Q&A with Charles Hansen of Ayre Acoustics Page 2


"Basically we want to do as little as possible inside the DAC chip."

What elements of a DAC chip do you like to see implemented in the chip and which do you like to see implemented outside the chip?
Basically we want to do as little as possible inside the DAC chip. There are two reasons for this:

a) Any things happening like including a digital filter will cause both radiated RFI and power-supply conducted RFI such that the digital signals will contaminate the audio signals.

b) Everything in the world of consumer electronics is done for the lowest possible price. This means we can always do a better job by doing it ourselves outside of the DAC chip. External current-to-voltage conversion is absolutely critical for good sound quality. We can implement any kind of digital filter we want in an external FPGA, including minimum phase, with whatever rolloff and window we want. Simply put, we can just do it better ourselves.

Can you discuss the cause and effect of Pre-Ringing and Pre-Echo?
It's really quite simple. Any filter steeper than 6 dB/octave (first order) will ring when a transient event comes along. The ringing can be minimized to any arbitrary degree by making the transition as gentle (as opposed to a sharp transition) as desired.

"...by the time you get to quad-rate sampling (176.4 kHz or 192 kHz), the compromises are practically non-existent. One can have flat frequency response to 40 kHz or 50 kHz, and still have a filter with little or no ringing..."

It is not practical to have a gentle transition with single-rate audio (such as found on CDs). You only have 2 kHz (20 kHz to 22.05 Hz = Fs/2) to get at least 96 dB (16 bits) of attenuation, so there will always be a lot of ringing. We can minimize it by letting the rolloff start at (say) 18 kHz instead of 20 kHz. Already that cuts the problem in half as now you have twice the bandwidth to perform the filtering, which means half as much ringing. There are additional compromises that can be made, but they will always be compromises.

On the other hand, by the time you get to quad-rate sampling (176.4 kHz or 192 kHz), the compromises are practically non-existent. One can have flat frequency response to 40 kHz or 50 kHz, and still have a filter with little or no ringing (in the case of the moving-average filter).

When a linear-phase filter is used, there is no phase shift at all, but then half of the ringing occurs before the impulse, and the other half occurs after the impulse. This never happens in the real world. In the real world there are always echoes (sound reflections from nearby objects), but they always occur after the impulse. It is impossible for echoes to occur before the event. This is one reason that standard digital technology tends to sound un-natural.

On the other hand, a minimum-phase filter has the same total amount of ringing, but it is all moved until after the impulse. This type of filter does have some phase shift, but it is very small and only at very high frequencies. It is like moving your head 1/2" or so closer to or further from the speakers.

inside the Ayre QA-9 USB ADC

There seems to be some confusion between Upsampling v Oversampling. Could you explain the differences?
No, I cannot. The standard technical term is actually interpolation, which simply means to calculate interpolated data points between the original data points that were captured during the original recording. In video this process has traditionally been called upsampling, and in audio it has traditionally been called oversampling.

Then about a decade ago a company that made sample-rate conversion boxes found that if they used this box between the transport and the DAC box, that is changed the quality of the sound. They called this "upsampling" because there was already a digital oversampling filter in the DAC box.

Nobody ever came up with an explanation for why this made any difference in the sound, because it is impossible to add any actual new real data (ie, real resolution) once a recording has been made. And adding another oversampling filter to the existing one is nothing new, as virtually all oversampling filters are a concatenation of several 2x filters in series. For example 99.9% of the time an 8x filter is made from three 2x filters in a row, simply because it is the cheapest way to do it.

Probably the best explanation is simply that if you "upsample" the data by (say) 4x and then "oversample" the data by (say) 8x, you end up with 4 x 8 = 32x oversampling, and changing the oversampling rate changes the sound. Technically, the higher the oversampling ratio, the easier it is to filter out the "image" frequencies.

"What this means in real life is that the "steps" in the stairstep waveform output of the DAC chip are smaller, which means the non-harmonically related frequencies represented by the steps are at a higher frequency and easier to filter out from the desired original signal."

What this means in real life is that the "steps" in the stairstep waveform output of the DAC chip are smaller, which means the non-harmonically related frequencies represented by the steps are at a higher frequency and easier to filter out from the desired original signal.

Over the years, some people have decided to call interpolation to a non--integer multiple of the original rate (eg, 44.1 kHz to 96 kHz) "upsampling" and to call interpolating to an integer multiple of the original rate (eg, 44.1 kHz to 88.2 kHz) "oversampling". But it is all just marketing terms and not technical terms, so people can call it anything they want.

In general, interpolation by a whole integer (eg 2 x 44.1 kHz = 88.2 kHz) sounds better than interpolation by a non-integer (~2.176870748 x 44.1 kHz = 96 kHz), probably because it is a simpler operation with less error in the calculation.

Some people claim that Non-Oversampling (NOS) DACs have a distinct sound as compared to DACs that employ oversampling and digital filters. What is your experience with the NOS approach and are there benefits to non-oversampling that cannot be achieved in any other way?
The lowpass playback filter in a traditional DAC (technically called a "reconstruction filter") filters out the steps in the waveform, and therefore more accurately recreates the original waveform. But since most digital audio is played back at the CD sample rate of 44.1 kHz, this requires a very steep filter with a lot of ringing.

A non-oversampling DAC removes the reconstruction filter entirely. This obviously eliminates any chance for the filter to ring, but it also leaves the large "stair steps" in the playback waveform. Quite often the analog stage of this type of DAC will incorporate an analog low-pass filter in the form of a transformer.

"But it doesn't matter whether a digital filter is used or an analog filter is used -- there is always a tradeoff between the sharpness of the filter (amount of ringing) or the gentleness of the filter (amount of "aliasing", or leakage of stairsteps)."

But it doesn't matter whether a digital filter is used or an analog filter is used -- there is always a tradeoff between the sharpness of the filter (amount of ringing) or the gentleness of the filter (amount of "aliasing", or leakage of stairsteps). The only difference is that an analog filter is always a minimum-phase type, while a digital filter is almost always implemented as a linear-phase type.


The Ayre QB-9 DAC has been very well received and employs "single-pass" 16x oversampling and a Minimum Phase filter. Can you talk about the benefits of this approach?
These are two completely independent parameters with two completely independent benefits. Performing all of the calculations in a single pass (rather than a concatenation of 2x stages) requires more computational "horsepower". We use an FPGA (Field Programmable Gate Array) with hundreds of thousands of gates that can be easily configured to suit our needs. In the past this would have been prohibitively expensive, but today the prices for these parts are quite reasonable (although still far too expensive to be used in a mass-market product).

"...a minimum-phase filter has all of its ringing after the impulse. This is the only way that echoes occur in nature, and therefore a minimum phase filter generally sounds more natural than a linear phase filter."

The advantage of performing the computation in a single pass is that there are always rounding errors at each oversampling operation. If you perform it all at once, the rounding errors are minimized. But if you perform 16x oversampling as a series of four 2x stages (the normal way, as it is the cheapest way), then the rounding errors are compounded four times.

As mentioned before, a minimum-phase filter has all of its ringing after the impulse. This is the only way that echoes occur in nature, and therefore a minimum-phase filter generally sounds more natural than a linear-phase filter.

I think that linear-phase filters became popular because around the time that digital audio was being commercialized, there was a trend to develop linear-phase loudspeakers and therefore linear-phase was perceived to be a generally good thing. Since it is trivial to make an FIR (Finite Impulse Response) digital filter either linear phase or minimum phase, and somewhat cheaper to use a linear-phase design, linear-phase filters have dominated since the beginning of digital audio.

You've mentioned that Ayre will be adding DSD capabilities to the QB-9. Do you have a planned rollout date and how will existing QB-9 owners implement this change?
We almost never have planned rollout dates. We do have an order in which we will tackle various projects, but the thing is that all of our projects incorporate new technology that has never been used before. This is our greatest strength and also our greatest weakness. It means that when we release a product it is always a leading edge design that breaks new ground and will have a long lifetime in the market, but at the same time usually takes longer to develop than we anticipate. So we only end up developing two or three products a year, and they almost always take longer than planned.

"All of our USB DACs already have DSD-capable DAC chips, so it will be relatively easy to add this to our products via firmware updates."

When your "new product" consists of the same old technology in a new box, it is very easy to predict how long it will take to complete. But when you are doing something that has never been attempted before you continually run into problems that have never been solved before. Sometimes you can find a solution in a few days and sometimes it takes a few months to solve a problem. The end result is that we rarely work to a strict schedule.

In the case of sending DSD signals over the USB connection disguised as PCM, Gordon Rankin and I first discussed this several years ago when we found out that Sony had introduced a format called "DSD-Disc". This is basically an SACD, but without the copy protection so that it can be played on any computer.

We discussed that conceptually it would be quite easy to implement, but in the end decided that there wasn't any real reason to do so. At that time the only way to obtain DSD files was to (illegally in this country) rip a copy-protected SACD to your computer. But in the interim several small recording companies have introduced DSD music downloads. dCS was the first company to announce a standard for what is now called DoP (DSD over PCM). They made it an open standard and Andreas Koch, now with Playback Design, but one of the original developers of the Sony SACD recording hardware worked hard to perfect the standard, with a lot of input from Gordon Rankin.

All of our USB DACs already have DSD-capable DAC chips, so it will be relatively easy to add this to our products via firmware updates. We may or may not make other upgrades at the same time; we haven't decided yet.

"The advantage of DSD is that there is no filtering, and therefore no ringing on the record side."

This leads us to the question of DSD and PCM playback. Do you see a benefit to one approach over the other?
The advantage of DSD is that there is no filtering, and therefore no ringing on the record side. There is some relatively gentle filtering on the playback side with minimal ringing. The disadvantage is that above 20 kHz the noise climbs very rapidly, so that at 100 kHz the S/N ratio is only about 30 dB. Well, in a recording of music (as opposed to a recording of, say, bats) there is absolutely no signal at 100 kHz that is within 30 dB of the full-scale output.

So the effective bandwidth of DSD is really only 30 to maybe 40 kHz. It also requires completely new equipment, not only for playback, but also for the entire recording chain.

In my opinion, the best solution is to use quad-rate PCM (176.4 kHz or 192 kHz) with very gentle filters that exhibit little or no ringing. It is that lack of ringing that gives DSD its sonic benefits and this can be obtained with quad-rate PCM, but there is no problem with out of band noise, and standard equipment can be used for recording and playback. This is what we have done in our new QA-9 A/D converter.

In the old days, to play a high-res recording you had to use either an DVD-Audio player or an SACD player. There were a few audiophile-grade SACD players made, but almost no audiophile grade DVD-Audio players. So there wasn't much point to buy the discs, because the proper playback equipment wasn't available. But with the advent of computer-based digital audio playback, high-sample-rate PCM is trivial, and even DSD playback is possible. So the entire game has changed. The format war is a thing of the past and it is easy to download high-res audio files with a broadband internet connection.

"It would be a lot of fun to make some $50,000 "statement" monoblock amps, and it would be just as much fun to make some entry-level equipment that more people could afford."

Can you talk about any new products that are on the horizon for Ayre?
Gosh, at any given time we have at least a dozen or two products to choose from. We are finishing up an integrated amplifier that sounds incredible, we would like to make a proper headphone amplifier, and maybe later we will make one that includes a DAC for those that don't already have a computer audio system. We still get requests for loudspeakers ever since I left Avalon twenty years ago, we really want to make a multi-input DAC and we have some great new ideas about how to do that. I personally would like to make a set of headphones, as I have some different ideas about how those could be improved. It would be a lot of fun to make some $50,000 "statement" monoblock amps, and it would be just as much fun to make some entry-level equipment that more people could afford. This list is practically endless, and the only question is what order to do them in.

For more on Ayre's Minimum Phase filter, check out their - MP Filter white paper.
Share | |
Rob McCance's picture

Great QnA Charles and Mike. Interesting to hear the technical discussion.

I'm a EE and a Audiphile (with a music background) and I no longer even attempt to explain anything to Vinyl/Analog biggots who will drone on forever about things they have no clue about. I typically just nod my head now and hope that they soon pipe down so I can hear the music.

Had one the other night at a listening session try to explain to me about how digital was just ON and OFF and therefore there was no way it would ever sound as good as analog. 

Finally I said, "before you get overly excited, know that this vinyl you are listening to right now was recorded and mixed on digital gear before being squished onto that platter."

He quickly got another beer than changed the subject.

And so will I..

Currently, I'm using the Metrum Octave who (best I can tell) has tried to make a NOS DAC using extremely fast chips to tackle some of the issues. 

I'm always on the lookout for innovative designs attempting to solve the digital playback issues. Very interesting the angles Ayre are using.

slim's picture

makes we wish for a DAC that offers all the discussed options in one unit, switchable, to get an aural grip on the sonic differences implied by the choices made:

- choice of digital input format: DSD, PCM, DoP

- choice of sampling: NOS, 2-, 4-, 8-, x-fold oversampling

- choice of filter: minimum phase, linear phase, apodizing, etc.

I would not need to own such unit, it should help to narrow down the choices to what would likely suit one's preferences.

Charles Hansen's picture

We already did all of that work so that you wouldn't have to.

Regarding the input formats, there isn't any performance difference between DoP (DSD over PCM) and he traditional DSD formats. So there isn't any point in comparing these two. Our A/D converter offers the two currently existing formats, SDIF-2 and SDIF-3. The next version of the DoP format should cover A/D converters, so we will offer a software upgrade for that after that standard becomes availaable.

We listened to NOS, 4x, 8x, and 16x and 16x was the clear winner, so that's what we used. That was at the single-rate sample rate. The data is fed to the DAC chip at the same overall rate for each sample rate, so dual-rate data uses 8x oversampling, and quad-rate data uses 4x oversampling. All of these arrive at the DAC chip at the highest rate it can handle, 706.4 kHz for multiples of 44.1 kHz and 768 kHz for multiples of 48 kHz.

In all of our tests, minimum-phase filters sounded better than linear-phase filters, so we use them for all of the filter choices.

Apodizing filters require a steep rolloff (with more ringing) to filter out any pre-ringing that may be present from the recording equipment. These are incorporated into our "Measure" filters, while the "Listen" filters all use slow rolloff filters with minimal ringing. You can listen to that and ssee which you prefer, but every review I've ever read agrees with our choice of using the "Listen" filters. The only reason that switch exists is that in some countries, products that don't measure perfectly are crucified in their audio press. So we put the switch there so that the reviewers in those countries won't give our products horrible reviews if the frequency response is down -0.5 dB at 20 kHz.

Our belief is that we are not trying to suit one's preferences as much as we are trying to build the most natural and realistic equipment we can. As noted above, we do the work so that you don't have to. I think any other approach is essentially saying, "We don't really know which is best, so why don't you try it and see what you think?

slim's picture

and for having done the work so that I don't have to.

Yet, being a natural scientist way more than an audiophile, I sympathize a lot with the approach mentioned in your last statement

"We don't really know which is best, so why don't you try it and see what you think?"

I do know that "best" is relative in the audio world, and let me make clear that the product I am wishing for is a hypothetical one, sort of a "probe DAC", and I do not expect any company to ever manufacture it.

Charles Hansen's picture

There are only two ways I can think of that the average enthusiast can try various digital filters to hear what effects they have on musical playback. The first is to borrow or purchase a product with one of the recent Wolfson DAC chips. I believe that these have five different digital filters that (in some products, at least) are selectable by the user.

I am not really familiar with these DAC chips as they are voltage output devices, and as I explained in the interview, I have little to no interest in them. The critical questions would be:

a) When selecting digital filters are all other parameters held constant? If so, then a fair comparison can be made. If not, then one doesn't really know which parameter results in the change of sound quality.

b) Does the unit offer the specific comparisons in which you are interested? Perhaps you want to compare a linear-phase slow-rolloff filter against a minimum-phase slow-rolloff filter. Those two options may not be offered by the DAC chip.

When we ran our experiments, we used FPGA's to implement the digital filters. This allowed us to program virtually any type of digital filter we wanted. We would just change one variable at a time, and switch between the two filters via a toggle switch on the rear. This is the only fair way to conduct such a test.

The other way would be to purchase an older non-MP version of the Ayre C-5xe. It had two user selectable digital filters -- a sharp rolloff ("brickwall") and a slow-rolloff (for improved transient response. This would allow you to compare these two parameters. Then you can send the unit back to the factory for the MP upgrade. We had basically the same two filters, but now they were implemented as minimum-phase types (hense the C-5xeMP desgnation). This test would not quite be fair, as we also bypassed the digital filter built into the DAC chip, and the "Measure" (sharp rolloff) filter was reduced slightly in frequency to filter out any ringing from the equipment used in the recording process (ie, "apodizing" filter).

But it would give you a good idea of what the differences were. You can also search the online forums from around that time to get some responses from some of the Ayre owners who had their units upgraded to MP status, such as this one:


Good luck in your quest for better sound!

labjr's picture

Will the QB9 be upgradable to decode DSD128 ?

Charles Hansen's picture

You have to remember that the only reason that DSD even exists is that the patents for CD were expiring. This used to amount to a $1 billion per year royalty stream for Sony and Philips, and they did not want to lose this income. They needed to come up with a system that they could patent and license. It's ironic because SACD failed in the marketplace, in large part because it is not readable by computers. It is only now that Sony has released "DSD-Disc" without copy protection that will play on computers, that there is a resurgence of interest in DSD.

DSD-128 is probably what DSD should have been all along. "Normal" DSD is 64x the standard CD sampling rate. Since it has only 1 bit, they use a 7th-order noise shaper to get reasonable performance out of the system.The problem with "normal" DSD is that while the signal to noise floor in the audio band is 120 dB (roughly 20 bits of resolution), above 20 kHz the noise rises very sharply. This has two effects:

a) The practical frequency response is limited to 30 or maybe 40 kHz at the most. Above that, any audio signal is going to be swamped by the noise.

b) You can only perform one conversion to "normal" DSD without losing a significant amount of performance. There is some mathematical equation that shows that more than "x" number of conversions to and from DSD will result in more noise than signal, and "x" is something relatively small, between 4 and 8 as I recall.

So for recording in the studio, they almost are forced to use double speed DSD (DSD-128) to have the ability to convert the master recording to PCM for editing, EQ, fades, mixing, adding reverb or other effects. After that it is converted back to DSD a second time, almost invariably at the "normal" DSD rate, since there is almost no hardware that will play back DSD-128. Remember, they have to sell product to stay in business. There really isn't much use for it as a playback format currently. Some of the problems include:

a) Sending DSD disguised as PCM over the USB connection requires a 176.4 kHz connection. Going to DSD-128 would require a 352.8 kHz connection. While USB 2.0 Class 2 Audio can easily handle this, it puts more stress on the signal chain. Everything has to be completely up to snuff, including the USB cable itself.

b) Download times are doubled, as are storage capacity requirements.

c) I don't know if the DAC chips Ayre uses (Burr-Brown DSD1792A in the DX-5 and  DSD1796 in the QB-9) will accept and decode a DSD-128 signal. They were designed before this format was in use and the data sheets are very spotty in this regard. We could try it and see, but I kind of doubt that they would work.

d) There is barely any DSD-64 software available at this time. DSD-128 is even rarer. I think 2L has a demo track or two but I don't think that this is something that will ever gain widespread support.

The best use for DSD (in my opinion) is as a digital playback format for analog tapes made before the digital era. But as I noted in my interview, I think that quad-rate PCM, when done properly, gives all of the sonic advantages of DSD with none of the practical disadvantages.

So the short answer to your question is "Maybe, but most probably not."

CG's picture

Charlie has a really great point.  Much of audio realism seemingly has to do with the implicit and explicit filtering in the overall system.  How much energy is devoted to that aspect of the design process?


One wonders how Charlie's observations might apply to Pono.  What kind of digital conversion will they be using?  How about all those filters?

At one time people raved about HDCD recordings.  Were they better because of the math?  Or because of the really high quality ADC needed for the process designed by Keith Johnson for Pacific Microsonics?  Same questions here.  I hope they get it right.

Charles Hansen's picture

We don't know much about the details of PONO yet, so all we can do is speculate. As far as HDCD goes, I would  guesstimate that probably at least 80% of the sound improvement was due to Keith Johnson's superb ADC box.

There were three other features. The Low-Level Extension only kicked in at levels below -40 dBFS. This only happened rarely, during a few very quiet passages in classical music. The Peak Extend was a compansion system, but was not recommended for use with pop music becausse undecoded, it would further compress already over-compressed music to make the overall sound worse. And there were two different anti-alias filters used that automatically switched in depending on the spectral contect of the music being recorded, but there was only one playback filter for both record filters.

The Pacific Microsonics ADC's are still highly prized for their sound quallity, indicating just how advanced Keith Johnson's ADC design was.

pbarach's picture

Really interesting article that left me (a non-engineer) with a couple of questions:

1. Why does a sharp filter produce more ringing than a gentle filter?

2. What's wrong with op-amps? I see a lot of criticisms of them, but I'm not sure exactly what negative effects they are supposed to have on sound quality.

Charles Hansen's picture

These are two really great questions, Thanks for asking them.

I was having a tough time with the first one. Richard Feynman, the late Nobel Laureate in physics, was once asked by a Caltech faculty member to explain why spin one-half particles obey Fermi Dirac statistics. Rising to the challenge, he said, "I'll prepare a freshman lecture on it." But a few days later he told the faculty member, "You know, I couldn't do it. I couldn't reduce it to the freshman level. That means we really don't understand it."

(However, he also once said, "If I could explain it to the average person, I wouldn't have been worth the Nobel Prize." But since this isn't worth the Nobel Prize, I'll give it a shot.)

At the website for the University of Oregon is a page that talks about electrical resonances:


It says, " Resonance in electrical circuits is a difficult and very non-intuitive subject. The way it is typically taught is through analogies to mechanical systems which exhibit the same kind of behavior." and shows an animated GIF picture of a simple resonant mechanical system:

Physical Analogy

In this mechanical analogy to an electrical resonance, the moving block has mass (which represents inductance), the spring has compliance (the opposite of stiffness, which represents capacitance), and the friction of the block sliding on the surface (also called damping, which represents resistance).

The compliance of the spring together with the mass of the block will determine the frequency of resonance. But in this discussion we are concerned with the sharpness of the "knee" between the flat part of the frequency response and the part that rolls off. The sharpness of this knee is set by the damping or resistance. A lower resistance in the electrical circuit creates a sharper knee, which corresponds to less friction in the mechanical system. In this case, one can intuitively see that if the block is disturbed, that it will oscillate (ring) for a longer period of time than if the friction is high (high electrical resistance). I am hopeful that this will help to understand why a sharper "knee" or "corner" in the frequency response will lead to more ringing in a filter.

The way to make the slope of the electrical rolloff steeper is to add additional sections to the circuit. In the mechanical analogy, this would be like adding additional springs and blocks to the first set, all coupled together like railroad cars in a train. Again, I am hopeful that once can see that if there are multiple springs and blocks oscillating that they will oscillate for a longer period of time before the movement dies down to zero.

(If this explanation is unclear, please let me know and I will try again.)

Regarding integrated circuits (IC's) in general (of which op-amps are a sub-type), there are many advantages to them, which is why they are so widely used in modern electronics. They are small, light, and cheap compared to a circuit constructed from discrete parts (eg, transistors, resistors, and capacitors). However, very few of these advantages have anything to do with performance. When an IC is used, it is similar to baking a cake from a box of Betty Crocker mix. There is very little that one can do to optimize the values and quality of the parts inside. In contrast, a master chef making a cake from scratch can select the quality of the ingredients and adjust the proportions to attain better results than an off-the-shelf box. And just as with cake mixes, some IC's offer higher performance (taste better) than others. But just as with cakes, the highest performing circuits are typically made from discrete parts, where the designer can select the precise values and quality of the individual components.

Most audio components use a special type of IC called an "operational amplifier", or op-amp for short. As the full name indicates, these circuits were first used in the '50s for analog computers before digital computers became practical (with the advent of IC's). The most basic operations that an op-amp can perform are to add (sum) two signals or subtract (take the difference) of two signals. Almost universally it is the latter that is done in audio circuits, where the output signal of the op-amp is compared to the input signal to the op-amp.

Since no circuit is perfect, there will always be noise and distortion of the output signal compared to the original input signal. By comparing the two, an error signal is generated and the error is subtracted from the input signal in a way that tends to cancel steady-state distortion. This is called "negative feedback" and is used almost universally in all audio products, whether based on tubes, discrete transistors, or IC's. But an op-amp uses negative feedback as its fundamental mode of operation. It is not possible to use an op-amp without also using negative feedback.

From a mathematical (theoretical) perspective, this works perfectly (as long as certain rules are obeyed). But many listeners prefer the sound of circuits with little or no feedback, despite what the theory says. At this point nobody has any proof as to why this should be. One argument (and in most cases it will just lead to further arguments!) is that negative feedback can only correct an error after it has occurred. Obviously this is true, but the theorists will claim that the correction (when done properly) happens so quickly that it is impossible for the human ear/brain to hear it.

But the theorists also claim that all cables, amplifiers and CD players sound the same, too. So in the end the best thing to do is let your own ears be your guide. It is always best to listen to music you are familiar with, and in a system and listening room that you are familiar with. This is most easily done if you have a quality dealer nearby that will let you borrow a component for a short while to listen to at your home. (Please note that it is considered bad manners to borrow equipment from a dealer and then purchase it elsewhere.)

I hope this helps, and have fun discovering which components bring you the greatest musical satisfaction!

Enter your AudioStream username.
Enter the password that accompanies your username.