Even the best software app can go only so far to overcome the basic limitations of hardware that was specifically optimized for audio acquisition and reproduction while being as cheap as possible.
The advent of inexpensive standardized 16- and 24-bit audio I/O hardware for personal computers inspired a rush of oscilloscope simulating applications. Many of these “Sound Card Scopes” are quite capable, providing input scaling, triggering and timebase options, frequency analysis, data file storage, and built-in signal generators.
But even the best software app can go only so far to overcome the basic limitations of hardware that was, after all, specifically optimized for audio acquisition and reproduction while being as cheap as possible. Among those limitations are:
In response, many hardware remedies have appeared in the literature. Basic buffers and variable attenuators are available that improve input impedance and range, while an ingenious design based on the AD583 S&H (by Doug Mercer in Analog Dialogue: “Front End Turns PC Sound Card into High-Speed Sampling Oscilloscope”), stretched the upper bandwidth limit (for repetitive waveforms) to 50MHz!
This Design Idea for a sound card scope front end is a little different. Referring to the schematic (Figure 1), it combines megohm input impedance with switched X1-X10-X100 attenuation, but then adds extension of the bottom end of sound card bandwidth by more than a factor of 10. For dual-channel (stereo sound card) scopes, the circuit is simply duplicated.
Figure 1 Sound card oscilloscope front end schematic.
The front end begins with the cascaded resistor network around attenuator switch S1. It provides 1MΩ (minimum) input impedance and selectable decade attenuation without using resistors higher than 2MΩ (the point where precision resistors start to become expensive), with a simple ON–OFF–ON three position toggle switch.
The non-inverting 9051 buffer amplifier level shifts the incoming signal to Vdd/2, and applies adjustable (one-time calibration for the particular sound card hardware being used) low frequency correction via the C1(R1+R2) feedback network. This is how that works.
Essentially all sound card CODECs have AC coupled inputs, and even though the rated cutoff frequency of the associated RC time constant may be as low as 10Hz (as was the case for the CODEC used to record the green trace shown in Figure 2), the resulting distortion (“droop”) of common waveforms of interest (e.g., Figure 2’s 20Hz square wave), can be extreme and unacceptable.
Figure 2 Low frequency response correction, showing the uncorrected CODEC response (green) versus the corrected response (red) with R1 adjusted for 22ms RC.
The fix consists of adjusting (R1+R2) so that the feedback time constant equals and cancels that of the CODEC input: 22ms in this example. With this one-time calibration in hand, a typical improvement is shown by the red trace in Figure 2, producing a quantitatively accurate reproduction of the original waveform, and of all similar inputs. Compensation isn’t quite perfect, because eventually the 9051 will run out of headroom, and because CODEC high-pass filtering is sometimes more complex than a simple single-pole RC. But as Figure 2 illustrates, the improvement is significant and useful.
Of course, as we add endless enhancements to what began as ubiquitous, simple, cheap, and cheerful sound card hardware, eventually there must come a point where the proverbial lily has been over-gilded and its cost effectiveness has been lost. Hopefully this design doesn’t cross that line.
This article was originally published on EDN.
Stephen Woodward‘s relationship with EDN‘s DI column goes back quite a ways. In all, a total of 64 submissions have been accepted since his first contribution was published in 1974.