Real time digital signal processing using Matlab

Increased usage of electronic devices and the fast development of microprocessors has increased the usage of digital filters ahead of analog filters. Digital filters offer great benefits over analog filters in that they are inexpensive, they can be reprogrammed easily and they open up whole new range of possibilities when it comes to Internet of things. This thesis describes development of a program that can sample music from the computer's microphone input, filter it inside the program with user built filters and reconstruct the music to the computer's headphone output meaning that the music can be played from the speakers. All of this is to happen in real time. The program is developed for students studying at the department of ``Signals and Systems" and the program is supposed the be one of the educational tools to make sense of signals and filtering. The program works well and filters the sound with satisfying results. It is easy to create filters and filter the signal. Since it is music that is filtered constructing perfect filters with minimum ripple, minimum or linear phase is quite difficult to achieve. The program could be improved by improving the user interface, making the environment more interactive and less difficult to construct good filters. Some improvements could also be made to the implementation; as of now the program might run a bit slow on startup on slower computers.