From: sehari@iastate.edu (Babak Sehari)
Subject: Re: FM stereo in TIME domain. (Was Re: How does stereo FM radio work?)
Originator: sehari@lisa.ecss.iastate.edu
Organization: Iowa State University of Science and Technology, Ames, Iowa.
Distribution: na
Date: Fri, 2 Oct 1992 23:44:24 GMT

In <1992Oct1.165659.8131@bnr.ca> mwandel@bnr.ca (Markus Wandel) writes:

>In article <1992Sep24.125938.3630@research.nj.nec.com> markl@research.nj.nec.com (Mark Lee) writes:
>>I don't know if this is a FAQ, but I have never seen this question asked
>>here and I want to know.  How do they broadcast and receive the two
>>channels of information needed for stereo on FM radio?  ...
>>
>>I have asked several people who know high-frequency electronics about
>>this and never gotten an intelligible answer.  I'd appreciate it if
>>someone could point out to me a decent text or article, appropriate for
>>someone like me who already understands basic mono AM and FM.

>I have read all the responses that were posted to this so far.  I have
>read technical descriptions of how stereo receivers work.  I have read
>communications textbooks, I've asked people, and always I got the frequency
>domain story of the frequency shifted L-R signal, frequency doubling
>and signal multiplication and so forth.  It sounded very complicated
>and elaborate to implement in practice.  So why was it that the whole
>bloody stereo demultiplexer circuit in a receiver that I had to fix it in,
>consisted only of a few transistors, a couple of inductors, and some
>diodes?  Well, this is how it works in TIME domain.

>Say that everything was based on square waves, and that you started out
>with a 38KHz clock waveform.  Based on this clock, you alternately select
>the left or the right channel for transmission, transmitting a sequence
>of samples L, R, L, R, L, R, etc.  You also add a 19KHz pilot tone at
>half the clock frequency.

>The receiver reconstructs the 38KHz clock from the 19KHz pilot tone, and
>routes the samples to the appropriate channels.  One gets L, ,L, ,L, ...
>the other gets  ,R, ,R, ,R, ... interpolating with low-pass filters to
>reconstruct the audio.  That's it.  A mono radio lacks the sample separating
>circuitry and instead feeds the low-pass filter L,R,L,R,L,R,... producing
>an average of the two channels and thus mono audio.  Now it sounds very
>simple, doesn't it.  The sample separator in the receiver I looked at was
>just a transformer feeding a switching matrix of diodes.  The compound
>audio was fed in and also a large amplitude clock, which alternately cut off
>one set of diodes or the other, allowing the audio only into the channel
>whose diodes conducted at the time.  Recovery of the 19KHz pilot tone is
>done with a bandpass filter, frequency doubling by fullwave rectifying it
>and filtering again.  The stereo lamp lights when the 19KHz is present.
>When it's not, no 38KHz is generated, no "clock" goes into the matrix,
>no diodes are cut off at any time, and both channels get the audio all the
>time.  Hopefully with this description you could pretty much DESIGN the
>stereo decoder.

>Incidentally it all corresponds to the frequency domain version (L-R, L+R
>signals, DSB-SC modulation of the L-R subcarrier, etc.).  The various
>lowpass filters make reality match the theory.

>Markus Wandel
>markus@pinetree.org <-- NOT the source of this posting.

Nothing new with your design.  The telephone company multiplex many channels
this way. After the start bit, or synchronization bit, the first bit belongs
to channel 1 the second belongs to channel 2 and so on...

Another way, which is called statistical multiplexing, sends each PCM with
it's address. This way if one channel does not have any data it does not
wast the time of transmitter by sending 0s all the time.  Disadvantage
with this technique is it has overhead for addressing.

                              With highest regards,
                                   Babak   Sehari.