Sharpening of the multistage modified comb filters

This paper describes the application of filter sharpening method to the modified comb filter (MCF) in the case of decimation factor, which is product of two or more positive integers. It is shown that in the case of multistage decimation with MCF, filters in each stage are also MCF. Applying the sharpening to the decimation filter in the last stage provides very good results, with savings in the number of operations comparing to the case of sharpening of the complete filter. Direct-form FIR polyphase filter structure is proposed for the filters in each stage

Design of Multistage Decimation Filters Using Cyclotomic Polynomials: Optimization and Design Issues

This paper focuses on the design of multiplier-less decimation filters suitable for oversampled digital signals. The aim is twofold. On one hand, it proposes an optimization framework for the design of constituent decimation filters in a general multistage decimation architecture. The basic building blocks embedded in the proposed filters belong, for a simple reason, to the class of cyclotomic polynomials (CPs): the first 104 CPs have a z-transfer function whose coefficients are simply {-1,0,+1}. On the other hand, the paper provides a bunch of useful techniques, most of which stemming from some key properties of CPs, for designing the proposed filters in a variety of architectures. Both recursive and non-recursive architectures are discussed by focusing on a specific decimation filter obtained as a result of the optimization algorithm. Design guidelines are provided with the aim to simplify the design of the constituent decimation filters in the multistage chain.Comment: Submitted to CAS-I, July 07; 11 pages, 5 figures, 3 table

Third order CMOS decimator design for sigma delta modulators

A third order Cascaded Integrated Comb (CIC) filter has been designed in 0.5μm n-well CMOS process to interface with a second order oversampling sigma-delta ADC modulator. The modulator was designed earlier in 0.5μm technology. The CIC filter is designed to operate with 0 to 5V supply voltages. The modulator is operated with ±2.5V supply voltage and a fixed oversampling ratio of 64. The CIC filter designed includes integrator, differentiator blocks and a dedicated clock divider circuit, which divides the input clock by 64. The CIC filter is designed to work with an ADC that operates at a maximum oversampling clock frequency of up to 25 MHz and with baseband signal bandwidth of up to 800 kHz. The design and performance of the CIC filter fabricated has been discussed

Programmable CMOS Analog-to-Digital Converter Design and Testability

In this work, a programmable second order oversampling CMOS delta-sigma analog-to-digital converter (ADC) design in 0.5µm n-well CMOS processes is presented for integration in sensor nodes for wireless sensor networks. The digital cascaded integrator comb (CIC) decimation filter is designed to operate at three different oversampling ratios of 16, 32 and 64 to give three different resolutions of 9, 12 and 14 bits, respectively which impact the power consumption of the sensor nodes. Since the major part of power consumed in the CIC decimator is by the integrators, an alternate design is introduced by inserting coder circuits and reusing the same integrators for different resolutions and oversampling ratios to reduce power consumption. The measured peak signal-to-noise ratio (SNR) for the designed second order delta-sigma modulator is 75.6dB at an oversampling ratio of 64, 62.3dB at an oversampling ratio of 32 and 45.3dB at an oversampling ratio of 16. The implementation of a built-in current sensor (BICS) which takes into account the increased background current of defect-free circuits and the effects of process variation on ΔIDDQ testing of CMOS data converters is also presented. The BICS uses frequency as the output for fault detection in CUT. A fault is detected when the output frequency deviates more than ±10% from the reference frequency. The output frequencies of the BICS for various model parameters are simulated to check for the effect of process variation on the frequency deviation. A design for on-chip testability of CMOS ADC by linear ramp histogram technique using synchronous counter as register in code detection unit (CDU) is also presented. A brief overview of the histogram technique, the formulae used to calculate the ADC parameters, the design implemented in 0.5µm n-well CMOS process, the results and effectiveness of the design are described. Registers in this design are replaced by 6T-SRAM cells and a hardware optimized on-chip testability of CMOS ADC by linear ramp histogram technique using 6T-SRAM as register in CDU is presented. The on-chip linear ramp histogram technique can be seamlessly combined with ΔIDDQ technique for improved testability, increased fault coverage and reliable operation

Design and VLSI implementation of a decimation filter for hearing Aid applications

Approximately 10% of the world’s population suffers from some type of hearing loss, yet only small percentage of this statistic use the hearing aid. The stigma associated with wearing a hearing aid, customer dissatisfaction with hearing aid performance, the cost and the battery life. Through the use of digital signal processing the digital hearing aid now offers what the analog hearing aid cannot offer. Currently lot of attention is being given to low power VLSI design. More and more people around the world suffer from hearing losses. The increasing average age and the growing population are the main reasons for this. The decimation filter used for hearing aid applications is designed and implemented both in MATLAB and VHDL. The decimation filter is designed using the distributed arithmetic multiplier in VHDL. Each digital filter structure is simulated using Matlab and its complete architecture is captured using Simulink. The resulting architecture is hardware efficient and consumes less power compared to conventional decimation filters. Compared to the comb-FIR-FIR architecture, the designed decimation filter architecture using Comb-half band FIR-FIR contributes to a hardware saving and reduces the power dissipation

Design exploration and performance strategies towards power-efficient FPGA-based achitectures for sound source localization

Many applications rely on MEMS microphone arrays for locating sound sources prior to their execution. Those applications not only are executed under real-time constraints but also are often embedded on low-power devices. These environments become challenging when increasing the number of microphones or requiring dynamic responses. Field-Programmable Gate Arrays (FPGAs) are usually chosen due to their flexibility and computational power. This work intends to guide the design of reconfigurable acoustic beamforming architectures, which are not only able to accurately determine the sound Direction-Of-Arrival (DoA) but also capable to satisfy the most demanding applications in terms of power efficiency. Design considerations of the required operations performing the sound location are discussed and analysed in order to facilitate the elaboration of reconfigurable acoustic beamforming architectures. Performance strategies are proposed and evaluated based on the characteristics of the presented architecture. This power-efficient architecture is compared to a different architecture prioritizing performance in order to reveal the unavoidable design trade-offs

Design Considerations When Accelerating an FPGA-Based Digital Microphone Array for Sound-Source Localization

The use of microphone arrays for sound-source localization is a well-researched topic. The response of such sensor arrays is dependent on the quantity of microphones operating on the array. A higher number of microphones, however, increase the computational demand, making real-time response challenging. In this paper, we present a Filter-and-Sum based architecture and several acceleration techniques to provide accurate sound-source localization in real-time. Experiments demonstrate how an accurate sound-source localization is obtained in a couple of milliseconds, independently of the number of microphones. Finally, we also propose different strategies to further accelerate the sound-source localization while offering increased angular resolution