ECE 3551 – Microcomputer Systems 1 Fall 2010 Siobhan Ireland

 Overview of Project  Generating Musical Notes  Filtering Audio Input  Improvements  Conclusion

 The initial idea for my project was to use the BF533 EZ-Kit Lite board to create an electronic vocal coach  This project could have real world application in helping people train their vocal ability  I implemented the project by splitting it into two parts: generating musical notes, and filtering audio input

 I calculated note frequencies over a range of four octaves and from each octave chose one note – corresponding with the four push buttons  The frequencies were found using the equation:

 The algorithm for generating the notes is:  Which in code translated to: y = 7*x*sinf[w0*n] if(note==1)//note C3 { n %= 367;//nmax y=7*x*sinf( *n); y0=(int)(y); //send output to left and right audio channels iChannel0LeftOut=y0; iChannel0RightOut=y0; }  I had a few difficulties getting the code working  Once working I tested various values for amplitude

 The code for generating musical notes was successful  Push button PF8 = note C3  Push button PF9 = note D4  Push button PF10 = note F5  Push button PF11 = note A6  Demonstration

 The initial purpose for this section of the code was to accept user input in the form of singing and analyze which octave the frequency lies in  I chose first to test the code using input in the form of a song from my I-pod  The filter values were chosen based on the calculated octave frequency values

 The algorithm for IIR filters is:  The filters were created using MATLAB fdatool  IIR filters were chosen over FIR filters because they have less coefficients and produce clearer, crisper filtering

 The c code for the IIR filters is given below: void low_lowpass_filter(void) { int i = 0;//reset loop (row) counter //set input equal to iChannel0LeftIn shifted by 8 //iChannel0LeftIn>>8 shifts the original audio from 24 bits to 16 bits x_input = iChannel0LeftIn<<8; for(i=0; i<13; i++)//for row 0 to row 13 { //implementation of algorithm //calculate input from d[m] = x[m]-a2d[m-2]-a1d[m-1] D_low[i][0] = x_input - (low_lowpassA[i][2]*D_low[i][2]) - (low_lowpassA[i][1]*D_low[i][1]); //calculate output from y[m] = b2d[m-2]+b1d[m-1]+b0d[m] y = (low_lowpassB[i][0]*D_low[i][0]) + (low_lowpassB[i][1]*D_low[i][1]) + (low_lowpassB[i][2]*D_low[i][2]); x_input = y;//set x_input to previous value of y D_low[i][2] = D_low[i][1];//shift values stored in array D_low[i][1] = D_low[i][0];//shift values stored in array } //shift value of y (typecast as int) by 8 bits from 32 bits to 24 bits //send output to left and right output channels iChannel0LeftOut=(((int)y)>>8); iChannel0RightOut=(((int)y)>>8); }

 Initially I applied band-pass IIR filters with frequencies corresponding to the octaves. There was no audible change to the input  Next I applied band-pass IIR filters with large frequency ranges. There was still no audible change  I then switched to low and high-pass IIR filters with original frequency values. Still no change was heard  Lastly I applied low and high-pass IIR filters with large frequency ranges. They successfully altered the audio input

 There was no point at this stage of development to add a microphone, as the filters wouldn’t work as originally planned  No button, - LED 9 - original audio  PF8 pressed - LED 4 - low range filter  PF9 pressed - LED 5 -low-mid range filter  PF10 pressed - LED 6 - high-mid range filter  PF11 pressed - LED 7 - high range filter  Demonstration

 Lab access  Equipment access  Improve filters  Add microphone  Design user feedback  Combine codes  Eventually extend number of notes played

 Overall I consider the project successful  The project utilized and built upon knowledge gained through lecture and lab  Generating musical notes was new to me and was implemented successfully  Filtering audio was adapted from lab work and while it works correctly, it needs further development to work as originally planned