FILTER Digital Khairul handono
Dedicated hardware processor Frequency Translate Digital Filter 512 Point FFT x(n) X(f) Frequency Translate Digital Filter 1024 Point FFT X(f) Dedicated hardware processor Data Select Filter Select Data Select FFT Select Data Select Data Buffers Digital Filter Data Buffers FFT Data Buffers x(n) X(f) Micro programmable signal processor hardware
Programmable Signal Controller Data Storage x(n) Programmable S P Input/ Output X(f) Distributed Programmable Data Storage Data Storage x(n) Input/ Output X(f) Data Communication Controller x(n) Input/ Output X(f) Processing element Processing element
Converting analogue signals to digital
In the process of measuring the signal, some information is lost.
Aliasing
The high frequency signal is sampled just under twice every cycle
The high frequency signal is sampled twice every cycle
Antialiasing
The Impulse respons of the reconstruction filter has a clasic : sin (x)/ x shape
Quantisation
An analogue signal which is held on the rising edge of a clock signal
Block Diagram
Spectrum
Sistematika Disain Analisis Diskrit Transformasi Z Finite Regst DSP Linier Sistem Diskrit Infinite Impulse Respons Digital Filter Finite Impulse Respons Digital Filter Multirate DSP FFT DFT Adaptive Filter Disain Digital signal
Methodology System Design Step 1 User/customer driven Develop system level Signal processing Non signal processing System level documentation Requirement specification Interface design specification System Requirements Defiition Step 2 Signal Analysis Step 3 Define input signal Types Parameter Noise sources & distribution Data rates Sisgnal Processing Design Step 4 Resource Analysis Dev SP graphs for each procss Specify primitive operation Initial partitioning Arithmetic analysis Iterative process Results in architecture approach Acceptable No Yes Step 5 Configuration Analysis Final partitioning of process Memory, Control, bandwidth Acceptable Perform resource analysis Configuration HW No Yes
Infinite Impulse Response (IIR) Disain prosedure: Menggunakan formula disain untuk analog yaitu penentuan pole dan zero pada Butterworth, Chebyshev dan Elliptic Formula transformasi bidang frekuensi Transformasi bilinier, dg pemetaan pole pada bidang-s ke pole bidang-z
LPF Digital dan Analog
HPF LPF BSF BPF
Keuntungan Digital Filter Stabil thd Panas: Perubahan temperatur pada R,C dan L tidak terjadi, karena menggunakan Adders, multipliers, dan sift registers Presisi: akurasi, stabilitas, respons frekw.dg menggunakan processor register. Mudah Penyesuaian: dapat lebih tepat dan dapat diprogram sesuai kebutuhan Kelipatan: dapat dilipatkan untuk mendapatkan rangkaian yang lebih efisien.
Kerugian Digital Filter Bandwidth terbatas: dengan hasil proses sampling dari analog ke digital (A/D converter), bandwidth signal terbatas setengah dari frekuensi sampling. Keterbatasan register: implementasi sistem waktu diskrit pada perangkat keras dengan penggunaan khusus terjadi penurunan performance, karena terbatasnya jumlah bit.
Sistem Waktu Diskrit
Fungsi Transfer orde-N Inverse Z-tranforms
Lowpass Butterworth Filters
Respons Frekuensi
Analog Lowpass Chebyshev Filter
Analog Lowpass Elliptic Filter
Transformasi Band Frekuensi Design normalized analog filter of order N Perform Freq. Band Transformation analog to analog Desired Digital Filter Digitize filter Design normalized analog filter of order N Perform Freq. Band Transformation analog to analog Desired Digital Filter Digitize filter
Transformasi Bilinier
Pemetaan Frekuensi dari transformasi bilinier
Digital Lowpass Filter Disain
Lowpass transfer function
LPF First order
Butterworth Low Pass Filter fp = 500 Hz fs = 750 Hz Ap = 0.1737 dB As = 40 dB Ap As fp fs
S-plane Pole dan Zero Z-plane Pole dan Zero Zero Pole No. Real Imaginary Real Imaginary 1 0.00000 0.00000 - 0.1564345 0.9876885 2 0.00000 0.00000 - 0.4539906 0.8910065 3 0.00000 0.00000 - 0.7071068 0.7071067 4 0.00000 0.00000 - 0.8910066 0.4539905 5 0.00000 0.00000 - 0.9876884 0.1564344 Z-plane Pole dan Zero Zero Pole No. Real Imaginary Real Imaginary 1 -1.0000 0.00000 + 0.1370099 + 0.844767 2 -1.0000 0.00000 + 0.1092149 + 0.607474 3 -1.0000 0.00000 + 0.0931414 + 0.411143 4 -1.0000 0.00000 + 0.0841441 + 0.238471 5 -1.0000 0.00000 + 0.0800774 + 0.078200
Koefisien orde 2 Numerator Denominator Stage A1 A2 B1 B2 1 2.00000 1.00000 - 0.2740197 0.7324039 2 2.00000 1.00000 - 0.2184297 0.3809528 3 2.00000 1.00000 - 0.1862828 0.1777139 4 2.00000 1.00000 - 0.1682881 0.0639486 5 2.00000 1.00000 - 0.1601547 0.0125276 IIR NORMALIZING FACTOR : C0 = 0.00125 STAGE 1 NORMALIZING FACTOR: C1 = 0.11360 STAGE 2 NORMALIZING FACTOR: C2 = 0.25799 STAGE 3 NORMALIZING FACTOR: C3 = 0.32522 STAGE 4 NORMALIZING FACTOR: C4 = 0.36908 STAGE 5 NORMALIZING FACTOR: C5 = 0.35624
Frequency Response
Buku Referensi Digital Signal Processing A System Design Approach By: David J Defatta Josepth G Lucas William S Hodkins Digital Signal Processing Principles, Algorithms & Application By: John G Proakis Dimitris G Monolokis
Correlation
Correlation is a maximum when two signals are similar in shape, and are in phase (or 'unshifted' with respect to each other).
Three different types of signal
Autocorrelation
Cross correlation to identify a signal
Convolution
If one signal is symmetric, convolution and correlation are identical
Fourier Transforms
FIR
FIR design by the window
IIR
The Z Transform
Poles and Zeroes
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