180 Gram Vinyl Records

 The vinyl record resurgence has been fun, but I've been mystified by one development, and that's the obsession with 180-gram new pressings. Seriously, why do we need to use that much vinyl? Isn't vinyl made from petrochemicals, which are supposedly environmentally unfriendly, and increasingly scarce?

We're told that heavy vinyl sounds better. Theoretically that extra mass provides more resistance to the stylus following those wildly modulating grooves, and the extra angular momentum smooths out wow & flutter and damps out noise.

Um, no. First, the stylus is a very low mass object, and it doesn’t require that much force to accelerate it, even at the highest audio frequencies. The intrinsic shape and stiffness of the vinyl ought to be more than enough to guide the stylus through it’s gyrating journey.

As far as wow & flutter are concerned, unless the record player has a very flimsy turntable, a heavier record isn’t going to make a significant contribution to the overall angular momentum. Frankly, if someone is listening with such flimsy equipment, they’re probably not concerned that much with sound quality anyway. If you need more support and inertia, a more effective solution would be to buy a high-density turntable mat once and for all.

In fact, a record doesn’t need to be much thicker than about twice the maximum groove depth, to play back correctly. Maybe some additional stiffness would help with handling, but that’s about it.

I’ve also heard superstitious claims about warpage, but I have heard conflicting claims about whether thicker or thinner vinyl are more prone to it. I have hundreds of records, and warpage simply hasn’t been an issue. Excessive heat is the worst culprit. Store in a cool dry place, and you’ll be fine.

I have some RCA Dynaflex records from back in the ‘70s that sounded fantastic and still do. Dynaflex records were lighter and more flexible than most. RCA was criticized at the time for cheaping out on the vinyl, but I thought it was brilliant. That was in the middle of the oil crisis, and it seemed like a good way to economize on raw materials, without having to raise the price of the music. Other record labels used recycled vinyl, which sounded like Rice Krispies in milk. Snap, Crackle, Pop.

I think we should re-introduce Dynaflex.

Boston -- More Than a Feeling

We were in the depths of disco hell in 1976. I was a sophomore or junior in college, and most pop radio was just a pain to endure. Even Paul McCartney & Wings were jumping on the disco pukewagon. 

But there were a few bright spots: Heart, Queen and Boston. In the midst of all the same-tempo repetitive disco drivel, along comes this amazingly well recorded, sophisticated music -- harmonically, sonically, and the vocals were just superhuman. 

I was (and still am) an audiophile, and even on heavily compressed & peak limited FM, this song sounded great. Maybe Tom Scholtz mixed it for the FM medium. I don't know. But it was one of the few times commercial FM radio actually sounded like high fidelity.

Producer Rick Beato lays it out for you. By the way, Rick's 12-string acoustic sounds gorgeous in this video.

I'd like to have lunch with Rick Beato sometime. What a musical mind.

The Big Giant Piano

Ever since I was a baby, my grandparents had a baby grand piano sitting in the corner of their living room. While I was growing up, I used to noodle around on it from time to time. One thing I noticed was that the middle of the keyboard sounded very musical, but the bottom octave basically just growled. It was hard to tell which note was being played. Each note sounded different, but the actual pitch was indistinct. 

It wasn't a bad piano. It's just that the bass strings were too short to sound the fundamental very well. And uprights and spinets, well, forget about it. The problem is made worse by the fact that the hammers hit the strings close to the bridge, which tends to excite more harmonics than fundamentals. 

Even concert grands lack really clear fundamental bass notes. It's basic physics, captain. 

When I was in high school, I learned to play 'Joy to the World' by Three Dog Night on our family's little Wurlitzer Spinet. I didn't think anything of it, until one Christmas, I was at my Great Uncle's house, and he had a concert grand in his living room. (He was a session musician in Hollywood, so of course he did!) I sat down at that piano, and started pounding out JTW by TDN. I was blown away by how good it sounded on that piano. For $40,000.00 in 1960s dollars, it better sound good!

Well, here's a piano that would put that tinkle box to shame! Fundamental bass, baby! It's all about that bass.

Oh and JTW by TDN, yeah. This is how boy bands sounded when I was a teenager...

Here's the story about that fabulous piano, by the way. 

That Phat Tube Sound

I worked as a chief engineer for several radio stations in the late '70s and early '80s. At that time, there was still a lot of tube electronics in service, given that industrial grade electronics can be expected to last 30 or 40 years, if it is well-maintained. And it was only recently that solid state technology made it into high powered transmitters. Given that a 50KW transmitter could cost $250,000.00, there's a strong incentive to make it last.

I had occasion to work on much more powerful tube amps than most hi-fi hobbyists were able to. Our 1KW blowtorch, a Gates BC-1T had a 1200 watt push-pull audio power amp, modulating the final RF stage of the transmitter. Frequency response, noise and distortion were all very good. Replace the modulator transformer with an 8-ohm transformer, and it could have driven a loudspeaker to ear-shattering hi-fi loudness.

The amplifier consisted of four 807 drivers (pictured lower rear), and two 833A finals, operating in class AB push-pull (V42 & V43, rear). The two knobs at the front adjust the bias idle current.

Anyway, during my tenure as a broadcast engineer, I had occasion to build numerous audio amplifiers. Line amps, voltage controlled amps, headphone amps, studio monitor amps. They were all solid state amps, because I wanted them to be, well, modern. And being an audiophile, they all measured and sounded great, if I do say so myself. I was especially proud of my studio monitor power amps.

So, fast forward to 2016. I thought it would be a fun idea to build a little hi-fi amp from two 50C5s and two 12AV6s, having spent most of my childhood listening to the radio on All-American-5 radios made from those tubes. As you know, the AA5 radios were the culmination of reducto absurdum, in terms of building a radio from the least number of parts.

AA5 radios had no power transformer, and got their high voltage rectified directly from the AC power line. When the filter caps were new, they got maybe 150 volts. So the tubes barely had enough voltage to operate adequately. The filaments were connected in series, so the voltages added up to 121 volts. And because of all the rampant hum, the output transformer and speaker were specifically designed for poor low frequency response. The 50C5s were operated in full pentode mode, single-ended, cathode bias, at 10% distortion. They put out maybe 1 watt (while using about 50).

Well, you get the idea. What about building a nice, ultra-linear power amplifier using two 50C5s and two 12AV6s? Two 12AV6s are identical to a 12AX7, if you ignore the extra diodes in the 12AV6. I could put those four tubes in series, and power the filaments right off the power line (at 124 volts -- close enough). And I could get a power transformer to run the B+ safely, with a true ground reference, and full wave rectifiers, i.e., a safe and decent power supply. Yes, I would use silicon rectifiers for power.

But after studying this out, I came to realize why no sane person designs with tubes anymore. There are so many constraints, and extra considerations to construct a high quality tube amplifier, that it isn't worth it. First of all, my series-connected heaters would have caused terrible hum problems, and there's no way I could have fixed that, without resorting to a DC filament supply. Now, things are starting to get weird. Not only that, but getting an output transformer that would provide the correct taps for 50C5 ultra-linear operation would be difficult. I can't even find out what percentage tap is optimal for a 50C5. I guess nobody has ever attempted it.

It would have been a fun project, but I could design a really great transistor amplifier for less cost and less effort. And if I wanted "tube sound", I could build an amplifier with field-effect transistors. They're basically just like tubes, but without the filaments, and they can run at safer voltages.

Vinyl Stereo Encoding -- Not That Anybody Cares

Audio on vinyl is having a resurgence among audiophiles and millennials, despite the fact that CD audio, and higher resolution audio on DVDs and Blu Ray is orders of magnitude better in every measurable way. It is also more convenient, generally skip-proof (in memory and solid state drives - and if you don't like MP3 compressed, there's always lossless).

So why is vinyl still popular? Well, it's fun! I enjoy listening to my old records. I bought them in the 1960s, '70s and '80s, and never replaced them with CDs. Many were never even available in digital formats, except the ones I transcribe myself. And frankly, most of them still sound as good as the day I bought them. I always played them on decent equipment. But there's more to it than that.

By the time digital audio hit the scene, vinyl (or analog) recording state-of-the-art had reached a very high level of sophistication. Nothing like the 96 dB dynamic range, wider bandwidth, ruler-flat frequency response, and vanishingly low distortion of digital, but dynamic range on vinyl is in the high 70s, save for the intrusive clicks and pops from damage or defects in the vinyl itself. The human ear is quite tolerant of distortion and frequency response errors, so we have that going for us.

While I never bought into the lie that vinyl sounds better than digital (even moderately compressed MP3s), vinyl does sound pretty darned good. Some records sound almost as good as a CD. The trouble is, some CDs sound really bad, and some vinyl sounds really bad, making comparison hopelessly subjective. It depends on how much care went into the engineering and manufacturing.

Which brings me to today's topic: How is two channel stereo encoded into one groove of a vinyl record? Well, after some really horrible ideas involving dual tonearms (which would have been completely incompatible with mono, and would have taken up twice as much space, and would have had intolerable phasing problems at high frequencies), they finally settled on the +/-45 system.

There are several misconceptions about the +/-45 degree encoding scheme, and I intend to clear those up today. To begin with, we need to understand that the cross section of a record groove is a 'v' shape, with the walls separated by 90 degrees. In other words, the outer groove wall is +45 degrees from a vertical line bisecting the groove, and the inner wall is -45 degrees, for a total of 90 degrees.

+/-45 Degree cutting head (upside down)

The stereo encoding is sometimes described as, the outer wall is modulated with the right channel, and the inner wall is modulated by the left channel. Well, sort of. But that is more of a side-effect than a specification. Another way of stating this is that the right channel is modulated on a plane -45 degrees from vertical, and the left channel is modulated on a plane +45 degrees from vertical. It is possible to separate the two channels on playback, because vectors that are separated by 90 degrees are orthogonal -- independent.

Neumann VMS 80 Record Cutting Lathe
(If you look closely, you can see the cutting head just 
above the right side of the turntable.) (Click to embiggen.)

So it seems this orientation will modulate the groove walls independently with each channel, but what about mono? A mono cut means that both left and right channels are identical, and a +/-45 degree cutting stylus will move laterally, in the same plane as the record. Both groove walls move together in the same direction. With a real stereo program, the cutting stylus moves in all directions. If you were to look at an image traced out over time, it would look like a Brillo pad. So the groove walls are not really independent.

In fact, a better way to visualize this, is to imagine L+R being recorded laterally, and L-R being recorded vertically. If both channels are equal, L-R goes to zero, and you have a mono record. I am not aware of any record cutters that work this way, but actually, it might make the stylus motion and groove excursions easier to control. Incidentally, L+R/L-R (also called mid-side, or M/S) is totally compatible with +/-45. The vector sum of the stylus motion is identical with either matrix.

Stereo phono pickup (cartridge)

The main reason that phono pickups (cartridges) are all +/-45 designs, is that they can decode the left & right channels directly, whereas a M/S cartridge would require additional decoding. Not that it's difficult; stereo phono preamps could have had this capability since day one, if anybody had thought of it. But it would have added a few cents to the cost of each unit sold. So +/-45 it is. But any record cutter could switch to M/S at any time (even on the same record!), and still be totally compatible with all +/-45 pickups (decoders). As I say, the vector sum of the stylus motion is identical.

The fact is, there is good reason to want to record lateral and vertical components separately. Lateral recording is more resistant to distortion, and can be recorded "hotter" than vertical can. So limiters could handle each plane separately, possibly having less audible impact on the sound quality of the end product. It might also be advantageous to engineer a cutting head optimized for separate lateral and vertical excursions, vs. two identical left and right coils, which could be a compromise for vertical and lateral cuts.

Not that anybody cares about this anymore. I don't think there are many new breakthroughs in record cutter designs, these days. The payback would be marginal, and maybe what we have is good enough for the market share that vinyl has, in the big picture.

Android Audio Stutter -- Fixed!

Motorola DROID RAZR M

Not long ago I finally bit the bullet and replaced my old flip phone with a smart Android phone. I had several reasons: I had an increasing need to respond to people texting me, and the ability to check email and sync up work and personal calendars on one device was attractive too. I also figured I could take advantage of 32GB SD storage to put my entire music collection on the device, and have it with me always. Or so I thought.

Android audio playback was a disappointment. The audio quality was fine, if you can call stuttering, choppy audio fine. It would be playing along perfectly; I'd be enjoying the music, not thinking about the hardware, when all of a sudden it would stop for a half second or so. Very jarring -- disconcerting even! It doesn't take much to pull all of the enjoyment of music out of an experience when you get yanked back into reality every few minutes at random.

This pattern occurred with all three music apps I use: Amazon MP3, Pandora and Play Music. Choppy audio is usually due to multi-tasking other applications. I tried shutting down all the other running apps, trying to find the culprit. Unfortunately, even with all apps disabled, it kept happening. Then I stumbled upon the Power Control Widget. The Power Control Widget allows me to disable the sync feature (it turns off the periodic email, calendar and other data fetching functions). It appears the sync feature is what stutters the audio.

Android Power Control Widget
Left to Right: Wireless, Bluetooth, GPS, Sync, Screen Brightness

The sync feature may be a poorly written app, or maybe it really needs the hardware to run at a higher priority than keeping the audio buffers full, or feeding audio from the buffers into the D/A converters in real time. Whatever the reason, disabling sync seems to be the solution.

So, I set up a home screen with all of my audio apps, and with the Power Control Widget at the bottom. Now, when I go to play audio, I simply turn off sync. I admit, it's a pain to remember to do that (especially turning sync back on when I'm done listening), but it's better than the alternative. Maybe someday, the Android developers will fix this, but in the meantime I'll live with it as-is.

Is 16 Bits at 44.1 KHz Enough?

Is 16 bits at 44.1 Khz enough? Yes, if...

Spectral cleanup with dither

If you're just listening, and if the source material is properly dithered, and properly mastered, 16 bits at 44.1 KHz is good enough. Decades of double-blind testing have proved it time and again. But that's a big if.

There are a number of things that will affect the sound quality and the need for more bits and more samples. The modern CD loudness wars, which compress, limit and clip the audio into a quivering mass of Velveeta would get by with 8 bits and practically nobody would notice. A well-mastered and well-engineered CD at 16 bits and 44.1 KHz sample rate can sound fantastic. A poorly-mastered and poorly-engineered one will sound like crap.

Are there ever times when more than 16 bits and 44.1 KHz sampling would be better? Yes. In a recording environment, where the incoming live sound is unpredictable, it's better to leave some headroom for unexpected, accidental peaks. If you're using all 16 bits, and a louder peak comes along, it will overload the A/D converter. Eight additional bits provide 48 dB of additional headroom, which is itself more than the dynamic range of the vast majority of commercial CDs being sold today - especially the loudness war victims.

In addition, when mixing tracks, each additional track adds about 6 dB to the sum (assuming each track is mixed at full scale - a worst-case scenario). Mixing eight tracks will consume all 8 additional bits of headroom. To avoid overload, recording additional tracks means the level of all tracks will have to be attenuated in the mix. If they're 24 bit tracks, they can be turned down without any loss of resolution (they do have to be re-dithered however - in fact, audio should always be re-dithered when it is re-scaled - up or down).

For live recording and mixing, you can do better with 24 or even 32 bits (Audacity uses 32-bit floating point by default). But for commercial distribution via e.g., CD, 16 bits really is enough. Trust me on this.

So, what about sample rate? Well, that's much less important with modern oversampling sigma-delta A/D and D/A converters. Humans can hear at best, out to 20 KHz. Most of us are lucky if we can hear 15 or 16 KHz, although it really depends on how loud the signal is. I suspect more humans could hear 20 KHz if it's loud enough. For music, there just isn't much significant energy above 15 KHz. A sample rate of 44.1 KHz means that the digital recording system can accurately capture up to a maximum of 22.05 KHz: more than most anyone can possibly hear in a musical setting. The Nyquist Shannon sampling theorem tells us that we absolutely, positively need not sample more than twice the highest frequency of interest. This is not speculation; it is one of the most well-validated results in the field of information theory.

Aliasing

In order to prevent ultrasonic frequencies, above 22.05 KHz, from getting into the A/D converter and getting aliased back down into the audio frequencies, the input should ideally be filtered. As a practical matter, the filtering requirement on A/D converters is fairly weak, because most music doesn't contain a lot of energy at or above 20 KHz, and certainly not far enough above to be aliased down to frequencies that we hear readily. Still, most good A/D converters do provide high quality "brick wall" anti-aliasing filters. It is very difficult to build analog brick wall filters, but modern (i.e., since about 1995) oversampling sigma-delta converters can do this using digital filters with mathematical precision. They hypersample internally, but they output e.g., 44.1 KHz or whatever the desired sample rate is.

The filter function is different for D/A converters, but the requirements are about the same. The output filters are used to remove the ultrasonic sidebands that are generated by the sampling process. Although we can't hear them, those ultrasonic frequencies can wreak havoc with amplifiers and tweeters, and they need to be removed. It is probably sufficient to merely attenuate the ultrasonic sidebands with a softer roll-off that won't over stress downstream equipment. But modern (i.e., since about 1995) oversampling sigma-delta converters can provide digital "brick wall" reconstruction filters with mathematical precision.

Back when ADCs and DACs ran at the nominal sample rate, we had to filter using analog elliptical filters, and I could justify a bit of "slack" between the stop band and the sample frequency. It reduced the constraints on the analog filter, which were critical and hard to keep in spec. In those days, sample rates of 48 or 50 KHz were common. But for today's oversampling sigma-delta ADCs and DACs with their built-in digital filters, any sample rate above 44.1 KHz is simply an outdated concept.

Do we need to sample at 96 KHz? Certainly not! It doesn't make the filtering job any easier. All you're adding is more load on the CPU and wasting storage space. If you can't hear 20 KHz, you're never going to hear 48 KHz, or anything in between. And 192 KHz? That's four times higher than 44.1 KHz! If you can't hear 20 KHz, neither you or your dog are going to hear 96 KHz. Why on earth store all that inaudible chaff? It's processor abuse!

High Fidelity Hobby Becomes Necessity

I have been an audiophile all my semi-adult life, getting into it in a big way while I was in high school. I just bought the most sophisticated sound system that I have ever owned: a Phonak Audeo Q90-312 system. That's right, they're deaf-aids. I have been struggling with hearing loss for about a decade, and I finally got fed up with not hearing offhand remarks in meetings, and asking people to repeat themselves. My music enjoyment was never affected much, if I turned it up loud enough. But my family objected to that.

I resisted getting hearing aids because I figured the fidelity would be low; optimized for speech intelligibility over fidelity. I had already decided what I wanted: Something like the Dolby S encoder for cassette tape. And it had to be high fidelity, wide band and low distortion.

The Dolby S encoder is essentially a multi-band, equalized compressor, whose job it is to keep low-level signals above the tape hiss, backing off once the input signal gets above the noise level. In my case, the noise is tinnitus - ringing in the ear. For me, it's like narrow band pink noise, centered on about 4 kHz. Sounds need to be louder than that, in that spectral region, in order for me to hear them.

One of the features I like about Dolby S is that the design was based on the principle of least action (Ray Dolby borrowed the term from physics, but it makes sense in a different way, in this context too). As the signal gets above the noise, further expansion isn't necessary, so Dolby S gets out of the way, and lets the system behave normally. Of course, my hearing is compromised, so while the encoder (hearing aid) brings the signal up above the ringing in my ears, there is no corresponding decoder to undo the encoding. Or is there? As a matter of fact, there is: my brain! I'll get back to that.

Hearing Impaired Musician

I started researching hearing aid technology, asking in particular, what do hearing impaired musicians use? Well, they use high-fidelity, WDRC-TILL (wideband dynamic range compression with treble increase at low levels) devices (or they did a decade or so ago). So, armed with this information, I sallied forth to my local audiologist, and told her that I'm her worst nightmare: a well-informed electronics buff, critical listener and opinionated & demanding audiophile. That news didn't seem to upset her in the least.

She took me back to the soundproof room and we did a hearing test. She keyed the data into her desktop computer, which displayed my hearing chart and calculated the necessary correction. We discussed various models of deaf-aids, and I explained that I would require the highest possible fidelity, and that I was very skeptical that anything could meet my requirements. She finally selected a sample and put them in my ears. She then radioed the prescription to the devices, configured them with the necessary gain, equalization and compression (in 20 bands). I could instantly hear better, and they sounded great (w00t)!

These devices have FM transceivers, so they can communicate with each other, and with a Bluetooth ComPilot that I can wear around my neck if I want to use my cell phone hands-free, or listen to a TV or iPod through my deaf-aids. In normal operation, they radio audio information left and right, to continuously triangulate on stereophonic directional cues, just as our brains do. So they can classify noise sources, as well as speech sources, that the digital signal processor on each ear can then sort out in real time. It's quite amazing, and all that in a package that is so small that the casual observer cannot even see I'm wearing unless I point them out.

The electronics (7/8" long) tuck behind each ear, and the tiny loudspeakers, pictured above right, on the other end of the wire, go in each ear canal. They have excellent fidelity - at least for frequencies above 300 Hz or so. Below 300 Hz, where I don't require any assistance anyway, the direct sound just goes right past them. They're driven by a super high-efficiency class-D amplifier, so battery life is about six or seven days (if I turn the devices off at night when I'm not using them). Batteries are mercury free air-zinc type 312.

For the first two weeks, I was delighted that I could understand speech so much better, even in noisy restaurants, and in conferences at work. But I did notice some distracting artifacts: the stereo image seemed to move around, and I noticed a strange "warble", kind of like the sound I would hear when I was a kid, and my brother and I would talk to each other through a rotating fan. Some people describe the sound as being like talking in a corrugated pipe.

The first artifact was undoubtedly due to the devices trying to make voices more intelligible, by changing their directional characteristics. It's great for speech, but for music, not so much. The second artifact, I found after doing a bit of research on the web, is probably caused by the anti-feedback algorithm being set too high. This chipset (code named Spice) has a characteristic sound when feedback cancellation kicks in.

So when I went in for my initial two-week tune-up and oil change, I told the audiologist that I need a "stable platform" for music; I described the warble, and what I had found on the web. She pulled up the configuration software, and I was pleased to see, there was a "music" program right on the menu. So she added it to my configuration options. When I listen to music, I can push the button behind my ear to select the music mode.

The audiologist also noted that anti-feedback was set rather high. She turned it off, and we experimented with various feedback-inducing scenarios, none of which set them off, so we disabled anti-feedback, and the warble is gone. My hearing loss probably doesn't require enough gain to risk feedback. That could change if my hearing continues to decline, but for now it isn't a problem.

When I first started wearing the devices, it was like having new glasses: everything sounded freakishly clear, and I noticed details that I hadn't heard in years. This is like listening to a Dolby S-encoded cassette without the decoder turned on, only more so. But after a while I became accustomed to the sound, until now it just sounds natural - except that I can understand speech and hear soft music. My brain has adapted and provides the necessary decoding.

The fidelity is spectacular - and I'm speaking as an audiophile now. I can listen to music and enjoy it more, and the hearing aids only enhance the experience - they don't get in the way, as I feared they might. Low-volume live music is audible and enjoyable again. I have noticed zero distortion or overload with loud music, and no compression artifacts with sudden dynamic changes. My family appreciates that I don't have to turn up my sound system or the TV to deafening levels. Now I wonder why I waited so long to do this.

Confessions of a Frustrated Audiophile

Magnavox Portable Stereo Record Player

I have been an audiophile ever since my Grandfather gave me a homemade cabinet containing an Electro-Voice SP12B speaker with an Atlas horn tweeter. At the time (1970), I had a portable Magnavox stereo.

I think the Magnavox had about 3 watts per channel into those 3x6 oval speakers shown in back. Well, I found some RCA plugs and just paralleled the outputs together and hooked them to my Grandfather's single mono speaker (not recommended for audiophile work). But it worked, and the amplifier didn't seem to mind. The sound blew me away. The high end and the low end were like I had never heard before, even from the "hi fi" my parents had in the living room. I was hooked.

The next task was getting back to stereo. So I hunted down a source of EV SP12B speakers. Unfortunately, the one my Grandfather gave me was from the late '50s, and the new ones were of a slightly different design. No matter. I only had $50.00, so I could only afford one. I built some cabinets from a project in Popular Electronics (I think it was), that used some mail-order tweeters from Mouser Electronics. The cabinets were designed for some Radio Shack woofers, but the specs were similar enough to the EVs that I decided to go ahead. I finished building the cabinets, and now I had full range stereo. At 3 WPC. With a ceramic cartridge. Still, I remember it sounded great! I got a lot of enjoyment playing Nilsson Schmilsson, Abbey Road, Ram, Straight Up, The Best of the Guess Who, and ... and I think that was the extent of my record collection then. I had some singles, Joy to the World, American Pie, Albert Flasher, a few others.

BSR 610 Automatic Record Changer

I started reading Stereo Review, and from there I realized that I would need a magnetic cartridge. Also, I started to notice thumps and rumble coming from the record changer in the little Magnavox. Plus, I wished it could play a little louder. My fascination with electronics had led me to get on the Heathkit mailing list, and they always had the most beautiful audio gear (for the time). It was time for an upgrade (such as my meager income could afford). So I bought a Heathkit AA-1214 amplifier and a BSR 610 record changer (with a Shure M71 cartridge). I eventually got matching woofers and purchased EV "building block" midrange horns, tweeters and crossovers.

BIC 980 Belt Drive Turntable

The BSR turntable also had thumps and rumble, so eventually I upgraded it to a BIC 980 with a Shure V15 Type III (I was making more money by then). I used the BIC until about 1984, when CDs started coming out. I bought a Magnavox 4X oversampling CD player (which was actually a Philips, which contained very good electronics and converters). In those days, it was nearly impossible to build a 16 bit D/A converter that was actually accurate to 16 bits. But Philips made 14-bit oversampling converters. The 4X oversampling produced the additional 2 bits by duty cycle modulating bit 0 of the 14-bit converter for 16-bit accurate output. It was ingenious, and it sounded great. So much so, that I started to wish I had a better turntable. Which brings me to the subject of this article. (That's the longest intro I ever wrote).

Dual 505-2 Belt Drive Turntable

Here's my confession: I love my Dual 505-2 turntable! I think it cost me about $300.00 in the day. The salesman talked me into a Denon high-output moving coil cartridge. Moving coils were all the rage in those days. But the Denon never sounded great to me. For one thing, my Heathkit AA-1214 had kind of a noisy phono preamp, and even though the Denon was "high output" for a MC design, it was lower than any moving magnet cartridge. So I purchased an Ortofon OM-20, which is one of the highest output moving magnet cartridges out there. I was kind of a Shure bigot, though. I only went with Ortofon because of the high output, and I had read that Ortofon and Dual had teamed up to match the 505-2 tonearm with the OM-20.

Ortofon OM-20 Phono Pickup

When the cartridge arrived in the mail, I hooked that baby up, and put on my Mobile Fidelity copy of Abbey Road. Wow! I was hooked. That was about 1986. My vinyl collection didn't get as much use in the CD era, or in the MP3 era, for that matter. But what I didn't have in digital format, I have played on vinyl. Over the past few weeks, I have been spinning a lot of vinyl, and really enjoying it, on a turntable and cartridge that is now 27 years old. The OM-20 stylus is still in good shape (I used to replace my old Shure stilii every year or two - don't ask me why the Ortofon has stood up so well). Fortunately, OM-20 replacements are still being made.

Here's the bottom line: Audiophiles and vinylphiles are quick to poo-poo any turntable that costs less than $1000.00, and is built with anything less than unobtanium parts and magical wire, broken in for three months. Well, the fact is, that old Dual is still kicking (after a belt replacement and a few other maintenance repairs), and it is capable of producing sound that is as good as can be stored on vinyl. A 50-lb platter just doesn't turn any more evenly or quietly than a well-engineered lighter one.

Thorens TD 235 Semi-Automatic Turntable

And here's a really interesting tidbit. I was looking at Thorens* turntables with the thought of updating (just because). Guess what I found? One of the midrange Thorens turntables is a dead ringer for the Dual 505-2, and I'm not just a-woofin'. Check it out: The tonearm is identical (right down to the gimbals and headshell), and the semi-automatic operation is as well. The plinth as been redesigned slightly (possibly for easier manufacture), but the look is nearly identical. MSRP: $1000.00 US.

I might buy the Thorens TD 235 someday, but for now, I still love my Dual 505-2 and that sweet sounding Ortofon OM-20. As long as I can still get parts.

* Thorens makes very high-end turntables for the rarefied audiophile community.

Lousy Sound -- Capacitors

I am an audiophile, but not an audiophool. I don't buy magic elixirs and oxygen-free copper, especially not in mains cords and speaker wire. The difference between capacitor sounds and amplifier sounds, tubes and transistors, digital vs. analog, have to have some basis in physics, and be demonstrable in a blind A/B comparison, or it doesn't exist. If a blind A/B difference can be detected, then we need a scientific hypothesis, and an explanation with reproducible results, or it's mere superstition.

There are some differences that are so gross, so obnoxious that you don't need a blind test to detect. We all know that. It sometimes happens when a component fails, and all of a sudden there's a hum that wasn't there before, or some kind of distortion that is just painfully obvious, and it goes away when you fix the problem.

Case in point: I have a pair of JBL 500 bookshelf speakers that I purchased for my office about 15 years ago. They're not what I would call hi-fi, but they are adequate for use with my computer sound system, playing the various bleeps and boops, and sometimes an AAC or MP3 here and there. They sounded OK for the past decade or so, but lately they have just started sounding obnoxious. I simply could not stand to listen to music through them. I would have to turn them off. "Obnoxious" was definitely the operative word here. And it wasn't just while I was thinking about it. Background music was distracting me from my other work. It wasn't subliminal.

I was able to rule out the source and most of the amplifier by switching to my AKG K-501 headphones. I still wondered if the amplifier was behaving badly, maybe oscillating, or under-biased with nasty crossover distortion (I usually listen at low volume). But no, the amplifier bench tested well within spec, and all the voltages and currents were normal. It was definitely the speakers -- or my hearing. But no, the headphones sounded OK. It was the speakers. But how? I've never experienced speakers going bad before.

I finally popped the back off the speakers, and at one glance, I had my suspicions. The crossover. It is the simplest kind of crossover that would possibly work, using the cheapest components possible: one (ferrite core) inductor and one (electrolytic) capacitor. The capacitors had gone bad in both channels. They were five microfarad bipolar electrolytics, and if their values changed, if they shorted in one direction, they could become quite non-linear. And of course, a change in value would make the crossover operate at the wrong frequency, and hence, change the frequency response and distortion characteristics of the speakers. I had to replace the caps.

So, what to replace them with? Electrolytics? Well no, but not with audiophool caps either. Electrolytics can deteriorate over time. Film caps generally do not. Five microfarad film caps are available for building crossovers. They're more expensive, and they're physically larger than electrolytics, which is why JBL didn't use them on entry-level bookshelf speakers, which outlived their warranty by some 14 years anyway.

So I paid my three dollars each, and put them in. The main problem was, they wouldn't fit on the crossover circuit board. I had to wire them on, and then use hot glue to secure them in place so they wouldn't vibrate and rattle around. But I got them installed.

The speakers sound much better now, thank you very much. Not obnoxious. I can work once again, undistracted, with music playing in the background. There is definitely a difference when a component is malfunctioning or operating out of spec. Capacitors can have a "sound", especially when they're used as a frequency-discriminating element in a filter, such as a speaker crossover, where voltages are gyrating all over the place. Bypass and coupling capacitors most likely do not have a "sound", unless they're inadequately specified, or if they're malfunctioning.

Lousy Sound

I have noticed a trend -- lousy sound. I have been a music lover and audiophile (not audiophool, which is unscientific superstition, like anthropogenic climate change) since high school.

Listening to Pandora, I was struck by a new kind of distortion. It's an obnoxiousness in the upper midrange, almost like a harsh, noise modulation being kicked up by things like electric guitars and vocals. It isn't harmonic, and it isn't normal intermodulation. And no, it's not this.

What a long strange trip it's been, getting to the bottom of it. At first, I figured it was Pandora's standard AAC bitrate. So I paid for the subscription version so that I could get the high quality feeds. No joy. Besides, AAC actually sounds really good at reasonable bitrates.

So then I suspected my speakers. Maybe they were just obnoxious, so I started listening with headphones -- three different ones. No joy. Next was the sound card, so I tried a number of different ones on various machines I have at my disposal. They all exhibited the same obnoxiousness.

Heathkit AA-1214

Crossover distortion in my amplifier? I replaced the finals in it a while back, because they went into Vce breakdown and burned up. I used the original part, but with a higher breakdown voltage that wasn't available in 1972. Did I get the bias wrong? It turns out, one of the bias resistors had changed value. So I re-biased both channels. Distortion measurements are well within their original spec. Besides, this amp has never sounded bad to me in the 40 years since I first assembled it. The obnoxiousness persists.

My ears? Maybe... I do have tinnitus... but no, it doesn't happen on every song. I don't notice it with my old personal CD or vinyl collections (although the distortion does sound somewhat similar to vinyl damage from a worn out stylus). But I have been familiar with that phenomenon since the early '70s, when my ears were brilliant. Although I miss with great anguish the acuity of my youth, I do know how to listen. This ain't it.

To reiterate: it doesn't happen on every song. I have heard of the loudness wars that CD producers have been engaged in. It's the same idiotic loudness wars that broadcasters have been engaged in for many years, except it's even more idiotic when we're dealing with a medium that so doesn't need any additional processing to sound good. There's more than enough headroom on a CD for the dynamic range of any music known to humans. There's no reason to pack it all into the top 6 dB of the 96 dB available.

Not only do they compress the snot out of it, they clip it too. If this clipping occurs before the A/D converters, the higher harmonics that would cause aliasing get filtered out. But if they clip after the A/D converters, it will cause aliasing, and aliasing sounds nasty.

Aliasing has no correlation with the harmonic structure of the music (in fact, it's inverted). But it will kick up as noise whenever the digital clipping occurs. I can't imagine that recording engineers and record producers would be so stupid as to clip in the digital domain (without using the proper anti-aliasing filters), but maybe they do. If anybody knows, please drop me a line, or post a comment here.

I would not expect this obnoxiousness to occur with the classics, but with so many classics being remastered and reissued, I expect that the new releases of old material are also getting the same treatment. It certainly sounds like it is. Even our proud legacy is being mutilated.

Realtek Sound Improving?

A favorite pastime among gamers and audiophiles is to bash Realtek audio codecs. This may have been justified at one time. A build I did several years ago using an Intel D945GCCR motherboard with the Realtek ALC883 audio codec certainly deserved the drubbing. Even an old Sound Blaster 16 PCI was a dramatic improvement.

But a recent build using an Intel DH67BL motherboard with the Realtek ALC892 audio codec was a very pleasant surprise. In A-B listening tests, I was unable to discern the difference between the Realtek and the Asus Xonar DG (a very well regarded audio board).

It's possible that the ALC892 sounds better than the ALC883 chipset. However, the published specs of the two chipsets are too close to quibble, and frankly the horribleness of the sound of the D945GCCR motherboard would have made any specification pointless. The thing that curled my eyelids was the harsh upper midrange distortion, which if anyone measured it, would have pegged the needle. It was that nasty.

You want to know what I think? I think both Realtek chipsets are just fine. I think Realtek is a victim of industry-wide poor motherboard design. I think the problem is with the analog portion of the D945GCCR motherboard, which was designed by Intel. Intel are digital logic designers, not analog audio designers. I think they botched it with the D945GCCR, but they finally got it right with the DH67BL. The former requires an add-in sound card. The latter really is a pleasure to listen to.

I think Realtek's problem is that it's their face on the audio subsystem -- their volume controls, equalizers, effects controllers, media players (if you use them). If Realtek were smart (or a bit smarter anyway), they would let the motherboard manufacturers put their logo on those apps. Then the blame (or praise) for the sound quality would go to the board manufacturer, which actually has more to do with the sound quality anyway.

The sound really depends on the board layout, the analog design and choice of components, whereas if you get the audio codecs right, they're right -- period, end of story. It really isn't any more expensive to make a good audio codec than a poor one, so might as well make a good one. If you get a justifiable reputation for making poor audio codecs, your bottom line will suffer terminally, regardless.

Now, having said all that, I should point out that the Realtek chipsets are not as quiet as the Xonar (with noise levels in the -90 dB range, the Realteks are comparable to 16-bit conversion), whereas the Xonar approaches -120 dB, which consistent with 20 ~ 24-bit conversion. They all claim 24-bit resolution, but only the Xonar approaches actual 24-bit performance.

Make no mistake though, unless you're using this for professional studio work, 16-bits is far more resolution than you'll be able to discern with your naked ear in anything but an anechoic recording studio. As a point of reference, the noise level of a good vinyl analog recording corresponds to about 8 ~ 10-bit resolution.

Classic Amp -- Updated PCB, and Why Not?

Ok, so I updated the PCB yet again. Check it out:

Classic Amp PCB - Click to Enlarge

What did I do? Well, I flipped the transistors Q7, Q9, Q11 and Q13 on the bottom half the board, so that they're a mirror image of Q6, Q8, Q10 and Q12 on the top half. It is now beautifully symmetrical, and the positive and negative pulling devices will have as similar a layout as is physically possible. There's just one problem: Q12 & Q13 are facing opposite directions. Too bad the transistor manufacturers didn't consider this when they made complementary devices - duh! I have it figured out, though: I'm going to sandwich both devices between two heat sinks. Pure symmetry, Captain! You'll just have to wait for the snapshot of the prototype to see what I have planned.

Classic Amp -- Guess What?

That's right folks, it's another iteration of the PCB layout. I wanted to get the drivers closer to the finals, so I rotated the big emitter degeneration output resistors so I could put the finals closer in. That necessitated refactoring both power supply planes. While I was in the market, I put most of the traces on the ground side so they would interrupt the power supply planes as little as possible. Finally, I added a 0.1uf bypass on the power supplies.

Refactored PCB - Click to Enlarge

There is a lot to be said for multi-layer boards. Life would be simpler if I could put the power and ground planes on an inner layer.

Classic Amp -- Plagiarism

Some of you, looking at my schematic, might have noticed similarities to this, this, or especially this. The last one has some marked similarities, right down to my R11. I rarely see emitter degeneration in the VA. In my final design, I might make it vanishingly small, or jumper it out altogether, but I wanted to experiment with it, and see what kind of effect it has on distortion, gain and bandwidth. I actually didn't see that schematic until a couple of days ago, when it turned up in a Google search for Sziklai parasitic oscillation.

Classic Amp - My Schematic - Click to Enlarge

Although I have the ESP site bookmarked, along with Douglas Self's and Marshall Leach's sites, I promise you that I was working independently when I did the schematic capture.

Of course, there are only a few ways to configure a "classic" amplifier and still have it be, well -- classic. So you should expect some similarities. As I said in my original post, I wanted to include a number of enhanced features, without over designing. There is only one way to make a differential amplifier that has a current source and a current mirror. There are only two typical output configurations: Darlington and Sziklai. The rest is all calculating values, parts selection, testing performance and optimization.

The area that lends itself to originality is the PCB layout. Although there's really nothing original about ground and power planes, I have never seen a power amp use them. I'm using Eagle Light for this project, so I can only do two layer boards. Otherwise, I would have done a four layer board, with power and ground on the two interior layers (V+ and V- planes could almost meet in the middle), and signal traces on the two outer layers.

But I think the other thing that makes this board unique is the ability to make the temperature compensation track by fastening together Q8 & Q10 and Q9 & Q11.

Classic Amp - My PCB - Click to Enlarge

I think that the power and ground planes might make this amplifier more stable, requiring less tweaking (fewer caps, or lower values) to control oscillation. Stay tuned. I believe this will be an unique amplifier.

Classic Amp Updated Board IV

Just when you (meaning I) thought it couldn't get any smaller or cleaner, I spy another opportunity for optimization.

• I moved Q6 & Q7 in line with the drivers (Q10 & Q11), which shortened the traces considerably, and made it possible to shrink the whole design once again.
• The size of the board being determined by the big filter caps, I decided to drop two of them, and increase the value of the remaining ones. Four 2200uf are now three 3300uf (which actually increased the total from 8800uf to 9900uf per side). The Panasonic data sheet says 3300uf fit in this footprint.
• I also added a power ground pad next to each power input pad. Both ground lines will go back to the transformer center tap, but this way, the ripple current will go to the associated filter caps, and not have to flow across the ground plane. 
• While I was moving grounds around, I decided to move the speaker ground where it would be centered on the board, so the current flow would be more symmetrical through the ground plane. 

Great Grandson of Updated PCB - Click to Enlarge

Anyone want to wager on how many more iterations there are?

Updated Schematic - Click to Enlarge

One final thought: There might be a bit of ripple at Q12 & Q13, or R14 & R15. I don't think this will be a problem, since the output voltage will be governed by Q10 & Q11. Any perturbations in the supply to the output devices would be compensated by the local feedback inherent in the Sziklai configuration. Still, I would have preferred to have the raw rectifier output hit a capacitor before it hits my output transistors. Even with the big fat power supply rail, there is still some resistance between P+ and C6, et. al. For this reason, there might be some justification for a bit of off-board filtration. I'm just sayin'.

Classic Amp Updated Board III

I have made some more optimizations to the board.

• Shortened leads by dodging corners
• Compacted the power amp section to shorten leads
• Changed the feedback capacitor (C1) from axial to radial, dramatically saving board real-estate.
• Pulled in the input section to fill in the vacant real-estate. 

Grandson of Updated PCB - Click to Enlarge

I could make it smaller yet, but then the silkscreen parts nomenclature would become more of a mess than it is already. This is probably small enough. The filter caps are as compact as they're going to get anyway. So unless we want to abandon the power plane and bypass idea, I think we're at our physical limit, for the most part.

Classic Amp Updated Board II

I made some more changes to the board layout:

• Moved the power planes to the top side of the board, so I could make the ground plane fill the entire bottom side, and so I could significantly widen the power planes. It also allowed the power planes to flow to the output transistor power connections, instead of having to route to it (higher current capacity).
• Widened speaker output trace (higher current capacity).
• Increased clearance to output transistor pads.
• Eliminated vias -- now using component mounting holes to switch sides if needed. 

Son of Updated PCB - Click to Enlarge

Classic Amp Updated Board

I fixed up some awkward routing, and a couple of dimension errors on some parts. It's always a good idea to double check the spec sheets before releasing a PCB to manufacturing. Once it's etched and drilled, it's too late.

Updated PCB - Click to Enlarge

Classic Amp Schematic and Board

Schematic - Click to Enlarge

I have updated the schematic to include the filter capacitors. Rather than a single large capacitor, I chose to use four smaller ones on each side of the power supply. I also used large power busses and a ground plane, so the power and ground are rock solid, and held in place by multiple capacitors on each side. If additional energy storage is needed, you might be able to squeeze 3300uf caps on there, according to the spec sheet, or one could always add more off the board.

Board - Click to Enlarge

Note also the placement of Q8 and Q10, Q9 and Q11. As I mentioned in my previous post, I intend to fasten these devices together for extremely accurate temperature tracking and stabilization.