Fixing a very tiny buzz

Every now and then I put my ear up to the Acoustats to see if a tiny buzz has been created by recent changes.  Sometimes I think I hear a sound but most often it's the air conditioning compressor (either mine or my neighbor's which is just outside the living room window), the refrigerator, or something else.  I pride myself nowadays on having a very quiet system.  Despite it's complexity.  This is partly because of design, and partly hard work like I just did today and yesterday. 

It wasn't always that way.  For 30 years I just took a small amount of hum and noise for granted.  Most often I could hear a quiet hum distinctly from the listening position with nothing playing.  Then around 2000--my 31st year of audio foolery--I finally started taking hum and noise seriously, and I strangely found that the biggest source of hum in my system then was not my MC225 tube amplifier, which I had long suspected of needing a refurb, but my pristine looking solid state 4 way analog crossover, the highly esteemed Pioneer "Series Twenty" D23.   Apparently because of a failing power supply, it was injecting hum directly into the audio signal.  It was not, that time, a ground loop.   I never fixed the D23, instead I tried many other approaches to doing speaker crossovers, and ended up using DSP around 2006 and ever since.  No matter how much DSP you do, and no matter how many DSP processors, there should be no hum and noise added.  Though you might get ground loops.

A few days ago I was thinking this time I was hearing a very tiny buzz, just barely audible.  It didn't sound like the air conditioning compressor or other noises happening right then, and it was clearly at it's loudest right in front of the Acoustats.  It wasn't easy to be sure, for awhile I thought it was just my hair rubbing against the Acoustat "socks".  I went back and forth between the speakers hearing something similar in both.  I was thinking to myself this could be the Acoustat power supply failing.  So, I shut off the Hafler 9300 amplifier.  It was pretty clear this made the tiny buzz go away.

I then though of doing something I hadn't done in awhile.  I hooked my Fluke 8060 A DVM to the Acoustat terminal with nothing playing.  For a lot of reasons, the Fluke is not necessarily an optimal way to measure audio signals.  But it's very convenient.

I was shocked to see 2.23 mV on the meter, way more than I would have expected.  

So then I put shorting plugs into the Hafler 9300, and measured 0.00 mV.  The Hafler not only was not at fault, it is one quiet amplifier.

So then I tried muting the Emotiva Stealth DC Dac driving the Hafler.  The buzz measured barely different.  While it was still muted, I pulled out the XLR connector for the AES/EBU digital signal.  And then again the noise measured 0.00 mV.

Clearly this noise is a ground loop from the source of the digital signal to the Emotiva.  I was immediately thinking I knew the Emotiva wasn't perfect wrt AES input.  I was thinking AES should not have a hum and noise problem because it's balanced, but the Emotiva implementation isn't very good.  I'm not sure if it uses a transformer.  I remember maybe reading a review which complained about something like that.  (But the Emotiva's AES transformer or lack of transformer is not actually at fault I concluded later, as I will explain.)

I started thinking I should buy a better DAC.  (And perhaps I should.)   But meanwhile I could switch to one of the DACs I have in storage, including an Audio GD Dac 17, and an available Denon DVD 9000.  As far as the Audio GD, I need to do some retesting on it, after a wrongly blamed it for causing the Krell FPB 300 to shut down frequently.  I want to be sure I know how good or bad the Audio GD is before putting back online.  Meanwhile if I used the Denon or most other DACs I have in storage I'd need to convert the AES signal to Coax.  I have several converters for that, but do I know if THEY are isolated or will just propagate the ground loop via the shield of the Coax?  With any of these converters, or even just the Emotiva itself, I might be tempted to use Toslink, which would certainly eliminate the ground loop problem.  So, if I'm going to use Toslink with some other DAC, why don't I just use Toslink with the Emotiva DAC that's already set up?

I tried Toslink and indeed it reduced the noise down to 0.00 mV with the Emotiva muted.

But I don't like Toslink.  I think it's a weak connection very much more subject to jitter than coax or AES.  So after reading online about ground loop issues with AES connections, I decided to do something radical that several people suggested, except perhaps not the way they would do it.  I took the 3 foot AES cable I had been using, and cut out the shield for about 1/2 at the end using wire strippers and cutters.  Along with the shield, there was a drain wire I cut too.  I patched it up with lots and lots of white electrical tape.

Sure enough, this also reduced the noise down to 0.00 mV with the Emotiva muted.  And it worked just fine apparently playing music.

Ground loop fixed!  It rarely gets this good, noise going down from 2.3mV (horrible) to 0.00mV (perfect, so good it's unreal) with one little change costing nothing.

Then with nothing playing, and in fact the Tact preamp itself was muted, I unmuted the Emotiva.  I was shocked to see the noise rise up to around 1 mV (sometimes as high as 1.5mV).  It was still silent at the speakers, but it almost seemed like 1/2 of the measured noise had come back.

Perhaps this was because I hadn't cut the #1 wire also, I wondered.  (Actually, the shield and drain wire carry the #1 pin wire, so in fact I had cut those as well, though possibly not a connection between the Emotiva chassis and the shell of the XLR connector.)

I tried the Toslink unmuted and sure enough, it had slightly more of this noise voltage, over 1.6mV.  That a Toslink connection also had this noise, with no XLR connected to the Emotiva, proves that the noise is not being "introduced" by my having cut the shield on the cable.  The noise appears to be related to the digital signal itself interacting with the Emotiva.

I now figured this was probably very high frequency noise generated by the DAC, perhaps caused by the ASRC, even for the supposedly muted signal.  Perhaps the Toslink connection had even more because it added jitter on top of jitter.

I was still uncomfortable about this new measured noise the next day and I decided to do some more tests.  I bypassed the ASRC containing miniDSP units, and the Crystal 8620 containing Behringers (I'm not sure if they enable ASRC or not) and plugged the DAC straight into the muted (and with no signal playing on anything either) Tact 2.0 RCS preamp, which uses fully synchronous digital receiving and transmission.  The spurious noise was about the same.  So it was not being "caused" by the ASRC's.

But I still really wanted to be sure this was not hum or buzz or any audible noise like that, even if it might not be audible simply because it was much lower than before (and it was barely audible then).

So I got out my Meguro Noise Meter, which has an A Weighting filter enabled.

I ran these tests with the shield-broken XLR cable.  With the Emotiva muted, I measured 0.076mV at the output of the Hafler.  With the Emotiva unmuted, I measured 0.13mV on the Meguro.

So the A weighted noise does not show the dramatic increase to 1.5mV shown in the Fluke's semi-wideband measurement.  This proves in fact the noise remaining, after fixing the ground loop issue, is essentially inaudible.  It is likely to be supersonic noise.

I can see some strange noise on the spectrum display of the Behringer DEQ that does midrange EQ after a miniDSP does the crossover.  Even when the Tact is muted, there is noise around -140dB or so shown on the Behringer display.  THAT may be the effect of dither and/or the ASRC in the miniDSP's.  But the ultrasonic noise we are measuring is much higher than -140dB, compared to the full output of the amplifier (around 34 volts) it is -106dB down.

So it's probably mostly above 20kHz, the cutoff on the Behringer spectrum display.

Not much to worry about.  The big thing this time was the ground loop which was easily fixed.  I grabbed another XLR cable, without the broken shield, and measured noise at 1.8mV on the Meguro.

I got the same results with Toslink and AES disconnected, so once again it's not my broken shield, though for that I repeated the Toslink only connection with the Fluke, and consistently measured just over 1.6mV, about the same as my Coax connection.

BUT, wait...I tried having no input connected to the Emotiva at all, and still measured the same everything...

Apparently this is just the wideband noise of the Emotiva DAC when it is operating, and it is slightly quieter in audible noise when it is muted, but way quieter in ultrasonic noise.  Here I should add that I have the Emotiva gain control set to +2.5dB.  The Gain of the Hafler 9300 is 29.  So the noise appears to be about 0.0016V / (29 * 1.4), which would only be 94dB below 2V.  This seems rather high, though I've never paid much attention to the "Unweighted" noise specs of DAC's if I've ever seen them at all.  Virtually all of the specifications we read are "A weighted," which makes most everything sound better than it is.  The weighted measurement on the Meguro is just over 20dB better, which would be a bit above 114dB, which is a bit below spec.

It's strange, very strange to me, that the DAC has way more wideband noise than my amplifier.  Way more audible frequency noise too, though so low in level it's hardly audible in practice.  I would have never expected this.  I wonder if this DAC is failing.  I already had the Emotiva DAC for one living room amplifier fail (so I can't run ABX amplifier tests right now, I have to disconnect input cables to switch amplifiers).  I'm going to have to look at other DACs.  But the audible portion of this noise problem appears miniscule, so it's not a high priority as was fixing the ground loop.  Back when I didn't care about hum and noise as much, I had the opinion that a little bit of hum and noise might actually make things sound better.  That's still a possibility I haven't fairly tested.  But now I presume that noise should be reduced as much as easily possible (without doing something that might increase distortion, etc).

It occurs to me that whenever you have a DAC with an AES connection, and outputs to coax audio, it is certain to propagate a ground loop.  The ground of the coax output is going to be referenced to the shield ground of the incoming AES.  It therefore makes sense to me that Pin 1 should be completely disconnected on AES devices having coaxial output.  AND the XLR shell should be isolated as well.

Or, maybe it's just that the Emotiva isn't grounded to it's AC input.  Perhaps gear with AES input MUST BE grounded, so that there is a separate chassis ground to which the AES shield is ultimately referenced, which is barely in contact with the active circuitry except perhaps through a small capacitor and/or resistor.  

The other possible solution is to have a cable (like mine now) with the shield and ground connections completely broken.  Rather than crudely strip off the shielding as I did, it would be better to make the cable this way from the beginning.  But so far as I can see, nobody sells a shield-broken AES cable, only a pin 1 lifter (which might not work if the shells are connected).  The shield should be broken on the male XLR pin side (outgoing signal).

Meanwhile, the importance of the tiny buzz I fixed, which could be faintly heard with ear right up to the Acoustat socks, is seriously questioned by the following line of reasoning.  This buzz problem that I just noticed and fixed must always have existed to a greater or lesser degree when I used an AES connection to the Emotiva to a single ended amplifier connection.  That means when I was comparing the Krell FPB 300 and the Hafler 9300 and in blind level matched ABX tests, and found no audible differences in music, I was comparing an amp which had this problem (the 9300) and an amp which didn't, because of its balanced input connections (the Krell).  If little buzzes like this make a difference at all, at least THAT should have been audible in the blind testing.  But it wasn't.

So, this would seem to show that faintly audible noises that you can just barely hear with an ear up against the speaker, are not important in ordinary listening.  The buzz measured approximately 2mV at the speaker terminals.  The rated output of the amplifier is about 34V.  Thats about 85dB down from peak level, or about 55dB down from soft passages.

What does that say about noises that are -110dB or -140dB ?  Perhaps not even worth thinking about ever.

Bit Perfect And Not

 SPDIF digital signals are almost universally decoded nowadays by a certain class of digital input receiver known as Asynchronous Rate Converter (ASRC).  These are most often integrated circuit chips, such as the Crystal Semiconductor CS 8420 (that particular chip is widely used but has a well known bug which is worked-around in higher quality equipment).

These make sense inside DACs as the best-possible way of handling jitter (source jitter, and carrier jitter) from the digital source clock and SPDIF itself.

Assume you have a DAC with a near perfect clock, but what you are receiving is a SPDIF input signal that contains jitter.  Even if the digital source had an absolutely perfect clock, the SPDIF signal from it will contain what I am calling carrier jitter because the actual clock transitions, embedded in the signal, are affected by the signal itself.  The ever varying digital signal itself will microscopically shift the "zero crossing" point enough to affect the determination of the precise clock--where the transitions occur--by enough to produce just below 200 ps of jitter.  This is the price paid for not having a separate signal for the clock itself--which would not be contaminated by the signal.  But note that all serial interfaces in the computer realm have this problem, and parallel interfaces which did not have virtually disappeared...because it's much cheaper to make serial interfaces that are good enough.

If all you have is 200 ps jitter, that is so benign you might as well leave it be.  But source clocks themselves are never perfect, so there will be additional jitter from that, and any mismatch in the speed of the source and receiver clocks will cause sample time units to be gained or lost--and that is not good.

Much has been made of the jitter issue in the subjectivist audiophile press, despite lack of evidence that jitter performance in decent equipment is audible in double blind testing, and much reason (including a major AES investigative report) to believe it would not be.  Often detailed graphs of the distortion sidebands down to -160dB resulting from jitter are shown, and people obsess over jitter sideband peaks which sometimes reach -110dB.  Meanwhile sometimes the same people may brush aside THD+N as high as -50dB (lovers of SET's for example) or -10dB aliases just above 20kHz (lovers of NOS for example) that can and do intermodulate downwards causing massive modulated noise.  If the noise process you are concerned about only causes noise peaks at -110dB, you should not be much worried about it.  Strangely, one of the best sounding DACs to me, the Denon DVD-9000 (which uses dual differential Burr Brown 1704's with an "AL24" digital filter including HDCD digital operators) has a lot of hashy looking sidebands above 10kHz starting just below -110dB and mostly just below -120dB.  Most likely those neither contribute to nor detract from the good sound.  And even those are likely not caused by the synchronous (I believe) digital interface but by the BB 1704's (the best R2R chip ever made, but not as low noise as the best Sigma Delta chips not too long after) and the fancy Denon-proprietary digital filter.  Does R2R have a fundamental advantage?  Well first it should be said the 1704's were quite good and better than the mainstream sigma delta chips for some time in ordinary SINAD measurements.  But secondly, the exact dynamic performance benefits, if any, of R2R chips may need a different kinds of analysis than frequency spectrum analysis to be apparent.  I've drawn a blank on that myself and all the converters I use on a daily basis are very low noise and distortion Sigma Delta dacs--which do sound good to me.  But I'm keeping the DVD-9000's and maybe more just for future tests....and currently for HDCD decoding as well  There are two reasons I'm not rolling DACs anymore.  I've been very happy with the Emotive Stealth DC-1 sonically--very pure sounding--and convenient size, price, and adjustability.  My system needed 3 identical DACs until last year.  Since adding the miniDSP's which convert everything to 48kHz or 96kHz for the supertweeters, the time delay will now stay fixed at different INPUT sampling rates, so I can more easily roll DACs again.  But I don't think it's as rewarding as speaker testing and adjusting.).

The landmark study published in the AES concluded that jitter would have to exceed 10,000 ps to be audible.

Anyway, jitter paranoia has guided design of digital interface receiver chips since the early 1990's.  And it was concluded around then that the best way to handle incoming jitter was to interpolate between the incoming digital values as they come in.  In this way a precise clock at the receiver gets values from the digital interface, but they are not the original numbers from the source, they are numbers interpolated from the input but at the new clock instants.  It's like you have a little computer examining the input values, and educated-guessing what the values would be at the new clock instants.  A fringe benefit of this approach is that it could intrinsically change from one sampling rate to another.  You could extract either a higher sampling rate or a lower one, just by asking the little computer for the guessed values more or less frequently.  This kind of digital input receiver is therefore known as an ASRC.

This approach provided lower distortion than the approach of using a sloppy Phase Locked Loop (PLL) like those used previously to "lock on" to the clock of the incoming digital signal with a small buffer for the digital values.  In that approach, the clock of the DAC itself would be made to approximate the incoming clock, but somewhat smoothed (to remove as much jitter as reasonably possible, including carrier jitter) but follows the clock embedded in the signal close enough so that buffer overrun or underrun never occurs.  So you have to speed up and slow down the clock of the DAC itself, which is hard to do without adding more distortion.

I was long skeptical of the benefit of ASRC until I measured it myself with my Emotive Stealth DC-1 DACS, which allows you to select either the PLL mode (called "Synchronous") and the ASRC mode (called SRC).  The default is SRC and when you select Synchronous, distortion rises from 0.0003% to 0.0004%.  I wouldn't lose any sleep over this, but it proved to me the effect is real, and that ASRC's are pretty damned good.

But this simplistic view does not account for the possibility that not all digital audio devices are either sources or DACs.  What about digital volume controls, EQ processors, Crossovers, Dynamic Filters, Displays, Limiters, and Storage Devices?

When any ASRC is used for the digital inputs of these devices, the values they are starting from on are not the original values.  The are Bit Perfect Not.

If you had a long chain of such devices, all set to "flat" or no change, the digital values passed through the system would be changed by each one cumulatively.  So each one may add only 0.0003% distortion+noise to the signal, but it keeps on adding up.

And if you are storing a digital signal, there is simply nothing better to do than store the original values in it.  Anything else is second (or third, etc) best.  The original values are the best values, having zero noise+distortion added to them, and no process which transforms those values can achieve or beat that.  Meanwhile we don't care if the digital storage process takes slightly more or less time on the order of nanoseconds or more.  It can wait, at zero cost in performance, for each value to come in, whenever it comes in.

My experiments indicate that ASRC's may have other even more insidious problems.  There is a potential for overload from inter-sample-overs.  The best-guessed new values between digital sample values can actually be above 0dB in some instances, especially with extremely highly compressed recordings.  Inter-sample-overs can be as much as +6dB which is severe digital clipping.  To be sure you will always avoid this problem, each digital processor must lower the digital signal going through it by 6dB.  This means the dynamic range keeps dropping by 6dB for each digital stage having an ASRC input.  (The presence of inter-sample-overs shows that the interpolation method used by ASRC's is not the linear interpolation we learned in High School, but a higher order interpolation method.)

This is little problem in a DAC where you can have more than the carrier number of bits operating inside the DAC itself.  You can have 32 bits for an incoming 24 bit signal, giving you way more dynamic range than needed.  In effect you have headroom above the headroom of the carrier signal.

But it is a huge issue for a chain of SPDIF connected DSP and storage devices (for which, BTW, the carrier jitter stays roughly the same no matter how may SPDIF interfaces are in the chain...at the end of 10 devices I still measure about 200ps jitter--same as from the input--because that's just inherent to the SPDIF carrier itself (as described above) and each synchronous interface reduces it internally to near zero with a PLL, then it goes back up to 200ps at each SPDIF output because of the SPDIF carrier itself...which all goes to show how much of an over hyped concern jitter is).  Each one is going to reduce the potential dynamic range by 6dB because it has to output back into the 24 bit domain.

Fortunately, many of my devices are old school synchronous SPDIF.  That especially includes my 2000 vintage Tact digital receiver, which even boasts about locking on to the clock of any of its digital inputs.  And it's true of my vast army of Behringer DEQ 2496's, each an extremely flexible DSP and Display device which is sadly now discontinued.  And, especially, my Alesis Masterlink, which records the exact digital values send to it, thanks to using a synchronous digital interface (CS 8416 I think). 

SADLY, most new digital processing and storage devices do NOT have synchronous interfaces for SPDIF.  This includes the fairly ubiquitous (as a replacement for Alesis Masterlink) Marantz PMD-580.  The Marantz uses an ASRC to accept whatever digital signal is provided, from 32kHz to over 96kHz, and convert it to whatever rate you choose to record at, maximum 48kHz.  There is no way to turn this "feature" off, and that is typical of digital recorders you see nowadays.

The TASCAM DA-3000, their current 2 channel flagship, does allow you to turn the SRC on or off.   When I saw that, I knew I had to have this unit to replace my extremely cumbersome Alesis Masterlink, and my ASRC-centric Marantz PMD-580,  to make digital recordings.  (I use my Lavry AD10 as analog-to-digital converter, and I suspect it may still be better than the DA-3000--though limited to 96kHz which is fine by me--and then pipe the AES/SPDIF signal to the digital recorder.)

It took considerable time for me to figure out a way of confirming that the DA-3000 makes bit perfect recordings.  And then my first measurements...which suggested it did not...were incorrect.  Finally I have concluded it does make bit perfect recordings at 48kHz when fed that signal from the Marantz PMD-580.

But with one strange caveat.  Once every minute or so there is an extra sample added or subtracted. 

I figure now that is because the "Synchronous" option works best when you also use a separate word clock signal.  I have ordered a Word Clock cable to check out this theory.  If that is the explanation, I'm good.

Another possibility is that this extra sample being added or subtracted is some kind of watermark.  I suspect it's not audible, but I would not be happy about it.

Using the same test procedure, I found the Alesis Masterlink is indeed Bit Perfect, though it might take one sample or so for it to lock on to a new signal.  (The DA-3000 takes 7-14 samples to lock on.)  The Alesis neither has nor requires a clock input or output.  At the 48kHz sampling rate, it just works perfectly without one.

Method

I recorded 30 seconds of music from the FM radio recorded on the Marantz PMD-580 at 48kHz.  I then played this over and over through 18 feet of Belden coax (2 pieces joined with an gold RCA double barrel...this is the "line" I have always used to play the PMD-580 on my system...but obviously not "ideal") into different digital recorders also at 48kHz, 24 bits.  I transferred the files (either on CD24 or CF) to my Mac.  There I used Sox to trim the leading zeros, and trim the end to the maximum shared length, and then to convert to a .DAT file having all the samples as numbers in a DOS text file.  Using the text editor Emacs I edit this file in several ways.  Primarily I edit to remove the column of time numbers which differ slightly between runs.  Just after the "signal" is starting in the digital recording, I look for the first matching line (pairs of values representing the sample value in each channel) in the two files being compared.  I edit out the earlier lines which result from one recording not starting as early or as fast as the other (depending on how fast the recorder locks on to the digital signal).  Then I also edit out mismatching lines after they both go back to all zeroes as that ending segment will vary depending on how fast the recording was stopped.  I then compare two files (typically both from the same recorder) using "diff" (terminal command in bash) to see any mismatches.

I have not yet tried using a word clock cable, but the current results suggest it is necessary for the DA-3000.

Results

For the Masterlink, once an initial sample or so is removed, two separate recordings match perfectly.

For the DA-3000--and with SRC turned off, once 7-14 samples are removed from the beginning of one recording or another (because of slow locking) they match perfectly, Except for about 1 sample added or subtracted every minute.  

I could not and did not need to the Marantz PMD-580 digital recorder because the SRC cannot be turned off and so it never records the original values but always interpolated values.  The specs for the AD converters aren't very good (and don't sound good to me) but no specs are given for the ASRC and I think it's fine as far as they go.  I haven't yet figured out which chip it uses, but obviously an ASRC chip.  I have never noticed bad sound when recording the output of the Lavry AD10 running at 96kHz converted to 48kHz by the Marantz, and actually the 48kHz recordings are smaller and more convenient, so it's wonderful for recording FM radio while using a better ADC (like the Black Lion) in front of it.  I'd hope for something better for recording vinyl, and it just bugs me that I cannot in principle make bit perfect recordings with this recorder.  It will always be subtracting something, measurably but likely imperceptibly, with the ASRC, when it's not really needed for recording to a flash card.  Therefore it is never giving a truly honest account of the digital signal, it is always part of the mix.  For the purposes of relaxed listening, that may be ok, but for the purpose of Audio Investigations, it is not acceptable. 

Conclusion

The digital interface receiver of the Alesis Masterlink appears to be the best I have tested, providing perfect results even in the less-than-perfect test setup.  I sure hope the errors with the DA-3000 with the SRC off go away when I add a word clock cable, otherwise I'm returning it.

Chips

Digital interface chips using ASRC

CS 8420 widely used but has well known bugs, must sometimes be restarted, so requires a design with microprocessor that can manage that, despite spec sheet alleging otherwise.  It can occasionally go into "garbage mode" or "muffled mode" and when that happens it needs to be restarted.  Some have characterized it as "evil."  It can be used in one of 9 different modes, including several which are PLL only (no SRC).  So you can give the user a choice.  But the PLL only mode may be inferior to that on actual PLL interface chips--it's just a pre-smoother for the ASRC.  The chip is supposedly discontinued (still widely available it seems) but current designs inspired by it are legion.

I see now that my beloved Behringer DEQ 2496 Ultracurve Pro units use this very chip.  I will have to investigate whether they use a PLL or an ASRC mode.  Somehow it seems they do preserve input sampling rate at the output.  The Ultracurve itself was discontinued in 2021.  I know because I've been ordering 1 or 2 a year.

AD 1896 an early reference standard following from the pioneering  AD 1890

WM 8805

Nowadays some DAC chips have the digital interface built in, and if so, you can bet it's ASRC.  PLL's require more ancilliaries which was why, historically, there was a separate chip receiver.  ASRC makes it possible to make everything cheaper. 

Digital interface chips using only PLL (Synchronous) only

CS 8416, widely used a decade and a half ago and still, has 8 digital inputs, often only 1 used.

CS8414, a slightly inferior predecessor to 8416 (though some say the reverse).

CS8412,  earlier generation, regarded by some as the best of the 84xx series.

AK 4117, possibly very slightly better than 8416 if you can find it.  I'm not sure of the design of later AK interface chips.

https://www.diyaudio.com/forums/digital-source/73446-dac-design-first-step-spdif-receiver-print.html

TI DIR9001  Possibly the best PLL chip of all at least when it was introduced.  People were saying this wouldn't be available after 2005, but it appears that you can still buy it new online.  So, it seems now, the designs and redesigns that were inspired by the widely feared impending demise of this chip were wrongly inspired.  HOWEVER, it is limited to 96kHz digital inputs.   So I say, what's wrong with that?  What high end gear that is not terminal (like a DAC) should do is use a real PLL chip for the PLL modes, not the crappy PLL of a chip mainly designed as ASRC.  And then if PLL doesn't bother with anything above 96kHz, like the great PLL chips of old (and still now apparently) that's fine by me.

The Alesis Masterlink uses AKM chips so it seems likely it uses the AK 4117 receiver (or earlier generation of same) which is said to be better than any in the Crystal Semiconductor CS841x series.  I wonder if the Tact 2.0 RCS uses the TI DIR9001 which is a high performance chip made by an American company like the Tact itself.  I have never directly compared the two but generally they both work well.  I've only rarely used the digital inputs on the Masterlink (unlike the PMD 580 it has decent sounding built in analog to digital converters that I've used instead), or even used it at all (I don't make recordings very often), whereas for 20 years I've endlessly used and tried to do impossible things (such as 100 foot cables) with the Tact, so I know the Tact is slightly less robust in locking to 88.2kHz than at 96kHz with long cables, possibly having nothing to do with the chips used.

TI DIR1703 was an early buggy version of DIR9001 from the mid-to-late 1990's.

[Update December 7, 2021]

Tascam DA-3000 Passes Bit Perfect Test with Clock Cable

I finally had time on December 5th and 6th to retest the Tascam DA-3000 using a clock cable, as I correctly surmised that it needed.

I could not do a repeat of the previous tests, except with the clock cable, as I had originally planned (and why I bought a 15 foot clock cable).  

What made the test impossible is that the Marantz PMD-580 does not have any clock inputs and outputs.  The Marantz is built with the philosophy that an ASRC operating all the time is just dandy.  It is not possible to turn it off.  You never get the exact original bits, but the Marantz meets its noise and distortion specs using ASRC.  For many purposes (except high end audio and audio investigations) that is Good Enough.

So I had to set up an even more elaborate test using the Lavry AD10, which is the important thing for me in actual use.  What I wanted to do all along is want to record the bits from the Lavry AD10, a very respected and nice sounding analog to digital converter which is still being sold as a new product for about 50% more than the DA-3000 itself.  I have always used the Lavry in the Living Room system to digitize vinyl records into my digital front end just for playback.  It was the best analog to digital converter I could afford (and still).

Since I already digitize vinyl simply for playback on my system, I would have thought an inexpensive device could simply capture those bits perfectly from SPDIF and record them to a Compact Flash card (or better yet, a USB memory stick).  Unfortunately such an inexpensive device does not seem to exist.  Inexpensive recorders actually tend to have no digital inputs or outputs at all, and definitely not AES balanced digital inputs and outputs.  Earlier generations of the Tascam had no way to turn off the SRC, just like the Marantz PMD-580.  The only commercial bit-perfect recorder that I know of other than the DA-3000 is the Masterlink ML-9600, which hasn't been made in almost 2 decades and is cumbersome to use, requiring the burning of a CDROM for every data transfer.  It's also too noisy to use in the Living Room, I decided years ago (though I now understand a tricky DIY SSD upgrade is possible, and that is said to eliminate the noise which I previously believed was caused by a non-removable fan).

I did use the Masterlink for all my earlier vinyl transcriptions, before I had vinyl playback in the Living Room.  In the bedroom, I arranged the masterlink so it's vent holes (I figured there was a fan there, but it might be just the harddrive) pointed away from the turntable, or at a different elevation, so it was never much a problem.  In the living room, the noisy vent holes of the Masterlink would be inches away from the turntable, not a good situation.  That was what led me to acquire the Marantz PMD-580.  It was only after I acquired it on eBay that I discovered you could not turn the ASRC off, which is contrary to my goals.

I had figured I'd repeat the PMD-580 to DA-3000 test, but just with a clock cable.  That would be holding everything else constant, except for the clock cable.  And then I'd test copying bits from the Lavry, the real goal, after that because it's a more complicated test.

But I had to skip straight to the more complicated, but also more important (like actual usage) test.

To do that, I connected the DA-3000 clock output to the Lavry clock input, with "Word Clock" selected on the front panel of the Lavry.  When it is receiving an external clock signal it can handle, the corresponding sample rate light lights up.  It is pointless to preselect the sample rate when you are using an external clock.

The AD10 has a Clock Input which accepts either Word or AES clock.  The DA-3000 has word clock inputs and outputs.  I do not need to change the clock setting of the DA-3000 since the work clock output is always active at the sample rate currently selected.

The AES digital output of the Lavry already goes through a Henry Engineering 4 way AES splitter (which operates like a little line amplifier buffer with zero delay) so that I can play on my system while recording, previously on the Marantz.  But now, to record on the Tascam, I ran AES cables from the splitter to both the DA-3000 and the Masterlink ML-9600.

So, whatever I record, I record it identically to both recording devices.  If the recordings turn out to be identical, then they must both be bit perfect.  I already proved the Masterlink is bit perfect, but even if I didn't know that, identical recordings on Masterlink and DA-3000 would prove they were both bit perfect, since they could not match if they weren't.

The pink noise track from the Stereophile Test CD 2 was playing over and over on the Oppo BDP-205, and the balanced analog outputs of the Oppo were feeding the Lavry analog inputs, with the Emotiva XSP-1 in between doing level setting and buffering.  So I'm not recording the actual bits on the CD, I'm recording the bits produced by the Lavry converting the analog signal which originated at the disc player.   Every recording will be different from the previous one, because the recording never starts at the exact same instant of the signal, but if both recorders are bit perfect, the corresponding recordings made from the two machines machine in the same session should match.

I recorded this track 7 times on both the DA-3000 and the Masterlink.  I primed both machines by pressing their record buttons.  (On the masterlink, I initially had to select playlist 1, then playlist edit, then every time I also had to press new track, before pressing record.  As I said, the Masterlink is cumbersome to use.)  When there was a gap in the playback of noise, I pressed Play on both machines (which is what you do to start recording, because the record button only engages "Record Pause").  Then when the track ended, I pressed Stop.  And so on, seven times.  I did it more than once in case I messed something up, and it's easy to keep doing once you get rolling.

As it happened, when I made the transfer CD24 disc on the Masterlink (containing all the 24/96 digital recordings) there had been one song from the previous tests still in the playlist.  I thought to myself "no problem, I'll just remember that when comparing."

As it turned out, by the time I was actually messing with all the files on my Mac, I forgot about that difference in track numbering.  And as a result I spent 30 minutes trying to make two files that would never match (because they were different recordings from a virtual analog source) line up because I could not find matching pairs of numbers.  I got very frustrated and angry.  But then I remembered I had to test a higher number Song from the Masterlink to the one from the Compact Flash which came from the DA-3000.

Once I got the two digital recordings to line up in time, and then removing the time counters from the files, using the method I've described before, they matched perfectly.  40 seconds of digital data at 24/96, almost 4,000,000 24 bit samples for two channels without a single difference.

The terminal on my computer screen looked like this:

(The two pound signs shown after the diff command were my initial faulty attempts to do a screenshot.)

Having passed the test, on the evening of December 6 I removed the Masterlink for storage and "permanently" set up the Tascam DA-3000 as my living room recorder.

Mysterious Problem Solved on Thanksgiving

[See Update 2 at bottom.  It appears a Sonos update the next day fixed this problem for now.]

[Nov 27 update I filled in some more steps in the testing.]

On the evening of Thanksgiving 2021, I was playing the living room system through the Sonos Connect known as "Living Room Sonos."  Or perhaps I should say...the Sonos Connect soon to be formerly known as "Living Room Sonos."

I wasn't playing THAT loud, I had the Tact level set to -10dB or so.

There was terrible breakup on a recording from Pipes Rhode Island, the Frank Chorale in B Minor.

I was using the Sonos Connect instead of the Oppo BDP-205 because I had previously been playing the Kitchen FM Tuner in the Living Room, and Sonos has analog inputs which let me do that (the reason I selected Sonos in 2005).

I tried other things before finally determining the Sonos Connect was at fault.  I was first afraid the speakers had an issue, very afraid.  But it was hard to test that...so instead I switched amplifiers to the Aragon 8008 BB.  I hadn't done that in awhile.  I figured it *could* be an amplifier problem, the Hafler has about half of the peak output current of the mighty Aragon--which has 1100VA transformers for each channel.  It was still breaking up.

I was getting more and more afraid that the speakers were indeed at fault.  So I did something I haven't done in many years.  I got out my Koss ESP-950 headphones and listened to the same song.   Electrostatic headphones are like an auditory microscope...you hear everything.  The Koss transformer box takes coax line inputs, so I fed it from the coax line outputs of the Emotiva Stealth Dac.  I got the same breaking up sound.

OK, so maybe it was the digital processing.  I was now fearing that my FIR based digital input filters might be introducing some kind of artifacts.  ("F" stands for finite...it's an approximation over finite time...that's how it can do so many amazing things...because it's not exact--in principle.  But audio is never exact anyways.)

So then I fed the headphones (which take line inputs to the transformer box) from the output of my Tact digital preamp.  It has analog outputs (using an optional high quality DAC on an internal card) which I don't normally use because I only use the XLR AES/EBU digital outputs...which feeds my chains of digital processors).

Same breaking up sounds. 

Then it occurred to me to try the Oppo, and while streaming the file through the Oppo it sounded fine (except for a few "ticks" which I hear so consistently, even on my dynamic speakers, I believe they are in the recording itself.  These ticks are nothing like the hashy breaking up sound I was getting on the Sonos Connect).

Wondering how I could check out the digital output of the Sonos Connect (I only use the digital outputs on most Sonos nodes)  I grabbed my Sencore digital meter.

The Sencore showed nicely low jitter (just over 200ps) and no errors or any other problems with the Sonos digital output.  And when I first checked the levels, and after turning off volume leveling on the Oppo (which I play through Roon, whereas in this discussion I'm playing the Sonos boxes through the Sonos S1 Controller), the levels were identical.  While doing these tests, I was not playing anything over the speakers because I had the digital signal piped into the Sencore instead.

But then later I was doing a test when the Oppo was playing through the speakers, but running the test meanwhile on the output of the Sonos.  And then I got a bogus level of -0dB where I shouldn't have (the peak level in the recording is -0.7dB).

I switched to the other nearby Sonos node I have been using the play HDCD's through the converter of the Denon DVD-9000 by sending the coax digital from that node to the Denon.  That Sonos node happened to be the earliest model, the ZP80, which I first bought 4 of in 2005.

The ZP80 did not have the breakup issue either, so THAT is now going to be the Sonos node I'm connecting to the living room stereo.  I have rarely used the HDCD path (which sends digital audio from Sonos to the Denon DVD-9000 for HDCD conversion) and I could always temporarily change the coax output of that Sonos ZP80 back to feeding the Denon if I do want to decode an HDCD that way, that takes little more time than just turning on the Denon 9000 and selecting coax input.

I really don't know that it was the fact I was playing music (possibly causing vibration or putting noise on the AC lines) that messed up the "Living Room Sonos."  It might have been something else.  When it first happened, I thought it was because I had just clicked mute and unmute in the Sonos controller.  I had been wondering if there was a bug in which, under some conditions, the Sonos Connect does digital amplification when it should not be doing so (such as when the level output is fixed).  I had thought I had seen such a bug before.  But the test examining the level from the Sonos where it, at first, exactly matched the Oppo, disabused me of that theory for awhile.

It could have even been that the coax connection to the Sonos Connect was slightly loose.

All I really know is for some reason, that one Sonos Connect was messing up it's digital output, sometimes.  It could be the fact it is also processing the tuner input for the rest of the house.

Maybe it's better to have one Sonos Connect to handle the FM tuner input, and a second Sonos node of one kind or another to feed the living room stereo anyway.  I also trust the ZP80's as much as, if not more now, than the Connects.  I tested the ZP80's extensively, as did John Atkinson.  I never actually tested the Connects, I just presumed they were an upgrade.  I can't remember when I first started having strange issues that seem like digital gain and clipping, was it when I got the Connects or before.  Also nothing stays the same because of Sonos software updates, which may be more important than whether a node is a ZP80 or Connect (which are almost identical in appearance and apparently identical in intended function).

And most of the time, I'll just be using the Oppo through Roon anyway to play music in the living room, which is my best digital pathway.  But...I had been using the Sonos Connect to run tests, and I very deliberately did not add the test files to the Roon list of music folders for that reason.

That calls into question some previous tests...including the test where I decided I had to lower the input of the midrange miniDSP by -6dB to prevent inter-sample-overs.  Perhaps the real issue was the bug in the Living Room Sonos Connect.

So from now on, what I'm going to do for some important tests is to temporarily add the test folders from Roon when doing tests, and that way  I can use the Oppo for doing tests, without fear of selecting test recordings at other times through Roon.  And then removing the test folders from Roon afterwards.

Because of worries about forgetting to turn such test folders off, perhaps I'll still use a Sonos node for most tests, but the ZP80 which hasn't yet shown any issues, instead of the Connect.  Until there is a weird issue (like the apparent inter-sample-overs on the miniDSP's) for which the ZP80 might be at fault, which I should have thought of when I encountered the apparent inter-sample-overs on the miniDSP's while using the Connect.

It's here I might also confess that the primary issue that made me quit using my Audio G_D DAC 17, which was the Krell FPB 300 shutting down, perhaps was never a problem with the DAC 17 at all.  It was only many months after that I traced the many shutdowns to a bad transistor in the Krell itself, at least by all appearances (I can see the hot spot in IR photos for example).  And ultimately it became impossible to use the Krell anymore because it would shut down before even coming fully on, and therefore before I could zap it with a big enough signal to stay on for awhile.

It was sad that I didn't imagine that it was principally the Krell at fault.  I also blamed the speakers and other things, perhaps wrongly in other cases too.  I spent many months trying to "fix" the speakers so they "wouldn't cause" the Krell to shut down.  I still need to re-test the DAC 17 and I might even use it in the future now that my system has fixed digital rates for each way...making it possible to do the time alignment for sources of any sampling rate even with different DACs...since all sources get converted to 48kHz or 96 Khz (for the tweeters) anyway.

The Franck Chorale is sufficiently complex that despite hearing it endlessly, it sounded different every single time.  This is the actual problem with subjective equipment tests...it's not so much that things don't sound different, but that you never actually hear the same thing anyway, and then you have to decide which differences (if any) are traceable to the change in equipment (which is pretty much determined by preconceptions).  And repeats especially sounded different at different levels.  With large level differences, it sound like a different composer.

Updates: I have tested reconnecting (and tightening) the digital cable from the Living Room Sonos Connect.  I still heard the breaking up.  I'm beginning to think this particular unit could be defective.   And the line inputs might be defective too.   About the same time as I replaced the original Living Room ZP80  with this Connect node I also noticed the line input from the tuner wasn't sounding as good.  According to all I've read just now, the only difference between ZP80 and Connect (same as ZP90) was a different and better radio.  If I remember correctly, I made this change last December, about a year ago, when I was incorporating the Marantz recorder and making other changes to improve the sound of FM, and it did sound much better in the living room, but not over Sonos, where it sounded worse.  I've had this theory that the better FM antenna, which reduced FM noise, was also making it sound more crispy.  But it sounded fine in the living room because I added the Musical Fidelity X10D to that path.  It merely sounded worse over Sonos.

Also the ability to control the line level input levels was removed from the computer software.  You must use the App (on a smartphone) to control this feature.  Or originally you could use the CR100's to control everything, including the line levels and the "Uncompressed" feature, but then CR100's were unsupported, which made me very angry and I still miss the convenience of the CR100's as compared to using a smartphone.  With the CR100 the pause and play buttons were always active, whereas with a smartphone you have to do a lot of fiddling to do anything.

My tests and those of John Atkinson showed the ZP80 to be bit perfect and it performed very will in the jitter test and other tests.  But neither I nor John Atkinson have tested the ZP90 aka "Connect."

I had been disappointed that Sonos never included high resolution into the original system (I believe it is available in the new S2 units) and went the direction of making midfi convenience speakers and such, rather than sticking to true hifi interconnection units (now called "streamers") and reasonably good amplifiers.

I see now that even my old "Connect" units cannot be used with the latest Sonos software.  I hardly care, if anything, I'd like to return to the original Sonos software of 2005, which did everything I needed it to very well.  I did later get a Logitech streamer, but it was such a hassle to use I never much did use the Logitech streamer, despite the fact it had high resolution.

But I can see the name of the game for many people is supporting all the different streaming services.  That is the excuse for the new incompatible S2 Sonos units and software.  I hardly care if Sonos keeps up with streaming services because I use Roon for that purpose, and for me it makes much more sense to me that interconnection units be dumb and the smarts be put into the central software as Roon is doing it.  I'd long hoped to roll my own dumb Pi3 interconnection units that were Roon Ready, until Roon started supporting Sonos, and then I had no excuse (other than modest monthly cost) not to get Roon and use with my existing Sonos network.  But Roon has never had any ability to use the Sonos Line Inputs.

Update 2:  This morning I googled "Sonos Uncompressed" and found you can set that feature (which defaults to "Automatic") using the app not on a node-by-node basis, but on a whole system basis.  Reportedly this feature only affects line-in playback, but the App doesn't say so.

I went to check the current setting in the app, but it wouldn't even let me view the current settings without updating my whole system.  I'm not sure when this update became available.  So I updated my system, and then it appeared that the option had already been set to "Uncompressed," as I now remembered doing sometime in the distant past (but possibly when this particular Connect node was offline or before I even bought it?).

I had already been testing the Sonos Line Input between the two different living room Sonos nodes.  I think they are identical.  But anyway I had moved the tuner output to the ZP80.

So now, with these two changes (updating Sonos, and moving the line input to the other node--which might be unimportant) the Living Room Sonos Connect sounds fine, no breakup playing the Franck chorus.

The best theory now is that somehow the Sonos Connect had a bug which was fixed by the update.  After the update, no more problem.  Though perhaps it wasn't even the update but a reboot caused by the update.

A related possibility is that Roon can fiddle with the Sonos level settings and left them in the wrong settings.  I sometimes play the Living Room Sonos Connect with Roon.  Roon has two relevant settings for each Sonos node, "fixed level" vs "device level" and then, when you select fixed level, you can set volume settings: min and max.  Last night I went through all my Sonos nodes in Roon and made sure all the ones that are supposed to be "fixed level" are set to "fixed level" in Roon.  I had never done that before.  Indeed the Living Room Connect was set to "Device Level," but Sonos S1 controller always showed it as "Fixed Level" anyway.  But I don't recall that changing this last night made any improvement in the sound.  Perhaps it required to update (and reboot?) to get it changed back correctly.  Perhaps the update makes sure the settings in Sonos are the ones in effect when you are doing playback in Sonos, regardless of what some other streaming software may have been doing previously with that node.

I still consider the Connect to be questionable enough that I'm going to use the ZP80 for living room playback, and either use the Connect for line input only, or in some other room where I'm not depending on the Fixed Level setting to remain in effect.  The bug has gone away for now, but there's no telling when it might come back.

Still testing the Sonos Connect, I moved the tuner connections to the inputs on this node again, and there is still no problem.

Both Sonos nodes are now working fine, but I'm still planning to switch the living room stereo to the ZP80 which may be better...though it's possible and perhaps even likely the problem was a software bug that just happened to the Connect based on how it had been used.  It may have been, for example, that I pressed the volume up button on it, and if Roon had told it to use Device Volume sometime in the past, that may have overridden the Sonos settings when using Sonos, though it shouldn't have.

But actually I'll run the FM tuner outputs to the Connect unit because it has a better radio in case one of the ethernet links goes down.  And it will interfere less with the ZP80 that will mainly be used for tests on the living room stereo (though I've never noticed any interference including in these tests).  And the Connect will feed the Denon for HDCD encoding, so if the bug occurs again it will be noticeable as the HDCD light won't light if the bits are tampered with.

ISY-994i programming

I use an ISY-994i controller for my home automation.  It is very easily user programmable, and powerful.

However, the "Programs" for the ISY-994i have a particular weirdness that I've struggled with for a decade and now, thanks to discovering this informative post, I have a handle on.

https://forum.universal-devices.com/topic/4376-wrapping-my-head-around-how-the-isys-else-works/?tab=comments#comment-33946

It's not obvious what the "trigger" conditions for running a program are sometimes, and they have to be understood separately from the "condition" which determines whether the THEN or ELSE clause runs.

If a condition is something being switched, the program is only triggered when it is switched THAT way.  So if that is the only condition, the ELSE clause is never run.  A way around this is to add an "is not" clause because they must be evaluated for any change.

OTOH, if a clause is about the status of something being on or off, the program is run for ANY status change, and then the THEN or ELSE may apply.

A program is run if any of the conditions is triggered, so then the THEN or ELSE may apply.

Gain Structuring

 For complicated reasons I'll describe below, I decided to turn the gain up on all 3 Emotiva Stealth DC-1 DAC's I use in the living room system by 4dB.  The current levels are

Tweeters: +10dB

Subwoofers: -11dB

Panels: +2.5dB

It's actually become extremely convenient that these DAC's have 0.5dB resolution gain switching that goes way down and up to +12dB.  Functionally it's like getting a preamp with your DAC.  Though purists might rather have their own high end preamp, the gain here is provided by an impeccable LM4562 with a 2005 generation digital level control (that was when they got about as good as an LM4562).  If I ever switched to another DAC w/o gain, it might be difficult.  But now thanks to fixed digital rate of 48kHz from the miniDSP (and 96kHz for the tweeters) I can simply time align the DAC's (and the speakers) for these fixed rates and it's done.  (Previously, where the rates could vary from 44.1kHz to 96kHz...causing different delays in different DACs, it was impossible.)  So, I can fairly easily use different DACs on each way of the system...but few have the nice digital level controls of the Emotiva and with 12dB extra gain.

The actual tweeter (crossing in at 20kHz) levels are so low that boosting the gain there is never going to cause the DAC to overload...or the following amplifier.  The SVS subs are just very sensitive and I have their local level controls turned down also.  The DAC feeding the Hafler 9300 driving the panels had one time been at -6dB so I could set the higher gain Aragon to 0dB.  The Aragon is currently offline because the DAC feeding it now needs repair.  Just before this latest adjustment shown above, the DAC feeding the Hafler had been turned to -1.5dB.

Well I decided to do this after seeing that even turned all the way up, to no digital attenuation on my Tact, I was getting peak levels at -87dB C weighted.  That's not very loud.  But you'd think at "0dB" it would be.

But when I was setting up the miniDSP for the midrange, I detected clipping problems if I set it's input gain control above -4dB (so ultimately I chose -6dB to be on the safe side of no digital clipping).  This is because the miniDSP use a 48kHz ASRC input for the digital input.  Such asynchronous sampling can be pushed into clipping by "inter sample overs" because there can be (and are, on many pop recordings) peaks up to 6dB (theoretical maximum) higher than the nominal level.  This is another reason why units with SPDIF or AES digital input and digital output should use SYNCHRONOUS not ASYNCHRONOUS transmission.  Asynchronous should only be used, if at all, in a terminal DAC unit which does not output digital.  But totally unwarranted fear of JITTER has led to the near abandonment of synchronous digital receivers, despite the fact that the better ones did a nearly impeccable job with typical low jitter levels and considerably beyond.

The 1999 Tact RCS 2.0 did in fact have synchronous digital in and out, at rates from 44.1kHz to 96kHz.  My living room system uses this as source pre-selector (to the input of the home-controlled switch, which switches between FM and everything else, and from there on to a splitter to the 3 miniDSP's) and level controller.  I don't find the DSP in the Tact system meets my needs, with a trifle re-programming it could.  The Tact preserves the input rate--and timing--from input to output.  There might be some smoothing in the timing, I'm not sure of the details actually.  But if I were to build something, I'd have it like this.  I'm unsure if the synchronous receivers are made anymore.  Or updated to 192kHz, which I don't consider of much importance.  Actually my miniDSP's are limited to either 48kHz or 96kHz anyway.  So it would be useless to lust over a front end controller ("digital preamp") with more than 96kHz anyway.  And as I've many times stated, I think all that's needed is 96kHz, everything above that is a WOMBAT (Waste of Money, Brains, and Time).  Peter Aczel and the near entirety of "Objective" audiophiles would say 44.1kHz is enough and that nobody has "proven" otherwise in reproduced blind testing.  But I'm allowing margin over 44.1, for various reasons, though I recognize that 44.1kHz sampling can be damned good (and likewise 16 bits) and the vast majority of what I listen to is just that, and I'm still often (and pretty much always nowadays) wowed and so on.

So, the unfortunate loss of 6dB gain in my digital processing means I have to make up the gain somewhere.  If I were to increase the output of the Tact, I'd get actual digital clipping (not just intersample overs) at the miniDSP inputs.  So the only thing to do is to raise the output "after" the digital conversion.  Which I can conveniently do with my Emotiva Stealth DC-1's because in fact the level control can be increased to +12dB.

But what happens with a digital signal at 0dB with the Emotiva at +12dB.  Distortion rises from 0.0005% to 0.02% which doesn't matter audibly.  That's because the output circuitry of the Emotiva can actually drive 25.65dBu (14.85V RMS) !  (Well, that's into balanced...) The output with the control at 0dB is 14dB, or 3.88v RMS, into 4 ohms, which is a bit low for some high end power amps, but you can crank it up.

I got that information from this Pro review.  Pro's are interested in these sorts of things, which often don't get tested by audiophile reviewers.  (John Atkinson always tests maximum output level, but it seems Amir did not, he simply tested with the level control at 0dB which is not the maximum.)

In unbalanced I likely have half the output, or about 7.92V, which is still almost 11.95dB higher than the standard output of 2V with the level at 0dB.  So I can crank the level up to at least 11.5dB without concern basically (at least as far as the DAC is concerned), even if perchance there were a maximum digital level signal.  (Perhaps to allow for inter-sample overs in the DAC itself...it might not be a bad idea to keep the level below +6dB for ultimate fidelity.)  In my case, where the signal itself is deliberately -6 dB, I'm good with the 11.5dB level even before intersample overs become an issue.

The amplifier has "headroom" (above it's 1.1V rated input for full output) and besides such signals are way below the average level, due to both the mandatory -6dB processing reduction, and the crossover itself.  If the amplifier is working too hard, I will hear it, just like someone with an all analog setup.  What I strongly want to avoid is "invisible" digital clipping from ASRC for example.

Basically the signal is never going to go to the peak level (for one thing it can only go to -6dB on average) and the panels only get frequencies above 120 Hz, which also reduces the level.  If there ever is a transient peak (which is all it could be above the -6dB level because of my miniDSP attenuation) it won't stress the analog at all.

Well now that I've checked the Emotiva capabilities, I'm thinking of cranking up the DAC's another 2dB.

What to do with CD's after Ripping

 It seems that hardly anyone recognizes the ethical and legal issues regarding first ripping CD's, then selling or donating them.  Essentially two copies of music have been created, but only one was legally paid for.  This issue gets only a tiny bit of discussion (while being dissed everytime it comes up) in this thread.

Some keep their old CD's for this reason or other reasons, while others donate or sell.

I had been planning to keep all my old CD's.  But as I acquire more and more, this is looking untenable.

I was hoping someone had already "solved" this problem with "recycling" service for CD's.  You send boxes of CD's in, and they recycle the paper and plastic and return you a receipt for all the titles you have recycled, which you can then save as proof of ownership.

Short of that, I was thinking of taking photos of 100 or so CD's at a time laid out flat, then boxed, then trashed.

Though I suppose, old CD's are only the tip of the iceberg of my junk problem.

Phono Frequency Response

I had a nagging to play vinyl on Saturday Night.  On top of some pile I can't remember buying I found Queen's Greatest Hits.  Having heard these same songs on digital, in some cases high resolution digital, this was different, having more presence, transparency, liquidity, and magic (and some noises).  But also, seemingly, less bass.

I think the less bass part is primarily a problem with this "Greatest Hits" pressing, which may not have the extended bass of the originals.  But also I suspected it could be my turntable system.  Often when there is less bass, we hear it as greater presence and transparency.

So I got out Hifi News and Record Review Analog Test LP, and played Track 7 Side 2 which is a sweep tone.  One of the very annoying things about this record is that each band is walled off from the others.  In order to play Track 7, you can't just cue to the space in between 6 and 7, because that just reverts back to 6.  You have to pass the wall at the beginning of Track 7, and cue precisely there.  It's hard to hit that point precisely, and when you do, the tonearm is still stabilizing at the beginning of the track.  That might, partly expaln the bass below 30 Hz in this recording shown here in Audacity.  There's also a matter of intermodulating with the tonearm and antiskate system resonances..

 The bump at the bottom where it more or less stabilizes is 30 Hz.  Up from there to the midrange level represents about a 1dB increase.  So it appears the bass below 100 Hz is gradually rolled off by 1dB at 30 Hz, then looks funky.  (The scale above is linear so it shows changes that look larger than they would on a logarithmic dB scale.)

I wouldn't think that 1dB loss would be that big a deal, I routinely make changes of that amount or more in the bass level, and in fact I tried adding 5dB (by changing the DAC output level from -15dB to -10dB) and the Greatest Hits recording still sounded like no bass, until the last track.

The instability below 30 Hz might be a problem, but possibly worse there is a peaky resonance just below 20khz which shouldn't be there.

Another measurement I made indirectly shows the instability below 30 Hz is not a matter of initial tonearm/table stabilization.  I happened to drop the needle at 30Hz instead of 20 Hz, and the reproduction is pretty clean, only showing the barest overlay of the needle drop stabilization, which disappears after 13 cycles of 30 Hz (just over 1/3 second).  That initial instability could also relate to the slight peak (small fraction of a dB) at 30Hz shown in the previous measurement.  This means that the instability shown above is caused by something else, presumably interaction with tonearm vertical and horizontal resonances that needs fixing for good bass below 30 Hz.  I'm showing higher magnification in the selected portion just after the needle drop (also shown) for clarity.

Digging through my files, I found a measurement with my Mitsubishi LT-3 linear tracking turntable, with damping added to tune the bass just as with the Linn (though it was an earlier job I didn't do as much or as well).  They both have a Dynavector 17D3 playing through an Emotive XSP-1.  I'm showing this somewhat aligned with the Linn result above.

The Mitsubishi is on top here.  I think it has slightly flatter bass below 120 Hz (perhaps 0.5dB loss down to 30 Hz vs 1.0dB for the Linn), it lacks a weird resonance in the Linn around 120 Hz, and the bass extends possibly a tad farther before it becomes unstable below 30 Hz--but they are so little different in that regard it's hard to tell, in both cases the bass becomes unstable just below 30 Hz.  They both seem to have a resonance just below 20kHz, I'm wondering if that's coming from the cartridge body or the screws.

I wonder if the resonance around 120 Hz in the Linn is the infamous flexible subchassis resonance and "flabby midbass."  From the day the LP12 was introduced people wondered why they were getting such a thin and flimsy subchassis with an expensive turntable, when even much cheaper turntables like the AR Turntable had very heavy gauge metal.  HP of Absolute Sound denounced both the subchassis resonance and the flabby midbass in his first review of the LP12.

Famously Linn doesn't have a subchassis upgrade except for the very expensive Cirkus bearing.  I figure this is because the original bearing has a noise that is absorbed by the flimsy subchassis.  The expensive Cirkus bearing eliminates that resonance, so you can even have specially cast and machined subchassis with it.  Such were never offered for the original bearing (except by outside vendors...and they were not universally loved).

The flimsy subchassis could have eliminated a bearing noise.  But the price may be that funny ripple in the frequency response around 120 Hz.  Remember the cartridges are the same model.

Although the Mitsubishi turntable yields better frequency response, I don't think it sounds as good.  It has larger and more undamped arm/cartridge resonances and/or motor and bearing noises.  The movement of the linear tracking tonearm subtly affects tonality.  I could possibly fix the larger arm/cartridge resonances, with the application of more damping material, and the electronic motor is probably due for a tune-up.  My Linn Valhalla board has been replaced at least twice, the Mitsubishi has it's original electronics and it's a wonder they're even still working.

Testing the M50

In May 2012, I decided to reward myself for something or other, and bought the M50 I have long planned to buy.

The Earthworks M50 is the king of measurement microphones nowadays, with calibrated (and quite flat) response to 50kHz.  It costs far less in real terms than the standards of old, such as B&K's with their custom amplifiers and such.

I think John Atkinson of Stereophile uses an M50, either that or an M25.

Until now, I've had no proof that my system even has frequency response above 20kHz or so, something I've invested thousands of dollars and endless fine tuning to achieve.  I have no idea what the proper calibration for my previous custom 25kHz calibrated Dayton Audio above 25kHz.  Given the different calibration curves for different angles, it's also hard to know which of several calibrations to use.  I didn't really trust it to 20kHz let alone 25kHz (but in fact I was wrong...it matches the M50 amazingly closely!  Perhaps I needn't have bothered.  Nah.  Anyway I plan to use my old mike for general purposes and the M50 for calibration and "final" measurements.)

For the measurements here, I recall I put the microphones straight forward.

Here is my system response with the 25kHz Dayton Audio microphone:

Here is the response with the M50, which I finally figured out how to calibrate correctly (the calibration they supply cuts off below 700 Hz..for ARTA I had to add in fake points at that exact same level).

The M50 shows a slightly cleaner HF peak at 22kHz, and about 3dB more output at 40khz, and that's about it.  If I had an accurate calibration above 22kHz for my Dayton I could probably make them match even better.

I think the speaker system response, shown in much higher resolution that you usually see above 100 Hz (and less below, thanks to FFT techniques) is pretty good.  It slopes nicely downward from 1kHz (though perhaps a bit too early and there is more loss than desirable above 8kHz or so.  The inaudible 22kHz peak merely restores the highest treble to the midrange level (and less than the deep bass), and also makes possible the response extension to 40kHz where there may still be more output than at 14kHz.  Many highly esteemed tweeters have much larger ultrasonic peaks from fancy construction.  My 21mm Dynaudio D21AF's have factory response curve showing flat from 1200Hz to 39kHz measured with close microphone but they don't account for a lot of variables here, including the polar response dilution of a cloth dome tweeter (which is good enough to even make 40kHz response possible--which can't be done with many materials including of course my electrostatic speakers).

Anyway, there clearly IS ultrasonic response at useable level at 40kHz, which has been my goal, and now proven with a top calibrated microphone.

Instead of huge peaks and valleys in the bass in the unequalized bass response, or where the wall bounce isn't properly corrected (I dodge that with my 4 foot wall spacing and crossing subs at 125hz with 8th order linear phase crossover) it's pretty flat 80-800 Hz (amazingly flat in my building and measuring experience), with just a soft rise below 80 Hz (which I may have reduced with subsequent adjustments as I'm writing this months after these measurements were recorded.)  It sounds natural.  1/6 octave omni measurements on my phone show less difference below and above 80Hz than these FFT measurements.

Some measurements last year were made before I figured out how to run the supertweeter miniDSP at 96kHz.  So it was running only at 48kHz like my other DSP's (the 48kHz plug in is more flexible than the 96kHz...but that flexibility is not an issue for my supertweeter...so I run only the supertweeter crossover at 96kHz where the highest sampling rate matters).  So measurements through the middle of last year showed a steep dive at 24kHz, the Nyquist Frequency of 48kHz sampling.  It took me awhile even to figure out that sampling rate was a problem, and until I did I greatly feared that I was basically getting zero response above 24kHz.  I remembered that fear into the present.  By that time I was thinking the microphone calibration explained the rolloff above 22kHz...but in fact the peak there, and the general response, shown by the Dayton...extending the highest calibration to all higher frequencies...is very close to that shown by the M50, with hard-to-see difference at 40kHz where useable response is ending with a precipitous roll off.  Also the Dayton curve shows a little notch where the calibration ends, whereas the M50 shows a smooth peak at 22kHz.

Automating Music Reproduction

Entry Switchplate (Labeled)

After decades of thinking and dreaming about such things, I've finally automated my main audio system.  I think it's the best upgrade ever, and I'm still working on making it even better.

Automation, in the context of audio reproduction, means two things (and preferably both):

1) Powering up and making whatever other connections, protocols, or other actions are required for the system to begin to play music.

For a personal device system, this includes:

1) enabling device, getting past authentication

2) dealing with personal device issues necessary (upgrades, identity issues, expirations, selection menus, etc) to run app, and running music app.

3) Connecting headphones to device, and headphones to ears.

Good luck automating that.  This is also an illustration of why I have kept personal device centered audio, and even USB (which means some sort of authenticating computing environment) away from my living room system.  Computer based systems nowadays are designed to authenticate and then spy on you, while gaslighting you to gauge your preferences ever better.  The whole point is to keep you making choices through actions.  Being able to bypass all that crap and just press a single button to start the music would be antithetical to that primary point of such systems to their makers.

I want a realm if not an eternity away from the general issues of such devices, which are anything but relaxing and inviting in my opinion.

I deal with the much conceptually simpler situation of automating the powering up my living room audio system, which I can and do listen to from all other rooms in my house.  Especially from the Kitchen, which also serves as my home office, audio and video mastering center, as well as kitchen, dining room, and snack bar.  Nowadays it's the single room I spend the most waking hours within, so it's also where I listen to the most music in Background.

The kitchen has it's own fairly high end audio system, featuring Revel M20's plus a subwoofer, AND surround speakers.  It's a fine system for mastering or watching video.  But as Background Music, I find playing the living room stereo much much better.  It is of course fundamentally a better sounding system, because full range electrostats combined with subs and supertweeters and far more and better fine tuning.   I lose the stereo image from the Kitchen.  But for Background Music, having no stereo image makes it less distracting, and that's the key.  If it were more distracting, I couldn't tolerate it and get other things done.  Not only is there no stereo image, I do strongly get the feeling that the music venue is in an adjacent room.  Therefore it can't get in my way, but it can create an ambient vibe that helps me feel better, like being in your own private room at the club.

I consider background music just as important as serious listening.  Not because serious listening isn't a wonderful peak experience (when everything works and it clicks) but it's comparatively rare event in one's life, at least in my experience--I never had very much time for it.  Background music may ideally be something for nearly every waking hour, except when you need absolute silence for perfect concentration or talking to somebody.  But if it's not easy to start up your main system for this purpose, you might hardly ever enjoy it for such. 

If all you ever use your main system for is serious listening (and/or serious listening or other tests), then I suppose a fairly time consuming startup procedure may be tolerable, or even help prepare the mind.  Many audiophiles have startup procedures taking many minutes.  I've always wanted to make system startup as quick and easy as possible, in the hopes that this would get me actually listening to music more, and I started using home automation devices for this purpose inconsistently and with varying success since the 1980's.  Mostly, recently, I've had no automation whatever, and had to power up the amp and fiddle with preamp and other components before starting, and that almost always involved getting down on my knees to fiddle with the preamp buttons, often waiting to scroll through volume levels for an appropriate volume...

2) Selecting the music, playlist, or station to begin playing.

This is really the important thing, and I believe, ultimately the most difficult thing.  And I mean in merely selecting what to play next, though in some cases the mechanics of getting that thing playing may be considerable also, though I usually count those actions as part of step 1 above.

When I started on this article in January, I had no idea I was going to be able to construct an automatic playlist generator this year, if ever.  Having a computer program select the music seemed too far fetched an idea.

Partly it's because my dreams were too far fetched.  What if the computer, acting as your personal concierge, would sense you mood from moment to moment, and pick the perfect music to elevate your spirit?  Well, yes, that would be nice.  Closer to earth, algorithms in Pandora are supposed to select similar music, but in continuous varying mood, because that's supposed to be better.  Perhaps.

My own finding is that given the set of music I like as both background and partly as foreground, that is particularly to me music without words, or too much stress, which can be classical, jazz, electronic, or ambient, I can just mix it up at random--done properly without replacement to minimize repeats and let you hear everything in shortest order, and these simple playlists have revolutionized my music listening more than anything.

With the new playlist generator I am programming now, I can listen to automatically generated playlists designed to play so I hear everything in my music library with as few repeats as possible--that is until everything has been played (or at least auto-selected and possibly skipped).

Roon automatically loads my playlists.  For other programs, though I have not tested this for other audio programs yet,  I can also create virtual directories of soft links which take very little space, and works unless the audio folders are moved, which isn't an issue for a near daily playlist generator, and would be the case for a normal playlist also.

I haven't yet set this to automatically make a new playlist every every day.  But generally it takes very little action on my part, at least when the still under development playlist generating program is fully working despite yesterday's changes.  [As of June 2021, the playlist generator has a known bug when a too high percentage of albums has been played, which I haven't quite reached yet, but will require a major rewrite to fix.]

If I ever work this up for Rock music as well, I'l have to improve the customizeability.  In that case it might be more like the computer program making suggestions I can accept.  I think this approach is far better than the old logo what do you want to do today.   I don't know what music I want to play today.   It's great to have a program make suggestions if not the choices...and in many cases the choices themselves ...or perhaps the choices following a chosen scheme.  Making more choices is hard work.  Too hard to bother with in the course of daily living.  Let the computer do the choosing.  If I'm in the kitchen or phone I can skip an unwanted track or album in the Roon interface.

Automated System Start Up

This is something that seems of zero interest to any of my audiophile friends.  They've simply not responded to my self congratulation.  I've had great difficulty writing about it, because among other reasons I want to capture why I believe it is most important, yet others can't see why it would be necessary or even useful.

In the pursuit of audio perfection, many if not most audiophiles have the opposite of automation.  They have start up procedures that may take many minutes, perhaps even an hour or more for total warmup.  Of course they believe that all these steps they must take before playing music are absolutely necessary (though skeptics may wonder if they aren't superstitious rituals).  And while many of these audiophiles have been devotedly making the circuitry in their system components ever "simpler," through adept acquisition or modification or construction, this often means even more work in starting up each day.  Or switching from one source to another.  They may have to replug whole chains of equipment to switch from analog to digital sources (which I'm including in the "startup" part of the equation).

Many audiophiles also believe any attempt to make anything automatic must involve compromises at some level.  They would be utterly opposed, for example, to adding switch devices in any AC path.  I will have to explain why the switching I do is perfectly fine, at least in the living room system, yet I'm sure many audiophiles will never believe me.  However, I should note that I've been unable to add any kind of automated AC switching device in the kitchen, where due to multiple outside antenna grounds, AC grounds, and whole house network connections, and low power devices on micro switching supplies, is a virtual RF oven in the AC path of the power amp.  Anything but the most highly shielded AC cord introduces hum and noise.  But the living room has no such problems, despite 3 AC circuits, and network connections from the Kitchen.  This is an issue I hope to get fixed sometime possibly by moving some things around.)

Now you might think that turning on the stereo to play FM would be a fairly simple process taking about 5 seconds, as it once was for me, long ago, when I used a Marantz 2270 receiver (purchased in 1974).  To play FM, I would press the ON button and turn the selector switch to FM and possibly readjust the volume control, with all those controls being in one fairly convenient place.

But in fact it hasn't been that simple since the days when I had my Marantz 2270 right next to my bed, in my college dorm room, which was long ago and not for very long.  And of course for anything other than FM there was a matter of deciding what media object to play, obtaining said object, preparing it, mounting it on the playing device, and initiating playing.  So playing anything beyond a favored FM station can be a lot of work, which is impossible to have the time and energy to do, simply for background music.

Before long my Marantz receiver, and/or the Kenwood KT-7500 tuner I later plugged into it for better FM reception, was not right at hand, but typically across the room from my bed or chair.  I did that mostly by something like necessity, but often rued the day when I had to give up direct reach control of my system for "passive" line level switching and attenuation boxes.  (I denounced passive as lossy sometime around 1998 and stick to that now btw, but I was a passive fan from 1980-1998.)

By the early 1980's I owned a house with stereos in both living room and bedroom.  Partly because of inconvenience, and partly due to lack of free time, I didn't actually use either one very much.  The living room stereo had grown (and kept growing) into many components.  I once figured I spent far more time modifying audio equipment than actually using it.  I had become similar to what one friend calls a build-o-phile from 1980-1982 when I finally built my own tube preamp after having been a tube modifier for 5 years.  Well, so what, if I was having fun that way?  But it would have been nicer to run that show with music but in fact I rarely did.  One a handful of occasions by the late 1980's I had enjoyable audio parties.  But it was only on a small multiple of other occasions did I have time for extended serious listening.  And it almost never worth the bother to switch on my tri-amplified system of 1985 just to play background music on FM.  If I did switch to FM, it was only after already having played something else, and not being able to decide what to play next, I put on FM, which more often took more work than I wanted to bother with.  It should be the other way around, being easy to pop on FM at the spur of a moment, or perhaps the beginning of the day, and then switch to some collected music in higher fidelity form when time and energy permits and desire calls for it.

Some time around 1985, I set up my most beloved FM tuner, which was then a Fisher FM80R which always sounded marvelous (though in mono, I didn't care) atop the side table near my listening seat.  This was a marvelous innovation, as once again I could tune in different FM stations without having to get up.  And that was particularly important then as there were two fairly nice, if still far from perfect, FM stations, but between the two of them I could usually find something good to listen to.  But generally I didn't start my system to play FM.  If I started it to play a CD, and the CD ended, I have to walk across the room and fiddle with several controls to get it to play the FM tuner right next to my listening chair playing again.  Which I rarely did.  But at least I did that more often than switching on the system just to play FM.

I spent much time imagining a system which would let me switch from CD to tuner without having to get up, or maybe do anything at all.  I imagined a system which would sense that the music was not playing anymore, and then automatically switch to the FM tuner.  I figured I could simply add some relays somewhere, and I think I purchased a couple relays, but that was as far as this kind of automationproject ever got, until now, 35 years later.

But even with these early insights, I could begin to see the essential ingredients in effective automation:

1) It should be convenient, eliminating the need to get up and do things, such as with a push button reachable from where you are sitting without getting up.  It could be even more convenient, such as turning on music from a background source automatically when the primary source has stopped, or perhaps even when you get up in the morning, etc.

2) It should simplify, reduce, or eliminate the need to make decisions about what to play next.  Channels such as FM radio stations can do this, as do streaming channels or Pandora.  (One complication may be that FM stations are not always good to listen to and you may want to choose another FM station or source.  I am very fortunate in having a very good listener supported Classical Music FM radio station I love.  But it's not merely good fortune, because it's also something I've contributed to for about 30 years.  Some sort of personal contribution is probably necessary to make any such channel good.  Recently is the second time in 25 years I've been nagging them to clean up their sound, now they have a digital skipping problem which is apparent on both their analog and digital feeds.)

So the two essential ingredients are (1) reducing/eliminating need to do stuff, and (2) reducing/eliminating the need to make decisions.  Frankly I believe that it is the decision making that is the hardest of all.  Yet I've never heard other audiophiles even think about it.  Perhaps it's something that's uniquely hardest for me.

Enter Sonos

I may have thought about it the automatic music selection problem, but didn't do much until I learned about the Sonos system in 2006.  I quickly obtained a Sonos ZP80 and CR100 and a Windows XP machine to work as my Music Server.  Sonos didn't originally support Macs, and my Mac was ancient anyway.  These support uncompressed digital over wireless or ethernet, produce bit perfect (when no volume control or other features used) reproduction over their digital outputs, and have fairly decent 44.1 uncompressed digital analog to digital conversion for one analog input.  I wouldn't bother with their amplifiers, speakers, or other devices.  I have "high end" component audio systems in every room.

With Sonos, I could control everything (except turning my components on and off) with the convenient CR100 controller, which I kept on my bedside table.  Up until 2009, my primary audio system was the one in the bedroom.  The living room stereo was lower spec components I used only when playing the living room TV, which I only did for parties.  In 2009 I had a dental near death experience which opened me up to splurging on my dream system with electrostats and subs, as I imagined in 1992 when I bought the house.

Thanks to its combination of hard and soft buttons, the CR100 still represents a high water mark in semi-automated audio convenience.  I rued the day that Sonos no longer supported it.   Smartphones are not a very good substitute because you must do a lot of fiddling just to open the phone, select the Sonos app, and then get it to do some audio thing.  This is by necessity a two-handed operation which may even require putting your glasses on, and often results in failures and curses for an endless number of reasons, such as unsuccessful overnight smartphone update.  With the CR100 I could simply reach over and press the Play button to start when I woke up in the morning.  No securanoia.  And then press the Pause button when going back to bed.  Sonos occasionally did do updates, but back in the early days they went quickly and always succeeded.  No passwords or passcodes were ever required.

One of the reasons I bought Sonos devices was that I really lusted for the Random play mode, which Sonos had from the day I bought it.  I imagined this would finally solve the Selection problem, which had always been the key stumbling block to having music playing all the time, or even very often.  The other reason was line-in, which enabled me to play the FM tuner, which only worked decently on an indoor antenna in the living room, in the bedroom.  Running a cable between the two points was highly impractical.

I soon discovered it was very useful to organize my music files not just in one folder but in multiple folders.  When Sonos became available on the Mac, I created one folder for Classical and Ambient music suitable for background music in one folder, the original iTunes folder actually, and all the Rock and Jazz music which was too loud, jarring, or distracting for background music in another folder, which I called altTunes.  I imagined being able to play the background music randomly forever.

This seemed like a dream come true at first.  I also added an X10 Appliance Module* to start my power amplifier so I wouldn't need to get up from bed to turn it on.  The subwoofer plate amplifier has a music sensing auto switch.  

For a few years after 2005, X10 appliance modules were still working in my house.  But about then they started becoming more and more unreliable because they are sensitive to the noise created by DC power supplies, particularly things like cell phone chargers and computers.  The noise created by DC power supplies blocks X10 transmission.  There are various means of fixing this, such as using line filters and X10 signal extenders.  I clung to X10 for another 4 years accumulating all sorts of additional equipment like filters, extenders, and X10 signal vs noise meters.  Finally around 2014 or so I abandoned X10 for Insteon, which has been my chosen home automation device brand since then, working very reliably after being not-always-easy setup, and for which I also use a Universal Devices ISY994i controller, which is key to the way I do things in Home Automation now, and gets my highest recommendation of all.  It allows very flexible programming which I would not be happy without.  Universal Devices supports Insteon and some other automation device standards.

But back in the day, I could have done just about as much with simple X10 modules as with Insteon (or other systems) now,  if only X10 had just kept on working, using the fairly open computer programs and interfaces available.  Notably I do not at all believe Scenes are necessary, or in most cases even useful, except that certain things with Insteon (like full control of the lights on touchpads) require you to use Scenes.  The old Macros I used to cook up for X10 could have been about as useful as the combination of Programs and Scenes I use on Insteon now.  In my view, this is another area of geekdom where everyone says you must have this shiny new idea, but actually the old approach could have been made to work almost as well.  I think I would actually have hated Insteon because of all the more complex, but closed and ultimately more limited controllers made available from Insteon itself.  The magic piece is the ISY994i, which opens up the system to more capable programming.

I show the Living Room wall controller at the top.  This is right by the front door.  Most often I don't much bother to label the wall switches and mini controllers because I know which button does what (including a few choices that require 2 button presses).

Bedside Insteon 8 Button Module (unlabeled)

*****

But soon I found that simple "random" song playback as implemented by Sonos was unacceptable for various reasons.  Despite my using separate folders for different kinds of music, so I could have all classical/ambient music which is suitable background music in one folder, and loud talking music which demands your attention in others.  

Some of the reasons included:

1) The Sonos random play seemed always to start from a particularly tedious small file.  From this I concluded that Sonos implemented a deterministic randomization procedure.  It is much better for this purpose to seed the random number generator from a non-deterministic or at least less-deterministic source, such as clock time, or as exists in Unix and MacOs, the number of clock clicks since startup (which is a larger number and therefore better than the number of seconds since starting).  Then it should never start the same way or repeat the same random pattern.

2) Also the fact that Sonos random play started from the same song revealed another fault.  It implemented a simple with-replacement randomization strategy.  Thus no matter how many times a thing has been played before, it would still play as the first song at the beginning of a deterministic random play.  The with-replacement strategy also means you will hear other songs you had heard at the beginning of the last random play.  (They might have fixed this since then.)  A with-replacement strategy also has many other faults.  One is that there is no guarantee the same song won't be played twice in fairly close succession.  A Square Root of N Rule applies: you will hear a repeat about every square root of N times, where N is the number of songs in your library.  Finally, sometimes, without-replacement search rule is implemented, sometimes it may be lost every time there is a power cycle or power failure.  So then you're back to the same only forever heard songs.  Finally, the Square Root of N rule also has a dark side: as well as some songs being repeated over and over, there seem to be an endless number of songs which are never heard at all.

3) In my opinion anyway, whole album song play is better.  In other words, albums should be selected randomly, then all the songs in that album played in their correct order.  (If an album contains multiple Titles, it might be OK to play per title, so long as the titles are not too short.)

So while it seemed like Sonos random play ought to be the new millenium in automatic music, it wasn't.  I quickly grew tired of it.  (Cynically you might wonder if that wasn't the plan.)  Before long, I had subscribed to Pandora and quickly upgraded to Pandora Premium.  Pandora did a pretty good job of selecting new music to my taste and in different categories.  However, I would never use Pandora as a Featured source, it was just some level of MP3 compression.  I discovered a lot of new-to-me artists on Pandora.  But sadly, eventually even Pandora became tiresome as background music.

I haven't used Sonos Random Play in over a decade.  All services push Radio Stations which i haven't yet found acceptable in digital form.  I like my local classical radio station KPAC best broadcast in FM Stereo.

Just as my 1980's automated background music idea centered on FM radio, so did the beginning of my new one last year.

Actually, it started from an attempt to improve on the time shifting of my favorite program on KPAC, Performance Saturday.  Often I was at the Symphony itself, or doing something else on Saturday night.  So I recorded via in incredible path.  Starting from the indoor antenna in the magic spot for KPAC, to my Pioneer F-26 tuner, into the analog input of a Sonos Connect, through hardwired ethernet connections to my large DLink gigabit switch, on on the hardwired ethernet connections into the bedroom, through another Sonos Connect, and into my Nakamichi RX-505 and onto Chrome tape, with Dolby B.  This actually produced wonderful sounding recordings, I still say.  But I thought it could be better and simpler.

Plus, it would be useful to have a digital recorder deployed in the Living Room for making digital copies of LP records as well, and LP record tests (which is mainly what I have done so far, yielding hugh improvements in tonearm resonance control).  So, sometime in 2019 I purchased a Marantz PMD 580 recorder for the living room.  (Strangely, I was not happy with the digital recordings until many furthher changes.  Until now, maybe, it hadn't been as good as the previous system based on a Cassette recorder!)

But before I realized the PMD 580 had a crappy sounding analog to digital converter, even crappier than humble Sonos, I figured I was at the dawn of a new era in good sounding FM and recordings from it.  (Actually, I was at the bottom of a steep hill.)

While purchasing another impeccable Henry Engineering AES splitter from Markertek for splitting the digital from my Lavry AD 10 analog to digital converter off to the Marantz so I could use the better Lavry for making digital copies of Vinyl, and listen at the same time, I noticed another very interesting product from Henry Engineering.  A 3 way AES digital switch using relays.

This is a powered pushbutton switch.  You push one of the 3 buttons and it sets the relays for that particular input.  And the light on the button lights up (a very cool touch).

And, not only that, this 3-way switch can be controlled by the pins on a DE-9 connector.   That was the key ingredient to make automated background music from FM radio work.  Those pins work on contact closures which can be controlled by various means, but trivially easily from an Insteon home automation system using the marvelously flexible Insteon I/O module.  I use 3 of them (plugged into a  Hammond power strip connected to a non-audio AC circuit) for the 3 possible selections: Background Music (FM radio), Feature Music (Oppo), and Mute.

One absolutely essential thing for Automated Background Music is an independent volume control for the foreground and background sources.  You may have been listening to vinyl last night with the system cranked all the way up.  Or cd's.  The system has to set the volume for the background music source independently of that somehow to make automatic switching useful.

Modern Day Levinson preamplifiers feature something that would work.  You can assign relative and starting volume levels for different sources.  Sadly few other high end preamplifiers, or any preamplifiers, have such a feature.  My Tact digital preamplifier, just as costly as a Levinson when new, does not.

But using the Henry Engineering switch at the to switch the input to my miniDSP crossovers from the Tact to a new separate digital converter for the background music source, and I could use a Behringer DEQ 2496 to fine tune the digital level, aka volume, specifically for the FM tuner.

So that was the answer for me, with zero degradation, to switch the AES balanced digital stream (with relays) from one path to another.  I needed to create a new digital conversion path for the background music source, my Pioneer FM-26 tuner.

I had originally figured I'd use the Marantz PMD-580.  But by the time I'd purchased the Henry Engineering switch I'd already decided the PMD-580 conversion just didn't sound good enough.  I couldn't use my Lavry for this because if I were using the Lavry for something else, such as listening to Vinyl, then I couldn't just go to the FM signal at the push of one button.  Many changes would need to be made...changing the source input on the XSP-1 preamp, and changing the level.  I still can't automate that.

But in my closet I had a never-used-by-me nice looking (and it turns out, very nice sounding) Analog to digital converter.  It's a Black Lion Sparrow (now known as the Sparrow 1).  And in my testing it sounded even as nice as the Lavry on FM when buffered and stepped up slightly by minty Musical Fidelity X-10 V3.  The AES digital from the Sparrow then feeds a Behringer DEQ which trims the level.  Levels from -15dB to -6dB are useful, I generally like -12dB or so for background.  Sure I lose resolution by this attenuation, but only in the context of 24bit audio transmission.  Subtract 12dB from the 24 bit resolution of 144dB and there's still 116dB.

Sparrow ADC shown beneath Sonos Connect in crowded back of shelf

Marantz reccorder, Behringer EQ, and Musical Fidelity Buffer

The Fixed Output of the Pioneer F-26 FM Tuner goes to the Musical Fidelity X 10 V3 tube buffer.  The unbuffered output goes to a Sonos Connect, the buffered output goes to the Black Lion Sparrow Analog to Digital converter running at 48 Khz.  The AES digital output of the Sparrow goes to the Behringer, which fine tunes the output level in digital usually with about a 12 dB reduction.  The adjusted digital signal then goes out from the Behringer in AES balanced over mogami cable to the Henry Engineering 3 way AES switch.  The switch feeds an AES splitter which feeds the 3 miniDSP digital processors that make up my "crossover."  The Tact is completely bypassed when I'm listening to the tuner, and therefore the Tact digital level control has no effect on FM, only the "Feature" source.

The Optical Digital output of the Behringer is converted to coax by an adapter and fed to the Marantz PMD, which also accepts an balanced AES digital from the Lavry I used when digitizing vinyl through a Henry Engineering digital splitter (the other path is for my stereo itself).  In no case do I use the analog inputs (and therefore the analog to digital converters) of the PMD 580 which I consider inferior and grainy sounding.

This new pathway for the FM tuner, combined with now using an outdoor antenna run through the Kitchen have improved the sound of FM immensely.  This, and the electronically controllable an lossless audio signal switch, and the independent level control, were the essential ingredients of an automated background music switch.

But you'd still need to turn the power amp on and off.  I find, that on the living room circuit, and Insteon On/Off module works flawlessly, adds zero noise, grips the outlet and the plug very tightly, and has no detrimental sound quality impact.  This works perfectly with the Hafler 9300 which is well within the ratings of the device.  It did not work so well with the Aragon 8008 BB which has 2.2kW of primary power transformers.  The Insteon is rated at 1800W.  Strangely, however, in the radio frequency tunnel of my kitchen, the 2 inches of unshielded wire inside the Insteon On/Off module picks up noise big time, like you can't ignore it. That was true with unshielded power cords used there as well, on an otherwise equivalent Hafler 9300 I use for the Revel's in the Kitchen.

So now I can turn on the Living Room stereo to play the FM tuner, regardless of what it was playing just before, and even turn the amplifier on if needed through 3 wall keypads in my house and keypads by my kitchen chair and bedside.  And mute or turn it off too.  It turned out that when the Henry Engineering switch is set to an input with nothing conveniently, very conveniently the 3 miniDSP crossovers which are in line with it's output (through a Henry Engineering AES splitter) mute perfectly.  Even when the Hafler power amplifier is turned off the muting is important to keep the subwoofers and supertweeters from playing.

And then the third button (and corresponding Insteon I/O module) selects "everything else" other than the FM tuner.  I originally called the FM tuner input Background Music and the "everything else" input "Feature Music," thinking like a Featured artist is why you go to the concert, the Feature is what you must select, whereas the Background is preselected.

However it did not occur to me that the term Feature Music has the unfortunate initials FM, which could be confused for my FM tuner used as background music.

Anyway, the Feature Music designation itself is somewhat obsolete since now my "everything else" can be preselected by my computer program, not actually human selected/featured.

Now on the Living Room wall keypad, I've designated the two selections as FM (as in FM tuner) and DVD (as in my Oppo BDP-205, which serves as my high resolution disc player AND streamer with SPDIF 24 bit digital output to the Tact digital preamplifier) even though that would also be the selection for listening to Vinyl (with a different selection on the Tact).  It would be nice if I could control the Tact as nicely as I do then Henry Engineering source switch, with contact closures.  The Tact has a proprietary digital interface that its program on a 2000 era PC communicates with.  I very rarely use that since I do not use the Tact's room correction or even measurement anymore.

But it was still a bother to get up from the Kitchen to switch the Oppo in the living room on to listening to harddrive and streaming musing through the Oppo 205 digital output which feeds my Tact.  Those work best using the DC triggers.  So I have a pair of Insteon On/Off Modules which power regulated 12DC adapters which are connected to the trigger inputs of the Oppo 205 and the Emotive XSP-1.  The Oppo is automatically turned on whenever I select "DVD" in any of the wall switches and keypads in the house.  The Emotiva is best switched on by the ON button of the other On/Off module.  Both are automatically turned off for "bedtime" or when the mute button is pressed a third time.  (The first press is soft mute, which permits background music to come on in one hour, the second press is hard mute which keeps the music off indefinitely, and the third press is system off.)  The two Insteon modules are connected to an all metal noise reducing power strip which is connected to the "main" living room circuit and not any of the 3 dedicated audio circuits.  I bought that strip for reducing the pops in my stereo system created by switching the Tensor lamp for the turntable and other purposes, but it is also good for giving Insteon modules a true AC connection (not going through my Panamax) and not polluting my line audio devices with yet more DC power supplies.

My Playlist Generator for Roon (which is my current music program).  Sadly it seems you can't trust streaming services, or even subscriber-funded Roon it seems, to implement a decent playlist program.  Like everything else, Roon is always trying to sell you new music.  That's what Roon Radio does.  I find that when any program (except Pandora, which did it fairly well) tries to sell me new music, I hate nearly every song it selects.  Especially the first one.

But, fortunately, Roon does provide a way to import playlists, including from simple M3U files.  I've recently discovered what a powerful idea this is.

What needs to be created is a Daily Playlist.  That's not too often, I think just right.  Thanks to using an across-session without-replacement randomization strategy (based on a stored dot file listing all the files currently played) combined with automatic-replacement-reset-on-exhaustion, you can be sure that today's Daily Playlist won't repeat something on yesterday's, or in my case going way back, except in the rare case that still occurs right after a reset.  To overcome the reset problem, an additional reset window is applied for N/4.  For the first N/4 of random selection, nothing in the previous N/4 is permitted.

It is also critical to use an unbiased shuffling algorithm, like Knuth's (also given other names).  The  most intuitive shuffling algorithm is biased, meaning it can only reproduce a limited number of the possible shufflings.  I believe this can lead you to just feel what is coming up next, and make everything seem dull.

Roon applies a very nice level compensation on an album basis to keep the playlist at the same level without compromising the musical integrity of any album.  Roon also nicely stops and remembers its position when the Oppo is shut down through my automated system.  (I keep it going until I hit the sheets at bedtime.)  However, Roon does not restart the playlist when I restart the Oppo.  I wish it did that though I'd have it apply only to trigger-starts not front panel pushbutton starts.  Now I have to go into a Roon interface such as on my computer to restart the playlist playing, which is very easy since it brings it right back to the position when the Oppo was shut down.

Roon playing playlist generated by my program "mplay"

Roon doesn't list track numbers any  more, which is a misfeature in my opinion and others.  Track numbers have been replaced by "play" buttons.  Somehow I remember calling for visible play buttons in the past.  Perhaps I even said they could replace the "useless" track numbers.  Well now I find those track numbers are not at all useless to a serious music lover, for which we play a hefty fee.  If I asked for this, I was wrong, and I am sorry.  Somehow the track number should be made into a visible "play" button, combining both concepts.  Or just bring back the track numbers and have a Play dropdown menu if the combo I describe is impossible for some reason.