[BC] IBOC "secrets" and my opinions.

[Johnson, Richard]

On Tue, 27 Mar 2007, Robert Orban wrote:

> At 10:39 AM 3/27/2007, Richard Johnson wrote:
>
> Just a few comments:
>
>> For the technical issues, the problem with digital and AM within the
>> same spectrum space is that the digital bins (samples) need to fit in
>> between the AM bins. In the case of digital, the difference between a
>> wanted bin and an interfering (unwanted) bin needs to only be that the
>> wanted bin be greater. Certainly a ratio of 2:1 (6 dB) would result in a
>> completely clean digital signal. However, for the AM of conventional
>> design (peak detector), the ratio ends up being the same as the
>> S/N, i.e., 100:1 for 40 dB, 400:1 for 52 dB, etc., when you
>> use envelope detection. Digital detection with filtering works by
>> completely suppressing __anything__ that is not in the passband.
>> This means that anything removed from the carrier by more than the
>> audio passband of 10 kHz is gone. It is removed __before__ any of the
>> nonlinear effects of analog circuitry can occur. The nonlinearity is
>> the nonlinearity of the A/D converter at the head-end.
>
> No digital filter can completely suppress out of band power. Theoretical
> "brick wall" filters require infinite time delay, and this is true
> regardless of the technology implementing the filter -- it's a "laws of
> physics" issue. Digital filters have cost/performance tradeoffs like any
> other technology. They can be very effective, and the cost for making very
> selective filters keeps decreasing. But "completely suppressing?" Can't be
> done.
>

Well yes it can. It was described in my first post on this subject.
I also described a 16-bit 20 MHz system so the 96 dB was previously
qualified. "Completely eliminated," in this context means that there
are no bits exercised in the 16-bit system. It is done by doing a
DFT or FFT, zeroing out the offending bins, then performing an inverse.
To obtain sufficient resolution, one needs to use a decent window
(a slightly modified Bartlett window will do) and after all is done,
the result needs to be convolved with the same window. This is all 
possible because, as previously described, "realtime" is only 10,000 
events per second.

> Moreover, DSP technology for consumer radios (including digital IF
> filtering and detection) has been around for some time and available for
> purchase in some currently manufactured radios. Eventually, I expect that
> all radios will be DSP-based because the tech will get progressively
> cheaper as time goes on.
>

If are the same Orban that makes the digital stereo generators and
Optimod (I've been out of the broadcast industry for 30 years), then
you know that you can even generate a "perfect" FM stereo signal
with zero 38 kHz (completely suppressed) using modern technologies.

Also, the DSP-based, nowadays, doesn't involve a conventional DSP.
The digital algorithm in put into a gate array on the same chip
as the A/D and D/A. Receivers will soon consist of a chip with
some external ESD protection diodes. It's already been done for
military tactical radios and they are sensitive enough so that
satellites (TDRSS) are used for relay.

>
>> Of course a practical receiver would have some kind of AGC amplifier
>> before the A/D converter to prevent outright overload and to try
>> to exercise as many bits as possible so that you truly have 96 dB
>> of dynamic range. Nevertheless, the technology is readily available.
>
> 96 dB corresponds to a very specific word length: 16 bits.
>

Yes, as previously described.

>> Anybody who has "brought up" a ham SSB signal out of the noise
>> has first-hand experience how synchronous detection works. Digital
>> synchronous detection works even better because with two sidebands
>> containing the same information, you get an automatic 6 dB improvement
>> in S/N. Then, anything that is not the same on the lower sideband
>> as is on the upper sideband, inverted, will not be recovered. This
>> means that only the original audio, which was identical on both
>> sidebands, will be recovered. Any of the so-called "splatter" that
>> is not synchronous with the audio, and therefore is not identical
>> on both sidebands, will not be present in the output. Needless to
>> say, you can't use AM stereo in a detector designed to suppress
>> anything that is not the same in both sidebands.
>
> Assuming that the RF noise floor consists of random noise, the S/N gain
> will be 3 dB, not 6 dB. The noise in the sidebands is uncorrelated and thus
> sums in an r.m.s. manner, resulting in 3 dB greater than the noise in one
> sideband, while the signal sums to 6 dB because the signal in the two
> sidebands is correlated. This gives you a 3 dB improvement in S/N.
>

This is being picky. I was talking about getting rid of the IBOC junk
which I am fairly sure is not uncorrelated noise.

> If the interference under the sidebands is not random noise, then the
> amount of improvement will depend on the nature of the interference and can
> be quite remarkable if the interference is mainly localized under one
> sideband.
>

Correct.

> Bob Orban
>
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Richard B. Johnson
Project Engineer
Analogic Corporation


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