Enlightened Audio

Monitors and the home recordist (Part 2)

2012-02-13T09:47:00.000-08:00

Nearfields are the most popular speakers for critical listening sold these days. The reason probably has to do with the fact that a lot of those practicing mixing or actually working in audio production don't always have the budget to finance a purpose built control room and large main monitors. So the needs of the recording professional and amateur have changed towards a desire for smaller speakers that can output accurately and have enough SPL (sound pressure level) at relatively close listening distances (1 to 2 meters). If you're a home recordist this is almost certainly the type of thing you're looking for if you're anything like the average.

The trouble with smaller boxes is that they are limited by the laws of physics in how much of the audible frequency spectrum they can output at reasonably high SPL to the listening position. The bandwidth extension problem lies in the bass frequencies. In order for you to hear all of the bass region speakers must move a lot of air. That can be done with either several drivers splitting the work or fewer, larger drivers moving more air with each excursion due to their larger surface area.

Single drivers doing all of the work of the entire frequency spectrum as in single driver designs are not able to give enough detail from bass to treble. Multiple drivers are expensive to implement and difficult to design properly. Somewhat of a compromise in cost and design is the common two way design. A two way speaker is a box with two drivers, one for the low to mid frequency output and another smaller driver for the high frequency output (known as the "tweeter"). They are useful because the bulk of the sonic information that a listener will be hearing is found in the midrange, with critical details found in the highs. With the bandwidth split between two drivers each driver can be designed to manage audible material that best suits the driver's size and intrinsic capability. It increases efficiency over the single driver design. But it also adds complication.

In order to send program material to two separate drivers the signal must be split and the content must be filtered out so that each driver only passes audio it's supposed to pass and so that there is little overlap. This task of splitting the signal is handled by the crossover. A crossover is a filter that is designed to divide an audio signal into parts and feed each part to the appropriate driver. How a crossover is designed, the specific type of filtering and the specific frequency it begins to work at are all matters to be carefully considered by speaker designers; it's a matter of balancing sonic benefits and disadvantages with each design decision. The goal is for the crossover to be inaudible. The point being that when the listener hears the final output from the speaker it sounds like one signal, with no distracting or misleading distortion or anomalies. It's not always done well, which is one of the major complications of a two way design.

Another major complication is that of adequate bass output. Even with two drivers working there is still a lot of missing information in the low frequencies when using smaller sealed cabinet speakers. This is partly because a speaker cabinet has air pressure on the inside of it. When the driver connected to the inner chamber of a speaker cabinet moves inward it compresses that air mass. But the less the internal volume of the cabinet the more difficult it is for the driver to compress the air; the air mass is "stiffer" in a smaller cabinet. Air behaves much like a spring as a driver compresses it. The stiffer "air spring" prevents the driver from moving inward as much as it needs to. This reduces the distance of the driver's overall travel and ultimately it's ability to output enough SPL at low frequencies where the extra excursion is needed to adequately move enough air for the listener to hear it. That means less audible bass from a smaller sealed cabinet given a constant driver size...

Check back to catch part 3 when we cover porting and how it is used to extend bass in smaller sized cabinets

Happy New Year

2012-01-01T16:40:00.000-08:00

It's 2012.

This year I plan on updating the blog on a regular basis so that those that come to get new information on how to record better can have it, without the long wait. Forgive me for being so neglectful. I never knew people actually read this thing.

I have a not-so-new assistant (His name is Bailey) for my studio work and he'll be helping out with the blog duties as well. You will see him around from time to time on the blog. I hope we can make this thing into something useful for all interested in having consistent info on quality recording techniques.

I hope everyone has a safe and healthy year and I hope you all prosper and get closer to living your fullest potential.

Truly,

Enlightened Hand

Monitors and the home recordist (Part 1)

2011-05-28T04:25:00.000-07:00

I hear so much nonsense about monitors that I decided to comment on the state of common "wisdom" with regards to what matters with monitors and what doesn't.

What exactly are we talking about?

Monitors are speakers, pretty much like any other speakers except they are supposed to be very useful for allowing a recording technician to know that what they're hearing isn't being changed by the monitors to sound like anything other than what the playback material actually is. This is important because every decision in recording is affected by the monitoring system.

Whenever you want to know if the mic placement is right for example, it's not enough to go into the tracking space and listen, because you need to know what the mic hears, not what you hear from where you might be standing. So what do you do? You go in the control room and listen through the monitors and make adjustments accordingly. Those kinds of decisions abound when you are recording from the tweaking and tuning of source instruments to the sound of preamplifiers and other gear. Anything you track you'll have to listen through the monitors to know how it's going to sound recorded. That's a lot of responsibility for your speakers. Imagine if they aren't providing you accurate information. It could be very detrimental to your results.

So what does a "good" or "accurate" monitoring system sound like? The answer is actually pretty simple: It sounds as if you're not hearing any speakers at all but only live performers right in the room with you. That's it. That's also a tall order because there is a lot of sonic information that is passed to your ears when you're hearing a live performance. Currently no speaker design is good enough to give all of that information to the listener. But the best designs come very close to reproducing a live performance sound and feel, and it's important to have both.

Our relationship with sound

You might be thinking that "feel" has nothing to do with sound but actually it has quite a bit to do with it. Think of the times when you've been in the room near a live drummer playing their drum kit. Remember when they played the kick drum and how not only could you hear the sound of the drum but you could actually feel it's impact deep in your center. Remember the snare drum and how whenever it was struck you could feel it crack inside of your skull. This isn't just loud playing. It's all kinds of different information that is hitting your ears and body and creating the experience of an actual sonic event.

It's that information that allows us as listeners to determine where exactly a sound is emanating from, what exactly it is, it's tone, texture and it's relevance to our place in the moment. Sound is a very complicated and deep matter with humans and consequently we are very good at hearing and unconsciously interpreting subtle information about sound. It's actually a set of survival abilities that have developed from eons of evolution. Now we are so used to hearing things and instinctively knowing about the sound of things that we take it for granted. But monitor designers can't afford to take such aspects of our relationship with sound for granted if they want to design speakers that are worth using as a reference...(to be continued)

Check back for part 2 when I talk about the most common types of "reference monitors" sold and how good or bad they really might be.

Honestly we should never need a UAD-3

2011-05-22T06:03:00.000-07:00

I remember when the original UAD-1 came out almost (if not more than) 10 years ago. At the time is was ground breaking for the computer audio mix engineer that was interested in being able to mix in the box and use very high quality plug-ins that didn't completely drain the host CPU resources. I got one and never looked back. The UAD-1 was better than anything else available at the time that I can remember.

Now with the UAD-2 available and well established we see the possibilities have grown tremendously for DSP mixing as a single UAD-2 solo card is 2.5 times as powerful as the UAD-1 and that's just for the the lowest powered version. The quad card is a full 10 times as powerful as a UAD-1 on a single card (and you can use four of them at once). The funny thing about all of that is not only has outboard DSP power increased 10 times or better in a very short period of time, but native CPU processing has also increased to immense levels. One does not even need a UAD to get the processing one might want for mixing a particular project if a well specified computer is used.

What all of this processing power amounts to are virtually limitless options for the in-the-box mixing engineer in terms of signal processing selection and individual instances of processing per channel available. That might seem like a good thing and it is in many respects. But there is a downside.

The downside is that many inexperienced mixers are relying on too many plug-ins to get the job done and their mixes are suffering for it. Too much signal processing is really a problem with a person's idea about what is needed to accomplish signal processing goals. The irony is that even though we have so much power and selection at our disposal the best way to use that power and selection is as conservatively as possible.

The overuse of signal processing or instances of signal processors in a single project is something of a new problem for would-be professional audio engineers in general. Back in the day it was a rare thing to have anything near the amount of signal processing available in one studio as the average home recording enthusiast has available to them inside their computer today. Yet even without the kind of selection available back then mix engineers managed to come up with some excellent mixes. Why? Well firstly, and perhaps most importantly, because they were working with great recordings many times to start with. But also they knew that it was more important to give just enough and let the record speak the rest of the story all on it's own. That's a lesson a lot of new in-the-box mix engineers could stand taking to heart.

Too much processing is like using too much makeup. If a person has a pretty face they don't need much makeup to enhance what's already there. So the goal to getting the perfect makeup job is to have a pretty face to start with and then not tossing on so much crap that the face doesn't look realistic or recognizable anymore. Audio mixers are a lot like makeup artists. The best ones have a few key pieces of gear that they use on almost everything just to tactfully enhance the good qualities already present in the recordings and to carefully and subtly minimize any flaws. That's what it's all about. Just because you have 10 compressors available to use doesn't mean that you'll need to use more than one or two at best. Just because you can put an EQ on every channel doesn't mean that you actually need to do so. Sometimes the best thing one can do is as little as possible.

Plug-in manufacturers make a lot of money from convincing those that don't know that they don't have a complete set of tools until they buy the latest and greatest plug-ins or bundles of plug-ins. But it's important to realize that a lot of that stuff all does the same kinds of things that most people already can do. If more in-the-box mixers understood how to tweak their basic signal processors they would find that they could get by not only with using a lot fewer plug-ins per mix but also with not purchasing the big name, big dollar plug-ins that are so often pushed as "the solution" to all your audio woes.

I recently mixed a project for a customer who had apparently had someone else do a mix of the same song before me. The customer brought both their un-mixed tracks and their "rough mix" to me to listen to so that I could use the rough mix to get an idea of where they had wanted to go. As I listened to the rough mix I found several instances of overuse of plug-ins that had the mix sounding messy, crowded and overly processed. It didn't breathe and it lacked definition and punch. But apparently this mix engineer was convinced that they were really doing something cool with the mix by over processing everything. I ended up mixing that project using no more than a UAD-1, a few plug-ins from the Waves Native Power Pack and Cubase's stock plug-ins and I still had 50% DSP available from the UAD. I might have used 8 individual instances of signal processing on the entire 20+ track mix. In the end the mix came out wonderfully and it just goes to show that the key is restraint.

It won't be long before Universal Audio (a great company by the way) releases the next version of their UAD card, which will undoubtedly be exponentially more powerful than the current version, thus enabling users to have thousands of instances of simultaneous, even real time plug-ins at their disposal. But even and perhaps especially then it will be most important for the in-the-box mix engineer to use restraint. The whole thing reminds me of the high sample rate craze that has been propagated by gear manufacturers. The great irony is that these days digital recording and signal processing technology is at the point that if used well it's every bit as good as analog for all typical end listeners so we really shouldn't ever need orders of magnitude more DSP or more plug-ins to do what we can already do quite well now. What we need is to learn how to make good use of what we already have.

Things I like: The M-Audio DMP3

2011-01-30T05:30:00.000-08:00

It's funny how the best deals fall through the cracks

There is an obsession with finding the perfect, affordable preamplifier. It's understandable. A lot of people that buy pro audio gear are completely consumed with thoughts about: 1.) Their own gear not being good enough to do the job to a world class standard, and 2.) Wondering if any affordable piece of kit can actually do near the job that the esoteric, expensive ones can do; After all they wouldn't be pricing that stuff so high just because they can, would they? Well, I've found that not all of them would. But throughout my investigation I have started to suspect with certain items, the preamplifier being one of them, that most companies do price exorbitantly indeed. Thus my personal search for a cost effective preamplifier that does excellent work has most recently lead me to the M-Audio DMP3.

The DMP3 is a two channel, "clean" style microphone preamplifier with instrument direct inputs (that I didn't test for this review). It's a simple design, housed in a rugged, black steel case. Each channel independently possesses a smooth gain potentiometer that can be toggled between "low" and "high" gain ranges by pressing a button on the front face of the unit. Each channel also has a low cut filter that is fixed at 75Hz, and is also activated by pressing a button on the front. Both channels have an attractive VU meter and a clip indicator LED. Phantom power is activated on both channels simultaneously by pressing a single button on the right of the front face, underneath the power button. On the rear of the unit, from right to left each channel has a microphone XLR input and a high impedance TRS instrument input. Both channels have polarity invert buttons, as well as balanced TRS line outputs. Power is supplied by an external wall wart unit that plugs into the leftmost socket on the rear.

BUILD QUALITY

Priced at $160 (street) my initial impression of the DMP3 was one of skepticism about it's build quality. The unit felt solid in my hands but the first thing I did was check the feel of the gain potentiometers as they are turned. I found that on my unit the knobs got harder to turn the higher they traveled up their range, until about the last 20% of turning. That's not a big deal, and it's probably something that will be less of an issue as the knobs are used, but it's the kind of small detail that brought my reservations about "cheap" gear to the forefront. I then checked the buttons to see how they felt. Thankfully they were sturdy feeling, with good resistance and a satisfying click, so no issues there. In fact the only real problem I found with the DMP3 was with it's power supply connector.

I don't know what the resistance is from manufacturers of affordable gear towards including solid, locking type power cable connectors. They are not the kind of design feature that greatly increases the cost of any unit. They don't affect the sound and they are entirely practical and helpful to implement. I sincerely wish all companies made simple locking connectors a standard. I know I would gladly eat the added cost. With the DMP3 M-Audio falls victim to the ridiculous implementation of a chintzy power cable connector that not only doesn't lock, but at least on my unit, didn't even appear to be pushed in all of the way when it could be inserted no further. This is silly and upsetting as it is far too easy to accidentally unplug. I know that at $160 there's not too much that can be expected but I think a simple robust power cable input lock is asking very little.

One thing I don't have much to say about with the DMP3 is the metering. It certainly has attractive VU meters and ostensibly helpful clip indicators, but in practice with the kinds of recording that will most likely be done with the DMP3 the VU meters are virtually unnecessary. They do help with determining the average output signal level from the DMP3, but they don't necessarily correlate with the input meters of any particular user's sequencer (the most important meters to take notice of for tracking anyway) and so the user is stuck with having to determine the appropriate average level by using only their sequencer meters, which means that perhaps M-Audio could have saved the cost of the VU meters and done something more to improve the DMP's overall functionality. Who knows? All negative commentary aside though, I think the money that was spent in development was likely spent on getting the thing to sound right, a goal I think they've achieved.

THE SOUND

With a preamplifier the sound, headroom and noise level are where the "rubber meets the road", and with the DMP3 I think M-Audio is on to something special. I tested the DMP3 with a Shure SM57 and a Rode NT1a. I chose those microphones because they are incredibly popular and well known and they represent examples of a typical gain hungry dynamic and a typical low cost, decent quality condenser; the two types of microphones I suspect will be used by most potential users of the DMP3. All sonic descriptions are relative in that they only mean anything in comparison to either the original sound source or another piece of equipment's manifestation of that sound. To establish my interpretation of the sound characteristics of the DMP3 I compared it to the run-of-the-mill preamps in my Presonus Firestudio2626 interface. I suspect these are the kinds of preamps that most project studios and home recordists will encounter the most and so they made sense to me to compare to a dedicated preamp unit like the DMP3. Also, just for fun I compared the DMP to the Golden Age Project Pre73.

Using the SM57 to capture a male, pop-style, tenor voice on the DMP3 with the gain range toggle in it's default high gain position I only had to turn the gain knob up about halfway to get more than enough gain to average an incoming signal level of -18dBFS on Cubase's input meters. Compared directly to the Firestudio preamps the DMP3 was a touch more relaxed or smoother in the high frequencies, slightly fuller in the low frequencies and more realistic and open sounding in terms of dimension and dynamic detail. The Firestudio by comparison had a tendency to be a touch more aggressive or bright with it's presentation of the high frequencies: sibilance came across as definitely more noticeable. The low frequencies of the Firestudio were thinner by comparison, though not by much and the overall dimension of the sound was a bit more congested. The DMP3 made an SM57 sound very nice and natural, with a clear, full but not hyped or aggressive presentation. It also had the headroom advantage as I felt there was plenty of gain available to do the job with the DMP3 with the knob only turned halfway up, where with the Firestudio I had to turn the gain up to about 75% to get the same amount of gain, though the sound did not come across as particularly strained. I just happen to know from experience that there was only about 15% more gain I could squeeze out before noise became a problem. The DMP3 by comparison was very quiet and I feel I could amplify any dynamic and most ribbons with it and have clean headroom to spare.

While using a Rode NT1a I put the DMP3 in low gain mode and engaged the phantom power. With that microphone and the pre set to low gain I only had to turn the knob up about a third of the way to get enough gain to average around -18dBFS on the input meters in Cubase. The sound was quite impressive. Compared to the Firestudio preamps it was full and natural with again more relaxed high frequency presentation and a sort of smoothness that made things sound natural and "right". The sense of dimension was open and the low frequency content was full and present. Overall the DMP3 compared to the Firestudio sounded more "real" or authentic. It was as if the singer was right in front of me. The Firestudio preamps by comparison sounded a touch more bright and aggressive in the high frequencies. They were again thinner in the lows, but not by much.

One thing I noticed about the low mids and the midrange in general with the Firestudio preamps compared to the DMP3, when using the condenser mic was a very slight bit of a veiled sound: a kind of haze. It was very slight but it was indeed there. When you think about tracking multiple signals through the Firestudio and using them all in a mix ("stacking" as it's sometimes called) you can imagine how that slight haze and slightly more aggressive high frequency presentation, combined with a thinner low end could come across as slightly more cold and veiled overall. With the DMP3 tracks have a better chance of sounding real, full and unvarnished, even though not quite to the extreme of a ultra-high fidelity piece like my cherished Grace M201, but definitely more so than the likes of the run-of-the-mill preamps that are commonly used in devices like the Firestudio.

Comparing the DMP3 to the GAP Pre73 was a different story. The clean, natural presentation of the DMP brought into stark relief the obvious "character" of the Pre73, though not in a bad way. Both the DMP and the Pre73 have a way of relaxing high frequency content so that it's not harsh. But the Pre73 by comparison also has a slightly fuller low end and a more forward mid-range as well as huge headroom and a sense of dynamic impact that often comes with transformer based designs. If the DMP3 makes a microphone sound natural and classy then the Pre73 makes one sound downright special and interesting, all while maintaining a musical, pleasant quality. But compare the two and you'll notice the coloration of the latter. If you're looking for character at the preamp the DMP3 isn't the kind of piece you'll want.

CONCLUSION

In the end the M-Audio DMP3 is a fantastic deal of a mic pre. It sounds excellent compared to ordinary interface preamps and it's never harsh or excessively thin. When you consider that the thing costs $160 then I don't see any good reason for anyone in the market for a preamp not buying one. It's just such a steal. I only recently got interested in the DMP3 because of a track that I mixed for a customer who had recorded their lead vocal through it, a lead vocal that sounded fantastic. It was so good that I had to buy one and see if there really could be a very good 2-channel preamp for less than $200 that you didn't have to build yourself. It turns out that there not only can be, but there is and it's called the DMP3.

Why, I wonder, don't a lot more people seem to know about it?

GUITAR EFFECTS: TRACK THEM IN OR ADD THEM LATER?

2011-01-03T01:55:00.000-08:00

This used to be a very simple question for me and at one point I had figured that I'd gotten to the bottom of the issue. But now I think the answer is more in depth: It depends. I know nobody wants to hear that. But it's the truth. There is no ironclad rule about tracking effects for the guitar. Sometimes it works out best to get them in at the beginning and have them printed on. Sometimes it's best to leave the guitar part bare and work out the effects later. The key to deciding is knowing the consequences of both methods and knowing what your goals are in the first place.

DO YOU KNOW WHAT YOU WANT?

If the musicians who are recording the songs have a clear idea of exactly what they want their effects to sound like and they already have their settings worked out so that they can make those sounds with their gear then I think it can be very helpful to track in the effects from the start. As always it comes down to auditioning things before committing them to record. This is where it matters to take your time and listen to the way the guitars sound along with everything else in the arrangement. You as the engineer have to play the role of producer a bit in that you must decide if the effects that are desired by the musicians are feasible for the quality of the whole arrangement.

Where things can get sticky is with time based effects (reverbs and delays). Time based effects can sometimes sound very good on their own and yet screw up the entire mix because they don't fit with the timing of the rest of the song. Usually (but not always) you want your delays, for example, to be correlated with the tempo of the song they are used in so they don't seem so obvious, yet add a special depth that is there but difficult for the listener to put their finger on. But many recording musicians are notorious for not knowing that is important and so many times they simply slap on a delay that doesn't sound bad live, but sucks when it's recorded. In those situations it's up to you, the engineer to let the musicians know that you must work with them to adjust the timing of their delays so that they fit the tempo and rhythm of the song. It's a simple matter of stopping them before they track their parts and asking to audition their chosen delay settings. Then you can listen to the delay along with the metronome (and the rest of the track) and decide if things will actually fit. When they don't, then adjust them until they do. It's really that simple. But it's critical that you do that.

With reverbs things become a matter of judging if the verb is washing the sound of the mix elements out by being too present or too loud, or if the verb is making things sound too far away by being too long. Reverb can be a tricky beast to tame, especially if you're unclear about how you want the final product to sound. My rule of thumb is that if a reverb is being used by the instrumentalist on a guitar part and it's not absolutely perfect for the final mix arrangement as you track it and play it back, get rid of it and track the instrument dry. You can always add reverb later, but it's a real problem to try and take it away (it's damn near impossible).

As for other effects I'm pretty much cool with tracking them in. I think that recording should be an exercise in capturing what is actually there. There is no need to do everything (effects wise) during the mix phase. One thing you can always do to save yourself a potential headache though, if you're ever unsure, is take a direct line in of each guitar as well as a microphone input from their cab. That way you have a little insurance should you get to the mix phase and find that something just won't work at all. You can then re-amp if you must or you can rely on software amp simulators to save the day during such dire occasions. It's a great way to keep you out of a real problem.

Remember to always check to see if things sound good coming through your monitors and don't rely on the live room sound to judge effects or anything you're recording really. Your monitors are your window into how the recording is going to sound played back over speakers. So it's important to make your decisions about what works and what doesn't by relying on them (which is why they ought to be good monitors). Take your time. Audition effects in context before you commit. Listen through your monitors. Sync your delays to your track tempo and rhythm. Beware of excessive verb, and don't be afraid to track things dry if you're unsure. Also, keep a line in take so that you have insurance against utter failure. But most importantly, don't be afraid to experiment and track effects in from the start. It can really make mixing more easy and enjoyable, because you don't have to create every sonic event from scratch.

But the one rule of recording really can never be broken:
-What you get out of a mix will only ever be as good as what you tracked in.

That being said, be careful out there in effects land.

Things I like: Golden Age Project Pre73

2010-12-31T23:01:00.000-08:00

If not for comparisons it would be a modern classic

I'm not really all that into "character" or "mojo" preamps. I feel that most of the work of getting sonic coloration is best done at the source and mic and later during the mix stage. For me a preamp is much more of a tool than anything else. Also, I've been very spoiled by using super clean, high quality preamps by some very respectable designers, so I tend to notice the sound of a preamp doing it's thing when it's not staying out of the way. All of that being said though, there are times when it's useful to have a classic sounding preamp because when they do what they do, frankly it sounds very cool.

The problem with well made, classic designs is that they tend to be expensive for most people. Part of the high price tag is due to the cost of manufacturing, components and workmanship. But I'm willing to bet a significant part is due to the name on the faceplate (Honestly, some of the classic designs are, technically speaking from a designer's standpoint, some of the simplest stuff on Earth). In any case, when you actually find that you really want a good classic sounding pre there's no escaping the relatively high price tag, that is until Golden Age came and set the world as we know it on it's ear with the Pre73.

The Pre73 is a single channel, solid-state, transformer based preamplifier that is designed to be a knock-off of the preamp section of the famous Neve 1073 preamp/eq unit. The Neve 1073 is arguably the most renown preamplifier of the past 40 years. It's sound has contributed to innumerable recordings and it's a very desirable piece of kit. It's also hard to get working originals and they are ridiculously expensive. Even the re-issue from AMS-Neve is crazy expensive. There are plenty of companies that make it their business to clone the Neve 1073. Some are better than others at approximating the exact sound of the original. Regardless of whichever clone you might choose they are all rather costly. But a low price and an unusually high level of build and design quality are what make the Pre73 stand out from the pack.

What it is

As a single channel preamplifier the Pre73 ticks almost all of the boxes of useful features. On the front panel from left to right there is the power button, a direct injection instrument input and three mode select buttons: one for toggling the DI input as active or not, one for switching phantom power on and off and one for toggling the preamp impedance from either it's normal 1200 Ohms to it's low impedance mode at 300 Ohms. After the buttons there is the rotary switch gain control, immediately followed by the smooth output level potentiometer. After that there is the 4-segment level meter and below it sits the polarity invert button. The circuitry resides in a rather rugged, simple, painted steel case. The rear panel hosts the power supply input as well as the microphone and line input jacks; both of which are combo jacks. Finally there is the 3-pin, male xlr line output jack and a separate balanced TRS output.

When I opened the packaging on the Pre73 I was surprised by how large and heavy it is. Even the external "line lump" power supply feels heavy and robust. I opened up the steel case and took a look at the circuitry and I was impressed by the build quality. Everything was neatly done. Only discrete components are used in the signal path. It's really an impressive looking piece on the inside. Honestly if it wasn't marketed as a 1073 knock-off it would be a very good preamp in it's own right. Even the gain control knobs are held firmly in place. They don't wobble or move in any way they are not supposed to. This is solid engineering at work. It's definitely not cutting edge or even fancy. But what is on offer here is time tested, high quality design that can get the job done for years if the need be.

What could be improved

In fact I only really have a few gripes about the build and design of the Pre73. The first is the power cable jack. It's not a locking type and it absolutely should be. It's far too easy to remove the cable accidentally. The second is the lack of numbering on the the output pot. It's always helpful to at least have the extreme clockwise and counter-clockwise positions labeled, so you have an idea of how much you're boosting or attenuating.

Finally there is the gain switch labeling. For the microphone gain stage it's actually backwards. I've used a 1073 and if the switch is operated at the lowest gain setting (for the microphone gain stages) the switch would be pointed at "-80". The Pre73 however, has the lowest gain setting labeled as "-20" and the highest gain setting labeled as "-80". I found this utterly confusing. Adding to that confusion is the fact that the Pre73's switch, just like the original 1073, is labeled according to it's level of gain attenuation (since that's actually what the switch does with this type of circuit during the mic gain stages) and not according to how much gain it's "adding" to a signal. There's nothing wrong with that in theory, because it's accurate. But most newcomers have no idea about how most preamp circuits actually work and with the labeling being inconsistent with the switch operation added to the counter intuitive negative numbering it can easily cause confusion for those novices trying be exact about setting their gain levels.

There is also the fact that there is no low cut filter. That's not a problem in practice since every sequencer has the ability for you to do that. But it would be nice to have as an option, especially since the Pre73 lacks the equalizer section of the original 1073. What it doesn't lack however is a great sound and that's what matters most.

The sound

There is a lot of talk about how the Pre73 doesn't sound like an original Neve 1073, and it's all true. It doesn't sound exactly like an original Neve 1073. But it's important for everyone to know that NONE of the clones of the original 1073 sound exactly like the original, and yet they all cost a LOT more and are basically no better built than the Pre73. Some clones are close to the original, even very close. But there is always a subtle difference. None of that is the point however. The Pre73 isn't supposed to sound exactly like a 1073. It's supposed to have similar sonic characteristics of that pre and it certainly has those.

I would say that the Pre73 in general sounds like almost all of the "vintage" mic preamps in that it has the capacity to sound smooth, full and punchy, depending on how you use it. On the two vocalists that I recently used it on, one male and one female, both with sm57s as the mic, I found that the sound was quite clear. The preamp was very quiet. There was plenty gain available as well as plenty of headroom. Most of the high frequency content was very smooth. The pre lent a weighty character to the vocal performances that I appreciated. It's tendency to smooth out the high frequencies allowed sibilance to be minimized and room reflections to be less obvious. Compared to run-of-the-mill interface preamps the sound from the Pre73 was more open and larger and revealed less harshness if there was any there in the first place. I suspect the comparatively open sound quality is due to the generous amount of headroom provided. Dynamic changes leap out at you, but not in any obnoxious way. Backing down the output trim causes the pre to take on a more compressed, less open sound. But sometimes that's just what you want.

An area where the Pre73 really shines is on percussive sources. On a snare drum close mic-ed with an sm57 the sound was punchy, in that the transients were very vivid and seemed to jump out at me, which is typical of classic transformer based designs. There was also a sense of weight in the low frequencies. I felt I could really feel the drum in my gut as opposed to just hearing the impression of the drum. The same held true for kick drum and toms. But interestingly enough the same held true for anything that had any percussive impact. I recorded me striking a meditation chime, a piece of steel equipment rack blanking panel, and even a large hard-cover book. It really didn't matter. If there is an impact the sensation is heightened when amplified through the Pre73, however the effect is less so with higher frequency content. That characteristic as well as the sense of fullness and relaxed high frequencies is exactly the kind of thing one wants from a "vintage" preamp, be it a Neve, API or whatever. In this respect the Pre73 is excellent. It's every bit as good as any other classic piece. It just happens to have it's own sound that while similar perhaps to a 1073 still manages to be unique. I'm quite impressed by it.

On condenser microphones the Pre73 also performed like a champ. Using a Rode NT1a and a Schoeps mk21 on male vocals both speaking and singing I was thoroughly impressed. The Pre73, like any good classic preamp, had the ability to give enough headroom so that the sound remained open. The presentation remained forward enough to cut through a mix, and yet the pre could still still smooth things out just enough to provide a silky feel. It's everything one might need to give a special something to a recording that really can't be easily explained.

Conclusions

Overall the Pre73 is a winner and if buyers are smart about it they will not let the many rumors about needing to upgrade it or modify it convince them that it is "just another cheap pre that one has to change to make it right". That's nonsense. The Pre73 is well made, sounds excellent, has plenty of gain for any microphone duty and is every bit as good as any other well made preamp. It's not an ultra clean pre. It's definitely a "character" pre. But it's flexible and sounds good enough on it's own that it would be a silly waste of time to modify it. I'm sure there will be those that have problems with it here and there due to potential quality control issues. I haven't ever experienced any, and these days the notion that something being "made in China" automatically means that it sucks needs to be put to rest. There are some wonderful, affordable products coming from China and the Pre73 is definitely one of them. It has vibe. It has a classy sound and it doesn't break the bank. If you're in the market for an affordable preamp that will do the job of a classic pre and sound great, look no further. The higher priced contenders might want to consider themselves placed on notice. Apparently there is a new game in town and it's all about affordable quality. Well done Golden Age. Well done.

Things I like: KRK VXT 4

2010-12-22T17:10:00.000-08:00

Underrated, but really useful

I've had a pair of KRK E8bs for some time now that I've been using as my main mixing monitors in my home project studio. I love them. They are clear, balanced and detailed in a way that makes mix translation effortless enough for me to use day in and out without complaint. Of course it's always a matter of preference when it comes to selecting monitors, especially since there are now so many great performers to choose from. If you're like most people who record these days the only real limitation is budget and the space that you have to work in. KRK have taken that into consideration with their VXT series of active speakers. In short, they have created a series of balanced, detailed and accurate sounding speakers that can fit just about any project studio's size and budget.

I came across the VXT 4 model about 3 years ago, when I was looking for some small speakers to travel with me from location to location, so that I could track and mix. I wanted portability but I didn't want to sacrifice mix translation or clarity for tracking purposes. Because I was already intimately familiar with the KRK brand I decided to give the VXT series a try. I have to say that now a few years later, I'm glad that I did.

The VXT 4 is a compact, near-field, active, two-way, front facing bass reflex port design with a 4" woven Kevlar bass/mid driver beneath a 1" silk dome tweeter. The cabinet is filled with ABS structural foam that KRK claims eliminates internal cabinet resonances and allows for the speaker to be designed with no internal parallel walls, which helps to eliminate internal nodes. There are controls on the rear that include a ground/lift switch, a limiter and clip indicator and an auto mute toggle. There is also a volume level adjust knob, as well as a TRS/XLR combo jack, the power plug input jack and the power switch.

As small as the VXT 4 speakers are they feel quite heavy for their size weighing in at 14 pounds each. But their size makes them perfect for on location tracking and mixing or for use in a small space like a bedroom or small project studio recording area.

In Use

Compared to my much larger and much more powerful main speakers the VXT 4 model is not in the same league, as it shouldn't be. It's a small design, with near-field work as the primary use in mind. However, when used as intended the VXT 4 are some of the most useful small speakers I've ever worked with.

The biggest asset that these speakers have is clarity in the mid and treble range. They come across as well balanced, and sonic details like instrument placement and tone are very easy to identify and visualize. The stereo image is adequately wide and believable. Even the bass frequencies come across as realistic and full. The speakers do lack in amplifier headroom and so they can't really get terribly loud. But they are loud enough to hurt your ears when used in near-field at around 1 meter away and that's without causing the clip indicator to flash at all. If you need more volume than that then you shouldn't be working with such small speakers to begin with.

Another great asset of the VXT is that they have the somewhat uncommon ability to not sound at all "boxy". I'm not entirely sure if it's the cabinet design or the port design or a combination of both but the VXTs don't give a hint of boxy-ness when you listen to them and that's something that a LOT of speaker manufacturers in this price range could learn to do much better about.

If there is a weakness in the VXT 4 it could probably be found in the bass frequencies. To the little speaker's credit, it's a very realistic sounding bass that's not hyped or "one-note" at all. The problem is that with speakers of this size it's quite difficult to get very much bass and that might be a problem for some potential users because so many people are used to excessive bass in their casual listening. The good news is that you don't need the extra bass usually because the design of the VXT 4 is well balanced enough that mixing on them usually produces great translation even with their limited bass output. To me that's more important than feeling it in my chest. But if you must have more bass, for whatever reason, it's good to know that you can always compliment the VXT 4 with the KRK sub and then you'll have more than enough bass for just about any mixing occasion. I happen to use them regularly with a sub and I want for nothing when I mix on them although I'm just as happy to mix without the sub.

I suppose I could go on and speak to the faint touch of grit that I hear in the upper mid frequencies from time to time, usually on vocals, when listening to playback on the VXTs. It's actually something that I wouldn't expect anyone who wasn't used to critical listening to notice at all. But I do notice it, although it doesn't bother me one bit. I suspect that it might have something to do with the amplifiers or whatever. In any case it's not a problem so much as it is a kind of sound characteristic of these speakers. I've never found it to interfere with a mix decision and it never affects translation of mixes either. In fact most people would probably not have any idea what I was talking about if they were to listen for it. I just thought I'd mention it in the interest of full disclosure.

Conclusions

Overall the KRK VXT 4 speakers are wonderful to work with. They are clear, balanced, detailed and they translate mixes quite well. Any perceived shortcomings were really a matter a preference and perspective and more or less came down to nit picking. But I'd be happy to recommend the VXTs as a fantastic bargain and great speakers for mixing on any project studio budget. They absolutely perform above their price level and I don't think anyone can ask for much more than that. You should definitely audition them if you're in the market for the specifications that they offer. They will surprise you with their quality.

So you want to work in a "real" studio?

2009-12-06T10:03:00.000-08:00

It's not really all that surprising that almost everyone that I meet that has taken the time to invest a couple thousand dollars or so in their recording equipment at some point or another really have the desire to work in a "real" recording studio. You know the ones with the giant consoles and really nice sounding live rooms and all of the fancy, expensive outboard signal processing gear. It's so cool to think of getting paid to record other people and tweak their sounds, all the while relaxing in a comfortable and secluded environment where you can be creative and listen to music all day. What a dream.

Unfortunately the reality is that it's VERY hard to get a job in a place like that; if by "job" you mean actually being a staff engineer that makes enough money to pay their rent and exist. I hate to break it down to you and to burst any dreams of graduating recording school and in a few short months transitioning to mixing the next platinum record, but the truth is that it just doesn't work that way. All of the creative arts are difficult to get paid reasonable wages to do. As a creative person you're pretty much stuck with forging your own path in the wilderness of the real world. In the real world you can very easily find yourself broke and lonely without any hope if you don't have a realistic plan, an extrodinary work ethic, exceptional amounts of dicipline and a whole lot of skill to back it up.

So how do you get from where you are to where you want to be? The specific answer for you in your situation is unknown, but the general idea is to think in business terms. You see this wonderful studio that you would like to work at. It's in your local area (or not that far away) and you figure that you'd be great as an engineer there. Why wouldn't they hire you? Because they don't need you. Any studio that exists today is much more concerned with keeping afloat and keeping work coming in the door than furthering your career. That's why if you want to get something from them you're going to have to give them something that they really want (and they don't want you). What they want, what they ALL want,...is more work coming through the door. What does this have to do with you? If you want to work at a studio then you're going to need to start consistently bringing work in the door.

Consider the idea of the internship. Back in the day before everybody and their uncle had their own project studio in their bathroom closet, you could actually go to a studio and try to get a position as a gopher. Basically you'd be their bitch and you'd keep your mouth shut until they felt like showing you something. If you were tenacious enough then you might become an assistant one day and things might very well move onward from there. But those days are long gone. Any decent studio has at least 200 people trying to be an intern and the only way that you're going to rise to the top of that pack is by bringing them work. So how do you do that? Well the first step is for you to sharpen your skills on your own.

BECOME A MASTER OF YOURSELF

Can you really record and mix? I mean honestly, are you any good? If you are within your first couple of years of this or if you're just out of recording school I'm going to save you the time spent on mental reflection of your skills and just tell you the truth: you suck. You do. You are not as good as the world class professionals that you want to work and live like and if you were you would know it so stop dreaming. It's okay though, because everybody sucks when they are new at this stuff. In fact I would go so far to say that even after a few years most people STILL suck and don't get it together until they really dig in and obsess about being better day and night. That's the only kind of passion and commitment that will make you any better than your peers. It isn't the gear that you buy or the school that you went to. It's about fixing it in your mind that you will learn how to use you ears and your brain to get the sounds that you hear in your head onto a recording whenever you desire. That's not an easy task for a novice to do but it can be done with unwavering commitment and desire. You WILL get better. But the problem that you have to deal with when looking for work is that everybody, and I do mean EVERYBODY is expecting you to deliver world class production results. That means that you're competing with people who have all of the gear and time and money in the world to do just what they like and you still have to make sure that anything that you put your name on as engineer sounds just as good or at least pretty damn close.

How do you get there? It's no secret and you already know: practice. All of the time you should be reading and studying everything audio production related. You should be recording people for free just for the chance to learn. You should be investing in a simple but flexible setup so that you can record anywhere, anytime. You should be a person possessed. There should be NOTHING that you don't work on. Master mic placement technique. Master knowing what each area of the frequency spectrum sounds like. Master the use of compression, EQ, limiting, time based effects and anything else you can get your hands on. You should literally go crazy learning to the point that you can track and mix with confidence and know that what you produce will come out sounding indistinguishable to what is commercially released. It can be done. But only if you're exceptional, so be exceptional in this regard. Why not be exceptional? Think about it for a moment. If everybody is already recording themselves then the ONLY reason anybody would want to actually spend money on you doing it for them is that you can do it better (and faster, and more efficiently, allowing them to concentrate on being musicians). If you can then you save them the headache and you get to further your career. If you can't then you're wasting your own time as well as theirs. Once you become a competent engineer then you can work on building a portfolio of work to demonstrate your skills to anybody that might be interested in getting their material recorded.

GET THEM TO THE "REAL" STUDIO

Now I haven't forgotten that the entire purpose of this course of action is to get you closer to that job at a studio. Assuming you have sharpened your skills and you have a few people that are interested in recording with you, now you have to get them to record in the big studio. Why do you want them in the big studio? Because you need to show the studio that you can bring in work from the street (remember that part?). If you bring them work then they will let you assist them and eventually run your own sessions. This allows you to learn how their studio works, and it gives you a back door way to having your own internship. As long as the work is coming in then you're officially an assistant engineer. You might be freelancing but I assure you that if you're bringing in work and keeping a good attitude, as well as learning how to work in the big studio then they will have no problems with you being around at all. It also teaches you the hardest part of being a successful studio owner: getting the work to keep coming in. If you can master the effort then you'll be much further along the path of being able to keep afloat on your own as a professional engineer. You should never rely on the studio or anyone else to get you work. You are the source and so long as you can keep the work coming you will have a job.

Another reason you want your artists to record at the "real" studio is simply because of it's inherent advantages. They already have a great sounding live room for tracking, so why not get your drums and vocals tracked in there? They have cool microphones and preamplifiers, so why not take advantage of their gear? If you learn how to be efficient in planning and executing your tracking so that you can get in and get the most acoustically critical aspects of the recording tracked in the big studio (usually it's the drums and anything else that is room dependent in order to get the right sound) in one or two days and do the rest in your own place, then you can keep costs down and still have great sounding raw tracks. Hell if you can get everything done well in the big studio in only a couple of days then you and your artists are pretty damn sweet (but it's unlikely that it will go that way most of the time).

The "big" studio presents a bit of a problem though. Big studios can cost big money. But never fear. You don't have to go to the most expensive place around to get the experience of being in a big studio. All you need to do is find one that you can afford that has what you're looking for. Trust me they are almost everywhere. Look for a studio that has a going daily rate of say $400-$800 and you're probably going to be looking at some pretty decent places to start out at.

So how does this work? Well the simple, straight forward way would be for you to call around and find a studio that you want to work at. Research their prices to make sure they are in your range and then get yourself in there for a tour of the facilities. It's important that you do that before you book any talent to record anywhere. You should know what you're getting into without any surprises. It's very unprofessional to have an artist plan to record with you in a place that you don't look comfortable in. You don't need to know it inside and out but you need to be able to look like you're comfortable there and know what's going on so that the artists can feel comfortable working with you there.

I have found that the best way to get your head around a new place is to go in there as an assistant on the tracking project that you bring in. Let the studio use their house engineer and you sit by, listen, ask questions and learn how to work their place. The point is for you to get that experience anyway, even if you have to do this a few times before you feel confident enough to lead the session just do it, because that's the whole point: to learn from and work at a real studio. So don't try to rent a place and run the show unless you know exactly what you're doing. Whatever you do, please, please, please don't go in to somebody's studio and try to show the staff how much you know about recording and/or mixing or anything for that matter. It's lame and nobody likes it and it will make them not want you around at all ever again, even if you do bring work. Just go there. Be humble. Shut your mouth for the most part. Pay attention and only ask relevant questions, at the right time. Your demeanor and attitude make all of the difference in the world when working at a studio. If you can be smart and silent, low key yet skilled, then you will get somewhere. But if you let your stupid ego get involved at any point you will mess it up. So behave well!

Once you have the right place, at the right price then the time comes to book a session with whoever you're planning to record. Tell the artists the daily rate of the studio and ask them what they can afford to do. At this point I suggest only recording 3 to 4 songs max. The point is to keep costs down and to keep the project easily manageable for you (you're still new remember?). 3 to 4 songs should not take more than a day or two to record if the artists can actually play well. Which brings up a very important point. I suggest open and honest communication with your artists. Always let then know what to expect. Tell them that they need to be confident that they can play well or they are not ready to go into the studio to record. If they go in unprepared they will end up wasting their money and your time. Assuming they are ready for the studio, if they can afford to pay a couple of days then you call the studio and book the time at the agreed upon dates between you and the artists.

If for some reason they can't (or don't want to) afford the daily rate of the studio then you have the option of entering into cutting them a deal. It sucks. But it can get you work if you can't get anybody with money. How it works is you simply put up a hundred or so of your own money towards the daily rate of the studio thus reducing the cost to the artists. Why on Earth would you do this? Because you benefit from getting them to record in the big studio and if it costs you a hundred dollars or so to get that chance to track there and learn on a short project then I think it's worth it. Nothing comes for free in life. If you're at the point where you really want to work in a real studio then you're going to have to make it happen, even if it costs a little bit of dough a few times to get the ball rolling. The good news is that once you have a portfolio of awesome mixes that you've done and you have a relationship with the studio where you can run your own sessions then you won't have to cut your own throat anymore. Once you have a few things going for you then you can uphold the studio's daily rate. That does lead to the question of how you profit from any of this. The short answer is that you might not profit for awhile. But there's always the chance that you can at some point charge the artists the cost for the studio's daily rate plus additional money for you to engineer their session. That's usually only an option if you are in demand. However you can also profit by charging the daily rate for tracking and then charging a separate rate for mixing. That usually works out if you have your own mixing room at your own place where you don't have to pay for time. I do it this way and it works well.

So there you have it: a quick outline of how to get your foot in the door at a real studio. Obviously your results may vary. But I've found that it's a pretty decent way to at least get yourself the consistent opportunity at the kinds of places you want to be at. At the very least you'll get to assist on a few sessions, possibly lose a few hundred dollars and come out with the knowledge of how business works at a real studio. At best you can end up running your own sessions at a real studio, more likely than not breaking even financially, and maybe, if you're exceptional, making a profit. In any case it can't hurt to give it a try. The success all depends on you and what you bring to the table.

Good luck

About Preamplifiers

2009-11-10T04:15:00.000-08:00

One of the most recent popular areas to tweak in a recording engineer's signal path is the preamplifier. In the last decade or so we have seen an explosion of different kinds that range in tone from ultra clean to everything else. Why is this and why does it matter to you? Well as usual, if you're going for the sound that is in your head and you want to deal with every possible variable then the preamp has some bearing on that. But what exactly is a preamplifier's role in recording audio? Let us examine this for a moment.

I'm not going to get into a bunch of technical specifics and terminology regarding the function of a preamplifier. Basically it boils down to you needing to know this: A preamp's job is to take the low output signal level of a microphone and increase it to levels that are suitable for recording. That's pretty much the entire story. Different designs of preamplifiers do this in slightly different ways, but overall they are all shooting at the same goal. What makes the difference in which preamp you end up choosing (besides budget and availability) is all about what kind of subtle tone you want imparted on your recordings. What I mean by the term "tone" is a sonic character that a preamp can add to your recorded tracks. These days preamplifiers are chosen often just for adding certain character. But all is not well in the land of preamplifier selection. This is because we have a major misconception about the amount of impact a preamplifier has on the overall quality of a recording. Usually, it's assumed to be more than it actually is in my opinion.

If I were to break down the amount of effect that a preamplifier has on the final sound of a recording in terms of percentages I would say that it's less than 10%. That means that if all other things are excellent then you could still get over 90% of the way to having the kinds of recordings you dream of with just "run-of-the-mill" preamplifiers. So why do we have so many novices sweating the quality of their preamplifiers? I think it's because they are emulating the kinds of decisions that experienced professionals make. But the parts that novices are often leaving out of those decisions: the recording space, the performance quality, the instruments chosen and the way they are tweaked, the microphones chosen and the placement of those mics, etc... are the ones that matter most. It just so happens that experienced professionals already know this and are taking this into consideration when they are obsessing about which preamplifiers. But those who don't know might very well be fooled into thinking that the one thing that is eluding them, that will make or break their "pro" sound, is the perfect preamplifier (or converter, but that's another discussion). The bottom line is that they're wrong.

So what does that have to do with you? It's pretty simple. Pick a preamp that isn't broken and that doesn't horribly distort and get started making recordings. From that point, at least until you have all of the other things sorted out, pay no mind to the type of preamp that you're using, because it's not going to make enough of a difference to be worth the cost. When you're recording things well the preamp makes a very small contribution to the overall whole. Now I'll be the first to admit that recording is ALL about making several small contributions that add up to a well done finished product. But it's important to remember that there is an order of priority to consider and that order goes something like this:

1. song arrangement
2. skill of musicians
3. quality of performance
4. quality of the tracking location
5. microphones chosen
6. microphone placement
7. gain settings at preamplifiers
8. type of preamplifier
9. type of A-D converter

So when you look at it, according to the list above the type of preamplifier is way down near the bottom of the priority list. Why is this? Because all of the things that come BEFORE the preamplifier are the most important. I guarantee that if you get all of those things that are on the list ahead of preamps correct you will have excellent raw tracks.

At some point in the past I told someone that the secret to professional sound recording is preperation and attention to detail. I still stand by that advice. It's all about those two orienting characteristics. If you are well prepared for reaching the goals you have in mind and you take the time to manage every detail well, then you're going to do well. Don't sweat the preamplifier until you've got everything else properly in place.

Things I like: The Presonus Firestudio (a quick review)

2009-10-10T07:40:00.000-07:00

Quick Summary
Good:
Plenty of ins, outs and digital expansion capability
Decent preamp and conversion sound
Useful monitor controller (optional)
Stand alone capable
Room for improvement:
Limited preamp headroom (only a problem on quiet sources with dynamic mics or with ribbon mics)
Limited metering
Phantom power selectable only in groups of four
No pads on inputs

Why do I like it?

Because it's an affordable 26 in 26 out firewire recording interface, with optional monitor controller that gets the job done without a lot of fuss and very good quality for it's price. I've had the opportunity to use one for about a year or so now and I really think it's a great buy. It's got eight preamps and digital expansion for up to 26 simultaneous inputs and outputs. Presonus is rather hit or miss with it's products in my opinion. But with the Firestudio I think that they've gotten a few things right. For starters there is the sound.

The Preamplifiers
I compared the preamps on the Firestudio to those that were in a Flickenger console that resides at a studio that I use from time to time. Flickenger is almost directly comparable to API in terms of sound character. We tracked drums, electric guitars and vocals of the same song through the Flick console and then through the Firestudio. The Flickenger preamps had a deep, weighty, punchy character to them that lacked a bit of high frequency fine detail. What worked well with the Flickengers is that the tracks were noticeably in-your-face. The Presonus preamps by comparison had a brighter, clearer and more finely detailed presentation. But they were obviously thinner sounding in the low to low-mid frequencies and they didn't have as much punch and were not nearly as in-your-face as the Flickengers. Considering the design and cost of the Firestudio's preamps I was pretty satisfied with the comparison results. Overall the Firestudio's pres were adequate and called no negative attention to themselves. If you want punchy, warm and characterful, these are not your go-to pres. But if you're fine with relatively neutral and clear. The Firestudio will work out just fine. The only thing that I can honestly ask for out of Presonus is more maximum gain. The pres top out at 55dB and that's fine for moderately quiet to loud sources, which, admittedly, is what most general recording will involve. But it won't work for ribbon mics and it won't do on specifically quiet sources.

The Conversion
I have found that the A-D/D-A conversion on most units of this price range and build quality is only very decent, despite the often hyped marketing claims and the Firestudio is no exception. I compared the Firestudio's conversion to the Digidesign Pro Tools 192 I/O converters in the studio at 48kHz sample rate. Overall the conversion on the Firestudio is adequate. It has a stable stereo image that seems to lack in the very subtle high frequency ultra fine detail resolution and there is a subtle lack of depth by comparison. It must be noted though that none of this is dramatically evident at all, just by comparison, and none of it will make your tracks sound bad. If you can't get great sounding tracks from these converters then it's not the converters that are the problem. However, tracks that were converted through the Firestudio sounded just a little bit more veiled by comparison to the HD interface, but again, not dramatically so. The D-A was equally adequate and nothing to get worked up about for better or worse. On the whole the conversion is useful, but not exceptional and I wouldn't expect it to be any different to be honest with any unit of this target market.
(NOTE: The listening comparisons that I did were not at all scientific, so don't rely on my results as objective evidence to support any claim of quality. Also, I must stress that I had to listen carefully to hear the differences. They were evident but VERY subtle, as differences in converters of similar quality often are.)

The ins and outs
The digital connectivity is one of the best aspects of this unit. If you want better preamps and conversion and you simply want to use the unit for an interface, it can do that. If you need several simultaneous ins and outs, it can do that. If you need S/PDIF or ADAT or even high sample rate SMUX data transfer in and out (up to 96kHz), it can do that. The only thing that the Firestudio lacks is AES. Which I think would have been great, but I can live without it. Further still there's wordclock in and out available if you need to distribute your word clock signal throughout a multi converter system (such as when linking up a couple of Digimax or other converter units). Really it's got your channel expansion covered quite well.

The monitor controller
An often overlooked addition to audio interfaces is monitor management. We all need to be able to calibrate our playback levels and to quickly mute or dim the playback as well as have talkback functionality and route multiple playback sources to multiple speakers in stereo, 2.1 or 5.1. The optional MSR controller allows this with the simple connection of a cat-5 cable.

Overall the Firestudio is a great all-in-one recording interface that has plenty of useful features to make it worth the money. I recommend it to anybody purchasing at it's price point.

Electric guitars without EQ or compression (part 4)

2009-04-21T07:34:00.000-07:00

HITTING HARD WITHOUT THE COMPRESSOR

One of the biggest things to get right when tracking electric guitars with much of modern music is getting them to feel as though they hit hard once things get heavy. Often times this means an increase in volume (but not always). The problem that I've noticed by a lot of home recordists is that they track (and usually double track at that) the heavy parts too hot and then quickly go to the compressor (or limiter) to work the excessive volume out. That's exactly what not to do. If you want your guitars to hit hard during the heavy parts the first thing that you must do is take a critical look at the song arrangement. A lot of how a song is received by an audience rests on the way it's arranged. If a song is supposed to be "punchy" for example, then it has to be arranged (and played) in such a way that it's "punch" is preserved and not washed out by having too much of the same, loud, distorted mush, or having the various instruments stepping all over each other at critical moments.

So for me the first step is to keep things simple. Try having your verses arranged a little quieter and played with fewer instruments. This may mean that if you're tracking two guitars only have one playing a relatively basic part during the verse while the other adds subtle accents or small melodic riffs here and there (nothing that draws too much attention away from the singer). The verse can be thought of as a time for you to highlight what's going down with your vocalist and many times this can also be an excellent moment to establish the groove with your drums and bass, because you won't have the guitars screaming all over the place. You may also want to try panning your verse guitars a bit tighter towards the center, but not necessarily directly in the center. You don't necessarily want mono. What you want is a subtle, yet distinct sense of movement in the guitars, so panning them tighter during the verse and then letting them be wide and enveloping at the chorus is a nice trick to get the listener to feel the extra punch of your chorus dynamics, without having to crank things up as loudly as possible. There is of course, a time for letting the guitars scream a bit and that is definitely during the choruses. But the key here is that less is often times more (much more) than you think and how you actually go about layering your mix can make all of the difference in rather you add just the right amount of raging chorus guitars so that things hit hard or if you push things too far and it all goes to mush.

I like to start with getting the drums and bass locked together in the mix first. Then I'll add the vocals. Then I'll bring the guitars in around that and tweak everything until it all fits just right. The reason I approach it that way is because it's very easy to put in too much guitar when you don't get your vocals in right away. I also like to get the verses mixed first when I really want my chorus guitars to hit hard. Because the verses are quieter and less populated with instruments I get used to the more subdued moment and when the time comes for the heavier chorus guitars to blast in I usually find myself thinking that they are too much and I don't bring them up too loudly, which is an important line to balance on.

Another thing to remember is to keep the gain settings appropriate for preserving the tone of your guitars in your recording. Lots of times people will crank the gain on their amps so that they can really get the feeling of distortion. But often when recording things (more often than not) it's the case that much of that distortion comes through as high frequency noise and it's not at all punchy or interesting. It's just loud, scratchy static. The key with recording distorted guitars is to start with the gain low and keeping things clean and then slowly increase the gain until you reach that point where the distortion is just enough and then back it off a taste. That's about all you really ever need. You can always record another track or so with the distortion as crunchy and raging as you wish and you can blend the two tracks together to create a nice effect. The point here is that too much gain, or rather too much distortion washes the tone out of your electric guitar recordings. It's not a case of more = better. You often times need a lot less distortion to record effectively than you would need if you were playing at a live performance. Interestingly enough the best sounding guitars I've heard recorded were very full and rich and only had a little distortion to make it sound powerful. But they all had plenty of low and mid frequency energy and a lot of tone.

Speaking of tone, another way to get your guitars to hit hard is in how the performer voices their chords. Instead of playing all of your chords in the most common voicing you might try using power chords for the heavier parts and then on separate tracks using higher voiced chords for the purpose of ultimately blending the two together as one take. It's also very helpful to sometimes tune down (using alternate guitar tunings like "drop D" or "drop C" or whatever works best for the song). Sometimes simply using alternate, lower tunings are useful for creating the low end punch that you might be lacking in your guitar tracks (the same thing goes for the bass guitar).

An interesting little "trick" you can use to get a hard hitting section to really stand out is to simply let it hit a bit harder at the preamps during tracking. But you must be careful not to overdrive your preamps (unless you're absolutely certain that's the sound you want, which most of the time it's probably not going to be). The key is to track the quieter sections just below line level, which in most cases is calibrated to be -18dBFS or 0dBVU. So what you can do is actually track at around -20 to -24dBFS for the the quieter parts and when the choruses or the hard hitting sections come in you can (on a seperate take) adjust the preamp gain so that it's a bit higher, like around -15dBFS to-12dBFS. That will automatically make those tracks "hit" harder when they playback, even when everything is at unity on playback. Trust me, those tracks will really hit hard then. But then you'll have to pay close attention to automation later in the mix process.

You can't be afraid of riding the faders. It's one of the purest skills of an audio engineer. You've got to be able to listen to a track and know how and when to pull things down that are too much and push things up just enough to get them all to sit well. That's the kind of thing that comes with a lot of practice (and something that we can cover in much more detail later). The point is that jumping straight away to the compressor or limiter is really the lazy way out (and sometimes it just doesn't sound all that great). Unfortunately there are many famous engineers that base their entire approach to mixing on the abilities of a compressor or limiter to subtly influence their 2 bus sound. But I think that it's become more of a crutch than a help for the newcomer. The most annoying thing I find going on with modern mixes is how the verse will be loud and the chorus will actually seem to get quieter, even though there is an abundance of distorted guitars and uptempo rhythms. That's pretty much the opposite of what you want. Everything is supposed to build up to the moment of release, which is the chorus in the vast majority of cases. So why would you slap a limiter on your heavy guitars and subsequently make them sound weak?

[RANT] Stop copying the non fundamental techniques of famous engineers. Engineers become famous because they happen to be involved with projects that happen to sell. It's really not about their skill in most cases. I'm not saying they're not skilled. They are and very much so. But you too would be famous if your mix (good or horrible) sold two million copies. Then people would flock with questions on how you did it. But that's not the way to become truly good at tracking and mixing. The way to be good is to master the fundamentals until you can pull them off in your sleep. Then you can experiment with esoteric practices that may or may not be what's needed on a particular project.[END RANT]

But back to guitar tone. Another thing to consider is taking a harsher sounding guitar track and blending it with a more rounded and mellow sounding track. The two extremes tend to compliment each other nicely in some cases. The bottom line here is that IT ALL DEPENDS. But what matters most is keeping in mind the context of the song, the arrangement and the intent in each section of the song and knowing how you can remain faithful to those goals while tracking and mixing. So the way to get your stuff to hit hard is simple: Stay out of the way. You want to make sure that you get things right at the source and then you want to make sure that you preserve that by doing as little as possible to make things sound just right during the mix process. Compression and limiting and even EQ are extreme techniques that require care and skill to pull of compotently. If you master not needing them so much then you will know how to be restrained when the time comes for you to actually use them. Which is in my opinion, the most effective way to learn. There is indeed a time and place for everything and I believe the that goes for compression and EQ. But start off without it and see how far you can really go. You might just be surprised.

HERE is an example of a song that I tracked and mixed using no EQ or compression/limiting at all on the guitars or the output buss (I did however raise the volume a bit with a limiter on the stereo file because it was simply too low to be comfortably enjoyed by most folks. However that's FAR from "mastering"). It's a large file because I didn't want to compress it to mp3, so it will take a high speed connection about a minute to download. It's also unmastered so the overall volume is relatively low. Take a listen. Incidentally, it was all done on very cheap gear. If anyone comments I'll reveal that gear.

Stay tuned for part 5 when we talk about tracking and mixing effects for the guitars.

Liz

Electric guitars without EQ or Compression (part3)

2009-04-09T23:30:00.000-07:00

TRACKING TOO HOT

Everybody wants their guitars to be as loud, punchy and exciting as possible. So naturally a lot of new engineers make the mistake of getting everything set up and then setting the preamp gain entirely too high and tracking things entirely too hot. This is the guaranteed way to make your tracks lose all punch and clarity in exchange for a mess. Why? Because your equipment isn't made to work that way, despite other folks telling you otherwise. It's useful to remember that getting volume out of a track is something that happens at the mastering phase and not while tracking and mixing.

Your preamplifiers are desinged to raise the level of an incoming microphone signal to a much higher (line) level so that you can actually use that signal to record with. Preamps are just circuits and if pushed too hard they can fail. In addition to that, preamp circuits begin to fail as you approach their design limitations. So a signal will distort as you begin to push the preamp too hard. Of course the better the preamp the less distortion such as this is a problem. But every preamp has it's limit (and that's nowhere you want your tracks to be) and tracking too hot is the way to reach that limit. Tracks that are too hot can sound less clear, thin, weak, harsh, cold, two dimensional and generally unpleasant to the ear. If you don't want your tracks to sound like any of that do not track too hot.

Preamplifiers aren't the only things that can suffer as a result of tracking too hot. If you are tracking digitally and you track too hot then you are also running the risk of pushing your A-D (analog to digital) converters too hard. Just like a preamp a converter is designed to work at line level and when you start to push it too hard it's analog stages can start to distort and if the level exceeds the digital full scale then you're going to get ugly digital distortion. So what can you do? It's simple, stop tracking too hot.

You should be tracking at or around an average of line level for your system. What is "line level"? It's really just a reference point for audio recording. On an analog VU (volume unit) meter it's the point at which the meter reads 0dBVU (that's zero deci-bels volume unit). In digital recording it's the point at which the digital meters read -18dBFS (that's negative eighteen deci-bels full scale). Now the point to remember is that those two scales are related in that way. The analog (VU) scale is different than the digital (FS) scale but the two are related in that 0dBVU is equal to -18dBFS (In most cases. Some converters are calibrated differently, but probably not yours). What all of this means is that when you turn up the gain on your preamp as you're setting your tracking levels and you take a look at the meters in your sequencer program they should be reading no higher than somewhere around -18dBFS. Following this advice is one of the best ways to ensure that you don't track too hot.

Will it sound quiet? Yes, if you've gotten used to tracking too hot. But in reality you've go so much usable dynamic range with 24bit recording (which I'm assuming is the bit depth that you're tracking at) that you don't need to worry about that. In actuality it's not quiet at all. It's actually the point at which your system was designed to operate at. To be on the safe side I like to track things at or around an average of -20dbFS. That pretty much eliminates all possibility of me tracking too hot. Because I'm aware that there are going to sometimes be transient spikes that get well above line level but they are not a problem as long as I have plenty of room before my signals distort (headroom). If as a rule you are tracking at or around line level you will be preserving headroom to the point of being able to not sweat little peaks here and there and your recordings will begin to sound better, much better. This is especially true with electric guitars. The electric guitar really starts to sound terrible when it's tracked too hot and you don't want to do that to yourself.

With the electric guitar (and this is not exclusive to the electric guitar, it just happens to be that's what we're focusing on) part of what you have to do is make sure that when you track things that have different volume levels (like clean parts and then distorted, heavy parts) that you take the time to re-adjust the preamp gain accordingly. This is not a "set and forget" method. It requires that you take a look at what's going on in the session and make changes when necessary to keep your tracks at or around an average of line level. The good news is that you can always break a take into seperate tracks so that you can take the time with each track in the take to make sure that the preamp is at the right gain setting. Then upon playback you don't have the problem of the takes suddenly getting entirely too hot in certain sections.

Stay tuned for part 4 when we explore how to get the guitars to hit hard without a compressor.

Liz

Electric guitars without EQ or compression (part2)

2009-04-05T07:32:00.001-07:00

A GOOD PLACE TO STICK A MIC

On a typical loudspeaker, like the ones found in your speaker cabs, there are three visible parts. The dust cap, the diaphragm and the surround. The dust cap is the (usually convex) protruding circle that's in the center of the speaker. The diaphragm is the rigid, concave, conical structure that is around and extending outward away from the dust cap and at the edge of the diaphragm where it would meet the enclosure is where you'll find the surround, usually made of a flexible, rubber like substance, (if not actually rubber itself). Mic placement with regards to a guitar speaker cab is pretty straightforward and simple. The sound at the center of the speaker, directly in front of the dust cap, is the most biting and harsh sounding and the sound gets less harsh and more smoothed or washed out as you move away from the center towards the surround. I have, through trial and error, found what I usually consider a good "neutral" spot to place a microphone and that is distanced approximately an inch away from and pointing directly at the meeting place of the dust cap and the diaphragm, so that the seam that joins them visually bisects the center of the microphone's diaphragm. That spot is where I've found a good blend of harshness and smoothness consistently. I always start my mic placement there when I'm tracking and close mic-ing a cab. But it's important to note that when you're close mic-ing things distances as small as a centimeter or two can make a huge difference in the sound that you pick up. The idea is to start at a familiar place and tweak as you go from there if the need be. The reason that I go for neutral right off the bat is because there are so many parameters to be adjusted that until I'm/we're sure what tone and effects we want to track I don't want mic position on the cab to be a big influence on the decision. Once you've gotten things worked out with your tone and effects then you can fine tune things a bit by angling the microphone towards the diaphragm or the dustcap or upwards or downwards or leave it strait on axis or whatever. The idea here is to take your time and experiement until you get the best possible sound that you can with your setup.

Speaking of close mic-ing, it's good to remember that distance is an important factor. If you want more of the sound of the room in your tracks then you must move the microphone farther away from the loudspeaker. Exactly how far is up to you and your specific goals. I tend to like to close mic speakers and add any extra reverberation artificially, after the fact. But there have been times where the idea was to make everything sound like it was in a specific room and thus I had to distance mic the cabs.

Stay tuned for part 3

Liz

Electric guitars without EQ or compression (part1)

2009-04-04T20:25:00.000-07:00

I track and mix mostly rock music these days. I'm young. I'm all about keeping current. I know about the "sound of our day". But I've just gotta ask, why are we always screwing up our electric guitars in modern rock recordings?

I seems to me that in a majority of modern recordings there's entirely too much nonsense going on with the treatment of the electric guitars with respect to compression and equalization (as well as other choice processes). The odd thing to me is that people tend to think that it's necessary to "do something" to a recording to "make it sound good". That idea is part of a larger problem that I see with modern audio engineering in general. As I see it, quality recording isn't about "doing something" to a track in order for it to sound good. It's more about getting the best possible sound to be captured in the first place, so you won't have to "do" much of anything at all. Those kinds of recordings make for the best mixes in my opinion.

AUDITIONING THE EQUIPMENT AND DIALING IT IN
(First the really obvious stuff)

So with the electric guitar there are a few things to consider. The first of which is the actual sound of the guitar itself, through an amp, with no effects. Here's the thing to do. Audition it. Make sure that the guitar sounds good all by itself with a good amp. This can easily be done in a retail store, as they usually have plenty of amps for you to plug your prospective guitar into to check out. That must be done to know rather or not you actually like the sound of the guitar. Now of course this also is to see which amp sounds best for the kind of tone that you're going for with your guitar. Once you're clear on which guitar and amp you like, then you can focus on getting used to setting up the sound of the guitar with effects. This is important as you have to dial your effects and your tone in just right to get anywhere with an electric guitar recording. It's not about anything if it's not about dialing it in right.

Dialing in an electric guitar's tone and effects is the holy grail of getting a great core sound. A lot of not-so-talented musicians screw this part up and a lot of impatient and inexperienced engineers do also. I'll admit it. While it happens to be the case that I play acoustic guitar, I don't really have that much knowlege on the esoteric details of electric guitar effects and tones. What I do have though are very good ears and a good taste for what sounds good. That and a dose of patience mixed with determination to get the sound that you're after is all anybody needs (because figuring out what each knob and button does isn't rocket science). Amps will vary in exactly how they work. Some have knobs that affect parameters in a cascading pattern (like Mesa amps). Others will be different. The point here is that a guitarist isn't ready to record for a commercial release if they don't have a good sense of how to use their own equipment. They should be able to get at least in the general area of the tone that they're looking for and then you as the engineer should be able to take them the rest of the way through taking the time to tell them to track a scratch take, listen in the monitors, stop and adjust, track another scratch, listen in the monitors, stop and adjust some more etc... This process, of course, takes time and it's the hard work that nobody seems to want to do with recording. But to be honest it's the work that pays off the most IMO. Your guitarists should be coming into the control room with you and listening to all of the scratch takes that have real potential so that they can be confident that the sound that's going down is going to be the sound that they want. But there's more. Let's not forget mic positioning and the room that you're recording in.

Stay tuned for part 2

Liz

The Price of Admission

2009-03-19T11:03:00.000-07:00

In a word, expensive.

There is not really an easy way to tell the truth to somebody who might not want to hear it. So if you're somebody that might not want to hear the truth about the cost of high quality DIY recording then I suggest you skip ahead to the next part. But the bottom line is that getting high quality results with the practice of audio recording and mixing is (relatively of course) expensive. I'm talking in the $5k to $10k range for a reasonably high quality, flexible modern digital project studio input and monitoring chain (and that's not including all of the peripherals) that somebody with skills and experience can make sound fabulous.

I know some people might be thinking, "There is no way anybody these days has to pay that much for a home recording set up." In a way that's true. You can always cut out some flexibility, expandability and/or overall quality to get a rig that cost less than a grand. But will you be doing it the way that you likely envision doing it? Probably not. Most people dream about having at least 16 high quality channels and a few really high quality channels on top of that to track with. They also think about having all the mics they need, all the cables and headphones they need and all of the other peripherals as well. They also think of having cue mix ability with talk back options and simply pressing record and everything working out just fine. They also envision mixing with a killer monitoring system and creating a product that translates to anywhere with a high degree of consistancy. All of these things and more are on the wish list of every wanna be recording engineer. But they all come at a cost and unfortunately for most people that cost is rather high.

Oddly enough one of the questions that people ask me over and over goes something along the lines of, "What's the best monitoring system for $300 or less?" That question is really a difficult one for me to tackle. The reason being is that all of the monitoring systems for $300 or less pretty much suck. Can they be used to make a mix on? Sure. Is it an easy or practical goal for a newcomer to expect to be able to do? Absolutely not. The best a newcomer can do with a $300 monitoring system is simply get by until they can do better or only practice. Even if that's the goal (practice) it will be next to impossible to make any real strides in educating their ears without properly made and implemented acoustic treatment for their listening space and that means more money indeed. So the real answer to the $300 or less question is that there is no such thing as a useful $300 or less monitoring system. Firstly, because an untreated room is part of that system and it will wreck any opportunities to work with $300 monitors and secondly because (like I said) $300-a-pair monitors suck compared to only the very next distinct step up in quality.

This really leads to a question of priorities. Why are you tying to record for yourself? If you imagine that you're going to make a go a being a professional recording engineer (or at least releasing professional quality results) then you should know that this stuff isn't a piece of cake like it may seem to be. It's expensive and time consuming and there is a lot of education involved in getting it right. Even some of the professionals get it wrong sometimes and a lot of them have much more money to waste than you do. So this isn't a dip-your-toe-in kind of activity. If you're going to do it then you are best off being aware of exactly what you're getting in to and how much you're going to have to invest to get it right.

If you're only interested in getting involved for the purpose of tracking an album or two, then I recommend investing that money into time at a well equipped and properly staffed professional recording studio. Things will go a lot smoother and you'll likely be a lot happier with the results. Don't think that you can learn how to record yourself in a few months. You are far more likely to learn how to record at a high level of quality in a couple to a few years (along with thousands of dollars and hours invested). I'm not trying to kill any dreams here. I'm just trying to put everything in perspective.

But if you are up for the long haul, then welcome. I'm all for somebody commiting themselves to a process that will, in all likelyhood, take some time to come to fruition. I've embarked on that journey and almost ten years later, I haven't quit searching yet. Good luck to you and good travels along the way.

Liz

Auditioning Monitors (loudspeakers)

2009-03-16T17:12:00.000-07:00

This is about the most thorny topic I know of to discuss with experienced folks because opinions abound. But I'm going to try and save you a little trouble by keeping it simple. First of all what is a monitor? A monitor is a speaker that is supposedly designed to help a person hear what was recorded and subsequently allow a person to hear the changes that are happening to the recorded material as it is going through signal processing, either digitally with plug-ins or in the analog realm with actual hardware processors.

That's it. There's really not much else to a monitor's mission than that. But the problem is that's a very difficult goal to achieve. Because sound is a complicated thing and designing and building a device to accurately reproduce sound is not a task for the faint of heart. That being said many have tried to design a quality loudspeaker and just about everyone has gotten it wrong at some point with some or all of the the design.

These days loudspeaker technology and design is at an all time high. There are some very good models to choose from. The problem is that along with the good comes the bad and there are many, many well marketed, yet very poorly designed or manufactured (or both) models that are misleading the wanna be recording and mixing engineers into confusion about exactly what to do and which to choose. Which brings us to rule #1 in monitors.

RULE 1
No matter what loudspeakers you have, or however much they cost or how good they are, they will only be as good as the acoustic quality of the room that you're using them in.

That's a serious thing to know. You can go out an purchase a pair of Lipinski Sound L707 speakers, complete with stands and subwoofers and use the custom amplifiers that they have for them and you still will have a system that sounds like sh*t if it's in a room that sounds like sh*t. It's very important to know this. You've got to get the room right first before you can know how a loudspeaker system really sounds. That's just the plain old truth.

This brings me to my next bit of advice. I think that it's pretty much a waste of time to audition speakers in a store. The acoustics of the stores that most people frequent tell the listener nothing about the actual sound of the monitors in question. It's therefore important (if you're serious about this stuff) that you take the time and money to actually purchase a pair of monitors that you might be interested in, take them home to your (hopefully treated) listening space and audition them there for a few days. The point is that you'll know for certain what you're getting into with whatever monitors you're working with. You will not then be mislead by the acoustics of the store. This means that you're very serious about what you're doing. You don't take anybody's word for it besides general recommendations. You take them home and you listen for yourself. That's the way to get at the truth. That would of course mean that you must have a good return policy on the pair that you buy and that you treat them with the utmost care so that you can return them and get your money back should you need to do so. I know that this process can be a very time consuming one. But if you go about it correctly then you greatly increase your chances of picking the right monitors for your ears and for your room.

What you're looking for with the selection of loudspeakers is translation. Translation is the loudspeaker's ability to let you comfortably mix your music on them and subsequently play your finished mixes back on other systems and things sound just as you'd expect them to sound, no nasty surprises like all of a sudden your mix sounds super muddy or super harsh or a combination of mis- matched levels and effects. You need to be comfortable enough with your loudspeakers that you can know what your mixes will translate like on other systems. Why? Because that's the goal of mixing; Translation. You want your mixes to sound balanced, clear and lively on the widest variety of playback systems and the best way to achieve this goal is to mix in the most balanced and accurate mixing environment, with the most balanced, accurate and translatable system you can get. Getting to that point might be a hassle, but it's worth it in the end when people hear your mixes and think that you know exactly what you're doing. There are few more satisfying feelings than landing a 3pt shot at the buzzer to win the game, hitting a home run, serving an ace or producing a kick ass mix.

Liz

The basic elements of a modern digital studio

2009-03-16T16:53:00.000-07:00

Often times people ask me what the basic elements of a studio are. I'm always tempted to give a glib response. But in reality it's a very legitimate question. I remember when I was purchasing most of my gear. I had no idea where to start. So I've laid out a simple list of things that one can consider when buying gear for a digital studio.

Basic equipment:

1.) Microphones
2.)Pre-amplifiers
3.)Analog to digital converter (aka: A-D converter)
4.)Computer audio recording interface
5.)A capable computer
6.)Sequencer software program (i.e. Cubase or Pro tools or Samplitude or any of a host of others will do)
7.)Third party developed audio processing plug-ins (optional but very useful indeed. I favor Universal Audio's UAD series).
8.)Digital to analog converter (aka: D-A converter)
9.)Monitor controller (optional but very useful.)
10.)Loudspeakers designed for critical listening applications (aka: "studio monitors")

These days #'s 2, 3, 4 and 8 are often found in the same device that is simply labeled "audio interface". This is because it's usually cheaper to go this route in trying to score all of those pieces at once. But you can purchase each element separately. The price will increase dramatically though.

Another thing to consider is all of the peripherals that are involved with recording that you might not think about so much. Things like microphone stands, cables, a desk, chairs, cable snakes, racks, monitor stands, headphones, a control surface, a cue mix system etc... The list goes on and on of what you could buy.

The truth is that you can get carried away quite easily when trying to break into this stuff. The point is that you need to make sure that you consider each element in your signal chain carefully before you purchase it. Know that you need it. Know what it does, how it's connected to your signal path and how much you'll likely use it so that you don't end up wasting money. The time that you spend researching and creating a virtual signal path is time well spent and it can save you thousands of dollars and hours later down the line.

Liz

Get the room right first

2009-03-08T08:21:00.000-07:00

Recording is more about listening than anything. But we've got to be clear that listening isn't exactly the same thing as hearing. Hearing is simply the process by which you recognize compression waves in the air as sound in your brain from your ears. Listening is about discerning meaning in sound. That's the skill that you've got to refine. It can't happen though until you get your listening space right.

We tend to think that buying a new pair of expensive loudspeakers that happen to be labeled "studio monitor" is the key to getting our critical listening skills up to the task of producing recorded music. Nothing could be further from the truth. The truth is that firstly, every loudspeaker that's claiming to be suitable for critical listening isn't. Secondly, no matter what monitors you have in your possession, they can only be as good as your listening room allows them to be and finally, there's a lot of ear training that goes into critical listening for the purposes of tracking and mixing audio, not to mention the learning curve required for you to become intimately familiar with your loudspeakers. But it's important that you know before you can even take on the proper search for a loudspeaker system you've likely got a lot of work to do in your listening room first and it begins with broadband absorption aka: Bass Traps.

You can pretty easily go online and find your materials for building your own bass traps and by doing so you can save yourself quite a bit of money over purchasing them ready made. But in either approach be it DIY or purchasing pre-made traps, what you're generally looking to do is absorb the excess low frequency energy that is bouncing around in your listening room and misleading you in what you're hearing from your loudspeakers (monitors). An associate of mine (I like to think of him as a friend but I make no assumptions) likes to say "treat the low end first". -John Scrip, massivemastering.com

He should know. He's got years of experience and a great mastering room that he's been working with for quite a while. The key is to start with treating all four corners of your room first by straddling them from floor to ceiling (or as close as reasonably possible to the ceiling) with at least 4" thick bass traps. Usually this is done by having two traps in every corner, one placed directly above the other (so one is straddling the corner from the floor up it's own height (4') and the other straddling the corner right where the first left off, up the rest of the height of the corner) with each trap being 2' wide by 4' high and being constructed of a simple rectangular wood frame inlaid with 2'x4'x4" panels of acoustic absorption material. That material can be one of a few useful options, the most popular being Owens Corning 703 or Roxul Rockboard 60 (the rockboard 60 being cheaper but just as effective as the 703). You can look here for a source of information on absorption material. Also remember to stay away from foam products when treating the low frequencies of your listening space. They are not effective for that (though they are often marketed as a cure all for your acoustics problems) and they are very expensive. Typically when the low end of a problem listening space is treated first and done properly, there isn't much else to do. So don't get swindled. Save your money and a lot of time.

There is much more to learn about exactly how to finish treating your listening space and there are also people better suited to explain it than I am. I'm a recording engineer, not an acoustics expert. But I do have experience with the latter due to working in the field of professional music production. If you take nothing else away from me remember,

-Treat the low frequencies first.
-Stay away from foam products when treating the low frequencies of your listening room.
-When treating the low frequencies remember to tackle the corners of your room first.
-Your loudspeakers will only be as good as the room that you're listening to them in will allow them to be. (that one's from John Scrip @ massivemastering.com )
-Acoustic room treatment doesn't have to be expensive. You can do it yourself. Here's a good place to get some ideas on how.

I know room treatment isn't sexy. It's not like buying gear and getting to see what it does to your mixes. But the truth is that proper room treatment is one of THE BEST things you can do for your mixes. Your loudspeakers will finally be telling you what you need to know (provided they're any good at all). The stereo image will be clearer. The bass will be tighter. Everything will come into perspective with considerably less effort and second guessing. It's so worth it I can't possibly overstate it. Just take the time to do this right and you'll thank yourself for it in the near future. Besides, it makes your otherwise ordinary room look pretty cool.

Liz

In the beginning there was the source

2009-03-06T12:07:00.001-08:00

So your raw tracks sound bad.

That's a bummer. But you've heard somewhere that if you have a UAD2 omni quad card and the Waves Diamond bundle that you can make them sound killer. All you've got to do is spend the close to $10k in software and you're in business, right?... Wrong.

Well then it must be the preamps and conversion that you're using on your interface. If you could get your hands on 16 channels of Neve 1073 or maybe some API 312 and then run it through some Apogee conversion then your tracks would sound killer for sure, right?... Wrong again.

Okay, okay it must be the microphones. With some Neumann and Geffell mics you'd definitely be in business, right?... Once again, wrong, wrong, wrong.

It ain't the mics. It ain't the preamps or conversion. It ain't the plug-ins. What's missing from your equation isn't really so much about any of the things in your input chain. Even a humble input chain like an sm57 run into a Presonus Firebox and mixed on stock plug-ins is capable of professional quality. But the key to making something like that happen is the source material.

It's all about the "good rule".
Good song writing + good song arrangement + good musicians + good intruments + good tuning and settings on those intruments + good performance + good space to capture that performance in + good microphone placement on each source instrument + good gain settings on the preamps for each microphone = potentially good raw tracks. Add to that fantastic equipment or even better-than-average equipment and you will have absolutely AMAZING raw tracks that will stun you while you just have everything playing back with the faders all at unity gain.

Tracks like that don't need a lot of help from fancy signal processing. They already sound phenominal. Which is why your stock plug-ins would work out just fine mixing in a situation like that, provided that you have the skill to tweak them accordingly.

So the task at hand is to focus on getting everything that comes before the input chain to sound killer and then get a decent input chain (even a basic one will do), then lo and behold you'll have killer sounding raw tracks. It all comes from getting it right at the source.

Liz

Change: it's a good thing

2009-02-25T09:49:00.000-08:00

If you're in it for the music then you've never had a better opportunity to make things happen. We are experiencing a change in the way music is recorded and produced. Change can be good (and it's inevitable anyway) and with it comes lower costs and potentially higher quality. The most important thing to remember these days is that it's not about the gear that you have, since now anybody can use virtually the same stuff that world class facilities are using. The difference in quality from amateur to professional is going to come down to skill, experience, preparation and attention to detail. That's the whole ball of wax.

I'm of the new breed of professional recording engineers. I track to hard disk, or to a computer and I mix all digital, in-the-box. Plug-ins are the order of the day with me and I have no problem getting professional results with them. I've heard all of the analogue lover arguments about the detriments of in-the-box mixing. I don't buy any of it. The way to get things to sound good these days is the same way there was before, using your ears and your experience to tweak it just right and being sure to capture it correctly at the source. Nothing in that has changed really. So don't allow yourself to get caught up in which $2000 signal processing box and what $60,000 and up mixer is going to make your tracks sound good. None of that matters. What matters is you educating your ears and growing in experience enough to make the right decisions when tracking and mixing. That's all there is to it.

Stay in it for the music.

Liz