In Sony SoundForge, you can synthesize a sine up to 1/2 the
sample rate, minus 1. So the highest frequency I can generate is
22,049, which produces a beautifully nyquist-modulated waveform.
One could not ask for a better visual representation of
aliasing! The SoundForge feature for doing this is Tools /
Synthesis / Simple (SF also generates DTMF tones and FM waves).
ac
Allen Cobb
http://acobb.com
http://shakespeare.acobb.com
http://timbreproductions.com
-----Original Message-----
From:
Behalf Of Gianni
Pavan
Sent: Thursday, October 20, 2005 8:49 AM
To:
Subject: Re: [Nature Recordists] MT2496 digital input
Hi all,
I just discovered that with Adobe Audition, maybe with
other
editors too, it is possible to generate the Nyquist frequency I
used
for testing digital inputs.
In the panel Menu' > Generate > Tones you simply need to
generate a
pure sinusoid at half the sampling rate (48000 for 96k sampling
rate), set the Start Phase to 90, set the amplitude to the dB
level
you want, and press OK.
Then, looking at the waveform in great detail you'll see the
samples
alternating among min and max and the Nyquist sinusoid passing
through them.
After passing that signal through a resampling device you'll see
again samples with alternated values, but those values will be
different from the original values and very likely they will
change over time.
Of course you'll see a new Nyquist sinusoid passing through the
samples, but its amplitude will be different and very likely it
will
be modulated.
Gianni
At 01.29 20/10/2005, you wrote:
>Gianni Pavan wrote:
>
> >I create a 16 bit stereo .wav file and I fill it
> >with integer samples with values
> >-32000,+32000,-32000,+32000 ... with a program
> >written in Visual Basic. Such a sequence of
> >samples corresponds to a perfect sinusoid at the
> >Nyquist frequency (half of the sampling rate)
> >that can't be recorded from an analog source.
> >If you feed a non-resampling digital input the
> >values are just kept and stored in a new file.
> >But if the digital input device resamples the
> >data stream all those value are processed with a
> >resampling algorithm that produces a new series
> >of samples that don't match with the original
> >ones. A resampler keeps the original stream,
> >upsamples to a much higher frequency, performs a
> >digital filtering to cut frequencies higher than
> >the new sampling rate (that could be the same of
> >the input or a different one) and also attenuates
> >frequencies close to the Nyquist frequency, then
> >downsamples to the final rate. If your original
> >stream is 48k and the output of the resampler is
> >exactly 48k again, frequencies higher than
> >22-23kHz will be attenuated and the "Nyquist
> >signal" will be greatly attenuated.
> >In some cases the input stream has not exactly
> >the rate of the sound board clock and in this
> >case the "Nyquist signal" will appear at a
> >frequency that is not the Nyquist frequency of the new file.
> >
> >A "no resampling" digital input is able to lock
> >on the incoming rate and to transfer samples
> >without any further processing. If you open with
> >Audition the file created by the digital input
> >device you see very clearly the difference among
> >resampled and not-resampled "Nyquist signal"
> >
> >I hope I was clear enough,
> >Gianni
>
>That's a cool tool. I don't have any way to make it. Could you
put a
>test file in a place where people could download it?
>
>-Dan Dugan
>
>
>
>"Microphones are not ears,
>Loudspeakers are not birds,
>A listening room is not nature."
>Klas Strandberg
>Yahoo! Groups Links
>
>
>
>
>
>
>
>
>
>
>--
>No virus found in this incoming message.
>Checked by AVG Anti-Virus.
>Version: 7.0.344 / Virus Database: 267.12.4/142 - Release Date:
18/10/2005
--------------------------------------------------------------
Gianni Pavan
Email
Centro Interdisciplinare di Bioacustica e Ricerche Ambientali
Universita' degli Studi di Pavia
Via Taramelli 24, 27100 PAVIA, ITALIA
Tel +39-0382-987874
Fax +39-02-700-32921
Web http://www.unipv.it/cibra
--
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Checked by AVG Anti-Virus.
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