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    Studio manual

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    Studio plan

    RED front door

    YELLOW fire exitsORANGE - desk

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    Inside studio A

    1 2

    3

    4

    5

    6

    7

    8

    9

    1. Multi-channel recordings (replaces thecomputer) you can record audio through this

    to complete a track ready for mixing.2. These are sound modules. They have built in

    sounds ready for the computer to use. Theyare the analogue version for loops and

    samples in logic.3. An audio interface gives you the ability to

    connect professional microphones,

    instruments and other kinds of signals to acomputer, and output a variety of signals aswell. They each hold 8 inputs per device.

    4. MIDI Keyboard5. This is the mixing desk. Essential for mixing

    tracks, it contains several different channelstrips where effects can be applied and levels

    of individual tracks can be set so that noinstrument overpowers another during themix.

    6. Speakers7. Screen8. This is the patch bay. It is placed on the end of

    the mixing board and contains several plug-ins. This is where bantam leads are conjoined

    with the outboard in order to apply analogueeffects to individual tracks.

    9.

    This is known as the outboard equipment.This is where the effects such as EQ,compression, limiters etc can be added onto

    the track in an analogue style avoiding logicpresets. This works by connecting the patchbay on the mixing desk to the outboard using

    bantam leads.

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    Speakers/Monitors

    So what are speakers/monitors?

    Speakers are transducers that convertelectromagnetic waves into sound waves. Thespeakers receive audio input from a device such

    as a computer or an audio receiver. This input maybe either in analogue or digital form. Analogspeakers simply amplify the analogelectromagnetic waves into sound waves. Sincesound waves are produced in analog form, digital

    speakers must first convert the digital input to ananalog signal and then generate the sound waves.

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    What types of speakers are

    there?

    There are hundreds of different types of speakersall over the world. Some are used for differentreasons. For example, KRK E8s are classicmodels used in recording studio as they producethe best quality sound. Whereas Bose companion2 are used for a home bedroom set up as they areused purely for playing music not for high qualitysound mixing. Below are several models both highquality end speakers and low quality end. I willshow you a comparison between both and show

    which is the better model to use for the rightenvironments.

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    Comparison of speakers

    High-end quality for recording studios;

    Achieve a level of clarity and sonic accuracy that is unprecedented

    The E8B utilizes two discrete Class A / AB power amplifiers - oneeach for HF and LF - with symmetrical heat sinks built into the sidesfor cool, reliable operation.

    Consisting of a beryllium and aluminum tweeter that takes highfrequency performance beyond the physical limitations of moreconventional tweeter

    Multilayered Kevlar/Rohacell woofer for superior reproductionaccuracy.

    With its stunning, highly functional enclosure featuring thick, non-parallel, internal walls, a non-slip, sound absorbing rubberized base,and magnetic shielding for use in close proximity.

    1699

    KRK E8B

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    High end quality speakers

    Similar design characteristics of our Expose E8B

    Consists of three models, the VXT4, VXT6, and VXT8.

    Providing low resonance, improved structural integrity and

    extended low-endSlotted ports for reduced port turbulence,

    The domed tweeter and the new woofer design both havevented formers to remove heat and reduce the effects ofpower compression, enabling these monitors to maintain

    their sonic signature.

    455-459 per speaker or 920 eachVXT8s

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    Low end quality speaker

    Behringer MS20 monitors

    20 Watt field between both of them

    85

    Can connect straight to mp3 player, soundcard

    etc

    Tweeters and woofers are said to produce an

    ultra-line frequency response.

    Lots of headroom in the amplifiers

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    Difference between active and

    passive speakers.

    A passive speaker doesn't have a built-in amplifier.

    It needs to be connected to an amplifier through

    normal speaker wire. A passive speaker level signalhas been amplified enough to drive the speakers

    sufficiently. However, an active speaker has a built-

    in amplifier and are fed by a low-level (line-level)

    signal passed along an interconnect cable

    originating at a preamplifier or controller. Becausethe amplifier is an active electronic device it needs

    power, therefore, you have to put any active

    speakers near a power socket. For most audio

    systems, the subwoofer is probably going to be

    your only active speaker although you can also findpassive subwoofers.

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    Audio Interfaces

    What is an audio interface?

    Audio Interfaces take various audio inputs and convert them into digital

    audio data so a computer can read them. This piece of hardware willcertainly use nearly all the inputs provided in a recording studio. Digital

    audio, electrical analog audio, and MIDI data will all come together in thissoundcard as it will acts as the central hub for all the various audio signals

    in your setup. By handling all the audio input and output signals of yoursystem, it performs the same function as the soundcard placed within in atypical computer. The main differences are that Audio Interfaces are

    analogue whereas the built in soundcard is MIDI. Another difference isinterfaces typically have mic preamps onboard, providing phantom power

    for condenser mics, and have easy-access external inputs and outputs.

    The link below is a video on what audio interface is suitable for different set-ups. It also explains how to use an audio interface and why it is vital in your

    studio set up.

    http://www.youtube.com/watch?v=dvkbwy1nnyQ

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    What models are there?

    The 2408 is a compact unit with extensive metering on the front panel and all the

    necessary digital and analogue connectors on the rear.

    It contains unbalanced phono connectors, albeit gold-plated, for the analogue ins and outs.

    The converters are full 20-bit devices and the internal signal path is 24-bit with support for

    both 44.1 and 48kHz sample rates.

    There are three additional outputs comprising one pair of balanced jack analogue outs as

    well as a secondary and a stereo headphone jack. These pick up whatever signal is

    assigned to analogue outputs one and two for monitoring or recording purposes.

    829-840

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    More models

    2 x 2 USB 2.0 audio interface with 2x D-PRE and 192 kHz

    support

    https://www.steinberg.net

    The website provides all the information accompanied with

    the model and also features prices, compatibility, software

    etc

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    What is a mixing desk?A mixing desk is simply a console where

    sound signals are mixed during recordingor broadcasting. It is used for routing,

    changing the level, timbre/dynamics of

    audio signals. A mixer can mix analogue

    or digital signals, depending on the type

    of mixer signals. The role of a mixing deskis to

    The channel stripOn a mixing desk there are 36 channel

    strips, half the board used for recording

    and the other half used for mixing andplay back. Below is a detailed annotation

    for a single channel strip on the mixing

    desk. I have hyperlinked a website below

    where it shows the whole mixing desk

    annotated and everything youll ever needto know about the use of a mixing desk.

    Mixing Desk

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    Patch Bay/Outboard

    A patch bay is a device or unit featuring a number of jacks,usually of the same or similar type, for the use of connecting androuting circuits for monitoring, interconnecting, and testingcircuits in a convenient, flexible manner. In recording studios,patch-bays are widely used to aid the connection of differentdevices, such as microphones, electric or bass guitar, electronicinstruments, effects (e.g. compression, reverb, etc.), recordinggear, amplifiers, or broadcasting equipment. The effects are onthe outboard, which, once is connected to the mixer via the

    patch bay can be applied either in your recording or in your mix.The outboard effects include reverb, delay, phase, flange etc.They also have processors like compression, EQ, limiters, gatesetc.

    Outboard compressor;

    This is the same compressor as the internal one on acomputing software such a logic. The only difference is that thisis external as it is placed in the outboard. This means that it canbe applied in live sound situations rather than having to

    recording and apply when editing. A compressor is simply aplug-in or electrical device which reduces the dynamic range of asounds signal.

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    Channel Strip

    Section 1

    Gain - to adjust

    when recording.

    Phantom power+48v.

    Sub

    Low Cut Cutsbelow 100Hrtz

    Tape Volume -Additional volume

    when playing back.

    Line Whenrecording in an

    instrument or mic.

    PhaseCancellation

    Especially usedwhen recording a

    snare. Phase on

    for better signal

    Phase cancellationworks when the signal

    is going the same way.For example

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    Section 2 :

    Noise Gate

    Flip

    Range

    Channel

    In

    Frequency

    Threshhold

    Release

    Hold

    Open

    Close

    Hold

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    Section 3

    EQ

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    Section 4: AUX

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    Section 4 :AUX

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    Monitor Faders

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    Leads

    Leads can be Balanced or Unbalanced within a studio.

    Balanced Tip ring sleeve

    Provides a better signal with less noise.2 LinesXLR

    Balance leads consist of three wires, one ground wire like the unbalanced and the other two

    are copying the same signal. For example, when you are singing the two leads are carrying aduplicate copy of your voice, however, they are not phase corrected as they have a flippedpolarity. When they reach the end of the wire they flip back as any interference will cancel

    itself out.

    Unbalanced -One line in between the metal.Patch cord/ RCA Connection.

    Also has three wires.

    Unbalanced, unlike the balanced cable, only has two wires. One ground wire and one wire forthe signal. Because there is only one copy of the signal its more likely to have less signal and

    more noise. Therefore the signal to noise ratio might be quite high. Normally keyboards orguitars use unbalanced cables, RCAs or Patch Cords.

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    LeadsXLR;

    XLR cables are used with microphones

    because they have a durable construction.

    They can work plugged into an unbalancedinput but because microphones generate low

    signal levels, these low signal levels needlots of gain to bring the levels up. XLR cablescan conduct 48V of phantom power.

    RCA;

    RCA (can also be known as phonoconnectors) is an unbalanced mono

    connection used for analog and digital audioconnections. Red lead is for the rightchannel, White is for left and yellow is forvideo. Found on sound cards and mixingdesks.

    Thunderbolt cables;

    Helps with high speed connections like

    with hard drives and network interfaces.

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    LeadsJack Connectors/ Jack Plugs-

    They are also known as phone connectors, they

    come in two different sizes, 1/4 inch and 1/8 inch.

    The 1/8 inch mono is known as the TS because its

    just the tip and sleeve with one line around it, this

    is unbalanced. The 1/8 inch stereo is balanced, its called TRS

    because it has tip, ring and sleeve, with two linearound it. Mainly found with headphone

    connections and sound card connections.

    The1/4 mono is unbalanced, its an alternative to

    an XLR so found on microphone input

    connections, guitars and keyboard cables. This

    can also be used for DI. The 1/4 TRS stereo is balanced, can be found

    used with mixing desks, audio interfaces etc

    (recording equipment).

    !- 6.35mm TRS cable

    1/8 3.5mm mini jack

    Bantam leads

    Connects the outboard equipment to the patch bay on

    the mixing desk.

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    How is it all connected?

    It can be very confusing when connecting all the equipment together as you have

    to make sure you have it set up correctly in order for it all to function. First of all you

    need to ensure that all your computer, speakers, monitors and any other piece of

    useful equipment is plugged into the mains and is functioning. The musician's will

    record most instruments in the live room, so there needs to be a connection that can

    allow you to record from the live room into the studio. This is called a stage box, the

    stage box has as little as 16 inputs up to 32+ to allow as many instruments as you

    need. The leads for the stage box is connected to the outboard in the recordingstudio. From there you use the bantam leads to connect to the patch bay on the

    mixing desk to ensure each input is in the right place. For example the stage box

    would have an active input 1, which means the bantam lead connects the outboard

    input 1 to the patch bay input 1. Once is all connected you can then select the

    appropriate input on logic and can adjust the levels further on the soundcard. Once

    you have done this arm your track and you should be getting strong signal. Other

    ways of connecting could be through a DI box as previously mentioned. This can berecorded in the studio rather than the live room as it is direct injection to the

    equipment. This will show level on the logic channel and also the VU meter which can

    be adjusted if needs be.

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    Direct Injection

    A DI Box is a device typically used in recording

    studios to connect high resistant unbalanced

    output signal to a low resistant balance inputvia a XLR cable. DIs are frequently used to

    connect an electric guitar or electric bass to a

    mixing console microphone input.

    DI's can be used on electronic/acoustic

    instruments and pick ups that do not contain

    an XLR balanced output. An example being anelectric keyboard that needs to be connected

    to a mixer board or directly or through a jack

    lead. Another example is electric or bass guitar

    as it is mixed through a mixing consol through

    to a main monitor or mixer.

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    V U Meter

    and

    Signal levelsVU- A volume unit (VU) meter or standard volume

    indicator (SVI) is a device displaying a

    representation of the signal level in audioequipment. This can also be connected with

    the DI box or pre amplifier. In some cases theVU meter is built in to the DI box when using a

    bass or electric guitar so you can instantly seethe audio level coming through.

    Signal level is the amount of sound acquired

    from a form of audio. It can be shown on VU

    meters on audio interfaces or in softwares.There can also be signal to noise ratio when

    recording in a studio. This is the comparisonbetween the desired sound level and the ratio

    of the level of background noise. This can be

    bad if un-noticed as it can cause distortion andclipping within your recording.

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    Noise GateThe purpose of a noise gate is to mute signals with low volume. Noise gates are often used in live

    sound recordings as they eliminate background noise and bleed when the instrument or vocals are

    not generating signal. This could be before a song starts or when it finishes. Drum micing canproduce a lot of bleed from other drums resonating thats why noise gates tend to be used on

    drums. For example the kick drum mic would only be active during the kick hits while the snare drum

    gate would only be open for the snare hits so that it doesnt leak into the kick microphone. Gates

    work at a slower speed than compression as they are often closed for a long period of time. Gates

    do not eliminate noise altogether they just hide it only becoming active when it is most needed (inbetween songs) and is let through when mixed with signal.

    The controls included in a noise gate are;

    THRESHOLD When the signal falls below the threshold level the dynamic processing start andthe gates begin to close.ATTACK TIME This is the time it takes for the gate to fully close once the signal level exceeds the

    threshold level. Lowest being 10-100(microseconds) and highest being 200ms-1second.

    RELEASE TIME Most known as decay, it is the opposite of attack time.

    HOLD TIME this is the minimum time that the gate is left open after the signal level has fallen

    below the threshold. It prevents the decay of short low signal levels that start after releasing phase.GATING INDICATOR an LED light will usually show whether or not the gate is active.

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    What is Equalization?

    Equalization is the editing of frequencies by HI-FI equipment to increase or decrease volume, bassand treble. This can be done by adjusting the bandwidth or changing the decibels.

    Equalizers are used in recording studios and live sound reinforcement to correct the response ofmicrophones, instrument pick-ups and loudspeakers. Equalization is also used to eliminate unwantedsounds, make certain instruments or voices more prominent, enhance particular instrument's tone, or toavoid feedback in a live show.

    Shelf EQ

    Shelving EQ is a filter that can be used to reduce or increase signals above or below a set frequency.Shelving filters always use/affect the bass and treble in audio. They provide an adjustable boost or cut tofrequencies above or lower than a certain point. Shelving filters require not only a control for selecting theshelving frequency, but also one for selecting the amount of cut or boost applied.

    Parametric EQ

    Parametric EQ allow users to control the three primary parameters: amplitude, center frequency andbandwidth. The amplitude of each band can be controlled, and the center frequency can be shifted, andbandwidth ("Q") can be widened or narrowed. Parametric EQ specialise in making more preciseadjustments to sound than other EQs. They are commonly used in sound recording and live sound toprevent feedback.

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    Health and safety within a studio

    When carrying out any kind of project that includes electrical equipment, small spaces, tripping hazards or usingcomputers for extended periods of time it is important to be aware of the possible health safety implications.

    Cables: Cables are obviously a massive tripping hazard in a recording studio, due to the fact that microphones will be plugged in

    to the walls via XLR cable throughout the day and will be moved and swapped around constantly. This means that thistripping hazard will be constantly moving throughout the day so it is important to be aware whilst walking around the liverooms when instruments and microphones are in use to avoid personal injury and equipment damage.

    Sound Pressure: Drums and Amplifiers in small spaces can and usually do reach high air pressure levels that can lead to hearingdamage. It is very important that this is considered in a studio as musicians and producers require the use of their ears for

    their profession. This health and safety issue can be overcome by wearing noise cancelling headphones or alternatively earplugs to reduce the pressure entering the ears. Another option that can be considered is to sit the musician in a separateroom to where the noise source is when recording amplified instruments.

    Fire Hazards: Due to the vast amount of electrical equipment being used in a recording studios there is a small possibility of

    electrical fires if a particular piece of equipment were to overheat. The majority of recording studio walls are covered insound proofing foam and can have poor ventilation due to the lack of noise wished to be let in. These can be potential firehazards. It is important to know all of the studios fire exits so evacuation can be made easy if necessary.

    Use of computers: If using a computer for an extended period of time it is important that you take regular breaks to avoid

    damaging your eyes or even you back if your sitting posture is poor. Always sit up straight with the computer monitor at eye

    level to avoid straining any muscles. The same goes for listening to loud music for extended periods of time as this has the

    ability to reduce the range of your hearing over time if regular breaks are not taken.

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    Mic placements

    Drum kits

    Drum kit;

    The drum kit can consist of as many as 8 microphones or even more ifwanted. When micing up any instrument the placement of the mic is veryimportant and sometimes the polar pattern of the mic is considereddepending on what you are recording.

    The 8 include;

    Overhead pair; These are condenser microphones which can capture a goodambience within the room. These help to give the drum kit a more naturalfeel so when editing it can still maintain its human sound. Models ofoverhead pairs include the neumann k184. When micing these it isimportant that you measure the distance between both drums correctly as itcan cause a delay on your recordings. There are two main options torecord with overheads, an X/Y (or coincident) pair or a spaced pair. The X/Y involves placing the two overheads directly above the kit with the micsnext to each other. This is a good way of capturing the kit with no phasingproblems, as the sound of the kit reaches both mics simultaneously. Tocapture more reverb, spreading the mics apart as a spaced pair is moreeffective. However there is now a risk of phase cancellation this where the

    signal from the source reaches each mic at different times, therebycancelling itself out. To apply phase cancellation take two channels andusing the pan control to pan them separate ways.(3oclock and 6oclock). Ifout of phase, the two channels will sound hollow when heard together. Thesolution is to either move one of the mics until this improves, or reverse thepolarity of one mic. In the logic so the waveform can be inverted after it isrecorded for the same effect.

    Kick drum; This can miced upon personal preference. If you want more attackyou can put the microphone inside the drum althought isolating it will lackdefinition. However if you want a thicker, fuller sound you will place the mic

    directly in front of the hole in the skin or if no skin present just in front of thekick drum. The AUDIX F6 is the standard kick drum mic.

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    Mic placement

    drum kitsSnare drum;

    The snare would usually be micd like the toms, from the top,

    three fingers away and at the edge of the drum and pointing at thecentre. The three-finger rule can help keep the mic away fromunwanted sounds from the edge of the head and hoop, while aiming

    it at the center helps capture the main, thick sound at it willintroduce more 'air' or ambience into the sound. Moving it closer, ifusing a dynamic mic, will introduce low-end as a result of what'scalled the 'proximity effect'. This can be useful to a point, however ifyou get too close, the proximity effect will make the sound sound

    unnatural or unusable .A dynamic mic is usually the chosen

    microphone the Shure SM57 as it has ability to capture the mid-range and handle the high SPLs (sound pressure levels. There canalso be a mic used underneath the snare to capture the rattle. Thiscan cause phase cancellation so you have to ensure you take the

    same procedure like the overheads, of changing the direction of thewaveform so they are simultaneous rather than conflicting. This canby pressing a button in logic before or after recording. It only needsto be applied to one track.

    Hi-Hats;

    Mic placement is key for when micing up hi-hats. It has to 2-3

    inches and placed directly above the centre. Make sure it isnt to

    close as when the drummer is drumming and open hi-hat it could be

    in danger of hitting the mic and ruining a recording. Mics used forthe hi-hats are the AUDIX ADX51.

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    Mic placement

    Drum kitsRack toms;

    From the toms you'll hear their ringing

    resonating onto other tracks as other parts ofthe kit are hit. When micing the toms you canuse either dynamic mics or small condensers.

    Shure PG56S are usually used as they havehigher pressure levels. They should be placed

    facing the centre or slightly off the skin. Youcan get clip on microphones to make this

    easier than having to place them with a stand.

    Floor tom;

    These are the same as the rack toms inthe sense of the microphones used and theplacement.

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    Microphones used for live recording

    KSM44A SM57 SM58

    AUDIX D2 BETA 57A

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    Condenser and Dynamic mics

    Condenser Microphones;

    These are the most common microphones youll find in studios. They have a much greater

    frequency and transient response which means they have the ability to reproduce the speed of an

    instrument. They have a louder output but are much more sensitive to loud sounds. They are

    normally more expensive than dynamic mics as they are better quality. Condenser microphonesrequire phantom power 48 volts, a button. supplied on a mixing board. They can struggle to

    withstand high SPL (sound pressure levels) but are used for overhead micing of drum kits or

    choirs. They differentiate between a large diaphragm and a small diaphragm. A large is generally

    the choice for studio vocals and any instrumental recording where a deep sound is desired. It isessential that you use a pop shield as they can be so sensitive to the P and PH noises that willbe made. Small diaphragm condensers are used for when a solid wide frequency and transient

    response.

    Dynamic Microphones;

    Dynamic microphones tend to be more rugged than condenser microphones as they are resistant

    to abuse thats why they are largely used on stage. Dynamic mics like the Shure SM57 and SM58

    are not only for their good sound quality but the amount of abuse they can withstand. They dont

    require their own power supply like the condenser, although their sound is generally not asaccurate. They have a limited frequency response and can withstand high SPL so are suited forguitar amps, live vocals and drums.