[Lf] Re: lf digest, Vol 1 #146 - 1 msg

[Terry Fox]

As we talked about at tacos last Saturday, you will probably need a lossless
compression system.  They are generally limited to a 2:1 compression ratio at
best.  Any other system will throw away data that is needed.  As an alternative,
we can reduce the samples in frequency.  Since you want a narrow bandwidth
result,  if you resample at a lower frequency, you can drop the audio frequency
range to close to the bandwidth.  This gives the same frequency range
(remembering Nyquist), but much fewer samples.

For example, a 200Hz bandwidth centered around 1KHz would require at least a
2.2KHz sample rate.  Moving the samples down to 300Hz would only require a 800
Hz sample rate (I think).

Now, apply a lossless compression of about 2:1.  That would result in 400
samples per second.  If those samples were reduced from 16 bits to 12 or even 8,
then more data can be sent.  While frequency compression may lose too much data,
a non-linear amplitude compression method may be successful.

I think we can achieve some compression on the audio data without losing data
integrity.  If we can achieve a low enough data rate, we can also use radios to
connect to remote receivers in addition to the Internet.  Sending the audio as
data would be much better than just sending the audio itself.  It would not be
distorted by the retransmission method.
Terry

Vittorio De Tomasi wrote:

> Hi Frank and the list,
>
> here are my opinions on your problem:
>
> if you want to digitize a receiver outbut and still be able to detect
> very faint signals buried under the noise, forget about audio
> compression, like MP3, RA, GSM, and so on!
>
> The reason is quite simple: compression (as well as denoising!) is done
> designing a suitable mathematical model that is fitted to the signal.
> Best fit is obtained by minimizing the energy of the difference signal
> (input minus modeled signal): computationally efficient methods exist to
> compute the signal model that fits at its best the incoming data. The
> difference between input signal and the model is simply discarded, and
> the transmitted information is the one needed to build up the model.
>
> MP3 adds to the best fitting process a psychoacoustic model: suppose
> that you listen to a 1000 Hz signal. Now add a 1500 Hz signal, and
> increase its amplitude until you hear it. Turn it off, and do the same
> for a tone at 1010 Hz: you will need quite a larger amount of signal to
> become aware that the tone at 1010 Hz is played with the steady 1000 Hz
> tone. MP3 encoders recognize the presence of tone pairs, and apply
> dynamic compression, i.e. they use a coarser signal quantization when a
> weak tone is close to a strong one. Imagine this kind of processing
> applied to VLF signals....
>
> The best compression to do is Nyquist compression: if you have B Hz of
> bandwidth, sample them at a rate equal to 2B (maybe a little more, just
> to avoid aliasing), and use a suitable number of bits to get the needed
> dynamic range. So for a data stream of 16 kbit/s, you can transmit 500
> Hz of bandwidth with 16 bit (i.e. about 96 dB of dynamic range, not
> bad...).
>
> Amplitude data can be encoded into the data stream replacing say a
> sample every 16k, so getting an amplitude measurement every second. The
> "lost" sample will be silently ignored also by the most sophisticated
> DSP engine.
>
> However there is a more efficient method: why don't you simply transmit
> the frequency spectrum ?!? I suppose you have a two way data channel, so
> you can think of sending commands to the "DSP server" in order to
> transmit you the desired frequency slice. What do you think about this
> ?!?
>
> vy 73
>
> Vittorio IK2CZL
>
> > Today's Topics:
> >
> >   1. Digitized Audio (Frank Gentges)
> >
> >   ------------------------------------------------------------------------
> >
> > Subject: [Lf] Digitized Audio
> > Date: Sat, 24 Jun 2000 20:02:20 -0400
> > From: Frank Gentges <fgentges at mindspring.com>
> > Organization: K0BRA
> > To: LF at AMRAD.org, tacos at AMRAD.org
> >
> > Hi,
> >
> > At tacos today I mentioned that I was looking for a good method to
> > digitize the 300 Hz bandwidth CW output of the RX320 to transmit
> > remotely at something around 16 kilobits per second.  I have looked at a
> > few available options in WIN98 and at Xing's MP3 encoder.
> >
> > I would like to find the best option that will provide a good
> > spectrogram with Spectran at the remote end.
> >
> > If I were to set the RX320 BFO for a 250 Hz, then the band should extend
> > from 100 Hz to 100 + 300 = 400 Hz.  But, the RX320 at 300 Hz bandwidth
> > has quite a bit of energy beyond 500 Hz and you can hear the beat note
> > come through zero which means significant artifacts could creep into the
> > spectrogram.  Simply put, the 300 Hz bandwidth has quite a bit of
> > transition band beyond the 300 Hz edges before the signal is far enough
> > down to ignore.
> >
> > One option would be to set the BFO for a 1 kHz center frequency like we
> > do now for driving Spectran.  The signal could be digitized and further
> > filtered digitally in real-time yielding a 16 kilobit per second
> > stream.  A reverse process could then be used on the remote end.  If
> > this could have limited processing load it could be done in the PC.
> > While we are at it we need to multiplex into the stream the RX320 signal
> > strength data, but lets not get ahead of ourselves.
> >
> > Another option would be to use a streaming audio process like MP3 or the
> > like to encode the audio.  MP3 is an open specification and we should be
> > able to use it freely.
> >
> > RealAudio might be an option but it is proprietary and does not seem to
> > have a low rate option.  Neither do we know the impact on Spectran of
> > its artifacts.  It would be nice to know how much we might be missing
> > here.
> >
> > In the end, I would like to be able to put a remote RX320 and computer
> > anywhere in the world and with a modem based internet (or modem direct)
> > connection, be able to listen to the LF band.
> >
> > Any thoughts?  Even better, any volunteers to work on this problem so we
> > can put your solution in our handbook?
> >
> > Frank
> > --
> > Frank Gentges
> > K0BRA, ex AK4R, W3FGL
> > Check out our LF web page at <http://amrad.org/projects/lf>
> >
> >   ------------------------------------------------------------------------
> > _______________________________________________
> > lf mailing list
> > lf at amrad.org
> > http://www.amrad.org/mailman/listinfo/lf
>
> --
> *************************************************************************
> Vittorio De Tomasi         ik2czl at amsat.org
> Home page:                 http://space.tin.it/scienza/vdetomas
> My DSP page:               http://www.freeyellow.com/members/padan
>
> "Wir muessen wissen; wir werden wissen" (David Hilbert)
>
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