FM stereo

I have a digital armband radio, for jogging.

It has a FM stereo detector, with internal threshold. When the signal gets weak or noisy, it switches to mono, and the noise drops significantly.

How does FM stereo decoding work, and why is it so much noisier than mono?

-- Rich

Reply to
RichD
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"Rich Dope"

** You never heard of Google ??

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..... Phil

Reply to
Phil Allison

The short answer is that the vast bulk of the power being transmitted goes to L+R using regular old FM as you see any textbook describe. When stereo was added, they just stuck a subcarrier at 38kHz in the baseband (which is ignored by monoaural receivers that cut off at 15kHz of thereabouts) and encoded L-R nearby... using a small fraction of the total signal power.

Since you need L-R to recover the left and right channels separately (L+R added to L-R gets you L; subtract the two and you get R), as the station's signal gets weaker the SNR of L-R drops into the noise more quickly than that of L+R and hence stereo broadcasts become noisy even when monaural (L+R) is still doing OK.

Wikipedia or Google can find quantitative results for things like "vast bulk of the power" if you're so inclined. :-)

---Joel

Reply to
Joel Koltner

It's here:

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The pilot tone is being decoded and when its level is to the receiver's liking it switches to stereo.

--
Regards, Joerg

http://www.analogconsultants.com/

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Reply to
Joerg

"Joerg"

** The 19kHz pilot tone contains no information - it's purely a reference of frequency and phase for the stereo decoder.
** The level of the pilot tone is fixed at 10% max modulation of the carrier.

Automatic stereo/mono switching is carried out on the basis of the signal strength in the IF stage.

.... Phil

Reply to
Phil Allison

The last ones I've dealt with used the SNR of the pilot tone to do exactly that :-)

--
Regards, Joerg

http://www.analogconsultants.com/

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Reply to
Joerg

** Utter BOLLOCKS.

Piss off - WOG MORON.

..... Phil

Reply to
Phil Allison

Oh yeah?

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Quote "The pilot detector output is designed as an open collector output, therefore an external pullup resistor is needed. To force the decoder to "MONO" Pin 19 has to be clamped to a voltage below 0.8V."

--
Regards, Joerg

http://www.analogconsultants.com/

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Reply to
Joerg

"Joerg" = f****it WOG from HELL

** Giant HUH ???

This anencephalic must be posting from a mental asylum.

FUCK off - YOU WOG MORON.

.... Phil

Reply to
Phil Allison

a

o.

t,

I see Phil is showing off his intelligence again. ;-)

Reply to
miso

On a sunny day (Mon, 23 Nov 2009 16:07:30 -0800 (PST)) it happened RichD wrote in :

FM encoding uses a 38kHz carrier on which the L-R and R-R signals are QAM modulated. To sync the carrier, a 19 kHz (f/2) pilot tone (above audible) is transmitted with the audio. The audio is normally transmitted as L+R.

After decoding, the signals can be combined by simple addition into L and R. The noise decreases when you switch to mono, because the whole decoder is switched off, and that part of the spectrum that contains noise, the sidebands around 38 kHz, is no longer fed into the signal matrix. All this IIRC.

Reply to
Jan Panteltje

On a sunny day (Mon, 23 Nov 2009 16:07:30 -0800 (PST)) it happened RichD wrote in :

FM encoding uses a 38kHz carrier on which the L-R and R-R signals are QAM modulated. To sync the carrier, a 19 kHz (f/2) pilot tone (above audible) is transmitted with the audio. The audio is normally transmitted as L+R.

After decoding, the signals can be combined by simple addition into L and R. The noise decreases when you switch to mono, because the whole decoder is switched off, and that part of the spectrum that contains noise, the sidebands around 38 kHz, is no longer fed into the signal matrix. All this IIRC.

PS:. I did some experiments with L+R and L-R encoding in mp3 and AC3, mainly originating from a request for a very narrow bandwidth audio link that would drop to mono if even less bandwidth was available:

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So you can have a mono mp3, that is L+R, and then send L-R over a second channel, even with a different, or LESS, bandwidth, and at the other end combine the signals, and have good stereo again. And if the second L-R channel fails, it will just drop back to mono. Works very well, even with [in] highly compressed formats. The 'subtract_wave' program I wrote to do this is here:
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All Linux GPL code.

Reply to
Jan Panteltje

YRNC it would seem! There's no QAM in the commonplace Zenith-GE pilot-tone system, just DSB-SC AM of a 38 kHz tone by the L-R difference signal. The L+R sum signal is provided by the monaural, baseband FM signal. L and R are separated by simultaneous equations and the 19 kHz pilot tone is obtained by dividing the frequency of 38 kHz tone by two.

Chris

Reply to
christofire

On a sunny day (Tue, 24 Nov 2009 20:30:16 -0000) it happened "christofire" wrote in :

You are right, it is double sideband surpressed carrier.

are

Yes, but in the decoder the 38 kHz is obtained by multiplying the 19 kHz by 2 I think, and that 38 kHz is then used to demodulate that DSBSC signal. This right?

Reply to
Jan Panteltje

... hence DSB-SC!

Yes, the obvious way.

Chris

Reply to
christofire

reference=20

signal=20

Alas, Phil A. is full of baloney on this one. The stereo decoder makes that decision, and it cares about S/N of the pilot tone itself. I have built many with my own hands. Even designed one or two.

Reply to
JosephKK

stereo.

=20

signal=20

exactly=20

Oh gosh. You are talking about IC's that do it all for you. I have done one with tunable "cans" (resonant transformers similar to IF transformers), and one with a 76 kHz PLL. The PLL performed much better. About 40 years ago though.

Reply to
JosephKK

QAM modulated.

transmitted with the audio.

Not QAM (at least not originally) just a double balanced modulator thus no 38 kHz carrier.

and R.

contains noise,

It was actually set up so that the math worked correctly for TDM decoding at 38 kHz or FDM subcarrier and matrix decoding. Some performance tradeoffs were observed later. It turned out that decoder solutions that used a bit of both techniques had the best performance. All known 40 years ago.

Reply to
JosephKK

"christofire"

RichD

are=20

L=20

matrix.

and=20

obtained=20

kHz=20

One obvious way. It however required carefully tuned "cans". The 76 kHz PLL method ended up far simpler and more reliable. It is what all the ICs implement.

Reply to
JosephKK

Why not ?

The mono (L+R) signal extends to 15 kHz, while the lower sideband of the difference (L-R) starts at 23 kHz extending to 38 kHz, while the upper sideband goes from 38 to 53 kHz and there may be a data carrier at 57 kHz and some auxiliary audio (SCA) in some countries.

The noise density from the FM demodulator increases at least with second power of audio frequency until the IF filter passband again drops the noise density.

Thus, it sufficient to detect the noise power in some "quiet" frequency band. For example the 15..23 kHz band is supposed to be quiet with only the 19 kHz pilot in the middle, so it should be possible to measure the pilot power and compare it to the total noise power of the frequency band. Alternatively notch out the 19 kHz and measure the noise power in that band.

As long as the IF amplifier has sufficiently gain to drive the FM demodulator into saturation at low signal levels, the FM demodulator pilot tone output level is indeed constant.

However, when the IF amplifier is no longer limiting, the pilot starts to drop. Even with plenty of IF gain, the FM demodulator is finally going to be saturated by the noise and hence also the pilot is finally lost.

However, the dropping of the pilot tone level occurs at such low audio levels that it usable only for completely muting even the mono reception and the signal has been useless for stereo reception a long time ago.

Reply to
Paul Keinanen

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