Audiophile System Strategy

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

Fidelity Potential Index


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. 



The FormatsAnalog or DigitalDynamic RangeFrequency ResponseEqivalent Sampling Rate (Hz)Equivalent BitsBits per SecondFidelity Potential Index
Wax Cylinders analog 20dB
160 - 3kHz
160 - 3kHz
AM Radio analog 48dB 50 - 6kHz 12,000 8 96,000 1
Shellac 78 analog 30dB
60 - 7kHz
60 - 7kHz
78 rpm Record analog 40dB 40 - 11kHz 22,000 6.7 147,400 1.5
FM Radio analog 70dB 40 - 15kHz 30,000 11.7 351,000 3.5
45 rpm Record analog 45dB 40 - 11kHz 22,000 7.5 165,000 1.7  
The Vinyl LP 33rpm analog 50dB
30 - 25kHz
30 - 25kHz
30 - 25kHz
Reel to Reel Tape Recorder analog 60dB
20 - 18kHz
20 - 50kHz
Cassette Tape Recorder analog 45dB
40 - 15kHz
40 - 15kHz
8 Track Tape analog 45dB 40 - 8kHz 16,000 7.5 120,000 1.2
The CD Compact Disc digital     44,100 16 705,600 7.1
DTS digital     96,000 24 2,304,000 23
Dolby Digital digital            
SACD digital         3,500,000 35
DVD Audio digital     88,000
Dolby True HD digital     96,000 24 2,304,000 23
Satellite Radio (mp3) digital            
iPod (mp3) 16 kbs 320 kbs digital         16,000
wave files
16bit, 32k
23, 44.1k
24, 96k
digital     32,000



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.


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.


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.


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.



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, 




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



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. 


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. 



Noel Keywood, publisher Hi-Fi World 



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! 


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. 

Acoustic Room Treatment

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. 

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

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. 

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

Settings Overview


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.


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.



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!!!



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


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. 




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. 

Quick Fix Guide


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.

Break-In Process

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. 

Unpacking and Setting Up





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



 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. 


Digital Amp Packages


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. 












Audio System Setup


The proper scale of the music soundstage is an important part of creating a realistic sense of "being there". Looking up at a large beautiful high def image on the video screen begins to seem unnatural after a while if you are listening down to a knee high musical soundstage. 

Match the scale of the soundstage to the visual image!  Tall LineSource speakers will produce the full height perspective of the original event.


Their time has come. Front projectors are making huge strides in their image quality and along with LED, LCD and plasma TVs, have removed the big box (rear projection or CRT) sticking out from between the speakers.

This development is important to music lovers because a ceiling mounted projector and front screen get rid of that huge mass of vibrating panels and reflective surfaces between the speakers which is a rear projector or CRT TV. In other words, getting rid of the big box in favour of a wall mounted screen will distinctly improve soundstage depth and focus to the benefit of both music and home theater sound.   This also means the big cabinet housing a TV should go as well.


If you are tuning your system for music, get the main speakers as far out ahead of the plane of the scrren as possible to reduce reflections. If you are listening to music with no video, you can experiment with ways to try and neutralize the degrading effects of the screen. Moving the speakers out, or draping a comforter over the TV should improve things.  

Remember, you are not trying to reproduce the sound of a movie theatre in your home because the sound in a movie theatre is vastly inferior to most good high end home systems. Going to a movie may be more fun as an event but technically, the home systems are pulling away from the large theaters.


R630v3, R645v3 (Full Range) Designed to be placed some distance out from the back wall (3’ to 6’) and the side walls (2’ to 4’). All of these distances are relative to the hardness or softness of the room. More air for hard rooms, less for soft. The ample bass output of the R645s allows them to be located far from the walls for maximum depth of stage. The listening position should be between 1 and 1.5 times from the speakers to the distance between the speakers. 

LineSource Monitor 

Exactly the same placement as for the R630s except these will be setup for soundstage only. The bass will come from sub-woofers which will be set up separately. This is the ultimate in flexibility. In small rooms, the LSM may well have enough bottom end for most music applications especially when placed near a wall in corners. 

LineSource Reference 

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. These absolutely must be 2+ feet out from the front wall to avoid a cavity effect - trapping side radiation against the front wall. 

Being modular and totally scalable, ceiling height is the only limitation. If you have a big room with a tall ceiling, these will likely light it up better for you than any loudspeaker system on the planet. 


The subs should provide solid bass to less than 20Hz in room. The key elements in integrating a sub seamlessly are picking the location with the fewest room modes (i.e. smoothest and deepest response at the listening seat) and crossing over low enough (40-60Hz) to avoid muddy mid-bass and localization of the sub. A sophisticated electronic/digital crossover is very valuable in this area either in your processor or in the subwoofer itself. Or, externally in the case of the Behringer DXC2496 or DEQX systems. Experiment with reversing the phase in every different location. This will help eliminate ‘fat mid-bass’ and maintain the speed of the system.   Another very easy and first class solution?  ROOM CORRECTION!  Multiple subs spread around the room will produce smaller room modes, be naturally smoother and make far less work for the room correction system and the amps.


Avoid placing the listener’s head close to a hard rear wall. Reflection from the wall will play havoc with both imaging and bass response. If the listening position must have a wall right behind it, cover the wall with Sonex, a heavy curtain or a tapestry laid over fiberglass or foam acoustic insulation etc. Whatever the method, the listeners ears must not be subject to a strong, direct rear reflection. Don’t place one speaker beside a reflective wall and one with open space to the side. Try to make the acoustic floor plan acoustically symmetrical for both right and left sides. If there is open space on one side try to simulate space on the other with absorbent material on the wall, plants etc. Avoid putting large objects in between the speakers. If a large TV or equipment rack (especially with glass doors is placed between the speakers, try to have the object recessed as far as possible. If the object is close to the same depth plane as the speakers, both horizontal and depth elements of the soundstage will fall short of the speakers potential. 

Don’t make assumptions or expect the speakers to work well just because they are setup roughly the same way they were in a friend’s sound room or the dealers demo room.  Your room is unique. It has different dimensions, furnishings and its boundary walls are made of different materials. 

Once you understand the trade-offs, the key to success is experimentation. Try various positions and listen for the differences on different pieces of music. Even a change of two inches one way or the other can result in dramatic improvements at the listening position. But be careful when moving the speakers, they are tall and heavy! Don’t install the spikes until the best position is found. If you run into problems, call Newform for ideas. With the vast majority of rooms, a 90% setup can be achieved very quickly with the final 10% coming with small tweaks over the break-in period. 

Once you have optimized placement, it will be possible to forget about the loudspeakers and enjoy the music to its fullest. 

Happy Listening! 

See also: The Right Loudspeaker.

Moore’s Law


Motorola announced in January 2001 they have developed a digital audio amplification process, dubbed Symphony, which B&K Components is using in their upcoming DA-2100 amplifier. The DA-2100 will do the Symphony processing in a Digital Signal Processor (DSP) program using Motorola's 56300 DSP family. Motorola plans to integrate Symphony processing into future DSPs, much like Dolby Digital, HDCD, AC3, and other algorithms are integrated now. A vendor need only add output power transistors and an output filter to get a high power and high efficiency digital amplifier.


The first trend to note is the proliferation of high fidelity digital audio amplification techniques and hardware fast enough to implement them. Until recently, digital amplifiers did not have enough distortion free bandwidth for use over the entire 20Hz to 20KHz spectrum and were relegated to subwoofer use only. 

The second trend is the integration of these amplifiers with DSP chips to create the first end-to-end digital audio solution for consumer audio. The audio signal stays entirely in the digital domain up to the speaker inputs. These inexpensive integrated modules and chipsets will allow companies to easily implement designs that have better sound and are cheaper to produce than traditional mixed analog/digital approaches. Let's refer to this type of solution as Advanced Digital Audio (ADA). 

In the next few years ADA is going to be a disruptive force in all segments of the audio industry, from portable MP3 players to high-end home stereos. Moore's law, the doubling of the number of transistors on a given chip every 18 months, will quickly push down prices and improve the sound quality available from consumer audio gear.


Digital amplifiers are a type of switching amplifier. Switching amplifiers rapidly switch the output devices on and off at 100KHz or higher, and then usually low-pass filter to recover the audio portion. Older Pulse Width Modulation (PWM) switching amplifiers, called Class-D, controlled their switching with analog circuits. These designs suffered from poor fidelity and high Radio Frequency Interference (RFI). 

A digital amplifier generates its switching signal using digital logic. Purists would say that digital amplifiers must accept a digital input signal and do all processing in the digital domain. By controlling the switching signal with digital logic, advanced signal processing can be employed to compensate for the switching distortion. Most digital amplifiers avoid the Class-D moniker to distance themselves from analog switching approaches. 

Switching amplifiers have been pursued with interest since they offer higher power amplifiers at a lower cost than traditional class A or A/B amplifiers. Switching amplifiers' output devices are switched entirely on or off. This means that the output transistors do not have to dissipate power that is unused at low volume levels as they do in Class A and AB amplifiers. A Class AB amplifier may be 50% efficient at maximum output power while a switching amplifier can achieve 90% efficiency. The story is better than the specs indicate since at low power levels a digital amplifier could be as much as six times more efficient than Class AB. The increased efficiency allows for amplifiers with smaller power supplies and smaller heatsinks with equivalent output power to non-switched designs. Both of these components are costly and bulky, so shrinking them reduces the size and cost of the whole amplifier. 

Until recently though, they were not suitable for high fidelity applications. In the past few years companies like TacT, Spectron, Sharp, and Bel Canto have released digital amplifiers with sound quality on par with traditional analog amplifiers. In fact, some feel that their fidelity surpasses traditional audio amplifiers. While these digital amplifiers cost upwards of $2,000, they are only the vanguard of the coming revolution. 



The second trend highlighted by Motorola's Symphony announcement is the upcoming ADA phenomenon. In 2001 major semiconductor companies will start to release digital chipsets that can handle every audio processing task including amplification. Currently, only Texas Instruments has their solution available in volume. These chipsets will be inexpensive, especially when compared to the many analog and digital parts they replace. For instance, the Pulsus chips go for $7 to $10. 

This phenomenon is being driven by the rapid convergence of cheap powerful DSPs, advanced digital PWM control theory, accurate psychoacoustic models, pervasive digital audio, MP3, and home theater. Market forces and recent technical advancements are making ADA audio technically possible, cost effective, and in demand. 

Not convinced it's happening? Lets see what some of Motorola's competitors have been shopping for lately: 

JUL 1999, Tripath licenses DPP to STMicroelectronics 

Tripath nonexclusively licensed their digital amplification process for use in commodity markets. In return, the company obtained favorable wafer prices and wafer supply availability. 

27 JUL 1999: Cirrus Logic Acquires AudioLogic; Gains Revolutionary PWM and Low-power Audio Technologies 

AudioLogic has several patents on low power DSP and feedback techniques for digital amplifiers. It appears that AudioLogic's feedback scheme is being incorporated into Cirrus's amplifier. AudioLogic's low power DSP could be useful for portable applications but may not have much bearing on amplifier performance. 

16 MAR 2000: Texas Instruments Acquires Danish Toccata Technology 

Toccata developed the EquiBit PWM ampifier process as used by TacT in their $10,000 Millenium amplifier. TacT has a layman's description of the process and its benefits here. 

02 OCT 2000: Cirrus Logic Expands PWM Technology Portfolio Through Purchase of Patents From B&W Loudspeakers 

Cirrus will introduce 5 amplifier chips using sigma-delta techniques. Four will produce a 110-dB dynamic range, and the highest-power product will have a 120-dB range. 

02 DEC 2000: TI releases 4-chip solution for digital audio. 

TI is the only vendor currently making volume shipments of an ADA solution.  

07 FEB 2001: STMicroelectronics gets exclusive license for Apogee's All-Digital DDX Amplifier Technology 

As you can see, digital amplification techniques have become very popular purchases. Philips, MicroSemi, Linear Technologies, and National Semiconductor are not included here because they seem to be using older analog PWM control techniques and focusing on lower cost car and portable applications. STMico has Apogee's technology but has not announced any high power products yet. 

It is odd that Analog Devices, who has a huge presence in audio codecs and DSP, has not made any announcements about a digital amplification strategy. Their SHARC DSPs are popular in audio products like the Sony TA-E9000 ES, Bose Lifestyle, and Denon AVR3300 home theater boxes. Their digital to analog converters and asynchronous sample rate converts are well respected. Unless they've got something in the labs, perhaps they should snap up Tripath or Korea's Pulsus. Tripath, being a mixed signal design, might prove too hard to put onto a single chip with a SHARC DSP. 



Why are large DSP and digital audio houses scrambling for digital amplification intellectual property? Maybe it's because they need it to compete for a piece of the projected $3 billion dollar market in two years time. 

"PC audio applications represented the largest opportunity in 1998 with nearly half of the $1.55 billion market. According to market research firm Forward Concepts, consumer applications will represent the largest segment by 2003, with a compound annual growth rate of 24 percent and a total available market (TAM) of nearly $1.9 billion. The firm predicts that the overall market, including PC, consumer and professional applications, will represent an opportunity of more than a $3 billion by 2003." 

If PC, consumer, professional, and high fidelity products can all use similar chips then the ADA vendors can target most of the audio market with a only a few chipsets. Can you say "economies of scale?" TI and Crystal already have fixed function cores specifically specifically for digital audio. 

It's heartening to see that the large semiconductor companies feel that high fidelity reproduction is worth pursuing. The variety of digital amplifier techniques is also good news since it will give vendors a choice and allow market forces to weed out low fidelity approaches.


Some audio hobbyists will argue that ADA robs them of the chance to mix and match their favorite speakers and amplifiers. Also, some will undoubtedly prefer to use tube electronics. Perhaps the industry can accommodate these segments while still bringing the unprecedented benefits of ADA to High Fidelity's mainstream. The benefits of ADA in high fidelity include cheaper front-end electronics, higher fidelity input to speakers, and much more speaker design flexibility. 

1. A penny saved.... 

No more sinking money into huge transformers, massive heat sinks, expensive crossover components, multiple chassis & power supplies, or exotic cables. In many high-end loudspeakers the inductors in the bass crossover alone can cost more than the digital audio chips we're talking about here. 

ADA will replace many components of a traditional audio system and allow systems designers to shift budget to speaker drivers. Most designers will agree that using high quality drivers is money well spent. While ADA will be priced like any commodity computer chipset, good speaker drivers will remain high price items. Advanced magnetics, high precision mechanical assembly, limited markets, and hand fabrication will conspire to keep quality driver prices relatively dear. 

Even if ADA only equaled the quality of our current rat's nest of preamps, amps, DAC's, cables, and crossovers, the increased expenditure on drivers alone would improve fidelity at any given system price point. This is only the tip of the iceberg. 

2. Garbage In, Garbage Out 

There's a school of thought that says that a speaker can only sound as good as the electronics feeding it. Replacing the speaker correction DSP, DAC, preamp, amplifer(s), and analog crossovers with a tightly integrated digital solution will provide higher signal fidelity at lower cost. With ADA there is no preamp, no analog parts variance errors, no analog parts drift, no compression in analog crossovers, less loop area for RFI problems, etc. Digital amplifiers are receiving good reviews, and they will only get better as the market focuses its resources. 

Some high enders are so concerned with signal fidelity that they use outboard power supplies like the $1000 PS Audio Power Plant 300. These devices resynthesize an AC waveform from household AC to get a more constant and noise free voltage for sensitive electronics. Some digital amplifiers, like Cirrus's, monitor power supply voltage and take it into account when calculating the output. Yet another $1000 to be spent elsewhere in an ADA system since a $15 chipset will remove the need for the component. 

3. Synergy is the speaker designer's friend. 

Digital audio pioneer Meridian has been producing highly acclaimed active loudspeakers for years. Meridian's designs exploit the advantages of integrating DSP, DAC's, and (analog) power amps into the speakers. There are no exotic drivers or revolutionary amplifiers and yet their systems are some of the highest rated in the industry. 

By performing all signal processing in the digital domain and designing each stage to work in tandem with the next, Meridian is able to extract a high level of performance from the components used. The same chips, amplifiers, and drivers used in more traditional stand-alone equipment would result in a system of lower performance at a higher cost. 

ADA will allow companies without digital hardware expertise to perform similar feats but with digital amplification and at lower costs. Sony experimented with digital input audiophile speaker systems back in 1998, as did Dunlavy Audio at the 2000 Consumer Electronics Show. Dunlavy showed a modified version of their SC-IVa with S/PDIF digital inputs, digital crossovers, and a Spectron 600 Watt PWM amplifier for every driver. 

There are many advantages in designing a multi-amplified system with active crossovers, some of which are listed here. There are extra perks to be gained from implementing some or all of an active system in the digital domain. 

 Digital amps are smaller, provide more power, produce less heat, and at a lower cost than their analog cousins. There's an inefficient driver you really like the sound of? Go ahead and use it. Want to biamplify or triamplify? Do it for the cost of a single analog amp. 

 Broader transducer choices for designers. DSPs can provide transparent EQ and crossover flexibility. Know of a driver with great time domain performance, but it's not completely flat? Flatten its frequency response with the DSP. Doing the same complicated EQ in the analog domain can be tricky. 

 expensive, and degrade sound quality. If ADA becomes pervasive, we may even see new drivers that tradeoff frequency response flatness for low non-linear distortion. 

 Crossovers impossible for analog designs are possible with digital crossovers. You need 4th order crossover slopes, but like the linear phase properties of 1st order designs? With digital FIR filters you can have both. 

 More decor friendly packaging could boost sales. ADA solutions could be designed with less boxes and cabling since there are fewer components. Speaker enclosures can be made more acceptable to consumers. Dedicated high power amplifiers can reduce bass cabinet size, and DSP delayed signals can align drivers' output without unusual baffle designs. Wireless networking such as IEEE-1394 could even remove the need for cables from the audio system to speakers with integral amplifiers. 

 With crossovers determined by coefficients in the DSP code, designers can test many more crossover shapes much faster than before. It's quicker to change coefficients than solder up a new board. Designers could A/B test crossover curves at the press of a button. 

See Meridian's site for additional ideas on digital integration. 

4. Upgradeable & Customizable speakers. 

This could actually create a new trend in the industry. Currently designers spend (we hope) a lot of time measuring and listening to the crossovers before going into production. How many times have we seen a manufacturer release a Mk II, or a factory upgrade to change crossovers in production systems? 

With DSP based crossovers, if a revision is warranted it could be published online and downloaded instantly at no cost to consumer or manufacturer. Perpetual Technologies has already started down this path with downloadable codes for their P-1A speaker correction DSP box. A company could treat the initial release of their speakers as a "beta" release. Hundreds of listeners in the field could then provide their input for the next crossover release. 


Say we go from the 50% efficiency of class AB amplifiers to 90% for our fancy digital PWM amps. We want 500 Watts RMS per channel so we have plenty of headroom. We still need fairly expensive transformers and capacitors for a 550+ Watt linear power supply. 

Can't we use a switch mode power supply (SMPS) for its cheaper, smaller transformer and better regulation under load? Accepted in the mass market and pro audio worlds, switching supplies do not have much presence in the high fidelity segment. It's difficult to suppress switching noise, and switching RFI plays havoc with nearby circuits and wires. Implementing a low noise SMPS requires much more engineering expertise than a quiet linear supply. However, some recent integrated SMPS controller chips have made high quality designs much easier to implement. 

Power semiconductor companies like International Rectifier are producing integrated solutions that incorporate "soft switching" logic such as Zero Voltage Switching (ZVS) and Zero Current Switching (ZCS). Both techniques have better power density, lower RFI, and reduce stress on the switching transistors improving reliability and product lifetimes. Integrated controllers can also provide Power Factor Correction (PFC), reducing AC current demand by 40%. The semiconductor companies are already developing versions of these components tailored for digital audio. 

In September 2000, Cirrus logic and International Rectifier announced they are working together to optimize power supply design, and high power MOSFETS optimized for the PWM output stages of digital amplifiers. Crystal's digital amplifier chipset has a sync-lock to easily integrate with a switching power supply. 

Clearly, the time for switching power supplies in high fidelity amplifiers has come. They have many sound quality advantages over linear supplies if implemented correctly, and can be cheaper. QSC Audio has been using a low EMI resonant switching supply they designed in-house for their acclaimed PowerLight and PLX professional amplifiers for years now. High-fidelity companies will start finding it possible and profitable to move to switching supplies since the new integrated SMPS controllers will lower costs and reduce required design expertise. 

It may already be happening. TacT's high fidelity PWM amp uses a switching supply. The highly regarded model 10 & 12 analog amplifiers from Jeff Rowland use a ZVS/ZCS switching power supply. California Audio Lab's CL-2500 MCA also uses a ZVS design to squeeze 5 x 500 Watt channels into a single chassis. These are all expensive products, but there is no barrier to prevent the technology to trickle down to mid priced units. 

Some advanced SMPS controllers use digital domain PWM control as do the audio amplifiers discussed in this article. After all, a regulated power supply is really just a type of amplifier. Digital amplifier designs may eventually integrate the PWM audio signal generation directly into the SMPS control logic, lowering the component count even further. Such designs would process the incoming AC power directly into the desired output signal without any conversion to DC. With this level of integration, it wouldn't be surprising to see a 300-Watt consumer ADA box for $ 300 or less in the next 5 years. 

The increased demand for advanced SMPS designs is due to demand for higher efficiencies and lower EMI/RFI. New European RFI regulations, longer battery life, efficient motor controllers, and computer & networking equipment are just a few of the driving factors. The worldwide market for integrated SMPS controllers and power semiconductors will grow rapidly in the next few years, and will result in lower costs, better specs, and tighter integration. All consumer audio has to do is sit back and enjoy the ride.


Moore's Law and rapid commoditization will drop the prices of ADA solutions to a point where they will be far cheaper to implement than the traditional DAC, amplifier, and crossover combination. The high-end industry will have an inexpensive solution to every part of the reproduction chain but the speaker drivers and cabinetry. 

So far it looks like Texas Instruments, Motorola, and Cirrus logic are in the lead for high fidelity ADA solutions. Don't count out consumer electronics powerhouses like Sony and Sharp, but their solutions are more likely to be used in-house. Things could get even more interesting if Analog Devices gets on the bandwagon this year. 

To accelerate the acceptance of ADA in the high-end market, some enterprising manufacturer could produce a branded or OEM multi-channel ADA unit for use in digital active speakers. Add some filter design software and half the industry would be knocking on your door. QSC has the right idea with their DSP-3 module and filter design software for their analog amplifiers. 

Speaker design houses could integrate ADA boxes into their designs, or they could be purchased separately by the consumer and the appropriate software downloaded from the speaker manufacturer's website. The convergence of digital audio makes it likely that this type of product will be developed by both amplifier companies and digital audio companies alike since the cost of adding one to the other will be low. 

High end amplifier companies will need to enter the ADA ring as even mid-priced digital power amplifiers will challenge their fidelity. Some high-end companies will probably become nothing more than valued brand names and distribution channels for repackaged OEM amplifier modules. However, when everyone is using the same five amplifier chipsets the problem for companies will be differentiating themselves from the pack. Since high performance will be easier to achieve, manufacturers may try to add value with unique packaging and hardware/software features targeted at niche markets. For instance, ADA boxes with integrated high quality Analog to Digital Converters like might be a selling point for those with large record collections. 

Luckily the coming transition to ADA won't force companies to abandon their current customer base. With the inclusion of an analog to digital converter ADA boxes could be used just like traditional amplifiers. Sharp is doing this with their SM-SX100. Manufacturers will be able to serve different audio market segments with the same electronics. 

Most of the digital amplifier chipset vendors are building products at three power levels. Portable, consumer, and Professional / Audiophile. This is a sensible strategy as the market exists today. However, if the cost and quality of ADA electronics causes the audiophile market to transition to active speakers, there will be reduced demand for high power amplifiers. With multi-amped active speakers, high fidelity companies may end up piggybacking on the midpower mass market segment. 

In multi-amped designs it is common to see every driver's amplifier under 100 Watts. A 30 Watt module, bridged to provide 60 Watts for a driver would satisfy many active driver requirements. With clipping distortion in digital amplifiers mitigated by transparent signal processing, designers will be able to size the system's amplifiers better, and without the risk of expensive warranty repairs for blown drivers. Mid-power mass-market modules could offer the incremental cost savings of high volume parts. 

With the DSP power and programmability of consumer products like Perpetual Tech's P-1A and Sony's TA-E9000 ES, and innovation from pro audio companies like QSC we're already halfway to high quality ADA active speakers. We can't be sure exactly how events will play out, but high fidelity enthusiasts have a lot to look forward to in the years ahead. 

Want to add new information to the pot or just argue a point? Post a message to the discussion group. To get posting rights and sign up for future news bulletins join WaveFront by following the link under the Members box on the upper left of this page. 

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