There are a great many instructional articles on loudspeaker design and speaker building from do it yourself manuals for audio hobbyists to the nuts and bolts of dome tweeter and woofer design for professional engineers. Speaker crossovers, capacitors, inductors, wiring and cabinet construction all figure in clean sheet speaker projects but the comments below look beyond the nuts and bolts of speaker building. 

Loudspeaker design strategy has to do with how the speaker is going to be used. What kind of room will they be placed in and where in the room will they be installed? Who and how many people will be listening to them? Where will these people be sitting? 

Different loudspeaker designs have different dispersion patterns and these work for specific rooms and listening positions. A D’Apolito speaker configuration is meant for the seated listener and won’t do for people who like to stand up and move around a great deal. Ditto short Ribbons. 

Dome based systems will bounce a great deal of sound off the ceiling while linesources and panel speakers won’t. Is that an issue in your installation? 

Taking the long view, recognize that room correction and digital crossovers are practical and affordable now.  Some receives and pre-amp processors even have both capabilities built in.   You may want to keep passive crossovers external or allow for their easy removal from the cabinet. 

Digital signal manipulation will not cure everything but it can help immensely in specific areas of phase and amplitude. Design your loudspeakers and the room setup to allow these devices to fully apply their strengths. Keep your designs simple and go with drivers with broad and smooth frequency response. See that the radiation pattern is similar for all drivers. And don’t try to squeeze the last bit of output from the bottom end by porting the speakers. Ported speakers have resonances that will always be heard no matter how skillful their implementation or how sophisticated the digital correction. 

Digital correction cannot compensate for poor drivers, disparate radiation patterns and hopeless placement. 

If your room has a particular acoustic “character”, make sure your chosen design doesn’t exacerbate it. Are you willing to acoustically treat the room for a more even response? 

Lastly, design in lots of dynamic headroom. As higher resolution recordings are now the norm (SACD, DVD audio, True Digital etc. and high res downloads) dynamic range has increased dramatically. Also, good, cost effective amplifiers are becoming more common. Design your speakers and your system to take advantage of these factors. 

Obviously Newform feels that tall, wide dispersion Ribbons and LineSource midbasses offer the best design approach for most home music systems but tall loudspeakers aren't practical or affordable for every application. 

Whatever your audio project, it is best to stand back and apply some strategic planning before buying components and cutting wood. Design your speakers for your room, your audience and to take advantage of cheap and acoustically transparent digital signal processing. 

Good speaker building!

AUDIOPHILE UPGRADES

 Quote “ .. In the 5 years I’ve since I got the 645s and the Behringer/Panasonic package, I’ve spent all of my money on music. I’m off the audiophile treadmill!” 

Get off the audiophile treadmill! 

Experimenting with different components, system configurations and room layouts has always been part of the high fidelity hobby. But the core of the hobby is music and enjoying listening to it regardless of the state of your system. Stereo and home theater sound systems always have a number of weakness in them. These can stem both from the music reproduction chain behind the speaker diaphragms and from the listening environment in front of the diaphragms. 

The electrical chain starts with power cords and ends up with the loudspeaker crossovers. Keeping noise out of the system and avoiding format changes (digital to analog and visa versa) are the main issues here. The old concept of matching components, so important in the days of tubes and analog devices, is something for the scrapheap as finding two digital components with offsetting design problems is highly unlikely. 

Where the electrical response stops, the acoustic response begins and room/speaker interactions are paramount. 

So the first order of business is identifying a problem and the second is reducing it. In the audiophile world, anything less than perceptually perfect can be described as a problem. In reality, it is simply sub-optimal. Improvements are always possible. Perfection is unachievable. 

There are those who will purchase $20,000 cd players to improve their systems. Hopefully they will have dealt with the major listening room/speaker interactions first. Very, very close to the same sound quality can be had from a good $400 DVD player as long as the digital outputs are used. In some cases, the most expensive products lag the cheaper mass produced units as demonstrated in a number of carefully conducted lab tests. The last 2% of high fidelity can be hard to nail down and trying is very expensive. 

But even assuming top $ means top performance, the digital differences are small. Analog outputs are another story - generally the high end gear is way ahead of the cheaper stuff - but do you really need to use analog outputs? The $19,600 difference can be much better spent on room treatments, amplifiers, loudspeakers and digital crossovers - the areas with the largest errors and where the greatest improvements can be made. 

Upgrading your system to accommodate the playback of higher resolution formats will also produce a large improvement although these formats are anything but convenient or consistent at the moment. Improvements are being made however and now upsampling high resolution format music servers are dropping in price. No surprise, they are digital and the price/performance ratio and the friendliness factor are moving in the audiophiles’ favour. 

Read our pages on Acoustic Room Treatments and Audiophile System Strategy. The key is to recognise where the opportunities for the largest improvements in sound can be derived. Understand your room and it’s fundamental acoustic properties and then pick loudspeakers which will best complement those properties. 

At some point in time, all systems will incorporate room correction and digital crossovers. Understand what a digital crossover and room correction can do for you in your room with your loudspeakers. Digital correction of room issues and some loudspeaker flaws can provide very large benefits with few acoustic drawbacks. 

But room correction and digital crossovers will not cure all ills. Dispersion mismatches with the room, dynamic limitations, diffraction and high levels of distortion are fundamental problems which will simply not be cured by digital means. 

However, there is a price to be paid in system complexity when installing a digital crossover. Double the amplifier count is required and, once you get into home theater, the system becomes basically unmanageable for the less than technically devoted. We have great things to say about the Behringer DCX 2496 digital crossover and the DEQX in a stereo system but these are awkward to incorporate into a multi-channel setup. 

The Tact Audio amplifiers with digital crossovers built in are an obvious solution to this complexity (one stereo amplifier per channel would be required) but they cost over $4000 each. However, a Tact amplifier ( or other world class amp with digital crossover) in every corner is something which will do a great deal for any audio system. And it is possible to spend a far more money on far less effective approaches. 

The sonic advantages of digital crossovers are very substantial both in terms of eliminating the inherent shortcomings of the passive crossovers (large inductors, capacitors and resistors) in loudspeakers and in extremely fine tuning speaker response. They offer precision and flexibility that passive crossovers simply can’t approach. 

Digital crossovers also offer some basic room correction with no loss (to our ears) in transparency. Rather than complete spectrum treatment, digital crossovers, by taking out the 2 or 3 worst bumps or dips, can dramatically improve system spectral balance. 

Room correction devices flatten response and possibly also add phase correction at the expense of some transparency. Transparency loss - very noticeable in early systems - has been dramatically reduced in recent years even in the cheaper receivers. However, some still remains. 

Basically, the less processing the better but if you have room problems and lack the means to tackle them directly, room correction can be a very large overall plus. 

On the electrical end, every component performs better when fed clean power and when static is reduced and grounding is properly done. Ground loops result in a loss of dynamics and transparency which can be fairly dramatic but it can sneak up and be hard to identify. 

Special electrical circuits and power conditioners can deal with many problems but listen to the power conditioners in your system before buying as some can actually degrade sound quality. High end power cables actually can offer audible improvements but they are expensive. 

Expensive digital cables are highly questionable as are extremely expensive speaker cables. Some of these can cost more than the digital crossover/extra amplifier route which offer vastly higher fidelity returns. 

Last but not least, many systems built up over the years have become very complex. Try simple. Unplug and disconnect the unnecessary, particularly processing and switching devices, and see what you can hear. 

Most home theater systems with Newform Ribbon systems sound better without centre channels. We also hear this from other high end speaker manufacturers. Try your system without the center channel. 

Less can be more and higher profits for the equipment vendors do not automatically mean higher fidelity for you. You got into this hobby for the music didn’t you? Experiment on faith. Buy on results. Relax. The latest and the best won’t be the latest by the time you walk out the door and it won’t be the best for much longer. 

Don’t allow fascination and then frustration with various components to interfere with your connection to the music. 

Loudspeakers Unsurpassed in Soundstage, Transparency, Detail and Dynamics in High End Stereo and Home Theater Systems 

Knowing what you are aiming for makes it a great deal easier to determine the best way to get there. Despite the blizzard of audio formats, electronic devices and forest of loudspeakers, building a great audio system is easier today than it ever has been. 

Sound sources, amplifiers and loudspeakers are all greatly improved. Like automobiles, you now have to go out of your way to get something truly awful. 

But working your way towards superb high fidelity still takes some thought. 

Pick an acoustically good room and match the loudspeakers to it. Check our Acoustic Room Treatment page. You can’t change your house but you can make the best choice possible of loudspeakers which will work the best in your chosen listening room. 

Make an effort to understand dispersion issues. Which dispersion pattern best suits your room? How many people will be listening at most? Critically or casually? Where will they be sitting? 

A high ceilinged room with log walls and wood floor will be a better environment for a dome tweeter based loudspeaker than a room with a 7' ceiling, tile floors and glass walls. 

Once you have nailed down the room/speaker options you can look at the electronic issues. Chances are your choices will be wide open. If you have chosen very low impedance speakers or low dynamic speakers then your choices are more narrow but still abundant. 

Don’t make the mistake of thinking it is possible to substitute a large budget for good planning. There are many absolutely superb audio systems out there which cost under $10,000 everything in and there are many (but fewer) $100,000 horrors. 

Who will be operating the system? What kind of complexity will the least technical operator tolerate? 

Is the system going to start out as stereo and grow into home theater? What is the final architecture going to look like? What is the ultimate system to which you can realistically aspire? 

Here are the ultimate options. An acoustically ideal room with perfectly matched loudspeakers driven by first rate amplifiers fed by digital crossovers and all controlled by a perfectly transparent room correction preamplifier. The sources will range from vinyl LP to FM radio to an upsampling high resolution music server. 

Where on the complexity and cost curves are you going to get off? 

Now that a large amount of the audio chain is digital, it is safe to say performance will continue to go up while the cost goes down. If you want to have the best hifi system you can afford installed all at once, by all means buy the best possible. If you are intending to build your system over a period of time, relax and take your time because high fidelity value will only get better the longer you wait. 

Don’t buy state of the art electronics and expect it to be relevant 5 years from now. 

Loudspeakers Unsurpassed in Soundstage, Transparency, Detail and Dynamics in High End Stereo and Home Theater Systems

Pono, IPOD, MP3, CD, LP, SACD, Vinyl - WHAT SOUNDS BETTER AND WHY

The evolution of high fidelity has followed a generally upward trend with the occasional sidestep into poorly thought out or poorly supported formats. 

As the means of reproducing music has burgeoned, so too has the variety of formats with the consequences of confusing media incompatibilities and redundant software. 

Amid this blizzard of formats, delivery systems and exploding playback options, the holy grail of the past 80 years of audio enthusiasts of ever higher fidelity has been largely sidelined in the scramble for market dominance and "accessibility". 

But no matter what the format or the listening environment, sound quality will ultimately have a huge impact on the enjoyment the listener will get from the music. So to put the evolution of music into perspective and evaluate the stages, it is important to compare the fidelity potential of the various formats whether iPod, MP3, SACD or DVD Audio.  The latest effort to reverse the mediocrity of MP3 comes from Neil Young's Pono.  As you will see from the chart below, the Fidelity Potential in the formats Pono has adopted (they didn't invent their own) is far higher than the ubiquitous MP3.

Pono Formats:

•    CD lossless quality recordings: 1411 kbps (44.1 kHz/16 bit) FLAC files 
•    High-resolution recordings: 2304 kbps (48 kHz/24 bit) FLAC files
•    Higher-resolution recordings: 4608 kbps (96 kHz/24 bit) FLAC files 
•    Ultra-high resolution recordings: 9216 kbps (192 kHz/24 bit) FLAC files

Comparison between analog and digital is difficult. However, it is possible to establish ranges of equivalence for comparisons among the formats. Below we list different formats and quantify their potential to deliver sound accurately and fully to the listener. 

Expressing digital in terms of mathematical quantity is simple but not so for analog whose limits are possible to ballpark but not to pinpoint. 

Also, the different formats have different weaknesses making exact comparison even less precise. However, in broad strokes, comparison is possible and long overdue. 

The ongoing debate over the past 25 years as to which format - analog or digital - "vinyl or CD" - sounds better has been conducted in the fog of ignorance and marketing hype. The first digital format, the CD, was billed as "Perfect Sound Forever" - fidelity so high no one human could perceive anything better. 

Many people knew at its introduction this was marketing hyperbole and now everyone knows it. Despite the many hoary flaws in analog playback that the public found extremely frustrating, the new CD system clearly had limitations of its own and they weren't all due to poor implementation. 

But the move to digital represented a complete direction shift for playback systems and perhaps we should not have expected the new system to be superior in every respect to the old. 

All things being equal, the more information a format can transmit, the better the sound will be. So here are the formats broken down into their bare bit potential some with high and low ranges. There are a huge number of caveats and remarks about the formats' various weaknesses but the Fidelity Potential Index gives a reasonable approximation of the fidelity a particular format is capable of delivering. 

FIDELITY POTENTIAL INDEX (FPI) TABLE

CALCULATION METHOD

Converting analog performance levels to a digital equivalent involves developing bit rate (sampling frequency) and bit depth (bits per sample) from the analog data. 

Since the sampling frequency for the CD format is 44.1 kHz - roughly double the highest frequency (20kHz) it can reproduce, the analog equivalent sampling frequency is calculated to be double the highest frequency that medium can deliver. 

For the bit size figure, a 6dB difference in dynamic range is taken to be equal to 1 bit so an analog medium with a dynamic range of 60dB has an equivalent bit size to a 10 bit digital signal. 

The bit depth times the sampling rate per second equals the number of bit per second the medium can deliver. This number divided by 100,000 for brevity is its Fidelity Potential Index. 

How fully the fidelity potential of each medium is exploited by the format structure and electronics limitations could be covered only by an extremely drawn out discussion so here, briefly below is a very truncated list of caveats.

FORMATS AND SPECIFICATIONS NOT INCLUDED

Many formats both analog and digital were not included. Digital formats like Dolby ProLogic which are lossy (ie they drop bits and then try to re-construct the signal to make the signal more compact) are not included due to the a huge amount of guess work involved. 

We have not included frequency response and dynamic range figures for the digital formats - only their sampling frequencies and bit rates.

COLUMN DESCRIPTIONS

Bit Depth - a sample of the musical waveform at one point in time can be represented by one single byte of information. The resolution of this byte (the number of bits that it can have) dictates the dynamic range of the signal. The more bits, the greater the number of possible levels which means louder loud passages and quieter silences. The range of the dynamics in the music can be much better represented by a 24 bit system than an 8 bit system. 

Sampling Frequency - how often the bits are represented. The more often they are represented the higher the frequency they can represent. Sampling 2,000 times a second cannot represent a 5,000 Hz signal. A waveform must be represented by at least 2 data points per cycle so the minimum sampling frequency required to cover the highest level of human hearing (20,000Hz) would be 40 kHz.

FORMAT DESCRIPTIONS AND CAVEATS

A sound signal starts out as an analog waveform - the original musical note - and finishes as an analog waveform - the sound that is reproduced for the listeners ears. The fidelity of a recording format is dependent not only on the raw ability of its core engine to capture high dynamic range and broad frequency response but on its ability to handle analog to digital conversions and processing of the recorded signal. 

The potential inherent in one medium does not guarantee sound quality superior over another medium of lower potential capability as music production standards vary immensely as does implementation of high standards of engineering in the recording and reproduction equipment.

Dynamic range is not signal to noise. Digital systems are inherently noise free. Any noise comes from their associated electronics, not their media. Analog systems, with their different types of mechanical noise (tape hiss, record ticks and pops) have a signal to noise level far smaller than their ultimate dynamic range.

Digital systems use various forms of filters in their recording and playback processes. These filters can introduce distortions in the audible frequency range. One of the most famous examples of this is the "brick wall" filters used above 20kHz on CDs. Early implementations of this introduced various phase anamolies down as far as 10kHz or even lower.

COMMENTARY FROM THE WEB

John: 

Of course, there are lots of ways to measure noise -- weighted, unweighted, and on phono recordings, whether you measure the pops of surface noise, or just the average. 

I can give you ballpark estimates of dynamic range based on my experience. High quality vinyl LP: 60-65 db Average vinyl LP: 50-55 db cassette (excluding noise reduction) 45-50 db. Add 8-10 db with properly functioning noise reduction professional reel-to-reel quarter-inch 2-track 15ips: 60-70 db (depending on tape formulation) 78 rpm shellac: 30-40 cylinder (vertical modulation) perhaps 20-30 35 mm optical ("academy" cinema, pre-Dolby) 40-50 db 

I have measured some of these -- reel-to-reel, vinyl test LPs. The others are what I would call educated estimates, based on what it sounds like to me over the years, in comparison to the other media. 

I should give a heads-up for one of your caveats, in case you are not aware, that a numerical S/N figure, or a firm number on distortion, is not really possible on perceptually-based bit-compression schemes, such as mp3, ATRACS, Real Audio, Windows media, etc. These encoding systems will give near-perfect results on steady state tones, normally used to measure analog systems. They end up wrecking the signal depending on the complexity of the waveform. The idea behind these systems is an algorithm based on what in listening tests people could hear, and what would be "masked" by other sounds, based on spectral content from moment to moment. The encoder then throws away the data representing the parts that people supposedly will not miss. E.g. a 96 Khz mp3 file throws away more than 85% of the data of a CD quality 44.1 KHz stereo PCM datastream. 

I am not aware of any reliable quantitative measurements of the quality of bit-compressed systems. They are all based on blind listening tests. 

With strict uncompressed PCM, there is of course a direct mathematical correspondence to S/N radio and dynamic range. 

Hope I have not belabored something you may already well know. 

Best regards, 

George 

FIDELITY POTENTIAL INDEX

Hi John, 

I'm a long time owner of 645's and have used the 45" ribbons in a variety of ways and systems, so I'm on your newsletter list. I posted the link to your "Fidelity Potential" page to a private group of audio guys as we were roughly on the topic of MP3's and how "kids today" don't care about quality etc. I won't name names, but some of these are guys you would have heard of. One of them asked: "Could someone further indulge me on the derivation for this theoretical "Fidelity Potential?" so I thought you may want to have a whack at that. If you do I will forward to the group and keep you posted. 

Personally I found it interesting, but also could not imagine how you might have quantified the "anecdotal", I think you referred to a conversion to sampling rate if I recall (I read the page a few days ago). 

Best regards, 

Dave King

Dave, 

There is a fair amount of discussion of the method on the page and there will be more when some posts are put up. Several posts will elaborate on what I'm saying here and include their own estimates of analog format capability. So ranges are important to include. 

The "anecdotal" are judgments about how far into the noise floor and ceiling one can hear. These extend the range of the hard specs for the format say vinyl. Digital has a hard, brick wall limit on dynamic range. It has no noise of it's own within that dynamic range. Analog formats have lots of noise sources and the signal to noise is less than the dynamic range. However, one can still perceive signal into the range of noise and this is what effectively extends the dynamic range of analog sources. 

Yes, these are personal estimates about what sounds good and why but they affect the range of the rating not the core value. 

If we could arrive at similar ranges for the effects of digital artifacts, I'd establish ranges on that basis as well. Of course, these would reduce the digital format FPIs whereas the analog FPIs are increased by the process. 

Particularly it would be interesting to assign reductions for various compression schemes and for room correction methods which may increase amplitude correctness but reduce "transparency". 

Let me know what you think! 

If you are playing with the R45s, - do a Coaxial Ribbon LineSource - build a system with 4+ good 7" drivers in sealed enclosures and stick the R45s in front of them, hopefully using a digital crossover and re-arrange your audiophile benchmarks. This is a step up from anything you have heard guaranteed. 

Cheers, 

John M. 

Dear John 

Your rating is an interesting project! The ratings look sensible on a wide window. I tend to disagree however on certain basic assumptions. 

Sampling rate is assumed to be identical to high-level bandwidth (eg. 0 dB -10 dB). So you assume vinyl as 50’000 kHz sampling rate. This is maybe true for high level signals (maybe even worse). But, with a good low inductance cartridge and a capable stylus, like hyperelliptical, VdH I&II, Gyger I&II, micro-ridge, Paroc etc. at least 100’000 kHz sampling should be assumed. There is even proof that 75 kHz signals are traced with LPs that were cut with DMM (Direct metal mastering). I think in real life there is a wide variability in the amount of ultrasonic content on LP. But the fact is, there is considerable energy above 20 kHz available in LPs (not always the recorded signal...), And there is traceable energy up to 75 kHz. Then there is the roll-off frequency and order of roll-off in analog systems compared to digital, which makes even a (IMO wrongly) assumed “analogue sampling frequency” of 50’000 Hz audibly different to a digital one, with it’s sudden, high order drop-off vs. the more “natural” analogue roll-off, which behaves more closely related to real-ear experiences with acoustical phenomena. Which most probably is audible in supersonic “inaudible” regions, specially when “linear phase” pre-ringing oversampling filters are involved. The problem in lining up digital and analogue systems sonically is the problem of comparing apples with oranges. High-level linearity (Freq. Resp. And distortion) vs. low-level etc. 

Sonically I would not totally disagree with 320 kBs MP 3 vs. Cassettes being in a similar range of “fidelity”, still it’s my feeling that one can get (considerably) more involved in the music and the sound with a superb (and expensive) cassette tape recorder. To my ears there is a certain aspect in the sound of digital compressed formats reminding of bad main’s, sucking out some of the bounce and communicating warmth and “energy” of the music. You don’t have this with cassette, and, BTW, good cassettes register information above 15 kHz (-20 dB bandwidth), contrary to MP3. This is audible too – eg. in PRAT... 

My ears told me on any DVD vs. SACD comparison I made (Sony SACD player, Audio Synthesis DAX Discrete) that even 24/96 PCM(DAD) sounded potentially more alive and natural than SACD. DVD-Audio sounds good too but I haven’t had the experience of totally locking into the performance with it, as was possible with optimal 24/96 DADs. Less data processing? 192 kHz / 24 Bit is promising, haven’t really heard it yet. And it’s a huge storage consumer. SACD is (for me) a theoretically impressive and brillant format which was promoted with kind of an audiophile-underground marketing hype, but which is, contrary to the hype of being “most analogfue-like”, highly feedback processed (high-order noise-shaping) and somehow in the end sounds kind of like it. It is definitely not on the level of 24/96 kHz for me, and is even a slight sonical trade-off compared to good CD. PRAT is in favour of CD, bass is heavier (hifi impressive) on SACD, and the top octave has considerably more “air” though. A further inherent problem of all these high data rate formats when burned/pressed on optical media is the considerable higher speed and motor forces involved in the process of reading DVD & Blue Ray (I think too) compared to CD. And even in CDs this problem is audible. An interesting observation when playing different data formats on my iBook and MacBooks: When you look at the processor load you see that the non-lossy compressed format ALC needs about 50% more processor work compared to AIFF or WAV. This is to my ears slightly audible on both the computer and an iPod Touch. Processor work is in the end analogue current and shapesdigital power supply noise, which in reality can not completely be blocked out by any measures IME. 

You were looking for “trouble” with that rating, didn’t you ? ;-) 

Best wishes 

Christoph Mijnssen

Arbelos Elektroakustik 

Note by Andrew Marshall on vinyl LP frequency response. - the quadraphonic systems may have included response up to 100kHz to manage the signal steering but they never worked well probable because at 100k, the LP playback system is simply not reliable. Also, cassette tape can go up to 20kHz with the right tape and noise reduction system. 

Hi John, 

"Civil, informed and humourous comments will be given preference". Er, quite. But the idea behind your your is rather simplistic and you have to understand may well provoke an irate response. You end up with 24/194 PCM sounding 7 times better than LP, a result so different from reality it doesn't really bear any further comment. 

However, what George has to say about compression systems being impossible to measure meaningfully with steady tones (multi-tones do yield a result) yet wrecking music is very true. More amazingly they are contrived on listening tests alone - often crude ones - a point few people understand. Read what Karl-Heinz Brandenburg and the Fraunhofer Institute say. So thumbs up to George on this! 

Finally, the idea of a controlled listening area freed, to an extent, from room effects, as provided by a line array is an interesting one little talked about. This also brings in ribbons, which most of us admire. Hope we can cover your products sometime. 

So good luck with your Index. Time to strap on the tin hat methinks. 

regards

Noel Keywood, publisher Hi-Fi World 

Noel, 

I knew there would be criticism but it has not turned out to be as severe as anticipated. I may be a little beaten up but not unhorsed. 

The FPI is not intended to be a linear representation of the fidelity a human is capable of perceiving, but rather simply a method of putting raw capability in a numerical order. So day in, day out, the 192/24 can be counted on to sound better than the LP. But, as you say, not 7 times better or even 2 times better. Just distinctly better on most recordings. 

Compression systems and the judgement of being able to listen into the analog noise floors and ceilings can skew the hard numbers. Also, from my own point of view, room correction systems flatten amplitude but result in some loss of transparency - I'm not sure at all as to how this happens or how to represent it numerically. 

And the flaws in digital which effectively reduce the dynamic range or add noise/distortion are not represented in the FPI either. Maybe all in good time. 

Thanks for your comment! 

Cheers,

John M. 

by Dennis Burton 

CD vs. Vinyl with some other stuff thrown in. 

When I was a child, the hifi debate was over small vs. large speakers (really) and transistors vs. vacuum tubes. These arguments never end because people use music systems for different reasons and also hear different things. For some people dynamics seem to be important beyond all reason, and so that is what they will notice. Others may worship at the shrine of tonal purity and musical pitch. Most of these people are quite in denial, claiming that, in fact they want it all; imaging, dynamics, vast bandwidths, seamless crossover, low coloration etc. etc. But this is not necessarily true. 

Each component in a music system has a “sound”. Now we are getting simple. A reel to reel tape machine will not “sound” like a turntable, or a cassette machine-all technical limitations and considerations aside. Transistors have a sound. In the end you go with what you like best. One man’s fluid sugary midrange from tubes is trumped by another man’s huge expansive bass line from a huge Class A transistor. Or is it the other way around? 

I listen to cassettes, open reel quarter track and half-track tape, MP3’s, CD, vinyl, and FM, and have heard quite acceptable results in all of them. Oh I am a blasphemer aren’t I? The ultimate source? Well it may well be reel to reel. But the machines are hard to use, tapes have to be rewound to prevent print-through, there is hiss, there can be dropouts, transport noise etc. This is not convenient. So then vinyl, only the big catch is that I simply cannot afford the equipment necessary to have that happen. I use a Linn Sondek LP12 and it is a nice turntable, but of course I would need between $30 and $60 thousand for a really good turntable and would then be off hunting for a suitable and expensive cartridge for it. Let us not discuss the drub pressings foisted on us over the years, which simply cannot be rescued by any known technology, or the fact that a cutting lathe has rumble figures you simply would not accept in a turntable of any price. 

My very first CD player cost less than the phono cartridge in my turntable. I knew then that the CD was inevitable. But more than that, at $175 that Philips CD player smoked anything and everything in a turntable at that price range. It’s not about money you say; it is art, purity, subtlety, and poise. Oh, I suppose so, but personally I enjoy music. I have heard an MP3 mastering from a 45RPM 7” single of the Chiffons singing He’s So Fine coming out of the hideous 3” speaker of my kitchen clock radio and been thoroughly delighted. The audio mangling was scarcely describable, with multipath from the FM into the bargain and the fridge itself adding noise to the background. Music, as with Art, demands that we bring something to the table or we are simply being entertained for good or bad. The whole idea of HIFI is to enhance the “transportation” of that music. With listening, I am not talking surrender, I am talking engagement. That is not the HIFI bit, that is your bit. 

Does a CD sound better than a vinyl record? Well that depends. If it is Classical Orchestral music recorded in a great hall, then I suspect that Half Track Reel To Reel or vinyl is the winner. If the music is Ultra Chilled Down Tempo Electronica then why use vinyl when all the samples are 16 bit anyways-CD wins there. And that is just two examples folks. Some people love songs and listen to the words, others adore virtuosity and listen obsessively to the playing, ignoring whatever words there might be. Still others surrender to melody and yet others to rhythmic structure. Still others love tempo and arrangement. Some thrill at production values and sonic volcanoes. And there are still others and others and others, but they will all tell you they want it all, and none of them (or you!) can pretend to be a final judge of anything we all might or might not care to actually hear. 

Years ago at the dawn of digital, I listened to a test involving a Classical vocal quintet in a very large room singing. It was a recording studio and we had two Neumann microphones optimally placed and we used a pair of very expensive microphone preamps and simultaneously routed the signal to a Dat recorder, which was then a new recordable digital medium, also the signal went to a very large Sony PCM reel to reel, and 

finally also to an Otari Studio Half Track analogue machine. All three recordings “sounded” different. Most noticeable however was that the analogue half track recording, although having a bit of noticeable hiss provided a very interestingly pleasant insight into the harmonic structure of the blending of the vocals as presented by the room itself. This was absent from both digital recordings, which presented as comparatively “dry”, although I hasten to add, both of them sounded wonderful with the Sony sounding superior to the Dat. We could not actually speculate as to what might be happening except to surmise that perhaps 16 bit 44.1 KHz was insufficient to render the complexities of the harmonic structure. Does this mean that the analogue was better? 

Well, to whom? I mean, the choir members were listening to their performance and couldn’t understand our obsession with some tiny detail of the sound. They thought it all sounded great. The choirmaster loved the room information and chose the analogue. The Sony reel-to-reel mastering machine was worth about $110k at the time and the Dat was $799 and the analogue half-track was about $34K. For the studio, the money certainly mattered, but quality is number one. But was the absolute silence of the digital to be opted for over resolution of harmonic structure? Which best was best? They ended up buying the Sony, but they never sold the Otari. They figured that for multi track recordings where lots of blending and stacking of tracks happens, then the edge goes to the digital because it does this very well. Co-incident pair purist Classical recordings were offered the choice between analogue and digital and for most of them, they either ignored the faint hiss or never heard it and went analogue. Which is best? 

OK, so you’re saying that someone who loves all this stuff has to be the judge, the final arbitrator so that for everyone else there is a reliable standard to follow, and if we can’t agree on or figure out what is best, then we aren’t worth our salt as experts. If asked any expert on anything is sure of herself or himself. So if I do not know wines, I can ask an expert and know that the given advice is based on vast experience and therefore worthy. But hey! It is the same for them-there is no best. It just depends on exactly what you want, who you are, where you are and maybe a few hundred other things. 

There is the story of the Buddha walking through a small village, and as he was passing the Butcher he happened to overhear a customer ask the butcher which of his cuts of meat was the best. The butcher replied that they were all the best. At these words the Buddha became enlightened. 

Common wisdom in the world of high end audio has long held that the two most important components in a high fidelity audio system are the loudspeakers and the room. In an era of very high quality digital sound sources and mid-priced receivers which can blow the doors off many mega $ audiophile amplifiers of just 10 years ago, this is more true than ever.

Loudspeakers are dealt with extensively on our site. Obviously we feel wide dispersion Ribbons, whether in classic two way or the ground breaking Coaxial Ribbon LineSource configurations, are the best answer for most listeners in most rooms. The LineSources in particular can work well in some exceptionally large or irregular spaces. 

But great loudspeakers can’t overpower terrible room acoustics and it is here that a little thought can yield great sonic payback. Room acoustics is really the management of reflected sound and the minimization of room dimension dictated “modes”. Some reflected sound is good but too much and from the wrong direction can be degrade soundstage coherence and create listening fatigue. 

Limited reflected sound from the side walls can enhance the high fidelity experience while sound bouncing off the ceiling, floor and back wall almost always degrades the soundstage and results in listening fatigue.

One of the reasons large Ribbon and electrostatic loudspeakers have gained their vaunted reputation in the audiophile world is they have very limited vertical dispersion so they automatically minimize floor and ceiling reflections. This is contrasted to dome tweeter based systems which radiate hemispherically and therefore push sound in all directions almost equally. 

There are a number of ways to tame room reflections. By tame, we mean arrive at a satisfying ratio of direct to reflected sound, not create a totally dead anechoic chamber. Managing reflected sound actively is done by damping the surfaces which reflect the sound. Passive management is achieved by moving the listening position and loudspeakers closer together so the path length from speaker to reflective surface to listener becomes much longer than the speaker to listener path distance. In acoustics, the further a wave has to travel, the weaker it gets. 

There is a listening distance which is best for each room and each listener so experimentation is necessary. Too much reflected sound is unlistenable and too little is unnatural. 

Reflected sound should be equally balanced from each side. This is often the most difficult thing to do since many rooms have an open side or the sound system has to be installed off centre. If reflected sound can’t be equalized, then the overall side to side balance will have to be tailored using speaker toe-in and possibly the balance control. Acoustic symmetry is the key. 

Sound reflections occur at all frequencies. Treatment for low frequencies (long waves) is different from that for midbass and higher frequencies. Very often effective room treatment can take the form of putting furniture, plants, bookcases and tapestries in the right position. Or by installing heavier drapes or blinds. 

Stuffed furniture and bookcases (filled with books) absorb sound, hard furniture breaks up reflections. Plants do a little of both whereas drapes form variable dampers as do doors which can be opened and closed depending on the sonic effects they create. 

Beyond these common in-room devices, home made damping can be provided in the form of blankets propped up by 2x4s or hockey sticks and small mattresses and pillows placed for maximum effect. (Usually behind the listening position if a rear wall is close). 

Of course, there are professional products readily available to do a first class job once you have determined the final approach. These products will also come with expert opinions if you purchase at the right place and hence will be vastly more useful. 

There are many room treatment companies. Here are three. You can discuss bass traps and slap echo damping. 

A company with a lot of experience and a great product selection. 

www.auralex.com 

If you are looking to design your room from scratch, ASC will have some material and advice useful for soundproofing. 

www.asc-home-theater.com 

And the old audiophile favourite which has helped many an audio company over the years (including Newform) wrestle audio show room acoustics to the ground. 

www.echobusters.com 

A prime consideration in room acoustics is keeping your walls quiet. If the drywall or the floor moves or rattles, they will add resonances which are very hard to deal with. Keep your walls, floors and ceiling solid! 

Of course, there are nearly ideal rooms which require an absolute minimum of work. If you are lucky enough to have a rectangular room maybe 14' to 18' wide x 24' or longer, consider yourself acoustically blessed. Concrete floors, walls and ceiling? Even better. 

A nice long room guarantees that rear reflections will be weak in which case the soundstage will really have a chance to become well defined. Damping on the rear wall can go a long way to sonically replicating a long room. 

Room modes are dependent on the placement of the speakers and the location of the listening position. Varying these can either minimize the mode or shift it out of the listening area. In some rooms though there can be wicked peaks and suckouts which will not be dealt with by mere speaker placement techniques. 

We find that only about 25% of our audiophile customers have great rooms. The rest of us have to work a little harder. But treat your listening room appropriately and it will treat you to many years of musical bliss. 

Loudspeakers Unsurpassed in Soundstage, Transparency, Detail and Dynamics in High End Stereo and Home Theater Systems

DIGITAL AMPLIFIER SYSTEMS

The Behringer is an extremely complex piece of gear when viewed for the first time. Learning how to move around in the menu system makes things much easier. Call us for a quick runthrough. Once you have gotten used to the Behringers controls, you will be able to dial in your system with a degree of flexibility and precision you will find amazing. You won't be going back to bass and treble controls. 

The Behringer DCX2496 has one Input which can either be digital into Input A or analog which is input into A and B (left and right). To change the type of input press the Input A button and then the setup button (right of knob) and then parameter down to the bottom where you have a choice of either analog or AES/EBU (digital). Reports are that digital sounds better - distinctly so in some cases. 

The Outputs are analog and we use 2 and 5 as the midbass outputs for left and right and 3 and 6 as the Ribbon outputs for left and right. 

There are 5 pages of menus for the inputs and 8 pages of menus for the outputs. You go to different pages by using the page arrow keys to the left of the round control knob. You go to different fields on the page by pushing the parameter buttons under the page buttons. You change the values of those fields by turning the round knob. 

We recommend you leave the dynamic EQ alone - we have no recommendations on it and dynamics are not in short supply with this system. You have to free the outputs so, for example, changing the high rolloff point of output 2 (left midbass) does not automatically change the low rolloff of output 3 (left Ribbon). The channels will still be linked however so pushing channel 2 (left midbass) will generate a solid green light on that channel and a flashing green light on channel 5 (right midbass). Changing a value on one channel will automatically change it on its linked channel.

DIGITAL AMPLIFIER SYSTEMS

You have to make sure the mutes are off. Red lights mean mutes are on so push the mute button and press the cancel button for the appropriate fields input and output. When the red lights are off, you are ready to roll. 

Input A (and B if you are running analog inputs) is reduced in gain by 10dB to keep the output levels compatible with a consumer product input, in this case the Panasonic XR45. You will have to experiment with this to get it right for your system. 

The recommended midbass crossover is a 12dB Butterworth (But 12) at 957Hz and the Ribbon is rolled in with a 6 dB Butterworth (but 6) at 2.11kHz. The rising output of the Ribbon at lower frequencies is the reason for this higher electrical crossover. The acoustic crossover is effectively around 1kHz. 

The midbasses have their gain reduced 3.5dB to match the sensitivity level of the Ribbon. 

Also, there are three filters used. The first boosts the bass output for a total of 5 dB starting at 53Hz. The second takes a bump out at 581Hz and the third takes a bump out at 1.07kHz. 

The phase for the Ribbon has been shifted 50 degrees. 

These filters and settings were determined by testing in our big 23x22 foot room, (35x22 with 10' ceilings when you add in our large openings) and your room will almost certainly differ greatly. You can experiment by turning these filters on and of one at a time or all at the same time. You can also dial in different frequencies and boost and cut levels and hear the results in real time. You can add filters to the point where you run out of processing power. You currently have 26% left which will allow quite a few extra filters to take care of most nasty nodes in your listening room.

DIGITAL AMPLIFIER PACKAGES

OVERVIEW

The EQ filters are parametric which in practical terms means you can chose the exact frequency at which you want to centre the filter, the exact level you want to boost or cut and pretty much exactly control the bandwidth of the filter with the "Q" control. The other types of filters are simple high and low pass (like bass and treble controls). We'll talk to you about it. 

Experiment but always make good notes on the settings and call us for a live runthrough on the Behringer before you use the system!!!

BEHRINGER

Inputs 

- minus 10 dB, no EQ or delay on the inputs. 

Outputs 

Low pass -midbasses - (2 and 5)

-3.5 dB page one, input source A 

Page 2 

957Hz But 12 - Filters on right side - high end rolloff. 

Page 3 

- EQ 

Filter 1 53Hz, 5.5 dB, LP, 12dB

Filter 2 581Hz, - 5.1dB BP, Q 2.5

Filter 3 1.07kHz -2.3dB, BP, Q 6.3 

- Dynamic EQ etc off 

High Pass -Ribbons- (Outputs 3 and 6) 

Page 2 

But 6 at 2.11kHz - Filters on left side - low end rolloff. 

No EQ 

Page 7 

Phase normal

Phase 50 degrees. 

DIGITAL AMPLIFIER PACKAGES

OVERVIEW

PANASONIC XR45

DVD 6 Channel 

Front speakers large No subwoofer No other enhancement filters or modes 

The trick with the Panasonic is to make the mains large and turn the subwoofer off. Only then will the XR45 feed full frequency into the main speakers. Otherwise there is a 100Hz rolloff and a beautiful bottom end is lost. 

HOW DO YOU FIX YOUR AUDIO SYSTEM?

By eliminating problems:

Symmetry - keep the reflections from each side of the room even. One hard wall and one soft wall will make it very difficult to achieving great soundstaging. 

Rear reflections - usually it is best to minimize them so either keep the seated listening head away from the rear wall or apply considerable acoustic damping to the wall. 

Space - get the speakers out into the room for depth of soundstage. 

Sub Placement - this is critical to the proper integration of the system. As with all of the above considerations, EXPERIMENT AGGRESSIVELY. 

Having problems? - Call us  or email  a sketch of your room see the Room Planner. Look over our Room Set-ups page to get an idea of the issues involved.

The break-in experience varies widely. Some owners report the speakers sound great out of the box and they did not hear significant differences over time. Others found harshness and restricted bottom end to be very distinct. 

Most owners find that the speakers become noticibly smoother after 3 or 4 days of moderate volume playing time. Most of the benefits of break-in are to be had in the first 3 weeks but reports vary. 

There are however, some clearly defined effects on break-in time. The longer the speakers are played and the louder they are played will determine their break-in status. Eventually, the changes slow down and cease to be audible. Loudspeakers are mechanical devices so not only will time and energy be factors but so will heat and humidity. If you are using a new amplifier, CD player or cables their performance will be changing as well. 

The fastest way to break-in the speakers is to leave them on at moderately high levels when the house is empty. This might not be recommended for owners with tube amps but for conventional solid state gear, there are not likely to be heat or instability issues which will harm the amps or the speakers 

You hear differently from day to day depending on atmospheric changes and the condition of your sinuses. As you become accustomed to the speakers and the system, you stop listening to them and listen through to the music. 

When the time comes that you only hear music when you turn the system on, the speakers are broken in, your electronics are broken in and your ears have determined that they really do like what they are hearing. 

Our new Coaxial Ribbon LineSource designs come in at higher price levels than we have occupied before but they offer significant improvements in both fidelity and practicality over most loudspeakers, regardless of price- conventional or planar - in most listening rooms. They are just as electronics friendly as our other speakers and thus, for $15,000 total system cost, it is possible to attain ultra system performance. Breakin will be the same for the LSRs as any of our loudspeakers but the midbass will be smoother from the start due to reduced room modes and therefore, breaking of the midbasses will be more difficult to detect. 

Open up the top and take out the top packaging. Needle nose pliers to pull the staples and a knife for the tape will make this easy.

Take out the top packaging. Place the carton gently on its side and open the bottom. Make sure flaps are spread out and raise the carton to an upright position. 

 Lift off the main carton and pull out the corner posts and take the cabinet out of the inner carton. 

Lay the cabinet on its back on the Ethafoam pad and line up the predrilled holes with those on the bases. Use a Phillips bit to  tightly fasten the bases to the cabinet bottoms. 

Note that the base must be attached securely to the bottom of the cabinet before the tall, heavy Ribbon is mounted, otherwise the system will be unstable.

Screw in spikes (if required) after the final location in the room is determined. Note that the inserts on the bottom of the bases are not flush so they will scratch a hard floor surface. Keep the speakers on pieces of carpet when moving them on hard surfaces. 

Need to spike on a hard surface? Look at http://www.superspikes.com

 Cut the tape around the seams of the carton. 

Open the hinged (R30) lid. In the case of R45, the top comes off completely.

 The Ribbon carton holds 2 ribbons in the case of the R30 and one in the case of the R45. The ribbons are heavy and slippery. Hold them firmly from the back and the bottom and avoid putting pressure on the front screens.

The hardware is all in one Ribbon carton and consists of: 8 spikes (¼" - 20 thread), 8 base mounting screws (1 ½" #10 wood screws), 4 small Ribbon bolts (10-24) and 2 large Ribbon bolts (5/16" - 18) plus 2 Nordost Flatline Gold Ribbon interconnects. 

 Insert the two smaller bolts (10-24 machine screws) in the lower holes and line up with the keyhole slots in the bracket on the mid-bass.

Insert the heads through the slots and then insert the larger bolt through the open slot in the top of the bracket and screw into the Ribbon back. Snug the larger bolt with your fingers but use only light pressure. The lower bolts do not need to be tightened as they are there for alignment. Also, they are tricky to get at due to the binding posts behind them.

Fold the flat conductors of the Nordost cables over each other and slide them into the holes in the 5 way binding posts on the top of the enclosure and tighten the plastic hex nuts with your fingers. Run the cables up to the binding posts on the Ribbon and repeat the procedure. 

Attach your amplifier cables to the binding posts on the rear of the midbass enclosure and you are ready to play music!

Low Waste Packaging!  Please Reuse and Recycle!

The cardboard boxes and the corner posts as well as the bracket tube are recyclable. The Ethafoam packing can be extremely useful for other uses. For instance, the bottom pads given that they are waterproof and insulating make excellent seats at outdoor events. 

The Break-in Procedure 

Breaking the speakers in will result in a smoother sound and greater bass extension and openness. Typically there is a noticeable improvement after 3 or 4 days will full breakin occurring in 3 or 4 weeks. Breakin time is a function of volume and time played.

 If you are 5' 2" tall, this is how you look against the R645.

 If you are biwiring, the top set of binding posts is for the Ribbon and the bottom for the midbass drivers. 

Take the gold grounding straps off if you are biwiring or biamping. Note how they are aligned, as the straps are tricky to get on again.

One of the most popular tweaks is upgrading the capacitor to Hovlands or Thetas type. The new capacitors will be attached directly to the positive Ribbon binding post and the positive lead from the amp. This bypasses the standard Ribbon crossover inside the enclosure. This is an easy tweak to do if you are biwiring. 

PANASONIC-BEHRINGER - SYSTEM OVERVIEW

This system is now 6 or 7 years old but for those interested, it can still provide great sound for very few dollars.  If you can't find a Panasonic receiver, a good Onkyo will do nicely.   BTW, virtually all the receivers we've tested from major manufacturers over the $1500 price point have good sounding amplifier sections.   With Onkyo, this seems to start at $600.

Note that some digital amps have crossovers built in like Hypex and Tact.  Several Onkyo receivers have digital crossovers built in like the Onkyo 818 and the new Emotiva pre-processor is supposed to feature crossover capability.  When combined with room correction as in the Onkyo and Emotiva, the results can be superb.  Note however, that crossover settings flexibility varies widely in these components and may not be suitable for your particular speaker system.

The digital output from the CD or DVD player is connected to input A of the Behringer. 

The Behringer crossover then separates the signal into high and low frequencies for both right and left channels. 

Outputs 1, 2 and 3 are for the left channel. Output channel two feeds the midbass (up to 1kHz) and channel 3 feeds the Ribbon (1kHz and above). 

Outputs 4, 5 and 6 are the right channel outputs. Output 4 feeds the midbass and 6 the Ribbon. 

Outputs 1 and 4 are not used in our system. They could be used in the future if we wanted to incorporate a subwoofer into the design making it a 3 way. 

The Behringer crossover feeds the Panasonic receiver which is used strictly for its amplifiers. The input is the 6 channel DVD analog inputs which allows us to use 4 of the 6 inputs. The front channel inputs (left and right) are used for the midbasses and the surround inputs are used for the Ribbons. The Panasonic must be set to DVD 6 channel mode via the button beside the power button on the remote. 

The Volume control on this system is the remote control for the Panasonic. 

You could use the Behringer input and output level controls as well but these should be set and left once the proper level for your room has been arrived at. 

If you are using a preamplifier, the analog outs (left and right) would be plugged into the Behringer inputs A and B. The setup of the Behringer will have to be changed so the input A is analog rather than our setting of digital (AES/EBU). 

Note that the Panasonic receiver is being used only for its amplifiers. It becomes a "dumb" power amp. Once set up in this configuration, you can’t plug other inputs into it or run surround speakers from it. The Panasonic must be driven by the Behringer only.