Message-ID: <01C51A66.BD219C10.actalbot@southsurf.com>
From: Andy <actalbot@southsurf.com>
To: "'rsgb_lf_group@blacksheep.org'" <rsgb_lf_group@blacksheep.org>
Date: Thu, 24 Feb 2005 11:48:17 -0000
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Subject: RE: LF: Re: Off Topic  Ft101zd
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For a normal A/D sampling at a rate approaching the Nyquist bandwidth (Fsample 
/ 2) , Signal to Quantisation noise ratio is approximately 6.N - 1.7 - (peak to 
mean ratio).  Where N is the number of bits.  So for the telephone network with 
8kHz sampling, if we need 46dB S/N ratio for voice with a typical 10dB 
peak-to-mean, then somewhere around 9 - 10 bits would necessary. So, to achieve 
this performance on the telphone network using 8 bits / sample, non-linear 
sampling (mu-law or A-law coding) is used which has the effect of compressing 
then expanding the audio dynamic range to effect the increase in S/N ratio for 
practical signals.

NOW, f you go to a lower number of bits but sample at a much higher rate, then 
other tricks become possible.  For example, delta modulation used many years 
ago for a simple digital transmision over radio used 1 bit sampling, 
transmitting a '1' if the amplitude has increased from the previous sample, and 
'0' if it decreased.  For speech a sampling rate of 30 to 50kHz gave acceptable 
fidelity.

In modern Codecs this technique is carried further, but made transparent to the 
user who sees only a 'classic' A/D converter operating at a lower rate.  For 
example, the codecs used in most soundcards actually sample the audio at many 
MHz at a low resolution (1 or 2 bits often) in a simple but accurate A/D.  Then 
sucessive samples are combined digitally and processed to give an effective 
sampling rate at the familiar values of 11025 - 44100Hz.   The disadvantage of 
this approach is that the A/D converter can never truely digitise a DC signal. 
 It can get arbitrarily close, but never down to 'true' DC.   This is one 
reason why a good Software Defined Radio using two channel I/Q processing 
cannot be made out of a soundcard / PC combination.   There will always be a 
hole right in the middle of the spectrum corresponding to the DC response. 
  There is also the issue of capacitor coupling that reduces DC response on 
practical hardware, but that's a matter for manufacturers.

The same techniques can be used at higher frequencies, but to achieve MHz 
effective sampling rates needs GHz sampling speeds in the first place - which 
is pushing sample-and-hold technology to its limits.   The arguments are 
exactly the same as described in earlier posts on this topic.   When a wideband 
signal is filterd and decimated (the sampling rate reduced) the S/N ratio of a 
narrowband signal can be increased way beyond what would have been possible at 
the original sampling rate.

Modern approaches can use combinations  eg a high specification 4 or 8 bit 
converter coupled with down sampling to give 16 bits or more.   But what goes 
on inside modern A/D chips is probably proprietory information and we see only 
the outside, effective, conversion performance - and go 'WOW' when we see the 
specs!

Andy  G4JNT


On Thursday, February 24, 2005 11:24 AM, Claudio Pozzi [SMTP:smtp01@email.it] 
wrote:
> On Thursday 24 February 2005 02:18, Alexander S. Yurkov wrote:
> > Dear Alberto,
> >
> > On Wed, 23 Feb 2005, Alberto di Bene wrote:
> > > Do really exist A/D converters capable of 24-bit resolution at 30
> > > MHz bandwidth ?
> >
> > Why one need 24-bit? Though 16 bit is adequate for most SW,MW,LW
> > recievers. 16 bit yelds abt 90 dB dynamic range (every bit exept
> > sign bit yelds 6dB).
>
> This means that 1 bit give a sufficient MDS? The recovered audio is
> ON/OFF tone?
>
> > But this is dynamic range if there is no
> > filtering. When you make banwidth narrow by DSP then dynamic range
> > will be improved because noise decreases. If frontend sampling rate
> > is, say 100 MHz (there is such an ADC) and bandwidth of DSP filter
> > is about, say 10 kHz this yelds 40 dB of noise decreasing. Thus RX
> > dynamic range to be about 90+40=130 dB!   Realy DD is not 130 dB of
> > cose. But it much less due to analog (!) effects in ADC absolutely
> > similary as in conventional analog RX. This is not digital effect!
>
> But without a single channel roofing filter a 130 dB outband signal
> make the AD converter overload, despite any sampling rate. Or not?
>
> I have some unanswered question:
>
> 1- How many bits are needed for a good audio reproduction, i.e. a S/N
> ratio of 10 dB with some voice dynamics? Telephone signals usually
> are sampled 8 kHz 8 bits for a telephonic quality (with no
> compression). May be that 4 bits are sufficient but 1 bit I don't
> think.
>
> 2- Those bits are to be taken in account for calculating the dynamic
> range? i.e. 16 - 4 = 12 bits =72 dB dynamic range.
>
> 73 de Claudio, ik2pii
>
> -
> Claudio Pozzi  -  Happy Linux User  -  http://www.qsl.net/ik2pii
>
>
>
>
>
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