Designing Optimal Filters for Speech Denoising

After conducting time-frequency analysis of the noise-corrupted speech signal, design an appropriate filter for denoising using digital signal processing techniques. This typically involves implementing filter design algorithms (such as FIR or IIR filters) with proper frequency response characteristics to attenuate noise components while maintaining speech integrity. Following denoising, perform time-frequency analysis of the processed signal to validate the filter's effectiveness. Subsequently, we can design a reverberation system utilizing four comb filters and two all-pass filters (as shown in the diagram below) that generates echoes and reflections. This implementation requires proper parameter tuning for delay lines and feedback coefficients, with additional equalizer integration for frequency compensation and attenuation. Furthermore, we can investigate incorporating additional filter types (such as low-pass or high-pass filters) within the reverberator structure to produce more diverse audio effects. This approach provides extensive experimental opportunities to deepen understanding of signal processing principles and their practical applications in audio engineering.