That’s pretty much it. Now, there are multiple words used to describe different errors made in each of these areas by stereo equipment. For definitions of those various words, see the Glossary: Sound Quality in the Audiopedia section of TheAbsoluteSound.com.

But the words all tie back to these basic observable elements of listening to music, defined by the physics of music and the psychobiology of hearing. The words aren’t gobbledygook or an advertisement, they point to observable phenomena that humans (i.e. you) can perceive and describe.

If you are curious why we don’t use quantitative measures as the primary means of communicating sound quality, please see our Audiopedia article on our review methodology.

For convenience, we classify the ranges of pitch into groups so we can talk about them more easily. So, you will see The Absolute Sound reviews mentioning these regions, which mean approximately the frequency ranges shown below:

• Low Bass: 10-40 Hz
• Mid-Bass: 40- 90 Hz
• Upper Bass: 90-180 Hz
• High Bass: 180-300 Hz

• Low Midrange: 300-600 Hz
• Mid Midrange: 600-1200 Hz
• Upper Midrange: 1200-2500 Hz

• Low Treble: 2500-5000 Hz
• Mid-Treble: 2500 to 10000 Hz
• Upper Treble: 10000-20000 Hz

The numbers here are excessively precise, but they give you a sense of the ranges being discussed.

Now, to help you learn to connect these numbers and words to actual musical sounds, we offer a chart we have found especially useful. It shows the frequency ranges of a variety of instruments. It conveniently shows both the fundamental and overtone ranges of each instrument. Instruments and vocals are based on resonators (strings, diaphragms, reeds, etc), and all resonators have a basic or fundamental frequency and a range of “harmonics” which are simple multiples (2x, 3x, 4x) of the fundamental frequency.

Clicking the image below will navigate you to a helpful interactive version of the chart.

So, it certainly isn’t clear that value for money is the issue; in fact, it seems quite clear that value for money is better in 2025. But there is something bothering people about the availability of more advanced, exotic, and expensive gear. It might just be that people resent what they haven’t got, but we wanted to know more about how exceptional and “insane” audio is or is not at the high end.

With that in mind, we created the following chart. In column A, you see various things you can spend money on to get an experience. In column D, you will see an estimate of the cost per hour of that experience. We haven’t been able to think of a way to adjust for the quality of the experience, but if you like listening to music, we’d say music stands up pretty well on the quality front.

As a reference for these comparisons, we used a $100,000 stereo system. That is a nice round number, and most people would agree that it is a fairly big number. (For those who wonder, “Who can afford this stuff?” just look around: approximately 1 million cars in or above this price range sell in the U.S. per year; that’s about 5% of new vehicle sales.)

Now for the punchline.

Note that Audio for Music is very close to the lowest cost per hour for the experience delivered:

What we are talking about here are distortions where the magnitude of the distortion is difficult to hear, per se. These are Below-threshold Distortions in the sense that they are generally there, but if the BTD is the only change, it is hard or impossible to hear and thus is below the threshold of hearing.  

Given this inaudibility, per se, some people choose to view BTDs as irrelevant or even foolish or idiotic, or deceitful to discuss or consider. That is a philosophical stance more than a logical or empirical one, as we will discuss, but listeners have to make the call. 

Our view is that BTDs are worth considering, but under specific circumstances. To explain, we start with an analogy. Imagine that salt being added to soup you are making is a BTD. You add one grain of salt to 3 gallons of soup. “No detectable difference” is the likely result of a taste test. However, we would suggest from experience that concluding that “salt makes no difference” would lead to inferior culinary results. Perhaps obviously, a pound of salt would render the soup inedible. And, logically useful for our purposes, there is some number of grains of salt where the saltiness becomes noticeable. The salt level becomes “above threshold”.  

It is easy to imagine that something similar could happen in audio. If we have a hearing detection level that requires distortion of level 1, and if we have a component with distortion of level 0.7, then we will not be able to hear the distortion of the component. But if we keep employing components of distortion level 0.7, these distortions may add up to greater than level 1 distortion. In simple terms, 0.7 + 0.7 = 1.4, which in our simple model is audible.  

Experienced listeners will suggest that there is audio equipment where this threshold phenomenon is important. Our favorite example is with cables. Our experience is that a single cable change is often inaudible. But we often find that changing the entire cable loom (all the cables in a system) makes an audible difference.  

Now, it is important to add that we think addressing BTDs is logically a later step in audio system development for most consumers, especially if the BTDs cost money to address. If you have problematic speakers and a limited amplifier and are using compressed music sources, you have bigger fish to fry than BTDs. The same with setup. If you have an undamped room and poor speaker location, and a deep lateral null, you have bigger fish to fry. And, addressing BTDs often will cost as much as addressing these issues. On the other hand, if you have a well-developed $100,000 system, some attention to cables and cable routing and vibrations might make sense, and the cost may be relatively small.  

We summarize that BTDs are generally in Quadrant 4 of our “Priorities for Audio Improvement”: 

We close with a potentially important point. If you are addressing BTDs, we think you would want some sense that products you are considering might actually do something in the world of physics. That is, you aren’t interested in magic. When reviewing BTD-related products, we will often: 

a. Skip listening observations of the effect of the component, because obviously these are impossible without convolving effects of other changes

b. Provide some engineering logic for why the equipment might make a difference, based usually on the observations of engineers  

Neither a. (obviously) nor b. (logically) proves that BTD-related gear works. If you need “proof,” we simply suggest that you probably won’t get it and that there are plenty of other equipment and setup adjustments to consider, or you can just listen to music as is. BTD-related gear is not required for good results.  

If you have adopted an anti-BTD stance as a religion, that’s another matter. Good luck to you. If you are agnostic, we simply suggest test-driving the logic above. And be suspicious of (often histrionic) claims without supporting evidence or logic.  

In music audio, on the other hand, it is harder to experiment and the methodology for approaching the audio side of things is less rigorously worked out on average. We have worked on this for decades and have learned quite a bit. But in a nutshell, we use the sound of real music and real musical instruments (the absolute sound) as the reference standard. An audio system that can reproduce a guitar or a singer or a jazz band or a symphony so that it sounds believably real, will tend to be more satisfying for most listeners most of the time. Again, this latter point is our experience from over 50 years of listening to live music and audio reproduction of music across hundreds of reviewers. Note that this isn’t the same as saying you need to listen to a particular kind of music. The point is that the ability to render music that has a known sound believably is a benchmark for understanding how well audio systems will work across a wide range of musical types.

Why Believability?

Besides its predictive power for musical satisfaction across a range of music types and recording techniques, we use believability for another reason. We want to free listeners from a worry about something that has been called ‘accuracy’. The concept of accuracy, for some listeners, can lead to an excessive focus on small details that aren’t knowable (what instrument with what mic was used in the recording), and that aren’t essential to limiting musical distractions. ‘Believability’ is intended to indicate the reduction of distracting distortions to a level that gets audio distractions below a threshold.

The 6 Major Problems of Audio Believability

We assert, based on years of observation, that audio engineering is often focused on the refinement of established work. This makes sense in that audio engineering, like most engineering, will work on solving problems known but incompletely solved in a last generation of development. As a result, since audio engineering has focused on reduced distortion in the signal path from input to output of the consumer audio system, distortion reduction there will get maximum focus. We are approximately 100 years into this work, and we observe that distortion reduction in the input/output signal path is still valuable. And we observe that so much progress has been made that ongoing progress there will likely be gradual and perhaps of incremental impact.

On the other hand, issues outside the core and well-known distortions in the signal path tend to receive less attention. We believe it is time to highlight some of the opportunities beyond the classic input/output distortion model. We call these the six major issues with audio believability. These generally aren’t addressed in the standard model of measurement (science is often confined to phenomena that can be modeled with manageable mathematics). And they get less engineering effort because they are less well understood and because they often involve multiple participants in the overall music chain.

Two of these major issues of audio are largely outside the consumer audio equipment realm:

1. the problem of visual images
2. the problem of recording standards

But the other four are in the audio equipment wheelhouse but involve system level and psychoacoustic knowledge:

3. the problem of spatial imaging
4. the problem of bass in real rooms
5. the problem of dynamics
6. the problem of digital distortions

The problem of visual images can be described as a key difference between concerts and music at home. With a concert, there is a visual presentation happening that is generally missing in the home environment. There are questions about why and whether this is important, but in most of the audio world it goes undiscussed and unaddressed, particularly when the focus is on music with high quality recording.

The problem of recording standards refers to the wide variations in frequency balance, compression and imaging management used across recordings. At some level, these different approaches can’t all be right. At another level, it is completely unclear that the distortions present are genuinely helpful.

The problem of spatial imaging is simply one of presenting performers in a way that makes them distinct and believably present in a believable space. This does not seem to be a problem of stereo per se, but a problem of psychoacoustics and a problem of execution vis a vis emerging psychoacoustic knowledge. It may be easier or harder to address this issue by going outside the standard stereo architecture, but empirically this is not even close to obvious, in fact quite the contrary.

The problem of bass in real rooms is that residential listening rooms (whether purpose-built or not) have resonances and reflections and noises (environmental and music-generated) that lead to distortions of the music emanating from the audio system. These distortions are complex and very specific to particular rooms. These distortions are generally unlike what happens in clubs or concert halls.

The problem of dynamics is that the sound of real instruments and voices encompasses a range of levels that often extends outside the linear (and often total) capability of audio equipment. This is also a problem with recordings and with listening rooms.

The problem of digital distortions is that A/D and D/A processing leads to mathematical errors that are quite unlike the basic music signal and thus are both obvious and distracting. The sources of these errors seem either poorly understood or involve system architectures that individual engineers do not control.

What are we trying to accomplish here?

I repeat: Be smart here, give this paper to a licensed electrician and let him do the work.

Please know that all items in this paper have been tested over 20+ years to identify definite and significant audible results. Among those people that have done this upgrade, no one has said merely “yes it’s better”; virtually all feedback is disbelief! Most people say “this is the sound I have always wanted in my system! Why isn’t anyone talking about this?”.

The main goal is thicker wall wire to the audio system. This may seem too simple but please read on.

People very often tell me “I have 20-amp dedicated lines” and assume they have done everything they need to do. By US electrical code definitions, a “20-amp dedicated line” will have 12-gauge wire in the wall. So, while you may have a “dedicated line”, 12-gauge wire is absolutely insufficient for high end audio systems. We are recommending 10-gauge or thicker wire here. It is the subject and goal of this paper. The gauge of the wire is FAR MORE IMPORTANT than the fact that the line is “dedicated”.

The subject of this paper works on the theory that the varying musical demands of your amplifier with its very big internal power supply are actually modulating the incoming power line, divorced from the utility (power company) by some amount of resistance (ohms). Even at 10-15 feet from the wall outlet to the breaker panel, 12- or 14-gauge wall wiring has too much resistance for audio purposes. The noise coming from your utility is often lower than you suspect, and the gauge of the wire is far more important because the amplifier itself is modulating the actual line it is plugged into. The amplifier demands current directly up and down moment to moment as determined by the music. This happens at audio frequencies that are of course above and below the 60 cycles from the power company. These demands are impressed on the line wavering the incoming voltage and so the amp is re-ingesting its own noise. It is also making the line dirty for itself AND the preamp, DAC, server, etc. This is possible because the wall wiring back to the breaker panel has some degree of resistance depending on the length of the run and the wire gauge (12-gauge or sometimes even 14-gauge).

Power conditioners and certain power cord designs help because they make an effort to “shunt” this noise (short it out and kill it) and consume the unwanted frequencies. These are dumped as waste heat.

But the other problem with 12-gauge wire is the amplifier starving for current moment to moment with the music. Not only are dynamics reduced, but starving the amplifier power supply causes blur and loss of detail across the audio spectrum especially in the precious mids and highs. A better answer is to reduce the resistance back to the breaker panel making it difficult for the amp to modulate the power at all, and at the same time and importantly, getting maximum moment-to-moment power for the amplifier power supply.

So, there are two benefits to reducing the resistance back to the breaker panel:

1. stop the amp from fluctuating or modulating the wall power,
2. provide the amp with maximum current (amperage) moment to moment as the music is playing providing an extraordinary increase in clarity.

Please Note: The single biggest goal of this paper is to install 10-gauge wiring up to 15-25 ft feet and 8-gauge beyond 25-35 feet, and then 6-gauge wire beyond 35 ft feet. In a recent test I have a detailed report from an audiophile who followed the paper exactly. He installed two 6-gauge lines and three 8 gauge lines from the Eaton CH sub panel at 35 feet. He tested his 200-watt class A/B amplifier into both the 6-gauge and 8-gauge lines. He reported that the 6-gauge line delivered better bass solidity, deeper extension, better dynamics and improved clarity. Everything else in this paper is there to be sure you get the maximum result.

Apply Silver Paste

In this next step, our objective is to further reduce the electrical resistance between the amplifiers and the breaker panel. Here we advise using special silver paste at key connection points. But read the details below, because you don’t want to use the paste on every connection.

These silver (actual Ag, not just color) pastes are called “grease” but are rather thick. Be wary of any others you find that are actually somewhat fluid. I have reports of migration of at least one audiophile silver grease that, because of voltage potential across the Line and Neutral, the paste attempts to physically migrate and close the gap between the hot and the ground. In one local attempt it burned up an outlet. MG Chemicals products are probably the thickest but do be careful in your judgement. It is almost crumbly and less likely to migrate upon inspection years later when used at distances between the poles of 1.5” or more. It does not seem to oxidize over time. There are reports of “audiophile” silver pastes that do oxidize and so the oxidized silver becomes worse than not applying silver at all.

1. If you are building power cords DO NOT try to use this stuff on plugs on each end of power cords. Again, the silver paste can migrate because of the voltages if the poles are too close together as they would be on a power cord plug.
2. Used on each side a wall receptacle is probably OK. Do so at your own risk.
3. The silver paste is OK in the breaker panel on the breakers and wire to the breakers.
4. DO NOT use the silver paste on interconnects, or speaker cables or any signal cables. While putting it on signal cables seems like a good thing, the paste is impossible to control and it smears around in use because it never really dries. The result is a partial or complete short across signal hot and ground. I added this comment based on one audiophile who tried it on his interconnects, got no sound in one channel, weak distorted sound in the other, and spent hours washing his RCA plugs and cleaning his input/output jacks on his components. I don’t t think it can be completely cleaned out and he should have replaced the jacks and plugs. In other words just don’t do it.
5. It is better to clean your interconnects and jacks once or twice per year (highly recommended). You can Google “interconnect cleaning” and “XLR input cleaning” and see procedures and chemicals to deoxidize the jacks on your equipment. Consult your equipment manufacturer for their recommendations also.

Main Point:

You will use the conductive paste at every AC power connection that is made starting by removing the breakers and applying it to the inside of the clip on the back of the breaker. No need to apply it to the busbar connection especially since these are always electrically live and fatally dangerous!! The breaker clip will transfer the paste to the live busbar.

1. I recommend new breakers if they are older than one year or so (they are cheap compared to the cost of your audio system). If you get the original equipment circuit breakers (like Square D, Siemens, etc.), from an electrical supply house (not Home Depot or Lowes), you will likely get silver-tungsten contacts inside the breaker. Cheap off-shore replacement breakers may have copper contacts which have higher resistance and will oxidize over time raising the resistance further defeating what we are trying to do here. Research with your local electrical supply for your brand of breaker panel and ask them to look up the breaker contact material to confirm it is silver or silver tungsten.
2. You will also use the silver paste on the wires where they enter into the screw terminals both at the breakers and the wall outlets. A thin film is all that is needed on all these connections and the silver actually performs better as a thin film. (more…is NOT better here).
3. The silver paste tends to get on the fingers and then everywhere else so be sure to clean up with Goo Gone or some such solvent since the silver paste is like liquid wire. It can be a finger- shock hazard if you are sloppy with it, so be sure to clean up any excess or smeared film with a solvent like Goo Gone EVEN IF YOU CAN’T SEE IT!
4. Note: Your electrician will have a non-corrosion paste that he always uses to preserve and prevent oxidation on an aluminum or copper connecFon but does not reduce the resistance of the connection anywhere near as well as the silver paste. The non-corrosion paste’s intent is to slow oxidation over time and nothing more. The electrician will say “oh I already have some stuff I use”. Let’s be very clear here: The electrician’s paste is not suitable for our purposes. The silver or a silver-loaded copper compounds are the only choices.

Now back to AC wiring!

Wire Gauge

I recommend at least two 20-amp 120 volt circuits run on 10 gauge wire up to 25 feet, 8 gauge wire from 25 to 35 feet, and 6 gauge wire from 35 to 60 feet. 6 gauge or 8-gauge may require a jump down to 10 gauge in a junction box somewhere near the wall outlets for the wire to fit into most wall outlets. Audiophile wall outlets may accept a bigger gauge. I have a report that Furutech outlets can accept 8 gauge and maybe 6 gauge. Please check other brands to confirm.

Breakers

Remember that 30-amp breakers do not pass any more power or have lower resistance. They simply kick off at 30 amps instead of at 20 amps. A 30-amp breaker could actually be more dangerous possibly causing a fire in the event of a true short circuit (very rare but that’s what the breakers protect against).

Install:

1. one dedicated circuit for all front-end equipment, and
2. one circuit for each amplifier and
3. one circuit for one (or two) subwoofers. (subwoofers are even worse than amplifiers at making the power lines dirty).

Try to find good outlets (something like the PS Audio Power Port, or Furutech, or Wattgate models. Generic commercial grade outlets are not a good substitute. Low- and medium-priced audiophile outlets are a good investment since they are heavier copper, better plated and really grip the blades of your power cord plug.

Sub Panels

If the distance between the wall outlets and the breaker panel is very far, like 60- 75 feet, one might consider a subpanel close to the listening room. A subpanel is a must at 80R-150R. I recommend the Eaton CH series panels using copper buss bars with silver plating. The internal power paths are highly conductive, again, with less resistance. In keeping with the purpose of the paper, ask the electrician to uprate the wire gauge between the main panel and the sub panel. 2-gauge should be minimum with 1-gauge or even 2/0-gauge being considered above 80 feet. Again, let’s remind the electrician we are not wiring for high current draw, we are wiring for the lowest possible resistance. The electrician will better understand our goals if we say the performance of our equipment will dramatically improve if we have the lowest possible resistance back to the breaker panel, even if we are not pull much continuous current.

Grounding

Also consider 2 or 3 ground rods 6 -10 feet apart fed together running back to your main breaker panel. Try to put the ground rods into the dampest place in the soil if possible. With some modern codes, the house ground to the re-bar that is embedded in the concrete foundation and concrete floors of the house. Your electrician will know the best procedures in your area.

AC Phase

For 120-volt circuits: MAKE SURE ALL EXISTING AND NEW CIRCUITS THAT YOU USE ARE ON THE SAME ELECTRICAL PHASE. I have had several direct experiences with an audio system connected on the opposite house electrical phases and the dual 120 volt feed from the electrical grid seems to make a good antenna to pick up RF. Connecting your system to only one electrical phase seems to prevent any RF issues that can damage equipment in areas with high RF. (No… you have no way to know if you are in a microwave path, or TV/radio transmission path, so just do it!) Usually, every other breaker in the stack is the same phase. In other words, starting at the top (first) breaker in the left column you will have “A” phase. The next breaker down (second) will be “B” phase, and then the next (third) will be “A” phase again, etc. So the two (or more) dedicated lines should be spaced every other breaker apart to be on the same phase. Another way to say this: On the left hand side of the panel, all of the odd numbered breakers on one side will be on one of the phases, all of the even numbered breakers will be on the other phase. Then on the right side of the panel the opposite may be true meaning the first breaker on the leR is on the same phase as the second breaker on the right, etc. On either side every other breaker will be on the same phase. Some newer panels may have one phase all on the left, and the other phase all on the right. If you don’t know and don’t know how to test, have an electrician help or do the work.

Warning!! Fatal voltages are exposed here. Decide if you are competent with an AC voltmeter and if you will not be dangerous to yourself and accept the sole responsibility for this decision.

If you have experience with an AC voltmeter measuring wall power and you feel you are competent then you can test between any two wall outlets to prove they are on the same phase by testing for AC voltage across the two shorter slots in the respective wall outlets (the longer slot is always the neutral, the shorter slot is always the “hot” or “line”). Measuring between the two outlets probing their respective “hot” (short slots) you should have a reading near zero volts and maybe floating around several millivolts (mv). Use an extension cord on one outlet to get near the other outlet to let your meter leads reach if necessary. Remember, the long slot is neutral, the short slots are hot or “line”. If your reading is 220-240 volts between the two short slots on the two outlets then the two outlets are on separate circuits on the opposite phases and should be corrected. This is done by moving one of the breakers to another position in the breaker panel so both outlets are now on the same phase. If you are not competent with a voltmeter, ask you electrician to determine the phase of your wall outlets. You accept responsibility to decide to do this test and know it can be lethal. Get your electrician to do it!

Tight Connections

Also, it is a good idea to ask your electrician to go around the breaker panel when he is done and tighten all of the set-screws that clamp the wires. This is especially important on the heaviest cables that feed the panel since these large feed cables are aluminum so tightening the large screw even a liNle bit refreshes the metal contact to the aluminum wire. These screws will be LIVE and lethal! They cannot be turned off so ask the electrician if he has the proper voltage-rated insulated tools to do this and if he is comfortable doing so. It is his decision and responsibility. Electricians will often do this any time they service a panel. Caution!! Lethal voltages are exposed here. DO NOT pick up the tools you own with the plastic or rubber grips and think you can do this yourself. Your tools are not voltage rated for this procedure, and it is fatally dangerous if you make a mistake, so DO NOT be tempted. Let an experienced, qualified, licensed, insured, electrician do this.

Incoming Noise

I use a meter made by Greenwave at $135 (https://greenwavefilters.com/product/emi-meter/) to get some scale of the electrical noise coming from the utility. Electrical noise can vary WIDELY across the country, affected by cell towers, radio and TVs stations etc. It can be very high frequency not only conducted on the copper wire but also following the outside of the wire like an antenna. The only way to know if electrical noise is present is to measure it. Use the meter during the day, evening, and late evening. Especially if your system sounds different in the evening. Use the meter direct into the wall outlets of your system with nothing plugged into any of the outlets. If audio gear is plugged in, then the reading will be lower because the audio gear is absorbing the noise. The reading that way will be meaningless. The noise is expressed in millivolts (mv). The mv reading is low below 50-100 mv, high around 400-500 mv and very high above 1000 mv up to 2000 mv which is the top range of the meter. I strongly suggest you use your ears to judge any power treatment. The meter is a guide. I advise not to chase small increments like 50mv compared to 40mv as you may not hear anything. Look for bigger changes and be sure to use your ears. You can measure power conditioners with nothing plugged into them but be aware certain components in some conditioners can show higher readings because of reflections between the power conditioner and the meter. The higher measurement from the conditioner may not be of concern. Again, use your ears with and without the conditioner. If the sound is better with the conditioner, trust that result.

Line Conditioners

I don’t generally recommend line conditioners for amplifiers when the system is done as described above. It is generally better to go straight into the wall. Again, use the Greenwave meter as additional information here. If you do use a line conditioner on an amp look for a claim of NO CURRENT or WATTAGE LIMITS so that it is likely a straight-through design with any filtering elements ACROSS the line. If it does have a wattage or current rating then it would indicate some sort of treatment in SERIES with the line which is almost never good for amplifiers and may even choke off lower power gear like front ends depending on the design of the condiFoner. Some conditioners claim to have no current limiting. I suspect a claim of no current restriction is a relative claim meaning the current restriction is very small and it could be true for our purposes. Let me say here to use your ears in all cases. I do recommend conditioning for all front-end equipment. For front ends which tend to draw little power compared to the amp, you might pursue a well-designed conditioner but be observant about power limiting. (consider all of the foregoing as theory, and in the end use your ears).

If you must run only one wall power line, plug the amp direct into the wall and then the front end into your line conditioner. It is infinitely better to install at least two lines (which must be on the same phase) because the amplifier will modulate the wall power fluctuating by the demands of the music and actually make noise on an otherwise quiet wall power line. Plug the amp directly into one and a line conditioner into the other which you will then plug your front end into. We have experience with the Audioquest 5000. I found the “high current” outlet best for DACs. I have less experience connecting things like servers to the dielectric outlets 5 and 6.

Our experience is to connect the DAC into the “high current” outlets intended for amplifiers. It gave us the best balance between transient current delivery and noise filtering. For the DAC, the other outlets seemed to restrict the current too much making the sound lackluster. Again, I often prefer the amps straight in the wall and do try line conditioners on the front end judging by ear. In any case, try to audition line conditioners before you buy. The AQ Niagaras are not the only choice; we have not tried others so do compare other brands.

Conditioning is probably most important for DACs and digital sources because of the digital clocks running at “insane” accuracies. Bad power can easily make the DAC clock jitter worse in addition to adding ground noise to the analog circuitry and signal cable connections.

240 Volt Connections in a 120 Volt Country

Some high-end amps can be switched over and run on 240 volts and I recommend it. If you absolutely cannot change your wall wiring to a heavier gauge, and your amp allows it, you may be able to use the existing wiring to convert to 240 volt then using a 240 volt outlet and a 240 volt plug on your power cord. Ask your electrician if you can convert an existing 120 volt line to 240 volt. Be sure the amp has the feature that enables it to be changed to 240 volts and be sure the changeover is done correctly. Transformer primaries and the transformer core will run slightly more efficiently yielding lower impedance so the internal DC amp supply might appear slightly “stiffer” to the amp’s audio circuits (always a good thing). Because the amp is now running at twice the voltage, the amperage (current) the wall wiring “looks” twice as thick to the amp as it does at 120 volt (ohms law). Now the amp will make even less audio noise on the line and it then rejects its own line noise better as per the goal of this paper. The 240 volt outlet can be a standard 15 amp (240 volt rated), with 10 ga. wire up to 60 feet then 8 gauge beyond that.

For the 240-volt lines, the electrician may, or may not know about a NEMA receptacle and plug number that is the same size and form as our common Edison duplex 120 volt receptacle but the wide blade of the plug is on the opposite side as the 120 volt duplex. Hubble or commercial Leviton works fine for 240 volt, and the 6-20 series looks less industrial in your home.

It is NEMA plug number 6-20P www.stayonline.com/detail.aspx?ID=6756 and NEMA receptacle number 6-20R or 6-15/20R http://www.stayonline.com/detail.aspx?ID=6756

Here is the NEMA chart. www.stayonline.com/reference-nema-straight-blade.aspx

“Stayonline” is a good source ( www.stayonline.com/default.aspx)but your electrician may like a local supplier.

*REMEMBER: BE SURE TO CHANGE OVER THE AMP INTERNALLY IF YOU DECIDE TO RUN 220 VOLT !*

Twisting the Wire

Twisting the Romex or individual wires almost completely rejects high frequency noise which consists of induced electrical fields from radio/TV stations, cell towers, microwave relay antennas, etc. Nearby low frequency noise from nearby AC wiring is also rejected. Conversely, the wiring being twisted will not emit a field or induce nearby wiring. Ask the electrician to twist the flat Romex cable one twist every 6-12 inches or so. He can do this by stretching out the length of the intended run and tying each end to something like a broom handle. Pull the wire taught and twist. I have done this using a big drill with a 1/2” chuck. The twist will be uneven which is a good thing attenuating a wider band of noise. Yes, it is tedious but critical to do so. If the lines are running in parallel each line should be alternately twisted relative to the one next to it and ideally the twists of two lines should be at different intervals meaning twist one line at 6 or 7 inches the another at 10-12 inches and another at 8-9 inches, etc . This prevents any coherent coupling between them.

Keep the lines away from each other by minimum 4 inches. It is perfectly OK to cross them at a right angle.

If you are running individual wires in conduit then twist the black (hot) and the neutral for each circuit. Multiple circuits can be run in the same conduit. I prefer PVC conduit if your local code allows so there is no inductive coupling to the (steel) conduit. Do not include the mechanical ground in the twist. Just run a lighter gauge insulated green ground wire for each circuit laid loose in the conduit.

In a recent untended test between twisted vs non-twisted wiring, two customers used the same electrician. The electrician was asked to twist the wire as described. It was discovered he did not do that step. He was required to come back and do it over, twisting the wire as described. Independently, both customers could not believe the difference in the sound. With TV/radio stations, microwave transmission towers and now 5g cell towers everywhere, these fields affect our equipment more now than we know. The most susceptible may be the highly accurate clocks in all DACs but these fields do affect all equipment, their signal paths and signal grounds.

Communicating with Your Electrician

If your electrician has any concerns about all of this, be aware he is always planning for CONTINUOUS current draw and rates everything and splits up the loads between each of the house phases, like the air conditioning, the electric dryer and electric hot water tank for the available amperage. Please explain to him that we are designing for incredibly short peak current pulses and we need the resistance from the wall outlets back to the utility as low as possible for the best amplifier and overall system performance. The continuous draw might be 4 -10 amps and is negligible from the electrician’s standpoint. Larger wire does not violate any codes in the US as far as we know but you and your electrician are responsible to be sure this is true in your state, county, and city. In summary, we ask the electrician to understand we want the lowest possible resistance between the wall outlets and the breaker panel. It is all about incredibly short-term current demands that are typical of music. These very high peaks can’t be measured with a conventional ammeter. Also, these very high, very short-term current peaks can be well above the rating of a 20-amp breaker. These peaks are so fast the breaker does not even see them and therefore they do not trip the breaker.

Summary

Here is the way to think about the goals and descriptions of this paper: The goal and ongoing process in our hobby is to find the choke points in our systems and upgrade them. The electrical power should be the first consideration in our systems and hence it is most often the first “choke point”. The system can never sound better than the worst power it is fed. There are two considerations: (1) increasing the wire size to improve the current starving and power modulation (noise), caused by the amplifier itself as previously described. (2) The noise coming from your utility and the surrounding airborne RF noise needs to be suppressed by twisting the wiring as previously described. Well-designed power conditioners are almost always a benefit.

Again, use your ears to decide.

Power conditioners fix some of the noise created by the previously described “amp starving” but it is better not to starve the amp and create the noise in the first place because amp starving often creates worse noise on the system that is measurably and audibly worse than anything that might be sent by your utility.

In many cases depending on the oxidation of your existing wall power connections, age of the breakers, but ESPECIALLY the length and gauge of the wall wiring, the above wall power changes in your home system are often a bigger improvement than any component that you can buy, especially with big solid-state amps that have long runs back to the breaker panel. But even people with tubes report improvements if not huge improvements.

Without exception over the last 20+ years, comments from those that have done the above heavy gauge wire wall power mods say there is audible improvement in dynamics while making the sound even more detailed, yet much more relaxed with dark backgrounds leaving only the notes and music. I was very surprised the first time I did this house power mod. I expected the bass and dynamics to improve which they did. I did not expect the mid-range and the highs to clean up and become far more coherent as much as they did. The improvements were at least as much as a “great” new component. The improved clarity is always surprising and obviously well worth the effort.

If you are new to this, it is perhaps controversial and involves detailed logic, so we invite readers to consider our methodology and logic and decide for themselves whether it suits their purposes. We aren’t making a moral argument, so readers could certainly want something else. If you do, at least you know we aren’t aiming to provide it. And that we have professional, logical reasons for not doing so.

We should say that we have learned that commonly assumed shortcuts to desired audio results often don’t work very well. What is often assumed to work, doesn’t work. What does work requires some effort and thinking. We think that is a feature not a bug, but not everyone will agree.

1. What is the point of even having a careful, considered approach to audio performance for music?

a. Musical Enjoyment. It is worth reminding ourselves and readers that a main part of the goal here is musical enjoyment. We love music well performed, and assume our audience does too, so we are always searching for enhancements to that experience. We often say “your sensitivities may differ from ours, so we offer insights into what equipment and software and set-up procedures do, but you have to conclude how you value those.”

b. A Quest. One of the great things about the audio-for-music hobby is that it can be conducted as a quest. To support this, we set forth a goal of “believable musical performance”. Such a goal is one that, arguably, cannot be fully achieved, which makes audio-for-music an almost ideal quest. Audio-for-music is also ideal as a quest because it is logically and scientifically complex, which makes it potentially rewarding over decades. A nice feature of the “Quest” element is that it can work for everyone. The Quest can be conducted at almost any level of expense, though perhaps not any level of effort. The Quest is intended to be a universal model of the musical life for all music-lovers who are interested in the search for the truth of the artist’s intent or inspiration.

c. Appreciation of Progress. Our experience doing this over more than 50 years is that great progress has been made toward believable musical performance at home. We, and those who join us, enjoy following the development of science and engineering and art in service of the core goal. The people involved in this progress are often interesting and their thinking is interesting and their effort is admirable. We examine all kinds of products and procedures and technologies and histories and personalities as part of this. We are hopeful that the sometimes high price of innovations is simply the beginning of a technological wave that cascades down or up the price ladder.

2. What Is the purpose of reviewing audio equipment?

Basically, our experience is that better audio equipment can lead to great musical enjoyment. This is actually not as obvious as this statement might seem to imply. For some listeners, good musical satisfaction can happen with, say, an iPad. For other listeners, the narrow bandwidth and compression of an iPad or a Bluetooth speaker are pretty much unlistenable. We would add that the general tendency to find a correlation between musical engagement and audio quality is rather dependent on whether you listen to music as a primary activity, or whether music is a background experience. We are reviewing audio equipment with quality-sensitive listeners in mind who view music as a foreground experience. This isn’t a moral judgement, it just specifies our intended audience.

The problem our intended consumers face is that there are many choices in audio equipment and there are many combinations of different types of gear. Our reviews are intended to help music lovers find equipment likely to perform well and suitable for their needs (living environment, budget, musical tastes).

In this effort, we expect listeners to participate. We try to describe how equipment affects musical playback. Consumers have to bring or be willing to investigate what their needs are.

This idea of “listener participation” isn’t some demand we make. It is the natural outgrowth of the “Quest” element of our (actually our target audience’s) philosophy. We are exploring “what works” to generate musical enjoyment. We do that as part of a conversation or stimulus for the search for “what works” that our readers and viewers are doing.

As you will see below, our review methodology is specifically geared to audience participation. We think our reviews are useful to people who “just want to know what to buy”, but that is not their primary design goal.

Such a quest has an important ongoing character to it. By this we mean that the quest to discover “what works” will be an ongoing effort. As technology changes and as industry and listener learning occurs, we and listeners will develop an improved model of “what works”. So, what we thought worked and what we observed as “believable” at one point is certain to change over time. Our readers and viewers should expect (and enjoy) this. The listener who must have incontrovertible and unchanging information is playing a different game; one we can’t support and which we believe probably isn’t possible.

This logic goes with the assumption that listeners are actually interested in musical enjoyment, not in pursuing tangential projects related to audio but not really to musical enjoyment. Examples of these tangential projects include:

• Audiophiles aiming to prove that they are superior people because they have a “better stereo”. To do this, one perhaps needs endorsements or measurements or comparative technological analysis. Since we aren’t focused on the relative status goal, we don’t deliver much supporting material.
• Audiophiles aiming to politicize audio. Philosopher Agnes Callard defines “politicization” as interactions where “instead of focusing on the merits of an argument or idea, the conversation becomes about which “side” someone is on, reducing potentially productive discussion to a battle for dominance or esteem between opposing parties”.We specifically do not support politicization efforts as we think they get in the way of the quest, properly understood.
• Audiophiles interested in “virtue signaling” by finding alleged technical “errors” or “scams” or “reprehensible behavior” and making that the focus of discussion rather than the search for musically meaningful audio. The data is rarely if ever available to judge these issues, and we have neither the skills or the time to do so.

So, we aren’t trying to support such projects. We have reason to believe such issues are a distraction from the mission we believe listeners are pursuing. Regardless, supporting such tangential interests is a distraction from sound-quality oriented reviewing, so we elect not to do it. Rather, we want to focus on helping listeners form a short list of gear to check out. We then encourage listeners to go experience these items at dealers or shows.

Along these lines of tailoring our methodology to the differences between listeners, we don’t know the budget of each listener. So, we review gear at many price points. Some of this is expensive, but that works for some people. And it helps to know what is possible at the state of the art in order to decide how much of your budget to allocate to audio equipment. If you don’t know “how high is high?” it is hard to know where to fly. However, we do not suppose that there is some price level that is necessary for musical enjoyment. Ideally, that might be a very low price, but so far the technology doesn’t work that way for everyone. But certainly, you don’t need to spend $1 million or $100,000 or $10,000 to have a satisfying experience. Sometimes if you spend more, you will have a better experience, but sometimes you won’t. We review equipment to up your odds, if you decide to try new things at whatever price point works for you.

3. How do you deal with the differences between listeners and their tastes?

We start with two related ideas.

First, we assume that the artist’s intent for the music will deliver the best results. That is, we assume that the consumer/listener is not the artist and can’t improve on what the artist wanted.  We’ll talk later about how the artist’s intent is unknowable, generally, so the logical outcome of this point is that we want to have audio equipment that lacks distortion, because we don’t want to modify what the artist did because that will make the result worse. A debatable point, but not obviously wrong as a logical starting point.

Second, we observe that all audio equipment has distortions. Now we don’t want distortions per the preceding point, but we have them in the real world. So we start with the idea that consumers pretty much want the same thing (to hear the music as the artist intended) but that there are enough distortions in the recording and reproduction process that two reasonable people could prefer different distortion profiles.

We think these different preferences for distortion profiles are partly a function of musical preferences (say musical styles listened to) and partly a function of usage (e.g. apartment living room vs suburban listening room) and partly a function of genetic and learned musical sensitivities (e.g. to transient accuracy or tonal composition).

So, we, again, aim to describe what the equipment does sonically, but evaluating this requires the reader/viewer to know what their musical needs and usage and sensitivities are. To make this slightly less complex (there are many sonic variables we describe) we try to characterize where equipment fits in a simple segmentation scheme.

The words here are approximate summaries, but over time if you correlate our descriptions of sonic details and this matrix you may find it helps you understand how the details form a larger picture given components.

4. Why do you do subjective reviews?

We don’t. Or for the most part we try not to make that the core of our reviewing. We aim to do observational, objective reviews. Now, there is some confusion about terminology in which “quantification” is “objective” whereas human “observation” is “subjective”. But this is wrong. That notion incorrectly glosses over a critical distinction. “Subjective” in the dictionary means human reactions that primarily involve feelings. But humans are also capable of observing objectively. Losing sight of that second source of objectivity, the use of careful observations, comes at a price, as scientists generally know, but regular conversation seems to forget. It is wise not to let sloppy conceptual categories direct your thinking.

A simple example may help make some sense of this important distinction. If your car is parked next to your house and we ask “which is taller?” you will observe that your house is taller than your car. It isn’t that you feel your house is taller, it is that you are fully capable of objectively observing the height differences. Humans can do this with facial recognition, color identification, bird calls, food smells, voice association and many, many other perceptual tasks.

Professional audio reviewers train to make these observations about audio equipment and report them. Professional reviewers compare notes and very frequently find listening notes that match almost perfectly. But we observe that most new listeners, as well as more experienced consumers, can make these observations too. The training of reviewers tends to involve knowing the signals that may trigger different observable behaviors and being able to use a standardized lexicon to communicate these findings.

The scientific method generally is said to start with observation. Scientists do observations of, say, planetary motion or the proverbial apple falling on Newton’s head, and then form hypotheses, which are tested in various ways. In physics, where scientists are seeking universal laws that can be used for prediction, often the goal is to find a mathematical formula (like F=MA) that applies in many circumstances. This is then verified (or not) with measurements that necessarily must be quantified. The physics model of mathematical formulas and the resulting quantification of things like spacecraft flight is a wonderful development where it can be done (low complexity phenomena). It also seems to lead to much confusion about what people are trying to do outside of physics. Attempting this is scientism, not science.

An extreme example is what we are trying to do when reviewing audio equipment. We are trying to observe the macro behavior of speakers and amplifiers and DACs. You can then form hypotheses about the sound and gather data about the performance of such devices that seem to match you needs. When you get a good match, you buy the product and you’re done. We are not trying to develop a mathematical description of the McIntosh 2800 or the KEF Blade. We and you do not have a mathematical hypothesis that needs quantified data to verify its predictive power.

Engineers also are practiced observers. Many measurements currently in use came to be invented or deployed after observing a phenomenon and then figuring out what measurements might shed light on the phenomenon. Note also, that the existence of a measurement that sheds light on a phenomenon is not the same as saying the measurement completely characterizes the phenomenon. So, measurement is often partial, although helpful to engineers. And so, observational listening is a frequent and often the final step in design work.

Now, some may feel that observations are “subjective” in the sense that human observations are partial representations of reality because the human perceptual apparatus is necessarily selective. But, of course, all measurements are like this too, whether conducted by humans or conducted by test equipment (which is why we need so many metrics and even then we don’t know everything). The other idea that seems to motivate the sense that human observations are subjective is that human observations have distortions (e.g. memory). When we are trying to objectively observe (i.e. not feel or judge merit) we are using perceptual apparatus that indeed has distortions, though some of those distortions are common across people. I don’t see ultraviolet light and neither do you.  The distortions are plausibly relevant to how the observed phenomena (i.e. music) will be perceived by most people. And, in any event, test equipment has distortions too, though these are often more conceptual than executional (Q: “What is the measurement of soundstage width?”  — A: “Oh, we don’t have one and we don’t know what measurements add up to that variable”).

5. Why not just measure equipment?

The biggest issue is that the measurement suite (the list of required measurements) needed to characterize equipment is large and complex. In simple terms, one needs to do a lot of measurements, and, very problematically, those measurements are hard to add together into a mental profile of what a piece of gear sounds like if you are a consumer (our audience). What you gain in apparent precision, you lose in meaning (for consumers). Consider for example: frequency response, polar response, power response, time domain response vs frequency, interference, diffraction, harmonic distortion with frequency, intermodulation distortion with frequency, impulse distortion, dynamic compression, sensitivity, impedance with frequency, changes in each variable with level and temperature, etc.

Those are just examples from the speaker world. Amplifiers and DACs are just as difficult. To be clear, by “difficult” we mean difficult to comprehend and difficult to add up into a musically meaningful sonic picture. Most consumers don’t know what half or more of the measurements are measuring or what sonic phenomena they point to. And even if they did know (and surely some do), the individual measurements and their many data points are measuring very specific, narrow technical phenomena that are difficult even for experts to connect with musical results. Note that this “measuring narrow phenomena” element is handy for the engineer who has to solve discovered problems by tracing them to the source. But it doesn’t work for consumers.

That doubt about any consumer’s ability to integrate a full audio measurement suite into a useful understanding of how equipment will perform on music may seem insulting. It may seem as if we doubt your ability to do the integration. This is not the case. What we doubt is that anyone can do it (and most audio engineers would agree, which is why they insist on listening as the final frontier of knowledge). What is behind this is something known in the philosophy of science as “the knowledge problem”. There are many predictive tasks where the knowledge base needed to do passable prediction is simply not available. An example of an unpredictable task that falls under the knowledge problem is a simple coin flip. It is not the case that coin flips are random. Most scientists would agree that coin flips are completely deterministic: they follow the rules of Newtonian physics. The problem is that we don’t know all the Newtonian initial conditions and inputs, so we call the phenomenon “random” when we mean “practically unpredictable”. See this article for more on the knowledge problem, a constant issue in many aspects of life.

A common attempted solution for consumers, then, is to reduce the measurement suite used to a few popular items (e.g. on-axis frequency response or THD). This simply gives a vastly incomplete picture (that is still difficult to connect to the sonic experience except by poor rules of thumb). History shows that this approach often leads to disappointing results for consumers.

Note that in addition to the difficulty of integrating a vast data set to predict sonic results, measurement also has the problem of reference standards. If we have a measurement result from a piece of equipment, what is the goal or reference to which we should compare our result to establish its goodness or badness? As a simple example, is flat on-axis frequency response the goal? Under what measurement conditions, for example what mic distance? If not, what is the reference goal? And how should power response data look? Is a -1db monotonic rolloff desirable? Why? From what frequency? If, instead, the power response rolls off at 1.2 db/octave, how does this affect what we want on axis? We could go on for these two measurements and then continue in this vein for another 30 or so metrics. Measurement results are extremely complex and standards are few and debatable, thus audio engineers spend years mastering measurement, creating some of their own proprietary standards along the way. It, again, seems impractical for consumers to come even close to the required knowledge to render this meaningful. And even if consumers were willing to do the significant coursework needed, there remains the reality that measurements need to be connected to sonic results, generally by…objective observation.

Note also, perhaps ironically, that the reference standard for measurements is often established by…observation (a.k.a. listening to music and then characterizing preferences quantitatively). The observations used to set references are often old and/or developed under debatable conditions (“average listeners” and “typical equipment” and “current recordings”). This isn’t to say that all references are meaningless, but that they require deep understanding for interpretation.

We also must mention the six major issues with audio believability that TAS has identified. These generally aren’t addressed in the standard model of measurement (science is often confined to phenomena that can be modeled with manageable mathematics). Two of these major issues of audio are largely outside the consumer audio equipment realm (the problem of visual images and the problem of recording standards). But the other four (the problem of spatial imaging, the problem of bass in real rooms, the problem of dynamics and the problem of digital distortions) are in the audio equipment wheelhouse but lack quantitative measurement standards. The scope of quantified audio measurement is simply narrower than the scope of observational capability. (It is also the case that measurements can detect phenomena we may not be able to observe; -150 db s/n is measurably different from -145 s/n, but is that observable? – and note that if it isn’t observable we can ask if it is meaningful whereas the inverse is not true).

If we contrast this with objective observations, communicated via words or diagrams, we gain several advantages. We gain quite a bit of simplicity because the characterization of musical audio involves 8-10 key concepts (frequency balance, octave-to-octave output variations, micro and macro dynamics, soundstage, sound space, harmonic and a-musical distortions). While these may benefit from some study, they are relatively intuitive as concepts. They are relatively intuitive because humans have extensive hearing practice (due to a life in which hearing is a survival skill) and humans have extensive vocabularies used day to day for describing experiences. Humans come into audio with some expertise and they can easily learn to expand or sharpen their vocabularies. This is not so simple with reading impulse response measurements or phase diagrams.

Adding to the simplicity in the objective observational approach, we note that sound quality observation concepts tend to be mostly common across all types of equipment. In contrast, amps and DACs and speakers tend to have different measurement parameters, leading to perhaps 100 or more measured parameters characterizing a full system.

With objective observation, we gain the advantage that our observations are conveyed with musically meaningful terms much like those people naturally use to describe that they hear. This, we believe, greatly aids understanding for a greater number of consumers. Consumers can also verify our observations by listening themselves, something they cannot generally do with comprehensive measurements.

And, with objective observation, we can test for all musically relevant phenomena, not just those we happen to be able to measure at the current state of the measurement arts. This allows us to address real but not fully characterized phenomena, for example our “major problems of audio believability”. Since these are extremely important to consumer satisfaction, a methodology that includes them has significant advantages over one that doesn’t.

6. But I like numbers and they seem more “solid”, why do you hate them?

We like numbers too. Some of our reviewers are engineers and some are scientists. And we use measurements frequently for setup purposes and other narrow applications. We’re simply pointing out that the subjective feeling that numbers and measurements are more reliable comes at a price.

Limited Information

One price is limited information delivery. Or, as we said above, with measurement “precision comes at the expense of almost all meaning” in audio for consumers. And meaning is what we must have or we have nothing.

Broader Understanding

With objective observation, we gain the advantage that our observations are conveyed with musically meaningful terms much like those people naturally use to describe that they hear. This, we believe, greatly aids understanding for a greater number of consumers.

Inviting More Audiophiles Into The Investigation

Consumers can also verify our observations by listening themselves, something they cannot generally do with comprehensive measurements. If this is a quest, we want all listeners to be able to participate. Measurements, for most listeners, are disempowering. We acknowledge that processing observational data takes somewhat more effort, and if you just want a simple way to decide what to buy, our observations may not provide it. But if meaningless ease of decision-making is the goal, you can throw darts.

Eyes On The Prize

We do note that one preference for measurements may be to “settle the issue” by focusing on numbers that attempt to resolve the psychological uncertainty of what equipment is “good”. Since we are trying to serve listeners who want a good musical experience, not listeners who want psychological quiescence about purchase decisions, we aren’t interested in measurement for the latter purpose. Related to use, measurements may be useful to fuel tribal battles online. Understood, but that’s not what we’re here for.

All Musical Phenomena

And, with objective observation, we can test for all musically relevant phenomena, not just those we happen to be able to measure at the current state of the measurement arts. This allows us to address real but not fully characterized phenomena, for example our “major problems of audio believability”. Since these are extremely important to consumer satisfaction, a methodology that includes them has significant advantages over one that doesn’t.

Entertainment

Some of the interest in measurements and technical description may come from the entertainment value of these things. While we are aiming at providing useful information for consumers trying to improve their audio experience, we intuit that some people want reviews and articles and videos largely as entertainment. We’re working on understanding whether we could occasionally serve that need and how we could do it without interrupting the primary mission. Properly done measurements have the difficulties we’ve described and doing them properly is very expensive. But if such information were for entertainment purposes alone, it might be simplified and become economically feasible if potentially misleading.

We add that the deck isn’t completely stacked in favor of objective observation. The advantages of using objective observation skip over an important issue with the application of words to describe sound quality: words are harder to stack rank when comparing two products. If two speakers both have “tight mid-bass”, it is a little hard to know if speaker A has tighter or less tight bass than speaker B. So, we’ve gained meaning but lost precision. It is frustrating, and we think The Absolute Sound needs to do some work advancing this art. However, you can go to hear the products under consideration and evaluate them for yourself (using TAS reviews to get to a short list).

It helps with regard to comfort with observation to understand that your powers of observation are excellent, an understanding which we think is enhanced once you drop the idea that quantified measurements would be better for evaluating sound quality by definition. Once you start to view measurements as borderline meaningless (for you, though valuable to engineers) and start to view objective observation as an audio superpower that you have, and can further develop, this process becomes more attractive. We find that many consumers assume that they can’t hear differences in sound quality and thus assume that quantitative measures will solve the knowledge problem. We think both assumptions are incorrect. You can hear differences and measurements won’t add meaningfully to your insight.

7. You talk about objective observation as a superpower, but how is your opinion objective?

To use observation as a meaningful measurement technique, you must have a reference standard. This is the case with quantified measurements too, just as it is the case with objective observation. Comparing what we (or you) hear to a reference gets us out of the realm of opinion (subjective feelings). As we said above, it isn’t your opinion that your house is taller than your car. It is an observable fact. It is an observable fact whether a guitar sounds like a guitar, and if it doesn’t, to what degree and in what way.

Reference: the absolute sound

In music audio, we use the sound of real music and real musical instruments (“the absolute sound”) as the reference standard. An audio system that can reproduce a guitar or a singer or a jazz band or a symphony so that it sounds believably real, will tend to be more satisfying for most listeners most of the time. This latter point is our experience from over 50 years of listening to live music and audio reproduction of music across hundreds of reviewers.

Now, it is true that different violins or different Martin D-18s will sound somewhat different, and recording practices will affect this. We address this issue in two ways. First, and a critical point, is that the distortions we are covering in audio gear are vastly larger than the differences between instruments, generally speaking. And, second, we don’t rely on a single data point of a violin recording or a guitar recording; rather, we use hundreds of tracks to find the patterns of distortion that we report on. We find that tiny differences are usually not issues of believability. Large errors make it obvious that the virtual sound is virtual or artificial. You can do this too, if you know what instruments sound like and how ensembles present themselves on a stage.

The alternatives

To be clear, “it sounds good” or “I like it” are pretty much meaningless subjective points to you. Unless you know what my reference is. The existence of a reference shifts observations from subjective to objective, at least if care is exercised. But if my reference is personal, it is still hard for you to use and disempowering to you. “It sounds good” is not a very useful observation.

Note, again, that with quantified measurements we have to go through two steps to develop a reference: first, coming up with a quantified standard to compare each measurement to and, second, compare the proposed standard to observed musical results to calibrate it to something meaningful. This critical second observational step is often overlooked as a point of error by consumers when thinking of the quantified measurement approach. And sometimes the basis for establishing these references isn’t as clearly meaningful to music as you might think. Similarly, you might imagine that establishing these references may involve either the limitations of the state of science at the time of standard setting, or opinions of the humans setting standards, or practical engineering limitations. In contrast with this problem, music as a reference is pretty close to automatically relevant to music listening. At least if your goal is a believable portrayal of the real thing.

You may not agree with the objective observational approach, but we invite you to try it. And if you don’t agree, you can still use our methodology by understanding what distortions you prefer and comparing equipment sound quality to your reference distortion profile. And you can read our reviews and look for products that fit with your profile (though if your profile is quite far from believably real, we may pass over equipment that you would prefer as we search for gear to review).

8. Why do you have to talk about sound with so many gobbledygook words?

If you assume that there is no methodology used in generating our descriptions of audio phenomena, then perhaps the words look like gobbledygook. But if you consider that this is an objective and carefully executed methodology, then words are useful, relatively simple and probably more powerful than the alternatives.

In addition, if you can learn the hardware parts of a stereo system, you can learn the descriptors of sound quality. The number of basic terms isn’t that vast. And the observational terminology is pales in comparison with the number of concepts needed for proper measurement.

We’ve tried to make the terms relatively simple, but it may help to do a little study of the terms we use for reviews. With this in mind, we have prepared a Glossary of Sound Quality terminology that is within our Audiopedia.

Now, as is common in English, there are plenty of synonyms. Since these are fairly standard words, once you have the basics down, it isn’t hard to understand the synonyms. And, as we pointed out above, the number of basic concepts is much smaller and easier to grasp than with the vast array of measurements.

That said, learning all of this is easier when you look up a few terms and then listen to some music and then repeat.  It can help to have a musical instrument to experiment with as well (piano or guitar for example). And attending concerts can be informative too. You will hear the phenomena we describe and then be able to attach the words to them to give the words richer meaning.

It also helps to spend some time listening to different equipment, noting what you hear and comparing this with review descriptions. Words describing phenomena that you’ve never experienced are tough to understand and warm up to.

9. Why Do You Use Old People As Reviewers When They Can’t Hear Most Of The Musical Spectrum?

There are frequent questions about how The Absolute Sound (and other publications) can use older listeners to review equipment (our reviewers typically range in age from mid-20s to mid-70s). The logic, for those open to logic, is easy to understand if not intuitive (for those in a hurry skip to items 1 and 6 in the list below, but to address common misconceptions about the need for “perfect hearing” be sure to read the whole list):

1. Everyone Has Hearing “Loss”. So-called age-related hearing loss is the proposed issue. Some roll off in high frequency hearing typically is noticeable in humans at the age of 20. The amount of high-frequency sensitivity loss generally increases with age, but varies person by person. Some 40 year olds have more roll-off than some 60 year olds.

2. Loss is Really Roll-Off. The effect of age-related hearing “loss” is actually a reduction in hearing sensitivity at higher frequencies. It is a roll off, not a brick wall filter.

3. Main Impact At Very High Frequencies. Typically, the largest effect is at frequencies above 8-10 kHz. Because the effect is a rolloff of sensitivities, not a brick wall filter.

4. Tones Are Not Everything. It is commonly said that hearing extends from 20 Hz to 20 kHz. This range is based on the use of sine wave tones. But recent (post-2000) studies have revealed that the ear/brain is also a discriminator of time, and that timing discrimination extends up above 100 kHz. Assuming similar age-related sensitivity loss, one could expect older listeners to hear time-based signals above 40 kHz or higher.

5. Less Than 0.1% of Music Affected. Musical energy is not linear with frequency, it is logarithmic. Because of this, as measured and reported in AES papers, less than 0.1% of musical energy is at frequencies above 8 kHz. Note that the highest fundamental note of the piano is 4180 Hz, violin is 3520 Hz, piccolo is also 4180 Hz although technique can raise that a bit. 2nd harmonics will thus generally be below 8 kHz. For further reference, middle C on a piano is 262 Hz.

6. Relative Measurement. A core methodology of The Absolute Sound is to compare the sound of stereo equipment to the sound of real music in real spaces. When listeners are trained in this approach, they are comparing two inputs, each subject to any rolloff in individual frequency sensitivity. The results, therefore, should be relevant to listeners of different hearing frequency sensitivities. Example: If listener A, age 40, listens to a violin, and the top note (A7, 3520 Hz) is played at 90 db, the listener might hear it at 70 db. If the same violin sound is recorded perfectly and played on a perfect speaker, this listener will hear A7 at 70 db. He or she will be able to say that the speaker sounds “like a real violin”. Now, if listener B, age 60, listens to this violin, the A7 may be heard at 50 db. And the perfect recording/speaker will reproduce it at…50 db. Listener B will say “this speaker sounds just like a real violin”. Just as listener A hears it.

7. Ear is Not a Microphone. Thus, a logical mistake is to assume that reviewers are using the ear like a calibrated microphone where the voltage from the mic must be the same for all frequency inputs (an absolute reference). But that isn’t what we are doing. We are comparing device sound to a known reference sound so that observations can apply across listeners. Listeners with different sensitivities will hear the same relative response between the equipment and real instruments.

8. Study Required. To be good at doing this, reviewers must study the sound of real instruments in real space. Basically, this means attending many concerts and doing this regularly. Our reviewers do this and pay attention to learning the sounds of real instruments. Some also play instruments and some keep several instruments on hand to check and update references. This is critical and either not a function of age or a skill that improves with age.

9. Audio Knowledge Required. In addition, reviewers must be able to attend to the many details of audio distortion. This requires study of and familiarity with the sound of resonances, harmonic distortion, intermodulation distortion, filter error, pre-ringing, soundstage stability, timing coherence, octave-to-octave frequency errors, balance errors, roll offs and more. Again, it takes time to learn to observe all of these phenomena. We have frequently experienced our younger reviewers (ages 29-35) spend hours trying to identify some quality that reviewers with decades of experience can identify in less than a minute with no prior input.

10. Ear and Brain Involved. It helps to understand that hearing is not simply a mechanical phenomenon, it is a combination of ear and brain workings of enormous complexity. The role of the brain is so significant that it indicates a role for learning in the process of hearing. This learning process plays to older listeners simply because it takes time to learn. Research indicates some compensation for hearing roll-off by the brain with age.

11. Test Music Crucial. Reviewers also need to know the music that triggers various distortion phenomena in equipment. Reviewers don’t just play some music, they play test music that is revealing of errors, a library of which is built up over years. Of course, younger reviewers have these libraries too, but time and experience are helpful.

12. Everyone Can Have A Golden Ear. Note that these last few points suggest that many reviewers have hearing capabilities that exceed those of typical consumers. That is true in a sense, but this is not a genetic difference, it is the product of study and work and this work can be done by anyone. So, we like to say, everyone has hearing as a superpower, but just like athletes can develop capabilities like running and jumping and throwing, every listener can develop a “Golden Ear”.

10. Why do you insist that I play specific music that you call “the absolute sound” but which seems to be confined to acoustic music performed live? The worthiness of stereo equipment does not relate to the “absolute” nature of that being reproduced. The equipment is good if it can produce the sound for what it is, as it was intended by the creator to be.

We don’t insist on or even recommend that. What we call “the absolute sound” simply defines test signals that we have found useful in characterizing audio equipment distortions. If you’ve followed the methodology we use as outline above, we need references to assess equipment performance. The references are known musical sounds (because we can’t use unknown sounds) as test inputs. And music is used largely because it invokes ~the full ear/brain system and triggers the resulting audible phenomena that are central to “produce the sound” and of necessity to perceive the sound.

Now, very important: there is no presumption that such specific signals must be used by the consumer in his or her listening. The absolute sound is not the music you play, it is the source of the test signals we use that have known attributes that allow the signal to reveal distortions added by equipment.

Now we add the observation that very often the distortions revealed by a broadly challenging set of test signals (music) will apply to music that isn’t part of the test suite. This makes the approach potentially useful. And there is an observation that lower distortion of many types helps to get closer to a satisfying result (i.e. produce the sound for what the creators intended it to be — as far as we can know).  This makes the approach meaningful, for some.

We allow that there could be listeners for whom lower distortion is irrelevant or for whom lower distortion of certain types is not important. They can cherry pick our comments or ignore them. And we allow that listeners to music that is not well recorded may want to ignore this model or may want to cherry pick it if they think it possible to define compensating distortions (e.g. for the 1950s and 1960s standards for bass roll off). The focus of our work is on observing the distortions added by devices. You need to interpret that in light of your sensitivities and musical preferences.

You can of course “just” buy what sounds good to you. Everyone should buy what sounds good to them, we’d say, but “just” doing that is the issue we try to help with.  Many of us tried doing “just” buying what sounds good without any framework for how to proceed or additional input on what works well, and we found it to be too much of a random walk (too many unsatisfying purchases and too many trade ins). So we built The Absolute Sound.

11. If you claim to use ‘the absolute sound’ as your reference for evaluating equipment, why do you have ‘reference equipment’ as well?

Reference equipment is used by reviewers because all equipment has distortions. The more that reviewers can understand and limit the characteristic distortions of the other equipment used in a review, the more the review can focus on the performance of the equipment under test. Conversely, the more the associated equipment is unknown, the more the reviewer doesn’t know what is causing a given distortion from the system being listened to (since only systems can be listened to). Ideally reference equipment has wide bandwidth and very low distortion. This is why our reviewers of (plausibly) the lowest distortion equipment tend to have reference equipment that is of a very high standard.

12. I want proof that equipment does what I want it to do. I need to have validation of your observations by measurements and listening panels. The listening panels and measurements must be certified. I need detailed descriptions of what measurements mean in terms of sound as I would perceive it. The measurements and observations must be backed by specific engineering and scientific logic. Academic or published references must be cited for all theory.

This is not what we are trying to do. In fact, we don’t believe it is possible in the real world. A typical example of the difficulty is the one mentioned above: we know of no algorithm that can integrate a suite of measurements into a set of listening phenomena like tonal balance or sound staging or edginess or blur. In fact, we know of no algorithm that can integrate a suite of measurements into a figure of merit or figures of merit, except by oversimplification.

So we are not aiming to prove things to readers and viewers. We aim to give you a good, useful sense (but not a perfect or complete or exact or unconditional sense) of the character of the sound of equipment under typical conditions. If your usage is unusual or your conditions are unusual, our observations may not apply. And since your perceptual sensitivities may differ, we ask you to learn what those sensitivities are and apply them in judging whether our observed qualities would work well for you. And, for equipment that seems interesting, we strongly suggest that you validate its performance by listening.

Now, it may help to point out that we are aiming for a decent representation of what the engineers might have intended, not aiming for literal accuracy to the input signal (we can’t know when we have literal accuracy). We use the term ‘believability’ to differentiate from ‘accuracy’. The believability idea is also takes into account that we are looking for relatively large/meaningful distortions and accuracy seems to imply any tiny deviations are of interest. Large distortion really means significant distortion, and significant really means “upsetting the sense of a believable lifelike performance”. Measurements, which almost always reveal many deviations from “accuracy” can distract the listener into an obsession with small things that we don’t know how to address and away from solving the big issues. What seems like an advantage (“I want measurements to reveal what I can’t hear”) is to a large degree a disadvantage of distraction in a world where there are so many easily audible issues.

13. Why don’t you do double blind testing?

There are several reasons. Double blind testing to some degree presumes that the comparison of interest is between two pieces of equipment. But, as we have seen, our comparison is with the sound of real instruments and voices in real spaces. For a designer interested in comparing two prototype designs, double blind listening might be useful. In our case, writing for consumers evaluating equipment, there are so many possible comparisons that it is impractical to do them at all, much less double blind. And double-blind listening tends to drag the conversation into relative benchmarking (“speaker A has more bass than speaker B” – okay, but we still have to compare with the absolute sound, so what did we accomplish?).

Double blind testing is difficult to execute well. In particular, level matching is a requirement and is quite time consuming, which means costly. Speakers need to be, presumably, in the same position and this is quite hard without complex mechanical systems. And are we level matching at 1 kHz or level matching the integrated power response or?

Then there is the question of whether such testing should be done with A/B switching. A/B switching has arguable deficiencies that render it possibly misleading. These include memory effects, stress states that do not mirror conventional listening and errors introduced by the switching equipment.

In principle the equipment must not be visible, and, again, that is impractical to arrange in the home environment where listening evaluations occur.

There are certainly potential errors made with objective observational listening too. But adding significant time and cost to the process to generate questionable gains simply doesn’t make sense. This is a common thread we’ve touched on, but to say it bluntly: if you understand what we’ve said above, we could indeed do measurements and double-blind testing, but we’ve argued that this wouldn’t add much value, might be misleading and it would distract consumers from developing their powers of observation. It is unclear how the very large costs of doing this well would be funded in a free media world, as well.

14. Why don’t you explain circuit designs and loudspeaker structures in detail?

 As described above, we think the most effective and efficient way to understand audio equipment is with listening evaluations. Measurements add an order of magnitude to the complexity of this process, and make the process accessible to far fewer listeners. Explaining circuits and structures and materials goes another step further back in the reasoning chain, so that one must attempt to reason along an even more complex path:

Circuits/Structures/Materials>Measurements>Listening Results

As you can imagine, this is much harder than a measurements>results chain, and it leaves even more listeners on the sidelines. We want to invite more people into the audio world.

We should add that there is really no practical way to have a team of reviewers (or one reviewer) who knows as much as practicing engineers about all the circuit topologies and design details and structures and design tools and materials for all the component categories. And, then, somewhat obviously, if we imagined using practicing engineers (as if they’d have time) to do the work, we find that practicing engineers disagree.

We do, however, often mention what manufacturers claim is the differentiating element or elements of the products we review. We do this not because we think we or our viewers can reason from design features to listening results. We do it because we want to explain our motivation in listening carefully to certain products. We find that sometimes a product really sounds different and better than others in its class and to help bring readers up to speed on such discoveries it seems helpful to explain what got us interested in the first place. Sometimes this is “just” a demonstration (high initial sound quality got us to wondering is sound quality would hold up under more extensive evaluation). Sometimes this is a technical feature. Sometimes it is a measurement.

A big point for understanding our reviews, which bears repeating, is that these reasons for investigating a product are not attempts to prove by reasoning (along the chain above) how a product performs in listening evaluations. People with restricted listening experience or people with extensive design experience will often be confused about our purposes here because they are accustomed to attempting to work on the above reasoning chain. That reasoning is fun and perhaps helpful for them, but we are not here primarily to support it. We are not capable of supporting it fully. We don’t think any third party is.

Not only do our reviewers not know everything about every technology, but the designers and manufacturers often do not provide more than basic information about their designs. Sometimes the designs are patented, sometimes they are felt to be proprietary, sometimes the explanation is extremely complex and the manufacturer doubts the ability of consumers to understand so they don’t provide the information. Again, when we have such information and it seems motivating and connected to sound quality, we try to pass it on, especially if we can add value from interviews or discussions with the designers. When such information seems to be mainly “reading the internet to viewers” we skip it or do brief summaries.

We should add a final word about correlation and causation. If we attempt to describe what motivated our interest (and might motivate your interest) in a product, and then we find that the product performs well in certain ways, this is not the same as saying the feature that motivated our interest is the cause of the listening result. The world just doesn’t work that way. There are far too many simplistic reasoning chains whereby a certain type of capacitor or power supply or driver configuration leads to good or bad results. If only things were so simple.

Summary Thoughts

In the end, it seems that some of these understandable questions come from a basic lack of trust in humans and human faculties. There is a thread connecting envy with suspicion which seems to undermine trust. Paul Ricœur described an approach to interpretation championed by three of modernity’s giants: Karl Marx, Friedrich Nietzsche, and Sigmund Freud. While differing in areas of interest and philosophical positions, Ricœur pointed out that they shared a “decision to look upon the whole of consciousness primarily as ‘false consciousness.'” In short, each of these thinkers posited that our own consciousness was fooling us with illusions, and thus hiding from us our own true beliefs, motivations, and intentions. Hence, we become deeply suspicious. Here, we simply point out that such a stance is rooted in a questionable intellectual framework.

As an alternative, we hope we’ve made it clear that it is possible to design a methodology using human observation that is sensible and practical and more useful than quantitative measurement for consumers. We probably haven’t fully addressed the trust issue, but we hope we’ve made it clearer that the seemingly obvious problems are not the problems they seem to be, and the solutions are not as obvious as they appear because they are not the solutions they appear to be.

From Lao Tzu in ‘The Tao Te Ching’:

…The further one goes, the less one knows…

SOUND QUALITY (SQ) TERMS

Accuracy

A problematic concept taken literally, but sometimes meant to indicate believability. See Realism, Believability, and the absolute sound.

Bass

Lower musical frequencies, from approximately 20 Hz to 200 Hz.

Believability

When we judge audio equipment we are fundamentally asking if the rendition of sound is “believable” in the sense that real instruments in a real performance space might sound like this. Believability integrates the instrumental and vocal sounds captured, along with the effects and mixing and mastering processing applied to the recording. See also: the absolute sound.

Black Background

By ‘black background’ we generally mean the myriad elements of noise and distortion that mask or blur the rendition of small signals (e.g. reflections in a concert venue or harmonics from an instrument). So, when a speaker or DAC or amplifier is introduced into a reference system and resolution or ambience or inter-transient silence or depth of image or soundspace rendition are increased, these phenomena can fall under the ‘black background’ term.

Bright

Refers to an elevated level of treble, generally somewhere in the range between 4 kHz and 10 kHz. See Frequency Balance.

Definition

See Resolution

Dynamics

The ability of the device we are discussing to produce soft and loud sounds without distortion. Dynamics are affected by timing: when a device doesn’t respond to an input as quickly as the input requires, the transition from soft to loud or back can be delayed leading to a “heavy” or “slow” or “soft” sound. Well executing dynamic timing leads to terms like “punch”, “drive”, “quickness”, and “pace”.

Frequency

Musical instruments and voices produce sound by vibrating (strings, instrument bodies, vocal chords, oscillators, horns, drum skins, etc). These vibrations can be characterized by their frequencies. Frequencies are measured in terms of cycles per second or Hz (1 Hertz is 1 cycle per second). A “low” frequency would be for example the low string (E1) of a bass guitar, which is 41 Hz. A “middle” frequency would be A4 on a piano keyboard, which is 440 Hz. A “high” frequency would be the top note of a piccolo, C8, which is 4186 Hz. An important thing to understand when using instrumental examples is that all acoustic instruments and voices vibrate (resonate) at the fundamental frequencies mentioned above and at multiples of the fundamental. So, a piano will resonate at 440 Hz (fundamental) when A4 is played and at 880 Hz (second harmonic) and 1320 Hz (third harmonic) and 1760 Hz (fourth harmonic) and so on. So, the sound of real instruments extends well above their highest fundamental tone. See: Bass, Midrange, Treble.

Frequency Response

When discussing sound quality, we often mention a set of terms related to frequency response. Frequency response is the output level of the device (speaker, amp, DAC, etc) for each relevant frequency (generally 20 Hz to 20 kHz, but possibly higher). A signal of constant level is fed into the device at each frequency, and we measure the output. Since the input is level with frequency, we generally want the output to be level with frequency or “flat”. But there can be cases where we do not want exactly flat response, particularly with speakers where on-axis and off-axis measurements may differ and, e.g., flat on-axis response may not sound accurate.

Frequency Balance (overall)

We often characterize the frequency response of a device using the term frequency balance. When we are speaking about overall balance, we usually mean the basic shape of the frequency response curve: is it tilted up in the treble or up in the bass or scooped (depressed) in the middle or rolled off in bass and treble?

Frequency Balance (octave to octave)

We may discuss frequency balance in octave-to-octave terms. Octaves are just a doubling of frequency. So the octave above 41 Hz extends to 82 Hz, and the octave above 440 Hz extends to 880 Hz. Octave-to-octave frequency balance is a useful way to communicate if local regions of frequency response are smooth and even or bumpy or peaky. The smoother an octave and the next one are, the more instruments in that range sound right, because the fundamentals and harmonics are in balance. It helps to understand the the harmonic character of an instrument is how we know a guitar from a cello or a piano from a clarinet.

Midrange

Middle frequencies of musical instruments and voices, generally from approximately 300 Hz to 3000 Hz.

Noise Level

This term usually does not refer to audible noise in the sense of noise you hear explicitly as you would with wind noise in a car. Rather, reviewers and audiophiles estimate the very low-level noise of audio devices by observing how the device affects small signals (instrumental overtones or venue reflections). Sometimes a low noise level is referred to as a “black background”.

PRaT (Pace, Rhythm and Timing)

While it is easy to think of musical sounds in what is called the “frequency domain”, meaning in terms of the musical notes with their frequencies (e.g. A4 on piano has a fundamental of 440 Hz), we also need to think of the output of audio equipment in terms of time. Almost all audio equipment has some amount of time distortion, meaning that certain tones that should have occurred at time t=X will completely or partially occur at t=X+.01 seconds or t=X+.2 seconds. This timing error leads to observational terms like “blur” and “overhang” and “slowness” and “softness”. As a way of summarizing how well a device limits time error, some reviewers use the term PRaT (Pace, Rhythm, and Timing) to capture the accuracy or lack thereof in the time dimension.

Realism

The idea that reproduced music sounds, or does not sound, as it would when performed. A difficult concept because of the listener’s generally limited knowledge of the original performance environment, and the heavy use of studio techniques in which the music is not performed in a real space at one time. See: Believability.

Resolution

By visual analogy, resolution is the ability of a device to produce separate aural images of closely spaced objects (could be closely spaced in location or in time). A high resolution sound field has clarity, depth, and little blur. We perceive resolution as definition of sounds.

Soundstage

The reason for having stereo (2 channel) or potentially even more channels is to present the music as occurring in a 3-dimensional space. Soundstage refers to the 3-dimensional presentation of performers, primarily in the left-right and front-back elements of their positioning on a virtual stage. We often refer also to the overall dimensions of the virtual stage that is presented, articulating whether it is wide or narrow and shallow or deep.

Soundspace

This term refers to the sense that an audio device gives of the size and shape of the overall space in which the virtual performers appear. Remember that the instruments and voices resonate to make sounds. These sounds travel out from the instruments and if the performance were in a large concert hall, the reflections would occur perhaps 1 or 2 seconds after the initial sound from the instrument made its way to your ears. The time delay of the reflections is used by your ear/brain to sense the size and shape of the concert hall. If the performance were in a small club, the reflections would occur more quickly and your ear/brain would sense the smaller venue.

the absolute sound

This term refers to the sound of real instruments and voices played in a real space. The idea of the absolute sound is to create a reference or a standard for observationally objective evaluation of the sound of audio equipment. If the sound of audio equipment believably resembles the absolute sound, we judge that it is performing well. If the sound does not believably resemble the absolute sound, then we judge that it is performing poorly. Since nothing is perfect, these evaluations must necessarily comprehend the tradeoffs that all real audio equipment make. Listeners can learn what the absolute sound means by attending concerts, especially those involving acoustic or limited-amplification instruments and voices. Without such a reference communication between listeners is rendered limited if not impossible. See: Believability.

Tilt

See Frequency Balance.

Transparency

Transparency is used to indicate the sense that a device can transmit a signal faithfully. In music reproduction, this means the device gets closer to what we imagine happened at the live (studio or concert) event. Transparency is mostly a combination of resolution and naturalness. We add the naturalness criterion because at times, practically speaking, there are some artificial distortions that can seem to enhance resolution at the price of naturalness.

Treble

Upper frequencies of instruments and voices, generally from approximately 4000 Hz to 20,000 Hz or higher.

Warm

Refers to somewhat elevated mid-bass and lower midrange, generally somewhere in the range from 80 Hz to 400 Hz.

LOUDSPEAKER TERMS

The world of high-performance loudspeakers has cultivated a language all its own to describe not only the various configurations and types of speaker and drive units, but also their performance characteristics.

Active

Loudspeaker systems that contain or partner dedicated electronics – power amplification plus electronic crossovers and equalizers, some of which can be entirely in the digital domain.

Bandwidth

The range of frequencies with defined upper and lower limits over which a system operates.

Bass

Lower part of the audible frequency range. Can be subdivided into deep bass (below 40Hz), midbass (40Hz–100Hz), and upper bass (100Hz–250Hz).

Baffle

The front face of a loudspeaker. Its role is to hold the loudspeaker drivers securely, while preventing the sound emanating from the front of the loudspeaker interacting with any emanating from the rear.

Bracing

The inside of a loudspeaker cabinet can flex and resonate, adding its own colorations. Judicious and careful use of cabinet bracing can help stiffen the cabinet and reduce unwanted distortions.

Brilliance

Alternative terminology for the highest audible frequencies from 6kHz–12kHz.

Co-Axial

Literally ‘symmetrical about a common core’, as in shielded aerial cable or loudspeaker drive units (such as those made by KEF or Tannoy).

Coloration

A general term used to describe the audible effects of a whole range of different distortions in different hi-fi components, but especially record decks and loudspeakers.

Crossover

More precisely described as a dividing network, the electrical circuitry inside a loudspeaker, which apportions the drive signal to the individual drive units.

Decibel (dB)

A logarithmic unit used to express relative loudness.

Distortion

Literally any deviation from the original, though often specified to particular mechanisms. Also known as ‘nonlinearities’.

Drive Unit or Driver

The sources of acoustic output in a loudspeaker; includes woofers, tweeters, and so on.

Dynamic Drivers

Loudspeaker drivers that create compressions and rarefactions in air by means of a pistonic drive unit operating at audio frequencies. These are typically cone-shaped for drivers operating in the bass and lower midrange, and dome-shaped for upper midrange and high frequency drivers.

Dynamic Range

The ratio (dBs) between the loudest and softest sounds a system or component can handle.

Electrostatic

A principle employed in some exotic loudspeaker and headphone transducers, whereby a large sheet of thin material (typically Mylar) is induced to vibrate (at audio frequencies) across its whole area by an electrostatic charge.

Enclosure (a.k.a. Cabinet)

The rigid mounting for the loudspeaker drive units, often also containing the crossover network, and – in some active loudspeaker systems – even the amplifiers. In most cases, the term is self-explanatory (the enclosure encloses the drivers, crossover, etc.), but can also notionally be applied to the frame housing planar magnetic or electrostatic panels.

Filter

An electrical circuit used to limit the bandwidth of a signal, and one of the principle properties required of a crossover.

Frequency Range/Spectrum

Can refer to any spread of frequencies, but most commonly the Audio Band of human hearing, from 20 cycles per second (20Hz) in the extreme bass to 20,000 cycles per second (20kHz) in the highest treble.

Frequency Response

The variation in output across a specified range of different frequencies.

Harmonic

Harmonics are the whole number multiples of a base frequency called a fundamental.

Harmonic Distortion (Thd)

The addition of unwanted harmonics to a signal.

HF

High frequency (i.e., treble). Often used in terms of describing loudspeaker drive units (‘HF’ directly equating to ‘tweeter’).

Horn

As the name suggests, a design using an acoustic horn – often with a specialized compression drive unit – to increase the efficiency of the loudspeaker system. This is one of the earliest examples of loudspeaker technology, as the basic concept predates electrical loudspeaker driver design.

Hz (Hertz)

Unit of frequency of vibration, 1Hz = 1 cycle per second.

Impedance

Measure of the electrical resistance (and reactance) of a component’s inputs and outputs.

Infinite Baffle (a.k.a Sealed Box)

In theory, the sides and rear of a loudspeaker cabinet act as extensions of the front baffle in trying to keep rear-radiation from the loudspeaker drivers at bay. When the cabinet is fully sealed, preventing any rear-radiating sound in the process, it is considered an infinite baffle.

kHz

1000Hz or vibrations per second (1kHz actually corresponds to a tone nearly two octaves above middle C).

LF

Low frequency (i.e., bass). Often used in terms of describing loudspeaker drive units (‘LF’ directly equating to ‘woofer’).

Materials

Materials science has caught up with the world of loudspeakers in all three places, but especially in enclosure material (which can often be aluminum, carbon-fiber, or one of a wealth of mineral-filled resins) and drive unit materials (which can be also be made from aluminum or carbon-fiber, but also ceramic, industrial diamond, beryllium, a number of different plastics, as well as composites conjoined by lightweight foam.

Midband or Midrange

The middle range of audio frequencies, where the ear is most sensitive. Can be subdivided into lower midrange (250Hz–500Hz), midranges (500Hz–1kHz), and upper midranges (1kHz– 2kHz).

Monitor

High quality (usually standmount) loudspeaker.

Moving Coil

A transducer system that changes mechanical energy into electrical energy or vice versa, used in high quality pickup cartridges and in conventional loudspeaker drive units.

Noise

Random unwanted low-level signals.

Octave

Span of frequency or pitch that represents a doubling or halving of frequency.

Ohm

Unit of electrical impedance or resistance.

Port

In reflex loaded loudspeakers, the opening which is ‘tuned’ to the box size and main driver characteristics to improve output at low frequencies.

Presence

Alternative terminology for the high frequencies between 4kHz–6kHz.

Reflection

Higher frequencies can be very directional, and their output can easily ‘bounce’ off reflective walls and ceilings, interfering with the sound directly from the tweeter itself. Room acoustics experts recommend placing absorption at the ‘first reflection points’ either side of the loudspeaker to limit this interference.

Resonance

A physical property where one vibrating system causes another system to ‘sympathetically’ vibrate at specific frequencies. These resonances can happen inside the loudspeaker cabinet, along the walls of the cabinet.

Sensitivity

The amount of output (loudness, expressed in decibels) for a given electrical input (usually 1 watt).

Separation

The separateness of the left and right channels of a stereo audio system.

Signal-To-Noise, S/N

The difference between maximum level of a signal and the background noise left when the signal is removed.

Snake Oil

A term used by consumers to describe products that involve technological principles that are not well understood by the consumer. Examples of such technologies include EMI, decimation mathematics, image creation in the brain, bandwidth of the ear, phase effects, pre-ringing and reference measurement parameters. Snake Oil is a term of approbation which strongly implies that what is not understood is not valuable, rather than focusing value judgements on results achieved.

Stereo

Literally ‘solid’ – a system which uses two loudspeakers (or a pair of headphones) to create solid spatial sonic images.

Subsonic

Below the audible frequency range, commonly considered to be anything below 20Hz.

Top Octave

Very high frequencies in the 10kHz–20kHz region.

Treble

Upper part of the audible frequency range. Can be subdivided into lower treble (2kHz–3.5kHz), treble (3.5kHz–6kHz), and upper treble (6kHz– 10kHz). Also see Presence and Brilliance.

Transmission Line

Instead of a conventional sealed or ported enclosure, a transmission line takes the sound generated from the back of the bass speaker through a long and labyrinthine damped pathway within the speaker enclosure itself.

Tweeter

Small loudspeaker drive unit used for higher frequency (treble) sounds. Commonly a pistonic dome design but can be anything from a planar magnetic or folded ribbon of metal foil to the corona discharge of high-energy electrical plasma. As this last can produce hazardous levels of nitrogen oxides and ozone in a living room, plasma tweeters are relatively rare!

Ultrasonic

Frequencies above the notional limits of audibility, but still considered important in high-resolution audio systems. Typically, in the region from 20kHz–100kHz.

Watt

Unit of electrical power (the product of voltage and current).

Woofer

Loudspeaker drive unit that handles lower frequency (bass) sounds.

DIGITAL AUDIO TERMS

Perhaps no single category in all of high‑end audio has spawned a more convoluted ‘alphabet soup’ of technical terms and abbreviations than digital audio. Indeed, the topic has given rise to so many TLAs (three-letter acronyms) that at times it seems almost impossible to keep them straight in one’s mind. We present here a minimalist glossary that, while by no means exhaustive, covers at least a few of the more common acronyms and terms you are apt to encounter when you go shopping for digital audio components.

AAC

This acronym stands for ‘Advanced Audio Coding’, which is one of several coding standards for lossy digital audio compression (see ‘Compression’ in this glossary for more details). AAC was originally developed as the successor of MP3, which is another form of lossy compression. AAC is generally thought to deliver somewhat better sound quality than MP3 for any given bit rate.

AAC comes up often in product specifications sheets because it is the default audio format for such popular products and services as: YouTube, iPhone, iPod, iPad, iTunes, and the Sony PlayStation 3.

ADC

The acronym ADC (sometimes also shown as ‘A/D’) is shorthand for ‘Analogue-to-Digital Converter’. Realistically, not many audiophiles own, or would have any reason to own ADCs, but it is worth bearing in mind that recording studios and production houses use ADCs in order to create the digital audio music files that most of us enjoy.

ADCs receive analogue audio signals, sample those signals at very high frequencies (under the control of extremely accurate clocks) and then generate digital bit-streams (that is, multi-bit words of digital audio data) that represent the sampled analogue audio signals as accurately as possible. As with any other type of audio equipment, ADCs are not created equal, and some have audibly superior performance capabilities to others.

AIFF

This acronym stands for ‘Audio Interchange File Format’, which is a digital audio file format developed by Apple. AIFF stores audio data in uncompressed pulse-code modulation (PCM) format and is therefore lossless. Because they are both uncompressed and lossless, AIFF files require more data storage space than compressed audio files would do, but the trade off—one that many audiophiles happily embrace—is that AIFF introduces no sonically deleterious ‘compression artefacts’ of any kind.

ALAC (and ALE)

The acronym ALAC stands for ‘Apple Lossless Audio Codec’, which is sometimes alternatively called ALE (for ‘Apple Lossless Encoding’). In short, ALAC is a method for compressing digital audio data in a completely lossless manner (meaning all of the original audio data is preserved).

ALAC was initially a proprietary Apple standard, but as of 2011 Apple made the codec available as open source and royalty free software. Both iTunes and iOS devices support ALAC (whereas Apple systems and devices typically do not support other lossless standards), so that ALAC has become the de facto lossless compression standard for audiophiles who use Apple computers and/or iOS devices.

Note that AIFF and ALAC are not the same things. AIFF digital audio data is not compressed at all and therefore is inherently lossless; ALAC digital audio data is compressed but can be decoded for playback in a lossless manner. ALAC digital audio files are roughly one half the size of equivalent uncompressed files.

Bit

One unit of digital data, typically represented by voltages either above or below a clear-cut threshold and by convention held to represent a ‘1’ or a ‘0’ as used in binary numbers. Typically abbreviated as a lower-case ‘b’ – as in, “My DAC can handle PCM digital audio files at resolutions up to 32-bit/384kHz.”

Bit-rate

The speed, expressed in number of bits per second, at which digital audio data is processed or transferred from one device to another or playback. For example, one of the better sounding and more popular forms of MP3 transfers data at 320kbps (kilobits per second).

Byte

An 8-bit ‘word’ of digital data, abbreviated with a capital ‘B’ – as in, “I store my digital music library on a 2TB drive” (where 2TB means ‘2 Terabyte’). The digital word lengths used in digital audio are typically multiples of 8-bits: hence, 16-bit, 24-bit, or 32-bit words are frequently discussed.

CD

The acronym stands for ‘Compact Disc’, a physical storage format for digital audio commercially launched in the early 1980s by Philips and Sony. CDs are polycarbonate discs that incorporate a highly reflective metallic layer upon which ‘pits’ can be etched along with shiny spaces in between the pits, known as ‘lands’. The pits and lands effectively represent the ‘1s’ and ‘0s’ inherent in digital audio data.

By convention, CD standards are set forth in the so-called Red Book, which calls for the digital audio data to be stored in 16-bit words of data sampled at a rate of 44.1 kHz. When writers talk about ‘CD resolution’ digital audio files, they will often refer to them as ‘16/44.1’ files. While CDs are arguably the most popular digital audio format on the planet, other storage formats are now on the rise, many of them offering resolutions (and, in principle, sound quality) much higher than that of CDs.

“The ear is extraordinarily sensitive to timing and thus can readily differentiate between clock errors.”

Clock

Digital clocks are extremely important in digital audio, both when encoding and decoding or playing back digital audio files. Since clocks govern the precise time intervals at which digital audio files are captured, and then later played back, it is critically important for clocks to be stable and accurate so that the intervals between clock beats are maintained with extreme precision.

The human ear is remarkably sensitive to clock timing errors, so that errors occurring down at the picosecond lever are thought to be audible. The more accurate, stable, and precise a clock is, the better the sound of the component will be (all other things being equal). Some very high-end components use extremely exotic Rubidium (or ‘atomic’) clocks to achieve the ‘nth’ degree of sound quality.

Codec

A codec is a software or firmware program that can encode or decode a digital audio stream. The term ‘codec’ represents a condensation of the more cumbersome phrase ‘encoder decoder’. Some popular codecs you may have heard of include MP3, MP4, ALAC, FLAC, Ogg Vorbis, and many more.

Compression

Compression is a data manipulation process where digital audio files are condensed in order to conserve data storage space. It is useful to think of compression, as it applies to digital audio, as a two-part process. First, digital audio files are compressed to reduce them to a more compact and manageable size for storage; then, later on, the compressed files are decoded or de-compressed for playback. There are many types of audio compression algorithms, but they generally fall into two categories: lossy compression and lossless compression.

Lossy compression algorithms do the most efficient job of compressing data, but with the tradeoff that—when it comes time to decode the lossy files—only part of the original digital audio data is restored, while some is irretrievably lost (hence the name ‘lossy’). Two of the more popular lossy compression codecs are AAC and MP3.

Lossless compression algorithms are less efficient than lossy algorithms in terms of conserving storage space, but they have the benefit that—when it comes time to decode the files—fully 100% of the original digital audio data is restored. Most audiophiles perceive lossless compression to offer audible performance benefits vs. lossy compression (although there is some debate on this topic).

As broadband internet speeds continue to increase and very high-capacity storage devices have become less expensive and more commonly available (even in small, portable, handheld devices) there is less pressure on audiophiles to conserve storage space, so that over time lossless compression algorithms have become increasingly popular. Two of the more popular lossless compression codecs are ALAC and FLAC.

DAC

This acronym stands for ‘Digital-to-Analogue Converter’, with the DAC serving as an essential ingredient in any digital audio playback device. In simple terms, the job of the DAC is to receive digital audio data at extremely precisely clocked intervals and to convert that data into an analogue output that mirrors (or is proportionate to) the numerical values of the digital audio data received.

DACs can be, and often are, condensed to fit on single integrated circuit chips, with popular DAC makers including firms such as Burr-Brown, ESS, Texas Instruments, Wolfson, and many more. However, it is possible to create DACs from individual, discrete parts—an approach some audio component manufacturers have pursued in the interest of superior sound quality.

Either way, it is important to understand that the DAC devices used in a given component do not necessarily define or determine the component’s characteristic sound (other circuit elements also play a major role in determining sound quality).

DSD

The acronym stands for ‘Direct Stream Digital’, which is a digital audio encoding and decoding system developed by Philips and Sony as the format of choice for use in their higher-than-CD-resolution Super Audio CD discs (commonly called SACDs).

Unlike, PCM (pulse code modulation) formats, which store digital audio data in the form of 16-bit, 24-bit, or even 32-bit words sampled or clocked at rates ranging from 44.1 to 384 kHz, DSD is a single-bit, delta sigma modulated encoding process, but with extremely high sampling rates of 2.8224 MHz (known as DSD64) or 5.6448 MHz (known as DSD128). In principle, DSD files are extremely easy to decode for analogue playback, requiring only a basic low-pass filter. Some critics argue that DSD files have high frequency noise issues to contend with and that the delta sigma process has some inherent errors that are difficult to overcome. Proponents of DSD, however, argue the DSD achieves a smooth, free-flowing, analogue-like sound that is often difficult for PCM to achieve.

While SACD discs have never achieved the popularity of conventional Red Book CDs, their underlying DSD file format has won widespread popularity in recent years, since many music lovers now prefer listening to files downloaded or streamed from the Internet (or a local network). DSD files can be streamed or downloaded via a transfer process called ‘DoP’, which stands for ‘DSD over PCM’. This process does not convert DSD files to PCM format, but rather temporarily stores DSD data in PCM ‘data containers’ in order to simplify file transfers.

DSP

The acronym stands for ‘Digital Signal Processing’, a topic that comes up often in discussion of digital audio. One of the beauties of digital audio is the fact that, once analogue signals are converted into digital formats, they can be processed in ways that would be difficult if not impossible to achieve solely through analogue means. For example, DSP can be used to implement complex digital filtering systems that can shape the sonic character of the ultimate playback presentation in extremely subtle and potentially desirable ways. Likewise, DSP makes possible certain elaborate equalization (EQ) systems that would be very difficult to execute with a purely analogue EQ system. Finally, DSP allows designers greater control over various sonic variables including noise, transient response, resolution, etc. as well as greater control over various processing/ playback artefacts.

Dynamic Range

In audio, dynamic range is the difference between the smallest and the largest usable signal that can be passed through a transmission or playback system; this difference is expressed as a ratio and typically is quoted in dB (decibels). The human ear is said to have about 140dB of dynamic range (which is also, in rough terms, about the same dynamic range as some of today’s best microphones).

Since digital audio inherently involves creating digital representations of analogue sound waves, one question that arises is this: “Does the digital system have more or less dynamic range than the analogue signals it is attempting to represent?” All other things being equal, digital components with greater dynamic range often offer superior sound, in part because they do not lose low-level signals in noise, nor do they overload on very high-level signals.

Part of today’s emphasis on higher-than-CD-resolution digital audio files involves the fact that 24-bit files offer dramatically higher dynamic range than do the 16-bit files found in CDs.

FLAC

The acronym stands for ‘Free Lossless Audio Codec’. FLAC is one of the most popular and widely supported lossless audio codecs in use today, in part because it is an open-source, royalty-free software package, but also because FLAC readily supports metadata tagging, complete with storage of album cover art and the like.

Jitter

As mentioned under ‘Clocks’, above, timing is absolutely crucial in digital audio with particular emphasis on maintaining absolutely identical time intervals between clock pulses. Unfortunately, nothing is perfect so that small variations or errors between intervals can and do occur—errors called ‘jitter’, which will usually be quoted as worst case timing variations (for example: ‘Jitter: </= 9 picoseconds’).

As mentioned elsewhere in this glossary, the ear is extraordinarily sensitive to timing and thus can readily differentiate between clock errors, even when those errors are measured in the parts per million vs. clocks with errors measured in the parts per billion. The point is that all other things being equal, the digital playback system with the lowest jitter almost invariably sounds best.

kbps and Mbps

The former acronym stands for ‘kilobits per second’ and the latter for ‘megabits per second’; both terms are used to express data transfer speeds. ‘kbps’ figures often come up in discussion of lossy compression codecs as a means of comparing the net amount of audio data one codec can supply vs. another codec (typically, the higher the data rate, the better the lossy codec’s sonic performance will be).

You might, for example, see digital downloads offered in two types of lossy formats: ‘MP3 (CBR at 128 kbps) or MP3 (VBR at 320kbps)’—where CBR stands for ‘constant bit rate’ and VBR is short for ‘variable bit rate’. In this case, the MP3 128kbps digital audio file would take up less storage space, but the MP3 320kbps digital audio file would offer markedly superior sound quality.

One small tip: In talking or reading about acronyms like these bear in mind that a lower case ‘b’ denotes ‘bits’, while a capital ‘B’ denotes ‘Bytes’.

Metadata

Literally ‘beyond data’, metadata is information about the data itself. For example, in an audio file, this might mean the title track, the artist, the composer, the genre, date of recording, date of composition, the album cover, band members, and more. This information about the music is generally ‘embedded’ within the file itself, to be read and displayed by media players and music servers alike. Metadata is enormously useful for listeners, simply because ‘Good Vibrations’ is a more memorable file name than ‘a156e03c’ to humans. Older file formats (such as WAV) are less robust in preserving metadata than their more modern counterparts.

MP3

MP3 is one of the oldest and most widely supported lossy digital audio compression codecs in the world. Over time MP3, which was created by the Fraunhofer Institute in the early 1990s, has emerged as a free ISO (International Organization for Standardization) standard that has also been incorporated by the MPEG (Motion Picture Experts Group) as part of both the MPEG-1 and MPEG-2 Audio Layer III standard.

MP3 was instrumental in the explosive growth that personal digital audio devices have enjoyed over the last 15 years or so, because it offered a means of substantially compressing large digital audio files so that even fairly large music libraries could be condensed to fit in devices with limited storage capacity (for example, early generation iPods).

MP3 also served, for many listeners, as an introduction to ‘perceptual coding’, where the general idea is to reduce the amount of data used to represent aspects of sound thought to be beyond the perceptual resolution of most listeners, while devoting data to the aspects of sound most readily heard and perceived. The concept was to reduce dramatically the amount of data that needed to be stored while still appearing to deliver full fidelity sound for most listeners, most of the time. Naturally, the idea of throwing out potentially useful sonic data did not sit well with most audiophiles and has been a topic of controversy and heated debate ever since.

Networked Audio & Network Streaming

Music stored on a computer can be removed to devices distributed across a home network (more accurately, a LAN or Local Area Network). This typically involves storing music on a computer or network attached storage device, which also runs some form of music server program to store and order these music files. The music itself is played through a ‘media renderer’ in your audio system that is also attached to the same computer network.

Functionally similar to internet streaming, networked audio distributes your own music library within the local network, instead of relying on online providers to stream their own music. While the popularity of personal libraries stored locally looks set to wane as online services proliferate, the networked audio system is a great way to store all your existing music collection in one easily accessible place.

PCM (and LPCM)

The former acronym stands for ‘pulse-code modulation’, while the latter stands for ‘linear pulse-code modulation’; both are means of representing analogue audio signals in a digital format. Many audiophiles use the terms PCM and LPCM interchangeably, though in fact the terms do not mean the same thing. PCM/LPCM is by far the most popular digital audio encoding format in use today.

Both PCM and LPCM sample the amplitude of analogue signals at precise and identical timing intervals. When each sample is taken, the amplitude of the signal is quantized and recorded as a multi-bit digital word. The difference between PCM and LPCM involves the manner in which signal amplitude is quantized; in PCM, samples are quantized to the nearest value within a range of possible digital steps, whereas in LPCM, samples are quantized to steps that are uniform in level.

The quality of PCM and LPCM encoding is largely controlled by two factors: the sampling rate (that is, the rate at which samples are taken) and the bit-depth of the samples taken (that is, the length in bits of the digital words used to represent each sample). As a general rule, all other things being equal, higher sampling rates and greater bit depths equate to better sound quality. Thus, a 24-bit/384kHz file of a song would likely sound superior to a 16-bit/44.1kHz file of the same song, assuming the master recording captured high levels of sonic detail and nuance in the first place.

“All other things being equal, higher sampling rates and greater bit depths equate to better sound quality.”

Resolution

In simple terms, ‘Resolution’ is the catchall phrase most audiophiles use to describe the amount of digital audio data used to represent analogue audio signals. As a general rule, the less data used the lower the resolution (and sound quality) will be, while the greater the amount of data used the greater the resolution (and sound quality) will be—up to a level where a perceived ‘point of diminishing returns’ is reached.

Generally speaking, lossy compression codecs yield what are considered low-resolution digital audio files. CD files, captured at 16-bits/44.1kHz are considered the standard, and files with higher-than-CD bit-depths and/ or sampling rates are considered to be high-resolution files.

Can listeners hear the difference? In a word, yes. The only area where there is room for discussion involves the question, ‘When is high resolution high enough?’

Servers

This term is the shortened form of the term ‘music server’. Typically, music servers provide a means of storing large quantities of digital audio files along with user interfaces that facilitate loading, organizing, and playing digital audio files. As a general rule, servers are typically thought to be self-contained units that not only store digital audio files, but also can deliver them for playback on demand.

Snake Oil

A term used by consumers to describe products that involve technological principles that are not well understood by the consumer. Examples of such technologies include EMI, decimation mathematics, image creation in the brain, bandwidth of the ear, phase effects, pre-ringing and reference measurement parameters. Snake Oil is a term of approbation which strongly implies that what is not understood is not valuable, rather than focusing value judgements on results achieved.

Streamers

By definition, streamers are network-attached devices that may offer Ethernet, Wi-Fi, and/or Bluetooth connectivity, or any combination of the above. The primary purpose of the streamer is to allow digital files from a music streaming service (e.g. Qobuz, Tidal, Spotify, Apple Music etc) to be located, selected and converted from internet protocol format to a format readable by an audio device like a digital-to-audio converter (DAC). Streamers are usually connected to the internet via an RJ 45 connector on an Ethernet cable connected to a switch or router that is part of your home network (hard wiring limits droputs and allows hi-resolution signals, unlike Bluetooth). DACs usually accept USB, S/PDIF, AES/EBU, I2S or Optical inputs.  Streamers may or may not have storage of their own for local files (in which case they would properly be called ‘streamer/servers’). Streamers often have an input or inputs to accept external files, for example on a memory stick or a portable SSD. Streamers have user interfaces to allow their owners to view, choose, and play audio content from the available network resources at hand. The compatibility of streamers with various user interface applications (e.g. Tidal Connect or Roon) and the provision of interfaces for switching between services (e.g. BluSound OS allows choosing between 25 services and selection of multiple output devices) is a point of differentiation between streamers.

UPnP/DLNA

UPnP (Universal Plug and Play) and DLNA (Digital Living Network Alliance) are similar sets of interoperability guidelines, allowing digital media devices to work together with little or no need for complex ‘handshaking’ protocols. Devices that fall under one (or more usually, both) standards are designed to be compatible with one another as standard, and fall into three broad categories for audio systems: control point (which might be an app on a tablet), media renderer (the network-attached DAC or streamer), and media server (that might be a computer or NAS drive).

WMA

This acronym stands for ‘Windows Media Audio’ a family of audio data compression codecs developed by Microsoft that together are part of the Windows Media framework or ‘ecosystem’. The are four WMA codecs:

• The original WMA codec is a lossy compression algorithm comparable to MP3.
• The WMA PRO codec supports multichannel or surround sound files (with up to eight discrete channels) and supports ‘high resolution audio’ (at up to 24-bit/96kHz levels).
• The WMA Lossless codec is a lossless compression algorithm.
• The WMA Voice codec is a low bit-rate, lossy compression algorithm focused specifically on conversational voice content.

WAV (or WAVE)

This acronym stands for ‘Waveform Audio File Format’, which was developed by Microsoft and IBM, and which is an uncompressed and therefore lossless file format that typically uses LPCM encoding. In theory, WAV supports compressed audio as well, though this is rarely seen in actual practice.

WAV and AIFF files are compatible with Windows, Macintosh, and Linux operating systems.

In simple terms, WAV—much like AIFF—is all about preserving maximum sound quality while eliminating compression artefacts of any kind. Two drawbacks are that WAV files take up considerably more storage space than files encoded by lossless compression codecs and that WAV files do not lend themselves to storage of album/song-related metadata. Recognizing the sonic potential of WAV, many manufacturers of ripping and/or music server software have come up with workarounds to allow WAV files to be stored with associated metadata.

ANALOG AUDIO TERMS

This brief section is intended for those who have little or no experience with analog audio and are eager to learn the basics. Treat this information as a set of foundational building blocks you can build upon later on.

JUST THE (ANALOG) BASICS

LPs, Records, ‘Vinyl’, or ‘Vinyls’

The whole idea behind analog audio is to achieve musically satisfying playback of vinyl phonograph records. Records are sometimes also called LPs (for ‘long play records’) or called ‘vinyl’ by the older generation or ‘vinyls’ by the younger buyers (as in, “I picked up some great new vinyls at the record shop today”). Vinyl LP records are relatively thin, flat vinyl discs, almost exactly 12-inches in diameter, with music—captured in the form of undulating grooves—pressed into their front and back sides. Traditionally, LPs rotate at 33 ⅓ RPM, although an increasing number of audiophile pressings now include multiple 45 RPM records, treating the LP as if it were a collection of 12” singles. In contrast, the single has commonly spun at 45 RPM and was often sold as either a 7” or 12” record. A small number of 10” extended play (‘EP’) records have also been produced, but – like the single – are rarely pressed today. By convention, the spiraling grooves in the record surface start at the outer rim of the record and move inward toward the record’s center. When the last piece of music on the record side is complete, the groove—no longer containing music—spirals inwards a bit further to a so-called ‘run-out groove’ where the stylus of the phonograph cartridge quietly rests, waiting to be lifted from the groove when the listener is ready either to turn the record over or to shut off the playback system.

Critically Important LP/Record Factoids

Staying within the (Straight) Lines: Masters lacquers for vinyl records are made on record cutting lathes where the lathe’s cutting head travels in a straight line from the outer rim toward the center of the master disc. In an ideal world, we would want the styli of our phono cartridges to follow this exact same straight line during playback, so that the phono cartridge/ stylus would remain perfectly tangent to the record grooves at all times. In practice, though, it is rarely possible to achieve true straight-line motion or perfect stylus-to-record-groove tangency at all times, so that engineers must create compromise solutions that position the phono cartridge stylus so that it remains nearly tangent to the record groove, most of the time.

Spacing Out: The spacing between record grooves is not constant, as some suppose. If you think about it, quieter musical passages require only very low amplitude modulations in the record grove, whereas loud and dynamic passages require groove modulations so high in amplitude that they are sometimes visible to the naked eye! Given this, record-cutting lathes can vary groove-to-groove spacing to allow for the dynamic swings that inevitably occur in music. This means that as the tonearm, phono cartridge, and stylus play the record from the outer edge to the innermost groove, their lateral motion is not absolutely constant, but rather varies in response to groove spacing variations.

Record Players

Some listeners (especially newcomers) sometimes use the informal term ‘Record Player’ to describe a complete record playback system, including a turntable, tonearm, and phono cartridge. However, audiophiles almost always discuss these playback components individually, as each has a separate role to play.

Turntables

Turntables are the devices we use to play or ‘spin’ vinyl records. The turntable’s job is to both support and rotate the record at a precise speed (typically either 33 ⅓ RPM or 45 RPM) during playback, while contributing as little noise and as few speed fluctuations as possible. (The human ear is extremely sensitive to speed fluctuations, because they translate directly into musical pitch fluctuations.) Some people use the word “turntable” to mean the whole record player assembly, but most serious audiophiles use the term to refer only to that part of the record player that is responsible for spinning the record.

Phono Cartridges

Phono cartridges are the devices tasked with ‘reading’ or tracking the grooves in the spinning record and then converting the physical movements involved in tracking the grooves into electrical signals that can be amplified for playback in our hi-fi systems. Phono cartridges have three basic elements: a stylus, a cantilever, and a motor (or signal generator mechanism) of some type. The stylus is the part of the cartridge that makes physical contact with the record groove and tracks the undulations in the grooves. Styli (the plural of stylus) are almost invariably made of extremely small, precisely shaped, and finely polished diamonds. The cantilever is a miniature rod or tube that forms a connection between the stylus and whatever type of electrical signal generator or motor the cartridge happens to use. The cantilever is typically supported by a flexible suspension system that serves double duty as both a ‘spring’ that supports the cartridge and as a damper to help control the motion of the stylus/cantilever mechanism. The motor of the phone cartridge translates the movements of the stylus in the record groove into an electrical signal that is analogous and proportional to the music encoded in the record grooves.

Tonearms

The tonearm’s job is to position the cartridge over the surface of the record and to hold the cartridge in place while the stylus is tracking the record grooves. This description sounds straightforward enough until you consider that the tonearm’s design brief can at times seem like a contradiction in terms.

For example, we want the tonearm to hold the cartridge’s body (or outer shell) almost perfectly still as the stylus, cantilever, and signal generating mechanism rapidly move in response to the groove modulations in the record. But at the same time, the tonearm cannot and must not hold the cartridge body in a rigidly fixed position; on the contrary, the tonearm must allow the cartridge freedom of movement in both the vertical (up and down) and horizontal (left and right) axes. These degrees of freedom of movement are necessary for three reasons.

First, tonearms must allow the phono cartridge to move so as to stay centered directly above the inwardly spiraling record grooves. Second, tonearms must allow cartridges to deal with the fact that many records are at least slightly eccentric, meaning the inward spiral of the groove is not necessarily smooth and continuous. Sometimes, listeners encounter records that require the tonearm to swivel back and forth (from left to right) as the record rotates, even if only very slightly. Third, many records are at least slightly warped, meaning the tonearm must allow the cartridge to move up and down to maintain a stable position relative to the surface of the record—a surface that, when viewed from the side, may at times appear to be ‘bobbing’ up and down as the record rotates.

Stated simply, the mission of the tonearm is to hold the cartridge in a stable position relative to record groove, while at the same time allowing the cartridge freedom of movement where necessary.

MORE ADVANCED ANALOG TERMINOLOGY

Anti-Skating Systems/ Skating Forces

The majority of tonearms on the market today are pivoted, non-tangential designs and the geometry of such arms makes for a condition where the cartridge stylus tends to be pulled inward toward the center of the record. This inward pull is called skating, and its result is that there is more stylus pressure on one side of the record groove than the other.

Ideally, we would want equal pressure on both sides of the record groove and to achieve this result many tonearms feature so-called anti-skating mechanisms that apply a compensatory force that is intended to offset skating forces.

Note that skating forces can and do vary with the amount of tracking force applied to the stylus, and also vary from one stylus shape to another (because styli of different shapes may have more or less ‘drag’ within the record groove). For these and other reasons, setting anti-skating forces is not an exact science and in fact some manufacturers advise against applying any anti-skating forces at all. In any event, adjustments to anti-skating force should—as with everything else in high-end audio—be verified by ear.

Arm Lengths/Stylus-to-Pivot Lengths

Phono cartridges mounted in pivoted tonearms move in an arc over the record and by following an arc the cartridge/stylus can achieve true tangency to the record groove at two points per record side. But at all other points the cartridge/ stylus assembly will experience some degree of tracing error, meaning the stylus will be just slightly askew to the ideal tangent-to-the-groove position.

This is where tradeoffs come into play and tonearm length looms large as a design variable. Generally speaking, the greater the length of a pivoted tonearm the lower its geometric tracing error will be—provided other length-induced design tradeoffs can be properly managed. However, increasing tone arm length is not a panacea, because longer tonearms may have potential problems with structural rigidity, unwanted resonance, cumbersome size, and excess mass.

These days the most common tonearm length is in the range of 9-inches from the pivot point to the stylus—a length that offers a good set of compromises in terms of structural rigidity, relative freedom from resonance, manageable mass, ease of handling, and reasonable physical size. At the same time, designers and listeners recognize that longer tonearms can and do reduce tracing error (because their arc shaped travel paths more closely approximate the theoretically ideal straight lines). For this reason, the analog world has in the past several years seen a resurgence of interest in 10-inch and 12-inch tonearms, with at least one manufacturer offering a turntable fitted with a 14-inch tone arm!

Azimuth

Azimuth refers to the degree of left/right tilt of the phono cartridge stylus as it rests in the record groove, where the ideal is for the stylus to be positioned exactly vertically in the record groove as viewed from the front.

One tricky factor, however, is that there is no guarantee that the stylus is perfectly aligned relative to the phono cartridge body, meaning that technically correct azimuth alignment might in fact require the cartridge body to be tilted just slightly to the left or right.

Not all tonearms (and especially not many inexpensive tonearms) offer provisions for making azimuth adjustments, but many mid and upper-tier tonearms do. Many enthusiasts have discovered that a very useful and simple tool for setting azimuth is a device called the Fozgometer (named for the veteran audio designer Jim Fosgate), which can used in conjunction with a set of recommended test records to check, revise, and adjust azimuth settings. It is also possible to use a test record and an oscilloscope for precision adjustment of azimuth, although this requires a considerably higher degree of user expertise… and the purchase of a test record and an oscilloscope!

Are the benefits of proper azimuth alignment audible? In high-resolution systems they most certainly are, making for a heightened sense of focus, clarity, and freedom from mis tracking on complicated musical passages.

Cartridge Overhang & Alignment/ Cartridge Adjustment Protractors

As stated above, the theoretical ideal would be for the phono cartridge stylus to move across the record surface following the same straight-line path followed by the record cutting head when the original master lacquer for the record was made.

The majority of turntables are fitted with pivoted tonearms that cause the phono cartridge/ stylus to swing in an arc across the record, rather than following a true straight-line path. Since an arc can only intersect a straight line at two points, the stylus can only achieve perfect stylus-to-groove tangency at two points on the record, meaning it will be slightly out of tangency at all other points on the record. To achieve best results with pivoted arms, two adjustments are critical: cartridge overhang (the exact distance from the arm pivot to the stylus) and cartridge alignment (the left-to-right angle of the cartridge relative to the tonearm and the record).

To help users adjust these two variables, many manufacturers offer cartridge alignment protractors, which are designed to slip over the turntable spindle and to rest temporarily on the turntable platter. Protractors provide markings that show where the stylus should be positioned in terms of overhang (X marks the spot) and that show how the cartridge/stylus should be aligned.

To use such protractors, listeners first loosen the fixing screws for their cartridges, then gently and carefully move the cartridges fore and aft and from left to right, following a gradual trial-and-error process until the desired overhang and alignment positions are achieved. Once the cartridge is correctly positioned, the fixing screws can be tightened to lock the cartridge in its properly aligned position.

Note that so-called straight-line or tangential-tracking tonearms also require overhang and alignment adjustments, but with the important difference that, once properly adjusted, they maintain perfect stylus-to-groove tangency across the entire record surface.

Cartridge Suspension & Dampening Systems

As noted above, the stylus/cantilever/motor assemblies used in all phono cartridges require some sort of suspension system, which in most cases will also double as a dampening system or ‘shock absorber’ of sorts. Many designs use either an elastomer ring or suspension block for this purpose, and as you may surmise the exact dimensions and compositions of these suspension/dampening elements are critical to performance.

If the suspension of the cartridge is too stiff or over damped, compliance will be reduced, and resonance problems may be introduced. On the other hand, if the suspension is too soft or under damped, compliance will be too high, and other types of resonance problems may arise (not to mention the potential problems of increased fragility and possible cartridge collapse). For obvious reasons, then, the idea is to achieve a carefully judged blend of appropriate compliance levels and damping characteristics that best suit the intended playback application.

It is worth noting that, in some moving coil cartridges, designers sometimes add a supplementary suspension/damping ‘tie-wire’ at the rear of the cantilever assembly to provide additional support and resonance control.

Cartridge Types

Phono cartridges tend to be classified by the types of signal-generation systems or ‘motor’ mechanisms they employ.

Moving iron & moving magnet: Moving iron and moving magnet cartridges are conceptually similar. In both cases, either a small magnet (moving magnet) or small ferrous metal tip with adjacent stationary magnets (moving iron) is fitted to the cartridge cantilever and positioned near a set of stationary coils of wire. As the stylus tracks the groove, the magnet or ferrous metal tip (acting as an induced magnet) is set in motion and generates a voltage in the cartridge’s signal coils. In most but not all cases, moving magnet and moving iron cartridges are considered high output designs and therefore should be used with phono stages that have a standard gain, moving magnet (“MM”) phono input.

As a general rule, moving iron cartridges are thought to offer better transient response than moving magnet designs, because their ferrous metal tips are lower in mass than equivalently sized magnets.

Moving coil: As their name suggests, moving coil cartridges feature cantilevers typically fitted with tiny cruciform frames around which are wound coils of wire positioned near sets of stationary magnets. As the stylus tracks the groove, the cruciform frame and coils are set in motion (within a fixed magnetic field), thus generating an audio signal. In the majority of cases, moving coil cartridges are considered low or mid-level output designs and therefore should be used with phono stages that have a high(er) gain moving coil “MC” input.

As a general rule, moving coil cartridges are thought to offer superior transient speeds and higher levels of detail than moving iron/ magnet cartridges, because their moving coils of signal wire are considerably lower in mass than moving magnet or moving iron signal generators. However, this theoretically superior performance comes at a price.

Generally speaking, moving coil models are more complicated to build and more costly to make and to buy than moving magnet/iron equivalents. Some moving coil models are prone to high-frequency resonances, which means designers must pay extra attention to damping schemes to mitigate potential problems. Finally, moving coil models typically require more costly high-gain/low-noise phono stages. With all this said, however, the majority of today’s top-tier phono cartridges are moving coil designs.

Optical: Optical phono cartridges use an optoelectronic mechanism to modulate a voltage supplied from an external power supply/ equalization box. In typical optical designs, which at this point are comparatively rare, the cartridge cantilever is fitted with a tiny light-permeable screen. When the stylus moves in the record grooves, the screen moves in response. An LED illuminates the screen, while an opto-electronic photodiode sensor located behind the screen ‘reads’ the light (as modulated by the moving screen) to produce an output signal.

Two theoretical advantages of optical cartridges are that their moving mechanisms are very low in mass, making for excellent clarity and transient speed, and they can in principle be very low in noise. One potentially significant drawback, however, is that they must be used with their own companion power supply/equalization boxes, which also serve in lieu of traditional phono stages.

Strain Gauge: Strain gauge-type cartridges are based—you guessed it—on strain gauges, which are flexible materials whose resistance to current flow changes as the materials expand and contract. In a stereo strain gauge cartridge, the cantilever is connected to two such strain gauges, with the strain gauges typically serving as both the suspension for the cantilever/stylus assembly and as the signal modulation mechanism.

Like optical cartridges, strain gauges require an external power supply box, but interestingly they do not require traditional RIAA equalization; this is because—unlike moving magnet, iron, or coil designs—strain gauges are not velocity sensitive transducers (where the signal depends upon how fast the stylus is moving), but rather are displacement-sensitive transducers (where the signal depends upon how far the stylus moves).

Advantages of strain gauges include the fact that their moving mechanisms are very low in mass and that their stylus/cantilever assemblies are directly and mechanically connected to the strain gauges that modulate their output signals. Three possible drawbacks are that strain gauge cartridges are costly to manufacture and to buy, are thought to be comparatively fragile, and they require use of a dedicated external power supply box.

Counterweights

Moveable counterweights are used at the back ends of tonearms, primarily to balance the arms once phono cartridges are installed, but also—in some but not all designs—to apply tracking force on the stylus. Also, for some unipivot tonearms, counterweights are deliberately eccentric in shape, so that the weights not only can move fore and aft, but also can rotate side to side for purposes of making azimuth adjustments. Typically, counterweights are made of relatively dense materials such as brass or, in some instances, even tungsten.

Headshells

The headshell is that element of the tonearm to which the phono cartridge is affixed, and which traditionally would provide a finger lift, if one happens to be used on the tonearm in question. Headshells may range from ultra-minimalist on through to quite elaborate designs that, in some instances, provide within-the-headshell adjustments for azimuth and for stylus rake angle.

Headshell designs can either be fixed (that is, permanently attached to the tonearm wand or perhaps even fashioned as an integral part of the wand) or detachable—usually via a locking collar of some kind. Proponents of fixed headshells cite their potentially superior strength, rigidity, structural integrity, and freedom from resonance, where proponents of detachable headshells emphasize the fact that detachable headshells facilitate cartridge swapping (because users are free to mount spare cartridges in separate headshells, thus making it possible to switch cartridges with a minimum of set-up hassles).

Motors

A wide variety of motors can be found in turntables, but some of the more common types are AC synchronous motors (motors that are in essence locked to the frequency of the mains), low-noise DC motors, and so-called ‘Hall Effect’ direct-drive motors (where in essence, the platter serves double-duty as the ‘armature’ of the motor).

Each type of motor has its ardent proponents, and each can, if well executed, give sonically superb results. The main points to grasp are that motors need to drive their associated platters at precise, unvarying speeds with as little noise as possible and with virtually no tendency to show speed fluctuations (not even extremely minor ones) in the presence of large or small-scale dynamic variations in the music.

Platters, Sub-Platters, Main Bearings, & Spindles

Platters: Platters are the relatively heavy, disc-like elements upon which records rest and rotate while in play. Ideally, we would want platters to be perfectly flat, perfectly round, and to be fitted with spindles that are perfectly centered in the platter’s top surface (the spindle is a round vertical post used to center the record upon the platter). Further, we would want platters to offer sufficient mass that, once in rotation, they would have enough inertia to be able to resist speed fluctuations—even when playing records where timing accuracy is hyper-critical (e.g., certain piano passages) or where there are wild dynamic variances over time (think of Tchaikovsky’s classic 1812 Overture). Finally, we would want platters made of materials that offer good internal damping and provide a solid, neutral sounding support surface for the record. It is common to see platters made of machined aluminum, glass, brass, copper, composite materials or combinations of the above.

Sub-Platters: Depending on the design brief being followed, some turntable designs feature platters that rest upon smaller sub-platters to which the turntable drive mechanism is connected and to which the main bearing of the turntable is attached.

Main Bearings: Main bearings must support the weight of the platter while allowing it to rotate as smoothly and quietly as possible. It is important to bear in mind that any noise— even seemingly very low-level noise—from the main bearing can be passed upward through the platter and the record, to be picked up by the phono cartridge. For this reason, precision-made main bearings are an absolute must for optimal sonic results to be achieved. It takes a great deal of expertise to design and to manufacture top-class main bearings, but the effort pays huge dividends in terms of sound quality. Indeed, one of the biggest differences between good vs. great turntables lies in the quality of the main bearings used.

Some common main bearing types include shaft and bushing designs (with or without continuous recirculating oil baths and with or without inverted bearing shafts), shaft and ball designs, air bearings (where the weight of the platter is borne upon a cushion of pressurized air), and opposed magnet supported bearings, where sets of opposing magnets are used to partially ‘levitate’ the platter thus relieving physical pressure on the bearing assembly. Bearings can be made of hardened tool steel with or without jeweled contact surfaces or balls, sintered bronze, other exotic metal alloys, ceramics, composites, specialized plastics/ polymers, and other man-made materials.

Spindles: Spindles are precision-made circular posts, typically made of metal, which protrude from the top center surface of the platter. Spindles are made to an industry standard diameter and their primary purpose is to act as a centering-pin for records, when records are placed on the platter for playback (and yes, there is a corresponding, industry standard, spindle-sized hole in the center of all LP records). But one other purpose for the spindle is to provide a gripping surface to which optional record clamps, if any, may attach.

Plinths

Plinths are the externally visible housings or structural frames for turntables. In some designs, the plinth is essentially an outer shell to which various sub-frames or assembles (for example, motor mounts) are attached—or from which they are suspended.

In other designs, however, the plinth basically is the frame of the turntable, to which the turntable’s tonearm, main bearing/platter assembly, and in some cases even the drive mechanism or motor is attached.

Can plinths affect sound? Recent Hi-Fi+ reviews of aftermarket plinths for popular turntables such as the Linn LP12 suggest that plinths can have a surprising high level of impact on the turntable’s overall sonic presentation.

For this reason, it is important to respect plinths as significant elements of turntable design and not as an afterthought.

Record Clamps and Vacuum Hold-Down Systems

Many analog audio experts think that it is desirable to clamp records firmly to the platters upon which they rest during playback and for this reason a number of turntable makers and aftermarket accessory manufacturers offer specialized record clamps, which typically are attached via the platter’s spindle.

Others go even further, suggesting that, since many records are very slightly warped, it is desirable not only to clamp records at their centers, but also around their outer perimeters (so that the records will lie perfectly flat upon the platter’s top surface). Accordingly, a handful of manufacturers offer ring-shaped clamps, typically made of metal, which slip over the outer edges of the record and turntable platter, thus coupling the record firmly to the platter, flattening out any warps in the record surface as a result.

Finally, it is worth noting that not all analog experts are devotees of record clamps, mostly out of concern that clamps might put undue pressure on the platter main bearing while potentially creating unwanted stresses in the record surface.

One way of achieving the benefits of clamping systems, but without actually using clamps, is to build turntables that incorporate vacuum-powered record hold-down systems. Turntable manufacturers such as SOTA and TechDAS have done just this, with very good results. The only drawbacks to the vacuum hold-down approach involve complexity, costs, and the need to manage the noise produced by the requisite vacuum pumps.

RIAA (and other) phono EQ curves

A fact little known among laymen is that records as pressed do not have flat frequency response. On the contrary, during the record mastering process specific equalization curves are applied curves that reduce the amplitude of bass frequencies and boost high frequencies. The typical EQ curve used is called the RIAA curve, where the acronym stands for Recording Industry Association of America. There are also other phono EQ curves that provide similar functions, although they are far less common than the RIAA curve. Alternate phono EQ curves include those from CCIR/Teldec, Columbia, DMM, and Decca/EMI. Arguments continue to rage today as to whether record companies switched wholesale to the RIAA curve when stereo arrived in 1958, or whether recordings cut in the 1960s or later used the alternate EQs derived in the monophonic era.

Why is phono equalization necessary? The answer is that bass content, if cut into the record with flat frequency response, would require record groove modulations so extreme that it is doubtful that even the finest phono cartridges could properly track them. What is more, the modulations would be so large in amplitude that they would force unfeasibly wide spacing between record grooves, which would severely limit the amount of content that could be included on each record side. At the other end of the audio spectrum, high frequency material, if cut into the record with flat frequency response, would potentially be so low in amplitude that it might get masked by naturally occurring groove noise.

Thus, phono equalization, complete with boosted highs and trimmed-back low frequencies, is always applied during the record mastering process. However, in order to restore flat frequency response when playing vinyl records, inverse phono equalization is applied during the playback process via a specific type of preamplifier called a phono stage. All phono stages provide inverse RIAA equalization, but some of today’s more elaborate, upper tier phono stages may also provide six or more specialized phono EQ curves, as mentioned above.

Rumble

Rumble is a measure of the detectable noise generated by turntables as they rotate, so that you could think of rumble as being the turntable world’s equivalent of the signal-to-noise-ratio in conventional audio electronics. Rumble is typically quoted as a negative dB figure (for example, -64dB) where—as with signal-to-noise ratios—the higher the negative number of dB, the quieter the turntable will be.

As with audio electronics, lower rumble in turntables may not necessarily be perceived as ‘lower noise’ (although it is just that), but rather as ‘enhanced low-level detail’ in the music.

Snake Oil

A term used by consumers to describe products that involve technological principles that are not well understood by the consumer. Examples of such technologies include EMI, decimation mathematics, image creation in the brain, bandwidth of the ear, phase effects, pre-ringing and reference measurement parameters. Snake Oil is a term of approbation which strongly implies that what is not understood is not valuable, rather than focusing value judgements on results achieved.

Speed Controls

It is impossible to overstate the importance of proper speed control in turntables since even very minor speed fluctuations can, under the right circumstance, be painfully audible (long, sustained piano chords are extremely revealing in this respect). For this reason, many designers have developed precision outboard power supply/speed control regulation boxes that serve to tighten up the speed accuracy of their associated turntables.

Is this just an example of ‘gilding the lily’? No. Proper speed control can make all the difference between a good turntable and a great one.

Stylus Profiles

The exact shape and dimensions of the phono cartridge stylus have much to do with how well the phono cartridge will track the record grooves. Some common stylus shapes you will encounter are the following:

Conical/Spherical: As the name suggests, conical styli are cone-shaped, but with rounded, hemispherical tips. Conical/spherical styli are the easiest to make and are the least finicky about set-up, but they have performance limitations in that they are comparatively high in mass, have relatively large tips with respect to the dimensions of the record grooves, and also provide relatively small ‘contact surfaces’ (analogous to the ‘contact patches’ of automotive tires) between the stylus and the groove.

Elliptical: An elliptical stylus represents an improvement over the conical/spherical because, rather than having a large round tip, the elliptical stylus offers a tip with an elliptical profile whose narrower edges face to the sides and directly contact the record groove. Two benefits accrue. First, the elliptical stylus is lower in mass than an equivalent conical stylus would be, and second, the elliptical stylus’ narrower but more elongated contact surface offers a better fit for purposes of tracking the undulating contours of the record groove (those narrow radius contact points can much more readily track high-frequency details, for example). Elliptical styli require somewhat more attention to set-up, but are still relatively forgiving.

Shibata: The Shibata stylus, named after its inventor, represents an even more radical step forward from the elliptical stylus in that it has an even narrower tip shape that, under a microscope, looks somewhat like the blade of a garden trowel turned so that the flatter side of the blade is facing the viewer. The side-radius of the Shibata tip is even smaller than that of an elliptical stylus so that the contact surface is not merely a somewhat elongated ellipse (as with typical elliptical styli), but rather is a much taller and narrower ellipse that almost resembles a vertical line. Relative to elliptical styli, Shibata styli offer three compelling advantages: significantly lower tip-mass, even narrower side-radius dimensions for superior tracking of high frequencies, and—somewhat unexpectedly—an increase in contact area with the record groove (meaning that even if higher tracking forces are used there is still less stylus pressure per square centimeter than with an elliptical design). Because the side-profile of the Shibata stylus is narrower and more blade-like than with elliptical designs, greater care must be taken to make sure that the stylus rake angle is properly adjusted.

Line Contact/Fine Line: Line contact/fine line styli, often attributed to the designers A.J. van den Hul and Fritz Geiger, represent an even further advancement along the same lines that inspired the Shibata stylus. The general idea is to pare away yet more stylus tip mass while narrowing the side-radius of the stylus tip, so that the stylus contact area becomes an extremely narrow and elongated ‘fine line’. But don’t let the shape and dimensions of that fine line mislead you; the fine line/line contact shape still offers plenty of stylus-to-groove contact area, so that stylus pressure per square centimeter still remains reasonable. Once again, improvements are noted in high-frequency tracking and in overall ability to trace fine, small details in the record grooves. More so than other stylus types, line contact/fine line styli are sensitive to set-up and to stylus rake angle adjustments.

Stylus Rake Angle

Stylus rake angle (SRA) refers to the front-to-back tilt angle of the phono cartridge stylus vis- à-vis the record grooves (whereas azimuth is the side-to-side tilt angle of the stylus in the groove). Unlike azimuth, however, the optimal stylus rake angle is not dead vertical (90 degrees), but rather is thought to be in the range of 91.5–92 degrees (depending upon which experts you consult), with the stylus tipped back just a bit, as if ‘scooping’ into the oncoming groove by 1.5–2 degrees.

Why is this very slight tilt back desirable? The answer is that the cutting head used to produce the lacquer master for the record also had a similar degree of tilt back. As always, for best sonic results the ideal is for the phono cartridge stylus to come as close as possible to following both the horizontal path and the vertical ‘angle of attack’ of the original cutting head.

It is possible to adjust SRA by ear, but an even more foolproof method is to use a USB microscope to observe and adjust the stylus rake angle as the stylus is resting upon the record.

Note that not all tonearms make provisions for SRA adjustments and note too that many audiophiles and even some experts tend to use the terms ‘stylus rake angle’ and ‘vertical tracking angle’ (VTA) interchangeably—even though they aren’t precisely the same thing. Sonically speaking, though, SRA is the adjustment you want to get right.

Turntables with Suspensions vs. Mass-loaded Turntables

Almost all turntable manufacturers seek to isolate key elements of their playback systems from both mechanical and airborne vibration, but there is much divergence of opinion as to how best to achieve that result.

Some designers believe in using mass loading to prevent (or at least suppress) transmission of unwanted vibrations and their designs typically use fixed, solid plinths to which the turntable platter and tonearm assemblies are firmly affixed (though turntable motors/ drive units may, in such designs, be mounted in separate housings or ‘pods’ that stand apart from the main plinth). In such mass-loaded designs, there usually is no suspension at all, apart from feet that may, in some instances, provide built-in elastomeric or spring-loaded suspension elements.

Other designers, however, strongly believe that it is best to have the turntable platter and tonearm mounted on sturdy sub-chassis that is suspended and—to a degree—isolated from its surrounding plinth. For even greater noise isolation, such designs very often attach the motor to the turntable plinth and then use an elastic belt-drive system to transfer power from the motor to the platter.

As a general rule, mass-loaded turntables are sometimes more prone to mechanically induced noise and vibration transferred via audio furniture or the floor, while suspended turntables tend to offer somewhat better vibration isolation, but at the expense of considerably more elaborate initial set-up procedures and a certain tendency to drift out of adjustment over time.

Tonearm Types

In broad strokes, there are three main types of tonearms you might encounter, although pivoted tonearms are by far the most common types. The other two types of arms are radial-tracking/ straight-line tonearms and tangential-tracking tonearms.

Pivoted Tonearms: Pivoted tonearms may feature straight or curved tonearm wands with either fixed or detachable cartridge headshells at the front end, a bearing assembly toward the rear, and a counterweight at the back end. In a pivoted arm, the cartridge/stylus always moves in an arc across the record surface, though tracing errors can be mitigated by careful adjustment of cartridge overhang and alignment angles.

Radial-tracking or ‘Straight-line’ Tonearms: Radial-tracking or straight-line tone arms almost invariably feature comparatively short, straight tonearm wands with either fixed or detachable cartridge headshells at the front end, a bearing/arm carrier assembly toward the rear, and a counterweight at the back end. What sets straight-line tonearms apart, though, are their distinctive bearing/arm-carrier assemblies, which significantly allow the tonearms to move straight sideways—not swinging in an arc as pivoted arms do. In this way, the arms realize the ideal goal of having the stylus move in a perfectly straight line across the record, always maintaining perfect tangency to the record grooves. The downside of straight-line tonearms, however, is that they are complicated to design and build, costly, and can in some instances prove difficult to set-up and to keep in proper adjustment.

Tangential-tracking Tonearms: Tangential tracking tonearms are conceptually a cross between pivoted tonearms and radial-tracking tonearms. On one hand, tangential-tracking tonearms are pivoting designs, but with one crucial difference: their cartridge headshells are not locked in a fixed position on the tonearm wand, but rather are position on an articulated mount that—get this—allows the cartridge alignment angle to be continuously adjusted during playback to maintain stylus-to-groove tangency all the way across the record. To achieve this desirable result, most tangential-tracking tonearms are built with a main tone arm wand and a secondary control arm that rides beside the main wand and that is responsible for making continuous alignment adjustments as needed. When viewed from above, tangential-tracking tonearms and their associated, articulated headshells look something like slender, elongated trapeziums. For obvious reasons, tangential-tracking tonearms must be crafted with extremely tight-tolerance bearings for the arms’ several articulated joints.

Tonearm Bearing Systems

As mentioned above, it is very important for tonearms to offer nearly friction-free movement, while preserving tonearm/cartridge/stylus geometry with great precision. To this end, designers have devoted a lot of attention to the types of bearings used. Some types commonly encountered are as shown below:

Air bearings: Air bearings are typically shaft-and-sleeve bearings where the sleeve is fed pressurized air from an external source so that the shaft never makes metal-to-metal contact with the sleeve, but rather rides on a virtually friction-free cushion of air. This type of bearing is used in a number of straight-line tone arm designs. Examples would include the Bergmann Magne, Kuzma Air Line, or Walker Proscenium Back Diamond V tonearms.

Ball/Gimbal bearings: Precision-made ball bearings are popular for use in tonearms, often via gimbal-type mounts where one pair of bearings handles horizontal axis motion, and the other pair handles vertical axis motion. Ball bearings are often graded using ABEC (Annular Bearing Engineering Committee) ratings where the higher the ABEC number the tighter the bearing tolerances are.

Knife-edge bearings: Some tonearm designs have used so-called knife-edge bearings for vertical axis applications. A knife-edge bearing consists of a knife-like blade that rides within a corresponding, precision machined V-shaped trough.

Multi-point/Kinematic bearings: Multipoint or kinematic-type bearings, as used by a handful of manufacturers, combine the precision of ball/gimbal-type bearings but offer the promise of even lower friction and essentially zero ‘free-play’ in the bearings. The general idea is to precisely locate the center of motion typically using just three or four contact points. Examples would include the Kuzma 4 Point and Wilson-Benesh ACT-series tonearms.

Thread-type bearings: Some tone arms forego traditional, metal rotational bearings and use threads not only to suspend the tonearm but also to afford it both horizontal and vertical motion. Examples would include the Well Tempered tonearms or the Funk Firm F6 tonearm.

Unipivot bearings: As their name suggests, unipivot bearing feature just a single point of contact—an idea appealing in its simplicity. Such bearings typically feature a spike (with or without a jeweled tip) that rests in a cup (again, with or without a jeweled contact surface). One point to note, though, is that arms fitted with unipivot bearings must be balanced from side-to-side in order to achieve proper azimuth alignment.

Tonearm wands/tubes, etc.

As mentioned above, tonearms must position phono cartridges precisely without introducing resonance problems. For this reason, arm wands/tubes must be strong, rigid, well damped, and as resonance-free as possible.

Most tonearm wands are constructed as tubes that can be made of metal, plastics, composites, or hybrid combinations of materials. Many manufacturers enhance tubular tonearm designs either by adding internal stiffeners or by adding dampening materials, or both.

Lately, several manufacturers have begun to experiment with 3D-printing techniques for arm wands, some using plastic-type materials and others using metal materials. 3D printing allows complex shapes/designs that could not be made via traditional machining techniques.

Tracking Force

Tracking force is the amount of downward pressure applied to the phono cartridge stylus and that is necessary in order for the stylus cleanly to track demanding material encoded in the record grooves. Above all, the intent behind using the proper amount of tracking force is to make sure the stylus remains in contact with the walls of the record grooves at all times, yet without applying so much pressure that the groove walls are damaged or subject to undue wear.

When a stylus does break contact with the record groove, even if only to a slight degree, that condition is called mis tracking, which is audible, unpleasant-sounding, and hard on the record grooves. Typical tracking forces for most modern phono cartridges will range from the mid-one-gram range to the mid-two-gram range, in accordance with published specifications for the cartridge. The general idea is to use sufficient force to eliminate mis tracking, but not more force than is necessary.

Contrary to popular assumptions it is preferable to use slightly too much tracking force than not enough. While heightened tracking force does increase record wear to a degree it also tends to help prevent mis tracking, which can be even more damaging to one’s record grooves.

Turntable Drive Systems

Turntables are often classified by the drive mechanisms they use. Some common drive mechanism types are described below.

Belt drive: In belt drive turntables the motor stands separate from the platter assembly, while a precision-made belt (typically, but not always made of elastomeric materials) transfers power from the motor drive pulley to the turntable platter (or to a sub-platter beneath the main platter). Some designs use thread or magnetic tape in lieu of an elastomeric belt. The belt is thought to decouple the platter from the motor, keeping motor noise from being transferred into the platter where it could be detected by the phono cartridge.

Direct drive: In a true direct drive turntable the ‘armature’ of a Hall-effect motor is embedded within the platter, while other parts of the motor are contained in the turntable plinth. In other words, the platter is essentially its own motor. If properly designed, direct drive turntables can be extremely quiet as their motors, by definition, rotate at platter speed and thus do not introduce higher-frequency vibrations. Also, direct drive tables—again, if properly designed—also allow extremely tight speed control.

Early generation direct drive turntables sometimes got unfavorable reviews because their designs allowed some degree of audible motor ‘cogging’ and because their speed control mechanisms sometimes introduced noise and micro-variations in speed. More contemporary designs typically address and solve both problems.

Idler-wheel drive: Idler wheel drive, sometimes confusingly called ‘direct drive’, involves a motor with a drive wheel and an idler wheel that transfers motor power to the platter. Almost the opposite of belt drive designs, idler wheel designs forge a direct coupling between the motor and the platter, so that it is imperative to base such designs on extremely low-noise motors (typically very high-quality DC motors). Proponents of idler-wheel drive praise their dynamic immediacy and solidity as well as their freedom from such micro-variations in speed as can be introduced by elastic drive belts.

Magnetic drive: Magnetic drive offers another method for transmitting power to the platter while at the same time physically decoupling the motor, per se, from the platter. In this system, the motor typically drives a substantial sub-platter, which is magnetically coupled to a physically isolated platter positioned directly above the magnetic coupler. When the subplatter rotates, its magnets attract those in the platter above, causing the platter to rotate.

Vertical Tracking Angle (VTA)

Many audiophiles and experts use the term vertical tracking angle to describe what should properly be called stylus rake angle (SRA). See above.

Wow and Flutter

The terms ‘Wow’ and ‘Flutter’ refer to two undesirable types of speed variation in turntables. Wow is a slow, gradual fluctuation that might yield a slow “Wow” sound as speed gradually increases and then decreases. Flutter is a more rapid speed fluctuation with would produce vibrato or tremolo-like sounds as speed rapidly increases or decreases. For obvious reasons, it is desirable to have turntables that produce as little wow or flutter as possible, though of the two types of speed variation flutter is arguably the more noticeable.

AUDIO CABLE TERMS

High-end audio cables, much like other categories of audio components, have gradually developed a specialized vocabulary all their own. And, as sometimes happens with other types of audio products, ‘cable speak’ can at first seem confusing if not dauntingly obscure to the uninitiated. But not to worry; help is on the way. The Absolute Sound team has assembled this Glossary of Audio Cable Terms to explain cable terminology in a manner that interested laymen will be able to understand (or at least that’s the plan). Enjoy.

Analogue Interconnects (or Interconnects)

Analogue interconnects are audio cables specifically designed to carry low-level analogue audio signals from source components to amplification components, or from preamplifiers to power amplifiers.

Typically, analogue interconnects come in two forms: single-ended cables (in most cases fitted with RCA jacks at both ends) or balanced cables (usually fitted with a male three-pin XLR plug at one end and female three-pin XLR socket at the other end).

Balanced Interconnects

The majority of interconnects are single-ended cables that have two conductors—one carrying +/– signals and the other serving as a ground.

Balanced cables, however, are different in that they have three conductors—one for the + signal, one for the – signal, and one serving as a ground.

When properly executed, balanced audio circuits offer either higher output than or lower noise levels than equivalent single ended circuits, which allows longer runs of cables and that is why pro-audio equipment is almost universally balanced in operation. However, balanced circuits are inherently more complex to design and manufacture than equivalent single-ended circuits, and likewise balanced cables are more complex (and usually more costly) than their single ended counterparts.

Some common balanced connector types include XLR connectors (much like the connectors you might see on professional microphones), TRS or ‘tip-ring-sleeve’ connectors (which look like ¼-inch phone plugs and are more commonly seen in pro-sound rather than high-end home audio applications), and AES/EBU connectors (which are used for balanced digital audio applications).

Bi-Wiring

Some loudspeakers are configured to allow bi-wiring, which means that instead of having just one +/- pair of connection terminals, the speakers—usually, but not always, two-way designs—instead have two sets of terminal, where one set is for the low-frequency driver and the other for the high-frequency driver.

When choosing to bi-wire, users would run two complete sets of speaker cables to each loudspeaker—one routed to the low frequency driver terminals and the other to the high-frequency driver terminals. In theory, this practice can yield a purer, clearer, and more tightly focused sound overall.

Several technical explanations are offered to explain the ostensible benefits of bi-wiring, but opinions on the efficacy of bi-wiring can and do vary among high-end cable designers.

When choosing not to bi-wire, users would instead run a single primary set of speaker cables to their loudspeakers—typically to the terminals for the low-frequency driver, and then would run a set of short ‘jumper’ cables (ideally identical in configuration to the main cables) from the low-frequency driver terminals to their adjacent high-frequency driver terminals.

Capacitance, Resistance, & Inductance

These three electrical characteristics are the basic building blocks of all high-end cable designs; they are the essential variables that cable designers seek to manipulate in their quest for higher performance and better sound.

Capacitance is the ability of a cable (or a capacitor) to store an electrical charge.

Generally speaking, most designers consider that lower capacitance is better. The train of thought is that one does not want an audio cable to absorb and store an electrical charge from the music signals being passed through the cable, because such charges will inevitably be released (or dissipated) later on in time, thus ‘smearing’ the sound of the music.

Inductance is the property of cable (or an inductor) to resist changes in current flowing through the cable through the process of inducing an electromotive force (EMF), which actively resists current changes. Generally speaking, most designers consider that lower inductance is better, since ideally one would want cables to allow current changes to occur in a natural or free-flowing manner as required by changes in the music signal.

Resistance is a measure of the difficulty to pass an electrical current through a conductor—in this case a cable. Generally speaking, most designers consider that lower resistance is better, since the lower the resistance the less energy is dissipated within the cable when driving current through the cable. This factor can be especially important in designing cables that are meant to conduct very low-level audio signals with minimum signal loss and distortion.

Coaxial Cable

A type of cable construction often used in digital or single-ended interconnects with a central +/- signal conductor surrounded by an insulating (dielectric layer), in turn surrounded by an outer conductive shield or sheath used as a ground or ‘return’, with a protective insulation jacket on the outside. The central conductor and the conductive sheath both share the same axis; hence the term ‘coaxial’.

Conductors

Technically, conductors are materials that permit electrons to flow freely and that allow electrical current to flow in one or more directions. Wires, in turn, are conductors that can carry electricity over their entire length. Conductive materials used in audio cables include copper, silver, gold, rhodium, and in some recent exotic designs, palladium and graphene. At least one manufacturer uses liquid metal conductors made from gallium, indium, and tin.

Depending on which designer one asks, the exact composition of wires, both in terms of the conductive materials used, the metallurgy of the wire, and even the cross-sectional characteristics of the conductors, are thought to have significant impact on sound quality.

Stranded-Core designs: In many cases the wires used in audio cables are composed of multiple, bundled, small-diameter strands of conductive materials—collectively known as stranded-core designs.

Solid-Core designs: In other typically higher-end audio cables, wires use solid-core conductors that are considerably larger in cross-sectional area than the tiny conductive strands used in stranded-core designs. The size and shape of the solid-core conductors used are thought to have an impact on sound.

Thus, at least one famous cable manufacturer touts the use of ‘rectangular solid core’ conductors, while another uses solid core conductors whose also rectangular cross section uses so-called ‘Golden Section’ proportions.

In a ‘big picture’ sense, the better the conductors an audio cable employs, the better it will sound.

Crystal or Monocrystal Conductors

The overwhelming majority of audio cables use metal conductors, but what few listeners realize is that the wires within those cables have a crystalline structure (many equate ‘crystals’ with gemstones, but metals are crystalline, too).

Under normal circumstances, drawn metal wires contain numerous metal crystals butted up against one another and many audio purists believe that the junctures between these crystals have a subtle, adverse effect upon sound quality.

However, one important development is the advent of manufacturing techniques that allow wire makers to produce monocrystal wires, where one metal crystal spans the entire length of the wires (meaning there are no crystal-to-crystal junctions to affect the sound in any way).

Cables featuring monocrystal conductors are highly prized for high-purity/high-accuracy applications, even though they are typically more expensive to make than conventional multi-crystal conductors.

Dielectrics

In simple terms, dielectrics are insulators—the materials or other related systems used to provide insulation for the conductors found in audio cables.

Dielectrics are important because they have much to do with the cable’s capacitance and thus resulting sound quality (see ‘Capacitance/ Inductance/Resistance’). The ideal would be to have dielectrics that absorb no electrical charges at all.

Some common dielectrics include fluorinated ethylene polypropylene (FEP), polyethylene, polytetrafluoroethylene (PTFE, aka Teflon), and others—many of which are available either as solid or as “foamed” materials. Several manufacturers have experimented with insulation systems that use air or a vacuum as dielectrics (because, in theory, a perfect vacuum would be an ideal insulator, though for obvious reason vacuums are very difficult to manage in a cable context).

Dielectric Bias System (DBS)

DBS is AudioQuest’s trade name for a system (co-developed with loudspeaker designer Richard Vandersteen) for applying a bias voltage (via a small battery) across the dielectrics of audio cables, effectively making them highly resistant to accepting music-induced electrical charges. One claimed advantage of DBS is that it obviates the need for lengthy cable ‘break-in’ periods.

Digital Interconnects

Audio cables specifically designed for carrying low-level digital signals (or files) from digital source components (e.g., a CD transport, music server, or streamer) to a digital audio component capable of decoding those signals.

At first glance, it is tempting to think of digital interconnects as being ‘just like’ analogue interconnects, but in fact the two cable types have significantly different ‘mission profiles’. Analogue cables must accurately convey analogue signals ranging in frequency from a few Hz on up into the kHz range.

Digital cables, instead, are expected to transfer square wave signals (representing digital ‘ones’ and ‘zeroes’) in the MHz range, loading into digital components whose input impedances are potentially quite different to analogue components.

Some common digital interface types include:

• AES/EBU (Audio Engineering Society/European Broadcasting Union)—a quiet, balanced digital audio interface that uses XLR-type connectors.
• Ethernet—a reliable, well-documented, wide-bandwidth multipurpose digital connection borrowed from the computer world, which typically uses RJ-45-type connectors and sockets.
• S/PDIF (Sony/Philips Digital Interface Format)—a popular and robust digital audio interface that typically uses coaxial wires with RCA-type plugs.
• TOSLINK (Toshiba Link)—a popular and robust digital audio interface that, instead of wires, uses fiber-optic connections that typically use EIAJ/JEITA RC-5720 optical connectors. Note: TOSLINK is essentially a fiber-optic implementation of the S/PDIF standard.
• USB (Universal Serial Bus)—an enormously popular, multi-purpose digital interface that has in recent years come to be the digital interface of choice for many high-end (and not-so-high-end) digital audio components. The USB specification allows for many types of connectors, but the ones most commonly seen in audio applications are: USB Type A (as found on many PCs and other digital sources), USB Type B (as found on many high-end audio DACs), USB Mini A – USB Mini B and now USB C (used on many smartphones and portable digital audio components).
• Lightning (Apple)—Since Apple removed the 3.5mm mini-jack from its popular iPhone range, the company’s own connector has become increasingly important in digital audio replay on the move.

Directionality in cables

Although the subject is considered somewhat controversial, the fact is that most if not all audio cables (or more accurately, the conductors within those cables) exhibit directionality—meaning that signal flow works and sounds better running in one direction than the other. The technical explanations behind this are somewhat complex, but according to AudioQuest founder Bill Low:

“All drawn metal has a directional impedance variation at higher RF/EMI noise frequencies. By ‘law’, energy must follow the path of least resistance, so we employ this impedance variation as a mechanism for consciously directing noise either to Earth or to whichever attached circuit is less vulnerable to noise. The key is to direct noise to where it will do the least damage.”

What is more, some cable designs use asymmetrical shielding schemes (where noise blocking outer sheaths might be, for instance, connected to ground only at one end of a given cable), adding a further directional element.

Given this, expect to see markings (arrows, marker rings, and the like) on many high-end audio cables to indicate the preferred direction of signal flow. Some speaker cables, for instance, even provide terminations marked ‘speaker end’ or ‘amplifier end’.

Gauge (or Wire Gauge)

The gauge of a cable, typically expressed as AWG (American Wire Gauge), is an indicator of the cross-sectional area of the wires used in the cable. AWG ratings are arranged so that the lower the AWG number, the more cross-sectional surface area the cable possesses. A giant power cord, for instance would have a very low AWG number, while the tiny run-out wires in a tonearm headshell would have a very high AWG number. Note: AWG numbers are considered useful indicators of a cable’s current carrying capacity (the lower the AWG or gauge number, the higher the current load the cable can bear).

Hospital Grade Power Plugs/Sockets

In discussions of American AC power distribution, we often encounter references to ‘hospital grade’ mains sockets and plugs. The reference is to specifications for mains sockets and mains cable plugs designed for use in ‘mission critical’ hospital applications (you wouldn’t want an AC plug to fail on a respirator, now, would you?).

Hospital grade sockets and plugs specify materials that can withstand both chemical and physical abuse and, in the case of plugs, also specify relatively tight-fitting connector pins that, by design, are difficult to dislodge.

There is no direct UK equivalent to the ‘hospital grade’ socket (in part because the three-pin socket used in the UK is hard to dislodge), but audiophiles in the UK often opt for unswitched 13A designs in place of standard switched models.

There is much debate over whether hospital grade mains connections are necessary or beneficial for audio applications, but many purists choose to use them (both for mains cables and for power distribution components)—if only as a precautionary measure.

“The most common result of skin effect is a tendency for a cable’s AC resistance to increase at higher frequencies.”

Litz wire

Litz wire is a specific cable configuration that uses bundles of multiple small-diameter, individually insulated strands of conductors, where the strands are typically twisted along the length of the cable. The main intent behind Litz wire is to mitigate the sonic problems associated with skin effect (see ‘Skin Effect’).

The most common result of skin effect is a tendency for a cable’s AC resistance to increase at higher frequencies, potentially causing at least some degree of audible treble roll-off. Happily, Litz wire overcomes this problem for the most part.

A few power amplifiers designed to be used with conventional stranded loudspeaker cable have been known to ‘struggle’ with the low resistance of Litz wire. Fortunately, in every cases we know of, these problems were resolved in the 1980s and are now historic.

Mains or Power Jacks & Plugs

Often, we think of our own AC connections as the norm, forgetting that there are actually numerous international standards for power distribution voltages, frequencies, and the sockets and plugs to deliver electrical power. The US Department of Commerce International Trade Association has identified 15 specific types of power plugs/ sockets in use worldwide (these plug socket combinations are assigned identifying letters from A through O).

The tricky part, however, is that various countries and regions use these 15 types of power plugs, some grounded and others not, in sometimes unusual or unexpected combinations.

One upshot of all this diversity is that high-end audio power cable manufacturers must potentially create very broad ranges of models in order to address the needs of the worldwide market.

Ohno Continuous Casting (OCC)

Under ‘Crystal/Monocrystal Conductors’, we mentioned that ‘monocrystal conductors are highly prized for high-purity/high-accuracy applications’. The man who successfully developed the manufacturing process that makes it possible to fabricate monocrystal wires is Dr Atsumi Ohno, and his famous process is called Ohno Continuous Casting, typically abbreviated ‘OCC’, not to be confused with the familiar psychological acronym, OCD.

Plugs, Lugs & Jacks for analogue audio cables

Audio cables use a wide variety of connectors, with certain connectors optimized for interconnects and others for speaker cables. When thinking about connectors it is helpful at times to remember that for plugs and lugs there is always a corresponding jacket, socket, or terminal to complete the connection.

Banana plugs and jacks: Banana plugs are extremely popular as terminations for loudspeaker cables. (The spring-loaded connector surfaces of the male Banana

plug look somewhat like miniature, metal ‘bananas’—hence, the name.) Banana plugs typically connect to loudspeaker cable-binding posts that, by design, have banana jacks bored into their outer ends. Banana plugs are very easy to use, allowing simple push-to-connect, pull-to-disconnect operations.

Banana plugs typically make a ‘press-fit’ connection with their associated sockets. Note, however, that some banana plugs are ‘locking’ designs, with thumbscrews that, when tightened, clinch the plugs for an extremely tight fit within their jacks.

BFA connectors: Built For Audio/British Federation of Audio terminations are a variation on the theme of the 4mm banana plug (effectively built inside out and coated in ABS), designed to express safety concerns raised because the similarity of this plug to the live and neutral terminals in a EU ‘Schuko’ AC terminal. The 4mm banana plug is (notionally at least) ‘banned’ in the EU, which is why amplifiers include little red and black inserts that prevent their use, but you can remove these inserts with a penknife and continue to use banana plugs as before.

BNC connectors: Male BNC (Bayonet Neill Concelman) connectors are sometimes used on coaxial interconnect cables for use with components fitted with female BNC connectors, although BNC interfaces are relatively uncommon in high-end audio applications and components. Male BNC connectors use a quickconnect, quick-disconnect, twist-to-lock collar or ‘nut’ that latches on to two bayonet locking pins found on the female BNC connector.

BNC connectors are desirable in settings where it is important (or even critical) that cable connections do not work loose and where a ‘fail-safe’ locking mechanism is therefore required.

RCA plugs and jacks: RCA plugs are the de facto standard terminations for analogue interconnects and for coaxial S/PDIF digital interconnects. Corresponding, RCA jacks are the standard socket fitments for single-ended analogue and coaxial S/PDIF interfaces on audio components. RCA plugs provide a central post, carrying +/- audio signals, and an outer sleeve that serves as a ground, or ‘return’.

As with banana plugs, RCA plugs make press-fit connections with their associated sockets. However, many audiophile-grade RCA plugs feature ‘locking’ mechanisms, most of which work on the principle of firmly clamping the plug’s outer sleeve against the mating surface on the RCA jack.

Spade lugs: Spade lugs vie with banana plugs as the most popular terminations for loudspeaker cables. Spade lugs, as their name suggests, look almost like miniature, metal garden implements. Typically, spade lugs provide a sturdy wire receptacle at one end (where the cable’s conductors attach to the lug), and a flat, thick, two-pronged metal connecting surface at the other end, which is designed to fit around the central shaft of a traditional loudspeaker binding post.

Loudspeaker cable-binding posts have threaded metal shafts, traditionally fitted with beefy metal locking nuts or collars. To make firm connections using spade lugs, one would first back off the binding post’s locking nut, then insert the spade lug so that its prongs fit on either side of the binding post shaft, and finally tighten down the locking nut or collar as firmly as feasible to clinch the spade lug in place.

Some contend that spade lugs offer inherently superior connections to banana plugs owing to their robust construction and large surface area, but one point to bear in mind is that it takes two hands to connect spade lugs properly—one hand to hold the spade lug in place against the binding post shaft while the other tightens the locking nut. Also, users should be aware that—depending upon cable positioning—the weight of the speaker cables can apply torque on the spade lugs, causing the binding post locking collars to become loose over time.

XLR connectors: XLR connectors are the de facto standard for use in all types of balanced analogue and digital interconnects. In traditional, loudspeaker-based audio systems, the most common variant would be three-pin XLR connectors where, as noted under ‘Balanced Interconnects’ and ‘Digital Interconnects’ above, one pin carries the + signal, another carries the – signal, and the third serves as the ground, or ‘return’.

By convention, three-pin XLR output jacks provide a socket with three outward-facing pins, while XLR input jacks provide a socket with three pin receptacles. To accommodate this convention, XLR cables are invariably set up with different connectors on each end, with a distinct signal input end (providing receptacles for the pins from the audio component’s XLR output socket) and a signal output end (providing outward-facing pins that plug in to the receptacles of the audio component’s XLR input sockets). Virtually all XLR sockets and plugs features spring loaded mechanical latches that lock the connectors firmly in place (typically the latches feature thumb-actuated release catches).

In headphone-based systems, however, one might encounter both three-pin or four-pin XLR connectors, where the four-pin variant is a stereo (two-channel) connector, providing two sets of +/– connections pins. Some higher-end headphones ship with balanced signal cable sets terminated either with dual three-pin XLR connectors (as used, for example, on the Abyss AB-1266) or with single four-pin XLR cables (as used, for example, on top-tier Audeze or HiFiMAN headphones).

“If you could only improve one cable in your entire system, it should be the mains cable that runs from your wall sockets.”

Power Cords or Mains Cables

You might think all mains cables are created equal (or nearly so), but in our experience, high-performance mains cables can and do have a profound effect on sound quality. Indeed, several leading-edge cable designers would say that, if you could only improve one cable in your entire system, it should be the mains cable that runs from your wall sockets to whatever power distribution component you choose to use.

The key differentiators between ‘garden-variety’ power cords and the high-performance models we recommend include: higher gauge conductors, conductors fashioned from superior and very high-purity materials, more sophisticated dielectrics, superior internal geometries (often focused on blocking noise), superior shielding schemas (again, focused on blocking noise), and ultra high-quality plugs at both ends of the cables.

Purity of Conductors

High-purity conductors are thought to have a direct and significant impact on sound quality and for this reason a number of purity-related acronyms and terms have come into play. Here are three you might encounter frequently:

HPC (high purity copper): Manufacturers who use copper conductors and have been selective in their choice of materials suppliers will often say that their cables feature HPC conductors. Caveat emptor: The term HPC implies that care was used in choosing sources of copper wire, but it does not tell you precisely how pure the copper actually is (although some manufacturers might clarify this point with additional specifications).

OFC (oxygen free copper): Oxygen is one of the most common ‘contaminates’ of pure copper, so manufacturers who have taken steps to source copper that is very low in oxygen content will often tout their use of OFC conductors. In many cases, references to OFC conductors will feature supplementary specifications to indicate the exact-level of purity.

‘Five-Nines’ or ‘Six-Nines’ conductors: These slang terms indicate levels of purity, expressed as, for example, 99.999% or 99.9999% pure metal, whether referencing copper, silver, or other metals. Obviously, more ‘nines’ describe conductors of higher purity, higher cost, and—it is thought—higher sound quality.

Skin Effect

Skin effect is the tendency for an alternating current (AC), or an alternating music signal, to flow or become concentrated mostly near the outer surface (or skin) of a conductor. The higher the frequency of the signal the thinner the functional depth of the skin being used to pass the signal, which means that the AC resistance of the cable tends to increase at higher frequencies. This is why some cables can exhibit a certain degree of treble roll-off.

Certain cable geometries (for example, woven Litz wire geometries) can, however, mitigate the problem of AC resistance increasing at high frequencies owing to skin effect. The point is that it pays to seek out cables whose designs minimize or eliminate skin effect problems in the audio range.

Snake Oil

A term used by consumers to describe products that involve technological principles that are not well understood by the consumer. Examples of such technologies include EMI, decimation mathematics, image creation in the brain, bandwidth of the ear, phase effects, pre-ringing and reference measurement parameters. Snake Oil is a term of approbation which strongly implies that what is not understood is not valuable, rather than focusing value judgements on results achieved.

Speaker Cables

Some audiophiles draw a distinction between ‘signal-bearing’ cables (namely, interconnects) versus ‘power-bearing’ cables (namely, speaker cables). Stated another way, speaker cables are responsible for delivering the

often high-wattage output of amplifiers to our loudspeakers and doing so with high bandwidth, minimum noise, low distortion and coloration, and maximum delivery of current as demanded by the loudspeaker.

To meet these demands, speaker cables place the same emphasis on geometries, materials, conductors, and noise-blocking shields as interconnects do, but with the added demand of being able to handle potentially very high levels of power (power = voltage x amperage).

Some speaker cable terms you may encounter are these:

(Internal) Bi-wire cable: A speaker cable that internally has double runs of conductors, with a single pair of +/- connections at the amplifier end and a double set of +/- connections at the loudspeaker end. In this configuration, the double runs of conductors are housed within a common sheath or jacket.

‘Shotgun’ cable: A speaker cable that provides double runs of conductors, each housed in its own sheath or jacket, where there is a single +/- set of amplifier connections and a double set of +/- connections at the speaker end. The term ‘shotgun’ comes from the fact that the dual-runs of conductions, each in its own jacket, look somewhat like the barrels of a double-barrel shotgun.