Questioning Basics: FM Stereo Decoding Schemes

Trnsfmr -- On the early Fisher MPX 100 schematic, the schematic is in fact complete, but hardly indicated so very clearly: The "missing" cathode connection for V4-B is due to the fact that this cathode is internally electrically common with the cathode for V4-A, with both cathodes then appearing at pin 3 of the ECH84 tube.

I hope this helps!

Dave

Well, that being the case, I'm stumped as to how the early MPX100 works. I don't doubt that it did produce some kind of difference on the L/R outputs, I just cannot figure out how the 38khz is added to the composite to make the L-R and then that and its inverse added to the L+R to get a separate L and R out, all in 1/2 of an ECH84. I'll try again later.

There's an early MPX100 for sale on ebay, and...

This morning I realized that ebay could be a valuable source of information... Pictures. The Bogen PX60 multiplex adaptor has a Matrix Adjust pot on the back panel.
BogenMPX.jpg
I could not find a readable schematic anywhere. But...

I did find this most interesting picture of a HH Scott 335 MPX adapter chassis.
Scott335.jpg

I was able to find a schematic for it and it is indeed the chopper, with the double bridge rectifier circuit. I don't doubt that Scott was awarded a patent for this most innovative circuit. That might explain why Fisher sent out a bulletin saying they ceased production on the Chopper, which probably happened when the patent was awarded. I bet they then made a slightly different circuit that did the same thing, called a switcher, or sampler, to get around the patent. Common corporate stuff.

I can add the Bogen PX60 to the list of matrix decoders. There are probably others.

Wish I knew how to search the US patent office database, if there is such a database.
 
Regarding the MPX-100's operation:

1. 38 kHz energy is applied to G3 of the pentode section of V4 (section A), while the L-R component of the composite signal (i.e., as a suppressed carrier signal) is applied to G1. This stage operates as a "product" detector, so that a detected L-R signal appears at its plate.

2. This signal has its level adjusted appropriately, and then applied directly to the matrix network at R21, and also to the L-R plate follower unity gain inverter stage (V4-B) via R23.

3. The output of the L-R inverter stage is applied to the matrix network at R20 via C26.

4. The L+R signal is applied to the matrix at R17 and R18.

4. The L and R stereo outputs appear at the junctions of R17/R20, and R18/R21 respectively.

I hope this helps!

Dave
 
Interesting topic, albeit maybe a little too technical for most readers here ? About the (supposed) superiority of one demodulation process (matrixing) versus another (switching/time multiplex) both methods are capable of excellent performances (both sonically and technically) if correctly implemented and neither is inherently better or inferior to the other. Both have their virtues and issues. Also I would not follow the simplistic analog (matrix=good) vs digital (sampling=bad) debate which doesn't apply here as both methods are actually analog. Comparing time-multiplex switching used in stereo FM broadcasts with (modern) sampling techniques is a gross over-simplification and technically incorrect. (a.o there is no digitization of the signal in the mpx process and the switching signal is not a 2-level square wave but a sinusoidal waveform). Both methods were used for encoding at the transmitting side and leads to an identical mpx composite signal strictly in accordance with the FCC standards. (actually on a scope you can't tell the difference). Back in 1961 when the FCC adopted the G.E-Zenith multiplexing method most engineers were more familiar with matrixing techniques than time-switching multiplexing which was less know then, also it was easier to build simple matrix type decoders with less active elements (tubes) than a switching type decoder. If H.H Scott opted for the time-multiplex approach it was certainly not for economical or simplification reasons but to build the best possible decoder feasible with the technology available then, regardless of price. I've read that Daniel Von Recklinghausen (Scott's chief engineer and tuner expert) actually tested ALL the available methods before finally opting for the time switching multiplex circuit. The fact that SCOTT (335) mpx decoders are still highly regarded today and considered amongst the best sounding ones (but I know there are others) tend to prove they fully reached their goal. This doesn't mean that there are no good sounding matrix type decoders or that switching multiplex is the only (good) way. There are much more factors involved in the audio quality of a multiplex decoder than the basic demodulation process used, especially the care in the design of the output audio stages and how well the 19 and 38 KHz residues are filtered out, deemphasis accuracy, driving capabilities,etc... all in the (LF) audio domain after the L/R signals have been separated.

Back to the decoder used in your ST-97 (which is basically the same circuit used in their MX-99 stand-alone unit) I have no doubt it can have excellent sound and performances (especially after your upgrade and re-alignment work), it's quite an elaborate and original circuit (for a kit) but it has been already mentioned it is NOT a true matrix-type decoder but a clever combination of switching and matrixing (actually more switching than matrixing, with some matrix type reinsertion to improve performances).

Excerpt from the EICO MX-99 manual circuit description (p.4):

... V2A provides two outputs which differ in phase by 180° and which are alternately sampled by the ring modulator at 38kc rate. The two sampled outputs are added and amplified by V2B. Adding the two alternately sampled out of phase signals effectively produces an output which is the input signal multiplied by a 38kc switching function of zero average value and odd symetry...

Thus, if you like the sound of your ST-97 (and I think you actually do) maybe you should not consider sampling/switching as a bad evil. My main concern with the MX-99 (and ALL tube decoders with a user front panel separation control) is a noticeable warm-up drift which needs periodical re-adjustment of the control during operation to maintain good stereo separation. (which is a hit or miss empirical operation). I'm wondering if your parts upgrading/selection has cured this problem and if you have measured separation performances at switch on (unit cold) and after warm-up (without re-adjusting the control). Maybe the ST-97 having a bigger case is less prone to thermal issues and has better heat dissipation characteristics than the small enclosure MX-99 ?

But all this discussion is a little bit academic now as analog broadcasts are doomed to disappear soon and the biggest problem (as far as sound quality is concerned) is the poor quality and heavy un-artistic signal processing of today's modern FM broadcasts.

If you're interested reading more about the subject (and especially the matrix vs switching methods) I would suggest checking the excellent article titled: Fm Multiplex Stereo Detection Methods by Norman H.Crowhurst in the JAN 1962 issues of Electronics World (p.50) which can be downloaded here:

http://americanradiohistory.com/Archive-Electronics-World/60s/1962/Electronics-World-1962-01.pdf

But don't expect a definitive answer, the topic was already controversial 54 years ago...
 
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Regarding the MPX-100's operation:

1. 38 kHz energy is applied to G3 of the pentode section of V4 (section A), while the L-R component of the composite signal (i.e., as a suppressed carrier signal) is applied to G1. This stage operates as a "product" detector, so that a detected L-R signal appears at its plate.

2. This signal has its level adjusted appropriately, and then applied directly to the matrix network at R21, and also to the L-R plate follower unity gain inverter stage (V4-B) via R23.

3. The output of the L-R inverter stage is applied to the matrix network at R20 via C26.

4. The L+R signal is applied to the matrix at R17 and R18.

4. The L and R stereo outputs appear at the junctions of R17/R20, and R18/R21 respectively.

I hope this helps!

Dave

DC,
Great job on that. Dissecting Tube circuits is not my forte', and I don't think I ever would have figured it out.
 
Tubo, A lot in your posting there. I will just take various issues up one at a time.

You said "both methods are capable of excellent performances (both sonically and technically) if correctly implemented and neither is inherently better or inferior to the other."

I guess I would at the most basic level disagree with that, pretty much just from the idea that two completely different highly technical processes could end up with exactly the same analog result. Over and over again in audio, it is found that there are sonic differences, and people can hear those differences, and sometimes it's not subtle. Please, no offense, but to say that something is the same because it looks the same on a scope, or that there is no measurable difference is what I call "engineering arrogance". And I can talk about it that way, because I am an engineer and was taught that arrogance just like every other engineer. Hearing is tremendously more sophisticated than any measuring instrument, because the ear integrates and evaluates information from both the frequency and time domain with all kinds of emotion and meaning thrown in.

It's like saying "these two amps both have 0.2% harmonic distortion, so they must sound the same." Again, no offense, but that is engineering arrogance.

It's not just me who is amazed by the Eico tuner. Over the last few years, trying to figure this out, I have collected some quotes from postings on news groups. Here's some of them. Keep in mind that most of these are about stock unrestored units with the ceramic caps still in circuit. Most of these from AK.

“When I first got my ST-97, I was shocked how good it sounded. I let go all my SS tuners right after I got the EICO (those ones mentioned in my previous post). It even beat the McIntosh MR74 in stock form. “

“I've had tube tuners from almost all the major brands at the time. (Sherwood TOTL S2100II (FM same as S3000V). FM-100B, Scott 350B, EICO ST-97). The EICO is the best. “

“I bought mine for $90 cad. The old gentlemen did some restoring work on it. It sounds amazing. Better than most of the SS tuners that I had (Pioneer TX-9100, TX-9500II, TX-7800; Luxman T-117, T-530, Sony ST-5130). Much better than Sherwood S2100II (FM same as S3000V). Very close to Fisher FM-100B (ST-97 has better low end while FM-100B has edge in high end).

The only one bettered it in my setup was my fully upgraded McIntosh MR74.”

“I neglected to mention the part where this tuner blows away my Adcom GFT-1A, and I think that's a very respectable one, to say the least. I've owned it since new in 1985-86 and use it daily. The Eico really grabs those signals, very impressive for such a "cheap" tuner. Definately a sleeper.
smoking.gif


“One issue with the Eico are ceramic caps in the audio chain. Ceramic series capacitors are very bad news in an audio frequency application.”

“I got the 2 new 8.2mH coils today and installed them in the ST97...wow what
a nice sounding tuner..now I know what Jim Showker was talking
about.....nice bottom end, nice stereo separation...I'm not a huge fan of FM
radio , but this sound is amazing.... Thanks, I'm shocked how good it sounds...”

As I indicated earlier, the sampling bad, analog good argument, is not something that I agree with. However I will state unequivocably that sampling done at a 38khz rate is not going to be good.

As far as the Eico MPX circuit using a sampling techniques, I am somewhat mystified by those statements. Maybe that language was as much motivated by marketing as it was engineering, I don't know. They promoted their decoder in advertising by saying it provided the best of both techniques. Compromise appeals to a lot of people.

The same ring demodulator circuit is used in SSB communications equipment to demodulate a SSB signal and it is not called sampling.

It is essentially a balanced detector giving a smoother signal, in the same way that a full wave or bridge rectifier gives a smoother DC output.

An audible analog signal will result. In the ingenious balanced Eico circuit, the two out of phase 38 khz signals are nulled out when they go to the L-R amp, thus doing away with that problem with no filtering required. The audible L-R components are in phase and combine to go to the L-R amp. Notice that there is no smoothing capacitor as would be required for a sampling technique.

To me, this circuit demonstrates the same kind of brilliance as the Scott chopper circuit. I think they did apply for a patent.

I guess you could call it sampling, due to the very nature of detecting an AM modulated signal. It probably would have been better to use a higher frequency than 38khz.

If one insists on calling it sampling, this circuit, by double sampling two out of phase signals, with a balanced pair of two out of phase 38khz signals, and then the audible components of this sampling being combined in phase... this effectively doubles that sampling rate to 76 Khz, which, from that point of view, may be the key to the improved sonic qualities. In the same way that a full wave rectifier will give 120hz DC ripple from 60 hz AC.

I think all that I am saying here is pretty much the same thing as the paragraph you quoted from the Eico manual. Remember, the word "sampling" was a new electronics term in '61 and did not have the precise meaning it does today. One could say that a bridge rectifier "samples" the + and - components of the AC line and routes them to the appropriate connection of a power supply.

I think this circuit, that derives the L-R signal, is the key to the improved performance of the Eico decoder.

I'm certainly not the expert here. This whole quest for me has been to find out why the Eico MPX sounds so much better. I had to learn all about MPX decoding to even have this conversation.

Interesting what you say about the warm up affecting separation. Maybe I have skirted this issue entirely with my own preferences, by not running these units with the cover (cage) on. The separation just seems so solid and unaffected by the control anyway. Tube gear just runs so hot, that I practically always run it with no cabinet. I like the look of tubes, anyway.

I would suggest to you that you actually hear the Eico MPX decoder and then you might look at it differently. Of course that's not so easy to do.
 
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Great thread, but a reminder of the problem the average owner may have with assuring accurate alignment for top performance.
Also, where do the ubiquitous Dyna FM tuners fall here?
 
A few days ago, I emailed the head engineer at KWVA, our local college FM station, which to my ears, has the highest quality signal in our area...

Ali, the engineer, got back to me with info about their setup. They use an Orban 5500 processor, which is where the compression and limiting happens. It is a 5 band unit, with completely variable settings. This processor is digital, ie. it converts the input signal to digital, processes, and then converts back to analog at the output.

The 5500 samples its input at a 512khz rate and does all internal DSP processing at that rate. Orban has a patent on their "half cosine interpolation" level processing.

Their FM exciter, which is where the FM Stereo MPX signal is generated, is a Broadcast Electronics FM50, which is an analog exciter, ie. the multiplex signal is generated with analog techniques. BE advertises a 93db signal to noise ratio and harmonic and IM distortion so low as to be immeasurable.

They just got the Orban about a year ago, and that was when when I noticed a big improvement in their sound quality.

I'm sure the gear helps their sound quality, but I also think their choice to run fairly low compression, giving their signal a 6db or so less loudness level on the dial, really affects their quality also. This gives their signal the space for real dynamics.

So much for the digital vs. analog argument. How about both?;)
Orban.jpg FM50.jpg
 
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Interesting thread: I've never been able to obtain a good sound from my MX-99 adapter.

Much better are my two Scott's (LT-110 and 350) and my Fisher FM-200-B.

I have another one MX-99 and I'll try one day to replace only the minimum of components to compare with the other one.
 
Well, now you've really got me thinking. I was sampling many tuners that have the LC IF filters along with the diode matrix decoders, and as expected they sound really really good, very dynamic. So one would ask where in the modern tuner 1970s and more modern, does the dynamics get lost. I have a Stereotech 1200 receiver and am running it through some Infinity Intermezzo 2.6 speakers, Stereophile class A. And since the Stereotech has the good diode matrix tuner, it is definitely more dynamic than the Revox B760 which is on the fmtunerinfo.com creme de la creme list. However it is important to note that the Stereotech (McIntosh bastard child) has many ceramic filters in the IF section. So I do have a theory that the germanium diode matrix, since it turns on at 0.2 volts or so, would be more dynamic than a MPX chip which is silicon based. The silicon transistors turn on at about 0.6 volts. So perhaps the germanium diode matrix would be more dynamic since it the diodes turn on so close to zero. And perhaps the ceramic chip technology permits the nice wide separation that one could only hope for with the LC filters in the IF section. However I do think that the IF section, if run through ceramic filters do not have the imaging and spatial characteristics. I played violin in a semi pro orchestra for twenty years, and have an engineering degree also, and realize that the measurements are either measuring the wrong thing or are greatly inferior to a good set of ears!
 
It is interesting that you would like to see lower forward voltages on the diode bridge. My Harmon-Kardon Citation III-X uses a diode bridge with a parallel circuit of 10 Kohms and 600 pF in series with each diode. The purpose of this was to retain a voltage that would back-bias the diodes so they would conduct only at a higher voltage. If the diodes had zero forward voltage, even the slightest phase shift would cause the left and right to bleed into each other and the phase relation between the recovered pilot and the composite signal would have to be kept exact to avoid this. If there is a time when no diodes are conducting, the composite may occur anytime within that time without contaminating left with right and vice-versa. This particular demux has (and needs) no separation control because the dead time prevents any cross-contamination.
 
First, thanks to all for this discussion. I will throw in a subjective observation. I have an Eico Classic Series 2200. I have all the original documents from the kit if anyone wants them. It is relatively noisy, but the stereo imaging/soundstage on small ensemble music is just wonderful. I can't speak to how the circuit compares to the ST 97. My Eico was refurbished.
 
I do not have a TU-X711 schematic, but I have a TU-X701 schematic. It uses a PLL type FM demod and the Sanyo LA3450 stereo decoder.
For a stereo encoder i have on my list to do, using the same ckt as in the ST1000A, it is a simple design, seems to do the job sufficiently. I am to understand that having a clean low distortion 19Khz is adventageous. That is why i see some using a DDS generator for it and the 38Khz. With the DDS I believe that you can adjust the phases to be near perfect. More projects for me to work on.
the problem that I see with

is that it is inferior technology, as compared to a DSP based tuner.
More so now, since the FM band in my area is getting full. I even have a few stations transmitting on the same low power freq of 92.3M reserved for college type stations.
The test that I use is to having two adjacent carriers that are stronger signals than the one you want to receive or tune too. If you look at most tuner specs they do not even consider adjacent carriers suppression, they only provide alternate carrier suppression, which are much further away. Why no adjacent carrier suppression specs? because they are truly that bad.

I like what DSP does for reception, but the audio is not up to my requirements for sonic quality. I love DSP and SDR radios for DX work. 92.3 is not a college station frequency. It's in the commercial FM band. The NCE (non commercial-educational frequency band) is 88.1-91.9 Mhz on FM. adjacent carrier suppression is such a problem because of the Capture effect, where the stronger, closer signal overpowers the weaker, further away station.
 
FM band frequency assignments are controlled by the government agencies. In Canada, where I reside, it is the CRTC, akin to the FCC in the USA. So one country’s frequency assignments, usage, etc. are not necessarily the same as the others unless they are agreed to by means of a treaty or such mutual agreement.
 
RE: SCA. Subsidiary Communications Authorization. Was used from the beginning of FM for Background music for shops, businesses, and industry. And also for the control of broadcasting equipment where phone lines were not available. And also for backhaul of remote broadcasts, emergency use as a STL/TSL link (Studio to Transmitter, Transmitter to Studio link system) and also for paging use, and slow FSK data. Each FM station had two frequencies for SCA use. Also the last and most common use for SCA non public was using it for Radio Reading Services for the visually and print handicapped or challenged. Then the FCC followed Europe's lead on RDS (Radio Data System) which gave you the stations call letters/ the song playing and the artist, and emergency information.
 
I happened across this quote from Syd Smith from https://www.worldradiohistory.com/A...m-Tube-Valley/Vacuum-Tube-Valley-Issue-05.pdf in the history of the Marantz 10:


The multiplex section is something
that people don't really appreciate. I never
wrote an article on it. It's called "quasi-
vestigial sideband system," we used more
of the lower sideband than the upper side-
band by unbalancing the mix between the
two over the frequency range. The
phase/amplitude relationship of that is
very tricky, and we used a phase-linear fil-
ter to keep the separation high. I came up
with that. To set this up in production I
had to modify some of the available multi-
plex generators then. I had co "un -kink"
some of the distortion. Later at Sequerra
we designed our own.
 
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