Introduction
One of the most powerful tools in any recording studio is the Equalizer (EQ). It allows us to surgically cut or boost specific frequency ranges, such as Sub, Bass, Mids, and Highs. However, the leading professional rule in commercial studios is absolute: always strive to achieve your desired sound before you ever touch a piece of outboard gear, an equalizer, or a software plugin.
Before reaching for the EQ on your mixing console or within your DAW, remember that you already possess a highly potent “Acoustic EQ” right inside the live room: your microphone selection and Microphone Placement.
1. The Acoustic EQ: Tone Sculpting in the Live Room
Every decision you make during the tracking stage alters the frequency response of your signal. By mastering the physical properties of acoustics, you can solve frequency problems at the source, ensuring a cleaner mix later on.
The Proximity Effect
Placing a directional microphone—such as one with a Cardioid polar pattern—closer to a sound source will significantly boost unwanted low-end and bass frequencies. Conversely, backing the microphone away by just a few centimeters performs a natural, completely clean “Low Cut”.
The massive benefit of this acoustic adjustment is that it shapes the low-end without introducing any of the Phase Shifts that often occur when using electronic or digital equalizers.
Off-Axis Placement
If an instrument generates harsh, piercing high frequencies—such as cymbals, acoustic guitars, or a loud electric guitar amplifier—you don’t need to instantly reach for a digital low-pass filter.
Angling the microphone slightly away from the center of the sound source (Off-Axis) softens the high-end response organically. This physical adjustment captures a smoother, more balanced tone right at the capsule, keeping your signal chain pristine.
The Philosophy of the World-Class Mix
When you study microphone theory and recording techniques deeply, you discover a fascinating reality: the highest-tier, world-class audio engineers are capable of crafting a complete, balanced, and vibrant mix solely through the strategic selection of capsule types—whether Dynamic, Condenser, or Ribbon—and by precisely shifting their placement within the studio space. They can achieve a professional tonal balance without a single touch of an EQ knob.
Therefore, in a truly professional studio environment, the equalizer remains a court of last resort for tone sculpting. Yet, despite being the final step in the tracking chain, it stands as one of the most vital Processors in audio history.
2. Etymology and Function: What Does an Equalizer Actually “Equalize”?
To master this tool, we must look at its literal definition. Etymologically, the word “Equalizer” stems from the root word Equal . Consequently, an equalizer is quite literally a “balancer” or a “leveler” [סעיף 2].
But what exactly is this processor balancing or equalizing?
An equalizer balances the Amplitude (electrical or digital volume) of different frequencies—measured in Hertz (Hz)—across the audio spectrum. Its core engineering purpose is to adjust these individual frequency levels to restore, correct, or enhance the natural balance of the original acoustic signal that may have been lost during transcription.
3. The Evolutionary Physics of Frequency: Human Speech vs. Nature’s Noise Floor
Our reliance on specific frequencies to transmit information is primarily an evolutionary adaptation designed to separate human communication from the noise floor of the natural world.
If you analyze the background noise in a studio, on a city street, or deep in nature, you will discover that the dominant acoustic energy of the environment—such as wind, ocean waves, running water, earthquakes, hardware hums, and air conditioners—is almost exclusively concentrated in the Low Frequencies (Bass spectrum).
To ensure humans could pass vital information to one another without their voices being swallowed by environmental noise, human speech frequencies—especially consonants, which carry the actual data of language—developed outside this low-end spectrum. Instead, human communication resides firmly in the Mid to High Frequencies.
The Psychoacoustics of Whispering and Childhood Development
This specialized auditory capability is crucial even at extremely low volumes. For instance, whispering while hiding, or speaking softly in an environment where others are sleeping, requires an auditory system capable of extracting the high frequencies of that whisper directly out of the silence.
This also explains a fascinating developmental milestone: young children often struggle to whisper or speak very softly. This difficulty occurs because their sophisticated hearing mechanisms have not yet fully matured. They have not yet developed the advanced neural decoding required to extract microscopic details and information from surrounding ambient noise, whether at low or high volumes.
4. Spatial Hearing and Panning: How the Brain Decodes Direction
How does the human brain determine exactly where a sound originates in a three-dimensional space, even when our eyes are tightly closed?
Our psychoacoustic system achieves this through a continuous, real-time comparison of the acoustic signals arriving at each ear. The brain relies on two primary physical mechanisms to determine stereo panorama (Panning) and directional positioning:
- ITD (Interaural Time Difference: The precise time delay between when a sound wave reaches one ear versus the other.
- ILD (Interaural Level Difference: The difference in volume between both ears. This difference is created because the human head acts as a physical “acoustic barrier,” attenuating and shadowing the high-frequency signals arriving at the farther ear.
If a sound wave hits both ears at the exact same fraction of a millisecond and with the identical amplitude (zero difference), the brain decodes the source as being on the median plane bisecting the body—meaning, dead Center.
5. Why Sub-Bass Has No Direction: The Wavelength Law
This brings us to a rigid physical and acoustic law that exerts a dramatic influence on how we build our mixes and position our speakers: At low frequencies (Bass/Sub-Bass), the human ear and brain are fundamentally incapable of detecting the physical direction of the sound source.
In acoustics, low frequencies behave as an Omnidirectional / Non-directional Sound field. The root cause of this phenomenon is purely mathematical and physical, dictating the strict relationship between a sound’s Wavelength and the physical distance between our two ears—which averages roughly 17 to 20 centimeters.
High Frequencies (kHz) vs. Low Frequencies (Hz)
At High Frequencies (Treble)
The acoustic wavelengths are incredibly short, measuring just a few centimeters or millimeters. Because the human head is significantly larger than these short wavelengths, it acts as a physical barrier, blocking the sound wave and casting an Acoustic Shadow.
The farther ear receives a substantially weaker signal, enabling the brain to instantly decode the lateral direction of the sound source using Interaural Level Differences (ILD).
At Low Frequencies (Bass)
The sound waves are massive and slow-moving. For example, a fundamental sub-bass frequency of 50 Hz has a massive wavelength of nearly 7 meters (approx. 22 feet). A sound wave of this magnitude is gargantuan compared to the size of a human skull.
Instead of being blocked by the head, the wave simply bends and flows around the skull without dropping in amplitude or being deflected. This physical wave-bending phenomenon is known as Diffraction.
As a direct result of diffraction, a low-frequency wave floods both ears at virtually the exact same phase and with the identical sound pressure level. Because there is no measurable discrepancy in arrival time (ITD) or volume (ILD) between the left and right ears, the brain is starved of directional data. It has absolutely no electronic or physiological tools to decipher the precise physical location from which the low frequency originated.
6. Practical Implications: Subwoofer Placement and Mono Bass Structure
This wavelength law results in two critical practical implications that every engineer must follow in the studio and when designing live sound reinforcement systems:
1. Subwoofer Placement
Because audio frequencies below 80 Hz carry no directional data for the brain, you can position your subwoofer cabinet almost anywhere within the control room (while accounting for room modes and standing waves). Despite its physical location on the floor or in a corner, the listener’s brain will still perceive the sub-bass as originating perfectly from the center or directly from the main studio monitors.
2. The Golden Rule of Mono Bass
In a professional mix, sub and bass frequencies—such as the kick drum channel and the main bassline—must always be kept dead center in the stereo panorama (Mono C). Attempting to pan extremely low frequencies to the left or right will not create a wider sense of space.
Instead, it will introduce severe Phase Shifts , cause energy instability across your speaker cones, and induce a physical sensation of acoustic discomfort and imbalance for the listener.
7. The Historical Necessity and Scientific Visualization of EQ
Let us return to our historical and engineering starting point: the equalizer was born out of pure necessity to restore and compensate for acoustic energy lost during transmission.
When early telecommunication companies transmitted voice signals over thousands of miles of long copper cables, high frequencies naturally degraded over the distance. The original purpose of the EQ hardware was to “equalize” the signal—boosting the amplitude of those attenuated frequencies to bring the audio as close as possible to its original state so human speech remained intelligible.
Decoding the Graph: Frequency Response vs. Time Domain
To analyze this scientifically, we must understand the charts used in audio engineering.
A standard Sine Wave graph displays acoustic air pressure fluctuations over a time domain (where the Y-axis represents Amplitude and the X-axis represents Time). However, to analyze how an equalizer or a room alters sound character, we rely on a completely different visual tool: the Frequency Response Graph
The Ideal Linear Benchmark: The “Flat” Response
In an ideal scientific scenario where an electronic or digital system passes an audio signal without altering its tonal character, the resulting frequency graph will be completely flat (Flat Response) . This linear line indicates that every single frequency across the audio spectrum—from the deepest sub-bass to the highest treble—is transmitted with the exact same amplitude and volume.
However, in historical telephone copper lines—or when sound travels massive distances through the open air—the graph presents a relatively straight line in the low frequencies, followed by a sharp, aggressive drop as the frequency rises. The high-frequency treble energy is physically absorbed and lost along the path.
This loss is precisely where the original equalizer comes into play: its mechanical purpose was to isolate only that damaged high-frequency group, apply a Boost at the specific cutoff point, and “equalize” the curve back to a perfectly linear state.
8. Field Acoustics: Open-Air Frequency Distortion
The technical necessity for frequency compensation extends far beyond old telephone systems; it applies directly to real-world environmental physics.
Imagine a person shouting at you from a distance of 300 meters (approx. 1,000 feet) in the open air. The voice arriving at your ears will not sound bright, detailed, or “crispy”. You will primarily capture their Midrange energy, with almost no high frequencies remaining.
This acoustic degradation happens due to two strict physical phenomena:
1. The Inverse Square Law
As a sound wave radiates spherically outward from its source, its acoustic energy is forced to spread across an ever-increasing surface area. Consequently, its overall sound pressure level drops predictably over distance.
2. Atmospheric Absorption
Air molecules, humidity levels, and ambient temperature exert physical resistance against a moving sound wave. Because high frequencies possess exceptionally short wavelengths and lower kinetic energy per cycle, they tend to collide with air molecules and become absorbed and dissipated far more easily than massive, slow-moving bass waves.
As modern audio engineers, we actively use the equalizer to compensate for these natural acoustic losses—particularly during on-location field recordings, broadcast operations, or when capturing sound design on film sets.
9. From Consumer Hardware to Studio Precision: Fixed vs. Parametric Architecture
To harness the true potential of this powerful processor, we must master its electronic and algorithmic architecture.
The most elementary version, familiar to everyone from home stereo systems or car radios, is the Fixed EQ . In these consumer devices, users are restricted to controlling only two fixed parameters: Bass (low frequencies) and Treble (high frequencies). The manufacturer pre-determines the exact frequency cutoff points—for example, treating everything below 100 Hz as bass and everything above 10 kHz as treble. The user is left with only one operational choice: to raise or lower the volume of that rigid, pre-defined spectrum.
While a Fixed EQ is fine for casual listening, it completely lacks the surgical precision and dexterity required for professional studio tracking and mixing. To achieve commercial-grade audio, we must advance to vastly more sophisticated hardware and digital processors: Semi-Parametric and Full Parametric EQs , which grant us absolute, localized control over every microscopic segment of the frequency spectrum.
10. The Three Core Pillars of Parametric EQ
While consumer systems force engineers to operate blindly, professional parametric equalization demands a deep understanding of three interconnected electrical parameters that govern the shape of the audio filter:
- Frequency : The exact point selected within the 20 Hz to 20 kHz spectrum to apply the sonic adjustment.
- Gain : The amount of active amplification or attenuation applied to the target frequency, measured strictly in decibels (dB).
- Q Factor (Quality Factor): The bandwidth of the filter curve. This determines whether the equalizer cuts or boosts a singular, razor-thin frequency or influences a broad, surrounding neighborhood of musical harmonics.
The Golden Studio Axiom: Narrow Cuts, Wide Boosts
When manipulation of the Q Factor is required, professional engineers adhere to a foundational studio axiom:
- Surgical Cutting : Attenuating frequencies should almost always be executed with a narrow bandwidth (High Q). This allows you to surgically pin-point and eliminate problematic room resonances, microphone honk, or muddy frequencies without damaging the surrounding, healthy musical information.
- Musical Boosting : Amplifying frequencies should almost always be executed with a wide, gentle bandwidth (Low Q). Broad curves simulate the natural acoustic behavior of real instruments, introducing tone and character smoothly while maintaining absolute musicality throughout the mix.
11. The Psychology of the “Smiley Face” EQ: Decoding the Loudness Button
On many consumer hi-fi amplifiers and car stereos, you will find a dedicated button labeled Loudness. Average listeners tend to view this control as a simple sound “inflator” that carves out a classic “smiley face” EQ curve—aggressively boosting the bass and the treble.
But what is the actual scientific and psychoacoustic reality behind the existence of this button?
The answer is rooted in one of the most fundamental discoveries in audio engineering history: the Fletcher-Munson Effect (also known as the Equal-Loudness Contours).
Unlike a professional studio microphone, which captures and measures acoustic sound waves linearly and objectively at any sound pressure level, the human ear is a dynamic, non-linear, and constantly shifting biological mechanism. To ensure human survival, our ears must continuously adapt their sensitivity based on changing environmental circumstances.
- At High Volumes (High SPL): The human ear perceives sound relatively flat and linearly, naturally capturing more low-end bass and high-end treble information.
- At Low Volumes (Low SPL): The human ear loses massive sensitivity at the extremes of the spectrum. The bass and treble fade away entirely, leaving behind primarily the mid-range.
Psychoacoustics in Daily Life: Acoustic Reflex and Room Noise
The High-SPL Defensive State
When exposed to extreme sound pressure levels in a loud club or concert, the ear activates a physiological defense mechanism. The tiny muscles inside the middle ear contract to stiffen the eardrum, a phenomenon known as the Acoustic Reflex. When you step out of the venue, everything sounds muffled, quiet, and distant until the biological system recovers.
The Low-SPL High-Sensitivity State
Imagine falling asleep with a bedroom radio playing at a whisper-quiet volume. When you wake up in the dead of night, enveloped in absolute silence, that exact same volume suddenly sounds shockingly loud. The environmental Noise Floor has plummeted. In response, your brain and ears turn their biological gain up to maximum to catch any hidden threats.
12. The Science Behind the Loudness Button
When you turn down the volume knob on your sound system, the baseline sensitivity of your hearing shifts—but it does not shift equally across all frequencies.
The human ear does not stretch its energy to capture low frequencies (Bass) at quiet volumes. Evolutionarily, there is no such thing as a “quiet low frequency” in nature. There is no such thing as a “gentle earthquake” or a “soft thunderstorm.” Low-frequency events in the natural world are always high-energy, catastrophic events, meaning we will hear them regardless of biological tuning.
Conversely, the exact spectrum that the human ear must preserve at all costs during low-volume situations is the High-Mids zone (roughly between 2 kHz and 4 kHz). This precise frequency range holds the overwhelming majority of information, data, and hard consonants required to decode human speech and detect the rustle of a predator in the bushes.
Therefore, when listening at low levels, the sound naturally loses its rich bass and crisp treble. The Loudness button acts as a smart compensator: it automatically introduces a tailored boost to the low and high ends specifically at low volumes, artificially restoring the natural tonal balance that your biological hearing misses.
13. The Studio Reality: How Fletcher-Munson Destroys a Low-Volume Mix
Consequently, as the master volume drops, the human ear sheds its sensitivity to low frequencies (Bass) and high frequencies (Treble) at an aggressively accelerated rate, while stubbornly clinging to high sensitivity across the midrange.
This leads to a notorious studio phenomenon: a mix that was beautifully balanced and “cooked” at a high volume will suddenly sound incredibly “thin,” boxy, muffled, and lifeless the moment you listen to it at a lower playback volume.
This acoustic degradation is exactly why the Loudness button was engineered. Its core technical function is to actively compensate for the flaws of human auditory contours at lower sound pressure levels. When the hardware recognizes that the overall volume is low, the Loudness circuit engages, applying a targeted boost to both the bass and treble boundaries. This artificial compensation reconstructs a balanced sonic image for the listener, allowing them to experience a full, rich, and dimensional mix even during quiet late-night listening.
The Consumer “Smiley Face” Trend
Naturally, within the realm of consumer home audio, many casual listeners took this psychoacoustic mechanism to an extreme, transforming it into a cultural trend. This birthed the notorious urge to engage the Loudness button, crank the physical Bass and Treble knobs to their absolute maximum parameters, and introduce aggressive subsonic bass harmonizers—all to force a bloated, artificially pumped sound out of consumer speakers.
14. The Iron Studio Law of Mix and Mastering
Because the Fletcher-Munson effect governs our biology, it imposes one unyielding iron law upon modern mixing and mastering workflows: Never make final EQ decisions or element balancing adjustments at volume levels that are too quiet or too loud.
The Low-Volume Trap
If you attempt to mix your track at a whisper-quiet volume, your ears will be biologically blind to the bass frequencies. To compensate, you will instinctively apply massive, unnatural boosts to your low-end using your EQ plugins. The moment that track is played back at a standard operating volume, those bloated bass frequencies will completely flood the speakers, distort your headroom, and instantly destroy the song.
The 85dB SPL Industry Standard
To avoid this auditory deception, professional commercial studios operate under a strict monitoring standard: tracking, mixing, and balancing are conducted at an average listening level of 85dB SPL.
Scientifically, 85dB SPL is the optimal sweet spot where the frequency response curves of the human ear flatten out, becoming as linear and objective as possible. At this precise volume, you hear the true, unwarped relationship between your sub-bass, mid-range, and high frequencies.
The Proper Role of Low-Volume Monitoring
You should only drop your master monitor volume to a quiet level at the very end of your mixing process. This is done strictly as a final test to ensure that your lead vocals, main instruments, and essential mid-range elements remain distinct, cohesive, and fully intelligible within a compressed, low-energy environment.
15. The Audio Engineer’s Trap: Listening Fatigue and the Treble Loop
One of the most dangerous traps an audio engineer can fall into during a long mixing session is a lack of awareness regarding the ear’s shifting dynamics. When we sit in a control room working on a mix for 12 consecutive hours in front of studio monitors, our auditory system undergoes a gradual process of Listening Fatigue (עייפות שמיעה) and physical adaptation.
Just as the ear “closes down” and drops its overall sensitivity when stepping into a loud nightclub, the exact same biological defense mechanism triggers inside the control room over time. As the hours roll on, the exhausted engineer suddenly perceives their mix as dull or muffled, mistakenly diagnosing that the track is “lacking treble” (High-End). In response, they reach for an equalizer and aggressively boost the high frequencies.
The Destructive Feedback Loop
This high-frequency boost imposes an even heavier energetic load on the physical eardrum. In a direct physiological reaction, the ear tightens and closes down even further to protect its delicate inner mechanisms.
A few minutes later, the engineer once again senses a lack of brightness and clarity in the sound, driving them to boost the treble a second time. This destructive feedback loop continues layout after layout until the engineer finally steps out of the room for a short break.
During this break, the biological auditory system rests and recalibrates to a silent baseline. The moment the engineer walks back into the control room and presses play, they are horrified to discover that their mix sounds harsh, thin, piercing, and completely unlistenable.
16. The Fundamental Psychoacoustic Reality
As audio professionals, we must internalize a core psychoacoustic reality: What we hear and decode at any singular given moment in the studio is not identical to what we will perceive five minutes from now, nor is it identical to what we will hear at a different playback volume.
Under the jurisdiction of the Fletcher-Munson effect, the precise mathematical ratio between the extremes of the spectrum (Bass and Treble) and the core mid-range alters radically with every shift of the volume knob. If we attempt to carve and “cook” a mix at an excessive sound pressure level—or, conversely, at an overly weak whisper—we will inevitably hardwire a severe distortion into the track’s Tonal Balance
17. The Kitchen Analogy: Minimizing the Statistical Margin of Error
To minimize our statistical margin of error during a mix, we must implement an approach similar to a chef cooking in a commercial kitchen: we actively search for the most balanced “middle ground.”
If we engineer and mix a track at an extreme, thunderous volume, our biological hearing will naturally over-compensate for the extreme boundaries of the spectrum. Consequently, we will instinctively under-mix and decrease the actual bass and treble inside the DAW. When the end-consumer plays that song at a normal or low volume at home, our structural mistake will be magnified exponentially—the mix will sound ridiculous, thin, hollow, and completely starved of energy.
Stabilizing the Midrange for Translation
This is exactly why the rigid industry standard is to “cook” the vast majority of a mix at a controlled, moderate volume—specifically around 80 dB to 85 dB SPL.
At this precise auditory sweet spot, the ear’s sensitivity curves stabilize and become their flattest, allowing us to focus entirely on constructing a bulletproof, well-balanced Midrange.
By anchoring our engineering decisions in this optimal middle zone, we guarantee a safe statistical margin. Even if the consumer drastically turns the volume way up or down on their playback system, the deviation of error in our mix remains minimal. This ensures that the sonic balance preserves its integrity and Translates flawlessly across every listening environment on the planet.
18. The 2-Minute Sanity Check Routine
During long mixing sessions, you should adopt a strict, unyielding workflow habit: engineer your tracks for approximately 45 minutes at the recommended linear volume (82 dB to 85 dB SPL). Immediately follow this with a 2-minute “Sanity Check” at an extremely low volume level—equivalent to a quiet whisper.
At this whisper-quiet level, closely audit your mix to verify if your primary central elements—specifically the Lead Vocal and the Snare Drum—are still cutting through the midrange and remaining fully intelligible. If these core elements vanish into the background noise of the low volume, your mix suffers from a fundamental leveling and balance defect. You must correct this structural volume imbalance immediately, before your ears completely adjust and lock into a false perception of the mix.
19. Protecting the Midrange Core and Technical Exceptions
In every standard musical mix, you must guarantee that the most critical harmonic data of the composition is securely anchored in the center of the frequency spectrum—specifically across the Midrange and High-Mids. As established, this is the biological zone where the human voice, the fundamental frequencies of melodic instruments, and the defining speech consonants reside to provide lyrical clarity.
The only exceptions to this rigid engineering rule are tracks designed from the ground up for highly specialized, non-standard playback environments:
1. Cinema and Surround Mixing (Dolby 5.1 / 7.1)
In multi-channel theatrical formats, engineers utilize a completely separate, dedicated physical audio track engineered solely for handling high-energy, low-frequency acoustic effects like explosions, rumbles, and impacts. This specialized environment is known as the LFE channel (Low Frequency Effects). In a standard 5.1 layout, the “.1” mathematically represents this isolated sub-bass channel, ensuring that standard dialogue frequencies are never cluttered by catastrophic low-end energy.
2. Club and Festival Electronic Mixing (EDM / Club Mix)
In modern electronic dance music engineered explicitly for festival stages and commercial dance floors, producers and mixing engineers intentionally push the sub-bass and bass regions to dramatically higher amplitudes than normal. Club sound reinforcement systems are physically built from massive horn-loaded subwoofers capable of moving heavy masses of air, and the audience demands a physical, tactile impact felt directly in the body from these low-frequency waveforms.
20. The End-Consumer Chain: Why the Smartphone Controls Your Mix
Therefore, excluding highly specialized environments like cinemas or music clubs, your entire mix must be structurally concentrated and flawlessly balanced across the Midrange spectrum.
The reasoning for this law is rooted in the listening conditions at the very end of the consumer chain—specifically, the real-world devices that the general public uses to consume your music.
Think about your smartphone speaker, a small portable Bluetooth speaker, or cheap earbud headphones. What kind of physical speaker driver is installed inside these devices? It is a microscopic element that, from a standpoint of physics and mechanical engineering, is utterly incapable of displacing the mass of air required to reproduce low frequencies.
- The Hard Physics Rule: If you route critical musical data (such as the core bassline or the fundamental frequencies of a lead vocal) and anchor it strictly below 150 Hz, the smartphone speaker will simply not reproduce that sound. Period.
- The Destructive Outcome: The song will be stripped entirely of its Context, the mix will sound ridiculous, and the artwork will fail to translate the original intention of the artist.
This is exactly why we must pause and comprehend these concepts rather than rushing blindly to twist equalizer parameters. Before touching a single piece of hardware or an EQ plugin, you must internalize this engineering equation: A truly professional mix is characterized entirely by its ability to Translate (להתרגם) flawlessly between the massive monitors of a commercial studio and the narrow, restricted driver of a mobile smartphone.
21. The Saturation Trick: Making the Bass Audible on Mobile Devices
How do you make a deep bass instrument—like an electric bass guitar or a sub-heavy synth bass—fully audible on tiny smartphone speakers without bloating your actual low-end energy?
The answer lies in a highly effective engineering technique: Harmonic Overdrive and Saturation.
By introducing subtle, controlled saturation or saturation-based distortion across the mid-range spectrum—specifically between 700 Hz and 1.5 kHz—you actively generate upper harmonics that the small mobile speaker can easily reproduce.
When the consumer listens on their phone, their brain captures these newly created upper harmonics within the midrange. Through a powerful psychoacoustic phenomenon known as the Missing Fundamental, the human brain automatically calculates and completes the deep bass frequency by itself, creating the illusion of a rich low-end on a speaker that cannot physically produce it.