Data acquisition techniques based on frequency-encoding applied to capacitive MEMS microphones

Mención Internacional en el título de doctorThis thesis focuses on the development of capacitive sensor readout circuits and data converters based on frequency-encoding. This research has been motivated by the needs of consumer electronics industry, which constantly demands more compact readout circuit for MEMS microphones and other sensors. Nowadays, data acquisition is mainly based on encoding signals in voltage or current domains, which is becoming more challenging in modern deep submicron CMOS technologies. Frequency-encoding is an emerging signal processing technique based on encoding signals in the frequency domain. The key advantage of this approach is that systems can be implemented using mostly-digital circuitry, which benefits from CMOS technology scaling. Frequencyencoding can be used to build phase referenced integrators, which can replace classical integrators (such as switched-capacitor based integrators) in the implementation of efficient analog-to-digital converters and sensor interfaces. The core of the phase referenced integrators studied in this thesis consists of the combination of different oscillator topologies with counters and highly-digital circuitry. This work addresses two related problems: the development of capacitive MEMS sensor readout circuits based on frequency-encoding, and the design and implementation of compact oscillator-based data converters for audio applications. In the first problem, the target is the integration of the MEMS sensor into an oscillator circuit, making the oscillation frequency dependent on the sensor capacitance. This way, the sound can be digitized by measuring the oscillation frequency, using digital circuitry. However, a MEMS microphone is a complex structure on which several parasitic effects can influence the operation of the oscillator. This work presents a feasibility analysis of the integration of a MEMS microphone into different oscillator topologies. The conclusion of this study is that the parasitics of the MEMS limit the performance of the microphone, making it inefficient. In contrast, replacing conventional ADCs with frequency-encoding based ADCs has proven a very efficient solution, which motivates the next problem. In the second problem, the focus is on the development of high-order oscillator-based Sigma-Delta modulators. Firstly, the equivalence between classical integrators and phase referenced integrators has been studied, followed by an overview of state-of-art oscillator-based converters. Then, a procedure to replace classical integrators by phase referenced integrators is presented, including a design example of a second-order oscillator based Sigma-Delta modulator. Subsequently, the main circuit impairments that limit the performance of this kind of implementations, such as phase noise, jitter or metastability, are described. This thesis also presents a methodology to evaluate the impact of phase noise and distortion in oscillator-based systems. The proposed method is based on periodic steady-state analysis, which allows the rapid estimation of the system dynamic range without resorting to transient simulations. In addition, a novel technique to analyze the impact of clock jitter in Sigma-Delta modulators is described. Two integrated circuits have been implemented in 0.13 μm CMOS technology to demonstrate the feasibility of high-order oscillator-based Sigma-Delta modulators. Both chips have been designed to feature secondorder noise shaping using only oscillators and digital circuitry. The first testchip shows a malfunction in the digital circuitry due to the complexity of the multi-bit counters. The second chip, implemented using single-bit counters for simplicity, shows second-order noise shaping and reaches 103 dB-A of dynamic range in the audio bandwidth, occupying only 0.04 mm2.Esta tesis se centra en el desarrollo de conversores de datos e interfaces para sensores capacitivos basados en codificación en frecuencia. Esta investigación está motivada por las necesidades de la industria, que constantemente demanda reducir el tamaño de este tipo de circuitos. Hoy en día, la adquisición de datos está basada principalmente en la codificación de señales en tensión o en corriente. Sin embargo, la implementación de este tipo de soluciones en tecnologías CMOS nanométricas presenta varias dificultades. La codificación de frecuencia es una técnica emergente en el procesado de señales basada en codificar señales en el dominio de la frecuencia. La principal ventaja de esta alternativa es que los sistemas pueden implementarse usando circuitos mayoritariamente digitales, los cuales se benefician de los avances de la tecnología CMOS. La codificación en frecuencia puede emplearse para construir integradores referidos a la fase, que pueden reemplazar a los integradores clásicos (como los basados en capacidades conmutadas) en la implementación de conversores analógico-digital e interfaces de sensores. Los integradores referidos a la fase estudiados en esta tesis consisten en la combinación de diferentes topologías de osciladores con contadores y circuitos principalmente digitales. Este trabajo aborda dos cuestiones relacionadas: el desarrollo de circuitos de lectura para sensores MEMS capacitivos basados en codificación temporal, y el diseño e implementación de conversores de datos compactos para aplicaciones de audio basados en osciladores. En el primer caso, el objetivo es la integración de un sensor MEMS en un oscilador, haciendo que la frecuencia de oscilación depe capacidad del sensor. De esta forma, el sonido puede ser digitalizado midiendo la frecuencia de oscilación, lo cual puede realizarse usando circuitos en su mayor parte digitales. Sin embargo, un micrófono MEMS es una estructura compleja en la que múltiples efectos parasíticos pueden alterar el correcto funcionamiento del oscilador. Este trabajo presenta un análisis de la viabilidad de integrar un micrófono MEMS en diferentes topologías de oscilador. La conclusión de este estudio es que los parasíticos del MEMS limitan el rendimiento del micrófono, causando que esta solución no sea eficiente. En cambio, la implementación de conversores analógico-digitales basados en codificación en frecuencia ha demostrado ser una alternativa muy eficiente, lo cual motiva el estudio del siguiente problema. La segunda cuestión está centrada en el desarrollo de moduladores Sigma-Delta de alto orden basados en osciladores. En primer lugar se ha estudiado la equivalencia entre los integradores clásicos y los integradores referidos a la fase, seguido de una descripción de los conversores basados en osciladores publicados en los últimos años. A continuación se presenta un procedimiento para reemplazar integradores clásicos por integradores referidos a la fase, incluyendo un ejemplo de diseño de un modulador Sigma-Delta de segundo orden basado en osciladores. Posteriormente se describen los principales problemas que limitan el rendimiento de este tipo de sistemas, como el ruido de fase, el jitter o la metaestabilidad. Esta tesis también presenta un nuevo método para evaluar el impacto del ruido de fase y de la distorsión en sistemas basados en osciladores. El método propuesto está basado en simulaciones PSS, las cuales permiten la rápida estimación del rango dinámico del sistema sin necesidad de recurrir a simulaciones temporales. Además, este trabajo describe una nueva técnica para analizar el impacto del jitter de reloj en moduladores Sigma-Delta. En esta tesis se han implementado dos circuitos integrados en tecnología CMOS de 0.13 μm, con el fin de demostrar la viabilidad de los moduladores Sigma-Delta de alto orden basados en osciladores. Ambos chips han sido diseñados para producir conformación espectral de ruido de segundo orden, usando únicamente osciladores y circuitos mayoritariamente digitales. El primer chip ha mostrado un error en el funcionamiento de los circuitos digitales debido a la complejidad de las estructuras multi-bit utilizadas. El segundo chip, implementado usando contadores de un solo bit con el fin de simplificar el sistema, consigue conformación espectral de ruido de segundo orden y alcanza 103 dB-A de rango dinámico en el ancho de banda del audio, ocupando solo 0.04 mm2.Programa Oficial de Doctorado en Ingeniería Eléctrica, Electrónica y AutomáticaPresidente: Georges G.E. Gielen.- Secretario: José Manuel de la Rosa.- Vocal: Ana Rus

Linearization of Time-encoded ADCs Architectures for Smart MEMS Sensors in Low Power CMOS Technology

Mención Internacional en el título de doctorIn the last few years, the development of mobile technologies and machine learning applications has increased the demand of MEMS-based digital microphones. Mobile devices have several microphones enabling noise canceling, acoustic beamforming and speech recognition. With the development of machine learning applications the interest to integrate sensors with neural networks has increased. This has driven the interest to develop digital microphones in nanometer CMOS nodes where the microphone analog-front end and digital processing, potentially including neural networks, is integrated on the same chip. Traditionally, analog-to-digital converters (ADCs) in digital microphones have been implemented using high order Sigma-Delta modulators. The most common technique to implement these high order Sigma-Selta modulators is switchedcapacitor CMOS circuits. Recently, to reduce power consumption and make them more suitable for tasks that require always-on operation, such as keyword recognition, switched-capacitor circuits have been improved using inverter-based operational amplifier integrators. Alternatively, switched-capacitor based Sigma- Delta modulators have been replaced by continuous time Sigma-Delta converters. Nevertheless, in both implementations the input signal is voltage encoded across the modulator, making the integration in smaller CMOS nodes more challenging due to the reduced voltage supply. An alternative technique consists on encoding the input signal on time (or frequency) instead of voltage. This is what time-encoded converters do. Lately, time-encoding converters have gained popularity as they are more suitable to nanometer CMOS nodes than Sigma-Delta converters. Among the ones that have drawn more interest we find voltage-controlled oscillator based ADCs (VCOADCs). VCO-ADCs can be implemented using CMOS inverter based ring oscillators (RO) and digital circuitry. They also show noise-shaping properties. This makes them a very interesting alternative for implementation of ADCs in nanometer CMOS nodes. Nevertheless, two main circuit impairments are present in VCO-ADCs, and both come from the oscillator non-idealities. The first of them is the oscillator phase noise, that reduces the resolution of the ADC. The second is the non-linear tuning curve of the oscillator, that results in harmonic distortion at medium to high input amplitudes. In this thesis we analyze the use of time encoding ADCs for MEMS microphones with special focus on ring oscillator based ADCs (RO-ADCs). Firstly, we study the use of a dual-slope based SAR noise shaped quantizer (SAR-NSQ) in sigma-delta loops. This quantizer adds and extra level of noise-shaping to the modulator, improving the resolution. The quantizer is explained, and equations for the noise transfer function (NTF) of a third order sigma-delta using a second order filter and the NSQ are presented. Secondly, we move our attention to the topic of RO-ADCs. We present a high dynamic range MEMS microphone 130nm CMOS chip based on an open-loop VCO-ADC. This dissertation shows the implementation of the analog front-end that includes the oscillator and the MEMS interface, with a focus on achieving low power consumption with low noise and a high dynamic range. The digital circuitry is left to be explained by the coauthor of the chip in his dissertation. The chip achieves a 80dBA peak SNDR and 108dB dynamic range with a THD of 1.5% at 128 dBSPL with a power consumption of 438μW. After that, we analyze the use of a frequency-dependent-resistor (FDR) to implement an unsampled feedback loop around the oscillator. The objective is to reduce distortion. Additionally phase noise mitigation is achieved. A first topology including an operational amplifier to increase the loop gain is analyzed. The design is silicon proven in a 130 nm CMOS chip that achieves a 84 dBA peak SNDR with an analog power consumption of 600μW. A second topology without the operational amplifier is also analyzed. Two chips are designed with this topology. The first chip in 130 nm CMOS is a full VCO-ADC including the frequencyto- digital converter (F2D). This chip achieves a peak SNDR of 76.6 dBA with a power consumption of 482μW. The second chip includes only the oscillator and is implemented in 55nm CMOS. The peak SNDR is 78.15 dBA and the analog power consumption is 153μW. To finish this thesis, two circuits that use an FDR with a ring oscillator are presented. The first is a capacity-to-digital converter (CDC). The second is a filter made with an FDR and an oscillator intended for voice activity detection tasks (VAD).En los últimos años, el desarrollo de las tecnologías móviles y las aplicaciones de machine-learning han aumentado la demanda de micrófonos digitales basados en MEMS. Los dipositivos móviles tienen varios micrófonos que permiten la cancelación de ruido, el beamforming o conformación de haces y el reconocimiento de voz. Con el desarrollo de aplicaciones de aprendizaje automático, el interés por integrar sensores con redes neuronales ha aumentado. Esto ha impulsado el interés por desarrollar micrófonos digitales en nodos CMOS nanométricos donde el front-end analógico y el procesamiento digital del micrófono, que puede incluir redes neuronales, está integrado en el mismo chip. Tradicionalmente, los convertidores analógicos-digitales (ADC) en micrófonos digitales han sido implementados utilizando moduladores Sigma-Delta de orden elevado. La técnica más común para implementar estos moduladores Sigma- Delta es el uso de circuitos CMOS de capacidades conmutadas. Recientemente, para reducir el consumo de potencia y hacerlos más adecuados para las tareas que requieren una operación continua, como el reconocimiento de palabras clave, los convertidores Sigma-Delta de capacidades conmutadas has sido mejorados con el uso de integradores implementados con amplificadores operacionales basados en inversores CMOS. Alternativamente, los Sigma-Delta de capacidades conmutadas han sido reemplazados por moduladores en tiempo continuo. No obstante, en ambas implementaciones, la señal de entrada es codificada en voltaje durante el proceso de conversión, lo que hace que la integración en nodos CMOS más pequeños sea complicada debido a la menor tensión de alimentación. Una técnica alternativa consiste en codificar la señal de entrada en tiempo (o frecuencia) en lugar de tensión. Esto es lo que hacen los convertidores de codificación temporal. Recientemente, los convertidores de codificación temporal han ganado popularidad ya que son más adecuados para nodos CMOS nanométricos que los convertidores Sigma-Delta. Entre los que más interés han despertado encontramos los ADCs basados en osciladores controlados por tensión (VCO-ADC). Los VCO-ADC se pueden implementar usando osciladores en anillo (RO) implementados con inversores CMOS y circuitos digitales. Esta familia de convertidores también tiene conformado de ruido. Esto los convierte en una alternativa muy interesante para la implementación de convertidores en nodos CMOS nanométricos. Sin embargo, dos problemas principales están presentes en este tipo de ADCs debidos ambos a las no idealidades del oscilador. El primero de los problemas es la presencia de ruido de fase en el oscilador, lo que reduce la resolución del ADC. El segundo es la curva de conversion voltaje-frecuencia no lineal del oscilador, lo que causa distorsión a amplitudes medias y altas. En esta tesis analizamos el uso de ADCs de codificación temporal para micrófonos MEMS, con especial interés en ADCS basados en osciladores de anillo (RO-ADC). En primer lugar, estudiamos el uso de un cuantificador SAR con conformado de ruido (SAR-NSQ) en moduladores Sigma-Delta. Este cuantificador agrega un orden adicional de conformado de ruido al modulador, mejorando la resolución. En este documento se explica el cuantificador y obtienen las ecuaciones para la función de transferencia de ruido (NTF) de un sigma-delta de tercer orden usando un filtro de segundo orden y el NSQ. En segundo lugar, dirigimos nuestra atención al tema de los RO-ADC. Presentamos el chip de un micrófono MEMS de alto rango dinámico en CMOS de 130 nm basado en un VCO-ADC de bucle abierto. En esta tesis se explica la implementación del front-end analógico que incluye el oscilador y la interfaz con el MEMS. Esta implementación se ha llevado a cabo con el objetivo de lograr un bajo consumo de potencia, un bajo nivel de ruido y un alto rango dinámico. La descripción del back-end digital se deja para la tesis del couator del chip. La SNDR de pico del chip es de 80dBA y el rango dinámico de 108dB con una THD de 1,5% a 128 dBSPL y un consumo de potencia de 438μW. Finalmente, se analiza el uso de una resistencia dependiente de frecuencia (FDR) para implementar un bucle de realimentación no muestreado alrededor del oscilador. El objetivo es reducir la distorsión. Además, también se logra la mitigación del ruido de fase del oscilador. Se analyza una primera topologia de realimentación incluyendo un amplificador operacional para incrementar la ganancia de bucle. Este diseño se prueba en silicio en un chip CMOS de 130nm que logra un pico de SNDR de 84 dBA con un consumo de potencia de 600μW en la parte analógica. Seguidamente, se analiza una segunda topología sin el amplificador operacional. Se fabrican y miden dos chips diseñados con esta topologia. El primero de ellos en CMOS de 130 nm es un VCO-ADC completo que incluye el convertidor de frecuencia a digital (F2D). Este chip alcanza un pico SNDR de 76,6 dBA con un consumo de potencia de 482μW. El segundo incluye solo el oscilador y está implementado en CMOS de 55nm. El pico SNDR es 78.15 dBA y el el consumo de potencia analógica es de 153μW. Para cerrar esta tesis, se presentan dos circuitos que usan la FDR con un oscilador en anillo. El primero es un convertidor de capacidad a digital (CDC). El segundo es un filtro realizado con una FDR y un oscilador, enfocado a tareas de detección de voz (VAD).Programa de Doctorado en Ingeniería Eléctrica, Electrónica y Automática por la Universidad Carlos III de MadridPresidente: Antonio Jesús Torralba Silgado.- Secretaria: María Luisa López Vallejo.- Vocal: Pieter Rombout

Interface Circuits for Microsensor Integrated Systems

ca. 200 words; this text will present the book in all promotional forms (e.g. flyers). Please describe the book in straightforward and consumer-friendly terms. [Recent advances in sensing technologies, especially those for Microsensor Integrated Systems, have led to several new commercial applications. Among these, low voltage and low power circuit architectures have gained growing attention, being suitable for portable long battery life devices. The aim is to improve the performances of actual interface circuits and systems, both in terms of voltage mode and current mode, in order to overcome the potential problems due to technology scaling and different technology integrations. Related problems, especially those concerning parasitics, lead to a severe interface design attention, especially concerning the analog front-end and novel and smart architecture must be explored and tested, both at simulation and prototype level. Moreover, the growing demand for autonomous systems gets even harder the interface design due to the need of energy-aware cost-effective circuit interfaces integrating, where possible, energy harvesting solutions. The objective of this Special Issue is to explore the potential solutions to overcome actual limitations in sensor interface circuits and systems, especially those for low voltage and low power Microsensor Integrated Systems. The present Special Issue aims to present and highlight the advances and the latest novel and emergent results on this topic, showing best practices, implementations and applications. The Guest Editors invite to submit original research contributions dealing with sensor interfacing related to this specific topic. Additionally, application oriented and review papers are encouraged.

Real-time audio spectrum analyser research, design, development and implementation using the 32 bit ARMR Cortex-M4 microcontroller

This thesis describes the design and testing of a low-cost hand-held real-time audio analyser (RTAA). This includes the design of an embedded system, the development of the firmware executed by the embedded system, and the implementation of a real-time signal processing algorithms. One of the objectives of this project was to design an alternative low-cost audio analyser to the current commercially available solutions. The device was tested with the audio standard test signal (pink noise) and was compared to the expected at-spectrum response corresponding to a balanced audio system. The design makes use of an 32-bit Reduced Instruction Set Computer (RISC) processor core (ARM Cortex-M4), namely the STM32F4 family of microcontrollers. Due to the pin compatibility of the microcontroller (designed and manufactured by STMicroelectronics), the new development board can also be upgraded with the newly released Cortex-M7 microcontroller, namely the STM32F7 family of microcontrollers. Moreover, the low-cost hardware design features 256kB Random Access Memory (RAM); on-board Micro-Electro-Mechanical System (MEMS) microphone; on-chip 12-bit Analogue-to-Digital (A/D) and Digital-to-Analogue (D/A) Converters; 3.2" Thin-Film-Transistor Liquid-Crystal Display (TFT-LCD) with a resistive touch screen sensor and SD-Card Socket. Furthermore, two additional expansion modules were designed and can extend the functionality of the designed real-time audio analyser. Firstly, an audio/video module featuring a professional 24-bit 192kHz sampling rate audio CODEC; balanced audio microphone input; unbalanced line output; three MEMS microphone inputs; headphone output; and a Video Graphics Array (VGA) controller allowing the display of the analysed audio spectrum on either a projector or monitor. The second expansion module features two external memories: 1MB Static Random Access Memory (SRAM) and 16MB Synchronous Dynamic Random Access Memory (SDRAM). While the two additional expansion modules were not completely utilised by the firmware presented in this thesis, upgrades of the real-time audio analyser firmware in future revisions will provide a higher performing and more accurate analysis of the audio spectrum. The full research and design process for the real-time audio analyser is discussed and both Problems and pitfalls with the final implemented design are highlighted and possible resolutions were investigated. The development costs (excluding labour) are given in the form of a bill of materials (BOM) with the total costs averaging around R1000. Moreover, the additional VGA controller could further decrease the overall costs with the removal of the TFT-LCD screen from the audio analyser and provided the external display was not included in the BOM

Acceleration Techniques for Sparse Recovery Based Plane-wave Decomposition of a Sound Field

Plane-wave decomposition by sparse recovery is a reliable and accurate technique for plane-wave decomposition which can be used for source localization, beamforming, etc. In this work, we introduce techniques to accelerate the plane-wave decomposition by sparse recovery. The method consists of two main algorithms which are spherical Fourier transformation (SFT) and sparse recovery. Comparing the two algorithms, the sparse recovery is the most computationally intensive. We implement the SFT on an FPGA and the sparse recovery on a multithreaded computing platform. Then the multithreaded computing platform could be fully utilized for the sparse recovery. On the other hand, implementing the SFT on an FPGA helps to flexibly integrate the microphones and improve the portability of the microphone array. For implementing the SFT on an FPGA, we develop a scalable FPGA design model that enables the quick design of the SFT architecture on FPGAs. The model considers the number of microphones, the number of SFT channels and the cost of the FPGA and provides the design of a resource optimized and cost-effective FPGA architecture as the output. Then we investigate the performance of the sparse recovery algorithm executed on various multithreaded computing platforms (i.e., chip-multiprocessor, multiprocessor, GPU, manycore). Finally, we investigate the influence of modifying the dictionary size on the computational performance and the accuracy of the sparse recovery algorithms. We introduce novel sparse-recovery techniques which use non-uniform dictionaries to improve the performance of the sparse recovery on a parallel architecture

Hardware-software design of embedded systems for intelligent sensing applications

This Thesis wants to highlight the importance of ad-hoc designed and developed embedded systems in the implementation of intelligent sensor networks. As evidence four areas of application are presented: Precision Agriculture, Bioengineering, Automotive and Structural Health Monitoring. For each field is reported one, or more, smart device design and developing, in addition to on-board elaborations, experimental validation and in field tests. In particular, it is presented the design and development of a fruit meter. In the bioengineering field, three different projects are reported, detailing the architectures implemented and the validation tests conducted. Two prototype realizations of an inner temperature measurement system in electric motors for an automotive application are then discussed. Lastly, the HW/SW design of a Smart Sensor Network is analyzed: the network features on-board data management and processing, integration in an IoT toolchain, Wireless Sensor Network developments and an AI framework for vibration-based structural assessment

A study on wireless hearing aids system configuration and simulation

Master'sMASTER OF SCIENC

Development of a sub-miniature acoustic sensor for wireless monitoring of heart rate

This thesis presents the development of a non-invasive, wireless, low-power, phonocardiographic (PCG) or heart sound sensor platform suitable for long-term monitoring of heart function. The core of this development process involves a study of the feasibility of this conceptual system and the development of a prototype mixed-signals integrated circuit (IC) to form the integral component of the proposed sensor. The feasibility study of the proposed long-term monitoring sensor is divided into two main parts. The first part of the study investigates the technological aspect of the conceptual system, via a system level design. This is to prove the technological or operational feasibility of the system, where the system can be built completely using discrete, off-the-shelf electronics components to satisfy the size, power consumption, battery life and operational requirements of the sensor platform. The second part of the study concentrates on the post-processing of the heart sounds and murmurs or PCG data recorded. This is where a number of different de-noising algorithms are studied and their relative performance compared when applied to a variety of different noisy heart sound signals that would likely be acquired using the proposed sensor in everyday life. This was done to demonstrate the functional feasibility of the proposed system, where the ambient acoustic noise in the recorded PCG data can be effectively suppressed and therefore meaningful analysis of heart function i.e. heart rate, can be performed on the data. After the feasibility of the conceptual system has been demonstrated, the final part of this thesis discusses the synthesis and testing of a 0.35 μm CMOS technology prototype mixed analog-digital integrated circuit (IC) to miniaturise part of this sensor platform outlined in the system level design, conducted in the earlier part of this thesis, to achieve the objective specifications – in terms of the size and power consumption. A new implementation of the multi-tanh triplet transconductor is introduced to construct a pair of 100 nW analogue 4th order Gm-C signal conditioning filters. Furthermore, a 7 μW digital circuit was designed to drive the analog-to-digital conversion cycle of the Linear Technology LTC1288 ADC and synchronise the ADC’s output to generate the Manchester encoded data compatible with the Holt Integrated Circuit HI-15530 Manchester Encoder/Decoder

Low noise amplifier design and noise cancellation for wireless hearing aids

Master'sMASTER OF ENGINEERIN

Impact of Automotive Glass Subwoofer Technology on Vehicle Interior Sound

An innovated design for the automotive subwoofer system is proposed where the rear glass functions as the dynamic driver of the subwoofer system. The rear glass is mechanically excited using two piezoelectric actuators located along the bottom edge. The glass is fixed along the top and is free to move along the other three sides. The actuators exert a force perpendicular to the glass surface which is proportional to the low frequency input signal taken from the audio system. A study was undertaken to evaluate and compare the acoustic performance and characteristics of the rear glass subwoofer system relative to a conventional subwoofer system. Acoustical properties including frequency response, total harmonic distortion, and loudness are characterized and compared for both subwoofer designs. A subjective evaluation was conducted to correlate with objective measurements. An evaluation procedure suitable for evaluating the glass subwoofer system performance is recommended for future implementation