Does anyone know a way to take an audio signal and output it multiple times with different amplitude

[Phil Allison]

"John Fields"
"Phil Allison"

** As usual, all YOU post are ignorant and unsupported ASSERTIONS

- aka Utter Drivel !!

There being NO CASE WHATEVER to answer

- no answer is needed.

**** off.



....... Phil

 

[Eeyore]

** Put up or shut up - you whining bullshit artist.

SOD OFF

 

[Eeyore]

One of the others is cheating, by using others' designs without
recompense to the original designer which, as I recall, both you and
Aylward endorsed as admirable. Kevin, at least, acknowledged the
source.

WTF are you talking about ?

 

[Eeyore]

They chose to specify it with only one channel operating at rated load,
when a more revealing spec would reflect what would happen with both
channels operating, full bore, into their rated loads, or bridged.

That's the way *I* do it. Although in fact, it IS measured that way, simply not
stated by that clot P.B. else the power into 2 ohms x 2 wouldn't be identical to
the bridged power into 4 ohms. Which it is.

Don't know much about audio do you ?

Graham

 

[Eeyore]

As usual, all you want to do is fight, instead of engaging in reasoned
discourse, so I'll disengage.

You're one to talk since you have misinterpreted every single power figure in
those specs.

Graham

 

[Eeyore]

---
That has nothing to do with wanting to fight.

All I did was misread a sentence or two and admit to the error when it
was brought up by Phil.

Interestingly, you both seem to think that admitting to an error is an
Achilles' heel of some kind and are loath to do it because you think it
puts you in a position of weakness.

Even more interesting is that you don't seem to realize that refusing to
acknowledge an error, when it's as plain as the nose on your face, makes
you look disingenuous and sullies your credibility.

What error ?

Graham

 

[Eeyore]

---
I cut my teeth on audio, and decided there were better things to do than
to follow it blindly for the rest of my life.

You, obviously, decided differently.

The point, which you seem to keep missing, is that THD was measured
under conditions which were optimum instead of,

No, the very reverse.

as I'd have done,
measured it with one channel operating fully loaded, both channels fully
loaded, and bridged, fully loaded, and published the results.

We only bother with both channels fully loaded. It's the pro-audio industry norm.

How long do you want the data sheet to be ? Users aren't interested in THAT much
gubbins. Suggest you read a few equivalent data sheets from other manufacturers in
the same business.

Graham

 

[Eeyore]

I cut my teeth on audio, and decided there were better things to do than
to follow it blindly for the rest of my life.

Me neither.

I have innovated in audio.

Graham

 

[Phil Allison]

"John Fields"

** I posted a reply.

You stupid autistic cretin.



....... Phil

 

[Phil Allison]

"John Fields"

All I did was misread a sentence or two and admit to the error when it
was brought up by Phil.

** Fucking LIAR.



...... Phil

 

[Angelo Campanella]

Does anyone know a way to take an audio signal and output it multiple times
with different amplitude and phases tia sal2

It's been too long since I have done a ny digital design, but, as delay
lines work, one needs to send an analog waveform, digitized in amplitude
to a sing-around circuit with 8 taps at the selected phase points, each
tap being the appropriate output. The sine wave remains conntained in
the sing-sround circuit. The frequency is determined by the clock speed
of the sing-around circuit.

Angelo Campanella

 

[Eeyore]

The companies I've worked for have generally preferred commercial
confidentiality plus you know how difficult it is to patent a circuit under the
'obvious use' or 'derivative' clauses.

I do have several patentable ideas in mind though. Both totally unrelated to
audio. Was thinking about one of them earlier today. Oh and (nearly forgot) one
audio one related to louspeakers.

Graham

 

[Eeyore]

It's been too long since I have done a ny digital design, but, as delay
lines work, one needs to send an analog waveform, digitized in amplitude
to a sing-around circuit with 8 taps at the selected phase points, each
tap being the appropriate output. The sine wave remains conntained in
the sing-sround circuit. The frequency is determined by the clock speed
of the sing-around circuit.

Which would therefore still only work at a unique single frequency at any one time.

Graham

 

[Jamie]

John Fields wrote:




The companies I've worked for have generally preferred commercial
confidentiality plus you know how difficult it is to patent a circuit under the
'obvious use' or 'derivative' clauses.

I do have several patentable ideas in mind though. Both totally unrelated to
audio. Was thinking about one of them earlier today. Oh and (nearly forgot) one
audio one related to louspeakers.

Graham

I thought I smelt something from over the pond.


http://webpages.charter.net/jamie_5"

 

[Angelo Campanella]

Which would therefore still only work at a unique single frequency at any one time.

But as I said, the clock rate of the shift register must be variable so
as to set the sine wave frequency. The way I see it, one generates, say,
an eight bit word constituting a full wave that is sent into the clocked
shift register that loops back to itself. The register is long, with
eight parallel-bit(8) output taps, each capable of reading the 8 bit
word in parallel, to be summed by a D to A converter to output an analog
sine voltage, which becomes the driver for that channel. The clock rate
for the looped shift register needs to be variable, so that your system
can set the frequency it needs.

Angelo Campanella

 

[Eeyore]

SINGLE FREQUENCY at any time.

CLOT !

 

[Eeyore]

He started by saying "an audio signal". Only later did he mention sine waves *as an
example*.

Therefore unclear. I take it you didn't do comprehension tests.

Graham

 

[Bob Monsen]

But as I said, the clock rate of the shift register must be variable so
as to set the sine wave frequency. The way I see it, one generates, say,
an eight bit word constituting a full wave that is sent into the clocked
shift register that loops back to itself. The register is long, with
eight parallel-bit(8) output taps, each capable of reading the 8 bit
word in parallel, to be summed by a D to A converter to output an analog
sine voltage, which becomes the driver for that channel. The clock rate
for the looped shift register needs to be variable, so that your system
can set the frequency it needs.

---
For one degree increments, your shift register would have to be 9 bits
wide and 360 stages long, and would have to be loaded serially using a 9
X 256 lookup table and some glue logic to switch the serial inputs from
the LUT to the 360° tap on the shift register.


a simpler way might be: (View in Courier)

CLK>---[COUNT]--+-[LUT1]--[DAC]--[FILTER]--[OPAMP]--->OUT1
|
+-[LUT2]--[DAC]--[FILTER]--[OPAMP]--->OUT2
|
+-[LUT3]--[DAC]--[FILTER]--[OPAMP]--->OUT3
|
+-[LUT4]--[DAC]--[FILTER]--[OPAMP]--->OUT4
|
+-[LUT5]--[DAC]--[FILTER]--[OPAMP]--->OUT5
|
+-[LUT6]--[DAC]--[FILTER]--[OPAMP]--->OUT6
|
+-[LUT7]--[DAC]--[FILTER]--[OPAMP]--->OUT7
|
+-[LUT8]--[DAC]--[FILTER]--[OPAMP]--->OUT8

JF

Can this scheme give a wideband phase shift?

One way to do a wideband phase shift is to do an FFT to amplitude/angle in
the frequency domain, add something to all the angle points, and then do an
inverse FFT back into time domain. That will require lots of processing
power, although a dsPIC might be able to do it for audio. A high end dsp
chip will definitely be able to handle it for audio. You also might be able
to do it with a PC, although there will be some latency there.

Regards,
Bob Monsen