Active audio filter doesn't work

[KlausST]

Hi,

I used the 7660 or equal with very high sensitivity circuit...OPAMP gain of 20 and a 16 ADC with 3VRef. Makes a resolution of 2.3uV/LSB.
A 3kHz analog signal filter and sampling with 10kHz with gives stable readings down to +/-1LSB.

At the 7660 output I have simple LC filters. Low ESR capacitors.
Four layer optimized layout hand routed).

****
So I´m sure there is a solution for your problem, too.
As said before. Capacitors are the one thing, PCB layout is the other. Unfortunately the worst of both will dominate.

Klaus

 

[Audioguru]

Maybe there is a voltage converter IC available that operates at a frequency higher than our hearing? Have you ever heard the switching from a high power car amplifier or from a class-D amplifier?

 

[ads-ee]

Maybe there is a voltage converter IC available that operates at a frequency higher than our hearing? Have you ever heard the switching from a high power car amplifier or from a class-D amplifier?

Yeah, ones like this dual supply from linear that use frequencies between 100 KHz and 2 MHz are pretty typical. Most of the designs that the RF guys in a previous company I worked at used >>100 KHz switching frequencies, that's pretty far outside of hearing range.

Or this one from Analog.

 

[David Kosman]

Hi,

One important question is: Do you need DC performance (input to output)?
If you can live with a lower cutoff frequency of - lets say - 10 Hz, then this makes all easier.. Then we use capacitors for coupling...

Could you please specify where and which the capacitor should be? (on post 13)

...

I would like to get it at least working, before I buy another thing. I worry how will the filters sound, because what I have heard from it was not what I was expecting...

 

[Audioguru]

The MAX1044 voltage converter should be used instead of the ICL7660 because it can operate at 35kHz and its whistle will not be audible. It costs a small fortune but it is only ordinary money that grows on trees in many places.

I do not know why you are boosting the bass like a boom-box and chopping off the highs like an AM radio with those filters.

 

[KlausST]

Hi,

For single supply you need to bias the signal with about half of the supply voltage.
At least at input, but most probably at output, too.

C = 1 / (2 × Pi * f * R)
f = cutoff frequency
R = here the impedance (input or output)

Klaus

 

[David Kosman]

Earlier I thought that the coupling capacitors can be used instead of the biasing circuit, but I was probably wrong?
(Klaus said that if I am okay with lower cutoff frequency, then we use coupling capacitors, it would be all easier)

I am really having trouble finding those C,R values, should it be calculated? I would be very glad if someone did it for me. The instructions are in english and thats bad for me.

Should the inverting op-amp bias circuit look like this?

 

[Audioguru]

When the opamp has a single supply then its (+) input must be biased at about half the supply voltage and the (-) input must be capacitor-coupled.
Your signal generator is shorting the (-) input almost to ground then the opamp amplifies it with its internal gain of a few hundred thousand and causes its output to be saturated at a DC voltage as high as it can go.
DOO it like DIS:

 

[David Kosman]

Thank you so much. I will also try the low ESR capacitors if they are not expensive.

 

[Audioguru]

This filter must be driven from a very low impedance like the output of an opamp.

 

[David Kosman]

I should also ask about the low-pass filter. Is this correct? I really don't know about the values.

 

[FvM]

Correct in which regard? I don't remember that you ever mentioned a specification.

- - - Updated - - -

At second sight, the Sallen Key filter parameters seems to indicate instability. In so far the design is not correct. How did you arrive at the component values?

 

[Audioguru]

You show a 3rd Order Sallen and key Lowpass filter. There are many tutorials and electronic calculators in Google. I sim'd it:

 

[FvM]

Yes, looks reasonable. I don't actually understand why my filter tool decided that the circuit would be unstable.

 

[David Kosman]

I was asking if the biasing components are correct. Those are R2,R3,R4,R5 and C3. And the crossed wire was there originally.

 

[KlausST]

Hi,

And the crossed wire was there originally.

If there was a wire... then don´t change it. (It confuses your filter characteristic)
Then probably in the orignal plan was no R4, R5, C3...You don´t need it.

R2 and R3 are correct as far as i can see.

***
Your output voltage is now biased to VCC/2. Maybe you need a serial resistor at the output.

Klaus

 

[David Kosman]

So I've finished both circuits. The volume is good with clean sound.
However the problem with the, somehow trimmed frequency response, persists. I will fully describe it when I have more time, but in very short, it seems that something doesn't let the filters to fully boost or damp the sound. Though I can hear that the high freq. roll-off starts at the right frequency very steeply, the high-end is still there very much.
And this is exactly the same with the bass boost. The 100Hz is, let's say +4dB, and all frequencies below it STAY at the same level.
I am very disappointed. What could be the problem?

 

[Audioguru]

I can hear that the high freq. roll-off starts at the right frequency very steeply, the high-end is still there very much.

Of course the high-end is still there very much. The response of the lowpass filter in post #73 shows that 12kHz to 15khzx is cut slightly, 20kHz is cut a little and higher frequencies that nobody can hear are cut more.

And this is exactly the same with the bass boost. The 100Hz is, let's say +4dB, and all frequencies below it STAY at the same level.

That is what the response of the bass filter shows in post #7 except the boost begins at about 120Hz and peaks at 35Hz with about +14dB down to below earthquake frequencies.