CHAPTER III

RESEARCH METHODOLOGY

This chapter describes the methodological approach applied in developing the

Digital Audio Filtering and Calibration System. It highlights the sequence of activities

undertaken by the researchers, which include system planning, prototype

development, integration of filtering and calibration features, as well as the testing

and assessment procedures carried out to evaluate the system’s effectiveness in

enhancing speech intelligibility.

Research Methodology

Drawing the work of Roy and Paliwal (2021) on adaptive speech

enhancement, the development of the Digital Audio Filtering and Calibration System

was guided by a structured and systematic approach. Their research highlights the

effectiveness of advanced filtering algorithms, such as Kalman filtering, in improving

speech intelligibility under noisy and reverberant conditions—circumstances

commonly encountered in classrooms and audio-visual rooms. Inspired by these

principles, the present study integrates both technical innovation and user-centered

design to ensure clarity and practicality in real-world settings.

The methodology focuses on aligning system features with educational needs,

addressing challenges in communication, and creating solutions that are both

functional and sustainable. Core elements of the system include modularity for

scalability, automated calibration for adaptive performance, and low power

consumption to ensure efficiency. In line with Roy and Paliwal’s emphasis on

robustness and adaptability, the design process combines expertise in acoustics,

electronics, and digital signal processing with practical considerations from classroom

environments.
 The development process followed iterative cycles of prototyping and testing.

Each cycle involved evaluating both hardware and software integration, then refining

the system based on feedback and performance results. This iterative approach

allowed the researchers to gradually improve clarity, stability, and usability, ensuring

that the final design remained reliable and adaptable to various acoustic conditions.

Evaluation was carried out through two main phases: usability testing with

educators and learners to measure improvements in speech clarity, and technical

validation to assess signal accuracy, stability, and reliability. Stress testing was also

performed to confirm system durability under extended operation. These procedures

reflect best practices in engineering design, emphasizing continuous improvement and

responsiveness to real-world conditions.

In addition, the methodology integrates concurrent development of both

product and process. Standards for quality control and calibration were observed to

guarantee consistent performance in educational environments. Feedback loops were

built into the development cycle so that user experience directly informed further

refinement of the system.

Finally, to ensure long-term relevance, the methodology incorporated

provisions for predictive maintenance and adaptability. Through monitoring and the

possibility of future upgrades, the system is designed to evolve alongside emerging

needs in educational technology. Documentation of processes and outcomes was

maintained to support knowledge sharing and innovation.

In summary, the methodology for the Digital Audio Filtering and Calibration

System applies the principles of adaptability, robustness, and user-centered design

identified by Roy and Paliwal (2021). By doing so, the system aims to provide an
effective and sustainable solution that enhances speech intelligibility in classrooms

and AVR environments.

Research Design

Richey et al. (2016) describe developmental research as “the systematic study

of designing, developing, and evaluating instructional programs, processes, and

products that must meet the criteria of internal consistency and effectiveness” (Seels

& Richey, 2016, p. 127). This approach emphasizes the close connection between

theory and practice, aiming not only to create a working product but also to contribute

to the improvement of design and development methods. Developmental research

may address specific, context-driven problems or explore broader processes to

develop more generalizable solutions.

In this study, the researchers employed a developmental research design to

create a Digital Audio Filtering and Calibration System intended to improve speech

intelligibility in classrooms and audio-visual rooms. Poor acoustics and background

noise remain major barriers to effective communication in learning environments. By

applying developmental research, the project focuses on systematically designing and

refining a prototype that integrates digital signal processing and adaptive calibration

features. The design prioritizes usability, reliability, and sustainability to ensure it is

practical for educational use.

The research design centers on developing and optimizing essential features

such as automatic calibration, test tone generation, and adaptive digital filtering.

These components enable the system to adjust in real time to varying acoustic

conditions, ensuring clarity and reducing distortion. The modular and scalable nature

of the design also allows future upgrades and modifications, reflecting the adaptability

required for long-term effectiveness.

 Automation and signal analysis are core elements of the design. By adjusting

variables such as filter sensitivity, calibration parameters, and acoustic response, the

system is tailored for maximum efficiency and accuracy. Iterative cycles of

prototyping and testing ensure that each development stage is informed by feedback

and performance evaluation, leading to a more reliable and user-centered output.

Roy and Paliwal (2021) emphasized the importance of adaptive speech

enhancement methods in addressing the challenges of noisy and dynamic

environments. Their findings informed the development of this system, particularly in

creating a design that dynamically adjusts to different acoustic conditions. By

integrating these adaptive principles, the study ensures that the system remains

effective and relevant across various classroom or AVR settings.

Ultimately, this developmental research design aligns with broader

educational objectives, particularly Sustainable Development Goal 4 (Quality

Education), which seeks to ensure inclusive and effective learning. By combining

advanced digital signal processing with iterative, user-centered development, the

project contributes to creating an accessible, sustainable, and impactful educational

technology.

Data Gathering Procedure

To develop the Digital Audio Filtering and Calibration System, the researchers

first conducted an extensive review of reliable sources such as textbooks on acoustics

and electronics, peer-reviewed journals on digital signal processing, technical

manuals, prior theses, and instructional videos. Online databases and other scholarly

platforms were also consulted to obtain up-to-date methods and practices in adaptive

audio processing and system calibration. These references provided both theoretical

and practical insights that served as the foundation for designing the prototype.
 The gathered information was not limited to technical specifications but also

included studies on real-world classroom and AVR conditions where speech

intelligibility is most affected. By examining existing systems and their limitations,

the researchers were able to identify which calibration techniques and filtering

methods would best address acoustic challenges in educational settings.

Literature Review & Data Collection

System Conceptualization
- Identify Components
- Define Features

System Design
- Hardware Layout
- Software Architecture

Prototype Development
- Programming

Testing & Calibration

- Room Response
Measurement
- Filter Adjustment

Evaluation & User Feedback

- Listening Tests
- Surveys

Refinement & Finalization

- Improve Performance
- Final Prototype
 The design process began with conceptualizing the system architecture. This

included outlining essential components such as the test tone generator, measurement

microphone, digital filters, calibration software, and microcontroller or DSP unit.

Draft layout and schematic diagrams were created to ensure that the intended

functions—particularly auto-calibration and adaptive filtering—were properly

integrated. Any potential inconsistencies identified at this stage were noted for

refinement.

Once the conceptual framework was established, the researchers proceeded

with iterative prototyping. Early versions of the system were subjected to preliminary

testing, where the accuracy of frequency response measurement, stability of filter

adjustments, and overall signal clarity were assessed. Identified issues were corrected

in successive design revisions, ensuring a gradual improvement in performance and

usability.

Product evaluation also involved technical testing under various noise levels

and acoustic conditions, simulating real classroom and AVR environments. This

ensured that the system remained consistent and reliable across different settings.

To complement technical validation, user feedback was also gathered through survey

questionnaires and controlled listening sessions. These responses provided valuable

perspectives on the clarity of speech before and after calibration, helping the

researchers fine-tune the prototype to meet user expectations.

Through this structured process of literature review, design conceptualization,

iterative prototyping, technical testing, and user validation, the data gathering

procedure ensured that the Digital Audio Filtering and Calibration System would be

accurate, reliable, and responsive to the needs of its intended users.

Block Diagram

The block diagram presents the overall architecture of the Digital Audio

Filtering and Calibration System. Each component works together to ensure that

speech intelligibility in classrooms and AVR environments is improved through

adaptive calibration and filtering. The system integrates both hardware and software

modules for real-time signal processing and user interaction. Below is the detailed

explanation of the system’s major components:

 Microcontroller/DSP Unit: Serves as the central processing unit of the

system. It handles signal acquisition, filtering, auto-calibration, and overall

system control.

 Microphone (Input Sensor): Captures the speech and test tones from the

room. This input is crucial for measuring the acoustic response of the

environment.

 Test Tone Generator: Produces calibration signals (such as sweeps or fixed

tones) that are used to measure room response and determine the necessary

filtering parameters.

 Digital Filter Module: Applies adaptive filtering algorithms to enhance

speech clarity. The parameters of the filter are automatically adjusted based

on room response analysis.

 User Interface (LCD/Tablet/PC): Provides the user with control and

feedback, such as calibration results, filter status, and system settings.

 Amplifier & Speaker System: Outputs the processed audio, ensuring

improved clarity and intelligibility.

 Data Storage/Cloud Module: Stores calibration results, system logs, and

performance data for later reference or remote monitoring.

 This block diagram demonstrates how the integration of input devices,

processing units, and output interfaces work in unison to create an effective, adaptive,

and user-friendly audio calibration system. By incorporating real-time processing and

feedback mechanisms, the design ensures that the system consistently provides

optimal speech intelligibility under varying acoustic conditions.

Microphone Digital Filter

(Input) Amplifier
Module

Microcontroller
-Signal Processing User Interface
-Auto-Calibration (Microcontroller LCD)

Test-tone Speaker System

Generator Data Storage (Output)

Figure 3. System Block Diagram of the Digital Audio Filtering and Calibration

System

Software Diagram

The software diagram illustrates the flow of operations in the Digital Audio

Filtering and Calibration System. The system begins by initializing the

microcontroller and connected modules, preparing both the hardware and software for

operation. Once activated, the microphone captures speech signals or test tones, which

are then digitized and sent to the microcontroller/DSP for processing.

The software first checks whether the system is in calibration mode or normal

operation mode. In calibration mode, the test tone generator produces signals that are

analyzed to determine the room’s acoustic response. Based on this analysis, the digital

filter parameters are automatically adjusted to compensate for distortions such as

echo, reverberation, and background noise. Once calibration is complete, the updated

filter settings are stored for real-time use.

In normal operation mode, incoming speech is continuously processed by the

adaptive digital filter. The filter enhances speech intelligibility by suppressing

unwanted noise while preserving the clarity of the voice signal. The processed signal

is then sent to the amplifier and speaker system for output.

Throughout the process, the user interface (LCD or PC application) provides

feedback to the user. It displays system status, calibration results, and error

notifications. If necessary, data such as calibration logs or performance results can

also be stored in local memory or uploaded to cloud storage for future reference.

This flow ensures that the software operates in a structured and adaptive

manner, enabling the system to maintain high-quality audio output regardless of

changes in the acoustic environment.

 Start/Initialize System

Check Mode:
Calibration or
Normal?

Generate Test Tone

Analyze Room Response

Adjust Filter Parameters

Store Calibration Setting

Capture Microphone Input

Apply Adaptive Filtering

Amplify & Output to Speakers

Display Status on User Interface

Save Logs/ Upload to Cloud

End/Loop

Figure 4. Software Flow Chart of the Digital Audio Filtering and Calibration

System
User Interface Description

The Digital Audio Filtering and Calibration System is built as a compact

device tailored for use in classrooms and Audio-Visual Rooms (AVRs). This section

provides the graphical user interface description of the Digital Audio Filtering and

Calibration System. Its design highlights portability, durability, and user convenience.

IMAGE NG GUI*

The system's graphical user interface (GUI) is implemented on a full-color

touchscreen display that acts as the main control environment for all audio processing

operations. Upon launch, the interface clearly identifies the system by displaying the

title "Audio Filter System" at the top of the screen. The seven filter modes—OFF,

LOW, HIGH, BAND, ECHO, BASS, and TREBLE—are represented by an organized

array of color-coded rectangular buttons that come below this heading. To improve

recognition, reduce selection errors, and enable users to easily reach their preferred

audio configuration, each button has a unique color.

Two horizontally aligned sliders that regulate the active filter's cutoff

frequency and audio loudness level are located beneath the filter selections. These

sliders have real-time updating numerical indicators that enable users to precisely

modify system parameters. Even when the system is operating continuously, the

sliders' positioning and size are intended to provide accurate touch input and

comfortable interaction.

A real-time status indicator at the bottom of the screen offers ongoing system

feedback. Important details like the active filter, current volume %, audio condition,

buffer count, and input channel are shown in this section. All essential controls and

feedback indications are available inside a single interface thanks to the consolidated,
high-contrast layout, which enhances usability, operational effectiveness, and

instantaneous visibility of system activity during audio processing.

*Supplies and Materials

The table below presents the list of supplies and materials that the researchers

utilized in the development of the Digital Audio Filtering and Calibration System.

Quantity Unit Description Total Cost

1 PC STM32F746G-Disco ₱

1 PC PL55 Plug Adapter ₱35

1 PC iRig Audio Interface ₱95

1 PC ST-Link Cable ₱200

Table 1. Supplies and Materials

Tools and Equipment

Table 2 lists the tools and equipment along with their purposes that the

researchers that contributed to the creation of this work. The following equipment and

tools were crucial to the creation of the Digital Audio Filtering and Calibration

System.
 Tools and Equipment Function

Microcontroller (e.g. STM32)  Acts as the brain of the system

that controls the filtering of the

noises in a room. It also displays

the calibration system that can

be manually adjust according to

the preferred audio

configuration.

Audio Interface (e.g. iRig)  Converts the incoming voltages

of the mic into the

microcontroller to lessen the

latency and filter the noise/audio

accurately.

PL55 Plug Adapter  It uses to connect the receiver of

the mic into the audio interface

of the system.
*Construction Procedures

The construction of the Digital Audio Filtering and Calibration System

followed a structured and organized process to ensure that the system would function

as intended. Each phase was carefully executed to integrate both the hardware and

software aspects of the project. The following steps were undertaken by the

researchers:

1. Component Preparation

The researchers gathered all necessary materials and components based on the

system’s requirements. These included the microcontroller (e.g., STM32), ST-Link

Cable, iRig Audio Interface.

2. Programming and Software Development

Using the STM Cube IDE, the team developed the program that would run on

the microcontroller. The code handled audio input processing, filtering algorithms,

auto-calibration routines, and manual-calibration routines. Relevant libraries were

used and customized to match the system’s operation.

3. Testing and Adjustment

Several tests were performed to evaluate the functionality of the system. These

tests included checking audio input/output quality, verifying the filter’s performance,

and testing the auto-calibration and manual-calibration feature. Adjustments were

made as needed—such as modifying code parameters, replacing defective parts—to

improve system performance.

4. Evaluation and Validation

Once the system was fully functional, the researchers conducted an evaluation

to assess the effectiveness of the filtering and calibration features. Feedback from
selected users was collected and analyzed. The results were used to make final

refinements to the prototype before documentation.

*Construction Time Frame

*Research Instrument

*Statistical Treatment