Chapter 3

The Mastering Studio

A Critical Listening Environment

A professional mastering studio (Figure 3.1) represents a flat and uncolored critical listening environment designed to reproduce full-range audio. It must be purpose-designed and functionally dedicated to the playback, assessment, and processing of audio. The absence of frequency or imaging anomalies allows the Mastering Engineer to confidently make accurate audio adjustments. This also ensures that masters created in the mastering studio translate accurately (sound the same or similar on other systems as they did in the studio) to downstream playback systems ranging from ear buds or headphones to car stereos and high-end audiophile systems. This chapter examines essential components of the mastering studio.

The Room

Common Design and Treatment Approaches

The ground-up design and implementation of a professional mastering studio, and/or the retrofitting of existing spaces, remains beyond the scope of this book. However, there are functional design concepts of the studio that a good Mastering Engineer must understand. The room that houses the mastering studio is ideally sound isolated so that environmental noise does not enter and distract critical listening, nor does playback audio escape and affect accurate frequency response or imaging. Room dimension, volume, and geometry represent vital considerations. This results from the physical length (and energy properties) of sound waves, especially at low frequencies—a 50Hz wave is 22.51 feet long, and a 20Hz wave is 56.26 feet long. This means that ideally, your listening room would be at least 23 feet long with a volume of at least 2,500 cubic feet to accommodate a 50Hz sound wave.1 For this reason, and to alleviate other common acoustic anomalies, acoustic treatments in the form of absorbers, bass traps, diffusers, and resonators are added for the accurate presentation of sound. Following is a brief definition of each.2

  1. Absorbers—Primarily absorb high frequencies. Strategically placed to absorb reflections from studio surfaces—walls, ceiling and floor—to the listening position. Quotidian examples are acoustic tile, carpeting, curtains, mineral wool, and fiberglass. Clouds are broadband absorbers that hang from the studio ceiling to dissipate and absorb diffracted sound waves.
  2. Bass Traps—A membrane absorber designed to minimize buildup of low frequencies. Usually placed at the back and in corners of the studio. Low-frequency issues are common due to the long length of sound waves below 50Hz.
    Figure 3.1

    Figure 3.1Mastering Studio 2 at Capitol Mastering. PBDAW (zone 1) is left, RDAW (zone 3) is center. The Sterling Modular desk holds all analog equipment (zone 2), which is in between the AD and DA converters.

    Source: (courtesy Capitol Mastering)

  3. Diffusers—Characterized by an irregular surface geometry. The relationship between these irregularities and the sound waves striking them determine the frequencies that are diffused. Used to create a large diffuse sound field for consistent imaging at various listening positions. Also constructed as an Absorptive Diffuser to combine attributes of both treatments.
  4. Helmholtz Resonators—Size-built air spaces with holes—analogous to a gallon jug resonating—to absorb specific resonant frequencies and standing waves via phase cancellation.

Professional acoustical treatment companies will measure, analyze, and photograph your room in order to generate a plan for recommended treatments to create a flat and focused listening environment (Figure 3.2). GIK, RPG, Primacoustic, and Vicoustic are companies that offer effective acoustical treatment options.

Acoustic Properties and Sound

Room dimensions—along with varying ceiling, wall, and floor surfaces—affect how sound presents in the mastering studio. These surfaces will have one or a combination of three basic acoustic properties: absorption (the absorption of sound energy which dissipates as heat energy), diffusion (the scattering or distribution of sound into other directions), and reflection (the sound energy bouncing off of the various studio surfaces). Larger rooms with high ceilings and reflective surfaces create a longer reverberation decay time, and smaller rooms will have a shorter reverberation time. A good acoustician can advise how ‘live’ the room should be, as it remains a matter of room accuracy and preference. Concrete, brick and glass are reflective, and can cause reflections and add reverberation and vitality to music. Conversely, carpets, curtains or fabric are absorptive and tend to smooth out mid or high frequencies. Wood generally offers a smooth and warm reflective quality, but wood types and finishes certainly affect presentation of sound. Correlating the range of surfaces and dimensions with the quality of sound presentation will inform both design approaches for the mastering studio and your assessment skills of recorded audio.

Figure 3.2

Figure 3.2Acoustical treatments in Mastering Studio 2 at Capitol Mastering. Dual pane windows behind the main speakers are angled up to reflect sound into absorptive material and frequency-tuned cloud absorbers in the ceiling. Fabric-covered fiberglass on walls manages reflections, and absorbers at incident angles to the speaker drivers alleviate image smearing and comb filtering at the listening position.

Source: (courtesy Capitol Mastering)

Common Acoustical Issues—Standing Waves, Flutter Echo, and Comb Filtering

The mastering studio should be symmetrical left to right so as to present stereo speaker cohesion and accurate imaging. The geometry of the room should be designed to provide reflective and absorptive surfaces at the proper angles to create diffuse and predictable sound fields without problematic anomalies such as standing waves, flutter echo, or comb filtering.3 Reflective parallel walls must be avoided at the design stage or treated, as they create standing waves whereby certain frequencies are multiplied and resonate as a function of the sound wavelength, and the distances from the sound source to the walls and the listening position. Flutter echo is another issue caused by parallel walls, as sound will reflect between them, causing an undesirable repetitive echo sound. Comb filtering is caused by the delayed frequencies reaching the ear at slightly different times so that some frequencies are phase-cancelled and others are amplified, resulting in a representation that looks like a comb rather than a smooth curve (Figure 3.3).4 Image smearing describes an acoustic phenomenon whereby aspects of the audio image sound undefined and smeared due to the issues previously described occurring at the listening position. The Mastering Engineer must make accurate decisions and adjustments, so all acoustic anomalies must be rectified in order to achieve accurate presentation of the music. Managing these issues involves adding acoustic treatments to the mastering studio that allow for the true and uncolored presentation of full-range audio.

Figure 3.3

Figure 3.3Graph showing comb filtering due to phase cancellations at dimension-specific frequencies in a room. An accurate assessment of audio quality cannot be made in a room with comb filtering issues.

Acoustical Testing and Treatments

As discussed previously, some aspects of a room are undesirable. In order to accurately rectify these, acoustic measurements of frequency response are taken by playing pink noise5 through the speakers and placing a microphone connected to a spectrum analyzer at the listening position. This, in conjunction with user listening tests to highly regarded recordings, will reveal acoustical concerns that the acoustician can then plan to resolve. Once observations are corroborated and diagnosis of the issues is made, adding acoustic treatments to the room helps resolve sound presentation issues. As described previously under Acoustic Properties and Sound, these treatments possess one or a combination of the absorptive, reflective, or diffusive acoustical properties.

Speaker Monitor Calibration

It is essential to calibrate the speaker monitors and a midpoint volume setting on your monitor controller to your chosen sound pressure level (SPL) level. The accepted standard for most audio engineers is 83dB SPL. I usually listen about 70% of the time at 70–75dB SPL, and the 20% at 80–85dB SPL and 10% at 90dB. This calibration process requires an SPL meter and a pink noise generator (for example, the signal generator plug-in found in ProTools). It is a straightforward and practical way to check for optimal stereo imaging and playback performance from your speakers.

  1. Open a pink noise generator on the PBDAW and set for the operating reference level of −14dBFS.
  2. Set the monitor volume for a midpoint or standard listening setting and mark with a pencil.
  3. With a SPL meter set to C-weighting6 at the listening position (on a stand if needed), first mute the right speaker and adjust the left speaker’s power amplifier until the SPL meter reads 83dB. Then repeat the same process with the other speaker. Do this for all sets of speakers if you use more than one pair.

This calibration will allow for your monitor controller to have a standardized volume setting for audio at a playback level of −14dBFS to equal 83dB SPL from the speakers. It will help avoid ‘volume creep’ on long days, and therefore protect your hearing. Additionally, switching from near-field monitors to mains will be seamless, SPL-wise. Finally, the speakers will present recorded audio with the L/R imaging exactly as the mix engineer intended.

Front/Back Treatment Approaches

There are numerous theories for adding room treatments, but two common approaches utilize front/back designs. In order to improve imaging and manage any sound-reflection based issues, either the front or back of the room is live or treated with diffusers, and the other end is dead and treated with absorbers. In a live front approach, the front area where the speakers are has diffusive (live) treatments, and the back and sides of the room, are treated with absorbers and bass traps to minimize high-frequency reflections and manage low-frequency diffraction and accumulation.7 Conversely, in order to create a reflection-free zone between the speakers and the listening position, the front can be made dead with absorbers, and the back and sides can be treated with diffusers to maintain an optimal time-delay gap or ‘liveness’ to the room.8

Power and Grounding

A professional mastering studio is preferably AC power and ground isolated, and ideally powered by an independent electrical panel feeding 20-amp circuit breakers and outlets. This alleviates any ground loops or hums in the system, and allows the equipment to draw the requisite electrical current for optimal functioning. Note that high-wattage power amplifiers or audio equipment may draw more current to reproduce dynamic intensity or louder passages in the music.

Speakers and Power Amplifiers

The primary consideration with main speakers for mastering is that they reproduce full-frequency audio that the Mastering Engineer can relate to accurately so that the finished project translates well to other playback systems. Most main speakers are either two-way (a woofer and tweeter with a crossover network in the cabinet to separate the two frequency bands), or three-way (three separate speaker drivers with two crossovers that separate the three frequency bands). The extra driver(s) should add definition and detail in the corresponding frequency range, but there can be issues at the crossover frequencies to listen for. With some audiophile speakers, an additional tweeter is added for a four-way speaker. A subwoofer (or even two) is often implemented so the lower octave and any sub-harmonic frequencies are adequately represented. Additionally, as I discuss later in this section, speakers are either passive (with separate power amplifiers) or active (amplifiers included in the speaker cabinet). Selecting and testing main speakers is a critical aspect of effective mastering studio setup.

The main stereo speakers are placed as two points of an 8–10-foot sided equilateral triangle with the apex being the listening position. Standard speaker mounting approaches are: soffit-mounted (flush wall-mounted to minimize diffraction of sound and especially to direct low-frequency energy into the room), freestanding, or on stands (Sound Anchors™ can be decoupled with spikes and weighed down with sand or lead pellets to minimize speaker cabinet movement and maximize sound energy projected into the room). Acousticians often seek to decouple the main speakers from any structure in the room or building for that matter. Hence, extreme decoupling maneuvers such as stone speaker stands (completely independent of the building structures, which go directly into the earth), or industrial spring-loaded speaker boxes. In this way, the sound energy emanating from the speaker is not dispersed through the floor or walls, and comes directly through the speaker drivers to the listening position. These common options for installing main speakers into the mastering studio are discussed in greater detail following.


A soffit is a wall-like structure that is custom built to accommodate your full-range speakers at the front wall. There are several advantages to soffit-mounted speakers, and you universally see them in professional recording/mix control rooms. For one, they are solid, minimizing energy transfer and movement of the speaker cabinet when sound is played back. Second, they allow the drivers to be flush with the soffit wall (studio front), meaning there are no reflections from behind the speaker cabinet, and no wall behind the speakers to cause smearing and unwanted sound reflections and frequency nodes or cancellations. In a mastering context, there appears to be a trend toward freestanding speakers. This is because if soffit-mounted speakers are problematic sounding, the soffit structure must be rebuilt to accommodate adjustments.


Freestanding speakers either rest on the floor, or on heavy stands. Stands are often filled with sand and put on spikes to decouple them from the studio and diminish energy loss through vibration. The advantage of freestanding speakers is that fine-tuning speaker placement or replacing components is relatively easy. Acoustical treatments notwithstanding, when placing freestanding speakers, the distance from the drivers to the back wall, sidewalls, ceiling, floor and front wall are measured and checked so they are not multiples of each other—especially if there are reflective parallel or near-parallel walls. This avoids the acoustical issues discussed earlier in this chapter (standing waves, flutter echo, and comb filtering) at the corresponding frequency wavelengths.

Near-Field Monitors

Near-field monitors are bookshelf speakers with 6” or 8” woofers that may have frequency response limitations due to both cabinet and speaker size. They are helpful as a second reference for audio. Near-fields tend to roll-off in the low frequencies, which can accentuate or reveal the mid- or high frequencies. They can indicate if robust low frequencies apparent on the main speakers need adjustment for impact on smaller speaker playback. Near-fields can also be used to mimic real-world playback scenarios that are not ideal, like an inexpensive home system or a car stereo. The Yamaha NS-10 was a near-field recording control room staple for many decades, and many engineers grew accustomed to its sound—slightly edgy and rolled-off in the low frequencies. You may occasionally see them in a mastering application as an additional reference speaker.

Off-Axis Listening (OAL)

This is listening without placing your head at the apex of an equilateral triangle with its two remaining sides indicated by a set of speakers. OAL utilizes an ancillary set of near-field monitors such as Yamaha NS-10s or single-driver ‘speaker cubes’ such as Auratones that are on the floor or under the console. This is a more drastic method of mimicking real-world playback scenarios. The mastering studio represents a purpose-designed laboratory of sound—often from the ground up. However, real-world listening is not like this, and often drastically different. End-users can listen back to your labor of audio love on tiny earbuds, as a lossy .mp3 file, in a car, or on randomly placed bookshelf speakers in a kitchen, dorm room, living room, etc. To that end, it remains beneficial to have a set of off-axis speakers available so you can check how the master translates in a less-than-ideal playback scenario. You will immediately know if your mid-range is too light, or if your low frequencies are building up. Small adjustments like this revealed by off-axis speakers can make for a master that translates very effectively to real-world end-user playback settings.

The quality of your speakers is mission critical in mastering. They must reproduce the full range of audible frequencies, and even beyond. Depending on the manufacturer, mastering speakers are engineered to reproduce sound from 20Hz or below at the low end, and up to 20kHz or above at the high end of the audio spectrum. You must accurately relate to what you are hearing, and the audio you adjust must discernibly translate to all other environments. PMC, ATC, Tannoy, Bowers & Wilkins, or Lipinski are respected manufacturers of full-range mastering speakers.

Power Amplifiers

Standalone power amplifiers are used to power passive speakers. Today, many high-end speakers are active, meaning the power amplifier is inside the speaker cabinet, and matched to the drivers in regard to power and impedance specifications. This helps create a consistent sound for the speaker if you happen to move them between studios. Passive speakers require a power amplifier, and in mastering applications, it is common to have one power amplifier for each speaker, or monoblock power amplifiers. This is to maximize the efficiency of each speaker/amplifier relationship, and allow them to be independently responsive, power-wise, to transients and program information. Additionally, transient quickness is reproduced with clarity, absent the sag or flattening characteristic of insufficient amplifier power, especially if both speakers share one stereo power amplifier.

This is where audiophile amplifiers incorporating tube and transformer circuit design with point-to-point wiring have appeal. The sound reproduction is rich, with quick transient response and a vitality that contributes to a compelling listening experience. Manley Labs manufactures respected examples of tube monoblock power amplifiers, whereas Bryston, Classé, and Hafler are known for solid-state offerings.

Monitor Path/Monitor Control

For mastering and critical listening, always seek the simplest but highest quality path of components and wire between the DA converter and speakers. As discussed in Chapter 1 under Playback System Considerations, each component in the playback chain is like a pane of glass9 and each pane must be transparent so that the audio you hear remains unadulterated. A high percentage of the audio you will listen to, analyze and master will originate from a digital source file. The playback system example in Figure 1.1 illustrates five ‘panes’ (eight if you count the wire) between source file and speaker. Strive to avoid long cable runs or extraneous elements that can compromise sound such as patch bays, daisy-chained cables, or additional gain stages in the monitor path.

For mastering applications, a monitor controller has the following features: a stepped attenuator (so that there is a separate precision resistor set for each attenuator position ensuring accurate volume attenuation and L/R imaging), L/R mutes, mono button, L/R 180-degree phase inversion, and even an active/passive option. Examples are: Custom Manley (Figure 3.4), Crane Song Avocet (Figure 3.5), and Dangerous Monitor (Figure 3.6).

Figure 3.4

Figure 3.4Manley custom monitor controller for mastering. Features include selectable inputs, L/R mute, L/R reverse, mono, VU meters, stepped attenuator with precision resistors, meter and output trims, switch for two sets of monitors, passive or active operation.

Source: (author collection)

Figure 3.5

Figure 3.5Crane Song Avocet Discrete Class A Studio Controller. Features include selectable analog/digital inputs, L/R mute, mono, dBFS Meter, stepped attenuator with precision resistors, stereo or surround speaker selectors, talkback, dim switch.

Source: (courtesy Crane Song)

Figure 3.6

Figure 3.6Dangerous Monitor Controller Remote. Features include four analog inputs, mono button, stepped attenuator with precision resistors, three speaker selections and a subwoofer selector, 5:1 surround speaker mutes, talkback, dim switch.

Source: (courtesy Capitol Mastering)

Signal (Channel) Path

The signal path represents the mastering chain with which the audio is processed, enhanced, or otherwise adjusted. In mastering, analog equipment may be daisy-chained together as needed. However, diligence is necessary to avoid inadvertent loading or frequency roll-offs between analog equipment devices with incompatible circuit designs. This can be tested by sweeping test tones between 20Hz and 20kHz through the mastering chain and listening to/VU metering the end of the audio chain flat (all equipment bypassed), then engaging the equipment (with settings flat) and noting any roll-off in the high or low frequencies at the VU meter. After successfully sweeping 20Hz–20kHz with no issues, play music you are familiar with through the chain and verify that the flat audio matches the post-chain audio by A/B-ing the corresponding positions on the monitor controller.

Another way to set up the mastering signal path equipment is to have it on the bypass switches of a mastering console such as the Dangerous Master (and Liaison), Maselec MTC-1X, or a Crookwood Mastering Desk. This is advantageous, as it allows for the quick insertion or removal of an EQ or compressor from your chain to evaluate its effectiveness. Additionally, the buffer amplifiers for each switch alleviate possible loading/impedance issues described in the previous paragraph. The Liaison features three flip buttons to swap the sequence of equipment in the chain, and also a parallel blend fader for parallel processing, which is extremely handy. The selection and sequence of equipment in your chain remains of critical importance, so when a mastering chain sounds special … remember to document it!

Mastering Console vs. Point-to-Point Considerations

Some theoretical and practical considerations arise when deciding on a signal path approach. A console adds insert buttons and buffer amplifiers, meaning more electronics and potentially increased noise. Conversely, any loading/impedance issues between analog devices is eliminated. Also, ease of use, quick equipment auditioning, and the ability to swiftly recall mastering chains represent huge upside considerations. For many years, I bypassed a noisy console in favor of connected equipment point-to-point, and the results were great. In recent years, I have implemented the Dangerous Master/Liaison (Figure 3.7), and appreciate the user interface and the ease in accessing the array of equipment connected to it. With this setup, I only require two DA converters in my signal routing/monitoring setup, because the console features a multed input tap for monitoring of the flat signal. Prior to this, the point-to-point method required three DA converters for a standard mastering setup—one for the flat mix, one for the PBDAW output (mastering chain input), and one for the RDAW output.

Cabling/Interconnects/AC Cords

High-quality cabling and interconnects are advised for mastering applications. Use short runs of high-quality braided copper cables and interconnects: Canare, Mogami, and Van Den Hul are superior examples. These will connect all analog equipment in the channel path via console inserts or point-to-point (zone 2 of the mastering system), from each DA converter to the monitor controller, and from the monitor controller output to the power amplifiers and speakers. For International Electrotechnical Commission (IEC) AC connectors, I recommend the ESP MusicCord Pro. Research and audition cables, interconnects, and power cords (IEC connectors), but use common sense and look for cables with a durable build and gold connectors.

Figure 3.7

Figure 3.7Dangerous Master, Liaison, and Convert-2. These three Dangerous Audio devices function together as an effective mastering console. Master features include L/R input balance, input switch for two independent sources, S&M (Side & Mid) functionality for insert #2, in-out monitor switch for flat/mastered A/B-ing, input offset to gain match the flat source to mastered, S&M width, and output gain to drive the next stage of the system. Liaison features include device insert expansion which can be connected to one of the insert switches on the Master (1 or 3 is best, as 2 has S&M functionality), flip options to change sequence of equipment, parallel blend for parallel processing, and easy front panel access on insert 6 for additional equipment. Convert-2 features a high-quality DA converter, clock selector, onboard output calibration, four input selectors, dBFS Meter, external word clock input, and output trim.

Source: (courtesy Capitol Mastering)

Cables and connectors for a mastering studio are designed for routing either balanced or unbalanced analog or digital signals. The most common analog options are balanced XLR, balanced ¼”, unbalanced ¼”, and unbalanced RCA. In the digital domain: AES/EBU, S/PDIF (coaxial RCA), and BNC are the most common configurations.

Digital Clocking and Dither

Choosing between internal or external clocking options for each DAW or digital hardware piece in the mastering system remains a matter of preference and debate. Digital audio through each device is clocked at the chosen sampling frequency via its clock source (usually internal, external or digital input). Internal clocks are found in: I/O cards, AD and DA converters, and any device that performs digital signal processing (DSP). Stable digital clocking is critical for the accurate digitization of analog sound at an AD converter, or for its reproduction at a DA converter.

Two issues can result from an unstable clock—jitter and quantization error. Jitter results from any variation in the sampled signal and causes noise and distortion in the signal. Quantization error is distortion characterized by the difference between the actual signal value at time of sample, and the quantization interval value it is rounded to. As signal level decreases, this distortion increases and is rectified by adding a low-level noise known as dither.10 The quality of DA conversion, AD conversion, and the selection of dither for final rendering to 16bit from higher bit rates is within the purview of the Mastering Engineer, so take time to research and familiarize yourself with these devices and processes.

There is a subjective philosophy/trend that a dedicated external clock can enhance performance and sound quality of DAWs, AD converters, DA converters, and other digital devices. Hence, the ubiquitous word clock in and word clock out BNC11 connectors on most digital audio equipment. These same devices can function as a master clock, or an external master clock can be connected to clock all the digital devices in the mastering system—known as a ‘star’ clocking setup—as each clock destination represents a point of the star. Listening tests, A/B comparisons, and the complexity of the digital hardware setup all inform clocking preferences. It remains your choice whether to use a master clock for the digital equipment in your mastering system, or to allow the internal clocks of your DAWs and/or AD converter to clock the system.

The zeitgeist changes; for years, Mastering Engineers favored external clocking, but I’ve noticed in recent years that internal clocking has regained favor. It is increasingly accepted that the internal clock in an AD or DA converter performs best, and external clocking may provide a different sonic quality but actually reduces clock accuracy and increases jitter.12 I usually clock the PBDAW internally, but my RDAW is clocked by the AES/EBU output from the AD converter. I have also achieved excellent results using an external master clock for the entire system. This can involve high-quality sample rate conversion (SRC) of source audio to match the sampling frequency resolution set by the AD converter. The question remains if it improves the fidelity of the final product. Your own research will come into play in deciding how to best implement digital clocking in your mastering system. Prevalent master clocks are the: Antelope Isochrone Trinity (Figure 3.8) and Isochrone 10M, and the Drawmer M Clock Plus.

Figure 3.8

Figure 3.8The Antelope Isochrone Trinity Master Clock features an oven-controlled crystal oscillator and sampling frequencies up to 384kHz.

Source: (courtesy Capitol Studios)

Figure 3.9

Figure 3.9Block diagram of The Three Zone Mastering System. Coincides with Figure 3.1, and shows broad organization of the mastering system.

The Anatomy of a Professional Mastering Studio

The Three Zone Mastering System

The Three Zone Mastering System (Figure 3.9) refers to how the signal is routed through the studio for both processing and monitoring. It is also didactic and helps students conceptualize and understand signal flow in a mastering context. The zones are delineated by the PBDAW DA converter and the AD converter. In my experience, an ideal mastering system utilizes two DAWs to process audio for high fidelity optimization: one DAW for the playback of source audio (PBDAW), and one DAW to record the processed audio (RDAW). This affords the Mastering Engineer a great deal of flexibility in determining not only how, but where in the system to best optimize the audio. Additionally, it allows for audio playback at one sampling frequency and audio capture at a different sampling frequency without having to SRC the source audio; or conversely, it will support up-sampled source audio so that DSP can occur at a higher sampling rate and corresponding bandwidth.

My mastering system functions in three distinct zones for processing audio. Zone 1 is contained within the PBDAW (digital domain), where level adjustments or preliminary processing via plug-ins before the initial DA conversion occur. Zone 2 is between the DA and AD converters, and completely analog. At minimum, I use a high-quality analog compressor and an analog parametric equalizer before the audio is routed to the AD converter. Zone 3 is also in the digital domain and includes another instance of DSP, usually a brickwall limiter or digital processor, before the final capture of the mastered audio into the RDAW. Once in the RDAW, there may be some very subtle additional adjustments, but generally, the mastering is complete once the audio is captured as a digital file. It then needs to have the remaining zone 3 procedure of editing the tops and tails and being rendered into the correct formats for delivery and/or archive.


To plan, select, and assemble the components of a mastering studio remains both demanding and exhilarating. If the studio subsequently becomes the source of excellent sounding masters, the Mastering Engineer has reason to celebrate. It is additionally a fantastic exercise—one I assign my students—to become familiar with both the equipment offerings in the pro audio market and the numerous decisions that must be made in setting up a mastering studio. I recommend that the reader conceive, design, and outfit an imaginary mastering studio in order to gain valuable insight into the inner workings and functions of the anatomy of a professional mastering studio.


  1. Select the main equipment for an imaginary mastering studio. Utilize three different budgets: low, middle, and high. Include: speakers, a console, plug-ins, analog equipment, converters, and DAWs. Research and compile a list, and then explain your selections.
  2. You are designing a mastering studio in an existing 18’×18’ room. How exactly will you place the speakers and determine the listening position? The floors are linoleum, and the walls are sheet rock. Which acoustical treatments will you add to the room, and why? How will you test the room and speakers for optimal audio fidelity?
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