SQ Shadow Vector Soundfield Mapping

[Lynn Olson]

Now for a deeper dive into allpass phase shifters. What are these gizmos, what do they do to the sound, and why are they essential for SQ and QS encoding and decoding, but not used in EV4 or DPL-II?

An allpass phase shifter, or allpass filter, is an odd little device that shifts the phase around 180 degrees, 360 degrees, or more while retaining a completely flat response. Square waves and impulses look really weird after passing through these things, but oddly enough, it's barely audible at all, despite the horrific appearance on a scope screen. I've met a handful of audiophiles who claim to hear absolute phase, but it has to be kept in mind there is no standard for absolute phase, and there are plenty of records where the absolute phase is different from track to track, since they were recorded in different studios. In any event, the kind of steady phase rotation produced by an allpass phase shifter is remarkably difficult to hear, even on direct and immediate A/B switching. The physical device is a simple op-amp circuit, or done with discrete transistors. The allpass function can also be realized in DSP software with more precision than physical capacitors and resistors allow.

When the previous post referred to 0 and -90 degree phase shifters, that does not mean the 0 degree function is directly connected to the input. No, not at all. Both 0 and -90 degree outputs came from phase shifters, just with slightly different tuning. The pair are set up so there is a constant 90-degree difference between the two outputs, but compared to the original input, the phase spins all over the place. In fact, the original, non-phase-shifted input and the output of the phase shifters can never be combined anywhere in the decoder, or anywhere else in the audio chain, or there will be very deep nulls and peaks (called a "comb filter" in the world of audio). Compared to the input, the phase from the two shifters spins around and around, with more spins with more poles, but the phase relation between the two shifters stays at a constant 90 degrees though the audio band.

Why do decoders use these things? For QS decoders, it allows a smooth 360-degree pan without a phase jump of 180 degrees at Center Back. This phase jump happens in EV4 and DPL-II, so Center Back is usually avoided, or special techniques are used to encode the circular pan.

How does QS do this? At first glance, LB is made of +0.924L and -0.383R, and RB is made of -0.383L and +0.924R. In reality, each rear channel is shifted by 90 degrees, so LB is really 0.924L at +90 degrees and 0.383R at -90 degrees, and RB is made of 0.383L at +90 degrees and 0.924R at -90 degrees, so they are actually in phase with each other. The two front channels are at 0 degrees, but this is NOT the direct input, but the output from the 0-degree phase shifter. If the fronts don't pass through the 0-degree phase shifters, the phase relation between the front and rear channels will be completely random, and no image will be possible on the side walls.

SQ relies even more heavily on phase shifters, since the LB and RB channels are entirely made of the outputs of the 90 degree phase shifters, and as with QS, the front channels are derived from the 0 degree phase shifters, NOT the direct input. In both QS and SQ, when a signal is panned in a 360 degree circle, the pair of front speakers are in-phase with each other, the pair of rear speakers are in-phase with each other, and there is a 90 degree phase shift between the pairs of speakers on the side walls.

In both QS and SQ decoders, all of the audio that passes through the matrix, whether static or dynamic, passes through the sets of 0 and 90 degree phase shifters. If the matrix is fed a combination of direct, non-phase-shifted audio and phase-shifted audio, the result is massive cancellations in the audio band as well as random phase relations between the front and rear speakers. I've read that early versions of DPL-II actually used phase shifters on the rear channels only, which results in random phase in relation to the fronts, and prevents any sidewall images from forming.

The previous post describing the Shadow Vector mentioned 0,-90, 180, and +90 degree phase shifters. This is really simple. The actual phase shifters generate 0 and -90 degrees, and there are a pair of inverters that create the 180 and +90 degrees. Repeat for the other channel and you have a set of 8 signals the VCAs can use in the dynamic matrix.

P.S. Every SQ and QS recording has passed all of the audio through a set of phase shifters, with anywhere from 360 to 720 degrees of phase rotation across the audio band. So absolute phase may not too meaningful for these recordings ... which includes quite a few beautiful EMI classical recordings made in the Seventies.

 

[Lynn Olson]

Just listened to the second radio interview on New York Public Radio. That was a second interview I don't remember doing. This one was after the UK visit to the BBC Labs, EMI labs, and KEF Loudspeakers. The termination of the Shadow Vector project must have come within days of this interview ... I was kind of shocked to hear a firm price quote of $1400 and a rough parts count of 500 for the decoder, and mentioning in passing that there no chance of using the TATE decoder chipset. Charley must have made the decision right afterward ... he could be pretty impulsive.

For example, in the summer of 1979, he fired both the Marketing Director (Gene Still) and the engineer that designed their most successful and profitable product (Bob Sickler). By then, I was already gone, having resigned several months before, but I thought at the time that this was a bizarre and inexplicable act of self-decapitation, firing the two key people who had finally brought the company into profitability for the first time in its history, and leaving it with zero design or marketing talent. And that's what happened. There were no new products in the pipeline, the Berning computer-bias BA-150 tube amplifier was an abject failure, sales of the Space & Time Composer were slow, and a few years later, the Audionics name was sold to Jim Fosgate in Arizona. Maybe one or two people moved from Oregon to Arizona, I don't know.

But this is all too typical for small audio companies. The management is either a former car dealer that is 100% focused on marketing and sales, or a self-proclaimed genius who thinks they are the next Einstein. Or both. That's why I got out of audio ... too many of these people, and I didn't want it to rub off on me. It took a decade of decompression to even think about audio again. Bob Sickler, the brilliant designer of the CC-2 amplifier, which was a decade ahead of its time, never returned to audio ... he designed products for the medical-appliance sector in Oregon (which I suspect paid much better, and had more sane management than what you usually see in audio).

Setting all that drama aside (so glad that crap is in the distant past, and I now have an independent retirement income), the mention of a QS -> SQ converter strikes me as a exaggerated promise. Yes, you can do this by (1) an antiphase cross-feed, followed by (2) a set of 0 and -90 degree phase shifters, which then have a -90 degree cross-feed across the two channels. The first step converts QS to EV4, increasing LF/RF separation form 7.7 dB to 20 dB, and the second step rotates the Scheiber sphere 30 to 45 degrees so LF is then midway between LF and LB, and RF is midway between RF and RB. If you want to get fancy, there can be two knobs, the first controlling the antiphase crosstalk, which controls the frontal width, and the second knob, which controls the degree of rotation of the Scheiber sphere. The second knob moves the front channels down the sidewalls and starts to give some L/R separation across the rear channels. This is NOT the same as real QS decoding, since separation in the rear channels is only a few dB, and the front channels separation is maybe 12-20 dB at best. None of the imaging is really where it should be; a QS decoder is better, and there are no license fees today.

 

[Lynn Olson]

One aspect of hearing both radio interviews is the smoothing-out effect of memory. The really difficult emotional aspects of that period of time were simply removed from memory, so I didn't remember the radio interviews at all. I still don't. The clarity makes it sound like an in-studio interview, but I have no memory of going to New York City and doing an interview, never mind two of them! Yet I hear what is clearly my own voice, chatting away with great confidence about the soon-to-be-launched groundbreaking product. How very strange.

I do remember the failed US introduction in 1974, which was in upstate New York at a hifi dealer who was very well-connected with the East Coast audio press. Cliff Moulton, who designed the SV circuits, made an odd choice for the VCAs: FETs which were time-division switches, which let him program 1% to 100% gain responses from the VCAs. The drawback was getting rid of DC transients from the switched FETs (which consumed several months time), and the horrifying discovery, right before the planned public demo in front of the audio press, that the VCAs were very susceptible to RFI from nearby AM transmitters.

With no program material stimulating the VCAs, you could hear steady low-level crosstalk from 2 or 3 AM stations at once, and with actual SQ program material, the RFI bounced all over the place, and was different in each channel! Attempts to wrap the boards in aluminum foil, or wrapping the power cord and RCA cables, were completely futile. Cliff Moulton, Audionics chief engineer, was 3000 miles away in Oregon, and none of the phoned-in suggestions worked. The demo had to be cancelled, and the hifi dealer did a very gracious deflection to the assembled audio journalists and made a nice dinner party of it, with no mention whatever of the problem. I was pretty shaken, and not happy at all the RFI problem had not been detected (and corrected) earlier.

It's quite possible I did the first radio interview either directly before or after the debacle, but it has been expunged from memory. I'm not sure I could have been as calm and collected as I sounded in that first interview after the failed debut, so it must have happened beforehand.

Oddly enough, when I worked for the Spectrum Analyzer Group at Tektronix 10 years later, they had me take some in-house classes in RFI control, and the mistakes of the Shadow Vector prototype became very clear. The top of the aluminum box did not have a solid electrical seal with the rest of the box, which can make the whole top plate act as an antenna. The circuit boards, for reasons of easy prototyping, did not have ground planes, and were single-layer boards. That is not good practice for high-impedance FET switches operating at 50~100 kHz. Nor was there any RFI filtering on the RCA coax inputs, nor where the AC power cord came in to the box. But that was 10 years later (the wisdom of hindsight) and I was very appreciative that Tek sent me to the class, which was at the grad-school level and aimed at practicing engineers. I finally knew what had gone wrong at the New York demo, and how to make sure that didn't happen again.

The 1975 radio interview is even more of a puzzler. My memory was the Shadow Vector getting cancelled very soon after the return to Portland, Oregon, but as the interview reveals, several weeks had gone by after the trip to the UK, and I was confident the product was going into production quite soon, despite the advances of the TATE DES team in California. Weird.

As you might surmise, the prototype had development issues, and if the thing had gone into production, they would have had to be resolved (including the not-so-good idea of using FET time-switching at a time when it barely worked). Modern software is really so much better for signal processing.

 

[Lynn Olson]

A sentimental journey to the wild-n-crazy days of small audio companies ... this kind of thing was the norm, not the exception, for little companies in the audio biz. Just the way it is. Dolby Labs, THX, Bose, and Apple make the big moola, but the rest of us scramble along the best we can. So I've always got a soft spot for the small company setting up an exhibition booth at an audio show ... I've been there, and know what it's like.

This post is aimed at Chuck, Malcolm, and anyone else crazy enough to build SQ decoders from scratch. The rest of you can follow along.

There's a subtle asymmetry in SQ decoders, whether they are static, full logic, or dynamic. It's in all of them. When you look at the phase angle between pairs of speakers, it looks like this:

LF/RF = 0 degrees
LF/LB = 90 degrees
RF/RB = 90 degrees
LB/RB = 0 degrees
Diagonal Split = one split is in-phase, and the other is out of phase (I don't remember which). Note QS, EV4, and DPL-II decoders do not have diagonal splits; this is unique to the SQ system.

I'd have to have a block diagram of a standard static SQ decoder to puzzle out which split is in-phase, and which is out of phase. Does this difference between the two splits matter? On most rock recordings with no real ambience aside from an EMT plate, no, it doesn't. On a typical CBS classical recording made with a forest of accent mikes, it doesn't much matter either.

But ... on the superb EMI catalog of SQ classical recordings, many of which are "stealth" SQ and not labeled as such, yes, you CAN hear the asymmetry. Instead of a perfectly circular soundfield, it's kind of ovoid shaped, with a tilt in the direction of the inphase split. It takes a fair amount of listening, but once you hear it, it doesn't go away. (One of the hidden penalties of becoming a hardcore decoder geek.)

What to do? Well, what I did in the Shadow Vector days was reverse the phase of one of the rear channels (maybe it was RB, but I'm not sure). Here's what that does:

LF/RF = 0 degrees
LF/LB = 90 degrees
RF/RB = 90 degrees
LB/RB = 180 degrees
Diagonal Split = both splits are in-phase.

So you gain in symmetry and lose the back 30~45 degrees of the image ... no great loss, really, especially with classical or jazz recordings. Now if the OTHER rear channel is reversed in phase (not both at once, just one), then it's the same as above, but then both splits are out-of-phase as well.

I've done this comparison ... the standard SQ connection, the in-phase split connection, and the out-of-phase split connection. They sound like about what you'd expect: the standard connection is subtly asymmetric, but it's not there unless you're listening for it (and then it is), the in-phase connection suddenly snaps into focus and the SQ soundfield becomes circular and fully symmetrical (it actually sounds more like QS then), and the out-of-phase connection sounds bigger, more spread out, but not as cohesive as the in-phase version. On most recordings, I preferred the in-phase split, and didn't mind the small loss at the back of the soundstage. Part of this is because CB is difficult to localize in real life, and nearly impossible as a phantom image between two rear speakers. The out-of-phase split can actually sound kind of interesting on some recordings ... a bigger, more wispy soundstage, with less energy in the center of the room (that QS sound) and more energy on the periphery (more SQ-like).

All this takes are two separate phase inverter switches on the LB and RB outputs of the decoder. It's a little clunkier to do it at speaker level, but it gives a quick proof-of-concept test so you can hear it for yourself. Note: only ONE phase-inverter switch should be flipped, not both. You will definitely hear the difference, particularly on the more spacious recordings with a natural-sounding acoustic. Second note: do NOT do this for QS, EV4, or DPL-II decoders. These decoders have symmetric phase relationships between all 4 speakers, so don't mess with them.

 

[Lynn Olson]

I owe Malcolm and Chuck an apology. I was pestering both of them with a request for digital-only versions of their decoders, using S/PDIF interfaces. Well, that was wrong. S/PDIF only supports two channel PCM, up to 192/24, but still 2 channels. That's it. Which makes it OK for the input side, but definitely a no-go for the output side of the decoder.

Regrettably, the only multichannel digital inputs AVR's support are S/PDIF with Dolby Digital (yech) or DTS (OK, I guess) or HDMI (with exorbitant annual license fees). ADAT lives in a world of its own, and I haven't found any ADAT to HDMI converters. There seems to be a wall between the pro/semipro world and consumer electronics, no doubt there intentionally. The only ADAT interfaces I found were multichannel ADCs and DACs, which isn't what I'm looking for.

I really have a prejudice against a signal path that is digital source -> DSP -> DAC -> analog -> ADC -> DSP -> DAC -> analog. Much tidier to keep digital stuff in the digital domain, where DSP is easily applied, rather than flip-flop back and forth between digital and analog. And if the decoder is computer-based, computer sound cards are the worst-sounding (and measuring) DACs there are, thanks to high jitter levels and inherently noisy computer power supplies. Computers have USB outputs, so I guess a semi-pro USB multichannel DAC is an option, and that in turn could go into the analog inputs of the AVR.

I have mixed feelings about DTS 5.1. It's far better than 320 kbps Dolby Digital, which is about the same as 128 kbps stereo MP3 in quality, right at the bottom of the heap sonically. DTS 5.1 is roughly comparable to 320 kbps stereo AAC in quality, or maybe a little better, since the data rate is several times higher. Hard to say how high-rate DTS would compare to a cascade of DAC -> analog -> ADC -> DSP -> DAC -> analog conversions.

Just came across an EBU study of all the compression algorithms. I despise these things on general principles, but I must confess my collection of DTS CD's sound pretty much like an uncompressed stereo CD ... I can't really tell a difference, and I've got a pretty high-end system. Which seems to agree with the EBU study (refer to the conclusions at the end) that DTS 1.5 Mbps is pretty much transparent:

https://tech.ebu.ch/docs/tech/tech3324.pdf

 

[scharfis_brain]

Ha, my AVR (Yamaha Aventage RX-A2070) hasn't even an analogue Multichannel input.

Thus I needed to buy a rare Analogue Multichannel to 5.1 DTS SPDIF/TOSLink converter: The Creative DTS-610.

So I would suffer both issues: analogue AND compression losses.

Btw. DTS also supports center-back via Ex-Technology, as far as I remember

 

[Lynn Olson]

Looks like AVR manufacturers have gone to extraordinary lengths to make it difficult to get multichannel sources into an AVR. These are the only two sources I found for encoding DTS 5.1 ... and neither is a realtime encoder:

https://patrakov.blogspot.com/2011/09/i-wrote-dts-encoder.html

https://www.minnetonkaaudioshop.com...tPath=/Shops/MASIShop/Products/3001-00099-000

I hate to say it, but in the digital era, multichannel audio is actually worse for the consumer than the 1970's era of fighting analog formats of SQ, QS, and CD-4.

There's a maze of incompatible physical formats, AVRs with deliberately limited inputs (HDMI uber alles) and consumer-level encoding is very difficult to impossible. Multichannel SACDs trap the music in the player, forcing the user to use the analog outs, DVD-A's require a TV to even select the tracks and get music out of the player, and Blu-Rays are tightly bound into the HDMI ecosystem, which intentionally prices small-production-run audiophile manufacturers out of the market.

2-channel audiophile DACs now have USB inputs to access high-res streaming sources like TIDAL and Qobuz, but of course AVR's don't support USB inputs ... because the 2-channel audiophile markets and home theater markets don't talk to each other. Grrrr.

 

[scharfis_brain]

AC-3 @ 640kbps should be okay, too.

AC3Filter is able to encode it in realtime

 

[Lynn Olson]

Ugh. 640 kbps on a 30-year-old codec. Dismaying to think that's one of the very few ways to get a multichannel signal into an AVR. That's a long, long way from the professional 88.2/24 or 96/24 LPCM standard that's been around since the late Nineties (and available to the consumer on DVD-A, Blu-Ray, and Internet downloads).

But I guess you might be right. I couldn't find any realtime DTS 5.1 encoders. I also didn't find any ADAT -> HDMI transcoders, multichannel or otherwise. There's a real Chinese Wall between semi-pro and consumer protocols.

You'd think I2S (on-board internal protocol) to HDMI encoders would be a pretty off-the-shelf solution, but the info on these is thin on the ground. And of course the manufacturer, either Chuck at Involve or Malcom at Shadow Vector UK, would have to pay the annual fee to the HDMI Institute or whatever they call themselves.

 

[chucky3042]

Yep, its around $20 K per year plus the chips. We have just decided to scrap it from out Y4 surround sound system and supply an external decoder box. Problem to is they keep on "upgrading " it so our old Y4 is not compatible now to the 4K TV format. Its a safer proposition to buy an external extractor.

Basically the format is there to keep the little guys out of action. Frankly you minimise messy 5.1 connectors just by using 2 channel to 4/5 channel convertors like ours or the SV.

 

[Lynn Olson]

Hmm, thinking aloud here: Analog in the studio -> ADC -> Mixdown, Mastering, DSP, etc. -> CD or Internet download -> generic 2-channel DAC -> 2-channel analog -> Involve or SV UK RCA inputs -> ADC -> DSP -> DAC -> 6-channel analog -> AVR analog inputs -> ADC -> distance correction & bass management DSP -> DAC -> analog amplification -> speakers.

That's 6 conversions in and out of the digital domain, and only one of them on the original recording. I can hear the differences between old-school R2R DACs, generic sigma-delta DACs, the Burr-Brown hybrids, and the various versions of the ESS Sabre 9018, 9028, and 9038. They all sound different to me, none of them alike. The internal DACs of the Marantz are OK at best, nowhere close to the sound of the Monarchy M32 with the Burr-Brown PCM-63K converters, but acceptable. AVR DAC's, from what I can tell, are just above iPhone sonics, but not by much.

Setting the AVR into "pure" mode apparently bypasses the ADC -> DSP -> DAC conversion, so that's a possibility, and it gets rid of an unnecessary 2 stages of conversion. If the Involve or SV UK has a S/PDIF input (no licensing required), that gets rid of another 2 steps, while leaving the problematic outputs alone.

P.S. I agree that HDMI is a worst-of-all-worlds solutions from the perspective of the manufacturer. Obnoxious licensing fees intended to shut out the small-volume audiophile manufacturers (because they can), and constant and pointless changes to the video spec because that distinguishes this year's TV from last year's model.

I was in a bit of a sour mood today, thinking what the rollout of stereo would have been like if the same economic structure as today was happening back then. It would be 1962, and the only way you could play a stereo LP would be to get one of three incompatible "converters", each with a heavy license fee to the multinational corporation that came up with the system. No adapter, then all you ge to hear is a low-fi 78 in mono. No hi-fi for you, cheapskate. Same for radio: no adapter, all you get is mono AM. Oh, you want to hear hi-fi stereo? Then buy one of three incompatible adapters, again, with a logo and license fee for the manufacturer. Or you can buy a complex, expensive all-in-one "receiver" from one of several multinational corporations, guaranteed to be obsolete in two year's time, because the underlying "stereo" system keeps changing.

And we wonder why surround music isn't more popular. This is why. The decisions made back then (Westrex 45/45 stereo LP's, Zenith time-switching FM stereo) were made by industry-wide consortiums with no annual license fee. Nowadays, if Dolby, THX, DTS, and HDMI don't get their cut, it ain't happening.

 

[quadsearcher]

Not just manufactures, I believe it impacts the development of the pool of technical-minded people that form the future of technology. To extend the stereo adapter comparison, what if the very concept of shielded cables and connectors was restricted by licensing in about 1950. There would have been a tiny fraction of people exposed to important concepts in audio by experimentation and home building, some of the large group shut out would have been leaders in engineering that would have gone on to revolutionize advances in quality of sound reproduction that benefit everybody. Or perhaps led to other technological advances. But no, market restrictions would have prevented that from happening. It really appears to be an unfair market advantage, something like a monopoly that should be shut down for the good of humanity. Also the HD-SDI standard is much more robust, for example of an alternative.

Hmm, thinking aloud here: Analog in the studio -> ADC -> Mixdown, Mastering, DSP, etc. -> CD or Internet download -> generic 2-channel DAC -> 2-channel analog -> Involve or SV UK RCA inputs -> ADC -> DSP -> DAC -> 6-channel analog -> AVR analog inputs -> ADC -> distance correction & bass management DSP -> DAC -> analog amplification -> speakers.

That's 6 conversions in and out of the digital domain, and only one of them on the original recording. I can hear the differences between old-school R2R DACs, generic sigma-delta DACs, the Burr-Brown hybrids, and the various versions of the ESS Sabre 9018, 9028, and 9038. They all sound different to me, none of them alike. The internal DACs of the Marantz are OK at best, nowhere close to the sound of the Monarchy M32 with the Burr-Brown PCM-63K converters, but acceptable. AVR DAC's, from what I can tell, are just above iPhone sonics, but not by much.

Setting the AVR into "pure" mode apparently bypasses the ADC -> DSP -> DAC conversion, so that's a possibility, and it gets rid of an unnecessary 2 stages of conversion. If the Involve or SV UK has a S/PDIF input (no licensing required), that gets rid of another 2 steps, while leaving the problematic outputs alone.

P.S. I agree that HDMI is a worst-of-all-worlds solutions from the perspective of the manufacturer. Obnoxious licensing fees intended to shut out the small-volume audiophile manufacturers (because they can), and constant and pointless changes to the video spec because that distinguishes this year's TV from last year's model.

I was in a bit of a sour mood today, thinking what the rollout of stereo would have been like if the same economic structure as today was happening back then. It would be 1962, and the only way you could play a stereo LP would be to get one of three incompatible "converters", each with a heavy license fee to the multinational corporation that came up with the system. No adapter, then all you ge to hear is a low-fi 78 in mono. No hi-fi for you, cheapskate. Same for radio: no adapter, all you get is mono AM. Oh, you want to hear hi-fi stereo? Then buy one of three incompatible adapters, again, with a logo and license fee for the manufacturer. Or you can buy a complex, expensive all-in-one "receiver" from one of several multinational corporations, guaranteed to be obsolete in two year's time, because the underlying "stereo" system keeps changing.

And we wonder why surround music isn't more popular. This is why. The decisions made back then (Westrex 45/45 stereo LP's, Zenith time-switching FM stereo) were made by industry-wide consortiums with no annual license fee. Nowadays, if Dolby, THX, DTS, and HDMI don't get their cut, it ain't happening.

 

[Lynn Olson]

Chuck is right about the minimum-pain solution to HDMI: an external box, with a simple circuit board, that carries the HDMI chip-of-the-month. I presume the digital interface is I2S or something like it ... in the audiophile sphere, there is a quasi-standard for I2S transmission that uses an RJ-45 jack for the short run between the two devices. Power can be supplied through the RJ-45 jack or an external wall-wart supply.

 

[chucky3042]

Hi Lynn

I really like and respect the work you have done and its a real challenge to unravel SQ (Our Dave the ***** has not been the same mentally for years!) when we did the SQ version of the SM , from the little I have learnt we share many common ideas, just you got there first (I am only a pup at 61 years old). Involve heavily relies concepts of psychoacoustics like SV and uses multiple bands with various attack and decay times etc. My issue with SQ is the heavy bias of frontal left / right information making the detail of the rear information hidden within a few dB.

Re the multiple licensing of surround recording formats vs Stereo. This is the reason we have a license fee of ZERO for recordings made in INVOLVE format as we want it to become the default (as we claim there are no losers) see the recent Suzanne Cianni recording. We are interested in the customer end of the sale.



The reviewer says it was recorded in QS actually it was recorded in INVOLVE (we analyzed it).

 

[Lynn Olson]

I'm puzzled by this line: "My issue with SQ is the heavy bias of frontal left / right information making the detail of the rear information hidden within a few dB."

If the Involve decoder doesn't use pairs of 0 and 90-degree phase shifters, no L/R separation in the two rear channels can be extracted from (ideal) SQ at all. The standard 4 -> 2 CBS encoders had some phase drift in the sidewall locations, which could translate into some L/R separation for a QS/EV4 style of decoder (which don't rely on pairs of 0 and 90-degree phase shifters to extract F/R information). An ideal SQ encoder, though, will give zero L/R separation in the rear channels for a QS/EV4 style of decoder.

On the other hand, if the decoder is an actual SQ decoder with the mandatory pairs of 0 and 90-degree phase shifters, and uses full-logic gain-riding, TATE cancellation methods, or a Shadow Vector dynamic matrix, there's 25 to 35 dB of separation in the rear channels ... from each other, from the adjacent front channel, or from the opposing front channel. I've measured it myself, it's not an illusion. Separation in the SQ system is mostly limited by the precision of the phase shifters; with each degree of departure from 90 degrees eating up several dB. If the phase shifters are accurate to 1 degree, 40 dB is available in all the cardinal points.

The Shadow Vector retains this separation as the signal is swept across the intermediate locations; I don't know if the TATE does this or not, since I can't unravel the math in the TATE patent. I think it might not, but I don't really know.

I think the shared CB localization in both QS and SQ can give the illusion the two systems are partially compatible. Center Back is actually the only place in the back quadrant of the Scheiber sphere where the two systems overlap. From the point of view of an ideal QS or EV4 decoder, the SQ Left Back and Right Back encodings are in never-never land, with 90-degree phase shifts between the channels. A QS or EV4 decoder, whether dynamic or not, interprets either condition as a 4-speaker sum, with no localization at all, and an odd phase relationship between the 4 channels.

The best description of SQ would be a "phase matrix", and in a deeper sense than QS or EV4. In QS or EV4, the Z axis is represented by the proportion of in-phase to out-of-phase content. The 90-degree phase shifters on the encode end of QS let the recording engineer make an overhead pan from front to back with no sudden jumps in phase. (EV4 does not use 90-degree phase shifters for either encode or decode.)

SQ uses phases all the way from -90 degrees to 180 degrees to +90 degrees to accomplish a pan from LB -> CB -> RB. Remarkably, as this pan happens, there is NO change in L/R ratios; both channels stay exactly equal throughout the pan. A pan across one of the sidewalls, from back to front, gradually reduces the level of the opposing channel to zero (the phase angle no longer has meaning when only one channel is present). A pan across the front is simply a conventional L/R pan in stereo, with the phase angle between the two channels now at 0 degrees.

I agree, this is a screwy matrix, when you look at all the crazy things happening in the sidewalls and rear. That's why Shadow Vector is several times as complex as a Vario-Matrix, but ultimately similar in operation. However, Vario-Matrix could get away with opto-couplers doing the blend functions, but this is too inaccurate for Shadow Vector. Precision VCA's are required to get the phase angles where they should be. I wasn't thrilled with Cliff Moulton designing switching FETs to do the VCA function, but he wanted reasonable distortion combined with precision and speed, which can't be done with opto-couplers. No problem in the digital era, so long as 32-bit precision, combined with independent dither for each VCA, is employed.

But ... if the Involve decoder doesn't have the required set of four allpass phase shifters, with a precision of 3 degrees or better across the audio band, the SQ decoding will be an artifact of encoder errors, which I grant are abundant in the standard CBS encoder. The standard 4 -> 2 CBS encoder isn't "wrong" per se, it just mis-encodes the sidewall signals. The corners are exact, and a pan across the rear is accurate. The "forward-looking" SQ encoder (widely used by EMI) was quite a bit more accurate, the Ghent microphone pretty much spot-on, and the "Position Encoder" right on the money, although it couldn't do a smooth pan, just click-stops at certain locations.

I'll be honest; although the control signals in a Shadow Vector can be skewed to offset the standard encoder errors, I never did this. First, too damn complex when you do this in an all-analog environment, and also, the decoding was accurate enough so the skew was audible for what it was, simply a mis-location of the signal as it traversed the sidewalls. In that sense, it was a good tool for the studio, since you could easily hear when the decoder was mis-locating things a little bit. If the encoding was 100% accurate, the signal stayed pinned to the sidewall, and if the decoder was off a bit, the signal drifted into the room (a little) as it was panned from front to back. I think Malcolm has probably noticed this with his prototype; you can hear the encoder errors as slight mislocations during pans.

I also admit I never built a multi-system SQ/QS version of the Shadow Vector. If it had gone to production, absolutely, QS would have been in there. In QS mode, the -90/+90 sensing axis is not used, the signals going into the VCA's are simpler, and only a pair of VCA's are needed for each decoded channel. The extra precision of VCA's, combined with higher speed, would give a more precise and stable image than the Vario-Matrix.

 

[chucky3042]

The involve decoder uses 4 banks of 5 stage op amp based 90 degree shifters. I am reporting the sensitivity of the SQ matrix to level/ phase errors in the cartridge compared to QS

 

[Lynn Olson]

Ah! No problem with the phase shifters, then. Sounds like excellent design (10 poles per bank!), and not cheap if done in the analog domain (those damn 1% precision caps). Very impressed!

We found that axis correction for the phono cartridge was highly desirable. This was a simple pair of in-phase and anti-phase crosstalk from channel to channel, with separate adjustment for L and R incoming channels. This raised the separation from 22 dB or so to about 35 dB across the midband. There was drift above 5 kHz, thanks to cantilever rotation at high frequencies, and we contemplated a secondary adjustment at 10 kHz.

As for L/R balance affecting the rear channels, that might be a difference between the TATE and SV topology. TATE seems highly optimized for the corners, while the SV has uniform separation at all intermediate points. The result is that L/R imbalance results in a bit of crosstalk appearing in one of the front channels ... in effect, a mislocation of the image. Phono cartridges are rarely balanced any better than 1~2 dB, so that's another annoying adjustment that needs either a test record or a mono recording to adjust the setting.

A particularly evil thing that can happen in a phono preamp are phase differences in the midband as a result of small differences in the L and R RIAA equalizers in the preamp. The manufacturers usually check amplitude differences against a lab reference, but they almost never do a phase check between L and R channels. At Audionics, since we made our own preamps, this became a production test item; less than 1 to 2 degrees of phase spread across the midband. We found plenty of competitors that had 5 degrees or more of phase spread ... and that didn't show on their amplitude-based measurements of RIAA error. I suspect this is still a common problem in phono preamps, and can seriously degrade quadraphonic decoding, whether SQ o QS.

The nasty part are the 500 Hz and 2.2 kHz breakpoints in the RIAA curve; instead of a constant 5-degree error across the band, there's a frequency dependence, so the image wanders all over the place. All it takes are a mismatched pair of capacitors in the RIAA circuits, deep within the preamp. And feedback doesn't fix this, since the RIAA is often part of the feedback network itself. Hand-matching pairs of RIAA capacitors is both annoying and expensive in production, but we found no other alternative that gave us the phase-matching we considered essential to good image quality.

As a result of Shadow Vector (we only used 6-pole phase shifters), phase matching kind of became a religion at Audionics. The preamps had RIAA equalizers matched within 1 degree (in production), and our speakers were phase-matched against each within 3 degrees (in production). Even in stereo, this had a very positive effect on image quality.

 

[chucky3042]

So true Lynn
I did the Involve / QS flavor concept in the SM but I left our Dave the ***** do the horrid QS embodiment- so I am a bit shabby on its details- I might ask Dave to exchange thoughts with you next week on his findings. I do remember though that the basic core logic decoder section sounded like a bag of crap before we added the involve pre processing to the stearing! As I said he has not been a well puppy since.

 

[Lynn Olson]

Feel free to exchange emails, no problem.

At Audionics, we really went through the mill on this. It helped we imported Redford into the USA at the time, and were also embarked on a new preamp and power amp design, and of course, I later went on to speaker design. The rigor of the SV program came in handy for speaker design ... I was always aware of the acoustic phase relations between the drivers, and the way that would affect polar patterns in the vertical plane, as well as direct perception on the part of the listener.

The three trim adjustments: +/- crosstalk into the L channel, +/- crosstalk into the R channel, and L/R balance, are pretty much essential for adjusting out all the manufacturing variations of phono cartridges. Those stupid things are all over the place, and it's not consistent, either. They just can't build them accurately, no matter if it's Shure, Ortofon, an exotic high-end manufacturers, they are all over the place. They just have to be hand-adjusted at the decoder or preamp level.

Phono preamps are generally pretty badly designed. Not enough slew rate (I consider 15V/usec a minimum), poor layout that contributes to unwanted crosstalk, and poor phase-match between channels around the RIAA breakpoint frequencies. These problems are very, very common.

 

[Sonik Wiz]

Yep, its around $20 K per year plus the chips. We have just decided to scrap it from out Y4 surround sound system and supply an external decoder box. Problem to is they keep on "upgrading " it so our old Y4 is not compatible now to the 4K TV format. Its a safer proposition to buy an external extractor.

Basically the format is there to keep the little guys out of action. Frankly you minimise messy 5.1 connectors just by using 2 channel to 4/5 channel convertors like ours or the SV.

When you say "external decoder box" do you actually mean encoder box? As in 5.1 analog to encoded multi-channel HDMI? It doesn't apply to me but I have looked extensivly for a 5.1>HDMI box to do this. All I can find is 2 ch in, HDMI out. Can you tell me more?