Homemade Speakers
chrisfenton.com
chrisfenton.com
To my ears, the best and most musical sounding speakers I've heard have time-aligned drivers and very simple crossovers. Tannoy, Spica, Vandersteen, and other such designs are clearer and less fatiguing. I'm a musician and have recorded numerous albums, and those speakers are the ones that match what I'm used to hearing the best.
The fundamental problem is that we work on what we can measure. It's very easy to measure frequency response. It's very hard to measure phase alignment. So we fix what we can measure. If you want to see serious map-over-territory thinking, look at THD specs for amplifiers. It's super-easy to measure THD - just isolate harmonics of a sine wave a 1khz. Unfortunately, this has approximately zero to do with music, unless your idea of "music" is static sine waves. Recorded music has a 20-30db dynamic range (less in the case of modern pop) and covers ten octaves. Dynamic recovery behavior, intermodulation distortion, stuff like this is what gives amps their distinctive sounds - but it's nearly impossible to measure! So they sell what looks good on paper... THD. Sigh.
Under contrived situations such as square waves and headphones. However, in general, the ear is not particularly sensitive to phase. [0] [1] which both discuss, in particular, [2] and [3] as well as many others listed in their bibliographies.
The stated conclusions are, emphasis mine, "Given the data provided by the above cited references we can conclude that phase distortion is indeed audible, though generally speaking, only very subtly so and only under certain specific test conditions and perception circumstances." [1]
Reads may also be interested in an additional conclusion which reads thusly: "Room acoustics further masks whatever cues that the hearing process may depend upon to detect the presence of phase distortion." [1]
[0] http://www.silcom.com/~aludwig/Phase_audibility.htm
[1] http://www.audioholics.com/room-acoustics/human-hearing-phas...
[2] Lipshitz, Stanly P., Pocock, Mark, and Vanderkooy, John, "On the Audibility of Midrange Phase Distortion in Audio Systems,' J. Audio Eng. Soc., Vol. 30, No, 9, Sept. 1982, pp 580-595.
[3] Toole, Floyd E., "The Acoustics and Psychoacoustics of Loudspeakers and Rooms - The Stereo Past and the Multichannel Future," 109th AES Conv., Los Angeles, Sept 2000.
> the human ear is largely insensitive to moderate variations in volume/frequency... but it's incredibly sensitive to phase.
No, it's not. The human auditory system is sensitive to time variation. Phase shift may contribute a time shift but only at frequencies low enough that their wavelength has reasonably relation to the spacing of the ears. For example at 10Khz the wavelength is far too short for 'phase' to impact arrival time. Ultimately what actually matters in this regard with the human auditory response is group delay, not phase.
> Phase is how we discern directionality (stereo), among other things.
As mentioned above, what matters in the context of what your saying is group delay, but that is far from the only thing that matters for directionality. The human psycoscoutic system is very complex, part of directionality is group delay, part of it is frequency attenuation caused by sound wrapping around the ear from the ear, part of it is a half dozen other things. 'Phase' doesn't come even close to being a catch all cause.
> To my ears, the best and most musical sounding speakers
And thus we've devolved from science, as this usually goes. What on earth does 'musical sounding speakers' even mean?
> It's very easy to measure frequency response. It's very hard to measure phase alignment.
I have no idea where you came up with this statement but it is very easy to measure the response of a loudspeaker, both in amplitude and phase.
> If you want to see serious map-over-territory thinking, look at THD specs for amplifiers. It's super-easy to measure THD - just isolate harmonics of a sine wave a 1khz.
THD can be measured at only 1Khz, but it isn't something intrinsic to what 'THD' means. Measuring at only 1Khz is generally indicative of a crappy amplifier manufacturer looking to inflate their power numbers. Proper specs will provide a 20-20kHz THD rating.
> Dynamic recovery behavior, intermodulation distortion, stuff like this is what gives amps their distinctive sounds - but it's nearly impossible to measure!
If an amp has a distinctive sound, it has failed to achieve it's core design goal. I'm not sure what you think is impossible to measure, but I assure you it is not.
This is only true if you can remove biases.
Want an A/B test? Put your hi-fi in the same room with a real singer, or an acoustic guitar, or whatever musical instrument, and see how hard it is to tell them apart. Not very, I assure you.
Interestingly, we can pick up very subtle and useful musical cues from very poor recordings and reproduction. It's a complex thing.
It's the time-of-flight difference between when the one ear receives the wave front and the other from a change in the sound that gives us the directional information.
Absent any change all we have to go on is volume so we rotate our heads to the point where both ears receive the signal equally strong, then the source is somewhere in the plane that bisects the head of the listener.
Rotating one ear forward gives us the clue about whether the sound source is in front of us or behind us. (Fails to work when it is directly overhead.)
That's part of it, the other part is the head related transfer function is the change in response of sound due to diffraction effects caused by the head, upper torso and pinnae. The pinnae also help us to judge elevation.
What makes you think the design goal of an amp is always to have perfect sound reproduction? That may be true for a home-theater system or if you're playing wind/string instruments, not so for a guitar amp. We have the (digital) tools to get pure, 100% uncolored sound, and musicians hate it.
Of course not. High-end monitors are high-end precisely because they achieve indistiguishability. Between to pairs of speakers from different brands, you wouldn't be able to tell the difference.
(As far as I remember, headphones are actually harder to get precise response curves due to the interaction with the skull and precise physiology of the listener, but I may be mistaken in that.)
The problem with any claim from people that they can "easily" tell two pairs of headphones apart or two pairs of monitors apart is that most of the time these claims are not scientifically validated. There are numerous biases and gotchas involved in measuring audio fidelity, and one must be aware of these when designing experiments. (And the whole "audiophile" industry is based on the idea of selling snake oil technology to people with fat wallets and who think that they have better ears than anyone, and they go to great lengths towards denying the science.)
The mixing and mastering engineers aren't trying to make some perfect reproduction of natural sound. They're trying to make records that sound as good as possible on as many different kinds of reproduction systems as possible - not just "perfect" audiophile systems, but car speakers, iPods, etc. As such, rather than going for accurate speakers, mix engineers rely on speakers that they know very well, so they can predict results elsewhere more easily.
The most popular professional mixing speaker is the Yamaha NS-10. It's not "accurate". It doesn't even pretend to be accurate. In fact, it has a pronounced peak and significant harshness around 2khz, right in the most sensitive area for vocals and midrange melodic instruments. Why use it? Because if you can make it sound good on the NS-10, it'll sound good anywhere. Likewise, the second most popular speaker is the Auratone, a single driver with limited frequency range. The Auratone has two advantages. First, it reflects the limited construction of many real-world speakers. Second, because it lacks a crossover, there's no phase weirdness in the midrange, so it's actually very pure at the most musically critical points. Deep bass and sizzly highs aren't important. Midrange is important, and Auratones are brutally honest at that, more than speakers costing orders of magnitude more.
This is a critical aspect of crossovers- since they contain the same signal, any phase differences between the intersecting components are easy to pick up by the ear and very difficult to measure since it won't show up as a significant frequency or amplitude differential. In the highly acoustically sensitive area where the crossover occurs, the phase distortion between the two signals is hard to pin down but it's definitely there, it's easy to hear by A/Bing.
I mix on a multiple-monitor setup and the one that is most critical to me sounds like absolute garbage. However, it has a wonderfully "flat" response and allows you to get an idea of what proven aesthetically-unpleasing issues are present. It has no crossover. The goal is not to have perceived rumble, tinniness, or mud on any of the incredibly wide range of listening systems out there. I think this is what beat means by "musical"- flat speakers are actually not pleasant to the ears, but the most empirically useful.
The most famous secret of many respected mix engineers is the Auratone. Michael Jackson's "Bad" was mostly mixed on this little thing. It sounds like absolute garbage aesthetically but reveals more issues in a mix than anything else. It's like when you first saw things under a "blacklight" as a kid and got to see all the particulate matter covering everything, that you can't see with the naked eye.
Here's more about the Auratone. I hope my comment has helped to bridge this critical area of "musicality" vs. empiricism. http://www.trustmeimascientist.com/2012/02/06/auratone-avant...
Postscript- the listening environment is the most important aspect! Always! You can have the best mastering grade monitors on earth in an improperly treated room, and it's all for naught!
I wouldn't go that far. If you have two signals emitting from physically separate location and they are out of phase, you just get comb filtering. Is it audible? It can be, but its not the phase you're hearing, it's the drastic frequency notching in amplitude. More importantly, you're going to get comb filtering no matter what you do with a multi-speaker setup. Just move your head a couple inches out of the ideal sweet spot equidistance from each speaker and you'll have created an effective phase shift and get the same type of comb filtering. In other words even with a theoretically perfect pair of time aligned speakers with perfect cross-overs, you'll still get comb filtering if you take various measurements around the listening area. Just moving the mic 4 inches can have drastic effects on the measured response. Incidentally if you've ever seen someone taking a measurement with a sound meter and rhythmically moving the mic around in a strange fashion, they are doing that to try to even the effects of comb filtering.
> This is a critical aspect of crossovers- since they contain the same signal, any phase differences between the intersecting components are easy to pick up by the ear and very difficult to measure since it won't show up as a frequency or amplitude differential.
If you're taking a measurement in the crossover region and there is a phase shift between the drivers you most certainly will see an effect in amplitude as you'll have at least partial wave cancellation. You can certainly take measurements in locations what won't show this, but that is always true. You can take measurements of a perfectly time aligned speaker that make it look like it has phase issues too, if you put the mic in the right spot.
Likely the largest improvement you get from time alignment of the drivers in a multi-way speaker is the ability to control vertical lobe tilting, but you can 'fix' that issue with MTM layouts without time aligning the drivers as well.
A similar argument which I think more people would agree with is that reflex-loading a speaker hurts the low-frequency group delay, the lack of which is one reason suggested for the supposed clarity of the classic NS10 monitor: http://www.soundonsound.com/sos/sep08/articles/yamahans10.ht...
I'm a musician and a recording engineer. I've made records, I listen to real-world instruments every day, I've built speakers, and I've built amplifiers. And I've learned along the way that "accurate" is a big stinking pile of BS, and "scientific" is usually just a euphemism for magical thinking - "If we can't measure it, it doesn't exist". If you can't measure it, maybe you don't understand the problem as well as you thought.
You don't get to ignore the evidence of the senses of domain experts just because it makes you uncomfortable.
From an EE perspective most of the "obvious" ways to screw up THD sound truly horrible. Crossover distortion, clipping, bias problems, truly excessive hum or noise...
Also note that just because your final PA is a modern class D doesn't mean there's not some opamp input stage in the box that is perfectly liable to classical transistor bias problems. So you can't get crossover distortion out of a modern class D amp final device as an inherent part of how those switch the transistors, but that doesn't mean you can't screw up the input stage of the opamp that is inside the box of an amp that none the less has a D-type output final stage. In terms of circuit design, A guy who can design a nice class D stage is not necessarily (but often is) the same guy who can design a class A or AB stage.
"Necessary but not sufficient" is a good way to describe good THD numbers.
I would have to think for awhile how to get a good THD number with foul intermod numbers... Maybe if you fed in two noise signals outside the freq bandwidth that when nonlinear mixed gave a difference freq of 2 KHz that would sound horrible even if a pure sinewave input at 1K sounds awesome. It would take some work to screw up an amp this way but it could happen.
Anyway nothing says "tradeoff" like engineering and I wouldn't trade off a pretty basic core characteristic like THD for improved ... anything, except maybe power level. I'd rather hear something good than something a couple dB louder. Reporting good THD is valid in marketing as a way to tell that you did at least the very first homework problem... I agree totally with you that there are multiple steps beyond it, but there's no point in working on any of them if you can't pass the first simple THD test. (Edited to add, I missed the PERFECT analogy: Its like passing the UL Labs certification that it probably won't burn your house down. That's kinda required, and its also possible to aim a little higher)
The mainstream A/B transistors + heavy negative feedback architecture has the benefit of low manufacturing cost (no need to match devices when you just feedback the differences out) and good specs. But where does the distortion happen? At the Class A/B crossover point, somewhere in the first watt. Until the amp hits maximum power and clips, the WORST distortion is in the first watt, where all the small signals live.
To my common sense and my ears, this explains how good-spec amps can sound very harsh and cold, while bad-spec amps can sound sweet and warm. It's not "euphonic distortion", as the pseudo-scientists claim. Rather, it's that the warm amp is actually distorting less where it actually matters.
Easy-to-build bookshelf speakers, the Overnight Sensations--$136 with free shipping: http://www.parts-express.com/overnight-sensations-mt-speaker...
And the Amiga towers are great for a living room, more towards $300 for the pair: http://www.diysoundgroup.com/speaker-kits/amiga-kit.html
More Amiga info: https://sites.google.com/site/undefinition/diy/amiga
I have built both, they're both really great at what they do. If I ever have the free time and money, I'd love to build some Statements: http://speakerdesignworks.com/Statements.html
Personally, I replaced an ~$800 pair of speakers with the Amigas, and they're vastly better in terms of clarity/resolution. I have a decent tube amplifier already.
Past that point I'm pretty sure room setup/room treatment starts to matter much more than hardware...
With aftermarket tubes: http://psvanetube.com/wordpress/purchase/shuguang-treasure-s...
The amp is super heavy (40+ lbs), so shipping is expensive.
I like it a lot--definitely changes the sound to something warmer. It's also a beautiful object, especially in the dark. The aftermarket tubes helped resolve bass, but are probably overkill (I used them as a reward to myself for crunch on a contract gig).
Note that tube amps have a lot of downsides--you can't leave them on 24/7, they make a lot of heat (so can't stack with other electronics), you need to tune voltage during the first few months of operation, only stereo output with minimal AV switching, etc...
But yeah, if you're just looking for a cheap set of speakers, and don't care about the actual DIY angle, making your own might be feel like a frustrating time sink...
http://www.cs.helsinki.fi/u/ljlukkar/labsub/
I’m currently working on a Arduino based pre amp that hosts a rest api for remote controlling: http://tinypic.com/r/15yvj2g/8 http://tinypic.com/r/11vmhld/8 http://tinypic.com/r/jrzfk5/8
I don't think I'll be building speakers anytime soon though, but I'm still very much interested in the details of psychoacoustics and sound reproduction.
http://www.amazon.com/Acoustical-Foundations-Music-Second-Ed...
Most of all, I'd like the pre amp to automatically turn on the power amp (currently a Gainclone) when a signal starts coming in from the connected Airport Express.
http://diyaudioprojects.com/Solid/DIY-Lightspeed-Passive-Att...
Here's another arduino-based pre-amp. This one includes relay-based stepped attenuators and input selectors. The pcbs are all available as well as the schematics in case you just want to use it as a base for your own project.
Back in the 80's, designing and building your own speakers was a suitable project for high school students. Still is.
Where it's rewarding is taking available drivers and building a full speaker around them -- usually optimizing for performance vs cost. It's difficult and expensive to build really high quality loudspeakers. Add markup to that and the retail price of a well engineered speaker will quickly go into 5 digits.
The speakers in my living room are DIY (though they don't really look "DIY" as far as that might carry negative connotations) and they're an excellent value. They were also fun to build. In no way was I disappointed afterwards because I had not, say, wound the voice coils myself.
For a brief time, I had a pair of speakers on my desk made from old PC case speakers that I salvaged and just put on stands made from blocks of wood with a hole in them. I liked the look, and the sound was alright, but rather weak and tinny. Then one day I decided to build a box around them, and the difference was night and day - much louder sound with significantly more bass.
There is a big difference between just making a box, making a box that looks nice, and making a box that cooperates with the specs of the driver to make a good speaker.
You might be surprised how awful the specs are for a bare driver hanging by its wires in the air without an enclosure. It is very much like the analogy of a musical instrument reed.
I for one would have loved to do this kind of activity as a kid: it's a pretty simple concept that puts a whole lot of power in your hands and exposes interesting questions. It looks like Bose is doing something in this space at kids science camps [0]. If you can acquire/make an amp and a strobe light to replace their "blue box", there's a whole host of kid-friendly activities on the site to do with kids centered around making hand-made, origami paper drivers.
The "D'Appolito array" is covered in that book--a design employed in many store-bought speakers where two mid-woofers flank a tweeter at a specific distance to avoid some of the time alignment concerns raised here. This is a good candidate if you want to build smallish, narrow speakers for home. In the car it usually comes down to "where is there enough room for the drivers I want?"
I think the book also discusses various dual-driver arrangements for subwoofers, which is another interesting area where you can use twin drivers to eliminate undesirable effects. If you put that together with the D'Appolito design in a single cabinet, you will have something like what Infinity used to sell for $1000 each...but doing it well won't save you much money because good drivers are expensive.
If you're an engineer that's comfortable with differential calculus, I'd recommend the grandaddy of them all: Acoustics by Beranek [2] (I'm actually linking to Acoustics: Sound Fields and Transducers which is an updated version of Beranek's original Acoustics cowritten with Tim Mellow)
[0] http://www.amazon.com/Loudspeaker-Design-Cookbook-Vance-Dick...
[1] http://www.amazon.com/Master-Handbook-Acoustics-Alton-Everes...
[2] http://www.amazon.com/Acoustics-Transducers-Leo-L-Beranek/dp...
Is this as true as it was in the 1990's?
A lot of people have high-end small speaker systems with branding such as 'Bose' on the front. Maybe they have got old and their ears have changed, but even if they once had big-box speakers due to the physics reasons, they have moved to the small speaker 5.1 things and it is now okay to do that. The sub-woofer doesn't have to be stereo either.
I don't know, but has new thinking fundamentally changed how hi-fi is done from the twin speaker stacks of old with lots of tweeters/woofers/crossovers to these new-fangled mini-5.1 things?
A mantra thats repeated continuously, mostly by people selling mono subwoofers or uninterested in audio reproduction, is low freqs are non-directional.
Yet I go outside and listen to thunderstorms and passing unmuffled motorcycles and firearms and much bigger .mil weaponry and it sounds highly directional. I can point at the rumble of distant artillery as well as any other sound.
Surely if you build a directional subwoofer system and can only perceive a field, that'll work OK even if its an economic financial fail. But the other way around is an epic fail. So save money at the risk of sounding bad. Well, if the point is spending money to sound good, having an extra 3 dB of signal is never a loss...
Another puzzle is I know from experience that L-R and L+R sound different from FM subcarrier detection. Supposedly mono subwoofer people claim that magically only applies to higher freqs. I would think a single mono channel subwoofer would sound "better" fed off only L or R channel than some peculiar mix of unknown phase. The problem is which channel to connect to?
Phase relationships are a pretty big puzzle in audio once you get past simple response problems and simple distortion problems.
Of course you won't hear this outdoors because the sounds are not in an enclosed space.
Bose and other manufacturers have done a good job of developing tricks that fool undiscerning listeners into thinking their system matches the performance of a full range system. But there's no tricking out the physics of low frequencies. Baffles and complex tube resonators can have limited positive effects but they often cause problems in other areas.
That said, unless all you care about is sound quality, the aesthetics and performance of a Bose system can in many cases be very attractive.
http://micka.de/org/en/index.php
Hope that someone finds this useful.
He does a great job of designing a crossover and phase-inverting the tweeter to get an excellent step response. beat mentions that this is something most speaker manufacturers fail to account for.
It's about as good as you're going to get for a DIY design and a whole associated design guide.
The hi-fi speaker business is a highly competitive market. The invisible hand already drives prices to about where they can go.
Most woofers are designed so they have to be in an appropriate box in order to be useful. Without access to the process for designing custom woofers, the DIY'er has at our disposal two main variables for tailoring low frequency response: Woofer choice from off the shelf models, and box design. And there are some tradeoffs to manage, such as low frequency response and box size, so that there's no ideal box design for all uses.
Me, too. Just when speaker design seemed to be a solved problem, with a small set of uniform solutions, up springs the concept of co-designing speakers and musical instruments.
What frequency range/instrument are you working with? And what architecture/horns/boxes/tubes/drivers?
Interestingly enough, I think that the electric bass involves a sort of de facto co-design, because the bass acquired its "classic tone" during a time when speakers were relatively low performance, i.e, having -3 dB points far above the lowest fundamental. To this day, speakers with that tone quality tend to be preferred by bassists, which is a fortuitous circumstance because it liberates us from dragging around huge boxes capable of actually reproducing that E fundamental at full strength.