audio processing

Use Nyquist Sampling Theorem:

The signal that was digitally recorded will be band-limited in its analogue reversion, whereas the signal that is real-time will not be "as band-limited". For example, assuming that the digital signal was sampled at 44.1kHz, you could take FT of that signal and check to see how much power is beyond 22.05 kHz (rougly speaking). If there is "a lot" of power, the signal could not have been digitally processed, whereas if there is essentially none, then it probably was....unless....the original source of the signal is Barry White, in which case you will not see any power above 880 Hz, whether sampled or real-time.

Note that you have to know the sampling frequency to make this work.

-Le Chaud Lapin-

Do you ? I can't see how.

Graham

That'll work. ;-)

Graham

Let's assume that the recording wasn't made on a reel-to-reel, all-tube recorder with gold plated chassis, oxygen free PTFE insulated cables, all electrolytic caps replaced by polyester cotton, and mounted on vibration-proof gimbals.

The problem then becomes easy- any hi-fi buff will INSTANTLY be able to distinguish.

Paul Burke

The current term for " hi-fi buff " is " audiophool" ( tm) J.Woodgate.

Graham

"dspdspo" schreef in bericht news: snipped-for-privacy@f14g2000cwb.googlegroups.com...

Follow the cable. If it comes from the recorder, it is carrying the recorded signal. If it goes to the microphone, it is the real time signal.

--
Thanks, Frank.
(remove \'q\' and \'.invalid\' when replying by email)

but

maybe if you do the ol Nixongate tricks and compare the 50/60hz hum and see if its "genlocked" to your 50/60Hz

martin

I still sort of like the idea of a linear Marx generator. Have, say, a dozen mosfet ramp generators, all sitting on ground. Charge their supply caps to 800 volts apiece, then trigger them to start ramping together, with steering diodes to "erect" the string.

Transformers might be interesting, except for the 50:1 time spread.

John

pls explain... i didn't get what you said about "a lot of power" beyond

22.05kHz...?

i mean, shouldn't there be a difference in the frequencies of recorded and real time?

Be sure to digitize a good image of the source/tape switch!

Not until you get to the cut-off frequency for a particular arrangement. If you look below 22.05kHz, you will see essentially identical images. But for frequencies above 22.05kHz, the straight-analogue signal will show some power (you will see a bit of fuzziness there on the spectral analyzer), whereas for the digitally processed signal, you will see a drop-to-zero. Here's what's happening:

If you take the straight-analogue signal, and view its spectral content, you will most likely see components in the lower frequences (under 20kHz), as well as components at higher frequencies (above

20kHz).

If you look at the recorded signal, assuming that the spectral content of whatever was recorded also had frequency components higher than

20kHz, you will *not* see those higher components in the analogue output of the digitally processed signal. The reason has to do with filtering. To reconstruct an analogue signal perfectly, the rate of sampling must be at least twice the highest frequency component contained in the sampled signal. So if someone plays high-pitched music, and the highest frequency component in that music is 8 kHz, then one must operate the A/D converter at at least 16 kilosamples per second (kps) for perfect reconstruction which is equivalent to avoiding spectral overlap in the frequency domain. This is the essence of the Nyquist Sampling Theorem, and this theorem is directly related to what is really happing when you sample a signal and later convert it back to analog - when you sample the signal in the time domain, with the *intent* to regenerate the analogue signal, you should think in your mind, at each instant of sample, BAM!!! You are multiplying the signal with a sequence of impulses, where each impulse is a BAM! Then, to see what this combination of multiplied signals makes in the frequency domain, you must convolve the image of the sampled signal in the frequency domain with the Fourier Transform of the "BAM" signals, which, in the frequency domain, is yet another train of impulses, but scaled by a factor. To keep the resulting blobs from overlapping each other, the spacing between the impulses in the time domain must be very narrow, or equivalently, very wide in the frequency domain. But "very wide" is relative - if a signal is sufficiently band-limited for a particular spacing of the impulse train in the frequency domain, then no overlap will occur during convolution. To band-limit the signal, you must use an anti-aliasing (low-pass) filter before sampling to kill off any frequencies higher than 22.05 kHz, and another similar filter at the output.

So if you are listening to a signal that is analogue throughout the channel, it *could* have components higher than 22.05kHz. But if you're listening to a signal that has been low-pass filtered, digitally sampled (A/D), converted back to analog (D/A), and low-pass filtered again, you can be certain that components above 22.1kHz have been essentially killed off. For a digitally processed signal, anything beyond 22.05kHz will be due to noise and imperfections in the output filter itself, and should barely show up on a spectral analyzer.

It should come as no suprise to you that 22.05 kHz (44.1 ksps) is the cut-off frequency. It just happens to be near the threshold of hearing for many humans.

-Le Chaud Lapin-