scholarly journals Optimization of the FIR Filter Structure in Finite Residue Field Algebra

Electronics ◽  
2018 ◽  
Vol 7 (12) ◽  
pp. 372 ◽  
Author(s):  
Dmitry Kaplun ◽  
Denis Butusov ◽  
Valerii Ostrovskii ◽  
Alexander Veligosha ◽  
Vyacheslav Gulvanskii
Keyword(s):  
Impulse Response ◽  
Digital Filter ◽  
Digital Signal Processor ◽  
Residue Field ◽  
Digital Signal ◽  
Performance Estimation ◽  
Field Algebra ◽  
Recursive Filtering ◽  
Residue Ring ◽  
Hardware Costs

This paper introduces a method for optimizing non-recursive filtering algorithms. A mathematical model of a non-recursive digital filter is proposed and a performance estimation is given. A method for optimizing the structural implementation of the modular digital filter is described. The essence of the optimization is that by using the property of the residue ring and the properties of the symmetric impulse response of the filter, it is possible to obtain a filter having almost a half the length of the impulse response compared to the traditional modular filter. A difference equation is given by calculating the output sample of modules p1 … pn in the modified modular digital filter. The performance of the modular filters was compared with the performance of positional non-recursive filters implemented on a digital signal processor. An example of the estimation of the hardware costs is shown to be required for implementing a modular digital filter with a modified structure. This paper substantiates the expediency of applying the natural redundancy of finite field algebra codes on the example of the possibility to reduce hardware costs by a factor of two. It is demonstrated that the accuracy of data processing in the modular digital filter is higher than the accuracy achieved with the implementation of filters on digital processors. The accuracy advantage of the proposed approach is shown experimentally by the construction of the frequency response of the non-recursive low-pass filters.

scholarly journals A Method of Increasing Digital Filter Performance Based on Truncated Multiply-Accumulate Units

2020 ◽  
Vol 10 (24) ◽  
pp. 9052
Author(s):  
Pavel Lyakhov ◽  
Maria Valueva ◽  
Georgii Valuev ◽  
Nikolai Nagornov
Keyword(s):  
Signal Processing ◽  
Theoretical Analysis ◽  
Digital Signal Processing ◽  
Digital Filter ◽  
Digital Filters ◽  
Digital Signal ◽  
Digital Filtering ◽  
Filter Performance ◽  
Filter Architecture ◽  
Hardware Costs

This paper proposes new digital filter architecture based on a modified multiply-accumulate (MAC) unit architecture called truncated MAC (TMAC), with the aim of increasing the performance of digital filtering. This paper provides a theoretical analysis of the proposed TMAC units and their hardware simulation. Theoretical analysis demonstrated that replacing conventional MAC units with modified TMAC units, as the basis for the implementation of digital filters, can theoretically reduce the filtering time by 29.86%. Hardware simulation showed that TMAC units increased the performance of digital filters by up to 10.89% compared to digital filters using conventional MAC units, but were associated with increased hardware costs. The results of this research can be used in the theory of digital signal processing to solve practical problems such as noise reduction, amplification and suppression of the frequency spectrum, interpolation, decimation, equalization and many others.

Digital Signal Processing Approach of Dithered RLG Based on IFIR Digital Filter

2014 ◽  
Vol 989-994 ◽  
pp. 4195-4199
Author(s):  
Kui Zhang ◽  
Gong Liu Yang ◽  
Wei Zhen Zheng ◽  
Ren Dong Ma
Keyword(s):  
Impulse Response ◽  
Digital Filter ◽  
Group Delay ◽  
Finite Impulse Response ◽  
Digital Signal ◽  
Noise Attenuation ◽  
Computation Complexity ◽  
Ring Laser Gyroscope ◽  
Laser Gyroscope ◽  
Filter Approach

An interpolated finite impulse response (IFIR) digital filter approach was purposed to improve the demerits of traditional finite impulse response (FIR) digital filter in the case of noise attenuation for dithered Ring Laser Gyroscope (RLG). Concentrated on the time delay and computation complexity, a comparison of FIR and IIR digital filter was illustrated. By optimizing the stretch factor L, an IFIR digital filter was designed to reach the requirements of a typical RLG. The static experiment results show that the impulse number is decreased from 250 to below 0.15, attenuation of the dither noise is nearly 100 dB and group delay remains the same level by 11.25ms.

scholarly journals EFEK REVERB TIPE LECTURE HALL DENGAN PENDEKATAN TEORI SABINE BERBASIS DIGITAL SIGNAL PROCESSOR (DSPs)

2016 ◽  
Vol 3 (1) ◽  
Author(s):  
Marisa Premitasari ◽  
Hendi Handian
Keyword(s):  
Impulse Response ◽  
Digital Signal Processor ◽  
Digital Signal ◽  
Infinite Impulse Response ◽  
Lecture Hall ◽  
Reverberation Time ◽  
Signal Processor

Salah satu jenis efek musik adalah reverb, yang merupakan hasil tiruan dari refleksi bunyi di dalam ruangan dimana sebagian bunyi akan terabsorpsi dan kemudian terjadi pemantulan bunyi yang berulang-ulang . Sesuai dengan tipe ruangannya, lecture hall didisain supaya tidak mengganggu suara dosen ketika sedang berbicara dimana sebuah pendekatan bernama teori Sabine mempunyaidelay di bawah satu detik. Teori ini akan direalisasikan menggunakan DSPs dengan input gitar elektrik .Perancangan efek dengan pendekatan Sabine disusun berdasarkan tiga subsistem yaitu Early Refelction, Butterworth dan Reverberation. Masing-masing subsistem menggunakan pemfilteran Infinite Impulse Response(IIR) dan total hasil efek reverb dilakukan pemrosesan CombFilter. Realisasi dijalankan dengan menggunakan duah buah PC dimana PC pertama untuk mengeksekusi program dan PC kedua untuk analisis hasil keluaran yang dicatat melalui dua buah professional software. CUBASE SX3 mencatat hasil dengan spesifikasi frekuensi Sabine dan ADOBE AUDITION 2.0 mencatat hasil dengan spesifikasi waktu Sabine. Hasil Eksekusi program sementara menunjukkan terjadinya error terhadap pendekatan Teori Sabine. Untuk spesifikasi frekuensi Sabine (faktor penguatan) error terbesar terletak pada frekuensi Sabine 2000 Hz sebesar 2.57 (frekuensi input 4000 Hz) dan 2.64 (frekuensi input 9600 Hz) sementara spesifikasi waktu Sabine (reverberation time) menunjukkan error 9.057 dengan frekuensi input 8000 Hz.

scholarly journals Implementasi Filter Digital Finite Impulse Response Metode Penjendelaan Blackman pada DSP TMS320C6711

2018 ◽  
Vol 2 (2) ◽  
pp. 85-89
Author(s):  
Gunawan Ariyanto ◽  
N Nurgiyatna ◽  
Endah Sudarmilah
Keyword(s):  
Real Time ◽  
Impulse Response ◽  
Digital Signal Processor ◽  
Finite Impulse Response ◽  
Digital Signal ◽  
Signal Processor

lmplementasi   tapis  digital  secara waktu nyata (real-time)   tidaklah mudah karena dibatasi oleh beberapa kendala semisal kebutuhan akan komputasi yang cepat, kesalahan quantisasi pada  ADC-DAC,  kesalahan kuantisasai koefisien filter  dan kesalahan pada  pembulatan  ari/matika.   Tulisan ini  menyajikan  hasil penelitian tentang  implementasi   sebuah filter  digital  bandpass  dengan metode  penjendelaan Blackman ke dalam perangkat keras berupa Digital Signal Processor (DSP) TMS320C671 I.   Hasil pengujian   menunjukan bahwa filter  tersebut dapat bekerja secara real-time dan memenuhi spesijikasi yang telah ditentukan.  Filter digital metode penjendelaan Blackman  tersebut memilikiperformaberupafsl=258  Hz.fp/=328 Hz, fp2=3370Hz,fs2=3440  Hz dan bandstop attenuation sebesar -31 dB.

The NEC μPD7720 Digital Signal Processor — A Laboratory-Based Case Study

1984 ◽  
Vol 21 (1) ◽  
pp. 47-61 ◽  
Author(s):  
Trevor J. Terrell ◽  
Robert J. Simpson
Keyword(s):  
Signal Processing ◽  
Digital Signal Processing ◽  
Digital Filter ◽  
Digital Signal Processor ◽  
Design Method ◽  
Digital Signal ◽  
Single Chip ◽  
Signal Processor

The concepts of single-chip digital signal processing are presented via a student-oriented tutorial/laboratory case study. This involves the design of a highpass digital filter using the first-difference design method and its implementation using the NEC μPD7720 Signal Processor.

Digital-Filter-Based Compensation of Case Effect in Sound-Level Meters

2010 ◽  
Vol 56 (3) ◽  
pp. 263-266
Author(s):  
Andrzej Miękina ◽  
Andrzej Podgórski
Keyword(s):  
Digital Filter ◽  
Digital Signal Processor ◽  
Finite Impulse Response ◽  
Digital Signal ◽  
Sound Level ◽  
Remez Algorithm ◽  
The Family ◽  
On Chip ◽  
Level Meter ◽  
Case Effect

Digital-Filter-Based Compensation of Case Effect in Sound-Level Meters The methodology for the design of a digital filter, which should compensate the effect of reflections and diffraction from the sound-level meter's casing (the so-called case effect), is presented. The coefficients of the family of the finite impulse response (FIR) filters, which were selected to fulfill the requirements of the compensation, were obtained in the MATLAB environment using the Remez algorithm. The frequency response of the selected designed filter are given. The chosen FIR filter was implemented in an on-chip Enhanced Filter Coprocessor of a fixed point 24-bit digital signal processor of a sound-level meter.

scholarly journals A Novel Dentary Bone Conduction Device Equipped with Laser Communication in DSP

Sensors ◽  
2021 ◽  
Vol 21 (12) ◽  
pp. 4229
Author(s):  
Jau-Woei Perng ◽  
Tung-Li Hsieh ◽  
Cheng-Yan Guo
Keyword(s):  
Digital Filter ◽  
Digital Signal Processor ◽  
Bone Conduction ◽  
Processing System ◽  
Digital Signal ◽  
Vibration Signal ◽  
Piezoelectric Sensor ◽  
Conduction System ◽  
Signal Processing System ◽  
Audio Output

In this study, we designed a dentary bone conduction system that transmits and receives audio by laser. The main objective of this research was to propose a complete hardware design method, including a laser audio transmitter and receiver and digital signal processor (DSP) based digital signal processing system. We also present a digital filter algorithm that can run on a DSP in real time. This experiment used the CMU ARCTIC databases’ human-voice reading audio as the standard audio. We used a piezoelectric sensor to measure the vibration signal of the bone conduction transducer (BCT) and separately calculated the signal-to-noise ratio (SNR) of the digitally filtered audio output and the unfiltered audio output using DSP. The SNR of the former was twice that of the latter, and the BCT output quality significantly improved. From the results, we can conclude that the dentary bone conduction system integrated with a DSP digital filter enhances sound quality.

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