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DIGITAL FILTERS Source: ANALOG DEVICES  Part 2: Infinite Impulse Response (IIR) Filters
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
Bandpass and Bandstop Filters Designed from Lowpass And Highpass Filters
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)
Hardware Implementation Of Second-order IIR Filter (Biquad) Direct Form 1
IIR Biquad Filter Direct Form 2
IIR Biquad Filter Simplified Notations
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
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
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
Comparison Between FIR and IIR Filters
Decimation Of a Sampled Signal By a Factor of M
Decimation Combined With FIR Filtering
Interpolation by a Factor Of L
Effects of Interpolation on Frequency Spectrum
Typical Interpolation Implementation Efficient DSP algorithms take advantage of: Multiplications by zero Circular Buffers Zero-Overhead Looping
Sample Rate Converters
Adaptive Filter
Digital Transmission using Adaptive Equalization
Linear Predictive Coding (LPC) Model of Speech Production
LPC Speech Companding System
All Pole Lattice Filter
Estimation of Lattice Filter Coefficients in Transmitting DSP
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 Newark Farnell

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Digital Filters Part 2

  • 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
  • 19. Digital Transmission using Adaptive Equalization
  • 20. Linear Predictive Coding (LPC) Model of Speech Production
  • 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 Newark Farnell