Digital Audio Equalizer

Overview

Implement real-time audio processing demonstrating DSP skills directly relevant to RF employers. This project applies digital signal processing theory to a practical audio application using the Blackboard's audio capabilities.

The equalizer will use FIR filters to implement multiple frequency bands with real-time coefficient adjustment.

Requirements

• Audio input from Blackboard microphone (PDM) or line in
• Implement 3–5 band parametric equalizer using FIR filters
• Real-time coefficient adjustment via switches or potentiometer
• Audio output via PWM or I2S DAC
• Visual spectrum display on HDMI (optional stretch goal)

Skills Demonstrated

• FIR Filter Implementation: Multiply-accumulate in FPGA fabric
• Fixed-Point Arithmetic: Q-format number handling and scaling
• Real-Time Constraints: Meeting audio sample rate requirements
• DSP Resource Optimization: Efficient use of DSP48 primitives

Prerequisites

This project requires understanding of:

• Sampling theory and Nyquist frequency
• FIR filter design principles
• Fixed-point arithmetic and Q-format numbers
• Xilinx DSP48 primitive usage

Study Resources

• The Scientist and Engineer's Guide to DSP by Steven W. Smith (free online)
• Understanding Digital Signal Processing by Richard Lyons

Architecture

Architecture will be documented as the project progresses.

Implementation Notes

Implementation notes will be added during development.

Resource Utilization

To be measured after implementation.

Progress Log