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#Synopsis

This document details the design and implementation of a real-time, single-channel Active Noise Cancellation (ANC) system using an STM32F407VGT6 microcontroller and the Least Mean Square (LMS) adaptive filter algorithm. The project aims to effectively reduce unwanted acoustic noise, particularly in low-frequency ranges, and demonstrates the system's performance through measured improvements in Signal-to-Noise Ratio (SNR). The work highlights the potential of cost-effective microcontrollers for real-time digital signal processing applications.

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0% found this document useful (0 votes)
55 views5 pages

#Synopsis

This document details the design and implementation of a real-time, single-channel Active Noise Cancellation (ANC) system using an STM32F407VGT6 microcontroller and the Least Mean Square (LMS) adaptive filter algorithm. The project aims to effectively reduce unwanted acoustic noise, particularly in low-frequency ranges, and demonstrates the system's performance through measured improvements in Signal-to-Noise Ratio (SNR). The work highlights the potential of cost-effective microcontrollers for real-time digital signal processing applications.

Uploaded by

smsanjay189
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PDF, TXT or read online on Scribd

ANC USING ADAPTIVE FILTERS ACADEMIC YEAR: 2025-26

Abstract

Active Noise Cancellation (ANC) is a critical technique in modern audio signal processing
for attenuating unwanted acoustic noise. This project details the design, implementation, and
evaluation of a real-time, single-channel, feedforward ANC system. The system is architected
around an STM32F407VGT6 microcontroller, which features a high-performance ARM
Cortex-M4 core. The widely-used Least Mean Square (LMS) adaptive filter algorithm is
implemented to dynamically model an acoustic noise path and generate a corresponding anti-
phase signal. The algorithm's real-time performance is achieved by leveraging the CMSIS-
DSP library, with audio signal acquisition and output handled by a custom analog front-end
and the microcontroller's integrated peripherals. The system's effectiveness is validated by its
ability to significantly reduce stationary, broadband noise, with performance quantified by
the measured improvement in the Signal-to-Noise Ratio (SNR). This work demonstrates the
viability of using cost-effective, general-purpose microcontrollers for computationally
intensive, real-time DSP applications.

Introduction

With the rise in the use of industrial machines such as engines, fans, and compressors, issues
related to acoustic noise are on the rise. Although passive noise reduction techniques (such as
implementing sound-absorbing materials) are effective for high-frequency sounds, they tend
to be bulky, expensive, and inefficient for low frequencies.
Active Noise Control (ANC) systems are employed to address this issue. ANC operates
bygenerating an "anti-noise" signal that neutralizes the undesired sound through the principle
of superposition. An average ANC system consists of:
1. A reference microphone that picks up the sound,
2. A filter that adapts to process this signal,
3. A speaker that produces the counter-sound, and a microphone for error measurement
that assesses the amount of remaining noise, aiding in filter adjustment.
A widely utilized algorithm is the Filtered-X Least Mean Square (FXLMS) paired with a Finite
Impulse Response (FIR) filter.
Nevertheless, in practical situations, the ANC system frequently encounters issues because of
nonlinear distortions. These occur when elements such as microphones or amplifiers become
overloaded or saturated, particularly when the incoming sound is excessively loud. A frequent
problem is signal clipping, where the signal is truncated at a peak level.
The Filtered-X Least Mean Square (FxLMS) algorithm with a Finite Impulse Response (FIR)
filter is widely used. However, practical ANC systems may face nonlinear distortions due to
overload or saturation of microphones or amplifiers, leading to problems like signal clipping
when sound levels are too high.

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Literature Review

[Link] Learning-Based Active Noise Control on Construction Sites


Mostafavi, A., & Cha, Y.-J. (2023). Automation in Construction, 151, 104885. This paper
presents a feedforward, deep-learning ANC controller called CsNNet. It is designed for highly
variable construction noise and shows significant broadband reduction. It also offers practical
real-time use compared to traditional ANC.
2. DNoiseNet: Deep Learning-Based Feedback Active Noise Control in Various Noisy
Environments Cha, Y.-J., Mostafavi, A., & Benipal, S. S. (2023). Engineering Applications of
Artificial Intelligence, 121, 105971.
This work proposes a feedback ANC design that models primary and secondary paths and
accounts for acoustic delay with deep networks. It outperforms the classic feedback FxLMS in
complex, changing noise conditions.
3. Deep Learning-Assisted Active Noise Control in a Time-Varying Environment (2023)
Journal of Mechanical Science and Technology.
This study shows a DL-assisted method for updating secondary paths in real time. In duct
experiments with moving sensors, this approach achieves around 10 dB of broadband reduction
despite significant changes in the boundaries.
4. Low-Frequency Active Noise Control System Based on Feedback FxLMS (Hybrid ANC)

(2023). Electronics, 14(7), 1442.


This paper introduces a hybrid system that combines FxLMS-based adaptive control with an
audio-balance circuit. Simulations and hardware tests show reductions of around 21.8 dB and
better resistance to fluctuating noise.
5.A Variable Step-Size FxLMS Algorithm for Nonlinear Feedforward ANC (2025). Sensors,
25(8), 2569.
This research presents a variable step-size, neuro-fuzzy controller. It improves convergence
and stability in nonlinear ANC situations, with simulations surpassing conventional ANC
benchmarks.
6. Deep Generative Fixed-Filter Active Noise Control
Luo, Z., Shi, D., Shen, X., Ji, J., & Gan, W.-S. (2023). arXiv preprint.
This work addresses the slow convergence of adaptive ANC. It generates task-specific fixed
filters using a deep generative approach. This method enhances responsiveness to rapidly
changing noise.

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ANC USING ADAPTIVE FILTERS ACADEMIC YEAR: 2025-26

7. Effects of Active Noise Control in Earphones on Preferred Listening Levels (PLLs)


Hoshina, T., Fujiyama, D., Koike, T., & Ikeda, K. (2022). Journal of Audiology & Otology,
26(3), 122–129.
This human factors study shows that ANC reduces preferred listening levels in noisy
environments. Canal-type earphones with ANC maintain PLLs below potentially harmful
levels, highlighting the hearing protection benefits of ANC.

Objectives

 Develop a hardware prototype using STM32F4 Discovery board with analog signal
conditioning for a single-channel ANC system.
 Implement the LMS adaptive filter algorithm in C using ARM CMSIS-DSP library
for real-time processing.
 Configure STM32 peripherals (ADC, DMA, I2S) for non-blocking, interrupt-driven
audio data handling.
 Test system performance in attenuating stationary acoustic noise.
 Analyze results by measuring SNR improvement and showing time- and frequency-
domain performance.

Methodology
1. System Design & Component Selection
 Feedforward ANC architecture.
 STM32F407VGT6 microcontroller chosen for processing.
 LM358 amplifier modules used for analog signal conditioning.
2. Hardware Implementation
 Amplifiers connect primary (Speech + Noise) and reference (Noise Only) signals to
STM32 ADC pins (PA0, PA1).
 Audio output goes through onboard CS43L22 codec to 3.5mm headphone jack.
3. Firmware Development
 Developed in STM32CubeIDE using C.
 Configured peripherals include ADC1, ADC2, I2S3, I2C1, and DMA.
 Real-time processing is done with DMA double-buffering
4. Testing & Data Acquisition
 Input is stereo WAV (speech + noise).
 Output is streamed to a PC for recording and analysis.
5. Performance Analysis & Validation
 Analyzed using MATLAB (waveforms, spectrograms).
 SNR improvement calculated to measure noise cancellation.

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ANC USING ADAPTIVE FILTERS ACADEMIC YEAR: 2025-26

Hardware Components

Core Hardware

 STM32F4 Discovery Kit (STM32F407G-DISC1)


Quantity: 1
Purpose: This is the main processor(the "brain") that will run your adaptive filter code
Amplifier Modules

 LM358 General-Purpose Amplifier Modules


Quantity: 2

Purpose: One module amplifies the Speech + Noise signal, and the second module
amplifies the Noise Only signal before sending them to the STM32. These are the correct
modules with IN pins and no onboard microphone.
Cables and Connectors

 3.5mm Stereo to Dual Mono Breakout Cable


Quantity: 1

Purpose: This is the critical cable that splits the stereo audio from your PC into two separate
signals for your two amplifier modules.

 USB Mini-B Cable


Quantity: 1
Purpose: This provides power and programming/debugging for your STM32 board.

 Headphones
Quantity: 1
Purpose: To listen to the final, noise-cancelled audio output from the STM32 board.
Prototyping Essentials

 Breadboard
Quantity: 1
Purpose: To easily connect the amplifier modules to the STM32 without soldering.

 Jumper Wires
Quantity: 1 pack
Purpose: To make all the pin-to-pin connections between the modules and the STM32 board.

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ANC USING ADAPTIVE FILTERS ACADEMIC YEAR: 2025-26

Conclusion

Active Noise Cancellation technology uses the principle of destructive interference to


effectively minimize unwanted sounds, enhancing audio clarity and user comfort. Its
applications range from headphones and vehicles to industrial and healthcare environments.
While performance may vary for sudden or irregular noises, continuous improvements in
adaptive algorithms and hardware are making ANC more efficient, reliable, and
widely accessible.

Expected Results

[Link] Noise Reduction – Reduction of constant, low-frequency background noise by


generating an inverted sound wave.
2. Improved Audio Clarity – Enhanced quality of speech, music, or important signals in noisy
environments.
3. Better User Comfort – Reduced ear strain and fatigue during prolonged exposure to noisy
surroundings.
4. Wide Application Potential – Effective performance in headphones, vehicles, offices, and
industrial setups.
5. Adaptive Performance – Ability to adjust to varying noise levels for consistent results.

References
 S. M. Kuo and D. R. Morgan, Active Noise Control Systems: Algorithms and DSP
Implementations. New York: Wiley, 1996.
 S. D. Snyder and N. Tanaka, “Active control of vibration using a neural network,”
IEEE Trans. Neural Netw., vol. 6, no. 7, pp. 819–828, Jul 1995.
 S. Doclo and M. Moonen, “On the output SNR of the speech-distortion weighted
multichannel Wiener filter,” IEEE Signal Process. Lett., vol.12, no. 12, pp. 809--811,
Dec. 2005.
 J. R. Treichler, C. R. Johnson, Jr., and M. G. Larimore, Theory and Design of
Adaptive Filters. New York: Wiley, 1987.

DEPARTMENT OF ECE, JSSATE-B PAGE NO: 5

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