Predict the future by creating it

Why Microphones Can't Hear Like Humans: The Journey Behind the Creation of GCEDAC

By: Hardy Nanda

System Engineer, Programmer, Acoustician, and Architect with 25 Years of Experience

How to measure delay time

Dear colleagues and friends,

Generally, there are three established methods for measuring the delay between a subwoofer and a full-range loudspeaker:

1. Distance Measurement: Measuring the physical distance from the listener to the subwoofer, followed by the distance to the main speaker. The differential distance is subsequently converted into the time domain and inputted into the Digital Signal Processor (DSP).

2. Acoustic Analysis: Utilizing a measurement microphone and acoustic analysis software to derive the required delay, the results of which are then programmed into the DSP.

3. Empirical Estimation: Estimating the delay based on practical experience and intuition.
   
   

Elaborating on the second method, the underlying mechanism is as follows: acoustic energy is captured by the microphone (comprising a diaphragm and voice coil assembly) and transduced into electrical energy. This analog signal is subsequently captured by an audio interface (via an Analog-to-Digital Converter), processed into discrete numerical values, and plotted as a visual graph. Analytical conclusions are then drawn from this graphical representation to configure the DSP.

In this context, the FRESL Impulse, which is the subject of my research, can yield an objective parameter when subjected to an inverse transform. This parameter functions to evaluate whether the sound reinforcement system (specifically, the time alignment between the subwoofer and full-range speakers) is genuinely coherent, based entirely on psychoacoustic principles. Conceptually, this metric operates similarly to the Speech Transmission Index (STI); however, it is specifically designed to assess phase coherence relative to the human auditory and perceptual systems investigated in my research.

Through the GCEDAC software, which I developed based on this FRESL Impulse principle, we are now presented with a fourth method.

The Procedure: An impulse signal is routed via the GCEDAC software to the target sound system. The listener then critically evaluates whether the low-frequency or high-frequency acoustic energy reaches their auditory perception first. Following this, the delay settings within the DSP are continuously adjusted until both the low and high frequencies are perceived simultaneously by the human auditory system.

This method may appear unconventional. Indeed, it is highly unconventional, as it represents a novel approach to determining time alignment that relies purely on psychoacoustics.

I hope this brief explanation provides clarity and circumvents any unnecessary debate. Thank you for your attention.

UNDERSTANDING THE FRESL CHARACTER

I'll share a little about the FRESL (frequency range envelope sound loudness) parameter, which is closely related to the "center time" of human hearing and processing sound perception. I've shared here, which is one of the fundamentals of the GCEDAC I created. Several images below visually display changes in the "center time" of sound perception with respect to frequency and room reverberation time. I'll show several visual images with categories of 50Hz with room reverberation times of 0.5 seconds, 1 second, and 2 seconds. Then 1000Hz with room reverberation times of 0.5 seconds, 1 second, and 2 seconds. This is an illustration of the center time of human perception of the frequency and reverberation time of space.

Based on my observations over five years of intensive research, the way humans hear involves a characteristic I call Frequency Range Envelope Sound Loudness, or FRESL for short (a term and independent hypothesis I invented, given the current lack of similar research literature). Based on this principle, it turns out that the center of time perception (center time) in the human ear can shift forward or backward in response to low and high sounds, depending on this parameter.

Why Your Sound System Looks Perfect on Paper but Sounds "Cold" to Your Ears

The Paradox of the "Perfect" Graph

In the high-stakes world of professional sound engineering, we have become dangerously fixated on the monitor screen. As architects of sound, we find a sense of security in seeing linear phase graphs, surgical impulse responses, and millisecond alignments that appear mathematically unassailable. Yet, there remains a persistent, haunting frustration: a system that looks flawless on a dual-channel FFT analyzer can still feel "cold," "hollow," or "lacking in soul" when the music finally begins to play.

This discrepancy exists because we are confusing two entirely different domains. A measurement microphone captures sound pressure levels (SPL)—a variable of wave physics—whereas the human brain processes an integrated "experience" through the lens of psychoacoustics. This gap between physics and perception was the catalyst for Hardy Nanda’s 25-year journey through acoustics and software architecture. After decades of following industry-standard calibration—aligning peaks and applying windowing to impulses—Nanda realized that technical accuracy on a graph did not equate to a unified sonic identity. This realization led to the birth of GCEDAC, a methodology designed to reconcile objective data with the most sophisticated processing algorithm on Earth: the human auditory system.

Takeaway 1: The "Blindness" of the Measurement Microphone

While a measurement microphone is a clinical, high-precision instrument, it is a fundamentally limited diagnostic tool for evaluating a listener’s experience. The microphone operates on the principle of "mechano-electrical transduction," where air pressure fluctuations vibrate a diaphragm to generate voltage. This process is purely data-driven and aesthetically "blind."

In contrast, the human brain performs what we call Auditory Scene Analysis (ASA). Unlike the microphone, which measures static pressure at a single point in space, the brain utilizes binaural integration. A microphone captures direct sound and room reflections as a single, overlapping entity of energy; for instance, it records comb filtering as a tangible destructive interference pattern at a single point. To a technician, this looks like a dip that needs "fixing." However, the human brain is capable of "hearing through" such interference. We use those very reflections to perceive "envelopment" and "room depth." When we focus solely on "chasing a straight phase line" to satisfy a microphone's linear requirements, we often strip away the spatial cues the brain expects, resulting in a sound that is technically correct but emotionally sterile.

Takeaway 2: The Cochlea’s Biological "Group Delay"

The human ear is not a linear sensor; it is a masterpiece of biological engineering that performs a "Biological Fourier Transform." Within the cochlea, a phenomenon known as Tonotopy dictates that high and low frequencies are processed at different speeds and locations. High frequencies are detected at the base of the basilar membrane, while low frequencies are detected at the apex.
"The human ear does not hear all frequencies at the same speed; there is a natural group delay in our cochlea."

This psychoacoustic reality is the root cause of the "bass lag" seen in traditional calibration. Standard methods often align speakers based on the voltage peak of a sine sweep. However, the auditory system is not a peak-detector; it is sensitive to the "energy package" or the "center of gravity of energy." Because the cochlea naturally introduces its own biological delay, aligning solely to the physics of the wave often results in bass that feels perceptually "behind" the transient attack of the high-frequency drivers.

Takeaway 3: The 97-Revision "Human-Centric Impulse"

To bridge the gap between wave physics and brain processing, the GCEDAC "Human-Centric Impulse" algorithm was developed. This was not a simple mathematical formula but a five-year iterative process involving 97 distinct revisions. This approach aligns with academic concepts such as "Gammatone Filterbanks," which attempt to model the auditory system's multi-resolution processing.
The development was defined by several critical stages:

• Identifying Fundamental Frequencies: Determining the specific spectral triggers required to stimulate the basilar membrane effectively.

• Harmonic Architecture: Humans do not hear pure sine waves as "musical." GCEDAC adds harmonic structures to act as "anchors," allowing the brain to recognize timbre and timing within reverberant spaces.

• Energy Balancing: Assigning precise gain to ensure the signal pierces the noise floor without masking delicate transient details.

• The Discovery of Non-Linear Delay: Perhaps the most significant finding was that delay per frequency is neither gradual nor linear. The human ear perceives harmony through specific, non-linear time shifts between frequencies.
  

By aligning the sound system to this "Human-Centric Impulse," we synchronize the audio to the Temporal Integration Window of the brain rather than the peak energy of a microphone.

Takeaway 4: The "Disappearing Subwoofer" Phenomenon

The ultimate goal of perceptual alignment is achieving a state of coherence where the hardware becomes invisible. In many professional systems, the kick drum feels disconnected—the high-frequency "click" of the mains and the low-frequency "thump" of the sub arrive as two distinct events.
The GCEDAC workflow utilizes an "Audition and Identification" process. By using the custom impulse, the engineer can clearly distinguish the temporal arrival of the two components. By adjusting the delay in the DSP until these two sounds fuse into a single, dense entity, we achieve a specific acoustic phenomenon:

"The subwoofer seems to disappear... the impact feels as if it is radiating directly from the full-range speakers."

When alignment is perceptually correct, the bass impact feels unified and focused, regardless of whether the subwoofers are physically on the floor or flown in an array.

Takeaway 5: Why Technology Must Remain the Servant

As architects, we must recognize a fundamental shift from "Physical Time Alignment" to "Perceptual Time Alignment." An over-reliance on Digital Room Correction (DRC) to achieve a flat line on a graph can be destructive. For instance, using extreme EQ to "fill" a physical room null forces transducers to work beyond their linear limits, creating harmonic distortion.

The human brain is exquisitely sensitive to this type of "unnaturalness." Even if the measurement graph shows a perfect straight line, the auditory cortex detects the distortion as a lack of fidelity. Technology should be a bridge, not a barrier; the ear must remain the supreme judge of the system's performance.

Conclusion: Reconciling Physics and Cognition

High fidelity is not defined by a 0 dB deviation on a computer screen. True fidelity is achieved only when a sound reproduction system successfully deceives the brain into believing a recorded event is happening live. To reach this level, we must move beyond the reductionism of raw electronic data and respect the biological complexity of the human listener.

Ultimately, we must ask ourselves: Are we calibrating our systems to satisfy a $500 microphone, or the most sophisticated algorithm on Earth: the human auditory cortex?

Perceptual Time Alignment

Why Microphones Can't Hear Like Humans: The Journey Behind the Creation of GCEDAC

In the world of professional sound engineering, we are often fixated on the screen. We see linear phase graphs, sharp impulse responses, and millisecond figures that look mathematically "perfect" on the computer monitor. However, one major question haunted me for years: Why do technically perfect measurement results sometimes still feel "lacking in solidity" to our ears?

About eight years ago, I found myself at a crossroads in my 25-year career in the acoustics and sound engineering industry. At that time, I was performing calibration using the standard industry method—using EASERA, applying windows to the impulse, and aligning the time between subwoofers and full-range speakers. In theory and on paper, everything was accurate. But when the music played, something was missing. The bass and mid-high frequencies didn't blend with a unified "soul."

As a programmer and architect, I am accustomed to seeking the logic behind structures. I realized one fundamental thing: Measurement microphones capture sound pressure, but the human brain processes the listening experience.

The Psychoacoustic Paradox

Our auditory system works in a unique way. The brain does not process low and high frequencies at the same speed. There is a biological delay time in the cochlea and a specific way the auditory nerves integrate sound energy.

This is the core challenge: If we align speakers based solely on the peak energy captured by a microphone (pure physics), we often neglect how humans "perceive" the impact of that sound (psychoacoustics). This is why many sound systems can sound "clean on the graph" but "cold in feeling."

The Birth of GCEDAC

Driven by this unease, I created GCEDAC—a methodology and software that integrates six main pillars: Gain, Crossover, Equalizer, Delay, Alignment, and Coherence.

The main difference with GCEDAC lies in how it performs time calibration. I developed a "Human-Centric Impulse"—a synthetic impulse sound designed to mimic how the human ear responds to sonic transients. With this approach, calibration is no longer just numbers on paper, but a synchronization that aligns with the human perceptual system.

The result? Sound that is more solid, powerful, and most importantly: Coherent.

Conclusion: The Ear is the Supreme Judge

My 25 years of experience have taught me that technology is a servant, not a master. As a system engineer, our task is not to satisfy measurement microphones, but to deliver the best experience for the human listener.

The professional audio world needs an approach that goes beyond mere electronic data. We need an approach that understands room architecture, programming logic, and the wonder of psychoacoustics. That is the mission I carry through GCEDAC.

Is there similar software to GCEDAC in the world?

The short answer is: Philosophically, your GCEDAC occupies a very unique position and is rarely found in a standard commercial software package.

While there are many professional calibration software solutions available, most of them still adhere to the principle of "Physical Time Alignment" (aligning waveforms objectively), not "Perceptual Time Alignment" (aligning based on how the brain processes sound) as you do.

Here is a comparative map of GCEDAC versus what exists in the global market today:

1. Standard Measurement Software

Focus: They are "Measurement Microphones." Their task is to provide raw data as physically accurate as possible (Impulse Response, Phase, Magnitude).

Weakness vs. GCEDAC: These tools provide objective data, but the interpretation is left entirely to the human operator. If the technician merely chases a "straight phase line" without understanding psychoacoustics, the result is often as you experienced: good graphically, but lacking in "solidity" sonically.

Latest Developments: some measurement software has begun introducing Wavelet Analysis, which attempts to display time-frequency data with a resolution closer to how the human ear works (multi-resolution). However, it has not yet evolved into a perception-based "auto-align" algorithm.

2. Automatic Correction Software

Focus: They use advanced algorithms to correct the Impulse Response and phase across a room.

Similarity: They pay significant attention to the time domain to create a sharp "phantom image."

Difference: They typically aim to make speakers behave as a mathematically "Perfect Point Source." They strive to eliminate the influence of the room. In contrast, GCEDAC appears more focused on synchronizing components (Sub vs. Top) by accounting for the latency of the human auditory system.

3. "Temporal Equalization" Technology

A another company has developed TQ (Temporal EQ) technology. They use DSP to correct a speaker's transient response to eliminate "smearing" (temporal blurring). This is the approach that comes closest to your idea of sonic "solidity." However, it is usually locked into their own proprietary hardware processors and is not general-purpose calibration software.

Why GCEDAC Could Be a "Game Changer"?

The discovery (creating an impulse sound that approximates how humans hear) is very close to the concept of a Gammatone Filterbank in academic research. The human ear does not hear all frequencies at the same speed; there is a natural group delay in our cochlea.

If GCEDAC successfully compensates for this, we have solved a classic problem in the audio industry:

The Problem: Using standard Pink Noise or Sine Sweeps often yields delay figures that are phase-"correct" but perceptually cause the "bass to lag."

GCEDAC's Solution: By using a psychoacoustically-based impulse, you align the speakers not to the microphone, but to the processing mechanism in the listener's brain.

Deconstructing the Acoustic Paradigm: The Divergence between Sound Pressure (SPL) and Auditory Perception in Profesional Audio System Calibration

Abstract

Audio calibration is often trapped in the reductionism of numerical data. Measurement microphones, as precision instruments, capture fluctuations in air pressure across the time and frequency domains. However, human hearing is not a linear sensor; it is an interpretive system involving complex neural processing—from mechanical transformation in the cochlea to cognitive integration in the auditory cortex. This article analyzes why a "flat" frequency response curve does not always correlate with emotional fidelity or perceptual accuracy. By dissecting the anatomy of hearing and the mechanics of microphones, we posit that true calibration must bridge wave physics with auditory psychophysics.

I. The Ontology of Sound: Pressure vs. Experience

In the realm of physics, sound is a longitudinal wave propagating through an elastic medium. A measurement microphone perceives sound as a variable of Sound Pressure Level (SPL), quantified in Pascals or decibels (dB). To a microphone, sound is purely data.

For a human, sound is an experience. This fundamental difference stems from the fact that the human brain does not merely receive data; it performs Auditory Scene Analysis (ASA). The brain segregates instruments, evaluates spatial dimensions, and even ignores certain distortions through neural adaptation mechanisms.

II. Mechanics of the Measurement Microphone (The Physical Sensor)

Measurement microphones (typically electret or externally polarized condenser types) are engineered with an ultra-light, rigid diaphragm to ensure clinical accuracy.

2.1. Mechano-Electrical Transduction

The microphone operates on the principle of capacitance. Fluctuations in air pressure vibrate the diaphragm, altering the distance between it and the backplate, which generates a change in electrical voltage.

• Linearity: The microphone is designed to be "blind" to aesthetics; it must respond linearly across the spectrum (e.g., 20Hz – 20kHz).

• Omnidirectionality: Most measurement microphones act as pure pressure transducers, meaning they capture pressure at a single point without accounting for the natural directional cues the human head provides.

2.2. Sensory Limitations of Microphones

A microphone lacks "memory" or "attention." It captures direct sound and reflections as a single, overlapping entity of energy. In a room, a microphone will record comb filtering (destructive interference) as a tangible drop in dB, whereas the human brain can often "hear through" such interference due to binaural localization.

III. Anatomy of the Human Auditory System (The Bio-Processor)

To understand why a microphone cannot replace the ear, we must dissect the auditory pathway from the outside in.

3.1. The Outer Ear and HRTF (Head-Related Transfer Function)

The human ear is not "flat." The pinna (outer ear) possesses complex folds that act as a passive acoustic filter. These folds create resonances and attenuations at high frequencies (above 5kHz) that allow the brain to determine if a sound originates from above, below, front, or back. A measurement microphone lacks a torso, head, and pinnae, thus losing the crucial spatial data the brain uses to construct "imaging."

3.2. The Middle Ear: Impedance Matching

The eardrum (tympanic membrane) drives three tiny bones (ossicles: malleus, incus, and stapes). This system acts as a mechanical transformer to match the impedance between the air (low-density medium) and the fluid inside the cochlea (high-density medium). Without this, 99% of sound energy would be reflected away.

3.3. The Cochlea and Organ of Corti: Biological Fourier Transform

The cochlea is a masterpiece of biological engineering. Inside, the basilar membrane possesses mechanical properties that vary from the base to the apex.

• Tonotopy: High frequencies are detected at the base, while low frequencies are detected at the apex. This is a physical manifestation of a Fourier Transform.

• Hair Cells: Outer Hair Cells (OHC) act as "biological amplifiers," sharpening frequency resolution. This allows humans to distinguish between two very close pitches—something a microphone merely sees as a blurred clump of frequency energy.

IV. Psychoacoustics: The Algorithms Within

Once electrical signals leave the cochlea via the auditory nerve, they enter the brainstem and eventually the auditory cortex. This is where "measurement" transforms into "hearing."

4.1. Equal-Loudness Contours (Fletcher-Munson)

Humans do not perceive all frequencies at the same volume. We are highly sensitive to the 2kHz – 5kHz range (the speech intelligibility zone) and remarkably insensitive to low frequencies (bass).

• Calibration Implication: If a microphone measures a flat bass response, the human ear perceives it as "anemic" or lacking. This is why target curves (like the Harman Curve) always feature a low-end elevation.

4.2. The Precedence Effect (Haas Effect)

The human brain has a temporal integration window of approximately 5 to 40 milliseconds. If room reflections arrive within this window, the brain does not hear them as separate echoes; instead, it fuses them with the original sound to add "spaciousness." A microphone, however, shows these reflections as peaks or dips on a graph, tempting technicians to apply EQ corrections that the brain has already "auto-corrected."

V. Comparative Analysis: Microphone vs. Human Brain

Based on the critiques found in Hardynanda (2026), we can distinguish the two systems as follows:

• Response Domain: The microphone measures static pressure at a single point in space, while the brain utilizes binaural integration (processing inputs from two ears simultaneously).

• Temporal Resolution: Microphones possess ultra-high resolution capable of seeing microseconds, whereas the brain integrates sound over time windows, prioritizing the "initial transient."

• Frequency Response: Microphones are strictly linear (aiming for 0 dB deviation), while the brain is non-linear (sensitivity changes based on the SPL/volume).

• Spatial Processing: Microphones capture all reflections as destructive or constructive interference; the brain utilizes reflections to perceive "envelopment" and room depth.

• Data Interpretation: Microphones provide raw data; the brain performs Auditory Scene Analysis (ASA) to separate music from noise.

VI. Toward Human-Centric Calibration

How do we calibrate a system if our tool (microphone) and our target (the brain) speak different languages?

6.1. Frequency Dependent Windowing (FDW)

Modern software employs FDW to mimic the brain. By limiting the analysis window, the software focuses on the direct sound and early reflections, ignoring the long-term room resonances that the brain naturally filters out.

6.2. Phase Coherence over Magnitude

The brain is exquisitely sensitive to time-domain alignment (phase). Phase distortions caused by speaker crossovers are often more jarring to a listener than a 2dB deviation in frequency response. Effective calibration must prioritize impulse response to ensure a stable "phantom center."

6.3. The Danger of Over-Correction

There is a risk in Digital Room Correction (DRC). Attempting to "fill" a null caused by room physics forces the speaker to work beyond its mechanical limits, increasing harmonic distortion. The brain will detect this distortion as "unnaturalness," even if the measurement graph shows a perfect straight line.

VII. Conclusion: Reconciling Physics and Cognition

The core thesis of GCEDAC serves as a vital reminder: a microphone is a diagnostic tool, not the final judge. High Fidelity is not about achieving a flat line on a computer screen. True fidelity occurs when a sound reproduction system successfully deceives the brain into believing a recorded sonic event is happening live in the listener's room. To achieve this, we must stop treating the ear as a microphone and start treating calibration as an attempt to satisfy the most sophisticated algorithm on Earth: the human auditory cortex.

Architectural Analysis and Psychoacoustic Methodology of GCEDAC 2.0 Software:

A Deterministic Approach to Professional Audio Time Alignment

Abstract

GCEDAC (Proprietary Audio Time Alignment Software) is a software platform designed to address phase anomalies and temporal discrepancies in multi-driver sound reproduction systems relative to the human auditory system. It utilizes a low-level ASIO library to achieve minimal latency and bit-perfect accuracy. This article examines the internal mechanisms of GCEDAC, from the generation of coherent pink noise signals to the wide-spectrum group impulse algorithm independently developed by Hardy Nanda.

I. Philosophical and Technical Foundations: Psychoacoustic-Based Perceptual Alignment

Unlike standard microphone-based automatic calibration systems, which often fail to capture human cognitive perception of transient integrity, GCEDAC adopts a Perceptual Professional Audio Time Alignment approach. This system recognizes that the human ear, through a process of temporal integration in the auditory cortex, has varying sensitivity to phase shifts at different frequencies.

GCEDAC not only measures the time of arrival of a wave (Time of Arrival), but also manipulates the harmonic structure and phase correlation between the left and right channels to ensure that the "acoustic center" of an audio system is precisely aligned with the listener's reference point.

II. Engine Architecture and Memory Management

Built for high efficiency, GCEDAC exhibits very low resource usage, as seen in the system log (Memory Usage: ~1098.72 KB). This is possible due to the use of Object Pascal compiled directly to binary machine code, avoiding the overhead typically found in managed code-based environments or virtual machine runtimes. Integrating an ASIO (Audio Stream Input/Output) driver, GCEDAC communicates directly with audio hardware using the ASIO protocol. This protocol bypasses the Windows mixer kernel to prevent resampling, dither, or latency introduced by the Windows Audio Session API (WASAPI) or DirectSound.

Buffer Stability and Synchronization: With a Buffer Size configuration of 256 samples at a sample rate of 44.1 kHz, GCEDAC ensures a stable data stream without buffer underrun or audio glitches, which is crucial when conducting highly sensitive phase correlation testing.

III. Test Signal Methodology: Pink Noise Mode Selection

GCEDAC uses a proprietary, calibrated variant of Pink Noise. Pink noise was chosen because it has constant energy per octave, which is psychoacously more representative of human auditory perception than white noise. This custom pink noise is very useful for determining the gain structure of professional electronic sound systems.

Coherent Mode: Identical signals are sent to both channels. Used to determine phantom center localization. If time alignment is inaccurate, the sound source will sound "wide" or shifted from the center point.

Inverted Channel (Left/Right): This feature induces deliberate phase cancellation. Under perfect alignment conditions, a 180-degree phase reversal in one of the channels should produce a null point, or a perceptually significant drop in sound level, at the transition frequency or crossover point.

IV. Impulse Mode Selection: Temporal Resolution and Transient Response

The Impulse Mode module in GCEDAC is designed to analyze the impulse response (IR) characteristics of the transducer.

Full Range Speaker vs. Subwoofer Speaker: The inertial characteristics of large drivers (subwoofers) require a different impulse approach than high-frequency drivers. GCEDAC adjusts the pulse width and spectral content of the impulse signal to ensure the speaker cone excursion provides accurate temporal information.

Critical Point for Special Purpose: This algorithm targets the critical crossover frequency where overlap between the two drivers often causes phase distortion.

Wide Spectrum Group Impulse: This is the pinnacle innovation in GCEDAC, where a series of wide-spectrum impulses are used to evaluate the group delay across the entire audible spectrum. The goal is to ensure that all fundamentals and harmonics of an instrument arrive at the listener's ears simultaneously.

V. Implementation Analysis on ASIO

In documented usage scenarios, GCEDAC works synergistically with the ASIO control panel. The Direct Monitor setting on the interface is crucial when using GCEDAC coherent mode to verify signal integrity before it is processed by the space reproduction system. Disabling Loopback and Auto Power Save ensures that the data stream integrity is not interrupted by the operating system's power management.

VI. Conclusions and Industry Implications

GCEDAC 2.0 is more than just a calibration tool; it is a precision instrument that bridges the gap between objective data from digital signal processing and the subjective experience of human hearing. With full compliance with Windows 10/11 x64 standards and optimization at the Lazarus compilation level, this software offers audio engineers a solution for achieving crisp sound imaging, an accurate soundstage, immersive subwoofers, and fast transient response.

Hardy Nanda's development, spanning over five years of research, reflects his dedication to Acoustic Science Research, positioning GCEDAC as an essential tool in the professional audio production chain that prioritizes uncompromising quality.

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