Tips for Controlling Vocal Sibilance - theproaudiofiles.com

Tips for Controlling Vocal Sibilance Posted on March 7th, 2012. Written by Rob Schlette.

Tips for Controlling Vocal Sibilance

Posted on March 7th, 2012. Written by Rob Schlette.

Vocal sibilance is an unpleasant tonal harshness that can happen during consonant syllables (like S, T, and Z), caused by disproportionate audio dynamics in upper midrange frequencies. Sibilance is often centered between 5kHz to 8kHz, but can occur well above that frequency range.
This problem is usually caused by the actual vocal formant, but can also be exaggerated by microphone placement and technique. This article will discuss some ways to control vocal sibilance, and keep the problem from becoming a musical distraction.

Sibilance at the Source (best read with sibilant whistle)

In phonetic terms, sibilance comes from a type of vocal formant called a fricative consonant. During these sorts of utterances, the airway (usually the mouth) is drastically constricted by two anatomical features, like the teeth, tongue, or palette.
This pressurization causes some amount of noise that forms the consonant sounds we would recognize from a phase like, “Sally sits sideways on the tennis trolley.” Sibilance is a very necessary feature of human speech, but when there’s (subjectively) too much noise created during these consonants, we get a very distracting harshness.
It isn’t really practical or productive to address micro-muscular vocal technique during a session, so your best bet to mitigate sibilance at the source is microphone selection and placement. Here are a few suggestions:

• Every vocalist is remarkably different, so don’t pre-suppose that anything you’ve tried before will or will not work again.
• Be sure to leave some space between your vocalist and the microphone. Twelve to eighteen inches would be a nice starting point.
• A pop filter won’t do anything to help with sibilance.
• Once you find a microphone and distance combination that helps, try angling the microphone downward 10 to 15 degrees to place the 0-degree axis toward the throat instead of the sibilant source. 

Audio Dynamics Processing

Vocal sibilance is a phenomenon of disproportionate dynamics within an isolated frequency range. In other words, it is a problem of too much loudness contrast within a small frequency range of a waveform that has a dynamic profile of its own.
‘De-essing’ is the classic compressor technique used to address vocal sibilance through processing. In fact, de-essing is just one example of many uses for compression that is conditioned on a limited frequency band, or a modified harmonic profile.

De-esser Signal Flow

Audio dynamics processors like compressors and expanders contain two signal paths:

1. The audio path, which is subject to conditional gain reduction and;
2. The sidechain or ‘key’ path, which the gain reduction is conditioned on.

In short, gain reduction happens (or not) in the audio path based on the interaction between the sidechain signal and the detector settings (i.e. threshold and time constants). By placing an EQ in the sidechain path, we can further condition gain reduction on user definable frequency conditions.
The de-esser technique typically uses a narrow peak EQ in the sidechain path to boost the most offensive sibilant frequencies. This EQ exaggerates the dynamic difference between the sibilant band and the rest of the vocal waveform, making it much easier to achieve gain reduction during those consonants (and only then).
A pre-configured de-esser may provide an interface as simple as a compressor threshold and the peak EQ center frequency. These often work just fine. For more detailed control, one could patch an EQ into the sidechain of a relatively fast compressor, or use any number of compressor plug-ins that provide detailed EQ in the sidechain path.
There are lots of great techniques based on this signal flow, so spend some time with it. Frankly, de-essing is the least of what you can do by adding frequency conditions to your gain reduction.

Other Precautions

When you’re recording a vocal performance that may have a sibilance problem, resist the urge to compress the signal in the channel path. Over-compression can exaggerate sibilance. Instead, try using a fader to level the vocal performance, or just record with an adequate amount of headroom.
The same applies to the mixing process. Once you’ve done your best to control vocal sibilance, try using a fader and automation to maintain a consistent vocal volume in the mix. If you simply must instantiate a compressor on every vocal track, keep the attack time slow (> 30ms), and the ratio low.
Finally, don’t listen too loudly when you mix. That’s good general advice, but quality control issues like sibilance highlight its importance. Try a control room volume of 78-83dB(C) SPL. You might be surprised how much detail you’re suddenly able to hear.

Source Post:
http://theproaudiofiles.com/vocal-sibilance/

Tips for Controlling Vocal Sibilance - theproaudiofiles.com

Recording: In The Studio: Tips For Mixing Toward Loudness - Pro Sound Web

Recording: In The Studio: Tips For Mixing Toward Loudness - Pro Sound Web

In The Studio: Tips For Mixing Toward Loudness
In order to deliver the most musically effective loudness, that goal must have been addressed in the mixing process

February 28, 2012, by Rob Schlette

studio
[Click to enlarge]
This article is provided by the Pro Audio Files.


Some people want their music really loud, and there’s nothing wrong with that.

If loudness is part of their aesthetic and the audience likes it, then I say let’s go for it.

In order to deliver the most musically effective loudness, that goal must have been addressed in the mixing process, but not as directly as you might think.

It’s important to remember that there are mix masters, and then there are replication or download masters. Your project isn’t finished until it has been mastered, so the relative loudness of a mix does not represent the final level of the project. Comparing the loudness of a mix master with a finished commercial CD is not particularly useful.

However, there are a lot of aspects of mixes that directly contribute to the eventual loudness of a finished master. So what should you be listening for while you’re mixing? Here’s an example scenario:

My client has brought me a set of five multi-track recordings to mix. The client is very concerned that her project should fit in with the latest release from “Artist X” as much as possible, including being equally loud.

Here are some things I would be sure to pay attention to while mixing her project:

1. The Loudest Instrument

What is the loudest instrument in Artist X’s mixes?

The answer is probably pretty consistent across the whole CD; and I’ll be sure to use a similar approach with my client’s project.

This may not seem like a pivotal factor, but the relative loudness relationships within a mix establish a lot about the eventual absolute volume of the mix (and the project). If one hip hop mix has a lot more vocal content than another, the relative loudness of the two mixes will be confused.

If I’m mixing in a drum-heavy genre, I’ll be careful to reference that primary balance benchmark. If my next project is a vocal-driven style, I’ll simply re-establish my benchmark. In either case, I’ve setup the balance relationships within my mixes so that they can directly compare with other albums in the presumed audience playlist.

2. Relative Loudness Relationships

What are the relative loudness relationships between kick, snare, bass, and lead vocal?

Before you start cranking around on piles of processing, set up these simple balance relationships. They go a long way toward establishing the fundamental structure of mixes in many musical genres.

Try starting by mimicking the balance between the lead vocal and the snare drum from your reference. Once that makes sense, add the kick at a level that is referenced from the snare drum. Finally, rough in the bass level relative to the kick drum.

Beyond just setting up the framework for balance within your mix, these balances establish a lot about how you will work with these instruments tonally. A loud vocal, for example, may get a totally different EQ treatment than one that is buried (however appropriately) in a sea of guitars.

Do some reference listening; you might be surprised.

3. Tonal Contrast

What is the brightest instrument in Artist X’s mixes?

Articulation is a big component of apparent loudness. Pay particular attention to tonal contrast between instruments to get the most impact out of any particular tone choice.

To put it simply, it doesn’t matter how bright the drums are; it matters how articulate they are compared with other instruments. A mix with built-in tonal contrasts can be more effectively managed in the mastering process.

4. Panning and Depth

Apparent dynamic range can have a lot to do with panning and depth.

Do Artist X’s mixes have a lot of subtle panning, or are they essentially 3-channel stereo? What is the contrast between the instruments that seem nearest to the listener and furthest away?

If your goal is blaring, ‘too loud,’ loudness, it’s important to note that these spatial contrasts survive aggressive mastering much better than subtle differences in level or tone do. It is not uncommon for even the most critical listeners to initially mistake spatial contrast for audio dynamics.

Mastering

These types of musically relevant aspects of mix structure will help you create consistent, engaging mixes that fit into a genre in a lot of fundamental ways. The mastering process can then more effectively finish preparing those mixes for their commercial audience, including addressing their market loudness.

Rob Schlette is chief mastering engineer and owner of Anthem Mastering (anthemmastering.com) in St. Louis, MO, which provides trusted specialized mastering services to music clients across North America.

Be sure to visit the Pro Audio Files for more great recording content.



Recording: In The Studio: Tips For Mixing Toward Loudness - Pro Sound Web

Recording: In The Studio: Tips For Mixing Toward Loudness - Pro Sound Web

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The Expensive Vocal Sound And Other Unicorns

The Expensive Vocal Sound And Other Unicorns

The Expensive Vocal Sound And Other Unicorns

Recording: Ensuring You’re Recording At The Proper Levels - Pro Sound Web

Recording: Ensuring You’re Recording At The Proper Levels - Pro Sound Web

Ensuring You’re Recording At The Proper Levels
You're probably tracking too hot. Here's why...

December 26, 2011, by John Scrip

This article is provided by
MASSIVE Mastering


I’m going to try to keep this very “fool proof”.

This was born out of the rantings of hundreds and hundreds of posts on a dozen or more audio forums exploding like a volcano recorded with lots of headroom.

I hope to instill a basic understanding of why certain trends and common beliefs are just plain bad.

And by the time you’re done reading, and perhaps doing a little experimentation based on this, you won’t need me to prove it. You’ll know it yourself.

Is this a “miracle cure” for bad recordings?

Normally, I’d say no.

But with the dozens and dozens - easily now into hundreds of e-mails, phone calls, letters, forum posts and other forms of communication I’ve received about how this advice has completely changed a persons view of recording recording, I figured this information is worth sharing with a wider audience.

The sad part is this should be common sense. To anyone that grew up “on tape” it probably is. To those brought up in 1’s and 0’s, it might not be so obvious.

So, if you’ve been struggling with recordings that sound “weak” or “small” or too dense or “just not ‘pro’ enough” then please, read on. If this is about you, you might think differently soon.

As a mastering engineer, I work on recordings from pretty much every level of experience. A few years ago, I noticed something unusual.

“Ultra rookie” recordings, that is those made by people with little or no experience, sounded fine. They didn’t know any better, so they didn’t have enough rope to hang themselves with.

Then, at the other end of the spectrum, “Pro” recordings sounded fine. They know what they’re doing and/or are using gear with obscene amounts of usable headroom (explained later).

The “middle of the road” engineers with a year or two - or much, much more experience—Those are the recordings that sounded “small” and spectrally challenged. So after quizzing these people over months and months, I came up with the following conclusion…

You’re probably recording too hot.

And it’s absolutely ruining recording after recording after recording. And it’s the simplest thing in the universe to correct.

I know, I know - “It says in the manual to record as hot as you can without clipping.” Well, I’m going to flat-out call that B.S. and I’m going to back it up with a simple (if not somewhat time-consuming) experiment.

Also as a mastering engineer, let’s get something straight - I don’t like the “loudness war” going on.

However, I’m as guilty as the next in contributing to it. I can’t fight it, as much as I try. Hopefully it’ll be over some day. Unfortunately, with the quest for loud, there are a lot of engineers out there shooting themselves in the foot before they even know how to aim.

They think that tracking loud and mixing loud contributes to a louder recording after the mastering phase. This is absolutely untrue and it’s generally the best way to make sure that your recording will not have the “loudness potential” of the average commercial release.

Clean recordings, or those made with low distortion and good spectral balance, are the ones that handle the “abuse” of the mastering phase with flying colors.

This article isn’t intended to give you some secret way of making louder recordings. But it will almost undoubtedly give you the ammo needed to make better recordings. And those better and cleaner recordings are the ones that can be louder recordings in the end.

First, let’s get through a little nomenclature,

dBFS
Deci-Bel (one tenth of a Bel) Full Scale, on the digital recording scale, -0dBFS is the hottest signal you can have.

“All ones.” Top of the scale, can’t get hotter, etc. Always “minus” as you can never go higher - So the reading will always be a specified amount below 0.

Line Level / 0dBVU
Just what it says. Line level. 0dBVU on an analog VU (volume unit) meter. Pro (+4dBu) or consumer (-10dBv) level, it’s line level.

We can also refer to this as RMS (Root Mean, Squared), or a level over a specific amount of time. You can go above or below 0dBVU.

It’s simply a nominal level to which basically everything audio is related to.

Headroom
The space between a nominal signal (in this case, line level) and the point where the circuit fails.

In digital, basically anything under full scale (-0dBFS) would be considered headroom.

In analog, it’s the space between 0dBVU and the point where the circuit clips (failing completely). In analog, there can be a big difference between “headroom” and “usable headroom.” We’ll get into that in a bit.

Steak
From the old Norse “steik” meaning “roast”. A slice of meat, typically beef, usually cut thick and across the muscle grain and served broiled or fried (thank you, Wikipedia).

So…

You have a microphone and a preamp going into a converter or sound card. Those converters are calibrated at line level.

In most cases, over the last several years, most I’ve seen are calibrated to -18dBFS = line level (or 0dBVU).

In other words, if you run a steady signal (a sustained note on a keyboard for instance) through a preamp and turn up the preamp gain until the VU meter reads 0dBVU, at the converter, it will read -18dBFS (or -18dBFS(RMS) - full scale, but measured over time).

This is where your gear is designed to run. This is where it’s spec’d at. You will have a decent amount of headroom, the lowest distortion, the best signal to noise ratio, etc., etc., etc. around this level or lower.

Some gear - usually very high-quality stuff, has a good amount of usable headroom above this level.

A lot of “budget friendly” gear does not. So all of this advice is more important if you’re using “okay” gear at the input. Even your digital converters are analog components up to the converter itself. They don’t want to be “beat up” all the time either.

Let’s look into headroom. Above that 0dBVU/-18dBFS range, digital headroom is simple - perfect, perfect, perfect, perfect, CLIP.

The signal is “what it’s supposed to be” until the point of failure. Analog gear (your preamp, compressors, outboard signal processors, etc.) isn’t like that…

It’s more like “Perfect, a little noisy, “tight” sounding, spectrally distorted, CLIP.

The converter’s job is simple - reproduce the signal it’s fed digitally, whether that signal is clean and dynamic or distorted and squishy.

The analog chain’s job is anything but - typically, you’re adding 20, 30, maybe 50dB of gain to the incoming signal.

The preamp is working - not just “passing” the signal.

And that signal can start to suffer from noise, distortion and dynamically dependent (varying along with volume) spectral imbalance (a skewing of the overall spectrum from an EQ standpoint).

In other words, a nice, thick, chunky guitar tone (for example) might have different characteristics depending on how hot the signal is.

The highs might be open and airy and then the signal gets loud for whatever reason and the highs either get swallowed up, or perhaps get very harsh and strident.

In any case, it’s an inconsistency that isn’t’ there when the levels are more “normal.”

Even though the analog gear probably has spec’d headroom well above digital’s full-scale, it doesn’t mean that signal actually has the integrity it should up to that level.

So what happens is simple, a signal is recorded that’s too hot (usually to “use all the bits” which again, is a bunch of BS).

It overdrives the input chain not unlike a guitar preamp overdriving a Marshall stack (well, not that much, but the premise is the same).

Now, after all the other tracks are recorded, all of them need to be attenuated by 12, maybe 15dB or more so the mix doesn’t clip. Those distorted, spectrally questionable, squishy, noisy tracks all get turned down.

Are you seeing my point yet?

When you take a steak and cook it until it’s burnt, it’s burnt.

If you pour ice cubes all over it, it doesn’t make it more rare - it makes it a cold, wet, burnt steak. No matter what you do, it’s still burnt. Just like if you record too hot.

But if you cook a steak a little too rare, you can always heat it up a bit later.

You can microwave it without it turning into leather. You can pan-fry it for a few minutes and it’s still a tasty, savory piece of steak.

When you use up all your headroom right away, you don’t get it back by turning it down.

It’s gone forever. Sure, you can increase mix headroom or the headroom at the buss - but it’s not going to make the track less distorted or fix the skewed S/N ratio.

Here’s Your Experiment
You’ll need a few Y-cables (let’s not get into the technical aspects of splitting a mic signal - It’s an experiment) and at least one stereo (2-channel) preamp.

Record a song using as many tracks as you feel fit. The more, the more apparent. You’re going to split the mic signal and record each twice simultaneously.

On one channel of the preamp, set the gain so it peaks between -18 and -12dBFS at the converters and record them to odd numbered channels.

On the other, set it as high as possible without clipping and record them to even numbered channels. Record some guitars, drums, maybe piano, of course some vocals, keyboards, go nuts.

Set all the odd numbered (“normal” level) channels to unity and toss up a rough mix to a stereo buss - which should be a piece o’ cake.

Switch over to the even numbered channels and figure out how much you’re going to have to attenuate them all so the main buss isn’t clipping constantly.

It might be a lot. Could be a 10-15dB cut on all channels before you can even think of starting to do anything else. Send those to a stereo buss. Solo the busses, one at a time, and try to match the levels between the mixes.

You’ll probably immediately notice that the “normal” mix is much more open, dynamic, airy, clear, clean, with much more “sonic space” between the instruments than the “hot” mix.

Now, add a limiter on the main buss. Run the “hot” mix into it and bring the level up until it starts to obviously distort and fall apart sonically.

Then switch over to the “normal” mix - which should now be “rammed” by the same amount. If your experience is pretty much like everyone else’s, the “normal” mix is still much more open, airy and dynamic with less distortion and more “crankability” than the other.

The “Dumbed Down” Version
Stop recording so hot. Instead of trying to get your tracks to peak at -2dBFS, have them peak between -20 and -12dBFS and your recordings will almost undoubtedly sound better.

Mixing will be easier. EQ will be more effective. Compression will be smoother, more manageable and predictable. You’re in the age of 24-bit digital recording - relax and enjoy the headroom.

Even if your only concern is the volume of the finished product (which would be a shame, but it happens), recordings made with a good amount of headroom are almost undoubtedly better suited to handle the “abuse” of excessive dynamics control.

Quieter recordings have more potential to be loud later. It’s because they’re usually better sounding recordings in the first place.

John Scrip is the Owner and Engineer at MASSIVE Mastering.

Recording: Ensuring You’re Recording At The Proper Levels - Pro Sound Web

How to Get the Best Performance From Your Vocalist | Audio Issues

This is a guest post by Kyle M. Bagley. If you want to submit a guest post, please see the guidelines in the author box below.

Getting a great recording from musicians often takes more than just choosing the right microphone or preamp. In the end, it is a great performance that will define the song and its impact.

Engineers use a series of simple techniques to get a great recording, making the singer most comfortable and using psychology on top of their engineering knowledge to craft the sound they’re looking for.
Who’s In The Room?

The first and most important consideration is who will be present for the recording. In general, singers are more comfortable with fewer people in the room, which can be a departure from their bandmates who may like to record in large groups.

Other factors, including what the singer hears in their headphones and the vibe of the room will greatly affect the performance’s energy level. It is your job to shape this energy to fit the mood of the song for the perfect take.
Lights

Lowering the lights in the room or booth will generally quiet a vocalist and bring the energy level down. Some musicians prefer this lighting, as it is easier on the eyes. However, be aware that for long sessions, a dark room might also make your singer tired.

Lamps give the singer ultimate control, especially if they can be pointed and/or moved to enable music and lyric reading without making anyone uncomfortable. They are also a solution for buzzing overhead fluorescent light fixtures. Singers do not usually know what is best, or how the light is affecting them, so have it set when they come in.

Headphone Mix

Creating a good mix for the singer to listen to while they track is crucial. Chordal rhythm instruments are the most important thing to hear. They should be panned center and be heard clearly of the rest of the rhythm. This keeps singers in tune and on pitch throughout a take.

Other rhythm instruments are only necessary to give your singer a sense of what they are used to hearing, for the balance of the band and to help them keep the form. Other melody instruments can either act as a crutch for your singer or be a source of confusion. Add them in selectively.

The overall headphone volume can have a similar effect on your singer to the lighting. Loud headphones will make them sing louder over the music, which can add a boost, but sometimes at the cost of a straining voice or reduced stamina. Always check to see if your headphones are bleeding too much into the microphone.

If the volume is not loud enough, many vocalists will end up singing flat, “under” what they hear. Again, these are not conscious changes to the singer, and it is your job to make the headphones just right for your singer.
First Timers

Singers going into the studio for the first time often have a hard time adjusting from how they sing at live performances. The stage and the studio have a completely different approach, and your singer may need time to adjust when he arrives.

Microphone technique should be discussed before you begin. Give tips on how close they should be to the mic and where to aim their voice.

I have engineered few sessions where a singer complained of a “delay effect” in the headphones. This is often not a software issue, but a discomfort that the singer has with hearing himself live in the headphones. If he still has trouble after a few takes, try having him sing with only one earpiece. That way he’ll hear himself in the room with the open ear alongside his headphone mix.
Tips For A Smooth Session

Just like every musician, singers usually need time to warm up before they are recording their strongest performances in the studio. Arranging a “set list” is usually a good idea. Start with the easiest tune, and once you know the singer is warm, go after the hardest ones. Songs with a wide range or complicated arrangement won’t get their full attention at the end of a long session.

Even if the singer knows all the words to a song, printing the lyrics for both the performer and the engineer can be beneficial. It helps them keep their place while tracking, and makes finding overdub spots go faster.

When starting on a new song, always always always track the first take! You can scrap it later if nothing comes out, but singers often exhibit raw energy and emotion that is lost as they go over the song multiple times.

Lastly, the most important thing you must do while engineering a session is to respond to the singer. Listen for problems and strengths, make changes if things aren’t working, and make sure they are comfortable throughout the session.

Kyle M. Bagley attended Berklee College of Music, and currently works as a musician and studio owner. See more of his work at http://www.kylembagley.com

Source Article:
How to Get the Best Performance From Your Vocalist | Audio Issues

How to Get the Best Performance From Your Vocalist | Audio Issues

- Eq-ing Vox : Recording Magazine -

Eq-ing Vox
Approaches to eq-ing vocals...
By Geno Porfido

By definition, the vocals are the focal point of a ‘vocal’ song. But getting a good vocal mix and a good vocal sound that cuts through isn’t always easy—especially when a lot of electronic sounds are being used. So here are a few quick eq tips and techniques to help achieve a better vocal mix

As a preface, remember the garbage in/garbage out rule: it all starts with a good recording of good vocals. But we’re not here to coach your singer or nag him or her to cut down on the cigarettes, booze, and so on. Nor are we discussing mic and mic preamp selection, compression, reverb, or delay. This is just about eq.

Beyond the performance

Unless you’re after an extreme vocal effect such as a supremely squashed vocal, rough edge/low tech sound, or that long-distance telephone sound, judicious use of eq can enhance your tracks tremendously. As a starting point, my preference is to start with a slight rolloff around 125 to 240 Hz, a few dB removed at 2.5 to 3.5 kHz to cut piercing frequencies, and a boost at 10 to 15 or 16 kHz to bring out the air.

All of the usual variables (singer, mic, etc.) will ultimately guide exactly what eq you use, but here’s the rationale for that starting point. Today’s trend is toward a very crisp lead vocal, crystal clear on top with nice unobtrusive bottom.

When trying to keep lead vocals proportionate to the band mix, for starters I roll off some bottom with a highpass filter. Some mics and most consoles have this built in; common frequencies range from 60 Hz to 200 Hz, depending on design or manufacturer, with varying degrees of sloped attenuation (example: 6 dB per octave rolloff, from 100 cycles down).

This filtering keeps excessive bass out of the vocal track and helps it fit in its proper frequency spectrum to make room for accompanying instruments. It also helps keep plosives (popped Ps and other hard consonants) from being obvious; that and a pop filter should eliminate them altogether. Most human voices don’t usually have a lot to offer below around 160 cycles anyway, so you’re not losing much by cutting in that range.

Next, the reason for removing 4 to 6 dB at approximately 125 to 240 Hz is to keep the vocal out of the upper bass drum/bass guitar/fat drums area. You always want to be aware that cutting too much out can result in a thin vocal, so take it slowly and remove a little at a time.

Again, the exact frequencies and cut amount will vary from singer to singer, even throughout a song. On some mixes a totally flat vocal or one that’s only been eq-ed slightly on input has worked as the main track. But there may be a section that changes drastically and the singer drops down or jumps up an octave for a few lines, requiring additional eq to match the rest of the song.

There are several ways to make eq changes on the fly that way, depending mainly on the type of equipment you’re using. With an analog board, the usual method is to mult (“Y”) the vocal track to two console channels, each with the different eq (and often volume fader) setting. Then you simply mute the appropriate channel while unmuting the other, or crossfade them.

If you run out of channels, it’s also feasible to insert outboard equalizers with the alternate settings. Inserts can easily be switched in and out from either the channel insert switch or the eq unit’s bypass control.

All this is a lot easier if you’re using an automated digital console, because you can automate the eq changes—it’s not necessary to mult the signal to two channels. Same with digital audio workstation software, which is how most of you will be working these days.

Regardless of the equipment, you want to pick a clear place in the music (if there’s no rest, the spot where the singer takes a breath is the most likely candidate) to make the change so there’s no pop or click. This can be done numerous times throughout the song as needed.

Top shelf

Whether you’re eq-ing vox or instruments, the goal is to help the part find its space in the tonal spectrum and try not to step on it with too much other information. The top end poses its own series of challenges.

Some recordings just have this incredibly bright vocal that’s still smooth and warm without being harsh. Others have this great edge, almost like a cliff, that pokes the vocal through even the thickest mix, yet sounds totally natural.

There are many ways to eq a track, but I like to approach it from two directions. One is a smooth shaping curve using a wide bandwidth (a.k.a. “Q”—the lower the Q number, the wider the bandwidth), very evenly boosted across the top. The other is to boost one or more frequencies in a tight bandwidth so the edge is purposeful and creates a bite.

For the wide bandwidth boost I generally use a shelving eq to add a few dB very high, somewhere between 7 and 12 kHz. Shelving eq starts at the selected frequency and continues out to the end of the spectrum, so if you ad d 3 dB at 10 kHz, from 10 kHz out to 20 and beyond the eq will be even in gain. This creates a nice, smooth, almost ethereal top end that brings the vocals up into an area where there’s less competition from other instruments.

Hopefully the whole top end is very smooth with no frequencies sticking out. Note that we’re safely above the sibilance (“ess” sounds) frequencies, which are usually around 5–8 kHz. Sibilance is a separate problem for another article, but once in a while you can just cut specific frequencies on singers and tame slight sibilance without having to use a de-esser (or de-essing technique). Usually the eq approach to this will bring too many side-effects, but it’s worth mentioning while we’re discussing eq.

The second high-end eq approach is to zero in on a couple of key frequencies, boosting or dipping them a few dB with a peaking eq. Peak eq centers around the selected frequency and slopes off on either side depending on what bandwidth you select. You can create an edge that just sounds incredible if you find the right places to boost. Unfortunately there’s no magic number here—you just have to listen to the vocal in context and find the spots. Sometimes the mic has an inherent boost, so you may not even need to do this.

Life and larger

Male and female vocals present different problems. In most cases I’ll leave the bottom untouched for female vocals unless they really need a corrective boost or cut to deal with a tonal problem.

With male vocals, some engineers like to get a larger, fuller sound. That’s okay too, but I’d still recommend a rolloff at least from 80 or 100 and below. Keep in mind that the big fat vocal track can eat up almost everything in its path, from kick drum to cymbals, and it may be harder to pull into the mix than a vocal that’s been eq-ed to fit a narrower frequency spectrum.

If you like to mix vocals first and then balance the band in around the vocal, a big fat sound could be the one for you. My preference is to go for fairly natural instrument sounds, though, and a singer has nowhere near the bottom end of a bass guitar or chunky Marshall.

When you’re dealing with a superstar vocal or an act whose focus is the vocal and nothing else, then a much larger-than-life sound is appropriate. With this sound the lead vocal is huge and full-bodied with lots of warm bass and clean high end, so it really stands above everything.

Some of the better tube mics available today will help fatten your track naturally, as will classics like a Neumann U87 or old U47. But if a flat recording yields a thin vocal take, you can try adding a few dB around 160 up to 300 Hz to thicken it up. Rolling off some upper mids may also be a good course of action, depending on whether you hear the problem as being a lack of certain frequencies or too much in one area.

Cutting and boosting both work, but remember to adjust the gain accordingly. If you boost 6 dB at 1 kHz on the track, for example, you may need to pull the track down to compensate and avoid distortion.

The best way to develop these skills is simply to listen to a lot of recordings and try to emulate the sounds you like (and avoid the ones you don’t like). Just remember to go slowly with eq—a little bit goes a lot farther than one tends to think.

Geno Porfido (talkback@recordingmag.com) is currently living and recording in San Francisco. He’s looking for projects to produce in the Bay Area, as long as they pay ridiculous sums of money or are located near a decent pizzeria.



Full Article Source:
- Eq-ing Vox : Recording Magazine -

- Eq-ing Vox : Recording Magazine -

KEYBOARDS AND COMPRESSOR/LIMITERS

Keyboard Magazine "In The Studio"

Article # 10

KEYBOARDS AND COMPRESSOR/LIMITERS

In this months column I'd like to discuss the use of compressor/limiters with keyboards.

Compression was introduced in recording to help the recording engineer cope with the problems recording live music. Because most symphony orchestras could well exceed 100+ db and tape could only handle around 75 db of dynamic range, an engineer had to be awfully quick at the faders ( his keyboard ) or be lucky enough to have a compressor and know how to use it.

It should be easy to understand what a compressor is being that it is a VCA. For the VCA in a compressor to react, a certain level has to be reached. This is called the threshold. When the input level of a compressor exceeds the threshold set, the compressor's VCA starts to reduce the input level that has exceeded that threshold. The amount the signal is decreased is determined by the ratio. If an input exceeds the threshold set by + 10 db, and the ratio is 2 : 1, the compressor's VCA will restrict the input level to + 5 db above the threshold set.

Most compressor limiters have at least three basic controls. They are the threshold adjust, the compression ratio amount, and the output gain. To learn how to get the best out of a comp/limiter we'll start with the threshold adjust. If for instance you would only like to trim off the peaks from your synthesizer that feed into your stage amplifier, you would play your loudest passages and then set the threshold to react to those levels. Next check the softer passages. Be sure the threshold indicators not being triggered. If it is flickering, you're threshold is set too low and you will be compressing the whole program. The next thing to decide upon is the amount of compression ratio. Notice that is the ratio is set to 1 : 1 nothing audible is happening to the program. If you turn the ratio up to 2 : 1 and the input is above the threshold you should hear the compression working slightly. The purpose of the ratio is the difference between compression and limiting. Lower ratio 's from 2 : 1 to 6 : 1 are considered compression and higher ratios above 10 : 1 are considered limiting. A good definition of a compressor vs. a limiter is as follows :

A compressor is a VCA amplifier whose gain decreases as its input level increases.

A limiter is a compressor whose output level stays the same, regardless of its input level.

The art of knowing when to use compression vs. limiting is something of a matter of taste. Acoustic piano and very percussive synth sounds sound excellent with their sometimes harsh peaks removed. But beware, if you set the threshold to low, you can remove all the percussive qualities from the initial attack. The better comp/limiters have adjustable attack and release times which are invaluable for situations mentioned above. If you are not fortunate enough to have adjustable attack and release on you unit, you're just going to have experiment with different threshold settings and ratios.

One of the biggest mistakes made concerning compression is having the threshold set too low! Although this can produce a very unique sound it can also add much unwanted noise. Lets take a look how to avoid this. First the reason we are using compression is because we want to smooth an instrument out. This will will make it more enjoyable to the listeners and allow for is volume to be louder. This can be great for a solo synth part. It has to be loud to be dramatic, but it shouldn't be offensive. Instead of just limiting the peaks we want to control the solo synth's overall dynamic range into a narrow amount of loudness. To do this we use a ratio around 2 : 1 to 4 : 1 and reduce the threshold so that almost all the notes played are above the threshold. But here's where the trouble can start. If that instrument's output is noisy, we can going to amplify that noise into the same dynamic range. So you have to be careful not to set the threshold so low that it picks up the noise. If a noise gate is placed in front of the comp/limiter lower thresholds can be obtained. I intend to cover noise gates in a future column.

Now the last control that we haven't covered yet is the output gain adjust. This is a most important function. If you have been reducing your input signal's output, it has to be brought up in level again. After you've smoothed out the peaks, you're going to have to raise the output amplifier to bring that compressed signal back into the mix. Be careful because once again this stage can add noise to your signal if you abuse it! If you don't have enough range to boost the signal, go back and check your input threshold. Try boosting the volume of the instrument you're compressing and then re-setting the threshold and output levels again. Too low of a level will cause noise. Whenever you have to amplify a signal excessively you are going to add noise in the process.

One of the side effects of using a compressor/limiter is called pumping or breathing. These effects can become audibly obtrusive especially when using higher compression ratios. Remember the higher the ratio the more the gain change. When a signal passes the threshold level the compressor turns on. Its VCA goes to work and starts to reduce the signal by an amount equal to the ratio's setting. This action is called gain reduction. When the input signal falls below the threshold, the compressor's VCA starts reducing the amount of gain reduction. With high input levels and high compression ratios the amount of gain reduction could easily be around + 10 db. If the gain reduction changes rapidly, a pumping or breathing effect would take place. On the other hand if the compressor you're working with has a slow release time, ( the time it takes for the compressor to reset itself ) the notes that follow and are under the threshold level will also be reduced in gain. The sooner you get to know your unit's capabilities, you can learn how to set the threshold differently to overcome pumping and breathing sounds. A most unusual effect can be created with a compressor with a very fast attack time and a slow release. This back-wards sounding effect is most noticeable on a drum set, especially on the cymbls.

One of the best uses of compression is on synthesizer bass parts. This is probably the best instrument to work with to get to learn the effects of compression. In the studio it is common practice to record the bass guitar or bass synthesizer with compression on it. Because of the low frequencies and the percussiveness of the higher frequencies, the bass is a perfect example. Experiment with high and low ratio amounts. Listen to the difference it makes on the low vs. the higher frequencies. Try also different threshold levels. Notice when the bass line really jumps out at you where you've set the threshold and ratio amounts and you will begin to understand why. Listen also for when the bass part sounds least percussive and take note of the threshold and ratio settings. It will begin to become very clear after a short while of experimentation, of how the threshold and ratio interact with each other. If you have a compressor with adjustable attack and release settings, try a slow attack. This can even bring those lines out more.

Where is the best place to insert a compressor/limiter into the signal path? Well if you only have one unit it might be best to use it for all your keyboards in a live situation. By inserting it between your mixer and amplifier is the most common. If you only need compression or peak limiting on only one or two instruments, try using one of the effects sends. If your console has more than one bus, assign all the keyboards that you wish to be compressed to that bus. If you have a stereo unit, you can have a stage setting and give your sound man a separate compression setting.

All in all, not only will this make your soundman's job easier but your overall sound will be smoother.

Article 10

KEYBOARDS AND COMPRESSOR/LIMITERS

The Signal Flow in a Basic recording studio

The studio tweak lives in a world of electrons, and is able to follow them through thick and thin. From the big, fat juicy voltages that make up analog signals to the tiny, rapid-fire ones and zeros of digital audio and MIDI, you need to learn to follow the path of the flow. In the good ol' analog days this was easy as there was always a cable connecting things on the outside and a signal trace inside the gear. Now it gets a little more complicated as the sequencer has virtual pathways through software mixers, effects, synths and processors. But in the end, it once again goes analog and goes to the speakers. The charts below are the basic building blocks. Inside each block there may be hundreds, even thousands of switches, but you need to get the core basic flow down first.

Full Article:
The Signal Flow in a Basic recording studio

The Signal Flow in a Basic recording studio

5 Stereo Mic Techniques for Acoustic Guitar (or ANY acoustic instrument)

5 Stereo Mic Techniques for Acoustic Guitar (or ANY acoustic instrument)


If you record acoustic guitar, then chances are you’ve either played around with stereo recording or at least thought about it. But maybe you’re not sure which techniques to use and the pros and cons of each.

I think I can help.

5 Stereo Mic Techniques for Acoustic Guitar (or ANY acoustic instrument)

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Home Recording Studio Feng Shui | Audio Issues

Home Recording Studio Feng Shui | Audio Issues

Home Recording Studio Feng Shui | Audio Issues

Shure Americas | How-To | Vocal Miking Problems: How To Solve Them

Shure Americas | How-To | Vocal Miking Problems: How To Solve Them

Miking Problems
How To Solve Them

The following article is an excerpt from Shure Notes, Issue #23.

Bill Gibson on Vocal Miking

Bill Gibson has spent the last 25 years writing, recording, producing and teaching music. He is well known for his production, performance and teaching skills.

In the early '80s, Bill began teaching recording classes at one of the pioneer studios in the Seattle area: Holden, Hamilton & Roberts Recording Studios. Ten years later, Bill was part of a team that produced one of the first instructional videos on the topic of audio recording.

Today, Bill is President of Northwest Recording and has authored over 20 books.

Does the directional characteristic make a difference in the sound of the mic?

Absolutely! Mics that hear equally from all directions (omnidirectional) have an open and natural sound. They’re used frequently in the studio any time the engineer wants to include the sound of the room in the recording.

The danger in using an omnidirectional mic is that any room sound (ambience) that is recorded is there to stay. For most people it’s safer to record with a more directional microphone (cardioid or hypercardioid) and add any ambient sound artificially during mixdown.

Proximity effect is the cause of the boomy, bass-heavy sound we hear when a voice or instrument is close to the mic (within less than a couple inches or so). Omnidirectional mics don’t suffer from the proximity effect like cardioid and hypercardioid mics do. Therefore, in the studio we often use an omnidirectional mic to record a lead vocal at close distance. Using this technique, we can capture a clean and open sound without the overwhelming low end that’s caused by the proximity effect.

The KSM44 is an excellent mic for studio vocals because the selectable pickup patterns let the engineer choose the texture, tone, and feel of the vocal track by simply changing between pickup patterns.

In a live setting, omnidirectional mics are the most prone to feedback. They don’t reject sound from any directional and are inappropriate for most applications. However, mics with a hypercardioid directional characteristic offer a bit of a compromise—they’re a little less boomy, or thick-sounding at close range than true cardioid mics. Their pickup pattern contains an area of sensitivity directly behind the mic, which affects the character of the sound as it is received at the mic capsule. In most cases, the vocal sound is a little more transparent and open-sounding when using a mic with a hypercardioid characteristic compared to a true cardioid characteristic.

In a live application, keep in mind that floor and stand monitor positions are usually different depending on the mic choice. When using a cardioid pattern, there is usually less feedback with the monitor directly in front of the vocalist. When using a hypercardioid pattern, the monitor should be placed slightly to one side or the other in front of the vocalist for minimal feedback. If you look at the polar response graph for the specific mic, you’ll notice exactly where the mic is least sensitive—that’s the right spot for the monitor.

A mic like the KSM9 is a great choice for vocals in a live setting. It sounds like a studio condenser mic and it offers a pattern selection — cardioid and supercardioid. The flexibility provided by selectable patterns makes a mic that would be exceptional if it only had one pickup pattern, at least twice as great.

Keep in mind that every singer is different. If you have a choice of mics and directional characteristics, simply select the pattern that sounds best for the vocalist or choose the pattern that provides the best feedback rejection.

Is handling noise really an issue? Aren't all mics about the same?

Many studio mics aren’t designed to be handheld. They’re mounted in specially designed shock mounts that protect them from vibrations, bumps, and thumps. However, mics that are used in a live handheld environment must contain ample internal shock mounts and vibration control.

If you line up ten different mics on stands, you’re likely to notice dramatic differences in the sound caused by simply removing each mic from its clip. Some mics even rumble in normal handheld use. They don’t sound good although they don’t sound terrible — but the amount of handling noise they produce makes them completely unusable. Just shifting the mic in your hand causes a dramatic rumble—the sound of putting them in and out of the clip is unacceptable.

Mics that exhibit excessive handling noise also pick up excess amounts of noise from anything that moves on, or vibrates, the stage, such as footsteps, the kick drum, the bass cabinet, dancing, and so on.

One of the reasons for the popularity of the SM and Beta series mics from Shure is excellent design of their internal shock mounting systems and their minimal handling noise.

How close should the lead vocalist be to the mic?

I get this question frequently and it’s usually borne out of the frustration that the sound operator feels when working with a singer who has bad mic technique.

It’s common for the sound operator to tell the singers to just stay close to the mic (within less than an inch). That’s definitely not the best approach, but it puts control in the hands of the sound operator.

Vocalists must learn to move closer to the mic when they are quiet and farther away when they’re loud—the actual distances depend on exactly how quiet and how loud. In addition, speaking too close to the mic can decrease intelligibility and clarity. The overall volume of the house mix, the size of the audience and the acoustics in the are also considerations in mic technique.

Work with each singer to determine the mic technique that works the best for him or her.

Determine three ranges of mic distances for three separate purposes:
The "I'm singing background" distance—usually 1.5 to 3 inches (2 fingers to 4 fingers).
The "I'm singing a quiet, intimate lyric" distance—usually 1 inch or less (1 finger or less).
The "I'm really belting it out and I don't want to hurt someone's hearing" distance—usually 6 inches to arm's length.

Our lead singer gets lost in the mix and yet there are times when she's way too loud. How can I get a smooth and even vocal sound, like the sound I hear on professionally produced recordings?

Considering that your singer has good mic technique and you’re riding the vocal levels to help with global differences between levels for speaking and belting, the sound you’re looking for is probably a result of compression. A compressor is an automatic volume control that responds to the strength of the incoming signal. The sound operator sets a threshold level. When the signal strength exceeds that threshold, a built-in amplifying circuit—typically a VCA (voltage controlled amplifier)—turns the signal down.

A compressor is essentially an automatic sound operator. Like you, it turns the signal down when it’s too loud and then back up to where it started when it’s not too loud. The attack time, release time, and ratio controls let you determine whether the compressor acts like a Masserati or the giant in Jack and the Beanstalk. Compressors only turn the signal down—they don’t boost levels. However, the effect of compression is to enable the nuance in the vocals to be heard better. Because the loud parts are turned down, the entire channel can be turned up. The gain reduction meter indicates the amount of gain reduction. If it shows that the channel is being turned down by 6 dB at the loudest parts of the performance, then the entire channel can be boosted at the channel fader or at the “Makeup Gain/Output” control on the compressor. This results in the loud passages being the same volume as they would have been but the softer passages, vocal nuance, and emotional inflections have been turned up by 6 dB—they are, therefore, more audible to the audience.

Setting up a compressor is really pretty simple:
Set the ratio control to determine how extreme the action is—typically between 4:1 and 7:1 for vocals. If the ratio is x:1, for every x dB that exceeds the threshold, the VCA will only let 1 dB through.
Set the attack time—typically between 5 and 10 milliseconds.
Set the release time—typically about 1/2 second.
Adjust the threshold so that there are times when there is no gain reduction and times when there are about 6 dB of gain reduction.
Boost the Makeup Gain/Output control to makeup for the gain reduction.

Often, recordings are extremely compressed. In a live setting, be aware that if the compressor is reducing the gain substantially during a performance, once the performance is over, the VCA will let the signal return to its original level—this can easily cause massive feedback. The amount of compression you use in a live performance is dependent of the amount of gain before feedback in your system. In a live application, it’s usually best to compress by 6 dB or less.

Should I always buy a mic with a flat frequency response curve?

No. Part of the reason for differences in response curves is the intended application. If you use a mic with a flat frequency response on a live, close-miked vocalist, the sound will be thick and muddy because of the proximity effect. If you use a mic that’s designed for close-miking, for instance, a distant mic on an acoustic ensemble, the sound will be far too thin and weak.

The SM58® or Beta 58 have frequency response curves that roll off in the low end with a presence peak in the high end. This fact doesn't make it a lower quality mic than a mic like the KSM32 or KSM141 that exhibits a flat frequency response—it just makes them better suited to close miking than distant miking.

When a handheld vocal mic is close to the singer’s lips—within a few inches—the proximity effect rounds out the lows so they are essentially flat. Low frequency response is determined by mic distance. The built-in presence peak helps provide a clear and understandable vocal range. Notice that these presence peaks are typically between 4 and 7 kHz—strategically positioned in the range of vocal sibilance and intelligibility.

Mics with a flat frequency response curve are best suited to distant-miking applications in which the mic is a foot or more from the source, and yet a full sound is desired. Many condenser mics exhibit a very flat frequency response; however, they often provide a low frequency roll-off switch to compensate for the proximity effect when used in a close-miking application.

What’s the difference between miking a vocalist in a live performance and miking a vocalist in the studio?
The difference is really much less than it used to be before the KSM9.

In a live setting, we use vocal microphones designed for close-miking. They have historically been moving-coil mics because of their dependability, ruggedness, and simplicity; however, moving-coil mics don’t capture the fine transient detail as accurately as condenser mics.

In the studio we have historically used large–diaphragm condenser mics for vocals. Since the acoustics are controlled in a studio and leakage isn’t a consideration, most studio vocals are recorded from a distance of 6 to 12 inches. Sometimes, the singer moves closer, but the mic might be set to an omnidirectional configuration so the sound isn’t too thick and muddy or the low-frequency roll-off might be applied to compensate.

The vocal sound is adjusted by moving the mic across a much greater distance range than in a live setting. In addition, many professional studios have excellent acoustics—the sound of the room blends very well with the vocal to provide a desirable character and personality.

The KSM9 utilizes a studio-quality condenser capsule that provides the type detail that’s expected in a studio sound. It is housed in a body that feels good in the hand and the capsule sits in a well-designed shock mount system—it sounds great and rejects handling noises and vibrations very efficiently.

Shure Americas | How-To | Vocal Miking Problems: How To Solve Them

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5 tips for better vocal recordings

Audio Geek Zine » Blog Archive » 5 tips for better vocal recordings

5 tips for better vocal recordings