GUI based Audio Equalizer - Simulink Model The system must run in real time for any input audio signal. Should be able to change equalizer settings while the song is being played to demonstrate its effectiveness. 15 bands with the following center frequencies in the 20 20kHz range: 25, 40, 63, 100, 160, 250, 400, 630, 1K, 1.5K, 2.5K, 4K, 6.3K, 10K, 16K The filters should be as low order as possible, but with as linear passband phase as possible. The transition band should be relatively narrow. This would depend on the frequency band, higher frequencies can have higher transition bands; however, the slope of descent should be roughly equal for each band. Determine appropriate Q values for the filters. Provide 20dB of gain / attenuation for each channel, to be controlled by the user. You can add bells and whistles, such as preset settings for certain modes (concert hall, jazz hall, discotheque, rock music, classical music, etc.), running histogram based spectrum, adding echo, reverberation, flange, voice removal, etc. You MUST have at least one meaningful major DSP design concept. There are infinite variations you can choose (polyphase filtering, near linear phase IIR design, minimizing finite word length effects, adaptive filtering for noise that covers signal spectrum, different structural implementations, new spectral estimation algorithms, wavelet denoising, independent component analysis (ICA) etc). Simple tricks such as adding echo, removing voice, flanging, pitch shift qualify as bells and whistles, and not major DSP design concept.

The final result will look like this: