Sudeep Audio : Sound Tips and Lessons for Sound Engineers, singers, musicians, students

1. A good quality microphone is the first thing that you should look into. There are several kinds of good microphones to start with. You can check a list of such microphones with their specifications here.

The better your microphones, the better sound you'll be able to get. The microphone is what really captures the sound. And all mics have colorations and non-linearities. Big recording studios use mics that cost anywhere between Rs. 35,000 to Rs. 1.5 lacs+! This is one way they can assure their clients of the best recorded sound possible. It follows that your choice of microphones will have a huge impact on the quality of your own recordings.

2. Something to make musical sounds with :-)

3. A reasonably good sounding, quiet space to record in is very important. The easiest way to make a bad sounding room sound good is to deaden it as much as you can, preferably with professional quality room treatments (like RPG Diffusors and Bass Traps). However, the nicest sounds come from big rooms where the sound can "breathe". If you have a loft space, you will need to shut out any traffic noises, and treat the room for unwanted resonances.

4. Yes, this room treatment thing is expensive. Your best bet is to deaden your small room as much as possible, and carefully sweeten your sounds with reverb and/or EQ when you mix down. SudeepAudio.com can help you with acoustics at affordable costs and superior quality.

5. A way of storing your music in a format that other people can listen to; at the very least, you'll need a cassette deck, though a DAT or CD-Recordable "burner" is much hipper.

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Basic Concepts of Digital Audio

As you know, computers can only work with binary data, or "0's and 1's". The zeroes and ones represent two states, either "off" or "on". This is like having lots of tiny switches that form a sort of super-fast Morse Code, which a computer uses to represent real world events (such as musical sounds) in what is known as binary code.

First, Analog Audio...

The audio we hear from our stereos and home entertainment systems is 'analog audio'.

This means that oscillating voltages are used to represent the original sounds. Here's how this works: A saxophone plays a note in a smoky basement jazz club. The vibrating air coming from the horn moves the air in the smoky room, and your eardrums vibrate back and forth with the vibration of the air molecules. We experience these vibrations as "sound". A microphone and an old-fashioned analog tape recorder are set up in the room. The saxophone vibrates the air around it, setting up a series of pressure changes that radiate through the air in the room. When these pressure changes reach the microphone's diaphragm, it shakes back and forth with the vibrations. The microphone "hears" these vibrations and converts them into electrical voltages that are an "analogy" of the air pressure changes that made the original sounds. The tape recorder's record head then stores these electrical voltages ("analog audio signal") on magnetic tape as magnetic fluctuations.

After the set is over, we take the tape recorder home and hook it up to our stereo system. Now we can play the recording back. We play the tape, the magnetic fluctuations on the analog tape are converted to electrical voltage changes (analog audio signal) by the tape playback head and the resulting voltages are sent to our stereo amplifier. The amplifier changes those fluctuating voltages into current fluctuations which move our stereo speakers back and forth, far and fast enough to create disturbances in the air of our listening room that are almost exactly the same as the original vibrations caused by the saxophone playing in the jazz club. That's High Fidelity analog audio!

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And now, Digital Audio...

So what happens in digital audio?

How is digital different than analog? First the original sound is converted to analog audio voltage fluctuations by the microphone(s). Instead of using an analog tape deck, we are now going to use a digital recorder. Let's use a DAT recorder as our example. The analog audio voltage fluctuations are fed to a circuit called the Analog-to-Digital Converter that changes the incoming voltages to digital "snapshots", 44,100 times a second. Each "snapshot" consists of 16 zeroes and/or ones. Each combination of zeroes and/or ones represents a different signal voltage. Using sixteen 0's and 1's in each "sample", one of 65,536 different voltage levels can be described by each sample. A DAT or CD uses a "sampling rate' of 44,100 samples per second (44.1kHz).

This means that 2,890,137,600 different analog audio voltage levels can be described each second -- and you're right, that's a lot. But some say that capturing audio with 16 bits, 44,100 times a second may not be enough to accurately describe what our ears can hear, so that's why there is now a push on to record everything in 24 bits, 96,000 times a second, or at "24/96 resolution"... When we want to actually hear the digital audio, the audio data has to go through a Digital-to-Analog Converter, which changes the binary code samples to analog voltage fluctuations that are then sent to a power amp and on to the speakers, which shake the air molecules enough for us to hear the original sound.

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Choosing a Microphone for your Home Studio Projects

Microphones convert the sounds you hear into electrical signals that can be recorded on to tape or hard disk. This means that choosing the right mic for the job at hand is critically important to getting the sound you want on your final tracks. No amount of EQ, compression or reverb can change the subtle signature that any particular microphone leaves on your audio tracks. So, how do you choose that perfect mic without first buying and auditioning everything out there? First, you should familiarise yourself with the basic microphone 'families':

Microphone pickup patterns:

Omnidirection
Cardioid
Figure-Eight

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Microphone types:

Dynamic
Condenser
Ribbon
Stereo

Omnidirectional:

This describes a microphone's pickup pattern; in this case one that picks up sound equally from all directions. 'Omni' mics tend to have very good bass response, without the artificial low frequency boost provided by the 'proximity effect' of a typical cardioid mic (see below). Really good omnidirectional condenser mics are great at capturing a sense of 'open space' and 'air', which makes them the first choice for critical reproduction of acoustic instruments in good sounding acoustic spaces, like symphonic orchestras, vocal choirs, pianos, or string quartets in concert halls. Some of the highest fidelity mics available are of the omnidirectional condenser type, such as mics from Schoeps, DPA (B&K) and Earthworks. A common use for dynamic omni mics is in TV and radio reporting (known as Electronic News Gathering or "ENG"), where you want to capture environmental sounds along with the reporter's voice.

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Cardioid

This describes a microphone's pickup pattern that is more sensitive to sounds directly in front of it than to sounds 90 degrees off to either side. A cardioid mic is even less sensitive to sounds directly behind it - in fact, cardioid mics practically cancel pickup of sounds emanating from directly behind the mic. This makes cardioid mics very useful for sound reinforcement (P.A.) and live recording use, as well as being the most popular choice for use in the imperfect recording environment of most home studios. To use a cardioid mic, simply aim the mic at the instrument you want to record, and the rest of the stage sound will be at least somewhat quieter than the desired instrument's sound. Most of today's most popular microphones have a cardioid pickup pattern.

There are a couple of variations on the cardioid pick up pattern. Supercardioid and hypercardioid mics are less sensitive to 90 degree off-axis sources than plain cardioids, meaning that they will do a better job of rejecting sounds from off to the sides. However, hypercardioids do pick up some sound from directly behind the front of the mic, making them a little bit like a 'figure-eight' mic.

Cardioid mics exhibit a characteristic called the 'proximity effect'. The closer a sound source is to a cardioid mic, the more the mic will accentuate that sound source's bass frequency output. This can add richness and fullness to a singer's voice or to a saxophone's sound, but it can also muddy the sound of a guitar amp or acoustic bass. When miking from a distance, cardioid mics have a tendency to sound somewhat thin in the bass when compared to omnidirectional mics. For this reason, cardioid mics are usually used for close-miking (with the mic placed less than two feet from the sound source), while omnidirectional or figure-eight mics are usually used when miking from farther away.

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Figure Eight

Figure-eight mics have the 'open' sound and good bass response of omnidirectional mics, with the added advantage that they reject sounds from either side of the mic. Since figure-eights pick up sound equally well from directly in back and directly in front, care should be taken that you don't capture undesirable reflections from low ceilings or nearby walls. A good place to use a figure-eight pattern mic is when you want to cancel reflections from side walls in a narrow-ish room but you want to capture a good sense of room ambience.

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Dynamic microphones

Dynamic mics use a 'moving coil' to sense the changes in air pressure that make sound waves. The wire coil is suspended over a permanent magnet. When moving air hits the coil, it moves over the magnet, which causes electromagnetic induction to take place. This causes an AC voltage to be formed that is an electrical 'analogy' of the original sound. The electrical signal that appears at the mic's output is a more or less faithful reproduction of the original vibrations in air, only in fluctuating AC voltages instead of air pressure changes. Small diaphragm dynamic microphones:

These are by far the most commonly used mics for P.A. and stage sound use. Dynamic microphones are typically very rugged and don't require a voltage source to work properly. Cardioid pattern, small diaphragm dynamic mics are most often used as handheld vocal mics (like the very common Shure SM-58) or as instrument mics for stage use (like the equally common Shure SM-57). Large diaphragm dynamic microphones: While similar to their small diaphragm cousins, large diaphragm dynamic mics are typically used for very loud, bass heavy instruments such as tom-toms, kick drums, and bass amp speakers.

The larger diaphragm allows these mics to withstand higher Sound Pressure Levels (SPL's) with ease, but the larger moving mass of the larger diaphragm will limit the high frequency response of the mic. These characteristics also allow low-distortion reproduction of very loud instruments such as trumpets, trombones and electric guitar amplifiers. Some popular large diaphragm dynamic mics are: Electro-Voice RE-20 - a favorite of radio announcers and a good mic for kick drums Shure SM-7 - similar to the E-V RE-20 Sennheiser MD-421 - commonly used on tom-toms and hand percussion

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Condenser microphones

Condenser microphones capture sound using a conductive diaphragm with a capacitative charged plate behind it. The charge is supplied by a DC voltage source like a battery or the 48 volt 'phantom power' supply present in most mixers and mic preamps. Air pressure changes meeting the conductive diaphragm cause it to move, which causes an analogous AC voltage to be formed in the charged plate. These small AC voltages are sent out of the microphone to be further amplified. Because the diaphragm can be made very thin and light, condenser mics tend to be more accurate than dynamic mics, especially in the midrange and treble frequencies. Because of this low mass construction, conenser mics do tend to be more delicate than dynamic mics.

Condenser mics are more commonly used for studio recording more than for live sound and P.A., but there are some road-worthy condenser mics (like the Shure SM-87). Small diaphragm condenser microphones: Small diaphragm condenser mics have the best high frequency response of all the commonly available microphone types. For this reason, small diaphragm condenser mics are most often used as drum set overhead mics (to faithfully capture cymbals and stick attacks), for acoustic stringed instruments like guitars and violins, and for hand percussion instruments like vibraphones, shakers, and marimbas. Another common use for small diaphragm condenser mics is as stereo pairs for ambient pickup of acoustic events in good sounding spaces.

Some popular small diaphragm condenser mics are: Shure SM-81 - very flat frequency response; commonly used on acoustic guitars and as drum kit overheads. Audio Technica AT-3528 - a cardioid model that is sort of a 'poor man's KM-84'. AKG C 1000 S - a budget favorite. Neumann KM184 - a truly professional recording mic. Oktava MC-012 - from Russia, this is another mic made to be similar to the KM-84, but for a lot less money.

Large diaphragm condenser microphones: Since condenser mics are intrinsically more sensitive to higher frequencies, it's possible to combine the warmth and fullness of a large diaphragm with the high frequency detail typical of small diaphragm condenser mics into a single microphone. These large diaphragm condenser mics are the mainstay of recording studios everywhere, especially for pop vocals and close miking horns. Some older vacuum tube based large diaphragm condenser mics, such as the Neumann U47, are collector's items prized for both their sonic warmth and their accurate reproduction of aural details. The 1960's vintage Neumann U87 is an FET-amplified, large diaphragm mic that is more of a modern classic.

Some popular large diaphragm condenser mics are: AKG CS 414 ULS - a standard for overhead drum miking and for general use; choice of cardioid, hypercardioid, omni, and figure-8 pickup patterns. AKG C 3000 B - a budget mic based on the design of the venerable CS 414; cardioid and hypercardioid only. Neumann TLM 103 - a new, lower-priced version of the famous U87; cardioid only. Audio Technica AT-4043a - a fabulous microphone for the price; cardioid only; great on saxophones. RODE NT1 - the new budget leader; cardioid only. CAD Equitek E-100 - another budget contender; super-cardioid only. Marshall Electronics MXL 2001-P - made in China, this is a very inexpensive, yet surprisingly smooth sounding mic; cardioid only. Oktava MC-219 - made in Russia, this is a classic budget 'sleeper'; cardioid only.

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Ribbon Microphone

When a wafer-thin, small aluminum ribbon is suspended between two mounting points inside a strong magnetic field, you get a microphone that is extraordinarily sensitive to vibrations in air (sound). Ribbon mics can really capture the thump of a plucked acoustic bass or the subtle dynamics of jazz drums. Unfortunately, ribbon mics tend to be extraordinarily fragile (one drop and they're history). Nevertheless, ribbon mics remain a favorite of recordists everywhere. Some common ribbon microphones: RCA BX-44 and BX-77 - the original classics. Coles 4038 - the standard in modern ribbon mics. Beyerdynamic M-260 - a budget ribbon mic; hypercardioid only

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Stereo Microphone

By combining two cardioid condenser elements into one chassis, a single point stereo microphone can be made. The most common is the X-Y type, where the two cardioid elements are pointed away from each other at a 90 degree angle. Some stereo mics are of the Mid-Side (MS) type, using a combination of a forward-facing cardioid element with sideways-oriented figure-eight element, which allows for remotely controlled adjustment of the stereo image width. Some common stereo mics: Audio Technica AT-825

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MIDI (Musical Instruments Digital Interface)

If you play piano well at all, you will want to use MIDI ("Musical Instrument Digital Interface") to record your music into the computer. Modern personal computers and MIDI sequencing software let you record audio over top of the MIDI tracks, with the audio and MIDI parts synchronized. This way you can record a MIDI synthesized backing band and then record yourself singing or playing an acoustic instrument over top of the synthesized orchestration. This is how almost all soundtracks for movies and TV commercials are made these days.

What you need for a good basic setup:

1) A synthesizer of some kind. These come in several different packages...

You can get an all-in-one keyboard synthesizer, which is a keyboard with a synthesizer built in, generally called a "synth". Make sure you get one with MIDI IN and MIDI OUT ports (MIDI THRU is nice, but not absolutely essential). Examples...

A Sound Module. This is the synthesizer part of a keyboard synth but without the keyboard. Using a sound module requires a MIDI controller (see below). A sound module will usually give you the best sound quality and versatility. A module can be used with any MIDI controller or keyboard, as well as with any computer-based sequencer with a MIDI interface installed, and they're portable. Examples of popular sound modules are the Roland SC-88 and JV-1010, and the Yamaha MU-10XG and MU-50.

A Digital Sampler. These function sort of like tape recorders that only record a few seconds of a sound (a "sample") and then map the resulting 'recordings' to MIDI note values so that the sampled sound can be played on a MIDI keyboard or whatever. These (expensive) gadgets are for the advanced synthesist who wants the most control he/she can get over the musical sounds to be used. Movie scores, music for TV commercials, and slick pop music all make extensive use of samplers.

A PC Sound Card with built in MIDI. Most inexpensive PC soundcards have bottom of the line synthesizers built in that don't sound very convincing at all. Some do have DLS 'samplers', but these are not very user-configurable. If you already have a soundcard with a WaveBlaster or compatible daughtercard header, you can add a MIDI wavetable daughtercard such as the Turtle Beach Cancun FX, Roland SCD-15 or the Yamaha DB50XG. Examples...

A major disadvantage of using a soundcard's MIDI is that a computer must be used to control the synthesizer, making it incompatible with other setups and hard to carry around with you. But a MIDI soundcard can be a viable choice for a self-contained home studio setup.

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2) A MIDI controller

Usually a piano keyboard in form, but there are also drum controllers, guitar controllers, and even wind controllers. The piano keyboard type is the least expensive, the most reliable, and the most versatile. Popular models...

A keyboard synth can be used as a controller for an external sound module, as long as it has a MIDI Out connector. Similarly, any Digital Piano with a MIDI Out connector can also be used as a keyboard controller.

Keyboard Action - You may have heard the terms "touch sensitive" and "piano feel" used to describe keyboards. Basically, this describes how well the keyboard's keys mimic the "touch" and "dynamics" of a real piano keyboard, and how that information is transmitted to the other devices downstream of the keyboard. Synth Action - means that the actual key is merely a switch, with little if any mechanical resistance to being pressed.

This is very similar to the action of a Hammond organ's keyboard. Weighted Action - means that the keys have a resistance to being pressed that is meant to mimic the feel of an actual piano keyboard. Some expensive controllers actually use a hammer action design borrowed from pianos, so that the feel of an acoustic piano is very convincingly reproduced. Cheaper "piano feel" keyboards have a sort of 'spongy' feel to them. A good weighted action helps the player convey more expressive dynamics in performance because the keyboard allows the player to "dig in" to the keys (for lack of a better way to describe it).

Velocity Sensitivity - describes the ability of the keyboard to transmit a whole range of dynamic touch in MIDI data transmission.

The basic idea is that a soft touch translates to a lower MIDI velocity number, which in turn translates to a lower volume for that note played. A harder touch will translate to a higher velocity number (i.e. louder). Better synthesizers change the envelope of a patch so that a higher velocity has a more intense attack than the same note played with a lower velocity. This can make for some very convincing sounding MIDI performances. Aftertouch - describes how some synth patches will change parameters if a key is held down after the note is first struck. For instance, some patches 'swell' if the player presses the keys while the notes are sustaining.

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3) A MIDI Interface

This allows your MIDI devices to talk to your computer. These range from very inexpensive models to really fancy things with all sorts of special features. The most important thing is to decide what exactly you want your MIDI setup to do, and then figure out how much you really need to spend to achieve your goal. I find that the more features a piece of gear has, the steeper the learning curve, and the more likely it is to keep you from just writing music, if that's your main goal. Of course, after you become a MIDI expert you can invest in all sorts of gadgets that will allow you to do lots of exciting, tricky stuff.

What to look for in a MIDI interface:

For all MIDI interfaces: You will need to decide if you can live with a "one in - one out" 16 channel interface (for a single sound module), or if you need a "two in - two out" 32 channel interface (for a fancier 32 channel sound module or two 16 channel sound modules). Having more channels available allows you to have more individual "voices" playing in your MIDI compositions, and gives you a wider choice of sounds to choose from.

If you're into composing symphonic pieces or other types of music that require a lot of different instruments playing at once, go for a 32 channel or even a 64 or 96 channel interface. For older Power Macintosh: Mac MIDI interfaces plug into the modem or printer port on the computer (these are 'serial' ports). It follows that MIDI interfaces for the Mac are always external. If you plan to use one serial port to hook up two devices (like your printer and your MIDI interface on the printer port) then you'll have to make sure the interface you get allows you to switch between devices with a 'pass-thru' switch. Popular low-end MIDI interfaces for the Mac are the MOTU FastLane, Opcode Midi Translator II, and the Midiman MacMan. You have to decide how many channels you need. A basic 16 channel interface will do for most people. A fancy 96 channel interface with SMPTE synchronization signal generation (like the Mark of the Unicorn Midi Express XT) will let you use the equivalent of six synthesizers synchronized with external audio and video playback. For iMac, G3 and G4: NEW! - The iMac, "blue" G3 and the new G4 Macintosh computers lack the serial ports that have been standard on the Macintosh since its introduction over fifteen years ago.

Instead, the latest Macs use the new Universal Serial Bus (USB). A new Mac will require a special USB MIDI interface. Roland, Mark of the Unicorn, Midiman, and Opcode have all come out with new USB MIDI interfaces. For Wintel PC: PC MIDI interfaces can either plug into an internal 'ISA' expansion slot, which requires opening up the computer's case, configuring the MIDI interface card for the correct IRQ and port address, plugging the card in, and installing the software drivers --or-- as an external box that attaches to the parallel printer port (LPT1), with the printer plugging into the MIDI interface's 'printer pass-thru' (similar to how it is with a Mac interface with two devices attached). Some PC external MIDI interfaces can connect to a PC's serial port (COM1 or COM2), though these are less common than the printer port variety. You can also use a typical PC multimedia soundcard as your MIDI interface. Most soundcards work just fine as MIDI interfaces, while a very few (like the old Creative Labs Vibra 16S) will cause stuck notes or other problems.

Avoid ISA bus soundcards as these are now obsolete. The PCI bus is the current standard for plug-in cards. NEW! - Universal Serial Bus (USB) MIDI interfaces have arrived! At this point, only Windows 98 supports USB well on the Wintel PC -- using USB is still problematic on many Win2000 systems and WinNT 4.0 doesn't support it at all -- but USB allows Win98 users with recent Pentium II or Pentium III systems to 'daisy chain' peripherals with the promise of true "Plug 'n Play" ease. Check out the Roland Super MPU-64 USB MIDI interface for example. Mark of the Unicorn, Midiman, and Opcode have also come out with new USB MIDI interfaces. You have to decide how many channels you need.

A basic 16 channel interface will do nicely for songwriting. A multi-channel interface with SMPTE will be necessary for making soundtracks for movies, commercials, etc. SMPTE allows you to synchronize external equipment such as tape decks or VCR's with your MIDI gear. There are even fancier interfaces available that can do all sorts of crazy things, from Opcode, Mark of the Unicorn, Midiman, and others. 4) A MIDI Sequencer software application - All the current MIDI sequencer versions allow you to record audio tracks that play along with your MIDI tracks. The only "professional's choice" used to be Mark of the Unicorn's Digital Performer (Mac), but there's no one program that completely rules the roost any more. Examples...

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Compressors with Acoustic Music

Most people believe that a Compressor over-smoothes the sound of acoustic music and squishes all the "acousticness" out of it. Some relate the use of compressors to the sound of TOP 40. It is true that compression plays a large part in creating that sound. However, a compressor can be used intelligently with acoustic music. This bias is based on more of a lack of knowledge of what a compressor does, than a true evaluation of its usefulness.

A compressor is a device that regulates the electrical signal from a microphone or pickup. If properly adjusted, the compressor allows the soft passages to be heard clearly but not necessarily more loudly, and the loud passages not to become too loud for the audience or too strong for the sound system. A good sound engineer should understand the desired effect of acoustic music and use compression in a way that won't make you sound like elevator-music or muddy up your sound. When used properly, the compressor greatly enhances the sound of acoustic instruments by allowing you to express yourself fully while allowing the audience members to hear the louds and the softs along with your playing intensity levels.

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High-Octane Optional Equipment

Mixing Board
DAT Monitors
Digital Audio I/O
Hardware DSP
CD Recorder
Outboard A-D-A Converter
Modular Digital Multitrack

Options that can really help you make better sounding recordings are:

A mixing board.
This will allow you to record from or play back more than one analog sound source at a time. You can adjust the volume levels (in other words, "mix") and use tone controls ("equalization" or "EQ") to change the timbre of your various MIDI, microphone, and line-level sound sources in real time. It is possible to do all of this in your computer, but it is usually difficult to control all of your devices from one interface, and requires a powerful computer with lots of RAM and lots of analog inputs. The most popular small studio mixers are made by Mackie Designs, Behringer, Spirit by Soundcraft, Yamaha, Tascam, Audio Centron, and MidiMan. The new digital mixers are taking over studios everywhere. First came the Yamaha ProMix 1, then the Yamaha 02R, now there are the Yamaha 01V, Panasonic WR-DA7, Fostex VM-200 and Tascam TM-D1000, with new models appearing all the time. A digital mixer allows you to record from microphones and other analog sources straight into the digital domain, where DSP effects can be applied and the waveforms can be stored digitally in "virtual tracks" on the hard disk recorder or DAW, or as actual digital tracks on an ADAT or similar. Then the tracks can be mixed down while still in the digital domain, with fully automated faders and all mix settings stored in memory for instant recall. Then you can master to DAT, CDR, Magneto-Optical disk, or whatever. All digital 'til the end listener plays back the final product! Kewl!

A DAT recorder, which is a GREAT thing to have, but does cost a bit more than the other audio gear mentioned here. Note that DAT is still hanging on as the industry standard stereo music production storage medium, although many people are switching to CD-Recordable.

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A really good monitoring system.
Basically this is a tonally accurate stereo system, especially designed for revealing the details and/or flaws in a recording. This is a critical part of any home studio setup. A good monitoring system will likely cost more than you expect, but you're "flying blind" without one. Most home studio setups will use small speakers that are meant to be listened to from no more than about four or five feet away. The idea is to form an equilateral triangle between the listener's head and the two speakers (e.g. the listener sits four feet away from either speaker, and the speakers are situated four feet apart from each other).

Speakers used in this manner are known as near-field monitors or simply "near-fields". Since most home studio setups have less than ideal acoustics, near-field monitors are a good way to keep sub-par room acoustics from interfering too much with the listener's ability to hear the playback accurately. When shopping for studio monitors for a computer-based home studio, remember to look for shielded ones. Magnetic shielding allows the placement of speakers closer to the computer's display, so that you can listen to and work on your audio data from the same position. Many newer monitors are also self-powered, with the necessary amplifiers built into the speaker cabinets. An example of a self-powered, shielded monitor is the Mackie HR824. The Yamaha NS-10 is neither shielded nor self-powered.

Other popular monitors are the Alesis M-1, JBL LSR-25P and Genelec 2029A, as well as others from Audix, KRK, PMC, Hafler, Tannoy, Dynaudio, Legacy Audio, Spendor, Vergence Technology, Dunlavy, Meyer, etc. Expect to pay at least Rs. 30,000 for a decent pair of studio monitors; more for really good self-powered speakers. If you're really strapped for cash, consider a good pair of "pro quality" headphones. Headphones cost a lot less than a good pair of near-field monitor speakers and an amplifier, but listening in headphones is quite different than listening to speakers in a room.

Mixing in headphones is therefore different than mixing on speakers, but it can be done well. Headphones are also a good tool to have around for a 'double-check' on your mixes, even if you have a good set of monitors.

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A digital audio I/O card.
This allows you to send digital audio to or from the digital input or output of a DAT, CD player, or Digital to Analog Converter ("DAC") directly in or out of your computer. The idea is to keep the audio signal from going through more than one or two digital-to-analog (D-to-A) or analog-to-digital (A-to-D) conversions during the entire process of recording your music. In digital audio processing, these conversions are where the worst distortions can occur. It's also a good idea to keep analog audio signals away from the inside of the computer, as all those clock crystals in there are generating lots of radio-frequency ("RF") noise, e.g. your microprocessor at 500MHz or higher, your PCI bus at 33MHz, your cool new AGP video card at 66MHz, and so on. Radio Frequency Interference ("RFI") does really bad things to analog audio circuits ("digititus" anyone?).

A digital audio workstation ("DAW") that includes advanced Digital Signal Processor circuitry (referred to as "DSP") in hardware. The hardware DSP can do all the audio processing without the need for use of the computer's CPU. The DSP can then be tweaked for best sonic results, while the CPU is left free to work on its normal computer operations. High quality hardware DSP costs a lot more than software DSP, but if you're a stickler for sound quality... The higher priced Digidesign ProTools 24 | MIX (MacOS/WinNT), Digital Audio Labs V8 (Win95/98/NT), and Sonic Solutions (MacOS) systems have DSP's built in to their digital audio hardware.

The TDM Plug-Ins take advantage of the ProTools DSP's. If you are running a Windows 95/98 PC, there are several higher priced digital audio systems that include advanced DSP chips. Examples of Windows 95/98 digital audio systems are Digital Audio Labs V8, Ensoniq Paris, Soundscape SSHDR-1, Creamware TripleDAT, Sadie, and MicroSound. Windows NT 4.0 is the best performing and most stable of the currently popular operating systems, but only a few Digital Audio Workstations support NT. Also be aware that MIDI is not as well supported in NT as it is in Win95/98 and the MacOS. Some examples of NT-ready DAW's are Digital Audio Labs V8, Soundscape Mixtreme, and Digidesign ProTools 24 | MIX.

The new 'all in one' DAW's like the Roland VS-880EX and VS-1680 come with DSP circuits built in. The VS-880EX has six balanced 1/4" mic inputs, with digital EQ, compression and reverb onboard, and a SCSI port for archiving your sessions to external hard drives or Jaz drives. It's portable, too!

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A CD-Recordable "burner".
These allow you to record Red Book-spec Compact Disc Digital Audio (CD-DA) on to CD-Recordable (CD-R) discs, so that others can hear your music in all its undiluted glory on their own home or portable CD players. CD-R recorders are available in SCSI versions for Mac or SCSI-equipped PC's, or in ATAPI (IDE) versions for plain jane PC's. SCSI burners are considered to be less problematic, though many are reporting success with IDE CD burners (I prefer SCSI). Look for 4X or 8X speed CD-R writing; this will speed things up considerably. SCSI CD burners from Yamaha and Plextor are generally thought to be the most reliable and best-sounding.

An outboard Analog-to-Digital-to-Analog Converter (or "ADAC" for short). This is a box that converts the digital audio data stream to analog audio so that you can hear it through a typical stereo amp and speakers. They can also take analog audio and convert it to digital audio data. Because an ADAC is a dedicated, single purpose device, it will usually sound better than the Digital-to-Analog Converters ("DAC's") that come inside CD players, DAT recorders, and consumer-grade computer audio hardware. Having a high quality Analog-to-Digital Converter ("ADC") can make your "live" audio tracks sound better. The MidiMan Flying Cow is a good low-end ADAC

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A Modular Digital Multitrack recorder ("MDM"),
like the Alesis ADAT LX-20, XT-20 or M-20, or Tascam DA-88, DA-98HR or DA-78HR 8-track digital tape recorders. MDM's are found in small studios all over the world, and are the standard for making demo recordings and broadcast audio. The Tascam's can record 108 minutes of 8 track, 24-bit audio on a single Hi-8 120 tape, while the ADAT's max out at 40 minutes of 8 track, 20-bit audio on a standard ST-120 S-VHS tape, or 62 minutes on a special ST-180 tape.

There is a whole market growing up around multi-channel Digital Audio I/O cards that route the 8 or 16 (or 24) channels of digital audio data between a personal computer and one or two (or three) MDM's. Examples of this kind of card are the Frontier Design Group Dakota PCI, Sonorus StudI/O, RME Project Hammerfall, Mark of the Unicorn 2408, Soundscape Mixtreme and Alesis ADAT Edit. The Alesis ADAT Lightpipe interface is much more widely supported than the Tascam T-DIF interface. For more info on digital audio interfaces and other soundcards, check out the soundcards page.

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How to take control of your SoundCheck (Part 1)

If you are lucky and you are booked at a venue with an installed sound system with an engineer, don't look at it as a chance to sit back and relax. This is an opportunity to create your sound with someone who knows how to utilize sound equipment. Every sound engineer has different habits but they should still be sensitive to your needs. By knowing the basic steps to running a soundcheck, you will be able to communicate more effectively with the sound engineer and be able to take better control of your sound. Here are some pointers& ..

1. Alway leave yourself plenty of time for sound check. It is recommended that sound check begins at least three hours before showtime and ends well before patrons have started to arrive. Even if the patrons are held in the lobby, you always want to consider not doing sound check when they are in the building. They will be still able to hear the sound check through the walls and floors.

2. Talk with the sound engineer before you get on the stage. This is a good time to get a feel of how easy it is to work with this engineer. It also gives the sound engineer an opportunity to find out any more information about your needs. Also, if music is playing when you arrive, walk around the venue to get an idea of the room reaction and general sound.

3 . If the sound engineer has not finished setting up microphone cables and such, allow him/her to finish before loading your whole band on the stage, (that is if you sent a technical sheet ahead of you).

4. If you travel with someone who understands your sound, have them walk around the auditorium while you are soundchecking. This will help later on when you are trying to communicate your needs to the sound person.

5. Tune all of your instruments and if you have pickups have them ready to be plugged in.

6. Have all the members of your band on stage and ready to play before you begin sound check. Bodies do have an effect on the sound and the monitors. If every one else is still in the green room while you are checking your guitar and then comes out on the stage, all your monitor settings will most likely have to change. It is advisable to make it a band rule, that fidgeting with instruments and talking during sound check are not allowed. This can be very distracting for the sound engineer and will shorten the amount of effective time spent adjusting your sound.

7. When everything and everyone is ready to play, begin your sound check.

8. At this point, the sound engineer will ask for what he needs you to do. Every engineer works a little differently and may ask for things in a different order than what you are used to, so bear with them. If you have a specific preference for the order in which the instrument and voices will be checked, communicate this to the engineer and they will let you run this part from the stage.

9. The sound engineer will bring the volume up and adjust the tone of every signal being sent to the board. Don't worry at this point if something sounds like it may be too loud or too soft. Also don't set your monitors yet.

10. When the sound engineer is finished setting all the channels, ask for different pairings of instruments and/or voices. In fact, you should ask to hear these pairings. Know what the relationship of the acoustic guitar and the fiddle should be. When a few different pairings have been listened to and adjusted, have the whole band play a few selections. Do not set your monitors yet.

11. Have the band play a few different selections that show the style, range and complexity of your music. It is a good idea to always use the same selections at all your soundchecks. This will give the sound engineer an idea of what to expect. This is also the best time to set effects such as reverbs and delays. You will be able to hear the main house speakers since your monitors are not on yet.

12. After your ears have adjusted to the room, and you have set the general levels and balance among the different musicians' instruments, then set your monitors. If you have an instrument that is particularly prone to feedback, begin with that one. This will set the maximum volume and general EQ of your monitors and avoid feedback during the show. If you need reverb in the monitors, ask the engineer. Not all sound systems are capable of this and it may lower feedback thresholds.

13. Once your monitors are set, play the SAME few selections that you played before. This will give the sound engineer an idea of how the monitors will affect your room sound. In fact, if you hear your monitors turning on and off or up and down, don't let it frighten you. The sound engineer is evaluating thresholds for feedback and checking the spillage of the monitors into your main sound. However, make sure that your monitors are left at comfortable levels before you end your sound check.

14. Have the people you have standing in the auditorium advice you, or the sound engineer, of which instruments should be brought up or down in general.

15. When you are satisfied with your sound and monitors, tell the sound engineer that you are all set. He/she may have a problem at their end that they are trying to work out and may ask that you play some more.

16. If you have time, it is suggested that you play/practice to the end of the time allotted for the sound check. This will allow you to become adjusted to the reaction of the room and the sound system. If you wish, ask the sound engineer to stay behind the board and ask them to warn you when time is up. Just remember that the sound engineer usually has to finish dressing the stage or may need to fix something.

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How to take control of your SoundCheck (Part 2)

LIGHTS ON OR OFF

Have all theatrical lighting on at their usual settings during sound check. This will bring your wooden acoustic instruments to the stage temperature and will keep your instruments from de-tuning during your performance. This also will show any hum and interference problems which may occur between certain types of pickups and lighting systems. It also allows your eyes to become adjusted to the lights. In accordance with all of this, it is advisable to re-tune your instruments and leave them on the stage if they are particularly sensitive to environmental changes. Then, have someone get your instrument 15 minutes before show time and bring it to you in an environment similar to the stage, re-tune and have him/her return the instrument to the stage.

THE AUDIENCE FACTOR.

You will be sound checking in an empty auditorium. Your overall sound will completely change when the venue fills up. Besides absorption of certain frequencies and reduction of room reverberation, audience members also add heat and humidity, which greatly affect sound transmission through the air. Try to run your sound checks with a main EQ curve that is hot in the mid to high range to allow for this. At the same time run your main volume slightly above the usual setting during soundcheck (for feedback thresholds) and adjust it downward based on the relative size of the audience.

It is best to set these parameters before the performance begins so the changes will not be as noticeable during the performance. However, some sound engineers prefer to adjust individual frequencies during the performance. This method may cause your sound to be muddy at the beginning of the show. With acoustic instruments it is better to start with a brighter, more midrange sound and only if needed, adjust these frequencies downward during the show. This is more comfortable for the audience to listen to and they may not even notice the changes. You will rarely have to re-adjust the main EQ during performances. How many shows have you been an audience member of where the show started off with a muddy sound only to finally have it clear up right in the middle of the third song? Hardly anybody complains that the sound was too bright at the beginning of the show. In fact, even in critical reviews, the muddy start is the first thing to affect the critic's reaction and is often a let down of their expectations of the show and affects the rest of their critique.

Critics are essentially audience members who write about their experience and there are two hundred or more of them sitting out there who just paid to see you and hear your music. With this in mind, be sure to listen to you instruments during soundcheck with the audience factor in mind and inquire as to how the sound engineer adjusts for the audience factor. If you agree with the theory ask them to use this method.

DON'T UNDER USE REVERBS.

They are a great way of separating similar instruments. Even if the room is very reverberant, a great deal of this disappears when the audience arrives. It is more difficult to adjust levels of reverb up when the show is running because there often has to be a change in line levels and balances between the effected sound and a dry sound and this may cause feedback during your show. If anything, start off with a more "wet"sound and let the engineer know that if the audience is smaller than expected, he/she should adjust the reverb balance to the levels you desire.

SOUNDCHECKING STAGE PATTER. Most acoustic/folk musicians use stage patter as part of their performance. Be aware of the difference between your speaking voice and your singing voice. In fact, sound check one of the stories that you tell with the usual volume that you use during your performance. This will give the sound engineer the settings that will have to be adjusted between songs. It is probably best that the reverb be muted while you are talking.

SET LIST FOR SOUND ENGINEER

Give the sound person your set list. Besides type of song or instrumental, each song should include notes that are relevant to your mix such as "big reverb on voices" or " mandolin solo". Don't overload each song with requests for a lot of changes. If you keep it simple, the sound engineer will be able to help you create the desired effect you are looking for. If you have a song with a great change in levels, it is advisable that you approach this during sound check.

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Looping (Acoustic Instruments) (for a DJ)

Here are some proposals to help colleagues who want to loop acoustic instruments like percussion. It is written for the ECHOPLEX, but should be similar for other machines.

The problems are: ·

The difficulty to hear the correct volume of the instruments and mix them correctly into the loop.
The danger of feedback. Even if the volume is not as high as to create a oscillation, sound from the monitor enters the mike again and is recorded again into the loop, deteriorizing the sound quality and making clean Replace impossible. Pressing OVERDUB really just while playing improves this situation a lot. The OVERDUB Mode SUS helps to do this.
Crosstalk from other instruments and noises into the loop. If the drums are playing next to the percussionist or loud in the monitor, the snare will be looped. Whether this is a problem or not depends on the music and the way the loops are used.
The physical distance of the instruments and the various postures of playing can make positioning of the pedal board difficult. ECHOPLEX pedal boards can be used in parallel. You can have the keys in several places, even in several forms (to operate by knee or elbow, for example)

Basically there are three ways to go:

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1.. A microphone/sound system only for the loop
This is the most simple, suitable for rehearsals, small shows:

· Connect a clip microphone directly to the ECHOPLEX and keep clipping it to the instrument you want to loop. You can prepare a piece of wood on each instrument so you know exactly where to clip the mic and how loud it is going to be.
Mark the correct position of the Input control for each instrument so you can adjust it quickly before you play. · Wind up the Mix control to "loop".
Connect the ECHOPLEX output to some amplifier (preferably not a guitar amp!) and regulate the volume, so the instruments appear about equally loud direct and from the loop. The sound will not be the same, but this can be interesting even.

2. Mixing on stage or by the band's sound man

The musician or a smart sound man controls the loops from the mixing desk. The sound can be equalized for each instrument and monitored. Thus, the difference between the original and the looped sound becomes small.
The ECHOPLEX is connected like a reverb to an Aux send and returns to a channel (remember to close the Aux in that channel!). The MIX control is way up to "loop". The problem is the position of the Aux send control in every channel. To optimize cross talk, the sound man should only open the channel that is actually going to be looped. With this setup you can for example maintain a groove on the congas (Aux closed) and throw a cymbal into the loop (Aux open) without having the congas looped (except cross talk).
The most perfect solution: Headphones (getting popular anyway!).

3. "Electric" percussion instruments:

Could be MIDI sounds, but that's too cold, sometimes.

Instruments like Korg WaveDrum are much better because they bring through details of playing techniques that cannot be recorded by MIDI, but very well in the loop, because there is no problem with noises and feedback. Maybe we should start inventing "electro-acoustic" percussion instruments in the sense of a electro-acoustic guitar: Little resonance and a pick-up in the right place. The sound can become richer, easier to amplify, and the instrument can be played very dynamically. Also, a simple piece of metal that has no volume but maybe an interesting sound can turn into a new instrument.
In general, this type of instruments will be lighter and smaller, needs less stands, less space on stage, can be accessed more immediately. Piezo pick-ups are cheaper than good microphones, need no clamps. Mixing desks with piezo inputs (impedance > 1 MOhm) are not common yet, but most ordinary ones can be modified easily. This is the most futuristic way. It will take time and efforts, but we will end up there.

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Looping (Feedback use techniques) - for a DJ

The Plex uses a 256 step value and filters it almost every sample so you can smoothly and quickly change it. It is strongly suggested to use a pedal. In longer loops you maybe want to grow only a part of it: For example: Open Overdub and reduce Feedback while opening the volume pedal so the sound you hear from the Loop will be replaced next time around by the one you fade in now. Not very difficult to imagine how it will sound.

Then as your note fades, you open Feedback again and have a phase of the loop as it was before. Replace is a function we have for this, but is to hard for most applications because it chops off/on. With the FB pedal, you do it more creative and smooth. Sometimes in long loops (like 25sec), start increasing the dynamics every turn around, rather taking back one part and then crescendo in to the full part... As it does not make sense to infinitally increase the content of the memory, we reduce automatically the FB a little while Overdub is on.

This prevents from the worst noises when somebody forgets that Overdub is on. When you reduce FeedBack, reduce loop time, too! (Million times executed experience - how it works for me): Most music (and stories in general) has its static phase (contemplation, solo) and its dynamic phases (walking, discovering). Obviously, FB open is for the static and reduced for the dynamic phase. Since in the static phase you have time, you will multiply and increase loop time to make the loop more interesting, maybe less obvious.

Then, when you enter a dynamic phase, its a drag, because changes take to long, or take a too radical reduction of FB which cuts the flow. So you reduce FB little, but also reduce loop time! If the loop is rather an educated one with a harmony sequence, built with Multiply, you will apply Multiply by 1 or 2 when the basic harmony comes back. The loop stays on this base, maybe 4 or 8 times shorter, which gives you the chance to change it gradually and then build (use Multiply again) a new harmony sequence. If the loop is rather of the anarchistic/ambient kind, you can reduce it with Unrounded Multiply, which is called by the RECORD following the MULTIPLY key. This way you can cut out any bit, as short as you want, maybe even applying Unrounded Multiply 2 or 3 times in a row, to really chop up the worm before the part with the heart grows again with more heads even.

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Overcoming Stage Fright

From butterflies to panic attacks, stage fright is nothing more than a fear of the unknown. How will the audience react? Will I forget the lyrics or sing out of tune? Will my voice hold out? Since none of these questions can be answered beforehand, anxiety builds. Preparation can help.

If you are well rehearsed and in good physical condition, any reasonable person would expect to perform well. But stage fright is not a rational fear, and performers are not reasonable people. It does not matter if it's all in the mind; dwelling on worst-case scenarios puts a real clamp on the voice.

Trying to talk yourself out of these mental tail-spins only makes things worse. What's important to remember is that anxiety means you care. Apprehension is