I used to *dream* of having 512K! We only had 16K on the dsPIC, and we thought we were lucky.

To actually answer the question, I've got a Raspberry Pi but I haven't done anything audio with it yet. Is there anything online about audio on Pi yet?

T.

_______________________________________________
Synth-diy mailing list
Synth-diy at dropmix.xs4all.nl
http://dropmix.xs4all.nl/mailman/listinfo/synth-diy

The information contained in this e-mail message is privileged and
confidential and is for the exclusive use of the addressee. The person
who receives this message and who is not the addressee, one of his
employees or an agent entitled to hand it over to the addressee, is
informed that he may not use, disclose or reproduce the contents
thereof, and is kindly asked to notify the sender and delete the e-mail
immediately.




From Peter.Ullrich at kapsch.net  Thu Mar 14 16:47:15 2013
From: Peter.Ullrich at kapsch.net (Ullrich Peter)
Date: Thu, 14 Mar 2013 16:47:15 +0100
Subject: [sdiy] is raspberry pi the future?
Message-ID: <B156A86831A33949A0977840648802478B1623E0EB@S060VS20.kapsch.co.at>

Maybe interesting:

Synthesizer with the Pi:

http://www.raspberrypi.org/archives/1878

Ciao
Peter

-----Urspr?ngliche Nachricht-----
Von: synth-diy-bounces at dropmix.xs4all.nl [mailto:synth-diy-bounces at dropmix.xs4all.nl] Im Auftrag von Tom Wiltshire
Gesendet: Donnerstag, 14. M?rz 2013 16:42
An: dan snazelle
Cc: *SYNTH DIY
Betreff: Re: [sdiy] is raspberry pi the future?


On 14 Mar 2013, at 15:04, dan snazelle <subjectivity at hotmail.com> wrote:

> Has anyone here gotten one of these yet to do audio work? it seems like such a powerful platform with so much ram!! 
> 
> 
> 
> 512mb instead of the 512k often found on controllers.

I used to *dream* of having 512K! We only had 16K on the dsPIC, and we thought we were lucky.

To actually answer the question, I've got a Raspberry Pi but I haven't done anything audio with it yet. Is there anything online about audio on Pi yet?

T.

_______________________________________________
Synth-diy mailing list
Synth-diy at dropmix.xs4all.nl
http://dropmix.xs4all.nl/mailman/listinfo/synth-diy

The information contained in this e-mail message is privileged and
confidential and is for the exclusive use of the addressee. The person
who receives this message and who is not the addressee, one of his
employees or an agent entitled to hand it over to the addressee, is
informed that he may not use, disclose or reproduce the contents
thereof, and is kindly asked to notify the sender and delete the e-mail
immediately.




From neil.johnson97 at ntlworld.com  Thu Mar 14 17:07:51 2013
From: neil.johnson97 at ntlworld.com (Neil Johnson)
Date: Thu, 14 Mar 2013 16:07:51 +0000
Subject: [sdiy] is raspberry pi the future?
In-Reply-To: <BLU0-SMTP230D6940684AB514B9ADF4BC8EC0@phx.gbl>
References: <BLU0-SMTP230D6940684AB514B9ADF4BC8EC0@phx.gbl>
Message-ID: <CAEaByMScr-Zub3f4A_OXwo+jrmqr7zvskQSOt3rW=fweE4u54w@mail.gmail.com>

Hi,

> Has anyone here gotten one of these yet to do audio work? it seems like such a powerful platform with so much ram!!

The PWM audio output does work, mostly.  It's not exactly CD-audio
quality, more like 11 bits on a good day.  Better option would be a
USB soundcard.

Neil
-- 
Modules and more: http://www.cesyg.com
Homepage: http://www.njohnson.co.uk


From cheater00 at gmail.com  Thu Mar 14 17:25:08 2013
From: cheater00 at gmail.com (cheater cheater)
Date: Thu, 14 Mar 2013 17:25:08 +0100
Subject: [sdiy] is raspberry pi the future?
In-Reply-To: <CAEaByMScr-Zub3f4A_OXwo+jrmqr7zvskQSOt3rW=fweE4u54w@mail.gmail.com>
References: <BLU0-SMTP230D6940684AB514B9ADF4BC8EC0@phx.gbl>
	<CAEaByMScr-Zub3f4A_OXwo+jrmqr7zvskQSOt3rW=fweE4u54w@mail.gmail.com>
Message-ID: <CA+9GZUjnYqWSEb-CtmyujE0SBZrY7y9Bi8026T9NqznqmqVsMg@mail.gmail.com>

The Raspberry Pi has GPIO. Surely, for building a synth you don't want
to have the shittiness of USB?

http://elinux.org/RPi_BCM2835_GPIOs

Happy Pi Day everyone. Too bad it's only half of tau: http://tauday.com/

Cheers,
D.

On Thu, Mar 14, 2013 at 5:07 PM, Neil Johnson
<neil.johnson97 at ntlworld.com> wrote:
> Hi,
>
>> Has anyone here gotten one of these yet to do audio work? it seems like such a powerful platform with so much ram!!
>
> The PWM audio output does work, mostly.  It's not exactly CD-audio
> quality, more like 11 bits on a good day.  Better option would be a
> USB soundcard.
>
> Neil
> --
> Modules and more: http://www.cesyg.com
> Homepage: http://www.njohnson.co.uk
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy


From neil.johnson97 at ntlworld.com  Thu Mar 14 17:28:18 2013
From: neil.johnson97 at ntlworld.com (Neil Johnson)
Date: Thu, 14 Mar 2013 16:28:18 +0000
Subject: [sdiy] is raspberry pi the future?
In-Reply-To: <CA+9GZUjnYqWSEb-CtmyujE0SBZrY7y9Bi8026T9NqznqmqVsMg@mail.gmail.com>
References: <BLU0-SMTP230D6940684AB514B9ADF4BC8EC0@phx.gbl>
	<CAEaByMScr-Zub3f4A_OXwo+jrmqr7zvskQSOt3rW=fweE4u54w@mail.gmail.com>
	<CA+9GZUjnYqWSEb-CtmyujE0SBZrY7y9Bi8026T9NqznqmqVsMg@mail.gmail.com>
Message-ID: <CAEaByMSNCm47DPL172WVO2tP932A+06qYE+Z99xz9X13o=v25w@mail.gmail.com>

Hi,

cheater cheater wrote:
> The Raspberry Pi has GPIO. Surely, for building a synth you don't want
> to have the shittiness of USB?
>
> http://elinux.org/RPi_BCM2835_GPIOs

A crappy USB interface would likely be better than the on-board audio outs.

Neil
-- 
Modules and more: http://www.cesyg.com
Homepage: http://www.njohnson.co.uk


From yo at vacoloco.net  Thu Mar 14 18:23:57 2013
From: yo at vacoloco.net (Paul Maddox)
Date: Thu, 14 Mar 2013 17:23:57 +0000
Subject: [sdiy] is raspberry pi the future?
In-Reply-To: <CAEaByMSNCm47DPL172WVO2tP932A+06qYE+Z99xz9X13o=v25w@mail.gmail.com>
References: <BLU0-SMTP230D6940684AB514B9ADF4BC8EC0@phx.gbl>
	<CAEaByMScr-Zub3f4A_OXwo+jrmqr7zvskQSOt3rW=fweE4u54w@mail.gmail.com>
	<CA+9GZUjnYqWSEb-CtmyujE0SBZrY7y9Bi8026T9NqznqmqVsMg@mail.gmail.com>
	<CAEaByMSNCm47DPL172WVO2tP932A+06qYE+Z99xz9X13o=v25w@mail.gmail.com>
Message-ID: <CADm3=9SaOX5YFHh+_n1FObpKC6pJNrZ6NHcdUEmANT78tupeuQ@mail.gmail.com>

the Raspi has some nice features, but I don't think I'd use it for
audio generation.
Perhaps control of a synth, for example, a raspberry-pi with an FPGA,
a few DACs and some VCFs would make one hell of a polysynth.


From thomas at pdp7.org  Thu Mar 14 18:31:33 2013
From: thomas at pdp7.org (Thomas Strathmann)
Date: Thu, 14 Mar 2013 18:31:33 +0100
Subject: [sdiy] is raspberry pi the future?
In-Reply-To: <B156A86831A33949A0977840648802478B1623E0E7@S060VS20.kapsch.co.at>
References: <B156A86831A33949A0977840648802478B1623E0E7@S060VS20.kapsch.co.at>
Message-ID: <51420975.7010401@pdp7.org>

On 14.03.13 16:46, Ullrich Peter wrote:
> Some hints on audio:
> 
> http://dbader.org/blog/crackle-free-audio-on-the-raspberry-pi-with-mpd-and-pulseaudio

This looks like just the thing I need to get back into trying to get
something useable together with the Pi. But the fact that the onboard
audio output was unuseable from puredata turned me off the whole idea
pretty soon. Maybe it's time for another attempt ...

	Thomas


From thx1138 at earthlink.net  Thu Mar 14 18:35:15 2013
From: thx1138 at earthlink.net (Terry Shultz)
Date: Thu, 14 Mar 2013 10:35:15 -0700
Subject: [sdiy] is raspberry pi the future or Cortex M series
In-Reply-To: <CADm3=9SaOX5YFHh+_n1FObpKC6pJNrZ6NHcdUEmANT78tupeuQ@mail.gmail.com>
References: <BLU0-SMTP230D6940684AB514B9ADF4BC8EC0@phx.gbl>
	<CAEaByMScr-Zub3f4A_OXwo+jrmqr7zvskQSOt3rW=fweE4u54w@mail.gmail.com>
	<CA+9GZUjnYqWSEb-CtmyujE0SBZrY7y9Bi8026T9NqznqmqVsMg@mail.gmail.com>
	<CAEaByMSNCm47DPL172WVO2tP932A+06qYE+Z99xz9X13o=v25w@mail.gmail.com>
	<CADm3=9SaOX5YFHh+_n1FObpKC6pJNrZ6NHcdUEmANT78tupeuQ@mail.gmail.com>
Message-ID: <A3FC0A35-4BAB-432C-A488-3BAFF945A867@earthlink.net>

Hi Paul,

These little Cortex M3/M4 ARM devices are great to do simple audio projects with.

The Arduino add-ons are plentiful and tools are getting pretty good to work with.

I have not seen any real Mac based tools, but I use WinXP on my MAc Book Pro using VM Fusion and so far all my PC tools work fine.

As I am no longer Freescale employed, I have been trying out other kits.

The Cypress folks just came thru town with a seminar on the CY8CKIT-050 PSoC 5LP and I have general Midi and a MP3 player up and running already.

The kit came with a Keil Compiler for 1 year Free. Also they gave us the kit for Free!!!

Just can't beat Free when you are unemployed.


Terry Shultz
thx1138 at earthlink.net



On Mar 14, 2013, at 10:23 AM, Paul Maddox <yo at vacoloco.net> wrote:

> the Raspi has some nice features, but I don't think I'd use it for
> audio generation.
> Perhaps control of a synth, for example, a raspberry-pi with an FPGA,
> a few DACs and some VCFs would make one hell of a polysynth.
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy



From wackyvorlon at me.com  Thu Mar 14 18:35:39 2013
From: wackyvorlon at me.com (Paul Anderson)
Date: Thu, 14 Mar 2013 13:35:39 -0400
Subject: [sdiy] is raspberry pi the future?
In-Reply-To: <A61FF618-7614-4D4D-8E4A-32042FDB5815@electricdruid.net>
References: <BLU0-SMTP230D6940684AB514B9ADF4BC8EC0@phx.gbl>
	<A61FF618-7614-4D4D-8E4A-32042FDB5815@electricdruid.net>
Message-ID: <57275E47-9711-4B31-8433-B5251AADF567@me.com>

It should be able to run csound or supercollider. 

--------
Paul Anderson -- VE3HOP

On 2013-03-14, at 11:42 AM, Tom Wiltshire <tom at electricdruid.net> wrote:

> 
> On 14 Mar 2013, at 15:04, dan snazelle <subjectivity at hotmail.com> wrote:
> 
>> Has anyone here gotten one of these yet to do audio work? it seems like such a powerful platform with so much ram!! 
>> 
>> 
>> 
>> 512mb instead of the 512k often found on controllers.
> 
> I used to *dream* of having 512K! We only had 16K on the dsPIC, and we thought we were lucky?
> 
> To actually answer the question, I've got a Raspberry Pi but I haven't done anything audio with it yet. Is there anything online about audio on Pi yet?
> 
> T.
> 
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy


From thx1138 at earthlink.net  Thu Mar 14 18:47:49 2013
From: thx1138 at earthlink.net (Terry Shultz)
Date: Thu, 14 Mar 2013 10:47:49 -0700
Subject: [sdiy] is raspberry pi the future?
In-Reply-To: <51420975.7010401@pdp7.org>
References: <B156A86831A33949A0977840648802478B1623E0E7@S060VS20.kapsch.co.at>
	<51420975.7010401@pdp7.org>
Message-ID: <CCFE8DBA-0661-48E1-8011-EB1CC21FA301@earthlink.net>

Hi Thomas,

The Freescale i.MX53 or i.MX6x Boards have better audio connectivity .

Having been involved in those projects at Freescale, we designed the DSP audio interfaces onto the Cortex A8 AND A9 based products.

The same audio interfaces from our DSPx5672x and DSPx563xx products are incorporated into the ARM processors along with S/PDIF, ASRC, and ESAI audio I/O.

Sure more money, but much better hardware for Audio projects iMHO.

Now that I am on the rank of the un-employed I am playing with Sharc, Blackfin, dspPIC to see what nuggets they might have to offer.

Still tools are a bit of a cost factor but I am going the cheap and get it done route.

I'll post more on my projects as soon as I can get a Web page cobbled together .

Check out my friend Paul Beckmann's tools http://www.dspconcepts.com/products/audio-weaver .

He supports Sharc, BlackFin and ARM A8-A9 and M4 products with the Audio Weaver tools.

Terry Shultz
DSP and ARM Audio Technology & Applications
thx1138 at earthlink.net



On Mar 14, 2013, at 10:31 AM, Thomas Strathmann <thomas at pdp7.org> wrote:

> On 14.03.13 16:46, Ullrich Peter wrote:
>> Some hints on audio:
>> 
>> http://dbader.org/blog/crackle-free-audio-on-the-raspberry-pi-with-mpd-and-pulseaudio
> 
> This looks like just the thing I need to get back into trying to get
> something useable together with the Pi. But the fact that the onboard
> audio output was unuseable from puredata turned me off the whole idea
> pretty soon. Maybe it's time for another attempt ...
> 
> 	Thomas
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy



From thx1138 at earthlink.net  Thu Mar 14 19:02:04 2013
From: thx1138 at earthlink.net (Terry Shultz)
Date: Thu, 14 Mar 2013 11:02:04 -0700
Subject: [sdiy] Contest to win a Tek 465
In-Reply-To: <CACDGQT_WLByiT61aNG67YNj_3TRaxnMw-nV16X0KDKFTZvLm2A@mail.gmail.com>
References: <CACDGQT_WLByiT61aNG67YNj_3TRaxnMw-nV16X0KDKFTZvLm2A@mail.gmail.com>
Message-ID: <743C5E15-8C34-4B7C-90DF-9D15176AE4CC@earthlink.net>

Check out VintageTek web site for great deals as well.

I found 465/475 Tek scopes in abundance on Ebay.

Just be careful on burned displays.

I just bought a Tek 468 + Scope cart for $300.00 

The 468 is like a 465 + Storage function.


Terry Shultz
thx1138 at earthlink.net



On Mar 9, 2013, at 10:10 AM, Tim Parkhurst <tim.parkhurst at gmail.com> wrote:

> Cool contest - Register at this site, send in a picture of your messy
> / interesting workbench, do a few other things, and you're entered in
> a contest to win a classic Tek 465 scope. :)
> 
> http://www.scopejunction.com/author.asp?section_id=1835&doc_id=260090&&goback=.gde_2316414_member_221102380
> 
> 
> 
> Tim (I'm sure we have some VERY 'interesting' workbenches out there) Servo
> ---
> "Sire, the church of God is an anvil that has worn out many hammers."
> - H.L. Hastings
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy



From nvawter at media.mit.edu  Thu Mar 14 19:09:49 2013
From: nvawter at media.mit.edu (nvawter at media.mit.edu)
Date: Thu, 14 Mar 2013 14:09:49 -0400
Subject: [sdiy] is raspberry pi the future or Cortex M series
In-Reply-To: <A3FC0A35-4BAB-432C-A488-3BAFF945A867@earthlink.net>
References: <BLU0-SMTP230D6940684AB514B9ADF4BC8EC0@phx.gbl>
	<CAEaByMScr-Zub3f4A_OXwo+jrmqr7zvskQSOt3rW=fweE4u54w@mail.gmail.com>
	<CA+9GZUjnYqWSEb-CtmyujE0SBZrY7y9Bi8026T9NqznqmqVsMg@mail.gmail.com>
	<CAEaByMSNCm47DPL172WVO2tP932A+06qYE+Z99xz9X13o=v25w@mail.gmail.com>
	<CADm3=9SaOX5YFHh+_n1FObpKC6pJNrZ6NHcdUEmANT78tupeuQ@mail.gmail.com>
	<A3FC0A35-4BAB-432C-A488-3BAFF945A867@earthlink.net>
Message-ID: <20130314140949.20262t2jt2gjzdkd@webmail.media.mit.edu>

Quoting Terry Shultz <thx1138 at earthlink.net>:

> Hi Paul,
>
> These little Cortex M3/M4 ARM devices are great to do simple audio  
> projects with.

Seconded.  I've put up a walk-through for installing development tools  
for the STM32F4 on linux:
http://diydsp.com/livesite/pages/AboutUs




From jasonnanna at gmail.com  Thu Mar 14 19:09:58 2013
From: jasonnanna at gmail.com (Jason Nanna)
Date: Thu, 14 Mar 2013 13:09:58 -0500
Subject: [sdiy] is raspberry pi the future?
In-Reply-To: <BLU0-SMTP230D6940684AB514B9ADF4BC8EC0@phx.gbl>
References: <BLU0-SMTP230D6940684AB514B9ADF4BC8EC0@phx.gbl>
Message-ID: <CAH8amKvm+_ANaRtJmpFS2_uVfYoD3AG0L-OdDKGbNrax2ZnCEQ@mail.gmail.com>

Haven't done anything with the audio interface onboard, but I managed
to get 8 quantized 8-bit DACs (TLC7226) and 8 bi-directional
trigger/gate i/o from the raspberry pi.  Currently integrating it with
a music composition/performance environment a friend of mine is
working on in Python called Pippi:

https://github.com/hecanjog/pippi

The software runs on the Pi and is designed to interface with another
machine (or machines) running pippi, via OSC.  It's embedded into a
small briefcase synth.  Ironing out sync issues with the OSC server at
the moment, but it's close!

On Thu, Mar 14, 2013 at 10:04 AM, dan snazelle <subjectivity at hotmail.com> wrote:
> Has anyone here gotten one of these yet to do audio work? it seems like such a powerful platform with so much ram!!
>
>
>
> 512mb instead of the 512k often found on ucontrollers.
>
>
>
>
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy


From sdiy-mz01 at blauwurf.info  Thu Mar 14 19:21:08 2013
From: sdiy-mz01 at blauwurf.info (Michael Zacherl)
Date: Thu, 14 Mar 2013 19:21:08 +0100
Subject: [sdiy] is raspberry pi the future?
In-Reply-To: <57275E47-9711-4B31-8433-B5251AADF567@me.com>
References: <BLU0-SMTP230D6940684AB514B9ADF4BC8EC0@phx.gbl>
	<A61FF618-7614-4D4D-8E4A-32042FDB5815@electricdruid.net>
	<57275E47-9711-4B31-8433-B5251AADF567@me.com>
Message-ID: <CB78F1D8-554E-4AEA-955E-6D32B9000D6D@blauwurf.info>


On 14.3.2013, at 18:35 , Paul Anderson wrote:

> It should be able to run csound or supercollider. 

It already runs SC and PD!
But don't ask for performance. ;-)

m.



--
keep your ears open: http://blauwurf.at
http://soundcloud.com/noiseconformist





From ebrombaugh1 at cox.net  Thu Mar 14 20:13:33 2013
From: ebrombaugh1 at cox.net (Eric Brombaugh)
Date: Thu, 14 Mar 2013 12:13:33 -0700
Subject: [sdiy] is raspberry pi the future?
In-Reply-To: <CB78F1D8-554E-4AEA-955E-6D32B9000D6D@blauwurf.info>
References: <BLU0-SMTP230D6940684AB514B9ADF4BC8EC0@phx.gbl>
	<A61FF618-7614-4D4D-8E4A-32042FDB5815@electricdruid.net>
	<57275E47-9711-4B31-8433-B5251AADF567@me.com>
	<CB78F1D8-554E-4AEA-955E-6D32B9000D6D@blauwurf.info>
Message-ID: <5142215D.8080906@cox.net>

RPi is a significant step forward for low-cost / high-power embedded 
computing and I'm glad to see it becoming so popular. That said however 
it's not the last word in this space - TI has been providing embedded 
boards with more I/O and more modern processors for several years and 
isn't resting on their laurels.

The current low-cost / high I/O winner is the Beaglebone, which provides 
more than 80 pins of digital and analog interfacing, including 
high-speed SPI, I2C, I2S, LCD, parallel external memory and plain GPIO 
for less than $90. A low-cost Beaglebone with similar capabilities is 
right around the corner and is expected to cost about half that. See 
more about it here:

http://www.citizensinspace.org/2013/03/next-generation-beaglebone/?utm_source=rss&utm_medium=rss&utm_campaign=next-generation-beaglebone

For synth applications the built-in multi-channel ADC and I2S audio 
output protocols put it in a class beyond the RPi, and the new version 
drops the price point into the same neighborhood.

Well worth consideration.

Eric



From gil_we at yahoo.com  Thu Mar 14 21:32:48 2013
From: gil_we at yahoo.com (Gil W.)
Date: Thu, 14 Mar 2013 13:32:48 -0700 (PDT)
Subject: [sdiy] JP6 doesn't write presets. CR2032 over CR1/3N - OK?
Message-ID: <1363293168.8803.YahooMailClassic@web162205.mail.bf1.yahoo.com>

I replaced the dead, stock CR1/3N in the JP6. I then moved every single parameter and saved the preset. Then copied it to all 48 memory patches.

However, when I now try to save a preset, it keeps the previous settings in the saved memory patch instead of writing the new one in.

I wonder if it has anything to do with using a CR2032 instead of the stock CR1/3N battery ? AFAIK the CR2032 has more amps and the same voltage...

any idea ?


From cheater00 at gmail.com  Thu Mar 14 22:16:43 2013
From: cheater00 at gmail.com (cheater cheater)
Date: Thu, 14 Mar 2013 22:16:43 +0100
Subject: [sdiy] is raspberry pi the future?
In-Reply-To: <CAEaByMSNCm47DPL172WVO2tP932A+06qYE+Z99xz9X13o=v25w@mail.gmail.com>
References: <BLU0-SMTP230D6940684AB514B9ADF4BC8EC0@phx.gbl>
	<CAEaByMScr-Zub3f4A_OXwo+jrmqr7zvskQSOt3rW=fweE4u54w@mail.gmail.com>
	<CA+9GZUjnYqWSEb-CtmyujE0SBZrY7y9Bi8026T9NqznqmqVsMg@mail.gmail.com>
	<CAEaByMSNCm47DPL172WVO2tP932A+06qYE+Z99xz9X13o=v25w@mail.gmail.com>
Message-ID: <CA+9GZUhQDRbgypuH2x0Y6eip6zZWkMzumiZmvKdy0Lodkgzu3g@mail.gmail.com>

Hi Neil,

On Thu, Mar 14, 2013 at 5:28 PM, Neil Johnson
<neil.johnson97 at ntlworld.com> wrote:
> Hi,
>
> cheater cheater wrote:
>> The Raspberry Pi has GPIO. Surely, for building a synth you don't want
>> to have the shittiness of USB?
>>
>> http://elinux.org/RPi_BCM2835_GPIOs
>
> A crappy USB interface would likely be better than the on-board audio outs.
>
> Neil

Yes - but then you're basically programming a less powerful desktop
computer, not a more powerful MCU. Pulse Audio and GPIO-connected DAC
are two completely different methodologies and will mean your code
needs to be restructured quite a bit. It's pretty amazing that
nowadays both approaches are viable on the same piece of low-cost
hardware - this hasn't been the case since the 70s.

BTW, like the links in your sig.

On Thu, Mar 14, 2013 at 6:23 PM, Paul Maddox <yo at vacoloco.net> wrote:
> the Raspi has some nice features, but I don't think I'd use it for
> audio generation.
> Perhaps control of a synth, for example, a raspberry-pi with an FPGA,
> a few DACs and some VCFs would make one hell of a polysynth.

That's what I've been saying for some time now. Stop using crappy 40x2
LCDs with pushbuttons and deeply nested menus and all that! That's
really outdated. A raspi plus cheap old LCD monitor cost less than a
good character LCD! You can hook up a mouse and keyboard and have a
rich, awesome user interface to your synthesizer or any other
electronics project.

Eric,
love the beaglebone.

Terry, can't wait to see what you can come up with in your new endeavours.

Cheers,
D.


From cheater00 at gmail.com  Thu Mar 14 22:18:43 2013
From: cheater00 at gmail.com (cheater cheater)
Date: Thu, 14 Mar 2013 22:18:43 +0100
Subject: [sdiy] Contest to win a Tek 465
In-Reply-To: <743C5E15-8C34-4B7C-90DF-9D15176AE4CC@earthlink.net>
References: <CACDGQT_WLByiT61aNG67YNj_3TRaxnMw-nV16X0KDKFTZvLm2A@mail.gmail.com>
	<743C5E15-8C34-4B7C-90DF-9D15176AE4CC@earthlink.net>
Message-ID: <CA+9GZUjo+6-mHOuPFq4_wKOeQZDXADzqhZ15EY4f-PmewwMXrg@mail.gmail.com>

At some point last december I took the time to locate photos of nearly
damn every analog scope tek have made, and one thing's clear: the 465
has the coolest looks of them all. There simply isn't a cooler looking
scope.

D.

On Thu, Mar 14, 2013 at 7:02 PM, Terry Shultz <thx1138 at earthlink.net> wrote:
> Check out VintageTek web site for great deals as well.
>
> I found 465/475 Tek scopes in abundance on Ebay.
>
> Just be careful on burned displays.
>
> I just bought a Tek 468 + Scope cart for $300.00
>
> The 468 is like a 465 + Storage function.
>
>
> Terry Shultz
> thx1138 at earthlink.net
>
>
>
> On Mar 9, 2013, at 10:10 AM, Tim Parkhurst <tim.parkhurst at gmail.com> wrote:
>
>> Cool contest - Register at this site, send in a picture of your messy
>> / interesting workbench, do a few other things, and you're entered in
>> a contest to win a classic Tek 465 scope. :)
>>
>> http://www.scopejunction.com/author.asp?section_id=1835&doc_id=260090&&goback=.gde_2316414_member_221102380
>>
>>
>>
>> Tim (I'm sure we have some VERY 'interesting' workbenches out there) Servo
>> ---
>> "Sire, the church of God is an anvil that has worn out many hammers."
>> - H.L. Hastings
>> _______________________________________________
>> Synth-diy mailing list
>> Synth-diy at dropmix.xs4all.nl
>> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy


From yo at vacoloco.net  Thu Mar 14 22:59:45 2013
From: yo at vacoloco.net (Paul Maddox)
Date: Thu, 14 Mar 2013 21:59:45 +0000
Subject: [sdiy] is raspberry pi the future?
In-Reply-To: <5142215D.8080906@cox.net>
References: <BLU0-SMTP230D6940684AB514B9ADF4BC8EC0@phx.gbl>
	<A61FF618-7614-4D4D-8E4A-32042FDB5815@electricdruid.net>
	<57275E47-9711-4B31-8433-B5251AADF567@me.com>
	<CB78F1D8-554E-4AEA-955E-6D32B9000D6D@blauwurf.info>
	<5142215D.8080906@cox.net>
Message-ID: <CADm3=9RbuRTast613q75OQX8wNhBgdkAhs7dRo9y1PZjmgaSVQ@mail.gmail.com>

I've got a beaglebone, great little box.
Tonnes of IO and as I'm not likely to use a monitor I love that it
comes without HDMI :)

On 14 March 2013 19:13, Eric Brombaugh <ebrombaugh1 at cox.net> wrote:
> RPi is a significant step forward for low-cost / high-power embedded
> computing and I'm glad to see it becoming so popular. That said however it's
> not the last word in this space - TI has been providing embedded boards with
> more I/O and more modern processors for several years and isn't resting on
> their laurels.
>
> The current low-cost / high I/O winner is the Beaglebone, which provides
> more than 80 pins of digital and analog interfacing, including high-speed
> SPI, I2C, I2S, LCD, parallel external memory and plain GPIO for less than
> $90. A low-cost Beaglebone with similar capabilities is right around the
> corner and is expected to cost about half that. See more about it here:
>
> http://www.citizensinspace.org/2013/03/next-generation-beaglebone/?utm_source=rss&utm_medium=rss&utm_campaign=next-generation-beaglebone
>
> For synth applications the built-in multi-channel ADC and I2S audio output
> protocols put it in a class beyond the RPi, and the new version drops the
> price point into the same neighborhood.
>
> Well worth consideration.
>
> Eric
>
>
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy


From thx1138 at earthlink.net  Thu Mar 14 23:13:45 2013
From: thx1138 at earthlink.net (Terry Shultz)
Date: Thu, 14 Mar 2013 15:13:45 -0700
Subject: [sdiy] is raspberry pi the future?
In-Reply-To: <5142215D.8080906@cox.net>
References: <BLU0-SMTP230D6940684AB514B9ADF4BC8EC0@phx.gbl>
	<A61FF618-7614-4D4D-8E4A-32042FDB5815@electricdruid.net>
	<57275E47-9711-4B31-8433-B5251AADF567@me.com>
	<CB78F1D8-554E-4AEA-955E-6D32B9000D6D@blauwurf.info>
	<5142215D.8080906@cox.net>
Message-ID: <11193470-EC8B-406D-A1C6-4673E61C0F61@earthlink.net>

Hi All,

Eric may e correct on I/O on the Ti based solution. I am speaking mostly from a Freescale centric application point.

There again, it depends on how these pins are brought out. They are usually heavily multiplexed as most ARM devices are.

If An ARM Audio centric board is what is required. putting enough A/D-D/A channels and some GPIO pins, UARTS, USB, etc. is usually what makes these small footprints so non-desireable to most of my applications.

I work a great deal in Multi-channel Audio and 3-D Audio applications and need a bit more meat than a RPI.

Perhaps I should do a quick analysis of a TI , Freescale, Marvel based Low cost board to see what features are similar/different.

best regards,
Terry Shultz
thx1138 at earthlink.net



On Mar 14, 2013, at 12:13 PM, Eric Brombaugh <ebrombaugh1 at cox.net> wrote:

> RPi is a significant step forward for low-cost / high-power embedded computing and I'm glad to see it becoming so popular. That said however it's not the last word in this space - TI has been providing embedded boards with more I/O and more modern processors for several years and isn't resting on their laurels.
> 
> The current low-cost / high I/O winner is the Beaglebone, which provides more than 80 pins of digital and analog interfacing, including high-speed SPI, I2C, I2S, LCD, parallel external memory and plain GPIO for less than $90. A low-cost Beaglebone with similar capabilities is right around the corner and is expected to cost about half that. See more about it here:
> 
> http://www.citizensinspace.org/2013/03/next-generation-beaglebone/?utm_source=rss&utm_medium=rss&utm_campaign=next-generation-beaglebone
> 
> For synth applications the built-in multi-channel ADC and I2S audio output protocols put it in a class beyond the RPi, and the new version drops the price point into the same neighborhood.
> 
> Well worth consideration.
> 
> Eric
> 
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy



From neil.johnson97 at ntlworld.com  Thu Mar 14 23:22:16 2013
From: neil.johnson97 at ntlworld.com (Neil Johnson)
Date: Thu, 14 Mar 2013 22:22:16 +0000
Subject: [sdiy] is raspberry pi the future?
In-Reply-To: <CA+9GZUhQDRbgypuH2x0Y6eip6zZWkMzumiZmvKdy0Lodkgzu3g@mail.gmail.com>
References: <BLU0-SMTP230D6940684AB514B9ADF4BC8EC0@phx.gbl>
	<CAEaByMScr-Zub3f4A_OXwo+jrmqr7zvskQSOt3rW=fweE4u54w@mail.gmail.com>
	<CA+9GZUjnYqWSEb-CtmyujE0SBZrY7y9Bi8026T9NqznqmqVsMg@mail.gmail.com>
	<CAEaByMSNCm47DPL172WVO2tP932A+06qYE+Z99xz9X13o=v25w@mail.gmail.com>
	<CA+9GZUhQDRbgypuH2x0Y6eip6zZWkMzumiZmvKdy0Lodkgzu3g@mail.gmail.com>
Message-ID: <51424D98.10807@ntlworld.com>

cheater cheater wrote:
> Yes - but then you're basically programming a less powerful desktop
> computer, not a more powerful MCU. Pulse Audio and GPIO-connected DAC
> are two completely different methodologies and will mean your code
> needs to be restructured quite a bit. It's pretty amazing that
> nowadays both approaches are viable on the same piece of low-cost
> hardware - this hasn't been the case since the 70s.

GPIO is a bit limited on the RasPi.  If you want decent audio out of the 
2835 directly (rather than via USB) then a better option would be to 
hook up an audio codec to the I2S lines and enable the driver -- I think 
this is available for the RasPi, but you'd need to check with the 
Foundation.

I believe I can say that I have quite a good grasp of what the 2835 is 
capable of.

Cheers,
Neil
-- 
Modules and more :: www.cesyg.com
Homepage :: www.njohnson.co.uk ::


From moonmuseum at hotmail.com  Fri Mar 15 04:40:29 2013
From: moonmuseum at hotmail.com (3 vcos)
Date: Thu, 14 Mar 2013 20:40:29 -0700
Subject: [sdiy] JP6 doesn't write presets. CR2032 over CR1/3N - OK?
In-Reply-To: <1363293168.8803.YahooMailClassic@web162205.mail.bf1.yahoo.com>
References: <1363293168.8803.YahooMailClassic@web162205.mail.bf1.yahoo.com>
Message-ID: <BAY401-EAS84FAAFFBF843E252744EA8DAED0@phx.gbl>

Funny I had the same problem recently. I actually replaced my battery with a new cr1/3n. Afterwards I wasn't able to write presets. 
I still haven't figured this out. 

On Mar 14, 2013, at 1:35 PM, "Gil W." <gil_we at yahoo.com> wrote:

> I replaced the dead, stock CR1/3N in the JP6. I then moved every single parameter and saved the preset. Then copied it to all 48 memory patches.
> 
> However, when I now try to save a preset, it keeps the previous settings in the saved memory patch instead of writing the new one in.
> 
> I wonder if it has anything to do with using a CR2032 instead of the stock CR1/3N battery ? AFAIK the CR2032 has more amps and the same voltage...
> 
> any idea ?
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy


From hugh at blemings.org  Fri Mar 15 05:20:40 2013
From: hugh at blemings.org (Hugh Blemings)
Date: Fri, 15 Mar 2013 15:20:40 +1100
Subject: [sdiy] JP6 doesn't write presets. CR2032 over CR1/3N - OK?
In-Reply-To: <BAY401-EAS84FAAFFBF843E252744EA8DAED0@phx.gbl>
References: <1363293168.8803.YahooMailClassic@web162205.mail.bf1.yahoo.com>
	<BAY401-EAS84FAAFFBF843E252744EA8DAED0@phx.gbl>
Message-ID: <5142A198.6030201@blemings.org>

Hiya,

On 15/03/13 14:40, 3 vcos wrote:
> Funny I had the same problem recently. I actually replaced my battery
> with a new cr1/3n. Afterwards I wasn't able to write presets. I still
> haven't figured this out.

It's a looong time (sadly) since I played with a JP6, but I've a vague 
recollection that you had to do an implicit factory reset after 
replacing the battery.

Perhaps this step also ensured the RAM was written to a known state with 
any required metadata/checksums in place before a user patch write would 
work ?

Maybe you folk have already tried this, apologies if so!

Cheers,
Hugh




From rsdio at sounds.wa.com  Fri Mar 15 07:09:50 2013
From: rsdio at sounds.wa.com (rsdio at sounds.wa.com)
Date: Thu, 14 Mar 2013 23:09:50 -0700
Subject: [sdiy] is raspberry pi the future?
In-Reply-To: <11193470-EC8B-406D-A1C6-4673E61C0F61@earthlink.net>
References: <BLU0-SMTP230D6940684AB514B9ADF4BC8EC0@phx.gbl>
	<A61FF618-7614-4D4D-8E4A-32042FDB5815@electricdruid.net>
	<57275E47-9711-4B31-8433-B5251AADF567@me.com>
	<CB78F1D8-554E-4AEA-955E-6D32B9000D6D@blauwurf.info>
	<5142215D.8080906@cox.net>
	<11193470-EC8B-406D-A1C6-4673E61C0F61@earthlink.net>
Message-ID: <DEBC9E0E-B4F2-4C29-AF0C-4944851D7F3D@sounds.wa.com>

I'd say that an ARM board like Raspberry Pi is not the best choice  
for the future of Synth DIY. Personally, I would prefer a single  
board with a powerful DSP and on-board conversion. Good choices for  
DSP would be TMS320 or SHARC. On-board audio would be I2S, with  
available I/O headers for additional channels.

One exciting chip family, I think, is Texas Instruments' OMAP, which  
combines ARM + DSP.

In other words, there's no sense buying a cheap platform that doesn't  
have audio and tacking on a second board, when it seems easy enough  
to just make a single board with everything. I have noticed that  
Texas Instruments has several evaluation boards for DSP at the $50  
and below price point, where stereo input and output are already  
available, along with a USB port for development and programming of  
firmware.

Brian Willoughby
Sound Consulting



From cheater00 at gmail.com  Fri Mar 15 09:19:43 2013
From: cheater00 at gmail.com (cheater cheater)
Date: Fri, 15 Mar 2013 09:19:43 +0100
Subject: [sdiy] is raspberry pi the future?
In-Reply-To: <51424D98.10807@ntlworld.com>
References: <BLU0-SMTP230D6940684AB514B9ADF4BC8EC0@phx.gbl>
	<CAEaByMScr-Zub3f4A_OXwo+jrmqr7zvskQSOt3rW=fweE4u54w@mail.gmail.com>
	<CA+9GZUjnYqWSEb-CtmyujE0SBZrY7y9Bi8026T9NqznqmqVsMg@mail.gmail.com>
	<CAEaByMSNCm47DPL172WVO2tP932A+06qYE+Z99xz9X13o=v25w@mail.gmail.com>
	<CA+9GZUhQDRbgypuH2x0Y6eip6zZWkMzumiZmvKdy0Lodkgzu3g@mail.gmail.com>
	<51424D98.10807@ntlworld.com>
Message-ID: <CA+9GZUhUqwGqJ2FdLgYT0cY_DQuK4LLDS0Z5Kvat8-d7fpw=iA@mail.gmail.com>

On Thu, Mar 14, 2013 at 11:22 PM, Neil Johnson
<neil.johnson97 at ntlworld.com> wrote:
> GPIO is a bit limited on the RasPi.  If you want decent audio out of the
> 2835 directly (rather than via USB) then a better option would be to hook up
> an audio codec to the I2S lines and enable the driver -- I think this is
> available for the RasPi, but you'd need to check with the Foundation.

That's a great idea. I wonder if i2s has a standard linux driver of some sort.

> I believe I can say that I have quite a good grasp of what the 2835 is
> capable of.

Yes


From gil_we at yahoo.com  Fri Mar 15 09:55:02 2013
From: gil_we at yahoo.com (Gil W.)
Date: Fri, 15 Mar 2013 01:55:02 -0700 (PDT)
Subject: [sdiy] JP6 doesn't write presets. CR2032 over CR1/3N - OK?
In-Reply-To: <5142A198.6030201@blemings.org>
Message-ID: <1363337702.81502.YahooMailClassic@web162203.mail.bf1.yahoo.com>

Thanks all!

After investigating a bit further, I found it was actually my mistake ! 

I used the preset memory banks to "save" sounds... The patch switch needs to be off in order to save to the 48 sound memory locations.

Works fine now ! :)




--- On Fri, 3/15/13, Hugh Blemings <hugh at blemings.org> wrote:

> From: Hugh Blemings <hugh at blemings.org>
> Subject: Re: [sdiy] JP6 doesn't write presets. CR2032 over CR1/3N - OK?
> To: "3 vcos" <moonmuseum at hotmail.com>
> Cc: "Gil W." <gil_we at yahoo.com>, "synth-diy at dropmix.xs4all.nl" <synth-diy at dropmix.xs4all.nl>
> Date: Friday, March 15, 2013, 6:20 AM
> Hiya,
> 
> On 15/03/13 14:40, 3 vcos wrote:
> > Funny I had the same problem recently. I actually
> replaced my battery
> > with a new cr1/3n. Afterwards I wasn't able to write
> presets. I still
> > haven't figured this out.
> 
> It's a looong time (sadly) since I played with a JP6, but
> I've a vague recollection that you had to do an implicit
> factory reset after replacing the battery.
> 
> Perhaps this step also ensured the RAM was written to a
> known state with any required metadata/checksums in place
> before a user patch write would work ?
> 
> Maybe you folk have already tried this, apologies if so!
> 
> Cheers,
> Hugh
> 
> 
> 


From gil_we at yahoo.com  Fri Mar 15 10:01:50 2013
From: gil_we at yahoo.com (Gil W.)
Date: Fri, 15 Mar 2013 02:01:50 -0700 (PDT)
Subject: [sdiy] JP6 doesn't write presets. CR2032 over CR1/3N - OK?
In-Reply-To: <5142A198.6030201@blemings.org>
Message-ID: <1363338110.69493.YahooMailClassic@web162202.mail.bf1.yahoo.com>

Just for the record , the correct "reset" procedure for dead battery scenario would be:

1. Replace battery
2. power on and switch of the Patch switch
3. Move any switch on/off any tweakable sound parameter
4. Once done, save the sound to location A-1 and than copy it from there to all 48 location.

I haven't reset the patch banks yet but I assume that the same way you will have to turn on/off any switch in the Patch section (the bottom line of switches), save and copy to all 32 Patch locations...




--- On Fri, 3/15/13, Hugh Blemings <hugh at blemings.org> wrote:

> From: Hugh Blemings <hugh at blemings.org>
> Subject: Re: [sdiy] JP6 doesn't write presets. CR2032 over CR1/3N - OK?
> To: "3 vcos" <moonmuseum at hotmail.com>
> Cc: "Gil W." <gil_we at yahoo.com>, "synth-diy at dropmix.xs4all.nl" <synth-diy at dropmix.xs4all.nl>
> Date: Friday, March 15, 2013, 6:20 AM
> Hiya,
> 
> On 15/03/13 14:40, 3 vcos wrote:
> > Funny I had the same problem recently. I actually
> replaced my battery
> > with a new cr1/3n. Afterwards I wasn't able to write
> presets. I still
> > haven't figured this out.
> 
> It's a looong time (sadly) since I played with a JP6, but
> I've a vague recollection that you had to do an implicit
> factory reset after replacing the battery.
> 
> Perhaps this step also ensured the RAM was written to a
> known state with any required metadata/checksums in place
> before a user patch write would work ?
> 
> Maybe you folk have already tried this, apologies if so!
> 
> Cheers,
> Hugh
> 
> 
> 


From dougt55 at yahoo.com  Fri Mar 15 11:25:56 2013
From: dougt55 at yahoo.com (Doug Terrebonne)
Date: Fri, 15 Mar 2013 03:25:56 -0700 (PDT)
Subject: [sdiy] JP6 doesn't write presets. CR2032 over CR1/3N - OK?
In-Reply-To: <1363338110.69493.YahooMailClassic@web162202.mail.bf1.yahoo.com>
Message-ID: <1363343156.12197.YahooMailClassic@web181605.mail.ne1.yahoo.com>

Acually the correct procedure in this specific case is to install the Europa upgrade which stores patches in flash memory and supports sysex dumps. :)

Without Europa you can deal with loading the patches after a batt swap from tape/wave file with various degrees of succces. The JP-6 without Europa can also lock up and exhibit other weird behaviour if invalid corrupt patch data is in memory...

Doug 
synthparts.com

--- On Fri, 3/15/13, Gil W. <gil_we at yahoo.com> wrote:

> Just for the record , the correct
> "reset" procedure for dead battery scenario would be:
> 
> 1. Replace battery
> 2. power on and switch of the Patch switch
> 3. Move any switch on/off any tweakable sound parameter
> 4. Once done, save the sound to location A-1 and than copy
> it from there to all 48 location.
> 
> I haven't reset the patch banks yet but I assume that the
> same way you will have to turn on/off any switch in the
> Patch section (the bottom line of switches), save and copy
> to all 32 Patch locations...
> 
> 
> 
> 
> --- On Fri, 3/15/13, Hugh Blemings <hugh at blemings.org>
> wrote:
> 
> > From: Hugh Blemings <hugh at blemings.org>
> > Subject: Re: [sdiy] JP6 doesn't write presets. CR2032
> over CR1/3N - OK?
> > To: "3 vcos" <moonmuseum at hotmail.com>
> > Cc: "Gil W." <gil_we at yahoo.com>,
> "synth-diy at dropmix.xs4all.nl"
> <synth-diy at dropmix.xs4all.nl>
> > Date: Friday, March 15, 2013, 6:20 AM
> > Hiya,
> > 
> > On 15/03/13 14:40, 3 vcos wrote:
> > > Funny I had the same problem recently. I actually
> > replaced my battery
> > > with a new cr1/3n. Afterwards I wasn't able to
> write
> > presets. I still
> > > haven't figured this out.
> > 
> > It's a looong time (sadly) since I played with a JP6,
> but
> > I've a vague recollection that you had to do an
> implicit
> > factory reset after replacing the battery.
> > 
> > Perhaps this step also ensured the RAM was written to
> a
> > known state with any required metadata/checksums in
> place
> > before a user patch write would work ?
> > 
> > Maybe you folk have already tried this, apologies if
> so!
> > 
> > Cheers,
> > Hugh
> > 
> > 
> > 
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
> 


From marc.nostromo at gmail.com  Fri Mar 15 12:11:46 2013
From: marc.nostromo at gmail.com (Marc Nostromo [M-.-n])
Date: Fri, 15 Mar 2013 12:11:46 +0100
Subject: [sdiy] is raspberry pi the future?
In-Reply-To: <CA+9GZUhUqwGqJ2FdLgYT0cY_DQuK4LLDS0Z5Kvat8-d7fpw=iA@mail.gmail.com>
References: <BLU0-SMTP230D6940684AB514B9ADF4BC8EC0@phx.gbl>
	<CAEaByMScr-Zub3f4A_OXwo+jrmqr7zvskQSOt3rW=fweE4u54w@mail.gmail.com>
	<CA+9GZUjnYqWSEb-CtmyujE0SBZrY7y9Bi8026T9NqznqmqVsMg@mail.gmail.com>
	<CAEaByMSNCm47DPL172WVO2tP932A+06qYE+Z99xz9X13o=v25w@mail.gmail.com>
	<CA+9GZUhQDRbgypuH2x0Y6eip6zZWkMzumiZmvKdy0Lodkgzu3g@mail.gmail.com>
	<51424D98.10807@ntlworld.com>
	<CA+9GZUhUqwGqJ2FdLgYT0cY_DQuK4LLDS0Z5Kvat8-d7fpw=iA@mail.gmail.com>
Message-ID: <CAEQ8eXMBJT5kqOD63531wb0O_-hoZsUaQeVngFLM33fA+CWv+g@mail.gmail.com>

I don't know about using the RPi as base for proper synth building.
However as a platform for building inexpensive digital synth I love
it. Hook a cheap USB audio out and midi in and you are ready to go.

I've successfully ported my tracker to it

https://fbcdn-sphotos-a-a.akamaihd.net/hphotos-ak-ash4/341346_10151102704988284_1068931733_o.jpg

And it runs flawlessly.

I've also tried some onboard headless synthesis and it works quite
well as long as you don't overload it too much with heavy algorithms:

http://www.marc-nostromo.com/?p=158

Right now for diy synth I'm more into using the teensy, and for
dsp-based or sequencing the RPi.

Cheers,
/M

2013/3/15 cheater cheater <cheater00 at gmail.com>:
> On Thu, Mar 14, 2013 at 11:22 PM, Neil Johnson
> <neil.johnson97 at ntlworld.com> wrote:
>> GPIO is a bit limited on the RasPi.  If you want decent audio out of the
>> 2835 directly (rather than via USB) then a better option would be to hook up
>> an audio codec to the I2S lines and enable the driver -- I think this is
>> available for the RasPi, but you'd need to check with the Foundation.
>
> That's a great idea. I wonder if i2s has a standard linux driver of some sort.
>
>> I believe I can say that I have quite a good grasp of what the 2835 is
>> capable of.
>
> Yes
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy



--
http://marc-nostromo.com


From subjectivity at hotmail.com  Fri Mar 15 13:46:50 2013
From: subjectivity at hotmail.com (Dan Snazelle)
Date: Fri, 15 Mar 2013 08:46:50 -0400
Subject: [sdiy] is raspberry pi the future?
In-Reply-To: <11193470-EC8B-406D-A1C6-4673E61C0F61@earthlink.net>
References: <BLU0-SMTP230D6940684AB514B9ADF4BC8EC0@phx.gbl>
	<A61FF618-7614-4D4D-8E4A-32042FDB5815@electricdruid.net>
	<57275E47-9711-4B31-8433-B5251AADF567@me.com>
	<CB78F1D8-554E-4AEA-955E-6D32B9000D6D@blauwurf.info>
	<5142215D.8080906@cox.net>
	<11193470-EC8B-406D-A1C6-4673E61C0F61@earthlink.net>
Message-ID: <BLU0-SMTP105A12270CDFDEC74084E41C8ED0@phx.gbl>

WOW
great thread!!
One thing 
I have learned since yesterday....the rasp pi is not truly open source..i guess some elements are closed.


This is a deal breaker.


I need a platform which if i end up making something cool, i can transfer it 100% (software and hardware) to a pcb.


So....looks like i need to look into those TI soloutions!!! 

Thanks!

Sent from my iPhone

On Mar 14, 2013, at 6:13 PM, Terry Shultz <thx1138 at earthlink.net> wrote:

> Hi All,
> 
> Eric may e correct on I/O on the Ti based solution. I am speaking mostly from a Freescale centric application point.
> 
> There again, it depends on how these pins are brought out. They are usually heavily multiplexed as most ARM devices are.
> 
> If An ARM Audio centric board is what is required. putting enough A/D-D/A channels and some GPIO pins, UARTS, USB, etc. is usually what makes these small footprints so non-desireable to most of my applications.
> 
> I work a great deal in Multi-channel Audio and 3-D Audio applications and need a bit more meat than a RPI.
> 
> Perhaps I should do a quick analysis of a TI , Freescale, Marvel based Low cost board to see what features are similar/different.
> 
> best regards,
> Terry Shultz
> thx1138 at earthlink.net
> 
> 
> 
> On Mar 14, 2013, at 12:13 PM, Eric Brombaugh <ebrombaugh1 at cox.net> wrote:
> 
>> RPi is a significant step forward for low-cost / high-power embedded computing and I'm glad to see it becoming so popular. That said however it's not the last word in this space - TI has been providing embedded boards with more I/O and more modern processors for several years and isn't resting on their laurels.
>> 
>> The current low-cost / high I/O winner is the Beaglebone, which provides more than 80 pins of digital and analog interfacing, including high-speed SPI, I2C, I2S, LCD, parallel external memory and plain GPIO for less than $90. A low-cost Beaglebone with similar capabilities is right around the corner and is expected to cost about half that. See more about it here:
>> 
>> http://www.citizensinspace.org/2013/03/next-generation-beaglebone/?utm_source=rss&utm_medium=rss&utm_campaign=next-generation-beaglebone
>> 
>> For synth applications the built-in multi-channel ADC and I2S audio output protocols put it in a class beyond the RPi, and the new version drops the price point into the same neighborhood.
>> 
>> Well worth consideration.
>> 
>> Eric
>> 
>> _______________________________________________
>> Synth-diy mailing list
>> Synth-diy at dropmix.xs4all.nl
>> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
> 
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
> 


From cheater00 at gmail.com  Fri Mar 15 14:02:53 2013
From: cheater00 at gmail.com (cheater cheater)
Date: Fri, 15 Mar 2013 14:02:53 +0100
Subject: [sdiy] is raspberry pi the future?
In-Reply-To: <CAEQ8eXMBJT5kqOD63531wb0O_-hoZsUaQeVngFLM33fA+CWv+g@mail.gmail.com>
References: <BLU0-SMTP230D6940684AB514B9ADF4BC8EC0@phx.gbl>
	<CAEaByMScr-Zub3f4A_OXwo+jrmqr7zvskQSOt3rW=fweE4u54w@mail.gmail.com>
	<CA+9GZUjnYqWSEb-CtmyujE0SBZrY7y9Bi8026T9NqznqmqVsMg@mail.gmail.com>
	<CAEaByMSNCm47DPL172WVO2tP932A+06qYE+Z99xz9X13o=v25w@mail.gmail.com>
	<CA+9GZUhQDRbgypuH2x0Y6eip6zZWkMzumiZmvKdy0Lodkgzu3g@mail.gmail.com>
	<51424D98.10807@ntlworld.com>
	<CA+9GZUhUqwGqJ2FdLgYT0cY_DQuK4LLDS0Z5Kvat8-d7fpw=iA@mail.gmail.com>
	<CAEQ8eXMBJT5kqOD63531wb0O_-hoZsUaQeVngFLM33fA+CWv+g@mail.gmail.com>
Message-ID: <CA+9GZUgMmm96yjCYimTYe9vYtGXt62oBrJTpGkmVf5799kFiWw@mail.gmail.com>

On Fri, Mar 15, 2013 at 12:11 PM, Marc Nostromo [M-.-n]
<marc.nostromo at gmail.com> wrote:
> I don't know about using the RPi as base for proper synth building.
> However as a platform for building inexpensive digital synth I love
> it. Hook a cheap USB audio out and midi in and you are ready to go.
>
> I've successfully ported my tracker to it

Is porting something like a tracker to the Raspberry Pi when using its
onboard linux different than porting it to any other linux-based
system? Here I'm assuming you haven't developed your own kernel with
usb audio support.

I see a lot of people go and say they did this or that "on a raspberry
pi" but they're not using any special features it provides over any
other pc; it's kind of stupid, because if you look at all these
headlines...

Raspberry Pi as a Spotify server with MPD control
Using OpenCV with the Raspberry Pi
Building a touchscreen XBMC setup with the Raspberry Pi
Multibooting the Raspberry Pi
Playing with the Minecraft API and a Raspberry Pi

...and make them say what really is going on...

PC as a Spotify server with MPD control
Using OpenCV with a PC
Building a touchscreen XBMC setup with a PC
Multibooting a PC
Playing with the Minecraft API and a PC

...then it's easy to see how ridiculous hanging everything onto the
Raspberry Pi has become.

Those were from the front page of the search for "raspberry" on
hackaday. What multibooting, spotify, and xbmc have to do with
hackaday I don't know. It's all kind of sad.

D.


> https://fbcdn-sphotos-a-a.akamaihd.net/hphotos-ak-ash4/341346_10151102704988284_1068931733_o.jpg
>
> And it runs flawlessly.
>
> I've also tried some onboard headless synthesis and it works quite
> well as long as you don't overload it too much with heavy algorithms:
>
> http://www.marc-nostromo.com/?p=158
>
> Right now for diy synth I'm more into using the teensy, and for
> dsp-based or sequencing the RPi.
>
> Cheers,
> /M
>
> 2013/3/15 cheater cheater <cheater00 at gmail.com>:
>> On Thu, Mar 14, 2013 at 11:22 PM, Neil Johnson
>> <neil.johnson97 at ntlworld.com> wrote:
>>> GPIO is a bit limited on the RasPi.  If you want decent audio out of the
>>> 2835 directly (rather than via USB) then a better option would be to hook up
>>> an audio codec to the I2S lines and enable the driver -- I think this is
>>> available for the RasPi, but you'd need to check with the Foundation.
>>
>> That's a great idea. I wonder if i2s has a standard linux driver of some sort.
>>
>>> I believe I can say that I have quite a good grasp of what the 2835 is
>>> capable of.
>>
>> Yes
>> _______________________________________________
>> Synth-diy mailing list
>> Synth-diy at dropmix.xs4all.nl
>> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>
>
>
> --
> http://marc-nostromo.com


From gil_we at yahoo.com  Fri Mar 15 14:50:14 2013
From: gil_we at yahoo.com (Gil W.)
Date: Fri, 15 Mar 2013 06:50:14 -0700 (PDT)
Subject: [sdiy] [AH] Re: JP6 doesn't write presets. CR2032 over CR1/3N -
	OK?
In-Reply-To: <1363343156.12197.YahooMailClassic@web181605.mail.ne1.yahoo.com>
Message-ID: <1363355414.78527.YahooMailClassic@web162202.mail.bf1.yahoo.com>

Thanks Doug,

I will take this into account ;)





--- On Fri, 3/15/13, Doug Terrebonne <dougt55 at yahoo.com> wrote:

> From: Doug Terrebonne <dougt55 at yahoo.com>
> Subject: [AH] Re: [sdiy] JP6 doesn't write presets. CR2032 over CR1/3N - OK?
> To: "AH" <analogue at hyperreal.org>
> Cc: "synth-diy at dropmix.xs4all.nl" <synth-diy at dropmix.xs4all.nl>
> Date: Friday, March 15, 2013, 12:25 PM
> Acually the correct procedure in this
> specific case is to install the Europa upgrade which stores
> patches in flash memory and supports sysex dumps. :)
> 
> Without Europa you can deal with loading the patches after a
> batt swap from tape/wave file with various degrees of
> succces. The JP-6 without Europa can also lock up and
> exhibit other weird behaviour if invalid corrupt patch data
> is in memory...
> 
> Doug 
> synthparts.com
> 
> --- On Fri, 3/15/13, Gil W. <gil_we at yahoo.com>
> wrote:
> 
> > Just for the record , the correct
> > "reset" procedure for dead battery scenario would be:
> > 
> > 1. Replace battery
> > 2. power on and switch of the Patch switch
> > 3. Move any switch on/off any tweakable sound
> parameter
> > 4. Once done, save the sound to location A-1 and than
> copy
> > it from there to all 48 location.
> > 
> > I haven't reset the patch banks yet but I assume that
> the
> > same way you will have to turn on/off any switch in
> the
> > Patch section (the bottom line of switches), save and
> copy
> > to all 32 Patch locations...
> > 
> > 
> > 
> > 
> > --- On Fri, 3/15/13, Hugh Blemings <hugh at blemings.org>
> > wrote:
> > 
> > > From: Hugh Blemings <hugh at blemings.org>
> > > Subject: Re: [sdiy] JP6 doesn't write presets.
> CR2032
> > over CR1/3N - OK?
> > > To: "3 vcos" <moonmuseum at hotmail.com>
> > > Cc: "Gil W." <gil_we at yahoo.com>,
> > "synth-diy at dropmix.xs4all.nl"
> > <synth-diy at dropmix.xs4all.nl>
> > > Date: Friday, March 15, 2013, 6:20 AM
> > > Hiya,
> > > 
> > > On 15/03/13 14:40, 3 vcos wrote:
> > > > Funny I had the same problem recently. I
> actually
> > > replaced my battery
> > > > with a new cr1/3n. Afterwards I wasn't able
> to
> > write
> > > presets. I still
> > > > haven't figured this out.
> > > 
> > > It's a looong time (sadly) since I played with a
> JP6,
> > but
> > > I've a vague recollection that you had to do an
> > implicit
> > > factory reset after replacing the battery.
> > > 
> > > Perhaps this step also ensured the RAM was written
> to
> > a
> > > known state with any required metadata/checksums
> in
> > place
> > > before a user patch write would work ?
> > > 
> > > Maybe you folk have already tried this, apologies
> if
> > so!
> > > 
> > > Cheers,
> > > Hugh
> > > 
> > > 
> > > 
> > _______________________________________________
> > Synth-diy mailing list
> > Synth-diy at dropmix.xs4all.nl
> > http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
> > 
> 


From marc.nostromo at gmail.com  Fri Mar 15 15:21:20 2013
From: marc.nostromo at gmail.com (Marc Nostromo [M-.-n])
Date: Fri, 15 Mar 2013 15:21:20 +0100
Subject: [sdiy] is raspberry pi the future?
In-Reply-To: <CA+9GZUgMmm96yjCYimTYe9vYtGXt62oBrJTpGkmVf5799kFiWw@mail.gmail.com>
References: <BLU0-SMTP230D6940684AB514B9ADF4BC8EC0@phx.gbl>
	<CAEaByMScr-Zub3f4A_OXwo+jrmqr7zvskQSOt3rW=fweE4u54w@mail.gmail.com>
	<CA+9GZUjnYqWSEb-CtmyujE0SBZrY7y9Bi8026T9NqznqmqVsMg@mail.gmail.com>
	<CAEaByMSNCm47DPL172WVO2tP932A+06qYE+Z99xz9X13o=v25w@mail.gmail.com>
	<CA+9GZUhQDRbgypuH2x0Y6eip6zZWkMzumiZmvKdy0Lodkgzu3g@mail.gmail.com>
	<51424D98.10807@ntlworld.com>
	<CA+9GZUhUqwGqJ2FdLgYT0cY_DQuK4LLDS0Z5Kvat8-d7fpw=iA@mail.gmail.com>
	<CAEQ8eXMBJT5kqOD63531wb0O_-hoZsUaQeVngFLM33fA+CWv+g@mail.gmail.com>
	<CA+9GZUgMmm96yjCYimTYe9vYtGXt62oBrJTpGkmVf5799kFiWw@mail.gmail.com>
Message-ID: <CAEQ8eXNfiRv=+Rk0kpLYGMO31wVnVbaSchgkW1-XFr0Q1cn2Kg@mail.gmail.com>

>> I don't know about using the RPi as base for proper synth building.
>> However as a platform for building inexpensive digital synth I love
>> it. Hook a cheap USB audio out and midi in and you are ready to go.
>>
>> I've successfully ported my tracker to it
>
> Is porting something like a tracker to the Raspberry Pi when using its
> onboard linux different than porting it to any other linux-based
> system? Here I'm assuming you haven't developed your own kernel with
> usb audio support.

No I didn't but that's where I think lies it's power. Basically all I
had to do is to grab the source code and target Debian.

I am not at all implying it is a coup de force on my part.However, the
fact you can have some cheap yet valid digital processor - on top of
that easily embeddable because of the form factor - make it all
worthwhile for me. It's something you can, once you developed code for
it and interfaced properly, simply plug and play without having to
cope with the fact it's a pc or not.

I'm not trying to say it's rocket science. It just makes sense for me.

/M


From tom at electricdruid.net  Fri Mar 15 15:54:44 2013
From: tom at electricdruid.net (Tom Wiltshire)
Date: Fri, 15 Mar 2013 14:54:44 +0000
Subject: [sdiy] is raspberry pi the future?
In-Reply-To: <BLU0-SMTP105A12270CDFDEC74084E41C8ED0@phx.gbl>
References: <BLU0-SMTP230D6940684AB514B9ADF4BC8EC0@phx.gbl>
	<A61FF618-7614-4D4D-8E4A-32042FDB5815@electricdruid.net>
	<57275E47-9711-4B31-8433-B5251AADF567@me.com>
	<CB78F1D8-554E-4AEA-955E-6D32B9000D6D@blauwurf.info>
	<5142215D.8080906@cox.net>
	<11193470-EC8B-406D-A1C6-4673E61C0F61@earthlink.net>
	<BLU0-SMTP105A12270CDFDEC74084E41C8ED0@phx.gbl>
Message-ID: <C4531484-CF4D-4798-95D3-DE05D8B85285@electricdruid.net>


On 15 Mar 2013, at 12:46, Dan Snazelle <subjectivity at hotmail.com> wrote:

> One thing 
> I have learned since yesterday....the rasp pi is not truly open source..i guess some elements are closed.

Where did you see this, Dan? And which bits are closed source?

That seems to totally undermine the whole point of it, in my view.

T.



From neil.johnson97 at ntlworld.com  Fri Mar 15 15:57:23 2013
From: neil.johnson97 at ntlworld.com (Neil Johnson)
Date: Fri, 15 Mar 2013 14:57:23 +0000
Subject: [sdiy] is raspberry pi the future?
In-Reply-To: <BLU0-SMTP105A12270CDFDEC74084E41C8ED0@phx.gbl>
References: <BLU0-SMTP230D6940684AB514B9ADF4BC8EC0@phx.gbl>
	<A61FF618-7614-4D4D-8E4A-32042FDB5815@electricdruid.net>
	<57275E47-9711-4B31-8433-B5251AADF567@me.com>
	<CB78F1D8-554E-4AEA-955E-6D32B9000D6D@blauwurf.info>
	<5142215D.8080906@cox.net>
	<11193470-EC8B-406D-A1C6-4673E61C0F61@earthlink.net>
	<BLU0-SMTP105A12270CDFDEC74084E41C8ED0@phx.gbl>
Message-ID: <CAEaByMSk_sL3or4s4U0rVyhMO3jAev3Xz++mFxoWR8NVYDhMKQ@mail.gmail.com>

Hi Dan,

> One thing
> I have learned since yesterday....the rasp pi is not truly open source..i guess some elements are closed.
>
> This is a deal breaker.
>
> I need a platform which if i end up making something cool, i can transfer it 100% (software and hardware) to a pcb.

Your bigger problems will be buying the BCM2835 and the required PCB technology:
- Broadcom does not sell the BCM2835 through distributors, so you'll
have to buy them in LARGE quantities direct;
- 6- to 8-layer board and BGA footprints - you'll need a good PCB
assembly house to build it for you.

The closed-source bits are for the VideoCore, not the ARM - make your
board sufficiently the same as RasPi (with your extra stuff that makes
it your product) and it should run the standard software from the
Foundation.  But see the two points above before you got too excited.

Cheers,
Neil
-- 
Modules and more: http://www.cesyg.com
Homepage: http://www.njohnson.co.uk


From neil.johnson97 at ntlworld.com  Fri Mar 15 15:59:49 2013
From: neil.johnson97 at ntlworld.com (Neil Johnson)
Date: Fri, 15 Mar 2013 14:59:49 +0000
Subject: [sdiy] is raspberry pi the future?
In-Reply-To: <C4531484-CF4D-4798-95D3-DE05D8B85285@electricdruid.net>
References: <BLU0-SMTP230D6940684AB514B9ADF4BC8EC0@phx.gbl>
	<A61FF618-7614-4D4D-8E4A-32042FDB5815@electricdruid.net>
	<57275E47-9711-4B31-8433-B5251AADF567@me.com>
	<CB78F1D8-554E-4AEA-955E-6D32B9000D6D@blauwurf.info>
	<5142215D.8080906@cox.net>
	<11193470-EC8B-406D-A1C6-4673E61C0F61@earthlink.net>
	<BLU0-SMTP105A12270CDFDEC74084E41C8ED0@phx.gbl>
	<C4531484-CF4D-4798-95D3-DE05D8B85285@electricdruid.net>
Message-ID: <CAEaByMQpTan+npQbs1BhHv5xVkSOdvHK0FcoZge9qJuwr3oMeQ@mail.gmail.com>

Tom,

>> I have learned since yesterday....the rasp pi is not truly open source..i guess some elements are closed.
>
> Where did you see this, Dan? And which bits are closed source?
>
> That seems to totally undermine the whole point of it, in my view.

The firmware for the GPU is closed-source.  Everything on the ARM is
open-source.  If you feel that "...undermine the whole point of it..."
then I'm sure you'll find all you want to know on the Raspberry Pi
forums.

Neil
-- 
Modules and more: http://www.cesyg.com
Homepage: http://www.njohnson.co.uk


From cheater00 at gmail.com  Fri Mar 15 16:26:12 2013
From: cheater00 at gmail.com (cheater cheater)
Date: Fri, 15 Mar 2013 16:26:12 +0100
Subject: [sdiy] is raspberry pi the future?
In-Reply-To: <CAEQ8eXNfiRv=+Rk0kpLYGMO31wVnVbaSchgkW1-XFr0Q1cn2Kg@mail.gmail.com>
References: <BLU0-SMTP230D6940684AB514B9ADF4BC8EC0@phx.gbl>
	<CAEaByMScr-Zub3f4A_OXwo+jrmqr7zvskQSOt3rW=fweE4u54w@mail.gmail.com>
	<CA+9GZUjnYqWSEb-CtmyujE0SBZrY7y9Bi8026T9NqznqmqVsMg@mail.gmail.com>
	<CAEaByMSNCm47DPL172WVO2tP932A+06qYE+Z99xz9X13o=v25w@mail.gmail.com>
	<CA+9GZUhQDRbgypuH2x0Y6eip6zZWkMzumiZmvKdy0Lodkgzu3g@mail.gmail.com>
	<51424D98.10807@ntlworld.com>
	<CA+9GZUhUqwGqJ2FdLgYT0cY_DQuK4LLDS0Z5Kvat8-d7fpw=iA@mail.gmail.com>
	<CAEQ8eXMBJT5kqOD63531wb0O_-hoZsUaQeVngFLM33fA+CWv+g@mail.gmail.com>
	<CA+9GZUgMmm96yjCYimTYe9vYtGXt62oBrJTpGkmVf5799kFiWw@mail.gmail.com>
	<CAEQ8eXNfiRv=+Rk0kpLYGMO31wVnVbaSchgkW1-XFr0Q1cn2Kg@mail.gmail.com>
Message-ID: <CA+9GZUhGYYYqpE6U4=-W0Zbdo_ZX0r+c71r9d9DwANj2zjSADQ@mail.gmail.com>

On Fri, Mar 15, 2013 at 3:21 PM, Marc Nostromo [M-.-n]
<marc.nostromo at gmail.com> wrote:
>>> I don't know about using the RPi as base for proper synth building.
>>> However as a platform for building inexpensive digital synth I love
>>> it. Hook a cheap USB audio out and midi in and you are ready to go.
>>>
>>> I've successfully ported my tracker to it
>>
>> Is porting something like a tracker to the Raspberry Pi when using its
>> onboard linux different than porting it to any other linux-based
>> system? Here I'm assuming you haven't developed your own kernel with
>> usb audio support.
>
> No I didn't but that's where I think lies it's power. Basically all I
> had to do is to grab the source code and target Debian.
>
> I am not at all implying it is a coup de force on my part.However, the
> fact you can have some cheap yet valid digital processor - on top of
> that easily embeddable because of the form factor - make it all
> worthwhile for me. It's something you can, once you developed code for
> it and interfaced properly, simply plug and play without having to
> cope with the fact it's a pc or not.
>
> I'm not trying to say it's rocket science. It just makes sense for me.
>
> /M

I agree with you completely, using tiny, cheap computers like this
really makes sense when you're doing embedded stuff.

Cheers,
D.


From grant at musictechnologiesgroup.com  Fri Mar 15 16:27:03 2013
From: grant at musictechnologiesgroup.com (grant at musictechnologiesgroup.com)
Date: Fri, 15 Mar 2013 08:27:03 -0700
Subject: [sdiy] =?utf-8?q?is_raspberry_pi_the_future=3F?=
Message-ID: <20130315082703.c16e921f9c8655b42ea149fc1daa3492.f882898f55.wbe@email13.secureserver.net>


Someone mentioned ARM. The mbed boards look pretty cheap:

http://mbed.org/

http://mbed.org/blog/entry/mbed-enabled-Freescale-FRDM-KL25Z-board/ 
($12.95)

http://mbed.org/cookbook/Homepage#audio


GB


From tom at electricdruid.net  Fri Mar 15 17:18:33 2013
From: tom at electricdruid.net (Tom Wiltshire)
Date: Fri, 15 Mar 2013 16:18:33 +0000
Subject: [sdiy] is raspberry pi the future?
In-Reply-To: <CAEaByMQpTan+npQbs1BhHv5xVkSOdvHK0FcoZge9qJuwr3oMeQ@mail.gmail.com>
References: <BLU0-SMTP230D6940684AB514B9ADF4BC8EC0@phx.gbl>
	<A61FF618-7614-4D4D-8E4A-32042FDB5815@electricdruid.net>
	<57275E47-9711-4B31-8433-B5251AADF567@me.com>
	<CB78F1D8-554E-4AEA-955E-6D32B9000D6D@blauwurf.info>
	<5142215D.8080906@cox.net>
	<11193470-EC8B-406D-A1C6-4673E61C0F61@earthlink.net>
	<BLU0-SMTP105A12270CDFDEC74084E41C8ED0@phx.gbl>
	<C4531484-CF4D-4798-95D3-DE05D8B85285@electricdruid.net>
	<CAEaByMQpTan+npQbs1BhHv5xVkSOdvHK0FcoZge9qJuwr3oMeQ@mail.gmail.com>
Message-ID: <5F0EEAFD-B155-4994-BF75-0CFE289747C8@electricdruid.net>


On 15 Mar 2013, at 14:59, Neil Johnson <neil.johnson97 at ntlworld.com> wrote:

> Tom,
> 
>>> I have learned since yesterday....the rasp pi is not truly open source..i guess some elements are closed.
>> 
>> Where did you see this, Dan? And which bits are closed source?
>> 
>> That seems to totally undermine the whole point of it, in my view.
> 
> The firmware for the GPU is closed-source.  Everything on the ARM is
> open-source.  If you feel that "...undermine the whole point of it..."
> then I'm sure you'll find all you want to know on the Raspberry Pi
> forums.
> 
> Neil

Thanks Neil, that's what I needed to know. I guess if you regard it as a component, it doesn't matter if it's closed source or not. Just put one in your box and go.

I suppose that for what Dan wants to do - develop on it, then design a stripped-down board with just the bits he wants - he could simply use some similar ARM chip and leave out the GPU?

Tom

From subjectivity at hotmail.com  Fri Mar 15 19:01:30 2013
From: subjectivity at hotmail.com (Dan Snazelle)
Date: Fri, 15 Mar 2013 14:01:30 -0400
Subject: [sdiy] is raspberry pi the future?
In-Reply-To: <5F0EEAFD-B155-4994-BF75-0CFE289747C8@electricdruid.net>
References: <BLU0-SMTP230D6940684AB514B9ADF4BC8EC0@phx.gbl>
	<A61FF618-7614-4D4D-8E4A-32042FDB5815@electricdruid.net>
	<57275E47-9711-4B31-8433-B5251AADF567@me.com>
	<CB78F1D8-554E-4AEA-955E-6D32B9000D6D@blauwurf.info>
	<5142215D.8080906@cox.net>
	<11193470-EC8B-406D-A1C6-4673E61C0F61@earthlink.net>
	<BLU0-SMTP105A12270CDFDEC74084E41C8ED0@phx.gbl>
	<C4531484-CF4D-4798-95D3-DE05D8B85285@electricdruid.net>
	<CAEaByMQpTan+npQbs1BhHv5xVkSOdvHK0FcoZge9qJuwr3oMeQ@mail.gmail.com>
	<5F0EEAFD-B155-4994-BF75-0CFE289747C8@electricdruid.net>
Message-ID: <BLU0-SMTP142F7D917C3375CD0F3DA70C8ED0@phx.gbl>



Sent from my iPhone

On Mar 15, 2013, at 12:18 PM, Tom Wiltshire <tom at electricdruid.net> wrote:

> 
> On 15 Mar 2013, at 14:59, Neil Johnson <neil.johnson97 at ntlworld.com> wrote:
> 
>> Tom,
>> 
>>>> I have learned since yesterday....the rasp pi is not truly open source..i guess some elements are closed.
>>> 
>>> Where did you see this, Dan? And which bits are closed source?
>>> 
>>> That seems to totally undermine the whole point of it, in my view.
>> 
>> The firmware for the GPU is closed-source.  Everything on the ARM is
>> open-source.  If you feel that "...undermine the whole point of it..."
>> then I'm sure you'll find all you want to know on the Raspberry Pi
>> forums.
>> 
>> Neil
> 
> Thanks Neil, that's what I needed to know. I guess if you regard it as a component, it doesn't matter if it's closed source or not. Just put one in your box and go.
> 
> I suppose that for what Dan wants to do - develop on it, then design a stripped-down board with just the bits he wants - he could simply use some similar ARM chip and leave out the GPU?
> 
> Tom


That might be possible

I was actually really surprised to see that some arm CHIPS are as much as $11 a piece!!

I was investigating the SAM chip in the new Arduino Due




From gil_we at yahoo.com  Mon Mar 18 15:30:10 2013
From: gil_we at yahoo.com (Gil W.)
Date: Mon, 18 Mar 2013 07:30:10 -0700 (PDT)
Subject: [sdiy] Fatar keyboard replacement keys (moog Voyager)
Message-ID: <1363617010.50328.YahooMailClassic@web162203.mail.bf1.yahoo.com>

I wonder what other synths use the same keyboard used in the early Voyager versions ? I need a replacement key for a friends' moog and wonder where else I can get it... any idea what Fatar keyboard model they used ?



From tom at electricdruid.net  Mon Mar 18 16:34:33 2013
From: tom at electricdruid.net (Tom Wiltshire)
Date: Mon, 18 Mar 2013 15:34:33 +0000
Subject: [sdiy] Fatar keyboard replacement keys (moog Voyager)
In-Reply-To: <1363617010.50328.YahooMailClassic@web162203.mail.bf1.yahoo.com>
References: <1363617010.50328.YahooMailClassic@web162203.mail.bf1.yahoo.com>
Message-ID: <4BAF893F-9492-4097-8B2C-35AD931C1A55@electricdruid.net>

Where are you, Gil?

In the UK, key parts have some Fatar keybeds:

http://www.keyparts.co.uk/shop?page=shop.browse&category_id=8

In Europe, you can get them from Doepfer, on their Spare Parts page.

http://www.doepfer.de/home_e.htm

HTH,
Tom


On 18 Mar 2013, at 14:30, Gil W. <gil_we at yahoo.com> wrote:

> I wonder what other synths use the same keyboard used in the early Voyager versions ? I need a replacement key for a friends' moog and wonder where else I can get it... any idea what Fatar keyboard model they used ?
> 
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy



From gil_we at yahoo.com  Mon Mar 18 18:50:08 2013
From: gil_we at yahoo.com (Gil W.)
Date: Mon, 18 Mar 2013 10:50:08 -0700 (PDT)
Subject: [sdiy] Fatar keyboard replacement keys (moog Voyager)
In-Reply-To: <4BAF893F-9492-4097-8B2C-35AD931C1A55@electricdruid.net>
Message-ID: <1363629008.71552.YahooMailClassic@web162201.mail.bf1.yahoo.com>

Thanks Tom,

Syntaur.com seems to have them.




--- On Mon, 3/18/13, Tom Wiltshire <tom at electricdruid.net> wrote:

> From: Tom Wiltshire <tom at electricdruid.net>
> Subject: Re: [sdiy] Fatar keyboard replacement keys (moog Voyager)
> To: "Gil W." <gil_we at yahoo.com>
> Cc: "synth-diy at dropmix.xs4all.nl" <synth-diy at dropmix.xs4all.nl>
> Date: Monday, March 18, 2013, 5:34 PM
> Where are you, Gil?
> 
> In the UK, key parts have some Fatar keybeds:
> 
> http://www.keyparts.co.uk/shop?page=shop.browse&category_id=8
> 
> In Europe, you can get them from Doepfer, on their Spare
> Parts page.
> 
> http://www.doepfer.de/home_e.htm
> 
> HTH,
> Tom
> 
> 
> On 18 Mar 2013, at 14:30, Gil W. <gil_we at yahoo.com>
> wrote:
> 
> > I wonder what other synths use the same keyboard used
> in the early Voyager versions ? I need a replacement key for
> a friends' moog and wonder where else I can get it... any
> idea what Fatar keyboard model they used ?
> > 
> > _______________________________________________
> > Synth-diy mailing list
> > Synth-diy at dropmix.xs4all.nl
> > http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
> 
> 


From music.maker at gte.net  Mon Mar 18 22:06:40 2013
From: music.maker at gte.net (Scott Gravenhorst)
Date: Mon, 18 Mar 2013 14:06:40 -0700
Subject: [sdiy] Noob SMT Soldering Question
Message-ID: <201303182106.r2IL6e9G024567@linux7.lan>


I've done on .65mm pitch TSSOP solder job, still don't know if it works...

Anyway, that wasn't too physically hard and I want to do a .4mm pitch
package (TQFP I believe, either 64 or 100 pins).  

Any advice?

-- ScottG
________________________________________________________________________
-- Scott Gravenhorst
-- FPGA MIDI Synth Info: jovianpyx.dyndns.org:8080/public/FPGA_synth/
-- FatMan Mods Etc.: jovianpyx.dyndns.org:8080/public/fatman/
-- Some Random Electronics Bits: jovianpyx.dyndns.org:8080/public/electronics/
-- When the going gets tough, the tough use the command line.



From jim at cernproductions.com  Mon Mar 18 22:45:03 2013
From: jim at cernproductions.com (Jim Credland)
Date: Mon, 18 Mar 2013 21:45:03 +0000
Subject: [sdiy] Noob SMT Soldering Question
In-Reply-To: <201303182106.r2IL6e9G024567@linux7.lan>
References: <201303182106.r2IL6e9G024567@linux7.lan>
Message-ID: <72F2FFFE-A3FC-46B0-BF2E-98E39BF64610@cernproductions.com>

Hot plate technique works great for me. Usually needs a tiny bit of tidying up with solder braid mind you. 

Perhaps I'll get better at it! 

On 18 Mar 2013, at 21:06, Scott Gravenhorst <music.maker at gte.net> wrote:

> 
> I've done on .65mm pitch TSSOP solder job, still don't know if it works...
> 
> Anyway, that wasn't too physically hard and I want to do a .4mm pitch
> package (TQFP I believe, either 64 or 100 pins).  
> 
> Any advice?
> 
> -- ScottG
> ________________________________________________________________________
> -- Scott Gravenhorst
> -- FPGA MIDI Synth Info: jovianpyx.dyndns.org:8080/public/FPGA_synth/
> -- FatMan Mods Etc.: jovianpyx.dyndns.org:8080/public/fatman/
> -- Some Random Electronics Bits: jovianpyx.dyndns.org:8080/public/electronics/
> -- When the going gets tough, the tough use the command line.
> 
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy


From oakleylist at btinternet.com  Tue Mar 19 14:08:46 2013
From: oakleylist at btinternet.com (Oakley Sound)
Date: Tue, 19 Mar 2013 13:08:46 +0000
Subject: [sdiy] Oakley HVM - VP330 and Lambda voice module
Message-ID: <5148635E.20905@btinternet.com>

Hi Folks,

My latest DIY rack project is the Human Voice Module.

The HVM is an electronic audio signal modification device that attempts 
to replicate some of the sounds from two classic keyboards first seen 
and heard in 1979. The module consists of three sections, two band pass 
filter networks and one parametric equaliser. It is possible to use each 
section independently if built that way but the suggested panel design 
has them wired together so as to allow maximum flexibility with the 
smallest number of front panel controls.

The design of the bandpass filter network takes its inspiration from the 
beautiful haunting sounds of the Lambda's chorus ensemble and the 
VP-330's human voice presets. In the suggested panel design of the HVM 
these are labelled Chorus and Vox respectively. A simple toggle switch 
on the panel allows the user to switch between the two modes. The chosen 
output then goes to a wet/dry mixer control pot. This sets the mix 
between the untreated audio input and the affected sound. The output of 
this control is then fed to a parametric equaliser circuit that can 
boost (maximum +20dB) or cut (maximum -20dB) a selected frequency band 
between 50Hz and 11kHz. Bandwidth, Q, can be controlled with the 
emphasis pot and can be varied from around 1 to 25.

More details including photos, Builder's Guide and sound samples:

http://www.oakleysound.com/hvm.htm

Note: the HVM does not contain the ensemble unit from the VP-330. I will 
be working on a four channel ensemble unit shortly.

Tony


From neil.johnson97 at ntlworld.com  Tue Mar 19 15:27:47 2013
From: neil.johnson97 at ntlworld.com (Neil Johnson)
Date: Tue, 19 Mar 2013 14:27:47 +0000
Subject: [sdiy] Oakley HVM - VP330 and Lambda voice module
In-Reply-To: <5148635E.20905@btinternet.com>
References: <5148635E.20905@btinternet.com>
Message-ID: <CAEaByMQDQd6_7T7osDOrCXcEy5fYk0dcOCr==vmi7LD3b5GutA@mail.gmail.com>

Tony,

> My latest DIY rack project is the Human Voice Module.

Nice work!  That's your bring-along sorted for July then :)

Cheers,
Neil
-- 
Modules and more: http://www.cesyg.com
Homepage: http://www.njohnson.co.uk


From tim.parkhurst at gmail.com  Tue Mar 19 19:49:30 2013
From: tim.parkhurst at gmail.com (Tim Parkhurst)
Date: Tue, 19 Mar 2013 11:49:30 -0700
Subject: [sdiy] Duplicates and multiples in KiCAD
Message-ID: <CACDGQT_C9g5VbwDJbVpEgU+8WzEEAwk-sYTX_+MyRPvv9x7dOQ@mail.gmail.com>

So I've run into a situation where I need to do duplicates and
panelized boards in KiCAD, and I have to admit I have no idea how to
go about it, or even if it's possible.

By "duplicates" I mean duplicating most of a circuit several times on
a single board, For example two or three LFO circuits on a PCB, all
fed by the same power input and maybe featuring some other common
circuitry.

By "multiples" I mean duplicating an entire board several times to
make a panelized layout. The separate boards would then be cut apart
later.

Oooh, one more question: is there a way to import Gerbers into KiCAD?

Of course, I'm going to go fish around and find some KiCAD forums, but
I figure I'd ask here first, since this is where all the cool kids
hang out.


Tim (Copy and Putz) Servo
---
"Sire, the church of God is an anvil that has worn out many hammers."
- H.L. Hastings


From jasonnanna at gmail.com  Tue Mar 19 20:05:57 2013
From: jasonnanna at gmail.com (Jason Nanna)
Date: Tue, 19 Mar 2013 14:05:57 -0500
Subject: [sdiy] Duplicates and multiples in KiCAD
In-Reply-To: <CACDGQT_C9g5VbwDJbVpEgU+8WzEEAwk-sYTX_+MyRPvv9x7dOQ@mail.gmail.com>
References: <CACDGQT_C9g5VbwDJbVpEgU+8WzEEAwk-sYTX_+MyRPvv9x7dOQ@mail.gmail.com>
Message-ID: <CAH8amKv603xZi4Uz0C_q8SzSD=p6E=QXd2RvzfkdcMhm2WhkYg@mail.gmail.com>

You can use the python script Gerbmerge to merge your gerbers
together.  It's pretty manual, and requires you to set up offsets for
x/y panelization.

I once watched a youtube tutorial on duplicating sub-boards, which
involved using variable numbers for netnames. from what I remember it
was a pretty messy process, but it can be done.

No way to import gerbers into Kicad as far as I can tell.  What are
you trying to accomplish with that?  You'd obviously have no netlist.
I'd use gerbv if you want to merge PCB's.  Looking just now, it seems
as though gerbv also supports panelization.

On Tue, Mar 19, 2013 at 1:49 PM, Tim Parkhurst <tim.parkhurst at gmail.com> wrote:
> So I've run into a situation where I need to do duplicates and
> panelized boards in KiCAD, and I have to admit I have no idea how to
> go about it, or even if it's possible.
>
> By "duplicates" I mean duplicating most of a circuit several times on
> a single board, For example two or three LFO circuits on a PCB, all
> fed by the same power input and maybe featuring some other common
> circuitry.
>
> By "multiples" I mean duplicating an entire board several times to
> make a panelized layout. The separate boards would then be cut apart
> later.
>
> Oooh, one more question: is there a way to import Gerbers into KiCAD?
>
> Of course, I'm going to go fish around and find some KiCAD forums, but
> I figure I'd ask here first, since this is where all the cool kids
> hang out.
>
>
> Tim (Copy and Putz) Servo
> ---
> "Sire, the church of God is an anvil that has worn out many hammers."
> - H.L. Hastings
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy


From igg.debus at t-online.de  Wed Mar 20 16:58:51 2013
From: igg.debus at t-online.de (Ingo Debus)
Date: Wed, 20 Mar 2013 16:58:51 +0100
Subject: [sdiy] Noob SMT Soldering Question
In-Reply-To: <201303182106.r2IL6e9G024567@linux7.lan>
References: <201303182106.r2IL6e9G024567@linux7.lan>
Message-ID: <87BA78DD-E773-42A9-AD11-C73469D1AFF6@t-online.de>


Am 18.03.2013 um 22:06 schrieb Scott Gravenhorst:

> Anyway, that wasn't too physically hard and I want to do a .4mm pitch
> package (TQFP I believe, either 64 or 100 pins).  
> 
> Any advice?

Pizza oven and stencil. By now, we have soldered three PCBs with a 200 pins 0.5 mm pitch IC on them. The hardest part is placing the devices correctly on the PCB. For the big ICs we use a vacuum pick-up tool.

We use this oven and temperature controller:
<http://www.beta-estore.com/rkuk/index.html?PHPSESSID=qnp17kql6ohbsrpgv125vfjnh4>

Ingo

From music.maker at gte.net  Wed Mar 20 21:47:42 2013
From: music.maker at gte.net (Scott Gravenhorst)
Date: Wed, 20 Mar 2013 13:47:42 -0700
Subject: [sdiy] More SMT
Message-ID: <201303202047.r2KKlgRG026317@linux7.lan>


I'm considering this:

http://www.schmartboard.com/index.asp?page=products_qfp&id=70

To mount a 64 pin TQFP 10mm x 10mm IC.  The board supports "up to 100
pins", but little other information is given.  

Still being noobish with SMT, am I to assume that there are only so may
TQFP package sizes and that the board is designed to accept them all?

-- ScottG
________________________________________________________________________
-- Scott Gravenhorst
-- FPGA MIDI Synth Info: jovianpyx.dyndns.org:8080/public/FPGA_synth/
-- FatMan Mods Etc.: jovianpyx.dyndns.org:8080/public/fatman/
-- Some Random Electronics Bits: jovianpyx.dyndns.org:8080/public/electronics/
-- When the going gets tough, the tough use the command line.



From timr at circuitabbey.com  Wed Mar 20 22:35:29 2013
From: timr at circuitabbey.com (Tim Ressel)
Date: Wed, 20 Mar 2013 14:35:29 -0700 (PDT)
Subject: [sdiy] More SMT
In-Reply-To: <201303202047.r2KKlgRG026317@linux7.lan>
References: <201303202047.r2KKlgRG026317@linux7.lan>
Message-ID: <1363815329.49774.YahooMailNeo@web5801.biz.mail.ne1.yahoo.com>

I've used these before. Pretty nice product. They can take different size packages. It is the lead pitch you need to look for. The one in your link is 0.5mm pitch. 


Tim Ressel
Circuit Abbey
503-750-9331
timr at circuitabbey.com




>________________________________
> From: Scott Gravenhorst <music.maker at gte.net>
>To: synth-diy at dropmix.xs4all.nl 
>Sent: Wednesday, March 20, 2013 1:47 PM
>Subject: [sdiy] More SMT
> 
>
>I'm considering this:
>
>http://www.schmartboard.com/index.asp?page=products_qfp&id=70
>
>To mount a 64 pin TQFP 10mm x 10mm IC.? The board supports "up to 100
>pins", but little other information is given.? 
>
>Still being noobish with SMT, am I to assume that there are only so may
>TQFP package sizes and that the board is designed to accept them all?
>
>-- ScottG
>________________________________________________________________________
>-- Scott Gravenhorst
>-- FPGA MIDI Synth Info: jovianpyx.dyndns.org:8080/public/FPGA_synth/
>-- FatMan Mods Etc.: jovianpyx.dyndns.org:8080/public/fatman/
>-- Some Random Electronics Bits: jovianpyx.dyndns.org:8080/public/electronics/
>-- When the going gets tough, the tough use the command line.
>
>_______________________________________________
>Synth-diy mailing list
>Synth-diy at dropmix.xs4all.nl
>http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>
>
>?


From music.maker at gte.net  Wed Mar 20 22:59:54 2013
From: music.maker at gte.net (Scott Gravenhorst)
Date: Wed, 20 Mar 2013 14:59:54 -0700
Subject: [sdiy] More SMT
Message-ID: <201303202159.r2KLxsIt031224@linux7.lan>


Tim Ressel <timr at circuitabbey.com> wrote:
>I've used these before. Pretty nice product. They can take 
>different size packages. It is the lead pitch you need to look 
>for. The one in your link is 0.5mm pitch. 
>

Thanks Tim,

Yeah, the part I want to put on it says it's 0.5mm pitch, so I guess I am good to go on
ordering the board.

The how-to vid makes it look pretty easy to use.



>
>
>
>
>>________________________________
>> From: Scott Gravenhorst <music.maker at gte.net>
>>To: synth-diy at dropmix.xs4all.nl 
>>Sent: Wednesday, March 20, 2013 1:47 PM
>>Subject: [sdiy] More SMT
>> 
>>
>>I'm considering this:
>>
>>http://www.schmartboard.com/index.asp?page=products_qfp&id=70
>>
>>To mount a 64 pin TQFP 10mm x 10mm IC.? The board supports "up to 100
>>pins", but little other information is given.? 
>>
>>Still being noobish with SMT, am I to assume that there are only so may
>>TQFP package sizes and that the board is designed to accept them all?
>>
>>-- ScottG
>>________________________________________________________________________
>>-- Scott Gravenhorst
>>-- FPGA MIDI Synth Info: jovianpyx.dyndns.org:8080/public/FPGA_synth/
>>-- FatMan Mods Etc.: jovianpyx.dyndns.org:8080/public/fatman/
>>-- Some Random Electronics Bits: jovianpyx.dyndns.org:8080/public/electronics/
>>-- When the going gets tough, the tough use the command line.
>>
>>_______________________________________________
>>Synth-diy mailing list
>>Synth-diy at dropmix.xs4all.nl
>>http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>>
>>
>>?
>

-- ScottG
________________________________________________________________________
-- Scott Gravenhorst
-- FPGA MIDI Synth Info: jovianpyx.dyndns.org:8080/public/FPGA_synth/
-- FatMan Mods Etc.: jovianpyx.dyndns.org:8080/public/fatman/
-- Some Random Electronics Bits: jovianpyx.dyndns.org:8080/public/electronics/
-- When the going gets tough, the tough use the command line.



From Peter.Ullrich at kapsch.net  Wed Mar 20 23:33:11 2013
From: Peter.Ullrich at kapsch.net (Ullrich Peter)
Date: Wed, 20 Mar 2013 23:33:11 +0100
Subject: [sdiy] More SMT
In-Reply-To: <201303202159.r2KLxsIt031224@linux7.lan>
References: <201303202159.r2KLxsIt031224@linux7.lan>
Message-ID: <B156A86831A33949A0977840648802478B17AE8DF7@S060VS20.kapsch.co.at>

Hi!

I don't know how much the schmartboard type is, but maybe simpler versions that are only for the 100pin TQFP are cheaper:

http://www.ebay.com/sch/i.html?_trksid=p2050601.m570.l1313&_nkw=tqfp+100&_sacat=0&_from=R40

Ciao
Peter

http://www.ullrich.at.tt
________________________________________
Von: synth-diy-bounces at dropmix.xs4all.nl [synth-diy-bounces at dropmix.xs4all.nl] im Auftrag von Scott Gravenhorst [music.maker at gte.net]
Gesendet: Mittwoch, 20. M?rz 2013 22:59
An: synth-diy at dropmix.xs4all.nl
Betreff: Re: [sdiy] More SMT

Tim Ressel <timr at circuitabbey.com> wrote:
>I've used these before. Pretty nice product. They can take
>different size packages. It is the lead pitch you need to look
>for. The one in your link is 0.5mm pitch.
>

Thanks Tim,

Yeah, the part I want to put on it says it's 0.5mm pitch, so I guess I am good to go on
ordering the board.

The how-to vid makes it look pretty easy to use.



>
>
>
>
>>________________________________
>> From: Scott Gravenhorst <music.maker at gte.net>
>>To: synth-diy at dropmix.xs4all.nl
>>Sent: Wednesday, March 20, 2013 1:47 PM
>>Subject: [sdiy] More SMT
>>
>>
>>I'm considering this:
>>
>>http://www.schmartboard.com/index.asp?page=products_qfp&id=70
>>
>>To mount a 64 pin TQFP 10mm x 10mm IC.  The board supports "up to 100
>>pins", but little other information is given.
>>
>>Still being noobish with SMT, am I to assume that there are only so may
>>TQFP package sizes and that the board is designed to accept them all?
>>
>>-- ScottG
>>________________________________________________________________________
>>-- Scott Gravenhorst
>>-- FPGA MIDI Synth Info: jovianpyx.dyndns.org:8080/public/FPGA_synth/
>>-- FatMan Mods Etc.: jovianpyx.dyndns.org:8080/public/fatman/
>>-- Some Random Electronics Bits: jovianpyx.dyndns.org:8080/public/electronics/
>>-- When the going gets tough, the tough use the command line.
>>
>>_______________________________________________
>>Synth-diy mailing list
>>Synth-diy at dropmix.xs4all.nl
>>http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>>
>>
>> 
>

-- ScottG
________________________________________________________________________
-- Scott Gravenhorst
-- FPGA MIDI Synth Info: jovianpyx.dyndns.org:8080/public/FPGA_synth/
-- FatMan Mods Etc.: jovianpyx.dyndns.org:8080/public/fatman/
-- Some Random Electronics Bits: jovianpyx.dyndns.org:8080/public/electronics/
-- When the going gets tough, the tough use the command line.
The information contained in this e-mail message is privileged and
confidential and is for the exclusive use of the addressee. The person
who receives this message and who is not the addressee, one of his
employees or an agent entitled to hand it over to the addressee, is
informed that he may not use, disclose or reproduce the contents
thereof, and is kindly asked to notify the sender and delete the e-mail
immediately.




From timr at circuitabbey.com  Thu Mar 21 03:59:44 2013
From: timr at circuitabbey.com (Tim Ressel)
Date: Wed, 20 Mar 2013 19:59:44 -0700 (PDT)
Subject: [sdiy] More SMT
In-Reply-To: <B156A86831A33949A0977840648802478B17AE8DF7@S060VS20.kapsch.co.at>
References: <201303202159.r2KLxsIt031224@linux7.lan>
	<B156A86831A33949A0977840648802478B17AE8DF7@S060VS20.kapsch.co.at>
Message-ID: <1363834784.16045.YahooMailNeo@web5805.biz.mail.ne1.yahoo.com>

The cool thing about the schmartboards is the extra thick solder mask. It helps align the pins for soldering. And let me tell you: aligning the 0.5mm pins is not very easy. Also the thick solder mask helps prevent bridges. I think they are worth the money.



Tim Ressel
Circuit Abbey
503-750-9331
timr at circuitabbey.com




>________________________________
> From: Ullrich Peter <Peter.Ullrich at kapsch.net>
>To: "music.maker at gte.net" <music.maker at gte.net>; "synth-diy at dropmix.xs4all.nl" <synth-diy at dropmix.xs4all.nl> 
>Sent: Wednesday, March 20, 2013 3:33 PM
>Subject: Re: [sdiy] More SMT
> 
>Hi!
>
>I don't know how much the schmartboard type is, but maybe simpler versions that are only for the 100pin TQFP are cheaper:
>
>http://www.ebay.com/sch/i.html?_trksid=p2050601.m570.l1313&_nkw=tqfp+100&_sacat=0&_from=R40
>
>Ciao
>Peter
>
>http://www.ullrich.at.tt
>________________________________________
>Von: synth-diy-bounces at dropmix.xs4all.nl [synth-diy-bounces at dropmix.xs4all.nl] im Auftrag von Scott Gravenhorst [music.maker at gte.net]
>Gesendet: Mittwoch, 20. M?rz 2013 22:59
>An: synth-diy at dropmix.xs4all.nl
>Betreff: Re: [sdiy] More SMT
>
>Tim Ressel <timr at circuitabbey.com> wrote:
>>I've used these before. Pretty nice product. They can take
>>different size packages. It is the lead pitch you need to look
>>for. The one in your link is 0.5mm pitch.
>>
>
>Thanks Tim,
>
>Yeah, the part I want to put on it says it's 0.5mm pitch, so I guess I am good to go on
>ordering the board.
>
>The how-to vid makes it look pretty easy to use.
>
>
>
>>
>>
>>
>>
>>>________________________________
>>> From: Scott Gravenhorst <music.maker at gte.net>
>>>To: synth-diy at dropmix.xs4all.nl
>>>Sent: Wednesday, March 20, 2013 1:47 PM
>>>Subject: [sdiy] More SMT
>>>
>>>
>>>I'm considering this:
>>>
>>>http://www.schmartboard.com/index.asp?page=products_qfp&id=70
>>>
>>>To mount a 64 pin TQFP 10mm x 10mm IC.? The board supports "up to 100
>>>pins", but little other information is given.
>>>
>>>Still being noobish with SMT, am I to assume that there are only so may
>>>TQFP package sizes and that the board is designed to accept them all?
>>>
>>>-- ScottG
>>>________________________________________________________________________
>>>-- Scott Gravenhorst
>>>-- FPGA MIDI Synth Info: jovianpyx.dyndns.org:8080/public/FPGA_synth/
>>>-- FatMan Mods Etc.: jovianpyx.dyndns.org:8080/public/fatman/
>>>-- Some Random Electronics Bits: jovianpyx.dyndns.org:8080/public/electronics/
>>>-- When the going gets tough, the tough use the command line.
>>>
>>>_______________________________________________
>>>Synth-diy mailing list
>>>Synth-diy at dropmix.xs4all.nl
>>>http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>>>
>>>
>>> 
>>
>
>-- ScottG
>________________________________________________________________________
>-- Scott Gravenhorst
>-- FPGA MIDI Synth Info: jovianpyx.dyndns.org:8080/public/FPGA_synth/
>-- FatMan Mods Etc.: jovianpyx.dyndns.org:8080/public/fatman/
>-- Some Random Electronics Bits: jovianpyx.dyndns.org:8080/public/electronics/
>-- When the going gets tough, the tough use the command line.
>The information contained in this e-mail message is privileged and
>confidential and is for the exclusive use of the addressee. The person
>who receives this message and who is not the addressee, one of his
>employees or an agent entitled to hand it over to the addressee, is
>informed that he may not use, disclose or reproduce the contents
>thereof, and is kindly asked to notify the sender and delete the e-mail
>immediately.
>
>
>_______________________________________________
>Synth-diy mailing list
>Synth-diy at dropmix.xs4all.nl
>http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>
>
>?


From elektrodwarf at yahoo.se  Thu Mar 21 04:44:34 2013
From: elektrodwarf at yahoo.se (Karl Ekdahl)
Date: Thu, 21 Mar 2013 03:44:34 +0000 (GMT)
Subject: [sdiy] Duplicates and multiples in KiCAD
In-Reply-To: <CACDGQT_C9g5VbwDJbVpEgU+8WzEEAwk-sYTX_+MyRPvv9x7dOQ@mail.gmail.com>
References: <CACDGQT_C9g5VbwDJbVpEgU+8WzEEAwk-sYTX_+MyRPvv9x7dOQ@mail.gmail.com>
Message-ID: <1363837474.65888.YahooMailNeo@web133017.mail.ir2.yahoo.com>

To panelize/combine boards you can use "append board" from the file menu in the PCB editor to add another file (or the same file in this case), just make sure to move the pre-existing board down/up/left/right first as the new board will be appended in the same position as it's saved in the file you append.

As for the "duplicates"-thing, i've been wondering that myself. As there is a way of doing it in the schematics editor using hierarchies, maybe there's something similar for the PCB editor?

Karl


________________________________
Fr?n: Tim Parkhurst <tim.parkhurst at gmail.com>
Till: Synth-DIY <synth-diy at dropmix.xs4all.nl> 
Skickat: tisdag, 19 mars 2013 14:49
?mne: [sdiy] Duplicates and multiples in KiCAD

So I've run into a situation where I need to do duplicates and
panelized boards in KiCAD, and I have to admit I have no idea how to
go about it, or even if it's possible.

By "duplicates" I mean duplicating most of a circuit several times on
a single board, For example two or three LFO circuits on a PCB, all
fed by the same power input and maybe featuring some other common
circuitry.

By "multiples" I mean duplicating an entire board several times to
make a panelized layout. The separate boards would then be cut apart
later.

Oooh, one more question: is there a way to import Gerbers into KiCAD?

Of course, I'm going to go fish around and find some KiCAD forums, but
I figure I'd ask here first, since this is where all the cool kids
hang out.


Tim (Copy and Putz) Servo
---
"Sire, the church of God is an anvil that has worn out many hammers."
- H.L. Hastings
_______________________________________________
Synth-diy mailing list
Synth-diy at dropmix.xs4all.nl
http://dropmix.xs4all.nl/mailman/listinfo/synth-diy


From jasonnanna at gmail.com  Thu Mar 21 09:41:02 2013
From: jasonnanna at gmail.com (Jason Nanna)
Date: Thu, 21 Mar 2013 03:41:02 -0500
Subject: [sdiy] Duplicates and multiples in KiCAD
In-Reply-To: <1363837474.65888.YahooMailNeo@web133017.mail.ir2.yahoo.com>
References: <CACDGQT_C9g5VbwDJbVpEgU+8WzEEAwk-sYTX_+MyRPvv9x7dOQ@mail.gmail.com>
	<1363837474.65888.YahooMailNeo@web133017.mail.ir2.yahoo.com>
Message-ID: <CAH8amKtO__uJaxnojicQPw_o3COYJFtwOFiS+8VwCHhVMgc14g@mail.gmail.com>

Helpful:

PyEDA aims to provide EDA tools written in Python, and to link to
other relevant projects.

The first subproject is kipy (Python tools for KiCAD). Not really
documented at all yet, but you can download kipy via subversion:

svn checkout http://pyeda.googlecode.com/svn/trunk/kipy kipy

kipy is basically a library designed for easy use from other programs,
but the kipy/tools directory contains scripts which can do the
following:

Round-trip (read and write) Kicad .pro, .sch, and .lib files
Parse the .sch and .lib files and do additional rules checking
Output the netlist in a format suitable for use with ExpressPCB
Dump pin to net information for any reference designator

Kipy works under Linux and Windows and currently requires Python 2.5
or 2.6, as well as a KiCAD installation.

And here's the youtube clip I had referred to on cloning subcircuits
-- it utilizes the tools available above:

http://www.youtube.com/watch?v=mChTfL746Fs

On Wed, Mar 20, 2013 at 10:44 PM, Karl Ekdahl <elektrodwarf at yahoo.se> wrote:
> To panelize/combine boards you can use "append board" from the file menu in the PCB editor to add another file (or the same file in this case), just make sure to move the pre-existing board down/up/left/right first as the new board will be appended in the same position as it's saved in the file you append.
>
> As for the "duplicates"-thing, i've been wondering that myself. As there is a way of doing it in the schematics editor using hierarchies, maybe there's something similar for the PCB editor?
>
> Karl
>
>
> ________________________________
> Fr?n: Tim Parkhurst <tim.parkhurst at gmail.com>
> Till: Synth-DIY <synth-diy at dropmix.xs4all.nl>
> Skickat: tisdag, 19 mars 2013 14:49
> ?mne: [sdiy] Duplicates and multiples in KiCAD
>
> So I've run into a situation where I need to do duplicates and
> panelized boards in KiCAD, and I have to admit I have no idea how to
> go about it, or even if it's possible.
>
> By "duplicates" I mean duplicating most of a circuit several times on
> a single board, For example two or three LFO circuits on a PCB, all
> fed by the same power input and maybe featuring some other common
> circuitry.
>
> By "multiples" I mean duplicating an entire board several times to
> make a panelized layout. The separate boards would then be cut apart
> later.
>
> Oooh, one more question: is there a way to import Gerbers into KiCAD?
>
> Of course, I'm going to go fish around and find some KiCAD forums, but
> I figure I'd ask here first, since this is where all the cool kids
> hang out.
>
>
> Tim (Copy and Putz) Servo
> ---
> "Sire, the church of God is an anvil that has worn out many hammers."
> - H.L. Hastings
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy


From jasonnanna at gmail.com  Thu Mar 21 09:51:55 2013
From: jasonnanna at gmail.com (Jason Nanna)
Date: Thu, 21 Mar 2013 03:51:55 -0500
Subject: [sdiy] Duplicates and multiples in KiCAD
In-Reply-To: <CAH8amKtO__uJaxnojicQPw_o3COYJFtwOFiS+8VwCHhVMgc14g@mail.gmail.com>
References: <CACDGQT_C9g5VbwDJbVpEgU+8WzEEAwk-sYTX_+MyRPvv9x7dOQ@mail.gmail.com>
	<1363837474.65888.YahooMailNeo@web133017.mail.ir2.yahoo.com>
	<CAH8amKtO__uJaxnojicQPw_o3COYJFtwOFiS+8VwCHhVMgc14g@mail.gmail.com>
Message-ID: <CAH8amKts8RvEyOYA1kCfCX-07KgEosW9w9vdB5Y-PcLZmDw6sw@mail.gmail.com>

And this script, specifically
http://code.google.com/p/klonor-kicad/

On Thu, Mar 21, 2013 at 3:41 AM, Jason Nanna <jasonnanna at gmail.com> wrote:
> Helpful:
>
> PyEDA aims to provide EDA tools written in Python, and to link to
> other relevant projects.
>
> The first subproject is kipy (Python tools for KiCAD). Not really
> documented at all yet, but you can download kipy via subversion:
>
> svn checkout http://pyeda.googlecode.com/svn/trunk/kipy kipy
>
> kipy is basically a library designed for easy use from other programs,
> but the kipy/tools directory contains scripts which can do the
> following:
>
> Round-trip (read and write) Kicad .pro, .sch, and .lib files
> Parse the .sch and .lib files and do additional rules checking
> Output the netlist in a format suitable for use with ExpressPCB
> Dump pin to net information for any reference designator
>
> Kipy works under Linux and Windows and currently requires Python 2.5
> or 2.6, as well as a KiCAD installation.
>
> And here's the youtube clip I had referred to on cloning subcircuits
> -- it utilizes the tools available above:
>
> http://www.youtube.com/watch?v=mChTfL746Fs
>
> On Wed, Mar 20, 2013 at 10:44 PM, Karl Ekdahl <elektrodwarf at yahoo.se> wrote:
>> To panelize/combine boards you can use "append board" from the file menu in the PCB editor to add another file (or the same file in this case), just make sure to move the pre-existing board down/up/left/right first as the new board will be appended in the same position as it's saved in the file you append.
>>
>> As for the "duplicates"-thing, i've been wondering that myself. As there is a way of doing it in the schematics editor using hierarchies, maybe there's something similar for the PCB editor?
>>
>> Karl
>>
>>
>> ________________________________
>> Fr?n: Tim Parkhurst <tim.parkhurst at gmail.com>
>> Till: Synth-DIY <synth-diy at dropmix.xs4all.nl>
>> Skickat: tisdag, 19 mars 2013 14:49
>> ?mne: [sdiy] Duplicates and multiples in KiCAD
>>
>> So I've run into a situation where I need to do duplicates and
>> panelized boards in KiCAD, and I have to admit I have no idea how to
>> go about it, or even if it's possible.
>>
>> By "duplicates" I mean duplicating most of a circuit several times on
>> a single board, For example two or three LFO circuits on a PCB, all
>> fed by the same power input and maybe featuring some other common
>> circuitry.
>>
>> By "multiples" I mean duplicating an entire board several times to
>> make a panelized layout. The separate boards would then be cut apart
>> later.
>>
>> Oooh, one more question: is there a way to import Gerbers into KiCAD?
>>
>> Of course, I'm going to go fish around and find some KiCAD forums, but
>> I figure I'd ask here first, since this is where all the cool kids
>> hang out.
>>
>>
>> Tim (Copy and Putz) Servo
>> ---
>> "Sire, the church of God is an anvil that has worn out many hammers."
>> - H.L. Hastings
>> _______________________________________________
>> Synth-diy mailing list
>> Synth-diy at dropmix.xs4all.nl
>> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>> _______________________________________________
>> Synth-diy mailing list
>> Synth-diy at dropmix.xs4all.nl
>> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy


From philgallo at gmail.com  Thu Mar 21 17:30:35 2013
From: philgallo at gmail.com (Phillip Gallo)
Date: Thu, 21 Mar 2013 09:30:35 -0700
Subject: [sdiy] More SMT
In-Reply-To: <201303202047.r2KKlgRG026317@linux7.lan>
References: <201303202047.r2KKlgRG026317@linux7.lan>
Message-ID: <CAN4YM6c5VEqcyg+RRVkuaLSeHKRW4gVVAvc6Yd7zJU5=mYumow@mail.gmail.com>

Scott,

I use these boards quite a bit.
I get 'em retail at Fry's Elec.

I scrape paste flush across the solder pads,  make sure i mount the
part right-side up (in sync with nomenclature), solder corners and
then 'drag solder' across all the pins.  I then come back and slowly
scrape solderwick across the pins.

I then 'ring it out' from the pin to the board edge connection.

 http://www.sdiy.org/philgallo/photo.jpg

p


From music.maker at gte.net  Thu Mar 21 17:52:09 2013
From: music.maker at gte.net (Scott Gravenhorst)
Date: Thu, 21 Mar 2013 09:52:09 -0700
Subject: [sdiy] More SMT
Message-ID: <201303211652.r2LGq9Dh001138@linux7.lan>


Phillip Gallo <philgallo at gmail.com> wrote:
>Scott,
>
>I use these boards quite a bit.
>I get 'em retail at Fry's Elec.
>
>I scrape paste flush across the solder pads,  make sure i mount the
>part right-side up (in sync with nomenclature), solder corners and
>then 'drag solder' across all the pins.  I then come back and slowly
>scrape solderwick across the pins.
>
>I then 'ring it out' from the pin to the board edge connection.
>
> http://www.sdiy.org/philgallo/photo.jpg
>

Yesterday, I noticed a youtube video link on the page URL I posted:

http://www.schmartboard.com/index.asp?page=products_qfp&id=70

The video link:

http://www.youtube.com/watch?v=E6WrzdgwyIw&feature=related

This is a "how-to" video on soldering.  What I found fascinating is that the
person did the solder job using liquid flux and no additional solder.  There was
no need (apparently) for cleaning up with solder wick.  Definately worth the 5 or
so minutes to watch and I will definately try this method.  I do note that the
tip on the iron was very fine and very clean.  looked brand new.

-- ScottG
________________________________________________________________________
-- Scott Gravenhorst
-- FPGA MIDI Synth Info: jovianpyx.dyndns.org:8080/public/FPGA_synth/
-- FatMan Mods Etc.: jovianpyx.dyndns.org:8080/public/fatman/
-- Some Random Electronics Bits: jovianpyx.dyndns.org:8080/public/electronics/
-- When the going gets tough, the tough use the command line.



From philgallo at gmail.com  Thu Mar 21 18:14:20 2013
From: philgallo at gmail.com (Phillip Gallo)
Date: Thu, 21 Mar 2013 10:14:20 -0700
Subject: [sdiy] More SMT
In-Reply-To: <201303211652.r2LGq9Dh001138@linux7.lan>
References: <201303211652.r2LGq9Dh001138@linux7.lan>
Message-ID: <CAN4YM6eZgqNTbQmah45oL=GSum0D9V92Rzd_HT77qkHNf+jZUQ@mail.gmail.com>

Scott,

Yes, great video.  I didn't master the technique. The time saved was
spent uncovering connections that were 'true' only with the pressure
of my meter probes (when ringing it).

p



On Thu, Mar 21, 2013 at 9:52 AM, Scott Gravenhorst <music.maker at gte.net> wrote:
>
> Phillip Gallo <philgallo at gmail.com> wrote:
>>Scott,
>>
>>I use these boards quite a bit.
>>I get 'em retail at Fry's Elec.
>>
>>I scrape paste flush across the solder pads,  make sure i mount the
>>part right-side up (in sync with nomenclature), solder corners and
>>then 'drag solder' across all the pins.  I then come back and slowly
>>scrape solderwick across the pins.
>>
>>I then 'ring it out' from the pin to the board edge connection.
>>
>> http://www.sdiy.org/philgallo/photo.jpg
>>
>
> Yesterday, I noticed a youtube video link on the page URL I posted:
>
> http://www.schmartboard.com/index.asp?page=products_qfp&id=70
>
> The video link:
>
> http://www.youtube.com/watch?v=E6WrzdgwyIw&feature=related
>
> This is a "how-to" video on soldering.  What I found fascinating is that the
> person did the solder job using liquid flux and no additional solder.  There was
> no need (apparently) for cleaning up with solder wick.  Definately worth the 5 or
> so minutes to watch and I will definately try this method.  I do note that the
> tip on the iron was very fine and very clean.  looked brand new.
>
> -- ScottG
> ________________________________________________________________________
> -- Scott Gravenhorst
> -- FPGA MIDI Synth Info: jovianpyx.dyndns.org:8080/public/FPGA_synth/
> -- FatMan Mods Etc.: jovianpyx.dyndns.org:8080/public/fatman/
> -- Some Random Electronics Bits: jovianpyx.dyndns.org:8080/public/electronics/
> -- When the going gets tough, the tough use the command line.
>
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy


From aym-htnys at teaser.fr  Thu Mar 21 20:15:30 2013
From: aym-htnys at teaser.fr (Andre Majorel)
Date: Thu, 21 Mar 2013 20:15:30 +0100
Subject: [sdiy] More SMT
In-Reply-To: <201303211652.r2LGq9Dh001138@linux7.lan>
References: <CAN4YM6c5VEqcyg+RRVkuaLSeHKRW4gVVAvc6Yd7zJU5=mYumow@mail.gmail.com>
	<201303211652.r2LGq9Dh001138@linux7.lan>
Message-ID: <20130321191530.GA5548@aym.net2.nerim.net>

On 2013-03-21 09:52 -0700, Scott Gravenhorst wrote:

> http://www.youtube.com/watch?v=E6WrzdgwyIw&feature=related
> 
> This is a "how-to" video on soldering.  What I found
> fascinating is that the person did the solder job using liquid
> flux and no additional solder.  There was no need (apparently)
> for cleaning up with solder wick.  Definately worth the 5 or
> so minutes to watch and I will definately try this method.  I
> do note that the tip on the iron was very fine and very clean.
> looked brand new.

That guy has steady hands. I'm not sure I could pull that off
but it's certainly interesting to know it can be done.

-- 
Andr? Majorel http://www.teaser.fr/~amajorel/


From altitude at optrand.com  Fri Mar 22 02:57:47 2013
From: altitude at optrand.com (Altitude)
Date: Thu, 21 Mar 2013 21:57:47 -0400
Subject: [sdiy] More SMT
In-Reply-To: <20130321191530.GA5548@aym.net2.nerim.net>
References: <CAN4YM6c5VEqcyg+RRVkuaLSeHKRW4gVVAvc6Yd7zJU5=mYumow@mail.gmail.com>	<201303211652.r2LGq9Dh001138@linux7.lan>
	<20130321191530.GA5548@aym.net2.nerim.net>
Message-ID: <000001ce26a0$a74590a0$f5d0b1e0$@com>

Obviously this only works with HASL plated boards but I have been fond of
doing the same thing for ENIG plated contacts by just tinning the tip of the
iron with solder (and I mean VERY little), fluxing the contacts and the just
touching each pin with the iron, works like a charm.  You should only have
enough solder on the tip for 3-4 pins but once you get it down, TQFP stuff
is no problem and you can even use that method for the dreaded uQFN stuff

-----Original Message-----
From: synth-diy-bounces at dropmix.xs4all.nl
[mailto:synth-diy-bounces at dropmix.xs4all.nl] On Behalf Of Andre Majorel
Sent: Thursday, March 21, 2013 3:16 PM
To: synth-diy at dropmix.xs4all.nl
Subject: Re: [sdiy] More SMT

On 2013-03-21 09:52 -0700, Scott Gravenhorst wrote:

> http://www.youtube.com/watch?v=E6WrzdgwyIw&feature=related
> 
> This is a "how-to" video on soldering.  What I found fascinating is 
> that the person did the solder job using liquid flux and no additional 
> solder.  There was no need (apparently) for cleaning up with solder 
> wick.  Definately worth the 5 or so minutes to watch and I will 
> definately try this method.  I do note that the tip on the iron was 
> very fine and very clean.
> looked brand new.

That guy has steady hands. I'm not sure I could pull that off but it's
certainly interesting to know it can be done.

--
Andr? Majorel http://www.teaser.fr/~amajorel/
_______________________________________________
Synth-diy mailing list
Synth-diy at dropmix.xs4all.nl
http://dropmix.xs4all.nl/mailman/listinfo/synth-diy



From dlmanley at sonic.net  Tue Mar 26 06:33:24 2013
From: dlmanley at sonic.net (Dave Manley)
Date: Mon, 25 Mar 2013 22:33:24 -0700
Subject: [sdiy] MN3209 unobtanium?
Message-ID: <51513324.3010606@sonic.net>

I've got an old synth with a chorus circuit that doesn't work.  I haven't
got the scope out yet, but it uses 2x MN3209.  Is there any source for
these?

-Dave


From tom at electricdruid.net  Tue Mar 26 11:55:53 2013
From: tom at electricdruid.net (Tom Wiltshire)
Date: Tue, 26 Mar 2013 10:55:53 +0000
Subject: [sdiy] MN3209 unobtanium?
In-Reply-To: <51513324.3010606@sonic.net>
References: <51513324.3010606@sonic.net>
Message-ID: <7866E6E6-0963-45B6-9AB3-7687FF9FDE78@electricdruid.net>

I got some off AllPartsPipe on Ebay that seem to be what they say they are:

http://www.ebay.co.uk/itm/2-Panasonic-MN3209-Operation-Low-Noise-BBD-ICS-CHIPs-/390125216745

I was experimenting with flangers, so I was looking for shorter delay lines. These a) had the pins in the right place b) worked and c) gave the expected delay for the stated length. That's about as many tests as I could think of!

Regards,
Tom


On 26 Mar 2013, at 05:33, Dave Manley <dlmanley at sonic.net> wrote:

> I've got an old synth with a chorus circuit that doesn't work.  I haven't
> got the scope out yet, but it uses 2x MN3209.  Is there any source for
> these?
> 
> -Dave
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy



From jpdesroc at oricom.ca  Thu Mar 28 13:44:20 2013
From: jpdesroc at oricom.ca (Jean-Pierre Desrochers)
Date: Thu, 28 Mar 2013 08:44:20 -0400
Subject: [sdiy] Fast envelope follower circuit needed..
Message-ID: <324c9c6efbb87ef612fdd8cdc2ccad26@oricom.ca>

Hi List,
I'm sure this has been already discussed in the past
but I'm looking for a fast envelope follower circuit
schematic to rectify an incoming +/-2.5v AC waveform
to a rectified 0-5v wave.. The wave frequency could
go as low as 1 hz and sometimes will not be symetrical.
Any circuits welcomed !
JP


From tom at electricdruid.net  Thu Mar 28 14:21:59 2013
From: tom at electricdruid.net (Tom Wiltshire)
Date: Thu, 28 Mar 2013 13:21:59 +0000
Subject: [sdiy] Fast envelope follower circuit needed..
In-Reply-To: <324c9c6efbb87ef612fdd8cdc2ccad26@oricom.ca>
References: <324c9c6efbb87ef612fdd8cdc2ccad26@oricom.ca>
Message-ID: <7D00642A-814D-4BBC-B510-CDAC1630728D@electricdruid.net>

Hi JP,

The typical circuit is a precision full-wave rectifier (or sometimes not even a full wave rectifier) followed by a filter. It works pretty well, but is not flawless, especially at low frequencies. The many Auto-wah/envelope filter circuits have examples if you need a schematic.

You're asking for a lot if you want both "fast" and "down to 1Hz". After all, the rise time on a 1Hz sine waveform is 250ms, which in my view is "slow".

Have you considered using a digital solution? Shift the signal to 0-5V range, then feed it to the ADC input of a PIC/AVR (You're a PIC programmer, I know). Digital rectification is easy. Then you could do the smoothing in software. This might allow you to do some clever tricks that you couldn't easily do with analog, like have a filter that responds to the incoming signal frequency. Just a thought.

Regards,
Tom



On 28 Mar 2013, at 12:44, Jean-Pierre Desrochers <jpdesroc at oricom.ca> wrote:

> Hi List,
> I'm sure this has been already discussed in the past
> but I'm looking for a fast envelope follower circuit
> schematic to rectify an incoming +/-2.5v AC waveform
> to a rectified 0-5v wave.. The wave frequency could
> go as low as 1 hz and sometimes will not be symetrical.
> Any circuits welcomed !
> JP
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy



From jpdesroc at oricom.ca  Thu Mar 28 14:39:28 2013
From: jpdesroc at oricom.ca (Jean-Pierre Desrochers)
Date: Thu, 28 Mar 2013 09:39:28 -0400
Subject: [sdiy] Fast envelope follower circuit needed..
In-Reply-To: <7D00642A-814D-4BBC-B510-CDAC1630728D@electricdruid.net>
References: <324c9c6efbb87ef612fdd8cdc2ccad26@oricom.ca>
	<7D00642A-814D-4BBC-B510-CDAC1630728D@electricdruid.net>
Message-ID: <e656f4eb01e9ddf9ffb5c18e0107f273@oricom.ca>

You are right Tom,

I was thinking of a 'digital' solution using a PIC..
But I was slowed down in my thoughts because I 'try'
not to use that much digital stuff in my analog synth projects.
The digital way to solve analog problems is very appealing !
And for this function I think I'll need some 'inteligence'
in the way the envelope will be generated.
The average analog envelope follower circuits have flaws
and I'll have to think about what I can accept out of them
and what I won't.
JP

Le 2013-03-28 09:21, Tom Wiltshire a ?crit?:
> Hi JP,
>
> The typical circuit is a precision full-wave rectifier (or sometimes
> not even a full wave rectifier) followed by a filter. It works pretty
> well, but is not flawless, especially at low frequencies. The many
> Auto-wah/envelope filter circuits have examples if you need a
> schematic.
>
> You're asking for a lot if you want both "fast" and "down to 1Hz".
> After all, the rise time on a 1Hz sine waveform is 250ms, which in my
> view is "slow".
>
> Have you considered using a digital solution? Shift the signal to
> 0-5V range, then feed it to the ADC input of a PIC/AVR (You're a PIC
> programmer, I know). Digital rectification is easy. Then you could do
> the smoothing in software. This might allow you to do some clever
> tricks that you couldn't easily do with analog, like have a filter
> that responds to the incoming signal frequency. Just a thought.
>
> Regards,
> Tom
>
>
>
> On 28 Mar 2013, at 12:44, Jean-Pierre Desrochers <jpdesroc at oricom.ca> 
> wrote:
>
>> Hi List,
>> I'm sure this has been already discussed in the past
>> but I'm looking for a fast envelope follower circuit
>> schematic to rectify an incoming +/-2.5v AC waveform
>> to a rectified 0-5v wave.. The wave frequency could
>> go as low as 1 hz and sometimes will not be symetrical.
>> Any circuits welcomed !
>> JP
>> _______________________________________________
>> Synth-diy mailing list
>> Synth-diy at dropmix.xs4all.nl
>> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy



From subjectivity at hotmail.com  Thu Mar 28 14:55:28 2013
From: subjectivity at hotmail.com (Dan Snazelle)
Date: Thu, 28 Mar 2013 09:55:28 -0400
Subject: [sdiy] Fast envelope follower circuit needed..
In-Reply-To: <7D00642A-814D-4BBC-B510-CDAC1630728D@electricdruid.net>
References: <324c9c6efbb87ef612fdd8cdc2ccad26@oricom.ca>
	<7D00642A-814D-4BBC-B510-CDAC1630728D@electricdruid.net>
Message-ID: <BLU0-SMTP338D2FA8904A3F6D5D401C9C8D20@phx.gbl>



Sent from my iPhone

On Mar 28, 2013, at 9:21 AM, Tom Wiltshire <tom at electricdruid.net> wrote:

> Hi JP,
> 
> The typical circuit is a precision full-wave rectifier (or sometimes not even a full wave rectifier) followed by a filter. It works pretty well, but is not flawless, especially at low frequencies. The many Auto-wah/envelope filter circuits have examples if you need a schematic.
> 
> You're asking for a lot if you want both "fast" and "down to 1Hz". After all, the rise time on a 1Hz sine waveform is 250ms, which in my view is "slow".
> 
> Have you considered using a digital solution? Shift the signal to 0-5V range, then feed it to the ADC input of a PIC/AVR (You're a PIC programmer, I know). Digital rectification is easy. Then you could do the smoothing in software. This might allow you to do some clever tricks that you couldn't easily do with analog, like have a filter that responds to the incoming signal frequency. Just a thought.
> 
> Regards,
> Tom
> 



Though i work with c//arduino/AVRS not pics, this sounds very appealing! Can anyone recommend a place to learn about writing C code for an envelope follower? How do you do it? Shift signal to 0-5, then......


Thanks!!!



> 
> 
> On 28 Mar 2013, at 12:44, Jean-Pierre Desrochers <jpdesroc at oricom.ca> wrote:
> 
>> Hi List,
>> I'm sure this has been already discussed in the past
>> but I'm looking for a fast envelope follower circuit
>> schematic to rectify an incoming +/-2.5v AC waveform
>> to a rectified 0-5v wave.. The wave frequency could
>> go as low as 1 hz and sometimes will not be symetrical.
>> Any circuits welcomed !
>> JP
>> _______________________________________________
>> Synth-diy mailing list
>> Synth-diy at dropmix.xs4all.nl
>> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
> 
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
> 


From cheater00 at gmail.com  Thu Mar 28 18:19:33 2013
From: cheater00 at gmail.com (cheater cheater)
Date: Thu, 28 Mar 2013 18:19:33 +0100
Subject: [sdiy] Fast envelope follower circuit needed..
In-Reply-To: <324c9c6efbb87ef612fdd8cdc2ccad26@oricom.ca>
References: <324c9c6efbb87ef612fdd8cdc2ccad26@oricom.ca>
Message-ID: <CA+9GZUj1AK+hmgAKQx+Zi-tau1YPOc+B112XSw67=245sZ5RXw@mail.gmail.com>

So what should happen when a 1Hz sinewave, amplitude-modulated by a
0.01 Hz sinewave, is fed into your box? Do you expect to get a 1Hz
square wave? 2Hz square wave? 0.01 Hz square wave?

On Thu, Mar 28, 2013 at 1:44 PM, Jean-Pierre Desrochers
<jpdesroc at oricom.ca> wrote:
> Hi List,
> I'm sure this has been already discussed in the past
> but I'm looking for a fast envelope follower circuit
> schematic to rectify an incoming +/-2.5v AC waveform
> to a rectified 0-5v wave.. The wave frequency could
> go as low as 1 hz and sometimes will not be symetrical.
> Any circuits welcomed !
> JP
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy


From tom at electricdruid.net  Thu Mar 28 18:45:35 2013
From: tom at electricdruid.net (Tom Wiltshire)
Date: Thu, 28 Mar 2013 17:45:35 +0000
Subject: [sdiy] Fast envelope follower circuit needed..
In-Reply-To: <CA+9GZUj1AK+hmgAKQx+Zi-tau1YPOc+B112XSw67=245sZ5RXw@mail.gmail.com>
References: <324c9c6efbb87ef612fdd8cdc2ccad26@oricom.ca>
	<CA+9GZUj1AK+hmgAKQx+Zi-tau1YPOc+B112XSw67=245sZ5RXw@mail.gmail.com>
Message-ID: <DC5CD033-0328-4997-B4B7-7A9645555AE2@electricdruid.net>

If he's trying to get the envelope, he'd want to see a 100 second sine wave, wouldn't he, since that's what the volume curve looks like? Either rectified or not, depending on whether it was a 2 or 4-quadrant amplitude modulator you used.  Or am I missing something? - I can't imagine where you got square waves from.

That example wouldn't even be that difficult since you've got sufficient space (x100) between the modulation signal or envelope and the carrier, so you could filter at a very low frequency (say 0.02Hz) and recover the modulation signal.

T.

On 28 Mar 2013, at 17:19, cheater cheater <cheater00 at gmail.com> wrote:

> So what should happen when a 1Hz sinewave, amplitude-modulated by a
> 0.01 Hz sinewave, is fed into your box? Do you expect to get a 1Hz
> square wave? 2Hz square wave? 0.01 Hz square wave?
> 
> On Thu, Mar 28, 2013 at 1:44 PM, Jean-Pierre Desrochers
> <jpdesroc at oricom.ca> wrote:
>> Hi List,
>> I'm sure this has been already discussed in the past
>> but I'm looking for a fast envelope follower circuit
>> schematic to rectify an incoming +/-2.5v AC waveform
>> to a rectified 0-5v wave.. The wave frequency could
>> go as low as 1 hz and sometimes will not be symetrical.
>> Any circuits welcomed !
>> JP
>> _______________________________________________
>> Synth-diy mailing list
>> Synth-diy at dropmix.xs4all.nl
>> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy



From cheater00 at gmail.com  Thu Mar 28 19:00:15 2013
From: cheater00 at gmail.com (cheater cheater)
Date: Thu, 28 Mar 2013 19:00:15 +0100
Subject: [sdiy] Fast envelope follower circuit needed..
In-Reply-To: <DC5CD033-0328-4997-B4B7-7A9645555AE2@electricdruid.net>
References: <324c9c6efbb87ef612fdd8cdc2ccad26@oricom.ca>
	<CA+9GZUj1AK+hmgAKQx+Zi-tau1YPOc+B112XSw67=245sZ5RXw@mail.gmail.com>
	<DC5CD033-0328-4997-B4B7-7A9645555AE2@electricdruid.net>
Message-ID: <CA+9GZUhK+TbGQEM=ZOW0v27WSMqd-84-LuEf+0JLA-X=KFzuWA@mail.gmail.com>

On Thu, Mar 28, 2013 at 6:45 PM, Tom Wiltshire <tom at electricdruid.net> wrote:
> If he's trying to get the envelope, he'd want to see a 100 second sine wave, wouldn't he, since that's what the volume curve looks like?

That's the question: what is his definition of volume curve. Or let's
even start with: definition of volume.

There are many different ways to approach this subject.

> Either rectified or not, depending on whether it was a 2 or 4-quadrant amplitude modulator you used.  Or am I missing something? - I can't imagine where you got square waves from.

I read "to a rectified 0-5v wave" and interpreted that as it possibly
being a logic signal so I asked to clarify and possibly strike off
that idea.

> That example wouldn't even be that difficult since you've got sufficient space (x100) between the modulation signal or envelope and the carrier, so you could filter at a very low frequency (say 0.02Hz) and recover the modulation signal.

Yes. In fact, I am thinking that the best thing to simply have a
normal LPF-based envelope follower with adjustable cutoff.

You could have CV input for the cutoff which would react to the
frequency of the input. A simple way to use this would be to use a
frequency counter. A more involved way would be to have a vocoding
pitch detector. That's a vocoding bank, where each band outputs the
current level, and the levels get compared and based on that, the
final detected frequency is computed. This can be done completely in
analog, without any sampling or counting or PLL. I have never seen a
realization of this idea in hardware, though.

Cheers,
D.

> On 28 Mar 2013, at 17:19, cheater cheater <cheater00 at gmail.com> wrote:
>
>> So what should happen when a 1Hz sinewave, amplitude-modulated by a
>> 0.01 Hz sinewave, is fed into your box? Do you expect to get a 1Hz
>> square wave? 2Hz square wave? 0.01 Hz square wave?
>>
>> On Thu, Mar 28, 2013 at 1:44 PM, Jean-Pierre Desrochers
>> <jpdesroc at oricom.ca> wrote:
>>> Hi List,
>>> I'm sure this has been already discussed in the past
>>> but I'm looking for a fast envelope follower circuit
>>> schematic to rectify an incoming +/-2.5v AC waveform
>>> to a rectified 0-5v wave.. The wave frequency could
>>> go as low as 1 hz and sometimes will not be symetrical.
>>> Any circuits welcomed !
>>> JP
>>> _______________________________________________
>>> Synth-diy mailing list
>>> Synth-diy at dropmix.xs4all.nl
>>> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>> _______________________________________________
>> Synth-diy mailing list
>> Synth-diy at dropmix.xs4all.nl
>> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>


From tom at electricdruid.net  Thu Mar 28 19:24:19 2013
From: tom at electricdruid.net (Tom Wiltshire)
Date: Thu, 28 Mar 2013 18:24:19 +0000
Subject: [sdiy] Fast envelope follower circuit needed..
In-Reply-To: <BLU0-SMTP338D2FA8904A3F6D5D401C9C8D20@phx.gbl>
References: <324c9c6efbb87ef612fdd8cdc2ccad26@oricom.ca>
	<7D00642A-814D-4BBC-B510-CDAC1630728D@electricdruid.net>
	<BLU0-SMTP338D2FA8904A3F6D5D401C9C8D20@phx.gbl>
Message-ID: <541FA361-13D0-4D41-8296-E24F2DB98CB4@electricdruid.net>

Hi Dan,

On 28 Mar 2013, at 13:55, Dan Snazelle <subjectivity at hotmail.com> wrote:

> 
> Though i work with c//arduino/AVRS not pics, this sounds very appealing! Can anyone recommend a place to learn about writing C code for an envelope follower? How do you do it? Shift signal to 0-5, then......
> 
> 
> Thanks!!!

The first thing after that would be to do the full wave rectification. Either dump the sign bit, if you've got signed data, or invert the data if the highest bit is clear if it's not signed - same thing, different representation. Half-wave rectification just makes the signal more "lumpy" and isn't significantly easier digitally anyway.

For the smoothing, I can imagine a couple of approaches:

1) Genuine honest-to-goodness digital filtering.
There's a ton of stuff about this all over the web. But since we're talking about bottom-end chips, most of it is totally out of reach. We're not going to be running long FIR filters. But we could perhaps use an IIR filter, or a FIR with a very low sample rate, to keep it manageable.

2) Peak detection.
This might be a good way to go. Keeping track of what the last couple of samples are would be easy, and would allow us to work out the current slope of the waveform. When the current sample is less than the previous one, the waveform has started heading back down, so we'd hold the peak value.
Obviously we can't hold the highest value for ever, so there needs to be some "decay" parameter that allows it to fall. Multiply by 0.99 every sample, perhaps, or remove some small constant. This is good because the algorithm can respond instantly to peaks, but will smoothly and slowly fall for decays, which naturally tend to be longer. As ever, it won't do everything well - the envelope of short percussion hits might get stretched, for example.

There's probably other ways, or combinations of one and the other that would work. The more specific you can be about what inputs you're trying to deal with, the better a job you'll manage to do. If you want to solve the general case, you'll need to make more compromises.

Years ago I did quite a bit of work on autowah circuits, and the envelope followers for those can get quite specific since you know that the signal is coming from a guitar, and therefore what the lowest note and highest note you have to deal with are. But then your funky stomp box doesn't sound so great when your keys player puts his fender rhodes through it - you can't win, or you can't win all the time, anyway. Have you done an autowah for Snazzy FX? Maybe you've found the same thing.

HTH,
Tom



From cheater00 at gmail.com  Thu Mar 28 21:49:25 2013
From: cheater00 at gmail.com (cheater cheater)
Date: Thu, 28 Mar 2013 21:49:25 +0100
Subject: [sdiy] Fast envelope follower circuit needed..
In-Reply-To: <541FA361-13D0-4D41-8296-E24F2DB98CB4@electricdruid.net>
References: <324c9c6efbb87ef612fdd8cdc2ccad26@oricom.ca>
	<7D00642A-814D-4BBC-B510-CDAC1630728D@electricdruid.net>
	<BLU0-SMTP338D2FA8904A3F6D5D401C9C8D20@phx.gbl>
	<541FA361-13D0-4D41-8296-E24F2DB98CB4@electricdruid.net>
Message-ID: <CA+9GZUjfFaNFM1HnK51SwcgncCm2NmkTCrexeq3rDxfDSraiZQ@mail.gmail.com>

Hi Tom,
a few comments:

On Thu, Mar 28, 2013 at 7:24 PM, Tom Wiltshire <tom at electricdruid.net> wrote:
> Hi Dan,
>
> On 28 Mar 2013, at 13:55, Dan Snazelle <subjectivity at hotmail.com> wrote:
>
>>
>> Though i work with c//arduino/AVRS not pics, this sounds very appealing! Can anyone recommend a place to learn about writing C code for an envelope follower? How do you do it? Shift signal to 0-5, then......
>>
>>
>> Thanks!!!
>
> The first thing after that would be to do the full wave rectification. Either dump the sign bit, if you've got signed data, or invert the data if the highest bit is clear if it's not signed - same thing, different representation.

I strongly advise against doing this. It will create insane amounts of
aliasing. Do the rectification in analog. In fact full-wave
rectification is the most difficult thing to do when modelling analog
circuits digitally. Doing it anywhere near a level acceptable even for
the non-discriminating musician would require processing power orders
of magnitude larger than what a lowly pic can offer you. And in analog
it's just a few junctions.

BTW, you can't really rectify a signal that's unipolar already.. just
a minor nitpick ;)

> For the smoothing, I can imagine a couple of approaches:
>
> 1) Genuine honest-to-goodness digital filtering.
> There's a ton of stuff about this all over the web. But since we're talking about bottom-end chips, most of it is totally out of reach. We're not going to be running long FIR filters. But we could perhaps use an IIR filter, or a FIR with a very low sample rate, to keep it manageable.

You can't change the sample rate, unless you mean you would downsample
the signal first. But then you'll have to have steep bandwidth
limiting filters and a filter for the down-sampling itself, which
means you do ten times the work you think you're saving by using lower
sampling rates. If you don't downsample you get aliasing, and then
you're detecting the envelope very incorrectly. It can only get you
fairly small speed-ups anyways before it stops being useful at all.

An FIR with a fairly short impulse is fairly easy to manage, and on
something like a dspic you can do even longer ones. However IIR is the
usual way to go with envelope detectors if I'm not mistaken.

> 2) Peak detection.
> This might be a good way to go. Keeping track of what the last couple of samples are would be easy, and would allow us to work out the current slope of the waveform. When the current sample is less than the previous one, the waveform has started heading back down, so we'd hold the peak value.
> Obviously we can't hold the highest value for ever, so there needs to be some "decay" parameter that allows it to fall. Multiply by 0.99 every sample, perhaps, or remove some small constant. This is good because the algorithm can respond instantly to peaks, but will smoothly and slowly fall for decays, which naturally tend to be longer. As ever, it won't do everything well - the envelope of short percussion hits might get stretched, for example.

Depending on how one would fill in the blanks, you've just described
either IIR, or FIR, or a thing that results in loads of zipper noise
and incorrect enveloping. IIR and FIR really are the best approaches
here.

> There's probably other ways, or combinations of one and the other that would work. The more specific you can be about what inputs you're trying to deal with, the better a job you'll manage to do. If you want to solve the general case, you'll need to make more compromises.

Or add knobs :-)

Cheers,
D.


From subjectivity at hotmail.com  Thu Mar 28 23:02:20 2013
From: subjectivity at hotmail.com (Dan Snazelle)
Date: Thu, 28 Mar 2013 18:02:20 -0400
Subject: [sdiy] Fast envelope follower circuit needed..
In-Reply-To: <541FA361-13D0-4D41-8296-E24F2DB98CB4@electricdruid.net>
References: <324c9c6efbb87ef612fdd8cdc2ccad26@oricom.ca>
	<7D00642A-814D-4BBC-B510-CDAC1630728D@electricdruid.net>
	<BLU0-SMTP338D2FA8904A3F6D5D401C9C8D20@phx.gbl>
	<541FA361-13D0-4D41-8296-E24F2DB98CB4@electricdruid.net>
Message-ID: <BLU0-SMTP202122B4C79A5F00D7154F6C8D20@phx.gbl>



On Mar 28, 2013, at 2:24 PM, Tom Wiltshire <tom at electricdruid.net> wrote:

> Hi Dan,
> 
> On 28 Mar 2013, at 13:55, Dan Snazelle <subjectivity at hotmail.com> wrote:
> 
>> 
>> Though i work with c//arduino/AVRS not pics, this sounds very appealing! Can anyone recommend a place to learn about writing C code for an envelope follower? How do you do it? Shift signal to 0-5, then......
>> 
>> 
>> Thanks!!!
> 
> The first thing after that would be to do the full wave rectification. Either dump the sign bit, if you've got signed data, or invert the data if the highest bit is clear if it's not signed - same thing, different representation. Half-wave rectification just makes the signal more "lumpy" and isn't significantly easier digitally anyway.
> 
> For the smoothing, I can imagine a couple of approaches:
> 
> 1) Genuine honest-to-goodness digital filtering.
> There's a ton of stuff about this all over the web. But since we're talking about bottom-end chips, most of it is totally out of reach. We're not going to be running long FIR filters. But we could perhaps use an IIR filter, or a FIR with a very low sample rate, to keep it manageable.
> 
> 2) Peak detection.
> This might be a good way to go. Keeping track of what the last couple of samples are would be easy, and would allow us to work out the current slope of the waveform. When the current sample is less than the previous one, the waveform has started heading back down, so we'd hold the peak value.
> Obviously we can't hold the highest value for ever, so there needs to be some "decay" parameter that allows it to fall. Multiply by 0.99 every sample, perhaps, or remove some small constant. This is good because the algorithm can respond instantly to peaks, but will smoothly and slowly fall for decays, which naturally tend to be longer. As ever, it won't do everything well - the envelope of short percussion hits might get stretched, for example.
> 
> There's probably other ways, or combinations of one and the other that would work. The more specific you can be about what inputs you're trying to deal with, the better a job you'll manage to do. If you want to solve the general case, you'll need to make more compromises.
> 
> Years ago I did quite a bit of work on autowah circuits, and the envelope followers for those can get quite specific since you know that the signal is coming from a guitar, and therefore what the lowest note and highest note you have to deal with are. But then your funky stomp box doesn't sound so great when your keys player puts his fender rhodes through it - you can't win, or you can't win all the time, anyway. Have you done an autowah for Snazzy FX? Maybe you've found the same thing.
> 
> HTH,
> Tom
> 
> 
Thanks for the great reply!!
I will have to investigate with the AVR.


Yes a few Snazzy Fx products have analog followers in them.

Depending on your input, a follower can be perfect for one source but not another. 

I worked very hard on the follower circuit in the Wow and Flutter euro module which is coming out this week. 

The original Wow and Flutter was for guitar, the module is for synth, so I had to re-work everything.



From rburnett at richieburnett.co.uk  Thu Mar 28 23:22:02 2013
From: rburnett at richieburnett.co.uk (Richie Burnett)
Date: Thu, 28 Mar 2013 22:22:02 +0000
Subject: [sdiy] Fast envelope follower circuit needed..
Message-ID: <521iy28t6vwmd41f0pc7ymu0.1364508615243@email.android.com>

The Hilbert transform is a classic tool for envelope extraction using DSP. Envelope following is really what electrical engineers would call AM demodulation, and has existed in various forms since the dawn of radio.

Also, FIR filters with cutoff frequencies which are very low compared to the sample rate and result in really long impulse responses (hundreds or thousands of taps) can be used effectively if implemented as polyphase filters. A 2000 tap FIR filter only requires the same amount of DSP grunt as a 20 tap filter if you can drop the sample rate of the output by a factor of 1/10.

The CIC filter is also a very common filter in communications applications where a high frequency signal needs to be demodulated and the envelope is output at a reduced sample rate.

-Richie,



Sent from my Sony Ericsson Xperia ray

cheater cheater <cheater00 at gmail.com> wrote:

>Hi Tom,
>a few comments:
>
>On Thu, Mar 28, 2013 at 7:24 PM, Tom Wiltshire <tom at electricdruid.net> wrote:
>> Hi Dan,
>>
>> On 28 Mar 2013, at 13:55, Dan Snazelle <subjectivity at hotmail.com> wrote:
>>
>>>
>>> Though i work with c//arduino/AVRS not pics, this sounds very appealing! Can anyone recommend a place to learn about writing C code for an envelope follower? How do you do it? Shift signal to 0-5, then......
>>>
>>>
>>> Thanks!!!
>>
>> The first thing after that would be to do the full wave rectification. Either dump the sign bit, if you've got signed data, or invert the data if the highest bit is clear if it's not signed - same thing, different representation.
>
>I strongly advise against doing this. It will create insane amounts of
>aliasing. Do the rectification in analog. In fact full-wave
>rectification is the most difficult thing to do when modelling analog
>circuits digitally. Doing it anywhere near a level acceptable even for
>the non-discriminating musician would require processing power orders
>of magnitude larger than what a lowly pic can offer you. And in analog
>it's just a few junctions.
>
>BTW, you can't really rectify a signal that's unipolar already.. just
>a minor nitpick ;)
>
>> For the smoothing, I can imagine a couple of approaches:
>>
>> 1) Genuine honest-to-goodness digital filtering.
>> There's a ton of stuff about this all over the web. But since we're talking about bottom-end chips, most of it is totally out of reach. We're not going to be running long FIR filters. But we could perhaps use an IIR filter, or a FIR with a very low sample rate, to keep it manageable.
>
>You can't change the sample rate, unless you mean you would downsample
>the signal first. But then you'll have to have steep bandwidth
>limiting filters and a filter for the down-sampling itself, which
>means you do ten times the work you think you're saving by using lower
>sampling rates. If you don't downsample you get aliasing, and then
>you're detecting the envelope very incorrectly. It can only get you
>fairly small speed-ups anyways before it stops being useful at all.
>
>An FIR with a fairly short impulse is fairly easy to manage, and on
>something like a dspic you can do even longer ones. However IIR is the
>usual way to go with envelope detectors if I'm not mistaken.
>
>> 2) Peak detection.
>> This might be a good way to go. Keeping track of what the last couple of samples are would be easy, and would allow us to work out the current slope of the waveform. When the current sample is less than the previous one, the waveform has started heading back down, so we'd hold the peak value.
>> Obviously we can't hold the highest value for ever, so there needs to be some "decay" parameter that allows it to fall. Multiply by 0.99 every sample, perhaps, or remove some small constant. This is good because the algorithm can respond instantly to peaks, but will smoothly and slowly fall for decays, which naturally tend to be longer. As ever, it won't do everything well - the envelope of short percussion hits might get stretched, for example.
>
>Depending on how one would fill in the blanks, you've just described
>either IIR, or FIR, or a thing that results in loads of zipper noise
>and incorrect enveloping. IIR and FIR really are the best approaches
>here.
>
>> There's probably other ways, or combinations of one and the other that would work. The more specific you can be about what inputs you're trying to deal with, the better a job you'll manage to do. If you want to solve the general case, you'll need to make more compromises.
>
>Or add knobs :-)
>
>Cheers,
>D.
>_______________________________________________
>Synth-diy mailing list
>Synth-diy at dropmix.xs4all.nl
>http://dropmix.xs4all.nl/mailman/listinfo/synth-diy

From rburnett at richieburnett.co.uk  Fri Mar 29 00:16:24 2013
From: rburnett at richieburnett.co.uk (Richie Burnett)
Date: Thu, 28 Mar 2013 23:16:24 -0000
Subject: [sdiy] Fast envelope follower circuit needed..
References: <521iy28t6vwmd41f0pc7ymu0.1364508615243@email.android.com>
Message-ID: <575460E7A1FC43DA928C9C86AD8BF7FE@homepc002>

> A 2000 tap FIR filter only requires the same amount of DSP grunt as a 20 
> tap filter if you can drop the sample rate of the output by a factor of 
> 1/10.

Sorry, that should have said ...if you can drop the sample rate of the 
output by a factor of 1/100.

-Richie, 



From tim.parkhurst at gmail.com  Fri Mar 29 00:42:43 2013
From: tim.parkhurst at gmail.com (Tim Parkhurst)
Date: Thu, 28 Mar 2013 16:42:43 -0700
Subject: [sdiy] Fast envelope follower circuit needed..
In-Reply-To: <324c9c6efbb87ef612fdd8cdc2ccad26@oricom.ca>
References: <324c9c6efbb87ef612fdd8cdc2ccad26@oricom.ca>
Message-ID: <CACDGQT8=2x2zQ1w+kDg=MkVhd66KODc4jcsVKtnfVVVO1ii0Tg@mail.gmail.com>

I'm pretty sure a digital envelope follower won't sound as fat as an
analog one. And if it's surface mount, forget it. It will never have
that pleasing ninth-harmonic distortion that makes it 'sit it the mix'
just right.

On the other hand, if you're looking for inspiration from an
interesting circuit design, you might try this one from our very own
Harry Bissell
http://www.edn.com/design/analog/4344656/Envelope-follower-combines-fast-response-low-ripple

This one uses three peak detector circuits with slightly different
time constants. Could be useful, even just as inspiration for a
digital method.

Tim (sits in the mix to add more bottom end) Servo
---
"Sire, the church of God is an anvil that has worn out many hammers."
- H.L. Hastings


On Thu, Mar 28, 2013 at 5:44 AM, Jean-Pierre Desrochers
<jpdesroc at oricom.ca> wrote:
> Hi List,
> I'm sure this has been already discussed in the past
> but I'm looking for a fast envelope follower circuit
> schematic to rectify an incoming +/-2.5v AC waveform
> to a rectified 0-5v wave.. The wave frequency could
> go as low as 1 hz and sometimes will not be symetrical.
> Any circuits welcomed !
> JP
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy


From dlmanley at sonic.net  Fri Mar 29 01:11:35 2013
From: dlmanley at sonic.net (Dave Manley)
Date: Thu, 28 Mar 2013 17:11:35 -0700
Subject: [sdiy] Fast envelope follower circuit needed..
Message-ID: <bpng4b8lqkpky6jr93xneepf.1364515895087@email.android.com>

Go cheap and old school: series diode into parallel RC (look up AM detector).  Buffer output as needed. Replace R with fet or vactrol.  It might have character you wont find in a precision circuit.

It should at least be ' fast' to build (and discard?).  ;-)

-------- Original Message --------
 From: Tim Parkhurst <tim.parkhurst at gmail.com>
 Sent: Thu, Mar 28, 2013 04:42 PM
 To: Jean-Pierre Desrochers <jpdesroc at oricom.ca>
 CC: Synth-DIY <synth-diy at dropmix.xs4all.nl>
 Subject: Re: [sdiy] Fast envelope follower circuit needed..

>I'm pretty sure a digital envelope follower won't sound as fat as an
>analog one. And if it's surface mount, forget it. It will never have
>that pleasing ninth-harmonic distortion that makes it 'sit it the mix'
>just right.
>
>On the other hand, if you're looking for inspiration from an
>interesting circuit design, you might try this one from our very own
>Harry Bissell
>http://www.edn.com/design/analog/4344656/Envelope-follower-combines-fast-response-low-ripple
>
>This one uses three peak detector circuits with slightly different
>time constants. Could be useful, even just as inspiration for a
>digital method.
>
>Tim (sits in the mix to add more bottom end) Servo
>---
>"Sire, the church of God is an anvil that has worn out many hammers."
>- H.L. Hastings
>
>
>On Thu, Mar 28, 2013 at 5:44 AM, Jean-Pierre Desrochers
><jpdesroc at oricom.ca> wrote:
>> Hi List,
>> I'm sure this has been already discussed in the past
>> but I'm looking for a fast envelope follower circuit
>> schematic to rectify an incoming +/-2.5v AC waveform
>> to a rectified 0-5v wave.. The wave frequency could
>> go as low as 1 hz and sometimes will not be symetrical.
>> Any circuits welcomed !
>> JP
>> _______________________________________________
>> Synth-diy mailing list
>> Synth-diy at dropmix.xs4all.nl
>> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>_______________________________________________
>Synth-diy mailing list
>Synth-diy at dropmix.xs4all.nl
>http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>

From rsdio at sounds.wa.com  Fri Mar 29 01:14:15 2013
From: rsdio at sounds.wa.com (rsdio at sounds.wa.com)
Date: Thu, 28 Mar 2013 17:14:15 -0700
Subject: [sdiy] Fast envelope follower circuit needed..
In-Reply-To: <541FA361-13D0-4D41-8296-E24F2DB98CB4@electricdruid.net>
References: <324c9c6efbb87ef612fdd8cdc2ccad26@oricom.ca>
	<7D00642A-814D-4BBC-B510-CDAC1630728D@electricdruid.net>
	<BLU0-SMTP338D2FA8904A3F6D5D401C9C8D20@phx.gbl>
	<541FA361-13D0-4D41-8296-E24F2DB98CB4@electricdruid.net>
Message-ID: <309EB6A9-742D-4A9D-8099-69D08A50A07C@sounds.wa.com>


On Mar 28, 2013, at 11:24, Tom Wiltshire wrote:
> Hi Dan,
>
> On 28 Mar 2013, at 13:55, Dan Snazelle <subjectivity at hotmail.com>  
> wrote:
>> Though i work with c//arduino/AVRS not pics, this sounds very  
>> appealing! Can anyone recommend a place to learn about writing C  
>> code for an envelope follower? How do you do it? Shift signal to  
>> 0-5, then......
>
> The first thing after that would be to do the full wave  
> rectification. Either dump the sign bit, if you've got signed data,  
> or invert the data if the highest bit is clear if it's not signed -  
> same thing, different representation. Half-wave rectification just  
> makes the signal more "lumpy" and isn't significantly easier  
> digitally anyway.


You can't dump the sign bit, at least not with twos-complement data,  
which is what all digital signed data is. Maybe you're thinking of  
conceptually dumping the negative symbol, but that's not the same as  
dumping the sign bit. If you were to dump the sign bit, then the  
negative part of the signal would not be inverted, as needed, but  
would merely be shifted up into the positive range. That would make  
an ugly, spiked signal that would introduce significant errors in  
your envelope.

The easiest thing to do with signed data is use abs() to get the  
absolute value. Many 8-bit processors even have a native opcode to  
compute abs(), but they're not merely masking out the sign bit as  
would be done by and'ing with 0x7F.

You are correct that inverting the data below the midpoint will work  
with unsigned data.

You can probably use almost exactly the same code for either signed  
or unsigned.

Brian Willoughby
Sound Consulting



From tom at electricdruid.net  Fri Mar 29 01:39:12 2013
From: tom at electricdruid.net (Tom Wiltshire)
Date: Fri, 29 Mar 2013 00:39:12 +0000
Subject: [sdiy] Fast envelope follower circuit needed..
In-Reply-To: <309EB6A9-742D-4A9D-8099-69D08A50A07C@sounds.wa.com>
References: <324c9c6efbb87ef612fdd8cdc2ccad26@oricom.ca>
	<7D00642A-814D-4BBC-B510-CDAC1630728D@electricdruid.net>
	<BLU0-SMTP338D2FA8904A3F6D5D401C9C8D20@phx.gbl>
	<541FA361-13D0-4D41-8296-E24F2DB98CB4@electricdruid.net>
	<309EB6A9-742D-4A9D-8099-69D08A50A07C@sounds.wa.com>
Message-ID: <4F616975-1B30-4B10-A377-BD6CF0C6C23B@electricdruid.net>


On 29 Mar 2013, at 00:14, rsdio at sounds.wa.com wrote:

> 
> On Mar 28, 2013, at 11:24, Tom Wiltshire wrote:
>> Hi Dan,
>> 
>> On 28 Mar 2013, at 13:55, Dan Snazelle <subjectivity at hotmail.com> wrote:
>>> Though i work with c//arduino/AVRS not pics, this sounds very appealing! Can anyone recommend a place to learn about writing C code for an envelope follower? How do you do it? Shift signal to 0-5, then......
>> 
>> The first thing after that would be to do the full wave rectification. Either dump the sign bit, if you've got signed data, or invert the data if the highest bit is clear if it's not signed - same thing, different representation. Half-wave rectification just makes the signal more "lumpy" and isn't significantly easier digitally anyway.
> 
> 
> You can't dump the sign bit, at least not with twos-complement data, which is what all digital signed data is. Maybe you're thinking of conceptually dumping the negative symbol, but that's not the same as dumping the sign bit. If you were to dump the sign bit, then the negative part of the signal would not be inverted, as needed, but would merely be shifted up into the positive range. That would make an ugly, spiked signal that would introduce significant errors in your envelope.

No, you're quite right. My mistake. You need to invert the byte if the sign bit is set, otherwise as you say, the negative part is just shifted up, rather than flipped over.

In my defence, I've actually been working on a random generator algorithm the last few days which *does* just dump the sign bit to produce noise in the output range 0-5V rather than -5 to +5V. Since it's noise it doesn't matter. Hence the confusion.

T.



From dlmanley at sonic.net  Fri Mar 29 01:43:19 2013
From: dlmanley at sonic.net (Dave Manley)
Date: Thu, 28 Mar 2013 17:43:19 -0700
Subject: [sdiy] Fast envelope follower circuit needed..
Message-ID: <m8hqxhp8da12813qt1286fj8.1364517799724@email.android.com>

You could also discard all negative samples (half wave).  The output goes thru LPF smoothing anyway,  why bother with full wave?

-------- Original Message --------
 From: rsdio at sounds.wa.com
 Sent: Thu, Mar 28, 2013 05:14 PM
 To: Synth-Diy List <synth-diy at dropmix.xs4all.nl>
 CC: 
 Subject: Re: [sdiy] Fast envelope follower circuit needed..

>
>On Mar 28, 2013, at 11:24, Tom Wiltshire wrote:
>> Hi Dan,
>>
>> On 28 Mar 2013, at 13:55, Dan Snazelle <subjectivity at hotmail.com>  
>> wrote:
>>> Though i work with c//arduino/AVRS not pics, this sounds very  
>>> appealing! Can anyone recommend a place to learn about writing C  
>>> code for an envelope follower? How do you do it? Shift signal to  
>>> 0-5, then......
>>
>> The first thing after that would be to do the full wave  
>> rectification. Either dump the sign bit, if you've got signed data,  
>> or invert the data if the highest bit is clear if it's not signed -  
>> same thing, different representation. Half-wave rectification just  
>> makes the signal more "lumpy" and isn't significantly easier  
>> digitally anyway.
>
>
>You can't dump the sign bit, at least not with twos-complement data,  
>which is what all digital signed data is. Maybe you're thinking of  
>conceptually dumping the negative symbol, but that's not the same as  
>dumping the sign bit. If you were to dump the sign bit, then the  
>negative part of the signal would not be inverted, as needed, but  
>would merely be shifted up into the positive range. That would make  
>an ugly, spiked signal that would introduce significant errors in  
>your envelope.
>
>The easiest thing to do with signed data is use abs() to get the  
>absolute value. Many 8-bit processors even have a native opcode to  
>compute abs(), but they're not merely masking out the sign bit as  
>would be done by and'ing with 0x7F.
>
>You are correct that inverting the data below the midpoint will work  
>with unsigned data.
>
>You can probably use almost exactly the same code for either signed  
>or unsigned.
>
>Brian Willoughby
>Sound Consulting
>
>_______________________________________________
>Synth-diy mailing list
>Synth-diy at dropmix.xs4all.nl
>http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>

From rsdio at sounds.wa.com  Fri Mar 29 02:04:21 2013
From: rsdio at sounds.wa.com (rsdio at sounds.wa.com)
Date: Thu, 28 Mar 2013 18:04:21 -0700
Subject: [sdiy] Fast envelope follower circuit needed..
In-Reply-To: <m8hqxhp8da12813qt1286fj8.1364517799724@email.android.com>
References: <m8hqxhp8da12813qt1286fj8.1364517799724@email.android.com>
Message-ID: <A9E6F624-4FA5-42D2-A04D-1A6D88B653A5@sounds.wa.com>

Yes, you can do half wave, and many analog circuits do this. It's  
mostly about saving costs of components. Software is generally free,  
though, except for the cost of cycles at run time.

However, to answer your question about "why bother," the answer is  
that half wave will put less energy into your LPF, and thus the  
output level will be less than with full wave.

In additional, the half wave version will either be less smooth, or  
it will be approximately half as fast as a full wave.

The original request asked for 5 V output, and I meant to point out  
earlier that you'd need at least 2x gain to get 5 V output from an  
envelope follower that's being fed a 5 V peak-to-peak signal. That's  
because the signal's average value is around the mid point, and thus  
the envelope peaks will be half of the peak-to-peak range.  
Theoretical maximum would be 2.5 V, and in practice it will be less  
than that with analog implementations because it's difficult to get  
the capacitor to fully charge after a diode voltage drop and the time  
constant delay. Digital implementations might be able to produce a  
peak envelope of 2.5 V for a brief impulse, even for half wave.  
There's nothing wrong with boosting this by a gain factor of 2, or  
more, if you really want 5 V envelope output.

A half wave solution would merely need additional gain to reach 5 V  
output. However, there is no solution to make the half wave as fast  
as the full wave.

Brian Willoughby
Sound Consulting


On Mar 28, 2013, at 17:43, Dave Manley wrote:
> You could also discard all negative samples (half wave).  The  
> output goes thru LPF smoothing anyway,  why bother with full wave?
>
> -------- Original Message --------
>  From: rsdio at sounds.wa.com
>> On Mar 28, 2013, at 11:24, Tom Wiltshire wrote:
>>> On 28 Mar 2013, at 13:55, Dan Snazelle <subjectivity at hotmail.com>
>>>> Though i work with c//arduino/AVRS not pics, this sounds very
>>>> appealing! Can anyone recommend a place to learn about writing C
>>>> code for an envelope follower? How do you do it? Shift signal to
>>>> 0-5, then......
>>>
>>> The first thing after that would be to do the full wave
>>> rectification. Either dump the sign bit, if you've got signed data,
>>> or invert the data if the highest bit is clear if it's not signed -
>>> same thing, different representation. Half-wave rectification just
>>> makes the signal more "lumpy" and isn't significantly easier
>>> digitally anyway.
>>
>> You can't dump the sign bit, at least not with twos-complement data,
>> which is what all digital signed data is. Maybe you're thinking of
>> conceptually dumping the negative symbol, but that's not the same as
>> dumping the sign bit. If you were to dump the sign bit, then the
>> negative part of the signal would not be inverted, as needed, but
>> would merely be shifted up into the positive range. That would make
>> an ugly, spiked signal that would introduce significant errors in
>> your envelope.
>>
>> The easiest thing to do with signed data is use abs() to get the
>> absolute value. Many 8-bit processors even have a native opcode to
>> compute abs(), but they're not merely masking out the sign bit as
>> would be done by and'ing with 0x7F.
>>
>> You are correct that inverting the data below the midpoint will work
>> with unsigned data.
>>
>> You can probably use almost exactly the same code for either signed
>> or unsigned.
>


From tom at electricdruid.net  Fri Mar 29 02:20:44 2013
From: tom at electricdruid.net (Tom Wiltshire)
Date: Fri, 29 Mar 2013 01:20:44 +0000
Subject: [sdiy] Fast envelope follower circuit needed..
In-Reply-To: <A9E6F624-4FA5-42D2-A04D-1A6D88B653A5@sounds.wa.com>
References: <m8hqxhp8da12813qt1286fj8.1364517799724@email.android.com>
	<A9E6F624-4FA5-42D2-A04D-1A6D88B653A5@sounds.wa.com>
Message-ID: <58B1455E-612C-40B4-9A16-0531D49415EE@electricdruid.net>

+1. Half-wave leaves more ripple, or needs more filtering = slower response. In this case full-wave is definitely better.

On 29 Mar 2013, at 01:04, rsdio at sounds.wa.com wrote:

> Yes, you can do half wave, and many analog circuits do this. It's mostly about saving costs of components. Software is generally free, though, except for the cost of cycles at run time.
> 
> However, to answer your question about "why bother," the answer is that half wave will put less energy into your LPF, and thus the output level will be less than with full wave.
> 
> In additional, the half wave version will either be less smooth, or it will be approximately half as fast as a full wave.
> 
> The original request asked for 5 V output, and I meant to point out earlier that you'd need at least 2x gain to get 5 V output from an envelope follower that's being fed a 5 V peak-to-peak signal. That's because the signal's average value is around the mid point, and thus the envelope peaks will be half of the peak-to-peak range. Theoretical maximum would be 2.5 V, and in practice it will be less than that with analog implementations because it's difficult to get the capacitor to fully charge after a diode voltage drop and the time constant delay. Digital implementations might be able to produce a peak envelope of 2.5 V for a brief impulse, even for half wave. There's nothing wrong with boosting this by a gain factor of 2, or more, if you really want 5 V envelope output.
> 
> A half wave solution would merely need additional gain to reach 5 V output. However, there is no solution to make the half wave as fast as the full wave.
> 
> Brian Willoughby
> Sound Consulting
> 
> 
> On Mar 28, 2013, at 17:43, Dave Manley wrote:
>> You could also discard all negative samples (half wave).  The output goes thru LPF smoothing anyway,  why bother with full wave?
>> 
>> -------- Original Message --------
>> From: rsdio at sounds.wa.com
>>> On Mar 28, 2013, at 11:24, Tom Wiltshire wrote:
>>>> On 28 Mar 2013, at 13:55, Dan Snazelle <subjectivity at hotmail.com>
>>>>> Though i work with c//arduino/AVRS not pics, this sounds very
>>>>> appealing! Can anyone recommend a place to learn about writing C
>>>>> code for an envelope follower? How do you do it? Shift signal to
>>>>> 0-5, then......
>>>> 
>>>> The first thing after that would be to do the full wave
>>>> rectification. Either dump the sign bit, if you've got signed data,
>>>> or invert the data if the highest bit is clear if it's not signed -
>>>> same thing, different representation. Half-wave rectification just
>>>> makes the signal more "lumpy" and isn't significantly easier
>>>> digitally anyway.
>>> 
>>> You can't dump the sign bit, at least not with twos-complement data,
>>> which is what all digital signed data is. Maybe you're thinking of
>>> conceptually dumping the negative symbol, but that's not the same as
>>> dumping the sign bit. If you were to dump the sign bit, then the
>>> negative part of the signal would not be inverted, as needed, but
>>> would merely be shifted up into the positive range. That would make
>>> an ugly, spiked signal that would introduce significant errors in
>>> your envelope.
>>> 
>>> The easiest thing to do with signed data is use abs() to get the
>>> absolute value. Many 8-bit processors even have a native opcode to
>>> compute abs(), but they're not merely masking out the sign bit as
>>> would be done by and'ing with 0x7F.
>>> 
>>> You are correct that inverting the data below the midpoint will work
>>> with unsigned data.
>>> 
>>> You can probably use almost exactly the same code for either signed
>>> or unsigned.
>> 
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy



From pata at ieee.org  Fri Mar 29 04:11:13 2013
From: pata at ieee.org (Takuya Katayama)
Date: Fri, 29 Mar 2013 12:11:13 +0900 (JST)
Subject: [sdiy] Fast envelope follower circuit needed..
In-Reply-To: <58B1455E-612C-40B4-9A16-0531D49415EE@electricdruid.net>
Message-ID: <523238.74752.qm@web100715.mail.kks.yahoo.co.jp>

Hi Lists,

I think true-RMS circuit is better.
See ADI's tutorial
http://www.analog.com/static/imported-files/tutorials/MT-081.pdf

ICs are little expensive, so yu can use Ring Modulator and LPF.

Taku


--- On Fri, 2013/3/29, Tom Wiltshire <tom at electricdruid.net> wrote:

> +1. Half-wave leaves more ripple, or needs more filtering = slower response. In this case full-wave is definitely better.
> 
> On 29 Mar 2013, at 01:04, rsdio at sounds.wa.com wrote:
> 
> > Yes, you can do half wave, and many analog circuits do this. It's mostly about saving costs of components. Software is generally free, though, except for the cost of cycles at run time.
> > 
> > However, to answer your question about "why bother," the answer is that half wave will put less energy into your LPF, and thus the output level will be less than with full wave.
> > 
> > In additional, the half wave version will either be less smooth, or it will be approximately half as fast as a full wave.
> > 
> > The original request asked for 5 V output, and I meant to point out earlier that you'd need at least 2x gain to get 5 V output from an envelope follower that's being fed a 5 V peak-to-peak signal. That's because the signal's average value is around the mid point, and thus the envelope peaks will be half of the peak-to-peak range. Theoretical maximum would be 2.5 V, and in practice it will be less than that with analog implementations because it's difficult to get the capacitor to fully charge after a diode voltage drop and the time constant delay. Digital implementations might be able to produce a peak envelope of 2.5 V for a brief impulse, even for half wave. There's nothing wrong with boosting this by a gain factor of 2, or more, if you really want 5 V envelope output.
> > 
> > A half wave solution would merely need additional gain to reach 5 V output. However, there is no solution to make the half wave as fast as the full wave.
> > 
> > Brian Willoughby
> > Sound Consulting
> > 
> > 
> > On Mar 28, 2013, at 17:43, Dave Manley wrote:
> >> You could also discard all negative samples (half wave).? The output goes thru LPF smoothing anyway,? why bother with full wave?
> >> 
> >> -------- Original Message --------
> >> From: rsdio at sounds.wa.com
> >>> On Mar 28, 2013, at 11:24, Tom Wiltshire wrote:
> >>>> On 28 Mar 2013, at 13:55, Dan Snazelle <subjectivity at hotmail.com>
> >>>>> Though i work with c//arduino/AVRS not pics, this sounds very
> >>>>> appealing! Can anyone recommend a place to learn about writing C
> >>>>> code for an envelope follower? How do you do it? Shift signal to
> >>>>> 0-5, then......
> >>>> 
> >>>> The first thing after that would be to do the full wave
> >>>> rectification. Either dump the sign bit, if you've got signed data,
> >>>> or invert the data if the highest bit is clear if it's not signed -
> >>>> same thing, different representation. Half-wave rectification just
> >>>> makes the signal more "lumpy" and isn't significantly easier
> >>>> digitally anyway.
> >>> 
> >>> You can't dump the sign bit, at least not with twos-complement data,
> >>> which is what all digital signed data is. Maybe you're thinking of
> >>> conceptually dumping the negative symbol, but that's not the same as
> >>> dumping the sign bit. If you were to dump the sign bit, then the
> >>> negative part of the signal would not be inverted, as needed, but
> >>> would merely be shifted up into the positive range. That would make
> >>> an ugly, spiked signal that would introduce significant errors in
> >>> your envelope.
> >>> 
> >>> The easiest thing to do with signed data is use abs() to get the
> >>> absolute value. Many 8-bit processors even have a native opcode to
> >>> compute abs(), but they're not merely masking out the sign bit as
> >>> would be done by and'ing with 0x7F.
> >>> 
> >>> You are correct that inverting the data below the midpoint will work
> >>> with unsigned data.
> >>> 
> >>> You can probably use almost exactly the same code for either signed
> >>> or unsigned.
> >> 
> > _______________________________________________
> > Synth-diy mailing list
> > Synth-diy at dropmix.xs4all.nl
> > http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
> 
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
> 


From rsdio at sounds.wa.com  Fri Mar 29 07:17:00 2013
From: rsdio at sounds.wa.com (rsdio at sounds.wa.com)
Date: Thu, 28 Mar 2013 23:17:00 -0700
Subject: [sdiy] Fast envelope follower circuit needed..
In-Reply-To: <523238.74752.qm@web100715.mail.kks.yahoo.co.jp>
References: <523238.74752.qm@web100715.mail.kks.yahoo.co.jp>
Message-ID: <2B7685A3-0924-4E42-9946-B6537DEE00EF@sounds.wa.com>

I disagree.

RMS and True RMS are decidedly not going to give you the envelope of  
the signal. By definition, not even true RMS will give you the peaks  
of the signal.

When designing a power supply, or measuring the power delivered by,  
e.g., 60 Hz 120 VAC, true RMS is a laudable goal. But the assumptions  
there are that all you care about is the power delivered, and not the  
peaks or the envelope of the signal. One assumes that AC power does  
not have a changing envelope.

As D. cheater pointed out, what really matters is the definition of  
"fast" and the definition of "envelope" that Jean-Pierre is looking  
for. Unless true RMS is part of his goal, then a true RMS circuit is  
vastly incorrect. For one thing, true RMS will certainly not deliver  
a rectified 0-5 V wave from a +/-2.5 VAC waveform.

Brian Willoughby
Sound Consulting

p.s. that tutorial sure is informative, though!


On Mar 28, 2013, at 20:11, Takuya Katayama wrote:
> I think true-RMS circuit is better.
> See ADI's tutorial
> http://www.analog.com/static/imported-files/tutorials/MT-081.pdf
>
> ICs are little expensive, so yu can use Ring Modulator and LPF.
>
> Taku




From pata at ieee.org  Fri Mar 29 07:35:53 2013
From: pata at ieee.org (Takuya Katayama)
Date: Fri, 29 Mar 2013 15:35:53 +0900 (JST)
Subject: [sdiy] Fast envelope follower circuit needed..
In-Reply-To: <2B7685A3-0924-4E42-9946-B6537DEE00EF@sounds.wa.com>
Message-ID: <671747.50288.qm@web100716.mail.kks.yahoo.co.jp>

Hi Brian and the Lists,

Oh, I see what you say and I understand problem in my argument.
Thank you.

Then, does a Sampling Amp fit to the requirement?
Sample at few kHz sampling clock, just like sampling clock is amplitude modulated by original signal's envelope.
Cut off freq. of LPF can be one-third of the sampling clock so "fast" enough to a few Hz or sub-Hz signal.

Taku

--- On Fri, 2013/3/29, rsdio at sounds.wa.com <rsdio at sounds.wa.com> wrote:

> I disagree.
> 
> RMS and True RMS are decidedly not going to give you the envelope of the signal. By definition, not even true RMS will give you the peaks of the signal.
> 
> When designing a power supply, or measuring the power delivered by, e.g., 60 Hz 120 VAC, true RMS is a laudable goal. But the assumptions there are that all you care about is the power delivered, and not the peaks or the envelope of the signal. One assumes that AC power does not have a changing envelope.
> 
> As D. cheater pointed out, what really matters is the definition of "fast" and the definition of "envelope" that Jean-Pierre is looking for. Unless true RMS is part of his goal, then a true RMS circuit is vastly incorrect. For one thing, true RMS will certainly not deliver a rectified 0-5 V wave from a +/-2.5 VAC waveform.
> 
> Brian Willoughby
> Sound Consulting
> 
> p.s. that tutorial sure is informative, though!
> 
> 
> On Mar 28, 2013, at 20:11, Takuya Katayama wrote:
> > I think true-RMS circuit is better.
> > See ADI's tutorial
> > http://www.analog.com/static/imported-files/tutorials/MT-081.pdf
> > 
> > ICs are little expensive, so yu can use Ring Modulator and LPF.
> > 
> > Taku
> 
> 
> 


From cheater00 at gmail.com  Fri Mar 29 10:03:59 2013
From: cheater00 at gmail.com (cheater cheater)
Date: Fri, 29 Mar 2013 10:03:59 +0100
Subject: [sdiy] Fast envelope follower circuit needed..
In-Reply-To: <521iy28t6vwmd41f0pc7ymu0.1364508615243@email.android.com>
References: <521iy28t6vwmd41f0pc7ymu0.1364508615243@email.android.com>
Message-ID: <CA+9GZUgOP-ttuVHT4kNAy6Y1YV7o4yREpVeG+2kNWKyEJW4BBQ@mail.gmail.com>

Richie,

On Thu, Mar 28, 2013 at 11:22 PM, Richie Burnett
<rburnett at richieburnett.co.uk> wrote:
> The Hilbert transform is a classic tool for envelope extraction using DSP. Envelope following is really what electrical engineers would call AM demodulation, and has existed in various forms since the dawn of radio.

That's interesting - where can I read up on how this is proven
mathematically? I've never heard of this property of the Hilbert
transform. Really interesting to me. And in fact, very surprising,
since the Hilbert transform is kind of like an all-pass filter.. a
chorus you might say. Just a bit of conceptual difference and what you
get is completely different!

> Also, FIR filters with cutoff frequencies which are very low compared to the sample rate and result in really long impulse responses (hundreds or thousands of taps) can be used effectively if implemented as polyphase filters. A 2000 tap FIR filter only requires the same amount of DSP grunt as a 20 tap filter if you can drop the sample rate of the output by a factor of 1/100.

Very smart. I was sure you could do the resampling and enveloping in
one step - but I wasn't sure if I should recommend it, because I've
never done this in practice. Thanks for clearing that up!

> cheater cheater <cheater00 at gmail.com> wrote:
>
>>Hi Tom,
>>a few comments:
>>
>>On Thu, Mar 28, 2013 at 7:24 PM, Tom Wiltshire <tom at electricdruid.net> wrote:
>>> Hi Dan,
>>>
>>> On 28 Mar 2013, at 13:55, Dan Snazelle <subjectivity at hotmail.com> wrote:
>>>
>>>>
>>>> Though i work with c//arduino/AVRS not pics, this sounds very appealing! Can anyone recommend a place to learn about writing C code for an envelope follower? How do you do it? Shift signal to 0-5, then......
>>>>
>>>>
>>>> Thanks!!!
>>>
>>> The first thing after that would be to do the full wave rectification. Either dump the sign bit, if you've got signed data, or invert the data if the highest bit is clear if it's not signed - same thing, different representation.
>>
>>I strongly advise against doing this. It will create insane amounts of
>>aliasing. Do the rectification in analog. In fact full-wave
>>rectification is the most difficult thing to do when modelling analog
>>circuits digitally. Doing it anywhere near a level acceptable even for
>>the non-discriminating musician would require processing power orders
>>of magnitude larger than what a lowly pic can offer you. And in analog
>>it's just a few junctions.
>>
>>BTW, you can't really rectify a signal that's unipolar already.. just
>>a minor nitpick ;)
>>
>>> For the smoothing, I can imagine a couple of approaches:
>>>
>>> 1) Genuine honest-to-goodness digital filtering.
>>> There's a ton of stuff about this all over the web. But since we're talking about bottom-end chips, most of it is totally out of reach. We're not going to be running long FIR filters. But we could perhaps use an IIR filter, or a FIR with a very low sample rate, to keep it manageable.
>>
>>You can't change the sample rate, unless you mean you would downsample
>>the signal first. But then you'll have to have steep bandwidth
>>limiting filters and a filter for the down-sampling itself, which
>>means you do ten times the work you think you're saving by using lower
>>sampling rates. If you don't downsample you get aliasing, and then
>>you're detecting the envelope very incorrectly. It can only get you
>>fairly small speed-ups anyways before it stops being useful at all.
>>
>>An FIR with a fairly short impulse is fairly easy to manage, and on
>>something like a dspic you can do even longer ones. However IIR is the
>>usual way to go with envelope detectors if I'm not mistaken.
>>
>>> 2) Peak detection.
>>> This might be a good way to go. Keeping track of what the last couple of samples are would be easy, and would allow us to work out the current slope of the waveform. When the current sample is less than the previous one, the waveform has started heading back down, so we'd hold the peak value.
>>> Obviously we can't hold the highest value for ever, so there needs to be some "decay" parameter that allows it to fall. Multiply by 0.99 every sample, perhaps, or remove some small constant. This is good because the algorithm can respond instantly to peaks, but will smoothly and slowly fall for decays, which naturally tend to be longer. As ever, it won't do everything well - the envelope of short percussion hits might get stretched, for example.
>>
>>Depending on how one would fill in the blanks, you've just described
>>either IIR, or FIR, or a thing that results in loads of zipper noise
>>and incorrect enveloping. IIR and FIR really are the best approaches
>>here.
>>
>>> There's probably other ways, or combinations of one and the other that would work. The more specific you can be about what inputs you're trying to deal with, the better a job you'll manage to do. If you want to solve the general case, you'll need to make more compromises.
>>
>>Or add knobs :-)
>>
>>Cheers,
>>D.
>>_______________________________________________
>>Synth-diy mailing list
>>Synth-diy at dropmix.xs4all.nl
>>http://dropmix.xs4all.nl/mailman/listinfo/synth-diy


From cheater00 at gmail.com  Fri Mar 29 10:06:48 2013
From: cheater00 at gmail.com (cheater cheater)
Date: Fri, 29 Mar 2013 10:06:48 +0100
Subject: [sdiy] Fast envelope follower circuit needed..
In-Reply-To: <309EB6A9-742D-4A9D-8099-69D08A50A07C@sounds.wa.com>
References: <324c9c6efbb87ef612fdd8cdc2ccad26@oricom.ca>
	<7D00642A-814D-4BBC-B510-CDAC1630728D@electricdruid.net>
	<BLU0-SMTP338D2FA8904A3F6D5D401C9C8D20@phx.gbl>
	<541FA361-13D0-4D41-8296-E24F2DB98CB4@electricdruid.net>
	<309EB6A9-742D-4A9D-8099-69D08A50A07C@sounds.wa.com>
Message-ID: <CA+9GZUjOi+AjM3oegAonSR4eRQpwXNzrOHA+ZLGx9O9sVKd+DQ@mail.gmail.com>

Brian & Co.,

On Fri, Mar 29, 2013 at 1:14 AM,  <rsdio at sounds.wa.com> wrote:
> The easiest thing to do with signed data is use abs() to get the absolute
> value. Many 8-bit processors even have a native opcode to compute abs(), but
> they're not merely masking out the sign bit as would be done by and'ing with
> 0x7F.
>
> You are correct that inverting the data below the midpoint will work with
> unsigned data.
>
> You can probably use almost exactly the same code for either signed or
> unsigned.

why bother? The noise floor and inharmonic distortion introduced by
this process, done much more easily in an analog front-end, will be so
severe that the whole device will produce erroneous, un-musical
output.

Cheers,
D.


From tom at electricdruid.net  Fri Mar 29 13:03:54 2013
From: tom at electricdruid.net (Tom Wiltshire)
Date: Fri, 29 Mar 2013 12:03:54 +0000
Subject: [sdiy] Fast envelope follower circuit needed..
In-Reply-To: <CA+9GZUjOi+AjM3oegAonSR4eRQpwXNzrOHA+ZLGx9O9sVKd+DQ@mail.gmail.com>
References: <324c9c6efbb87ef612fdd8cdc2ccad26@oricom.ca>
	<7D00642A-814D-4BBC-B510-CDAC1630728D@electricdruid.net>
	<BLU0-SMTP338D2FA8904A3F6D5D401C9C8D20@phx.gbl>
	<541FA361-13D0-4D41-8296-E24F2DB98CB4@electricdruid.net>
	<309EB6A9-742D-4A9D-8099-69D08A50A07C@sounds.wa.com>
	<CA+9GZUjOi+AjM3oegAonSR4eRQpwXNzrOHA+ZLGx9O9sVKd+DQ@mail.gmail.com>
Message-ID: <4F6CAB3A-6477-4BFF-8100-27DBCBBAA68A@electricdruid.net>


On 29 Mar 2013, at 09:06, cheater cheater <cheater00 at gmail.com> wrote:

> Brian & Co.,
> 
> On Fri, Mar 29, 2013 at 1:14 AM,  <rsdio at sounds.wa.com> wrote:
>> The easiest thing to do with signed data is use abs() to get the absolute
>> value. Many 8-bit processors even have a native opcode to compute abs(), but
>> they're not merely masking out the sign bit as would be done by and'ing with
>> 0x7F.
>> 
>> You are correct that inverting the data below the midpoint will work with
>> unsigned data.
>> 
>> You can probably use almost exactly the same code for either signed or
>> unsigned.
> 
> why bother? The noise floor and inharmonic distortion introduced by
> this process, done much more easily in an analog front-end, will be so
> severe that the whole device will produce erroneous, un-musical
> output.

You made this claim previously, but I don't see the problem.

It's true that doing rectification in the digital domain will introduce aliasing - but so what? Imagine we sample at 20KHz, and we've got a 440Hz sine wave coming in. We rectify and get a waveform that looks like a series of little hills and valleys at 880Hz. Those little valleys are sharp spikes that will cause aliasing.
But why do I care? I'm not going to listen to the result. In fact, the only information I do care about is the height of those hills, and I haven't altered that at all. Once I derive the height of those peaks, I'm going throw that signal away. It won't ever see the light of analog day, and it certainly won't reach my ears, so I don't see how it can be "unmusical".

As to "easier in an analog front end", a typical precision op-amp full wave rectifier requires two opamps plus a few resistors and diodes. It isn't complicated, but it's complicated enough that (as has already been mentioned) several commercial implementations use a simpler half-wave rectifier despite the drawbacks.
Doing rectification in software is two instructions, or one if you've got a handy "abs" instruction. So, no, it's easier digitally. About the only argument I can see for doing it before the ADC is that it would provide protection against negative signals, which offsetting the input perhaps wouldn't.

Tom





From juzowen at gmail.com  Fri Mar 29 14:39:07 2013
From: juzowen at gmail.com (Justin Owen)
Date: Fri, 29 Mar 2013 13:39:07 +0000
Subject: [sdiy] Fast envelope follower circuit needed..
Message-ID: <5155997d.a163b40a.78a4.ffffb4ff@mx.google.com>

For a basic analog cook-book solution - look at the schem for the Polyfusion 2044 Envelope Follower.

It's a pretty simple set up - Half Wave Rectifier followed by LPF with switchable response. It also has a Gate output.

...but can someone clarify:

What is the LPF actually doing with regards to the follower output? Do the high frequencies cause some sort of unpleasant spike?

HTH/Thanks

J


-----Original Message-----
From: Jean-Pierre Desrochers [jpdesroc at oricom.ca]
Received: 28.03.2013 12:45:04
To: synth-diy at dropmix.xs4all.nl
Subject: [sdiy] Fast envelope follower circuit needed..

Hi List,
I'm sure this has been already discussed in the past
but I'm looking for a fast envelope follower circuit
schematic to rectify an incoming +/-2.5v AC waveform
to a rectified 0-5v wave.. The wave frequency could
go as low as 1 hz and sometimes will not be symetrical.
Any circuits welcomed !
JP
_______________________________________________
Synth-diy mailing list
Synth-diy at dropmix.xs4all.nl
http://dropmix.xs4all.nl/mailman/listinfo/synth-diy




From rburnett at richieburnett.co.uk  Fri Mar 29 15:32:10 2013
From: rburnett at richieburnett.co.uk (Richie Burnett)
Date: Fri, 29 Mar 2013 14:32:10 +0000
Subject: [sdiy] Fast envelope follower circuit needed..
Message-ID: <6owmerre9v8stkqr66ape3jv.1364567315150@email.android.com>

I agree with you Tom.

I'd be more concerned about the discrete sampling process not actually catching the peak value of a high-frequency signal than I would be about aliasing in the intermediate processing that follows. Remember, Nyquist's rule that you have to sample at at least twice the bandwidth of the signal only guarantees that you can reconstruct the original signal. It's not sufficient to guarantee that you'll get a sample at the true peak or trough position on your waveform. Or even anywhere near them! Your two samples per audio cycle could in fact be near the zero-crossings! Therefore you need to oversample considerably or use other tricks like interpolation to find the true peak amplitude of sampled high-frequency signals.

A search on the subject of 'Inter-sample overs' discusses how signals with high-frequency content can peak higher than the discrete samples that represent the signal in the digital domain. This is of interest to anyone designing a digital peak-reading audio level meter for example.

-Richie,

Sent from my Sony Ericsson Xperia ray

Tom Wiltshire <tom at electricdruid.net> wrote:

>
>On 29 Mar 2013, at 09:06, cheater cheater <cheater00 at gmail.com> wrote:
>
>> Brian & Co.,
>> 
>> On Fri, Mar 29, 2013 at 1:14 AM,  <rsdio at sounds.wa.com> wrote:
>>> The easiest thing to do with signed data is use abs() to get the absolute
>>> value. Many 8-bit processors even have a native opcode to compute abs(), but
>>> they're not merely masking out the sign bit as would be done by and'ing with
>>> 0x7F.
>>> 
>>> You are correct that inverting the data below the midpoint will work with
>>> unsigned data.
>>> 
>>> You can probably use almost exactly the same code for either signed or
>>> unsigned.
>> 
>> why bother? The noise floor and inharmonic distortion introduced by
>> this process, done much more easily in an analog front-end, will be so
>> severe that the whole device will produce erroneous, un-musical
>> output.
>
>You made this claim previously, but I don't see the problem.
>
>It's true that doing rectification in the digital domain will introduce aliasing - but so what? Imagine we sample at 20KHz, and we've got a 440Hz sine wave coming in. We rectify and get a waveform that looks like a series of little hills and valleys at 880Hz. Those little valleys are sharp spikes that will cause aliasing.
>But why do I care? I'm not going to listen to the result. In fact, the only information I do care about is the height of those hills, and I haven't altered that at all. Once I derive the height of those peaks, I'm going throw that signal away. It won't ever see the light of analog day, and it certainly won't reach my ears, so I don't see how it can be "unmusical".
>
>As to "easier in an analog front end", a typical precision op-amp full wave rectifier requires two opamps plus a few resistors and diodes. It isn't complicated, but it's complicated enough that (as has already been mentioned) several commercial implementations use a simpler half-wave rectifier despite the drawbacks.
>Doing rectification in software is two instructions, or one if you've got a handy "abs" instruction. So, no, it's easier digitally. About the only argument I can see for doing it before the ADC is that it would provide protection against negative signals, which offsetting the input perhaps wouldn't.
>
>Tom
>
>
>
>_______________________________________________
>Synth-diy mailing list
>Synth-diy at dropmix.xs4all.nl
>http://dropmix.xs4all.nl/mailman/listinfo/synth-diy

From mr at analogue.org  Fri Mar 29 16:08:39 2013
From: mr at analogue.org (Mattias Rickardsson)
Date: Fri, 29 Mar 2013 16:08:39 +0100
Subject: [sdiy] Fast envelope follower circuit needed..
In-Reply-To: <A9E6F624-4FA5-42D2-A04D-1A6D88B653A5@sounds.wa.com>
References: <m8hqxhp8da12813qt1286fj8.1364517799724@email.android.com>
	<A9E6F624-4FA5-42D2-A04D-1A6D88B653A5@sounds.wa.com>
Message-ID: <CAP92RBVdt7xFCQ04d6yBOB_P9X13UE8TUwJNAJFwedSU9ohdhQ@mail.gmail.com>

Speaking of half-wave vs. full-wave rectification... does half-wave
rectification capture the true amplitude envelope? Most waveforms are
unsymmetrical and have higher peaks towards one direction than the
other. The human voice being a good example. I'd argue that the peaks
are not what should be detected in an envelope follower, but even when
detecting the true "volume" of a signal - is it enough to just look at
the positive half of the signal, discarding the negative one?

Speaking of peaks... sample rate vs short spikes was mentioned, but if
a spike appears between the samples, isn't there a good chance that it
wouldn't contribute to the amplitude much at all? If it would happen
over and over again, like in classic aliasing, it's another problem of
course, so the upper frequencies should be limited.

/mr

Den 29 mar 2013 02:05 skrev <rsdio at sounds.wa.com>:
>
> Yes, you can do half wave, and many analog circuits do this. It's mostly about saving costs of components. Software is generally free, though, except for the cost of cycles at run time.
>
> However, to answer your question about "why bother," the answer is that half wave will put less energy into your LPF, and thus the output level will be less than with full wave.
>
> In additional, the half wave version will either be less smooth, or it will be approximately half as fast as a full wave.
>
> The original request asked for 5 V output, and I meant to point out earlier that you'd need at least 2x gain to get 5 V output from an envelope follower that's being fed a 5 V peak-to-peak signal. That's because the signal's average value is around the mid point, and thus the envelope peaks will be half of the peak-to-peak range. Theoretical maximum would be 2.5 V, and in practice it will be less than that with analog implementations because it's difficult to get the capacitor to fully charge after a diode voltage drop and the time constant delay. Digital implementations might be able to produce a peak envelope of 2.5 V for a brief impulse, even for half wave. There's nothing wrong with boosting this by a gain factor of 2, or more, if you really want 5 V envelope output.
>
> A half wave solution would merely need additional gain to reach 5 V output. However, there is no solution to make the half wave as fast as the full wave.
>
> Brian Willoughby
> Sound Consulting
>
>
> On Mar 28, 2013, at 17:43, Dave Manley wrote:
>>
>> You could also discard all negative samples (half wave).  The output goes thru LPF smoothing anyway,  why bother with full wave?
>>
>> -------- Original Message --------
>>  From: rsdio at sounds.wa.com
>>>
>>> On Mar 28, 2013, at 11:24, Tom Wiltshire wrote:
>>>>
>>>> On 28 Mar 2013, at 13:55, Dan Snazelle <subjectivity at hotmail.com>
>>>>>
>>>>> Though i work with c//arduino/AVRS not pics, this sounds very
>>>>> appealing! Can anyone recommend a place to learn about writing C
>>>>> code for an envelope follower? How do you do it? Shift signal to
>>>>> 0-5, then......
>>>>
>>>>
>>>> The first thing after that would be to do the full wave
>>>> rectification. Either dump the sign bit, if you've got signed data,
>>>> or invert the data if the highest bit is clear if it's not signed -
>>>> same thing, different representation. Half-wave rectification just
>>>> makes the signal more "lumpy" and isn't significantly easier
>>>> digitally anyway.
>>>
>>>
>>> You can't dump the sign bit, at least not with twos-complement data,
>>> which is what all digital signed data is. Maybe you're thinking of
>>> conceptually dumping the negative symbol, but that's not the same as
>>> dumping the sign bit. If you were to dump the sign bit, then the
>>> negative part of the signal would not be inverted, as needed, but
>>> would merely be shifted up into the positive range. That would make
>>> an ugly, spiked signal that would introduce significant errors in
>>> your envelope.
>>>
>>> The easiest thing to do with signed data is use abs() to get the
>>> absolute value. Many 8-bit processors even have a native opcode to
>>> compute abs(), but they're not merely masking out the sign bit as
>>> would be done by and'ing with 0x7F.
>>>
>>> You are correct that inverting the data below the midpoint will work
>>> with unsigned data.
>>>
>>> You can probably use almost exactly the same code for either signed
>>> or unsigned.
>>
>>
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy


From dlmanley at sonic.net  Fri Mar 29 17:53:04 2013
From: dlmanley at sonic.net (Dave Manley)
Date: Fri, 29 Mar 2013 09:53:04 -0700
Subject: [sdiy] Fast envelope follower circuit needed..
Message-ID: <0ma3xmb39jwps0ufdkrs1pu7.1364575984968@email.android.com>

What's the simplest peak detector/hold circuit?   In each sample period don't we really want the peak value?

-------- Original Message --------
 From: Mattias Rickardsson <mr at analogue.org>
 Sent: Fri, Mar 29, 2013 08:08 AM
 To: rsdio at sounds.wa.com
 CC: Synth-Diy List <synth-diy at dropmix.xs4all.nl>
 Subject: Re: [sdiy] Fast envelope follower circuit needed..

>Speaking of half-wave vs. full-wave rectification... does half-wave
>rectification capture the true amplitude envelope? Most waveforms are
>unsymmetrical and have higher peaks towards one direction than the
>other. The human voice being a good example. I'd argue that the peaks
>are not what should be detected in an envelope follower, but even when
>detecting the true "volume" of a signal - is it enough to just look at
>the positive half of the signal, discarding the negative one?
>
>Speaking of peaks... sample rate vs short spikes was mentioned, but if
>a spike appears between the samples, isn't there a good chance that it
>wouldn't contribute to the amplitude much at all? If it would happen
>over and over again, like in classic aliasing, it's another problem of
>course, so the upper frequencies should be limited.
>
>/mr
>
>Den 29 mar 2013 02:05 skrev <rsdio at sounds.wa.com>:
>>
>> Yes, you can do half wave, and many analog circuits do this. It's mostly about saving costs of components. Software is generally free, though, except for the cost of cycles at run time.
>>
>> However, to answer your question about "why bother," the answer is that half wave will put less energy into your LPF, and thus the output level will be less than with full wave.
>>
>> In additional, the half wave version will either be less smooth, or it will be approximately half as fast as a full wave.
>>
>> The original request asked for 5 V output, and I meant to point out earlier that you'd need at least 2x gain to get 5 V output from an envelope follower that's being fed a 5 V peak-to-peak signal. That's because the signal's average value is around the mid point, and thus the envelope peaks will be half of the peak-to-peak range. Theoretical maximum would be 2.5 V, and in practice it will be less than that with analog implementations because it's difficult to get the capacitor to fully charge after a diode voltage drop and the time constant delay. Digital implementations might be able to produce a peak envelope of 2.5 V for a brief impulse, even for half wave. There's nothing wrong with boosting this by a gain factor of 2, or more, if you really want 5 V envelope output.
>>
>> A half wave solution would merely need additional gain to reach 5 V output. However, there is no solution to make the half wave as fast as the full wave.
>>
>> Brian Willoughby
>> Sound Consulting
>>
>>
>> On Mar 28, 2013, at 17:43, Dave Manley wrote:
>>>
>>> You could also discard all negative samples (half wave).  The output goes thru LPF smoothing anyway,  why bother with full wave?
>>>
>>> -------- Original Message --------
>>>  From: rsdio at sounds.wa.com
>>>>
>>>> On Mar 28, 2013, at 11:24, Tom Wiltshire wrote:
>>>>>
>>>>> On 28 Mar 2013, at 13:55, Dan Snazelle <subjectivity at hotmail.com>
>>>>>>
>>>>>> Though i work with c//arduino/AVRS not pics, this sounds very
>>>>>> appealing! Can anyone recommend a place to learn about writing C
>>>>>> code for an envelope follower? How do you do it? Shift signal to
>>>>>> 0-5, then......
>>>>>
>>>>>
>>>>> The first thing after that would be to do the full wave
>>>>> rectification. Either dump the sign bit, if you've got signed data,
>>>>> or invert the data if the highest bit is clear if it's not signed -
>>>>> same thing, different representation. Half-wave rectification just
>>>>> makes the signal more "lumpy" and isn't significantly easier
>>>>> digitally anyway.
>>>>
>>>>
>>>> You can't dump the sign bit, at least not with twos-complement data,
>>>> which is what all digital signed data is. Maybe you're thinking of
>>>> conceptually dumping the negative symbol, but that's not the same as
>>>> dumping the sign bit. If you were to dump the sign bit, then the
>>>> negative part of the signal would not be inverted, as needed, but
>>>> would merely be shifted up into the positive range. That would make
>>>> an ugly, spiked signal that would introduce significant errors in
>>>> your envelope.
>>>>
>>>> The easiest thing to do with signed data is use abs() to get the
>>>> absolute value. Many 8-bit processors even have a native opcode to
>>>> compute abs(), but they're not merely masking out the sign bit as
>>>> would be done by and'ing with 0x7F.
>>>>
>>>> You are correct that inverting the data below the midpoint will work
>>>> with unsigned data.
>>>>
>>>> You can probably use almost exactly the same code for either signed
>>>> or unsigned.
>>>
>>>
>> _______________________________________________
>> Synth-diy mailing list
>> Synth-diy at dropmix.xs4all.nl
>> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>_______________________________________________
>Synth-diy mailing list
>Synth-diy at dropmix.xs4all.nl
>http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>

From cheater00 at gmail.com  Fri Mar 29 18:23:04 2013
From: cheater00 at gmail.com (cheater cheater)
Date: Fri, 29 Mar 2013 18:23:04 +0100
Subject: [sdiy] Fast envelope follower circuit needed..
In-Reply-To: <4F6CAB3A-6477-4BFF-8100-27DBCBBAA68A@electricdruid.net>
References: <324c9c6efbb87ef612fdd8cdc2ccad26@oricom.ca>
	<7D00642A-814D-4BBC-B510-CDAC1630728D@electricdruid.net>
	<BLU0-SMTP338D2FA8904A3F6D5D401C9C8D20@phx.gbl>
	<541FA361-13D0-4D41-8296-E24F2DB98CB4@electricdruid.net>
	<309EB6A9-742D-4A9D-8099-69D08A50A07C@sounds.wa.com>
	<CA+9GZUjOi+AjM3oegAonSR4eRQpwXNzrOHA+ZLGx9O9sVKd+DQ@mail.gmail.com>
	<4F6CAB3A-6477-4BFF-8100-27DBCBBAA68A@electricdruid.net>
Message-ID: <CA+9GZUh1UZ-KmwRQ3UzG-SVdwv+K7MQ1rBz=ZvB1cTOmPtpjag@mail.gmail.com>

Hi Tom,

I'm afraid I have to disagree with you on several points here.

On Fri, Mar 29, 2013 at 1:03 PM, Tom Wiltshire <tom at electricdruid.net> wrote:
>
> On 29 Mar 2013, at 09:06, cheater cheater <cheater00 at gmail.com> wrote:
>> The noise floor and inharmonic distortion introduced by
>> this process, done much more easily in an analog front-end, will be so
>> severe that the whole device will produce erroneous, un-musical
>> output.
>
> You made this claim previously, but I don't see the problem.
>
> It's true that doing rectification in the digital domain will introduce aliasing - but so what? Imagine we sample at 20KHz, and we've got a 440Hz sine wave coming in. We rectify and get a waveform that looks like a series of little hills and valleys at 880Hz. Those little valleys are sharp spikes that will cause aliasing.

And they contain wide-band content of very high energy. Bear in mind -
this is crucial - that an envelope follower is just an asymmetric
low-pass filter in its most basic form. Its response is going to be
very frequency dependent. Feeding it with a signal that's mutilated in
the frequency domain is going to do a lot of damage to its operation.

The Hilbert transform approach Richie mentioned above is also
frequency-dependent.

> But why do I care? I'm not going to listen to the result.

Oh, but yes you are. The aliasing doesn't only go above the base
frequency of the wave, but below it as well. You'll get rumble in the
envelope output.

But even at -36 dB, which is a good approximation for the amount of
inharmonic content introduced by the aliasing after the enveloping has
been applied, this error easily introduces inharmonic content into
whatever that detected envelope is going to modulate. If you're using
it to modulate an amplifier, then it's like adding a fairly quiet (but
not absent) oscillator at some unrelated pitch that'll break all your
harmonies. Similarly when frequency modulating a filter.

So you have rumble and high-frequency buzzing. And as your melody
progresses, that rumbling and buzzing will play another melody,
harmonically unrelated to it, messing up any sort of arrangement you
were trying to create.

> In fact, the only information I do care about is the height of those hills, and I haven't altered that at all. Once I derive the height of those peaks, I'm going throw that signal away. It won't ever see the light of analog day, and it certainly won't reach my ears, so I don't see how it can be "unmusical".

That's the thing, filters don't work this way. They take every part of
the signal into consideration, not "just the peaks". Those parts of
the signal end up in the output of the filter depending on their
frequency content and power. When the aliasing happens, all the
instantaneous power of the signal (and more, depending) is being fed
into the filter above its cutoff frequency. The filter will be unable
to reject it all (because you don't want to use steep filters for
envelope following) and thus you end up with the junk in your output,
modulating the rest of your sound.

> As to "easier in an analog front end", a typical precision op-amp full wave rectifier requires two opamps plus a few resistors and diodes. It isn't complicated, but it's complicated enough that (as has already been mentioned) several commercial implementations use a simpler half-wave rectifier despite the drawbacks.

You got that right. But do you know what it takes to do rectification
in software *properly*? I don't. It's still being figured out, expect
to see new papers every now and then. But even taking the most usual
good-sounding approach that's used in most new synths, i.e.
band-limited transient reconstruction: it's a shitload of work and
requires a huge amount of processing (huge compared to the little PIC)

> Doing rectification in software is two instructions, or one if you've got a handy "abs" instruction. So, no, it's easier digitally. About the only argument I can see for doing it before the ADC is that it would provide protection against negative signals, which offsetting the input perhaps wouldn't.

Of course it's easier to "do". But it's not easier to do.

Cheers,
D.


From gil_we at yahoo.com  Fri Mar 29 19:36:21 2013
From: gil_we at yahoo.com (Gil W.)
Date: Fri, 29 Mar 2013 11:36:21 -0700 (PDT)
Subject: [sdiy] FS: Parts for JD800 Matrix6 Korg-M1 Juno106 Polysix
	Hammond-B100
Message-ID: <1364582181.80509.YahooMailClassic@web162205.mail.bf1.yahoo.com>

Let me know if you need anything.



From tim.parkhurst at gmail.com  Fri Mar 29 20:28:58 2013
From: tim.parkhurst at gmail.com (Tim Parkhurst)
Date: Fri, 29 Mar 2013 12:28:58 -0700
Subject: [sdiy] CEM and SSM chips for sale
Message-ID: <CACDGQT8JtGDP1xS6QdSLRGUdU2se8aLv9xWDck2yF+qZXfXbrA@mail.gmail.com>

Okay, I have some CEM and SSM chips that have been sitting in my
drawer for almost 10 years. I have SO many projects to do that I don't
think I'll ever actually get around to using them, so I figure I'll
put them out there for someone else to use. :) Prices don't include
shipping. Thanks for shopping at TimmyMart.
4x 3372 VCF/VCA $30ea
2x 3396 wide $20ea
3x 3396 narrow $20ea
1x 3394 synth on a chip $50
1x SSM2044 VCF $35
1x SSM2040 VCF (legendary) $60

Tim (chips off the old block) Servo
---
"Sire, the church of God is an anvil that has worn out many hammers."
- H.L. Hastings


From tom at electricdruid.net  Fri Mar 29 22:14:41 2013
From: tom at electricdruid.net (Tom Wiltshire)
Date: Fri, 29 Mar 2013 21:14:41 +0000
Subject: [sdiy] Fast envelope follower circuit needed..
In-Reply-To: <CA+9GZUh1UZ-KmwRQ3UzG-SVdwv+K7MQ1rBz=ZvB1cTOmPtpjag@mail.gmail.com>
References: <324c9c6efbb87ef612fdd8cdc2ccad26@oricom.ca>
	<7D00642A-814D-4BBC-B510-CDAC1630728D@electricdruid.net>
	<BLU0-SMTP338D2FA8904A3F6D5D401C9C8D20@phx.gbl>
	<541FA361-13D0-4D41-8296-E24F2DB98CB4@electricdruid.net>
	<309EB6A9-742D-4A9D-8099-69D08A50A07C@sounds.wa.com>
	<CA+9GZUjOi+AjM3oegAonSR4eRQpwXNzrOHA+ZLGx9O9sVKd+DQ@mail.gmail.com>
	<4F6CAB3A-6477-4BFF-8100-27DBCBBAA68A@electricdruid.net>
	<CA+9GZUh1UZ-KmwRQ3UzG-SVdwv+K7MQ1rBz=ZvB1cTOmPtpjag@mail.gmail.com>
Message-ID: <F79ED645-76C5-4E22-B07A-410E2AE8AB08@electricdruid.net>

Hi Damian,

> I'm afraid I have to disagree with you on several points here.

I thought this might run for a while yet ;)

The key thing for me here is that flopping the bottom part of the signal up to the top doesn't change any of the peak levels, whatever happens to the frequency spectrum, and it's primarily the peaks I'm interested in. However, it does obscure which peaks are positive ones and which are negative, so maybe that's a problem. 

Perhaps I don't even need to bother rectifying the signal. Why not just keep a check on the negative peaks like the positive ones, and then use some average of the two as the final output. JP said his signal might well be asymmetric (which is not at all uncommon), so we can't assume the two envelopes will be the same.

It seems to me that most of your complaints about filtering rectified signals apply pretty much equally to analog as digital. Ok, so you haven't got aliasing to worry about, but you still converted a lot of low frequency energy in the signal into high frequency spikes which then get filtered away - after all, that's the point. And there will be some residual ripple - that's a fact, and that's something we see in the analog implementations.

Also it's worth pointing out that we don't have to follow the rectifier (if we use one) with some kind of filter.
Harry Bissel's analog envelope follower that Tim posted gives some clues to the sort of techniques I'd think about using:

http://www.edn.com/design/analog/4344656/Envelope-follower-combines-fast-response-low-ripple

This takes the highest output from three peak detectors and uses that as the output.
I'd adopt some similar idea - check for highest peaks, and store them, and then draw a straight line between those peak points. Do the same for the negative peaks. Invert the negative envelope so that it's positive, and then average the two to get your final output.

There you go! No rectification, no filters*, no problem!

Ok, so it isn't perfect, but neither is the analog solution we're trying to replace. In fact, far from it. The bar is pretty low here. Furthermore, the proposed digital solution won't need any multiplies, it's dead simple to code, it'll fit in an 8-pin PIC, and it'll probably do at least as good a job as an analog solution that needs many more chips. Ultimately, that's what counts. I could worry about aliasing and the frequency domain representation of a full-wave rectified signal, but I don't think it'll get it done quicker or cheaper.

Tom

*Some of these stages could be represented by filters of various types, but that wouldn't be the best way to implement them or think about them, so I choose not to refer to them as filters.





From music.maker at gte.net  Fri Mar 29 22:32:23 2013
From: music.maker at gte.net (Scott Gravenhorst)
Date: Fri, 29 Mar 2013 14:32:23 -0700
Subject: [sdiy] Fast envelope follower circuit needed..
Message-ID: <201303292132.r2TLWNhf006015@linux7.lan>


Tom Wiltshire <tom at electricdruid.net> wrote:
>Hi Damian,
>
>> I'm afraid I have to disagree with you on several points here.
>
>I thought this might run for a while yet ;)
>
>The key thing for me here is that flopping the bottom part of the 
>signal up to the top doesn't change any of the peak levels, 
>whatever happens to the frequency spectrum, and it's primarily 
>the peaks I'm interested in. However, it does obscure which peaks 
>are positive ones and which are negative, so maybe that's a 
>problem. 
>
>Perhaps I don't even need to bother rectifying the signal. Why 
>not just keep a check on the negative peaks like the positive 
>ones, and then use some average of the two as the final output. 
>JP said his signal might well be asymmetric (which is not at all 
>uncommon), so we can't assume the two envelopes will be the same. 
>
>It seems to me that most of your complaints about filtering 
>rectified signals apply pretty much equally to analog as digital. 
>Ok, so you haven't got aliasing to worry about, but you still 
>converted a lot of low frequency energy in the signal into high 
>frequency spikes which then get filtered away - after all, that's 
>the point. And there will be some residual ripple - that's a 
>fact, and that's something we see in the analog implementations. 
>
>Also it's worth pointing out that we don't have to follow the 
>rectifier (if we use one) with some kind of filter. 

POSSIBLE NAIVE POSTER WARNING...  And even if you did and used full wave rectification, the peak
detection part of the algorithm which would supply the filter with its input data removes the
spikes (and their associated transients) anyway - so aliasing due to transients becomes moot.  The
signal you filter is then a band limited signal with very small potential for alias artifacts due
to very low frequency harmonic components.

In my opinion, a digital approach to this sounds dandy.

-- ScottG
________________________________________________________________________
-- Scott Gravenhorst
-- FPGA MIDI Synth Info: jovianpyx.dyndns.org:8080/public/FPGA_synth/
-- FatMan Mods Etc.: jovianpyx.dyndns.org:8080/public/fatman/
-- Some Random Electronics Bits: jovianpyx.dyndns.org:8080/public/electronics/
-- When the going gets tough, the tough use the command line.



From rsdio at sounds.wa.com  Fri Mar 29 23:02:50 2013
From: rsdio at sounds.wa.com (rsdio at sounds.wa.com)
Date: Fri, 29 Mar 2013 15:02:50 -0700
Subject: [sdiy] Fast envelope follower circuit needed..
In-Reply-To: <F79ED645-76C5-4E22-B07A-410E2AE8AB08@electricdruid.net>
References: <324c9c6efbb87ef612fdd8cdc2ccad26@oricom.ca>
	<7D00642A-814D-4BBC-B510-CDAC1630728D@electricdruid.net>
	<BLU0-SMTP338D2FA8904A3F6D5D401C9C8D20@phx.gbl>
	<541FA361-13D0-4D41-8296-E24F2DB98CB4@electricdruid.net>
	<309EB6A9-742D-4A9D-8099-69D08A50A07C@sounds.wa.com>
	<CA+9GZUjOi+AjM3oegAonSR4eRQpwXNzrOHA+ZLGx9O9sVKd+DQ@mail.gmail.com>
	<4F6CAB3A-6477-4BFF-8100-27DBCBBAA68A@electricdruid.net>
	<CA+9GZUh1UZ-KmwRQ3UzG-SVdwv+K7MQ1rBz=ZvB1cTOmPtpjag@mail.gmail.com>
	<F79ED645-76C5-4E22-B07A-410E2AE8AB08@electricdruid.net>
Message-ID: <E30DE717-5170-4926-9E8B-65CFE70AFE86@sounds.wa.com>


On Mar 29, 2013, at 14:14, Tom Wiltshire wrote:
> Also it's worth pointing out that we don't have to follow the  
> rectifier (if we use one) with some kind of filter.
> Harry Bissel's analog envelope follower that Tim posted gives some  
> clues to the sort of techniques I'd think about using:
>
> http://www.edn.com/design/analog/4344656/Envelope-follower-combines- 
> fast-response-low-ripple
>
> This takes the highest output from three peak detectors and uses  
> that as the output.
> I'd adopt some similar idea - check for highest peaks, and store  
> them, and then draw a straight line between those peak points. Do  
> the same for the negative peaks. Invert the negative envelope so  
> that it's positive, and then average the two to get your final output.
>
> There you go! No rectification, no filters*, no problem!
>
> *Some of these stages could be represented by filters of various  
> types, but that wouldn't be the best way to implement them or think  
> about them, so I choose not to refer to them as filters.


Sorry, Tom, but that circuit has at least three rectifiers and four  
filters. A filter removes unwanted elements of a signal, and at least  
four capacitors in that circuit are serving to remove any non-smooth  
aspects of the signal. That makes them filters. The diodes in each of  
the peak detectors are rectifiers, although they're really only half- 
wave rectifiers.

I don't see how you've shown that any problems that might be due to  
rectification or filtering are in any way removed by that circuit,  
since it has all of the problematic characteristics of both.


p.s. cheater is correct that there will be aliasing, but I am  
convinced that the aliasing will not have any negative effect that  
wouldn't also be present in the perfectly non-aliased ideal.

First of all, the digital rectification does produce aliases of the  
original frequency. Second, aliased envelopes can affect the audio  
signal path. In the case of a VCA EG, even in the digital world  
amplitude modulation creates the sums and differences of all input  
frequencies because the two signals - envelope and audio - are  
multiplied. If the sum of any frequency in the envelope signal and  
any frequency in the audio signal would exceed the Nyquist frequency,  
then the resulting audio output would be aliased.

The big problem with aliasing is that you usually cannot filter out  
the aliased frequencies because they lie in between the non-aliased  
frequencies. Sure, you can use a low-pass, but any aliases that are  
below the cutoff will still be in the output.

I do not see how that is a problem with an envelope detector. The  
goal is to produce a control signal with basically "no" frequency  
components, or at least they should be as close to DC as possible. In  
practice, there will always be undesired low-frequency components in  
the envelope. My point here is that aliased components are no more  
problematic than non-aliased frequency components. The smoothing  
filter in the envelope generator will successfully remove all  
frequencies above a certain point, whether those frequencies are  
aliases or not. The remaining frequency components are all equally  
undesired, and it doesn't really matter whether they're aliases or  
not, so long as they are low enough in frequency.

The hard limit is as follows: Assuming your audio signal is limited  
to 20 kHz maximum, and your sample rate is 44.1 kHz, then your  
envelope signal need only have no aliases above 2.05 kHz. Bingo, no  
audio aliasing. I assume that envelope smoothing filters will have a  
3 dB cutoff frequency sufficiently below 2 kHz that there will  
basically be nothing left above 2 kHz in the envelope.

Again, even if you could reach the ideal of no aliased frequencies in  
the envelope, there will still be non-aliased frequencies that will  
ring-modulate the audio signal. That form of distortion is always  
possible in a gain block with non-constant (non-DC) gain. Because  
gain is multiplication, the only way to maintain the precise audio  
frequencies of the input is for the gain factor to be DC (constant) -  
an envelope is not constant.

Brian Willoughby
Sound Consulting



From rsdio at sounds.wa.com  Fri Mar 29 23:04:38 2013
From: rsdio at sounds.wa.com (rsdio at sounds.wa.com)
Date: Fri, 29 Mar 2013 15:04:38 -0700
Subject: [sdiy] Fast envelope follower circuit needed..
In-Reply-To: <201303292132.r2TLWNhf006015@linux7.lan>
References: <201303292132.r2TLWNhf006015@linux7.lan>
Message-ID: <EDE624C8-1881-4BBA-BB6F-D7D82C314476@sounds.wa.com>


On Mar 29, 2013, at 14:32, Scott Gravenhorst wrote:
> POSSIBLE NAIVE POSTER WARNING...  And even if you did and used full  
> wave rectification, the peak
> detection part of the algorithm which would supply the filter with  
> its input data removes the
> spikes (and their associated transients) anyway - so aliasing due  
> to transients becomes moot.  The
> signal you filter is then a band limited signal with very small  
> potential for alias artifacts due
> to very low frequency harmonic components.


The peak detection part of the algorithm actually creates aliasing  
rather than removing it. That's because peak detection allows the  
output to suddenly change if the input is greater than the hold  
value, and any time a digital algorithm allows a sudden discontinuity  
in its output you have the potential for non-band-limited square wave  
type aliasing. You are correct that peak detection will remove any  
downward-pointing spikes, but the tradeoff is that peak detection can  
introduce upward-pointing spikes. I suppose you could argue that the  
peak detector will never make a sudden jump unless the input signal  
already has a sudden jump, but the nonlinear nature of abs() destroys  
the continuity of the original signal.

If aliasing were a problem, then I don't think it's correct to assume  
that peak detection would avoid it. However, my previous message  
tried to make the argument that aliasing is not a problem for  
envelope control, and thus it doesn't really matter.

Brian Willoughby
Sound Consulting



From pfperry at melbpc.org.au  Sat Mar 30 07:24:24 2013
From: pfperry at melbpc.org.au (Paul Perry)
Date: Sat, 30 Mar 2013 17:24:24 +1100
Subject: [sdiy] Fast envelope follower circuit needed..
References: <324c9c6efbb87ef612fdd8cdc2ccad26@oricom.ca><7D00642A-814D-4BBC-B510-CDAC1630728D@electricdruid.net><BLU0-SMTP338D2FA8904A3F6D5D401C9C8D20@phx.gbl><541FA361-13D0-4D41-8296-E24F2DB98CB4@electricdruid.net><309EB6A9-742D-4A9D-8099-69D08A50A07C@sounds.wa.com><CA+9GZUjOi+AjM3oegAonSR4eRQpwXNzrOHA+ZLGx9O9sVKd+DQ@mail.gmail.com><4F6CAB3A-6477-4BFF-8100-27DBCBBAA68A@electricdruid.net>
	<CA+9GZUh1UZ-KmwRQ3UzG-SVdwv+K7MQ1rBz=ZvB1cTOmPtpjag@mail.gmail.com>
Message-ID: <52800F3374954107AC3E6D91E2C76012@ownerf0d9273f5>

A little philosophy on envelope followers:
It's easy to see what the envelope should look like - after the original 
signal has been! But if you take a typical input signal, graph it, and then 
cover it with a card, and try to draw an envelope as you slide the card to 
the right........... you might see that is is not just 'difficult', but 
actually 'impossible' to have a circuit that behaves as you would wish.

In my experience, there is no envelope follower that suits all applications. 
In particular, percussion and guitar are very different. I don't doubt that 
Harry's circuit is ideal for a guitar.. and that if he could have got away 
with fewer components, he would have (remember, a hex guitar has SIX 
outputs!)

On the bright side, I have also found that if you are using the derived 
envelope to control a filter or VCA, the actual audio effect can be quite 
musically useful, even though the envelope itself looks really unpromising 
on the scope. In particular, full wave rectification was not much better 
than half wave.

Real problems arise when the frequencies being enveloped are very low, in 
this case the low pass filtering is in conflict with the envelope being 
derived. Again, it is a matter of knowing what the results are going to be 
with a practical envelope, rather than knocking oneself out trying to 
produce an 'ideal' envelope.

paul perry Melbourne Australia




From cheater00 at gmail.com  Sat Mar 30 15:19:19 2013
From: cheater00 at gmail.com (cheater cheater)
Date: Sat, 30 Mar 2013 15:19:19 +0100
Subject: [sdiy] Fast envelope follower circuit needed..
In-Reply-To: <52800F3374954107AC3E6D91E2C76012@ownerf0d9273f5>
References: <324c9c6efbb87ef612fdd8cdc2ccad26@oricom.ca>
	<7D00642A-814D-4BBC-B510-CDAC1630728D@electricdruid.net>
	<BLU0-SMTP338D2FA8904A3F6D5D401C9C8D20@phx.gbl>
	<541FA361-13D0-4D41-8296-E24F2DB98CB4@electricdruid.net>
	<309EB6A9-742D-4A9D-8099-69D08A50A07C@sounds.wa.com>
	<CA+9GZUjOi+AjM3oegAonSR4eRQpwXNzrOHA+ZLGx9O9sVKd+DQ@mail.gmail.com>
	<4F6CAB3A-6477-4BFF-8100-27DBCBBAA68A@electricdruid.net>
	<CA+9GZUh1UZ-KmwRQ3UzG-SVdwv+K7MQ1rBz=ZvB1cTOmPtpjag@mail.gmail.com>
	<52800F3374954107AC3E6D91E2C76012@ownerf0d9273f5>
Message-ID: <CA+9GZUhhq-q4yWB5Tz2Rb8uUgnO6m7Pyb85qpn_F2EdMHrfS6w@mail.gmail.com>

Hi guys,
there are several points from Tom, Scott, and Brian here I'd like to
address. I hope it's ok, I would like to point out some errors to
spare you frustration in the future.

1. "the filter does not care about the spikes in the valleys"

that's incorrect. Bear in mind that this high-frequency content is not
removed in a low-pass filter, it is attenuated. Quite often the
rejection is less than the theoretical maximum for many reasons
(numerical error, incorrect algorithm implementation, etc). In effect,
that "rejected" signal is still there.

Especially if you use non-linearities later in your signal path (that
is, always), whatever program you modulate with such a tainted
envelope will sound worse than it could otherwise.

2. "analog rectification creates wide-band energy bursts anyways"

not an accurate picture of the situation. In analog, this sound will
be harmonically related to your input, which is a good thing. In
digital, it will work exactly against it, which is very bad. Besides,
the turn-on and turn-off speed is limited, it's not when using abs()
in a digital signal, but I don't know what the exact difference in
bandwidth limitation is in analog - it's unlimited in digital (with
the caveat that it gets mirrored around fs/2 and 0Hz, perhaps several
times)

3. "the peak hold doesn't care about the little valleys"

not true at all. I believe this comes from the conceptual
simplification of the filter, which could go like this:
(i) when the input signal is above the current output, raise the output to match
(ii) when the input signal is below the current output, drop the
output at a constant rate

that is extremely inaccurate. What happens is more similar to this:
(iii) measure the distance between the current output and current
input. Move so that during a period of time, say 1 second, you cover a
fixed percentage of this distance, directly related to a number called
the time constant. For example, cover 90% in 1 second.

What does this mean? Say your output is at 1V, and your input is a
sinewave oscillation between 0V and 0.01V. As your output is falling,
it'll fall relatively faster when the input is near 0V (because the
distance is higher), and it'll fall relatively slower when the input
signal is *closer* to the output signal (i.e. when the input is at its
0.01V maximum).

This is easily described by saying that the peak hold is just a filter
as well, but it doesn't quite picture it, that's why I described the
operation above.

You might ask: "But can't I just use max() of the input and output and
somehow use that?"

One might think this code would work:
if (output < input) {
  output = input
}
else {
  output -= 0.01
}

This is a beginner's mistake. This sort of design will create awful
amounts of zipper noise. You don't want that. You want to use a
filter.

Brian, you first correct Tom on his mistake, then make the same
mistake yourself elsewhere, then agree with Scott who has made the
same mistake :)

4. "because peak detection allows the output to suddenly change if the
input is greater than the hold value, and any time a digital algorithm
allows a sudden discontinuity in its output you have the potential for
non-band-limited square wave type aliasing"

Not true. You just use a normal (asymmetric) filter for the peak hold,
like you usually would, and set its time constant for rising signals
so that it reacts instantly.

Your logic is flawed because you're thinking in terms of the "if/else"
model I described above. It doesn't hold. The transition you describe
will not be abrupt. There will still be phase delay like in any usual
filter (that's a good thing), and this phase delay will give the
filter just enough time buffer for things to happen relatively
smoothly. It's a different question, though, whether your filter
aliases *as a whole*.

5. "Assuming your audio signal is limited to 20 kHz maximum, and your
sample rate is 44.1 kHz, then your envelope signal need only have no
aliases above 2.05 kHz. Bingo, no audio aliasing"

That's not how it works. If your signal picked up aliasing somewhere
earlier, it's not going to magically get rid of it by becoming
band-limited later on. If you have frequencies within your 2.05 kHz
that do not agree with the harmony of your program, you're going to
hear it, simple as that. If you said...

"multiplying two signals, such that the sum of their bandwidth is less
than fs/2, will not create new aliases"

...then I might be partial to agreeing, although due to numerical
inaccuracies that's not really true either. Depending on what you're
working with, YMMV.

6. "I do not see how that is a problem with an envelope detector. The
goal is to produce a control signal with basically "no" frequency
components, or at least they should be as close to DC as possible. In
practice, there will always be undesired low-frequency components in
the envelope. My point here is that aliased components are no more
problematic than non-aliased frequency components."

Only if you revel in muddy sound that becomes more and more occluded
and unclear the more processing you apply. It's a fact that
harmonically unrelated sounds present in your program will deteriorate
its quality. They're much worse than e.g. white noise, because they're
concentrated in single, discernible tones, which play melodies that go
completely against the sounds that you are playing on the keyboard.
I'm not sure how to better say this; if anyone here is not convinced
that having this is bad, please read the previous sentence 50 times.

7. "The remaining frequency components are all equally undesired, and
it doesn't really matter whether they're aliases or not, so long as
they are low enough in frequency."

It's exactly the other way around: aliased low-frequency components
are very undesirable and are glaring problems in the final sound.
Harmonically related frequency components don't work against the sound
you're building, and are often very desirable. At worst they are
harmless because they become invisible when surrounded by lots of
other stuff playing in the same harmony.

8. "Perhaps I don't even need to bother rectifying the signal. Why not
just keep a check on the negative peaks like the positive ones, and
then use some average of the two as the final output. JP said his
signal might well be asymmetric (which is not at all uncommon), so we
can't assume the two envelopes will be the same."

This is DSP, not analog fx design. DSP is a harsh mistress. You can do
a lot of learning by experimenting in analog electronics and it works
out well. The signal degradation which happens when you make errors is
very musical: nth harmonic distortion, homogenic noise, cross-talk of
content that is harmonically related. It's all nice and very
forgiving. In DSP, you have very, very bad degradation modes: i)
aliasing - takes your melody and makes it sound like it's just random
notes; ii) through-zero distortion - adds mud and reduces clarity in a
way that's not related to your sound in any way; iii) time delay -
makes sure that you will never be able to mix your sound correctly
again after you've split it; iv) numerical errors - similar to
aliasing, those create junk in your sound that is harmonically
unrelated to the sound you're trying to achieve (examples include bit
errors, zipper noise, clicks, hard clipping, etc).

Be warned: you really have to go by the book on DSP. There are very
good books online; Julius Orion Smith's website is the best,
most-recommended classic. You simply can't "common-sense" your way
through digital audio. It's daunting. You'll be making errors, and the
worst thing is that in analog when you make errors they're obvious
whereas in digital, when you make errors, you might not notice until
after you've shipped the box; in the end people figure it out though
and just don't like your design. Granted that doesn't matter if you're
selling for the masses that buy microcorgi and other junk like that;
but let's maybe leave this kind of talk off this list and try to think
about what it takes to do an honest job. As someone said, if
something's worth doing, it's worth doing well.

Cheers,
D.

On Sat, Mar 30, 2013 at 7:24 AM, Paul Perry <pfperry at melbpc.org.au> wrote:
> A little philosophy on envelope followers:
> It's easy to see what the envelope should look like - after the original
> signal has been! But if you take a typical input signal, graph it, and then
> cover it with a card, and try to draw an envelope as you slide the card to
> the right........... you might see that is is not just 'difficult', but
> actually 'impossible' to have a circuit that behaves as you would wish.
>
> In my experience, there is no envelope follower that suits all applications.
> In particular, percussion and guitar are very different. I don't doubt that
> Harry's circuit is ideal for a guitar.. and that if he could have got away
> with fewer components, he would have (remember, a hex guitar has SIX
> outputs!)
>
> On the bright side, I have also found that if you are using the derived
> envelope to control a filter or VCA, the actual audio effect can be quite
> musically useful, even though the envelope itself looks really unpromising
> on the scope. In particular, full wave rectification was not much better
> than half wave.
>
> Real problems arise when the frequencies being enveloped are very low, in
> this case the low pass filtering is in conflict with the envelope being
> derived. Again, it is a matter of knowing what the results are going to be
> with a practical envelope, rather than knocking oneself out trying to
> produce an 'ideal' envelope.
>
> paul perry Melbourne Australia
>
>
>
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy


From tom at electricdruid.net  Sat Mar 30 17:22:59 2013
From: tom at electricdruid.net (Tom Wiltshire)
Date: Sat, 30 Mar 2013 16:22:59 +0000
Subject: [sdiy] Fast envelope follower circuit needed..
In-Reply-To: <CA+9GZUhhq-q4yWB5Tz2Rb8uUgnO6m7Pyb85qpn_F2EdMHrfS6w@mail.gmail.com>
References: <324c9c6efbb87ef612fdd8cdc2ccad26@oricom.ca>
	<7D00642A-814D-4BBC-B510-CDAC1630728D@electricdruid.net>
	<BLU0-SMTP338D2FA8904A3F6D5D401C9C8D20@phx.gbl>
	<541FA361-13D0-4D41-8296-E24F2DB98CB4@electricdruid.net>
	<309EB6A9-742D-4A9D-8099-69D08A50A07C@sounds.wa.com>
	<CA+9GZUjOi+AjM3oegAonSR4eRQpwXNzrOHA+ZLGx9O9sVKd+DQ@mail.gmail.com>
	<4F6CAB3A-6477-4BFF-8100-27DBCBBAA68A@electricdruid.net>
	<CA+9GZUh1UZ-KmwRQ3UzG-SVdwv+K7MQ1rBz=ZvB1cTOmPtpjag@mail.gmail.com>
	<52800F3374954107AC3E6D91E2C76012@ownerf0d9273f5>
	<CA+9GZUhhq-q4yWB5Tz2Rb8uUgnO6m7Pyb85qpn_F2EdMHrfS6w@mail.gmail.com>
Message-ID: <D38401AC-5379-46F7-A221-9032544781AC@electricdruid.net>

So how would *you* solve the problem, Damian?

So far, you've spent a lot of effort telling everyone who's made a suggestion about the problem why they're wrong, and why it couldn't possibly work, but I'm not seeing much in the way of better suggestions.

Tom


On 30 Mar 2013, at 14:19, cheater cheater <cheater00 at gmail.com> wrote:

> Hi guys,
> there are several points from Tom, Scott, and Brian here I'd like to
> address. I hope it's ok, I would like to point out some errors to
> spare you frustration in the future.
> 
> 1. "the filter does not care about the spikes in the valleys"
> 
> that's incorrect. Bear in mind that this high-frequency content is not
> removed in a low-pass filter, it is attenuated. Quite often the
> rejection is less than the theoretical maximum for many reasons
> (numerical error, incorrect algorithm implementation, etc). In effect,
> that "rejected" signal is still there.
> 
> Especially if you use non-linearities later in your signal path (that
> is, always), whatever program you modulate with such a tainted
> envelope will sound worse than it could otherwise.
> 
> 2. "analog rectification creates wide-band energy bursts anyways"
> 
> not an accurate picture of the situation. In analog, this sound will
> be harmonically related to your input, which is a good thing. In
> digital, it will work exactly against it, which is very bad. Besides,
> the turn-on and turn-off speed is limited, it's not when using abs()
> in a digital signal, but I don't know what the exact difference in
> bandwidth limitation is in analog - it's unlimited in digital (with
> the caveat that it gets mirrored around fs/2 and 0Hz, perhaps several
> times)
> 
> 3. "the peak hold doesn't care about the little valleys"
> 
> not true at all. I believe this comes from the conceptual
> simplification of the filter, which could go like this:
> (i) when the input signal is above the current output, raise the output to match
> (ii) when the input signal is below the current output, drop the
> output at a constant rate
> 
> that is extremely inaccurate. What happens is more similar to this:
> (iii) measure the distance between the current output and current
> input. Move so that during a period of time, say 1 second, you cover a
> fixed percentage of this distance, directly related to a number called
> the time constant. For example, cover 90% in 1 second.
> 
> What does this mean? Say your output is at 1V, and your input is a
> sinewave oscillation between 0V and 0.01V. As your output is falling,
> it'll fall relatively faster when the input is near 0V (because the
> distance is higher), and it'll fall relatively slower when the input
> signal is *closer* to the output signal (i.e. when the input is at its
> 0.01V maximum).
> 
> This is easily described by saying that the peak hold is just a filter
> as well, but it doesn't quite picture it, that's why I described the
> operation above.
> 
> You might ask: "But can't I just use max() of the input and output and
> somehow use that?"
> 
> One might think this code would work:
> if (output < input) {
>  output = input
> }
> else {
>  output -= 0.01
> }
> 
> This is a beginner's mistake. This sort of design will create awful
> amounts of zipper noise. You don't want that. You want to use a
> filter.
> 
> Brian, you first correct Tom on his mistake, then make the same
> mistake yourself elsewhere, then agree with Scott who has made the
> same mistake :)
> 
> 4. "because peak detection allows the output to suddenly change if the
> input is greater than the hold value, and any time a digital algorithm
> allows a sudden discontinuity in its output you have the potential for
> non-band-limited square wave type aliasing"
> 
> Not true. You just use a normal (asymmetric) filter for the peak hold,
> like you usually would, and set its time constant for rising signals
> so that it reacts instantly.
> 
> Your logic is flawed because you're thinking in terms of the "if/else"
> model I described above. It doesn't hold. The transition you describe
> will not be abrupt. There will still be phase delay like in any usual
> filter (that's a good thing), and this phase delay will give the
> filter just enough time buffer for things to happen relatively
> smoothly. It's a different question, though, whether your filter
> aliases *as a whole*.
> 
> 5. "Assuming your audio signal is limited to 20 kHz maximum, and your
> sample rate is 44.1 kHz, then your envelope signal need only have no
> aliases above 2.05 kHz. Bingo, no audio aliasing"
> 
> That's not how it works. If your signal picked up aliasing somewhere
> earlier, it's not going to magically get rid of it by becoming
> band-limited later on. If you have frequencies within your 2.05 kHz
> that do not agree with the harmony of your program, you're going to
> hear it, simple as that. If you said...
> 
> "multiplying two signals, such that the sum of their bandwidth is less
> than fs/2, will not create new aliases"
> 
> ...then I might be partial to agreeing, although due to numerical
> inaccuracies that's not really true either. Depending on what you're
> working with, YMMV.
> 
> 6. "I do not see how that is a problem with an envelope detector. The
> goal is to produce a control signal with basically "no" frequency
> components, or at least they should be as close to DC as possible. In
> practice, there will always be undesired low-frequency components in
> the envelope. My point here is that aliased components are no more
> problematic than non-aliased frequency components."
> 
> Only if you revel in muddy sound that becomes more and more occluded
> and unclear the more processing you apply. It's a fact that
> harmonically unrelated sounds present in your program will deteriorate
> its quality. They're much worse than e.g. white noise, because they're
> concentrated in single, discernible tones, which play melodies that go
> completely against the sounds that you are playing on the keyboard.
> I'm not sure how to better say this; if anyone here is not convinced
> that having this is bad, please read the previous sentence 50 times.
> 
> 7. "The remaining frequency components are all equally undesired, and
> it doesn't really matter whether they're aliases or not, so long as
> they are low enough in frequency."
> 
> It's exactly the other way around: aliased low-frequency components
> are very undesirable and are glaring problems in the final sound.
> Harmonically related frequency components don't work against the sound
> you're building, and are often very desirable. At worst they are
> harmless because they become invisible when surrounded by lots of
> other stuff playing in the same harmony.
> 
> 8. "Perhaps I don't even need to bother rectifying the signal. Why not
> just keep a check on the negative peaks like the positive ones, and
> then use some average of the two as the final output. JP said his
> signal might well be asymmetric (which is not at all uncommon), so we
> can't assume the two envelopes will be the same."
> 
> This is DSP, not analog fx design. DSP is a harsh mistress. You can do
> a lot of learning by experimenting in analog electronics and it works
> out well. The signal degradation which happens when you make errors is
> very musical: nth harmonic distortion, homogenic noise, cross-talk of
> content that is harmonically related. It's all nice and very
> forgiving. In DSP, you have very, very bad degradation modes: i)
> aliasing - takes your melody and makes it sound like it's just random
> notes; ii) through-zero distortion - adds mud and reduces clarity in a
> way that's not related to your sound in any way; iii) time delay -
> makes sure that you will never be able to mix your sound correctly
> again after you've split it; iv) numerical errors - similar to
> aliasing, those create junk in your sound that is harmonically
> unrelated to the sound you're trying to achieve (examples include bit
> errors, zipper noise, clicks, hard clipping, etc).
> 
> Be warned: you really have to go by the book on DSP. There are very
> good books online; Julius Orion Smith's website is the best,
> most-recommended classic. You simply can't "common-sense" your way
> through digital audio. It's daunting. You'll be making errors, and the
> worst thing is that in analog when you make errors they're obvious
> whereas in digital, when you make errors, you might not notice until
> after you've shipped the box; in the end people figure it out though
> and just don't like your design. Granted that doesn't matter if you're
> selling for the masses that buy microcorgi and other junk like that;
> but let's maybe leave this kind of talk off this list and try to think
> about what it takes to do an honest job. As someone said, if
> something's worth doing, it's worth doing well.
> 
> Cheers,
> D.
> 
> On Sat, Mar 30, 2013 at 7:24 AM, Paul Perry <pfperry at melbpc.org.au> wrote:
>> A little philosophy on envelope followers:
>> It's easy to see what the envelope should look like - after the original
>> signal has been! But if you take a typical input signal, graph it, and then
>> cover it with a card, and try to draw an envelope as you slide the card to
>> the right........... you might see that is is not just 'difficult', but
>> actually 'impossible' to have a circuit that behaves as you would wish.
>> 
>> In my experience, there is no envelope follower that suits all applications.
>> In particular, percussion and guitar are very different. I don't doubt that
>> Harry's circuit is ideal for a guitar.. and that if he could have got away
>> with fewer components, he would have (remember, a hex guitar has SIX
>> outputs!)
>> 
>> On the bright side, I have also found that if you are using the derived
>> envelope to control a filter or VCA, the actual audio effect can be quite
>> musically useful, even though the envelope itself looks really unpromising
>> on the scope. In particular, full wave rectification was not much better
>> than half wave.
>> 
>> Real problems arise when the frequencies being enveloped are very low, in
>> this case the low pass filtering is in conflict with the envelope being
>> derived. Again, it is a matter of knowing what the results are going to be
>> with a practical envelope, rather than knocking oneself out trying to
>> produce an 'ideal' envelope.
>> 
>> paul perry Melbourne Australia
>> 
>> 
>> 
>> _______________________________________________
>> Synth-diy mailing list
>> Synth-diy at dropmix.xs4all.nl
>> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy



From wackyvorlon at me.com  Sat Mar 30 17:43:44 2013
From: wackyvorlon at me.com (Paul Anderson)
Date: Sat, 30 Mar 2013 12:43:44 -0400
Subject: [sdiy] Fast envelope follower circuit needed..
In-Reply-To: <CA+9GZUhhq-q4yWB5Tz2Rb8uUgnO6m7Pyb85qpn_F2EdMHrfS6w@mail.gmail.com>
References: <324c9c6efbb87ef612fdd8cdc2ccad26@oricom.ca>
	<7D00642A-814D-4BBC-B510-CDAC1630728D@electricdruid.net>
	<BLU0-SMTP338D2FA8904A3F6D5D401C9C8D20@phx.gbl>
	<541FA361-13D0-4D41-8296-E24F2DB98CB4@electricdruid.net>
	<309EB6A9-742D-4A9D-8099-69D08A50A07C@sounds.wa.com>
	<CA+9GZUjOi+AjM3oegAonSR4eRQpwXNzrOHA+ZLGx9O9sVKd+DQ@mail.gmail.com>
	<4F6CAB3A-6477-4BFF-8100-27DBCBBAA68A@electricdruid.net>
	<CA+9GZUh1UZ-KmwRQ3UzG-SVdwv+K7MQ1rBz=ZvB1cTOmPtpjag@mail.gmail.com>
	<52800F3374954107AC3E6D91E2C76012@ownerf0d9273f5>
	<CA+9GZUhhq-q4yWB5Tz2Rb8uUgnO6m7Pyb85qpn_F2EdMHrfS6w@mail.gmail.com>
Message-ID: <D05254C1-5C2C-4FA8-B8FF-1F83008AC7BA@me.com>

To add to these comments, the cut off on most filters is often defined at the point of 3dB of attenuation. 3dB roughly equates to a doubling or halving of amplitude. So a 3dB increase on a 6 volt signal makes it 12v. 

Filters look very simple, but the theory is surprisingly complex. 

Sent from my iPhone

On 2013-03-30, at 10:19 AM, cheater cheater <cheater00 at gmail.com> wrote:

> Hi guys,
> there are several points from Tom, Scott, and Brian here I'd like to
> address. I hope it's ok, I would like to point out some errors to
> spare you frustration in the future.
> 
> 1. "the filter does not care about the spikes in the valleys"
> 
> that's incorrect. Bear in mind that this high-frequency content is not
> removed in a low-pass filter, it is attenuated. Quite often the
> rejection is less than the theoretical maximum for many reasons
> (numerical error, incorrect algorithm implementation, etc). In effect,
> that "rejected" signal is still there.
> 
> Especially if you use non-linearities later in your signal path (that
> is, always), whatever program you modulate with such a tainted
> envelope will sound worse than it could otherwise.
> 
> 2. "analog rectification creates wide-band energy bursts anyways"
> 
> not an accurate picture of the situation. In analog, this sound will
> be harmonically related to your input, which is a good thing. In
> digital, it will work exactly against it, which is very bad. Besides,
> the turn-on and turn-off speed is limited, it's not when using abs()
> in a digital signal, but I don't know what the exact difference in
> bandwidth limitation is in analog - it's unlimited in digital (with
> the caveat that it gets mirrored around fs/2 and 0Hz, perhaps several
> times)
> 
> 3. "the peak hold doesn't care about the little valleys"
> 
> not true at all. I believe this comes from the conceptual
> simplification of the filter, which could go like this:
> (i) when the input signal is above the current output, raise the output to match
> (ii) when the input signal is below the current output, drop the
> output at a constant rate
> 
> that is extremely inaccurate. What happens is more similar to this:
> (iii) measure the distance between the current output and current
> input. Move so that during a period of time, say 1 second, you cover a
> fixed percentage of this distance, directly related to a number called
> the time constant. For example, cover 90% in 1 second.
> 
> What does this mean? Say your output is at 1V, and your input is a
> sinewave oscillation between 0V and 0.01V. As your output is falling,
> it'll fall relatively faster when the input is near 0V (because the
> distance is higher), and it'll fall relatively slower when the input
> signal is *closer* to the output signal (i.e. when the input is at its
> 0.01V maximum).
> 
> This is easily described by saying that the peak hold is just a filter
> as well, but it doesn't quite picture it, that's why I described the
> operation above.
> 
> You might ask: "But can't I just use max() of the input and output and
> somehow use that?"
> 
> One might think this code would work:
> if (output < input) {
>  output = input
> }
> else {
>  output -= 0.01
> }
> 
> This is a beginner's mistake. This sort of design will create awful
> amounts of zipper noise. You don't want that. You want to use a
> filter.
> 
> Brian, you first correct Tom on his mistake, then make the same
> mistake yourself elsewhere, then agree with Scott who has made the
> same mistake :)
> 
> 4. "because peak detection allows the output to suddenly change if the
> input is greater than the hold value, and any time a digital algorithm
> allows a sudden discontinuity in its output you have the potential for
> non-band-limited square wave type aliasing"
> 
> Not true. You just use a normal (asymmetric) filter for the peak hold,
> like you usually would, and set its time constant for rising signals
> so that it reacts instantly.
> 
> Your logic is flawed because you're thinking in terms of the "if/else"
> model I described above. It doesn't hold. The transition you describe
> will not be abrupt. There will still be phase delay like in any usual
> filter (that's a good thing), and this phase delay will give the
> filter just enough time buffer for things to happen relatively
> smoothly. It's a different question, though, whether your filter
> aliases *as a whole*.
> 
> 5. "Assuming your audio signal is limited to 20 kHz maximum, and your
> sample rate is 44.1 kHz, then your envelope signal need only have no
> aliases above 2.05 kHz. Bingo, no audio aliasing"
> 
> That's not how it works. If your signal picked up aliasing somewhere
> earlier, it's not going to magically get rid of it by becoming
> band-limited later on. If you have frequencies within your 2.05 kHz
> that do not agree with the harmony of your program, you're going to
> hear it, simple as that. If you said...
> 
> "multiplying two signals, such that the sum of their bandwidth is less
> than fs/2, will not create new aliases"
> 
> ...then I might be partial to agreeing, although due to numerical
> inaccuracies that's not really true either. Depending on what you're
> working with, YMMV.
> 
> 6. "I do not see how that is a problem with an envelope detector. The
> goal is to produce a control signal with basically "no" frequency
> components, or at least they should be as close to DC as possible. In
> practice, there will always be undesired low-frequency components in
> the envelope. My point here is that aliased components are no more
> problematic than non-aliased frequency components."
> 
> Only if you revel in muddy sound that becomes more and more occluded
> and unclear the more processing you apply. It's a fact that
> harmonically unrelated sounds present in your program will deteriorate
> its quality. They're much worse than e.g. white noise, because they're
> concentrated in single, discernible tones, which play melodies that go
> completely against the sounds that you are playing on the keyboard.
> I'm not sure how to better say this; if anyone here is not convinced
> that having this is bad, please read the previous sentence 50 times.
> 
> 7. "The remaining frequency components are all equally undesired, and
> it doesn't really matter whether they're aliases or not, so long as
> they are low enough in frequency."
> 
> It's exactly the other way around: aliased low-frequency components
> are very undesirable and are glaring problems in the final sound.
> Harmonically related frequency components don't work against the sound
> you're building, and are often very desirable. At worst they are
> harmless because they become invisible when surrounded by lots of
> other stuff playing in the same harmony.
> 
> 8. "Perhaps I don't even need to bother rectifying the signal. Why not
> just keep a check on the negative peaks like the positive ones, and
> then use some average of the two as the final output. JP said his
> signal might well be asymmetric (which is not at all uncommon), so we
> can't assume the two envelopes will be the same."
> 
> This is DSP, not analog fx design. DSP is a harsh mistress. You can do
> a lot of learning by experimenting in analog electronics and it works
> out well. The signal degradation which happens when you make errors is
> very musical: nth harmonic distortion, homogenic noise, cross-talk of
> content that is harmonically related. It's all nice and very
> forgiving. In DSP, you have very, very bad degradation modes: i)
> aliasing - takes your melody and makes it sound like it's just random
> notes; ii) through-zero distortion - adds mud and reduces clarity in a
> way that's not related to your sound in any way; iii) time delay -
> makes sure that you will never be able to mix your sound correctly
> again after you've split it; iv) numerical errors - similar to
> aliasing, those create junk in your sound that is harmonically
> unrelated to the sound you're trying to achieve (examples include bit
> errors, zipper noise, clicks, hard clipping, etc).
> 
> Be warned: you really have to go by the book on DSP. There are very
> good books online; Julius Orion Smith's website is the best,
> most-recommended classic. You simply can't "common-sense" your way
> through digital audio. It's daunting. You'll be making errors, and the
> worst thing is that in analog when you make errors they're obvious
> whereas in digital, when you make errors, you might not notice until
> after you've shipped the box; in the end people figure it out though
> and just don't like your design. Granted that doesn't matter if you're
> selling for the masses that buy microcorgi and other junk like that;
> but let's maybe leave this kind of talk off this list and try to think
> about what it takes to do an honest job. As someone said, if
> something's worth doing, it's worth doing well.
> 
> Cheers,
> D.
> 
> On Sat, Mar 30, 2013 at 7:24 AM, Paul Perry <pfperry at melbpc.org.au> wrote:
>> A little philosophy on envelope followers:
>> It's easy to see what the envelope should look like - after the original
>> signal has been! But if you take a typical input signal, graph it, and then
>> cover it with a card, and try to draw an envelope as you slide the card to
>> the right........... you might see that is is not just 'difficult', but
>> actually 'impossible' to have a circuit that behaves as you would wish.
>> 
>> In my experience, there is no envelope follower that suits all applications.
>> In particular, percussion and guitar are very different. I don't doubt that
>> Harry's circuit is ideal for a guitar.. and that if he could have got away
>> with fewer components, he would have (remember, a hex guitar has SIX
>> outputs!)
>> 
>> On the bright side, I have also found that if you are using the derived
>> envelope to control a filter or VCA, the actual audio effect can be quite
>> musically useful, even though the envelope itself looks really unpromising
>> on the scope. In particular, full wave rectification was not much better
>> than half wave.
>> 
>> Real problems arise when the frequencies being enveloped are very low, in
>> this case the low pass filtering is in conflict with the envelope being
>> derived. Again, it is a matter of knowing what the results are going to be
>> with a practical envelope, rather than knocking oneself out trying to
>> produce an 'ideal' envelope.
>> 
>> paul perry Melbourne Australia
>> 
>> 
>> 
>> _______________________________________________
>> Synth-diy mailing list
>> Synth-diy at dropmix.xs4all.nl
>> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy


From cheater00 at gmail.com  Sat Mar 30 19:24:54 2013
From: cheater00 at gmail.com (cheater cheater)
Date: Sat, 30 Mar 2013 19:24:54 +0100
Subject: [sdiy] Fast envelope follower circuit needed..
In-Reply-To: <D38401AC-5379-46F7-A221-9032544781AC@electricdruid.net>
References: <324c9c6efbb87ef612fdd8cdc2ccad26@oricom.ca>
	<7D00642A-814D-4BBC-B510-CDAC1630728D@electricdruid.net>
	<BLU0-SMTP338D2FA8904A3F6D5D401C9C8D20@phx.gbl>
	<541FA361-13D0-4D41-8296-E24F2DB98CB4@electricdruid.net>
	<309EB6A9-742D-4A9D-8099-69D08A50A07C@sounds.wa.com>
	<CA+9GZUjOi+AjM3oegAonSR4eRQpwXNzrOHA+ZLGx9O9sVKd+DQ@mail.gmail.com>
	<4F6CAB3A-6477-4BFF-8100-27DBCBBAA68A@electricdruid.net>
	<CA+9GZUh1UZ-KmwRQ3UzG-SVdwv+K7MQ1rBz=ZvB1cTOmPtpjag@mail.gmail.com>
	<52800F3374954107AC3E6D91E2C76012@ownerf0d9273f5>
	<CA+9GZUhhq-q4yWB5Tz2Rb8uUgnO6m7Pyb85qpn_F2EdMHrfS6w@mail.gmail.com>
	<D38401AC-5379-46F7-A221-9032544781AC@electricdruid.net>
Message-ID: <CA+9GZUhtSsSEgqhEq8KwMZczFFi_B9UGjowMit+CHOu-OdgpvQ@mail.gmail.com>

Hi Tom,
just like I said at the very beginning:

> Do the rectification in analog.

Then, use one of Julius Smith's filter cook-books to get a good
asymmetric low-pass filter, 2nd-order, and feed that directly from the
ADC. Set the rise time and fall time separately, call them "attack"
and "release".

If your PIC's ADC does not have one, use a brickwall filter chip in
front of it to prevent aliasing when digitizing the signal.

To improve quality you can up-sample the signal from the ADC, run the
asymmetric filter at that higher sample rate, and then lower the
sample rate.

With some thought (beyond my current rusty knowledge of filters) it
should be possible to have the upsampling, filtering, and downsampling
in one single filter, with very nice numerical properties when it
comes to stability and cycle efficiency. Listen to Richie's advice if
he offers any.

In fact, digital filters are a very good fit for such analysis tasks -
using the best of both worlds (rectifying in analog, filtering in
digital) you might get a superb envelope follower, unlike anything on
the market right now.

Of course, make sure to have a nice high-current output on the DAC side.

It doesn't hurt to use a dedicated ADC, since onboard PIC stuff would
be more suited to encoders than to audio. This means reduced quality
when it comes to fast (audio-rate) signals and usually cross-over
distortion.

Cheers,
D.

On Sat, Mar 30, 2013 at 5:22 PM, Tom Wiltshire <tom at electricdruid.net> wrote:
> So how would *you* solve the problem, Damian?
>
> So far, you've spent a lot of effort telling everyone who's made a suggestion about the problem why they're wrong, and why it couldn't possibly work, but I'm not seeing much in the way of better suggestions.
>
> Tom
>
>
> On 30 Mar 2013, at 14:19, cheater cheater <cheater00 at gmail.com> wrote:
>
>> Hi guys,
>> there are several points from Tom, Scott, and Brian here I'd like to
>> address. I hope it's ok, I would like to point out some errors to
>> spare you frustration in the future.
>>
>> 1. "the filter does not care about the spikes in the valleys"
>>
>> that's incorrect. Bear in mind that this high-frequency content is not
>> removed in a low-pass filter, it is attenuated. Quite often the
>> rejection is less than the theoretical maximum for many reasons
>> (numerical error, incorrect algorithm implementation, etc). In effect,
>> that "rejected" signal is still there.
>>
>> Especially if you use non-linearities later in your signal path (that
>> is, always), whatever program you modulate with such a tainted
>> envelope will sound worse than it could otherwise.
>>
>> 2. "analog rectification creates wide-band energy bursts anyways"
>>
>> not an accurate picture of the situation. In analog, this sound will
>> be harmonically related to your input, which is a good thing. In
>> digital, it will work exactly against it, which is very bad. Besides,
>> the turn-on and turn-off speed is limited, it's not when using abs()
>> in a digital signal, but I don't know what the exact difference in
>> bandwidth limitation is in analog - it's unlimited in digital (with
>> the caveat that it gets mirrored around fs/2 and 0Hz, perhaps several
>> times)
>>
>> 3. "the peak hold doesn't care about the little valleys"
>>
>> not true at all. I believe this comes from the conceptual
>> simplification of the filter, which could go like this:
>> (i) when the input signal is above the current output, raise the output to match
>> (ii) when the input signal is below the current output, drop the
>> output at a constant rate
>>
>> that is extremely inaccurate. What happens is more similar to this:
>> (iii) measure the distance between the current output and current
>> input. Move so that during a period of time, say 1 second, you cover a
>> fixed percentage of this distance, directly related to a number called
>> the time constant. For example, cover 90% in 1 second.
>>
>> What does this mean? Say your output is at 1V, and your input is a
>> sinewave oscillation between 0V and 0.01V. As your output is falling,
>> it'll fall relatively faster when the input is near 0V (because the
>> distance is higher), and it'll fall relatively slower when the input
>> signal is *closer* to the output signal (i.e. when the input is at its
>> 0.01V maximum).
>>
>> This is easily described by saying that the peak hold is just a filter
>> as well, but it doesn't quite picture it, that's why I described the
>> operation above.
>>
>> You might ask: "But can't I just use max() of the input and output and
>> somehow use that?"
>>
>> One might think this code would work:
>> if (output < input) {
>>  output = input
>> }
>> else {
>>  output -= 0.01
>> }
>>
>> This is a beginner's mistake. This sort of design will create awful
>> amounts of zipper noise. You don't want that. You want to use a
>> filter.
>>
>> Brian, you first correct Tom on his mistake, then make the same
>> mistake yourself elsewhere, then agree with Scott who has made the
>> same mistake :)
>>
>> 4. "because peak detection allows the output to suddenly change if the
>> input is greater than the hold value, and any time a digital algorithm
>> allows a sudden discontinuity in its output you have the potential for
>> non-band-limited square wave type aliasing"
>>
>> Not true. You just use a normal (asymmetric) filter for the peak hold,
>> like you usually would, and set its time constant for rising signals
>> so that it reacts instantly.
>>
>> Your logic is flawed because you're thinking in terms of the "if/else"
>> model I described above. It doesn't hold. The transition you describe
>> will not be abrupt. There will still be phase delay like in any usual
>> filter (that's a good thing), and this phase delay will give the
>> filter just enough time buffer for things to happen relatively
>> smoothly. It's a different question, though, whether your filter
>> aliases *as a whole*.
>>
>> 5. "Assuming your audio signal is limited to 20 kHz maximum, and your
>> sample rate is 44.1 kHz, then your envelope signal need only have no
>> aliases above 2.05 kHz. Bingo, no audio aliasing"
>>
>> That's not how it works. If your signal picked up aliasing somewhere
>> earlier, it's not going to magically get rid of it by becoming
>> band-limited later on. If you have frequencies within your 2.05 kHz
>> that do not agree with the harmony of your program, you're going to
>> hear it, simple as that. If you said...
>>
>> "multiplying two signals, such that the sum of their bandwidth is less
>> than fs/2, will not create new aliases"
>>
>> ...then I might be partial to agreeing, although due to numerical
>> inaccuracies that's not really true either. Depending on what you're
>> working with, YMMV.
>>
>> 6. "I do not see how that is a problem with an envelope detector. The
>> goal is to produce a control signal with basically "no" frequency
>> components, or at least they should be as close to DC as possible. In
>> practice, there will always be undesired low-frequency components in
>> the envelope. My point here is that aliased components are no more
>> problematic than non-aliased frequency components."
>>
>> Only if you revel in muddy sound that becomes more and more occluded
>> and unclear the more processing you apply. It's a fact that
>> harmonically unrelated sounds present in your program will deteriorate
>> its quality. They're much worse than e.g. white noise, because they're
>> concentrated in single, discernible tones, which play melodies that go
>> completely against the sounds that you are playing on the keyboard.
>> I'm not sure how to better say this; if anyone here is not convinced
>> that having this is bad, please read the previous sentence 50 times.
>>
>> 7. "The remaining frequency components are all equally undesired, and
>> it doesn't really matter whether they're aliases or not, so long as
>> they are low enough in frequency."
>>
>> It's exactly the other way around: aliased low-frequency components
>> are very undesirable and are glaring problems in the final sound.
>> Harmonically related frequency components don't work against the sound
>> you're building, and are often very desirable. At worst they are
>> harmless because they become invisible when surrounded by lots of
>> other stuff playing in the same harmony.
>>
>> 8. "Perhaps I don't even need to bother rectifying the signal. Why not
>> just keep a check on the negative peaks like the positive ones, and
>> then use some average of the two as the final output. JP said his
>> signal might well be asymmetric (which is not at all uncommon), so we
>> can't assume the two envelopes will be the same."
>>
>> This is DSP, not analog fx design. DSP is a harsh mistress. You can do
>> a lot of learning by experimenting in analog electronics and it works
>> out well. The signal degradation which happens when you make errors is
>> very musical: nth harmonic distortion, homogenic noise, cross-talk of
>> content that is harmonically related. It's all nice and very
>> forgiving. In DSP, you have very, very bad degradation modes: i)
>> aliasing - takes your melody and makes it sound like it's just random
>> notes; ii) through-zero distortion - adds mud and reduces clarity in a
>> way that's not related to your sound in any way; iii) time delay -
>> makes sure that you will never be able to mix your sound correctly
>> again after you've split it; iv) numerical errors - similar to
>> aliasing, those create junk in your sound that is harmonically
>> unrelated to the sound you're trying to achieve (examples include bit
>> errors, zipper noise, clicks, hard clipping, etc).
>>
>> Be warned: you really have to go by the book on DSP. There are very
>> good books online; Julius Orion Smith's website is the best,
>> most-recommended classic. You simply can't "common-sense" your way
>> through digital audio. It's daunting. You'll be making errors, and the
>> worst thing is that in analog when you make errors they're obvious
>> whereas in digital, when you make errors, you might not notice until
>> after you've shipped the box; in the end people figure it out though
>> and just don't like your design. Granted that doesn't matter if you're
>> selling for the masses that buy microcorgi and other junk like that;
>> but let's maybe leave this kind of talk off this list and try to think
>> about what it takes to do an honest job. As someone said, if
>> something's worth doing, it's worth doing well.
>>
>> Cheers,
>> D.
>>
>> On Sat, Mar 30, 2013 at 7:24 AM, Paul Perry <pfperry at melbpc.org.au> wrote:
>>> A little philosophy on envelope followers:
>>> It's easy to see what the envelope should look like - after the original
>>> signal has been! But if you take a typical input signal, graph it, and then
>>> cover it with a card, and try to draw an envelope as you slide the card to
>>> the right........... you might see that is is not just 'difficult', but
>>> actually 'impossible' to have a circuit that behaves as you would wish.
>>>
>>> In my experience, there is no envelope follower that suits all applications.
>>> In particular, percussion and guitar are very different. I don't doubt that
>>> Harry's circuit is ideal for a guitar.. and that if he could have got away
>>> with fewer components, he would have (remember, a hex guitar has SIX
>>> outputs!)
>>>
>>> On the bright side, I have also found that if you are using the derived
>>> envelope to control a filter or VCA, the actual audio effect can be quite
>>> musically useful, even though the envelope itself looks really unpromising
>>> on the scope. In particular, full wave rectification was not much better
>>> than half wave.
>>>
>>> Real problems arise when the frequencies being enveloped are very low, in
>>> this case the low pass filtering is in conflict with the envelope being
>>> derived. Again, it is a matter of knowing what the results are going to be
>>> with a practical envelope, rather than knocking oneself out trying to
>>> produce an 'ideal' envelope.
>>>
>>> paul perry Melbourne Australia
>>>
>>>
>>>
>>> _______________________________________________
>>> Synth-diy mailing list
>>> Synth-diy at dropmix.xs4all.nl
>>> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>> _______________________________________________
>> Synth-diy mailing list
>> Synth-diy at dropmix.xs4all.nl
>> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>


From tu at alphalink.com.au  Sat Mar 30 23:32:19 2013
From: tu at alphalink.com.au (Tristan)
Date: Sun, 31 Mar 2013 09:32:19 +1100
Subject: [sdiy] Fast envelope follower circuit needed..
In-Reply-To: <D05254C1-5C2C-4FA8-B8FF-1F83008AC7BA@me.com>
References: <324c9c6efbb87ef612fdd8cdc2ccad26@oricom.ca>
	<7D00642A-814D-4BBC-B510-CDAC1630728D@electricdruid.net>
	<BLU0-SMTP338D2FA8904A3F6D5D401C9C8D20@phx.gbl>
	<541FA361-13D0-4D41-8296-E24F2DB98CB4@electricdruid.net>
	<309EB6A9-742D-4A9D-8099-69D08A50A07C@sounds.wa.com>
	<CA+9GZUjOi+AjM3oegAonSR4eRQpwXNzrOHA+ZLGx9O9sVKd+DQ@mail.gmail.com>
	<4F6CAB3A-6477-4BFF-8100-27DBCBBAA68A@electricdruid.net>
	<CA+9GZUh1UZ-KmwRQ3UzG-SVdwv+K7MQ1rBz=ZvB1cTOmPtpjag@mail.gmail.com>
	<52800F3374954107AC3E6D91E2C76012@ownerf0d9273f5>
	<CA+9GZUhhq-q4yWB5Tz2Rb8uUgnO6m7Pyb85qpn_F2EdMHrfS6w@mail.gmail.com>
	<D05254C1-5C2C-4FA8-B8FF-1F83008AC7BA@me.com>
Message-ID: <1364682739.515767f34d66c@webmail.chariot.com.au>

A 3dB attenuation actually approximates a halving of the signal power, as power is related by the 
square of the amplitude. Changing a 6 volt signal to 12 volts is an approximately 6dB increase in 
amplitude.

/Tristan  

On Sun, Mar 31st, 2013 at 3:43 AM, Paul Anderson <wackyvorlon at me.com> wrote:

> To add to these comments, the cut off on most filters is often defined at the
> point of 3dB of attenuation. 3dB roughly equates to a doubling or halving of
> amplitude. So a 3dB increase on a 6 volt signal makes it 12v. 
> 
> Filters look very simple, but the theory is surprisingly complex. 
> 
> Sent from my iPhone
> 



From dixon at mail.ubc.ca  Sat Mar 30 23:51:09 2013
From: dixon at mail.ubc.ca (David G Dixon)
Date: Sat, 30 Mar 2013 15:51:09 -0700
Subject: [sdiy] Fast envelope follower circuit needed..
In-Reply-To: <1364682739.515767f34d66c@webmail.chariot.com.au>
References: <324c9c6efbb87ef612fdd8cdc2ccad26@oricom.ca><7D00642A-814D-4BBC-B510-CDAC1630728D@electricdruid.net><BLU0-SMTP338D2FA8904A3F6D5D401C9C8D20@phx.gbl><541FA361-13D0-4D41-8296-E24F2DB98CB4@electricdruid.net><309EB6A9-742D-4A9D-8099-69D08A50A07C@sounds.wa.com><CA+9GZUjOi+AjM3oegAonSR4eRQpwXNzrOHA+ZLGx9O9sVKd+DQ@mail.gmail.com><4F6CAB3A-6477-4BFF-8100-27DBCBBAA68A@electricdruid.net><CA+9GZUh1UZ-KmwRQ3UzG-SVdwv+K7MQ1rBz=ZvB1cTOmPtpjag@mail.gmail.com><52800F3374954107AC3E6D91E2C76012@ownerf0d9273f5><CA+9GZUhhq-q4yWB5Tz2Rb8uUgnO6m7Pyb85qpn_F2EdMHrfS6w@mail.gmail.com><D05254C1-5C2C-4FA8-B8FF-1F83008AC7BA@me.com>
	<1364682739.515767f34d66c@webmail.chariot.com.au>
Message-ID: <47813FB6E9D14D28ABAA8C5EB7BF78C7@galvanox01>

Yes, 3dB of signal attenuation is root(2)/2, or 70.7%. 

> -----Original Message-----
> From: synth-diy-bounces at dropmix.xs4all.nl 
> [mailto:synth-diy-bounces at dropmix.xs4all.nl] On Behalf Of Tristan
> Sent: Saturday, March 30, 2013 3:32 PM
> To: synthdiy diy
> Subject: Re: [sdiy] Fast envelope follower circuit needed..
> 
> A 3dB attenuation actually approximates a halving of the 
> signal power, as power is related by the square of the 
> amplitude. Changing a 6 volt signal to 12 volts is an 
> approximately 6dB increase in amplitude.
> 
> /Tristan  
> 
> On Sun, Mar 31st, 2013 at 3:43 AM, Paul Anderson 
> <wackyvorlon at me.com> wrote:
> 
> > To add to these comments, the cut off on most filters is 
> often defined 
> > at the point of 3dB of attenuation. 3dB roughly equates to 
> a doubling 
> > or halving of amplitude. So a 3dB increase on a 6 volt 
> signal makes it 12v.
> > 
> > Filters look very simple, but the theory is surprisingly complex. 
> > 
> > Sent from my iPhone
> > 
> 
> _______________________________________________
> Synth-diy mailing list
> Synth-diy at dropmix.xs4all.nl
> http://dropmix.xs4all.nl/mailman/listinfo/synth-diy



From rburnett at richieburnett.co.uk  Sun Mar 31 01:06:35 2013
From: rburnett at richieburnett.co.uk (Richie Burnett)
Date: Sun, 31 Mar 2013 00:06:35 +0000
Subject: [sdiy] Fast envelope follower circuit needed..
Message-ID: <pknylhgatv7n8fqtvnmndgjt.1364687583444@email.android.com>

The Hilbert transform produces an output where every spectral component of the input is shifted by 90 degrees. (Or more practically produces two new outputs where the spectral components in one output lag the other by 90 degrees over the frequency range of interest.) You then square the value of each output, add them together and take the square-root to get the amplitude envelope. Imagine a continuous sinewave input, phase shifted by 90 degrees to give a cosine wave. If you square the sine, square the cosine, add the squares together, and square-root you get the amplitude envelope. In this case it's a smooth line of unity if the input sinewave isn't modulated.

Provided the phase shift network maintains an accurate 90 degree phase shift over the frequency range of interest, the ripple in the calculated envelope output is very small.

People on this list have previously described various clever optimisations to ease the CPU burden of the magnitude calculation by either estimating the magnitude or speeding up the square-root calculation.

If you're interested in the multi-rate FIR filter stuff do a google search for "multi-rate filter design momentum data systems". If you're into DSP programming it's worth downloading the PDF and keeping these tricks in your back pocket for when you need then most.

-Richie,



Sent from my Sony Ericsson Xperia ray

cheater cheater <cheater00 at gmail.com> wrote:

>Richie,
>
>On Thu, Mar 28, 2013 at 11:22 PM, Richie Burnett
><rburnett at richieburnett.co.uk> wrote:
>> The Hilbert transform is a classic tool for envelope extraction using DSP. Envelope following is really what electrical engineers would call AM demodulation, and has existed in various forms since the dawn of radio.
>
>That's interesting - where can I read up on how this is proven
>mathematically? I've never heard of this property of the Hilbert
>transform. Really interesting to me. And in fact, very surprising,
>since the Hilbert transform is kind of like an all-pass filter.. a
>chorus you might say. Just a bit of conceptual difference and what you
>get is completely different!
>
>> Also, FIR filters with cutoff frequencies which are very low compared to the sample rate and result in really long impulse responses (hundreds or thousands of taps) can be used effectively if implemented as polyphase filters. A 2000 tap FIR filter only requires the same amount of DSP grunt as a 20 tap filter if you can drop the sample rate of the output by a factor of 1/100.
>
>Very smart. I was sure you could do the resampling and enveloping in
>one step - but I wasn't sure if I should recommend it, because I've
>never done this in practice. Thanks for clearing that up!
>
>> cheater cheater <cheater00 at gmail.com> wrote:
>>
>>>Hi Tom,
>>>a few comments:
>>>
>>>On Thu, Mar 28, 2013 at 7:24 PM, Tom Wiltshire <tom at electricdruid.net> wrote:
>>>> Hi Dan,
>>>>
>>>> On 28 Mar 2013, at 13:55, Dan Snazelle <subjectivity at hotmail.com> wrote:
>>>>
>>>>>
>>>>> Though i work with c//arduino/AVRS not pics, this sounds very appealing! Can anyone recommend a place to learn about writing C code for an envelope follower? How do you do it? Shift signal to 0-5, then......
>>>>>
>>>>>
>>>>> Thanks!!!
>>>>
>>>> The first thing after that would be to do the full wave rectification. Either dump the sign bit, if you've got signed data, or invert the data if the highest bit is clear if it's not signed - same thing, different representation.
>>>
>>>I strongly advise against doing this. It will create insane amounts of
>>>aliasing. Do the rectification in analog. In fact full-wave
>>>rectification is the most difficult thing to do when modelling analog
>>>circuits digitally. Doing it anywhere near a level acceptable even for
>>>the non-discriminating musician would require processing power orders
>>>of magnitude larger than what a lowly pic can offer you. And in analog
>>>it's just a few junctions.
>>>
>>>BTW, you can't really rectify a signal that's unipolar already.. just
>>>a minor nitpick ;)
>>>
>>>> For the smoothing, I can imagine a couple of approaches:
>>>>
>>>> 1) Genuine honest-to-goodness digital filtering.
>>>> There's a ton of stuff about this all over the web. But since we're talking about bottom-end chips, most of it is totally out of reach. We're not going to be running long FIR filters. But we could perhaps use an IIR filter, or a FIR with a very low sample rate, to keep it manageable.
>>>
>>>You can't change the sample rate, unless you mean you would downsample
>>>the signal first. But then you'll have to have steep bandwidth
>>>limiting filters and a filter for the down-sampling itself, which
>>>means you do ten times the work you think you're saving by using lower
>>>sampling rates. If you don't downsample you get aliasing, and then
>>>you're detecting the envelope very incorrectly. It can only get you
>>>fairly small speed-ups anyways before it stops being useful at all.
>>>
>>>An FIR with a fairly short impulse is fairly easy to manage, and on
>>>something like a dspic you can do even longer ones. However IIR is the
>>>usual way to go with envelope detectors if I'm not mistaken.
>>>
>>>> 2) Peak detection.
>>>> This might be a good way to go. Keeping track of what the last couple of samples are would be easy, and would allow us to work out the current slope of the waveform. When the current sample is less than the previous one, the waveform has started heading back down, so we'd hold the peak value.
>>>> Obviously we can't hold the highest value for ever, so there needs to be some "decay" parameter that allows it to fall. Multiply by 0.99 every sample, perhaps, or remove some small constant. This is good because the algorithm can respond instantly to peaks, but will smoothly and slowly fall for decays, which naturally tend to be longer. As ever, it won't do everything well - the envelope of short percussion hits might get stretched, for example.
>>>
>>>Depending on how one would fill in the blanks, you've just described
>>>either IIR, or FIR, or a thing that results in loads of zipper noise
>>>and incorrect enveloping. IIR and FIR really are the best approaches
>>>here.
>>>
>>>> There's probably other ways, or combinations of one and the other that would work. The more specific you can be about what inputs you're trying to deal with, the better a job you'll manage to do. If you want to solve the general case, you'll need to make more compromises.
>>>
>>>Or add knobs :-)
>>>
>>>Cheers,
>>>D.
>>>_______________________________________________
>>>Synth-diy mailing list
>>>Synth-diy at dropmix.xs4all.nl
>>>http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>_______________________________________________
>Synth-diy mailing list
>Synth-diy at dropmix.xs4all.nl
>http://dropmix.xs4all.nl/mailman/listinfo/synth-diy

From tim.parkhurst at gmail.com  Sun Mar 31 08:48:07 2013
From: tim.parkhurst at gmail.com (Tim Parkhurst)
Date: Sat, 30 Mar 2013 23:48:07 -0700
Subject: [sdiy] CEM and SSM chips for sale
In-Reply-To: <CACDGQT8JtGDP1xS6QdSLRGUdU2se8aLv9xWDck2yF+qZXfXbrA@mail.gmail.com>
References: <CACDGQT8JtGDP1xS6QdSLRGUdU2se8aLv9xWDck2yF+qZXfXbrA@mail.gmail.com>
Message-ID: <CACDGQT8iP_Uh=pmO4jALxxnR=7B-4u2y-FhC6tqFCOVASS8ULg@mail.gmail.com>

Hello All,

Just wanted to let everyone know that the CEM/SSM chips are all sold.
Thanks to all who asked, and I appreciate your patience. A fair amount
of interest in the chips, especially that 2040. Wish I had more!

Tim (putting the Dip in chips since 1995) Servo
---
"Sire, the church of God is an anvil that has worn out many hammers."
- H.L. Hastings


On Fri, Mar 29, 2013 at 12:28 PM, Tim Parkhurst <tim.parkhurst at gmail.com> wrote:
> Okay, I have some CEM and SSM chips that have been sitting in my
> drawer for almost 10 years. I have SO many projects to do that I don't
> think I'll ever actually get around to using them, so I figure I'll
> put them out there for someone else to use. :) Prices don't include
> shipping. Thanks for shopping at TimmyMart.
> 4x 3372 VCF/VCA $30ea
> 2x 3396 wide $20ea
> 3x 3396 narrow $20ea
> 1x 3394 synth on a chip $50
> 1x SSM2044 VCF $35
> 1x SSM2040 VCF (legendary) $60
>
> Tim (chips off the old block) Servo
> ---
> "Sire, the church of God is an anvil that has worn out many hammers."
> - H.L. Hastings


From cheater00 at gmail.com  Sun Mar 31 11:51:04 2013
From: cheater00 at gmail.com (cheater cheater)
Date: Sun, 31 Mar 2013 11:51:04 +0200
Subject: [sdiy] How I organize,
	was: Fast envelope follower circuit needed..
Message-ID: <CA+9GZUhWEA2trEM+=bGv7m9pLyfOb_SCj-fnnzDm8EcZdNbM2Q@mail.gmail.com>

On Sun, Mar 31, 2013 at 1:06 AM, Richie Burnett
<rburnett at richieburnett.co.uk> wrote:
> it's worth downloading the PDF and keeping these tricks in your back pocket for when you need then most.
>
> -Richie,

Yeah, I keep all my data in a fairly structured manner.. it's become
quite a lot by now..

$ find Documents/topics/ -type f | wc -l
150088

$ find Documents/topics/ -type d | wc -l
11103

Mind you, every saved web page adds a directory and all its assets in
a subdir that ends with _files, so there's actually quite a few less
directories. There are also source checkouts and website spiders on
there. With those retracted you're left with this many topics of
interest:

$ find Documents/topics/ -type d | grep -v _files\\\|\.hg\\\|Think\
Stats\\\|plan9front\\\|unix-jun72-read-only\\\|keyboardporn.com\\\|oh-my-zsh\\\|reaver-wps-read-only\\\|aircrack-ng\\\|scapy-2.2.0\\\|sicp-pdf\\\|www.ccs.neu.edu\\\|scs.stanford.edu\\\|dreamsongs.com\\\|\.git\\\|\\\|audioXpress\\\|orms-today.org
| wc -l

681

with this distribution of nesting level (after adjusting for from
where I'm running find):

$ find ........\\\|Documents\/topics\/\$ | while read line; do (grep
-o \/ <<< "$line") | wc -l | while read count; do expr "$count" - 1;
done; done | sort -n | uniq -c
     39 1
    441 2
    115 3
     50 4
     34 5
      1 6

and with this many actual files:

$ find ...... | while read line; do find "$line" -type f -maxdepth 1;
done | wc -l
6578


In parallel, I keep a moodboard. It's mostly images (anything that
creates an emotional response), videos, and poetry. There are this
many files in it currently:
$ find Documents/creative/ -type f | wc -l
9755

this many top-level "emotion" or "style" directories:
$ find Documents/creative/  -maxdepth 1 -type d | wc -l
10

and this many in total after all the bogus dirs are removed:
$ find .... | wc -l
167

with this distribution of nesting level:
$ find .... | while read line; do (grep -o \/ <<< "$line") | wc -l;
done | sort -n | uniq -c
     10 2
     88 3
     61 4
      3 5
      5 6


and this many files sorted according to the emotion they evoke:

$ find .... | while read line; do find "$line" -type f -maxdepth 1; done | wc -l
3056


I also keep a mood board on youtube where i sort all sorts of
different tracks into moods/styles/emotion, each has its own (private)
playlist. I've built a small python tool that backs up the names
(because the tracks like to disappear from the cloud).

It contains this many playlists (playlist names start with ----------):
$ cat 2013-03-31\ 11\:21\:10+02\:00.log | grep
\\\-\\\-\\\-\\\-\\\-\\\-\\\-\\\- | wc -l
117

and, after you remove empty lines that separate playlists, and the
playlist names, you get this many songs:
$ cat 2013-03-31\ 11\:21\:10+02\:00.log | grep -v
\\\-\\\-\\\-\\\-\\\-\\\-\\\-\\\-\\\|^\$ | wc -l
2417

most songs show up only once, many are duplicated across up to 6
playlists, one song is there 15 times:
$ cat 2013-03-31\ 11\:21\:10+02\:00.log | grep -v
\\\-\\\-\\\-\\\-\\\-\\\-\\\-\\\-\\\|^\$ | sort -n | uniq -c | cut
-c1-7 | sort -n | uniq -c
    823       1
    192       2
     87       3
     73       4
     40       5
     29       6
      5       7
      9       8
      3       9
      4      10
      4      11
      2      12
      2      13
      1      15

many moods/emotions/styles contain only one song, some contain quite a
few (the largest ones are usually collections by style rather than
emotion). I really wish Youtube allowed playlists longer than 200
songs:

$ cat 2013-03-31\ 11\:21\:10+02\:00.log | awk 'BEGIN{count = 0} /./ &&
!/----/{ count += 1; } /------/{if (count != 0) { print count }; count
= 0} END{ print count; }' | sort -n | tr '\n' ' ' | sed 's/\ /,\ /g';
echo
1, 1, 1, 1, 1, 1, 2, 2, 2, 2, 2, 2, 2, 2, 2, 3, 3, 3, 3, 3, 3, 3, 3,
3, 3, 4, 4, 4, 4, 4, 4, 5, 5, 5, 5, 5, 5, 5, 6, 6, 6, 6, 6, 7, 7, 7,
8, 8, 8, 8, 8, 9, 9, 10, 10, 10, 10, 10, 10, 11, 11, 12, 12, 13, 13,
14, 14, 14, 14, 14, 15, 15, 15, 16, 16, 16, 17, 17, 17, 19, 20, 21,
23, 24, 26, 27, 27, 30, 30, 30, 33, 34, 34, 36, 37, 39, 40, 42, 42,
42, 43, 43, 43, 43, 44, 44, 44, 45, 46, 49, 65, 71, 99, 133, 190, 191,


Hope this helps anyone who's looking to maximize their work output, be
it creative or not...


On Sun, Mar 31, 2013 at 1:06 AM, Richie Burnett
<rburnett at richieburnett.co.uk> wrote:
> The Hilbert transform produces an output where every spectral component of the input is shifted by 90 degrees. (Or more practically produces two new outputs where the spectral components in one output lag the other by 90 degrees over the frequency range of interest.) You then square the value of each output, add them together and take the square-root to get the amplitude envelope. Imagine a continuous sinewave input, phase shifted by 90 degrees to give a cosine wave. If you square the sine, square the cosine, add the squares together, and square-root you get the amplitude envelope. In this case it's a smooth line of unity if the input sinewave isn't modulated.
>
> Provided the phase shift network maintains an accurate 90 degree phase shift over the frequency range of interest, the ripple in the calculated envelope output is very small.
>
> People on this list have previously described various clever optimisations to ease the CPU burden of the magnitude calculation by either estimating the magnitude or speeding up the square-root calculation.
>
> If you're interested in the multi-rate FIR filter stuff do a google search for "multi-rate filter design momentum data systems". If you're into DSP programming it's worth downloading the PDF and keeping these tricks in your back pocket for when you need then most.
>
> -Richie,
>
>
>
> Sent from my Sony Ericsson Xperia ray
>
> cheater cheater <cheater00 at gmail.com> wrote:
>
>>Richie,
>>
>>On Thu, Mar 28, 2013 at 11:22 PM, Richie Burnett
>><rburnett at richieburnett.co.uk> wrote:
>>> The Hilbert transform is a classic tool for envelope extraction using DSP. Envelope following is really what electrical engineers would call AM demodulation, and has existed in various forms since the dawn of radio.
>>
>>That's interesting - where can I read up on how this is proven
>>mathematically? I've never heard of this property of the Hilbert
>>transform. Really interesting to me. And in fact, very surprising,
>>since the Hilbert transform is kind of like an all-pass filter.. a
>>chorus you might say. Just a bit of conceptual difference and what you
>>get is completely different!
>>
>>> Also, FIR filters with cutoff frequencies which are very low compared to the sample rate and result in really long impulse responses (hundreds or thousands of taps) can be used effectively if implemented as polyphase filters. A 2000 tap FIR filter only requires the same amount of DSP grunt as a 20 tap filter if you can drop the sample rate of the output by a factor of 1/100.
>>
>>Very smart. I was sure you could do the resampling and enveloping in
>>one step - but I wasn't sure if I should recommend it, because I've
>>never done this in practice. Thanks for clearing that up!
>>
>>> cheater cheater <cheater00 at gmail.com> wrote:
>>>
>>>>Hi Tom,
>>>>a few comments:
>>>>
>>>>On Thu, Mar 28, 2013 at 7:24 PM, Tom Wiltshire <tom at electricdruid.net> wrote:
>>>>> Hi Dan,
>>>>>
>>>>> On 28 Mar 2013, at 13:55, Dan Snazelle <subjectivity at hotmail.com> wrote:
>>>>>
>>>>>>
>>>>>> Though i work with c//arduino/AVRS not pics, this sounds very appealing! Can anyone recommend a place to learn about writing C code for an envelope follower? How do you do it? Shift signal to 0-5, then......
>>>>>>
>>>>>>
>>>>>> Thanks!!!
>>>>>
>>>>> The first thing after that would be to do the full wave rectification. Either dump the sign bit, if you've got signed data, or invert the data if the highest bit is clear if it's not signed - same thing, different representation.
>>>>
>>>>I strongly advise against doing this. It will create insane amounts of
>>>>aliasing. Do the rectification in analog. In fact full-wave
>>>>rectification is the most difficult thing to do when modelling analog
>>>>circuits digitally. Doing it anywhere near a level acceptable even for
>>>>the non-discriminating musician would require processing power orders
>>>>of magnitude larger than what a lowly pic can offer you. And in analog
>>>>it's just a few junctions.
>>>>
>>>>BTW, you can't really rectify a signal that's unipolar already.. just
>>>>a minor nitpick ;)
>>>>
>>>>> For the smoothing, I can imagine a couple of approaches:
>>>>>
>>>>> 1) Genuine honest-to-goodness digital filtering.
>>>>> There's a ton of stuff about this all over the web. But since we're talking about bottom-end chips, most of it is totally out of reach. We're not going to be running long FIR filters. But we could perhaps use an IIR filter, or a FIR with a very low sample rate, to keep it manageable.
>>>>
>>>>You can't change the sample rate, unless you mean you would downsample
>>>>the signal first. But then you'll have to have steep bandwidth
>>>>limiting filters and a filter for the down-sampling itself, which
>>>>means you do ten times the work you think you're saving by using lower
>>>>sampling rates. If you don't downsample you get aliasing, and then
>>>>you're detecting the envelope very incorrectly. It can only get you
>>>>fairly small speed-ups anyways before it stops being useful at all.
>>>>
>>>>An FIR with a fairly short impulse is fairly easy to manage, and on
>>>>something like a dspic you can do even longer ones. However IIR is the
>>>>usual way to go with envelope detectors if I'm not mistaken.
>>>>
>>>>> 2) Peak detection.
>>>>> This might be a good way to go. Keeping track of what the last couple of samples are would be easy, and would allow us to work out the current slope of the waveform. When the current sample is less than the previous one, the waveform has started heading back down, so we'd hold the peak value.
>>>>> Obviously we can't hold the highest value for ever, so there needs to be some "decay" parameter that allows it to fall. Multiply by 0.99 every sample, perhaps, or remove some small constant. This is good because the algorithm can respond instantly to peaks, but will smoothly and slowly fall for decays, which naturally tend to be longer. As ever, it won't do everything well - the envelope of short percussion hits might get stretched, for example.
>>>>
>>>>Depending on how one would fill in the blanks, you've just described
>>>>either IIR, or FIR, or a thing that results in loads of zipper noise
>>>>and incorrect enveloping. IIR and FIR really are the best approaches
>>>>here.
>>>>
>>>>> There's probably other ways, or combinations of one and the other that would work. The more specific you can be about what inputs you're trying to deal with, the better a job you'll manage to do. If you want to solve the general case, you'll need to make more compromises.
>>>>
>>>>Or add knobs :-)
>>>>
>>>>Cheers,
>>>>D.
>>>>_______________________________________________
>>>>Synth-diy mailing list
>>>>Synth-diy at dropmix.xs4all.nl
>>>>http://dropmix.xs4all.nl/mailman/listinfo/synth-diy
>>_______________________________________________
>>Synth-diy mailing list
>>Synth-diy at dropmix.xs4all.nl
>>http://dropmix.xs4all.nl/mailman/listinfo/synth-diy


From cheater00 at gmail.com  Sun Mar 31 11:51:44 2013
From: cheater00 at gmail.com (cheater cheater)
Date: Sun, 31 Mar 2013 11:51:44 +0200
Subject: [sdiy] Fast envelope follower circuit needed..
In-Reply-To: <pknylhgatv7n8fqtvnmndgjt.1364687583444@email.android.com>
References: <pknylhgatv7n8fqtvnmndgjt.1364687583444@email.android.com>
Message-ID: <CA+9GZUiYxyBJATRVj5eZ3Cd6jCbfP2y3JNYdgTy-E+LNA1SZcw@mail.gmail.com>

On Sun, Mar 31, 2013 at 1:06 AM, Richie Burnett
<rburnett at richieburnett.co.uk> wrote:
> The Hilbert transform produces an output where every spectral component of the input is shifted by 90 degrees. (Or more practically produces two new outputs where the spectral components in one output lag the other by 90 degrees over the frequency range of interest.) You then square the value of each output, add them together and take the square-root to get the amplitude envelope. Imagine a continuous sinewave input, phase shifted by 90 degrees to give a cosine wave. If you square the sine, square the cosine, add the squares together, and square-root you get the amplitude envelope. In this case it's a smooth line of unity if the input sinewave isn't modulated.
>
> Provided the phase shift network maintains an accurate 90 degree phase shift over the frequency range of interest, the ripple in the calculated envelope output is very small.

Thanks. You learn something new every day.

Cheers,
D.

> People on this list have previously described various clever optimisations to ease the CPU burden of the magnitude calculation by either estimating the magnitude or speeding up the square-root calculation.
>
> If you're interested in the multi-rate FIR filter stuff do a google search for "multi-rate filter design momentum data systems". If you're into DSP programming it's worth downloading the PDF and keeping these tricks in your back pocket for when you need then most.
>
> -Richie,
>
>
>
> Sent from my Sony Ericsson Xperia ray
>
> cheater cheater <cheater00 at gmail.com> wrote:
>
>>Richie,
>>
>>On Thu, Mar 28, 2013 at 11:22 PM, Richie Burnett
>><rburnett at richieburnett.co.uk> wrote:
>>> The Hilbert transform is a classic tool for envelope extraction using DSP. Envelope following is really what electrical engineers would call AM demodulation, and has existed in various forms since the dawn of radio.
>>
>>That's interesting - where can I read up on how this is proven
>>mathematically? I've never heard of this property of the Hilbert
>>transform. Really interesting to me. And in fact, very surprising,
>>since the Hilbert transform is kind of like an all-pass filter.. a
>>chorus you might say. Just a bit of conceptual difference and what you
>>get is completely different!
>>
>>> Also, FIR filters with cutoff frequencies which are very low compared to the sample rate and result in really long impulse responses (hundreds or thousands of taps) can be used effectively if implemented as polyphase filters. A 2000 tap FIR filter only requires the same amount of DSP grunt as a 20 tap filter if you can drop the sample rate of the output by a factor of 1/100.
>>
>>Very smart. I was sure you could do the resampling and enveloping in
>>one step - but I wasn't sure if I should recommend it, because I've
>>never done this in practice. Thanks for clearing that up!
>>
>>> cheater cheater <cheater00 at gmail.com> wrote:
>>>
>>>>Hi Tom,
>>>>a few comments:
>>>>
>>>>On Thu, Mar 28, 2013 at 7:24 PM, Tom Wiltshire <tom at electricdruid.net> wrote:
>>>>> Hi Dan,
>>>>>
>>>>> On 28 Mar 2013, at 13:55, Dan Snazelle <subjectivity at hotmail.com> wrote:
>>>>>
>>>>>>
>>>>>> Though i work with c//arduino/AVRS not pics, this sounds very appealing! Can anyone recommend a place to learn about writing C code for an envelope follower? How do you do it? Shift signal to 0-5, then......
>>>>>>
>>>>>>
>>>>>> Thanks!!!
>>>>>
>>>>> The first thing after that would be to do the full wave rectification. Either dump the sign bit, if you've got signed data, or invert the data if the highest bit is clear if it's not signed - same thing, different representation.
>>>>
>>>>I strongly advise against doing this. It will create insane amounts of
>>>>aliasing. Do the rectification in analog. In fact full-wave
>>>>rectification is the most difficult thing to do when modelling analog
>>>>circuits digitally. Doing it anywhere near a level acceptable even for
>>>>the non-discriminating musician would require processing power orders
>>>>of magnitude larger than what a lowly pic can offer you. And in analog
>>>>it's just a few junctions.
>>>>
>>>>BTW, you can't really rectify a signal that's unipolar already.. just
>>>>a minor nitpick ;)
>>>>
>>>>> For the smoothing, I can imagine a couple of approaches:
>>>>>
>>>>> 1) Genuine honest-to-goodness digital filtering.
>>>>> There's a ton of stuff about this all over the web. But since we're talking about bottom-end chips, most of it is totally out of reach. We're not going to be running long FIR filters. But we could perhaps use an IIR filter, or a FIR with a very low sample rate, to keep it manageable.
>>>>
>>>>You can't change the sample rate, unless you mean you would downsample
>>>>the signal first. But then you'll have to have steep bandwidth
>>>>limiting filters and a filter for the down-sampling itself, which
>>>>means you do ten times the work you think you're saving by using lower
>>>>sampling rates. If you don't downsample you get aliasing, and then
>>>>you're detecting the envelope very incorrectly. It can only get you
>>>>fairly small speed-ups anyways before it stops being useful at all.
>>>>
>>>>An FIR with a fairly short impulse is fairly easy to manage, and on
>>>>something like a dspic you can do even longer ones. However IIR is the
>>>>usual way to go with envelope detectors if I'm not mistaken.
>>>>
>>>>> 2) Peak detection.
>>>>> This might be a good way to go. Keeping track of what the last couple of samples are would be easy, and would allow us to work out the current slope of the waveform. When the current sample is less than the previous one, the waveform has started heading back down, so we'd hold the peak value.
>>>>> Obviously we can't hold the highest value for ever, so there needs to be some "decay" parameter that allows it to fall. Multiply by 0.99 every sample, perhaps, or remove some small constant. This is good because the algorithm can respond instantly to peaks, but will smoothly and slowly fall for decays, which naturally tend to be longer. As ever, it won't do everything well - the envelope of short percussion hits might get stretched, for example.
>>>>
>>>>Depending on how one would fill in the blanks, you've just described
>>>>either IIR, or FIR, or a thing that results in loads of zipper noise
>>>>and incorrect enveloping. IIR and FIR really are the best approaches
>>>>here.
>>>>
>>>>> There's probably other ways, or combinations of one and the other that would work. The more specific you can be about what inputs you're trying to deal with, the better a job you'll manage to do. If you want to solve the general case, you'll need to make more compromises.
>>>>
>>>>Or add knobs :-)
>>>>
>>>>Cheers,
>>>>D.
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