TGen
|
StoneAge
|
Contact
|
Purchase
|
About Us
|
Links
|
Home
|
Can you use
TGen instead of an audio generator to test audio equipment?
Absolutely!
|
What Kinds of Audio Tests Can TGen Do? |
|
The square waves it produces, are they rounded or sharp (90 degree) at the corners? |
This sounds
like fun to answer.
Let's get the straight story here then... as we all know,
there is no such thing as an exact 90 degree square wave. TGen is naturally limited by the
sample rate. No appreciable quantities of energy above 22 khz can make it out of the soundcard.
I.e. it takes at least the time of one sample for
the flank to rise (or fall). But this is not the only problem. The proper time
for the flank to rise may land
anywhere between two sample times. This forces some kind of distortion into the square wave.
I have in fact created 3 separate square waves to deal with
this kind of issue. They are
square, sharp square, and soft square. Square wave rises and falls
at the next sample after the
correct rise/fall time. Sharp square does the same, except the very first sample of each
flank is exaggeratedly high or low in order to compensate for the
filtering in
the system
that will tend to round off the corner at approx. 3db = 20 khz.
SoftSquare doesn't flank
all at once. If the proper flank rise time lies between two sample times,
softsquare begins flanking
at the sample before the proper rise time, and finishes flanking on the next
sample. Each of the three waveforms has its own style of
distortion. You get to take your pick. No pc digital based square wave
generator that runs at 44100 can offer any more.
For most octaves, flanking is just not really an issue, but
the last octave of square waves will tend
to look
more like sin waves when the signal gets out of your sound card. Some quantity
of correction
can be done
by re-amplifying with a comparator that triggers at half signal maximum
(especially for
the soft
square wave form). Another help may appear in running in inverted right channel
mode.
Keep in mind that grounding could become an issue here. This
is easily solveable by coupling the signal through capacitors. One for
each signal line. about 47 microfarad should easily deal with most
any amplifier, but you may be able to go as low as 2 microfarad or so with no
problems. Most amplifiers have high input impedances, and 2 microfarad would
be more than plenty.
If we invert the right channel, and combine both channels
into one, the sample rate will be effectively doubled for every
soundcard that uses only one dac (which is probably every soundcard). I must
admit though, I haven't yet created a waveform to properly take advantage of
this. Some advantage will automatically appear though.
Now comes the hammer, for audio, frankly, this is all,
generally speaking, a mute point. The reason for this is that audio amplifiers have their own
input filtering. You can wiggle the input signal as fast as you want, but
those quick moves will never
make it to the outputs. Amps typically start chopping the fast stuff out
at a little under 20
kiloherz. Yeah, technically, it is possible to force high frequencies
through the filter by
exaggerating them. But this idea is essentially nonsense.