FPGA Central

Jerry Avins wrote:
> Mark wrote:
> > The L and R are time division multiplexed by sampling each at 38 kHz.
> > The L and R samples are interleaved and FM modulated onto the main RF
> > carrier. This is spectrally equivalent to what you described. This is
> > two different descriptions of the same thing, one description looking
> > at the frequency domain, and one description looking at the time
> > domain. Actually there are two different design approaches to an FM
> > MPX encoder, the TDM approach and the L-R matrix approach, both create
> > the same output signal and are fully compatable.
>
> There is no sampling. 38 KHz is the frequency of the subcarrier that
> carries the difference signal. The sum signal is transmitted as in a
> mono signal, so that a mono signal is received as such and a mono
> receiver receives L+R. That's what's meant by compatibility. After the
> difference signal is recovered from by demodulating the subcarrier (with
> the aid of the 19 KHz pilot tone), the two signals are matrixed to
> create L and R. If the pilot tone is too weak, there is no matrixing.
>
> Which standard have you been reading?
>
> Jerry
> --


The 38 kHz sampling and the L-R matrix are different but EQUIVALENT
methods.

Read this then we'll talk again.

http://members.tripod.com/~transmitters/stereo.htm

Mark


> Engineering is the art of making what you want from things you can get.
> ¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯ ¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯

Mark wrote:
>
> Jerry Avins wrote:
>
>>Mark wrote:
>>
>>>The L and R are time division multiplexed by sampling each at 38 kHz.
>>>The L and R samples are interleaved and FM modulated onto the main RF
>>>carrier. This is spectrally equivalent to what you described. This is
>>>two different descriptions of the same thing, one description looking
>>>at the frequency domain, and one description looking at the time
>>>domain. Actually there are two different design approaches to an FM
>>>MPX encoder, the TDM approach and the L-R matrix approach, both create
>>>the same output signal and are fully compatable.
>>
>>There is no sampling. 38 KHz is the frequency of the subcarrier that
>>carries the difference signal. The sum signal is transmitted as in a
>>mono signal, so that a mono signal is received as such and a mono
>>receiver receives L+R. That's what's meant by compatibility. After the
>>difference signal is recovered from by demodulating the subcarrier (with
>>the aid of the 19 KHz pilot tone), the two signals are matrixed to
>>create L and R. If the pilot tone is too weak, there is no matrixing.
>>
>>Which standard have you been reading?
>>
>>Jerry
>>--
>
>
>
> The 38 kHz sampling and the L-R matrix are different but EQUIVALENT
> methods.
>
> Read this then we'll talk again.
>
> http://members.tripod.com/~transmitters/stereo.htm

That's an interesting page. I find a stated advantage of FM over AM,
"More distance on the dial between FM stations", a bit nerdy, but a
small point. What you loosely call "sampling" is called in the article
"time-delay multiplexing" and "chopping", hardly the same thing.
Moreover, the paper describes a stereo modulator, but I was describing
the demodulation process. My mistake there; you had mentioned the
encoder before I chimed in.

Jerry
--
Engineering is the art of making what you want from things you can get.
¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯ ¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯

Jerry Avins wrote:
(snip regarding FM stereo multiplexing)

>> The 38 kHz sampling and the L-R matrix are different but
>> EQUIVALENT methods.

They are equivalent if you ignore the filtering problem.

It might be that today one could generate a nice digital filter
to do it, but that would not have been possible when the standard
was being written.

>> Read this then we'll talk again.

>> http://members.tripod.com/~transmitters/stereo.htm

> That's an interesting page. I find a stated advantage of FM over AM,
> "More distance on the dial between FM stations", a bit nerdy, but a
> small point. What you loosely call "sampling" is called in the article
> "time-delay multiplexing" and "chopping", hardly the same thing.
> Moreover, the paper describes a stereo modulator, but I was describing
> the demodulation process. My mistake there; you had mentioned the
> encoder before I chimed in.

I find it more believable as a decoder. The filtering required for the
encoder would be very hard to do, except possibly with a digital filter
at a much higher frequency. Well, for a decoder the filtering isn't
easy, either. You have to at least filter everything above 53kHz
sharp enough to keep the tails of the SCA subcarrier centered at 67kHz
away. SCA is an FM subcarrier so it can have long tails.

The inputs to the encoder must be filtered to less than 15kHz, so
obviously sampling at 38kHz would be enough. Balanced modulators were
pretty easy to make, even with vacuum tubes. Sharp filters with a flat
passband were not.

For comparison purposes there are two descriptions of the NTSC color
subcarrier, and the choice of subcarrier frequency. In one there is a
180 degree phase shift between successive lines and for a given line
between successive frames. This tends to reduce the visibility of the
subcarrier on a monochrome receiver. In the other description, assuming
relatively low horizontal spatial frequencies in the video image, the
luminance signal has peaks at multiples of the line rate and the
chrominance signal has peaks in between those.

As with the FM stereo case, one is time domain and one frequency domain,
but the result is the same.

-- glen