Implementation of Finite Impulse Response Digital filter in Digital Signal Processor Kit for voice signal application

Author(s):  
Endang Djuana ◽  
Suhartati Agoes ◽  
R. Deiny Mardian ◽  
Revi Noviananda Nurmalasari
2018 ◽  
Vol 2 (2) ◽  
pp. 85-89
Author(s):  
Gunawan Ariyanto ◽  
N Nurgiyatna ◽  
Endah Sudarmilah

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.


2014 ◽  
Vol 989-994 ◽  
pp. 4195-4199
Author(s):  
Kui Zhang ◽  
Gong Liu Yang ◽  
Wei Zhen Zheng ◽  
Ren Dong Ma

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.


2016 ◽  
Vol 3 (1) ◽  
Author(s):  
Marisa Premitasari ◽  
Hendi Handian

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.


Electronics ◽  
2018 ◽  
Vol 7 (12) ◽  
pp. 372 ◽  
Author(s):  
Dmitry Kaplun ◽  
Denis Butusov ◽  
Valerii Ostrovskii ◽  
Alexander Veligosha ◽  
Vyacheslav Gulvanskii

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.


1984 ◽  
Vol 21 (1) ◽  
pp. 47-61 ◽  
Author(s):  
Trevor J. Terrell ◽  
Robert J. Simpson

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.


2010 ◽  
Vol 56 (3) ◽  
pp. 263-266
Author(s):  
Andrzej Miękina ◽  
Andrzej Podgórski

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.


2007 ◽  
Vol 17 (2) ◽  
pp. 470-473 ◽  
Author(s):  
Igor I. Soloviev ◽  
M. Raihan Rafique ◽  
Henrik Engseth ◽  
Anna Kidiyarova-Shevchenko

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