Speech Signal Acquisition and Processing System

Resource Overview

1) Speech signal acquisition using audio recording functions 2) Spectral analysis through FFT implementation 3) Design of high-pass/low-pass digital filters with frequency response visualization 4) Signal filtering using convolution operations 5) Comparison of pre/post-filtering waveforms and spectra 6) Original signal audio playback 7) Time-stretching algorithms for fast/slow playback 8) Echo effect implementation using delay lines 9) Pitch shifting techniques for high/low voice effects 10) GUI system design with interactive controls

Detailed Documentation

This article systematically explores key aspects of speech signal processing through the following implementation phases: First, speech signal acquisition is performed using audio recording functions (like audiorecorder in MATLAB) to accurately capture target audio signals with proper sampling rate configuration. Next, spectral analysis is conducted on the acquired signals using Fast Fourier Transform (FFT) algorithms to characterize frequency components and identify dominant spectral features. To optimize signal quality, we design high-pass and low-pass digital filters using windowing methods (e.g., Hamming window) and finite impulse response (FIR) techniques, visualizing their frequency responses through magnitude/phase plots. The designed filters are then applied to signals using convolution operations or filter functions (e.g., filter() in MATLAB) to remove unwanted frequency components or modify signal characteristics. Post-filtering analysis involves comparative evaluation of waveform morphology and spectral distributions before and after processing, using time-domain plotting and spectral overlay techniques for effectiveness assessment. Original signal playback employs audio device interfaces to output raw acoustic data, enabling perceptual validation of source material characteristics. Time-domain modification algorithms implement playback rate control through sample interpolation/decimation for slow/fast playback effects while maintaining pitch consistency or applying pitch synchronization. Echo effects are created using delay line structures with feedback loops, implementing recursive digital filters to generate customizable reverberation patterns. Pitch shifting effects utilize phase vocoders or time-domain pitch synchronous overlap-add (PSOLA) methods to alter vocal characteristics while preserving temporal structure. Finally, an intuitive GUI system integrates all functionalities using interactive controls (sliders, buttons, waveform displays) for parameter adjustment and real-time processing visualization, built with application development frameworks.