Patent Review

Patent No: US 3668542

Publication date: June 6,1972                                                                                                   Inventors: Daniel W. Stoffer C.      

Title : Audio Compression Circuit Using OpAmp and JFET

Summary :

         Audio Compression reduces the dynamic range of the any recording by bringing down the level of the loudest parts, meaning the loud and quiet parts are now closer together in volume and the natural volume variations are less obvious. This generally helps in mixing of musical tracks without a significant fight. The audio compression circuit mentioned in this patent promises a good frequency response and fast attack. The mentioned technique makes use of a p-channel JFET connected to an Op-amp input. The FET acts as a voltage controlled amplifier in the Operational amplifier feedback loop. This invention thereby aims to provide gain control not affecting the dc bias and also free of settling time limitations which otherwise  poses to be a major problem in most audio compression circuits.

Patent Link : https://www.google.com/patents/US3668542

Paper Review

Journal : International Journal of Advanced Research in Computer and Communication Engineering

Title : Analysis and Comparison of Audio Compression Using Discrete Wavelet Transform

Volume 86 – No 13, January 2014                                         

Summary :  Audio compression has become one of the basic technologies of the multimedia age. The change in the telecommunication infrastructure, in recent years, from circuit switched to packet switched systems has also reflected on the way that speech and audio signals are carried in present systems.

       Audio Compression reduces the dynamic range of the any recording by bringing down the level of the loudest parts, meaning the loud and quiet parts are now closer together in volume and the natural volume variations are less obvious.The given sound file is read. The vector is divided into smaller frames and arranged into matrix form. DCT operation is performed on the matrix. DCT operation is performed and the elements are sorted in their matrix form to find components and their indices.

Paper Link : https://www.researchgate.net/publication/272384288_Analysis_and_Comparison_of_Audio_Compression_Using_Discrete_Wavelet_Transform

Plagiarism Check Link :  https://drive.google.com/drive/folders/0BxzVWYbaSXFrMUtVRzNZWXpLeTQ

Operations using DSP Processsor

The operations were performed on a DSP Processor(TMS320F28335). The TI Code Composer Studio(CCS) was used for programming the DSP processor.The various operations carried out were Arithmetic Operations which included Addition Subtraction Multiply Divide, Logical Operations like And and Not and Shifting Operations like Logical Shift Left, Logical Shift Right, Rotate Right, Rotate Left. 

Fast Fourier Transform

In this experiment the FFT algorithm was applied to a 4 point and an 8 point sequence. The code was modified to calculate the number of real multiplications and real additions.

Fast Fourier Transform is an algorithm used to compute Discrete Fourier Transform of a sequence. An FFT rapidly computes transformations by factorising the DFT matrix into a product of a sparse factors. As a result the complexity of the computation is reduced.

FIR filter design using frequency sampling method

The aim of the experiment was to design digital filter using frequency sampling method. The magnitude and phase spectrum were plotted for LPF and HPF. 

It was observed that the phase plot is similar. Also, if the order of LPF and HPF are same, then the phase plot for both the filters is also same. The observed and calculated values of As and Ap were verified.

FIR filter design using windowing method

The aim of the experiment was to design digital filter using windowing method and study the spectrum. The input parameters were passband attenuation(Ap), stopband attenuation(As), passband frequency(Fp), stopband frequency(Fs) and sampling frequency(F). The magnitude and phase spectrum for LPF and BPF using Hanning window were plotted. 

The observed and calculated values of Ap and As were compared. Thus, the values were verified. The phase spectrum was observed to be linear.

Overlap add and Overlap save method

The aim of the experiment was to filter long data sequence using OSM and OAM. The inputs are very large in these methods. They are used for decreasing the output delay. Also, aliasing effect is present in OSM in the initial M-1 samples in the corresponding output block. This effect is absent in OAM.

Butterworth filter

This experiment was also performed using Scilab. The input parameters were passband attenuation(Ap), stopband attenuation(As), passband frequency(Fp), stopband frequency(Fs) and sampling frequency(F).The theoretical and observed values of Ap and As were compared.

It was observed that all the poles lie in the unit circle for high pass and low pass digital filter. Thus, the filters are stable. Greater the order of filter, greater is the accuracy in the theoretical and observed values. 

Chebyshev filter

This was the first experiment performed using Scilab. Digital chebyshev filter was designed using BLT method.

The imput parameters were pass band attenuation(Ap), stop band attenuation(As), pass band frequency(Fp), stop band frequency(Fs) and sampling frequency(F). The observed and theoretical values of Ap and As were compared.

The slight difference in the values were due to variations in constant values and inaccuracies in measurement. More the order of the filter, less will be the difference between observed and theoretical Ap and As values.