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MP3 Audio Compression Using DCT

This document discusses audio compression using discrete cosine transform (DCT). It explains that DCT expresses audio data as a sum of cosine functions at different frequencies, allowing most of the information to be retained while removing unnecessary data to reduce file size. The document outlines an approach using DCT to transform audio samples to the frequency domain, remove redundant information, then inverse DCT to reconstruct the audio. MATLAB code is provided and testing showed the DCT compression scheme works well, with potential for improvement by combining it with wavelet transforms.

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JheelRathod
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277 views

MP3 Audio Compression Using DCT

This document discusses audio compression using discrete cosine transform (DCT). It explains that DCT expresses audio data as a sum of cosine functions at different frequencies, allowing most of the information to be retained while removing unnecessary data to reduce file size. The document outlines an approach using DCT to transform audio samples to the frequency domain, remove redundant information, then inverse DCT to reconstruct the audio. MATLAB code is provided and testing showed the DCT compression scheme works well, with potential for improvement by combining it with wavelet transforms.

Uploaded by

JheelRathod
Copyright
© © All Rights Reserved
Available Formats
Download as PPTX, PDF, TXT or read online on Scribd
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MP3 AUDIO

COMPRESSION
USING DCT
Jheel Rathod 2017PH10819
Koduru Sudheer 2017PH10822
Introduction

■ Audio compression is used to solve the problem of reducing amount of data


required to represent digital audio. It follows the conventional approach of getting rid
of unnecessary duplicate data to achieve this task.
■ In our present work, we make use of Discrete cosine transform(DCT) to code and
implement an audio compressor.
■ Common audio formats like MP3, AAC and Ogg Vorbis employ Discrete Cosine
Transform (DCT) to transform the audio data to a form that lends itself well to
compression.
WHAT'S DCT AND WHY USE IT?

■ A discrete cosine transform (DCT) expresses a finite sequence of data points in


terms of a sum of cosine functions oscillating at different frequencies.
■ The DCT gives a real-valued frequency decomposition of a real-signal x(n).
■ In particular, a DCT is a Fourier-related transform similar to the discrete Fourier
transform (DFT), but using only real numbers.
■ DCT can be used for speech compression because of high
correlation in adjacent coefficients(which we are going to exploit in our upcoming
slides).
■ We can reconstruct a sequence very accurately from very few
DCT coefficients. This property of DCT helps in effective reduction of data
OUR APPROACH AND HOW WE
STARTED?
■ From previous slides, we know The DCT is like a ‘real-valued’ version of the DFT.
■ Its basic operation is to take the input audio data and transforms it from one type of
representation to another.
■ In our case, the signal is a block of audio samples.
■ We exploited DCT to transform a set of points from the spatial domain into
an identical representation in frequency domain.
■ Then, we identified pieces of information that can be effectively thrown away without
seriously reducing the audio's quality and got rid of them.
■ Finally we inversed the DCT and collected sample output.
■ The DCT most commonly employed and the one we are using is type-2 and the
inverse DCT used is type 4.
Working Principle

■ By Fourier analysis, any reasonable function f can be approximated arbitrarily well


on the interval [0, 2π] by a combination of trigonometric functions.
■ In DCT, the signal x(n) is given by a sum of cosine functions, so the DCT is a discrete-
time version of the Fourier series using only cosine.
■ The mathematical representation for one-dimensional DCT Is :
MATLAB CODE
MATLAB code for second method
Conclusion

■ To summarize, we proposed an audio compression scheme using discrete cosine


transform (DCT) and the performance test results indicated that the proposed
scheme is promising.
■ It is implemented using MATLAB.
■ As a future work the developed system can be improved by applying wavelet
transform, as first transform step, in order to decompose the audio signal, then
each one is passed through the above scheme to get better compression gain.

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