Editor's Note: Audiophile Style community member George Graves has kindly allowed us to publish his five part series on high quality audio recording. This series is a primer that many audiophiles will find interesting and educational. It isn't a treatise, textbook, or master class designed to cover every detail in depth. As a music lover and audiophile I want to understand a bit more about recording, but I don't want to become a recording engineer. This series is right in my wheelhouse, and I hope it's in yours as well. - CC

OK, you have chosen your recording medium, Digital Audio Tape, a laptop computer, a portable flash-memory recorder such as a Zoom H2 or H4n, or perhaps a DSD format Korg MR-1 or MR-1000. You have picked a suitable mixer or microphone preamps, and have bought yourself a few good microphones. Nice. Unfortunately, all of this fine equipment will do you little good unless you know how to use it. This week, we're going to look at how to use microphones to best advantage in a location recording situation. 

The Type of Music Dictates the Procedure

The first thing that has to be considered and understood is the fact that even though we humans tend to look upon microphones as surrogate ears, they really aren't. Our ears don't just pick-up sounds, they, with the help of our brains, both process and interpret sounds. We have the ability to isolate sounds from the background. That's why we can still listen to music in a noisy nightspot. We can pinpoint instruments in space, we can do any number of signal-processing functions without even thinking about it consciously. Still and all, we are applying intelligent signal processing to the sounds that our ears pick up. Microphones cannot do this. All microphones can do is pick-up, and convert to an electric current, whatever sounds occur within the space that the microphones occupy. This sound field is modified only by the characteristics of the microphone itself; it's frequency response, it's pick-up (or polar response) pattern, it's noise-floor characteristics and it's maximum sound-pressure level. That's it. Microphones make no "decisions" about what to highlight, what to suppress, or even what to include in it's pick-up. All of these decisions have to be made by you, the recording engineer, and they have to be made before the music starts. These decisions are a result of knowledge and experience. Here, I can start you down the correct path with some of the more universally applicable knowledge, and the experience, you'll soon pick-up for yourself by experimenting with the principles found here. 

The second thing to consider is that because of the above facts about microphones, different kinds of music require slightly different procedures to obtain the best results. Since the recordings that you are likely to make will be recorded in concert before live audiences, studio techniques usually do not apply. While classical music can benefit from a number different microphone arrangements and can give satisfying results from may different perspectives from relatively close-up, to fairly far back in the hall, jazz and pop music needs to be recorded close-up. Distant prospectives only tend to obscure the intricate play of instruments in jazz, and tend to make pop musicians seem far away. 

What We Want is Stereo

In the 1960's when multitrack tape recorders became widely available, a technique for recording that utilized 8, 16, 32, 48, 64, or even 96 tracks(!) gained popularity with many of the major record companies. This practice came from the world of rock-and-roll recording with their over-dubbs, sound-on-sound, sound-with-sound and other special effects. It was found that recording each individual instrument and voice to a separate, isolated track allowed for the flexibility to double-up on voices and instruments and add effects in post-production. This was a good move for pop music and can be heard to best advantage in the early works of guitar and recording technology legend Les Paul (with his then wife, Mary Ford). Listen to Ford perform a duet with herself on "How High the Moon" Rock innovators , The Beach Boys, used these techniques to great effect in their famous "Good Vibrations". But, as great as these multichannel techniques were for gimmicky pop and rock music, they were translated to classical and jazz recording for an entirely different reason and with very mixed, and often disastrous results. 

It was in the economic atmosphere of the late 1960's and the inflationary '70's where the idea of applying, to classical and jazz, studio multitrack techniques designed for pop music. Recording engineers like Bob Fine (Mercury) and Lewis Leyton (RCA Victor) used minimalist microphone techniques to capture classical performances so stupendously good that they are still revered by audiophiles to this day. In the jazz world, Rudy Van Gelder started recording the L.A. jazz scene with just a "portable" (that meant, in those days, that it had a luggage handle on it) tape recorder and single condenser mike. When he adopted  stereo, he still used only two or three microphones. His recordings, too are highly regarded, even today. But in order to record this way, it takes lot a work to set up the microphones to get the perspective just right. All the while, the players are there waiting for the engineers to make minute adjustments to their microphone placement. Somebody decided that this was expensive. Apparently, it  was much cheaper to set up empty chairs for the "eventual" musicians and place a microphone in front of each one. One microphone is erected for each instrument (and in some cases, for each group of instruments) and then when that is completed, you  called the musicians in, got the performance down on tape in as many tracks as possible and then sent the expensive musicians home. At that point the producer and the mix engineers can play with the balances between the instruments 'till their hearts' content. The result, when mixed down to two tracks, is, unfortunately, not stereo. It's exactly what it seems - multi-track mono. Jazz is almost worse. Each instrument is close miked, committed to its own separate track, and then mixed down to what is called "three-channel" mono. All the instruments are grouped into three groups and sent to either the extreme left, the center (by mixing the instruments equally into both channels), or the extreme right. Again, there is no real stereo image. 

But people hear stereophonically, or as the Greek word stereos denotes, three dimensionally. We not only hear musicians arrayed right to left, we also hear them front to back, and vertically. If the brasses at the back of the band are on risers, and thus higher than the players in front, the human ear can hear that, and in a real stereo recording, that information can be captured. One of the great disappointments of commercial recordings, is that so often, it's not captured. It's so easy to do that one wonders what commercial record companies are thinking. This is where you and I can better them. We, as amateurs,  can make recordings so lifelike as to be literally spellbinding. We want stereo, and this is how we're going to get it...

Real Stereo Recording

To record in real stereo, you theoretically need only two microphones. I say theoretically because this assumes ideal conditions. In less that ideal conditions, you might need more, but a good stereo-pair of microphones is the correct starting point. What is a stereo-pair? Well, basically it is two identical microphones arrayed in one of the classic stereo microphone configurations and placed in front of a group of musicians in such a way as to accurately capture the sound field that surrounds the performance. There are a number of ways to do this, and all can yield excellent results.

Coincident Pair

In the 1930's, both Bell Labs in the US and the BBC in England experimented with stereo sound. Bell Labs' famous experiments concluded that an accurate recreation of the stereo soundfield was possible with just two channels although it was first postulated that this would require many channels. They kept reducing the number of channels employed until only two remained. The soundfield was still stereo and still coherent. Two channels stuck. In England, a man named Alan Blumlein was working for the Columbia Grammophone Co,. Ltd. when he patented his "Blumlein pair" of figure-of-eight ribbon microphones where the mikes were mounted next to one another and and angled away from the center axis at 90 degrees to one another.

Diagram showing Alan Blumlein's co-incident Microphone technique from the early 1930's . His use of ribbon microphones creates a crossed pair of figure-of-eight patterns where the front lobes are in-phase and the rear lobes are out of phase. 

This basic stereo microphone technique is still used and is very viable, giving a sense of space and dimensionality that can make for stunning sound stage presentation. Today, this arrangement is most often referred to as a "coincident pair" and is usually done with cardioid pattern (unidirectional) microphones instead of figure-of-eight bidirectional microphones although the latter can still be used. This arrangement will not work with omni or non directional mikes (unless you want a monaural recording). 

Two small cardioid condenser microphones arrayed as a coincident pair. or modified Blumlien pair. Notice that the faces of the two mikes are set at a 90 degree angle to one another  and the two screens are almost touching. Notice also that the connectors on the mikes appear backwards. IOW, the right channel cable is connected to the mike on the left and vice-versa. This is because, obviously, the left-most mike is facing the right side of the pickup stage and the right-most mike is aimed at the left side of the stage, Playing with the angle from 90 to 120 degrees will widen or narrow the soundstage. 

X-Y Pair

The next stereo arrangement to consider is the X-Y pair. This is similar to the Coincident pair except that the mikes are not in the same plane. They are separated by about 7 inches (the distance between the two mounting points on a standard stereo 'T'-bar mount. Again, the point of departure is 90 degrees between the angle of the two microphone elements, but as in the co-incident pair, varying the angle between the mikes either broadens or narrows the stereo stage. Generally speaking, the X-Y arrangement is better for situations where the microphones must be some distance from the ensemble being recorded. Otherwise, if the mikes are right on top of the performers you're going to get the "spotlight" effect where one mike picks up too much of one instrument and not enough of another, ruining the imaging. If you must be close to the ensemble, or if the ensemble is small, the coincident method is better.  

Here we see the same two small cardioid microphones arrayed as an X-Y pair. This is an acceptable method for miking orchestras, bands, and solo concert grand pianos. Give the mikes a little space to form a coherent image and to pick-up a bit of hall ambience for best results

Middle-Side or MS Miking

M-S miking is hardest stereo microphone technique for most people to understand. It generally consists of two microphones that are coincident with one another (in this case it is imperative that they be arrayed one above the other, or ideally, as a single-point stereo mike such as the aforementioned Avantone CK-40. M-S requires that one of the two mikes (since they are aligned on the same axis vertically, it doesn't matter which) be set to the cardioid (or the omnidirectional) pattern and the other must be set to a figure-of-eight pattern. The cardioid (or omni) mike is aimed forward at the center of the ensemble as if one was making a monaural recording. The second mike is facing 90 degrees from the center of the cardioid mike's pattern. This means that the two lobes of the figure-of-eight are pointing to each side of the sound stage. Obviously one of the figure-of-eight's lobes is in-phase and the other lobe is out of phase. The cardioid or omni middle or 'M' mike is fed into one channel of the microphone mixer and is pan-potted to center, and the figure-of-eight or 'S' mike is split into two feeds and routed to two separate microphone inputs where S+ is pan-potted to the extreme left and S- is pan-potted to the extreme right. S+ is in phase and the S- is 180 degrees out of phase. The level on the cardioid mike is set normally and the two faders on the figure-of-eight mike are brought-up together. Obviously, if the faders on the figure-of-eight mike are closed, one gets pure mono. but as one brings-up the figure-of-eight side mike, the following matrix occurs:

Where M = the Middle or front-firing cardioid or omni mike and 
S+ = the in-phase lobe of the side-firing figure-of-eight microphone
S- = the out-of-phase lobe of the side-firing figure-of-eight microphone and R = right and L= left
L= M + (S+) 
R = M + (S-)

M-S or Middle-Side stereo microphone technique. Here, in the example, the front-firing or 'M' mike is an omnidirectional, but it can be a cardioid as well. Cardioid works best in a concert recording situation because it minimizes audience pick-up. 

The advantages of M-S miking are several. First of all, an M-S miked recording is perfectly phase coherent and therefore mixes to mono (for mono radio, for instance) perfectly, without loss. More importantly (in this day and age) is that it lets the recording engineer control the width of the soundstage from his recording console. The more Side mike that's mixed-in with the M mike, the wider the pick-up. The less Side, the narrower the soundstage, until, with the Side mike killed completely, one gets straight monaural sound. 

Spaced Microphones

The last stereo technique is probably the most obvious: Spaced microphones. This is one of the oldest of the techniques used to make stereo recordings and indeed, the famous Mercury records of the 1950's and 60's as well as the early RCA Red Seal stereo recordings were made this way. Bob Fine of Mercury started recording in the early '50's with a single omnidirectional condenser mike placed in front of the band or orchestra. When stereo came along he added two more omni-directional mikes for a total of three. At first, the center mike was just for mono compatibility (in those days, record companies put out dual inventory. Releases would be available as both stereo and mono titles). Fine would use the center track to cut the mono disk, and the two flanking ones to cut the stereo. At some point, he decided to mix the center mono track equally into both the left and right channels. This gave better center fill on the recordings and if you listen to one of the SACD multichannel releases of some of Fine's three-channel recordings (with a third speaker and amp, preferably identical to the your other two) you would be able to hear how well this works. Most of us can't afford to do that with the price of equipment these days, but I have heard it at an audio dealer and it is impressive. Mostly spaced arrays are done with omni-directional mikes, but it can be done with cardioids. With omni's it is possible to place the mikes farther apart than with cardioids without worrying about a "hole in the middle"effect. Under the best of circumstances, spaced mikes can give a pleasing result, although I've never heard spaced mikes give a pin-point stereo image like the other stereo techniques discussed here. The main reason why they were used in the "golden age of stereo" of the '50's and '60's, and continue to find some favor with record producers, even today, is the fact that omnidirectional microphones tend to have much flatter frequency response and and a lot better low end than do cardioids or figure-of-eight microphones. 

A spaced array pair of Microphones 

Spaced pairs work because instruments located halfway between the two mikes are picked up equally by both and therefore, during playback, locate themselves between the two speakers. The left microphone picks up the sounds from the left side of the ensemble and the right side of the ensemble with equal intensity, but the sound originating on the left, that is to say, closer to the left microphone, is received earlier than the sound originating on the right side of the ensemble (and vice-versa). This is what gives spaced pairs their stereo ability. In coincident and X-Y setups the differences are both in intensity and phase, and this is what gives the stereo effect. In terms of absolute location the human ear responds to phase and intensity differences more than it does to time delay. This makes the coincident methods better at presenting an accurate soundstage. 

Placing Microphones

Placing microphones is somewhat of an art. I know recording engineers who have been doing this for years and they still don't "get it". We're going to assume for sake of this discussion, that the ensemble being recorded is a college band. We are also going to use an X-Y pair of cardioid microphones. 

The first thing we need to do is to see where the band will set-up. Hopefully the chairs will be already arranged or you'll have to wait for the members to arrange them. Then, obviously, you have to find the center of the ensemble (usually where the conductor stands, but not always). This is where the stereo pair goes. At first it will help to have the mikes mounted on the T-Bar and to hold them in your hand pointed at the band. Set each mike 45 degrees off of straight-ahead, giving you 90 degrees between each mike "head". Now, sort of "sight" each microphone by looking over the top of the mike head to see where it's pointing. Each mike should bisect the half of the ensemble that it's looking at. In other words, the left mike should be pointing halfway between the outside, stage left edge of the band, and the center or the band, and the right mike should be pointing halfway between the outside, stage right edge of the band and it's center. If it doesn't, step either forward into the stage of back away from it until that line-of-sight is achieved. If, for some physical reason (like the front edge of the stage), you can't get that far back, you can move the mikes further than 90 degrees apart to achieve that sightline, but realize that if the angle is too acute (say, more than 120 degrees), you might get a dead zone in the middle where there is insufficient coverage. On playback, this will give you the dread "hole-in-the-middle" and will destroy the stereo image. Avoid the temptation to adjust the pair to less than a 90 degree angle. This tends to lessen the the stereo effect. If you find that you can only get the proper sight-line by going less than 90 degrees, change your microphone pickup technique from X-Y to coincident. That should give you the stereo pattern you want at a closer range.   

Ideally. the microphones need to be behind the conductor, and should peak over his head. Microphones do rather well looking down on the ensemble being recorded. The best positioning ploy is to hang them from the proscenium. This can often be done if the band is playing in an auditorium with a formal stage and stage overhead. Often you can affix the microphones with nylon fishing line and sometimes you can get the auditorium's stage crew to help you. The downside of this arrangement is that it takes a lot of microphone cable to go from the mikes up to the stage fly-area and then over to the side of the stage and down to your equipment. Best results are achieved with the mikes about 5-7 feet above the conductor's head and slightly behind him.

Most of the time, though, you won't have the luxury of being able to "fly" the microphones and you will need stands. This can be awkward especially if the band is situated so far downstage that the conductor's position is near the front edge of the stage. Then you'll have to place your stand on the floor in front of the stage, and in so doing, lose about three feet of your stand's height. This is why it is necessary for you to have access to a really tall studio stand with a counterweighted boom for your mikes. 

OK, let's assume that you have placed your stereo pair of mikes as indicated above. Are you finished? Not at all. Some instruments are going to need highlighting. 

Highlight Microphones

Sooner or later you are going to come up on a situation where there are vocal soloists, or where some instrument is not being picked-up well enough to register properly in the recording. Another situation where a single stereo pair isn't good enough is when recording, say, a small jazz ensemble such as a trio or quartet in a club venue. All of these situations are going to require highlight microphones.

As daunting as this might, at first seem, It doesn't have to be if you keep a couple of hard-and-fast rules in mind. First of all, think about what you are trying to do. You are trying to augment the pick-up of a single instrument or voice against the overall stereo backdrop provided by the main stereo pair. You are not (in most cases) trying to make that instrument or voice dominate. Remember, while our ears can select and home-in on a single sound against a background of other, perhaps louder sounds, the microphone cannot. If, through your monitoring headphones, you cannot hear a particular instrument or voice, or, if that instrument of voice seems swamped by the rest of the musicians, then this instrument or voice is a prime candidate for highlighting. 

Secondly, you want this highlight mike to pick-up as little of the instruments surrounding the highlighted instrument or voice as is possible. This is controlled in three main ways. First of all the highlight microphone needs to be a cardioid mike. Omnis and figure-of-eight mikes cannot be used because, obviously, they pick-up sounds from more than one direction. Next, you can control the pickup by proximity. Place the cardioid mike as close to the instrument (or voice) in question as possible. Because cardioid mikes tend to accentuate low frequencies when they are placed close to the sound source, you might want to invoke the low-cut filter on the microphone - this is especially important if we're talking about a male vocalist or a piano. Lastly, you control the pick-up of adjacent instruments by bringing up the highlight microphone's gain just enough to fit-in with the rest of the ensemble, and no more. We want to hear it, but we don't want it dominate or call attention to itself (unless it's a soloist, of course).  You will need to pan-pot the highlight microphone visually to the same place on the left-to-right sound field as the instrument actually occupies in real space. If you don't, the pickup from the stereo pair and the pick-up from the highlighted mike will clash, confusing the image. Never, stereo mike highlights. if you do that it will bring the instrument or voice forward to stage center and again, will confuse the stereo image and even cause phase problems which can be very unpleasant to listen to. These "rules" are, as you can see, just common sense, and with experience, the correct methodology will make itself clear to you. 

In this recording, you can see the overall stereo mike in the right edge of the picture. Less noticeable, probably. is the small "lipstick" cardioid mike just inconspicuously peeking over the edge of the piano next to the piano's lid strut. It was apparent in the headphones during this rehearsal in the actual auditorium where the performance was to take place, that the piano was not projecting well enough, even though the open lid is pointing directly at the stereo mike. 

Recording A Small Local Ensemble

The easiest thing for amateur recordists to get permission to record is a small ensemble of jazz or pop musicians playing in local bars, coffees houses, restaurants and other public night-spots. Usually, it's not necessary to get permission from the venue management (why would they object to a couple of more pieces of paraphernalia in the musical group's "space"?), but it is vital that you get permission from the musicians themselves. Once the musicians have okayed you recording them (about which more, later), it's not a bad idea to introduce yourself to the venue management as the group's "recordist". 

Sometimes a little "background noise" consisting of people talking, glassware rattling, and in general, those sounds that accompany a restaurant, bar, or coffee house will add a bit realism and intimate ambience to a recording of this type, but, you don't want too much. This means that you will want to mike each instrument as closely as possible and separately in addition to your overall stereo pair (as opposed to instead of it). Many will tell you to eliminate the stereo pair altogether and just mix the individual instrument's mikes together. This is a personal choice, of course, and this will work. But you will end up with a "three channel" mono recording instead of stereo. 

For this example, we will assume an overall stereo pair and individual mikes on a trio consisting of a piano, a saxophone and a string bass. 

Showing arrangement for a small ensemble recorded in a restaurant, bar or coffee house.  Ensemble consists of piano, saxophone and stand-up bass. Notice that the stereo pair occupy mike mixer inputs 1 and 2, while the three mono highlight mikes occupy inputs 3, 4, and 5. Highlight mikes are cardioid and are moved close to each instrument to avoid pickup of audience noise. 

As can be seen in the illustration, above, a stereo-pair is still used to give overall ambience and stereo imaging. The highlight mikes are cardioid and placed as close as physically possible to the instruments. The saxophone mike can be troublesome because most saxophonists move around a lot when they start to "swing". If you can get the sax player to stand still, facing the microphone, good for you. Otherwise you might have to back that mike off a bit, physically and use more gain. Still, you are going to get a much closer pickup than the stereo pair will produce. 

Correctly pan-potting the three close-up or highlight mikes is crucial to maintaining the proper perspective. Of course, the stereo pair is panned far left and far right. The piano, which, in the example above, is on the left, is likewise panned far left. The sax, in the center of the group, is panned to the center, that is to say, equally in both the left and the right channel. and finally, the string bass is panned to the far right. 

As far as levels are concerned, you don't want the highlight mikes "louder" than the overall stereo pair. My advice is to start at about 25% the level of the stereo mikes. You will likely have to trim each from that setting to get a proper spread and perspective. Ask the musicians to give you some levels, either individually or together before you start to record. If they plan to play more than one set, you might choose to throw-away the first set for experimental purposes. Be sure to explain that to the musicians. If they are accommodating enough to give you levels before you record, then you are good to go. If you plan to record this group more than once, make note of the final levels and especially where the microphones are placed relative to the players so that you can repeat the setup. 

If you want to eliminate the stereo pair, as an experiment, you might actually like the result. Ever since the dawn of the stereo era, it has been traditional to record jazz in what is called three-channel mono where everything is panned into three distinct groups: left, center, and right. Of course, the center channel is a "phantom" channel because we are recording to only two actual channels, but it works fine. What you'll get is a really close-up presentation with little of the venue's hall sound or any of the noise resulting from an audience who will be eating, drinking and talking.  Just don't try this with a classical chamber group. It will sound awful (you wouldn't be recording a chamber group in a bar or restaurant anyway). 

Getting Permission to Record

Finally, a word about getting permission to record. The most important thing to keep in mind when approaching musicians is that you can give them something they want: A CD of the group performing. Promise them that right off the bat. tell them that you will give them each a CD of that evening's performance and that your services are gratis. It's usually enough to get you in. You should also promise that the recording you are making is for your own use and will not be used for any commercial purposes. I have actually made up a general contract that I take with me. If they seem reluctant, pulling out a contract that binds you to deliver the CDs and to not use the recordings for any purpose other than your own enjoyment, might turn a "no" to a "yes". Do a good job, deliver the CDs to them in a timely fashion, and you might find yourself getting calls from other groups. Musicians usually don't play exclusively with one group. One musician might play with this band one night and that band another. If you're good enough, word will get around. 

Editor's Note: Audiophile Style community member George Graves has kindly allowed us to publish his five part series on high quality audio recording. This series is a primer that many audiophiles will find interesting and educational. It isn't a treatise, textbook, or master class designed to cover every detail in depth. As a music lover and audiophile I want to understand a bit more about recording, but I don't want to become a recording engineer. This series is right in my wheelhouse, and I hope it's in yours as well. - CC

One cannot make an audio recording of a live performance without a microphone. It's that simple. For stereo, you need two identical microphones. You also may need more than that for highlights and ancillary ensembles such as vocal choruses and soloists. 

Ask any ten recording engineers what microphone techniques they favor, and you'll get at least thirty answers. Everyone has their favorites, many are variations on the same ones, some might seem downright bizarre. But they must work, or they wouldn't be mentioned. When we get to that part of the discussion, I will tell you my favorites and explain all of the basic ones. Microphone placement is an art, an art that is learned through experience. But before you can fly, you have to walk, so we will start with microphone types suitable for recording.

Microphone Types

There are two basic types of microphones used in recording today, the condenser mike and the dynamic mike. Years ago, amateur recordists used several other types. One type was called a piezoelectric mike, often referred to as a crystal or ceramic mike and the fourth type of microphone that recordists used to use was the carbon-button mike, ubiquitous in the 40's, 50's and even the 60's neither of these latter types are used any more. 

Dynamic Mikes

Of the types of mikes used today, probably the easiest to understand is the dynamic mike. These microphones work by moving a membrane connected to a coil of wire in a magnetic field - like a loudspeaker in reverse. The sound strikes the membrane (or diaphragm) which causes it to vibrate in sympathy with the sound striking it. This causes the coil of wire to move with it. The coil, cutting across the lines of force set-up by a permanent magnet, induces a minute electric current to flow through the coil wire. This current is analogous to the sound waves striking the diaphragm. This current is amplified and can then be sent to a public address system or to a recording medium. Dynamics of this kind are often used as PA mikes or vocalist mikes. They are valued for their ruggedness and dependability and simplicity. 

A variation of this principle is called a ribbon mike. Instead of a diaphragm and coil, the ribbon mike uses a long, thin ribbon of some lightweight foil. The ribbon is usually folded like an accordion to get more surface area. The ribbon is suspended between the poles of a very powerful magnet. When the ribbon vibrates in reaction to being hit by sound waves, it, again, cuts across the lines of magnetic force and a weak current flows along the ribbon. Both ends of the ribbon element are connected to a transformer which steps up the tiny voltage created by this current flow and matches the ribbon's impedance (which is extremely low, usually less than an Ohm) to the microphone cable and amplifying circuit. Ribbons have such feeble output that they require a lot of amplification before they are useful. Even so, throughout most of the history of commercial broadcasting, ribbons were the preferred microphone for most radio stations. The RCA 44BX microphone, which is usually the mental image most people conjure-up when they think of a microphone, is, in fact, a ribbon mike.

The famous RCA 44BX. For almost two decades this mike was the voice or both radio and television broadcasting and most recording done here in the United States. To many people, this is what they think of when someone says the word "microphone". 

Ribbons are almost always bi-directional and pick up sound equally from both the front and the back. Due to the thinness of the ribbon, they are easily damaged by blowing into them or using them out of doors on a blustery day. While ribbons like those from Royer can be made with outstandingly flat frequency response (and concurrently high prices), most ribbons don't have a lot of top end. They are good for instruments with lots of midrange such as choruses and perhaps acoustic guitars. They are often valued for their very natural, warm sound and make beautiful announcer mikes (as long as the announcer doesn't blow into them).

Condenser or Capacitor Microphones 

Probably the most often used microphone for recording is the condenser or capacitor mike, as they are sometimes called. These mikes have always offered the highest performance and most accurate translation of sound to electrical current of any microphone technology. Although, condenser mikes have been around almost as long as has radio broadcasting and "electric" recording, really good ones, of the type we now associate with the technology, didn't appear until WWII Germany.

The Germans had developed two technologies during that war that absolutely dumbfounded the espionage types in England and other Allied listening posts. German radio stations would be broadcasting concerts of, say, the Dresden State Symphony Orchestra on nights when the Allies knew for a fact that Dresden (or Berlin, or Cologne, or Munich) was, at that moment. under heavy air attack. Two other characteristics of those "phantom" broadcasts puzzled the Allied listeners: The concerts were broadcast without breaks, tics and pops, and without the "scratchy" sound that usually accompanied transcription by phonograph record. The then only known method of recording sound for later broadcast. Add to that the fact that the recordings sounded so life-like and clear; they must be live!  It wasn't until after the war, when the Allies occupied Germany and went into their radio stations did they find the answer to this puzzle. The Germans had perfected the audio tape recorder, which, due to several German innovations (like AC bias for recording) turned out recordings that were unparalleled in their wide frequency response, low distortion and noise. The second thing they found was that the research arm of the German broadcasting ministry, Telefunken, in collaboration with the Georg Neumann company, had perfected the condenser microphone. Georg Neumann, another condenser mike pioneer,  had actually invented the modern condenser mike in the early 1930s. Called the Neumann "Bottle" it was used throughout Germany and developed all during the war. In 1947, Neumann came out with the famous U47 (also sold under the Telefunken name), which, along with the later U87 became the ubiquitous form factors for condenser Mikes. So much so, that most condenser mikes today, irrespective of what they cost or where they are made, look like one or the other.

After WWII, both Telefunken (Left) and Neumann (right) came out with the first successful condenser mikes for recording, the U47. Here in the USA, they unseated the RCA 44BX as the microphone of choice for recording and broadcasting, ushering-in the era of high-fidelity - just in time for FM broadcasting and the LP. 

Operation

The condenser mike works in one of two ways. Both types work because the diaphragm and the back plate of the microphone element form a small capacitor of about 40-50 picofarad. In the most common application, this capacitor is charged with a polarizing voltage which is used to control an amplifying device such as the grid of a tube, the base of a bipolar transistor or the gate of an FET. The sound striking the diaphragm varies the capacitance of the mike capsule coupling more or less voltage to the control element of the amplifying device causing that device to conduct either more or less and the resultant signal is analogous to the sound striking the diaphragm. The second method uses a similar capacitor to the above example, but instead of that capacitor controlling the coupling of voltage to the microphone circuitry, this second method has the capacitor as part of a tuned RF oscillator circuit. The audio, striking the diaphragm, changes the frequency of the oscillator, also as an analog of the sound being picked-up. At the output of the microphone, a detector circuit called a discriminator strips the RF frequency from the signal leaving only the audio modulation. This works a lot like an FM radio, and indeed,  is often called an FM microphone. While there are still some FM microphones being made, most condenser mikes are of the first type.

The biggest drawback to the condenser mike has always been it's requirement for an external power supply to polarize the capacitor mike capsule and to drive the electronics. Pre-solid-state, most condenser mikes relied upon dedicated power supplies and special cables with multi-pin connectors to connect the mike to the supply. These cables had to carry both the audio and the various voltages such as the B+ and filament voltage for the tube(s) as well as the polarizing voltage for the capsule. Generally speaking, the link from the power supply to the recording console was via standard XLR cables. Extension cables between the mike and power supply were non-standard and usually quite expensive. After condenser mikes became solid-state, in the late 1960's, the separate power supply was abandoned in favor of so-called phantom power. Phantom powering uses 48 volts (this came from the telephone industry) DC which is piggy-backed on the same wires that carry the audio from the microphone to the mixing console. The DC doesn't affect the audio and today, most mixers have the 48 volt phantom-powering power supply built in. More expensive mixers allow you to apply the phantom-power individually to each microphone input, while cheaper mixers merely allow one to turn it on or off globally to all microphone input circuits. 

The Electret Condenser Microphone

In the late 1960's after solid-state electronics became possible inside the microphone itself, the need for tubes and the high-voltages they require was obviated. This allowed practical application of a type of condenser microphone capsule called an electret. An electret is a stable dielectric material with a permanently embedded static electric charge. Due to the high electrical resistance and chemical stability of the materials used, the electret device will retain its charge for hundreds of years. Although the electret principle was discovered in the 1920's, it didn't become practical until two Bell Labs engineers designed one using a thin Teflon coated metal foil as the electret in 1962. The combination of the two maturing technologies, the materials technology for the electret itself, and the use of the FET transistor for the amplifier, suddenly made this kind of microphone practical. Due to the fact that an electret condenser microphone does not need a polarizing voltage, and the FET amplifier in the mike draws very little current, suddenly, small, cheap condenser mikes which ran on batteries became possible. The biggest application for these mikes, became, of course, in telephones to replace the century-old carbon button mike. The improvement in voice quality was apparent to everyone who heard one of the new electret miked phones.

Schematic of a typical Electret Condenser Microphone showing how simple it is. Often, these mikes can be powered by a single watch battery yet provide long battery life and better performance than the cheap carbon, crystal, or dynamic mikes that they have replaced. 

It wasn't long before companies like Sony, in Japan, were applying electret principles to recording microphones. In the early 1970's, Sony introduced the ECM-22p a professional quality electret that could be powered by either a nine-volt battery or via standard 48-volt phantom-powering. The ECM-22p sported rugged build quality, and surprisingly decent frequency response. Sony spec'd the mike at 40-15,000 Hz, but I've found that the two that I owned to be rather bass-shy. However they are great on the top end and midrange and make excellent drum mikes. After having owned them for almost 30 years, I can say that they still work as well today as the did when new (although the Eveready #206 9V batteries are somewhat hard to come by these days - thank Sony for making them standard phantom-power compatible).

Early pro electret condenser Microphone the cardioid Sony ECM-22P

Although a number of electret microphones for pro use have been made, most are small-capsule mikes designed for the low-end consumer market. When I was in Japan a number of years ago, I picked up a Sony electret single-point stereo mike (ECM-929) to use with my Walkman-Pro cassette recorder. This mike has been excellent with it's M-S pick-up pattern and adjustable soundstage width. But like most electrets, it's tiny capsules make it a bit shy on bass and definitely suited only for casual recording (see the second installment of this blog for a picture of this microphone photographed with a Sony MiniDisc recorder).  

Modern Condenser Microphones

For most of the post-war era, condenser microphones were considered the Rolls-Royce of microphones. They cost a bundle. Even today, A Neumann condenser mike like an M149 can easily cost close to $6000! When I was in the recording business in the 1970's I owned a pair of "cheap" Japanese condenser microphones from Sony called C-37Ps. These had an FET amplifier in them and a single capsule that was switchable between omni-directional (non-directional) and cardioid (uni-directional). They cost well over $1000 in 1975 dollars and were considered inexpensive compared to Neumann, AKG, and Beyer condenser mikes of the time. 

Sony C-37P FET Condenser Microphones were considered "inexpensive" at about $500 each in the mid 1970's. They are still very good microphones, by the way and command top-dollar when they show up on the used market. 

Enter the Chinese

From 1949 until the death of Chairman Mao Tse-Tung, China was a closed country. They did not trade with the west, and, essentially, had no consumer markets as existed in the west. Like the Russians, the Chinese reverse-engineered (read that "copied") essential technologies from western manufacturers and simply made their own. One of these technologies was microphones. All the best condenser mikes had "Chinese copies" and they were used in both broadcasting and recording. When the Bamboo Curtain fell in the 1980's, it was found that these copies of Neumanns, AKGs, Telefunkens, Scheops, Sennheisers, Beyers et al, were actually not half bad, and available for mere pennies on the dollar compared to their western counterparts. 

As soon as trade agreements with western companies became possible, many people decided to have their own microphone designs built in China. That brings us to the present glut of excellent and cheap condenser microphones which not only are well made, but actually perform very well. It is possible to buy Chinese-built microphones from firms such as Avantone, Behringer, Samson, Rode, and many others. 

Most of these mikes are better than the classic mikes from which they are copied. The reason is that the classic Neumann and Telefunken models (not to mention AKG and Beyer) had acid-etched brass diaphragms, which, while thin and light by the standards of their day, are today, so massive that they gave these microphones a peaky, rising top end which can sound harsh, especially when used to record digitally. Modern condenser microphones, including Chinese ones, have diaphragms made from a thin Mylar plastic which has been "sputtered" with an atom-thick coating of either aluminum or even gold. The metal coating on the Mylar makes the diaphragm a conducting capacitor plate without adding any weight. The resulting diaphragm is so low in mass that it's fundamental resonance (the characteristic that gives the older mikes such an aggressive top-end) is pushed way up into the ultrasonic region of the audio spectrum, where people cannot hear it. Modern mikes, even inexpensive ones, therefore tend to have a smooth, clean sound that shames most older designs. 

One Chinese made mike that I have used and found to be just about the best mike I've ever used  is the Avantone CK-40. This stereo mike has switchable patterns between omni-directional, figure-of-eight, and cardioid and is actually two microphones in one case. The top element can be rotated either left or right 90 degrees (for a total of 180 degrees relative to the lower element). Physically, it is a close "copy" of the famous (and fabulously expensive) Telefunken ELA-M-270 from the 1950's which is still made and can still be purchased new from Telefunken USA for a mere $16,000. The CK-40, on the other hand lists for about $600 and instead of being tubed like the Telefunken, features an ultra-quiet FET preamp. 

The Avantone CK-40 (right) can be said to be a virtual "Chinese copy" of the famous Telefunken ELA-M-270 (left shown without its shock mount). Having used both an original ELA-M-270 and the CK-40, I can tell you that while both microphones are good, the Avantone is much better and, in fact, is one of the best sounding microphones this writer has ever heard.

Choosing Microphones for Your Own Recording Set-up

There are a number of characteristics that one needs to keep in mind when choosing microphones for recording. Generally speaking, the larger the capsule diameter, the better the bass. Most decent condenser mikes, these days have at least a 1-inch capsule and I would consider this a minimum for any mike that will be used for general coverage of an orchestra, symphonic band, or any ensemble with a wide range of instruments needing a solid low-end foundation. Affordable microphones that meet this criteria are, happily, fairly abundant. The Samson CO1 at around $80 is an excellent entry into this type of so-called "big-capsule" microphone as is the SM-ProAudio MC01. Both of these cardioid-only microphones are excellent performers and superb values. For multi pattern mikes, the $150 Samson CL8 is a very good pick and offers a choice of cardioid, figure-of-eight, and omnidirectional patterns. Also in this range is the excellent Behringer B-2Pro. All of these mikes have excellent, wide frequency response and solid bass performance. Of course, I cannot heap too much praise on the aforementioned Avantone CK-40 stereo mike. This dual-head single-point stereo mike has large 35mm diameter capsules and among the best low-end performance that I have heard. When coupled with this mike's smooth midrange, clean, flat top-end and wide dynamic range. it's hard to beat - at any price.

Behringer B-2Pro multi-pattern, large capsule condenser microphone shown with included accessories

There are always going to be situations where you are going to need more than just a single stereo-pair of microphones to get proper coverage of the ensemble you are recording. When recording a jazz big-band recently, I found that the stereo pair was not picking up enough of the piano. Listening to the ensemble play from the audience perspective, I could hear the piano, but when listening to the mike feed on headphones I could not. This necessitated the use of an auxiliary microphone on the piano and mixing the result into the overall pick-up from the stereo-pair. To do this I used a single cardioid SM-ProAudio MC01. This mike was placed low so that it just "peaked" over the edge of the grand piano and was aimed at the center of the raised top. I pan-potted the mike to the extreme left (the piano was on the extreme left side of the band) and raised the level on the mixer input so that the piano could just be heard in the headphones. The results were perfect.

This means that you are going to need more than just one pair of microphones. It is not necessary that these mikes have the bottom end of the main mikes as they are usually just for accent. Sometimes there will be vocalists involved and these mikes will work fine for that as well. A typical microphone complement for a modest amateur recording "kit" might be:

1- Avantone CK-40 single-point stereo mike (or equivalent such as a pair of Behringer B2 Pros on a 'T-bar'.
2- Behringer B2 Pro multi-pattern microphones (or equivalent switchable pattern mikes).
2- Samson CO1 cardioid large capsule microphones or M-ProAudio MC01(or equivalent)
2- (a matched  pair) of "lipstick" small capsule microphones (Behringer C-2, C-4 or equivalent)

Author's Microphone complement for location recording. Each mike has proved itself to be rugged, well made, and a good performer. 

Needed Accessories

Missing in the above pictures are two very important components. The first is microphone cable. Since you never know where you are going to end-up, it is important to always have enough. My rule of thumb is to carry 4 -50' lengths, and eight 25' lengths. This looks like a lot, but I'd rather carry too much than too little. It is rare that you'll ever need more than three or four microphones for any one gig, but you never know what you might run into or how far away from your microphones that you'll be setting-up. 

Cables

Now, about cable. Naturally, it needs to be balanced microphone cable with decent quality XLR connectors; a male on one end and a female on the other. Other than that, if you buy from a reputable source such as Shure or Hosa, you can be assured of getting decent quality. Let me clear something up about cables right now. I spent years in the aerospace industry as a wiring/cable engineer for some of our country's most leading edge rockets and satellites. I have studied wire thoroughly and I can tell you this, just between you and me: at audio frequencies, wire is wire. Oh, I know, you can spend a fortune on cables from high-end audio cable companies, but it's all bling; stuff and nonsense. No double-blind cable test has ever revealed any difference whatsoever between expensive and cheap cables. Audio cables have no sound of their own. So buy well made, reliable cables from reputable sources and don't worry about the rest. One thing is important, however, and that is to use runs as short as possible and keep the number of connectors between the console and the microphones as low as possible. In other words, one long cable is better than two or three shorter ones connected together. 

Mike Stands

Now you need something to set the mikes on. Floor stands are fairly cheap, and the best kind for location recording are the folding kind. I use four of the Euroboom OS13 stands. These microphone stands have a folding, three legged base and a column which raises to about 63" and a boom that gives another 33" of extension. They weigh only about 5 pounds each, fold very compactly and can be bought online for about $30 each. For the big (and heavy) stereo mike, I use either a Euroboom with sandbags on the legs (for stability) or I use the foldable StuBoom from On-Stage ($120). This stand extends to 80" and the counterweighted boom gives another 82" of extension. This stand is heavy, and large. I only use it where I have the room for it and I don't carry the boom very often because it won't fit in my car very comfortably (being almost 7 ' long). But it is there when I need it. Check around, you can find some very decent mike stands out there. If you have a music store in your area, you might ask them to be on the lookout for used stands for you. I once bought a huge studio boom with a cast iron base for $20 at a local music store. New ones are almost $300 today. I ended up giving it away because I couldn't easily carry it around. Deals are out there.

Next time we'll talk about microphone pick-up patterns and how to mike various instruments for best effect.

Editor's Note: Audiophile Style community member George Graves has kindly allowed us to publish his five part series on high quality audio recording. This series is a primer that many audiophiles will find interesting and educational. It isn't a treatise, textbook, or master class designed to cover every detail in depth. As a music lover and audiophile I want to understand a bit more about recording, but I don't want to become a recording engineer. This series is right in my wheelhouse, and I hope it's in yours as well. - CC

Part One: Commercial Recording Quality (link)

Part Two: Recording Media (link)

Part Three: Microphone Mixers and Pre-amps

This time we are going to discuss how to get the microphone signal to the recording device. For this discussion, we are going to assume a two-channel stereo recording, but, keep in mind that these things apply to multi-channel recordings as well (whether they are mixed to two-channels for production or to 5.1 channels or any other surround-sound format.)

Professional quality microphones, whether condenser, or dynamic have one parameter in common: they all need to be amplified before they can produce a recordable signal. Usually, microphones need between 40dB (100X amplification) and 60dB (1000X amplification) to get their signals up to what is regarded as "line level" which is required to give maximum record volume. Some microphones, such as ribbon mikes have such low output that sometimes as much as 70 or even 80dB of gain is required. This means that pre-amps required for microphones must have the following characteristics: They must be quiet, have low distortion, lots of headroom and wide, flat frequency response. They also must have differential inputs to take professional "balanced" microphone cables.

Noise, is a major factor here. At these levels of amplification, the "self noise" generated by active components such as bipolar transistors, FETs,  Integrated Circuit (IC) operational-amplifiers (op-amps) and vacuum tubes becomes a major factor in the quality of the finished recording. Much of this self-noise is thermal. Active electronic components use the physics of electron attraction and repulsion to move signals around and to amplify them. Moving electrons through the devices creates a certain amount of random, or non-correlated noise (along with heat). In fact, vacuum tubes (or valves as they are sometimes called) operate by heating an element inside the tube hot enough to actually "boil" electrons off of its surface. The rest of the tube's elements (the grid(s)) control the flow of those electrons and actually determine how many of those boiled electrons get passed it and on to the plate where they constitute the amount of current that the tube conducts; I.E., less current flow represents small signals, more current flow represents larger signals. Since all of the tube's amplifying ability comes from the fact that it is a heat-operated device, the amount of random electron flow, and thus noise, is characteristically quite high.

Solid state devices work differently, and while they still create heat it is much less than a tube, and rather than the heat being the method of operation for these devices, it's more of a by-product with them. While noise is still an issue, generally speaking, solid-state electronics are much quieter than tubes. Does this mean that tubed electronics cannot be used to make modern digital recordings? Not at all. The reason is because there is another way to get voltage amplification of a microphone signal; transformers. Voltage gain, in a transformer, is largely a matter of the turns ratio of the transformer's coils between the "input" (the primary coil) side of the transformer and the "output" side (the secondary coil). As an example, if you put 10 turns on the primary windings (coil) of a transformer, and 100 turns on that transformer's secondary windings and apply a one-volt AC signal to the primary, you will get about 10 volts out of the secondary. This is an oversimplification, but it does show how the voltage gain of a transformer is determined by its turns ratio. Since the transformer is not an active component, it adds no noise, but it will amplify any noise in the signal applied to it right along with that signal. There is no free lunch, after all. So if we take the output of a good mike with decent noise characteristics, feed it into a transformer before applying it to an active microphone preamp, it is possible to get by with far less gain in the preamp itself. Less gain equals less thermal noise being added to the signal making it very feasible to use tubes in modern microphone preamps and still get signal-to-noise ratios that are compatible with even high-resolution digital recordings. Of course, there is a downside with transformers. Good ones, which have flat frequency response across the entire audio spectrum are expensive. Transformers also have problems with maintaining phase integrity at all frequencies and especially have problems coupling low frequencies through them. Most modern solid-state microphone pre-amps and mixers don't use transformers, but use a type of input circuitry called a differential amplifier. These are very good at rejecting noise that is common to both legs of the balanced interconnects from the microphones. These include, hum, air conditioning spikes (when the compressor cycles), noise created by the proximity of light dimmers in auditoriums and other public venues, etc. This ability is called common-mode rejection.

These days, a decent microphone preamp will give signal to noise ratios of somewhere in the region of  about -125 to -130dB, the greater the number, the quieter. Any microphone preamp with figures in this region will give recordings that are, for all intents and purposes, essentially silent. Even with the playback gain set quite high, one should hear nothing but blackness when no instruments are playing. 

Another characteristic of microphone preamps to consider is headroom. Live music can get quite loud. Cheap preamps can clip (distort) when fed a microphone level that's too high. Modern condenser mikes can handle sound pressure levels of as much as 150dB before clipping. It would seem like it would be nice to have a microphone preamp that had similar characteristics. Thankfully this isn't necessary. All microphone inputs have controls on them to vary the amplifier's gain and most have a light on them to indicate when that mike channel is clipping. A rule of thumb here is that the louder the source is playing. the less gain is required from the pre-amp. It is always advisable to ask the ensemble's leader to have them play the loudest part of their program before the recording starts so that you can advance the gain to clipping, and then back-off until the clipping indicator light goes out for all microphones. Then, back-off a bit more - just in case somebody wasn't really playing their loudest. This insures the highest possible signal-to-noise ratio without worry of overdriving the microphone preamps. Some recording engineers use a calibration box on each input to feed a signal of known amplitude into each microphone preamp. This might work in a studio situation where things like room-loading and individual microphone characteristics are well known, but for location recordings, I'd rely on the actual musicians to tell me how loud they're going to play rather than count on some unrelated "standard". After all, conditions will vary from venue to venue and musical group to musical group. No two situations will ever be the same. Of course, if you are in a position to record the same group in the same venue time after time, experience will guide you in setting your microphone preamp gain. Ultimately, the amount of overload protection built-into one's microphone preamps is down to their design. The higher the power supply voltage feeding the amplifiers, the more head-room they will have. On modern mixers, even fairly inexpensive ones, this shouldn't be a problem.

EQ

Many mixers contain, for each input, a group of controls called "EQ" or equalization controls. These are essentially, "tone controls". Usually there are at least three and sometimes more. They are usually marked "high", "mid" and "low". Sometimes the frequency at which they come into effect is also marked on the mixer and sometimes that frequency has its own control and can be varied somewhat. There is usually a set of these for each input on the mixer. If one is doing an 8 or a 16-track recording where the mix will be finalized at a later date and every instrument or instrument group has been assigned it's own microphone channel, then I can see where such controls would be very useful. On the other hand, most of the types of mixers that are used for location recording are "X" number of channels in but only two channels out and are designed for mixing on-the-fly while the performance being captured is actually occurring. So, no matter how many mikes you are using, the end result is two channels recorded to media, and that is cast in stone. There is no going back and "tweaking" this mike feed or that one. Since there is no way to "undo" an injudiciously applied amount of EQ in these cases, I tend not to use it. The exception would be if I had a certain microphone that was deficient in some way (like a ribbon mike that had little response above about 10 KHz) I might use a bit of EQ on that channel to accentuate the area that was a little lacking. It is also possible to judiciously add a little presence to a vocalist by lifting the midrange a bit, or to reduce the chestiness in a male vocalist's voice by reducing the bass on his mike's input. Other than that, I feel that it's best to leave these controls out of the picture. They're great to have when you need them as long as you keep in mind that a little goes a long way, and the results are not reversible.

Each microphone mixer or pre-amp has its own features such as built-in reverb effects, or busses for external effects and it is beyond the scope of this article to discuss them. But what I do think is necessary is to talk about the size of mixer needed for location recording. 

When I got back into recording after a long hiatus (see the first installment of this blog entitled "Commercial Recording Quality"), I figured that since all of my earlier recordings had been made with mostly two microphones, and when confronted with a chorus as well as an orchestra, a maximum of four, that a four microphone input mixer would more than suffice.

Behringer 1202 Mixer sports four excellent microphone inputs and 8 line-level inputs. These can be had for less than $120

When I bought the above pictured Behringer 1202, I was astonished by the street price of just a hair over $100 (US). My previous mixer, a TAPCO, had cost about $1200 and wasn't anywhere near as good. Behringer calls their microphone preamps in this line of mixers "Xenyx" pre-amps and they tout them as being very quiet. They are. I have made some astonishing recordings with this mixer. The circuitry sounds so good and is so quiet that instruments just "appear" out of a velvet black background. I realize that manufacturing this mixer in China (from a German design) is part of the reason for the low-cost, and the advancements in solid-state technology is responsible for the rest, but still, I was blown away by the quality. Soon I realized, however, that the types of recordings that I was doing required more than just four microphone inputs. Not the least reason was because I was using a stereo microphone in the M-S pattern (which will be discussed in another installment), and that required three microphone preamps to yield two channels. That left one. Realizing that I needed more microphone inputs, I went back to the Behringer catalogue and found the 1832FX. This mixer sported 6 microphone inputs as well as built-in reverb effects. I hesitated due to the size of this mixer (one more thing to carry). It is much larger than my 1202. But I figured that where the 4-input 1202 was sufficient, I could cary that, and where I needed more, I could carry the 1832FX. 

The Behringer 1832FX Mixer has 6 of Behringer's excellent "Xenyx" series microphone pre-amps and 8 line level inputs. It also sports built-in digital special effects and a graphic equalizer on the outputs. The street price on this mixer is around $250.  

While I have chosen Behringer mixers, that doesn't mean that there aren't others just as good, and while I find the Behringers excellent performers and suburb values, Mixers from Peavy, Mackie, Allen & Heath,   Edirol and Yamaha are probably just as good. Choose according your projected needs and keep in mind that you will have to tote around whatever mixer that you eventually choose. If you do find, at some point down the road, that you need more microphone channels than your current mixer can provide, that there is another alternative to buying a whole new mixer.

Add-On Microphone Preamps

Most mixers on the market today come equipped with a certain number of microphone preamp stages. In the case of the Behringer 1202, that number is four, and with the 1832FX it's six. The Peavy PV20, for instance, is close in price to the Behringer 1832FX and offers sixteen microphone stages but lacks the Behringer's comprehensive features list.  It is also immense. Many of these same mixers also have a number of line-level inputs. While these, lacking the gain, are not designed for microphones, but rather for other sources such as recorders (for mixing-in pre-recorded material and adding to it), and even other mixers. My Behringer 1202 and 1832FX both have 8 such inputs and other mixers may have similar. This is a perfect application for outboard microphone pre-amps. These devices can be had for as little as about $35 for a single-channel tubed unit from Behringer up to several thousand dollars. SM Pro Audio sells an excellent 4-channel solid-state microphone pre-amp called the Q-Pre4 which is available for a street price of less than $80. For my purposes, the Behringer, again, proved to have the most bang for the buck. Behringer's MIC100 is a solid-state, stand-alone, single-channel microphone preamp with a tube output buffer to impart "the tube sound" to the microphone being fed it. The 12AX7 vacuum tube used is not for gain and therefore adds no appreciable noise. You will, of course, need one for each extra microphone you connect. For either Behringer mixer, that means eight in total. I carry two in my recording kit and have even used them in place of an entire mixer when only two channels are required! They sound excellent and would, in fact, constitute a fine starter system. A pair of MC100s, a pair of decent big-capsule cardioid condenser mikes such as the Samson C01 at less than $80 each (street price) or a pair of Behringer Pro 2Cs (multi-pattern mikes) at slightly more along with a Zoom H2 solid-state recorder and you will be recording 24-bit, 96KHz stereo recordings for a a maximum investment of only about $500! This kind of price/performance combo would have been unheard of just a few short years ago.

The Behringer MIC100 Tube buffered Microphone preamp. At $35 street, it's excellent and hard to beat for flexibility and control

The SM Pro Q-Pro4 4-channel Microphone preamp. at around $80, it would be hard to go wrong having one of these in one's kit

Next time we'll discuss Microphone types and how to deploy them...

Editor's Note: Audiophile Style community member George Graves has kindly allowed us to publish his five part series on high quality audio recording. This series is a primer that many audiophiles will find interesting and educational. It isn't a treatise, textbook, or master class designed to cover every detail in depth. As a music lover and audiophile I want to understand a bit more about recording, but I don't want to become a recording engineer. This series is right in my wheelhouse, and I hope it's in yours as well. - CC

Part One: Commercial Recording Quality (link)

Part Two: Recording Media

For most of the "stereo era" the choice of recording mediums to use was limited to one and only one, magnetic tape. When the digital age came upon us, it was still limited to only one choice, magnetic tape, although this time it was video tape in the form of U-Matic, VHS or Beta and other digital magnetic tape formats such as DAT. Professional studios even used large reel-to-reel digital tape decks, some with 2-inch wide tape. But for most of the years following the Second World War, it was analog tape running at 15 or even 30 ips (inches per second) with wide tracks taking up fully half the width of the tape (1/2 of a quarter-inch-wide tape for each of two channels or three tracks covering a piece of 1/2 -inch tape or 35 mm magnetic "film") to guarantee reasonable signal-to-noise ratios and relative freedom from dropouts. The machines were big, heavy, complex and expensive, but that was all we had.

Today, it's quite different. The choices available to even the amateur recordist are so varied as to be almost bewildering. Digital recording has benefitted from the advances in computer technology which is moving at a breakneck speed. The first casualty of the computer age was the practice of recording digital audio to magnetic tape. Computer hard drives became the norm for studios who gladly replaced their stands of multitrack digital tape recorders with racks of computer hard drives. These are used for both audio capture and for long-term storage (with suitable back-ups, of course). Now, solid-state memory is making inroads into the hard-drive-based recording camp. Solid-state memories have the advantages that they have no moving parts, have long storage life and can be physically very small. Expect this trend to continue until it has replaced hard disk recording altogether.

Example of a rack-mounted Solid State Recorder/player 

As amateur recordists, we benefit from these computer-based innovations in a number of ways. There are portable devices available which use a myriad of formats and media that cover just about everything from MP3 recording from a live source, all the way to the very sophisticated and flexible DSD (Direct Stream Digital) with it's 2.8 or 5.6 MHz sampling rate. The equipment required ranges from laptop computers to tiny solid state recorders, some, not much bigger than a pack of cigarettes. Let's examine some of the more popular and accessible options.

Using a Computer as a Digital Recorder

Since most people have a laptop computer of some description laying around, this would seem to be the cheapest and easiest path to digital recording. Certainly, the software to do this is in abundance, and one of the better programs, Audacity, (http://audacity.sourceforge.net/download/) is even free. Available for Windows, Mac, and Linux, this program gives you all the tools you need, save one, to use your computer as a digital recorder. The one tool missing is an outboard analog-to-digital converter. It takes analog signals in from the mixer or microphone preamps and outputs a digital signal in the required format. It needs to communicate with the software on the computer, and for this the interface between computer and A/D converter needs to be via a two-way bus such as USB or Firewire. All modern computers have USB 2.0, but few use Firewire. Unfortunately, the current USB standard, USB 2.0, is bandwidth limited and even two channels of 24-bit, 96 KHz digital audio is a stretch. While USB is fine for 44.1 and 48 KHz digital audio, even at 24-bit, it's just too slow for the higher sampling rates. Some implementations of USB use "tricks" to defeat the master-slave protocol on USB to allow it to work at 24-bit, 96 KHz, but the conversion is slow, causing a delay between the capture and the digitization of the audio stream. This delay is called latency and it makes monitoring the digital signal very difficult because it can be many seconds behind the performance being recorded. Therefore, for anything greater than a 48 KHz sampling rate, it is advisable to use Firewire. Firewire 400 is faster than USB 2.0 in the continuous transfer mode because it is a peer-to-peer protocol that doesn't have the computer overhead of USB. This reduces latency and even allows for the transfer of two-channels of digital audio at 32-bit, floating point and 192KHz sampling rate.

However, the problem remains that most computers don't come with Firewire ports. If you use a desktop computer, you can always purchase a Firewire PC card to go in it, but if you are doing location recording (as opposed to having your own studio), this makes the prospect of live recording about as appealing as the days of lugging huge reel-to-reel analog recorders around. Laptops are the preferred computer here, but these days, only a couple of Sony models and Macbook Pros have Firewire ports built-in. Some of the more expensive Windows Laptops sport PCI Express slots and these allow one to add third-party Firewire support to one's laptop, but either buying a Macbook Pro laptop or a higher-end Windows laptop with a peripheral card slot is an expensive proposition, and there are other alternatives.

Also, the outboard A/D converter can be expensive but at least here there are excellent inexpensive solutions, available as well. For A/D (and D/A) conversion of so-called "CD quality" recordings (16-bit, 44.1 KHz sampling rate), there are a number of USB interfaced converters available from companies like Behringer, Lexicon and Samson, some for less than (US) $40. Behringer sells an excellent Firewire computer interface box that has A/D and D/A converters, called the FCA202 that will accommodate up to 24-bit, 96 KHz audio for less than (US) $100 street price. I have used this converter with an iBook laptop and Audacity to make some truly spectacular sounding recordings.

Behringer FCA202 Firewire combination A/D and D/A computer interface is excellent quality and affordable with street prices less than $100

Hi-Md

For years, Sony has sold its proprietary "Atrac" compression scheme in conjunction with its "Mini-Disc" transport and media system for portable recording and playback. While "Atrac" arguably sounds somewhat better than MP3, it's certainly not what most of us would use to "master" live performances. About 7 or 8 years ago, Sony came up with an improvement to the format called Hi-Md. The improvements were threefold. First, the Hi-Md disc capacity was increased to 1 Gigabyte. This gave just short of 8 hours of Atrac recording and playback at the highest 256 kbps "Hi-SP" Atrac3+ setting. Secondly, Sony improved it's compression algorithms to "Atrac3+" which was a distinct improvement in sound quality, and thirdly and most importantly for our purposes, Sony added a linear (non-compressed) 16-bit, 44.1Khz PCM recording ability to the format which gave 1 hr and 34 minutes of CD quality audio on one Hi-Md disc! These devices can make excellent recordings, but there is one drawback. If you didn't buy the top-of-the-line recorder/player, there was no way to physically transfer the resultant digital recording from the Hi-Md recorder to one's computer! All you could do was to output it as an analog signal and re-digitize it (in real time) as you write it to your computer's hard drive.

The Sony Walkman MZ RH-910 Hi-Md MiniDisc recorder/player shown here with an M-S electret condenser microphone

Solid-State Recorders

In the last five years, a new type of portable recorder has appeared. Taking their cue from digital cameras, these small, hand-held recorders use the same kinds of solid-state memories as cameras use to store the digital audio files. Lacking moving parts of any kind, these devices give excellent battery life and with tiny SD cards, are truly hand held. Most have their own microphones built-in, but will take external mikes. Some, like the Zoom H4 series, will take professional XLR microphone connectors and will even supply the 48-volt phantom power required by professional condenser microphones. Others will take only consumer grade outboard electret microphones connected via 1/8-inch "mini" phone plugs. All will accept line-level inputs from mixers and outboard microphone preamps, but most accept only consumer audio levels of -10 dBm. Many of these devices offer multiple recording formats ranging all the way from MP3 to 24-bit, 96 KHz PCM audio. But a lot of these recorders only record to 48 KHz, so one must be careful when buying. One of the smallest and most affordable of these devices is the Zoom H2 "Handy-Recorder". Capable of recording up to 24/96, this device sports four microphones of its own and can be used to record surround sound (although it is limited to 48 Khz sampling in this mode). At a street price of less of than US $200, one could use this device with an inexpensive mixer from Mackie or Alesis or Behringer and a couple of inexpensive big-capsule cardioid condenser microphones as one's primary recording system without any apologies to anyone. An even simpler system would be Zoom's H4n which has its own microphone preamps and powered XLR connectors. Then all you need is the microphones. Of course, you'd then limit yourself to only two, but the recorded results would be virtually identical to those obtained with the H2 and an outboard two-microphone preamplifier or mixer as the recording circuitry for the two units is identical.  

I use an H2 as a backup recorder and it has come in handy on several occasions. I just connect it to my mixer with a stereo mini-phone plug to stereo RCA cable, and start it before the program begins and forget it. If anything goes wrong with the computer setup, I still have a 24-bit, 96 Khz recording. 

These recorders are made by such companies as Alesis, Sony, TASCAM, Samson/Zoom, Korg, Marantz, Edirol, and M-Audio. 

The Zoom H2 "Handy-Recorder" is capable of making high-resolution recordings at 24-bit/96 KHz

Mixers With Built-in Recording Devices

An alternative to a separate Mixer or microphone preamp and recording device would be a solution that combine the two into one unit. Some, like the Korg D888, use and internal hard disk to store audio directly. Others, like TASCAM's 2488neo, use both a hard drive and a CD burner internal to the unit. The downside of both of these units is that one is limited to 44.1 KHz sampling rate, although the TASCAM will record 24-bit to the hard drive. Zoom's new R16 will will act as 16 track stand-alone mixer and has 8 microphone preamps. It records to solid-state memory, or can function as a DAW interface to a computer using USB which limits its sampling rate to 48 KHz. While these can be effective at reducing clutter and simplifying one's setup, I find that they are restricting in that one cannot upgrade any part of the system (let's say that down the road, you find that you need more microphone inputs, for instance) without replacing everything.

Zoom R16. An Example of an "all-in-one" solution. 8-microphone inputs, built-in recorder to SD Cards up to 32 Gig, Limited to 24-bit, 48 Khz. 

The Ultimate Recording Medium, DSD

Back around the turn of this century, Sony, responding to complaints from audiophiles that CD wasn't "good enough" came out with a new high-end format called "Super Audio Compact Disc" or SACD. The format used for recording these high-resolution discs was a departure from all other recording schemes then in existence. It was called DSD or "Direct Stream Digital" and instead of being the standard Pulse Code Modulation (PCM) used for CD and DVD-A (another high-resolution format), DSD used a single-bit process that employed a very high sampling rate of either 2.8 MHz or 5.6 MHz. SACD didn't succeed in the mass market, as most people felt that regular CD was "good enough", but it does have a following in the audiophile market and companies like Telarc, Mobile Fidelity, Virgin Records, and a number of others still record and release in this format. Korg, known for their recording and sound reinforcement equipment, sells three recorders that will allow the amateur recordist to capture performances in this ultimate of high-resolution formats. The three machines are called the MR-1, which is a small hand held field recorder about the size of a Zoom H2 (see above) the MR-1000 which is a larger field recorder about the size of a very thick paperback novel, and sports XLR microphone inputs,  and the rack-mounted MR-2000. The MR-1 retails for about (US) $900, while the MR-1 retails for about (US) $1500 and the MR-2000 is about (US) $2000. 

It is possible to regularly find MR-1s on E-Bay for less than $500 and I got mine for $299. Now, the DSD format and Korg's implementation of it needs a little explaining. Currently, there is no practical way for any home or amateur recording enthusiasts to make their own SACD discs. While there is software available for this, it is extremely expensive, the cheapest being around (US) $5000. However, the software that Korg supplies with their DSD recorders, called 'AudioGate', allows owners of Windows and Mac computers to "translate" the super high quality DSD master to any currently used digital audio format. This means everything from 24-bit, 192 KHz PCM all the way down to MP3. Think of the DSD format as the audio equivalent of the "Raw" format for digital cameras. One can store these DSD files on any hard disk and then transfer them to the MR-1 for playback in their native format, or make lower resolution copies to distribute to your "talent" (those who allow you to record them) or even to burn DVD-A's at 192, 176.4, 96 or 88.2 KHz sampling rates.

This is my preferred method of recording. The device is beautifully made, takes pro levels, has balanced line inputs and will transfer files over USB to one's computer. A couple of slight drawbacks of this device need to be noted here. One is that it uses an internal 20 Gigabyte hard drive. This makes the recorder somewhat fragile and it means that one cannot increase storage size. The built-in battery is also limited to about 2.5 hours of recording time and is not user replaceable. For my part, battery life is a non issue, because I always record in venues with electric mains power (after all the mixer needs mains power too). Also, unlike the MR-1's bigger siblings, it records at only 2.8 MHz, 5.6 MHz not being available on this unit. That's OK really as I have heard SACDs with samples recorded at both sample rates, and honestly, I defy anyone to hear the difference. Most commercial SACDs are mastered at 2.8 MHz. 

I prefer the Korg DSD recorder to my old method of using my computer as a recording device because the MR-1 is so much simpler to use than the computer (which is actually, quite complex in it's setup). Forget to do any one of a number of "rituals" with the computer, and you might find yourself, as I did fairly recently, recording a large symphonic band with the computer's little built-in voice microphone instead of with the fine stereo mike that was on the stand in front of the group. OOPs! Luckily, I had the Zoom H2 connected to the mixer and it did record the ensemble properly. Always have a backup, if possible. 

The Korg MR-1 can record in DSD, or PCM and is very flexible. It will make the highest quality recordings to which an amateur recordist can currently aspire and it's smaller and lighter than a laptop; simpler too! 

Next time we'll look at mixing and microphone preamp options.

Editor's Note: Audiophile Style community member George Graves has kindly allowed us to publish his five part series on high quality audio recording. This series is a primer that many audiophiles will find interesting and educational. It isn't a treatise, textbook, or master class designed to cover every detail in depth. As a music lover and audiophile I want to understand a bit more about recording, but I don't want to become a recording engineer. This series is right in my wheelhouse, and I hope it's in yours as well. - CC

Part One: Commercial Recording Quality

I don't know if any of my fellow audiophiles out there have noticed this, but even the best recordings always seem to "lack" something. Uncompressed digital (even RedBook), promises wide dynamic range, excellent frequency response and low distortion. It should be possible to make recordings so good that, given a halfway decent playback system, the musicians are in the room with you. It is technically possible and surprisingly easy to do this, but it rarely happens with commercial recordings. Why is it that still, in this digital age, audiophiles cling to performances recorded more than fifty years ago as the pinnacle of the recording arts? Recordings made in the late 1950's and early 1960's by such people as Mercury Record's C. Robert Fine, or RCA Victor's Lewis Leyton in the classical recording world, and Rudy Van Gelder of Riverside, and Impulse fame in the world of jazz are held in such high esteem, that even CD and SACD re-releases of their recordings still sell very well today. It's as if no progress has been made in the art and science of recording in the last 55 years or so. 

I have found in building my stereo system that this has become a dog chasing his tail endeavor. My playback equipment gets better and better and yet the recordings to which I listen, ranging from terrible to OK never get any better than just OK. Even so-called audiophile recordings from labels such as Telarc and Reference and Naxos, to name a few, never sound quite as good as I think they should.

This started me on a quest. If I can't buy reference quality performances to play on my high-end audio system, perhaps I could make some. I didn't come to this decision in a vacuum. In a previous life, I was a semi-pro recording engineer who used to record a major symphony orchestra for their archives and for broadcast. I had also professionally recorded, for broadcast on NPR's "Jazz Alive" series, such artists as Hubert Laws, Dizzy Gillespie, Stepan Grapelli, etc. Needless to say, in most cases, I kept masters of these recordings. The client either received copies or co-masters recorded on a tandem analog recorder. I gave up this pursuit because of the weight and amount (not to mention the cost) of the recording gear that I was forced to schlep around, necessary, in those days, to make a truly professional recording. But, still, today, CD transfers of these 25-year old 15 ips 1/2-track stereo analog tapes are among the best sounding recordings I have.

Things have changed. Today, excellent quality recording equipment is not only plentiful, but cheap. It is possible to buy excellent mixers for just a few hundred dollars. Recoding devices capable of 24-bit, 96 KHz performance are likewise very inexpensive. It is even possible to purchase small, portable recording devices that will actually capture audio in Direct Stream Digital (DSD), the 1-bit recording method used for SACD. And this equipment is small and light. One can easily carry an entire recording studio (less the microphone stands, of course) on the passenger seat of the family car! The mike stands go in the trunk, of course. 

On the microphone side of things, changes are even more profound. When I was recording semi-professionally, good quality condenser microphones were available from only a hand-full of suppliers such as Neumann, Sony, AKG and Telefunken and they were extremely expensive (especially the Neumanns and Telefunkens). Today, an excellent pair of big-capsule condenser mikes can be had from dozens of sellers for just few hundred dollars (Neumanns and Telefunkens are still tres cher, however). Companies such as Behringer, Audio Technica, Avantone, and Rode make microphones that have flat frequency response, low distortion and low noise. Today's microphone capsules use sputtered gold coated Mylar diaphragms which have such low mass that they move the microphone's fundamental resonance far above the audio passband. Back in the 1970's and 1980's most good mikes still used acid-etched brass diaphragms with frequency response peaks starting at around 6 or 7 KHz and peaking at 16 or so KHz. This worked OK with analog recording where magnetic tape self-erasure tended to roll-off the upper frequency extreme anyway, but when digital came along, it made for unnaturally bright and brittle-sounding CDs. This is, I believe, mostly where CD got it's bad reputation from audiophiles early-on. 

In future posts I will discuss some of these issues and make recommendations for a really good starter recording set-up. One that can be easily carried from place to place and yet will yield recordings that sound so much better than anything you can buy, that it will make you wonder what the pros are doing wrong! We will also discuss how to get local groups (whatever your musical preference) to allow you to record them. We will also discuss various microphone techniques, and how to choose the best arrangement for the individual ensembles you will encounter. 

We will also discuss playback equipment, of course. After all, this a two-ended process; capture and playback. Getting the most from good recordings requires a good stereo system. I think that we are going to have fun with this blog and I invite comments, suggestions and submissions from everyone.

A while back, over beer & conversation, a colleague and I thought it would be fun to write a couple of test files in MATLAB for our stereo systems. We ended up dumping the results to WAV files and burning them to CD-R -- next thing you know we had our own little Test CD.  A few years/beers later, I re-wrote same and added to them when I decided it was time to move on in my research from MATLAB to Python... now that "Test CD" has morphed into a collection of files (both PCM encoded as FLAC and DSD as DSF) packaged as an album suitable for use with a music server.  There's even an  accompanying booklet in which I attempt to explain in some detail the individual tracks.  The entire thing is licensed under a Creative Commons Attribution-NonCommercial-ShareAlike 4.0 (CC BY-NC-SA 4.0) International License, so feel free to use it under those terms. Perhaps you’ll find it interesting and of some use, if not amusement.  

Here is the CD booklet for people to browse through. It's included with the complete CD download below as well.

Download Link - https://audiophile.style/b (4.8MB PDF)

Here is the complete CD download containing the booklet, Disc 1 for PCM files and Disc 2 for DSD files. 

Download Link - https://audiophile.style/z (380MB ZIP) or try this link by right-clicking and select Save As, then remove the .wav at the end of the file when downloaded, so you can unzip the file.

The following is the introductory paragraph contained on page one of the booklet. 

Dr. Z’s Test CD

This PDF document serves as the liner notes (or ”booklet”) for Dr. Z’s Test CD, which is really just a collection of 24/96 files¹ in the FLAC format² and not a physical disc at all. But back in the day, my colleague Prof. John V. Olson and I used to dump files like these in WAV format onto a CD-R for use at home – such were the halcyon days before ”computer audio” was a household thing – hence; for historically sentimental reasons, a Test CD. That said, the ”disc” was authored for system evaluation and set-up, and is therefore not recommended for an enjoyable listening session. You may think that there are some typical test files missing or notice that there isn’t much in the way of traditional musical content – as to the former, a dedicated ”burn-in” track is pointless and the intent was listening, not to write tracks for use with an oscilloscope to directly test electronic components (perhaps a Test CD 2?);³ as for the latter, nothing musical is really missing, since that’s what your collection is for, isn’t it? This ensemble of tracks is primarily designed for testing and evaluation of loudspeaker placement, system setup, room characteristics and overall performance while ensconced in your favorite listening chair. You can use it with headphones, too, but with such devices you would be testing more for your hearing sensitivity or capsule isolation. Headphones are sufficiently unique beasts in their non-flat responses and they don’t really interact with the listening room. Bottom line: this is nothing more than a work of computational-audiphiliac Onanism.

I’m trained as a physicist, but do a fair amount of statistical signal processing in my work,⁴ particularly on digital records of infrasound generated from natural and anthropogenic sources (and in a previous life, of ground-based magnetometer data from the interaction of the solar wind with Earth’s high-latitude magnetic field). But long before that I just really enjoyed listening to recorded music and thinking about the equipment that made that possible. That fascination has remained a strong and continuous thread, making for a lifelong pursuit; this work is therefore but an ex- tension of it: to learn more about the digital aspects of music reproduction, as well as how we perceive that reproduction in real-world listening spaces. There are numerous other very fine Test CDs out there, so why make my own? I tell students that using a ”black box” is all well and good – I drive a truck and certainly wouldn’t want to build one – but from time to time you get more out of making your own box to better understand what’s inside and how it works. In particular, the best way to make sense of data is often to ”play” with it (using various digital signal processing tools), so what better playground than a project like this? I get to make the very data I’m about to play with! This particular project all started a few years ago over a beer and conversation... a few years/beers later and this is what I came up with. Perhaps you’ll find it interesting and of some use, if not amusement.

1. Several alternative bit depths and sample rates are used in a few of the tracks to test for such things; all the other tracks were rendered at 24-bit/96 kHz resolution.

2. Beginning with ver. 1.1, a second ”disc” of DSD files in the DSF format was included.

3. In ver. 1.1.2, some of those electro-analytical type files were added at Track.

4. To be honest, I do more managing scientists and translating science to other non-techncial managers than actual science these days, c’est la vie.

The reign of Tidal for audiophiles here in the US is nearly over. Yes, I can't believe I wrote the words reign and Tidal in the same sentence, but it's true. Tidal has been the only lossless streaming service supported on high end platforms here in the US for several years. Hold your letters, calls, and emails to notify me about the lossless service offered by the Scottish Nose Whistle label streaming at 32 bit / 768 kHz. For all intents and purposes in the US it has been Tidal or nothing (unless you wisely subscribed to Qobuz by circumventing the geographic rules). 

In preparation for the October release of Qobuz here in the US, a few of us at CA have been taking the service for a spin. So far, I love what I hear and see. However, the thought of switching from Tidal to Qobuz and manually marking nearly 1,000 albums as favorites again is too daunting. This is the point where everyone who purchases all their music laughs at me and says I told you so. 

Anyway, those of us who are tightly interwoven with Tidal should rest easy. The service called Soundiiz has all of us covered for less than a cup of coffee. Soundiiz can synchronize playlists between services for those of us who subscribe to several. It can also do what many members of the CA Community will happily pay for, copy favorite artists, albums, tracks and playlists from one service to another (Tidal > Qobuz). 

This is the part in the article where I tell all the Tidal users to bookmark this page for future reference. You'll switch to Qobuz (trust me) and this will save you so much time you can actually listen to your audio system rather than mess with a computer :~)

Step By Step 

First, signup for Soundiiz and pay the tiny bit extra for the Premium service that enables one to use the Platform to Platform feature.

Second, connect both Tidal and Qobuz accounts within the Soundiiz site / app by clicking on them and selecting connect. Very simple. If you skip this step, there will be another chance to login when conducting the Tidal to Qobuz copy. 

Select Platform to Platform Transfers from the left column, then select Let's Go.

Select Tidal as your source service.

If you didn't already login to Tidal, you'll get this popup window to do so.

Select Qobuz as your destination service.

Select which items you want to copy from Tidal to Qobuz. You can click the gear icon to specify within each category what you want to copy. Click Confirm My Selection when done. 

Make sure everything looks good and select Begin the Transfer.

You can watch the process or even leave the process and wait for an email from Soundiiz the the batch has completed. 

When everything is done you'll see the green bars up top and be able to scroll through your content for items that need further attention such as Not Found or partial playlists. 

That's really it. Soundiiz makes everything really easy. Now for some additional details.

A Note About High Resolution

Some of my albums from Tidal appear in Qobuz as the high resolution versions. This is a cool, possibly unintended feature of the transfer. One strange things happened with the transfer of Jose James' album yesterday I Had the Blues. I have it as a Tidal favorite, but the content is no longer available in Tidal. Soundiiz selected the 24/96 version of this album as my Qobuz version. Not the identical album, but I'll take it. A cursory look of my other albums shows several high resolution versions as Qobuz favorites (nothing to do with MQA).

Here is a shot of my newly created favorites imported from Tidal.

Jose James album not found in Tidal

Qobuz has the high resolution version of the Jose James album favorited, even though this wasn't the version favorite in Tidal and Tidal no longer has the album available.

A Note On Not Found Albums

Roughly 150 of 1,000 albums weren't found on Qobuz, but this could be the quality of the transfer software in finding the exact version or the fact that I favorited some special version in Tidal and only the standard version exists in Qobuz. Let's take a look.

Th first album on my list of Not Founds is 12 Years A Slave. Searching both Tidal and Qobuz yields the exact same album with the exact same number of available tracks. Perhaps the fact that the entire album isn't available for streaming has something to do with this one not being identified by Soundiiz. 

Searching for Traffic's Mr. Fantasy that Tidal calls the Remastered Remasters, I see both album in both services. Unfortunately Soundiiz couldn't translate the funky title even though both albums have 22 tracks identically named. 

Finally, searching for The April Maze's album Recycled Soul yields the expected result. It's available in Tidal but not Qobuz, thus the Not Found identifier in Soundiiz. 

After looking over several more, it seems that there's a mix of truly unavailable albums, stranger named albums, and different versions of albums between the services. Hopefully Soundiiz will improve at matching the same albums with slightly different titles. 

After the batch, as Soundiiz calls it, was complete I wanted to export a list of my Not Found albums. Or, at least export a list of albums so I could compare to see which one's I needed to look for in Qobuz. Souniiz's ability to export lists is less than stellar. 

To view completed batches click on My Batches, then the three little dots on the right for details.

First, there's no way to export the entire list of items copied from Tidal to Qobuz. Even though it appears in a nice spreadsheet-looking format within Soundiiz, the site can't export this exact view. I tried to work around this limitation by exporting a list of all my albums. I figured the list would contain a single line of all the albums not found in Qobuz and a double line (listing one entry for Tidal and one for Qobuz) for the albums found on both services. This ended in disappointment as well because Soundiiz exports a strange csv file that doesn't separate the Artist and Album when imported into Google Sheets. However, upon closer inspection, one must manually specify the separator as a semicolon rather than a comma during the data import. For some reason, Soundiiz uses a semicolon in its "comma separated value" exports.  

Playlist export

Albums export

If you don't click download this file, your web browser may just open the csv and it will look like this.

When you download the csv file and let Google decide what the separator is, the imported data looks like this

After telling Google to use a semicolon, the spreadsheet looks good and one can scroll through to find the non-duplicated rows. These are the row that are found in only Tidal as the duplicated rows show one row for Tidal and one for Qobuz. It would be much better if all the data visible on the batch screen was exported, but this will suffice. 

Lastly, I thought I could use the filters available within the Soundiiz site to only show the albums that weren't found in Qobuz. This wasn't the case. The one heading by which I needed to filter was called Score, but Soundiiz only enables filtering by Status and Types (artists, albums etc...). The only way I could get a logical list of items Not Found in Qobuz was to sort the Soundiiz batch by Score with Not Found on top. This enabled me to screenshot several pages of items that required further investigation. Far from ideal, but a first world problem indeed. 

Filtering options

Wrap Up

Although Soundiiz isn't perfect, it's a huge time saver for those of us who've spent years curating a Tidal collection of favorites and playlists. I highly recommend using it if you value your time and if you have more than a couple dozen items you'd like to appear in Qobuz when you switch over from Tidal. 

 Integrating Subwoofers with Stereo Mains using Audiolense

In this article, I walk through the steps of using Audiolense to create a digital crossover and time align dual subs with stereo mains. In addition, showing how to smooth the frequency response and reduce group delay at the listening position.

This results in a smooth frequency response (12 Hz to 22 kHz ±3dB on my system) with all direct sound arriving at my ears at the same time. The phase response and group delay is mostly flat at the listening position.

In addition, I walk through a time domain experiment designing two correction filters with the same frequency response, but one with time domain correction and one without.  I discuss the audible differences through listening sessions.

Why subs for music?

In Audiolense Digital Loudspeaker and Room Correction Software Walkthrough, I was able to smooth the frequency response of my JBL cinema loudspeakers, time align the drivers, and achieve relatively flat phase and group delay at the listening position. I am happy with the results, but something is missing…

My JBL speakers have solid output to 40 Hz and can extend in room response close to 30 Hz. Great high efficiency kick and punch, but missing a bit of weight on the bottom octave. I listen to rock, blues, and alternative music, most of which does not have deep bass (bass guitar low E is 41 Hz), so if I added subs, would I notice?

Enter Rythmik Audio. A company that has been around for many years with a good engineering and “no-hype” reputation. Being a tech geek, I was intrigued by their direct servo technology and the most extensive FAQ I have seen from any sub manufacturer. Most importantly for me, one of the few sub manufactures that publish their own measurements and validated by 3^rd party testers.

I purchased two F12 entry level music subs direct from Rythmik. They arrived in a timely fashion and extremely well packed – thick cardboard box, within another thick cardboard box, with the sub floating in high density foam. Could drop off the end of the truck with no damage. Very nice.

Sub Setup

Much has been written about setting up subs in numerous configurations. One can even use room simulation software to determine best placement. However, since we are using DSP, try whatever sub setup and configuration you wish. If your ears (and measurements) disagree with the sound, then some fine tuning of placement may be required. However, using this advanced DSP, it is likely the measured results and listening experience will be more than acceptable.

It is generally accepted for frequencies below 80 Hz, it becomes difficult to determine a sound's location. If you click on the link, you can try it in on your own system. As an ex-recording/mixing engineer, I can say for the recordings and mixes I worked on, low frequencies below 100 Hz, whether from bass guitar, drums, piano or synthesizer, were always in the center of the mix and never panned. This seems to be the case for most music, except from the 60’s and other recordings where it is intended as an effect.

So, why “stereo” subs? I wanted more sub output, without having to buy a bigger sub. Whether it is one or multiple subs, it is important to be able to individually control the frequency and timing response of each sub with respect to the mains. This is the key takeaway from the article. In my case, I have control over the frequency and timing response of all drivers in the system.

Setup:

Is this the best location for my subs (i.e. between the JBL’s and electronics)? Probably not. It is more about convenience than anything else. Once measured and listened to, they sounded good no matter where I am in the room. However, I am mostly interested in how they sound across my three seat listening area. 

Sub configuration:

I am letting Audiolense take control of adjusting delay/phase and XO duties for the best possible integration with the rest of the drivers. I level matched the subs to the mains with the volume control.

Configuring Audiolense

See my intro article to Audiolense for basic configuration and operation, as most of this article will focus on the differences, so as not to duplicate content.

I have taken my existing stereo two way biamp (XO at 630 Hz) speaker setup, and turned it into a three way triamp setup, crossing the subs at 40 Hz:

Why did I choose 40 Hz as the sub XO corner frequency? The JBL cabs with 2 x 15” drivers each, have solid bass output to 40 Hz, as will be seen in the measurements. I like the high efficiency punch and slam, but looking to supplement the bottom octave to give it the full weight.

The source material I listen to does not have much output below 40 Hz (we will see about that), which allows me to get away with smaller subs, as the JBL cabs take the majority of the bass signal. As it turns out, even with 525 watts @ 4 ohms into dual 15”woofers per side, the music beat can trigger the limiters first before the subs run out of gas (at 300 watts per sub). This is at concert level of ~105 dB SPL continuous output at the listening positon with peaks above that. That’s just for short term fun, as hearing impairment begins at around 5 minutes at this continuous SPL, even though it sounds perfectly clean.

Most of my critical listening is performed at reference level, i.e. the magic of 83 dB SPL. For lower levels, I calibrate using JRiver’s dynamic loudness control, which provides a more natural sounding volume control based on the frequency response characteristics of human hearing.

The main Audiolense screen shows the newly designed digital XO:

In my previous Audiolense article, there is more about XO choices, steep slopes, etc. Bernt has a really good section on XO choices in the help file. Also, Rod Elliott’s article on, “Phase, Time and Distortion in Loudspeakers” has a good read on crossover filters. 

Taking Measurements

The detailed steps of setting up and taking measurements are covered in my previous walkthrough. Here, I am simply taking the measurement:

A few things to note. See how the channel outputs are matched to which speaker. It is likely that you will need to enable Output Channel Override from the Advanced Settings menu. While I have a triamp config, a passive stereo mains with added sub(s) (i.e. biamp config), will require one or more additional DAC output channels, one per sub.

Note the frequency sweep ranges for each of the channels. Keep this in mind when looking at the frequency response charts, relative to the crossover slopes. When a correction filter is made, the speaker’s raw response is convolved with the corresponding digital cross over slope. In the case of linear phase crossovers, which these are, the advantage is that all direct sound frequencies arrive at the same time to the listener’s ears, in phase.

Check the time delay in the last column. This measurement has already been taken and these are the resultant delays between drivers, relative to the tweeter. If you look back at my previous article, you will see the same delay values for the midrange channels. While the subs appear to be in the horizontal plane, I did tape measure them, one can see the measured delays for each channel are different, which is why we want independent time domain control for each sub.

Below is the Audiolense filtered frequency response that better represents what we hear versus the raw, unfiltered measurement. I used a custom filter procedure with True Time Domain (TTD) correction turned on, as well as TTD per driver and selective preringing turned on. See the Audiolense help manual or my previous article for details for designing your own custom filter procedure:

I left the crossover slopes in so one could see how the each driver’s response would be convolved with their corresponding digital XO filter. Meaning, with the corner frequencies chosen, and steep XO slopes, each driver is working well within their normal operating range. Therefore, the acoustics slopes of the driver become the linear phase digital crossover slopes, and sum perfectly both in the frequency and time domain.

Let’s focus in on the subs (Rythmik L12) response. Here I zoomed in on the horizontal frequency scale and left the 40 Hz low pass XO displayed so one can see how the subs measured response will be convolved with the XO. I.e. the acoustical slope will become the digital XO slope after 40 Hz:

In my room, solid response down to 12 Hz (-3 dB). After 40 Hz, starting to see real room effects and then by 110 Hz, smoothing out and rolling off. Resembles the measured spec from Rythmik.

Now let’s look at the bandpass (JBL 2 x 15” ported cabs):

Good response to 40 Hz, which is the tuning frequency of the JBL vented cabs. Note in both the subs and bass measured responses, the left side of my system has nulls around 70Hz and 90 Hz and for the right speaker, non-minimum phase response at 110 Hz, 120 Hz and 140 Hz. In part, because my stereo is set up off center of the room so the left speaker is more in the corner and the right speaker positioned at the middle of the long wall. We will see Audiolense do its room correction job in these areas so both the timing and frequency response arriving at ones ears matches, even though the stereo is offset to one side of the room.

40 Hz looks to be a good XO point, again within the normal operating range of the woofers. Folks may choose a higher XO point if using bookshelf speakers. Same goes for 630 Hz XO point, well within the normal operating range of the compression driver. Let’s turn our attention towards the measured time domain (i.e. step response). 

In the Audiolense main form, select Impulse Response from the Chart View radio button group. Then from the Audiolense Analysis menu, select Measurement, then select Step Response: 

I have labelled the timing diagram to help identify which driver goes with which peak. We are looking at the direct sound plus the next 20 milliseconds of sound arrival at the microphone at the listening position. The first arrival at the listening position are the tweeters with positive polarity (i.e. the compression driver and waveguide). Second arrival is the bass and lower mids (i.e. the double 15” cabs) at .69 (left) and .71 (right) milliseconds later, again with positive polarity. Finally the subs arriving 2.75 (left) and 3.38 (right) milliseconds later after the tweeter, with negative polarity. The delay values are from the measurement window shown earlier. Note that Audiolense assigns 100 ms on the horizontal scale as the start of arrival of the sound. In our relative terms 100 ms = 0 ms.

That’s approximately 3 milliseconds of delay for the subs, even though they are in approximately the same physical horizontal plane as the double 15” cabs. If we were to visualize that, sound travels ~1 foot per millisecond and would be as if the subs were physically placed 3 feet behind the mains from their current location. 

Certainly relative to the tweeter peak, the subs are delayed and have long wavelengths. Now let’s expand the horizontal scale to see 40 ms (i.e. ~ over 40 feet of sound travel in the room).

Whoa! The left subwoofer has a huge reflection which shows up as an amplitude peak at 28 milliseconds (i.e. 128 ms on the chart). Its magnitude is bigger than the direct sound, which means it is maximum phase peak at 28 ms. Let’s look at 100 ms of sound travel to see if there are any other room issues:

The tweeters spike is just a sliver compared to the subs peaks. No other major issues after 100 ms of sound travel. That’s quite the peak at 28 ms. Something to ponder. Let’s see what Audiolense can do about that.

The other aspect of long sub wavelengths is where is the peak located? The peak can occupy several samples with the same amplitude values… and even peak higher later in time, as we see in the example above. Because we are using linear phase digital XO’s, the peak is half of the filter length in number of taps. In the case of our 65,536 tap filters, the peak would occur at sample position 37,268 – i.e. the peak of the waveform.  For a minimum phase XO, the peak would occur at sample position zero, which would be the start of the rise of the waveform. Audiolense will automatically calculate and align the peaks of each of the drivers and ensuring all drivers are positive polarity. 

As a side note, there are several techniques and software tools available to measure the time alignment of drivers in a loudspeaker system. As mentioned above, it gets tricky in sub territory due the long wavelengths involved, especially with an XO of 40 Hz. I have tried most of the tools and techniques available. I must say that Audiolense has exhibited the best accuracy with a high degree of precision that is both predictable and repeatable for time alignment. I demonstrate that here with two sets of measurements taken months apart and can replicate the exact time alignment with the woofers and tweeters (see previous article). The fact that the process is automated is a real time saver. Manually time aligning drivers requires many steps and is prone to carbon unit failure.

Finally, time alignment is not just for one mic location either. There are not enough pages here, but in my book I show time alignment of a three way triamped system maintains perfect time alignment after moving the measurement microphone to 14 different locations, covering a 6 foot by 2 foot grid area at the listening position. Basically the area of a 3 seat couch, whether sitting upright or back into the couch. Same goes for phase and group delay, virtually flat over the same listening area. 

Let’s see what Audiolense can do with this “typical” mess.

Designing a Custom Filter

Correction Procedure Designer (CPD):

As mentioned above, this is the same procedure as described in the Audiolense intro article and not going to repeat here. Rather, let me share some tips. It is worth the time to read Bernt’s help file on what each of the CPD controls do, as it makes a big impact on the sound quality. 

Essentially one is defining how much correction both in the frequency (i.e. dB of correction applied) and time (over how long in milliseconds) domains, using a user defined frequency dependent window and psychoacoustic filtering that best represents what our ears/brain hear. This offers considerably more flexibility that any other type of eq, plus time domain correction is being applied. Not only time aligning drivers, but correcting for room reflections. There is an example of that in the Audiolense intro article where the group delay in the bass frequencies were greatly reduced.  

As shown in the above step response measurement, the left sub has a huge amplitude peak at 28 milliseconds that is greater in amplitude (i.e. maximum phase) than the direct sound. We will see Audiolense correct for that.

It may take a few filter procedure iterations, similar to narrowing down the target response process, as described next, where one is happy with the sound quality. I encourage experimentation to try several graduated settings, generate/save filters and while listening to music, switch filters in real time and listen/compare. The workflow is fast, only taking a minute or two to cycle through it.

Target Design:

Let’s start with a quick tutorial on preferred target frequency responses. I have covered some of this in the Audiolense intro article and my series on measuring loudspeakers. In this article, I am going with Sean Olive’s and Floyd Toole’s research on The Subjective and Objective Evaluation of Room Correction Products and The Measurement and Calibration of Sound Reproducing Systems respectively. 

From Sean’s slide deck, is a preferred ranking of average magnitude responses, measured at the primary listening position:

The top preference (red trace) is a flat, but tilted measured response. If 0 dB is 20 Hz, then it would be a straight line to -10 dB at 20 kHz.

Note that this tilted measured response is perceived by our ear/brain as subjectively flat or a neutral response according to Sean’s research:

A measured flat in-room frequency response is not the preferred target. See how an objectively measured response of 20 Hz and straight line to -10 dB at 20 kHz is subjectively perceived as a neutral or flat response to our ears/brain (red trace overlaid in the above chart). Reading the articles linked above and JJ’s research on Acoustic and Psychoacoustic Issues in Room Correction (See PowerPoint presentation) explains further.

Armed with that knowledge, I designed a similar target response in Audiolense’s Target Designer:

0 dB = 20 Hz and a straight line to 24 kHz, so it is -10 dB down at 20 kHz. A good place to start and one can fine tune to taste by moving the 24 kHz red marker up or down in one or two dB increments at a time and listen.

I use a “bracket” method by first making one target sounding too dull and the other too bright. Might take a few tries to establish this. Then between the two targets, I move the 24 kHz maker up or down, in 1 dB increments, generate and save the filter. Using JRiver’s Convolution engine, as an example, I open the file dialog and select a FIR filter while the music is playing. There is less than a second gap of silence as the filter is switched. It is fairly easy to hear the spectral differences between filters using 1 dB increments. 

One can cycle through the process fairly quickly and in no time, be narrowing it down to a couple of candidates. Once you are down to two candidates, this takes a bit more time, as you cycle through more music, switching back and forth a number of times listening to more and more tunes. But after a couple listening sessions over a few days, one will emerge as your top preference. Whether your preference is for a neutral tone, or whatever your preference is, you can narrow it down quickly using this method.

As a side note, and not directly related to our subwoofer discussion, a loudspeakers directivity index and measured off axis frequency response is an important consideration when using DSP. The JBL 4722’s I use have a tight, but constant or controlled directivity polar pattern. It responds well to on-axis eq, as off-axis response is virtually identical, due to the constant or controlled directivity design of the waveguide used for this speaker. The Harman/JBL “spinorama” chart for my loudspeaker shows good constant directivity from about 400 Hz on up. With these speakers toed in an equilateral triangle of 10 feet, is just enough distance to illuminate the entire couch area with full range frequency response.

Note these constant directivity waveguides require high frequency compensation by design. No audible issues were heard with the HF compensation engaged and the measured HF distortion is well within the capability of the JBL Pro 4” large format compression driver.

One can also see that the target design follows the natural roll off of the subs in the room. This is best practice. 

It is worth spending the time on target design. For best accuracy, one has to zoom right in on the red marker (i.e. dot) to line it up exactly at what frequency and dB setting you want. I mean zoom way in. It will take multiple zooms by clicking down and dragging the mouse from top left diagonally to bottom right and releasing. And vice versa to zoom back out.

Final guidance relative to custom filter procedure design and target design. Pick one and optimize first. Only adjust one variable at a time in order to train ones ears to know what to listen for. Personally, I optimize the spectral timbre (i.e. frequency response) first. In other words, the target design. For me it is about getting that neutral sound. Pick a target response, like the Olive and Toole one referenced above, use it and if you don’t like it, then fine tune to your preference. With the recommended bracket procedure, it won’t take long to zero in on what you prefer.

If you look in my book or search online, there are several recording/mixing production guidelines, well known monitoring procedures and industry specs that one does try to attain as a professional in the industry. I spent 10 years in the recording/mixing chair and the “sound” is ingrained in my mind, as I heard each recording and mixes hundreds of times and on many systems outside the control room before committing to final mix for mastering. When your number one goal is to have the music “sound good” on a wide range of playback devices and environment’s, the pros try and mix and master on neutral speakers in a neutral environment (i.e. a control room acoustically designed to a specification). So whatever is artistically rendered, translates the intent as best as possible across a wide spectrum of sound reproduction systems.

While there is quite a bit of variability in the sound quality of recordings, mixes and masters, I find that the vast majority of recordings sound good across my system, including the mixes I made in the recording studio. I should know how they are supposed to sound, as I was there and mixed it! All meaning to say that one target response does work well for the vast majority of recordings I have. For example, all of these tunes sound great on my system. Rolling Stones Top 500.

It is not so much about the variability of sound quality that I object to. It is the excessive dynamic range compression that is crushing the ever living beat out of the music, is what I object to most.

Filter Generation and Simulated Output

Let’s look at the simulated frequency response, with the target:

The -3 dB points are 12 Hz and 22 kHz, and within ±3 dB variance across the frequency range, plus a tighter tolerance than that through most of the range. It did take a number of iterations of CPD tuning and listening to achieve this result. Just like in the intro article, Audiolense’s simulation is virtually identical to the measured response, using a 3^rd party acoustic measurement software like REW, for example.

Not let’s look at the timing (step) response. Here is the simulated step response plus target over 100 milliseconds:

Virtually text book timing response closely following the target. No preringing, perfectly time aligned, (that’s the vertical step showing no discontinuities) and the elimination of the maximum phase peak. Pretty much as good as it gets, for my speakers in my living room.

Let’s zoom the horizontal scale to show the first 40 ms of sound arrival:

A note on reading the chart, consider the 780 millisecond start of the step as reference = 0 milliseconds. Meaning the signal has reached the microphone at the listening position or our ears for that matter. This shows that all frequencies are arriving all at once (i.e. time aligned). Further, that nasty maximum phase peak at 28 milliseconds is gone and overall, the response follows close enough to the target for rock and roll ☺

A Time Domain Experiment

An experiment I performed is AB’ing two different FIR correction filters, one with time domain correction and one without, but both having the same frequency response. The exact same target and correction procedures were used, except for TTD correction is not enabled, nor is TTD per driver, but all other settings remain exactly the same. This effectively turns the time domain correction off, but has the same frequency correction (i.e. tonal response), so when switching between FIR correction filters in real time while listening to music, one can start to tune into the difference it makes when a system is time domain corrected, versus one that is not, especially with subs. Why? Subs introduce milliseconds of delay and even at low frequencies, we can still hear the overhang or lag in the bottom end.

Here we go, a new filter procedure, same as the previous filter procedure, but with time domain correction turned off, same target and all other settings identical. Here is the result:

As one can see, the frequency response is virtually identical to the same frequency response with the time domain correction. Check. Now let’s switch to the time domain and look at the step response over 100 milliseconds:

Does not track at all to the same target. Look back at the step response with the time domain correction turned on. Quite a difference. Let’s break it down a bit. I zoomed the horizontal scale to 60 milliseconds so we can see the time domain issues better:

Just like the measured step response at the beginning of the article of the non-time aligned speakers, here in our modeled simulation, we can see the tweeter arriving first, bass cabs second, and the subs some 3 milliseconds later, with a negative going waveform. Of course, the reflection from the left sub is still there, higher in amplitude than the tweeter, but not quite the same height as the woofers from original measurement. Why? Well, we have applied frequency correction, so there is going to be some effect on the timing, but as can be seen, not much compared to the original measurement. This is a great example of how one can’t fix time domain issues with eq. Not only does the timing response resemble nothing like the overall target, but also can’t fix the high frequencies arriving at my ears before anything else and the subs arriving late.

Given that the two frequency responses are identical, but the timing responses are not, means that Audiolense can adjust the timing response independently of frequency response. This is exactly what happens when Audiolense True Time Domain (TTD) is turned on. All drivers are time aligned and the timing response tracks closely to the target response while taming room reflection issues like that maximum phase peak in this article and reducing group delay at low frequencies as demonstrated in my previous Audiolense article.

The question… is any of this audible?

Personally, under blind conditions with my lovely assistant switching filters, I can distinguish between the two every time, even though it does take some concentration. It is not a night and day difference, but rather subtle. For me there are two audible tells. One is that more often than not, the tweeter or high frequencies are the first to arrive. This, to my ears, produces a more forward sound, being a bright brighter in tone, even though the frequency response is the same. That’s because the tweeter is almost always physically closer to ones ears than the other drivers. No amount of “eq” can fix the tweeter arriving first.  Second, one can hear the lag on the other drivers, especially the subwoofer. It does take a while to tune into what is happening when switching between filters, but when you hear it, it is hard to forget about it.

In addition, this may be what people perceive/confuse as “slow” bass, meaning bass overhang or simply the sub is still outputting, or not begun to output sound, even though the transient has passed and the mains have stopped outputting sound. It is easier to tell with transient material like drums that have a good high frequency cue and low frequency content, like a kick drum for example. Can you hear the click first and then the boom? Or does it sound integrated?

Given the sophistication and power of today’s audio DSP software, I am hoping that the industry revisits the time domain of speakers in rooms, as it seems to me to be the missing half of what constitutes accurate sound reproduction. If the goal is to accurately reproduce the waveform of what is stored on the digital media to ones ears, then there can be no frequency or time domain distortions added to the waveform. 

Just like in my previous article where I show the flat frequency, phase response and group delay of my Lynx Hilo converter, I should be measuring the same as the sound arrives at my ears. This ensures, whatever is stored on the digital media is reproduced as close to as possible to my ears with little frequency or time domain distortion. However, there are a few major transfer functions along the way that really mess with frequency and timing responses – mainly non-time aligned speakers of all types, in wildly variable room acoustics of all shapes and sizes.

This is where Audiolense comes into play as one can design a custom, high resolution digital FIR filter that contains the mathematical convolution of your specific speakers in your listening environment. This custom filter is designed to restore the music signal back to what is actually on the digital media, or as close as possible, again, specifically designed for your speakers in your room.

It helps if your speakers are time aligned with constant directivity characteristics. It helps if the speakers have been designed and engineered using science like Harman’s spinorama system for example. 

Lots of controversy over whether to correct frequencies above Schroeder or 500 Hz as an upper limit. In my case, I am trading a more ragged frequency response for having high efficiency (or dynamic speakers). These speakers respond well to eq as verified by the measurements below. If my speakers were Salon2’s for example, they would have a smoother response beyond 500 Hz and may require no eq at all, but they are at least 10 dB less efficient than the JBL’s. Audiolense’s partial correction can be set for any frequency and independently controlled in both the frequency and time domains.

Verification Measurements:

As mentioned in my previous article, Audiolense simulations are virtually identical to real world measurements using a 3^rd party acoustic measurement software like REW. While there is some variability based on smoothing algorithms used, my book shows in detail that these sophisticated DSP packages produce simulations that are virtually identical to their corresponding measurements.  I can also overlay exactly what the subs contribute versus just the JBL cabs. 

Frequency response:

-3dB points are at 12 Hz and 21 kHz and within a ±3 dB tolerance of the target design and better than that over most of the range. 

Step response:

All direct sound arriving at the same time and well behaved over 100 milliseconds of sound travel in the room. That nasty peak at 28 ms is gone.

Group Delay:

Mostly flat with natural rising delay at the very bottom of the response. A little ripple at 350 Hz.

Phase:

Again mostly flat with natural rising phase in the low end. Here we can see a bit of ripple a 350 Hz. Again unlikely it is audible, but I will investigate.

This Bruel & Kjaer application note on, Loudspeaker phase measurements transient response and audible quality, provides some insight for folks interested in this topic. The one limitation that is overcome with modern DSP software is the ability extract the minimum phase response, correct that, while independently correcting the excess phase response.

What is interesting is that not only the frequency and timing responses match well between channels, but so does the phase response and group delay. All of which are responsible for a speaker’s ability to completely “disappear”. All one is left with is the stereo illusion presented in 3D with a rock solid stereo phantom center image.

The most telling improvement is not only showing that the subs integrate seamlessly, both in the frequency and time domain, but how much the subs contribute to extending the bottom end of my JBL 4722’s:

The red and green traces are with the subs integrated. The blue and purple traces are without subs as measured in the previous article. I used the “bracket” method to determine my target preference for each setup. Even over a span of several months’ between measurements, and target designs, I arrived at virtually the same tonal response, except for the bottom end extension. This was starting with blank target designs in each case. It is interesting to me that I consistently end up with the same tonal response. I know what I prefer ☺

Finally, a zoomed in scale from 10 Hz to 150 Hz showing the difference it made to my setup integrating subs:

Listening Results

Wow, I shouldn’t have waited so long to add subs to my setup. The added weight in the bottom octave really compliments the double 15” impact, giving it the full club or concert sound I have been looking for. I love live music and anything I can do in my home system to give me the feeling like being at the concert or club is all good to me.

As a side note on the pic above. For critical listening, I move the coffee table out of the way, I took this pic while I was supposed to be working. Since this pic, I installed quiet curtains behind the speakers covering the windows. They do a really good job in quieting my overly live room to fit the upper RT60 spec limit for the size of my room.

I must say I am surprised how much music material I listen to actually has output below 40 Hz – virtually everything I have has some content below 40 Hz. I can listen with just the subs turned on. I can see why the Rythmik L12’s are recommended by folks with planar or electrostatic speakers. To borrow an Austin Powers or Jake Peralta word, toit!!

Here is a small sample of music I use to evaluate audio systems and simply enjoy the music. This subgroup contains tunes that have reasonably good dynamic range. But before I do, please allow me to make a short comment on the state of the recording industry:

Michael Jackson – Thriller is #1 in worldwide sales, and still is today, along with AC/DC – Back in Black, #2 in worldwide sales. Both are DR 12. I want more of this dynamic sound and less of the hyper compressed music that makes up, unfortunately, the majority of my music collection. 

While I get artistic intent, I feel the DR scales above fairly represent what is good and bad sound from a dynamic range perspective. See the bottom of the DR chart, where DR = punch and impact. Yah, more of that please. There is no excuse for overly compressed music today. It’s just wimpy loud sound.

If there is one thing we can collectively ask for as music consumers that would make the biggest impact on our sound reproduction systems is to allow the consumer to control the volume. Now back to our regular scheduled programming.

I could go on about each one, but I think I have gone on enough. Most were a new listening experience for me, discovering for the first time how much low frequency content was on each recording. If there is good low frequency content on the media, it is reproduced unlike what I have heard before from my system. Great fun.

Aside from a couple of later recordings, all of these recordings, mixes and masters are 16 years old or older. While my daughter bugs me that this is Dad music, it is sad to me that I have to go back 16 years or so to get a decently recorded, mixed and mastered rock album that has some dynamic range (check out the DR column in the playlist above). The more modern music I listen to on a regular basis, most unfortunately, is in the DR8 to DR6 range with too much dynamic range compression.

To be sure, my sub application is for music. However, the subs are fun with movies too. For example, the Jumanji remake with the drums and rhinoceros stampede shook my house so much my daughter came running out from her room wondering if there was an earthquake going on. Mission accomplished. However, these are light duty home theater subs, Rythmik has several larger subs designed for LFE HT applications. For music, I find these more than loud enough for my particular scenario.

Conclusion

While several audio DSP or DRC products can smooth the frequency response, Audiolense’s True Time Domain (TTD) correction is something to experience. Not only accurately and precisely time aligning drivers but also taking care of room reflections. I know of no other DSP on the market that can do this with this level of workflow automation. One can set up, take a measure, and be listening to a first good corrected response in under 30 minutes. Fine tuning after that is to one’s preference.

To me, accurate sound reproduction means the sound reproducing system (including room) is not altering the frequency or timing response arriving at my ears. Meaning a flat perceptual response within a ±3dB tolerance with no phase distortion or excess group delay. I want to hear the music arriving at my ears matching as closely as possible to the content on the recording.

Most loudspeakers are not time aligned and the timing response (i.e. delay and phase) gets worse when adding subs due to the long wavelengths involved. In addition, room reflections are inevitable due to the physical dimensions of our listening environments. While there are several subwoofer configurations that can help smooth out the bass, Audiolense DSP can pretty much smooth out the response. I show two subs integrating perfectly with my stereo mains and arguably achieving a smoother frequency response than adding more subs alone would do. Having time aligned subs with mains really shows off the transient impact of having the entire music wavefront arriving at ones ears at the same time. I can’t emphasize this point enough.

If you read JJ’s article linked earlier on, one can learn why we hear what we hear in small room acoustics. Audiolense takes advantage of this knowledge and programs the ability to control these parameters in a software DSP program. Includes user adjustable algorithms like frequency dependent windowing, which based on JJ’s research shows that the spectral balance (i.e. timbre) our ears care about is a blend of room interaction at low frequencies, and mostly direct sound in the mids and top end. 

Later arriving reflections have an influence on the perceived frequency response, and sometimes quite substantially. Therefore, a more psycho-acoustically correct frequency smoothing technique is used in combination. As a result, this is what Audiolense sees and corrects in the frequency domain. A frequency smoothing based on psycho-acoustic principles leads to a smoothed response that sits high in the comb filter region and avoids overcorrection of dips. 

These two psychoacoustic features are just the beginning of this very sophisticated and powerful audio DSP software program. At 64 bits of resolution, the calculations and FIR filter adds no distortion of any kind when convolved with the music signal.

If you are going the whole nine yards with digital XO everything with True Time Domain correction, Audiolense has automated most of the workflow. Once the XO’s are designed and satisfactory, then the workflow is basically the same as if one is working with a passive loudspeaker. It is quite the time saver.

I can recommend Rythmik subs and Audiolense to anyone looking to get the most out of their two channel or multichannel system. While I have been into DSP for quite a while, I should not have waited so long on adding subs. Using Audiolense, the subs integrate seamlessly with my mains as evidenced by the simulations and verification measurements – and my ears! Those subs are low frequency canons and really add weight below 40 Hz to give a deeper, but “toit” concussive sound quality. Those are Rythmik’s entry level subs. I am really impressed.

If you can achieve objective measurements similar to what I have shown in this article, I don’t think you would be disappointed with the sound quality. You may find correcting the bass in room is all the partial correction one needs, if the loudspeaker exhibits really smooth on and off-axis mid and high frequency response. The time domain correction can also be independently set for whatever frequency. It can be set to the same frequency as the partial correction above. This will correct the low end in the time domain, like the two examples of reducing group delay in the previous article and controlling reflections in this article. Or you may choose to apply an overall time domain correction, if you can always hear the tweeter arriving first, but just a partial frequency correction to correct below 500 Hz. Experimentation is encouraged.

Two other Audiolense features to be reviewed in a future article are user defined, mixed phase target design and multi-seat correction. For the latter, some folks feel is perceptually better than a single point measurement used for correction. Let’s see if that is true or not. Until then, enjoy the music!

Note: Mitch Barnett's previous article titled "Audiolense Digital Loudspeaker and Room Correction Software Walkthrough" can be found via the link below.

 I wrote this book to provide the audio enthusiast with an easy-to-follow step-by-step guide for designing a custom digital filter that corrects the frequency and timing response of your loudspeakers in your listening environment, so that the music arriving at your ears matches as closely as possible to the content on the recording. Accurate Sound Reproduction using DSP. Click on Look Inside to review the table of contents and read the first few chapters for free.

Mitch “Mitchco” Barnett.

I love music and audio. I grew up with music around me, as my mom was a piano player (swing) and my dad was an audiophile (jazz). My hobby is building speakers, amps, preamps, etc., and I still DIY today.

I mixed live sound for a variety of bands, which led to an opportunity to work full-time in a 24-track recording studio. Over 10 years, I recorded, mixed, and sometimes produced over 30 albums, plus numerous audio for video post productions in several recording studios in Western Canada.