Professional MATLAB source code with comprehensive documentation and examples
Comprehensive audio signal processing techniques including time-scale modification without pitch alteration, pitch shifting while maintaining tempo, and male-to-female voice conversion. Includes practical audio examples and implementation insights us
Insights on voice enhancement algorithms, including spectral subtraction adaptive filtering, auditory masking effects, and other techniques for improving speech signal quality.
Load speech file, generate random noise, synthesize noisy speech signal, and define parameter settings. Step 2: Frame segmentation with 50% overlap. Step 3: Apply Hamming window and perform Fourier transform to obtain power spectrum and phase spectru
Implementation of Speech Recognition with TDNN (Time Delay Neural Network) - Advanced Time-Series Pattern Learning for High-Accuracy Audio Processing
Detailed explanation of the UBM model MAP algorithm process with code implementation insights for speaker recognition systems
Implementation of Gabor transform for speech recognition, originally sourced from Cambridge Laboratory's website. Includes enhanced technical details about filter bank configuration, feature extraction methods, and recommended MATLAB/Python code appr
The RLS algorithm for multi-microphone speech denoising utilizes least squares adaptive filtering techniques, including comprehensive documentation and MATLAB code implementation.
A MATLAB implementation for speech frame segmentation in speech signal processing, featuring comprehensive signal processing tools and practical code examples
MATLAB implementation for audio digital watermark embedding and extraction with algorithm explanations
Voice extraction and recognition system implementing fundamental frequency extraction and analysis from speech signals using signal processing techniques
Exploration of double threshold-based speech endpoint detection, including signal quantization techniques, LPCC algorithm implementation, and comprehensive calculation of cepstral parameters.
Highly practical DAP source code for speech recognition applications - featuring robust feature extraction and acoustic modeling capabilities. Ideal for developers seeking to implement efficient voice processing systems.
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 implementatio
Implementation program for Hilbert-Huang Spectrum calculation, where extracted spectral information can be applied to speech recognition, fault detection, and other signal processing applications. The algorithm involves Empirical Mode Decomposition (
A self-developed digital equalizer implemented in MATLAB GUIDE environment. Supports playback of WAV audio files with 8-band equalizer adjustment, displaying original waveforms, frequency spectra, and processed waveforms/spectra. The compressed packa
Analysis of Audio Signal Amplitude and FFT Magnitude Spectrum
MATLAB source code implementing speech information hiding using wavelet transform. The algorithm applies 3-level wavelet decomposition to both original speech and watermark signals, then embeds the watermark's low-frequency components into selective
MATLAB code implementation using Kalman filtering for speech signal enhancement with added colored noise. Demonstrates excellent performance in noise reduction and signal quality improvement.
Implement speech signal decomposition with various wavelets in MATLAB, extract multi-level high/low frequency coefficients, visualize coefficient waveforms, and perform signal reconstruction.
Utilizing dual-tone multi-frequency technology to identify dial tones when audio input is available, with implementation approaches for signal processing and tone recognition