On Wed, Nov 06, 2002 at 03:57:05PM -0500, Lamar Owen wrote:
> Yeah, I'll google around for it. I just have to see which flavor of Bassman
> this one is (twin 12's with separate head). It is very old, that much I know.
If the covering is tweed, it's a real collector's item...
likewise if the cont
On Wed, Nov 06, 2002 at 03:57:05PM -0500, Lamar Owen wrote:
> On Wednesday 06 November 2002 14:47, Mark Rages wrote:
> > On Wed, Nov 06, 2002 at 02:09:50PM -0500, Lamar Owen wrote:
> > > cap already, in which case you have real problems, because the effective
> > > capacitance or two series caps is
On Wednesday 06 November 2002 14:47, Mark Rages wrote:
> On Wed, Nov 06, 2002 at 02:09:50PM -0500, Lamar Owen wrote:
> > cap already, in which case you have real problems, because the effective
> > capacitance or two series caps is equal to the reciprocal of the sums of
> > the reciprocals of the i
On Wed, Nov 06, 2002 at 02:09:50PM -0500, Lamar Owen wrote:
> On Wednesday 06 November 2002 12:50, Steve Harris wrote:
I'm an electrical engineer and I've been following this discussion with
interest.
> A series 1 microfarad capacitor in the 1kV breakdown range in series with your
> line input
On Wednesday 06 November 2002 12:50, Steve Harris wrote:
> On Wed, Nov 06, 2002 at 12:07:24PM -0500, Lamar Owen wrote:
> > As it also works as a transient recorder and spectrum analyzer (to
> The trick really is get get a sample off of it, 8bit might be enough, but
> I suspect 16 will be easier to
On Wed, Nov 06, 2002 at 12:07:24PM -0500, Lamar Owen wrote:
> As it also works as a transient recorder and spectrum analyzer (to 32MHz), it
> is a great buy. Even though it is an 8-bit device, it is useful in showing
> what the signal looks like, as your eye can't distinguish the difference on a
On Tuesday 05 November 2002 17:57, Steve Harris wrote:
> On Tue, Nov 05, 2002 at 08:26:40 +0100, Tim Goetze wrote:
> > Steve Harris wrote:
> > >The bad news is that it means going inside someones amp with a probe, I
> > >would have a go myself, but I've allready killed my last amp, and my
> > >elec
> i'm not up to understanding all implications of the fact that the
> incoming signal is not a pure sine;
This intrigued me -- I think the answer is that the
process is not linear -- you get sum and difference tones much as
in complex fm. I added an example to clm.html. Back when
Marc and I were
Paul Winkler wrote:
>
> input stuff -> (tube) --> (tube) --> power amp stuff
> ^
> |
> = large electrolytic capacitor rated at e.g. 220V
> |
>positive end of tap
>
>
> That should take ca
On Tue, Nov 05, 2002 at 10:54:02PM +, Steve Harris wrote:
> Unluckily it looks like to do a good job we need to tap those 100+ volt
> signals :(
I'm pretty handy with a soldering iron, if anyone has a clue how this
could be done. If I remember my moron-level analog electronics,
couldn't I just
On Tue, Nov 05, 2002 at 08:26:40 +0100, Tim Goetze wrote:
> Steve Harris wrote:
>
> >The bad news is that it means going inside someones amp with a probe, I
> >would have a go myself, but I've allready killed my last amp, and my
> >electronics skills are bad enough that I would probably fry myself
On Tue, Nov 05, 2002 at 12:01:57 -0800, Paul Winkler wrote:
> On Tue, Nov 05, 2002 at 07:32:21PM +0100, Tim Goetze wrote:
> > 8 octaves would be something like 40 - 10240 Hz, that should do
> > it.
>
> Truly godawful image quality, but this may be of interest -
> it's a plot of a celestion "Greenb
On Tue, Nov 05, 2002 at 07:32:21PM +0100, Tim Goetze wrote:
> 8 octaves would be something like 40 - 10240 Hz, that should do
> it.
Truly godawful image quality, but this may be of interest -
it's a plot of a celestion "Greenback" 12" gguitar speaker
in an anechoic chamber:
http://home3.netcarrier
Steve Harris wrote:
>The bad news is that it means going inside someones amp with a probe, I
>would have a go myself, but I've allready killed my last amp, and my
>electronics skills are bad enough that I would probably fry myself ;)
>
>Brave and/or stupid volunteers welcome ;)
looking at a fende
Steve Harris wrote:
>The impulse includes the cabinet response, but the IIR version you
>produced I'm not so sure about, I'd have to run the IIR through octave to
>find out I guess, but (looking at it simplisticly) I dont think the
>response is long enough to include the cabinet effect.
have you
Steve Harris wrote:
>On Tue, Nov 05, 2002 at 01:44:03 +0100, Tim Goetze wrote:
>> sure thing. is a sawtooth from the virus ok or would you prefer
>> an integer cycle frequency? still have to do the rewiring if
>> you need it from the box.
>
>It really needs to be integer period, and in sample syn
On Tue, Nov 05, 2002 at 01:56:21 +0100, Tim Goetze wrote:
> >> > For the record my current guess for an amp process is:
> >> >
> >> > .-> bandpass -> shaper -. .--- LP <--.
> >> > | | v |
> >> > input -> EQ? -+-> bandpass -> shaper
On Tue, Nov 05, 2002 at 01:44:03 +0100, Tim Goetze wrote:
> sure thing. is a sawtooth from the virus ok or would you prefer
> an integer cycle frequency? still have to do the rewiring if
> you need it from the box.
It really needs to be integer period, and in sample sync.
- Steve
Steve Harris wrote:
>On Mon, Nov 04, 2002 at 08:42:26 +0100, Tim Goetze wrote:
>> >I've attached some example code that generates a saw wave (p), applies a
>> >nonlinear function (x, p^2 * 0.3 + p^3 * 0.73 - p^5 * 0.1), a highpass
>> >filter and a delay (about 16.1 samples), then puts the transfer
Steve Harris wrote:
>On Mon, Nov 04, 2002 at 11:10:14 -0800, Paul Winkler wrote:
>> > For the record my current guess for an amp process is:
>> >
>> > .-> bandpass -> shaper -. .--- LP <--.
>> > | | v |
>> > input -> EQ? -+-> band
On Mon, Nov 04, 2002 at 02:58:03 -0800, Paul Winkler wrote:
> I'd like to get you some samples from my old Gibson at
> various settings (it's very versatile, almost sounds like
> a different amp depending on how you set it).
> What would I need to run through it?
I just tried using a constant freq
On Mon, Nov 04, 2002 at 08:42:26 +0100, Tim Goetze wrote:
> >There is an example here: http://plugin.org.uk/tmp/foo.png
>
> have you tried it with real-world data yet?
Just tried it, tested with a synthetic overdrive effect. The results look
plausible. but I haven't tried shaping anything with th
On Mon, Nov 04, 2002 at 02:36:07 -0800, Mark Knecht wrote:
> Make a pretty picture like that and make the shapers compressors and then I
> have a Waves C4, the best little end-of-the-line compressor around. ;-)
Funnily enough that was what made me draw it that way, I've been thinking
about multib
On Mon, Nov 04, 2002 at 10:27:39PM +, Steve Harris wrote:
> On Mon, Nov 04, 2002 at 08:42:26 +0100, Tim Goetze wrote:
> > have you tried it with real-world data yet?
>
> Nope, dont have any yet. I'l try grabbing something of a distoriton pedal
> or something, can you try one of your amp?
I'd
On Mon, Nov 04, 2002 at 10:31:37PM +, Steve Harris wrote:
> > Sure, if there is pre-shaper and post-shaper gain for each
> > band then that's your EQ. Lot of parameters to control,
> > though.
>
> Well, they would be preset for given amp types I guess.
Of course. But then there will be thin
, November 04, 2002 2:32 PM
To: [EMAIL PROTECTED]
Cc: Paul Winkler
Subject: Re: [linux-audio-dev] Re: Cheby amp code
On Mon, Nov 04, 2002 at 11:10:14 -0800, Paul Winkler wrote:
> > For the record my current guess for an amp process is:
> >
> > .-> bandpass -&g
On Mon, Nov 04, 2002 at 11:10:14 -0800, Paul Winkler wrote:
> > For the record my current guess for an amp process is:
> >
> > .-> bandpass -> shaper -. .--- LP <--.
> > | | v |
> > input -> EQ? -+-> bandpass -> shaper -+-> delay -
On Mon, Nov 04, 2002 at 08:42:26 +0100, Tim Goetze wrote:
> >I've attached some example code that generates a saw wave (p), applies a
> >nonlinear function (x, p^2 * 0.3 + p^3 * 0.73 - p^5 * 0.1), a highpass
> >filter and a delay (about 16.1 samples), then puts the transfer function
> >into a table
Steve Harris wrote:
>I've attached some example code that generates a saw wave (p), applies a
>nonlinear function (x, p^2 * 0.3 + p^3 * 0.73 - p^5 * 0.1), a highpass
>filter and a delay (about 16.1 samples), then puts the transfer function
>into a table (ignoring the delay). It can guess the corre
On Mon, Nov 04, 2002 at 06:39:33PM +, Steve Harris wrote:
> Do you have any thoughts on trying to gather transfer functions
> with saws by any chance?
Circular saw? Skillsaw? Seriously, this stuff is beyond me.
I know diddley-squat about DSP.
> For the record my current guess for an amp pr
On Mon, Nov 04, 2002 at 10:03:11 -0800, Paul Winkler wrote:
> On Mon, Nov 04, 2002 at 04:06:53PM +, Steve Harris wrote:
> > My current guess is different transfer functions. Do you know the phrase
> > "when all you have is a hammer, every problem looks like a nail" ;)
>
> I'd bet money that th
On Mon, Nov 04, 2002 at 04:06:53PM +, Steve Harris wrote:
> My current guess is different transfer functions. Do you know the phrase
> "when all you have is a hammer, every problem looks like a nail" ;)
I'd bet money that they use different pre- and post- filters, too.
It's very instructive to
On Mon, Nov 04, 2002 at 04:24:19 +0100, Tim Goetze wrote:
> >Yeah, I think thats difficult, and probably not neccesary, the "hard clip"
> >from a guitar amp doesn;t look very hard to me, so I recon you could just
> >apply the shaper, plus a bit of oversampling, a LP filter and it'd be
> >fine.
>
>
On Mon, Nov 04, 2002 at 02:42:49 +, Steve Harris wrote:
> I think its probably worth a try with saws, at least it will make
> extracting the transfer function simpler. I will try playing with
> extracting a transfer function from a known nonlinear process and see how
> much the phase shift mess
Steve Harris wrote:
>Well, IMHO the point of ivory towers is that you dont have to be connected
>to reality, so you'd think they would just go for the most accurate
>process. NB I haven't read the paper.
maybe the thought of amplitude-dependent harmonic relations
didn't occur to them, i don't kno
Steve Harris wrote:
>We'd also need to split up the incoming signal by frequency though, and
>that would make it expensive to run.
very much so.
>> been thinking about how to do a hard clipper with sinc some more
>> today, without real results though.
>
>Yeah, I think thats difficult, and probab
The sine -> waveshaper mapping I talked about in the previous followup to
this mail isn't going to work (I think) because of the huge asymetry of
the output waveform, after looking at line-sine-fade Tim mailed.
It might still be possible with saws as the input, but then you have to
worry about pha
On Mon, Nov 04, 2002 at 01:54:56 +0100, Tim Goetze wrote:
> >Thats interesting, cos liiing at your waterfall plot it doesn;t seem liek
> >that should be enough.
>
> afai understand the paper, they've been doing this as an academic
> exercise. ivory towers and so on ...
[sorry for the bad-even-fo
Steve Harris wrote:
>On Sun, Nov 03, 2002 at 07:48:55 +0100, Tim Goetze wrote:
>> >OK, why do they use two shapers? Or is it one cheby and one non polynomial?
>>
>> two chebyshevs, blend factor depending on incoming amplitude.
>
>Thats interesting, cos liiing at your waterfall plot it doesn;t see
On Sun, Nov 03, 2002 at 08:06:53 +0100, Tim Goetze wrote:
> >> i'm not up to understanding all implications of the fact that the
> >> incoming signal is not a pure sine; neither do i have a recipe for
> >> preparing the coefficient tables -- if we scale the individual
> >> coefficients by 1/sum the
On Sun, Nov 03, 2002 at 07:48:55 +0100, Tim Goetze wrote:
> >OK, why do they use two shapers? Or is it one cheby and one non polynomial?
>
> two chebyshevs, blend factor depending on incoming amplitude.
Thats interesting, cos liiing at your waterfall plot it doesn;t seem liek
that should be enoug
Steve Harris wrote:
>On Sun, Nov 03, 2002 at 05:08:33 +0100, Tim Goetze wrote:
>> the peak value of the chebyshev-shaped output will be the sum of all
>> coefficients calculated in this manner. the further the incoming sine
>> is scaled down (from [-1,+1]), the less the harmonic mix will match
>>
Steve Harris wrote:
>Did you try adding the 0th harmionic to the front of the table, and
>dropping the last harmonic? That made it sound pretty good for low notes.
i still think the harmonic mix doesn't sound right, and it's
becoming obvious why.
>> additionally, it seems that to tackle intermo
On Sat, Nov 02, 2002 at 08:12:10 +0100, Tim Goetze wrote:
> >In the instantaite block fixes it up more-or-less. Maybe even adds a bit
> >of compression (it boosts the gain to make it roughly 1:1 too).
>
> have you modified the lut in the meantime? i don't seem to be getting
> the right results wit
On Sun, Nov 03, 2002 at 05:08:33 +0100, Tim Goetze wrote:
> the peak value of the chebyshev-shaped output will be the sum of all
> coefficients calculated in this manner. the further the incoming sine
> is scaled down (from [-1,+1]), the less the harmonic mix will match
> the wanted amplitudes.
Hm
found out some interesting facts about the chebyshev. been playing
around a little with a chebyshev shaper, feeding it various harmonic
amplitudes and a sine oscillation, taking an FT afterward.
it seems that in order to get a harmonic of amplitude 0.5, you must
not pass 0.5 to chebpc for that har
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