Lab 4- Overlap add method and Overlap save method

Real-time signals converted into the digital sample are very large in the length which can make the processing speed much slower and complex.Therefore to make the processing speed faster the samples are taken in blocks and then output for blocks are taken one by one hence the process is called block processing technique. So in this experiment we perform two of the block processing technique
1. Overlap add method.
2. Overlap save method.

Lab 3 - Fast Fourier Transform

 Discrete time Fourier transform is computationally slow hence to increase the performance of the processors Fast Fourier Transform​ method is used gives the output with lesser number of calculations.The output was bit reversal of respective input signal.

       In fft we divided​ the signal in N/2 equal parts even and odd. The even portion of the signal is added by the odd part which is multiplied by twiddle factor.Hence computations are reduced as calculation were done for N/2 length of signal.

  The number of real addition,real multiplication, complex addition and complex multiplication was also calculated from the program using the counters.

Lab 1 - Convolution and Correlation

 Digital signal processing and processors the subject in which we first time got to know how the signals are generated and implemented in real world application.We first time got introduced to technology which is the foundation of today's digital world.
    In this experiment we performed convolution and correlation which are basic part of any digital processors which processes an analog signals.We performed this experiment using C programming.
      Convolution:In linear convolution the output signal has length one less of addition of input signals.In circular convolution the output was equal to size of maximum of input signals.
       Correlation:This method is used to find the similarities between input signals and the center value of signal gives the energy of the signals.

Lab 2 - Discrete Time Fourier Transform

Practical signals have infinite numbers of samples which is practically impossible to sample hence the Discrete Fourier Transform gives the approximate magnitude spectrum of the signal.DFT of the signals was calculated by the formula derived from original formula of DFT , because C programs  does not have the concept of imaginary numbers so the separate array for real and imaginary part was created.Length of the output was found to be same as input.Approximation error was decreased by zero padding.