The Virtual Recording Studio

If you are just starting out, you might find our recording primers helpful.

Digital Recording Primer
Digital Recording 101

Source->Room->Capture->Amplify->Convert->Save->Mix->Convert->Speakers->Room

Introduction:

Disclaimer
This tutorial is only a collection of the stuff I think I have figured out by doing some recordings. I'm sure I got some if it wrong. It is not meant to recommend any particular product, nor any particular way of doing things. Digital recording is a complicated subject, and I am aware that some of the generalities in this tutorial will not be true in all cases. But I had to start somewhere. You will find exceptions to many of the descriptions I have given here. Please do not email me to tell me about products that are exceptions - I already know. All I am trying to do here is give the reader a place to start their own investigation and understanding of digital recording.

I cannot be held responsible for any decisions you make regarding your own recording. You assume all risk if you continue to read this.

No warranty is expressed or implied...

Your Mileage May Vary...

My recommendation is that you take all of this worth a grain of salt...

I'd recommend you don't even read any further...

Turn Back!

Caveat Emptor - Use at your own risk.

Click HERE to search Google for Digital Recording

Why another recording primer:

I have been a musician for... Well, a long time.

Probably, like yourself, ever since I wrote my first song, I hoped I would get an opportunity to record my music for others to hear, and hopefully enjoy.

When the time came time to make my first record, I was excited, but quickly discovered that it was an expensive proposition to record my music in even a small project studio. I was ill prepared for the cost of my mistakes: In my arrangements, my performances, my production decisions, and even in the way I used studio time as a practice time. I did not anticipate that every production mistake was going to cost me money, and I just did not realize that making a record was going to be so different from performing live.

Chapters:

Other Useful Links:

Note: This is NOT an endorsement of these sites, nor any of their products. However, browsing through the sites below may give you even more information to help you understand digital recording, and recording in general.

After I burned through half of my budget, with little to show for it, I thought that maybe I could save some money by doing some of it myself. The studio that I was recording in was entirely digital, and the engineer was very knowledgeable. He took time to show me what he was doing, and how I could accomplish some of the same things at home.

I will always be indebted to him for his generosity - especially since I spent a lot of time asking him about rather mundane recording concepts, and bothered him quite a bit when I couldn't understand some of the problems I was having.

I made a lot of mistakes, and spent a lot more money learning how to record than I should have. But I learned quite a bit. When I asked my friend the engineer how good my record would be when we finished - he wisely suggested I look at my first attempt as a "learning experience". And it was.

During that first project, I spent a lot of time trying to find someone to explain some very basic things about digital recording to me. This tutorial is a small attempt at helping the new recordist get up to speed on terms, ideas, and concepts that they will run into when setting up a new home studio.

I am no expert by any stretch. I am not even really a very good recordist - but I did learn some things that I think would help others.

I hope this tutorial will help you to reach that goal of recording music that you have always dreamed of - and I hope this tutorial will help you avoid some of the mistakes I made.

The Source:

In the words of Glinda the good witch of the north: "It's always best to start at the beginning". In this case the beginning would be the source of our recording.

The source of any musical performance is the artist.

Any song that is poorly written or poorly performed will not sound good, no matter how great the gear is. A multitude of sins can be covered by great gear, and a talented engineer. But bad music is bad music, whether it's recorded well or not.

When a young artist comes to me to record their music. I ask myself a few questions.

Is the song well written, does it say something, does it evoke emotion? Does the artist perform the song well? Is the artist's instrument a high quality instrument, and does the artist use the instrument well? If the instrument is not good quality, or the artist does not use the instrument well, is that appropriate for the song? For instance, a beat up buzzing guitar might be just the right instrument for a down to earth blues song. The same instrument would not be appropriate for a baroque guitar piece. These things influence the recording as much or more than any recording gear decision.

With this in mind, I try to do my best to capture the artist. It is, after all, not my job to determine whether the song should be recorded or not, but rather to capture the song and the artist truly.

The Recording Room

Does the room sound good?

It is likely that you will be recording some live instruments. Whether it will be a singer, drummer, guitarist, bassist, or other performer. The performance will likely be recorded with a microphone (or microphones) placed in a room, to capture the performance. The properties of that room will influence the final sound that the microphone captures.

To understand how the room affects the captured performance, we must first understand a few basics about sound.

Sound travels in waves. Each wave has a speed, amplitude, wavelength, and frequency. I will not cover these individual properties of a sound wave, but every individual sound in your recording has these properties.

Sound waves, or "waveforms" also have some very interesting behaviors. One behavior is that when two waveforms with the same properties reach the microphone, an increase in loudness is perceived by the device. This is called "Constructive Interference".

Sound Wave 1	Sound Wave 2
Combined Sound Waves

A second (more interesting) behavior is that when two waveforms reach a microphone, that are exact opposites of each other they cancel each other out and no sound is recorded. This is called "Destructive Interference" or is sometimes referred to as "180 degrees out of phase". You might also hear someone refer to it as a "phasing problem".

This concept warrants some further investigation. Here is how it works: If we say that a waveform completes it's cycle every 360 Degrees, then we can also say that the first half of the waveform (0 to 180 degrees). will be an exact opposite of it's second half (180 to 360 degrees).

If we take two occurances of the same waveform, and compare their first two cycles, which would run from 0 to 360 degrees(Cycle 1) to 0 to 360 degrees(Cycle 2). We can say that a waveform of 0 to 180 to 360 degrees (the first complete cycle) will be an exact opposite of the 180 to 360 to 180 waveform. In other words, a delay equal to one half of the waveform (180 degrees), when overlaid with the original waveform will cancel the original waveform out.

Sound Wave 1	Sound Wave 2 (Offset by 180 degrees)
Combined Sound Waves (Cancellation)

But what might cause such a delay?

One of the most common causes of phase cancellation is when two microphones are placed in a room to record a performance. The sound wave(s) will reach each microphone at a different time. When both tracks are then mixed together, it can cause phasing problems. To overcome this issue, you can follow the "3-to-1 rule". I will leave this as an exercise for the reader to explore.

Waveforms can also bounce, or "reflect" off of the many surfaces (walls, tables, mic stands, etc...) in a room, causing "reflections". These reflections can bounce off of one or more surfaces, multiple times, back to the microphone, which will record the sum of all these waveforms as they are received by the microphone.

This is important, let me repeat it again: The microphone will record the SUM of all the waveforms.

The reflection process causes the waveform to reach the microphone with a decay and a delay. Depending on the room, and the location and type of microphone, the microphone may receive a single reflection or multiple reflections of the same sound.

When these delayed, reflected sounds are received by the microphone, they combine with the direct sound from the performer, and are recorded as "Natural Reverb". This is the property that gives a listener the perception that they may be listening to something recorded in a cavern, a basketball court, a large hall, or a small room.

This is why the room is important, and why the location of the mic (called "mic placement") is critical. Moving a microphone just a few inches (to find the "sweet spot") can make the difference between a good capture of the instrument, and a great one.

When recording in a new location, I ask questions like: Does the room have too much natural reverb? Does the room allow the fundamental and overtone frequencies to be captured in a way that makes the performance sound alive? Or does the room add-to or remove-from the performance?

Fundamental Digression
You might now be asking - What is a fundamental? What is an overtone?

A fundamental is the basic sound of the note that is played. The "overtones" or "harmonics" of a note are the other frequencies that the instrument generates (they are mathematical multiples of the fundamental). Overtones are the thing that give an instrument it's "timbre" (pronounced Tam-Ber) or sound. All instruments generate fundamentals and overtones.

A "good sounding" room, allows all of those frequencies to be captured well. The subject of room design is well beyond this tutorial (and my understanding), but information can be found at other web sites that discuss the math involved in designing a "good sounding" room.

Typically a room should be "dead". Meaning, it should not reflect a lot of sound. This is because it gives the engineer more choices later, when mixing the song. Let's say an engineer decides a certain vocal should have some reverb, and this engineer records the vocal with some reverb. Later on the artist decides that they do not like the reverb, or it is too much reverb. At that point nothing can be done (well, it can, but it usually doesn't work very well). If the engineer had recorded the vocal "dry" (without any effects), and then applied the reverb to the track at a later time, He could have gone back to the original track and re-applied less reverb to the vocal. This is why a "dry" version of a performance is often recorded with the "wet" (with effects applied) version of the track.

A room can be treated in many ways to allow for better recording. This can be everything from completely rebuilding the room with special construction and materials, to just installing some bass-traps and gluing some auralex to the wall. Bass-traps and Auralex are simply foam materials that absorb sound and prevent it from being reflected back. The subject of which materials should be used, and where they should be applied can be found on many other web sites that delve into this issue in greater detail. I will leave this investigation up to the reader.

Another question I ask myself is: Does the room have a good vibe? Does the room encourage the performance? Both in the way the room sounds, and in the way the room feels. Does the artist feel good in the room, or nervous? What emotions are being communicated in the performance? Does the room help the artist tap into those emotions?

Things like throw rugs, beanbag chairs and tapestries can improve not only the sound of a room (by cutting down or diffusing the reflections), but it's vibe also. Bobble-heads, Lava lamps, and other wacky props can add a sense of fun and life to a room. Which can, in turn, make the artist feel more relaxed and natural.

A Philosophical Digression
As I mentioned above. We measure waveforms from 0 to 360 degrees. That might look familiar to you. It is also how we measure a circle. And in fact waveforms are usually represented 2 dimensionally as a graph of points on a rotating circle as it travels through time.

Think of a dot painted on a bicycle tire. If we graph that dot as the bicycle travels through time, we get the same sinusoidal wave. In the physical world however, sound waves actually travel in a three dimensional plane, in an expanding sphere of pressure from the source.

Philosophically it is interesting to me, to ponder the relationship of sound to the cosmological arrow of time, and to space, energy, motion, lines, circles, and spheres.

For those that are interested, there are other web sites that illustrate and explain this much better than I ever could. And even though it will not improve your digital recordings, I do find it fascinating.

The Capture:

The performance will be transmitted from the artist (in the room) to a capturing device. This may be a pickup on a guitar, a microphone, or some other device that converts the performance into an electronic format.

Since the capturing device is really the first place where the performance is received to be converted into an electronic representation, it will have a significant impact on what is passed further downstream in the recording process.

For instance, even though the performance does not change, a dynamic microphone will create a different sonic picture of the performance than a condenser microphone, even if they are placed in the exact same spot.

Questions I ask myself about this part of the recording chain are: Is this capture device high quality? Does it capture the entire sonic palette of the performance? Is it a device that is honest, or is it a device that enhances the sonic qualities (timbre) of the performance? Is it neutral or colored? Does it introduce any noise to the recording? Does it capture the ambiance of the performance or strictly the performance itself (how sensitive is the device)?

The Capture:

The Amplification:

Very likely, the capturing device will need to be connected to another device to boost the captured signal, so that it can be properly recorded. This may be a direct out from an amplifier or it may be a microphone pre-amp. It may be a chain of several different amplifiers and pre-amps, or a chain of different pre-amps and capturing devices.

One common example would be a guitar, plugged into an overdrive preamp, connected to a guitar amplifier, that is close-mic'ed, connected to a mic preamp. Each Capturing device and amplifier in the "chain" will have influence on the sound of the recording.

Some questions I ask myself about the amplifier(s) in the chain are: Is the amplifier/pre-amp neutral or colored? Does it have other processing options to change the performance, such as an EQ, compressor, or effects? Can you mult the signal to record both the processed and unprocessed versions of the performance? Is the artists looking for a certain "sound" that only a specific model of amplifier or pre-amp can deliver?

One unique device that falls into this category is a "modeler". This is a device that pretends to be something else. For instance, rather than connect up a real Marshall stack (weighing several hundred pounds), a small modeler weighing a couple of pounds can emulate (model) a Marshall stack. modelers cannot exactly duplicate the sound of the specific amplifier they model, but they can approximate the sound very closely. They do not "push air", that is they do not, usually, connect to a speaker cabinet. They are usually DI (Direct In), and thus cannot really reproduce the authentic sound exactly.

The advantage to modeling (or to any DI recording) is that it is very quiet, because it is usually done with only headphones for monitoring. This is especially important to home recordists living in apartments who may be recording heavy distorted guitars in the middle of the night.

In some cases, a modeler can affect the performance itself. This is because some artists have an organic "conversation" with their instruments. They play off of the reaction they get from the amplifier, and that influences their performance. For instance, for a heavy metal guitarist, nothing can compare to standing in front of an eight foot stack, feeling the speakers' air push back at you, and the heavy bottom end in your chest. For this reason some artists prefer the actual amplifier to a modeler.

The Conversion:

The performance thus far has been converted from a physical event (singing, playing) to an electronic analog signal. The next step is to take that analog signal and convert it again, into a digital signal. This is called A/D conversion. A/D conversion is done via a computer chip in either the amplifier/pre-amp or in the sound card. In higher end studios it is usually done with a special purpose unit called, simply, a "converter", or an "A to D converter". Conversion is one of the most overlooked yet most important parts of the signal chain.

Digital audio data is "sampled". That is: It is not a continuous analog stream. Digital audio is created by saving thousands of snapshots of the sound per second. The number of samples per second and the amount of information stored per sample, greatly influences the sound quality. For instance, 24bit samples, at 96Khz (96 Thousand samples per second) will have much higher fidelity than 16 bits sampled at 44.1Khz. Due to the inherent mathematical qualities of sampling, a higher bit depth is usually more desirable than a higher sample rate.

It is also important to note that not all converters are created equal. A very high quality converter at 20Bits 48Khz, may actually have higher fidelity than a poor converter at 24Bits 96Khz. The quality of the converter is influenced by the algorithms used in the converter chip, by the number of bits in the sample, the sample rate, and by the "Clock".

The "Clock" or "Word Clock" is used to ensure that the samples are taken at accurate and consistent times. The more accurate the clock, the more accurate the samples will be, and thus the higher fidelity of the audio.

When recording I ask myself: Do I have good converters? Do I have a good clock? How many bits? What is the sample rate?

A Conversion Digression
As I mentioned above, digital converters (A/D and D/A) are usually built into your soundcard. They can also be external boxes (your soundcard may support a combination of both). You might decide you want to connect an external converter to increase the number of inputs to your DAW, or to improve the fidelity of your recordings, by installing a higher quality converter than the one built into your soundcard. External converters may or may not include their own pre-amps.

An external converter will need to interface to your soundcard. There are many interfaces used to connect external converters to a soundcard. The two most popular are S/PDIF and "lightpipe". S/PDIF and lightpipe are both digital interfaces that are used on many soundcards. S/PDIF is a two channel (stereo) interface, while lightpipe is an 8 (or 16) channel interface. Ensure that your soundcard can accept the same interface that the converter uses, including the same bit depth and sample rate (for example: 8 channel lightpipe, 24 bits at 96Khz).

When you shop for a sound card, this will be an important consideration. If your soundcard has two analog inputs and supports two S/PDIF inputs, the largest number of discreet inputs you will have is 6 inputs. If you anticipate needing at least 16 inputs, make sure that your sound card can accept a few simultaneous lightpipe inputs.

Saving The Data:

Once the audio data has finally passed through all of these stages, it can be stored on the computer's hard drive. The Hard drive does not influence sound quality. At this point, the data is already stored as 1's and 0's and the hard drive cannot influence the data in any appreciable way.

That is not to say that the hard drive is not important.

A poor quality hard drive can crash, or can have other problems that make your stored performance unusable. A high quality hard drive will pay for itself in reliability and in time that is not spent trying to recover a lost performance.

Hard Drive Data Digression
I have seen a hard drive crash completely destroy a recording session. In this particular case, the computer crashed, and the "session file" was corrupted. The session file is the file that compiles all of your takes, and/or punch-ins into each individual track. A single track can have a dozen or more different takes and punch-ins, interspersed with silence. These pieces of audio are called "regions". They are all placed on a timeline to playback that track. If you have a few dozen tracks in the session, you can easily have several hundred regions of audio that are organized into a session. If the session file becomes corrupt, it is nearly impossible to reconstitute the hundreds of audio snippets back into a usable session. It is a very irritating occurrence.

Multiple Regions in a track

For this reason I make it a habit to create an additional, single audio file for each of my tracks, by consolidating the regions into single complete tracks. I do this at the end of each session. I like to keep the original files to work with, and back up the consolidated tracks with the rest of the session, just in case I need them. This process also works well when I want to transfer tracks from one brand of DAW software to another.

Consolidated to a single region in a track

One other factor that influences your digital data is the speed of the hard drive. There are many factors that influence the speed of a hard drive (seek time, rotational speed, latency, transfer rate). These are outside the scope of this document and are technical in nature. Suffice to say that you should buy the fastest hard drive you can afford. A faster hard drive will allow you to mix more tracks. A slow hard drive may not be able to keep up with 15, 20, 30 or more tracks when you are mixing. It is very annoying to have to bounce several tracks down to one track because your hard drive cannot keep up.

All the files on your computer, including your audio regions are stored internally by the hard drive into one or more sections of the disk called clusters. When the hard drive cannot store the file in a contiguous section of clusters it breaks the file into sections called fragments. These file fragments will then be stored in any free cluster on the hard drive. This is done automatically by the hard drive, and cannot be controlled by the computer user. To the user (and to the DAW software) each audio region file is presented as a contiguous file, but internally the hard drive will store each audio region in several different places on the hard drive. This can cause the hard drive to slow down and to be unable to playback the tracks in real-time. This is because the hard drive and controller must search around the hard drive looking for each fragment of the data to present it as a contiguous file. The technical explanations for this involve things like rotational delay, seek time and latency, which are all beyond this simple tutorial. Fortunately, all you need to know is that you can minimize the effects of a "fragmented" hard drive by simply "de-fragmenting" your disk drive before each recording session. This will move all of the different parts of the audio data for each track in closer proximity to each other, and will allow the disk drive to operate more efficiently. Be sure to backup all of your data before doing a defragmentation. If the computer fails during a de-frag, you could loose all of your data.

Mixing the song:

There are so many different types of hardware and software that I cannot possibly discuss all the issues related to mixing a song. I will assume for the sake of space that your device has enough processing power, enough disk space and enough memory to mix the tracks that you have recorded.

Most mixing software allows you to apply effects to each track or to several tracks using "plug-ins". Plug-ins are software equivalents to effects boxes. In an analog world you might have a compressor or a reverb unit connected to a patch bay, which allows you to route a track or tracks to that particular unit so that you can change the sound of the track. With plug-ins, the same concept applies, but no physical device is ever connected. It is all done through software, by adding the plug-in to the track.

plug-ins can be applied to one or multiple tracks, and they can be automated. This makes them more flexible than their physical equivalents. However, it is accepted and true that plug-ins only approximate their real world equivalents, by using math. Therefore, their sound can be very good, but anyone that can afford a physical famous brand limiter, would likely prefer it over it's emulator plug-in.

Plug-In Digression
The subject of plug-ins could take an entire tutorial itself, but I do want to mention just a couple of things about them.

There are two basic kinds of DAWs: The "Host Based" DAW, and the "DSP based" (Digital Signal Processing) DAW. In a host based DAW the computer processor (the CPU) handles all processing for the computer operating system, the DAW software, and the plug-ins. In a DSP based DAW, the soundcard itself (or a DSP add-in card) processes the plug-ins. This frees the host computer CPU from having to process plug-ins, which can be very compute intensive. High end DAWs are primarily DSP based, allowing multiple cards to be installed in the DAW to increase the number of plug-ins that can be run simultaneously without the need for a faster host CPU chip in the computer.

Generally, plug-ins do not permanently alter the recorded track. They run in "real-time", altering the sound of the track as it plays back. The more plug-ins you have running, the more processing you will require from your DAW. If you have a host based DAW, and you are running out of processing power you can either route multiple tracks to an AUX track that has a single instance of the plug-in, or you can "print" the effect onto the track (permanently altering the track) and remove the plug-in. I will leave these subjects as further exercises for the reader. The main thing I want to point out is that the more processing power you have in your DAW, the more flexibility you have in real-time mixing. Once you print the effect it cannot be unprinted, so make sure you keep a copy of the unaltered track as a backup.

There are a number of different plug-in architectures (for example: RTAS, VST, or DX), and not all DAWs support all the different types. To complicate matters, some plug-ins can be more expensive in one architecture than another. I encourage the reader to understand this issue before deciding on which DAW software to purchase.

Mixing software allows you to completely automate the mixing of a song. Almost all aspects of the mix can be automated: Volume, pan, effects, etc. This is called "mixing in the box". I have spoken to professional mixers who swear by mixing in the box, and I have spoken to others that swear AT mixing in the box. It is a personal preference. For those that find mixing in the box to lack life and dynamics, mixing outside the box is preferred. When mixing outside of the box, the computer is used as a glorified tape recorder, and all recording and playback is routed through a traditional mixing board, where the actual mixing occurs. Those that swear AT mixing in the box will often refer to the "summing-bus", "mix-bus", or "2-bus" problem. The 2-bus problem refers to the lack of depth, clarity and separation that can sometimes occur when the digital tracks are mathematically summed (mixed down) to the stereo master fader in the DAW software.

Perhaps the greatest claim to fame for mixing software is its editing capability. This is clearly the greatest advantage of converting audio to digital format. Once the performance is converted and stored in digital format it can be manipulated in countless ways. A performance that is out of pitch can be electronically modified to be in pitch. The chorus of a song can be duplicated or moved to another part of the song, out of time drums can be edited to be in time, mediocre performances can be modified to sound better than the performer could ever play them.

This is also the curse of digital audio. Some would say that digital audio has destroyed talent and musicianship. Whether this is true or not, it is clear that many people that never would have been able to record their music, can do so now, because of the proliferation of inexpensive recording hardware/software. The robust editing capabilities of these systems allow a mediocre musician to compile or "comp" many performances into a single performance that they could never otherwise play. The process of making a bad performance or recording sound better is often called "turd polishing".

More Conversion:

Once the song is mixed down, it must be either listened to, or "bounced" to a single file that can then be burned to a CD.

If the song is to be bounced to a file, one must consider the necessary conversion to the proper CD format. CD format is 16bit @ 44.1Khz. Since it is likely that the tracks were recorded at 24 bits and at a sample rate higher that 44.1Khz, the final mix down must be "dithered". The process of dithering removes information from the mix down and replaces it with noise. This process hides the missing musical information in the mix down, so that it is not noticeable.

If the mix down is to be listened to (monitored but not bounced), it can still be dithered using a dither plug-in on the master fader, so that the person mixing the song can mix it as it will be heard on CD.

In order to listen to the mix down on our reference monitors, the digital audio must be converted back to an analog signal. This is called the D/A conversion. As discussed in the above A/D conversions, we must consider the quality of the converters we are using in order to ensure that we are truly hearing a good representation of the final mix down. Conversion into the digital system (recording) and then back out (monitoring) is often call ADA or A/D/A.

The Speakers (Reference Monitors):

Speakers, or more accurately "Reference Monitors", allow the person mixing and recording the song to ensure that an accurate representation of the performance has been captured and is being mixed. Monitors are probably the one thing that all novice mixers try to skimp on. Unfortunately this is often a major mistake.

Reference Monitors are the interface between the mixer and the recorded tracks. Reference Monitors should be "flat" across all frequencies to allow the mixer to hear everything that is going on in the mix. Some reference monitors are known for their harshness, but they "translate" well. That is: If you can get a good mix on them it will sound good anywhere.

The key here, is that all playback systems do not sound the same. When you mix a song, your speakers must allow you to mix in a way that will translate to many different playback systems such as boomboxes, home stereos, and auto CD players. The most obvious example of bad mixing is trying to mix on regular computer (gaming) speakers. Computer speakers are very "hyped" in the low end, so that computer games have a thundering low end. When mixing with these speakers, a mixer will compensate by pulling back the low end in the mix, so it is not so muddy. Once that mix is burned to CD, it will sound thin on a regular stereo, because all the low frequencies have been EQ'd out.

Another mistake novice mixers make is to mix using headphones. The Fletcher-Munson effect tells us that low frequencies sound louder at closer ranges. Thus, mixing on headphones will also sound boomy and will not translate well to other systems. In addition, most listeners will hear the song played back through speakers, and in order to mix the song accurately for the average listener, the mixer must hear the song as it dissipates through the air.

The Mixing Room:

To understand the importance of the Mixing Room, when monitoring your mix, we must first understand a few basics about sound.

Previously we looked at the Recording Room and how it can effect what is captured by a microphone, and how that can effect the recorded tracks.

The same concepts are important when discussing the Mixing Room. The difference is that when the sound is played back, the mixer will perceive the sound as it reflects in the mixing room. The mixer will then make mixing decisions based on the sound that reaches his ears.

The difference is subtle, but important. In the Recording Room, the room affects what is actually recorded. In the Mixing Room, the room influences what the mixer hears.

When monitoring or mixing a recording, a circumstance can occur in which the reflected waveforms we discussed above can be delayed at an interval which causes some of the waveforms to be in or out of phase, causing some frequencies to "drop out", and other to be boosted. This effect is called "Comb Filtering".

This will affect the sound that that mixer hears, and cause the mixer to compensate by boosting or lowering some frequencies by using an EQ, or by making some instruments louder or softer in the mix. Since it was the room itself that caused the mixer to change these instruments, the mix will not "translate" well to other sound systems. In other words, it will only sound good in that specific room. For this reason, the mixing room should be treated to minimize the effect of sound reflections, so that the mixer can more accurately monitor the mix.

A room can be treated in many ways to allow for better mixing. This can be everything from completely rebuilding the room with special construction and materials, to just installing some bass-traps and gluing some auralex to the wall.

Errata:

Several other issues affect the sound quality of your mix. I will not get into these individual issues, but they should be considered. They include: Ground Loops, cable quality, room treatments, speaker placement, isolation, EM interference, attitude, and talent.

Backup:

I backup my session and audio files after every recording session. I have never actually had a catastrophic hard drive failure, though I know other people who have. It may be because I update my computer pretty often. Or because I take good care of my computer. Or maybe I am just very lucky. Regardless, I still religiously backup my data.

I do this because the risk is not worth the cost. A musical performance is a once-in-a-lifetime thing. If the artist looses that performance because I didn't back it up, not only will I loose my reputation, and have to redo the session(s) for free - but I might loose the only inspired take that performer can deliver. It's just not worth the risk to me.

I also take regular backups to an offsite storage location. Sometimes bad things happen, and if I lost my studio to theft, flood, or fire. It would be nice to be able to recover my audio.

That is also why I also create consolidated versions of each track in my sessions for backup purposes. I discussed this in the "Saving The Data" section above. In a nutshell, if I have a single Wav file for every track, I could take that data to any other mixer I know, have them import the data into their DAW, and have them mix my project, even if my studio was completely destroyed. Has this ever happened to me? No. It's just insurance. Both for the work I have done so far, and for my reputation.

Mastering:

Genuine audio mastering is a subject that I know very little about. It is a magical realm left to those who have learned it's mystical secrets.

I mention it here, because I often hear the word "mastering" used to describe making the final mixdown of a song "louder" on a CD. Calling this "Mastering" is equivalent to calling Jimi Hendrix "some guy that played guitar", or referring to The Beatles as "an old band that had a few hits".

There are several standalone and plug-in software products that can accomplish the task of making your mixed song sound louder on CD. They usually have "presets" that allow you to know almost nothing about the process of mastering, and yet get pretty good results. While this is not truly mastering, you may want to consider some of these products to get quick and fairly decent results for yourself or your clients to make the final mix sound louder and more like commercial CDs.

Comments:

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