Analog is Better (4 hours of mastertape listening)

Tonight I spent 4 solid hours listening to low generation mastertape copies, on a state of the art music playback system at the Newport Beach T.H.E. show.  Never have I been more convinced that analog tapes like these (15ips, 2 track, current state of the art mods to Tascam deck by United Home Audio) have more musical information than any kind of digital recordings, and especially DSD recordings.  They sounded wonderful.  The playback system other than the tape deck was all MBL's best, including MBL's best omnidirectional speakers, and floating bias Class A/AB amplifiers.  I would say it was generally the best reproduced sound I have ever heard.  A solid group of dedicated audiophiles like me listened to most or all of the presentation.

I continue to believe that we need more than 16 bits at 48kHz as some objectophiles argue would follow from the Shannon-Hawkins corollary to Shannon's Information Theory, and tonight's listening was further personal confirmation.  I was not the only true believer...all the chairs were filled for the 4 hours I was there.

I have never heard any form of digital recording sound this good.  I think that either Shannon-Hawkins is wrong, misapplied to human musical perception, or misinterpreted.

Today I also heard the Audio Note exhibit which features a pure PCM DAC, with no oversampling or upsampling or filterning.  It sounded good generally, though I like my home system (using Onkyo RDV-1 as DAC) better.  Although this demonstration of an Audio Note system was far better than the one I heard in 2009 (which I could barely stand due to harshness and distortion) and I thought their current DAC was more revealing in some ways like mine than currently typical sigma delta DACs, there still was a whiff of harshness. I think my DAC, which uses PCM 1704 DAC and 8x oversampling, is better--it has no harshness at all.

I think that PCM dacs sound better than sigma delta DAC, including all forms of DSD.  But it is not necessary to remove overampling reconstruction filters (used in the very first PCM products, including the first CD player the SCD-1).  In fact, digital reconstruction filters are perfectly fine and likely better than analog equivalents (though we might be better off with higher rate digital--at least 48kHz--so as to make the reconstruction filtering easier).  The fly in the ointment, I believe, is with the sigma delta DACs, and the theory which says they are equivalent to PCM dacs.  Is that Shannon-Hawkins again?  So far, they don't sound equivalent to me.  Sigma Delta dacs have a characteristic smoothness which seems to be information lossy.  DSD has this same property, but adds just enough noise to make the resulting presentation about equivalent in brightness to pure PCM, but with a slightly fake quality.

Now, maybe an all MBL system like the one I heard tonight would sound equally good with high rez digital like 24/96?  I don't know, but the last times I heard sound equally good--was at previous mastertape demonstrations at the 2009 CES.  That guy had first generation safety masters which he himself had recorded of musicians like Frank Sinatra.  Digital, in all the various forms I have heard it, doesn't reach that level, but high bit rate PCM comes closer than DSD or sigma delta digital.

Another odd thing...I thought the analog tape recording of Kind of Blue sounded much more like the LP record than the remasterings done by Sony after they developed DSD.  CD masterings in the DSD era have sounded much brighter and slightly harsh.  The analog tape sounded neutral but not dark, only slightly brighter than the 1985 CD version but way more transparent than any version I've ever heard.  Stuff which doesn't seem to make sense in other recordings sounded real on tape.

Sounding Good Now…System 5

I've decided to give the current living room system a special designation: System 5.  This counts back various ways I did the living room system since I got the Acoustats.  A friend agreed the system was sounding better than ever.

The current system 5 includes special DAC for the Acoustats, currently the Onkyo RDV-1.  For what it's worth.  I suspect the PCM 1704 used BB's DF 1704 for digital filtering.  DF 1704 is an 8x oversampling unit.  The combination of 1704 and 8x oversampling was also used by the Chime DAC, which friend Tim says is the best ever DAC he's heard.

One of my subs has the Sledge Amplifier, which may sound more tuneful than the BASH.

I've re-tuned the bass EQ, not entirely eliminating boom around the perimeter, but letting enough through so bass tunes are not lost, and have a bit of weight and authority.  This is judgement call, but basically -8dB at 45 Hz instead of -11dB.  Plus the bass level is raised 1-2dB from before.  I dialed back in 4dB of 30Hz boost eq on one side and 2dB for the hallway sub (which seems to have large excursions…and dialing back 30Hz eq to only 2db helped).  The idea of 30Hz is that it adds about 2dB weight to 32 Hz, but also gives boost to the 20's, which aren't overdone here (they show up as very weak strangely on Android app).  The Q is 1.3 to 1.6, not sharp.  The corner sub also gets 1dB of boost at 20Hz.

Basically, the Bass is boosted slightly compared with System 4, which was max cancellation of boom.  But often turned off by mistake, because max cancellation was not very satisfying.  I'm also using LR24 vs LR48 crossover by necessity, since that's all I can create using DEQ for crossover which lets me use external DAC.  But the use of external DAC has one thing I keep forgetting to mention, perhaps because it reveals a weakness of previous DCX systems I didn't want to mention, perhaps.

The problem is that the high max output of the Behringer DCX 2496, 10V,  means that you waste an extra 10dB of digital resolution into home audio components.  This may not be a huge problem, looked at the standpoint of the 24 bit DAC's having more resolution that you might ever need: 144dB.  But of course they don't provide quite that much actual dynamic range, or S/N, or things like that.  Say if the converters have 120dB dynamic range, using them at -10dB and below means you effectively have only 110dB.  And so on.  So this is an important consideration.  The max digital level should correspond to a max output that you never get to, but come fairly close to.

Many people who hook the Behringer up differently complain about the noise, or whatever, resulting from the high max output.  But using digital inputs, I was not getting noticeable noise, running straight into my midrange amplifier, the Aragon 8008 BB driving the Acoustats.  The problem, if there was a problem, was not so obvious as hiss.  It was sound with less there there than there should have been.

Many people have solutions like external attenuators (which I use on the bass, but determined did not sound good on miss or highs).*  (The DCX outputs get increasing distortion above 2V output.  They are excellent 2V and below.  But near 10V distortion rises to near 0.1%.  That's what you see when you use an external attenuator.)  Or transformer outputs (I eschew ferromagnetism in the signal path).  Or a re-done output stage…YES that's what this unit needs.  I would say the max output should be set to 4V.  2V doesn't push a lot of amps to their maximum.  And the 4V output should be clean.  0dB digital should correspond to 4V output.

With my system 5, I'm getting about 2V max output in the midrange at 0dB digital.  This seems to play about as loud as I actually need, though it scares me that I might run out of digital headroom.  Anyway, I'm not wasting 10dB of dynamic range as I was before.  And that could be as big a difference as having an R2R DAC.  Or not, or both might be unimportant.

Anyway, the overall effect is a far more musical system than ever before.

NOS Dacs

I don't know how much oversampling is used with the Burr Brown PCM 1704 in my Onkyo RDV-1 I use for decoding digital for my Acoustat speakers.  The manual doesn't say.  I'm inclined to think that one of the advantages of R2R Dac's is that they don't require the huge amount of oversampling that Sigma Delta Dac's require.  (Even if they are not used in an oversampling circuit, Sigma Delta Dacs have at least 64x oversampling and usually much more now built in, that's how they work.)  But it may also be that R2R dacs simply have more resolution at each point they generate, and they sound better because of that, and the oversampling as such may be immaterial.  Sigma Delta DAC's have 1-5 bits of resolution, and all the rest is made up (faked?) with oversampling.  One other question I have: do Sigma Delta DAC's use feedback to get correct results, or is the oversampling all that's needed?

Update: I think it's quite likely Onkyo RDV-1 uses 8x oversampling, because Burr Brown made a digital filter, the DF 1704, specifically designed for the PCM 1704.  And the DF1704 does 8x oversampling.  It also accepts inputs up to 96kHz.  So it's a perfect match with what I've got.

Anyway, now that I've started to go off the Objectophile ranch (they say that all DACs sound the same, and also that there is no reason why R2R DACs should sound better, and they could be worse because they have poorer measured linearity, and they don't like that idea that you would go with something that measures worse because of your superstition, though they still maintain they all sound the same) with a R2R DAC, I've been thinking about the non oversampling approach too.  Many appear to confuse the two concepts, oversampling vs non-oversampling, and ladder R2R vs sigma delta.  But most ladder R2R Dacs (as equipment) do use some oversampling, they only raise the possibility of getting away with none.

Here's a great thread which lists a bunch of non-oversampling Dacs (equipment, not chips) you can buy.  Some of the ones mentioned at semi-affordible prices are:

47 Laboratory Shigaraki Series Model 4715 DAC
Audio Note DAC Kit 2.1, DAC Kit 3.1.
dB Audio Labs Traquility and Tranquility SE
Metrum Acoustics Octave and Hex
MHDT Laboratory Havana
Mojo Audio AD1865 NOS DAC
Promitheus Audio Solid State DAC and Tube DAC
Red Wine Audio Isabellina
TeraDak TDA1543 NOS DAC V3.1D
Wavelength Audio Brick v2
Vertex AQ Alethia DAC

Another one, sometimes considered a benchmark, is the Light Harmonic Da Vinci 384k ($20,000), not to be confused with another maker's Da Vinci product.

TeraDak also OEM's other products, like the Valab 1543, and those two units are among the least expensive NOS DAC's.  I don't like the fact that they use transformers for I2V and output coupling, though it is one way to get around the ultrasonics (a problem with a lot of NOS DAC's).

DAC's like NuForce and Peachtree claim to be NOS, but use Sigma Delta DAC's which use oversampling internally.

Here's a DIY thread about NOS DAC using 1541A.  This thread at DIYAudio has become quite popular, growing to over 500 pages!  NOS DAC's may be mostly "underground", but they are a very hot item among a certain class of DIY and other underground audiophiles!

I remember seeing one NOS DAC tested in the pages of Stereophile.  The objective measurements didn't look good.

Update: I believe I have now read some where that Sigma Delta DAC chips do indeed use feedback to get linearity.  So at the end of the day, this is glorified mud!  Throw enough mud on the wall until it weighs enough.  Not too much different from PWM systems.  I'm convinced now.  Give me real PCM!!!  Oversampled or not, it's still better than mud!

After time alignment, new DAC sounds even better

It turned out not to be so hard to set the time delays I needed to time alight the Acoustat panels with the subs and super tweeters using the Behringer 2469 DEQ I now use as crossover for the Acoustats.  I had previously been fooled by the fact that when you bring up the time delay dialog, and turn the big knob, it advances the "distance" (how it measures delay) in large 0.6 meter increments.  But on Wednesday night I discovered that pressing the knob in once (until it clicks) changes this default increment to 0.01 meters, which is good if not as fine as I would like (0.001 meters).

So I was able to dial in the same delays I had previously used in the Behringer 2496 DCX.  I didn't have to check my notes or this website, I just checked the DCX itself by turning the delays back on and checking the numbers I had added.

After this and other adjustments, another long unplanned listening session ensued because the sound was so good.  This time I also listened to different music, the DVD-Audio recording of Santana's Supernatural (one of my favorite sounding recordings of all, which previously I had been thinking was an SACD and therefore proof that SACD's could sound especially good…but then later pulling out the disc again, I see it wasn't an SACD after all..and now I can't think of a single SACD that is categorically better than CD, though I can think of lots of DVD-Audio discs which are).  I did not hear any of the dryness I had heard the previous night when listening to Pink Floyd Wish You Were Here, in 16 bit version through Sonos.  But I did continue to hear the marvelous dimensionality, layered imaging, and super high resolution I noticed the night before, and which I associate with the new DAC.

Additional tweaks I made were that since the new subwoofer is slightly farther back than before, I increased the panel delay by 0.05 meters (a couple inches) to account for that.  That seemed reasonable to do without measurement, but I will have to re-measure with Tact to determine the exact delays needed.  I also increased the delay on the tweeters by the same amount on average, but while doing so I made both tweeter delays the same (I hadn't remembered setting them different, but apparently at one point I had).

I messed with relative levels too, bumping the sub and tweeter levels up a dB or two, and then also moderating the massive -11dB notch at 45 Hz to a mere -8dB.  Bass had been sounding sucked out before.  But even moderating the notch just this amount, brings back massive resonance in the corners of the room and in the hallway.  But it really made the bass more enjoyable at the sweet spot.  Until and even when I can notch out the 45Hz modes more precisely, this has to be a compromise, though I also have some ideas on how the two subs can be tuned slightly differently as another strategy.

I eliminated all the sub boosts below 32 Hz on the left side sub.  They were making the bass sound congested, especially on another favorite of mine, Bass Erotica by Bass Connection, which I played on Thursday night.  Later, though, I dialed back in a 25 Hz boost on the left side in order to remove it on the right, and tuned both subs to have 2dB boost at 32 Hz.  It seemed light the right side sub was straining and shaking too much, while the left side sub, with new amplifier, seemed placid (despite having levels matched by playing Stereophile Test Disc 2).  Playing Bass Erotica was shaking walls, but I went outside to the sidewalk and nothing was audible there, or close to nothing, while the sounds of wind and AC compressors could be quite loud.  It seems like at least 15dB reduction from outside the front door to the front of the garage, and perhaps another 15dB reduction from there to the sidewalk.  I figured I could play even 10dB louder before reaching 40dB at the sidewalk.  I was playing with level at 82 (about -11dB digital) and inside levels were surprisingly low on Bass Connection…about 70dBa peak, but this recording saves all it's dynamic range for the deep bass, which doesn't show up as loud as it sounds in A weighting.

I tried temporarily rolling off the panels less by canceling one of the two 80Hz 12dB high pass cutoffs, then dialing it back in at various frequencies below 80Hz, then ultimately going back to 2 80 Hz cutoffs (for proper Linkwitz Riley 24dB/octave) which didn't sound any worse, and possibly somewhat better.

I also checked the 2nd bedroom (Queen's Room) which has "soundproof" wall and door, and you could hear that bassy music was playing, but someone not inclined to make trouble could just as well sleep through it, I decided.  So while I wish my soundproofing was better than it has turned out to be, it seems to have reached a minimally acceptable level of performance.  (Before the soundproofing, that wall would be rattling constantly at these levels, and the door might as well not be closed.)

So the new setup has been time aligned, tweaked, and tested, and it sounds good, and won't bothers others playing to 5am (as I did on Wednesday night).  A great success.

System back together, DAC's sound different?

A weekend earlier I took the pile of components on the left side of the living room stereo apart and moved the left Acoustat speaker out of the way so I could replace the plate amplifier in the SVS PB 13 Ultra subwoofer.  The original BASH amplifier had died after 6 years of usage.  I had already purchased the replacement Sledge amplifier, and already had it on hand, but it could not be connected to the speaker wires of my PB 13 because the connectors on my wires were too small for the new amplifier.  Disappointed about being unable to put everything back together, I moved on to other household issues, like cleaning the master bedroom floor with Scooba for the first time in 5 months.

I emailed SVS about the dilemma and they apologetically shipped me the "Bash Adapter" during the next week, so last weekend (May 18)  I was finally able to replace the subwoofer plate amplifier and put my system back together.  I noticed that the new Sledge amplifier has hiss that you can hear putting ear next to speaker.  I had noticed funny noise on that side of the room before but didn't remember that the subwoofer had been the cause.*  I checked the right subwoofer, and it also has some hiss, but about 4dB lower and with a lower frequency spectral balance.  This convinces me that I should be using external audiophile grade Class AB amplifiers for the subs, but it will take some time to develop and deploy an external amplifier solution with the required high pass filter and limiting.  (*What I remember now is that many times I heard faint noise or hum on the left side of the room, and couldn't figure out what it came from.  Perhaps one or more of those times I actually figured out the subwoofer was the cause, but that never registered permanently in my memory before.  It wasn't easy to disconnect the subwoofer from the crossover because of the deep corner location.  t also determined falsely or not many other presumed causes, such as an echo from the refrigerator in the kitchen, or the neighbor's AC compressor.  Now I have proven that the subwoofer makes audible hiss, and I don't think that is good, though a vintage amp with hum--as many of my older amps do--could be worse.  The hiss can't be measured because the background noise level in the room doesn't get low enough.  But it can be heard unambiguously with ear close to the speaker.  So there--measurements don't show everything!)

Saturday I went to a meeting of the River City Audio Society, had dinner at Earl Abel's with my lady friend, heard Beethoven and Prokofiev at the San Antonio Symphony, went to a reception at the Plaza Club, and then had an extended conversation with my lady friend mostly about my money borrowing and spending habits--I believe in borrowing and spending money on good things, within reasonable limits, it's good for me and good for the economy, and I will not be shamed or feared to worry about my small debts--that only adds to the problems of others through a deleveraging spiral.  We had some wine but she wanted to go home, but then 30 minutes later she called back and the conversation continued until 1:30am.  A wonderful long day, steam letting and all, but no energy left for working on subwoofer or stereo.

On Sunday I slept late, mowed the lawn, trimmed around the house, lady friend helped with dead tree branch, and then I had dinner until about 9:30pm, felt exhausted again, so took a nap until 12:30am.  By 5am I had bolted in the new subwoofer amplifier and had the system back together in a new way, though I had to cheat on one of the new digital connections.  For some reason, the optical connection from Tact preamp to Behringer DEQ (now being used as crossover for the Acoustat panels) didn't work, so I used an XLR-to-RCA adapter to permit connecting the DEQ via SPDIF coax.  The sound was obviously very detailed, but somewhat dry sounding.

On Monday the AES/EBU splitter had arrives, so I used that to more properly connect both the Behringer DCX and Behringer DEQ to the XLR AES/EBU output of the Tact.  With that configuration I was ready to do some serious listening to Pink Floyd "Wish You Were Here" over Sonos.

The sound was much higher resolution than I have ever heard before!  I heard shadow vocals, and multiple vocalists, where I had never done before.  There's a lot of stuff going on in the background on this album too.  I've listened to it hundreds of times, but this time was far more interesting.  I wasn't planning to listen to the entire album but couldn't stop.

At the same time, I wouldn't say I liked it better in every way.  There was still a peculiar dryness to the sound.  I wouldn't say it was harsh at all, just dry, but perhaps dryness is the first level of harshness.

Some of this may be that I really haven't gotten things set up yet.  Since I went a week without subwoofers, I turned off all my time alignment delays in the crossover.  (Normally, the panels are delayed the most since they are the closest.)  Now, with the new DEQ serving as crossover, I need to do the time alignment differently because the DEQ doesn't have the precise delays possible in the DCX.  Instead of having zero delay on the subs as I did before, I'll have to make a bigger delay on the panels and then do the fine adjustment with the delay on the subs.

But my guess now is that some of the dryness of the sound with an R2R DAC is unavoidable.  I think this is very much as with scaling old fashioned NTSC TV (480i) to HD (1080p).  The upscaled image often seems much smoother, and has no flicker.  But the upscaling can't possibly add resolution.  What it really does is reduce the resolution, just slightly.

Sigma Delta DAC's work by a process much like rescaling.  The input is oversampled to a much higher frequency, 64x minimum I believe.  In that way, sigma delta DAC's achieve great linearity on test signals.  But just as with rescaling video, some resolution from the original recording is unavoidably lost.  The result is a smoother sound, as if you had run a spline through the original 44.1kHz samples.  But that covers up the fact that there is less resolution still there.

Now I believe my Onkyo RDV-1 does do some oversampling anyway, just as most 1541 based CD players from the 1980's used oversampling too, typically 4x.  (Even Sony's first CD player used oversampling.)  But it can't possibly be doing 64x oversampling (otherwise it would have been DSD friendly).  Probably more like 16x or 32x.  Mainly, the Burr Brown 1704 chip achieves a large range of linearity the old fashioned way, with R2R ladder.

I still hope that as I re-do the time alignment, or make other changes, I will get some of the smoothness back.  But even if I don't, I am already hooked on the increased resolution.

IR over coax to emitters in multiple rooms

When playing FM tuners in the living room elsewhere in the house (through Sonos…that's one of the best features of Sonos is the ability to play sources in other rooms) it would be nice to have remote control over those tuners.  Several of my top tuners have Infrared (IR) remote control (Yamaha TX-1000, Kenwood KT-6040, Kenwood L-1000T).

I've tried this before…many times over the years.  It's been hard, very hard, with Radio Shack Remote Extenders (which have great difficulty sending signals more than 20 feet in my house).   The master bedroom to living room is the longest stretch, about 35 feet, and the Radio Shack extenders can't go that far (despite relative lack of big metal stuff in the way).  So long ago, I cut one of the LED wires and spliced in a 75 foot zip cord which still runs from the living room stereo all the way to the Radio Shack "Receiver" (using Radio Shack terminology, that means the device which receives the RF signal and emits IR, or allows you to attach an LED on a wire to emit the IR) in the Kitchen.  That specially located Receiver in the kitchen picks up the Transmitter in the master bedroom very well, especially now that I (many years ago also) attached a big external antenna to it (which is connected straight to the PC board inside the Receiver) mounted inside a kitchen cabinet.  Then I found that attaching this big wire degraded the direct IR output so much it couldn't control all the video components in the kitchen anymore.  AND, it didn't do a good job of getting the IR into the living room either.  This was going to be the big "solution", but after I did all the work I found it didn't work.  (It had seemed to work in mockup.)

And that's only one little story out of 100's in my long standing efforts to get the Radio Shack Remote Extenders to work as advertised.  If they actually did work as advertised (100 foot range or whatever it is they claim), I could control equipment in all different rooms in my house from any one room in the house.  That's "many to many", a kind of holy grail in home A/V distribution.  But I've never gotten to do so in more than 2 rooms at a time, and never the living room.  The problem appears to be that all these systems (not just Radio Shack, but all the RF remote extenders) are limited to a paltry 10dBm at around 400mHz, which barely rises above the RFI in my house.  I see now there's one with the same 10dBm which claims 325 foot range.  (I'm tempted to try it anyway, but I've already tried about 10 such systems, and they all work poorly, despite what friends and other say.)

I should have gone to a more "professional" IR distribution system long before all this.  But now I'm finding out that isn't necessarily easy either.  Even though now I have all rooms professionally wired with many different kinds of wire (as of last July), and I have currently unused coax running to all rooms.  Many IR distribution systems run on coax.  So I should be set, right?  Now it turns out that most such systems are only designed to emit IR in one room only.  Here's message I just posted to AVSForum (which has a Remote Control Area):

For IR over coax, I've looked at Xantech, ChannelVision, Channel Plus, and others. None of them show complicated systems, usually the the most simple system possible or close to it. The older Xantech manuals (not the latest ones) show reasonably complex setups, with IR receivers (targets, etc) in multiple rooms. But nobody shows LED emitters in multiple rooms. That's what I need. One recent Xantech manual I've seen has a disclaimer at the bottom that systems more complicated than those shown require professional installation. Question: how much would that cost? I already have spare (currently unused) coax running to all rooms. I worry that professionals wouldn't be interested in just adding IR to my existing system, I would think they like to do whole systems from the ground up because that makes them more money.

Xantech seems to have some of the best quality hardware for IR in general, and since I already have some of their stuff I'd like to stick with Xantech if possible. For coax distribution they make the INJ94 injector and CPL94 coupler (which permits the direct attachment of 4 LED's in 2 strings). It looks like you can have as many INJ94 injectors in a coax network as you'd like, as they each work independently. But nowhere in any Xantech documentation have I seen it say how many CPL94's you can have on the same coax. I'd suspect the answer is: only 1. Now you can have more than 4 LED's, but only if you have CPL10 (a therefore essential part that Xantech seems to be discontinuing) attached to a connection block, and the standard connection block can take 8 LED's (in 4 strings) and you can get amplified connection blocks that handle more.

Given what I know, I wonder if 2 CPL10's on the coax network would work better than 2 CPL94's. I'm thinking a CPL10 might put less load on the line (or injector) than the CPL94. But it's exactly this sort of information which is not available.

My current system needs emitters in two rooms, one room with 4 LED's and another with 8 LED's. I can imagine a future system might need emitters in a 3rd room as well. 4 rooms need receivers as well, but that part is well documented. This is many to many whole house system. Video is already handled with HDMI over Cat6 with a 4x4 switch, and audio is handled through Sonos. What I need to control in non-central rooms is mainly additional audio sources run through Sonos.

Another question: can IR over coax co-exist with composite video, or does video need to be modulated into a vhf/uhf channel?

Kenwood KT-6040: incredible resolution and liveliness

Last weekend (first Saturday in May) I modified my Kenwood KT-6040 FM tuner by replacing the 50uS deemphasis capacitors (since it was a European model…KT-6040 in US version so rare it may not even exist) with the 75uS deemphasis caps needed to bring it to the 75uS deemphasis required in the USA.  On Thursday night, I hooked up the newly modified turntable to my Sonos audio network.  The sound has remarkable transparency, resolution, and liveliness, listening to the KRTU college radio jazz station.  It sounds like it has more resolution, depth, and spaciousness than you might get from typical CD's.  It is very fun to listen to, and by the morning I was playing the station in all 3 Sonos equipped rooms in my house as background music.  Bopping away.

Previously used tuner on same antenna was Yamaha TX-1000 (US model).  It also sounded very good, and possibly less noisy, but not quite as transparent as typical CD sound.  Nice, but not as compelling.

The noise on KRTU is something like a background tone in both channels but out-of-phase.  It's not the pure 10kHz tone I've determined to be a station fault on KPAC, but it could be a similar kind of thing (intermodulation of the 19kHz pilot with subcarriers).  It's sounds a bit like (but clearly isn't) the groove noise you might get from a worn out record being hit with metal stylus and 20g stylus force.  It becomes obvious in quite parts, such as between words of the DJ, but then gets hidden in louder portions of the music.  I now believe this noise might be ameliorated with a better antenna (unlike the noise on KPAC, which doesn't go away even with antenna near the transmitter).  Given that it sounds louder than with the Yamaha, I'll have to check if I was using "high blend" or the like with the Yamaha.  Or it could be that it's more obvious on the Kenwood simply because of greater transparency.

Despite the noise, and occasional sense of harshness (probably resulting from the noise) I still like the Kenwood sound much better because it sounds so much more alive.  This is not merely a tipped up frequency balance in the highs (though the frequency balance might be tipped up a tiny bit too).  I'm considering this to be a big improvement, with only minor qualifications.  And I think I can hear the "teflon" sound in the super high resolution and spaciousness.

This project had been delayed for something like 4 years.  For quite awhile, I used the tuner with a Behringer DEQ 2496 to correct the deemphasis externally.  (I figured out exactly the required shelving filter needed to change the deemphasis…and this worked, though hardly elegant or optimal.)  Also, I used other tuners a lot, including the Pioneer F-26 (which now is assigned full time duty for the classical radio station…since it sounds so good on that tuner, better than the KT-6040 with digitally corrected deemphasis for sure).  Then, for over a year, I enjoyed the stellar Kenwood L-1000T which beat all my other tuners on every station, and it has two antenna inputs so I could use both antennas.  I took the Kenwood offline when it seemed to exhibit some power supply instability.  Most recently (well, just the past 2 weeks) I was using a Yamaha TX-1000 for the small college radio stations (keeping the F-26 on classical music station).  On the antenna that works best for the college radio stations, the TX-1000 is remarkably good sounding.  (It sounded horrible on those stations using every antenna position I tried in the master bedroom.)

But the plan had been to get the KT-6040 online for college radio duty asap.  I first hoped the job could be done without new parts and gave it to Luther for modification.  Luther however informed me that parts would be required, and that led to the most recent phase before Saturday…getting the needed parts.  Friend Tim determined which capacitors I needed to change and to what value (the schematic was very hard to interpret because of multiple country options).  What I needed to do was change 1500pF capacitors to 2200pF, right at the output of the MPX chip.  (This was not even suggested by the schematic.)

I first came across some Russian FT-1 PTFE (aka Teflon) 2200pF capacitors.  They were cheap enough that I straight away ordered 5, but with the fear that likely I would not be able to use them.  If they had steel leads, which Russian teflons (created for use in military equipment) were often known as having, I would be better off using something else, even if not quite as good as Teflon.  Then I also ordered some Phillips polystyrene capacitors, which I felt sure would not be using any ferromagnetic parts.

It took over 6 weeks to receive the Phillips caps from the seller in UK.  I got the Russian caps somewhat faster from Romania.  Actually, when I first saw that I was getting a shipment from Romania, I figured it was the custom turntable base I had ordered from Moldova, which at the time I was far more concerned about.  I missed the delivery attempt on Friday and went down to the Post Office on Monday.  They couldn't find the package.  At first they suggested it might have been shipped back.  They would find out and call me.  They never called, so I went down again the next day.  This time I spoke to the Postmaster, and he said he would call his associate at the central post office and be sure it was held and returned to the local post office.  But I was too anxious to wait, so I went to the central post office right then.  The central post office said they did not do any kind of customer pick up. However, they checked again, and said my package had just been delivered!  (Before going to the post office on the second day, I had signed a delivery release form and attached it to the door.)  So I hurried home, and couldn't see the turntable base anywhere.  I checked my mailbox, and there were the Russian caps from Romania, which by that time I had believed were likely useless.

It wasn't until waiting several more weeks waiting for the polystyrene caps that I got around to actually testing the Russian caps with a refrigerator magnet.  Nothing on the test cap showed any sign of ferromagnetism.  The next day I obtained a stronger magnet from Radio Shack and tried again.  Still no sign of ferromagnetism.  Some online discussion at DIYAudio suggested the grey colored leads (which are grey all the way through) are a silver/copper alloy.  That's about as cool as it gets in audio, Teflon caps with silver alloy leads.

Still, I figured the polystyrenes might be safer, so I continued waiting.  Finally I got the polystyrene caps, just before my eBay buyer protection was about to run out.  And to my shock I found the polystyrene caps were definitely ferromagnetic, they showed strong attraction even to the weak refrigerator magnet.

I've been surprised at the many audiophiles and DIYAudio's not aware of the fundamental ranking of capacitor dielectrics with regard to dielectric absorption, and the importance of dielectric absorption with regards to audio.  I've known about this since reading Marsh in 1980.  Here's a more recent article which notes the decline in availability of some dielectrics, especially polystyrene.  The rankings are PTFE (aka Teflon, the best), Polystyrene, Polypropylene, Polyester (aka Mylar), and way below these polymer films are the electrolytic dielectrics aluminum oxide and tantalum oxide.

Enjoying the second FM tuner and DAC

It was a week and a half ago that I reconstructed (?) the new pile of components on the left side of the living room stereo.  (And it looks like it will soon be necessary to deconstruct the pile soon to install the SVS Sledge amplifier on that side…)  Meanwhile, I have been enjoying the second tuner playing college radio from KSYM and KRTU.  But it was not until last Sunday that I got the new ladder DAC actually hooked up.  And it's not really hooked up in it's intended configuration.  Since my subwoofer setup isn't fully working anyway, I simply hooked the DAC to the SPDIF coax output of my Tact 2.0 preamplifier.  Then, if I turn off the subwoofer, I'm running the Acoustats full range for the first time in years, with a new arguably state-of-the-art DAC (a repurposed Onkyo RDV-1 from 2001, but having the ultimate ladder dac chips, Burr Brown 1704's as found in top Levinson, Wadia, and others in the day).  Whee!

I suppose I should be seriously listening to high resolution audio sources, but instead I've just been continuing with my FM tuner and college radio, mostly, and mostly as background.  FWIW I think the RDV-1 running through full range through the Acoustats is very transparent and also tuneful.

Meanwhile I've been exploring various hookup options for triamplification once I get the subwoofer fixed.

Connecting the Tact to the Behringer DEQ and DCX at the same time for my new configuration could be done in several ways:

1) AES to DCX and Toslink to DEQ.  This was the most obvious method, since no adapters are required, and conveniently the Tact has 3 parallel digital outputs: one each of AES, SPDIF coax, and Toslink.  And the DEQ has both AES and Toslink inputs.   But Toslink connection is generally considered inferior to coax and AES.  I might go with this if I could do the reverse, and run AES to the DEQ and Toslink to the DCX, since the DCX will only be handling super tweeters and sub woofers, which are not as critical as the nearly-full range panels.  But the DCX does not have Toslink inputs.

I did buy a short Sonicwave Glass Toslink cable so I could do this, as backup.

2) Coax to DCX and AES to DEQ, or vice versa.  This has the issue that the DCX input, in particular (though perhaps same as DEQ) has a lousy digital input receiver, one known to go into "dull mode."  To my knowledge this has never happened to me, but most likely it has never happened BECAUSE I only connect DCX and DEQ in all my systems through AES, which has 10 times the voltage of SPDIF/Coax.  So if I were to start doing using a simple XLR adapter or equivalent passive cable (XLR on one side, RCA on the other) I'd start getting the dull mode.  No thanks.

3) AES splitter.  This is what I have decided to do now.  I got a SESCOM AES Y splitter, with a second short XLR cable.  This way both DSP's are connected the best way, AES.

THEN, there's also the issue of how to connect the DEQ to my new "DAC" (Onkyo RDV-1).  For that I considered these options:

1) Toslink.  I purchased a 6 foot Sonicwave Glass Toslink cable, and also a Lifatec Toslink Silflex Glass cable with Optisilk jacketing.

2) AES to Coax transformer.  I spent a long time looking at these.  However, while the transformer converts AES balanced to Coax, it does not reduce the voltage from 5V to 0.5V.  Thus it might burn out a SPDIF input receiver.

3) Simple adapter cable (XLR on one side, RCA on the other), or XLR to RCA adapter with 6 feet of RCA fitted coax.  These are said to "usually work."  But the downside is the same or worse than for #2.  It could burn out the SPDIF input receiver.  That's a risk I really don't want to take with the rare Onkyo RDV-1.  (I could check the manual and see if that is permitted, which I haven't done.)

4) DC powered AES to SPDIF converter.  This is what I decided to get.  A Hosa Technology CDL-313 Data Link, along with a 3 foot AES cable.  I'll run 3' coax from the adapter to the Onkyo.

So I'm using AES as much as possible.  I bought an AES cable with each device above from Markertek (for the SESCOM) and B&H Photo (for the HOSA).  Then I got two 3 foot coax cables with Belden 1505F and Belden 1695A from Blue Jeans cable (in Blue and White colors for contrast from other cables, which are mostly AC cords).  And I also have at least two ways to go all Toslink for backup or comparison.

Sledge Amplifier arrives

I picked up the new SVS Sledge Amplifier on Monday night at FedEx.  I was immediately surprised at how light the things is, perhaps 5 pounds or so.  I was thinking a 1000W amplifier would have to be more like 15 or 20 pounds, plus a heavy metal plate to resist the sound waves.  Whatever the plate is made out of it's quite non-resonant, unlike a steel plate which would ring like a bell.

SVS had merely wrapped it in bubble wrap, no fancy box.  Well I shouldn't have been surprised…it is only sold for upgrades or replacements.

Non-resonance appears to have been a big concern.  Cables are wrapped in non-resonant sleeves.  And a yellow goo is spread over, under, or in between many parts.  The way this is done isn't very pretty, so I'd hardly call this amplifier "audio jewelry."  In fact, I was a bit concerned about the array of small capacitors toward the top of the main board.  They are tilting in many directions with the yellow goo underneath and between them.  Are they actually soldered in well, with the leads unbroken?  One can't verify that, and can only hope and try.

I still think my long run strategy is to use an external Class AB amplifier with meters.  But the new amp has a 5 year warranty, so I think I may just let that run out first.