On Sun, 15 Mar 2015 13:54:03 GMT,
N0S...@daqarta.com (Bob Masta)
wrote:
Yes, that's one method that barely works. The 2nd harmonic of the
common 44 KHz audio sampling frequency would bring the bias signal
into the audible region at 12 KHz. 48 KHz would be better with a 4
KHz mix. The problem is that the sampling frequency is usually quite
symmetrical, and therefore lacks sufficient 2nd harmonic energy to do
much mixing. 4 and 12 KHz are also in the audio region, making it
difficult to see under that recorded audio. Originating from a
computah clock, which might be dithered (spread spectrum clock) to
reduce emissions, the result is a wide, dirty, low level, and useless
mix. The 3rd harmonic of 44 KHz would be at a higher level, but it
mixes to 32 KHz, which might be better, but is out of the range of my
equipment. Might be worth a try.
In the distant past, I've detected the tape bias signal by slowing
down the original 7 ips tape to 15/16 ips. That kills most of the
audio, and shifts the 100 KHz bias signal down to 12.5 Khz, which can
be detected. Lots of problems with this method and more than a few
tricks involved, but it can be made to work. The big problem is that
it has to be done with the original tape, which is often unavailable.
Much better is a ferrite tape head that offers expanded frequency
response, typically to 1 Mhz. These are now commonly available but
were previously rather specialized devices. They're used in systems
that phase(?) lock onto the bias signal to provide an AFC (automagic
frequency control) to eliminate flutter and wow from the tape:
<
http://www.nab.org/xert/scitech/2012/radioTechCheck/RD020612.asp>
<
http://www.plangentprocesses.com> (See links at bottom of page)
>So if the bias is normally -60 dB on the tape itself, how
>much of it gets through the anti-alias filter when
>converting to digital?
I don't have any examples available, so I'm guessing from memory. I
would guess(tm) a 16 bit digitized bias level would be about -100 dB
below the peak audio level. That's fairly horrid when the audio band
noise floor is probably about -60dB. Trying to extract a signal 40 dB
into the noise is not my idea of fun. Yet, given time, it can be
done. Plenty of articles on detecting signals below the noise floor:
<
https://www.google.com/search?q=detecting+signals+below+noise+floor>
The easiest is a sliding narrow band filter, that slowly and
repeatedly scans across the 100 KHz area of interest, collecting
signal level data and bin counting. I built one of these which
produced its output on an x-y plotter. It was a crude autocorrelator,
that looked for coherent signals. After about 1,000 passes, I could
see a bump at the expected frequency, if the pen didn't rip the paper:
<
http://en.wikipedia.org/wiki/Autocorrelation>
It was also mechanically sensitive enough that I could see doors
closing and changes in air pressure on the plot. The catch is that it
could easily take a day or two to see anything meaningful.
Today, there are certainly better methods of audio forensics. I'm
mostly familiar with older analog techniques and am somewhat lost in
todays digital DSP world. The stuff I do today is far less
sophisticated. Mostly it's signature analysis by looking at waterfall
plots, spectrograms, sonograms, etc of radio transmissions, looking
for residual tones and junk that can help identify the source. I've
used your program (Daqarta) for this, but prefer Spectrum Lab for
signal analysis.
The big problem with using a computah to look under the noise is that
the FFT requires a huge number of samples. For example, with a common
44 Khz sampling rate, in order to resolve 0.1 Hz (to reduce the
noise), I would need 880,000 data samples.
>Or do they digitize with special
>setups that use *no* (or minimal) anti-aliasing specifically
>to allow these sorts of analyses?
I don't believe there's a single established method for extracting
such signals. There are too many different types of recorders, media,
and encoding schemes for one solution to work well with all of them.
The wide band tape head is probably the most useful, but only with
original media.
I don't know of any arrangement that does not use some form of
anti-aliasing filtering to keep the audio and the bias signals from
mixing. The better recorders place their cutoff frequencies quite
high (about 50 KHz) to prevent group delay problems at the high end of
the audio spectrum. Some people claim that they can hear phase
shifts, so preserving the original waveforms has become a requirement.