Scribd Logo
Upload

Welcome to Scribd!

  • 108 views

    Using Lms Algorithm and Matlab For Noise Cancellation in Audio Signal

    Uploaded by

    NITHISH KANNA
    This document discusses using the least mean squares (LMS) algorithm and MATLAB for noise cancellation in audio signals. It describes the LMS algorithm which uses an adaptive filter to minimize the error between a desired signal and the output signal by adjusting the filter weights. The algorithm is used to remove noise from a music recording by taking the distorted audio plus noise as one input and the noise alone as the desired signal. Over time, the LMS filter removes the noise from the input to output only the clean music. The filtering process involves convolution of the input and weights, calculating the error, and adapting the weights based on the error.

    Copyright:

    © All Rights Reserved
    Download as PPTX, PDF, TXT or read online from Scribd

    Using Lms Algorithm and Matlab For Noise Cancellation in Audio Signal

    Uploaded by

    NITHISH KANNA
    108 views9 pages
    This document discusses using the least mean squares (LMS) algorithm and MATLAB for noise cancellation in audio signals. It describes the LMS algorithm which uses an adaptive filter to minimize the error between a desired signal and the output signal by adjusting the filter weights. The algorithm is used to remove noise from a music recording by taking the distorted audio plus noise as one input and the noise alone as the desired signal. Over time, the LMS filter removes the noise from the input to output only the clean music. The filtering process involves convolution of the input and weights, calculating the error, and adapting the weights based on the error.

    Original Description:

    Noise filter

    Original Title

    LMS Noise Filter

    Copyright

    © © All Rights Reserved

    Available Formats

    PPTX, PDF, TXT or read online from Scribd

    Did you find this document useful?

    Is this content inappropriate?

    This document discusses using the least mean squares (LMS) algorithm and MATLAB for noise cancellation in audio signals. It describes the LMS algorithm which uses an adaptive filter to minimize the error between a desired signal and the output signal by adjusting the filter weights. The algorithm is used to remove noise from a music recording by taking the distorted audio plus noise as one input and the noise alone as the desired signal. Over time, the LMS filter removes the noise from the input to output only the clean music. The filtering process involves convolution of the input and weights, calculating the error, and adapting the weights based on the error.

    Copyright:

    © All Rights Reserved
    Download as PPTX, PDF, TXT or read online from Scribd
    Download as pptx, pdf, or txt
    108 views9 pages

    Using Lms Algorithm and Matlab For Noise Cancellation in Audio Signal

    Uploaded by

    NITHISH KANNA
    This document discusses using the least mean squares (LMS) algorithm and MATLAB for noise cancellation in audio signals. It describes the LMS algorithm which uses an adaptive filter to minimize the error between a desired signal and the output signal by adjusting the filter weights. The algorithm is used to remove noise from a music recording by taking the distorted audio plus noise as one input and the noise alone as the desired signal. Over time, the LMS filter removes the noise from the input to output only the clean music. The filtering process involves convolution of the input and weights, calculating the error, and adapting the weights based on the error.

    Copyright:

    © All Rights Reserved
    Download as PPTX, PDF, TXT or read online from Scribd
    Download as pptx, pdf, or txt
    of 9

    USING LMS ALGORITHM AND

    MATLAB FOR NOISE


    CANCELLATION IN AUDIO
    SIGNAL
    TEAM MEMBERS:

    K . NITHISH KANNA - 17BEC0919


    V . NISHANTH - 17BEC
    SURYA TEJA
    PROBLEM STATEMENT:

     In electronics, noise is an unwanted disturbance in an electrical


    signal. Noise generated by electronic devices varies greatly as it is
    produced by several different effects.
     In communication systems, noise is an error or undesired random
    disturbance of a useful information signal. The noise is a summation of
    unwanted or disturbing energy from natural and sometimes man-made
    sources.
    SOLUTION:

     A least mean squares (LMS) filter is an adaptive filter that adjusts its transfer function
    according to an optimizing algorithm. You provide the filter with an example of the desired
    signal together with the input signal. The filter then calculates the filter weights, or
    coefficients, that produce the least mean squares of the error between the output signal and
    the desired signal.
     This example uses an LMS filter to remove the noise in a music recording. There are two
    inputs. The first input is the distorted signal: the music recording plus the filtered noise. The
    second input is the desired signal: the unfiltered noise. The filter works to eliminate the
    difference between the output signal and the desired signal and outputs the difference,
    which, in this case, is the clean music recording. When you start the simulation, you hear both
    the noise and the music. Over time, the adaptive filter removes the noise so you hear only the
    music.

     Least mean squares (LMS) algorithms are a class of adaptive filter used to mimic a desired
    filter by finding the filter coefficients that relate to producing the least mean square of the
    error signal (difference between the desired and the actual signal).
     Algorithm

     Least mean squares (LMS) algorithms are a class of adaptive filter used to mimic a desired
    filter by finding the filter coefficients that relate to producing the least mean square of the
    error signal (difference between the desired and the actual signal).
     This example uses the least mean squares (LMS) algorithm to remove noise from an input
    signal. The LMS algorithm computes the filtered output, filter error, and filter weights given
    the distorted and desired signals.
     At the start of the tutorial, the LMS algorithm uses a batch process to filter the audio input.
    This algorithm is suitable for MATLAB, where you are likely to load in the entire signal and
    process it all at once.
     Filtering Process
     The filtering process has three phases:
    Convolution
     Convolution is the mathematical foundation of filtering. In signal processing, convolving two
    vectors or matrices is equivalent to filtering one of the inputs by the other. In this
    implementation of the LMS filter, the convolution operation is the vector dot product
    between the filter weights and a subset of the distorted input signal.
    Calculation of error
     The error is the difference between the desired signal and the output signal.
    Adaptation
     The new value of the filter weights is the old value of the filter weights plus a correction
    factor that is based on the error signal, the distorted signal, and the adaptation step size:
    MATLAB BLOCK DIAGRAM:
    TO BE COVERED IN THIS PROJECT :

     Plotting the ecg graph


     Calculating Heart Beat Rate in (beats/second)
     R-peak detection using Matlab

    You might also like