How to output a square wave on my soundcard

Question

I would like to create a digital (square) signal on my sound card. It works great if I generate high frequencies. But, since I can't output DC on a sound card, for lower frequencies the resulting digital bits will all slowly fade to 0.

This is what the soundcards high pass does to my square wave: http://www.electronics-tutorials.ws/filter/fil39.gif

What's the mathematical function of a signal, that, when passed through a high pass will become square?

Ideally, the solution is demonstrated in gnuplot.

Answer 1

The sound card cuts out the low frequencies in the waveform, so you need to boost those by some amount in what you pass to it.

A square wave contains many frequencies (see the section on the Fourier series here). I suspect the easiest method of generating a corrected square wave is to sum a Fourier series, boosting the amplitudes of the low frequency components to compensate for the high-pass filter in the sound card.

In order to work out how much to boost each low frequency component, you will first need to measure the response of the high-pass filter in your soundcard, by outputting sine waves of various frequencies but constant amplitude, and measuring for each frequency the ratio r(f) of the amplitude of the output to the amplitude of the input. Then, an approximation to a square wave output can be generated by multiplying the amplitude of each frequency component f in the square wave fourier series by 1/r(f) (the 'inverse filter').

It's possible that the high-pass filter in the soundcard also adjusts the phase of the signal. In this case, one might be better off modelling the high pass as an RC filter, (which is probably how the soundcard is doing the filtering), and invert both the amplitude and phase response from that.

Answer 2

Some of the previous answers have correctly noted that it is the high-pass filter (AC coupling capacitor on the soundcard's output) is what is preventing the low frequency square waves from "staying on" so they decay quickly.

There is no way to completely beat this filter from software or it wouldn't be there, now would it? If you can live with lower amplitude square waves at the lower frequencies, you can approximate them by sending out something like a triangle wave. From a transient analysis perspective, the theory of operation here is that as the coupling capacitor is discharging (blocking DC) you are increasing its bias voltage to counteract that discharge thus maintaining the square wave's plateau for a while. Of course you eventually run out of PCM headroom (you can't keep increasing the voltage indefinitely), so a 24-bit card is better in this respect than a 16-bit one as it will give you more resolution. Another, more abstract way to think of this is that the RC filter works as a differentiator, so in order to get the flat peaks of the square wave you need to give it the flat slopes of the triangle wave at the input. But this is an idealized behavior.

As quick proof of concept, here's what a 60Hz ±1V triangle signal becomes when passing through a 1uF coupling cap on a 1Kohm load; it approximates a ±200mV square wave

Note that the impedance/resistance of the load matters quite a bit here; if you lower it to, say, 100ohm the output amplitude decrease dramatically. This is how the coupling caps block DC on speakers/headphone because these devices have much lower impedance than 1Kohm.

If I can find a bit more time later today, I'll add a better simulation, with a better shaped stimulus instead of the simple triangle wave, but I can't get that from your average web-based circuit simulator software...

Well, if you're lucky you can get one of those $0.99 USB sound cards where the manufacturer has cut corners so much that they didn't install coupling caps. https://www.youtube.com/watch?v=4GNRzwfP7RE

Answer 3

Unfourtunately, you cannot get a good approximation of a square wave. Sound hardware is intentionally slew rate limited and would not be able to produce a falling or rising edge beyond its intended frequency range.

You can approximate a badly deformed square wave by alternating a high and low PCM code (+max, -max) every N samples.

Answer 4

You can't actually produce a true square wave, because it has infinite bandwidth. You can produce a reasonable approximation of a square wave though, at frequencies between say 10 Hz and 1 kHz (below 10 Hz you may have problems with the analogue part of your sound card etc, and above around 1 kHz the approximation will become increasingly inaccurate, since you can only reproduce a relatively small number of harmonics).

Tp generate the waveform the sample values will just alternate between +/- some value, e.g. full scale, which would be -32767 and +32767 for a 16 bit PCM stream. The frequency will be determined by the period of these samples. E.g. for a 44.1 kHz sample rate, if you have say 100 samples of -32767 and then 100 samples of +32767, i.e. period = 200 samples, then the fundamental frequency of your square wave will be 44.1 kHz / 200 = 220 Hz.

Answer 5

I found an application that I build on it.

http://www.blogger.com/blogger.g?blogID=999906212197085612#editor/target=post;postID=7722571737880350755

you can generate the format you want and even the pattern you need.

The code uses SLIMDX.