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Digital Filters Part 2
Digital Filters Part 2
Digital Filters Part 2
Digital Filters Part 2
Digital Filters Part 2
Digital Filters Part 2
Digital Filters Part 2
Digital Filters Part 2
Digital Filters Part 2
Digital Filters Part 2
Digital Filters Part 2
Digital Filters Part 2
Digital Filters Part 2
Digital Filters Part 2
Digital Filters Part 2
Digital Filters Part 2
Digital Filters Part 2
Digital Filters Part 2
Digital Filters Part 2
Digital Filters Part 2
Digital Filters Part 2
Digital Filters Part 2
Digital Filters Part 2
Digital Filters Part 2
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Digital Filters Part 2

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Introduction of infinite impulse response filters and adaptive filters

Introduction of infinite impulse response filters and adaptive filters

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  • 1. DIGITAL FILTERS
    • Source: ANALOG DEVICES
    Part 2: Infinite Impulse Response (IIR) Filters
  • 2. Introduction
    • Purpose
      • This module introduce the technology of Infinite Impulse Response (IIR) Filters, Multirate Filters and Adaptive Filters.
    • Outline
      • Introduce the technology of Infinite Impulse Response (IIR) Filters
      • Introduce the technology of Multirate Filters.
      • Introduce the technology of Adaptive Filters.
    • Contents
      • 23 pages
    • Duration
      • 15 Minutes
  • 3. Bandpass and Bandstop Filters Designed from Lowpass And Highpass Filters
  • 4. Infinite Impulse Response (IIR) Filters
    • Uses Feedback (Recursion)
    • Impulse Response has an Infinite Duration
    • Potentially Unstable
    • Non-Linear Phase
    • More Efficient than FIR Filters
    • No Computational Advantage when Decimating Output
    • Usually Designed to Duplicate Analog Filter Response
    • Usually Implemented as Cascaded Second-Order Sections (Biquads)
  • 5. Hardware Implementation Of Second-order IIR Filter (Biquad) Direct Form 1
  • 6. IIR Biquad Filter Direct Form 2
  • 7. IIR Biquad Filter Simplified Notations
  • 8. Review Of Popular Analog Filters
    • Butterworth
      • All Pole, No Ripples in Passband or Stopband
      • Maximally Flat Response (Fastest Roll-off with No Ripple)
    • Chebyshev (Type 1)
      • All Pole, Ripple in Passband, No Ripple in Stopband
      • Shorter Transition Region than Butterworth for Given Number of Poles
      • Type 2 has Ripple in Stopband, No Ripple in Passband
    • Elliptical (Cauer)
      • Has Poles and Zeros, Ripple in Both Passband and Stopband
      • Shorter Transition Region than Chebyshev for Given Number of Poles
      • Degraded Phase Response
    • Bessel (Thompson)
      • All Pole, No Ripples in Passband or Stopband
      • Optimized for Linear Phase and Pulse Response
      • Longest Transition Region of All for Given Number of Poles
  • 9. IIR Filter Design Techniques
    • Impulse Invarient Transformation Method
      • Start with H(s) for Analog Filter
      • Take Inverse Laplace Transform to get Impulse Response
      • Obtain z-Transform H(z) from Sampled Impulse Response
      • z-Transform Yields Filter Coefficients
      • Aliasing Effects Must be Considered
    • Bilinear Transformation Method
      • Another Method for Transforming H(s) into H(z)
      • Performance Determined by the Analog System’s Differential Equation
      • Aliasing Effects do not Occur
    • Matched z-Transform Method
      • Maps H(s) into H(z) for filters with both poles and zeros
    • CAD Methods
      • Fletcher-Powell Algorithm
      • Implements Cascaded Biquad Sections
  • 10. Throughput Considerations For IIR Filters
    • Determine How Many Biquad Sections (N) are Required to Realize the Desired Frequency Response
    • Multiply this by the number of instruction cycles per Biquad for the DSP and add overhead cycles (5N + 2 cycles for the ADSP-21xx series, for example).
    • The Result (plus overhead) is the Minimum Allowable Sampling Period (1 / fs) for Real-Time Operation
  • 11. Comparison Between FIR and IIR Filters
  • 12. Decimation Of a Sampled Signal By a Factor of M
  • 13. Decimation Combined With FIR Filtering
  • 14. Interpolation by a Factor Of L
  • 15. Effects of Interpolation on Frequency Spectrum
  • 16. Typical Interpolation Implementation
    • Efficient DSP algorithms take advantage of:
    • Multiplications by zero
    • Circular Buffers
    • Zero-Overhead Looping
  • 17. Sample Rate Converters
  • 18. Adaptive Filter
  • 19. Digital Transmission using Adaptive Equalization
  • 20. Linear Predictive Coding (LPC) Model of Speech Production
  • 21. LPC Speech Companding System
  • 22. All Pole Lattice Filter
  • 23. Estimation of Lattice Filter Coefficients in Transmitting DSP
  • 24. Additional Resource
    • For ordering the ADSP-21xx, please click the part list or
    • Call our sales hotline
    • For additional inquires contact our technical service hotline
    • For more product information go to
    • http://www.analog.com/en/embedded-processing-dsp/adsp-21xx/processors/index.html

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