sir, i am doing project about detecting artifacts in eeg by using ica
plz help me to complete independent component analysis code ....etc

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Audio Signals and Systems and its analysis



What is Audio?

Audio means of sound or of the reproduction of sound .
, it refers to the range of frequencies detectable by the human ear approximately 20Hz to 20kHz

Audio work involves the production, recording, manipulation and reproduction of sound waves

FFT for audio Signal Analysis

The FFT can be applied directly to finite duration energy signals to examine the spectral content of the signal.

The ....etc

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Software implementation of coherent demodulation of audio (speech) signals



Sharad Kumar
Student Id 4114356
EE 4541 Digital Signal Processing Project
FINAL REPORT

abstract

INTRODUCTION

Demodulating a signal decomposes it into a slowly varying temporal envelope and a rapidly varying cosine (or fine structure--related to the instantaneous frequency of the signal). The motivation in representing a signal by envelope and fine structure comes from our desire to understand and model ....etc

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implementation of pcm pulse code modulation on speech signals in matlab

Pulse-code modulation (PCM) is a method used to digitally represent sampled analog signals. It is the standard form of digital audio in computers, Compact Discs, digital telephony and other digital audio applications. In a PCM stream, the amplitude of the analog signal is sampled regularly at uniform intervals, and each sample is quantized to the nearest value within a range of digital steps.Linear pulse-code modulation (LPCM) is a specific type of PCM where the quantizati ....etc

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source code for enhancement of single channel speech signals using mvdr in matlab

Abstract

To start off a series of three tutorial-style dsp seminar on current single-channel source separation methods, the first talk will introduce the topic of classical methods used for speech enhancement. We will cover denoising methods such as spectral subtraction, wiener filters, and probabilistic estimators (if time permits). This will give an overview of the large range of literature on reducing background noise from speech.Given the nature of the appli ....etc

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subband coding of audio speech signal and decopression using matlab ....etc

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feature extraction from speech using mfcc matlab coding

function = mfcc( speech, fs, Tw, Ts, alpha, window, R, M, N, L )
% MFCC Mel frequency cepstral coefficient feature extraction.
%
% MFCC(S,FS,TW,TS,ALPHA,WINDOW,R,M,N,L) returns mel frequency
% cepstral coefficients (MFCCs) computed from speech signal given
% in vector S and sampled at FS (Hz). The speech signal is first
% preemphasised using a first order FIR filter with preemphasis
% coefficient ALPHA. The preemphasised speech signal is subjected
% to the s ....etc

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Abstract:

A digital audio broadcasting system that is capable of providing a large number of high quality stereophonic channels to mobile receivers in an environment with severe multipath delay and fading. Optimum combination of frequency and time diversity to guarantee robust performance in the mobile channel, with its multipath delay and frequency-selective fading effects. The system is based upon a dynamic single channel per carrier assignment of each stereo channel to many carriers. Inter symbol interference degradations caused by multipat ....etc

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Welcome to the world of intelligent speech processing!! Ordinary Automatic speech recognition systems use only audio information for speech processing. On the contrary, human speech is bimodal in nature. It consists of two main parts audio and video information. Audio information is the usual speech signal that we receive. Video signal consists of the lip movements and other visual information. The relevant data in a video signal is isolated and features are extracted from it. From a series of feature vectors, we make a set of higher set of se ....etc

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A New Bit-Allocation Scheme for Coding of Digital Audio Using Psychoacoustic Modeling

Abstract:
In this paper we present a modified adaptive bit-allocation scheme for coding of audio signals using psychoacoustic modeling. Statistical redundancy reduction (lossless coding) generally gives compression ratios up to 2. We increase this ratio using perceptual irrelevancy removal (lossy coding). In this work MPEG-1 Layer-1 standard for psychoacoustic modeling is used and compression is improved by modifying the bit-allocation ....etc

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