filtering after digital to analog convert

hi
what is the best low pass filter can we use after adc dac conversation ?
thanks

How many bits and what is the highest signal frequency?

How much do you need to filter the signal?

it's an 8 bit and i'm using dac800 i don't anything else about the frequency but i'm using it for human voice ....
my application is microphone->amplifier->adc->dac->amplifier->speaker but the sound is too awful and unclear so i need a good filter after the dac for it ....

I doubt that a filter will solve your problem, but try a simple RC low-pass filter to start with a corner frequency of about 10kHz to see if it helps.

What do you mean you don't know anything about the frequency? Who designed the circuit?

I agree with Crutschow. You could get 'aliasing' problems if the ADC/DAC speeds are too slow, and that can be reduced by filtering the input to some degree but 'awful and unclear' suggests something else is wrong. A filter after the DAC might remove 'sharpness' from the sound but it cannot add clarity. Can you post a schematic, or at least try temporarily removing the ADC and DAC and linking the amplifiers directly together to eliminiate the digital part as being the problem.

Brian.

What do you mean you don't know anything about the frequency? Who designed the circuit?

Click to expand...

as i told you i'm using my voice frequency in this circuit.
for the schematic i'm using most of the circuit in the datasheet:
as i told you before my diagram is as this microphone->amplifier->adc->dac->amplifier->speaker
for the amplifier i've attached the diagram in the file circuit.doc
the adc i used the one in page 11 of the datasheet attached.
the dac i used is in the first page in the datasheet.

I think most of us are familiar with the ICs. It's how you have connected them together we want to know.

I assume you have the ADC wired as in Figure 17 of the data sheet. How are you biasing the analog inputs?

Brian.

exactly fig 17 ....what do u mean with biasing ? i connected pin 7 to ground and pin 6 is the output of the amplifier. i connected leds to the output of the adc i found that the leds changes depending on my voice

When you build a circuit without understanding it, it's not surprising it doesn't work properly.;-)

If you look at the data sheet for the ADC0804LCN you will see that the input signal goes from ground to V+, thus you must bias a normal audio signal, which goes above and below ground, at 1/2 V+ to keep within this signal range.

For maximum dynamic range, the peak-to-peak audio signal for the loudest signal should go from 0V to V+.

The data sheet also shows that the free run conversion rate is 8888 conversions/sec. The Nyquist theorem states that the maximum audio signal you can convert is 1/2 that or about 4.4kHz (which is low, comparable to phone bandwidths). Any audio frequencies above that will alias into the passband causing noise and distortion. So you need a high order 4kHz low-pass active-filter (probably 4-pole or greater) at the ADC input to prevent this aliasing.

Also 8-bits is a low-number of bits for sending audio (16-bits is typical) and will give a low S/N ratio as well as some possible audible distortion.

i don't think so because i went to the lab and i had put a function generator as a sine wave to the input and changed the frequency from 500hz to 4khz and more without any changes...........

foxbrain said:

i don't think so because i went to the lab and i had put a function generator as a sine wave to the input and changed the frequency from 500hz to 4khz and more without any changes...........

Click to expand...

Then I guess you don't need our help. ;-)

Agreed - anyone who can defy the laws of physics either doesn't need our help or is beyond help!

Brian.