Gifts from the Cinema

On more than one occasion, it has occurred to me that movies are, in many ways, the visual equivalent of music.  When done well, the result is an emotional connection with the recipient of the message.  Repeated viewing, like repeated listening, provides opportunity to deepen the connection and hence, the appreciation.

The two art forms often work together, each enhancing what the other brings to the audience, without diminishing the ability of either to stand on its own.  Music can profoundly affect how a movie (or a scene within it) is perceived.  Movies can also provide another means of finding great music we might not hear otherwise.

Some great movies I’ve enjoyed over the years have in turn led me to some great music.  I am confident the impact of films like Alfred Hitchcock’s “Psycho”, “North by Northwest”, and “Vertigo”, Orson Wells’ “Citizen Kane”, J. Lee Thompson’s “Cape Fear”, François Truffaut’s “Fahrenheit 451”, or Martin Scorsese’s “Taxi Driver” would have been radically different without the music of Bernard Hermann.

Similarly, Nino Rota’s music played a significant role in the movies of Federico Fellini as well as in Francis Ford Coppola’s Godfather trilogy.  Perry Henzell’s “The Harder They Come” is known for the reggae music that comprises its soundtrack.  Carol Reed’s noir classic “The Third Man” is highlighted by the distinctive sound of a zither, played by Anton Karas, who composed the score.

Recently, I saw an extraordinarily beautiful film from Italy that was released in 2013: Paolo Sorrentino’s “The Great Beauty” (aka “La Grande Bellezza”).  Aside from the story, the characters, the acting, the cinematography, and the visuals of this movie, I found the soundtrack captivating, with a variety of musical cues from different sources, in different genres.  I purchased the soundtrack disc(s) as well as the DVD and have found it as difficult to stop listening as it has been to stop viewing this new favorite.

This movie is loaded with musical gems, from the opening a cappella “I Lie”, performed by the Torino Vocalensemble, to Danish soprano Else Torp’s performance of Arvo Pärt’s “My Heart’s in the Highlands” (with lyrics from the Robert Burns poem of the same name), to the Kronos Quartet’s rendition of Vladimir Martynov’s “The Beatitudes”.  In addition, the soundtrack features some more pop oriented music including a sensitive performance by Damien Jurado performing his “Everything Trying” and even some club-oriented dance music.  You’d have to see the movie to understand how it all works together.  Musically as well as cinematically, there is much to treasure.

Winds of Change

Almost eight years after we recorded the first release for my Soundkeeper Recordings label (documented in the December 13, 2013 entry in this blog, like the album, entitled Lift), I was once again joined by Art Halperin and his band, Work of Art, for a new project.

For a long time, Art and I had discussed a follow-up to Lift and now the time was right.  Art had written a great new collection of songs, which the band had been rehearsing.  I had recently made some new additions to the recording setup in terms of upgraded power and microphone cabling (see the previous entry in this blog, entitled New Connections).  And I found just the right recording locale for the project.

Instrumentation for the songs includes a wide collection of different guitars including both nylon-string acoustic guitars and Martin steel-string acoustic guitars, a Dobro type resonator guitar, a Guild 12-string guitar and a few electric guitars, one of which is the Fender Stratocaster given to Art by Eric Clapton.  Along with the guitars, a mandolin, pedal steel guitar, and ukuleles are also present on the recording, while double bass and drums accompany the voices throughout.  The rich vocal harmonies are a big part of these songs, some featuring up to four voices behind Art’s lead vocals.

For those interested in the recording setup, the equipment for these sessions was as follows:

Microphones: Earthworks QTC-1 (aka QTC-40, matched pair, separated by a custom designed baffle)
Mic cables: Nordost Tyr 2
Interface: Metric Halo ULN-8 (serving as microphone preamps, analog-to-digital converters, digital-to-analog converters, and headphone amplifier)
Laptop: Apple MacBook Pro
Software: Metric Halo Console X (including its Record Panel)
Power cables: Nordost Heimdall 2 (for interface) and Nordost Purple Flare (for laptop)
Power conditioner: Monster Cable HTS-400
Vibration isolation: Custom made base to support laptop and interface

One of the many nice things about this project was that the players, having already done one Soundkeeper Recording in Lift, were already familiar with the process and the fact that they would be together, hearing each other through the air, for real, as opposed to being separated by headphones and baffles and listening to an electronic mix via headphones.  Everyone knew they had to pay close attention to each other and to how their own sound blended with the whole.  They all knew we were capturing performances, without the ability to “punch in” later to fix any mistakes.

I selected a local 19th century church as the recording venue.  It is a stone structure with a wooden interior and a warm acoustic, providing a good sense of air around the players but maintaining a nice sense of intimacy, ideally suited to this music.

My expectation was that the stone construction of the church would result in a relatively cool interior, even for our late June recording sessions.  The good news is that we all had a great time, even though my thermal assumptions were off by a good measure.  In short, the music wasn’t the only thing that was warm.  Several large ceiling fans keep the air in the church circulating but these had to be turned off during recording, as the mics very clearly picked up the quiet hum they produced.  Next time at this locale, spring or fall would make optimal seasonal choices for the best indoor climate, free of the sounds heating or cooling systems would necessarily add.

We recorded in the church on two successive days and all the hard work Art and the band put in preparing for the sessions was clearly in evidence.  I have commented before on just how great the feel is in Art’s music.  It pleased me to no end to find that others noticed exactly the same thing upon hearing the early playbacks.  What surprised me at first, but upon reflection turns out to be no surprise at all, is how all the comments used the same word.  When my wife (and most trusted listening partner) first heard the playbacks, she said “This is such a joyful album!”  Others have used the very same adjective, including two of the players in subsequent independent communication with me about the sessions.  The word came up so frequently that one of my early candidates for the album’s title was “Joyful”.

The music and performances are certainly full of joy.  As it turned out, so were Art and yours truly as we listened to the impact the new cable additions brought to the results.  I mentioned in the previous entry in this blog that this project marked my first use of Nordost’s Tyr 2 cables to connect my microphones to the ULN-8’s mic preamps, as well as my first use of third-party power cables, in this case Nordost’s Heimdall 2 feeding the ULN-8 power supply and their Purple Flare feeding the laptop power supply.  As I said in that entry, both Art and I remarked that we’d never heard recorded acoustic guitars sound so much like the instruments themselves.  The speed and extension on the double bass too, matched the sound of the instrument at the sessions like we’d never heard before.  (Thank you Nordost, for taking my recordings to a whole new level!)  While I’d have been pleased with “Joyful” as the title, in the end we decided on an equally fitting one we like even more: Winds of Change.

The recording format was 24-bit, 192k sampling, captured by the ULN-8 to .aif files.    As has become the norm for Soundkeeper, we will release it in multiple formats, from 24/192 (.aif or .wav) files-on-disk, to 24/96 (.aif or .wav) files-on-disk, to 24/96 audio-only DVD (in DVD-V format), to CD-R, to pressed CD.

One other thing we decided to do for this project was document some of it on video, to share with Work of Art (and Soundkeeper) fans, some of the “behind the scenes” views of the recording sessions.  The videos will be completed once the audio mastering is complete and the album art is done.  There is still some work ahead of us before the album can be released.

Making a record is most definitely much harder work than most folks might realize, but making Soundkeeper Recordings has been, and continues to be, a delight.  How fortunate I am to know Art and his band, and to be able to produce and engineer this album.  For someone who loves making records, it doesn’t get better than this.

New Connections

It was almost a year ago, in one of the earliest entries in this blog, entitled The High End Arrives, that I recounted some of my first experiences with better gear.  In both of the specific instances mentioned, my expectations were toppled.  First, a different turntable changed my thinking from “turntables just turn” to having a greater appreciation for just how much more is involved in retrieving music from the spiral groove.  In the second instance, a change of speaker cables taught me that everything the signal passes through has an impact on the final sound.

That was a valuable lesson, particularly, as I came to learn later, when applied to making recordings, not just playing them back.  While I was reading about debates regarding whether cables could make an audible difference, I was bringing my own to work when I started mastering for CD.  I’d found that replacing the “pro” cables in the studio (which connected the output of the master tape playback machine with the input of the analog-to-digital converters) with “audiophile” cables let more of the musical information in those tapes get through to the CD master.  It wasn’t that the cables I installed were making the sound better.  They just did a better job of getting out of the way.

How odd, it seemed to me, that in some quarters, folks were actually trying to legislate audio, lobbying New York City’s Commissioner of Consumer Affairs at the time, in an effort to make audio cable advertising illegal. (!)  It is one thing to listen and not hear any difference.  It is also understandable that one might not comprehend what mechanisms could possibly be responsible for the sonic differences others hear.  I certainly wouldn’t want to force anyone to use cables they don’t want to use.  But by the same token, please don’t take mine away because you don’t hear what I’ve been enjoying.

When I started Soundkeeper Recordings, I sought to use the simplest, highest quality signal path to make my recordings.  To this end, I tried replacing my professional microphone cables with a set of balanced cables from an audiophile manufacturer.  If cables made such important differences in playback systems and helped me create more faithful CD masters, I was interested in hearing what they did at the very front of the signal chain, connected to my microphones.  In retrospect, I am not surprised this turned out to be one of the more obvious places where doing a better job of getting out of the way resulted in more Life getting to the recording.  They made the pro cables sound coarse, grainy, and closed in by comparison.  In short, they revealed the sonic fingerprint those pro cables superimposed on everything.

In a post from November of last year, entitled Three Decisions (Part 1), I talked about my first experience with cables from Nordost.  When I first built my own studio, after spending a number of months auditioning a wide variety of candidates for cabling, I kept returning to Nordost cables as they always allowed me to feel like I was hearing past them, to the recording itself—which is exactly what I sought from the monitoring system in the studio.  Where other cables I’d used sounded “good” (something I consider to be a coloration), these seemed very clearly to allow the sound of the gear being connected—and ultimately, the recording—to pass without editorializing or superimposing their own sonic fingerprint.  I listened to a number of different products within their line and found a family resemblance insomuch as that ability to get out of the way.  The more expensive models just seemed to take it further.  And the balanced interconnects, used as microphone cables, showed me that my microphones were even better than I’d previously thought they were.  Price being a major consideration at the time, I started with their least expensive speaker cables and interconnects, which replaced cables that cost three times their price (and which, in terms of getting out of the way, they sonically left in the dust)!  Over the years, I’ve stepped up to more elaborate models within the line.

Cut to the present time.  I have used different cables over the years and have enjoyed continual improvements in each one’s ability to get out of the way and let more of the music through.  For the past several months, I have been using a new set of cables, covering the signal path from my microphones all the way to my loudspeakers.  I have also been using some types of cables that I’ve never tried before.  For example, I learned years ago that better loudspeaker cables and better interconnects (both for analog and digital signals) made for great strides in the quality of a recording or playback system.  What I’d never tried yet though, were replacements for the AC cables that came with some of my gear.  (I’d also never tried using a better HDMI cable for video or anything other than a basic USB cable to connect the hard drive that houses my music library.)

Most of the new cables are from Nordost’s Heimdall 2 series.  While I was curious to hear the whole system with the new cables in place, I was intrigued by the AC cables, so I started by replacing only the AC cables on the components that did not have captive cables.  The first AC cord went from the wall outlet from one of the dedicated lines feeding the studio, to the power distribution block.  The next one went to the power supply feeding my Metric Halo ULN-8, which serves as my digital-to-analog converters in the studio (and also as my microphone preamps, analog-to-digital converters and headphone amplifier during recording sessions).  Others went to the studio power amplifiers and subwoofers.

Experience has taught me not to assess any audio component until it has played music for at least a week—and with loudspeakers, many weeks.  While the basic character might be evident right out of the box, maximum performance does not occur until the component has been in use for a while, until it has been “burned in”.  (I have read a lot of theory on why this is the case, as well as arguments from some quarters as to why it cannot be.  Not surprisingly, the latter come from the same folks who would say I’m imagining the differences I hear between cables.  All I can say is, if I’m imagining this, I imagine it every single time my assistant switches to these cables without my seeing which are installed.  And I’m having a great time!)

As one who has long appreciated what good cabling can do for a system, I was surprised it took me so long to try replacement AC cables.  And I was absolutely thrilled at how much more alive the system sounded.  By then however, my curiosity about what Heimdall 2 would do for the rest of the system came to the fore and I replaced the speaker cables, analog interconnects, and digital interconnects (S/PDIF from the CD transport and the USB cable from the hard drive housing the music library for the server).  The system was now wired with Heimdall 2 all the way from the AC outlets to the loudspeakers.  I put the CD player on continuous repeat and left the studio, only returning to occasionally grab a listen or switch to a different disc.  I wanted to give the system plenty of time to get wherever it was going.

By the time I started the serious listening, it was one of those events where you want to listen to recording after recording (and can’t hear them all fast enough) to find out what the new changes reveal about them.  If the AC cables brought a new and previously unheard sense of “snap” and life to the system, upgrading the rest of the cables forced a reevaluation of the system’s limitations.  I am hearing the Magnepan 3.7s do things I didn’t think Magnepans can do.  Specifically, there is now a dynamic “slam” within the system’s capabilities that I had long thought was just something I had to trade in exchange for the multiplicity of wonderful things the speakers can do, that make me love and admire them so much.  The AC cables are certainly a big part of this but bringing all the other cables in the system to Heimdall 2 solidified it even further.

The other major change I noticed with the new cables is how much easier it is to hear individual parts in a recording, particularly with complex passages played by large ensembles but also with simpler arrangements played by smaller groups.  It is just so much easier than before, to focus the attention on an individual voice in a choir or an individual horn in a section, etc.  And the system was no slouch about this before.  It has just been elevated a couple of steps.  Big steps!

In addition to the Heimdall 2 that has transformed the system in the studio, I am using a pair of Nordost’s Tyr 2 balanced interconnects as my new microphone cables.  I had the opportunity to give these a real test a few weeks back, when I recorded what I expect will be the next release on Soundkeeper Recordings.  In addition to the Tyr 2 cables on the microphones, this was the first time I made a recording with the new Heimdall 2 AC cable feeding the power supply for the Metric Halo ULN-8 (again, serving as the microphone preamps and analog-to-digital converters during recording sessions, not to mention the digital-to-analog converters and headphone amplifier for monitoring during the sessions).  Also on hand was a Nordost Purple Flare (figure-8 type) AC cable, which replaced the stock cable on my Apple MacBook Pro laptop, where the captured audio was stored.

Back in the studio after the sessions, I heard the same benefits mentioned earlier, captured in the recordings.  How much of this was the result of the different AC cables and how much was contributed by the stellar Tyr 2 cables on the microphones, I don’t know.  What was obvious to me though, and to the artist too when he first heard the playbacks and voiced exactly what I’d been thinking, is that we’ve never heard recorded acoustic guitars sound this way, i.e., so much like the instruments themselves.  The artist and his band utilized a wide variety of guitars on this project, both acoustic and electric, from nylon stringed classical instruments, to various Martin steel stringed guitars, to a 12-string Guild, to a resonator guitar (along with a number of electric instruments).  The sound of each, as well as that of the mandolin, double bass, percussion and other instruments, was captured as we heard them during the sessions, to a degree that is new to both of us.

As I’ve been listening to these cables for a good while now and have been reporting my music and audio experiences in this blog, I wanted to share some of this but had no intention of writing a “review”.  There are a number of other models further up Nordost’s own line.  Based on my previous experience with the ones I’ve heard, I would expect each of those to take it up another step or two from what I’ve been thrilling to each time I listen.  Meanwhile, the new connections have taken my recordings and my listening to a whole new level.

 

Toward a definition of high resolution audio

We are starting to see the idea of high resolution audio gain some traction beyond the audiophile world, where it has been enjoyed for the past several years.  Some of the major labels, perhaps seeing new opportunities for commerce, have formed a working group to define exactly what high resolution audio is.

I think we’re going to see a wide variety of perspectives on this one.  The labels are using variations of the term “Master Quality”, with different designations, depending on whether the original source is an analog tape, a CD master, or some other digital format.  My take is this can be quite vague, particularly in view of the fact that there is such enormous variation from recording to recording, even within one of the above source formats.  In some ways, the sonic differences between recordings can far exceed the sonic differences between formats.

Another definition, which seems to come up a lot in the hobbyist fora, is “anything better than CD quality ”, meaning anything where the digital audio is encoded with a word length longer than CD’s 16-bits and a sample rate higher than CD’s 44.1k.  (Word length and sample rate are discussed in the previous entry, Is “too much” not enough?)

Sometimes I think terms like “Master Quality” or “CD quality” are oxymorons, like “the sound of silence”, “jumbo shrimp”, “living dead”, or “civil war”.

Personally, I would differentiate between “high resolution” and “not as low resolution”.  (How’s that for a selling point?  “This new album is not as low resolution as the previous one!” ;-} )

As I hear it, going from 16/44 to 24/44 is an improvement, as is going to 16/48 or 24/48, but I wouldn’t refer to any of these as “high resolution” for the simple reason that to my ears, they are not.  24/44 does not do as much damage to low level information as 16/44 but in my view, it still suffers from an inadequate sampling rate, as does 24/48.  The anti-aliasing filter (also discussed in the previous entry) is still way too close to the top of the audible range and its consequences reach down well into the audible range.  (Yes, I know some claim otherwise.  I’m still waiting to hear the audible evidence to support such claims.  So far, it speaks otherwise.)

As we get to the 2x sample rates (i.e., 88.2k or 96k), there is much less damage and perhaps I’d refer to these as “intermediate resolution”.  I say this because of what I perceive as the critical threshold that is crossed when 4x rates (i.e., 176.4k or 192k) are properly done.  While “properly done” still seems to describe the minority of devices carrying these numbers in their spec sheets, those that do achieve it do something I’ve never heard from any other format, including the best analog—and that is what I have been referring to as “getting out of the way”.  This alone makes the 4x rates, to my ears, a bigger jump upward in quality over the 2x rates than the latter are over standard CD.  And this alone differentiates them in my mind as being true high resolution.

While the intermediate resolution rates can sound very good, this is exactly what I believe prevents me from thinking of them as high resolution:  they sound.  I don’t want gear or recordings or formats that sound “good”, “detailed”, “smooth”, etc., I want them to not sound.  I want them to get out of the way, leaving the sound to that which is being recorded, played and listened to—the performance.

I’m reminded of how the video world defined intermediate resolutions—those better than standard but not really high—with the term “extended”.

Personally, I’d place anything at 1x rates (or with a 16-bit word length) in the SRA (“Standard Resolution Audio”) category.  This would include 16/44, 16/48, 24/44 and 24/48.  With the latter two, my experience has been that while the added word length helps, the limitations of having the low-pass filtering so close to the audible range—and thus, its effects within the audible range—mean that in the end, these are all effectively just minor variations of “CD resolution”.  (I would ultimately consider 16/96 or 16/192 SRA also.  There is, in my view, no good reason to record with less than 24-bits and if the release is going to be at one of these sample rates, I would deem word length reduction to 16-bits counterproductive and just plain silly.)

The above is at odds with what appears to be the more common “anything better than CD” definition of high resolution.  To me, that is like saying anything better than a Big Mac is filet mignon.  Or anything better than Night Train is Dom Perignon.  I don’t think so.  I think there are intermediate levels and that it takes more than being better than mediocre (or just plain bad) to quality as “fine”.

Any 2x recording (88.2 kHz or 96 kHz) again, at 24-bits, I would refer to as ERA (“Extended Resolution Audio”).  Now we have a real improvement in fidelity to the input.  It doesn’t quite get out of the way, but to my ears, it is noticeably better than SRA.

HRA (“High Resolution Audio”), I would reserve for 4x recordings (176.4 kHz or 192 kHz) again, at 24-bits.  Properly done and played back on gear that can actually perform at these rates, we have the first format in my experience that is truly capable of getting out of the way.  This is what high resolution audio is about.

By these definitions, I would consider Soundkeeper Recordings’ CDs as well as our CD-Rs to be SRA.  The latter is certainly closer sounding than the pressing but ultimately, they’re both 16-bits.  I’d call our 24/96 DVDs and 24/96 files-on-disc releases ERA and our 24/192 files-on-disc releases HRA.

Of course, as I’ve long said, my belief is that 90-95% or more of a recording’s ultimate sonic quality has already been determined by the time the signals are leaving the microphones.  The delivery format just determines how much of that original quality is available for playback.  I’d rather hear a CD (or even an mp3) of a Keith Johnson recording than a 24/192 (or the original masters) of recordings from a lot of other engineers.  But best of all, is the HRA version of Keith’s work.

Is “too much” not enough?

As digital audio and the means of playing it back mature, there is an increasing divergence of perspectives to be found on the Internet.  Some revel in the sonics of music heard at high resolution, while others argue that the CD standard is not to be audibly improved upon and still others want even higher resolution.  All this while Joe and Jane Average download one song at a time at resolutions that throw away at least 75% of the information contained on a CD.

There are new efforts from some quarters to show Joe and Jane what they’re missing and to elevate what the download services offer.  The idea is to, at the very least, deliver 100% of what the CD offers and at best, deliver true high resolution.  Yet these efforts have spawned Internet “papers” and articles in effect, ridiculing the very idea of high resolution and arguing the supposed inaudibility of its benefits, or worse, suggesting that high resolution by definition will sound worse, not better.

I can’t speak for what others find but I can say that whatever these folks are reporting is quite the opposite of what I experience.  I’m hearing fidelity such as I’ve dreamed about for years and when I read those stories, they strike me much as though the authors are trying to convince me there are no colors in a rainbow.

The arrival of high resolution digital has the potential to fulfill the promise digital audio first made more than a quarter century ago.  Back then, astute listeners wondered at the marketing mantra “perfect sound forever” while cringing at the dry, bleached and airless sounds delivered by the first CD players.  While a great deal of progress has been made during the intervening years, the inherent limitations of the format remain.

Looked at in the most rudimentary fashion, the specifications for CD would, on the surface, appear to be all that is needed to perfectly reproduce anything that can be heard.  Human hearing is nominally sensitive to frequencies from 20 Hz through 20 kHz (i.e., 20 cycles per second through 20,000 cycles per second).  As we age, the top end limit decreases and most adults would be lucky to hear 15 kHz.  With CD, music is sampled 44,100 times per second.  That is, the digital recorder “looks at” the sound 44,100 times every second and captures a sample.  According to the theory, all frequencies below half the sample rate, (in this case, all frequencies below 22,050 cycles per second) will be captured accurately and since this is well beyond what most folks can hear, it all sounds quite neat.

These digital samples are each a series of digital bits, with each bit representing one of two binary states or values, often thought of as “ones and zeros”.  Each sample is stored in a digital word.  The CD standard uses 16-bit words, where each sample contains 16 values.  The particular combination of ones and zeros represents the level (i.e., volume) of each sample.  A series of 16 zeros (i.e., 0000000000000000) would be the lowest level that can be encoded and represents complete silence.  A 16-bit word representing an intermediate level might look like this: 0111011110101110.  The highest level would be 0111111111111111, a zero followed by 15 ones.  (For technical reasons which are beyond the scope of this entry, the loudest value is not a series of 16 ones.)

A word length of 16-bits allows up to 65,536 different levels to be represented.  The difference between the loudest sound that can be captured and the noise floor of the format is called the signal-to-noise ratio.  Signal-to-noise ratio is measured in units of loudness called decibels (dB).  For a 16-bit format like CD, the signal-to-noise ratio is approximately 96 dB, which means the noise floor (the inherent noise of the format) is 96 decibels below the loudest sound that can be captured.  This is much quieter than vinyl or analog tape.  Any hiss heard on a CD is captured from the source and is not inherent in the medium.  Many folks confuse the signal-to-noise ratio specification with dynamic range (the difference in level between the loudest possible sound and the lowest sound).  We’ll come back to this later and see why this is misguided.

The problems start when we move from the theoretical to the practical.  (Someone, perhaps it was Yogi Berra, once said “In theory, there is no difference between theory and practice, but in practice, there is.”)  When digital audio is recorded, any frequencies above half the sample rate can cause problems – they engender aliases or aliasing distortion, false frequencies that are not part of the program material.  In order to avoid aliasing, when digital audio is encoded, as well as when it is played back, most digital processors use a filter to ensure that no frequencies above half the sample rate can pass.  These anti-aliasing filters have audible side effects, manifesting in the time domain – the signal gets smeared in time.  Some designers will use gentler filters to minimize the time smear but in doing so, they cause the higher frequencies to fall off prematurely.  A number of modern playback devices have user-selectable filters where the listener can select between steep filtering and its associated time issues or gentler filtering and its associated frequency issues.

So, while CD can capture all the audible frequency range, the requisite filtering means the frequencies delivered to the listener are not all arriving on time or are not all arriving in the same proportion in which they were captured, or some combination of both of these.  One great advantage of the higher sample rates is that the anti-aliasing filter is moved far above the audible range.  This allows gentler filtering to be used without affecting the audible frequency range.

In recent years, thanks in no small part to formats like DVD and others, which are capable of storing more information than will fit on a CD, digital audio has grown up from the 16-bit words and 44.1 kHz sample rates by which sound is encoded for CD.  We’ve had 24-bit audio with sample rates of 96 kHz, 176.4 kHz and 192 kHz.  For reference, a 24/96 (24-bit, 96 kHz) version of a given recording contains more than three times the information contained in the same recording at 16/44 (16-bit, 44.1 kHz).  A 24/192 version contains more than six times the information.  And where a word length of 16-bits allows up to 65,536 different levels to be represented, going to 24-bits increases the dynamic resolution 256 times, allowing up to 16,777,216 different levels to be represented.

The widespread use of computers (and computing devices) for audio playback has enabled the proliferation of high resolution audio and emancipated music from the confines of silver discs and the limitations imposed by the process of retrieving music from these in real time.  (Separating the processing “overhead” from the playback will provide higher quality playback.)  Good as the best disc players and transports can be, my experience has been that there is invariably a loss of focus and fine detail, often subtle, sometimes not so subtle.  It is only via proper computer playback that I’ve heard results that I find indistinguishable from listening to the master used to create those silver discs.

This is good news, even for music at CD resolution, because the listener at home can now hear what is effectively the CD master itself.  However, while the limitations of playback from molded disc have been removed, the limitations of the format remain.  In addition to the frequency and time-related issues brought about by having the anti-aliasing filter just above the audible range, there are the consequences of inadequate word length.  Although the noise floor with a 16-bit medium like CD is 96 dB below the loudest possible sound that can be captured by the format, many often confuse this signal-to-noise ratio with dynamic range.  The assumption is that if the noise floor is 96 dB below the loudest sound, sounds just above the noise floor will be captured with the same fidelity, providing a range of dynamics as wide as the signal-to-noise ratio.  In fact, with a 16-bit medium, the fidelity plummets at lower levels.

The full resolution, in this case 16-bits, is only realized for sounds near the top of the volume range.  Each bit captures about 6 dB of the dynamic range (about 6.02 dB to be more precise but let’s use 6 in this example to keep things simple), so in a 16-bit system, sounds lower in level than 6 dB below the maximum will effectively be captured at less than 16-bit resolution.  To wit, if this lower level information is say, 12 dB lower in level, it will be encoded at what is effectively approximately 2 bits less than the full resolution of the format (i.e., 14 bits in a 16-bit recording, 22 bits in a 24-bit recording). If it is say, 36 dB lower in level, it will be encoded at what is effectively approximately 6 bits less resolution (i.e., 10 bits in a 16-bit recording, 18 bits in a 24-bit recording).

Some information, such as the trailing end of reverb as it fades away, or the higher harmonics of musical instruments, can be well more than that 36 dB lower in level than the loudest sounds and will be encoded with resolutions corresponding to fewer bits.  This results in the thinned, bleached and coarsened instrumental harmonics in even the best 16-bit recordings, as compared to a good 24-bit recording (or of course, the original sound in real life).  It also results in the defocusing of the spatial information and in the relative airlessness in the 16-bit recording compared to a good 24-bit recording (and real life).

While the level meter may show a peak on that 16-bit recording that is within the top 6 dB, this, like the waveform views shown by some computer software, is only a view of the “top” part of the musical waveform — the loudest part.  Sounds and components of sounds that are underneath the top part (i.e., in the background) are not captured as faithfully.  Accordingly,  when considering the dynamic range of the format, it is a good idea to take into account the relative distortion at different levels within the range.  If increasing distortion is not desirable, the real dynamic range potential is going to be considerably less than what the spec sheet might suggest (or is often echoed in the audio press and in some places on the Internet).  Note that even with low level information as in the examples above, a 24-bit recording still delivers more resolution than a 16-bit recording at its best.

Why then, would someone publish a “white paper” against higher resolution or declare that resolutions like 24/192 are “pointless” or worse?  A few possible reasons come to mind:

  1. The higher sample rates place significantly increased demands on the gear used to record and play them back.  For example, digital gear contains an internal clock to control the timing as the device encodes or decodes the stream of digital samples.  Spacing between the samples must be kept accurate or the reconstructed analog waveform that we hear will not have the correct shape and hence, will not provide the correct sound.  Irregularities in timing are referred to as jitter.  Higher sample rates also mean the analog stages of the gear must be able to perform at the wider bandwidths.  Perhaps the folks complaining about high resolution are using gear that does not have clocking that is up to the task and analog stages that can perform at high bandwidth.  Such will either not reveal any benefits or will actually sound worse than they do at the easier, lower rates like 24/96. (This is true of a number of “professional” units as well as those sold to audio enthusiasts.  A  built-in, $250 “soundcard” simply won’t do it, regardless of what the specs claim.  In today’s market, it may cost 10 times this amount for a device truly capable of revealing the potential of these sample rates.  Maybe it is no wonder these folks hear little or no difference.)
  2. It could be possible that the rest of the system these folks are using isn’t up to resolving a wide band recording.  Or it could be that these folks are just not sensitive to these particular differences.  I’ve always found that different folks have different sensitivities to different aspects of sound.
  3. Perhaps they believe CDs (or 24/96) already sound identical to the input signal.  If that is the case, I can understand that anything more would seem wasteful.

Sample rates like 176.4k and 192k don’t, as some have erroneously suggested “have more jitter”.  Sample rates don’t have jitter.  As stated above, higher sampling rates do place greater demands on clocking accuracy (just one reason why buying a DAC
(digital-to-analog converter) “by the chip” is at best a foolish enterprise).  They also place greater demands on the analog stages surrounding the digital stage.

Why some would see these characteristics as “flaws” (and write papers or articles on the subject), I don’t understand.  I’ve always gone with empirical evidence over theoretical analysis; that is, when “theory” and direct experience are at odds with each other, I’ll tend to seek a new theory.  (As I see it, theory should explain the experience, not the other way around.)

All this to say, when a firmware upgrade enabled 192k capability in the converters I use for my work, I approached it conservatively — even continuing to do a few recording sessions at 96k because I was familiar with it and could be confident in the results.  But then I started running tests at 192k and quite quickly found I had to get my jaw up off the floor: for the very first time in my experience, I was hearing (with this device anyway) a recording device “disappear”.  I had never heard that before, even with the best analog recorders and most certainly nothing close with the best digital recorders, even with this very device when used at 96k.

Now I felt a threshold had been crossed (I’ve read similar words since then from one of my favorite audio engineers, Keith Johnson).  The results no longer sounded like “great digital”; they no longer sounded “digital” at all.  They didn’t sound like “great analog” either.  The jump from 24/96 to 24/192, when done well, is to my ears a much more significant jump than the one from 16/44 to 24/96.  It’s all about that threshold; this is the promise digital made in 1983, finally and for real.  (While it certainly sounds more faithful to the input signal than 16/44 does, 24/96 doesn’t yet, to my ears, “get out of the way”.  Having the anti-aliasing filter moved well up and away from the audible range definitely helps but it is the rates like 176.4k and 192k where I find the threshold is crossed.  Interestingly, to me, while many speak of the treble response, I find some of the greatest benefits to be in how much more lifelike I find the bass.)  24/192 pointless?!?  Only if real progress in music reproduction is pointless.

While some decry the higher sample rates, either declaring them to offer no audible sonic differences from the lower ones or to offer inferior sonics (!) to the lower ones, other voices are talking even higher numbers.  We are seeing marketers talking 32-bits and sample rates like 384 kHz.  As long as there are customers who are taken in by sheer numbers, there will be those who see an opportunity for commerce, who will accommodate them with sheer numbers.

I get a chuckle out of these things because my experience has been that in reality (that is, with the hardware or software one can go out and buy today, as well as the recordings one can purchase to play on these), I see gear that isn’t particularly clean at 24-bits.  I see other gear which, when presented with 4x sample rates (176.4k or 192k), performs worse than it does at 2x rates (88.2k or 96k).  Yet the spec sheets and ads say “24-bit” (or more) and they say “192k” (or more).  And the reviewers simply echo the numbers.

In the here and now, if it is a minority that can achieve the performance potential of 24/192, I take claims of higher numbers as a joke at best and cynical marketing at worst.  Just my opinion of course but with so few showing they can design for 4x rates, why would anyone think those same few could deliver 8x rates (or more)?  I find it interesting that those claims are not coming from the designers of gear that can achieve the potential of 4x rates.  (We have the equivalent of makers of 2-cylinder subcompacts claiming to make cars that an outrun a Lotus!)

The numbers game isn’t limited to hardware.  I’ve seen one company release CDs they claim were made with “32-bit mastering” and another claim “100 kHz resolution”.  (Do they have 100k gear or are they rounding up from 96k?)  Does anyone think those CDs are anything other than 16-bit, 44.1k?  If there are such folks, I have a fine bridge in New York City to sell.  The tools I use to create a CD master have 80-bit data paths and I’m working at 192k.  The higher quality tools do result in a higher quality CD but should I then say they are made with “80-bit mastering”?  Or that they exhibit “192 kHz [or 200 kHz] resolution”?  I’d rather make records than sell bridges.

The finest 24/192 I’ve heard to date has given me back recordings I have not yet been able to discern from the direct input from my microphones.  (To be clear, I am referring to gear that actually seems to achieve the potential of these numbers and not merely gear that sports them on a spec sheet.)  Would 32/384 sound better?  I suppose I’d have to hear the flaw(s) in properly done 24/192 first.  And second, if the first condition was met (and in my experience, it has not even been challenged yet), that 32/384 gear would have to actually achieve the potential of that resolution and not merely claim it on a spec sheet.  For me, right now, it is just marketing.  Someday perhaps, we’ll have the audible evidence.  Perhaps.  Right now, I’m trying to imagine how it might be better than what is (so far) indistinguishable from the input signal.

24/192?  32/384?  64/768?  Or should I wait for the 128/1536 version?
Is the best of today’s 24/192 too much?  Is it not enough?  I think it is just right.

 

Can you hear what you’re doing? (Part 2)

Last time out, in Can you hear what you’re doing? (Part 1), I talked about setting up monitoring to maximize its ability to “get out of the way” and allow better access to the recording itself.  This time, we’ll take it to the next step and talk about addressing the room itself, whether a studio control room, a mastering room or a home listening room.

Those who have experienced a listening space with properly treated acoustics will understand why I said monitoring is more than just the speakers.  It is the room in which one listens.  It is where in the room the speakers are located.  And where in the room the listening position is located.  And where just about everything else in the room is too.

By determining speaker (and listening position) placement first, before anything else is added to the studio or listening room, we can maximize the ultimate potential of the monitoring by minimizing the contribution from the room.  This is the starting point that determines how effective acoustic treatment can be when going to the next step in monitoring accuracy, which is getting the room out of the way.

The two main things to consider when addressing the acoustics of a typical studio or listening room are resonant modes in the bass and early reflections in the treble.  A quick search of the Internet or the printed literature will reveal a wide variety of strategies.

Before going into what I’ve found that works, it is important to mention the idea of so-called electronic “room correction”.   The basic idea behind room correction (sometimes also called “speaker correction”) is to measure the frequency response of the system in the room and then apply complementary alterations to the signal itself.  In other words, special test signals are played through the system while a microphone captures the sound.  The microphone output is sent to a device (nowadays, a computer) that will analyze the signal and provide a graphic representation of how the combination of the system and room responds at different frequencies.

In a typical frequency response measurement, all frequencies are fed to the device or system at the same intensity.  Ideally, that device or system will then deliver all these frequencies at the same intensity.  This would be called a “flat” frequency response.  If the device or system caused certain frequencies to be exaggerated, the response would show bumps or peaks that correspond with the amount of exaggeration.  If the device or system caused certain frequencies to be diminished, the response would show troughs or dips that correspond with the amount of diminution.

With electronic “room correction”, if the response shows a peak at a frequency of say, 100 Hz in the bass (i.e., the frequency of 100 Hz is being exaggerated), the input to the system will be altered so that it is fed a correspondingly smaller amount of 100 Hz.  In theory, the end result will be equalized — the correct amount of 100 Hz is delivered by the system.

So much for the theory.  I know a lot of folks who love the results of using such an approach to dealing with their rooms and I would never argue with whatever brings anyone their listening pleasure.  I’ve listened to a number of designs for this electronic tack and to my ears, while the results always sound very different, they don’t sound better.  They often sound quite a bit worse.  After a closer look at the assumptions being made with the electronic approach, I found this isn’t at all surprising.

We’ll start with the assumption that the microphone hears the same way a human being hears.  This might be true if the human had an extraordinarily narrow head with a single ear in the middle of their face (or perhaps on top of their head).  Many electronic recipes attempt to deal with this by taking several measurements with the microphone in a different position for each, then averaging the results to arrive at the final response.

A larger assumption is that what the listener hears is the sum of the direct sound from the speakers combined with the sound of the room.  It would be easy to get fooled into thinking this is the case if one looks at the sound in only one dimension — that of frequency response.  But sound cannot exist independent of time.  Without time, there is no sound.  In fact, the direct sound from the speakers, having the shortest path to the listener’s ears, always arrives before any contribution from the room.  The first thing we hear is the direct sound from the speakers.  Then we hear the room responding.  They are two things, not one.

Let’s look at what this means.  With perfect speakers and a room resonance at say 100 Hz, the electronic system will drop the level of 100 Hz in the signal fed to the speakers.  The formerly perfect speakers will now deliver a direct sound that has a dip at 100 Hz — the sound will be thinned, losing fullness and body.  Since there is less of the frequency that excited the room, there will be less of the resonance from the room.  This would be fine if the sound from the speakers and the sound from the room summed algebraically into a single entity and were not, in fact, separated in time.

When I listen to such a system, instead of the problem of the room being fixed, I hear instead, the creation of an additional problem.  The electronic mode of addressing the room has taken one issue and turned it into two issues.  The first arrival, the direct sound from the speakers, determines the overall character.  In this case, I hear the loss of fullness and “meat” resulting from the diminution of 100 Hz in the input.  Next, while somewhat lessened, the room is still resonating at 100 Hz.

This leads to another assumption, which is that those peaks and dips in the frequency response are the problems with the room.  Here again, looking at the issue only in terms of frequency response is to miss the bigger picture entirely and that is this: room issues are time-based, not amplitude-based.  They occur over time and are not merely changes in the level at which different frequencies are delivered to the listener.  When the room resonates at 100 Hz, it is “holding onto” that frequency, “ringing” at that frequency, after that frequency has already stopped in the signal.  The peaks and dips in frequency response are symptoms of room issues, not the issues themselves.

In the attempt to address the symptom, the source of the issue remains and an additional problem is created.  While the level of 100 Hz in the example above might be diminished, as long as the resonance is being excited it will cause the time-based issue to exist.  There will be less 100 Hz but what is there will still continue to ring after this frequency has already stopped in the input signal.  The ringing fills in the quieter parts of the signal, obscuring low level detail and disturbing the pace of the music.

As I see it, attempting to address a time-based problem by using an amplitude-based “solution” is like trying to fix a broken arm by wearing a different hat.  Much better, in my view, to address the problem at its source: the room itself.  Earlier in this entry, I said that when addressing room acoustics, we need to consider resonant modes in the bass and early reflections in the treble.  (Another thing about the electronic method is that it doesn’t even attempt to address early reflections.  It is possible to imagine how this might be accomplished but it would involve even more degradation of the direct sound from the speakers.)  The most effective means of acoustic control in my experience have been the modern day iterations of the “functional sound absorbers” engineer Harry Olsen first proposed more than half a century ago.  These cylindrical “traps” are available as commercial products, Art Noxon’s excellent Tube Traps from ASC (Acoustic Sciences Corporation) or you can build variations on the theme yourself with instructions that can be found on the Internet.  Cylindrical traps can have additional benefits in the treble range, which we’ll cover next.

The example above mentioned a single room resonance.  Typical rooms will have several.  There will be a fundamental resonance frequency for each room dimension (length, width and height).  There will also be harmonics at double each of these frequencies and, depending on room dimensions, at quadruple these frequencies too.

In Can you hear what you’re doing? (Part 1), I mentioned an article I wrote a while back called Setting up your monitoring environment.  From that article:

“Controlling early reflections is as simple as putting absorbent material at the points where the reflections occur.  There are two such points, one for each speaker, on each wall as well as on the ceiling (and uncovered, hard flooring).  Imagine the walls and ceiling of the room are mirrors.  From the listening position, you’d see a reflection of each speaker on each of the walls and on the ceiling.  If the floor is not covered with a carpet or rug, you’d see a reflection of each speaker here too.  An easy way of finding these points is to enlist the aid of an assistant who will hold a mirror up against each wall while you sit in the listening position.  With the mirror at your (seated) eye level, the assistant moves the mirror along the wall until you can see the reflection of one of the speakers.  When you can see one of the speakers in the mirror, you’ve found the point on the wall to place the absorbent material.  Have the assistant continue moving the mirror until you see the other speaker.  Now you’ll have the second place on that wall that will need the absorbent material.  Do this for each wall as well as the ceiling if possible.  A carpet or rug will work well to prevent early reflections from the floor.”

“While there are many types of foam sold for this purpose (and they are better than nothing at all), a much better solution lies with the modern iteration of the “functional” traps mentioned above.  The best commercial designs as well as the better DIY (do it yourself) variety will be cylinders that have half their surfaces covered with a material that is reflective in the treble while the other half is fully absorbent.  Too often, the use of foam to control reflections results in a dead feeling in the listening space that is neither natural sounding or comfortable.  The common mistake is in the perception that if a little is good, more must be better.  In fact, what is needed is absorption of the early reflections without affecting the later ones.  Further, diffusing or scattering these later reflections contributes to the naturalness and comfort of the room.  Cylinders with reflective halves allow the sound to be tailored as each cylinder is rotated.  When placed at the early reflection points, the absorbent half can be oriented toward the speaker it is nearest, while the reflective half will help diffuse the sounds arriving later in time, in this way maintaining the natural ambience of the listening space.”

In addition to addressing room modes in the bass and early reflections in the treble, diffusion is the third element in room treatment.  Many photographs of studios and listening rooms published in magazines or on the Internet show diffusors installed near the speakers and some manufacturers even advise their customers to do things like this.  Placing diffusion near the loudspeakers (or behind the listener’s head or on the middle of the ceiling) will only ensure the early reflections that should be absorbed (and thus prevented from reaching the listener) are instead scattered, guaranteeing they will reach the listener’s ears. This hardens the sound, shrinks the soundstage and defocuses images.  To avoid these, diffusion should be used only for late reflections.  Place diffusors so they are facing away from the speakers.  Other photographs will show studios with foam or other absorbing material completely covering the walls.  To avoid a “dead”, closed-in sound, absorption should be used only at the early reflection points.

With cylindrical type sound treatments, I orient them so the absorptive half of the cylinder is aimed at the nearest speaker, thus, the “live”, reflective half is aimed away from the speaker.  Traps that are equidistant from both speakers are oriented so their absorptive half faces the center of the room.  These cylinders perform bass trapping, early reflection absorption and diffusion, all in a single device.  With some placed at the early reflection points, the soft side performs the absorption and the “live” side only gets sound that has been around the room already.

By the way, there is another application for these which can work wonders in smaller recording studios.  Using a loose ring of traps around what is being recorded, with the soft sides facing the center of the circle, the diffusive sides keep sending reflections from the room back to the room.  When this is properly done, the mic(s) inside the circle “think” they’re in a larger room than they are really in.

Lastly, a few ideas for super inexpensive room treatment.  These won’t work to anything like the same extent as the cylinders mentioned above but they will provide an idea of what proper treatment can do:

  1. Gather at least a dozen cardboard boxes (the light brown cartons in which items are delivered to supermarkets or in which mail order items are shipped).  Ideally, these will be approximately 16” x 16” x 24” (~40 cm x 40 cm x 60 cm) in size.  Two dozen boxes would provide an even better idea.
  2. Fill all of the boxes as tightly as you can with individually crumpled sheets of newspaper.
  3. Tape the boxes closed and stack them to create columns at least 6 feet (~ 1.8 meters) tall.
  4. Place a column in each corner of the room.  If you have gathered more boxes, also place a column at the half-way point along each wall.
  5. To avoid reflections from the columns, cover them with thin towels, blankets or other absorbent materials.  For a neater appearance, cover them with any soft, absorbent fabric you find attractive.
  6. The 5 steps above will (begin to) address the room’s resonant modes in the bass.  To address the early reflection points, find them using the information from earlier in this entry and cover them with thin towels, blankets or other absorbent materials.  Here again, an absorbent wall hanging may have more eye appeal.
  7. Now just listen to some music and notice the increased pitch definition in the bass, the increased sense of “punch” and tightness of timing, smoother treble, more easily audible low level detail and (if the recordings contain it) the improved sense of three-dimensionality and focus.

With speakers and listening position optimally placed, treating the acoustics of the room by addressing resonant modes, absorbing early reflections and diffusing later reflections, helps get the room out of the way, making it even easier to hear past the monitoring all the way to the recording itself.

Can you hear what you’re doing? (Part 1)

If there’s one thing the advent of digital audio has accomplished, it is what I call the democratization of record making.  Unlike the days when musicians needed to interest a large company in order to get a record made, today many have small studios of their own and with the help of the Internet, self-release their music to the world.

While the technology and the means have certainly become much more widely available than they used to be, the information on how to use the technology has not proliferated to anything like the same degree.  There are plenty of magazines, both print and Web-based and several Internet fora where recording enthusiasts gather and these provide “how to” instructions. What is missing however, is the reasoning behind the how: the why.

So the new studio owner buys the hardware and software they read about and proceeds to turn the knobs, real and virtual, and then wonders what went wrong.  Not in every single case of course but from what I’ve heard over the years, the ones who are truly pleased are the exceptions.  Perhaps they sought something that did not sound like a true representation of themselves and their instrument(s).  Nothing wrong with that.  What is “good” or “better” or “best” depends entirely upon precisely what one seeks.

For those who seek to make recordings that sound less like recordings and more like musical performances (real or imagined), the standard recipes won’t work.  They are designed to achieve certain types of sound.  They are not designed to “get out of the way”.  (I use that phrase often lately when discussing audio gear or setups or recordings that I’ve found particularly involving. To my mind, they work because they “get out of the way” and allow the listener better access to the music.)

This blog entry will be the first in a series written with the hope of helping musicians and other recordists who are interested, like myself, in studio setups and recordings that get out of the way.  The series will not necessarily be consecutive in terms of publication (there may be other topics interspersed along the way) but the goal will be to raise some issues not raised elsewhere.  If these provide food for thought and perhaps inspiration for trying something different, I’ll consider them successful.  For those that don’t make records or don’t play instruments but who comprise the audience, the listeners, I hope there is something here of interest for you as well.

Above all, my recommendation is to not simply take my word for what you can expect to hear, since I can only report on how sounds strike my ears.  I encourage all to listen for themselves and draw their own conclusions.  Remember that asking any three audio folks a question will result in at least four different answers (five of which may well be wrong).  Only listening for yourself will tell you how something sounds to you.

In some earlier entries in this blog, I’ve mentioned something I’ve called “The Questions”.  To quote from one of those entries, “These are questions that need to be asked if one is ever to arrive at answers.  They are the questions I’d never seen mentioned in any of the books on recording I’d ever read or in any of the magazines.  They are the questions I was never taught to ask when I was an assistant engineer, the questions that students in today’s “audio engineering” schools never encounter.”

Let’s start our exploration by asking the first question I always ask about any studio: “Can you hear what you’re doing?”  This can be rephrased to accommodate listening setups as well as recording setups: “Can you hear past the system, all the way to the recording itself?”  Seems like an obvious question – at least it should be – but the fact is, in my experience, monitoring is all too often the weak link in most studios I’ve visited.  Since every decision regarding the sound at every step in the process of record making is based on what the monitors tell us, if you can’t hear past the monitoring all the way to the recording, if you can’t hear what you’re doing, you can’t determine how your recording is going to sound.  You can’t make it sound the way you want it to because you don’t know how it sounds.

Many studios have different sets of monitors and these all have very different presentations.  (This is discussed in the entry called Why doesn’t it sound (in here), like it sounds out there?)  Folks will often take a reference out of the studio to “see how it sounds” on some other system or even in the car(!).  Each provides its own view, like lenses with different tints or like prisms but more often than not, none simply gets out of the way.  From the blog entry cited above: “After all, if the engineer can’t hear what they are doing, the best they can do is attempt to blindly steer in the desired direction but the results are effectively left to happenstance.  It occurred to me that adjusting sound while referencing typical studio monitoring is like mixing paint colors while wearing sunglasses.  Over the years, a few folks have claimed to be able to hear “around” the monitors but the audible evidence always tells a different story.”

Certain types of systems will always apply certain types of colorations to how the recording is presented.  That will not change.  A system that gets out of the way, allowing access to the recording itself, removes any questions about what has been captured and how well (or not) it has been captured.  A recording that sounds right on such a system will sound its best on the greatest number of other systems.

Does that mean that everyone needs to buy a certain type of speaker and all will be well?  That would be nice but unfortunately, it doesn’t work that way.  Monitoring is more than the speakers.  It is the room in which one listens.  It is where in the room the speakers are located.  And where in the room the listening position is located.  And where just about everything else in the room is too.  The good news is that by paying attention to all these things, just about any speaker can be helped to get just a little more out of the way.  While the basic character, the basic potential of a given speaker design won’t be changed, in most instances, whatever that potential is can be a lot more fully realized.

This is a big subject and there is much to be said.  This time out, we’ll just start with a few ideas to experiment with.  To start, let’s talk about acoustics. To keep it simple, we’ll break the subject into two areas: bass acoustics and treble acoustics.  Every enclosed space, meaning every listening room, every studio and control room that isn’t outdoors, will have resonant modes.  These are frequencies in the bass where the room tends to “sing”.  When the speakers present content with these frequencies (or their harmonics), the room will tend to “hold onto” these parts of the sound, even after they have stopped in the input signal.  In addition to causing these frequencies to linger too long, filling what should be quieter parts of the signal, some of these resonances will cause certain frequencies to be disproportionately louder (or softer) than they are in the input signal.

While proper acoustic treatments can make important differences in the sound (and will be covered in a future entry in this series), the starting point will determine their effectiveness.  If the goal is to get the monitoring out of the way, a key part of this is getting the room itself out of the way.  Better to minimize any excitation of room resonances from the start.  Placement of the monitors plays a big part here.  As we approach a room boundary, resonant excitation increases.  As we approach places where boundaries meet, such as corners, excitation increases further.  In most studios, we find the listening position behind the console (i.e., mixing board) and the console placed toward the front of the room.  This common placement tends to put the monitors in positions that are very good at stimulating room resonances.  Moving the monitors away from boundaries results in less interference from the room.  I have often said “Every foot from the wall adds $1000 to the sound.”

In terms of acoustics, bass issues manifest themselves in the room’s resonant modes.  In the treble, it is reflections that cause acoustic issues.  The most harmful are called “early reflections” because they arrive at the listening position just after the arrival of the direct sound from the loudspeakers.  These slightly delayed sounds will alter instrumental timbres and smear stereo imaging, in effect, defocusing the audio “picture”.  Here again, proper acoustic treatment of early reflections can make significant differences in the perceived sound but here too, placement is the first step in ensuring the system and room get out of the way to the greatest degree.

Early reflections can occur from room boundaries and from objects in the room, especially from objects between the monitors and the listening position.  Consider the large reflective surface that is the console in most studios.  It is common to see loudspeakers placed atop the meter bridge of the console.  Sounds bouncing off the console reach the engineer’s ears just slightly behind the direct sound from the speakers.  The reflected sound combines with the direct sound and at these distances, one of the results will be a dip in the midrange (a weakening of sounds in the “presence region”).  In an effort to remedy this, the engineer tends to reach for the equalization controls to boost the level of midrange frequencies and “restore” the missing presence.  The problem is, the “remedy” is being applied to a recording that isn’t missing anything.  Because the monitoring has not gotten out of the way and is instead providing false information, something that is not contained in the recording but is in fact an artifact of the monitoring setup, the engineer is being misled and a recording that doesn’t need a thing is being arbitrarily brightened.  Played on a system that doesn’t suffer from the same reflections, the recording now has an artificial, hardened “edge”.

With all the above in mind, we’ve started to answer the questions “Can the room affect what I hear from the speakers?” and “Can where I place the speakers and what I place near them affect what I hear from the speakers?”  If monitoring is the crucial aspect of setting up a studio, where to start?  My experience has been that it is best to start with a clean slate.  For any studio or listening space, rather than fill the space and see what’s left for the monitoring, I find it best to start with the monitors themselves and place everything else afterward.  I’ve already mentioned staying well away from room boundaries.  In the middle of the last century, engineer Peter Walker determined that room excitation can be minimized by placing the monitors near 1/3 points along the room’s diagonals.  In other words, as a start, find the points that divide the room’s length and the room’s width in three.  Placing the monitors near these points will excite the room the least.  I have had good success in several rooms and studios by leaving 1/3 the room’s width between the speakers and 1/3 the room’s length behind them.  (For more on this subject, see Setting up your monitoring environment.)

For now, before placing other items in the room, set the listening position at a point just slightly farther from a line drawn between the speakers, than the center of each speaker is from the center of the other speaker.  In other words, if the center of the left speaker is for example, 72 inches (~1.8 meters) from the center of the right speaker, place the listening position so that your head is slightly further than this distance from either one of the speakers, say perhaps, 80 inches (~2 meters).  Aim the speakers at a point just behind the listening position.

To those not familiar with such a setup, having speakers near the 1/3 points can seem like the speakers are “in the middle of the room”.  But listen to how much easier it is to hear past the speakers, to the recording itself.  Now you hear the bass contained in the recording and not the sympathetic, out of tune “woof” of room resonance.  The sound becomes freed from the confines of the speakers and has a depth dimension (if the recording contains this — more on the subject in a future entry).  The sense of the speakers getting out of the way is increased as the speakers themselves become less obvious sources of the sound.  The part of the room behind the speakers simply comes alive with the stereo “soundstage”, as determined by the recording itself.

Having a monitoring setup like this doesn’t just increase how much you can hear from the recording.  It changes how you go about making recordings.  Now you can hear what you’re doing.