Tech Tip Of The Day: UPS Mania | ProSound News

The many types and applications of uninterruptible power supply systems, and clarifying terminology…
by PSW Staff
sweetwaterQ: I’ve been getting quite a few questions from clients lately about UPS (uninterruptible power supply) systems.The problem is, most either want to run out to their local office supply store and buy what’s cheapest, which I’ve insisted probably isn’t the best idea, or they’re an audiophile who’s heard that on-line UPS systems are the way to go.

What can I tell people who don’t understand the value of the (many) other types of UPS systems, or don’t even know they exist?

A: No doubt about it, you’re certainly stuck between a rock and a hard place. As you mentioned, it’s widely believed that there are only two types of UPS systems, namely Standby and On-Line.

And, to make matters far worse, these two terms are often incorrectly applied to many UPS systems on today’s market.

As to what can be done I’ve always found that the best defense in your situation is a good offense, as many misunderstandings about UPS systems are cleared up when the different types of UPS topologies are properly identified.

Often time when clients are presented will ther entire picture (which often involves you educating them a bit), they realize which unit is best for their situation themselves.

To assist in your UPS client education, the major systems have been defined below (Note: All references to kVA sizes are general.). Hopefully this helps!

Standby UPS
This is the most common UPS type used for personal computers. The transfer switch is set to choose the filtered AC input as the primary power source and switches to the battery/inverter as the backup source in case of a failure of the primary source (AC).

In the case of power failure, the transfer switch must operate to switch over to the battery/inverter backup power source. The inverter only starts when the power fails, hence the name “standby.” Benefits of this model include high efficiency, small size, and low cost.

Line Interactive UPS
This is the most common UPS used for small business, web, and departmental servers.In this design, the battery-to-AC power converter (inverter) is always connected to the output of the UPS. Battery charging is provided by operating the inverter in reverse during times when the input AC power is normal.

When the input power fails the transfer switch opens and the power flow is from battery to the UPS output. The fact that the inverter is always connected to the output provides additional filtering and yields reduced switching transients when compared with the Standby-type UPS.

The inverter also provides regulation, operating to correct brownout conditions, which would otherwise force the UPS to switch to battery operation. This allows the UPS to operate at sites with very poor power. The inverter can be designed such that its failure will still permit power flow from the AC input to the output, which eliminates the potential of single point failure and effectively provides for two independent power paths.

This topology is inherently very efficient, which leads to high reliability while at the same time providing superior power protection. High efficiency, low cost, high reliability coupled with the ability to correct low- or high-line voltage conditions make this the dominant type of UPS in the 0.5-5 kVA (volt-ampere) power range.

 

Standby On-Line Hybrid
This topology is used for most UPS under 10 kVA (volt-ampere) that are labeled “on-line.”

Just as with a Standby type, the standby converter from the battery is switched on when an AC power failure is detected; additionally ,the small battery charger resembles a Standby UPS.

However, the Standby On-Line UPS will exhibit no transfer time during an AC power failure.

The most misunderstood part about this topology is the belief that the primary power path is always “on-line,” when in fact the power path from the battery to the output is only half “on-line” (the inverter), while the other half (the DC-to-DC converter) is operated in the standby mode.

This design is sometimes fitted with an additional transfer switch for bypass during a malfunction or overload.

Standby-Ferro UPS
This was once the dominant form of UPS in the 3-15 kVA (volt-ampere) range. The design depends on a special transformer that has three windings (power connections). The primary power path is from AC input, through a transfer switch, through the transformer, and to the output.

In the case of a power failure, the transfer switch is opened, and the inverter picks up the output load. In the Standby-Ferro design, the inverter is in the standby mode, and is energized when the input power fails and the transfer switch is opened.

The transformer has a special “Ferro-resonant” capability, which provides limited regulation and output waveform “shaping.” The isolation from AC power transients provided by the ferro transformer is as good or better than any filter available, but the ferro transformer itself creates severe output voltage distortion and transients, which can be worse than a poor AC connection.

Even though it is inherently a Standby UPS, the Standby-Ferro generates a great deal of heat because the ferro-resonant transformer is inherently inefficient. Standby-Ferro UPS systems are frequently represented as on-line units, even though they have a transfer switch, the inverter operates in the standby mode, and they exhibit a transfer characteristic during an AC power failure.

High reliability and excellent line filtering are the strengths of the Standby-Ferro design. However, it has very low efficiency combined with instability when used with some generators and newer power-factor corrected computers, which has caused the popularity of this design to decrease significantly.

Double Conversion On-Line UPS
This is the most common type of UPS above 10 kVA (volt-pmpere). Double Conversion On-Line UPS is the same as the standby UPS except that the primary power path is the inverter instead of the AC mains.

In the design of Double Conversion On-Line operation, failure of the input AC does not cause activation of the transfer switch, because the input AC is not the primary source, but is rather the backup source.

Therefore, during an input AC power failure, on-line operation results in no transfer time. The on-line mode of operation exhibits a transfer time when the power from the primary battery charger/battery/inverter power path fails. This can occur when any of the blocks in this power path fail.

The inverter power can also drop out briefly, causing a transfer, if the inverter is subjected to sudden changes in the load, or if the inverter experiences an internal control “glitch.”

Contrary to popular belief, Double Conversion On-Line UPS systems do exhibit a transfer time, and in actual installations may transfer as frequently as Standby-type UPS systems; however, on-line UPS transfers are not related to AC input power failures as they are in a Standby UPS. Both the battery charger and the inverter convert the entire load power flow in this design, which causes undesirable heat and results in reduced efficiency.

Due to practical design constraints, UPS below 10 kVA that are represented as Double Conversion On-Line UPS are almost always actually of the Standby On-Line Hybrid UPS. The Double Conversion On-Line UPS provides nearly ideal electrical output performance.

However, the constant wear on the power components can reduce reliability, and the energy consumed by the electrical power inefficiency is a significant part of the life-cycle cost of the UPS. Also, the input power drawn by the large battery charger is often non-linear and can interfere with building power wiring.

Delta Conversion On-Line UPS
This design is more recent, introduced to eliminate the drawbacks of the Double Conversion On-Line design and is available from 5 kVA and greater. Like the Double Conversion On-Line design, the Delta Conversion On-Line UPS always has the inverter supplying the load voltage.

However, the additional Delta Converter also contributes power to the inverter output. Under conditions of AC failure or disturbances, this design behaves identically to the Double Conversion On-Line. During steady state conditions the Delta Converter allows the UPS to deliver power to the load with much greater efficiency than the Double Conversion design.

A simple way to understand why this works is to consider the energy required to deliver a package from the 4th floor to the 5th floor of a building. It obviously saves energy to carry the package only the difference (delta) between the starting and ending points.

The Double Conversion On-Line UPS converts the power to the battery and back again whereas the Delta Converter moves much of the power from input to the output directly. In the Delta Conversion On-Line design the Delta Converter can also charge the battery so the overall design is no more complex than the Double Conversion approach and it provides the same output characteristics.

In addition, the Delta Conversion On-Line UPS offers reduction in energy losses and costs by approximately a factor of 4. As a side-effect of the design, the input power quality of the Delta Conversion UPS is also superior, particularly in the large kVA sizes.

For more tech tips go to Sweetwater.com

Electric Guitar Sound | ProSoundWeb

Good starting points for capturing and recording the sound you want…

by Barry Rudolph

guitar mic techniquesBecause of its fundamental importance in popular music, the electric guitar is the subject of intense scrutiny and wide differences of opinions. Just what makes a good guitar sound?

Compared to all the subtle and not so subtle sounds that come out of an electric guitar amp, fidelity judgments of vocal sounds are easy to make!

With good knowledge of the different guitar and amplifier sonic capabilities, coupled with good microphone techniques,we can achieve the ultimate guitar sound that “fits” the guitar part, song and production genre.

A good guitar sound starts with a good player with the right amp and guitar all working together. It’s unrealistic to rely on an engineer to make poor gear sound wonderful in the control room. Microphone choices and miking techniques are good starting points for capturing and recording electric guitar amp sound.

Microphone Selection
Microphone choice (for me) is as big a part of the guitar amp’s recorded sound as the amp and guitar used, volume played and player choice, because the mic type and placement will greatly influence the player’s performance and tone.

Dynamics
The Shure SM57 cardioid dynamic is the most common microphone used to record electric guitar. This started back when all the more expensive microphones had already been used in big tracking sessions. Engineers were left with the “lowly” SM57 to handle those loud, cranky, noisy guitar amps.

But it turns out that the SM57 is perfect for the task; its frequency response, originally tailored for speaking, matches the mid-range “voice” qualities of the guitar. Italso has a compression effect on loud sounds – it squashes nicely, facilitating the engineer’s job of maintaining consistent recording levels.

You’ll see engineers push a SM57 right into the grill cloth of an amp cabinet, taking advantage of the proximity effect, which boosts low frequencies when the mic is placed close to a sound source. The SM57 locks in a certain “size” for the electric guitar, maintaining its appropriate place in the mix without additional EQ or compression.

The Sennheiser MD 421U cardioid dynamic is also popular, offering a wider frequency response (more high and low frequencies) than the SM57. A five-position rotary switch adjusts the frequency response from the flat position, called M (for music), all the way to the contoured S (for speech).

Generally, I find the 421 brighter with less of the compression effect than the SM57. These mics are also more directional, which is important for isolating the sound coming from one speaker in a multi-speaker cabinet.

Condensers
Condenser microphones also work great, but care must to taken to not get an overly bright sound. Your guitar player might complain that his amp sounds brighter than usual (compared to a SM57) and feel he must readjust his recording knob settings. As a result, I place them further away from the speakers.

Figure 1, click to enlarge (Photos Courtesy Of The Oliver Leiber Collection)

Figure 1 shows a Shure KSM44 condenser about 20 inches from a 1960s vintage straight Marshall 4X10 cab. (I was auditioning the three cabinets in the picture).

Condensers pick up more low frequencies from the amp, and this may or may not be a good thing. Pushing a lot of air might work in a heavy metal track, but it also might be inappropriate for a lighter pop song.

I’ve also noticed certain condensers sometimes add distortion when close-miking extremely loud amps. Occasionally, the metal wind screen can get loose and vibrates. Always use the attenuator pad and maybe the low frequency roll-off. The Neumann U 87 and U-47FET, Shure KSM44 and Audio-Technica AT4041 are all good choices.

Condenser microphones also provide the opportunity to experiment with different polar patterns such as omnidirectional and Figure-of-Eight (see my article about this here). Omnidirectional mics do not exhibit the proximity effect and will pick up more of the total sound of the amp and room tone, rather than one particular speaker.

I like omni mics for more of an ambient guitar sound or for room mics. Figure-of-Eight mics also pick up more of the room, but only from directly behind the mic’s body opposite the front side.

Speaking of Figure-of-Eight, one of my favorites for guitar is the ribbon microphone. I have a pair of Royer R-121 Figure-of-Eight microphones that offer a whole new range of warm electric guitar sounds.

Big, cumbersome and old ribbon mics have been around for years, but using them on loud instruments carried the fear of knocking the fragile ribbon element off their suspension mounts. The smaller, lighter Royer mics can handle huge volumes without the worry.

One engineer remarked to me,“when you switch from the Royer back to the 57, you wonder where half the guitar sound went.” Grantedm a big, fat and warm guitar sound on its own might sound ideal, but does it fit into your song production?

The Royer mic picks up sound from two opposing sides in what is also called a bi-directional pattern, and you can take advantage of this to get more of the recording space (room) in the sound. Sound entering the rear of the mic is 180 degrees out-of-phase with that coming into the front.

Placement

Various mic placements on guitar amps are closely guarded trade secrets amongst recording engineers, but here are some tried and true methods that’ll make good starting positions.

From The Front
Figure 2 shows an SM57 pointed at a very rare 1960s Gibson recording amp.

I aim the microphone exactly at the center of the speaker driver inside the amp. (For those who don’t know this amp, the speaker is not mounted in the exact center of the cabinet, but I’m acting as though it is).

To facilitate seeing through the grill cloth, use a flashlight.

This position produces the most high frequencies and moving the mic closer increase both level, low frequencies and reduces the cabinet’s contribution to the overall sound.

Some guitarists and engineers say there is nothing coming out of the exact center of a speaker that’s worth recording. (I could make a funny retort here too easily.)

This is true for certain speakers, but test this position yourself to judge its usefulness.

If you want less highs and more warmth, move the mic sideways, parallel to the floor, toward the outside of the speaker.

Move in 1-inch increments, with someone in the control room you trust listening for sound changes. With a mic positioned inches from a speaker cone, small increments make a big difference.

Figure 3, click to enlarge

At An Angle
Figure 3 shows an SM57 at a slight angle on a Fender Brown-Face Deluxe. I find this a better position most of the time since you seem to get plenty of highs and more tonality than straight on.

It’s a good idea to try from both sides and from the top or bottom. If the mic ends up on the floor pointed at the speaker, and sounds good, nail it down! The floor is going to trap bass frequencies and also add tone especially if it is made of wood and is built on a raised foundation.

Tilt The Amp
If the floor seems to “close down the sound,” try tilting the amp back. Fenders have chrome legs on the sides of the cabinets for that purpose, and it also projects the sound up at the musician. VOX amplifiers, like the AC30 and Super Beatles models, come with tilting carriage stands that completely isolated the amplifier from the floor.

If possible, I like to set the amp on a folding chair.

Figure 4, click to enlarge

Figure 4 shows an MD 421U on a vintage Fender Tweed Deluxe. Here, the amp is tilted back so that the bottom of the amp’s cabinet couples less with the floor.

All of these amps are open-backed, so a wall or open space directly behind them greatly affects bass response.

Front & Back
Figure 5 shows an open-backed amp, the Matchless DC30, with an SM57 on the rear and a KSM44 on the front.

Figure 5, click to enlarge

This set-up produces a very unusual tone and allows mixing the two mics for a mono track or using two tracks placed left and right in the mix. The phase of one of the mics should be flipped.

Try moving the mics very close to the speakers and processing the rear mic through a very short delay – less than 3 ms (milliseconds). Also, try NOT flipping phase.

Multiple Miking
This is a good way find out which speaker sounds the best in a 4X10 cab, and you can put each mic on a separate track and pan them left and right.

Be careful with phase, as some speaker manufacturers deliberately wire multi-speaker cabinets in and out of phase.

Some engineers will use a condenser on one speaker and a dynamic or ribbon on another, but I like to find the best speaker and use two mics on that speaker.

Even then, I sometimes hear a little “phasiness” between the microphones.

A very popular setup is an SM57 up close and then, four or five feet away, a U 87 or the tube U 67 for a cabinet mic.

In The Middle
In Figure 6, I use a Royer R-121 equidistant between a 1960s 4X10 Marshall straight cabinet and a reissue 4X10 slant cabinet.

Figure 6, click to enlarge

Both cabinets are driven by a single top, and it won’t work unless one of the cabinet’s speaker cable is wired out of phase, (or you have determined the cabinets are already out of phase with each other).

With both speaker cabinets pushing and pulling together and the warm sound of the ribbon in the middle, you’re not going to get any fatter tone, especially for clean sounds. (The VOX cabinet was not in play here… but who knows?)

Scientific Starting Point
Figure 7 shows a scientific approach to mic placement on a Beatles AC30 VOX amp. The tech in the picture is using test instruments rather than his ears to find the “sweet spot”.

Figure 7, click to enlarge

Sound waves coming out of a guitar amp emanate in all directions, setting up patterns much like a stone produces when thrown into a very still and flat pond. Like in the water, sound waves interact producing multiple reflections and standing waves where sonic energy coalesces. The idea is to place the mic where sound waves add together.

Determining this position requires a way to measure output level of a microphone while it’s moved around in front of the amp. First, set up for a guitar overdub with your microphone choice and fix the headphones so the musician can hear himself – unplug all headphones.

Then send a steady 700 Hz oscillator tone into the guitar amp’s input jack and place a large VU meter so it can be seen from wherever the guitar speaker cabinet or combo amp is located.

If this is impossible to do, plug a voltmeter into the cue system and read the level there. Setting the meter to read mid-scale with the cue system level control and with ear protectors on, (let’s make this science project as painless as possible), the mic should be moved around the front of the amp.

In doing this, you’ll find many peaks and dips in level. Also, I try to stand behind the speaker so that my body doesn’t affect the measurement process.

Obviously, as the mic gets closer to the speaker the level goes up, but if you keep a fixed distance and then move the mic left, right, or up and down, you’ll find a peak level. This is where you should set the mic.

Other Methods
These are some other ways to accomplish the task of recording guitar amps. Whether used alone or in combination with the aforementioned, all require much more experimentation and studio time.

Room Mics
Placing mics around the room is useful for effect, especially if you are recording in a great sounding space. This is an easy way to dress up a boring guitar sound, and there’s a lot of room to experiment! (Pardon the pun…)

Try different distances and heights, microphone types, EQ, compression and delay. I sometimes tape pressure zone microphones to the control room glass, or leave the studio doors open, or put a mic down a hallway. If you have an elevator shaft or stairwell handy, use that – put the amp at the bottom and a mic at the top.

Figure 8, click to enlarge

Recording The Electric, Acoustically
Figure 8 shows a Sennheiser e865 handheld condenser miking the “holy grail” of electric guitars: a 1959 Les Paul Sunburst.

Obviously, the amp should be placed in another room for this to work. But I have recorded amps and mics on separate tracks and it makes a very unique combo. Great for arpeggios!

Direct Recording
Direct recording, also called “DI,” from the British term “direct injection,” is simply recording the output signal of the guitar itself. Electric bass guitar is almost always recorded this way because of its super present and clean sound.

Guitarists like to play through some type of stomp box or pedal just to “vibe up” this very pristine sound. There is a whole category of guitar pedals, processors and effects, all meant to be recorded directly.

Whether you use a stomp box or not, here are a few caveats for getting the best sound from your guitar or bass.

Remember that the typical output signal of an electric guitar is very low level and very high impedance. Low level means that it will need amplification, and the high impedance requires matching to your recording system.

Guitars are equipped with either passive or active pickup systems. Passive is where the strings vibrate in the magnetic fields of the pickup and generate an electrical signal that is processed by passive tone control circuitry and then fed to the output jack.

Everything is the same with active pickups except that the electrical signal from the pickups is electronically amplified within the tone circuitry and then passed out the output jack.

Be aware of which system is used in the guitar when recording direct. Vintage guitars, because of the older and weaker pole-piece magnets in the pick-ups, put out an even weaker signal, and this is why a lot of people replace those pickups with newer and hotter pick-ups.

Further, when attaching a direct box (or anything else such as a digital tuner), the output impedance is lowered (if the attached device is not of sufficiently high impedance) diminishing the guitar’s signal further to say nothing of the tone, particularly the high frequencies. Only use a direct box with a very high input impedance to avoid an effect called “loading.”

On the active side, there is less stress since the built-in amplifier inside the guitar (that always seems to need a new battery) jumps up the level and lowers the impedance. As a result, whatever it’s connected to has less affect.

The guitar’s super-output level, if not treated correctly, may become a mixed blessing because it affects your amp’s tone and settings and the direct recording sound.

When direct recording guitars with active pickups, I use a simple, good quality matching transformer, (such as a Jensen), that converts to the standard microphone pre-amp input impedance of about 150 ohms.

I hardly ever use an active direct box for active guitars – too much activity! For passive guitar pickups, I like to use active direct boxes, such as models from Raven Labs and Countryman Associates, or the Demeter Tube models. These are all electronic isolation amplifiers with extremely high input impedances that will not load your guitar’s output signal and change the tone.

Speaker Jack Direct
Another way to record direct is by way of the speaker jack. A cable from the external speaker jack is connected to a matching transformer and that signal is recorded. The difference is that the sound of the amp, (with all its personality), and the guitar are recorded.

There are several models of power soaks or power resistor loads available that take the place of the speaker and provide an isolated recording line output. You can also put a mic on the speaker and mix in the direct signal. You’ll have to check phase again and probably use an equalizer.

Musicians In The Control Room (Or Not!)
For an electric guitarist, there’s no better way to achieve a great sound than by sitting or standing next to the amp. There is an important interaction between the guitar’s body and the vibrations set up by the amplifier.

Increased and unique sustain qualities, feedback effects and tone enhancement due to acoustic reinforcement only happen if the player is located next to the amp. When I listen to some of the newer rock records, I miss that sound.

In the recording studio, it may not be practical for the guitar player to be near his/her amp since the wide volume differences of all the sound sources in the same room (drums, bass, other guitars and singers) preclude the use of higher guitar amp levels during live band tracking sessions.

So the guitar amp or speaker cabinet is placed in an isolation booth for maximum control and separation. The guitarist can sit out in the studio with the drummer and the rest of the band, or even in the control room with the producer and engineer, but must rely on a good monitor mix to play along.

In this scenario, the guitarist loses all sonic interaction between his guitar and amplifier retaining only the basic tonal qualities of his amp.

Typically when a player is remote from the amplifier, the amp top is set next to the guitarist, and heavy-gauge cables connect it to the speaker cabinet.

Figure 9, click to enlarge

In the case of one-piece combo amps, running super-long guitar cords from the guitar to the amp is unacceptable, since the distributed capacitance along the long cable acts as a high frequency roll-off filter on the guitar’s delicate output signal. I recommend using a small buffer amplifier for this job that “conditions” or buffers the guitar’s signal for long cable runs.

The Little Labs PCP Instrument Distro, shown in Figure 9, is a distribution amplifier and switching matrix for sending electric guitar signals to multiple amps and/or at the same time, to +4dBv professional outboard signal processing gear.

The Distro also provides a clean, direct signal for recording directly to your recorder.

Barry Rudolph is a veteran L.A.-based recording engineer as well as a noted writer on recording topics. Be sure to visit his website, and also check out his related article, “A Wide Variety Of Microphone Techniques For Recording Drums”.

In The Studio: The Importance Of Pre-Production | ProSound News

by Joe Gilder

recordingOK, so you’re getting ready to start recording that next hit album. You’ve got the musicians lined up, you’ve scheduled your sessions. Now what? Do you jump right in and start recording? Easy there, Tiger. There’s an important step that needs to happen next: pre-production.

What Is It?
Pre-production is the planning phase for the entire album. It’s where you take a song in its simplest form (usually a guitar-vocal or piano-vocal) and plan out how you will build the recording around it.

This is a crucial phase of the entire process. You can liken recording an album to starting a business. You need a business plan. If you jump in without a plan, you’ll most likely end up scratching your head at the end of the project, wondering why you hadn’t planned things out more carefully.

Live & Die By The Arrangement
Pre-production becomes most important when dealing with the arrangement of the songs. I’m not referring to the order of the songs on the album. While that should be taken into consideration, it’s not crucial in the pre-production stage.

By arrangement I mean the outline, or structure, of the individual songs. Should there be two measures between the chorus and the verse or four? Should the bridge be longer? How long should the intro be? Should there be an intro at all? Will the tempo of the song remain constant throughout the song or does it need to change?

You need to ask these types of questions, because as soon as you dive into that first tracking session, you’re locked in to whatever arrangement you record. It becomes near impossible to change the arrangement of a song once you have several parts recorded.

The arrangement is like the backbone of the song. You can change the instrumentation as you see fit throughout the recording process, but the basic layout remains the same. That’s why it’s so important to get it right from the outset.

Pre-Production Steps

Perhaps this whole concept of pre-production is new for you. You may wonder what it looks like in real life. Here are some suggested steps you can take:

1) Determine a tempo. There are varying opinions on whether or not you should record to a metronome, or click track. Some would argue that music needs to breathe and flow, and that using a strict tempo inhibits creativity. While there is certainly some truth in this, a lot of people who complain about playing with a click track are simply complaining because they can’t play at a steady tempo!

If you’re recording an entire band, and everyone is playing at once, at least try using a click track. If the performance is really suffering, get rid of the click and let the musicians play. It’s better to capture an emotional, dynamic performance that waivers in tempo than a stiff, rigid, lifeless recording.

However, if you’re taking an overdub approach, recording only one or two parts at a time, it becomes really helpful to record everything at a set tempo. This is helpful for a number of reasons.

A) It allows you to record several takes of a single part. If they are all recorded at the same tempo, then you can go back and use different takes as needed.

For example, perhaps the guitarist nailed the verse but made a mistake in the chorus on his first take. Instead of hounding him to play one, single, perfect take. Have him record the entire song a couple of times. Chances are you’ll come away with one solid performance, and if not, you’ve got backup takes you can copy and paste as needed.

B) If and when you decide to add MIDI instrumentation to a song, having the song already mapped to a tempo grid makes it easy to quantize MIDI parts and even loops to the tempo of the song.

C) Having a steady tempo can keep the musicians from running away with the song, playing it too fast. You’ve heard this happen plenty of times. The musicians tend to speed up as the energy of the song increases. I’ve done this myself a lot. When I play a song at a show, I almost always play it a little too fast. In the studio, I let the click track reign in the speed of my performance.

2) Create markers. Most recording software platforms have some sort of marker system. This is a simple way to label sections of the song. These can be very simple — Intro, Verse 1, Chorus, etc.

However, you can get creative with these markers, especially during pre-production. Let’s say you’ve got the arrangement of the song figured out, but you’re not sure where you want the drums to come in, or where to put background vocals. Create markers for your different ideas. This way you won’t forget them before the next session.

In fact, Pro Tools lets you save notes with your markers (see pic), so you can document what you want to try in each section of the song. This gets saved with the session, so you won’t worry about keeping up with your notes on pen and paper!

3) Record scratch tracks. These are basic, quick recordings of the song that you use as your guideline. Typically it’s a guitar or piano track and a vocal track, recorded to the tempo of the song. (I name these tracks with an ‘X’ for ‘scratch’ in the beginning — X Guitar, X Vox, etc.)

The sound quality of the scratch tracks is not all that important. Most likely you’ll re-record these parts later. The purpose of scratch tracks is to allow you to listen to the arrangement and tempo of the song to decide what works.

I’d recommend dedicating your first recording session to simply recording scratch tracks. You can listen to them over the next few days and decide if anything needs to be changed in the arrangement.

Since these tracks aren’t that important, you can cut or add sections freely. It doesn’t matter if the transition doesn’t sound smooth when you copy and paste the different sections around. What matters during this phase is getting the arrangement right.

Once you have the arrangement like you want it, you can now begin to record the “real” parts. Eventually you’ll mute (and inactivate) the scratch tracks.

Give It Time
When finished with pre-production, it can be easy to want to just jump in and make that hit record. However, be sure to schedule in some buffer time between pre-production and your first recording session. Give your scratch tracks time to “simmer,” especially if you’re going to be recording relatively new songs.

Whenever I finish writing a song, I’ll usually make a quick recording of it. Over the next few days I’ll listen to that recording. Almost every time I’ll come up with things I want to change. Giving myself some time allows me make these changes before committing them to the final recording.

I know it may seem like a waste of time to do all this preparation, but it will save you time and frustration down the road. All of your favorite albums were not recorded on a whim. They were carefully planned. We don’t hear the planning in the music. We simply hear a well-performed, well-arranged album.
Joe Gilder is a Nashville-based engineer, musician, and producer who also provides training and advice at the Home Studio Corner. Note that Joe also offers highly effective training courses, including Understanding Compression and Understanding EQ.

In The Studio: Playing With The Click—Advice From A Top Session Drummer | ProSound News

Insights from Brian MacLeod, the pulse behind recordings by a “who’s who” of top artists…

by Bobby Owsinski
recordingAnyone who reads me regularly knows how much I love, admire and respect great drummers. My good buddy Brian MacLeod certainly falls into this category, having been the pulse behind recordings of Sheryl Crowe, Madonna, Christina Aguilera, Seal, Ziggy Marley and many more.

I remember the first time I heard Crow’s “All I Wanna Do” at a party as it was climbing the charts, and the only thing I could think about was “Wow, does the drummer ever groove on this.” And that’s exactly what Brian does—make the song groove and feel good in a way that few can.

Here’s an excerpt from an interview with Brian from The Drum Recording Handbook (which was written with the great engineer Dennis Moody) where he talks about playing to a click and gives some great advice.

There’s a defining moment for every player when they finally “get it.” What was yours?

Brian McLeod: That’s an interesting moment for me. I think it was on my first trip to LA on my first kind of big session. It was with Patrick Leonard (producer for Madonna, Rod Stewart, Jewel, Bon Jovi, Elton John and Pink Floyd among others). He flew me down from San Francisco, where I was teaching drums and playing live. I had toured a bit and done a few albums in England at that point.  Pat was starting a band and was auditioning me to be the drummer.

Now this is back in the days of tape. He played me a track, which was already finished, that had a click. I played that song and then another and he said, “Hey Brian, can you come in here for a minute” [to join him in the control room]. I thought to myself, “That’s my audition. I guess I’m outa here,” so I actually grabbed my stick bag and zipped it up so I wouldn’t have the embarrassment of having to walk back out into the studio to get it. I figured that the door out of the studio was in the control room and if he was firing me, I just wanted to be able to leave as fast as I can.

I zipped up my stick bag and walked into the control room and he looked at it with a confused look on his face, like “What are you doing?”. So I go, “What’s up?” and he says, “I love the way you play to the click. You know how to lock in without making it sound mechanical. I want to hire you to do this record.” Then he says, “Oh, by the way, listen to a track I just finished,” and he cranks up Madonna’s “Like A Prayer” at full volume. That was the moment I felt like, “Wow, I think I get this.” I realized at that moment that I really could play to a click and make it breath at the same time, and that really is an important thing for drummers to learn. If you play to a click, don’t be so focused on it that you lose sight of the fact that your actually playing a song.

What are you looking for in the phones when you record?

That’s a great question. Generally it depends on how we’re tracking. If I’m tracking with a bass player and we’re doing overdubs to an existing track, I’ll try to get a nice even level so it sounds like a record with the vocals and the bass player just above the music. I want to hear the bass player so I can be sure to lock my kick drum with him. Then if I’m tracking live I want whoever is the leader of the song to be above the track. Like if the guitar player has written the song, he might be doing some important inflections that I need to hear. If it’s a vocalist who has written the song and they’re evoking some emotion that they really want, I’ll make sure that is above everything else. So I latch on to whatever the main instrument of the tracking date is, or what is the biggest concern seems to be when laying down the basics.

I’ll also have the click at an ungodly level, which can drive producers and engineer’s crazy, so I like to use closed headphones for that. I’m still looking for the perfect set of headphones because you don’t want the click track leaking into the song. On Christina Aguilara’s “Beautiful,” you can hear a bit of the drum machine on her vocal track, but the vocal was so amazing that they just went with it.  You have to be careful especially on endings of songs. I try to get the engineer to cut the click off so that the cymbal sustain doesn’t have any click bleed. I’ll even punch in the ending of a song if they can’t catch it at the right time.

What kind of a click sound do you like?

In the old days I used to be very specific about it. I used to like a cowbell or some sort of side-stick sound with a shaker doing 16ths or 8ths depending on the feel of the song. I have to say that’s still my favorite click track, but I’m getting used to just the Protools click. I’ve adjusted over the years, but my preference still is the cowbell and shaker.

Do you any mic preferences on your drums?

Depending on the engineer and the producer, if they have a preference I’ll go with what they want, but I gotta say I really love a [Neumann] FET-47 on the outside of the kick drum. That’s one of my favorite mics. I like ribbon mics a lot for room and overheads. I like the Beyer M-160 ribbon on the hat. That warmed it up a lot.

I did a session the other day where we used Sony C-37s on the toms (which haven’t been made since the late 60s) and they sounded amazing. The producer said, “If you weren’t the drummer, I wouldn’t put them up,” because they’re so fragile that you have to be afraid of hitting them. That was really quite a compliment. Then again, some people get great results from Sennheiser 421’s.

I don’t generally do top and bottom mics on the toms. I don’t like too many mics on the drum kit unless the producer and engineer are really paying attention to the phase cancellation, but I have had good results with people who have done it that way.

I walked into a session with a metal producer who shall remain nameless, and he had the kit miked up with what looked like 40 microphones. I thought, “This is ridiculous,” but I played the track and it sounded amazing. Then sometimes I’ll work with just three mics on the kit and it will sound great too.

Everybody has their own technique and I try to be flexible because most of the people that I work with are so high end that I trust them to get my drums sounding the way they want them to sound.

You mentioned before about Patrick Leonard inviting you to LA to record. Would you consider that your big break?

I think so because after we finished that I record I was pretty much planning on moving back to the Bay area, but Patrick said, “Hey Brian, if you lived in LA I would use you on the records I work on.” Ironically the engineer/co-producer on that record was Bill Bottrell (who eventually went on to produce Sheryl Crow, Michael Jackson and Shelby Lynn) and he said the same thing to me. So I had two top-of-the-line producers tell me that if I lived in LA they’d use me on their records. It became a no-brainer for me to run up to the Bay area, pack my things in a U-Haul, and get my butt to LA. Then it kind of expanded from there.

I had no delusions of moving to LA before those sessions. I was too content up in the Bay area where I had a nice life teaching drums and playing live almost every night. It was wonderful, so I really didn’t want to move to LA unless there was a good reason because I didn’t just want to try to break in the way everyone seems to do it. It would have been too frustrating for me.

Do you have any other advice for a young drummer just starting out?

Yeah, I’d say try to play to a click as much as you can so you can learn to play with it yet lose sight of it at the same time. You want the feel of the click track to become like intuition, so it doesn’t make you feel shackled to it.

Also, when you work with a producer, be as flexible as you can be. Don’t be stubborn and trust the people you work with. If the engineer or producer has a suggestion, trust their advice. I was talking to a producer the other day about he’ll sometimes have a drummer come in that will insist on playing his own kit.

If I work with a producer that wants me to play his old vintage kit, of course I’ll play it because I think it’s important to be flexible. Even if you show up with your gear, if he has his kit miked up and he knows what it sounds like, I’ll generally do that. If they’re not satisfied after that, then I’ll use my drums.

Another thing, if you have any ideas, make the suggestion if the time is right because it’s all about teamwork and you’re on the team.

Bobby Owsinski is an author, producer, music industry veteran and technical consultant who has written numerous books covering all aspects of audio recording. Get The Drum Recording Handbook here.

In The Studio: Singer/Mic Positioning & Monitor Mixing | ProSound News

The tradecraft of recording vocals — techniques and approaches for “the most important thing”…

by Barry Rudolph
imageWith mic choice and signal chain dialed in, listen to your singer over the monitor speakers to determine the correct distance from the microphone.

You should have a fixed the initial distance back when you set the microphone height for the singer out in the studio. Some engineers will open a hand and use the distance from the end of the thumb to the end of the baby finger as the starting distance.

The volume or level of the human singing voice operates in ways very predictable. I’ve found (and then I’ve never recorded operatic singers) that generally pop singers are loudest at near the top of their full-voice range and lowest in volume at the bottom of their range.

Falsetto doesn’t count here, but singers with a falsetto louder than their full voice are in a blessed minority. In the studio this is a consideration since this vocal dynamic range must be handled so that both the quieter moments and big and loud moments are accurately recorded without noise or distortion. This is accomplished by either changes in the singer’s mic distance, a change in microphone pre-amp gain by the engineer, or the best solution, a combination of both.

If your vocalist sings loud all the time and all through the song, you’ll want to pick a fix distance from the mic for your singer to stand – maybe mark an “X” on the floor with tape.

For loud singing, this distance balances the mic’s direct sound with the room’s ambient contribution. Experiment with this distance because the room will contribute a feeling of size to a loud vocal that is energizing it.

At all times I try to keep my singer aimed at the mic’s capsule. Singing only inches off beam will result in radical change in “mic presence” – the components of the sound that, to our sense of hearing, builds a solid mental image of the singer performing in front of you.

In addition, you’ll change the tonality (relationship of top, middle and bottom of the vocal sound) and decreased intelligibility—important for understanding the words of the lyrics and for the vocal to cut a dense track in the final mix.

If your singer is a live performer used to backing off the mic for loud moments and then “eating the mic” for the quieter bits, you can make use of this in the studio.

But in the more sensitive studio, only a few inches in distance makes a huge difference. Too far off-mic sounds distant and too close builds low frequencies due to proximity effect and changes the vocal sound radically.

I’ve always preferred using the proximity effect on vocals if it sounds good. Proximity on soft vocals produces that “pillow talk” or “whispering in your ear” kind of close intimacy. To reduce the low frequency build up from excessive proximity, you will have to roll out LF or use the mic’s roll-off filter or do both.

Proximity fattens thin-sounding voices nicely and I try to keep most singers on mic even when they hit the loudest, highest and full-voice moments where there can be a tendency for their sound to thin out.

For rangy melodies that require huge leaps of volume to hit high notes and big chesty gulps of air to power the low notes, I like to work with the singer—learn the ebb and flow of the vocal performance and invisibly “ride” changes in mic gain that will not affect mic presence noticeably.

I’m changing the gain staging slightly – 1 to 4 dB maximum. This is the audio level going into the compressor so that the amount of compression from the quietest to the loudest remains about the same—perhaps a little more squash on the loud notes if it sounds good and causes the vocal to ride well within the track.

Monitor Mixing
During vocal overdubs the control room monitor mix should contain only track elements that will be in the final mixed version of the song. Apart from possible loud headphone spill (Of these tracks) on the lead vocal track, logically the singer should not have to sing around music parts that may not be used.

Depending on where the vocals are added, the monitor mix could be just a simple rhythm track to a fully sweetened production. Some artist perform better hearing everything around them and others not.

There are two considerations: a simple backing track gives more freedom to the singer for adlibs, improvisational vocal parts and customized “excursions” from the song’s written melody. Changes in the song’s arrangement and future production are often built stemming from these moments.

On the other hand, a properly pre-arranged, fully realized track production (brass, strings, solo section, backing vocals) affords the lead vocal to be produced and recorded emotionally and dynamically to “fit” the track perfectly.

As with all other critical overdubs, it’s important to hear both the track and vocal more or less in the context of a final mix. I do prefer the overdub vocal(s) to ride above the track while working on it so that I can clearly hear the actual beginnings and endings of sung notes. In a lot of cases this mix works well for the singer’s cue mix.

Cue Mixing
While the control room’s monitor mix might fly for the singer’s headphone mix as is, it is wise to be able to turn up the singer’s vocal track even more in the phones. Call the “more me” control; if I’m running the cue post fader from the monitor mixer, I just turn up the vocal track more.

I’m also careful about vocal effects like reverb and delays too. I prefer none but that’s not always the case with the artist—whatever minimal amount you can get away with, the better.

The concern is loss of a pitch reference when there is too much “me.” This is the first place to check if the singer starts to get pitchy.

Playing to many “pitch ambiguous” instruments and noises is also counter-productive. Sure the mix sounds cool with all that stuff flying around but for the business at hand; I think your singer will stay in tune better hearing only sonorous, in tune tracking instruments such as pianos and well-intonated guitars.

Tracks with wide chorus or flanger effects and loud atonal sound effects tend to disturb the ear’s pitch recognition abilities.

Happy Vocal Recording!
This is the third in a series of articles on recording vocals. View part 1 and part 2.
Barry Rudolph is a veteran L.A.-based recording engineer as well as a noted writer on recording topics.

In The Studio: Gain Structure And Recording/Mixing Paths | ProSound News

Every process a sound goes through will change the tone of the sound, even if the process is a little bit of gain change…

by Bruce A. Miller
recordingThis article is provided by BAMaudioschool.com.
Signal path refers to the path that sound makes while being processed.

Recording signal paths include:

Sound Source > Capturing Device > Wire From Capturing Device To Console Channel Input > Channel Volume, EQ, Etc. > Channel Output To Recorder Track

Mixing signal paths include:

Recorder Track To Console Channel Input > Channel Processing, Volume, Pan, Etc. > Channel Output To Stereo Bus > Main Stereo Output Master Fader > Final Mix

In addition, mixing signal paths include:

Channel Output To Audio Bus Through Aux Sends > Aux Channel With The Send Feeding Into It > Processing > Aux Channel Output To Stereo Bus, Etc.

Gain (volume) refers to the way that the volume of a sound will increase and decrease as it goes through the different stages of a recording or mixing path.

Gain structure refers to the input and output levels of each stage of the path. While it’s possible to set gain/volume knobs to anything and at the very last knob turn things down if necessary, you’ll most likely have inefficient gain structure that can cause extra noise to be added in one stage, and even overloading in another.

Unity gain means that a path or even stage of the path has the same volume going out as it did coming in. Although setting a knob or fader at zero will usually provide unity gain, once you start to process a sound you end up changing the volume within that particular stage of the path. You’ll then most likely have to adjust a different stage of the path to compensate.

Many do not understand gain structure and end up overloading early stages, turning the sound down later and not knowing why their meter level is “in the green” but the sound is distorted.

Changing gain means decreasing it (which can be done passively but is usually done using electronics) or increasing it (using electronics to amplify it). Every process a sound goes through will change the tone of the sound, even if the process is a little bit of gain change.

Imagine pouring water from glass to glass in a long row of different sized glasses, and you only have the first glass of pure water and a hose with dirty water. As you pour water from the first glass into the second and from the second into the third (and so on), some of the glasses will change the overall amount of water that is being transferred.

One glass may be wider so the water poured in from a narrower glass would not be enough to fill it, and then hose water would be needed to top it off. This would be the same as turning up a fader or knob on a channel to make a quiet sound louder. The amount of what you’re working with has increased to a desirable amount, but the purity of what you’re working with has been diluted by the dirty water or sound changes of the audio components.

Another glass may be filled with rocks and so the water it cannot hold spills out, reducing the amount of water you have. This is the same as using a fader or knob to make a loud sound quieter. Note that the purity of what you have is not changed. There are audio components that reduce sound (using passive resistors) without changing the sound, unlike active amplifiers that increase sound and noise.

Thus the volume will change throughout the chain, and so will the purity at different stages. Of course you want your first glass (the first volume control the sound source hits) to be as full as possible so you have a better chance or retaining sound purity through to the end of the chain.

However, be careful. All stages have maximum allowable volumes at their input (and within their processing) that will result in distortion that will be passed down the chain. Imagine overloading your second glass, which adds red dye to the water. No volume changes down the chain will ever get rid of the red dye.

Big processing (even volume) means big changes in sound. With a good gain structure, there’s usually not much need to add volume in the chain, helping to keep sound as “pure” as possible. And it’s important to start with a good-sounding first stage, which is where most of the volume increasing should be done.

Finally, always look out for distortion at every stage of the audio chain. You’re better off dealing with the extra noise on a quiet clean sound than the distortion on a good volume sound.

Bruce A. Miller is a veteran recording engineer who operates an independent recording studio and the BAM Audio School website.