Transistorized FM stereo decoder question

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neazoi

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Good question. It looks like D1,D2,D3,D4 form a bridge mixer phase detector to demodulate FM. T1 and T2 transistors amplify the FM signal to generatate an LO signal big enough to switch the diodes. L2/C2 form a Phase Shift network so that a sample of the FM signal is sent to the mixer diodes, but as the instantaneous frequency deviates +/-, a corresponding phase shift happens, which the diodes detect and send out as an audio signal.

Sound reasonable?
 
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    neazoi

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The purpose of the first two transistors apart from amplifiyng the signal is to double it's frequency. They are actually tuned to the 19KHz pilot and double this to 38KHz, to drive the diodes.

Are you sure about the L2/C2? What is this tuned circuit's resonance (I am figuring out chat coil should I plug there.)
 

I think L2/C2 are a 19KHz trap. The pilot tone is extracted and doubled to 38KHz by the transformer and diodes between T1 and T2 and the audio is re-inserted in the diode switch after L2/C2 have removed the pilot from it.

Brian.
 
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    neazoi

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I think L2/C2 are a 19KHz trap. The pilot tone is extracted and doubled to 38KHz by the transformer and diodes between T1 and T2 and the audio is re-inserted in the diode switch after L2/C2 have removed the pilot from it.

Brian.

So at the quad diode mixer, you mix 38KHz with the 19KHz coming out of the L2/C2 trap?
Where is the FM signal to be recovered comes in then?
 

The four diodes in a ring are not a mixer. To understand how it works you must first realize how the two channels of a stereo signal are encoded. The way they are transmitted is complicated but if I over simplify it to explain:

The left and right channels are both chopped at a fast rate then selected alternately. So the transmission (I'm ignoring the transmitter carrier offsets and differential channel signals) is basically a short burst of left channel then a short burst of right channel repeated continuously. This maintains compatibility with mono receivers which hear a combination of the two channels as though they were one. The chopping and selecting has to be done in the receiver at exactly the same speed and phase as it was in the transmitter so one channels doesn't 'leak' into the others time slot. The switching has to be done fast enough that it can't omit complete cycles at the highest audio frequency or cause beat notes as both would degrade the audio quality. The problem with FM transmissions is the higher frequency in the modulation is, the wider the overall bandwidth becomes. The requirement to send a fast enough switching tone and still use a reasonable bandwidth can't be met in the normal domestic VHF broadcast frequency allocations. So a trick is used - the pilot is sent as a low level 19KHz signal, just too high for most people to hear and too quiet to cause noticable interference to the audio. At the receiver, the 19KHz is picked out of the received audio by using a filter, then the frequency is doubled to 38KHz which is then used as the channel switching frequency.

In the circuit you show, the input is already converted from FM and is the recovered audio itself. It is split into two paths, one is the audio through a parallel tuned circuit (L2/C2) which removes the 19KHz pilot leaving only the audio, the other is the path through T1. The tuned circuit in the base of T1 is to accept the 19KHz and amplify it (remember it's at low level compared to the audio). The tuned circuit in T1's collector works just like a full wave rectifier in a power supply. The transformer is tuned to 19KHz and its secondary is center tapped so the output from the two diodes is predominantly twice the input frequency at 38KHz. L3/L4 is tuned to 38KHz and the baseband audio is fed to the center tap of the secondary. The top and bottom of L4 carry 38KHz, one in phase and the other inverted, these alternately turn on one or the other pair of diodes in the ring so they act as switches to route the audio down one path or the other. This gives the separated channels out. Left and right are recovered.

Now the bit where I over simplified it - Ignore this if you want to - The signals are actually transmitted as L+R and L-R alternately, T3 does a kind of "invert and add" operation to do the math needed to make L and R really come out properly.

The only other point to note is that on most mono FM receivers, the discriminator (FM demodulator) is normally followed by a de-emphasis stage. If you build this circuit you will have to extract the audio before the de-emphasis is carried out. The reasoning is this: In order to improve the signal to noise ratio, the higher audio frequencies are boosted relative to the lower ones at the transmitter, rather like turning the treble tone control up before broadcasting. This is called pre-emhasis, to re-level the response at the receiver, de-emphasis is applied which does the reverse, its like turning the treble down again. It improves the SNR because the noise in the system is independent of the audio and reducing the high frequencies at the receiver makes it less obtrusive. Boosting before sending and lowering again the receiver gives an overall flat audio response while still reducing the noise. This introduces a problem though, because the de-emphasis circuit in the receiver reduces higher audio frequencies, it also reduces the pilot level and makes it more difficult to extract it cleanly. The fix is simple, feed the circuit from the discriminator without de-emphasis and apply it to the left and right channels after they have been decoded instead. This is what R3/C3 and R2/C4 do in the circuit.

I hope that makes sense - I'm writing it after a long hard day and my brain is slipping into 'standby mode'

Brian.
 
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    neazoi

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Brian, I really appreciate your excellent explanation, it really answers all my questions. Now I understand why the article says that the input signal comes from the FM detector.

I am building this receiver **broken link removed** (schematic is on the last page).
Where should I take the output from and drive the stereo decoder? Would that be right after R25?

I really appreciate for the help
 

Across C23 pads. C23 IS the de-emphasis (very crude!) but removing it altogether might let IF reach the audio stages so I would leave something there, perhaps 470pF or so to work as an RF filter.

I would take the guy who drew that schematic out and shoot them! It has to be one of the worst drawn diagrams I've ever seen.

Brian.
 
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    neazoi

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This circuit has not got the ability to switch between mono or stereo. I think the mono signal would be much attenuated. Do you think I should include a bypass switch to bypass the stereo decoder, when mono operation is chosen?
Any thoughts will be helpfull.
 

The easiest way to force mono is to stop the 38KHz reaching the diode switches. I think shorting out the 33K bias resistor in the base of T1 should do the trick. The volume shouldn't change much.

Incidentally, unless you really want it, the whole EM87 "magic eye" can be left out of circuit, it is only a pilot level indicator and plays no part in the decoding process.

Brian.
 


T3 won't do any harm on it?

Also, the de-emphasis C23 in the main receiver circuit, should be changed back to it's original value for mono operation, or just leave the 470pF there as you suggested?
 
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