Weekend Project: Build a 108″ DIY Projector Screen for $28

This weekend I built a super-cheap, no-frills, 108″ projector screen for my family room home theater. A modest setup consisting of a Pioneer 7.1 AVR, some DIY speakers and sub I built years ago and a 50″ Panasonic plasma HDTV. We’ve got the usual array of sources as well – a Panasonic Blu-ray player, Xbox 360 and the ubiquitous Home Theater PC (long live Windows Media Center!). I’ve been toying around with the idea of a garage theater for the last year, so I do have a projector, but have yet to bite the bullet on formalizing some kind of air conditioning out there, so in the summer months, movies in the garage come at a fairly high cost of nearly sweating to death.

I always thought it would be fun to just bring the projector in the house and project on a wall so we could watch movies in the (air conditioned) house, but there was just no good place to put it where there was easy access to the existing home theater equipment and a good wall to hang a screen. That’s when I started concocting a plan, a plan which involved making a super cheap screen which I could actually hang from my existing entertainment center in front of the TV/stereo/center thus making the transition from TV time to full-blown-movie-going extravaganza as easy as hanging a picture on a wall. And since I wasn’t sure how this would work out overall, I started out thinking this would just be a proof of concept only, so I did it as cheaply as possible. This whole thing only cost $28. So far it’s worked out so well, I see no reason to upgrade or change it out for something “better”. So without further adieu, I offer up this tutorial and a few pictures for you so you too can transform your family room home theater into a bigger, better movie experience for less than it costs to take the kids to Chick-fil-A.

List of materials:

Qty (3) yards Roc-Lon Blackout Fabric (white/white or white/beige) from Joann’s (54″ wide) – $12 (with a 50% off coupon) normally $7.99/yard – don’t forget the coupon!
Qty (4) 3x1x96 inch furring strip pine lumber – $6 – The Home Depot
Qty (6) 2×4 inch Simpson Strong-Tie Mending Plate – $4 – The Home Depot
Qty (2) D-Ring Frame Hangers – $2 – Walmart
Qty (2) Small Caribiner – $2 – Walmart
Pack of 3/8″ Heavy Duty Staples – $2 – Walmart

List of Tools/Supplies:

Staple Gun
Scissors
Measuring Tape
Miter Saw
Titebond II Wood Glue
Orbital Sander and 60 grit sandpaper
Cordless Drill and Drill Bits
Phillips Screwdriver
Square

First and foremost, let’s talk about the Roc-Lon Black Out fabric. They sell 4 different colors as well as mixed front/back colors but you want to note the color of the backside of the fabric because that is the side you will project onto. The frontside, the side that is rolled outwards, has a bit of a texture to it, but the backside is smoother, has almost no texture and is more rubbery – that is the side you care about. My Joann’s had a white/white, a beige/white a beige/beige and an ivory/white (front/back) to choose from. I ended up getting the beige/white because the roll had more fabric on it and looked like it was in better shape. I was worried about whether or not wrinkles would iron out or not stretch out properly. You just want to make sure that the backside is white, but the frontside doesn’t matter as much.

Turns out that you can iron this fabric and small wrinkles will stretch out so you don’t have to be too fussy about the condition. Quick tip though, bring your own 2″ cardboard role and after they cut it have them re-roll it onto your own role, otherwise they will fold it 3-4 times over which creates creases. They did this with my fabric and I didn’t think any of it at the time, but it creased badly, which meant I had to spend 30 minutes trying to iron out all the wrinkles, which like I said you can do, it does work, but it’s extra time that would otherwise be unnecessary if I had just been a little more prepared going to the store. They may have extra rolls there too you could ask them nicely if they’d roll it for you. Also, don’t try and iron the rubbery side, aka the projector side or backside, it doesn’t work well and will ruin the texture on that side, you have to iron the frontside/fabric/texture side. It’s still not easy to iron a huge 54″ x 108″ piece of fabric though, without creating creases as you iron out creases just trying to manage this beast of a piece of fabric on a tiny ironing board.

So I had purchased the Carl’s Place Sampler Pack a couple of months ago and was planning on buying one of their premium screen materials, so I was able to compare some of their samples to Joanns’ Roc-Lon blackout cloth and here’s what I can say about the similarities. The “frontside” of the Roc-Lon blackout is very close to “backside” of the very popular Carl’s place standard white Blackout Cloth. It’s got the same type of fabric-like texture to it, it’s almost the exact same color white and has similar stretching properties although the Carl’s Place Blackout cloth is slightly more flexible/stretchier. But the backside of the Ron-Lon fabric looks and feels almost exactly the same as the frontside of the Carl’s Place Blackout cloth, which is why if you buy this fabric, you will want to project on what is effectively the backside with respect to how it’s rolled, since it matches the definition defined by Carl’s Place as “the smooth texture side with a rubber coating faces the audience”. Both materials are the same thickness as well measuring 0.014 inches [0.35mm] thick. Visually, using projected content, the Roc-Lon and Carl’s Place blackout cloths are nearly indistinguishable in terms of white/black levels and color balance. I observed no obvious hot spots or changes in texture or sheen that would be observable or distracting in any way.

I’ve got some pictures of the fronts and backs of both fabrics in both natural light and projected light so you can see the differences. I’d probably have to give the edge to the Carl’s Place Blackout if I were being super picky about the texture, because the fabric texture bleadthru from the other side is ever so slightly less obvious on the CP Blackout cloth than with the Roc-Lon Blackouut but we’re talking nanoinches. And it depends on which location in the fabric you’re looking at. Click on the two pictures above and you can see what I’m referring to. This is with natural light, it is far less obvious with projected light. Of course the most obvious disadvantage to the Joann’s stuff is you’re limited to a 54″ height. Which works out to about 53 usable inches vertically (once you wrap and staple it to a wooden frame) and 94.25″ horizontally which makes for a 108″ total diagonal screen with a 16:9 aspect ratio. Carl’s Place doubles that to 110″ allowing for some extremely large screens.

My initial thought was a 108″ screen would too small (I wanted 120″), turns out I was dead wrong. This thing is HUGE in my living room. Seating arrangements are about 8-10 feet from the screen which is seriously just about as close as you’d want to be for this size screen. But if you wanted to go bigger, you could do a cinematic 2.35:1 aspect ratio screen which would result in a 53″ x 124.5″ or 135″ diagonal screen. I couldn’t even imagine that big, but making the trade between vertical dead space on the sides for 16:9 stuff or “letterbox” dead space for everything else, I opted for the standard 16:9 since it’s arguably the most common ratio, top and bottom bars bug me less than vertical ones, kids watch a lot of Disney movies which tend to be 16:9, plus we watch a lot of TV which is also more 16:9 and with the Xbox for playing games, the 16:9 format is easily the standard there. Not to mention buying the materials for building the frame is easier because you only need to buy four of the 1×3 common pine in 8 foot pieces and don’t need to go to the 12 foot suckers. Trying to fit those in your car without them poking out a back/front window is a nightmare.

So let’s build the frame. I bought more 1×3 pieces than I needed but fortunately had the extra pieces because after bringing one of the 8 foot pieces home, the next day it looked like it had warped a good two inches. Since this frame essentially hangs on a couple of hooks, there’s really nothing to keep the frame square, or true in the z-axis, so to speak. If a board warps inward or outward, even after the frame is assembled, it will likely still have this shape to it which may negatively affect your movie experience. You only need four boards so be super picky about the quality of each one. Home Depot/Lowe’s have different grades of common pine and you pay for the nicer stuff, which you’re more than welcome to do. I bought 2 pieces of the common pine which is about $3 each and then went back and bought a few more from the cheap-o bin (actually called furring strips) and just hand-picked my pieces and then got one extra (because I didn’t realize that one board cut it half makes the side pieces perfectly). More on that later.

You want to make sure the boards are straight along the face and the edge. Pick up the board and give it a good stare-down, holding one edge up by your eye and looking down the length of the board, then rotate 90 degrees and do the same. If the board warps more than ~1″ in any direction, start a discard pile and pick another. Watch out for big knots because they will tend to be the points where bending and warping occur most. Also look for edge quality, you need at least one good edge where the fabric will wrap and staple. If the edge is over-cut or chamfered too much, discard it and pick another. Split boards should be left in the bin as well as any board that twists even a little bit.

Now this is where I’m going get the most flack from anybody whose worked with wood, or considers them self a carpenter, but this was meant to be a prototype, so bear with me on this one. I used six Simpson Strong-Tie 2×4 mending plates to join the 1×3 lumber at each joint. This honestly worked out about as well as a 71 cent piece of metal could have worked. So, basically like crap. In theory this would have been the easiest, cheapest and quickest way to attach the vertical and horizontal boards. If you’ve got a Kreg Pocket Hole Jig, use it. Or a bunch of crazy-long pipe clamps, use them. Even a L-shape/T-shaped steal plate and some screws would have worked out better, but at 10x the cost so I made due with these crappy strong-tie plates and a bunch of glue.

The problem with the all-in-one tie plates is that they split two of my boards and they don’t hold the boards flat against each other on the edge. Once I had all six plates tacked down, the bottom/top boards sat at about a 20 degree angle to the other boards. I’m not talking in the X-Y direction, I used my square to make sure the angle was correct but because the metal is pounded in with a hammer, it tends to want to pull the board away from the edge of the mating board. To combat this I put glue on the edge of each board and then added weights to the entire frame to hold it flat so the boards weren’t bending upwards. I used various objects from around the garage to keep the frame flat overnight while the glue dried. There’s a few 1-star reviews on this product and people say you have to use a press to install these, not a hammer. I can second that. So if you want to spend more money or have a better carpentry solution to joining 5 boards to each other, then by all means do it the best way you know how. Before attaching the fabric run a sander across all of the edges to “round over” where the fabric will be wrapped to prevent the sharp edges/corners from cutting through the fabric causing it to tear.

So the material width from the roll I got ended up being 54.5″ wide and this stuff is not very stretchy but it will stretch about 0.25″ over its width and 0.5″ over its length if I just had to guess based on how much I had left over. Still I opted for a 53″ screen which allowed a non-stretched fabric fit across its front and over each frame edge the thickness of the board, or 0.75″. I think it worked out just fine to only have the fabric wrap to one side and be stapled but if you want the fabric to wrap all the way to the backside, then you’ll need to make the frame smaller to support that size. But I wouldn’t bet on “stretching the fabric to fit”, make the frame the size you want based on the fabric unstretched because it does not have a significant amount of give to allow you to “oversize” the frame with the hope of stretching the fabric to fit. I pushed it to 53″ only because my fabric was 54.5″ wide, but if you get a roll that is 54″ on the nose, or less, make sure you account for that and take off at least 1.5″ for wrapping and stapling.

So other thing to consider is that the boards are slightly longer than 96″ which allows you to make two of the uprights boards from one ~96.1″ 1×3 board. Each board is 2.5″ x 0.75″ inches so if we want a 53″ frame and we are using butt joints to connect each piece, then the three upright pieces need to be 48″. I ended up cutting each one at 47-15/16″ because each board was about 96-1/4″ long which allowed me to get two pieces given the 1/8″ scrap from the miter saw. So trying to make the height of the frame any larger would just result in more scrap and extra cost. So all you need is 4 boards, two for the horizontal pieces and two for the vertical pieces which includes one in the center. Then you end up with one scrap 48″ piece but you can use some of it to create a block to attach a picture frame ring hook that makes it easier to hang the screen to a hook on the inside of the entertainment center. Still, not much wasted wood. And the top pieces are cut to 94.25″ which corresponds to a 16:9 aspect ratio. Note that I did not add a black felt boarder. If you intend on adding a self adhesive type boarder, then you will need to change the dimensions of the frame to ensure that inside of the boarder is still 16:9 after applying the felt tape. My plan is to add an exterior 1×2 that is wrapped in black felt, making the frame actually larger instead of reducing the overall viewing size.

So this is how I stapled the fabric to the frame. This takes two people so get a friend, spouse or family member to help. With the frame on the ground, I laid the fabric with equal distance overhanging each side. I put two staples into just one corner (no more than 2″ down on each side) on both the vertical and horizontal boards without stretching anything. Then I lifted the frame upright with the wife’s help so the fabric was just hanging down while holding onto the non-stapled end. I pulled just the top portion as tight as I could without trying to rip out the fabric from the few staples in the other side. Then with just this top piece fairly taught, I ran a row of staples across the edge of the fabric making sure to match the overhang of the fabric to the board edge along its length, which in this case was zero overhang. Then we pulled the fabric taught along the short side of the frame in similar fashion to doing the top and stapled it in place. Now flip the frame over.

So at this point we only have the top (now bottom) and one side pulled tight and stapled. Then we moved to the opposite corner of the two stapled sides and just pulled in both directions until all the fabric was tight in each direction. This is basically your chance to get out any waves or soft spots. Once it’s pulled taught and looks good, pop a couple of staples in both the sides bust just at the corner, again no more than 2″ towards the center. Next we pulled the fabric taught along the top basically trying to match the tension we had applied the first time we did this and dropped a staple every 2″. There was just a little bit of fabric left over which you can cut off or staple to the backside of the board as desired. We did the short side last and once again pulled it just tight enough to match the tension in the fabric from the other sides. You can’t really work out any large wavy spots at this point however so you’re really just pulling the fabric tight enough not overtight, or tighter than you’ve been pulling the other sides, just trying to match the tension already applied. Staple it down every 2″ with the 3/8″ staples and then stand back and admire your work! This is arguably the hardest part and also extremely hard to explain how we did it in words. The only saving grace is that the fabric is so cheap, that if you do happen to screw this up, it’s not going to brake the bank if you have to go back to Joann’s and pick up more fabric. Also, don’t forget to make sure the white, non-textured, more rubbery side is facing out. That is the side you want to project to.

The last thing I did was install a pair of ring-style hooks to the top of the frame to make it easy to hang to the entertainment center. I added a small screw to the inside facing of the entertainment center and then hung a small caribiner from each one which allowed me to then hang the hooks from the screen directly to the caribiner. This was actually harder to do with the fabric taught over the frame compared to when I fit checked it earlier that day. So I added a small framing clip to the caribiner which made a hook for the ring loop to easily attach to. I can setup and teardown the screen in only a few seconds. Plus when the screen is down, the entertainment center and TV leave no remains of there even being an option to add a screen in this manner.

And that’s about it, take a look at the pictures and click on any of the links herein to take you to some of the product pages used to make this screen. I haven’t sat through an entire movie yet but it looks completely awesome just viewing still images and short clips of movies/shows. Also my projector is a modest Panasonic PT-AX200U 720p 3LCD unit but it’s got a brand new bulb as of a few weeks ago so it’s looking better than ever and with the new white screen, the image is super bright and the small 108″ size makes even 720p content look crisp and sharp. Screen gain is best I can guess 1.0 since comparing it to the Carl’s Place Blackout screen provides nearly identical white balance and gain which they state is a 1.0 gain screen. Anyway, I’d love to hear your comments or questions about this project. Worth the $28 and a few hours to make? Would you build this for your home theater?

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Decoding the Signal Path of My 2-Channel Stereo

Last week I changed around my stereo a little bit just for fun and wanted to share a few of the things I learned in doing so. I didn’t do anything too crazy, just removed the Schiit SYS passive pre-amp and replaced it with a Denon AVR-1801 receiver. The receiver has been part of my garage theater which is currently undergoing some modifications. So I thought this would be a good opportunity to add it to my 2-channel stereo setup so I’d have an actual active pre-amp in the mix. Or rather, an active 5.1 receiver that happens to have a pair or RCA pre-outs (not something every receiver has these days I’ve come to find out) that should double nicely as a dedicated pre-amp. Acting as a pre-amp only the Denon holds its own quite well. A plethora of inputs, optical, coaxial, analog, even a phono input. Right now I’ve got two modest sources, an Apple Airport Express and a Sony DVP-S300 DVD/CD player. It’s worth mentioning that I bought the Sony unit from Goodwill for $6, the Airport Express from a guy on Craigslist for $20 and the Denon unit from a different guy on Craigslist for $50. Total investment at this point – $75. I do audio on the cheap, I know. But good gear doesn’t have to cost a fortune either. And good used stuff is even better.

Now back to the setup. Since the Denon has both analog and digital inputs, I was left with two connectivity options for each source: connect the CD player and Airport Express to the analog CD/DVD inputs using a couple pairs of RCA cables, or connect the S/PDIF or Toslink output from the CD player/Airport to the S/PDIF and/or Toslink inputs on the receiver. Either way you get sound coming out the other end. What difference would either option make? Was there not an obvious better way to connect up this equipment? If so, what was it? And was there a way to know which option would be better without actually listening to them?

So my first instinct was to connect everything up using only the digital interfaces. I figured the Denon probably has a better digital-to-analog path in it and therefore would sound better. This means the conversion to analog is bypassed on both the Apple Airport and Sony units to allow the Denon unit to do all the work. I have nothing against running analog in’s and out’s between stereo gear, but if you can keep things in the digital domain longer, and not flip flop between them too often, then it just seems like that should be better. Especially if you have reason to believe one of the two potential DACs and subsequent analog parts in your signal path is significantly better in one piece of gear than in the other. Imagine having two pairs of speakers connected to your amp via an A/B selector switch and selecting the crappier speakers all the time without even realizing it. Little did you know that just by switching to “speakers B”, you could have been listening to a much better pair of “speakers”, you just had to pick the better pair and put those in your signal path. Just about every stereo setup anyone owns probably has an unused DAC in there somewhere. Missed opportunity perhaps?

After connecting the CD player to the Denon via the S/PDIF port, I popped in some CDs and had a listen. Up until this point my setup was such that I was using the DAC in the CD player to convert the digital 1’s and 0’s to analog, but now I was feeding those same 1’s and 0’s to the Denon unit and allowing it to decode and convert those bits into analog sound. So what did I think? Did it sound better? Actually, my first impression was that it did sound better. Inexplicably better, not definitive, but somehow just better. Maybe I was just hearing things? Probably, but could I prove that one setup was superior to the other? Sure, why not? So I decided to do a little deep dive into the physical signal path differences between these two setups and compare them. What is actually different? Which configuration looks better on paper? What DAC chips are actually in these units? Were there other obvious defining differences? Read on to learn what I found out.

For now this is just going to be a paper analysis. I’m not going to actually measure anything. So if you’re hoping for FR and phase plots, THD and FFT plots, sorry to disappoint. But this is an easy analysis just about anyone can do with their own stereo, though your mileage may vary depending on which gear you’ve got, brand and how old (or new) it is. The internet is a huge resource, but even then, some old parts are just not going to turn up. And some of the newer and boutique audio shops just aren’t going to publish that kind of info. No sense revealing the Colonel’s secret 11 herbs and spices.

You can download a pdf of the information I put together for this chain analysis here which includes excerpts of schematics and block diagrams for both units. Datasheets for all of the components can be found by clicking the hyperlinks within this post.

I started out by searching for service manuals for both the Sony DVD/CD player and the Denon AVR. Service manuals are great for digging into this kind of thing and I found a ton of super useful information on both units. Such as schematics, part numbers, troubleshooting tips, you name it, these service manuals had it. That’s when the fun began! I traced out the digital and analog signal paths for each unit starting at the optical transport and finishing at the pre-out of the receiver. The one thing I was interested in initially was what DAC was in each of these units (but I actually learned quite a bit more). I was hoping from there I could then find out some specifications that might indicate which one should be better. My search didn’t stop at just the DAC though. But I’ll jump right into that part, since it’s the most relateable component. And if you don’t end up reading this entire article, you will at least have learned that much.

The Sony DVP-S300 contains a Sony CXD8750N-T2 2-channel DAC. There is absolutely zero information about this DAC available online. It is most likely one of Sony’s proprietary designs so they aren’t inclined to publish anything on it which means there’s no info out there to get any sense of what the DAC can do. Searching the part number online reveals that it appears in units with model numbers M35/S300/S305/S315/S500D/S705D/S715. So it was a popular DAC in its day used in a wide variety of now discontinued models. Not much else go go on there.

So let’s move on to the DAC in the Denon receiver. This is where it got a lot more interesting. The DAC in a Denon AVR-1801 is an AKM AK4527 “High Performance Multi-channel Audio CODEC”. AKM have been making DACs for ages and have a wide portfolio of DAC and DAC-related products. The the most interesting thing noted in the datasheet for the AK4527 is they claim it supports multi-bit. The first paragraph in the data sheet states,

“The DAC introduces the new developed Advanced Multi-Bit architecture, and achieves wider dynamic range and lower outband noise.”

There is no other mention of multi-bit in the rest of the datasheet and in most cases there are references to sigma-delta processing if it does come up again. I have no idea if this DAC does in fact support multi-bit as it’s commonly referred to today or not. This data sheet is dated 1999. Other specs tout 128x oversampling, 96 kHz sampling rate, 24-bit 8x digital filter, 106 dB S/R and high jitter tolerance. EEtimes.com wrote a short article about this DAC when it was released back in 1999 and here’s a short excerpt from that article:

“The six D-to-A channels use a new multi-bit sigma-delta architecture with 128X over-sampling. The dynamic range in this direction is 106 dB. This part brings a new performance level to multi channel codecs,” said Ahmad Nowbakht, applications engineer at AKM Semiconductor, a subsidiary of Asahi Kasei Microsystems Co. of Tokyo. He suggested the AK4527 could be used with compressed audio applications such as AC-3, DTS or other multi-channel standards. “It could be used in DVD players, DVD-RAM drives, or home audio receivers along with many other multi channel audio systems,” he added. The AK4527 is packaged in a small 44-pin LQFP, 10-x-10-mm, package with the same pinout as AKM’s previous two-input/six-output codecs. Prototype quantities are available now, priced at $6.12 each for 5,000 pieces.

So there you go, what’s the real takeaway? That they use the term multi-bit and sigma-delta in the same sentence? Like, which is it? How about the fact that this is a $6 DAC? Either way, if I had to blindly pick a DAC in my setup, either the Sony unit or the AKM unit, based on the information available, I’d probably pick the AKM DAC over the Sony DAC just because it’s got some decent specs against a part that has, well, no specs. It offers good dynamic range with 24-bit/96 kHz processing whereas the Sony DAC probably only supports 16-bit/44.1 kHz audio, seeing as how it is designed for a regular CD player, ideally there’s no reason for anything higher. Besides the bit depth and sample rate, are there other less obvious sonic benefits to one DAC over the other? Perhaps other features built into each chip make a larger difference, like better digital filtering, better tolerance to jitter, better DC biasing (power supply and power supply filtering) and better analog filtering. Specs alone hardly tell the whole story. But it’s certainly a starting point.

So what else did I learn about the signal path in my search? Let’s take a look at the op-amps. Both the Sony and the Denon filter the analog DAC output via a pair of standard operational amps. Likely configured in a simple low-pass filter and some form of gain stage to bring the output to line-level standards. The Denon service manual was scanned in with too poor a resolution to really tell what resistor/cap values are used. But that’s what the Sony op-amps were configured for. The Denon uses a pair of NJM2068 low-noise op-amps which are designed for high-performance audio applications. The Sony contains NJM4558V op-amps which are basically spec’d for general purpose applications and they don’t claim to be low-noise or suited for high-end audio applications. In fact, I thought it was funny that the datasheet for the NJM2068 claims it is actually better than the NJM4558. Ha! The two op-amps I have in this very setup. Must be a popular pair? Here’s an excerpt from the datasheet below:

“The NJM2068 is a high performance, low noise dual operational amplifier. This amplifier features popular pin-out, superior noise performance, and superior total harmonic distortion. This amplifier also features guaranteed noise performance with substantially higher gain-bandwidth product and slew rate, which far exceeds that of the 4558 type amplifier. The specially designed low noise input transistors allow the NJM2068 to be used in very low noise signal processing applications such as audio pre-amplifiers and servo error amplifier.”

So the op-amps in the Denon are “sold” as being better than the ones in the Sony. Okay, fair enough. A/B comparing the datasheets reveals identical equivalent circuitry, with the exception of one extra cap on the NJM2068 across the gate input of one of the transistors. Perhaps one of the means by which they claim lower noise? Here’s some other features that favor the NJM2068 op-amps in the Denon; better common mode rejection and power supply rejection, faster slew rate, higher bandwidth, lower noise voltage and they actually spec THD at 0.001% whereas the NJM4558V doesn’t spec it at all. I’ll take a good spec over no spec any day.

So once again, on paper, the Denon op-amps appear to be the winner in this signal-path shootout so far. So to sum up what we found: the Denon has a presumably better DAC and definitely better spec’d analog op-amps than the Sony. If if you had to pick which signal path to choose, would you choose the Denon or the Sony? For now I am thoroughly enjoying allowing this old $6 Sony DVD/CD player to act as a transport-only thus allowing the Denon unit and its superior AKM DAC and NJM op-amps to decode and process the digital and analog signals. It sounds great to me, though mainly I just enjoy knowing what parts lie between the player and the pre-outs. It’s surely just a placebo effect by me visualizing the signal running through some crappy Sony DAC and a couple of 69 cent op-amps compared to the signal being fed to a decent AKM DAC and NJM audio-grade op-amps (don’t look up how much they cost though). My brain is telling me one path is better than the other. I guess I’m just a dork like that. But wait, I’m about to dork out even more on this. I wasn’t satisfied just tracing out the digital to analog portion of the signal path, I wanted to trace out the entire signal path, from CD to speakers. Here’s the jist of what I learned, starting with the optical pick-up all the way to the speakers. Buckle up, we’re in for a wild ride.

The Sony DVP-S300 contains a KHS-180A/J1N optical transport. One that was used in a vast array of Sony units back in the day. A guy on Ebay sells them for $165 a pop because apparently it’s the same unit used in the Sony SCD-777ES and SCD-1 Super Audio CD Players. Units that sold for $2500 in their heyday. Hey cool, not a bad find for $6 at Goodwill. We’re starting off with presumably a half-way decent optical pickup and if you get right down to it, the optical pickup/transport is really only responsible for one thing, getting information off the disc optically and converting it into an electrical format with which the rest of the electronics can actually process/amplify/filter/decode as appropriate. This signal from the optical block looks nothing like you might think, it’s not really a digital signal at all, it’s a modulated RF signal! Using a modulation type known as eight-to-fourteen (EFM) the “bits” coming off the disc form an eye pattern but don’t actually look like 1’s and 0’s yet at all. Check out the patent for more information on the subject. Pretty interesting stuff!

In the Sony unit the modulated RF signal feeds what’s called a “CD RF AMP”. The model number is CXA2555Q-T4. I found the datasheet for this unit and what it does is basically amplify, filter, and EQ this RF signal in preparation to send it off to the ARP unit for demodulation (it performs other functions like laser tracking and focus). The next part in the signal path is the ARP unit, model number CXD1865A. This device is actually responsible for demodulating this crazy-looking eye pattern RF signal (that yes, contains our audio signal buried inside) and creates a binary data stream into two much more common formats: S/PDIF and I2S. Aha, now we’re getting somewhere. We all know what S/PDIF is and many fewer of us are familiar with I2S but at this point we at least have an intelligent digital data stream that represents audio from the CD with bit-for-bit-accuracy. But is it really bit-for-bit? We haven’t even got to the DAC yet but how confident are we that we have a digital representation that actually matches what is on the disc? My take is this, and I could be wrong, but demodulating a NRZI or EFM encoded compact disc is not new, in fact, every CD player in every computer you’ve ever owned is capable of demodulating every bit off that disc with perfect precision, because it was designed to do just that. Here’s a quick excerpt from one of the service manuals on the details of just what’s inside the ARP unit:

“In the ARP, first RF signal processing such as asymmetry correction, adaptive equalization, and sync clock extraction by the RF-PLL are carried out so that the signal becomes binary data synchronized with the PLL clock. This data is…EFM demodulated (CD-DA and video-CD) in the demodulator, subjected to frame/sector sync detection, address detection, and protection, and sent to the buffer memory controller.

…This is the same for the CDDA. The ARP is linked to the built-in and external memories, and CDDA signal processing block to carry out error correction, output data flow control, etc. The signal is then muted and corrected by the CDDA signal processing block, and then sent to the IC203AV decoder (L64020).

For all of these disks, RF jitter is calculated by the RF signal processing block, and this information is used for adaptive control of the servo DSP via the CPU.”

Forget about what DAC you’ve got, I want to know what ARP you’ve got! I can’t even figure out what ARP stands for. There is no datasheet out there for the CXD1865A. I don’t hear a lot of talk on the street about this fancy demodulation and error correction device either. Seems like it’s pretty dang important though. I think most people just think everything is 1’s and 0’s, all the time, I know I did, until I started researching this whole signal path thing. Kinda feels like I fell into a rabbit hole a little bit though. Did you know that a CD is actually an analog medium? Yeah, stew on that for minute or two. Do we even know what digital really is?

Some things worth noting about what goes on this ARP unit, and presumably every single CD player out there, is error correction and jitter management. When I read about people using a CD player as a transport, such as the way I am using it in my system, you are effectively removing as much of the processing of both digital and analog signals as you can and offloading that work to an external DAC but you’re still dependent on at least two IC’s in that “transport/CD player” to create a perfect binary representation of what is on that disc. EFM modulation contains error correction. The engineers at Sony and Phillips knew that an optical disc being read by a laser would be susceptible to errors, so they created a modulation pattern with built-in error correction, much like almost every data transferring scheme developed in the modern world today. Nothing gets through perfectly every time, errors occur, through various means, and when they do, there has to be a mathematical means of re-creating lost data. And a way to do it perfectly. But what happens when too much data is lost? And what constitutes too much lost data? And if data is lost, what is reconstructed in its place? The ARP unit is responsible for all of that (and more) and yet it’s never mentioned in most stereo systems.

I think I’ve lost track, let’s circle back to another day. You can check out this site if you want to read more about this topic or this site here for some really good general info on the compact disc. For now let’s assume we have all-new, pristine CDs with no scratches, damage, pits, or other defects that would otherwise make that ARP have to do more work than necessary and heaven forbid, actually have to make up stuff that isn’t on source medium. Let’s follow the signal paths of our two signals, we’ve got S/PDIF and I2S at this point coming from the ARP where both signals are fed to the AV Decoder part number L64020-D-QC-27. I was able to find the datasheet for this device and the only noteworthy thing I took out of it is that is passes the S/PDIF and I2S signals straight through without any additional processing. The signal names bounce around a bit in the schematic, and they never call these digital signals “I2S” by name but presumable it meets the interface standard because all there are 3 data lines that track throughout the circuit and the signal names are _DATA, _BCK and _LRCK which coincide with I2S naming convention for a multiplexed data signal, the bit clock and left-right clock. So it’s definitely I2S. Might there be a way to pipe this out directly to the MiniDSP? Hmmm, a topic for another day.

The S/PDIF signal is fed to a TC7S08FU AND gate then to a simple transistor buffer stage before terminating into a standard female RCA plug (and don’t forget the orange center plug!). So surprisingly not much going on there and very little processing to get from the compact disc surface to something we can finally start to decode and turn back into an audio signal. In short working backwards, the S/PDIF signal comes from the AV Decoder (bypassed) which is fed by the ARP unit which demodulates the EFM signal from the RF AMP which amps, filters and EQ’s the RF signal from the optical block which is using a photo sensor to pick up the pulsing pattern of a laser beam bouncing off the surface of the CD encoded with pits and lands that make up a digital representation of audio. Wow. And this my friend is ~36-year old technology. As complicated as it is, it’s amazing. The time and energy that was put into developing the standards behind the audio compact disc in a day and age where computers as we know them today did not even exist. It boggles the mind that you can seriously walk into any Goodwill and find a half a dozen old CD/DVD players selling for under $10. It’s just crazy.

Alright, let’s go back to my original analysis, let’s follow the I2S path and see where it takes us. The I2S standard, unlike S/PDIF is not really meant for long runs or runs in cables between stereo equipment. It requires 3 signal lines plus ground and is more suited towards internal interconnects either on single circuit board between devices or between boards with very short runs where impedances can be controlled in order to maintain good signal integrity. I won’t get into the I2S specification, you can download it online. It was developed by Phillips back in 1986 and they describe it as follows:

Standardized communication structures are vital for both the equipment and the IC manufacturer, because they increase system flexibility. To this end, we have developed the inter-IC sound (I2S)bus – a serial link especially for digital audio.

Hey, we all like common interface specifications, it just makes things so much easier to mix and match and let hobbyist and hardware developers do what they do best. So back to the Sony DVP-S300, the I2S signal is fed to the infamous nothing-can-be-found-about-this Sony DAC, the CXD8750N-T2. The DAC without a name or any information available online. C’mon Sony. Okay, so to be fair, this thing is old as mud, I’m not surprised there isn’t any info out there. We’ll just assume it’s a run-of-the-mil digital-audio converter with 16/44 bit/sample rate just good enough for your average Joe. The DAC output feeds the two NJM4558V op-amps which look to be configured as a low-pass filter and a gain stage similar to the Denon. The second op-amp likely bumps the output level to standard line-level and then over to the pair of RCA jacks. And that’s our analog output path from the Sony CD player. Tracing back to the split-point between the S/PDIF and the I2S we can see that the only real difference between the digital out on this unit and the analog out is one DAC and two op-amps. Which is exactly how it should be, and which is probably how most CD players are architected.

This post is a million miles long, man how did I get here? I haven’t even traced out the digital and analog paths within the Denon unit yet! So let’s finish off the analog path in the Denon receiver where we left off coming out of the DAC and from the NJM2068 op-amps to the actual pre-outs on the back of the receiver. And then we’ll come back to the digital path from the S/PDIF input. And lastly we’ll talk about what the analog path looks like coming into the receiver, such as through the CD input and I’ll tell you why you should never trust the analog CD input on a home audio AV receiver like this for 2-channel audio. You probably already know, but I won’t ruin the surprise.

So back in the Denon the analog signal from the NJM2068 op-amps following the DAC is fed to a switch which provides selection of different sources (since it’s a receiver with like 8 inputs). The part is a Toshiba TC9162 Analog Switch Array. Noteworthy specs are 0.001% THD and 1.0 uVrms noise voltage. Frequency response looks like it extends to 50 kHz, though they don’t spec it directly. The device is probably as transparent as any analog switch and sort of a necessary evil if you want more than one source. Which brings us to the volume control. I was happy to see that the volume control is done completely in the analog path and that it is digitally controlled. The chip is a Toshiba TC9482 which is designed to be used as a volume controller in high-end audio equipment. It basically incorporates a resistor array that offers 95 dB of range in 1 dB steps. THD is spec’d at 0.005% and noise voltage is 1.2 uVrms. BW is not spec’d though test conditions indicate 20-20kHz and there’s no indication of amplitude flatness.

So the volume control is sort of another necessary evil. One could design the volume control in the digital domain, but then you run into issues with dynamic range and using the “total available bits”. And I don’t even want to get into that whole discussion. After the volume control, there’s another NJM2068 op-amp. Man these things are everywhere inside this unit! I guess if you’re going to design a receiver, might as well pick a decent op-amp and just litter it everywhere you need one. Keep the unique BOM count low and get the best price breaks on a single device. Oh and the NJM2068 you can buy for only 60 cents. 45 cents if you buy 50 or more. If you’re Denon and buying thousands of these things, they probably get them for a quarter or less. Yes, that’s what audio-related electronics cost in the commercial world. Like I said, the dollar value consumers associate with a piece of audio gear does not always reflect its actual value or the true cost to manufacture it. But anyway, moving on…

Following the NJM2068 op-amp, the signal splits, one path feeds yet another NJM2068 op-amp tied in with a Tone Control device which is a Toshiba TC9184. The other path bypasses both the op-amp and the tone control device. A switch on the other end allows the tone control to be “defeated”. A nice feature for the “purist” who doesn’t want either the option to adjust bass and treble thus ensuring a perfectly flat response, or because defeating the device all together makes you feel better about the signal path, that you’ve somehow eliminated some unnecessary extra circuit. Sure, you can just set both the Bass and Treble knobs to “0” but is that really the same as clicking the fancy “Tone Defeat” switch?

Turns out it’s not the same. Well at least it’s not a gimmick, the switch does in fact defeat the tone control chip and supporting components, so if there were even the slightest hint of degradation caused by the part, it would be defeated as stated. I run with the tone defeat switch on because, well, I simply enjoy knowing that this little extra bit of circuit is not part of the signal path and I don’t really need to add additional bass or treble boost since the MiniDSPs are already configured for a flat in-room response. No sense mucking all that up with course bass and treble knobs. But even then, looking at the datasheet for the TC9184 it’s about as transparent as can be. It’s basically a massive ladder array of different value resistors in conjunction with passive external low and high-pass filters that are then summed with one of the NJM2068 op-amps. But it’s a digitally controlled part, though the entire the signal path is all analog and mostly passive. Specs for the tone control part state typical THD of 0.005% but no other noteworthy parameters for the signal path otherwise. My guess is the tonal quality within this path is probably more heavily influenced by all the caps used in the filter network than the bulk array of resistors. But you can see, setting the Bass and treble knobs to “0” is not the same thing as hitting the “tone defeat” switch from a signal path perspective. Now ask me if I can tell the difference between the two? Nope. Not one bit. Unless either of the bass or treble knobs is adjusted any amount off of “0” since the step size is a whopping ±2 dB. That I can hear.

Anyway, that’s about it. The signal goes from the tone defeat switch straight out the RCA pre-outs and then, in my setup, off to the pair of MiniDSPs, the Emovita UPA-700 and then finally to the DM-4 speakers. I’m not going to trace out the signal path any more than this for today. I’ve gone on far enough that you get the idea. Ultimately the signal that we just spent so much time converting to an analog is going to be converted once again back to a digital signal in the MiniDSP, where it will be processed digitally (performing filtering, EQ, gain/attenuation and delay) before it’s converted back to an analog signal for the last time. After all this you can understand why there’s a case for not having an active crossover/DSP unit in this location in the signal path. The extra ADC-DAC conversion feels unnecessary and some may say sets an “upper limit” or boundary of sorts for how good my system will ever sound. Nothing will ever sound better than the weakest link in your setup. I’m not saying the MiniDSPs are not great, they are awesome little units, but a part of me is itching for a pair of the new MiniDSP HD 2×4 units. At least the MiniDSP does support I2S, so it’s possible to skip the extra AD conversion, unfortunately my setup doesn’t really support doing so in this manner, and since MiniDSP (the company, not the product) discontinued the MiniDIGI, I’m not quite sure how to improve this setup beyond just replacing the MiniDSPs one day or attempting to tap into the I2S directly off the CD player. But once again, a topic for another day.

The last thing I wanted to share is why after researching this why you should be careful using the analog “CD/DVD/VCR” inputs with an AVR in a 2-channel setup. The reason? Because the analog signal is most likely being converted back to digital for additional DSP. In my setup the analog inputs eventually make their way over to the AK4527 DAC (which actually has a 2-channel ADC in it as well) before being converted back to analog. That’s right, there is one more ADC-DAC conversion for those inputs. Though it’s not clear what settings on the receiver invoke this path, because there’s another path the goes straight from the analog inputs the Switch Array TC9162 device, joining back up at the same spot where the AK4527 DAC feeds. So as long as you can guarantee that you haven’t hit some button on the receiver (like all the fancy surround sound modes, Pro Logic or music effects that are included, blah) that causes the signal to be fed back into the AK4527, then the signal path stays all analog from input to output.

However just be to sure, the best all-analog path is the 5.1 inputs since those are fed directly to the switch array and bypass all other digital devices in this AVR, thus eliminating even the option to accidentally re-encode-decode the signal by selecting some special sound mode. So for fun, imagine the worst-case setup – the signal path if we used the analog outputs from the CD player (that’s one DA conversion) to the CD input on the receiver set to a fancy sound mode where it’s converted back to digital and then back to analog (that’s two DA conversions plus one AD conversion) and then is fed out to the MiniDSP where it’s converted back to digital-from-analog, processed via DSP and the converted back to analog (that’s two AD conversions and now three DA conversions) before being amplified and fed to the speakers. It sounds terrible on paper, I know, all those conversions back and forth and back and forth from digital to analog to digital etc., though I honestly wonder, would anyone be able to tell the difference? I might try out some A/B tests of such a setup, because it would be pretty easy to do.

And that pretty much sums everything up for this episode of Decoding My Signal Path. In the end it was a fun exercise and as I begin my hunt for additional audio gear, I’ve realized that unless I can find this level of detail in the components within the components, I’m not satisfied just relying on reviews from folks claiming the ability for a particular piece of gear to make music sound like pink fluffy unicorns dancing on rainbows. I want cold hard facts man. And I only looked at signal path and didn’t even consider biasing paths. I could dive deeper into this, and I know there’s people out there that do. They redesign input/output stages or swap op-amps, change out capacitors, add supply filtering, add shielded inductors, you name it, people have done it. Until then, happy listening! And if you’ve got a setup similar to mine, let me know how you’ve got it connected and what you like about it. See ya!

Click here to download the Complete Signal Chain Analysis with Schematics and Block Diagrams for the Sony DVP-S300 and Denon AVR-1801

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I Started a YouTube Channel – “How to Play” Songs on the Piano – Go Check it Out

So a few weeks ago I pulled out my camera tripod, I set it up hovering over my keyboard and with my phone attached to the end, I hit record. I started playing a song, it was Piano Song by Erasure. A song I learned how to play some twenty-plus years ago. A song I taught myself how to play just by listening to the song over and over. A CD player on repeat with my parent’s old upright in the living room of my house growing up. This is something I’ve enjoyed doing ever since I was in grade school to this very day. As the phone recorded while I played through the oh so familiar chords I thought, what if I were to try and teach someone how to play this song? How would I do it? How long would it take to explain? How well do I know the names of all the chords in this song? Do I even play the song correctly? Would people care if it’s not note for note? Would people want to learn how to play this old song from so long ago? Should I post this on YouTube? Aren’t there already a million tutorials all over the Internet already? What difference would one more video really make?

After a couple of takes, hitting record and walking through a simple explanation of the chords for Piano Song, I finally ended up with a take that I thought was decent. Not perfect. But not garbage. I flubbed the chords only once or twice and I didn’t say something completely stupid on that take. I realized it is quite difficult to play piano and talk at the same time (totally different from playing and singing at the same time) especially when the talking breaks up the rhythm and timing of just playing the song beginning to end. And trying to play the piano and speak coherent sentences is no walk in the park either. The main reason it took me so many takes to get just one right. But I was really just playing around at this point. This song would be my “proof of concept” song. I would record it, edit it, add chord descriptions using some form of video editing software (Windows Movie Maker anyone?) and then post it to YouTube and see if I got any bites. This would be the first song, one of many I planned on doing depending if people liked it or not. If it was well received, then maybe I’d do another and another. What if made a video of every song I’ve ever learned how to play? Would people actually watch them? Would people like them? I asked myself all these questions, but ultimately knew there was only one way to find out. I’d have to post it and scariest of all…make it public.

I have to admit though, I am terrified of YouTube. And by that I mean YouTube as a platform for content creators not just for the casual viewer. YouTube is a melting pot for so many different types of people with complex varying interests from all over the world. They get billions of views per day and their market reach is near endless. And the worst part about YouTube is…the comments section. Here you can find the cream of the crop for some of the worst things ever said on the Internet. I just wasn’t interested in playing on that playground. Like the little grade-school boy inside me who liked to play piano instead of sports, the kids here are just too mean. Not to mention the fact that most of the stuff you can find on YouTube showcase people that have spent (what must be the better part of) their entire natural lifetime honing in on a skill that makes them seem almost super human. They have an unnatural ability to outflip, outrun, outplay, outperform, outdive, outrock, out-you-name-it, any talent you actually thought you had. I’m pretty sure there’s a 4-year-old kid that can wipe the floor with my sorry piano-playing skills. Not to mention that one kid who can pick out a Eb-add9-flat5-maj7-11th-over-C# just by hearing the one chord even without any other context (freaking amazing by the way). Makes you pretty much want to crawl back in bed and ask yourself, “What am I doing with my life”? Where comments like, “Kid can do this…I find a Cheeto stuck to my face” are not uncommon. Believe me, I’ve been there, I live there. Not necessarily with the whole Cheeto thing, but as someone who ditched Facebook a decade ago and don’t regret it one bit, I’m definitely not immune to the culture of social comparisons and mentally ranking where I fall in this vast world of talented individuals.

Which brings me back to the piano tutorials. I basically decided to throw caution to the wind and just go for it. I mean, why not? I’ve got about 100 songs in my back pocket that I’ve played for years that I could record today. And easily hundreds more I plan on figuring out just to grow the channel into hopefully something worthwhile. I don’t admit to being an awesome piano player. And I’m sure I will get comments where people tell me I suck or whatever. But ultimately I just decided, so what? I enjoy playing piano and I really enjoy playing all of the songs I’ve learned how to play. And if I can help even one person learn a great song, a song from a band that we both know and love, then it will be totally worth it to me. And as far as everyone else is concerned, well what can I say, haters gonna hate.

Anyway, the plan is to try and upload at least one video per week so that in one year from now I have 52 reasonably-produced instructional videos showing how to play different songs on the piano. For now my main interest will be to do songs that are originally played on the piano and/or are written for the piano. I don’t like adapting songs played on the guitar or other instruments and converting them into a piano arrangement. While I have done that for a handful of songs, I don’t typically enjoy doing them as much and am never quite happy with the way they turn out. So for now if it’s played on the piano and even better, it is primarily played only on the piano with very little other accompaniment, then that’s my jam. Especially because it’s usually easier for me to pick out subtle nuances in chords and chord variations (like inversions and augmented/diminished chords) if it’s just a solo piano playing without a lot of extra instruments backing it.

So without further adieu, I present the latest in YouTube greatness, Dan’s Piano Tutorials – An instructional video series teaching you how to play all your favorite tunes from the 80’s, 90’s and today. Take a minute to check out my channel which as of this writing has a whopping 5 songs including bands like Erasure, Morrissey, Matt Nathanson, Death Cab for Cutie and Coldplay. Click Subscribe to get updates on new videos and be sure to comment on which ones you like best! Also feel free to shoot me suggestions for songs you’d like to see me cover, they can be obscure titles, rare B-sides, deep cuts, or maybe just videos for songs no one has ever done before. I’m always open to a good “figure out this song” challenge. As always, thanks for supporting this blog and I look forward to seeing everyone over on YouTube!

https://www.youtube.com/user/danaktion/

Sample Playlist of my most recent songs from the Dan Marx Piano Tutorial Channel

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Setting up a 3-Way Speaker with MiniDSP 2×4 using REW

It’s been about a month since the completion of these speakers and it’s about time I do a write-up on how I set up and tuned these speakers. I’ve spent the last several weeks and many hours setting up and dialing in these speakers and overall I’d say I am very happy with the results of this setup. I doubt this will be my final tuning/crossover setup with the MiniDSP but for now I’m just enjoying the new speakers and how awesome they truly do sound. Because overall the speakers sound fantastic. Better than I could have hoped for. Of course, my opinion is somewhat/mostly biased. But they literally sound just how the response plots would indicate – flat and neutral. No frequency range sounds out of balance, neither too forward nor too timid. But before I get into the subjectivity that is describing hi-fi sound, let me share the setup.

The above block diagram details the setup for measuring and tuning the speakers. The setup starts with a pair of laptops, one running windows 7 with the MiniDSP 4-Way Advanced Crossover Plug-in while the other is running Lunix Mint Cinnamon and REW 5.19 Beta 4. The sound card is a Behringer UCA222 connected via USB. My measurement mic is a Behringer ECM8000 using the cal factors available from Behringer’s website. Home Theater Shack also has a popular cal table for the ECM8000 (newecm.cal) but I felt like it over-corrected above 10 kHz and wasn’t realistic of the actual response of my microphone. When I use it, it causes almost every speaker I own to have this rising tail at 20 kHz that I just don’t believe. Behringer’s correction factor is more subtle and even though I am sure it still does not perfectly represent my microphone, I feel like it’s close enough for casual home use. See my comparison plot below.

Also the Behringer cal table includes phase. I think the main reason people use the newecm.cal is because the correction factor goes all the way down to 5 Hz to satisfy all the bassheads need for insanely low frequency measurements. Since the mic needs 48V phantom power, I use a Behringer XENYX 1202FX mixing console (which is normally part of my home recording studio) between the mic and the UCA222. A Schiit Audio SCH-13 Sys Preamp allows simple volume adjustment of two different sources. For my modest setup at the moment, I have an old Sony DVD/CD player and an Apple Airport Express. Better sources are in the works, including a real pre-amp, but for now, this is what I’ve got. The Airport Express is more for just convenience of listening than any kind of critical listening. Check out this post here on my whole house audio system.

So I debated doing a really nice all-aluminum housing for the MiniDPs and in the end just installed everything into an old VCR enclosure. The big spring-loaded slot in the front makes for easy access to the USB cables that I keep tucked away in there when not in programming mode. It’s not super glamorous, but is functional and practical. Both MiniDPSs are mounted on standoffs along with four LM317 linear regulator kits I picked up for $2 a piece on Amazon. I had an old 15 Vdc wallwart that after tearing it apart realized it was complete garbage. It claims an output current of 1000 mA which is just about right for two MiniDSPs (they draw 150 mA each) while maintaining some overhead. But with only a single transformer, a full-wave bridge rectifier and one capacitor, this would not be my first choice in powering any audio gear. But it’s nothing we can’t clean up nicely with some decent linear regulators. The wallwart provides 15 Vdc and feeds the first LM317 which takes the voltage down to 10 Vdc. The second LM317 takes the voltage down to 5 Vdc and over to the MiniDSP. Each LM317 claims roughly 65 dB ripple rejection so long as Vin-Vout > 5 Vdc at 150 mA. So with two in series, we probably get around 100 dB rejection which should be enough for even the most demanding audio needs. Since the MiniDSP outputs feed the UPA-700 amp with a fixed 29 dB of gain, everything in the signal path and DC bias paths needs to be dead quiet in order to provide quiet, hum-free and hiss-free experience. I think I succeeded because even though there exists what I would call traditional background white noise in the setup, you only hear it from less than 1 foot away from the tweeter with your ear directly in front of the speakers. Once you back up a couple of feet, everything is dead quite. And this is with the original MiniDSP kit which only claim >98 dB SNR. It’s definitely plenty for me. And yes, the MiniDSP’s are on all the time. For curiosity sake, I plugged my little VCR/MiniDSP setup into a Killawatt meter and measured a measly 3 watts (or 0.003 kW/h) which if left on year-round amounts to only 26 kW/year (24*365*.003). I pay about $0.1/kWh at my house on average which amounts to a grand total of $2.60/year. So to leave on a pair of MiniDSP’s 24-7-365 costs less than a side of fries at Five Guys. In other words, it’s basically peanuts. Which oddly enough, you can also get at Five Guys.

I made six sets of speaker cables that are each 14 feet in length from a 100-foot spool of 12 gauge OFC speaker wire from Parts Express. I bought 12 sets of black/red screw-style banana plugs which dressed up the cables nicely and provide a clean transition from speaker cable to the binding posts at the back of the speakers. I wanted to do something fancier on each side, like bundle each set of cables into some nice black expanded cable sleeving but just ended up leaving the speaker cables exposed for now. It’s functional and does the job. No complaints here.

Here’s a quick shot of the back of the upper cabinet with the nice Dayton Audio Premium gold-plated binding post terminals and aluminum face plate from Parts Express. Normally I wouldn’t go all out and buy such fancy binding posts, but for this build I decided they would complement the high-end feel of the speakers overall. They are super nice and provide a robust quality and feel to the speakers. I definitely recommend them for that great premium look.

Here’s the basic microphone setup for doing near-field measurements of just the midrange driver. I moved the microphone around a lot during my measurements. I did near-field plots of each driver at 0.5″ distances and then 6″ and 12″ and 30″ and 1M distances (approximately) as well as in-room measurements. I haven’t done a lot of off-axis measurements yet. But I do plan to take more measurements as time permits. This mic orientation and distance do not represent all of the measurements taken/shown here.

So this is sort of how the setup looked. The old Dell Inspiron was used to connect to each MiniDSP and provide instantaneous crossover and PEQ changes while the HP Elitebook on the right was running REW for taking and recording all the measurements. I only played with one speaker at a time, so I could make all the adjustments to just one MiniDSP unit. Once I was satisfied with the settings, I saved a configuration file and then loaded the settings onto the second MiniDSP. I did some casual listening of various types of music with each setting. Then I would go back to making adjustments as needed and redo the process of tweaking one unit, taking a measurement, loading the second MiniDSP and then doing some listening. I basically did this for about a month. What I’ve really been doing with this whole experience is educating myself on not only the basics of crossover design but some of the more complex elements as well. I figured I have a really nice set of speakers built into a great enclosure, now is my opportunity to really understand the benefits of different slope rates or crossover types on a really nice set of speakers. I have at my disposal the ability to test multiple filter types and slopes and take tons of measurements to see just what makes one filter better than another.

So the measurement and tweaking process is definitely on-going. One of the banes of active crossovers I suppose. The constant fear that just another tweak will make it sound so much better, or 10 times worse. I had started with a Linkwitz-Riley, 24dB/octave, 200 and 2,000 Hz crossover with almost no PEQ and 3 dB padding on the mid and tweeter with a 0.08 ms time delay on the tweeter. The slope rate and frequency was mostly just pulled out of the air, though loosely based on tweeter Fs and cone break-up modes of the RS-180P-4. I think this woofer works admirably up at 2,000 Hz. My near-field measurements matched very closely with PE’s measurements showing 1st, 2nd and 3rd cone break-up modes at 4 kHz, 5.3 kHz and 6.4 kHz and then rolling off sharply after that. I picked 200 Hz for the mid because it is simply 1/10th the tweeter crossover frequency. That’s really the only reason. The overdamped sealed enclosure actually starts to rolloff at around 100 Hz naturally (according to my measurements) so 200 Hz is basically double that. Mainly, I didn’t want the speakers to be a 2.1 way where the lower cabinet is just a subwoofer producing only bass. It seems like anything between 200 and 400 Hz would be appropriate for this setup though I have no real reason to limit the upper frequency response other than perhaps to match the natural frequency where baffle step losses occur. From my measurements baffle step losses look to occur between 400 and 200 Hz and is about -5 dB.

As far as tweeter minimum crossover point, I remember from times gone past the rule of thumb being 2x the fs of the tweeter minimum with a 12 dB octave slope. Though this may work in general, distortion levels will rise as the tweeter crossover frequency approaches fs and power handling will decrease. So from that perspective, it’s better to maintain a minimum crossover of 3x the fs and 18 dB / octave slopes or higher, but only as a rule of thumb. I still would like to experiment with the RS28F-4 down at around 1,200 Hz with 24 dB/octave slopes and see what it sounds like. So far the lowest I’ve run is 1,700 Hz, which is where it is at right now with 24 dB/octave Butterworth slopes. Besides setting the overall level of the three drivers (in amplitude), I feel like the crossover frequency has the largest overall impact in providing the desired tonal quality of the speaker. So while it may seem like an arbitrary selection point, it can make or break your speaker system. But how do you exactly determine the best crossover point? For me it’s about trial an error. Try something out, give it a listen for few days/weeks and then try something else. Ultimately whatever crossover point sounds best is invariable the best point. Though it’s doubtful that any single point or crossover type, or slope is “the best”, there must be dozens if not hundreds of viable options. I know many textbooks tend to favor crossover points that will sum acoustically with the natural roll-off of a driver. Thus allowing an acoustical slope that is steeper than the electrical slope of the crossover alone. These crossover points have to be very specific to coincide with the natural roll-off of the driver so as to sum appropriately and provide the desired response. But does this actually sound better than simply picking any other appropriate frequency regardless of the driver’s natural roll-off?

I look at it somewhat theoretically and ask myself, which driver do I feel has the best opportunity to reproduce a given frequency range the most accurately? In the broader sense this is probably obvious from looking at the natural (without crossover) frequency response plots and distortion plots of each driver. 1″ tweeters do a horrible job reproducing bass/mid frequencies and woofers can’t produce treble/highs worth a darn. Obviously, but what is more subtle to me is the overlap frequencies of the different speakers. Take the case in these 3-way speakers where the 100-500 Hz region could easily either be covered by the dual 8″ woofers or the 7″ midrange. As is the case with the 1,200-2,400 Hz frequency region, could either be played by the 7″ woofer or the tweeter. Yet different crossover points even within these small frequency regions will certainly cause significant tonal balance and sound quality to change. Something a simple on-axis frequency response plot or phase response plot probably wouldn’t even capture. However distortion plots may reveal a little more into which drivers can produce which overlapping frequency range better, in addition to off-axis response plots. I bet the the 1-1/8″ dome tweeter produces 2,500 Hz tones “better” than the 7″ woofer when you consider phase, on and off-axis frequency responses, impulse response and distortion. Good crossovers are about the creating predictable polar response patterns as a result of multiple drivers at different Y and Z axes and have them sum appropriately. If I had enough time I could compare such measurements of each driver in the ranges where they have natural overlapping usable ranges and pick which one does it best. At some point the crossover point that is selected should be based upon the crossover of where the woofer begins to sound better than the tweeter and vice versa.

Here’s a quick shot of my little Schiit Sys passive pre-amp. I bought this mainly as a stop-gap until I made a decision on what active pre-amp I really wanted to purchase long-term. As a passive little unit, it works quite nicely, it basically provides a completely neutral response to any fixed-output analog signal and allows attenuation of that signal with reasonable resolution. Once I received the unit, I ran the Schiit Sys through REW because, well, why not! I was really curious about overall flatness, roll-off and left-right amplitude balance. I wanted to see just want this thing actually looked like electrically. I uploaded some of the plots here.

Basically to sum up the plots, at full volume (the 5 o’clock position) the SYS is about 0.4 dB down and nearly perfectly flat from 20-20,000 Hz with only a subtle low-frequency roll-off of about -0.05 dB at 20 Hz. Left-right balance is also excellent differing less than 0.05 dB between the two channels. I then took ten additional measurements with the dial at the 4 o’clock, 3 o’clock, 2 o’clock, etc. locations and continued to watch for frequency flatness and channel imbalance. The left-right balance began to diverge once the dial was moved off of the 5 o’clock position (full volume). At its worst the balance is separated by only 0.6 dB left to right while the flatness remains about -0.2 dB down at 20,000 Hz. I ran into noise floor limitations at low frequencies in my setup once I exceeded about 30 dB attenuation or the 10 o’clock position. Notice in the plots I had to zoom way in to even see anything substantial. At first glance the plots look like the loop-back reference measurement. At minimum volume the attenuation is basically infinite, equal to disconnecting the cable, but the limitation of my setup was around 70 dB (above 1,000 Hz and only 30 dB below it).

Overall the volume adjustment feels natural and increases appropriately based on dial location. The flatness performance is quite exceptional, at its worst being only -0.2 dB down though the L-R imbalance was less than perfect, it’s likely mostly inaudible. This kind of performance is quite good actually considering the simple use of just a wiper-style potentiometer. Also, the switch which selects between inputs causes no click or pop in the output, even with the output driving a fixed-gain amp straight to the speakers. So I am happy with this unit and feel it was a worthwhile addition to my little setup and can recommend it easily. It’s got a solid feel to it, construction is nice, and it’s really small, so it fits just about anywhere.

So I have no real stereo rack, everything just sits on the floor underneath an end table next to the couch. This is my best chance at having this system pass the WAF.

Another shot of the setup. An Airport Express ($25 on Craigslist) and a Sony DVP-S300 DVD/CD player (a $6 steal from Goodwill), the Schiit Sys, two MiniDSP 2×4’s in a VCR and an Emotiva UPA-700 amplifier. Quite compact and mostly unobtrusive.

And here’s a shot of the whole room. The speakers are close to the walls but it’s the only practical location for this room.

I’ll start off with a final response plot of the three drivers with the crossover settings at 170 Hz and 1700 Hz, they are Butterworth filters at 24 dB/octave on all drivers. The response is smoothed with 1/24th octave smoothing. Measurements are taken in-room at approximately 20″ from the top of the midrange driver. There is some active PEQ on just the two 8″ drivers and some subtle PEQ settings for the entire speaker. Overall, the speaker itself really needs no PEQ. Most of the EQ I added was just to compensate for room modes, not necessarily driver response imperfections. I grabbed a couple of screen shots of the REW auto-tuning mode. These pics don’t represent the final settings, but show you how it’s possible to create a target response from measured data and what PEQ settings in the MiniDSP are required to achieve it. I balanced the output of the drivers by adding 4 dB of attenuation to both the mid and tweeter equally. This roughly accounts for most of the baffle step losses and provides the overall flat on-axis frequency response you see here.

The above plots show multiple overlays of various different settings used while dialing in the PEQ, levels and crossover settings. The measurements are also done at varying distances. For any given tuning session I would quickly fill all 30 measurement locations in REW with multiple measurements from multiple settings. Either changing the slope type, rate or other settings. Even this graph only shows a handful of settings. Most of the time I don’t even write down what the settings are for a particular measurement. Again, I’m just trying to see what settings affect what and how it changes the what can be measured in REW. I used the target filter response option in REW to auto-generate PEQ settings and then just typed them into the MiniDSP. It does a remarkable job of predicting the new response each time. I was quite impressed.

This graph shows the basic final on-axis frequency response plot of the entire speaker with 1/24th octave smoothing and includes the phase response. Frequency response is roughly 30-20,000 Hz ±2 dB. I unwrapped the phase for a better visual depiction of what the phase response actually looks like. Phase is nice and linear (though not flat) and changing at a rate of about 120°/octave from 20-2,000 Hz. Then from 2,000 – 20,000 Hz the phase begins to reverse slightly and flattens out somewhat changing no more than 100° over this entire range. The tweeter alone pretty much produces this kind of phase response no matter what EQ/crossover settings I used. The tweeter phase is not inverted from the woofers and there is no electrical time alignment on any of the drivers. Since most people don’t talk about what phase actually sounds like, I don’t really have a good feel for whether or not this kind of phase response is good or bad or just otherwise unremarkable. I’m going to start playing around with the timing and crossover slopes to see if I can shift this phase response around and see what that sounds like.

Lastly, here’s a few screenshots of the 4-way Advanced Crossover Application from MiniDSP. Here are some of the windows available for setting up the MiniDSP, such as the crossover freq, slope rate and type, the PEQ for the woofers and the output level settings for the woofer and midrange which shows the -4 dB for the mids. The tweeter is also -4 dB with no time delay on any of the drivers. Nothing too fancy otherwise. The interface is functional and intuitive and changes are made instantaneously.

So that’s basically it. Not much more I can say about these things. Though there is one other thing I can share that I did that I thought was kind of fun and hadn’t seen anyone do. I put the mic in the exact location that I took most of these measurements and then recorded an entire song through the mic at a reasonable volume and imported the recorded song into a track in Audacity along side the original .wav file. Then I bounced back and forth between the recording and the actual song with a pair of headphones and I was honestly quite amazed at how similar they were. Aside from the recording sounding more “live”, everything else was spot on, the bass, treble, midrange, vocals, they all sounded fantastic and balanced nicely with the original source. If my room had more acoustic treatment on the walls and floor, I bet I could have narrowed the gap even more between the recording I made and the original, since that was really the only defining different between the two recordings. That and the fact that I converted the song to mono since I was only capturing a single speaker with the mic.

But you be the judge, click play on the audio clip in the media player below to listen to a song recorded with a mic in front of the speakers against the original source CD. The first 15 seconds is the original CD while the the next 15 seconds is a recording of speakers and so forth. The audio alternates every 15 seconds between the source music and the speakers until the song ends. The song is I Only Have Eyes for You by Harry Connick Jr. It’s an okay song for speaker auditioning, though arguably I could have picked something more jazzy and upbeat, it’s got some good vocals, a decent bass track, some piano and strings that can help showcase a speaker’s overall range. Put some headphones or use some good computer speakers and have a listen. See if you can spot the transitions without watching the time. Enjoy!

I Only Have Eyes for You – Harry Connick Jr., Recorded with the DM-4 Speakers

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DM-4 Reference Speaker Build Part V – Finally Complete!

The speakers are done! I’m super excited to finally unveil the completion of the DM-4 Reference Loudspeaker. It’s been crazy though, the last few months have been such a roller coaster of different, unexpected projects, as is evidenced by my last few posts. It seemed like with each new project this speaker build kept getting put off more and more. At some point I said to myself, I just need to finish a couple of these other things on the to-do list and then hit the speaker project full-time and just get them done. This has probably been the longest time I’ve spent doing a single speaker build. From conception to completion this project taken almost 3 years. It got the point where in the middle somewhere I completely lost interest in the whole project and began designing a completely different set of speakers. But alas I persevered and now I can say that I am super happy I did because these speakers turned out amazing! Well, in looks at least. As of this post I haven’t hooked them up and actually listened to them yet aside from my initial tuning of lower enclosures which I did a few months ago. The method for creating the crossovers and doing the measurements will have to be saved for another post. For now I just wanted to get the rest of the build pics up and of course the final shots of them sitting in my living room just looking awesome.

It’s funny because as with so many DIY projects, at some point you just have to accept the little flaws in a project and recognize that this thing is made by the hand of an imperfect person. I know I get way too picky about the fit and finish of the final product that can’t enjoy it for what it is. Overall these speakers went together without too much fuss. I’ve been working with wood and MDF and building speakers for many years so I’ve come to understand what it takes to get the job done without making too many mistakes. But when it comes to the final paint and finish work, I’m a complete mess. For this project things were looking good all the way up until I added the Polycrylic clear coat. Staining the birch was a piece of cake, it went on like any stain, really cleanly and looked flawless once complete. I went out on a limb and did the black surfaces with just spray paint and to my surprise that also went on really well. I think I did 3 coats of gray primer and then 2 coats of flat black. I had masked off the birch while spray painting and when I lifted the masking for the first time I got a glimpse of what these speakers were going to look like and I was super excited.

The gray stain contrasting the black just made a killer combination that just looked different from the usual speaker. So I was excited and pumped to put on that first coat of clear coat to give it that clean, smooth, subtle shine to it. This is where it all fell apart. I seriously have to find a better method for clear-coating speakers. There has to be a better way. I won’t go into the details, but I ended up with 4 coats before I just gave up and said it’s as good as it’s going to get. After each coat I’d find a drip, or a spot where I’d brushed it again when I shouldn’t have, or a spot I missed completely. So I’d sand it down and put on another coat, only to miss another spot, over-brush a different spot and end up with a run somewhere else. After sanding down that coat and doing it again and again I just quit and said it is what it is. So that’s when I decided you know what, they’re not perfect and overall they look as good as I ever could have expected them to look so I’m just going to say that I’m happy with them for now. But if and when I learn how to do a proper clear coat, I’ll remove all the drivers, sand this top coat down once again and finish them proper. But that’s down the road, like, way down the road.

Well I should do a write-up on the final build of these speakers. They definitely deserve it, there are so many little details that went into every aspect of making this speaker. You can probably gleam how they were built from all the pics but a few words won’t hurt. I’ve already written quite a bit leading up to this point. A lot of finishing these speakers is just duplicating what I did to the lower cabs. I had just finished building the upper cabinet when I stopped late last year. So I will pick up this post with adding the 1/4″ MDF to the front/top/back/bottom and cutting the 45° chamfer around the edges. The 1/4″ MDF worked out great and gives a clean finish to work with as far as painting goes. Plus it allowed a perfect flush fit for the Dayton Audio drivers. I used Parts Express’s sweet Jasper Jig Model 240 to router perfect holes for the woofer and tweeter. This took a lot of trial and error though! I think I went through 5 different boards before I found a size that allowed the drivers to fit just right without the gap being too big or too small. I was shooting for a gap that was at least 0.005″ but no more than 0.015″. That’s basically a baffle diameter that is 1/32″ larger than the diameter of the driver basket. This allows for some growth from paint/clear coat but no so much that the driver can actually move around. The smaller the gap, the better the finished speaker looks, in my opinion. I did error on the smaller side for the tweeter and I had a tough time getting it fit after the 4 coats of clear coat that lined the inside edge. But since it is probably the only driver anyone is really going to look at, I wanted to make sure the transition from the tweeter’s faceplate to the baffle was absolutely seamless. Plus with the Jasper’s fairly course 1/16″ increments (or 0.062″) I didn’t have much choice with the tweeter as one size was spot on at 0.000″ and the next size bigger left a whopping 0.0315″ gap around the whole tweeter. That was more than double what I had already agreed was reasonable from an aesthetics perspective. So I cut it with no gap and then sanded it slightly to get the tweeter to fit. I then just tried to be careful with the paint and clear coat so as to not decrease that size too much. It worked out, because the tweeter fit by the skin of its teeth and the gap is nearly zero.

The next step was adding the acoustic insulation – I reiterate what I had stated about doing this insulation technique on the lower cabinets that it was an absolute pain and was super time consuming but I think in the end it is going to provide superior damping and absorptive properties than some of the traditional (and simpler) methods. I took a bunch of pictures of each layer going into the cabinet and then made a “sample coupon” on a separate piece of MDF just so show a cool cutaway view of the built-up layers of materials. In short each upper cabinet is covered with 2 layers of 1/2″ (6-lb.) standard carpet pad following by 4 layers of a 80/20 cotton/poly blend fabric/fill and then 4 layers of a 100% poly batting material. The total thickness ends up being almost 3″. Each layer is glued onto the previous layer with spray glue to create one cohesive material that won’t move around and hopefully provide consistent absorptive properties throughout the enclosure and in both enclosures. The upper cabinet is sealed and will have an approximate high-pass filter of around 200-400 Hz so for the most part I was shooting for a nearly-full enclosure, but still ended up with a solid pocket of just air with no fill in just the middle portion. So it’s probably like 75% filled if I had to guess? The lower cabinet only has the walls lined and is probably only 15% filled.

Alright so now the cabinets are done it’s time to start working on the finish. I picked up a can of Winwax “Classic Gray” and put 3 coats of stain the on birch sides for all four speakers. I didn’t bother masking the bare MDF since I was planning on painting it anyway. I should have stained those raw MDF edges better though because the spots I overbrushed with the stain actually sanded nicely and cleaned up that rough MDF edge look and then painted even better. But I did what I could to sand the bare MDF edge so it didn’t look like crap. In the end they are not perfect, I could have done some better edge prep, but they look fine overall and am not going to worry about it too much. After letting the stain dry I masked off the sides and began preparations to paint everything else. I picked up 3 cans of Painter’s Touch 2X gray primer and 3 cans of the flat black. Over the course of the next couple of months I went out to the garage and painted a coat of the primer, then sanded it, then did another coat and sanded it, until I had a really nice flat base finish. Then I moved on to the flat black paint. I also sanded between coats until I had a near-flawless black finish. This spray paint went on really well and I would definitely use this spray paint again. The flat black is extremely flat and I wouldn’t recommend it as a final top coat. It needs a satin or semi-gloss clear coat of some kind to finish it off.

Once the paint was all dry I brought the speakers in the house for the first time. I didn’t want to do the clear coat/top coat in the garage because seriously anything that sits in my garage for more than 5 minutes is immediately covered in a layer of dust/saw-dust/paint-dust/you-name-it-dust. It’s not like I’ve got a 10,000 clean room at my disposal so the house will have to be good enough. I masked off the floor and my coffee table and set up the speakers for their first coat of Minmwax Polycrylic Satin clear coat. I’ve used this stuff before on countless projects and usually praise the ease with which this product goes on. It’s usually forgiving of brush strokes, provides a really consistent sheen across its surface and it’s water based so clean-up is a breeze. But I realized that painting over a solid black surface is not as easy. Every blemish in the coat was readily visible. Any imperfection in the base finish showed through and any area in the top coat that wasn’t applied perfectly evenly was visible when the light shown off it at certain angles. Thus began the paint-sand-paint-sand-paint-sand-paint nightmare that lasted about a week. And yes, with the speakers sitting in my living room just like this pretty much the entire time. Anyway, I called it good enough after that last coat and accepted the fact that once again I’ve nearly ruined a perfectly good project with a shoddy top coat. I started looking at past speaker projects and realized that none of them are perfect and that surely after a while I’d soon forget all about it anyway. I can say that birch sides look great. The clear coat went on and after only the second coat looked great. So I will say that real wood, with a grain and some texture to it, is way more forgiving of the irregularities in the top coat. But that solid flat black was something else. Especially since the fronts have the holes for the drivers, brushing around the driver openings and trying to get it look seamless was just a pain. I’m doing a spray top coat next time. I’m sure that goes on easier.

Alright so now to the last part, the installation of the drivers! This part was probably the most fun. I took my time on this part, I did only a pair of drivers a day for basically a week until all four speakers were complete. I made up the internal wiring using 14 gauge 100% OFC copper wire and crimped them to a set of 0.250″ gold plated female disconnects from Parts Express. I went all out and bought the fancy Dayton Audio 5-way binding post kits with the black aluminum faceplates from Parts Express. These things look super nice and are definitely a step up when compared to a good old fashioned terminal cup. I won’t claim they offer much sonic benefit other than being a robust, gold-plated, low-loss connector, but they look stinking awesome and definitely match the high-end look of the speakers overall.

I attached each disconnect to its appropriate spot ensuring I maintained proper polarity on each connection. I’ve been know to look right at a red connector and plug it straight onto the black connector like a moron. I took pictures of each connection just so I could be sure. Nothing’s worse than trying to diagnose polarity issues in a speaker system via measurement techniques alone. It is possible though, but I’d rather not get into that situation where I don’t know if the polarity is inverted or not or double-inverted and therefore not inverted. The MiniDSP allows inversion of any of the 4 outputs so there’s flexibility in the processing to get the polarity right but it’s better that everything be correct from the get-go.

I applied a strip of the 1/8″ x 1/2″ open-cell foam gasket to backs of the 7″ and 8″ drivers and applied two strips to the back of the tweeter, since the faceplate is thinner. The basket thickness of the woofers is about 0.210″ and my baffle thickness is 0.250″ which leaves 0.040″ (about 1mm) of uncompressed gasket to provide a perfectly flush fit. The gasket I received from Parts Express was actually about 0.140″ thick and was not easily compressed to 0.040″. (I had purchased another role of gasket but it was only 0.110″ thick and was significantly softer so I didn’t use it). It could compress more under the pre-load provided by the six (6) 8×1 screws but since I shooting for a flush fit I only tightened the screws enough such that the basket sat flat to the surface. This is the only downside to using just a piece of 1/4″ MDF to flush mount the drivers. The drivers are not seated tight against the baffle, I mean they are tight, but there’s still 1 mm of foam between the driver and the baffle. Normally you would tighten the screws super tight, or reasonably tight such that the gasket reaches “full” compression. Whatever that is, you decide. It’s not going to zero no matter what, so when you do design for a flush-mounted driver you have to account for some amount of gasket under the driver that makes it sit higher than just the thickness of the basket. As it is this material has a fairly soft durometer, it’s probably in the 20-30 region based on a Shore 00 rating, if I had to guess. Uncompressed the thickness is 0.140″ and and final compression in my design it is 0.040″ which represents a compression ratio of 71%. Honestly I couldn’t turn the screws much more to get that compression any higher without stripping them out. After all it’s only MDF with a pilot hole, I did not use any kind of hurricane nut or t-nut (I just find them way more hassle then they are worth for small drivers such as these). The drivers are definitely in there and the seal is surely air-tight. Taking them out later on will be a pain since I’ve found this soft gasketing material tends to bond itself to both surfaces once compressed and proved quite the challenge when removing drivers years later.

Anyway, that pretty much brings me to today. The drivers are installed, the speakers are set up in their (probably) final location in my living room and so I snapped some pictures so I could show them off. Oh yeah, the other thing I ended up doing for the feet was I bought a set of sixteen (16) of the Penn-Elcom rubber feet from Parts Express. They are 1″ in diameter and 0.375″ tall. They were only 57 cents a piece and worked out awesome. I have wood floors so moving them around on the floor is a piece of cake plus they stick quite nicely. The top cabinet is definitely not going anywhere and the height it just perfect. So if you’re debating on getting a full set of toe spikes at 10x the cost, my recommendation is just go with the rubber feet. I doubt they provide any measurable difference in sound in the end over anything else but they are super easy to install and are essentially invisible in the final product.

Here’s the gallery of pics starting with the upper cabinets and putting on the 1/4″ baffle boards forward to completion. I can’t believe this took five blog entries to complete. And I wrote a lot. I’m sorry for being so long-winded, but hopefully some of this information is helpful if you are working on speaker project of your own. There is so much that goes on behind the scenes with a good speaker design. I’ve only really touched on a handful of design concepts and ideas that I considered while making these speakers. Not to mention I still have to measure them and design the crossover for them, sot that’s all to come. Until then here’s a bunch of pictures of the completion of the DM-4 Reference Loudspeaker System!

And if weight provides any indication of the build quality of a speaker, here’s the stats on these puppies: the lower cabinet weighs in at 96.0 pounds and the upper cabinet is 45.6 pounds bringing the total weight per speaker to 141.6 pounds! No wonder I nearly bust a gut every time I have to move these beasts.

Click here to go to a complete gallery of final build pictures!

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How I Built an Awesome Classic-Style Bookcase from Scratch with Mixed Lumber from The Home Depot

So I’m trying something a little bit different. I’ve been blogging for a long while now and while I feel like it’s still a great venue for sharing DIY projects and what-not, the bigger and better trend these days is vlogging – the video version of blogging. YouTube is way bigger than Blogger ever was and for most people it’s the sharing platform of choice. Me, I honestly hate shooting video of my projects. It just takes extra time, it distracts me from the work I’m doing, and most of the time I just don’t know what to say on the video. But for this project I made a conscious decision from the very beginning, for good or bad, that I was going to turn this bookcase build into a video blog. Just to try it out, see what I could turn it into. So in addition to a very short write-up on my blog, I’ve created a 4-part video series of this entire project from beginning to end for your viewing enjoyment!

Honestly though this will probably be the last time I do this kind of thing. As much fun as it’s been working with one of the most useless and frustrating video editing tools known to man (Windows Movie Maker), I really don’t think that many people are going to sit down and watch this thing front to back. But I feel like if you are interested in building a project similar to this, then I hope there are some nuggets of useful information in the 30+ minutes of video I pieced together of this project. I will say that did have a lot of fun throwing in old songs I had written, some from as old as 20 years ago as well as some songs I wrote just last month, to back the videos and at least make it somewhat more watchable (though depending on your music tastes, that’s debatable). I’ve got hundreds of these old songs I’ve composed over the years that I never know what to do with, most are just synth beats with some melody and a bassline, reminiscent of synth pop music from the 80’s. My son enjoys listening to them, and so for some reason I keep writing them even if they are lame. Anyway, I hope you enjoy the soundtrack to the Benjamin Bookcase Project. I think I will leave it at that for now. I will do a more complete write-up here in the future. For now I’ll just post a few pictures as well as links to all 4 videos on YouTube.

As always, thanks for stopping by and I hope you enjoy the pictures and my first ever video blog of the Benjamin Bookcase Project – A classic DIY bookcase featuring victorian styles including crown and fluted mouldings mixed with old-fashioned farmhouse comfort with the shiplap backing and clean white features. This is an all-white, 6-shelf, classic-style bookcase measuring 12-1/2″ deep by 70″ wide by 100″ tall. This bookcase looks great in almost any home mixing a little bit of old with a little bit of new to make a unique center piece for any living or family room. More info about this project will be posted here. Pin it if you love it!

Benjamin Bookcase Video Part I

Benjamin Bookcase Video Part II

Benjamin Bookcase Video Part III

Benjamin Bookcase Video Part IV

Mega-Picture Gallery

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