Reconstruction Algorithm for Speech Signals Based on Compressed Sensing
A speech signal reconstruction algorithm based on compressed sensing, implementing the Backpropagation (BP) neural network algorithm for signal recovery and reconstruction.
Explore MATLAB source code curated for "语音信号" with clean implementations, documentation, and examples.
A speech signal reconstruction algorithm based on compressed sensing, implementing the Backpropagation (BP) neural network algorithm for signal recovery and reconstruction.
A MATLAB program utilizing wavelet transforms for speech signal denoising through threshold-based noise removal, featuring parameter optimization capabilities
Precise computation of one-third octave bands for speech or noise signals with implementation insights
Perform cepstral analysis on speech signals to generate signal waveforms and cepstrum graphs with MATLAB implementation details
Implement speech signal decomposition with various wavelets in MATLAB, extract multi-level high/low frequency coefficients, visualize coefficient waveforms, and perform signal reconstruction.
Speech Signal Processing - Implementing endpoint detection and pitch trajectory tracking for input voice signals using autocorrelation function (ACF), average magnitude difference function (AMDF), and combined ACF/AMDF methods with code implementation details.
Communications Principles Course Project: Simulation of Speech Signal Baseband Communication Transmission System Using DPCM Coding and Cyclic Code Error Correction
A MATLAB-developed software interface for speech signal processing featuring audio mixing, playback functionality, and time-frequency domain waveform visualization capabilities
MATLAB source code implementation of Baum algorithm for training speech signals in Hidden Markov Model (HMM) based speech recognition systems
Speech Signal Analysis, Processing and Design. The MATLAB-based voice changer operates by modifying input audio frequencies to alter timbre and pitch, creating perceptually distinct output sounds. Voice changers employ dual-composite modifications of both timbre and pitch characteristics. For self-generated vocal inputs, formant frequency alteration is achieved through resampling techniques, enabling diverse vocal effects.