2
\$\begingroup\$

sinc compensationSchematic

The post equalizing circuit is shown on above. How do I determine the value of the capacitance? I couldn't find any equation to determine it.

It have been advised to use LT6202 to construct it, data sheet here.

The data sheet didn't show any post equalizing circuit, so how do I determine the connection of resistor and capacitance to the IC?

Amplitude vs Frequency Graph

The frequency response of DAC.

Pinout for LT6202:

LT6202 Pinout

\$\endgroup\$
4
  • 1
    \$\begingroup\$ It's important to show the frequency response graph of the DAC before and after filtering with this circuit. The op-amp won't have this type of application mentioned in the data sheet. It's also important to state what your target frequencies are so that folk can recalculate the new value of C and R based on your circuit. \$\endgroup\$ Commented Mar 4, 2014 at 11:08
  • \$\begingroup\$ my target frequency will be 40MHz.the frequency response of DAC after filtering \$\endgroup\$ Commented Mar 4, 2014 at 15:56
  • \$\begingroup\$ What sampling frequency is being used by the DDS system? My answer guesses at 100MHz but I could be mistaken completely. You can't design the filter without knowing Fs. \$\endgroup\$ Commented Mar 4, 2014 at 16:05
  • \$\begingroup\$ fs=100MHz.and then the gain for the opamp= 1+ r2/r1 =1.9, if I need gain=10, then the resistance value will be r2=10k,r1=1.1k? thus it will amplify my amplitude of 0-40MHz to 10 V? \$\endgroup\$ Commented Mar 4, 2014 at 16:57

1 Answer 1

3
\$\begingroup\$

The output signal level from any DAC has a constant RMS value until the input signal's frequency is half the sample rate - then aliasing begins because the first image and the desired output start to align.

The RMS level remains constant but, as the input sinewave becomes sampled fewer times per cycle, the DAC output waveform progressively contains more high-frequency sampling artifacts. Because the RMS is constant, the fundamental of the sinewave must reduce as f approaches half-sample-rate. This is what happens: -

enter image description here

All DACs do this; most of us have seen the shape of the signal as f approaches \$\dfrac{f_S}{2}\$: -

enter image description here

If you totalized the squares of all the samples over the period in which they align (for convenience) then divided by the number of samples then took the square root you would find that the RMS is 0.7071 i.e. exactly the same as the original analogue sinewave but it contains energy from the first image and possibly higher order images as well.

RMS = \$\sqrt{\dfrac{(+0.7071)^2 + (0)^2 + (-0.7071)^2 + (1)^2 + (-0.7071)^2 + (0)^2 + (+0.7071)^2 + (-1)^2}{8}}\$

= \$\sqrt{\dfrac{4}{8}}\$ = 0.7071 i.e. same as a sinewave of peak amplitude 1

So, to counter this effect you can add what is called a "sinc-compensation filter" to the output of the DAC - the filter attempts to reduce the falling amplitude of the fundamental by countering the formula shown on the top diagram.

To design a filter (example shown in the OP's question) you need to know the sampling rate that the system is using and adjust the value of C appropriately. Here's the full picture: -

enter image description here

The picture informs us that the sampling frequency is 10MHz and, if the sampling frequency for the OP's system was 100MHz then C would reduce to 8.2pF BUT don't uderestimate the effects that the op-amp can cause and, without knowing the precise sampling frequency used it is pointless trying to design and test a compensation system.

\$\endgroup\$
7
  • \$\begingroup\$ for my case,fs=100MHz,therefore the value of capacitance will be 8.2pF..and then the gain for the opamp= 1+ r2/r1 =1.9, if I need gain=10, then the resistance value will be r2=10k,r1=1.1k? thus it will amplify my amplitude of 0-40MHz to 10 V? \$\endgroup\$ Commented Mar 4, 2014 at 16:13
  • \$\begingroup\$ While both ADCs and DACs need some filtering, the need is often greater with ADCs. In an ADC, any high-frequency signals that aren't filtered out will get mirrored to audio frequencies. A DAC will generate spurious high-frequency signals, but if either filtered or passed through cleanly by the output amplifier they generally won't be audible. The biggest problem caused by such signals is that amplifiers may distort them in such fashion as to produce audible artifacts. Filtering such signals enough to avoid such distortion is often easier than filtering out everything over Nyquist. \$\endgroup\$ Commented Mar 4, 2014 at 16:23
  • \$\begingroup\$ @Andy aka, I have go through the data sheet of TL6202, the R1,R2,and C will be connected to pin 2 and then pin 3 wil be connect my sinwaveform and for the supply voltage pin 4 and 7 will use 6.3V as the supply voltage according to datahseet is 12.6? is it correct?. \$\endgroup\$ Commented Mar 4, 2014 at 16:32
  • \$\begingroup\$ i think the chip needs 5v. That is the DDS supply you use? \$\endgroup\$ Commented Mar 4, 2014 at 17:14
  • \$\begingroup\$ that means I connect the 5V supply to V+ and V- of the opamp?ya,DDS used 5 V supply. \$\endgroup\$ Commented Mar 4, 2014 at 23:49

Start asking to get answers

Find the answer to your question by asking.

Ask question

Explore related questions

See similar questions with these tags.