EL – 413 Digital Signal Processing

Experiment # 12

FIR Filter using DSK 6713

Performed on:

Student Name:
Roll Number: Section: E

Maximum Marks Performance = 05 Viva = 05 Total = 10

Marks Obtained
Remarks (if any)

Experiment evaluated by
Instructor Name: Engr. Muhammad Waseem Zeeshan Ashrafi

Signature:

Copyright © Department of Electrical Engineering – Usman Institute of Technology

Objective
 Effects on audio signal using three FIR Low-pass Filters.
 Two Notch Filters to Recover a Corrupted Speech Recording
 Voice Scrambling

Required Equipment
Software: Code Composer studio 3.1, 6713 DSK Diagnostic tool
Hardware: TMS320C6713 DSP starter Kit (DSK), Headphone, Aux cable

Effects on Voice or Music Using Three FIR Low pass Filters

In this part of the experiment, three FIR low pass filters with cutoff frequencies at 600,
1500, and 3000 Hz respectively and sampling frequency 11360 Hz (same as audio signal
provided in lab) are implemented. The coefficient of three low pass filters were designed using
MATLAB (FDATool). The coefficient file LP600.cof represents 81 coefficients of FIR low pass
filter with a 600 - Hz cutoff frequency, using the Kaiser Window function.

Procedure:
1. Launch CCS by double - clicking on its desktop icon.
2. Create project fir3lp.pjt
3. Add source files fir3lp.c, c6713dskinit.c, vectors-intr.asm, c6713.cmd and library files
csl6713.lib, rts6700.lib and dsk6713bsl.lib in project.
4. Connect the microphone to the LINE IN via Aux cable and headphone to the HP OUT
sockets on the DSK.
5. Build the project ( Adjust Build option as performed in previous lab)
6. Load the executable file fir3lp.out
7. Select Debug→ Run.
8. Select File →Load GEL and load the file fir3lp.gel (in folder fir3lp).
9. Select GEL→Filter Characteristics→filter. Filter can be varied through gel slider while the
program is running.
10. Vary the filter and note the effect of the three different low pass filters while playing any
song by aux cable and listening on the headphone.

Lab Task:
Implement two band stop FIR filter in series to recover a speech recording corrupted by the
addition of two sinusoidal signals at frequencies of 900 and 2700 Hz in dsk6713 kit provided in
program notch2.c and corrupted audio file.

Open Ended Task:

Implement voice scrambler in MATLAB on a given audio file speech.mp3.

Copyright © Department of Electrical Engineering – Usman Institute of Technology