DSPP BLOGS

This blog presents normal discussion about the architecture of dual mode audio reconstruction filter which can be used for AC-3 and MPEG.It utilises the real FFT (Fast Fourier Transform), the efficient MPEG windowing. The digital audio which is an integral part of any video or multimedia application substantiates the need for an efficient compression algorithm due to its considerable consumption of the overall bandwidth. The MPEG audio coding algorithm is the first international standard for the compression of audio signals. Both MPEG and AC-3 make no assumptions about the source of the input signal, and they can be applied to both speech and high fidelity audio. Link:  http://ieeexplore.ieee.org/document/628679/

The aim of this experiment was to perform operations using DSP Processor.  We learned programming using DSP Hardware. In this experiment, we took two, 4 point signal values performed addition, subtraction, multiplication and division using Processor. We also performed bitwise logical operations and shifting of signal values. After performing each of these operations we verified the register values and compared them with the values of these registers before the execution of these instructions. We observed the practical implementation of the processor and its instructions. 

The aim was FIR Filter design using Frequency Sampling Method. We designed the digital filter using frequency sampling method. The input specifications were given as For LPF / HPF filter Design : (1) Pass band Attenuation (Ap) (2) Stop band Attenuation (As ) (3) Pass band Frequency (Fp) in Hz (4) Stop band Frequency (Fs) in Hz (5) Sampling Frequency in Hz  For BPF / BSF filter Design : (1) Pass band Attenuation (Ap) (2) Stop band Attenuation (As ) (3) Pass band Frequency (Fp1, Fp2) in Hz (4) Stop band Frequency (Fs) in Hz (5) Sampling Frequency in Hz        In this experiment we observed that phase response will be same for low pass and high pass filter if the orders are kept same. We also verified the values of Ap and As.

The aim was Linear Phase FIR Filter design using window function. In this experiment we designed a digital filter using windowing technique and studied the spectrum of the filter. The input specifications were given as For LPF / HPF filter Design : (1) Pass band Attenuation (Ap) (2) Stop band Attenuation (As ) (3) Pass band Frequency (Fp) in Hz (4) Stop band Frequency (Fs) in Hz (5) Sampling Frequency in Hz  For BPF / BSF filter Design : (1) Pass band Attenuation (Ap) (2) Stop band Attenuation (As ) (3) Pass band Frequency (Fp1, Fp2) in Hz (4) Stop band Frequency (Fs) in Hz (5) Sampling Frequency in Hz    F or the phase spectrum we concluded that it is linear for FIR filter. Also the observed values of As and Ap are close to the input values

The aim of this experiment was designing analog and digital Chebyshev Filter .  In this experiment we designed a digital Chebyshev filter from analog Chebyshev filter using BLT. The input specifications were given as (1) Pass band Attenuation (Ap)  (2) Stop band Attenuation (As ) (3) Pass band Frequency (Fp) in Hz (4) Stop band Frequency (Fs) in Hz (5) Sampling Frequency in Hz In the end we concluded that in both low pass and high pass filters, poles are inside the unit circle and hence they are stable. For low pass filter there is a definite zero at z=-1 while for high pass filter there is a definite zero at z=1. The values of Ap and As as input are approximately same.

The aim of this experiment was designing analog and digital Butterworth filter. We designed a digital filter from analog filter and studied the aliasing effect due to sampling in Impulse Invariant Method and the frequency warping effect in BLT Method. The Input Specifications were given as (1) Pass band Attenuation  (2) Stop band Attenuation  (3) Pass band Frequency (4) Stop band Frequency  (5) Sampling Frequency        We arrived at a conclusion that for both low pass and high pass filter, poles lie inside the unit circle. Hence both the filters are stable. But we also observed in the result that the values of Ap and As are not approximately same. Hence for better stability the order of the filter needs to be increased.