Related Experiment Video
Updated: Oct 29, 2025

06:04
Systematic Hearing Performance Evaluation Process for Adolescents with Cochlear Implantation at Early Ages
Published on: March 24, 2023
534
Hardware-efficient auto-reconfigurable hearing aids using 3-level octave interpolated filters for auditory
1APJ Abdul Kalam Technological University, Thiruvananthapuram, Kerala, India. tomsondevis@gmail.com.
Physical and Engineering Sciences in Medicine
|July 6, 2021
Summary
This study introduces a novel, low-complexity reconfigurable hearing aid with auto-adapting capabilities for diverse hearing impairments. The device efficiently matches audiograms, reducing setup time and enabling cost-effective solutions.
Area of Science:
- Electrical Engineering
- Biomedical Engineering
- Signal Processing
Background:
- Traditional hearing aids require hardware modifications for different hearing disabilities.
- Manual audiogram matching is time-consuming and requires expert intervention.
Purpose of the Study:
- To propose a novel, low-complexity, auto-reconfigurable hearing aid.
- To enable automatic adaptation to various hearing impairments (mild to severe).
- To reduce the time and complexity of hearing aid fitting.
Main Methods:
- Dividing the audio spectrum into three regions with four adaptive schemes each.
- Automatic selection of optimal schemes based on individual hearing thresholds.
- Utilizing octave and fractional interpolation with a Parks-McClellan filter for sub-band generation.
Main Results:
- The proposed structure uses only 18 coefficient multipliers, achieving up to 92% savings.
- Delay and matching errors are within acceptable global limits.
- Hardware implementation on FPGA confirmed a compact and power-efficient design.
Conclusions:
- The auto-reconfigurable hearing aid effectively caters to a wide range of hearing disabilities.
- The system minimizes manual intervention, significantly reducing fitting time.
- The low complexity enables cost-effective manufacturing of advanced hearing aids.
Related Concept Videos
Active Filters
1.1K
Active filters are electronic circuits that use operational amplifiers (op-amps), resistors, and capacitors to filter out unwanted frequency components from a signal. A first-order low-pass active filter is designed to pass signals with a frequency lower than a certain cutoff frequency and attenuate frequencies higher than that cutoff frequency. The transfer function for a first-order low-pass active filter is:
1.1K
Design Example
419
The innovation of touch-tone telephony revolutionized the telecommunications industry by replacing the traditional rotary dial with a dual-tone multi-frequency (DTMF) signaling system. This system uses a matrix-style keypad with buttons arranged in four rows and three columns, creating 12 distinct signals each assigned to a pair of frequencies. Each button press results in a simultaneous generation of two sinusoidal tones – one from a low-frequency group (697 to 941 Hz) and one from a...
419
Passive Filters
745
Passive filters are utilized to shape the frequency spectrum of signals across a diverse array of applications. These filters, using only passive elements like resistors (R), inductors (L), and capacitors (C), are capable of selectively allowing or blocking certain frequency ranges without the need for external power sources.
Low-Pass Filters
Low-pass filters are designed to transmit signals with frequencies lower than the cutoff frequency, ωc, and attenuate those above it. The cutoff...
Low-Pass Filters
Low-pass filters are designed to transmit signals with frequencies lower than the cutoff frequency, ωc, and attenuate those above it. The cutoff...
745
The Cochlea
48.0K
The cochlea is a coiled structure in the inner ear that contains hair cells—the sensory receptors of the auditory system. Sound waves are transmitted to the cochlea by small bones attached to the eardrum called the ossicles, which vibrate the oval window that leads to the inner ear. This causes fluid in the chambers of the cochlea to move, vibrating the basilar membrane.
48.0K
Reconstruction of Signal using Interpolation
425
Signal processing techniques are essential for accurately converting continuous signals to digital formats and vice versa. When a continuous signal is sampled with a period T, the resulting sampled signal exhibits replicas of the original spectrum in the frequency domain, spaced at intervals equal to the sampling frequency. To handle this sampled signal, a zero-order hold method can be applied, which creates a piecewise constant signal by retaining each sample's value until the next...
425

