Audio Engineer Recompilations

Studio Acoustics and Soundproofing Basics

2011-05-01T12:15:00.000-07:00

Sonex 600 Acoustic Foam

The science of acoustics is something that tends to alternately baffle and intimidate most of us. Outside of a handful of highly trained individuals, the aspects of what makes a room sound a certain way is looked upon as a sort of black art. Performance venues and upscale recording studios routinely include acoustic designers in their construction budgets, spending considerable sums of money in pursuit of sonic perfection.

But for the average musician, budgeting for acoustic treatment has traditionally ranked well below the more tangible fun stuff like instruments, mics, recording gear, plug-ins, toys and more toys. Even if you’re at liberty to physically alter your space without incurring a landlord’s wrath, budgeting for two-by-fours, sheetrock and caulking doesn’t tend to hold the same appeal as that new channel strip plug-in or twelve-string you’ve been pining for.

Fortunately, the same technological revolution that has brought multitracking into spare bedrooms and one-car garages has also created low-cost solutions for many of the common acoustical issues facing the average project studio. In this month’s Studio Basics we’ll look at some ideas to smooth out your sonic nightmares.

Just Scratching the Surfaces

Let’s start off with a disclaimer: the purpose of this article is not to give you an education on acoustics. There are plenty of authoritative books on the subject, among them F. Alton Everest’s classic “How to Build a Small Budget Recording Studio from Scratch,” as well as a wealth of great articles and web posts. Rather, our goal here is to talk about some of the most common issues we encounter in our musical spaces, and some of the means available to address them.

That said, let’s divide the concept of acoustic treatment into some basic categories. There’s insulation, which usually entails keeping the sounds of the outside world out, or keeping your own sounds in. Closely related is isolation – the art of keeping individual sounds from bleeding too heavily into each other.

The other challenge is a bit more subtle, and has to do with how our rooms affect the sounds we’re creating in them. In any given space, the characteristics of that space have a direct effect on what we’re hearing. That’s why an instrument will sound different in a large hall than it will in a small club. It’s also the reason your mix sounds so different in your home studio than it does when you’re squirming in your chair in that A&R guy’s office.

The average home studio or rehearsal space rarely does well in addressing any of these issues. Most times we’re dealing with a spare bedroom, converted garage, basement or loft, none of which boast construction aspects that are in any way conducive to good sound. Thin, parallel walls, boxy shaped rooms, low ceilings and rattling window frames are only some of the enemies we face.

Even a few short years ago, the only way to address these issues involved massive amounts of money, materials and frustration. While the ultimate solution is still to plan and construct a purpose-built environment from the ground up, these days there are a number of ways to markedly improve your odds of making your workspace sound better without having to sell your instruments or smash your fingers.

Bass Traps

Soundproofing and Insulation

One of the most frustrating aspects of sound is that it will go where it wants to, and find its way through any space via any available path. That’s why it’s so important (and so difficult) to block any potential points where sound can leak through. In all cases, mass is your friend – the thicker and more dense your walls are, the better they’ll be at stopping sound.

Even more effective is mass combined with air. The most common construction technique is what’s known as a “floating room,” where an entirely new set of walls, floor and ceiling are built within the existing space, detached and separated by several inches from the outside walls (and, in the case of flooring, by rubberized “floaters” that lessen the transfer of vibrations). If you’re constructing your own space, there are companies that offer soundproofed doors and windows, as well as soundproof wall panels in pre-set or custom sizes.

Even if you don’t have the luxury of new construction, sealing areas of potential leakage in your existing structure will go a long way toward keeping the inside sounds in and outside out. For doors and window frames, look for the thickest, most dense weatherstripping that will fit in the allotted space. Use caulking to seal around areas like heating and air conditioning ducts, electrical outlet boxes, lighting fixtures, unfinished drywall joints and, if you’ve got them, tiled ceilings. While there are countless varieties of commercially available caulks and sealants, consider a latex sealant designed for acoustical applications.

You can also accomplish a lot by adding sound blocking layers to your existing walls. Several companies offer low-vibration materials which are exceptionally dense but surprisingly thin and lightweight.

If You Can’t Do the Whole Room…

For many of us, especially those who can eschew live drums, the toil and expense of insulating the entire room can be avoided by simply isolating only those elements that need it. In traditional studios, isolation booths have long been used to separate the vocalist or drummer during a live take. While these tend to be of the permanently-constructed variety, a number of companies offer various sizes of portable, lightweight “iso-booths” that can be assembled quickly and easily when and where you need them. Alternatively, you can search the web and find plans to build your own.

Another variation on the iso-booth that has become increasingly popular is the amplifier chamber. These can vary from small, soundproofed boxes just large enough to hold your guitar amp and a mic stand, to cabinets with speaker and mic (XLR) jack built in.

Your Biggest Fan

Sonex Computer Case

Your computer can be one of the biggest contributors of noise in your studio space. Particularly if your room is otherwise relatively quiet, the background hum of one or more computers can adorn your delicate acoustic tracks with all the ambience of a runway at Heathrow.

If you’re reasonably computer-savvy (or know someone who is), replacing your computer’s stock fan with a whisper-silent one is a quick way to reduce the noise. Another option is to look into sound-dampening cases with quiet cooling systems, which can knock off several decibels of noise, as well as cabinets that will completely enclose your computer’s CPU.

Semi-Isolation

In many cases, complete isolation is neither necessary nor desirable. As anyone who has ever recorded a live band will tell you, a little leakage can be a good thing, adding a natural sounding element that’s sometimes lost by separating things too much. Sometimes a bit of baffling between players and/or amps is all that’s necessary to provide enough separation for a decent recording.

This is typically accomplished with a gobo, a small portable wall panel around four or five feet tall. Many people build their own, sometimes covering one side with carpet or other absorbent material, the other with a reflective surface like parquet, and putting them on wheels for easy maneuvering. You can also find pre-manufactured versions of these, as well as transparent acrylic panels to surround the drummer but still allow for that all-important eye contact.

Fixing the Vibe

Let’s shift gears now and talk about the other major challenge in any studio: controlling the sonic characteristics of your space. Every acoustic environment’s sound is dictated by a number of factors, including the distance between walls, the height of the ceiling, the angles at which the walls meet and the materials comprising the surfaces, not to mention the composition and placement of tables, pictures and other surfaces, furniture, curtains, etc.

For the vast majority of us, our creative environments end up being places like basement rooms, garages or second bedrooms – typically smallish boxes with parallel walls. These types of spaces tend to encourage the buildup of standing waves, resonant frequencies and other sonic anomalies that can substantially color what we’re hearing, rarely for the better. The hard surface of a side or rear wall can create reflections that can significantly change the sound of your mix.

Step One – Identify the Problem

Many of today’s software programs offer tools to help identify some of the most common issues. Spectral analyzers, also known as Real Time Audio meters (RTA’s), are basically meters that break the sound down by various frequency groups, and can tell you a lot about what your room is (or isn’t) doing to your mix. By using a reasonably sensitive microphone in various spots throughout the room, an RTA can help to identify areas where there’s an excess buildup of certain frequencies. Some audio software applications have RTA’s built into the program. You can also get dedicated software or hardware units that can perform the same function.

One important caveat here: meters can be invaluable when used correctly, but meters don’t mix music – your ears do. Trust your ears first and foremost. Listen and compare, then use the meters to verify what you’re hearing.

Stop and Reflect

Generally, your best defense against unwanted reflections is to attack problem areas with a combination of absorption and diffusion. Absorptive materials prevent or greatly reduce reflection, while diffusers break up the reflection, scattering the waves in a multitude of different directions and greatly lessening their impact.

Bass Bin Trap

Much can be accomplished using common sense and everyday materials. The rear wall of my office/project room has a large, floor-to-ceiling bookshelf, fully stocked. Heavy carpeting and thick, theater-style curtains also work well, and you’d be surprised at the difference a strategically placed overstuffed sofa can make. But a number of commercial (and slightly less unwieldy) products are also available, including acoustic foams, fiberglass panels and blankets.

Also available are a number of diffuser products – geometrically-shaped panels and materials that, attached to your flat surfaces at strategic locations, can go a long way toward breaking up and eliminating reflections. And a number of companies offer products created of dense, uneven materials that will both absorb and diffuse sound waves, giving you the best of both worlds.

Bass traps, also known as barrel diffusers, are another popular means of addressing specific areas of your environment. Their typically cylindrical shape and uneven, absorptive finish work wonders to break up reflections in problem areas of your room. I’ve seen people construct these from plastic trash cans, though less inelegant versions are available commercially. Many companies offer bass traps that also perform as speaker stands, studio furniture, and even entire modular environments.

Conclusion

As I mentioned at the top of this article, the science of acoustics can be wide-ranging and confusing. While we know a lot about how sound behaves and what to expect out of a given space, there are always enough variables to keep it interesting. A new instrument, more bodies in the room, even changes in the weather….everything can influence the way things sound. What works for one situation may not be ideal for another, and the best we can do is to try and create as neutral and objective a listening environment as possible. Arm yourself with good monitors, meters and spectral analyzers, identify and correct obvious problem areas, and listen to as many different types of music, mixes and instruments as you can. But at the end of the day the most important tools you have are your ears – if it sounds good, it probably is good.

Source: http://www.uaudio.com

hypersmash.com

Noise Colours & Types

2011-04-28T07:59:00.000-07:00

Certain noises are described by their colour, for example, the term "white noise" is common in audio production and other situations. Some of these names are official and technical, others have more loose definitions. These terms generally refer to random noise which may contain a bias towards a certain range of frequencies.

Black Noise A term with numerous conflicting definitions, but most commonly refers to silence with occasional spikes.

Blue Noise Contains more energy as the frequency increases.

Brown Noise Mimics the signal noise produced by brownian motion.

Gray Noise Similar to white noise, but has been filtered to make the sound level appear constant at all frequencies to the human ear.

Green Noise An unofficial term which can mean the mid-frequencies of white noise, or the "background noise of the world".

Orange Noise An unofficial term describing noise which has been stripped of harmonious frequencies.

Pink Noise Contains an equal sound pressure level in each octave band. Energy decreases as frequency increases.

Purple Noise Contains more energy as the frequency increases.

Red Noise An oceanographic term which describes ambient underwater noise from distant sources. Also another name for brown noise.

White Noise Contains an equal amount of energy in all frequency bands.

Note: Some of these definitions refer to "all frequencies". This is only theoretical — in practice this means "all frequencies in a finite range".

Source: http://www.mediacollege.com

Best & most popular (DAW) Digital Audio Workstation Software of 2011

2011-04-27T11:31:00.000-07:00

This could not have been very difficult as you can simply ask this question in the top recording forums or even start a poll/survey. But potential problem could be that forum users can be paid by the software company to promote their products by answering polls and post in forums. Bear in mind those users in the home recording/audio forums are not true representative of the entire DAW user population so the result are not entirely accurate.

Therefore, to find out the reality aside from doing a survey/polls/asking a question is to get it from the most reliable data source – Google trends and searches tool. Google Inc. takes care in providing the most accurate data as possible. The results are also worldwide so it’s pretty a good representative of the entire DAW user population.

The first thing I did is to list all the known DAW commercial software available in the market. I came up with these lists:

1.) Ableton Live

2.) Acid Pro

3.) Adobe Audition

4.) Apple Garageband

5.) Apple Logic

6.) Cakewalk Sonar

7.) Cockos Reaper

8.) Cubase

9.) FL Studio

10.) Magix Samplitude

11.) Magix Sequoia

12.) Mixcraft

13.) Nuendo

14.) Pro Tools

15.) Propellerhead Reason

16.) Reaper Cockos

17.) Sony Sound Forge

The next thing is to get their search volume in Google using this tool:

https://adwords.google.com/select/KeywordToolExternal.

This shows how many users are actually looking for this DAW in Google search engine. This is a monthly figure and the higher this number, the more popular is the DAW. Below is the result:

It’s surprising and sometimes hard to believe that FL Studio is the most popular DAW based on popularity by search volume. It overtakes Cubase, Adobe Audition and Ableton Live in terms of popularity. Personally, I didn’t expect FL Studio to be this popular. I don’t know exactly the reason. Maybe it’s due to its price, features, ease of use and popularity among hip hop producers which of course one of most popular type of music genre today. I always thought either Cubase or Pro tools command the DAW popularity because they already been there in the business for some time already. And also take note that Pro tools have been regarded as the industry standard in DAW (http://bit.ly/hgZutw). The above data also shows that the top 5 DAW hold approximately 80% of what the users are looking for (see the cumulative column).

So what happens basically in the past 7 years? How did this came to happen? You can take a look at the details by using Google trends: http://www.google.com/trends. Let’s plot and analyze the trend of the top 5 performing DAW (FL Studio, Cubase, Adobe Audition, Pro tools and Ableton Live):

Based on the data it clearly reveals that in the year 2004 to 2009, Cubase holds the DAW overall popularity and is the choice for most users. FL Studio at the time (in 2004) is still in the bottom of top 5. Protools and Cubase did hold a significant share in the popularity in year 2004 to 2009. But things happen really slowly, FL Studio continuously becoming popular starting in the year 2005 until now (as shown by the increasing popularity trend.).

Adobe Audition and Ableton Live has similar share in the user’s popularity. Sad to take note that Cubase popularity went down significantly in the last 7 years and it was overtaken by FL Studio and Protools sometime in the year 2010.

Conclusion and Recommendations: Does being popular also means it’s the best? It’s not true at all times. But another question is why FL Studio became so very popular? Why did Cubase popularity went down significantly in the last 7 years only to be overtaken by FL Studio? Some might answer because the price is lower which means it is very affordable. Some might answer because it is relatively easy to use and some will say they have great documentation and manuals as well as community support. Some will also testify that the features are complete for the very low price they paid (best bang for your buck). Or some users might also answer that FL Studio is a very light program and take very little amount of system resources to operate. Does this imply that FL Studio is now the best DAW software? You decide. Wait; let’s see what will happen in the next couple of years.

Source: http://www.audiorecording.me

7 Common Recording Mistakes in Pro Home-based Music Production to Avoid

2011-04-26T08:14:00.000-07:00

MISTAKE #1: Using onboard sound card when recording music to your computer

Onboard soundcard has lot of limitations that could prevent you from creating high quality recordings. It is because they have very low signal to noise ratio it means that the noise created will be substantial over the recordings. The second primary reason is that onboard card will not allow you to record at highest sampling rate/bit depth as possible which is crucial for professional sound recordings. Most onboard cards only support 16-bit/44.1Khz or 48Khz which is not optimum or recommended. The last reason is that they have limited connectivity; onboard card is designed not for professional music productions but for other less audio-intensive apps like gaming and chatting. So if you need to record two instruments simultaneously, you just can’t. Much worse if you are tracking/recording drums :) Instead; invest in high quality audio interfaces such as Tascam US1641 USB 2.0 Audio and MIDI interface

In this case, you really do not need a soundcard or an outboard audio mixer. All you need is an audio interface and connect it to your computer using USB 2.0 technology. They accept several inputs and is ideal for recording several instruments at once which includes drums. These audio interface cost around $300 dollars, so if you are on the very tight budget and plans to use a soundcard. You can start with M-audio Audiophile 2496 which allows recording at 24-bit/96Khz format and only cost $95.

MISTAKE #2: Using Computer/Laptop multimedia speakers for monitoring audio.

These speakers are not designed for professional audio monitoring. They do not have flat frequency response. As a result, you won’t be able to monitor the details and assess the quality of your recordings objectively. Common multimedia speakers such as Creative, Altec, etc are designed for gaming applications and not suited for serious music production. One of my most favorite entry level professional studio monitor is Yamaha HS80M Studio Reference Monitor:

Reference monitors allows you to assess the quality of your recordings accurately because they have a flatter frequency response compared to speakers designed for other applications. These are “powered” studio monitors under $500 and they have exceptionally flat frequency response.

MISTAKE #3: Not doing pre-production or recording production plan

If you are aiming to produce the best sounding album as possible, crucial planning is needed. You need to examine what musical instruments or instrumentation is needed to be added to the song to make it sound great. Test things in advance before recording the tracks. In this case, do some pre-production runs, let the band perform and experiment with different arrangements to decide what is good or not.

Then you make a plan and write it on a paper. Sequence your multitrack project in advance, so you will decide how many guitar tracks you need to record. How many vocal takes, back up vocal is needed. Or whether you need to hire violinist to fit the song, etc. Once you have completed that solid plan, then start the recording sesssion.

MISTAKE #4: Recording and Mixing in UN-treated room acoustics

Your room that you are recording or mixing has a HUGE impact on the results of your music production. In this case, you need to treat your room properly so that it won’t unncesssary bounce sound waves that could bias your mixing/recording decisions. You can read this tutorial on mixing studio setup acoustic design. This is more in-depth and complete tutorial on home studio acoustics that basically covers everything you need to learn.

MISTAKE #5: Recording everything in stereo

Some tracks will only be highly necessary to be recorded in stereo (such as a solo instrument). In a multitrack project, everything should be recorded in 24bit/96Khz mono since these tracks will be mixed and then summed up into a two-channel (left and right) signal known as stereo mixdown.

The file sizes are also less compared to a stereo signal. You can read this post on the advantages of recording mono compared to stereo.

MISTAKE #6: Do not have a “trained” ear

If you are working in a studio both as an engineer or a producer, it is a requirement that you have “trained” ear. Your ear is the most powerful studio equipment. This means you can spot out of tune recordings easily, perceive minor changes in volume level, changes in tempo, pitch, noise, etc. There is no overnight success formula to have this asset. Instead you need to trained your ear on a continual basis so that you can sort out what sounds good and what sounds bad. In this case, you need to undergo ear training development exercises for recording/mixing engineers. Do not forget to monitor at reasonable level because consistent loud volume can damage your ears in the long run.

MISTAKE #7: Not recording in high resolution

A common newbie mistake is to record at 16bit/44.1Khz. This is not optimal since mixing and mastering needs digital audio sampled at much higher rate such as 24bit/96Khz for best results. It offers a much higher signal to noise ratio and your recording sounds cleaner and with depth.

Source: http://www.audiorecording.me

Tips in Mixing Electric Guitars using "Double Tracking" Technique

2011-04-25T09:21:00.000-07:00

One of the key elements in rock mix is thick and heavy guitar sound. One of the effective ways to accomplish this sound in the mixing process is through a technique called as “Double Tracking”. In this post I will illustrate how to double track guitars in the mix with the objective of making it heavy and thick.

Bear in mind there a lot of ways to thicken the guitar sound. Double tracking is one of the easier ways. Alternatively you can do:

Compression on guitars to make it sound thick.
Applying effects such as maximizer to increase loudness.
Parallel compression.

If the guitar sounds thin and weak, it will tend to affect the commercial appeal of the song especially if it is being marketed as a pure rock or alternative music. It is highly essential to mix things right but…

The following are the important requirement before you can double track the guitar in the mix:

Electric guitar photos

The recording of the guitar should be free of noise and normalize to the maximum volume.
If the guitar is recorded twice, it should also be clean and normalized. But it is not required to record it twice.
Record with the best distortion tone you need. Do not record it yet if you are not yet convinced of the distortion tone. Much better to experiment with a live band before starting to record the guitar. The overall purpose is to have a clean and final recording ready for mixing. Remember it is not advisable to fix the distortion tone in the mix; it makes the mixing process to be complicated.
Double check the tuning of the guitars, even slightly out of tune guitars can be problematic since if you double tracked it will tend to worsen the out of tune guitars.

It is also highly important particularly in the recent pop rock music trend to achieve not only thick guitar sound but it is also a wide guitar sound. This will achieve the “airy” sound of the distorted guitars.

So how do we start the mix?

Start with placing the 1st track in the Track one of the mixing session.
Place the other guitar track in the Track two of the mixing session. If you are recording only once, just copy and paste the wav file in the Track one to Track two.
Pan the Track one to -75 units (left). Depending on your recording software, this could be in %, for example if the maximum left pan setting is 100% so it will be 75/100 or 75%.
Pan the Track two to 75 units (right).
Now to get that wide thick sound, you can apply 5ms delay to one of the guitar (either left or right) (mix 100%)
To even make it heavier, do not anymore apply reverb on any of the tracks ( it is highly important that the reverb is from the room and amp based reverb that will be realized during the recording process). It is because if you start applying reverb on the guitar, it will tend to sound weak and far. Since you are mixing for rock, it is important to get the “in your face” guitar sound.
EQ it properly, do not cut too much bass in the distorted guitar, it will help add the heaviness sound.
Cut 1000Hz and 800 Hz on any guitar to make sound so clean and avoid the cracking sound.
Adjust the track one and track two volume and stop when it is loud enough for the guitar tracks to be heard, not dominating the vocals.
Cut 3000Hz with around -6dB and Q of 1.0 for both guitar tracks.
If your effects are arrange serially below are the sequence of effects that will be placed in each guitar :

a. Parametric Equalizer
b. Compressor
c. Reverb (optional) necessary only if the guitar tracks is too dry.
d. Delay (only on one track)

It is highly important to rely on your ears to do the settings. Do not believe in holy grail settings of compressor, EQ, they are there to serve as a guide and it is important to stick with the basic principles in double tracked mixing such as above.

Source: http://www.audiorecording.me

Using Pan, Volume and effects

2011-04-22T10:48:00.000-07:00

You have probably noticed on your mixer there is a "pan" control on nearly every channel. No, this does not refer to the frying pan the significant other menaced you with after your last trip to the gear store. Pan is short for "pan pot". And Pan Pot is short for Panoramic Potentiometer. (A potentiometer, by the way, is a fancy word for "knob".)

Panning is critical to the makeup of your stereo image. A stereo image has two basic perspectives, left to right and front to back. Pan pots control the left and right axis. Volume, reverb, delay, filtering and ambience create the front and back.

Simple Panning Tricks for Licks

In this day of totally staggering possibilities with plugins we often forget how powerful, and critical, the pan knob is to attaining an excellent stereo image. The main thing here is to keep instruments out of the way of each other so the listener can hear them clearly. Perhaps the most obvious example of this problem is with 2 electric guitars, particularly if distortion is used. Even using one will fill the audio bandwidth significantly, but two turns it quickly into a metal junkyard of cacophony. It will help immeasurably to pan these two so they are out of each other's way. Also make them take turns sometimes. But you'll see, if you try this, that panning about 30% will really help things.

The Image of the Band

Pretend you are in the audience. Where's the keyboard player? Always on the right. Or extreme left and right if there are two of them. The Drummer? Dead center. The Guitarists are usually at 10 and 2 o'clock, and the vocalist is dead center, in front of the drummer. Of course you have seen that a million and a half times. So set up your classic rock mix with that as a guide. Center the kick and snare, let the cymbals go a little to the side. If their is a conga player, put them on the left end. Use the pan controls to bring a focus to the perspective from the audience.

Front and Back

It may seem obvious, but we need to say it anyway. Instruments that we perceive to be closer are louder and have more of a direct, rather than reflected, sound. The elements of the mix that are important are up front and we hear them most clearly. Those in the back may have more early reflections infused into the main sound of the instrument. In your mix, you might create a reverb just for these early reflections that is separate from the main, hall reverb. Why is that? Consider being at a concert hall. The loud elements may bounce off the back wall and ceiling even though they are up front. Yet the softer instruments in the back may be imbued with reflections but very little of the sound energy may actually bounce off the back wall. Using 2 reverbs helps in this situation.

Creating a "longer" reverb.

An old trick is to first run signals through a digital delay, then to the reverb. We used to have to do this because digital reverb times were shorter than they are today, but the trick still works. In fact, it has been done on so many recordings that it is a bit of a standard. Its just the thing for ambient type soundcapes and may be used to mask imperfect vocal performances, as the delay tends to help mask off pitch notes.

Advanced Texture Mix Tip

Ever wonder why some mixes just jump out at you? It seems like the sound is deep and wide and almost 3 dimensional. There's a number of ways to achieve that, some good, some bad. The most dramatic is reversing the phase on one channel of a stereo mix. Sound just leaps out, but there is a problem. Sum to mono and the whole image disappears, what we know as phase cancellation. Another way to do this is with a combination of a delay and pan controls. You hard pan the mix left and right and add a tiny, infinitesimal delay to one channel. I mean really tiny or the mix will get lopsided. Our ears, conditioned by thousands of years warding off wild animals, can appreciate subtle shifts in the direction a sound comes from. As you add the delay, listen for the sound to "open up". It will if you do this right. Just another thing you can do with simple pan controls.

Panning the Orchestra

There is no absolute way to create a sonic image of an orchestra, but it does make sense to follow a classic seating chart which helps create a balanced, uniform sonic image. Note in the example below, how frequency ranges of the instruments (i.e., how bassy, mid range or treble-like the instruments are) tend to avoid conflict. The Bass Drum is far from the double basses. Also note how they reinforce each other. The Cellos and Violas can play one part distinctly on the right while the violins play a different part of the left. When they all play together there is a pleasing wash of sound, sometimes called a "pad" in electronic lingo. Note that the woodwinds, perhaps the most melodic of the orchestra, are centered. As you go to the right, the sound goes from soft to hard, from sweetness to bratty trumpets and tubby tubas. As you go left, it gets more delicate, with soft horns, piano or harp. In the back, you have your short and louds, like Piatti (cymbal) Snare, Bass Drum and Timpani. In the front, you have the long and softs, the strings.

To pan your MIDI orchestra, 0 should be far left and 127 is far right. You rarely want to set any instrument to an extreme value. For example, Harp, might be set to 20, French Horn to 40, Flute to 60, Oboe to 70 and double basses to 110. The Front strings might be at 40 and the Celli at 89. Don't read these numbers as absolutes, they are just an estimate. Every piece of gear sounds a little different. While all synths have 128 theoretical pan values, many of these values do not do anything to the sound. Some only change the actual sonic position every 3, 7, 15 values, some even 31 values. So experiment, move things around "a little" and hopefully the sounds will fall into their pocket.

Tascam Gigastudio 3 Orchestra Sampling Software (Windows)

Less is Often More

Effects should be used minimally. If a stereo effect is so great that you can no longer pinpoint the instrument, you used too much. Another tip here is doubling and detuning. You can make any instrument dramatically wide, yet centered, by putting the same instrument far right (127) and far left (0) and slightly detuning them by about 5-7 cents. This is a great technique for "wall of sound" like mixes that has strings that appear to "float" on the mix. Use it sparingly though, as hard panned doubles can easily take up sonic space where other instruments need to go.

Its good advice to work up a mix without any effects and apply them sparingly in the final stages. After your ears become accustomed to hearing the "in your face" mix, you will notice that as you add effects the mix will become darker, muddier, and less defined. Again that is a sign that you are going overboard.

A Final Point

What i have hoped to show in this article is simply that conservative settings often play a role in strengthening a mix. You rarely have to pan anything 100%, you rarely have to max any one fader or effects send out. Just little bits of signal going to alternate audio paths goes along way towards giving you a breathtaking sonic image.

Source: http://www.tweakheadz.com

Basic Music Mixing Panning Of Channels

2011-04-22T10:40:00.000-07:00

This article is as basic as it gets. It's for someone who has never used a mixer and panned channels. In later articles we'll show you how to use two or three lead vocal tracks, how to pan them, delay them, etc. This article is the bare basics.

In the stereo field you have a left speaker and a right speaker. Panning a channel puts that sound somewhere within that field. If you pan a channel hard left (L90), you will hear the sound playing only out of the left speaker. Pan a channel center (C0) and you'll hear the sound coming from directly between the two speakers, right in the center.

Typically in music, certain instruments and sounds consistently appear in the same areas of the stereo field. Technically, you could pan things anywhere. But your goal is to pan instruments and vocals in common recognizable areas that leave space for each other. You could pan every instrument in the song dead center, but if you did, you'd have a train-wreck of noise all on top of each other. Each instrument needs to have its own space in the stereo field (and in the frequency field).

Back in the day, the Beatles panned their vocals hard left and the drums hard right in some of their songs. That wouldn't work today (or back then either). While listening on an Ipod, no one would want to hear a guy singing only in their left ear for an entire song. The industry quickly scrapped that panning experiment.

Here Are Your Basic Panning Starting Points

Note: C=Center, L=Left, R=Right

Lead Vocal - C0 (double and triple vocals are panned in multiple areas)

Snare - L5, or C0, or R5

Kick Drum - C0

Hi Hats / Wood Hit, Clicks, Snaps, Etc. - Between L35 to R35

Cymbals - Between L10 to R10

Bass Guitar - Between L10 to R10

Lead Guitar - Could be anywhere, but usually "at least" 20L or 20R off of center.

Keyboards, Piano, Horns, Violins - Between L80 to R80 or stereo (L90 and R90). All depend on the song and the arrangement. Many times these instruments are stereo, but in a full mix sometimes the piano or a horn is only on one side, around L45 or R45. Also, different musical melodies are sometimes played. One violin melody could be playing on the left while a different one is being played on the right. This will be explained in detail in our future "stereo field" and "music arranging" articles.

I never pan anything L90 or R90 (I'll go L80 or R80) unless its a stereo track whose material doesn't reach the very outer edges. Anything that is panned this hard sometimes is exaggerated when using an Ipod. The sound could be annoying and hot in one ear.

The best way to learn where instruments are panned is to listen to commercial artists whose music style is similar to yours. Listen to different songs and take notes on where the instruments appear in the stereo field. This will at least give you some idea of what's going on.

Note: When multiple vocals or stereo instrument tracks are playing it could be hard, if not impossible, to tell exactly where they are panned. In future articles, we will explain in detail different advanced panning techniques that will help you quickly decipher what's going on in your favorite artist's songs. Which means you can emulate these commercial panning techniques.

Source: http://cdmusicmastering.com

Audio Expansion Basics

2011-04-19T18:04:00.000-07:00

Audio expansion means to expand the dynamic range of a signal. It is basically the opposite of audio compression.

Like compressors and limiters, an audio expander has an adjustable threshold and ratio. Whereas compression and limiting take effect whenever the signal goes above the threshold, expansion effects signal levels below the threshold.

Any signal below the threshold is expanded downwards by the specified ratio. For example, if the ratio is 2:1 and the signal drops 3dB below the threshold, the signal level will be reduced to 6dB below the threshold. The following graph illustrates two different expansion ratios — 2:1 and the more severe 10:1.

Input Level vs Output Level With Expansion

An extreme form of expander is the noise gate, in which lower signal levels are reduced severely or eliminated altogether. A ratio of 10:1 or higher can be considered a noise gate.

Note: Some people also use the term audio expansion to refer to the process of decompressing previously-compressed audio data.

Source: http://www.mediacollege.com

Audio Limiter Basics

2011-04-19T18:02:00.000-07:00

A limiter is a type of compressor designed for a specific purpose — to limit the level of a signal to a certain threshold. Whereas a compressor will begin smoothly reducing the gain above the threshold, a limiter will almost completely prevent any additional gain above the threshold. A limiter is like a compressor set to a very high compression ratio (at least 10:1, more commonly 20:1 or more). The graph below shows a limiting ratio of infinity to one, i.e. there is no gain at all above a the threshold.

Input Level vs Output Level With Limiting Threshold

Limiters are used as a safeguard against signal peaking (clipping). They prevent occasional signal peaks which would be too loud or distorted. Limiters are often used in conjunction with a compressor — the compressor provides a smooth roll-off of higher levels and the limiter provides a final safety net against very strong peaks.

Source: http://www.mediacollege.com

The Basics of Reverb

2011-04-18T09:59:00.000-07:00

Reverb is arguably one of the most often-used effects in modern recording, and probably one of the most misunderstood. It’s interesting to consider the fact that, as with so many things, we’ve spent decades perfecting different ways to imitate something that occurs on its own in nature.

This month we’ll take a look at one of modern recording’s favorite effects – how it has evolved, its use and its misuse. Let’s start with a little bit of history.

Early Reflections

In the earliest recordings, the only reverb was what occurred naturally in the recording environment. The sound of the room itself was picked up by the microphone (and in most cases it was just that – one microphone), and rooms with great sonic characteristics (mainly theaters, symphony halls and the like) were sought after as recording environments. This worked fine for the recordings of the day, which were mainly of the orchestral and operatic genres.

In the post-WW2 Big Band era of the late 1940s and early 1950s, radio began to play an increasingly important role in how audiences consumed recorded music. Improvements in microphone technology and the advent of audio tape made it possible for recording engineers of the day to experiment with mic placement, increasing consciousness about reverb, if not necessarily options. One of the first documented uses of natural (ambient) reverb to intentionally enhance a recording was by engineer Robert Fine, who introduced ambient mics on some of the early “Living Presence” recordings on Mercury Records.

The first use of artificial reverb.

It was none other than Bill Putnam, Sr., founder of Universal Audio, who pioneered the use of artificial reverb in recordings in 1947. Putnam converted his studio’s bathroom to create one of the first purpose-built echo chambers, placing a speaker in one corner and a microphone in another, and mixing the sound with a live recording. The unique sound of his Universal Records label’s first recording, “Peg o’ My Heart” by The Harmonicats, was a runaway hit, and Putnam went on to design reverb chambers for his studios in Chicago and Los Angeles. Other studios followed suit (including the still-active chambers under the Capitol Records building in L.A.), and the sound of echo chambers dominated the recordings of the 1950s.

As groundbreaking as Putnam’s echo chamber concept was, it still utilized the natural ambience and reverb of a real space. It wasn’t until 1957 that the German company Elektro-Mess-Technik (EMT) unveiled their EMT 140, the first plate reverb. The famed EMT 140 (and subsequent units) worked by attaching a small transducer (loudspeaker) to the center of a thin sheet metal plate; vibrations from the speaker were sent across the surface of the plate, and were picked up by one or more small pickups attached to the edge of the plate. The result was a dense, warm sound that emulated a natural room echo but was uniquely its own. And while the EMT plate reverbs were large and unwieldy, they were still a cheaper and more versatile alternative to building a dedicated echo chamber.

Another technology that emerged during the 1950s was spring reverb. Essentially, a spring reverb works in much the same way as a plate, but substitutes springs for the metal plate. Because springs take up far less space, spring reverbs became popular in applications where plate reverbs were impractical, including early guitar amps (Fender’s being the most well-known) and Hammond organs.

Lexicon 224: The quintessential '80s reverb.

The advent of digital technology in the late 1970s and early 1980s changed the face of most things audio-related, including reverb. Digital reverbs made it possible to create “programs” that emulated the natural ambience of any space, as well as the sound of plate, spring and other electronic reverb sources. In almost no time at all, a veritable flood of digital reverb and multi-effects boxes appeared on the market. Some of the most popular units included the EMT 250 and Lexicon’s 224 and 480, and Yamaha’s Rev7 and SPX90. Now it was even possible to modify the parameters of those programs to create effects that don’t occur naturally, including artificially altering early reflections (the first reflected sound), pre-delay (the time before the first reflected sound is heard), and even reverse and gated reverb (probably one of the most overused snare effects of the 1980s).

Less is More

In the early days of recording, the only reverb on a record was that of the room the recording took place in. Studios were prized for their natural ambience. As multitracking evolved, studios were designed to be fairly “dead” and mics were placed close to each instrument to capture as much direct sound as possible, with minimal reflections from the room. A single reverb device (usually a plate or chamber) was then used to create an artificial “room” ambience.

In today’s DAW-oriented world, signal processing is cheap and plentiful. Even entry-level recording programs offer a multitude of reverbs, and today’s recordings typically employ one or more reverbs on each instrument. Now the challenge is no longer which reverb to use, but what combination of reverbs works to create a cohesive and natural sound.

Not surprisingly, it’s easy to overdo it. In fact, excessive or poorly used reverb is one of the most common mistakes inexperienced recordists make. An instrument’s direct sound is important in establishing directionality and clarity. Add too much reverb and your mix can easily become a lush pool of mush. One general guideline to consider is that, unless you’re intentionally after a special effect, the best use of reverb is typically when it’s almost imperceptible within your mix.

Anatomy of a Reverb

At first look, many of the parameters of reverb units can be pretty confusing. We can simplify things by breaking it down to basic physics.

Like throwing a stone into a pool of water, sound emanates from the source in waves. Those waves eventually hit multiple surfaces (walls, ceiling, floor, seating, whatever) and echo back, mixing with the original sound. The way we hear that sound depends on several factors — how far away those various reflective surfaces are, what they’re made of, where our ears are located in relation to the original and reflected sound waves, and even other subtle factors like temperature, humidity, altitude and more. In most cases, what we hear is the product of thousands of echoes, reflected many times.

Our brains decode this information in various ways. The first echoes that occur when sound waves hit surfaces (early reflections) and the amount of time between the initial sound and those first reflections (pre-delay) work together to tell us how large the space is, and what our position is within the space.

The length of time until the echoes die away (decay) also helps determine the size of the space, but the way that decay interacts with the early reflections also makes a difference. For example, a small but reflective room (e.g., a tiled bathroom) can have a decay time similar to a larger hall, but the smaller room’s early reflections will arrive sooner.

Old School: a custom "Men's Room" echo chamber.

The tonal color of the reflections also plays a critical role. The reverb in that tiled bathroom will be considerably brighter sounding than a larger room with wood or fabric-covered walls. Larger halls will also attenuate different frequency ranges at different rates, and the combination of which ranges last longer also affects our perception of the space.

Other factors also affect our perception, including density (how tightly packed the individual reflections are) and diffusion (the rate at which the reflections increase in density following the original sound). A large room with parallel walls will usually have a lower diffusion rate than a similarly sized room with non-parallel or irregularly shaped walls.

As you can imagine, creating a natural sounding ambience is a complex, multi-faceted process that involves programming dozens of interdependent parameters. For the most part, it’s best to find a reverb program that comes close to what you’re looking for, and keep the tweaking to a minimum.

What Works Where

As with most effects, there are no hard and fast rules, other than the age-old adage “trust your ears.” But here are a few general guidelines to start with.

As stated earlier, less is more. You’ll achieve more natural sounding results using few reverbs, rather than several. One short, bright program (small room or plate) and a larger, warmer program (large room or hall) will often be enough to cover most of your mix. For best results, insert reverbs into an effect or aux buss, rather than directly into a signal chain. This will enable you to use the same reverb for multiple tracks, while varying the amount of send for each source.

The EMT 140 Plate Reverb plug-in for the UAD platform.

Drums and other percussive sounds typically sound more realistic with small to mid-sized rooms (shorter reverb tails, shorter pre-delay), or plate programs. A longer pre-delay can create the impression of a “phantom” doubled attack, while a longer reverb decay can affect directionality and clarity. Too much high-frequency content can create a harsh, brittle sound, particularly on snare drums. Lower density settings can also sound coarse and unnatural on drums. Higher densities and warmer reverbs will generally deliver better results.

Acoustic instruments like strings, woodwinds and some vocals can benefit from larger room and hall settings and longer pre-delay times, which can help smooth and add depth. Those larger spaces can also be useful in widening a stereo image. Overused, a large room sound can “blur” an instrument’s attack and create a “swimmy” sounding mix that lacks definition and directionality.

One trick for helping to define, rather than blur, the imaging in your mix, is to use reverb in combination with delay. Pan the original sound slightly to one side. Delay the reverb return slightly (try anywhere from 3 to 10 ms) and pan it to the opposite side. This works particularly well to help separate sounds in similar tonal ranges, like multiple stacked guitar tracks.

The UAD-2 Powered Plug-In emulation of the classic EMT 250 Electronic Reverb unit.

Vocals can be particularly susceptible to losing definition with larger room settings. Especially with shorter pre-delay times, the reverb can “step on” the vocal, robbing intelligibility. Using a longer pre-delay before the actual reverb kicks in allows the vocal’s clarity and impact to cut through, but gives it a natural “tail” that rings out without blurring. Background vocals are somewhat less critical in this respect, and can often benefit from a larger room setting, which can smooth and blend multiple parts.

Be Creative

We’ve spent most of this column talking about the best ways to use reverb naturally. And for the most part, that’s a good idea. In fact, in most instances, the best use of reverb is to create a mix where its use is pretty much indiscernible.

But as with most effects, experimentation can lead to some great surprises, so don’t be afraid to bend the rules. Try combining a couple of different instances of the same reverb with slightly different parameters and panning them left and right. Or try adding a subtle chorus or distortion to a reverb. Again, subtlety is key here – a little bit of something unusual, buried deeply in the mix, might be just the thing to give your mix that special “something.”

Source: http://www.uaudio.com

Studio Monitor Basics

2011-04-15T09:07:00.000-07:00

Powered vs. Unpowered Monitors

Q: Which is better, Powered or Unpowered monitors?
A: The answer, of course, is that there are benefits to either, and that it depends on your situation. A "powered" monitor is one that is self-powered, or has its amplification built into the speaker cabinet thereby relieving you of purchasing an amplifier separately (and the headaches involved). An "unpowered" monitor is not self-powered which necessitates purchasing a power amplifier.

Passive / Unpowered Monitors & Amps

To operate Passive / Unpowered Monitors, you simply connect the line-level outputs of your mixer to a power amp and then run speaker wire to the monitor. If you already own a power amp, then passive monitors may be your ticket to saving money. Simple right? Well, yes and no. Now you have to deal with two separate pieces (actually several when you consider cables and connectors) - the monitor and the power amp. Monitors are fairly straight forward, but while figuring out a power amp is not rocket science it's not super easy to set up for the beginning studio owner. Here are just a few issues you'll need to address when using a power amp:

Ensure Proper Cooling: If you rackmount your power amp, DO NOT block the front rear or side air vents. The side walls of your rack should be a MINIMUM of two inches from the amp and the back of the rack should be a MINIMUM of four inches away from the back of the amp. Without proper airflow, your amp will not function properly which can cause damage to both the amp and your speaker.
Proper Cables: Take time to figure out your inputs and outputs on your amp and purchase the correct cables with proper gauge (at least 22-24 gauge to your amp input, 16 gauge or better to your monitors, depending on distance). Your amp may have balanced or unbalanced XLR, balanced or unbalanced 1/4-inch connectors; or you may find banana plugs, spade lugs or even binding posts. Be sure to reference your owner's manual for specific information.
Use care when making connections, selecting signal sources and controlling the output level.
Remember that amps have a sonic character all their own. Just as you might combine the sonic characteristics of a microphone and preamp, you need to consider the combination of the sonic character of your reference monitor and separate amplifier. In other words, the same passive / unpowered monitor will not necessarily sound the same when juiced by different amplifiers.
A general rule of thumb when searching for amps to drive your passive / unpowered monitor is to purchase an amp that delivers twice (2 x) the wattage necessary for the monitor (this allowance is for headroom). So if you need 300 watts at 8 ohms, purchase an amp that is rated for 600 watts at 8 ohms. Again, this is just a rule of thumb and is not necessarily true in all cases.

Active / Powered Monitors

The benefits that come from investing in Active / Powered Monitors on the surface is that you simply don't have to deal with any of the above-mentioned issues. Many of us don't want to know about ohms, watts, damping, overload protection, crossovers, and the like - it's enough to know that the monitor works, it sounds great and all I really have to do is plug it into my mixer or computer audio interface. Besides, we'd really like to get back to making or recording music. If, on the other hand, you'd like to know more about the technical benefits of Active / Powered Monitors, we suggest you call your Sales Engineer.

Studio Monitor Placement Guide

Where do you aim the speakers to give you the smoothest and most consistent sound, and how far apart do you place them to give you a good stereo image? The basic rule is to follow the layout of an equilateral triangle, which is a triangle with all three legs the same length. The distance between the two monitors should be roughly the same as the distance between one monitor and your nose in the listening position where you are leaning forward on the console armrest. The speaker axis should be aimed at the half-way point between your furthest forward and the furthest rearward listening positions. This is typically a range of about 24" (600mm). If you can, you also want to try to get your ears lined up with the vertical speaker axis (half way between the woofer and the normal listening position lined up in the best spot possible. If this would have you resting your chin on the console or desktop, you could tilt the monitor back slightly. This keeps your head in the sweet spot whether you're leaning forward adjusting level or EQ, or leaning back and listening to the mix. Don't go crazy trying to get this exact to three decimal places, within an inch or two gets you into the game.

You will also want to keep your monitors upright and vertical even though you'll be tempted to place them on their side to give you a better line of sight behind them. With the monitor on its side, moving your head horizontally means that you are now moving through all those rays, or lobes, where the wavefront from the woofers and tweeters interfere with each other. The midrange frequency response will be different for each head position. It is our opinion that all two-way component monitors, no matter who manufactures them, need to be used with the multi-driver axis vertical (that's just the way it has to be when you're in the near-field).

What is "Bi-Amplification"?

When a passive system's single amplifier must reproduce the whole audio spectrum, low frequencies rapidly "use up" the amp's headroom. As higher frequencies "ride along" on lower frequency waveforms, they can be chopped off or distorted even though the high frequencies themselves would not be clipping. Separating highs from lows via an active electronic crossover lets a bi-amped system use two different amplifiers. Each is free to drive just one transducer to its safe maximum limit without intermodulation distortion or other interaction between the two drivers.

How does price relate to sound quality?

Q: Is there really a difference between monitors that are just a few hundred dollars and the ones that I see for a few thousand?
A: Sure, just as you would find qualitative differences in microphones, guitars, preamps, keyboards, etc. that varies in price, so it is with reference monitors. The old adage that "the devil is in the details" is still true. Generally speaking, manufacturers with monitors costing more, such as Genelec, Focal and Mackie (among many) have spent more time developing a better design and use higher quality components. This equates to a more accurate imaging, smoother frequency response, extended low frequencies and clearer high frequencies and consistent quality at different dynamic levels. In other words, better mixes, faster. Saying that, today's crop of monitors from M-Audio, Samson, Edirol (and others) that come in for just a few hundred dollars are a tremendous value for many desktop audio professionals who aren't necessarily planning to finish their mixes (or master) on their own. Our advice is to purchase the best set of monitors your budget can afford - your mix and your ears will thank you.

Source: http://www.sweetwater.com/

Recording Acoustic Guitar

2011-04-14T08:27:00.000-07:00

Taylor 314CE Acoustic Guitar

Taylor Guitars

Most home recording engineers are singer/songwriters - recording vocals and acoustic guitar at home. And as any of them will tell you, getting a good acoustic guitar sound can be hard! In this tutorial, we'll take a look at recording the acoustic guitar, one of the most difficult instruments to get right!

Microphone Selection

The first thing to do before you start recording is to select the microphone you'd like to record with. For acoustic guitar, you can do two different techniques: a single, or mono, microphone technique, or a two-microphone, or stereo, technique. What you do is completely up to you and what resources you have available.

For recording acoustic instruments in the highest quality, you'll want to use a condenser microphone rather than a dynamic microphone. Good condenser microphones for acoustic guitar recording include the Oktava MC012 ($200), Groove Tubes GT55 ($250), or the RODE NT1 ($199). The reason you want a condenser microphone rather than a dynamic microphone is very simple; condenser microphones have much better high frequency reproduction and much better transient response, which you need for acoustic instruments. Dynamic microphones, like the SM57, are great for electric guitar amplifiers which don't need as much transient detail.

Microphone Placement

Take a listen to your acoustic guitar. You'll find that the most low-end build-up is near the sound hole itself; the higher-end buildup will be somewhere around the 12th fret. So let's look at the two types of microphone placement I mentioned earlier.

Single Microphone Technique

If using just a single microphone, you'll want to start by placing the microphone at about the 12th fret, about 5 inches back. If that doesn't give you the sound you want, move the mic around; after you record it, you might want to give it extra body by "doubling" the track - recording the same thing again, and hard-panning both left and right.

When using a one-microphone technique, you might find that your guitar sounds lifeless and dull. This is generally fine if you're going to be mixed into a mix with many other elements in stereo, but should be avoided when the acoustic guitar is the primary focus of the mix.

Two-Microphone (Stereo) Techniques

If you have two microphones at your disposal, put one around the 12th fret, and another around the bridge. Hard pan them left and right in your recording software, and record. You should discover that it's got a much more natural and open tone; this is really easy to explain: you have two ears, so when recording with two microphones, it sounds more natural to our brain. You can also try an X/Y configuration at around the 12th fret: place the microphones so that their capsules are on top of each other at a 90 degree angle, facing the guitar. Pan left/right, and you'll find that this gives you a more natural stereo image sometimes.

Using The Pickup

You might want to experiment using the built-in pickup as well, if you've got the inputs to do it. Sometimes taking the acoustic guitar's pickup and blending it with microphones can yield a more detailed sound; however, it's totally up to you, and in most cases, unless it's a good quality pickup, it'll sound out of place on a studio recording. Remember to experiment. Each situation will be different, and if you don't have any microphones to record with, a pickup will do fine.

Mixing Acoustic Guitar

If you're mixing acoustic guitar into a full-band song with other guitars, especially if those guitars are in stereo, you might be better off with a single-mic technique, because a stereo acoustic guitar might introduce too much sonic information into the mix and cause it to become cluttered. If it's just you playing guitar and vocals, a stereo or doubled mono technique will sound the best.

Compressing acoustic guitar is subjecting; a lot of engineers will go both ways. I personally hardly ever compress acoustic guitar, but a lot of engineers do. If you chose to compress, try to very lightly compress it - a ratio of 2:1 or so should do the trick. The acoustic guitar itself is very dynamic, and you don't want to ruin that.

Remember, any of these techniques can apply to other acoustic instruments, too!

Source: http://homerecording.about.com

Audio compressor basics

2011-04-13T14:11:00.000-07:00

Compression is a very important aspect of audio production. You need to have an idea of what the audio compressor does and what all those buttons do. Everybody uses it differently and everyone has a method. These methods are created after mastering the tool that compression is.

You can't have a specific method to compression if you don't know how to instinctively use it. You can't only know what one button does and ignore the others. That's no method.

Therefore, let me introduce you to all the typical buttons an audio compressor has. What do they do? How do they interact with each other. When you get a grasp on which button to push, knob to turn or slider to move, you have an easier time getting that sound you want.

Audio compressor

Basically, compressors compress the audio signal you want to process. Easy right? Well, that doesn't really tell you anything. Compressors even out the audio signal at a certain level, called the threshold and compress the level above at a certain ratio you determine. We push the loud levels lower and therefore have a less dynamic signal.

Now, let's get into all the buttons and sliders and whatnot. As an example, I'm using Logic's Audio Compressor. Any compressor you have will almost certainly have the same buttons.

Let's start with the most important parts first. If you have a incredibly basic audio compressor on your hands, chances are you only have these two parameters to work with. Threshold and Ratio.

Threshold

The threshold determines at what level the compressor starts acting. Say you have a signal that has peaks at around -1dB on the meters of your fader. If you have your threshold at -10, the compressor will start working when your audio starts going over the -10dB threshold.

Now, if you have a weak signal that never goes over -15dB and you have your compressor on -10dB there's no compression going to take place. Maybe, if you have a swanky cool compressor it will give you a nice color to your sound, but as for actual compression, nada.

The signal doesn't reach the threshold, and therefore none of the other parameters of the compressor are going to start working. But once that signal goes over your threshold it will get compressed. How much will it get compressed? Well that brings us to our next button.

Ratio

The ratio is where you determine how much compression you are going to apply to a signal that goes over your threshold. For every signal that goes over the threshold, it gets compressed according a certain ratio.

Example: For a compressor with a threshold at -10dB and a 3:1 ratio, a nice starting point for vocals. If you have a semi-constant level of the vocal at -1dB it will become compressed so that it only reaches -7dB.

Why?

Because after going over the threshold the vocal reaches its peak 9dB after -10dB, or at -1dB. We take those 9dB and divide them by three, since the ratio is at 3:1. Out of that we get 3dB which we add to the threshold at -10dB. A compression of 6dBs reaching its peak at -7dB. Let's illustrate this with a simple formula:

In this formula you can see the basics of calculating the output of a compressor.

If we take the example above and apply it to this formula, we get this:

So you see, that if we have a higher ratio, we compress the signal more resulting in less signal coming out of the output.

Say we have an example of a loud kick drum that's peaking at +4dB but we have a threshold at -20dB and a ratio of 8:1. That's a lot of compression but serves to illustrate a point.

We have a dynamic range of 24 dBs, from -20dB to +4dB. We are compressing everything that goes over -20dB by a ratio of 8:1.

Let's plug those numbers into the equation:

The highest peaks of the kick drum that are reaching +4dB before are now only reaching -17dB! That 24dB dynamic range we had from -20dB to +4dB has been reduced to 3dB. Talk about over-compression!

After studying these formulas and basics behind the relationship between threshold and ratio, I think we can move on to the next phase of our compressor journey.

The limiter

A limiter does things a little differently when it comes to the ratio and how it reacts to sounds that go over the threshold. Instead of compressing the peaks that go over the threshold, a limiter simply cuts them off. Which can actually sound better sometimes!

The knee

The knee on the audio compressor has a relationship with the ratio. It applies compression gradually increasing the ratio until a certain point. See the link below for a detailed explanation.

Attack & Release

Every plugin seems to have an attack and release of some sort. And they often don't even mean the same thing. Let's dive into how the attack and release on the audio compressor work.

Attack

The attack, measured in milliseconds, is how fast the compressor starts acting on a signal. With a fast attack the compressor starts working right away on the audio, often dulling the sound of the transient.

What's a transient? Transients are the first seconds, or attack of the envelope of a signal. Huh? The first peaks of a signal are called the transients ok? Drums have fast and loud transients !Whack! !Whack! But a cello might have a slower transient, or attack.

By using a fast attack you make the audio compressor chomps down right away on a signal, but with a slower attack time the initial attack, transient or punch gets through before the rest of the signal is compressed.

In practice:

For a punchier kick drum have a slower attack so you get the untreated sound of the beater. But if you want a thumpier and more rounded kick drum, have the attack at a fast setting.
If you have a bad bass part and you get a lot of uneven notes jumping out at you, having a fast attack setting can help dull out the unexpected pops from the bass player. You can even put the ratio into limiting by having it at 10:1 or higher if you are dealing with a really troublesome part.

Release

In contrast, release is the parameter that determines, in milliseconds, how long the audio compressor will continue acting on a signal once it goes under the threshold again. If the release is too fast you run the risk of the compressor letting go too early, but if you have the release too slow you can get a pumping effect. This pumping effect is a clear sign of over-compression because the compressor never stops compressing. It is too slow too react, even after the signal has gone under the threshold.

Thus the compressor compresses the next signal too, even though it is under the set threshold. This can work on some instruments with a slow transient response if you want an even compression..

In practice:

Try to make your snare drum breathe by making the release go in time with the track. You can watch the time of the release in the gain reduction meter.
A quick release on the kick drum is good since the signal of a kick drum is so short. That way you can be sure that it doesn't continue into the next kick drum hit.

Now that we've covered the basics of the attack and release, let's make our way to the most important tool you have when working these aforementioned parameters.

Gain reduction meter

This is the most important visual meter you have when compressing. It shows you, in dB how much you are really compressing. With it you are able to gauge the effect you are making, both by making sure that your signal is reaching the threshold and also seeing how fast the attack and release are working.

If you don't have the threshold low enough you won't see it working at all, since the audio compressor isn't compressing your signal.

If you have your release too slow you can be sure to see how the meter never really goes down. By taking a visual cue from the GR meter you can tweak the release in time with the track.

Obviously it's good practice to use your ears when compressing, but being able to see the amount of reduction and compression visually is a very effective way in quickly finding the correct sound you are looking for. If you only want to control the peaks, it's easier to see on the GR meter when it is only compressing the peaks instead of trying to listen to the actual audio signal.

Makeup-Gain

Lastly, makeup gain is that last parameter we need to worry about. Since an audio compressor turns down the volume of certain parts of our audio signal, we need a gain knob to increase the average volume to where it was before we started compressing. If we are always compressing around 3dBs we will need to turn the gain up 3dB in order to make up the loss in volume by compressing.

Reference: Gibson, Bill.(2007). Instrument & Vocal Recording. Hal Leonard Books.

Buss compression

You can crank up the sound of your drums by using buss compression in your tracks.

By using parallel compression underneath the dynamic drum tracks you can create a larger than life sound.

Multiband compression

Multiband compression is used for many purposes, and can be a handy tool for mastering your tracks. By using multiband compression you can define the specific frequency areas you want to compress, using different compression values on separate areas of the frequency spectrum.

Conclusion

Now that we've gone through the knobs you can find on your typical audio compressor you are all set to start compressing. If you need further tips on compressing, or want to share your cool compression tips, please share it with the rest of us here below. Everybody is always looking to enhance their understanding of compression.

Source: http://www.audio-production-tips.com

Ultimate Guide for Mixing Drums

2011-04-12T08:25:00.000-07:00

Get a great sound from your acoustic drums.

Your drum sound is one of the most important aspects of your mix. Mixing drums is therefore a number one priority for laying that solid foundation to your tracks, guaranteeing you a solid rhythm section.

Drums can be one of the most problematic instruments to get right in a mix. The complexity of recording drums is equally complex when it comes time to mix them.

If you did a great job recording the drum kit, then mixing your drums can only be a pleasurable experience.

But where to start?

Kick Drum Sound

Mixing drums starts with the foundation of the kick drum. The sound of the kick drum, along with the snare will be the defining factors of your drum sound. If you leave the kick drum sounding bad, the whole foundation of the song will lose its footing. The kick drum needs to be tight and punchy, with enough low end to fill up the bass range and enough mids to cut through the mix.

EQ

It's important to emphasize the low end of the kick with EQ. If you feel there isn't enough bass to your kick drum, a low shelving boost around 80 – 100 Hz normally does the trick.

A boomy kick drum can also cloud up the clarity of your kick drum sound, so it's normally a good idea to cut around 200 – 250 Hz if you feel there is too much muddiness in your kick drum sound. A boxy kick drum sound is also a common nuisance, which can be fixed with Eq'ing out the boxiness that resides in the aread around 300 – 600Hz or so.

If your kick drum is all thump and no snap then we need to bring out the sound of the beater. We can usually find it around the 2 – 4 Khz area. Depending on the genre of the song, and the type of beater used, different frequency boosts in the beater area generate different sounds. A boost at 2.5 Khz is more of a typical rock sound as opposed to a narrower boost at around 4 Khz, which results in a Hardcore Metal type snap.

Compression

When mixing drums, along with everything else, using compression is a subjective subject and everyone has an opinion on how things should be compressed. That said, there are a few guidelines you can follow to get a steadier kick drum sound.

How much gain reduction you want from the compressor depends on the genre, the steadiness of the drummer and the feel of the song. I usually start with a ratio of 4:1 or 6:1 and lower the threshold down until I'm compressing around 6dBs.

Then I adjust the attack and release depending on what sort of sound I want. A fast attack clamps down on the transient of the kick drum, dulling the initial attack down somewhat, but a slower attack lets the attack of the beater break through before the compressor starts working.

I try to time the release in time with the beat, so that the compressor has stopped compressing before the next hit. It's easy to do this in modern DAWs because you are able to see the gain reduction meter working, enabling you to tweak the release perfectly in sync with the song.

Snare drum sound

Partner in crime with the kick drum, the snare drum is the other defining rhythmic factor to the song. “It's all about the snare” an experienced engineer once told me, because it's what supplies the song with that steady backbeat. Since it's such an important aspect of mixing drums, there needs to be a lot of care taken with getting the best sound possible.

EQualization

EQ-wise, there is not an awful lot you need below 100 Hz, so you can start by high-pass filtering all the low end away.

The body of the snare can be brought forward with a little boost at around 150Hz, if you feel like it's lacking some thickness.

I like thick snares so I often catch myself adding a little weight to the snare around that area.

If your snare has ringing frequencies that you find annoying you can try pinpointing them by boosting a specific frequency band with a high Q and sweeping the spectrum until they pop out. I find that sometimes the snare needs a little cut in the mids, either resulting from boxiness at 500 – 800 Hz or too much of a nasally attack from the area around 1 Khz. Enhance the attack of the snare with a broad boost around 2 – 4 Khz and search for the sizzle of the snares in the higher frequencies.

Compression

Like I do with the bass drum, I try to make the snare compress in time with the song. By timing the attack and release I can get a nice steady snare sound that breathes with each hit. I normally leave the attack at a medium to slow setting so that the snap of the snare is unaffected, and time the release so that it stops compressing just in time for the next hit.

I start with a ratio of 3:1, often going way higher as it depends on the genre how hard I want the compressor to be pumping. You can adjust the threshold so that it is only lightly compressing the peaks for a subtle sound, or you can push the threshold down harder for a heavily compressed sound.

Snare compression is perhaps one of the most argued about subjects in audio production. Every engineer has a certain method to mixing drums, and I think it's up to you to experiment and get acquainted with the knobs and sliders on your audio compressor so that you can create the sound that you want.

Reverb

You can create a completely different snare sound by just applying an interesting reverb to it. Whether that's a rock arena reverb, subdued room or even a spring reverb, different reverbs can transform the sound of your snare drum.

Go through your reverbs and see what type of reverb sounds best with the song you're mixing. Are you going to add a bright plate reverb to make it stand out, or will you be mixing it into a specific room with a small room sound? If you are in a particularly adventurous mood, you can try adding some gated reverb to your snare.

Mixing the toms

EQualization

If the toms are playing a big part in your drum sound, mixing them so that they sound punchy and powerful is crucial to a great drum sound.

Get them punchy with EQ. The best way to EQ toms is to find the unflattering frequencies with your equalizer. Normally, these are the middle frequencies, from 300 – 800 Khz or so.

Find the boxy and unwanted frequencies, cut them out and then add low end power and high end punch as needed.

When mixing drums like toms, sometimes you need to finely cut a few adjacent frequencies instead of scooping out a big portion of the frequency spectrum.

Compression

By adding a generous amount of compression to your toms you can get a larger than life sound out of them. You can fatten them up considerably with some tight compression, and with the addition of a little reverb you can make them sound huge and powerful. If that's what you want to go for. The same rule of subtle compression applies as well to toms if you only want to control the peaks and lightly color their signal.

Overheads

The overheads might be the most important microphones on the kit. The overheads are the microphones that are supposed to pick up every drum and give a complete sound to your drum kit. There are two ways of mixing drums with the overheads; you can either use them as the primary sound, sculpting every drum around the overhead sound or you can use them to primarily accent the cymbals and air around the kit.

By adding the overheads to the mix early on, you can get a better sense of the full sound of the kit, making your drum mixing easier. Just notice how different a snare drum microphone sounds compared to a snare that's coming from the overhead mics.

By adjusting the overheads with the rest of the close miked drums you can get a different sound. By focusing on the overheads you can get a roomier sound, but if you want a close in-your-face drum sound you would rather use the overheads as complimentary to the rest of the drums, mainly using them to accent the cymbal sounds.

The Hi-Hat

Mixing drums is a selective process, meaning that certain elements of the drum-kit only need specific frequency ranges. You only need a specific frequency range from the hi-hat. Considering that the hi-hat microphone is probably picking up a lot of bleed from other drums, some heavy high-pass filtering is in order. Filter up to 250 Hz at least, even higher if you feel that you aren't losing anything from the hi-hat sound with higher filtering.

Now if you feel that there is something lacking from the hi-hat, or that you want to bring out the gong sound, you can find it in the 200 Hz area. So if your hi-hat needs a little more gong to it, you will have to sacrifice that aggressive filtering. Like everything else, just filter until you start hearing the sound becoming compromised and then back off a little bit.

Cutting at 1Khz can reduce the cheap jangly sound from the hi-hat, but you can enhance and give it some sparkle with a boost from 7 Khz or so. Use a high shelving EQ if you want to enhance the high end with some air, but a parametric bell EQ if you just want to accent a specific frequency area.

Room mics

Room microphones give a different sound to the drum kit than the regular overhead mics. Due to the distant miking technique most room mics are recorded with, we get a full sound of the drum-kit as well as a great amount of the reverb of the room it was recorded in. Which, depending on the sound of the room, can either sound amazing or horrible.

Mixing Drums With a Roomy Sound

But let's assume our recording room is great. With a nice room mic picking up the complete kit we can try a few different techniques. We can apply some heavy compression to the room mics to get an even punchier sound. We can EQ the kit as to draw out the most important elements, such as kick and snare and we can add it underneath an already great drum sound for that final touch.

Mixing drums into a room

If the drums weren't recorded in a nice sounding room and sound quite dead when they come from the recording stage, it's time to add some space to our drum tracks.

A good way to add some ambience to our drum tracks is to add a 0.5 second drum room reverb. You can add a a nice amount to the overhead tracks, and maybe even a slightly different reverb to the snare to make it stand out. Go through your reverbs to try to find the best sound to your particular track.

Mixing drums – Drum Replacement

If you have ended up with badly recorded drums that you can't get even get close to a good sound out of, maybe it's time to look elsewhere. Drum replacement is a great way of mixing drums in order to keep the dynamics of a real performance with the sound quality of amazing sounding drum samples. Audio recording software such as Pro Tools and Logic Pro have great drum replacement options when you are stuck with crappy sounding drum tracks.

Conclusion

Mixing drums is a challenging but enjoyable aspect of audio production. Since there are so many different ways of getting the drums to sound with EQ, compression and other mixing tricks there is no actual right way of mixing drums.

The only solid piece of advice I can give you for mixing drums is to experiment with all the tools you have on hand. Get every element to sound as good as possible and then try to mold them together to make them sound like a complete whole.

As always, there are trends in the music industry as to what sounds good right now, but being able to get whatever sound you want, whether it's huge 80's toms or a 90's arena rock snare is an important aspect of being a well rounded mixing engineer.

Source: http://www.audio-production-tips.com

Drum recording

2011-04-12T08:19:00.000-07:00

Drum recording - Good mic placement

Drum recording can be a difficult task. Not only is it about selecting the right microphones and placing them, but you have to be aware of how each and everyone of them interacts with each other and the room around the drum-kit. This is no easy feat, as each drum is different and you are not recording one instrument but rather recording a group of instruments together.

Different types of music warrant different types of microphone placement and selection and in this article I'll be going into detail about how I recorded a drum-kit for a vintage blues project.

Blues has a more natural sound than the modern drum sound in rock recordings. I'm going to go into how I aimed for a more natural and organic drum recording.

Mic selection

I wondered what types of microphones to use, as vintage blues was often recorded with one or two mics I didn't want to go overboard in mic placement. Still, I decided to use microphones on each drum just to make sure. I could always discard them later, not putting them into the mix.

It also meant I could make two different mixes, a vintage one using only the overheads and room mics and a modern one using all of the resources I had at my disposal. That way, the extra work spent during the drum recording phase didn't go to waste.

Note: If you are confused by the microphone terms I am using, check out the recording microphones section, and then promptly return here to read more about drum recording.

So in the end, these are the microphones I used:

For the inside of the bass drum I used a Sennheiser PZM electret microphone.

For the outside I used the AKG D112.

On top of the snare I used an AKG 414 large condenser. This made the snare come to life and was much fuller than if I would have stuck a SM57 on there.

To get the rattling of the snares I used a small diaphragm AKG C391.

For both toms I used a pair of Sennheiser MD-421.

For overheads I used a pair of AKG-414 large condenser. I decided on the large diaphragm in order to get a more full range thicker sound that I wouldn't have gotten if I had used a regular pair of small condenser.

To catch the full kit and a little bit of ambience I used a Neumann U-87 as a room mic. It captured the full range of the kit and although the room didn't sound that good it worked out nicely with the rest of the kit.

Now, I know for some of you, those microphones aren't within your reach budget wise. A Neumann U-87 is a classic and a very expensive mic to buy, as well as the AKG 414, let alone three of them!

If you aren't able to use these exact mics, don't worry, try to get as close as you can using the mic selection you have at your disposal.

For instance, if you can't get two 414s as overheads, at least try to use large diaphragm condensers instead of small ones. It certainly gives a distinct sound, regardless of the model you are using.

Microphone placement

Now in order to get the best drum recording possible obviously I had to place the microphones well. The room was not a great sounding room as it was quite dead and didn't have anything special to offer. So I resorted to close miking everything as best as I could and get a great room sound in the mixing phase.

Here's how I placed the mics.

Bass drum

The bass drum was using two mics, one inside and one outside. The PZM was put very close to the inside head in order to get the sound of the beater.

The D112 was positioned right outside the sound hole.

Snare

The 414 on the snare was positioned very close and at 45° pointing at the center. It was set to hyper-cardioid pattern in order to make it more directional.

Having it more directional decreases bleed from the hi-hat as it picks up less sound from the side, where the hi-hat was positioned.

Under the snare I positioned the C391 directly pointing to the snares. Using two microphones on the snare, pointing in opposite directions you have to be aware of phase problems.

If you don't reverse the phase of this microphone the snare will most surely sound dull and lifeless.

Hi Hat

I put another C391 pointing away from the drum-kit on the hi-hat. Positioning it on the outside border and away resulted in a very clear sound.

Good mic placement is crucial to your drum recording

Toms

The toms were a typical 45° angle pointing at the center of the toms to get a little bit of attack from the sticks. Nothing super exciting here, just the simple and normal way to do it.

Overheads & Room microphones

The overheads were positioned quite closely as I wanted to minimize the room sound. You can see on the picture below that they are fairly close to the drum kit.

It's crucial that the overheads be positioned at the same distance from the snare drum because you could end up with annoying phase problems when the snare sound hits the overheads at different times.

So as you can see, they are both tilted towards the center of the kit, in order to minimize any problem I might have with the snare.

Lastly, the Neumann U-87 was positioned in front of the drum-kit. It picked up the whole kit nicely, and being in cardioid pattern didn't pick up a lot of the room. Although, it being farther away than the other mics, the room sound was definitely a factor.

Listening to the drum recording in the control room, leveling off the faders it was obvious that we had accomplished a pretty good drum sound.

Conclusion

Although these microphones and placements were used for blues, it does work for other styles as well of course. As somebody probably says, a good sound is a good sound is a good sound. It pretty much depends on the performance of the player and the style of music that's being recorded what type of sound you will get.

Source: http://www.audio-production-tips.com

Delay Basics

2011-04-11T08:09:00.000-07:00

Understanding the basics of delay is very important as its one of the most commonly used effects. Once you've got the basics nailed you can start to create your own sounds and techniques. In this tutorial i'll cover the basic parameters that are commonly found on delay plugins. The plugin i use in this tutorial is the Kjaerhus audio classic delay which i think is very good. If you want to know more i have written an article on it here.

What is delay?

Delay is a an time based effect, which means it creates its effect by manipulating the way the sound behaves over time. What delay actually does is take the audio signal that's fed into it and repeats it over time. The resulting sound is like an echo of the original sound, just like if you shouted into a big cave your voice echo's back at you. This was first achieved with tape delay where there was a loop of tape and different record and playback heads which would record and playback the signal as it went round the tape loop. Usually the sound gets quieter everytime its repeated and some times deteriates tonally (this is what old tape delays did). There are different types of delays like tape delay, analog and digital which all have different tonal characteristics. In this tutorial were focusing on software plugins that usually emulate the different sounds.

Parameters

Understanding what each parameter does is important to being able to use delay properly and also gives you the power to manipulate it to create your own sounds. The actual names displayed here may differ from plugin to plugin but they all do the same thing.

Time

The time parameter is probably the most obvious. Time changes how long the delay time is, in other words how long between each delay is heard. Time is usually displayed in milli seconds but more and more plugins sync to the tempo of the sequencer or host, so it might be displayed in fractions of a beat like 1/4, 1/8 and 1/16 meaning the higher the second number the faster the delay.

Feedback

The feedback or repeat parameter controls how many of the delay repeats are heard, almost like the decay of the delay. For example if feedback is set to 100% the delay will be never ending. What the feedback parameter actually does is feed the delay signal back into the plugin to create more delays, hence the name.

Low & high cut filters

Not all plugin delays will have this parameter but many good ones do. Basically they allow you to filter out any low or high frequencies. I really like these controls as they allow you to mimic old tape and analog delays which had high pass filters and naturally filtered out frequencies.

Mix

The mix parameter controls the mix between the dry (uneffected signal) and wet (effected signal) signals. This parameter is useful when you use the delay as an insert on a audio track as you can control how much of the original signal is heard. If you are using the delay as an effects send you should have the mix parameter set so only the wet signal is heard as you hear the dry signal from the original track.

Gain

Very simply put it allows you add extra gain to the effect if its too quiet. Not all plugins will have this parameter.

Conclusion

So now you the basic principles and parameters of delay you can go and make your own presets and sounds. Once you've mastered these you can go on the the intermediate and advanced techniques of using delay. The name of the game with delay is to experiment and see what you can get out of it. Check out the related tutorials below to get more information on techniques and the different types of delay.

Source: http://www.live-laptops.com

All about Studio Compressors

2011-04-10T17:06:00.000-07:00

You have no doubt heard people talking about compressors and recording. Perhaps you heard of albums or tracks being "compressed" to make the sound better. You may have also heard about audiophile albums boasting that "no compression" was used as a positive thing. Huh? What gives here? We'll get to that and many other issues in this article, designed to make you fully conversant about the compression process and where to use and how much to use and when not to use. We'll end up with a discussion of software vs. hardware compressors and when software is appropriate.

Compression, ideally, is an "invisible" sort of effect that can bring your audio material up to spec with professional recordings. Most audio professionals do use compressors in every piece and sometimes on nearly every track in every piece.

And sometimes compressors are overused. Ever listen to a radio broadcast talk show and notice that when no one is talking you hear noise and hiss coming through until someone talks? That's a compressor doing that. Radio stations, especially those with weak transmitters, pump the gain so they can get every ounce of volume out of FM radio's limited bandwidth. They know that the loudest channel will attract and keep more listeners than the ones at lower levels.

And the same is somewhat true of the music we buy and listen to. Top 40 music is always compressed, polished and buffed so when it comes across the radio or TV, even on tiny speakers, it's fully listenable and accessible.

3 Ways to Use a Compressor in your Studio

There are 3 places in the audio chain where compression can be used to enhance your work of art. They are the recording chain, the tracking chain and finally the mixdown chain. We'll spend a little time on each one.

The Recording Chain

Here the compressor is put on a direct out or insert of the mixer which takes the microphone signal after it is boosted by the preamp. Other methods are to place the compressor "in between" a mic preamp and an audio interface, or on the inserts of an audio interface or preamp.

The purpose here is to optimize the material for the recorder. You want to make sure all low volume passages actually do have a strong enough level where they won't bring in noise later, and you also want to stop and loud "peaks" from overloading the recorder's input, which will ruin the track. That is compressor theory 101.

However, there is a strong bias among those recording to computer sequencers not to record with with compression, but to record at 24 bits. The idea is that 24 bit audio offers such a significantly lower noise floor it is best to simply record at at full dynamics (louds and softs) at a level so low that the highest peak will never approach 0db fs. When you have the audio recorded as pristinely as possible, then you apply compression in the digital domain, usually, with a plugin.

Even recording to analog tape, or 16 bit files, you can decide to avoid compression while recording, if you are good at riding the gain or you have performers that understand how to position themselves with the mic. (That is, they back off a few feet before letting out the loud, and eat the mic when they whisper). However, the more out-of-control your performers are, the more likely you will need compression as you record. Its also true that some people like to record through compressors because they want to work that way. Finally, if you are recording live audio direct to a 2 track stereo feed, say, for live TV, you may simply have to have a whole lot of compressors working for you, particularly on the vocal channels.

There are many products specifically designed for the task of compression. If you see a mic preamp on a single channel compressor, these are designed for this part of the chain. Sometimes these are called vocal compressors. But like any other gear, you can use it for other uses too, such as guitars, acoustic instruments, etc.

ART Pro VLAII 2-Channel Compressor

The Tracking Chain

Once you have your audio tracks recorded on your computer or multi-track, you will be in the process of tweaking each track to make it sound the best it can, in reference to all the other tracks. Here the compressor is added as an insert on a mixer. That is, the signal goes out of the fader, goes through the compressor, then goes back to the fader's channel. If you recorded your vocals and acoustic instruments without compression, and you are mixing on an analog board, you almost certainly have to use one here to get the track up to spec. This can be done in the computer sequencer's mixer with a plugin, in the multi track if it has onboard compressors, or you do at at an analog board on inserts or busses. No matter how you mix, the idea is to get the tracks uniform, so you don't have instruments or vocals suddenly dropping out because they went soft on you.

You may also need to clamp down on those pesky peaks. Compression helps. If you have a single guitar note that peaks 15 db higher than the rest of the material, for example, your whole track will have to be mixed 15db down which will definitely put it in the background. The compressor, by clamping down on that peak, allows the whole guitar track to be boosted higher in the mix, where it can, at least, be heard.

A classic compressor such as the UA LA2A is a nice choice for a vocal track. it helps keep the vocal above the band in a very pleasing way. But let me tell you you won't be finding too many of these at your local pawn shop. That's to software modeling you can have an authentic replica of the LA2A on your sequencer track. Or you can get it in hardware

Universal Audio LA2A Classic Leveling Amplifier

Universal Audio now announces the rebirth of the Teletronix LA-2A, a Universal Audio Classics product. Painstaking care has been taken to ensure that every new LA-2A provides the performance and characteristics of the original. Each unit is hand built, each component carefully evaluated for authenticity. No expense has been spared to guarantee that this LA-2A will bring that classic sound to your recording. Demand the original. Accept no copy.

Compressors can also be used as effects in their own right on drum tracks. Drums are "peaky" by nature and by clamping down on the peaks, you can make the drums louder and fuller sounding. If you have ever heard any strong rock drums on the radio, you are hearing drums squashed down with compression and then boosted with volume. Drums without compression cannot hold up next to screaming vocals and distorted guitars. The same is true even for light jazz, where the engineer might only compress enough to tame the peaks, but not affect the transparency of the audio.

SM Pro Audio TB202 2-Channel Tube Microphone Preamp/Compressor

The Mixdown Chain

In the mix, a variety of compression techniques may be used. Compressors can be put on busses or even on sends and returns to affect (and effect!) certain parts of the mix.

An advanced mix technique is often called Parallel Compression, where the uncompressed source tracks are mixed in with the compressed signal coming back on a return or on a bus. The advantage here is that the compressor fattens the overall sound yet the peaks (which come from the source signal) remain clear and "on top" of the compressed signal. Parallel compression can work for drums and vocals, or anything really. It can also be done with groups of tracks.

But sometimes there is a temptation to put the compressor on the master bus (the main outs) particularly among newbies.

OK, a compressor may be added here too, and can have a dramatic affect, for better or worse. Some professionals advise against using compression here. Particularly if you are sending the mix to a mastering house for cd replication, let them use their gear. However, if this is a home cd production, you will have to master it yourself. But again, more cautions. See if your mastering software has any software tools for the finalizing task. Mix to wave without compression and use a mastering processor there. But if you are mixing down direct to a cd recorder or DAT and this is the last stop, then go ahead, compress the mix. If done properly, the whole thing will come out louder and stronger.

There are some exotic compressors like the Fairchild, which has been modeled by UAD and Waves, that is designed to be strapped on a 2 channel mix. These impart a character on the whole mix in a pleasing way. Universal Audio has released a hardware replica of their famous 1176 Limiting amplifier. You can also get an 1176 in software in the UAD2 system.

Universal Audio 2-1176 Twin Vintage Limiting Amplifier

Finally, a true stereo 1176! With over 2000 units already sold to date since the year 2000 re-birth of Universal Audio's 1176LN, it is clear the public's love affair with the legendary sound of the 1176 is still as passionate as ever.

Dynamic Range at Mixdown

This is a good thing, right? Not if you want to be loud. Dynamic range is the difference between the softest passage and the loudest passage in a song. Compression shrinks dynamic range. it makes the soft part louder and the loud part the loudest it can be. So. Got to be Loud? It's at this point where you would consider multi-band compressors, like the TC electronics Finalizer and brick wall limiters, like the Waves L2, or the UAD Precision limiter. These will let you use every bit of space in the audio bandwidth and you will be able to maintain consistent loudness. Because those writing top 40 hits all seem to do this, you may need to go this route if that's your bag. Want to be soft and loud? You might consider not using compression or just extremely light limiting at all at this stage and preserve the dynamics of the material. Orchestral and ambient works benefit from this approach as it makes for great dramatic passages when the orchestra does get loud. This is where some producers boast, "no compression was used". Of course, they are not aiming to get played on car radios around the globe.

TC Electronic C400XL Dual Gate Compressor

Tweak: A compressor with presets? You bet. For those that don't want to learn how to tweak a compressor, the TC C400XL offers presets to get you in the right ballpark.

Hardware vs. Software Compressors

Just a few notes here. Most modern sequencers have software compressors these days. These come in many different styles and types and many of them sound quite good. However, these are mainly for post-recording. You apply them to an audio track or soft synth as a plugin, after the recording has been made. Software compressors do not help as you record, so they cannot limit the peaks coming off the microphone through the preamp and into the converter. Hardware compressors, on the other hand, when setup correctly, modify the signal before it is recorded, thus preventing the overloads that can ruin a take. If you don't want to use a hardware compressor here you simply have to be careful about overloads. With 24 bit recording you can record at a lower level to avoid overloads, however, it is a great idea to have the protection of a hardware compressor all the same.

The UAD-1 Fairchild 670
Software Compressor is modeled after the famous Fairchild compressor designed for compressing the signal for vinyl LP. You can use it in your software sequencer and it can impart an interesting sheen over the mix. It is one of my favorites.

There are some great software compressors available today. As usual, the better ones will cost you. Take a look at the Universal Audio card which allows you to run software models of the classic 1176LN, LA-2A, Pultec EQP-1A, and Fairchild 670 hardware devices. Also check out the Waves Platinum compressors. But as always, use what you have first, get as much as you can out of them, and then consider upgrading. One of the more unique compressor bundles you can get today is the Focusrite Liquid Mix which is a combination of software and hardware DSP in a unified control surface. The package includes 20 different EQs and 40 classic compressors you can use right inside your sequencer, yet it uses a firewire connected DSP for the horsepower.

OK, you should be up to speed on what a compressor is and why, how and where they are used.

Focusrite Liquid Mix FireWire Mix Processor

Each of Liquid Mix’s 32 channels provides EQ and Compressor emulations selected from a huge pool of high-quality vintage and modern day classics. 20 EQs and 40 compressors

Source http://www.tweakheadz.com

EQ Principles

2011-04-09T08:48:00.000-07:00

What is an Equalizer?

An equalizer, or EQ, is a filter that allows you to adjust the volume level of a frequency, or range of frequencies, in an audio signal. In its simplest form, an EQ will let you turn the treble and bass up or down, allowing you to adjust the coloration of, let’s say, your car stereo or iPod. In pretty much any audio application—including live sound, recording, broadcast, and corporate installations—equalization is a sophisticated art and is critical to a good mix.

When used correctly, an equalizer can provide the impression of nearness or distance, “fatten” or “thin” a sound, and help blend or provide separation between similar sounds in a mix, allowing both to be heard as intended. It can also be used to adjust a sound system to account for the acoustical response of a room or an outdoor venue.

Parametric EQ

The parametric EQ and semi-parametric EQ are mainstays of recording and live sound because they offer continuous control over their parameters. These types of EQ offer continuous control over the audio signal’s frequency content, which is divided into several bands of frequencies (most commonly three to seven bands).

FIG 1

Both parametric and semi-parametric EQs typically provide control of the gain (boost/cut) for each frequency band, the center frequency of the midrange bands, and the cutoff frequency for the low and high bands (see Fig. 1). The difference between fully parametric and semi-parametric EQs typically is that the fully parametric EQ offers continuous control of the bandwidth, which determines the range of frequencies affected, or control over the Q, which is the ratio of the center frequency to the bandwidth (see sidebar “Who or What is Q?”). For most purposes, a Q control accomplishes the same thing as a bandwidth control but they are not identical. Most PreSonus EQs offer Q control; the exception is the EQ3B 3-band parametric EQ, which has a bandwidth control instead (see Fig. 2).

The EQ in the Eureka™ is a good example of a fully parametric hardware EQ, offering control of gain, center frequency, and Q for all of its three frequency bands. With more than three bands, you can get even more precise, as with the ProEQ plug-in for Studio One™ (see Fig. 3).

In a true semi-parametric EQ, the gain and frequency are adjustable but the Q and bandwidth are fixed at a preset value. A variation on the semi-parametric is the quasi-parametric EQ, which typically provides full frequency and gain adjustment but only two or three Q settings.

FIG 2

A good example of the difference can be seen by comparing the fully parametric EQ in the Fat Channel section of the StudioLive™ 24.4.2 digital mixer, which provides continuous Q control, with the quasi-

parametric Fat Channel EQ in the StudioLive 16.4.2 digital mixer, which offers a simple choice of high or low Q settings. We usually call the StudioLive 16.4.2 equalizer a “semi-parametric EQ” because that’s a much more common term and is accurate in a broad sense, but strictly speaking, it’s quasi-parametric.

Shelving EQ

FIG 3

A shelving EQ attenuates or boosts frequencies above or below a specified cutoff point. Shelving equalizers come in two different varieties: high-pass and low-pass. Low-pass shelving filters pass all frequencies below a specified cutoff frequency, while attenuating all the frequencies above the cutoff. A high-pass filter does the opposite, passing all frequencies above the specified cutoff frequency while attenuating everything below. Usually, the frequencies beyond the cutoff are rolled off, following a predetermined curve, not cut off sharply, as with a “brickwall” filter.

Some multiband parametric EQs offer low and high bands that can be switched to shelving filters. In others, such as the EQ in the Studio Channel, the low and high bands are shelving filters, while the mid band is fully parametric.

Graphic EQ

FIG 4

A graphic EQ typically consists of a bank of slider controls used to boost or cut fixed frequency bands (see Fig. 4). A well-designed graphic EQ creates an output frequency response that corresponds as closely as possible to the curve displayed graphically by the sliders. Designers of analog EQs must carefully choose the bandwidth of the filter and decide how the bandwidth should vary with gain and how the filters are summed or cascaded. In general, narrower bandwidth signifies a more precise EQ.

In general, most graphic EQs have between 7 and 31 bands. Professional sound-reinforcement graphic EQs generally have 31 bands, and the center frequency of each band is spaced 1/3 of an octave away from the center frequency of the adjacent bands, so that three bands (three sliders on the front panel) cover a combined bandwidth of one octave. Graphic EQs with half as many bands per octave are generally used when less precision is needed. You will often find this 2/3-octave design on monaural, 15-band or less graphic EQs in guitar amps, bass amps, and some stompboxes. In traditional graphic EQ designs, the center frequency of each band is fixed.

Some high-end digital graphic EQs offer greater precision. This is the approach PreSonus took with the StudioLive™ 16.4.2 mixer: the graphic EQ) is a pool of shelving filters from which coefficients like cutoff frequency, bandwidth, and gain are extracted through a process of curve-fitting. The curve entered by the user is first oversampled. The system then works with an internal curve made up of 128 bands to find coefficients for the first shelving filter that, when subtracted from the user’s curve, will produce the flattest possible response: 0 dB. The resulting response is then used to find coefficients for the second shelving filter, using the same optimization process. Coefficients for all available shelving filters are found through a recursive process.

Unlike conventional designs, the frequency and bandwidth of the “bands” depends on the curve entered by the user. This allows for much tighter matching of that curve. Because of this innovative design, the curve fitting-process is capable of very steep transitions, and unlike conventional, analog graphic EQs, what you see is what you get. With a carefully drawn, smooth curve, the StudioLive EQ will have almost no frequency ripple.

Graphic EQs are generally used to fine-tune the overall mix for a particular room. For instance, if you are mixing in a “dead” room, you may want to boost high frequencies and roll off some of the lows. If you are mixing in a “live” room, you might need to lower the high-midrange and highest frequencies. In general, you should not make drastic amplitude adjustments to any particular frequency bands. Instead, make smaller, incremental adjustments over a wider spectrum to round out your final mix. To assist you with these adjustments, here is an overview of which frequencies affect different sound characteristics:

Sub-Bass (16 Hz to 60 Hz). These very low bass frequencies are felt, rather than heard, as with freeway rumbling or an earthquake. These frequencies give your mix a sense of power even when they only occur occasionally. However, overemphasizing frequencies in this range will result in a muddy mix.

Bass (60 Hz to 250 Hz). Because this range contains the fundamental notes of the rhythm section, any EQ changes will affect the balance of your mix, making it fat or thin. Too much emphasis will make for a boomy mix.

Low Mids (250 Hz to 2 kHz). In general, you will want to emphasize the lower portion of this range and deemphasize the upper portion. Boosting the range from 250 Hz to 500 Hz will accent ambience in the studio and will add clarity to bass and lower frequency instruments. The range between 500 Hz and 2 kHz can make midrange instruments (guitar, snare, saxophone, etc.) “honky,” and too much boost between 1 kHz and 2 kHz can make your mix sound thin or “tinny.”

High Mids (2 kHz to 4 kHz). The attack portion of percussive and rhythm instruments occurs in this range. High mids are also responsible for the projection of midrange instruments.

Presence (4 kHz to 6 kHz). This frequency range is partly responsible for the clarity of a mix and provides a measure of control over the perception of distance. If you boost this frequency range, the mix will be perceived as closer to the listener. Attenuating around 5 kHz will make the mix sound further away but also more transparent.

Brilliance (6 kHz to 16 kHz). While this range controls the brilliance and clarity of your mix, boosting it too much can cause some clipping so keep an eye on your main meter.

How to Find the Best and Leave the Rest

In the “Making the Cut” section are some general frequency principles to guide you through the wonderful world of equalization but these are far from set in stone. So how do you find the best and worst each instrument has to offer and adjust them accordingly?

Here is a great starting place:

First, solo just the instrument with which you are working. Most engineers start building their mix with the drums and work from the bottom up (kick, snare, toms, high hat, overheads). Each instrument resonates the most in a specific frequency bandwidth, so if you are working on your kick drum mic, start with the lowest band of the EQ. Tune in the best-sounding low end and move on to the attack. It is not uncommon to hear an annoying ringing or a ‘“twang’” somewhere mixed in with your amazing-sounding low end and perfect attack, so your next task will be to find that offending frequency and notch it out. Once you are satisfied with your kick drum, mute it, and move on to the next instrument.

Taking your time with equalization is well worth the effort. Your mix will have better separation and more clarity when each instrument’s EQ is set so that it shines through the mix.

A few general words of wisdom:

You can only do so much. Not every instrument can or should have a full, rich low end and a sharp attack. If every instrument is EQ’d to have the same effect, it will lose its identity in the mix. Your goal is not individual perfection, it is collective perfection in the mix.
Step away from the mix. Your ears get fatigued just like the rest of you. If you are working particularly hard on one instrument, your ears will be quite literally numbed to that frequency range.
Your memory is not what you think it is. Comparing a flat EQ and the curve that you’ve created allows you to see exactly what you’ve done. So be honest with yourself. Sometimes that EQ setting you’ve been working on for 15 minutes is not the right choice, so move on.
Never be afraid to take a risk. The best EQ tricks were found by mad scientists of sound. “Playing” applies to engineers as well as musicians.

Source: http://www.presonus.com

How Does Vocal Pitch Correction Work?

2011-04-08T05:07:00.000-07:00

A vocal performance is a series of notes in different pitches. When holding a long note (word) that isn't supposed to change pitch, it should stay relatively around the middle of the key. If it stays in the middle then sharply goes up or down near the end of the note, that's when you hear an obvious off-key break.

When we edit pitch, we can bend the end of the note so it stays near the middle. Now it's perfectly in tune. We can also increase or decrease vibrato.

Sometimes singers run out of breath at the end of a line and those words are weaker, or sometimes a singer can't belt out certain words. In this case we can make the words 3-4db louder during vocal editing. We also cut down the volume on breaths that are too loud. We de's vocals too and can add effects.

When submitting a track, we need only the vocal track, no music. You can't edit vocals with music playing along with it.

I like to say it moves everyone up one grade. In some cases a grade and a half. If you give a "C" performance, adjusting pitch and vibrato can make it a "B" maybe "B+." The most important thing you must have is good time and some feeling in your vocals. Pitch and vibrato correction can do the rest.

Source: http://cdmusicmastering.com

Music Production and Mixing Tips & Tricks

2011-04-08T04:56:00.000-07:00

What makes a pro recording pro? What is the "sound" that the pros get and how can you make your recordings sound more professional?

The simple answer is - there's no simple answer. But with careful listening and a little experience you can create excellent results with modest equipment.

Good mixing starts ear

The first and most important item of equipment is - who knows? Anyone? It's your ears! Sorry to tell you this, but listening to ten hours of Rave at 110dB will do nothing for them and you might as well give your mix to a turtle as try to mix with misused ears.

Listen to commercial recordings of mixes you like, analyse them, listen for the effects and get to know what constitutes the sort of sound you're after.

Mixing secrets

There's no hidden secret to getting a good sound, but if we had to sum up the secret of mixing in two words it would be this - EQ and compression. Okay that's three words.

These are probably the two most important tools used by professional producers. However, like any tools, if you don't know how to use them you'll be carving Habitat tables instead of Chippendale chairs.

That's where your ears and experience come in. Here we have assembled some production ideas, suggestions, tips and tricks but they can only be guidelines and need to be adapted to suit your material. There are no presets you can switch in to make a bad recording sound good. And if your original material has been poorly recorded not even Abbey Road could salvage your mix. But follow these suggestions and see how much your mixes improve.

Get the level right

You can't push the levels when recording digitally as you can when recording to tape but you still want to get as much signal into the system as possible. This means watching the levels very carefully for clipping, and recording at an even and constant level.

Some recording software lets you monitor and set the input level from within. Some expect you to use the soundcard's mixer while others have no facility for internally adjusting the input level and expect you to set this at source.

Monitors

Your ears are only as good as the monitors they listen to. DO NOT expect to produce a good, pro mix on tiny computer speakers. It may sound fine on a computer system, but try it on a hi fi, in a disco and through a car stereo.

Oddly enough, you don't necessarily need the most expensive Mic. Many top artists use what some might call "average" Mics because they work well and get the job done. You can spend a wad on a large diaphragm capacitor Mic (yes, they're good for vocals) if you have the lolly but check out dynamic Mics which are much more affordable and can be turned to several tasks.

Mixing MIDI and audio

One of the great things about computer-based recording is that the parts can so easily be changed, edited and processed. It's also so easy to combine MIDI and audio tracks and many musicians use a combination of sample loops, MIDI parts and audio recording.

Audio recordings are generally guitar and acoustic instruments such as the sax and vocals. Incidentally, the best way to record guitars is by sticking a Mic in front of its speakers. You can DI them and process them later and this may be cleaner but for a natural guitar sound a Miced amp is hard to beat.

It's not necessary to record drums live and, in fact, it's difficult to do and retain a modern sound. You can buy off-the-shelf MIDI drum riffs and audio drum loops, or program your own. The quality of the gear which makes drum noises these days is such that anyone with a good riff can sound like a pro.

Mixing MIDI

As MIDI and audio parts appear on the same screen in modern sequencers, it's very easy to arrange them into a song. However, when you come to mix everything down there's another consideration. If you are recording to DAT you can simply route the audio and MIDI outputs through a mixer and into the DAT machine.

However, if you want to create a CD you must first convert the MIDI parts to audio data. The entire song can then be mixed to hard disk and burned to CD. Converting MIDI to audio can have another benefit and that's the ability to process the MIDI tracks using digital effects.

Effects

There are three positions for effects known as Master, Send and Insert. Use the Master for effects you want to apply to the entire mix. These will often be EQ, compression and reverb.

Although giving each channel its own Insert effects is kinda neat, each one uses a corresponding amount of CPU power. So if your computer is struggling and if you're using the same effect on more than one channel, make the effect a Send effect and route those channels to it.

Many pieces of software let you apply an effect Pre or Post fader. With Post fader, the amount of sound sent to the effect is controlled by the fader. With Pre fader, the total volume level of the signal is sent. Post fader is the usual default and the one you'll use the most.

EQ

EQ is the most popular and the most over-used effect. Yes, it can be used to try to "fix a mix" but you can't make a silk purse out of a sow's ear as me Gran used to say and what she didn't know about mixing could be written in the margin of the book of honest politicians.

But before you start messing with EQ - or any other effect for that matter - make sure you have a decent set of speakers. Have we said that already? Oh, must be important, then.

There are plug-in effects such as MaxxBass which can psychoacoustically enhance the bass frequencies to make it sound better on smaller speakers. However, this is by no means the same as getting a good bass sound in the first place by observing good recording principles.

EQ can enhance a mix to add gloss, fairy dust, shimmer, sheen, a sweetener or whatever you want to call it to the final production. It can be done with enhancers and spectralisers, too, although these tend to mess with the harmonics which some producers don't like. However, don't dismiss them out of hand.

General EQ lore says that you should cut rather than boost. If a sound is top-heavy, the temptation is to boost the mid and bass ranges. But then what usually happens is you start boosting the upper range to compensate and you simply end up boosting everything and you're back where you started - only louder!

The reason why cutting is preferred is that boosting also boosts the noise in the signal which is not what you want. Try it. Boost every frequency and listen to the result. If you think it sounds okay, fine. What do we know?

But when you're fiddling, do keep an eye on the output meter. Boosting EQ inevitably means increasing the gain and it's so-o-o-o easy to clip the output causing distortion which does not sound good.

Finally, check EQ changes to single tracks while playing back the entire piece. In other words, listen to the tracks in context with all the other tracks. It may sound fine in isolation but some frequencies may overlap onto other tracks making the piece frequency rich in some places and frequency poor in others.

Reverb

Reverb creates space. It gives the impression that a sound was recorded in a hall or canyon instead of the broom cupboard. Recording lore suggests that you record everything dry, with no reverb, so you can experiment with a choice later on. You can't un-reverb a track once it's been recorded.

The more reverb you apply, the further away sound will seem. To make a vocal up-front, use only enough reverb to take away the dryness. Vocals don't want to be mushy (lyrics can be mushy) so use a bright reverb.

A common novice error is to swamp everything with different types of reverb. Don't - it sounds horrible!

Mixing down

You've done all the recordings, done the edits, applied the effects and now it's time to mix everything into a Big Number One Hit! Before you do, go home and have a good night's sleep. Have two. In fact, sleep for a week.

Yes, we know you're hot and raring to go but your ears are tired. They're falling asleep. Listen carefully and you might hear then snore!

There is a phenomenon known as ear fatigue and consistent exposure to sound, especially the same frequencies, makes our ears less responsive to them. Goes back to the bit about spending your life in a Rave club - you'll never be a master producer. If you try to mix after spending a day arranging, your ears will not be as responsive, so do them and your mix a favour by waiting at least a day.

Source: http://www.articlealley.com

The Sound Spectrum & Analysis Tools

2011-04-07T06:48:00.000-07:00

Have you ever wondered why you can listen to one mix and hear every part and instrument clearly, but listen to another and struggle to figure out what's going on? Well, wonder no more ...

Understanding the Audio Spectrum

You probably already know that sound is made of a spectrum. The limits of human hearing in this spectrum run from about 40 Hz at the lower end to about 16,000 Hz at the upper end. At the lower and higher extremes the sound may actually be felt rather than heard, but we don't need to go into that just now.
You might already know this, but do you know how to make use of this knowledge so as to make it work for you in your mixing? The chances are you may never even have thought about it.
Different regions of this spectrum exhibit different characteristics. There's no absolutely arbitrary way of identifying precise boundaries, but the categories shown below are generally agreed to be there or thereabouts. Along with each category is listed its main characteristics.
Bass: from about 40 Hz to 200 Hz: Booming, full, solid
Low Mids: from about 200 Hz to 800 Hz: Body, fatness, fullness, warmth
Mids: from about 800 Hz to 5,000 Hz: Clear, present, forward
Highs: from about 5,000 Hz to 8,000 Hz: Bright, alive, brilliant
Ultra Highs: above about 8,000 Hz: Crisp, radiant sparkling
The way in which the sounds produced by different instruments use different permutations of these frequencies is one of the aspects which determines the sound of the instrument. One issue that you will need to address in mixing is how to prevent your different instruments (and voices) from fighting each other for ownership of the same frequencies.

Spectral Analysis Tools

There are a number of tools available to help you to do this. One category is the Spectral Analyzer. It is important to understand that unlike the variety of plug-ins that you already own - EQ, Delay, Reverb, Compressor, etc. - this type of plug-in does not in any shape, fashion or form alter or affect the audio signal to which it is applied. It merely gathers information about this audio signal and passes it back to you.
Two examples of such a plug-in are illustrated. These are the MultiInspectorFree and the Roger Nichols Digital Inspector. Both of these area available free. The freeware version of MultiInspector lets you compare the signals from up to three different tracks at the same time (the illustration shows two). There is also a more powerful version available for purchase that lets you compare up to 16 tracks at one time.
These tools can be useful many ways. One such use is to help you in identifying areas where you may need to resolve conflict between different track.
Consider our example of the use of MultiInspector. The illustration on the left shows this plug-in applied to analyse a vocal harmony between two voices. You can observe, for example, that there are some frequencies (for example around 250 Hz) where one voice will clearly cut through above the other. However, there are other areas (particularly from about 1,200 Hz to about 3,000 Hz) where they seem to be locked in mortal combat.
In our next article we'll look at what methods we have at our disposal for resolving this, and how to apply the EQ tool.

Source: http://www.brighthub.com

Tips on Setting Up Your Own Home Recording Studio

2011-04-07T06:46:00.000-07:00

Sound Design

Setting up your own home recording studio seems a lot more practical in today's world of social networking, independent record labels, and home audio mixing software. With a certain amount of home materials, computer equipment, microphones, and associated technology you can actually create a fairly sufficient home recording studio without having to use professional money and studio space. If you are really looking at what setting up your own home recording studio means it requires you first identifying what purposes you want, what type of home audio mixing software you want to use, how much money you are willing to spend, and how much space you are willing to give up. Here is a look at a few tips on setting up your own home recording studio.

Software and Computer

The computer, and the home audio mixing software you use, is going to be the primary focus for setting up your own home recording studio. This is going to process your sound design at all levels, and the rest of the sound equipment is simply to facilitate the capture of the sound. To do this effectively you need to take a look at the home audio mixing software that you want to use, which should dictate the technical requirements of your computer. Pro Tools, for example, dictates a certain computer requirement as well as a hardware component until the most recent incarnation. If you are buying the software and the computer at the same time make them congruent as this is going to set your limits on how in depth your sound design will be.

The Room

If you are setting up your own home recording studio for recording music tracks then you are going to have to have sufficient space that is designed to facilitate performance. Essentially this means that an entire medium to large room must be isolated just for the audio recording studio, which will also require creating a furniture infrastructure that will facilitate musical and recording equipment as well as soundproofing on the walls. Beyond the computer this will be the most significant and expensive part of setting up your own home recording studio, but without this kind of dedication you will not be able to record clean music or work professionally. This is really just the bare minimum, and it would be even better to try to remodel two rooms so that performance can happen in one while the technical recording and mixing process can happen in another. This will give a cleaner monitoring of the sound and allow the person doing the sound design to have a better conception of the audio mix.

Microphones

Many electronic instruments will plug directly into the recording interface, but acoustic instruments and voices will need high quality microphones to facilitate the recording. There are a lot of different options for this, but what you should remember is that the microphones used in recording in your home studio are going to be a lot different than what you want from microphones used for live performance or recording. Sensitivity and clarity should be the main focus of your studio microphones, so you may want to avoid condenser microphones in general and this may be one of the times when a ribbon microphone is a good choice.

Source: http://www.brighthub.com

Vocal Tuning & Pitch Correction 101

2011-04-07T06:38:00.000-07:00

Tuning & Pitch Correction

In the year 2011, the engineer’s ability to take an “off note” and bump it to a correct note is a well known fact. Made most popular by the use of Antares Auto-Tune by artists such as Daft Punk, Cher, and T-Pain, Auto-Tune has become a commonly understood (sometimes mis-understood) concept and has even shown up as an iPhone app.

Tuning software has been around for well over a decade, and has been used in records in far more transparent ways for awhile now. In fact, it’s been a lifesaver to many a recording. Sometimes you get that magical take (or only one take) that just has that slightly off moment. Tuning gives the engineer the choice to leave the take natural, or make it “perfect.”

Pitch Correction vs. Pitch Shifting

Pitch Correction is not simply finding the intended pitch and gluing the off note to it. That’s called Pitch Shifting. While Shifting and Correction have a lot of things in common – there is one fundamental difference. Instruments, particularly the human voice, have harmonic signatures. These signatures in conjunction with overtone patterns allow us to identify when an instrument is a guitar, a flute, a sine wave, or a voice. The voice in particular is manually shaped with various harmonic signatures, called Formants, that yield vowel sounds. Our natural overtone pattern, and the resonance shaping we make with our glottis, mouth shape, and nasal cavity come together to form our unique sound. When we sing higher or lower notes, some characteristics change, but our Formants (our harmonic signature) actually stays the same. After all, just because one note is lower than the other doesn’t mean our mouth or nose changes dimension in order to produce the lower note.

Pitch Correction accounts for this harmonic signature, these formants, and rearranges our sound to the desired pitch while preserving the harmonic shape.

How It Works

This part is pretty complicated, and the most technical bits are still outside of my understanding. However, I’ll lay down the basic info. Pitch Correction uses a variant of phase based vocoding. So, when people call Pitch Correction software “the vocoder” – they aren’t totally incorrect. Pitch Correction is essentially a very specific vocoder.

Phase Vocoding sounds complicated. It is and it isn’t. At the most basic level a vocoder isn’t terribly complex. It literally “codes voice.” Our voice has different amplitude shapes at different frequencies as we pronounce words (particularly with vowel sounds). The vocoder reads the incoming volume levels at different frequency bands and figures out the shapes. These shapes control a set of frequency filters in the vocoder which are then applied to a different signal. While we most commonly think of this as being done by a compuer, it can actually be accomplished in the analog world relatively simply. But, the more frequency bands and filters used, the more accurate the vocoding will be – and computers allow for extremely high numbers of bands and filters to work simultaneously.

Here’s the complicated part. In traditional vocoding, the voice may act as the modifying control for a simple wave shape or sound. In Pitch Correction, the voice essentially acts as the controller for a Pitch Shifted version of itself. Not only that, but the pitch shifted version is kept at the same or a similar time boundary as the original. This is done through mathematical algorithms that involve Fourier transformation and re-synthesis to get down to the basic structure of the sound, and re-create it at a different pitch without changing the rhythm – or making “intelligent” time changes that keep it as close as possible with minimal artifacts.

The bottom line: Pitch Correction changes the pitch, but keeps the harmonic signature of the original (or as close to it as possible).

Inside the Tuner

There are several tuning programs on the market, the three most common being Antares Auto-Tune, Waves Tune, and Celemony Melodyne. All three have a unique sound to them that can be most noted when the functions are set to do the maximum amount of tuning within the smallest increments of time – meaning the tuned sound stays unnaturally locked on pitch. Even though they are all doing essentially the same thing, the make-up of the formant encoder, the quality of the filters, the intelligence of the pitch shifter (choosing how to best change pitch with minimal effects to time), and the accuracy of the programs parameters, all amount to somewhat different sounds.

Auto-Tune – To be fair, my experience with Auto-Tune is more limited than my experience with Waves Tune or Melodyne. My impression has been that it is generally the most heavy handed of the three programs. Even when set lightly, it still seems to impart a coloring onto the sound. That being said, it can be a nice color – almost like an exciter. It’s also very easy to use. Automatic mode works very well the moment it’s turned on, and graphical mode is fairly intuitive when you want more control.

Waves Tune – Waves Tune is to my ear, the most easily transparent. It’s designed with transparency in mind. This is probably not the software to use if you want some kind of excited effect, though it can still be done. There is a mild dulling of the sound at times, in fact. It’s also less CPU intensive. If I were to take a blind guess, I would say there’s simply wider frequency bands being used in the encoding and re-synthesizing process. The controls are extremely intuitive, and allow for very detailed access to the sound with little fuss.

Melodyne – Melodyne is generally my choice of the three. It can be very transparent or very non-transparent, and the ways it can do it are more varied. That being said, it’s not the most intuitive – it takes a while to really master all of it’s uses. It’s probably not the best choice for a quick nip & tuck on a small moment – but for a sound where serious work is involved, you have more options, control, and great sound quality. What I especially like in Melodyne is that it allows for independent formant pitching, in addition to pitching the root tone itself.

Tools & Terms

Skillful use of Pitch Correction requires a bit more expertise than just figuring out the key and running the process. There are in fact some fine controls that can be used for subtlety, to add some slight distortion or excitement, or totally turn the sound into a robo-synth version of itself (and a number of ways to do the latter). Now, in each program these things are controlled and labeled differently, so I’m going to make up some of my own words here – this isn’t technical terminology, but I think it will help explain the process.

Tightness – There is generally a control that allows you to determine how much you want the sound source to stay locked on pitch. In Auto-Tune this is controlled by “Re-tune Speed”. In Waves Tune this is again controlled by a “Speed” control, but is also heavily influenced by the “Note Transition” function. In Melodyne there is “Pitch Modulation” which controls vibrato shaped pitch variance, and “Pitch Drift” which controls non-periodic pitch variance. Vibrato functions around the idea of the center pitch, whereas drift does not contain a pitch around which it modulates.

Transitions – In addition to how closely the sound source sticks to a given pitch, there’s also the action of transition from one note to another. As unrealistic as it is for something to not modulate around a pitch center – it’s just as unrealistic to have no slide between notes on fretless instruments and especially a voice (at least on a legato line). As long as the phrase is continuous there will be some degree of glissando between the notes. In Waves Tune you have a “Note Transition” function which allows for more natural or more forceful transitions – measured in milliseconds. In Auto-Tune, source sound is broken into notes and curves. The “curves” are the transitional points, and can be manually stretched and bent. In Melodyne, note transitions are controlled by an angle setting that kicks in when using the pitch altering handle and placing the cursor at the edge of a note segment.

Formant Control – A key element to all three is the ability to correct for formant displacement that would occur with regular pitch shifting. All three programs use different algorithms that can be modified by the pitch range of the sound source. While Waves Tune simply offers an on/off control and voicing range (bass, tenor, alto, soprano), Auto-Tune and Melodyne both offer some more interesting options. Auto-Tune comes with a “Throat” control which allows you to change the width of harmonic resonance that would occur from having a smaller or larger throat size. This algorithm is used in other Antares software, and can create the illusion of a persons head being larger or smaller! Melodyne has a more practical and interesting control – an independent harmonic shifter for the formants themselves. One of my favorite uses of Melodyne is not actually pitch correction, but lifting the formants as a way of exciting a vocal. This is good for people who’s voice could use a little excitement, or for a thinner voice to sound a bit breathier by shifting the formants down. Mind you, formant shifting does not actually change the pitch – it changes the placement of the harmonic bands where signature formants are occurring.

Draw Function - One of the cool things about Auto-Tune and Waves Tune is that you can actually draw the pitches you want to produce. This allows for pitch automation in a way that is otherwise unprecedented. The one drawback of Melodyne is that the company has rejected the implementation of this very useful feature – though separating notes and using more conventional pitch altering tools can certainly get you far (just not as quickly).

Unique Features – Different programs have embraced unique feature sets in anticipation of end user requirements. Melodyne can act as a rudimentary DAW, allows for advanced rhythmic correction and amplitude shifting – in case you want to rework a sound source in any way. Auto-Tune has a Vibrato creator that can allow you to put in pitch variance that didn’t previously exist, as well as a ‘Humanize’ function that varies the re-tune action based on the duration of the note being effected. In addition Auto-Tune can transpose as a traditional pitch shifter (no formant correction), while acting as a pitch corrector (with formant correction) in the same pass. Waves Tune, well, Waves Tune is just really fast and easy to use (and also allows for creating artificial vibrato). For the really ambitious user, notes can also be triggered by MIDI.

Techniques

There’s a number of techniques one can use as a mixer, producer, or editor when it comes to pitch correction. I’ll share a few of the things I’ve done and hopefully you will comment below and share some of the things you’ve done.

First, there’s really two applications of pitch correction.

As a subtle way to lock in an off-pitch in a performance
As a special effect to make someone or something sound like a robot from Cydonia.

To be subtle, it helps to understand that pitch is the perception of frequency. Meaning, to be on pitch, one does not have to actually be exactly on the frequency – the sound just needs to center around that frequency. You really don’t need to be heavy handed with the tightness (speed). A little nudge will at least lock the sound in enough to perhaps be “pitchy” for a moment, but not “off pitch.” For a natural performance, pitchy is not the end of the world. Also, don’t rely on the pitch corrector as your only tool in the arsenal. If there’s a really out moment, you might be able to clip a snippet of a similar note or phrase from elsewhere in the song and paste it in, or cut out the off moment and time stretch the moment before it to reconnect the sound. It’s better to use a variety of approaches in subtle ways then to rely heavily on just one technique. Also, listen closely to how legato phrased notes slide together – they shouldn’t sound like distinct glisses, but also shouldn’t sound like sudden jumps either – unless it was done purposefully or is a distinct part of the performer’s style.

For a natural performance, really assess the player’s/vocalist’s vibrato. String players and vocalists who are classically trained but still developing as musicians tend to be a little less controlled when it comes to vibrato – or they go the other way and get too stiff. A long sustained note generally falls into a vibrato as the muscles of the throat and diaphragm tire. If the vibrato is swinging more than a quarter tone in either direction, I would tuck it in a bit. If you have a performer or singer who is just super controlled and doesn’t waiver a drop – give it just a hint of quiver and see if that opens up the moment a bit.

Now, for the fun stuff. Here’s a few fun ways to mangle up a vocal and turn it into something weird – while straying from being totally cliche.

Fun Stuff!

Let’s say you want that “tuned” effect on a vocal, but you still want the vocalist to sound like a human being. The trick here is to have natural note transitions, but unnatural pitch tightness. Use graphic mode to make sure your vocalists voice moves like a human being’s, but sustains robo style.

Or, do the opposite – hard tune the transitions, but keep the actual sustain of the voice loose. This will add just a glimmer of synthy harmonic distortion – just something slightly surreal.

Here’s a fun one I learned from Chris Athens. Copy a track and take the cadences at the end of each phrase. Tune those cadences up to a different note that is still relevant to the chord the vocalist is on. Now your vocalist is harmonizing with her/himself. Tuck that harmony vocal way down, or mute it’s output but send the signal off to the main vocal’s reverb/delay. This will reinforce the implied harmonies and add a little excitement.

One thing I like to do with strings when working with a dance or modern hip-hop track is to hard tune them, sometimes twice, even using different tuning programs. Strings have a natural waver that occurs from the push of the bow changing the tension on the string. It happens really subtly and really fast, so it’s not really heard, it’s just part of the sound. But with pitch correctors you can take that waver out. You’ve made the natural string now sound like the best synth string patch ever heard. This can also be effective with horn instruments. Antares Auto-Tune has a specific “instrument” mode that you can experiment with, and I’ve heard great results from even tuning a fretless bass on an electronic style dance track to get that “perfect” pitch.

One last fun one I recommend playing with: Copy the track you are working with. Hard tune one track to the pitch, but go a couple cents sharp (just one or two). On the copy, hard tune to a couple of cents flat. Set at equal volumes and pan slightly apart. Suddenly your sound is very wide. It’s a unique chorus-type effect. The more they overlap, the more of the “swishies” you will get from the two versions going in and out of phase with each other – which can be cool. The more they are apart the exponentially wider the vocal will stretch, because those quick phase conflicts will trick the ear when they come from different directions.

Conclusion

Since pitch correction became popular, there has been a divergent rise in the appreciation, as well as distaste, for tuning. Pitch correction, like any other tool, is all about what you do with it. So please share some of your thoughts and techniques in the comments section below, and keep checking in for more great articles.

Source: http://theproaudiofiles.com/

Mixing 101

2011-04-07T06:34:00.000-07:00

There they are...8/16/24/48/96 tracks that you’ve painstakingly recorded, overdubbed, erased, and recorded some more.... so now what? You can’t play with your mixer’s knobs every time you want to hear your masterpiece! You need to blend all those tracks to (usually) a 2-channel, stereo mix. In other words, mixdown your tracks.

Great --- so what does THAT mean? Well... first - you need something to mixdown onto - a computer, a stand-alone hard disk recorder (such as the ubiquitous Alesis Masterlink), a 2-channel reel-to-reel, a HiFi VHS machine, a MiniDisc recorder, or even a cassette deck -- all are usable options (some more usable than others!) The pros and cons of each format are beyond the scope of this article, but suffice it to say, use the highest quality recorder your budget will allow.

I'm also going to have to presume you already have a decent monitoring chain in place. You can't make good sonic decisions if you can't properly hear what your tracks sound like! And it IS true - headphones are not considered a "decent monitoring chain." While they are useful for double-checking your mix, and proofing it for noise or artifacts that don't show up on the monitors - they are horribly ineffective as a primary mixing tool.

You also want to pay attention to the volume of your monitors as you mix. There are many different preference engineers take to levels, but it is commonly accepted that 85dB SPL is where human hearing frequency response is most flat, and this is typically where many engineers leave their levels at... it is good practice though, to vary listening levels while mixing to get a feel for the mix balance at different volumes. Mixing too loud almost always results in unbalanced mixes, mixing at softer levels usually produce more balanced results.

Next?
Mental preparation......... rest your ears, rest yourself... it’s virtually impossible to mix right after a tracking session, or after going to a concert.... you need your ears to be in top form for analyzing and objective & critical listening... what works for me is starting fresh in the morning, when my ears are most rested....

OK - I’m rested, my ears are rested -- let’s do this thing...........

Well.... throw up the faders and see what you have to work with.... yes, ALL of them.... what you want to listen for are the tracks that are working, and the tracks that aren’t. Yeah, the guitars are fighting during the choruses, the background vocals aren’t working midway through the verses, the lead solo starts too early.... and of course, you’re taking notes for yourself as you do this...

The plan here is to take out the stuff that’s not working together, and leave in the stuff that IS working.... there are a couple of ways to do this: 1) use EQ to tailor the portion of sonic spectrum a track will fit in, and 2) mute the track at the problem spots.

Let’s talk about muting....that’s right, you pull the track right out of there! If it doesn’t help the song it doesn’t belong - period. Mixdown is the time to be blatantly critical of every track that’s been put down... you don’t reduce the volume, you don’t bury it behind something else - this simply results in messy mixes -- it either works, or it doesn’t......... you will accomplish muting either by automation (via software or mixer hardware), or the old-fashioned follow-the-timelog-with-your-hands-on-the-mute-button technique. You don’t have to mute the track completely either - you can add interest by pulling it in and out of the mix at key spots (obviously not at the places where it’s causing mix difficulties!)

"But I like the track that doesn’t work!"
Let’s be honest... sometimes the track works, but is simply frequency-fighting with another track.... in this case, you probably don’t want to mute the track, but you need to shape it so that it doesn’t interfere with another track’s piece of the sonic pie. So....... what you reach for are the EQ knobs....

First, some EQ background.... it’s not cut-and-dry -- there are various ways to use EQ, and EQ decisions you may have made during tracking will affect the mix process.

The addition of EQ into the signal chain always results in "some" compromise of the waveform by introducing phase-shifts (time-based artifacts that can results in comb-filtering of the waveform) - especially when boosting frequencies. Cutting frequencies can result in less of these artifacts, so it is generally advisable to apply EQ by cutting frequencies you don't want, rather than boosting the ones you want to enhance - a practice known as subtractive EQ. The quality of the EQ itself also dictates the artifacts - cheap EQ gear means more artifacts, mastering-grade EQ means significantly less (for comparison, a Weiss EQ-1 used by mastering houses runs about $5500 US)...

So, using EQ to shape sound is a bit of a compromise - yes, it changes the signal, but it introduces "some" small signal degradation.... the obvious solution is don't use it--- er, at least, not until absolutely necessary. "But wait...", you say, "I need it, my guitar/bass/drums don't have that sound..."

Well ok... for line-level instruments such as synths, you certainly can use EQ to shape the sound.... but for mic'd sources, it's much better to use mic selection and mic placement to get the sound you're after, rather than reaching for the EQ knobs.... for example, don't brighten an amp by boosting your hi-shelf EQ - change amp settings, change guitar pickups, change amps, move that mic closer to the center of the cone... if you're not getting the sound you want, maybe you're using the wrong instrument/amp combination!

Another point to keep in mind - try to get your tracks sounding the way you want during tracking - if the tracks "to tape" are sounding the way you want them, then selecting sounds during overdubs become much easier. And even better, during the mixing phase, you'll find your tracks will blend better (since you've already blended them correctly in the tracking process!) Best guideline to follow: never "fix it in the mix" - fix it now - move a mic, change the mic, change the source, move the source, switch rooms. If none of these work, then reach for EQ!

During mixing - if you've done your tracking homework, there should be less work needed in getting the tracks to fit, since you've taken so much care during the tracking process. But very likely, there are still some tweaks you'd want to make.... I strongly suggest you adopt the subtractive EQ approach - cut instead of boost. If there are too few highs, remove some mids or bass to shape it. This does two things - minimizes phase-related artifacts, and more importantly, reduces unnecessary signal level that will eat into your mixer's headroom (since cutting will reduce the amount of frequency "space" a waveform will take up.)

Well - that’s the basic EQ theory... so now we’re going to shape the tracks of our mix with EQ. To do this you do have to have some concept of the audio picture you’re about to paint, within two frames of reference 1) the various frequencies of the tracks, and where they sit; and 2) the placement of the tracks in the soundstage in front of you (between the left and right speakers).

The charts below will give you some indication of the frequency ranges for various sound sources that will help guide your use of EQ.

1/3 Octave Frequency Charts **

Audio Octave Ranges

Frequency Range	When Used Produces This Effect	When Used Too Much Produces This Effect
16Hz to 60 Hz	Sense of power, felt more than heard	makes music muddy
60Hz to 250Hz	Fundamentals of rhythm section, EQing can change musical balance making it fat or thin	makes music boomy
250Hz to 2KHz	Low order harmonics of most musical instruments	telephone quality to music 500 to 1KHz horn-like, 1K to 2KHz tinny, listening fatigue
2KHz to 4KHz	Speech Recognition	3KHz listening fatigue, lisping quality, "m:, "v", "b" indistinguishable
4KHz to 6KHz	Clarity and definition of voices and instruments, makes music seem closer to listener, adding 6db at 5KHz makes entire mix seem 3db louder	sibilance on vocals
6KHz to 16KHz	Brilliance and clarity of sounds	sibilance, harshness on vocals

Key Frequencies For Instruments

Instrument	Key Frequencies
Bass Guitar	Attack or pluck is increased at 700 or 1KHz; Bottom added at 60 or 80Hz; string noise at 2.5KHz
Bass Drum	Slap at 2.5KHz; Bottom at 60 or 80Hz
Snare Drum	Fatness at 240Hz; Crispness at 1 to 2.5KHz; Bottom at 60 or 80 Hz
Hi-Hat and Cymbals	Shimmer at 7.5 to 10KHz; Klang or gong sound at about 200Hz
Toms	attack at 5KHz; Fullness at 240Hz
Floor Toms	attack at 5KHz; Fullness at 80 or 240Hz
Electric Guitar	Body at 240Hz; Clarity at 2.5KHz
Acoustic Guitar	Body at 240Hz; Clarity at 2.5KHz; Bottom at 80 or 120Hz
Piano	Bass at 80 or 120Hz; Presence at 2.5 to 5 KHz; Crispness at 10KHz; Honky-tonk sound at 2.5KHz as bandwidth is narrowed; Resonance at 40 to 60Hz
Horns	Fullness at 120 or 240Hz; Shrill at 2.5 or 5KHz
Voice	Fullness at 120Hz; Boominess at 200 to 240Hz; Presence at 5KHz; Sibilance at 2.5KHz; Air at 12 to 15 KHz
Harmonica	Fat at 240Hz, bite at 3 to 5KHz
Conga	Resonant ring at 200 to 240Hz; Presence and slap at 5KHz

These charts, however, don't tell you the whole story.... the two frames of references I mentioned earlier are related in terms of their effect on the sonic soundscape of a mix.

Think of a mix as a 3-dimensional space in front of you... you have control over the left/right, the high/low, and the front/back of the sound stage. The tools that let you manipulate this area are Panning (for left/right positioning), EQ (for high/low positioning), and Fader Level (for front/back positioning).

Homing in on EQ for the moment, keep in mind that as you shape your tracks, higher-frequency tracks will appear to come from higher up in the monitors than lower-frequency material. This can be useful in positioning guitar tracks - if a guitar track is fighting with something else in the mix, you can "move it away" from the offending track by removing some bass content in conjunction with panning.

As for Panning - it alone can also be used to separate mix elements into distinct positions in the mix. For example, panning a keyboard rhythm part off to one side while panning a complimentary rhythm guitar part to the other will result in a pleasant, wide and full rhythm section whose elements don't interfere with each other. When using panning, it is often helpful to envision a music stage in front of you, and place the tracks within that space as you would normally hear at a concert. You may not keep the tracks in this position as you build-up and further define your mix, but it does make a useful starting point.

And finally, Level -- faders allow you to control the level (and thus how close or how far away the source is) of the track in the mix. No tricks here except that you really shouldn't use level to hide a track -- if the track doesn't work, simply mute it..... you want a track to be more "in your face" move it closer to you, you want it to sound as if it's further away (like towards the back of the stage!), lower the level.

An extremely important point to remember is to maintain your relative levels so that you don't eat away at your mixer's headroom -- if you've set all the levels of the tracks and find you have to push your solo track level very high for it to cut through, then you've got all your other track's faders set far too high. Unless you've got 6-digit consoles that are more forgiving of "level-pushing", most mixers will start sounding pretty harsh if pushed too hard.

But wait a minute! What about reverbs, delays, and all those cool other effects -- you didn't mention a thing???
Well yes... effects play an important role in mixing - much like spices do in fine cuisine. It's all in the way you use them... and the topic merits its own full-blown article (maybe in the future), but for the moment, here's a brief overview.

My own approach is that unless a particular effect is an integral part of a track's sound (such as a chorus, or wah on a guitar part), I leave all effects until the mix stage. The tracks really should stand on their own merit without any effects added-in, then you make the mix bigger, fatter, wider, with careful and judicious use of things like reverb, delay, choruses, etc.... little touches such as timing delay and chorus settings to the tempo of a track really do a lot to make the tracks shine. Overuse of effects results in muddy, poorly-defined mixes, so much like EQ - less is usually more. As well, pay particular attention to reverb used for vocals... for most listeners, vocals are the component of the song that reach the people first, and poor effects, or bad EQ is immediately noticeable -- it's not uncommon spending hours on reverb selection for the lead vocal track!

One effect that is very misunderstood (and often poorly used) is compression -- this effect can be an important part of a polished sound both during the tracking AND mixing stages of the production process, again, if used properly. A colleague of mine wrote an excellent article (with examples) on the use of compression... check out Moshe Wohl's Description of Compressing/Limiting for some great notes on the subject.

So we've tamed our tracks with EQ, balanced fader levels, set up some nice ambient effects, you've made your muting/adjustment notes and everything is sounding great... this is the "work" part of the mix -- the previous tweaking was the "fun" stuff, now - depending on how many mutes, volume, pan and EQ adjustments you have to make (if you have mix automation, it's a no-brainer!), you'll want to rehearse your mix a few times before hitting Record on the mixdown unit.

In this process you treat the mixing console like an instrument - following your notes as you go along. At 01:10:25 drop level to -5db on track 3, 01:40:30 mute track 6, you get the idea... The point is to become familiar enough with the mix moves you have to make so that you're not looking at your notes so much that you miss something. For complex mixes, get your bandmates/family/girlfriend to help you with an extra pair of hands! Once you've got the mix moves down smoothly, fire up the mixdown recorder!

A quick note on mix levels -- it's a very good idea to keep your levels strong and balanced throughout the mixing process. Having to turn up the recording level on the mixdown unit because you've got the levels too low on your mixer usually results in unnecessary noise and possibly distortion if the levels are extremely diverse. Remember that all gear has an optimal signal level range and your best results are obtained by staying within it. Know your gear and what's it's capable of handling in terms of low-level noise and high-level distortion. In addition, use the meters as a guide, but always let your ears make the final decision -- you hear with your ears, not your eyes!

So.......... once you've got that mix take down and you like the way it sounds, you're done, right??? Wrong...... you've only just started! You can, of course, stop there - but you very likely won't be pleased with the result.... you've spent all that time and energy getting balances and levels right, only to find that when you play it in the kitchen boombox, it sounds very different. The fact is - you do have to learn how to translate your mixes from your monitoring system to other systems, and as well, learn what works for a mix everywhere, and what doesn't.... and there's no way to describe it - you have to learn by experience. What this means is you will follow this approach of performing a mix, checking the sound on various sound systems, coming back and readjusting/rebalancing some mix elements, performing another mix, etc... for a few iterations until you've honed it to a point where the mix sounds reasonably good on most systems.

Take heart though - as you learn mix translation with your system, it does become easier - and with enough practice, you'll develop a feeling for what works and what doesn't.

The art of mixing is very broad...
There are various mixing guidelines for all the many styles of music (you don't mix a big band song the way you would a pop song); and within each mix style there are commonly accepted practices. What I've tried to describe in this article was to provide an overview of some basic mixing practices that are common to all mixing styles, as well as some functional tips regarding one of the most abused mixing practices - EQ.

The best way to approach mixing is to read as much as you can about common approaches, while at the same time, practicing the techniques you've read about and adapting them to suit your own needs.

Some recommended reading related to the art of mixing are:

Modern Recording Techniques - Huber & Runstein

Mastering Audio - The Art and the Science - Bob Katz

Behind The Glass - Howard Massey
Behind The Glass Vol. 2 - Howard Massey

The Mixing Engineer's Handbook - Bobby Owsinski

The Recording Engineer's Handbook - Bobby Owsinski

The Art of Digital Audio - John Watkinson

Source: http://www.bluebearsound.com/

The Perfect Mix

2011-04-07T06:28:00.000-07:00

There are many ways to get your songs to final form. What matters is not how you get there, but that you do get there. Lets pretend you are enrolled in one of the world's fine universities and you are writing a Master's Thesis. This is not just "any" piece of drudge paperwork, but the culmination of you education. You know you have to write in excellent form, have to watch out for tiny grammatical imperfections, and make sure substance and style flows well. In short, you have to rewrite and edit, a lot. It may take several experiments to get this just right. You might be working for weeks, not going out to the clubs with your buds, even sending hopeful significant others away. Why? The darn paper is important--you have to do well!

Apply that same attitude to your mix and you will have a great mix. Tweak's axiom: The value underlying successful production is the same in all fields--art, architecture, music, quantum mechanics, even political science and business. Beauty has a tone. Its not a tone you hear with your ears or see with your eyes but that you realize on reflection. (That is, when you stand back and ask "what is this?"). When you sense the passion of the creator coming at you from the work of art they made for you, you begin to sense the piece at hand is great.

Lets assume, for this article, final form means a beautifully polished piece of music in 16 bit 44.1 khz digital audio (i.e., the "red book" cd audio standard) or a standard .wav or .aif file, perhaps at a higher resolution for later mastering. You need to start, of course, with a fully or almost finished song. This is the point where the writing ends and the TweakMeistering begins. I'm going to give you some hard earned tips on Mixing and Mastering in the old analog style.

Mixdown and Mastering, traditionally speaking, are two very separate processes. Mixdown is the art of leveling, equalizing and effecting all the various sources from many tracks down to a stereo Mix. Mastering is the process of taking the stereo mix and putting it in the final album-ready form. Recent software and hardware developments make these processes easier and less expensive than they ever have been in the history of making music. Given that much of the time we can stay in the digital domain we can add processing to our heart's content and maintain a high signal to noise ratio and achieve optimum dynamics for the piece at hand.

The Mix Process

Every mix is different. In reality, there are no formulas. None! But there are hundreds of "practices" a professional mixologist will do without even thinking about them. But I know where you are at and what you need. You need a map to get you started, and a flow of working. That is what this article is about. Please consider these parameters not as rules but a starting point for you mixes for the standard pop song or ballad using an analog mixer. (We will cover mixing in the sequencer in the next class) Of course the instruments change if you are doing techno or symphonies, or ambient stuff, but the reference may still be helpful.

Step one is always to calibrate the mixer however you can. You could use a test tone of 0dbVU (that's LOUD, so turn down the monitors). First, set each fader at the 0db marking on the board. When you apply the test tone, turn up the trim until the meter on each channel pegs at 0db. If you don't have a tone to use take the loudest sound that the channel does during the mix, Set the trims so that when this peak occurs, the meter pegs at 0db. Do this for every channel in the mixer. This gives you a reference. A zero dbVU signal (or your loudest signal on the track) should meter at zero db when the fader is at zero db. When you move your fader to -10 db, the meter should peg at -10db. Now you know what those numbers are for that are silk screened on your mixer! Do It!

Note: If you don't have meters on every channel then you have to use the main meters on the mixer for this. If you don't have a "solo-in-place" function on your mixer, you will have to mute every channel except the one you are calibrating. Yes, it takes time to do this, but it is well worth it.

Match the following instruments when soloed in place to the db markers on your mixing desk or your mixdown deck or software.

Kick drum 0db Eq to taste. No FX except maybe subtle ambience You will tweak the kick again, this is just to get you going. In an instrumental piece, the kick is the first and last tweaked. It's got to be just right.

Tip: If using a live drummer, you need to stop the kick drum from resonating too much. A pillow inside the drum may help. If you have an excessively ringing kick drum, you can add a gate as an insert to damp it.

Snare -2 db eq to taste in the frequencies above 4khz. Add reverb if the song calls for it. Do the best you can to keep it out of the way of the vocal, even if you have to pan it a few degrees. Near the end of the mix you need to come back here to perfect it.

Lead Vocal 0db use a low cut filter to eliminate rumble and plosive pops around 100-200 hz. Carefully enhance the delicate high end around 15khz to add air and sheen and don't overdo it! This is the trickiest adjustment and may often spell hit or dud. Perfectly center the vocal and, if this is a stereo track, pan it not with pan controls, but with very subtle left/right hi freq eq's. Put on the cans (headphones) and make sure its in the absolute center of your forehead.. Every word must be intelligible. Add reverb and delays but don't let it get smeared. Before you print to tape or DAT or whatever, check the vocal any make those tiny adjustments that are needed.

Cool trick: Split the main vocal track to two seperate faders. Compress the main vocal and send the secondary, uncompressed vocal to a reverb unit. This way the reverb stays out of the way until the vocalist gets loud. Hey that's they way it works in real life.

Note: It is often quite wise to use mono tracks for vocals simply because they they will stay centered better than stereo tracks, and are impervious to phasing anomalies that may occur with stereo tracks.

Cymbals -25 db Avoid letting these get in the way of the vocals. Pan them to 2 o'clock and remember their main function is to add the glue to a track to hold the music together--they do not have to be loud or present. Think about how horrible they will sound on your girlfriend's or boyfriend's car stereo if you let then get too loud. Remember, loud cymbals can wreck a whole mix.

Tip: Never let the drummer in the control room, except under extreme sedation, unless you want all your mixes to sound like Led Zepplin.

Synth pads -20 db Do these in stereo and hard pan left and right with generous effects if needed. However, keep them in the back. Pads indeed are beautiful additions to a song but don't let them overshadow any of the main elements of the song. Yet for a sense of dimensionality, let these create a "landscape" the listener can walk on.

Cool trick--you want a really BIG Pad? Delay one side of the Left/Right by about 10-12 microseconds. You'll be hearing a landscape if you do it right. Don't let any engineer tell you these have to be mono. Make him earn his pay by fighting the phase issues. Wassat? All you do is do a mono check on the mix and make sure the stereo pad didn't disappear.

Bass -10 db maybe hotter Always front and center. If you use FX restrict yourself to chorusing or a light flange--no reverb. Note that the quality we associate with "good" music is a tight syncopation of kick drum and bass. If you hear any duff notes make sure you fix them.

Cool trick: Bass does not have to hit exactly on the kick drum. But it a wee bit after so the listener hears the kick 1st. Do microseconds count? Yep. Ears are really good at detecting even tiny, tiny delays in what we hear. Are there more secrets in the micro-timing domain? Yer catchin' on dude--good work!

Big Bad Tip: Keep the bass and kick out of the way by giving each a different EQ. If the kick peaks at 65 HZ make sure the bass peaks somewhere else. You can use a spectrum analyzer to see where the loudest frequencies are for each.

Rhythm guitar -15 db pan off center eq: use a low cut filter to get rid of any bass and add a mid range eq for a slight narrow boost, but make sure it is not competing with the vocalist's sweet spot.

Hot tip: Bass KILLS, remember that. Get rid of ANY bass frequencies you don't absolutely have to have. "B-b--b-ut" you sputter, "my guitar now thounds like thiiit" Want cheese with your whine? Try it, the mix will sound better. Kill all the upper bass mud you can on any instrument you can do it on. These muddy frequencies around 250-400HZ build up fast and are a sure sign of an inexperienced mixologist.

Percussion -20db- put these elements off center unless they are essential to to basic beat. EQ in a tasteful way if necessary. I shoot to get a little skin sound on the hand drums if possible.

The Mix itself

Now, watch the meters when you play the whole mix through the board. On an analog board you should have peaks at no more than +3db. If what you have is more notch down every fader in 1 db increments until you get there. Shoot for 0db for the whole mix. Now because we put the kick and vocal at 0dbVU, when all the instruments are added the final level might be quite high, like +7db. So now we notch down every fader 7dbVU. When you are done with this exercise, you should see your whole mix peaking at 0dbVU. You still have headroom on your analog mixer. You want to get the signal in the mixer's "sweet spot". Now you can start nudging things a bit higher, a bit lower. You should have a sense of what the song is asking you to do.

Mono Check: Always check you mix in Mono and look for sudden drop outs or instruments that disappear. That's phase cancellation at work, and it happens with stereo tracks and effects.

No faders above 0dbVU rule: Remember, we calibrated the board so the loudest sound of each track pegged at 0db (or we used a test tone) and the board markers represent 0db. Never move your fader over that mark. That's right. Never. Cutting a signal is fine, go as low as you have to, but never add gain at the fader (unless you have an ultra premium board that can do this). If you follow this you can make a great mix even on a cheap $200 mixer. Just pretend that 10db of boost each channel has available does not exist and don't go there. If you find your vocal doesn't sound good unless its at +5db then move everything down 5 db. Conserve headroom. You don't want your mix compromised by that awful crackle at the peak of your song.

Side Note: When people say "Brand X's Mixer sounds like crap" its nearly always because they don't know how to mix and added too much gain. Yes, inexpensive mixers don't add gain well, but they pass through a signal without gain perfectly and are able to subtract gain better than they can add it. Even $4,000 mixers have this issue. There is only one place where gain should be added--at the preamp's trim knob--and only add as much as you need, never more. Every other pathway should either let it pass through or subtract gain. Be really stingy about adding gain.

Now you fine tune to taste. Listen for the quality to "lock". There is a definite point where this happens. Suddenly it all falls into place, given you have good material. A great mix of a great song will fill you with absolute elation. You'll be blown away and in awe. You will feel in love with it. No kidding. Might sound corny to the less mature among us, but I assure you its true. A great artist friend of mine puts it this way. Greatness in art depends solely on how much love you put in to a work. You put it in, it pays you back, your friends back, and everyone who listens. Moral of this lesson. Never take mixing and mastering lightly. The tiniest fader movements make a difference. Be exacting!

The Mix is a Dynamic, Moving Process

Assuming you are doing a real time mix, don't just sit there while your mix goes to tape, or hard disk, or DAT. If you are using a board, assign the faders to groups. For example, if you have 4 subgroups you might want to send your vocal tracks to groups 1 and 2 and everything else to 3 and 4. This way you can slightly alter the balance between the vocalists and the band as the piece goes to tape. This technique, while tricky, can yield outstanding results. You can give the vocalist a touch more edge just when they need that oomph and when the vocalist takes a break you can subtly boost the band a bit. If you have 8 busses you might dedicate 5 and 6 just to drums and 7 and 8 just to effects, nudging each as is appropriate. If you have a digital mixer, this is where you want to automate.

The Role of Compression at Mixdown

First of all, if you plan to have your material professionally mastered, don't add compression at mixdown. A professional mastering engineer will have a better compressor than you do and they cannot remove the layer of compression you add. Just get the mix sounding great without compression, record the mix so it's top peak is several db below 0db. Let them make it louder, that's their job.

But if you are not sending the piece off for mastering, and aren't going to add a pass later through mastering processors, then, yes, patch in the compressor at mixdown or do a separate pass later with the mixed file.

On it's way to the recording device, you can patch a compressor/ limiter/gate. The Gate simply cuts out any audio below a certain threshold so that any hiss or noise coming from your synths or mixer is eliminated before the music starts. The limiter keeps your peaks under a certain fixed level and will not let them go higher. A Compressor is a volume slope applied to the audio material going through it. It can amplify the "valleys" and attenuate the "peaks". Essentially compression reduces the dynamic range we have just struggle to achieve in our mix. You might wonder why you would want that. In many circumstances, you don't want it. However, in the majority of cases you will find it useful, especially if you want your music to be "hot", "have punch" "be as loud as possible", or have the consistency of a radio mix. The stereo compressor also helps balance the song and give it a uniform character we are so used to hearing in commercial music. It essentially gives you the strongest and smoothest mix and calms down some of the 'jaggged edges' that might disturb the casual listener. However, it is also very easy to make a mix totally lifeless with a compressor and reduce its dynamic power. What started as a powerful orchestral arrangement can end up a wimpy piece of Mall Muzak so be careful and bypass it frequently to make sure you like what you are tweaking up. I think compression works well to attenuate that occasional peak that rips through the roof of a digital audio recorder and ruins the track. Also if you have the cash for a fine analog tube compressor. or even a high quality compressor plugin, there is lots of magic you can do at this stage.

The Role of the Software/Hardware Mastering processor

Hardware mastering processors are becoming less popular, now that there are software models of classic compressors and eqs. We will cover those in one of the next classes. Yet the hardware processors are serious tools and are particularly useful for a hardware based recording studio with an analog mixer. If you have one, you might consider using that in lieu of a compressor at mixdown as mastering processors usually have all the functions and additional functions such as mastering eq, multi-band compression as well as compressors, limiters and gates. These mastering tools can go a long way to giving your music a unique sonic imprint. There are many uses. In addition to adding the refining touch to your mix as it goes to the recorder, it can be used to give all your songs on an album a consistent uniform character and balance the volume between widely different songs giving your project a professional touch.

Using narrow band mid range eqs can give you a very contemporary sounding presence and make your dance tracks come alive with freshness. Pumping the compressor a little at 50-60hz can give you the "kick in the chest" kik drum without wrecking the delicate dynamics of the high end vocals. There are many more applications such as using them to send midi tracks to your digital audio mixer compressed optimally, ducking for voice overs, de-essing, warming through "tape saturation" parameters and Hard Gate effects on individual tracks. Remember Tweakheadz rule of thumb: Any piece of gear can be used in any way as long as it enhances the quality of the final product.

Software Mastering and Post-Production

A good digital audio sequencer will let you master in the digital domain of your computer. ou can do it in any digital audio application that lets you add plugin processors. Its a good idea to use one of the major sequencers that has mix automation and you can automate your way to you master just as you did with your mix. Volume automation: The main thing is to be able to draw a volume envelope over the whole waveform. Rather than botch a fade 20 times on an analog mixer, simply draw in the perfect fade with the mouse. Where the piece loses intensity, notch it up a tad, to restore your intended dynamism to your mix. Splicing and Crossfading: Say you have the perfect mix except for one horrible "sp-p-p-lat" where your sequencer choked at bar 72. No prob. Just remix the offending bar again, cut out that piece in your sequencer and drop in the new one and let the automatic crossfading give you the absolutely perfect, digitally calculated crossfaded splice. Works! Need to touch up the EQ and do your compression in software? Tweak it in. It's all undoable, so your not going to ruin anything. Decided the mix you did last year really sux? You need to cut out a chorus or fade 5 seconds earlier? Say you did a trance piece but the kick is so wimp that it makes you cringe? Just drag in a looped 808 kik and paint it on the next track, setting the volume and compression to make the whole song whupass. :) Your sequencer has the tools. Its just a matter of knowing the right mouseclicks.

The Final Touch

You've worked hard on a song, maybe several weeks, maybe longer. Its now in final form, just a matter of the last transfer to DAT, Tape, Wave or CD. Here we enter into the subtlest, but arguably, most far reaching of your tweaks. Sometimes it makes sense to compare the quality of masters to metals. Do you want a mix of raw iron? Steel? Silver? Gold? Of course we choose gold for most things. Gold is firm, strong, yet soft, malleable, pleasing. This takes you right to the heart of the digital vs. analog controversy. And you no doubt have heard the advice "use your ears!". And perhaps you've heard of engineers said to have "golden ears", indeed a point of much pomp and egosity in the field. What does the golden eared producer have that you don't? Listen close now, here's a secret, your reward for reading so far. What they have is an aural image in their minds of how things can sound beautiful, and they have the gear that allows them to get the audio to that place in their heads.
Think about that OK? It's like religion. The believers all see a truth that is obvious that no one else can. Is your audio like velvet? Silk? Or is it more like uncomfortable rayon, or dull like flannel or harsh like sandpaper.

The final touch is never easy. You are also fighting with "the audience in your head" on how something should sound. Finally, you have been working on it so long you might NOT be hearing what it really is as your brain is conditioned to filter what you want to hear. If you can't nail it by the 3rd or maybe 4th play in a session, can it for the rest of the day. Bring everything up to spec as close as you can and come back tomorrow. The most important factor in the final touch is not gear; it's the interaction between your ear and your mind. Yet having good gear at this stage helps your ear and mind "find" that doorway to quality, where you blow yourself away into sonic ecstasy, and your final master communicates that to everyone who hears it. This, my friends is the "holy grail" of audio. It's where a song becomes more than a song, it's an adventure, a voyage, a statement. I wish you happy journeys.

Summing Up

Whether you are writing industrial hardcore or the darkest ambient, a 100 piece orchestra or a stark minimalist a capella mix, always keep your ears tuned to making an artistic statement, a work of unforgettable beauty. This is the bottom line. The more control your Mixer gives you, the better you can paint the overall image. Working with compressors and mastering processors gives you a shot a polishing that image much like we polish a stone to bring out its colors. Hope this article helped you get a handle on the concepts of the perfect Mix, mastering and post-production, and the Final Touch.

Source: http://www.tweakheadz.com/