THE TOOLS

Advanced technicians from different fields seem to share the ability to select and use their tools skilfully, achieving consistent high-quality results. Such proficiency commonly results from years of experience, although it may be mistaken for a natural talent. In fact, technicians that develop their expertise through practice may find it difficult to rationalise or explain their actions, as production appears progressively more intuitive and effortless.

Although technical knowledge alone does not guarantee successful results, it may provide the necessary foundation upon which experience may be built. With this in mind, it seems advisable for recordists to study the tools they work with in detail, setting out guidelines for their selection and use. To some, this process could initially be based on familiarity or external influences, e.g. using Neumann microphones to record vocals, as Geoff Emerick did to record the Beatles, although ultimately it would be more appropriate for such important decisions to be based on a balance between intuition, scientific knowledge, experience and the desire to experiment.

The following sections present possible strategies for the selection of recording tools.

MICROPHONES

Microphones are broadly described as transducers or converters of energy. These, alongside loudspeakers, arguably constitute the most important elements in the audio chain, as both perform vital roles at the two ends of the signal path. While the microphone selection process may seem simple and spontaneous to some, it is not uncommon for recording engineers to spend considerable time and effort to ensure the best device is selected for each given job.

While the criteria influencing the choice of transducers may extend well beyond the realm of technical specifications, for the sake of objectivity four main factors are presented here as most important for microphone selection. These are:

•    Transducer type – Bias vs. Fidelity

•    Diaphragm size – Output sensitivity / Bias vs. Fidelity

•    Directionality – Separation

•    Construction – Physical considerations.

MICROPHONE SELECTION CRITERION 1: TRANSDUCER TYPE (BIAS vs. FIDELITY)

A basic understanding of the physical principles of transduction on the part of the reader is expected in this section, where the main objective is to discuss the general characteristics and merits of different transducer types, which are described next.

Transducers may be classified according to their operating principle, falling into the following key groups:

•    Electro-dynamic or dynamic (moving coil, ribbon and printed ribbon)

•    Condenser (Including electret condenser and PZM or boundary)

•    Piezo or crystal-based

•    Other.

The following illustrations depict examples of moving coil and ribbon-based dynamic microphones that are commonly used in music production. The regulated phase example is included here for historical / illustrative purposes, as this type of transducer is not currently popular and its use has become somewhat rare.

Condenser Microphones

Condenser microphones rely on the varying distance between the two plates of a capacitor to operate (variable capacitance).

Condenser-based transducers:

•    Have very light moving parts, which allow for a fast transient response and high sensitivity.

•    Offer an extended high-frequency response.

•    Can be extremely transparent (flat frequency response).

•    Commonly require a (phantom) power supply to operate.

•    Frequently incorporate a built-in amplifier section (valve or FET), used to boost output gain and convert (lower) impedance.

•    May be placed at greater distances from sound sources.

•    Are commonly fragile.

The following pages contain examples of condenser microphones that are commonly used in music production.

Piezo Microphones

Piezo or crystal-based capsules are commonly employed in ‘contact’ microphones or acoustic instrument ‘pick-ups’.

Piezo microphones:

•    Are commonly biased towards high / high-mid pick up, with poor low and very high-frequency representation.

•    Minimize the effect of feedback originating from backline.

•    Work well to capture signals for subsequent replacement, e.g. snare drum contact microphones.

OTHER MICROPHONES

Other types of microphone exist, e.g. carbon, fibre-optic, laser, etc. although these are not commonly used in music applications at present.

MICROPHONE SELECTION CRITERION 2: DIAPHRAGM SIZE (OUTPUT SENSITIVITY / FIDELITY)

The influence of diaphragm size over microphone performance varies according to transducer type. Most authors question the use of large capsules in dynamic microphones, arguing that due to operating principle the benefits of such implementation are minimal, if not negligible. A few designers, such as Bob Heil, believe on the other hand that a larger diaphragm equates to improved transient response, even in the case of moving coil microphones, as long as the parts are made light enough. Regardless of the debate, capsule dimensions have a more tangible effect in the case of condenser microphones, where:

1.  Microphone sensitivity is commonly proportional to diaphragm size.

2.  Off-axis pickup fidelity is inversely proportional to diaphragm size.

As a general rule, the use of a larger sized diaphragm should be considered when significantly small variations in pressure must be captured with great detail, e.g. vocal performances where minute ‘mouth’ noises may add intimacy to a recording. At the same time, the use of small diaphragm transducers is indicated when off-axis sounds must be captured with great fidelity, e.g. large ensembles recorded with few microphones.

It is important to note that sound engineering authors do not seem to agree on the impact of diaphragm size over bandwidth. Considering the series of factors that may influence a microphone’s response, e.g. diaphragm tension, mass density and the sizes of the internal cavity and of the static pressure equalization vent, it does not seem reasonable to assume there is a simple causal relationship between diaphragm size and bandwidth of pickup, i.e. recordists should always check a microphone’s frequency response graph.

MICROPHONE SELECTION CRITERION 3: DIRECTIONALITY (SEPARATION)

Two important questions provide the basis for this criterion:

1.  How much separation between channels is required?

2.  How much natural reverberation is desired?

If in answering the first question it is established that a great degree of separation is required, omnidirectional or bi-directional microphones should be avoided or used with caution (unless the physical isolation between performers in the studio is achievable and practical). In such case, directional microphones, e.g. hypo, super, hyper or standard cardioid transducers represent a better choice, as these may offer considerable rejection of unwanted sounds. This is particularly important in the case of projects where extensive editing may be required, i.e. productions in which song arrangement or structure might change and different instrumental parts may need to be moved or copied between sections.

As far as the second question is concerned, transducers with wider pick-up patterns, e.g. omni and hypocardioid, may capture the natural sound of an environment with more accuracy. In favourable recording settings, the use of non-directional microphones may prove invaluable and should be explored.

It is important to mention that directionality will also affect frequency response (and therefore ‘fidelity’) for both on and off-axis sound pick-up.

The following pages describe common microphone polar patterns, including a brief examination of their use.

MICROPHONE SELECTION CRITERION 4: CONSTRUCTION (PHYSICAL CONSIDERATIONS)

This last criterion is the simplest, where selection is influenced or dictated by the overall physical dimension of the microphone, e.g. when difficult placement or restricted access may require the use of a transducer of small size. In such circumstances a compromise may be required, where sound quality may not be given priority over other criteria.

DI / DIRECT INJECTION BOXES

A DI or direct injection box is a device that allows for high impedance electric or electronic instruments to be connected to low impedance inputs, e.g. through direct injection, the output of an electric bass or guitar may be fed to a console’s microphone input, bypassing the use of an instrument amplifier (backline) and of a microphone.

Passive DI boxes lower the impedance of the source (the instrument) through the use of transformers. The latter also allow for the balancing of signals, i.e. a conversion from unbalanced ‘jack’ to balanced XLR connections. Active DI boxes utilise electronic circuitry to achieve the same impedance conversion as passive devices, while offering improved fidelity, i.e. higher gain and wider frequency response. Such devices are commonly fed by phantom power or by batteries.

DI boxes make it possible for:

•    High-impedance electric or electronic instruments to be recorded without backline amplification.

•    Instruments to be recorded in extreme isolation.

•    Signals to be transmitted over longer distances (through balancing).

•    The subsequent ‘re-amping’ of instrument signals.

It is important to point out that active direct injection boxes may not be perceived as ‘superior’ to passive devices by many musicians who prefer using the latter.

ANALOGUE AUDIO CABLES / CONNECTORS

‘Jack’ / Phone

The following are types of ‘jack’ or phone connectors:

•    3.5 mm / 1/8” Jack or ‘TS’ ‘Minijack’ Two terminals: tip and sleeve

•    3.5 mm / 1/8” Jack or ‘TRS’ ‘Minijack’ Three terminals: tip, ring and sleeve

•    Bantam / TT Three terminals: tip, ring and sleeve – ‘TRS’

•    BPO (BPO316) / ‘B-gauge’ Three terminals: tip, ring and sleeve – ‘TRS’

•    6.35 mm / 1/4” Jack ‘TS’ ‘A-gauge’ Two terminals: tip and sleeve

•    6.35 mm / 1/4” Jack ‘TRS’ ‘A-gauge’ Three terminals: tip, ring and sleeve

•    2.5 mm Jack (not commonly used in current manufacturing).

RCA / Phono

RCA or ‘phono’ connectors commonly contain two terminals (tip and sleeve – ‘TS’).

XLR / Cannon

XLR or Cannon connectors / cables are available in two, three, five and seven-pin configurations.

5-Pin XLR Connectors

5-pin XLR connectors are employed by some dual-capsule stereo microphones.

7-Pin XLR Connectors

7-pin XLR connectors are used for the connection of some valve microphones to their corresponding power supplies.

Speakon

There are three common types of speakon connectors:

NL2 – containing two terminals.

NL4 – containing four terminals.

NL8 – containing eight terminals.

‘Banana’

‘Banana’ connectors contain one terminal.

D-Subminiature / D-Sub

There are several types of multi-pin D-subminiature connectors, e.g. DE9, DA15, DB25, DE26, DC37, DD50. The most common type of audio D-sub connector is the DB25, which contains 25 terminals.

EDAC / ELCO

EDAC or ELCO connectors are available in 20, 38, 56, 90 and 120-pin configurations.

It is important for recordists to be able to assemble audio cables by soldering conductors to their respective connector terminals. Soldering is an invaluable skill, particularly for those working in isolated residential music production environments. The following pages describe the conductor / connector configuration for audio cables that are commonly used in music production.

NB All balanced output to unbalanced input connections depicted here correspond to impedance-balanced or transformer-based balanced devices feeding unbalanced inputs. In the case of active-based balanced devices feeding unbalanced inputs, the ‘ring’ of ‘jack’ connectors or ‘pin-3’ of XLR connectors should be left unconnected or ‘floated’. All unbalanced feeds to balanced inputs should use 3 conductor cables and incorporate a ‘cold’ to ‘ground’ link on the output side only.

DIGITAL AUDIO CABLES / CONNECTORS

TDIF (Tascam Digital Interface)

TDIF cables commonly terminate at 25-pin D-Sub connectors.

S/PDIF (Sony / Phillips Digital Interface)

S/PIDF protocol employs either coaxial low impedance (75 ohms) cables and RCA connectors, or fibre-optic cables and TOSLINK connectors.

ADAT Lightpipe

ADAT Lightpipe utilises fibre-optic cables and TOSLINK connectors.

AES / EBU

AES/EBU digital cables comprise low impedance (110 ohms) conductors that commonly terminate at XLR 3-pin or D-Sub connectors.

MADI

The MADI protocol employs low impedance (75 ohms) conductors and BNC connectors or, less commonly, multimode fibre-optic cables.

OTHER CABLES / CONNECTORS

Word Clock

Word clock cables utilise low impedance (75 ohms) conductors and BNC connectors.

MIDI

MIDI cables utilise three conductors and 5-pin DIN connectors, hence only three of the five pins are wired (pins 2, 4 and 5). MIDI cables should run for a maximum of 15 metres or 50 feet approximately.

MICROPHONE PREAMPLIFIERS

The main role of the microphone preamplifier is to raise microphone level signals to line level, although some units may offer extra features such as equalisation and compression. Microphone preamps differ in sound and while some devices may be broadly described as ‘transparent’, others alter the nature of audio material noticeably. In addition to that, equipment deemed as ‘transparent’ will commonly respond to quiet and loud sounds in a non-linear fashion, e.g. with increasing harmonic distortion.

A preamp’s alteration of timbre must not necessarily be seen as a negative side effect of amplification, as experienced engineers frequently attempt to ‘shape’ microphone signals before routing them to the multitrack recorder, e.g. via the use of equalisation. With that in mind, it appears as a sensible procedure for recordists to audition a few different preamplification units and to do so at ‘clean’, i.e. conservative input signals, and at higher, ‘overdriven’ levels before committing to a signal chain.

The following page contains a few examples of common microphone preamplifiers used in music production.

EQUALISERS

Equalisation may be described as frequency (band) specific level control and it is commonly used to alter the timbre of audio signals or the response of sound reproduction systems. Equalisers are frequently utilised to repair or enhance recorded material, improving tone and clarity (balance), and are also used creatively to personalise music productions. The following is an overview of equaliser types and their applications.

Filters

Filters are primarily used for the removal of unwanted signal content, e.g. rumble, hiss, etc., providing an increasing level of attenuation above or below a given frequency. The rate of this attenuation is given in decibels per octave (filter slope) and it commonly varies between devices. Low and high pass filters may present no variable controls, i.e. incorporating fixed cut-off frequency and slope, or may offer selection facilities over:

•    Frequency

•    Slope (mostly found in digital units, whereas analogue filters commonly present fixed slopes).

Shelving Equalisers

Shelving equalisers allow for (practical) constant attenuation or boost above (high-shelving) or below (low-shelving) a given frequency. Shelving EQ labelling methods vary considerably between manufacturers, although it seems sensible to adopt the ‘stop frequency’ as the nominal frequency of a given device (as AMS Neve and other manufacturers do), i.e. the frequency where boost or attenuation has reached a maximum ‘stable’ level.

Budget shelving units offer simple control over gain, commonly labelled as ‘bass’ or ‘treble’, while professional devices offer selection over frequency and gain.

Peaking / Peak Equalisers

Peak equalisers, sometimes also referred to as ‘bell’ equalisers, are usually found affecting the mid-range band of ‘budget’ or semi-professional consoles. This type of EQ offers variable control over gain, but no frequency or bandwidth selection facilities.

‘Sweep’ Equalisers

Also commonly referred to as ‘bell’ equalisers, sweep EQ units are similar to ‘peak’ devices, while offering added control over frequency selection. NB ‘sweep’ equalisers do not offer control over bandwidth.

Semi and Fully-Parametric Equalisers

Parametric equalisers allow for user selection of frequency, gain and bandwidth or Q. If the control over Q is continuous, devices are said to be of fully parametric design, while if the same control is non-continuous or stepped, they are labelled as semi-parametric.

DYNAMIC RANGE PROCESSORS

Dynamic range processors may be broadly described as gain riding devices, providing automatic control over the level of signals. This control may be effectively used to reduce (compressors and limiters) or extend (expanders and gates) the dynamic range of audio material.

Compressors / Limiters

Compressors / limiters are used to reduce the dynamic range of signals and may incorporate all or some of the following user-adjustable parameters:

•    Threshold – The level at which a compressor will initiate gain reduction, i.e. the level at which signals will start being affected by a compressor (possibly influenced by a ‘knee’ function).

•    Ratio – A gain reduction ‘intensity’ control (x:1), this parameter determines the proportion between input and output levels above the threshold (overshoot).

•    Attack – A time-related function determining the speed at which a compressor will reach its user-defined ratio of gain reduction (x:1) once input signals rise in level above the threshold. The primary function of the attack parameter is to determine the speed at which a compressor will start working once signals overshoot the threshold (although attack still applies for further increases in level above the threshold).

•    Release – A second time-related function determining the speed at which a compressor will diminish its gain reduction when input signals fall in level from previous readings above the threshold. The primary function of the release parameter is to determine the speed at which a compressor will stop affecting signals once their level drops below the threshold (although release will also affect decaying signals above the threshold).

•    Knee – This parameter may be described as a fine-tune control over threshold and attack, i.e. a soft knee function effectively lowers the threshold and slows down the attack of a compressor.

•    Make-up gain – A post-signal processing level control commonly found at the output stage of compressors.

•    Input level – On devices with no time constant controls, i.e. no attack or release, this function dictates the extent of compression by raising or lowering the level of input signals above or below a fixed threshold.

•    Output level – On devices with no time constant controls, this function dictates the overall output level.

Although the gain-reduction cell corresponds to a portion of a larger, complex system, compressors are usually described according to their gain stage, falling (mostly) within the following categories (see page 78):

•    Delta-Mu (Vari-Mu)

•    Opto-Electrical

•    FET

•    VCA

Page 75 depicts a comparison between the effects of compression and those of fader riding.

Expanders / Gates

Expanders / gates are used to increase the dynamic range of signals and may incorporate all or some of the following user-adjustable parameters:

•    Threshold – The level at which an expander / gate will stop gain reduction, i.e. the level at which a gate will ‘open’.

•    Ratio – A gain reduction ‘intensity’ control (x:1), this multiplier determines the proportion between input signal undershoot (the portion of the signal below the threshold or below threshold minus hysteresis) and effective output level.

•    Attack – A time-related function determining the speed at which an expander / gate will stop gain reduction completely (gate fully open).

•    Hold – The length of time in which an expander / gate will remain inactive, i.e. delay its gain reduction, once signals undershoot the threshold (or threshold minus hysteresis).

•    Release – A second time-related function determining the speed at which an expander / gate will realise its full ratio or range of gain reduction (gate fully closed).

•    Range – The maximum level of gain reduction applied to gated signals.

•    Hysteresis – A decibel value describing the difference between the threshold (the point at which the gate will open) and the point at which a gate will start attenuating signals (gate closing).

The following is an illustration comparing the effects of gating with those of fader riding.

MIXING CONSOLES

Audio consoles vary greatly in design and features while sharing the same basic functions, commonly working as:

•    Central routing devices – Allowing for signals to be routed between different (own) internal and external points with minimal need for patching, e.g. from microphone input to multitrack send.

•    Signal processing and amplification devices – Offering facilities for level matching and signal manipulation, e.g. equalisation.

•    Signal summing devices – Making it possible for multiple signals to be summed or combined.

Console Types

Mixing consoles may be classified according to design, falling into one of the following categories:

•    Line mixer – A very basic device with a single set of I/O (input / output) module faders and no microphone preamps. Line mixers are commonly used for the summing of line level signals, e.g. the back end of DAW-based mixing systems.

•    Split console – A mixer that accepts line and microphone level signals, while still offering a single set of I/O module faders. These are commonly operated with half the number of channels sending signals to the multitrack recorder, while the other half is used for monitoring (multitrack returns).

•    Semi in-line console – A console that accepts line and microphone level signals and offers two sets of faders per module, corresponding to the two different signal paths (channel and monitor or Mix B). Semi-in line consoles commonly present a ‘tape input’ that may be routed to monitor or Mix B paths (record) or to channel paths (mixdown).

•    In-line console – Similar to a semi in-line console, although a true in-line mixer has the output of its corresponding multitrack machine feeding both (channel and monitor) paths simultaneously. The use of a patchbay is usually required for the operation of this type of console.

•    Matrix console – Commonly used for live sound stage monitor mixing, matrix consoles may offer a considerable number of auxiliary busses, but no linear channel faders (not commonly used for recording purposes).

The illustrations that follow depict basic set-ups incorporating the different types of mixing consoles described in this section.

Console Elements

Mixing consoles may be very broadly depicted as a collection of channels and busses interconnected via routing with flexible monitoring capabilities. Channels commonly carry signals originating from individual sound sources, e.g. a microphone, while busses are summing points where multiple signals are combined. This allows for numerous sound sources to feed a single input device, e.g. two bass drum microphones routed to the same track, guitar and snare drum signals sent to the same reverb unit, multiple signals summed and sent to a headphone amplifier, etc. It is important to note that the monitor or Mix B paths of in-line consoles may be simply described and treated as alternative channels that carry line level signals that commonly originate from recording devices. This may be easier to understand when one works with a digital console, where signals ‘going to’ and ‘returning from’ the recorder are usually assigned to channels on different layers, e.g. Channels 1 – 24 for ‘send’ and Channels 25 – 48 for ‘return’. In the case of analogue consoles, these two ‘layers’ translate as two faders (or a fader and a rotary potentiometer) on the same I/O module.

The following sections contain a brief description of common mixing console elements and their functions. It is important to note that some digital mixers may offer extra facilities such as analogue-to-digital and / or digital to analogue conversion, internal effects processing, etc.

Input / Output (I/O) Modules or Channel Strips

Common I/O modules or channel strips incorporate the following sections:

1.  Input Section

The input section of a module corresponds to the entry point to its channel path. This section may include the following controls:

•    ‘Mic’ gain – Microphone preamplifier level control

•    Line trim – Line amplifier level control

•    Input flip – Channel input selection (microphone or line)

•    Sub-Group – This switch assigns the signal from a group buss to its corresponding channel path, e.g. group buss 1 will feed channel 1, etc.

•    48 V – Phantom power

•    Pad – Fixed microphone input level attenuation (– 20 to – 30 dB)

•    Polarity (also referred to as ‘phase’) – Used to invert the polarity of signals

•    High-pass filter – Commonly with fixed cut-off frequency if placed here.

2.  Auxiliary Section

The auxiliary section of the I/O module provides access to the auxiliary busses commonly used for effects sends, headphone mixes, etc. This section may contain some or all of the following controls:

•    Source selection, e.g. large or small fader / channel or monitor (Mix B) path

•    Pre or post-fader sourcing selection

•    Send to auxiliary on/off

•    Auxiliary send level controls (each corresponding to a given mono or stereo buss)

•    Auxiliaries pairing switch (which may turn a level control into ‘pan’).

3.  Insert Section

Inserts are commonly used as a means to access outboard equipment for the serial processing of signals that can be improved or enhanced, e.g. inserts may be used to equalise vocals that lack ‘sparkle’ or ‘air’, where the original ‘dull’ microphone sound is enhanced before it is recorded.

Insert sections should ideally offer:

•    Channel or monitor path (Mix B) assignment selection

•    Pre or post-EQ placement option

•    An ‘insert in’ or ‘bypass’ function, allowing operators to compare the original and the processed versions of a signal.

4. EQ Section

Professional mixing consoles frequently incorporate multi-band equalisers and cut filters in their I/O modules. The type of equalisation offered may vary according to frequency band, where shelving or sweep components are commonly used at the two ends of the spectrum (low and high bands) and semi or fully-parametric ones manipulate the mid-range bands (low and high-mid). EQ sections should ideally offer channel or monitor path assignment selection and a ‘bypass’ function, allowing operators to quickly A/B the original and the equalised versions of signals.

5. Dynamics Processing Section

High-end mixing desks may feature I/O module and output buss compressors, limiters, expanders, gates and duckers. In the case of in-line consoles, access to the dynamics section may be assigned to channel or to monitor (Mix B) paths.

The dynamics processors found in I/O modules may offer:

•    Assignment selection (channel or monitor path)

•    Placement options (pre or post-EQ)

•    Side-chain or key input access

•    A ‘dynamics on/off’ or a ‘bypass’ switch.

6. Fader Section(s)

The large fader section of a module is allocated to the channel path, while the small one is assigned to the monitor or Mix B path. This always holds true, regardless of global operation status (Record or Mix), unless a console is capable of operating in ‘fader reverse’. Fader sections commonly contain:

•    A large or a small fader

•    A solo switch

•    A mute switch.

7. The Routing Matrix

Routing matrices offer access to group and / or main output busses. Some professional consoles contain two matrices in each module, while others may only incorporate a single one, commonly placed within or near the large fader section.

The use of routing matrices varies according to console status. In ‘Record’, status group busses are commonly used to feed the multitrack recorder, while in ‘Mix’, group busses are frequently used as a means to access outboard processing equipment (working as ‘extra aux busses’). Routing matrix sections usually incorporate a pan pot for odd/even assignment.

The Master Section

The master section of audio consoles commonly incorporates all or some of the following elements:

1.  Master Group Section

This section may offer linear faders controlling the level of group busses. A small routing matrix and a pan pot may also be incorporated, allowing for the output of group busses to be routed to the main stereo output buss. Some mixers also offer access to the auxiliaries from the group busses and this facility may be explored for the creation of streamlined ‘cue’ mixes based on instrumental groups, e.g. drum groups, guitar group, etc.

2.  Master Stereo (Mix / Output) Buss Section

A linear fader is frequently used to control the main stereo output buss level. In some cases, two separate faders are found controlling the individual level of the left and right channels of the stereo buss. The master output buss section may also contain a ‘mute’ button.

3.  Auxiliary Master Section

This section contains the master controls over the auxiliary busses, commonly including:

•    Auxiliary buss master on/off switch

•    Auxiliary buss master level

•    Auxiliary buss master EQ (commonly shelving or cut-filters)

•    Auxiliary pairing switch.

4.  Effects (Echo / Rev) Returns Master Section

This section contains the master controls over the ‘effects’ or auxiliary returns and it may include:

•    Effects returns master level control

•    Effects returns master on/off switch

•    Effects returns master EQ (commonly shelving or pass-filters)

•    Routing matrix – Allowing for the assigning of effects to the main output buss and/or the headphone mixes.

5.  Tone Generator / Oscillator Section

This section provides tone generating facilities, allowing for equipment testing, alignment and troubleshooting.

6.  Meters

The master or central area of consoles may incorporate output, group and auxiliary buss meters alongside control room (two-track) metering facilities. The meter section may also include a global I/O module metering selection matrix.

7.  Control Room Monitoring Section

The control room monitoring section may include:

•    Monitoring level control – Main level to amplifier or active speakers

•    Mono function – Left and right output buss combining function (used for checking polarity coherence)

•    Dim – A monitor level attenuation switch

•    Monitoring source matrix – Selection of control room monitoring sources, including the Mix Buss, auxiliary busses, ‘cue’ mixes and external inputs

•    Speaker selection.

8. Communications Section

This section offers control over the communication between the control room and the studio (live room). Commonly referred to as the ‘Talkback’ section, this area may include:

•    A built in microphone

•    Level control

•    Talkback and return talkback on/off switch

•    Talkback and return talkback level controls.

Console Signal Flow (Block) Diagrams

The path that connects a transducer to the recording device may vary in size and, consequently, in number of gain stages. It is extremely important for technicians to have an understanding of signal flow, as without such knowledge it would be difficult, if not impossible, to optimise signals before committing them to the storage medium. Audio chains can be as basic as a single microphone connected to a computer interface with a built-in preamplifier or as complex as what is found in commercial multitrack studios built around large format consoles. In all cases, recordists must consider the variables that may affect sound quality and ensure that the path to the recorder is as short as possible and free of unreasonable gain changes in order to maximise the signal-to-noise ratio.

Home studio recordists who work with single sound sources in isolation commonly have little to worry about regarding signal flow, i.e. their signal paths are short, with one gain stage corresponding to the audio interface’s ‘mic’ preamplifier. The picture is considerably different when engineers choose to record entire groups simultaneously using a mixing console and outboard processing. In such cases, the use of block diagrams may be essential as such documentation frequently provides technicians with a clear and linear overview of signal flow from console input to multitrack recorder to two-track recorder to loudspeakers, etc.

Signal flow diagrams come in different shapes and sizes, utilising a variety of symbols that may change according to manufacturer. However, a few universal rules seem to apply and are observed in the majority of documents, such as:

•    Signals flow from the left to the right of a page.

•    Routing destinations are depicted as vertical lines.

The following pages present an overview of block diagrams:

PATCHBAYS

Patchbays allow for the easy interconnection between the various devices found in a music production set-up. The flexibility they provide may help increase the efficiency of the recording process significantly, where the sometimes demanding and time-consuming equipment patching procedure is avoided or simplified.

Patchbays vary in kind (as audio connectors) and in size, spanning from models with a few domestic ‘jack’ patch points to large ‘B-gauge’ or bantam professional models. The following are some connectivity options found in patchbays used alongside in-line consoles:

•    Microphone lines – Where the ‘live room’ microphone cables terminate, N.B. the term may appear confusing, as ‘lines’ is used here in place of ‘leads’ or ‘cables’ (not implying line level).

•    Microphone inputs – Access point to the microphone level channel input of the console.

•    Multitrack returns – Where the cables originating from the multitrack recorder output terminate.

•    Channel line inputs – Access point to the line level channel inputs of the console.

•    (‘Off-tape’) monitor input – Access point to the monitor path.

•    Channel insert sends – Where signals may be sourced from for serial processing, e.g. dynamics.

•    Channel insert returns – Access to a point where signals may be replaced with a processed version, e.g. equalised, compressed, etc.

•    Group outputs – Where group buss signals may be sourced from, e.g. for the feeding of effects processors in mixdown.

•    Multitrack sends / Off-buss monitor path input – Access point to the multitrack recorder inputs and to monitor paths (with a ‘send’ signal).

•    Auxiliary or effects sends – Where signals may be sourced from for parallel processing, e.g. reverb, delay, etc.

•    Auxiliary or effects returns – Access to especial console inputs, commonly fed by the output of effects processing units.

•    Main outputs – Where the main stereo (mix) buss signal may be sourced from.

•    Two-track input – Access to the two-track master mixdown recorder.

The following illustration is an example of a patchbay wired for a studio containing:

•    An in-line console offering:

-    Twenty-four I/O modules

-    Eight mono auxiliary busses

-    Four stereo auxiliary returns

-    One main stereo output buss

-    Two ‘external’ stereo control room monitoring inputs

•    Three external effects processors

•    One two-track recorder (for the recording of mixes)

•    A DAW used as a multitrack recorder.

The following pages describe the process of patchbay normalling graphically.

EFFECTS PROCESSORS

A number of effects processing devices are utilised in recording and mixing sessions. These may be grouped according to their principles of operation and split into the following categories:

•    Time-Delay-Based Effects

•    Amplitude-Based Effects

•    Filter-Based Effects

•    Waveform Distortion Effects

•    Other Effects.

The following sections describe common effects processors used in music production.

Time-Delay-Based Effects

Reverberation

Artificial reverberation devices are commonly used to add depth or dimension to signals. Some digital reverb presets are designed to simulate physical spaces, e.g. halls, chambers, rooms, etc., while others are emulations of electro-mechanical devices, e.g. spring and plate reverbs. It is common for reverb units to offer the following user-changeable parameters:

•    Size or decay – The overall dimensions of the space or of the electro-mechanical device being simulated, i.e. the length of reverberation.

•    Pre-delay – The time separating the ‘dry’ or direct sound from its first early reflection.

•    (HF) Damping – A level attenuator affecting the high-frequency content of a reverberation effect.

•    (LF) Damping – Similar to HF damping, LF damping allows for the control over the level of low-frequency content.

•    Crossover – The point where the spectrum is divided into high and low frequencies, this parameter affects high and low-frequency damping.

•    Density – The spacing between echoes and, therefore, the total number of reflections that fill up the reverberation decay time.

•    Diffusion – The overall geometry of the reflection cluster, i.e. how unevenly the spacing between reflections develops over time.

•    Mix – The relationship (ratio in percentage) between ‘wet’ and ‘dry’ sounds.

Echo / Delay

Although similar to reverberation, echo or delay effects are commonly used in a more creative or less realistic fashion, due to the discrete or discernible nature of the reflections they produce. The following are common echo / delay parameters:

•    Time or delay – The gap between echoes or copies of the ‘dry’ signal.

•    Feedback – The total number of echoes. NB negative feedback implies the echoes are opposite in polarity in relation to the ‘dry’ signal.

•    (HF) Damping – Same as in reverberation.

•    (LF) Damping – Same as in reverberation.

•    Crossover – Same as in reverberation.

•    Mix – Same as in reverberation.

Amplitude-Based Effects

Tremolo

Tremolo units alter the level of signals in a cyclical manner, generally producing a ‘stutter-like’ effect. Common tremolo parameters include:

•    Depth – The amount of amplitude modulation, i.e. the intensity of the effect, dictated by the level of the modulating LFO wave.

•    Rate or speed – The speed at which amplitude is modulated or altered, i.e. the frequency of the LFO.

•    Waveform – The shape of the LFO wave, e.g. sine, square, triangle, etc.

Ring Modulation

The principle of operation of ring modulators is similar to that of tremolo devices, although a high-frequency oscillator (or an upper range LFO) is used. Ring modulators generally produce metallic or robotic sounding results. The parameters found in ring modulators are similar to those of tremolo units.

Auto-Panning

Auto-panners move signals dynamically across the stereo field. Auto-panning device parameters include:

•    Depth – The intensity of the panning effect.

•    Rate or speed – The speed of panning.

•    Waveform – The shape of the LFO wave, e.g. sine, square, triangle, etc.

•    Width – How much of the stereo field is covered by the effect.

Filter-Based Effects

Phaser

Although sometimes described as another by-product of delay modulation, phasing is effectively achieved through the use of sweeping notch filters, as an altering of delay time would incur in pitch shifting (traditionally not a characteristic of the so-called ‘phaser’ effect).

Phasers also employ LFOs (used for the modulation of filter cut-off frequency) and present the following user-controllable parameters:

•    Depth/Sweep – How far the notch filters will be swept, i.e. the amplitude of the LFO.

•    Rate or speed – The speed at which the notch frequencies are swept, i.e. the frequency of the LFO.

•    Waveform – The shape of the LFO wave, e.g. sine, square, triangle, etc.

•    Feedback/Resonance – An internal feedback control, i.e. the output of the filters is fed back into their input.

Wah-Wah

The effect known as ‘wah-wah’ is accomplished through the use of a low-pass filter with a resonant peak. The filter’s cut-off frequency may be modulated ‘manually’, e.g. by a foot pedal, or automatically by an LFO.

‘Auto-wah’ effects processors may also utilize envelope following circuitry as a source of modulation, i.e. the cut-off frequency is made directly proportional to the amplitude of the input signal.

Vocoder

In the vocoder effect, the characteristics of one signal, e.g. vocals, are superimposed onto those of another, unrelated ‘carrier’ signal, e.g. a synthesiser.

Waveform Distortion Effects

Overdrive / Distortion / Fuzz

The original idea behind the overdrive effect was to provide a gain boost to instrument level signals, e.g. electric guitars, on the way to the amplifier. This gain boost was set so the input stage of the amp would be overdriven, resulting in distortion. Overdrive effects processors, e.g. guitar pedals, were devised to generate their own internal distortion.

Distortion and fuzz effects offer progressively more extreme manipulation of waveforms, where the source signal is clipped more radically in either a symmetrical or non-symmetrical fashion. These effects were originally devised to allow instruments to produce harmonically rich or very aggressive sounds at low(er) amplifier levels.

Other Effects

Other effects are found in use during recording / mixing sessions. These include aural exciters, Leslie cabinets, talk-boxes, pitch correction applications, etc. Some of them correspond to a combination of different manipulators into a complex effect, e.g. ‘exciters’ and Leslie, while others are unique in the way they operate.

MULTITRACK RECORDERS

Multitrack recorders (MTR) allow for operators to capture numerous sound sources simultaneously, while storing them in isolation, i.e. several instruments may be recorded at the same time onto their own unique track. This provides great control over productions during the subsequent mixdown stage, e.g. a unique level balance between instruments may be generated, individual sources may be muted, etc.

Analogue tape machines were the first ‘sound on sound’ multitrack recorders utilised in commercial studios, evolving from two to three, four, eight, sixteen, twenty-four and thirty-two track capabilities. With the advent of digital audio, different devices took the place of the analogue tape recorder. Some of them are standalone, e.g. ADAT, RADAR, etc., while some, commonly referred to as DAWs, represent a combination of hardware, e.g. computer/ audio interface, and software, e.g. Logic Pro, Pro Tools, etc.

TWO-TRACK RECORDERS

Two-track recorders are commonly used to capture ‘reference’ or monitor mixes during recording sessions. Such devices are usually fed with the output signal from the console’s main stereo buss and are commonly kept running at all times during production, so as to capture any spontaneous ideas, dialogue between musicians, etc.

It is important to note that many recordists prefer to record the two-track mix onto the multitrack recorder itself.

THE DIGITAL AUDIO WORKSTATION (DAW)I

‘Digital Audio Workstation’ is the collective name given to the components that make it possible for analogue audio to be converted and stored digitally. The number of separate elements that sum up to a DAW may vary from one (a standalone recorder) to many more.

The following are examples of possible DAW configurations:

•    Audio Interface > Computer (running Logic, Pro Tools, etc.)

•    Standalone A/D (and D/A) Converter > PCI Card > Computer

•    Standalone A/D (and D/A) Converter > Audio Interface > Computer.

The Computer / Software Application(s)

A number of software applications are currently utilised for recording purposes, e.g. Cubase, Digital Performer, Logic Pro, Pro Tools, Record, Reaper, etc. With the passage of time, these are becoming increasingly similar and at present the choice of program appears to be a simple matter of taste, i.e. a good recordist should be able to work using any of the popular DAW software applications running on either a PC or Macintosh OS-based computers.

NB Other devices may be found as part of DAW systems, e.g. word clock generators, control surfaces, etc.

THE MONITORING SYSTEM

The monitoring system, alongside the input to the signal chain, i.e. the transducers, should be given the highest priority in the audio path hierarchy. Without a fair and somewhat transparent evaluation of the material being recorded, production teams may work on the tracking stage of a project only to discover subsequently that the content gathered is less than ideal and does not correspond to expectations.

Monitoring systems may incorporate one or many of the following elements:

Power Amplifiers

Power amplifiers raise line level signals to loudspeaker level. This significant increase in voltage levels may be achieved in various manners and ‘amps’ can differ from each other considerably. Many of the factors that contribute to the aforementioned differences, e.g. topology, are beyond the scope of this book, which focuses on providing recordists with basic guidance.

The quest for clean, non-distorted sound reproduction should start with an investigation into amplifier and loudspeaker power. Equipment ‘spec sheets’ commonly display a number of different figures of varying relevance. One of these, ‘continuous power rating’ appears as the most important and realistic, as it describes the ability of a device to continuously handle ‘real-world’ program material. As a rule of thumb, the amplifier should present no less than twice (and ideally four times) the IEC continuous power rating of a connected loudspeaker with matching nominal impedance, e.g. a power amp with a continuous IEC power rating of 400 watts at 8 ohms should be used to feed a loudspeaker rated at 100 watts at 8 ohms (or at the very least a 200-watt power amp should be used). This ensures the amplifier will operate far below stress levels and with plenty of headroom.

Another important specification, known as ‘slew rate’ describes the ability of an amplifier to follow (react to) input transients and it is expressed in volts per microsecond. The slew rate dictates how high the input sensitivity of an amplifier (level) may be set to until it cannot reproduce transients faithfully. A very generic way to delineate suitable slew rates is:

Rate > 10 V/µs Rate > 30 V/µs

For amplifiers rated under 100 watts per channel For amplifiers rated over 200 watts per channel

Current high-quality power amps have slew rates that fall well above the described minimum range without presenting objectionable distortion or suffering from interference.

Loudspeakers

Loudspeakers are commonly the last and possibly the most important elements in the signal chain. Still, it is not uncommon for their selection to be heavily influenced by subjective criteria, where choice is based primarily on ‘character’ and not on fidelity (in a similar fashion to microphones). The topic of studio monitoring is a heavily debated one and it is approached with extreme caution by some, while others view it as a simple matter of taste. Despite the argument, it is nonetheless clear that different speaker models may be used effectively for music production as long as the technicians that employ them understand their behaviour.

A case may be made supporting the use of ‘flattering’ or ‘colouring’ monitors, considering that transparency may be virtually unobtainable for recordists working with restricted budgets and that ultimately most consumers will listen to music through uneven systems. With that in mind, it is not unusual for engineers to choose to work with loudspeakers that reproduce signals with a familiar bias or ‘colouration’, i.e. devices that superimpose a desirable uneven frequency response onto that of the program material. On the other hand, although it is possible for recordists to achieve good results with basic budget monitors, in the long run it seems advisable for them to seek for the ‘flattest’, most ‘truthful’ high-quality speakers, as such devices should lead to consistent results that translate well to a greater range of systems, e.g. from portable mp3 players to large PAs.

Headphones

Headphones play an important dual role in music production, being used not only by performers but also by recordists on occasion. As far as engineering with headphones is concerned, the following are advantages of working with such devices:

•    Headphones tend to provide recordists with a greater level of detail than ‘middle of the road’ / low-quality studio monitors.

•    Headphones allow for work to be carried out under ‘less than ideal’ control room circumstances or even within the performance space, i.e. the ‘live’ room.

•    It is very likely that current productions will be auditioned (by the general public) through headphones more extensively than through any other speaker-based monitoring system.

With that in mind, it is still important to also note the following disadvantages of working with headphones:

•    The stereo image of headphones is considerably different to that of loudspeaker systems (unless crosstalk and delay are introduced).

•    Engineers tend to work more ‘conservatively’ when mixing with headphones (lower relative levels, e.g. vocals, and narrower stereo image).

Ultimately, as artists and other production personnel will commonly expect to audition ‘takes’ or monitor mixes through conventional monitoring systems, headphones should be used temporarily and always alongside loudspeakers.

Monitoring Management Devices

Monitoring management devices aim to provide large format console, control room monitoring capabilities to individuals working in smaller, ‘budget’ DAW-based studios. This type of equipment usually offers functions such as output level adjustment, input selection, output (monitor) selection, talkback, headphone outputs, etc.