Digital Signal Processing Logic

PLLs Phase Locked Loops

FFT Fast Fourier Transform

CVSD Continuous Variable Slope Delta

Filters

• The data rates for audio are so low compared to what an FPGA is capable of, that processing the data bit serially (ie one bit at a time) allows for pretty complex circuits for amazingly little logic at the cost of a small delay. Larger FPGAs can be used to implement true real-time digital filters. The common implementations are
  • FIR/IIR parallel digital filters
    • See "Astronomical Issues Part 3: Filters and Undersampleing" by Ingo Cyiax in Circuit Cellar #107 pg 45
    • Difference between FIR and IIR
  • Majority serial filter which is easy, low-cost, and simple but inserts a delay.
    • VHDL sample
• digital twist
• Tony Fishers digital filter code generator
• Micro controller interfacing to analog sensors (such as the LM335A) Simple running average filter.
• A Simple Digital Filter

Phase Accumulators

• http://www.g4dvj.demon.co.uk/dds.htm
• Book Advanced Digital Signal Processing by Zelniker and Taylor
• Book The Scientist and Engineer's Guide to Digital Signal Processing   by Steven W. Smith is available (free) on-line but its worth it to buy the hard copy.
• http://www-users.cs.york.ac.uk/~fisher/ (cached 20000101) Source updated from GIF to PNG ploting by Nikolai Golovchenko

Also:

• PIC Microcontroller DSP
• Speakers
• Music
• Speech
• Audio
• Massmind Newsletter 04/04 - Sine Waves

See also:

• https://arachnoid.com/BiQuadDesigner/ Designer for Bi-Quad Filters. They can act as a lowpass, highpass, bandpass, notch, peak, low shelf and high shelf filter depending on the coefficients supplied. The web interface plots the response and returns the coefficients for the filter equation given the type of filtering desired, the center frequency, q, gain, and sample rate. The filter is implemented as:

  // perform one filtering step
  public double filter(double x) {
      y = b0 * x + b1 * x1 + b2 * x2 - a1 * y1 - a2 * y2;
      x2 = x1;
      x1 = x;
      y2 = y1;
      y1 = y;
      return (y);
  }

• http://kmt.hku.nl/~pieter/SOFT/RESP/html/PoZeTools.html Command line and CGI-BIN applications written in ANSI-C that: Given a set of poles and zeros, plots the frequency response, calculates filter coefficients, plots the impulse response and prints the linear difference equation.
• http://www.csdmag.com/newframe.htm
• http://www.goldwave.com/features.php

See: