ECE140

Lab #2

Sampling and Quantization

Introduction:

In this lab you will investigate the influence of the sampling rate and the number of quantization
levels on the quality of digitized sound.
The sampling rate standard for audio CD is 44,100 samples per second so the highest frequency
that can be sampled without aliasing, 22,050 Hz, is above the audible range of frequencies. Also,
the standard of 16-bit quantization yields quantization noise below the threshold of hearing.
Thus any ill effects of the limited sampling rate and finite number of quantization levels are
largely inaudible to humans. In this lab you will use a digital recording and manipulation
program (Audacity) and Matlab on the PC audio workstations. Audacity will enable you to
record and save digital audio as a ".wav "file as well as to convert the sampling rate. Audacity
does not have much flexibility for converting the number of quantization levels so we will use a
furnished Matlab file Quantlevels.m for this purpose. Sample rate manipulation is just the "tip
of the iceberg" of the capabilities of Audacity and this lab serves as a simple introduction to
using this program.
NOTE: If you have an equivalent computer sound recording program on your home computer
you may do this lab using your own software. Audacity may be downloaded for free from
http://audacity.sourceforge.net/.

1. CD quality Hard Disk recording:
Set up a microphone and use Audacity to record a few seconds of speech at 44,100 Hz and 16
bits, which are the default settings for Audacity. The lab TA will show you how to set up the
microphone and record if you have difficulty. To obtain the highest quality recording adjust the
recording level to achieve nearly full-scale for the loudest part of the recording; avoid overloads
and distortion. You may also use any of the prerecorded sounds on the computer for the
remainder of the lab or you may even record a few seconds of sound from an audio CD.
2. Sampling rate
After you record (or otherwise obtain) a short sound file you may use Audacity to change the
sampling rate. This may be accomplished by going to the Project Rate area in the lower left
hand corner of the project window. You are shown a pop up menu in which you may choose
among fixed sample rate options or you can type in any sample rate value you want.
Make sure that you start with sound files that are in 44.1kHz - 16 bit format.
Convert your files to 22,050 kHz, 11 kHz, 5.5 kHz, 2 kHz, and 1 kHz and comment on the
audible artifacts of reducing the sampling rate.

3. Quantization levels and distortion
In this part of the lab you can use the furnished Matlab program Quantlevels.m to manipulate
the number of quantization levels of an audio file (as determined by the number of bits). Here
are the first few lines of that program.

ECE140
%

Lab #2

M-file to convert number of quantization levels of a wav file

clear
close all
Nsec = 3;
bits = 12;

% Number of seconds of the sound file to process

% Number of bits for requantization

[x,R, nbits] = wavread('salinas.wav');

% Read in wave file, use any file you like

The number of quantization levels may be adjusted by changing the value of bits. On the
following line put in the name of the wav file that you want to manipulate, salinas.wav is a
guitar recording I posted on the website. This is a good recording with which to experiment
because there are many quiet intervals between the notes. Also experiment with your
recording of yourself speaking. Reduce the number of bits by 1 or 2 bit increments all the way
down to 1-bit.
For both recorded speech and the music files - at what bit resolution do the quantization
effects become audible? Describe the audible effects in both the spoken and music files at the
different bit-resolution levels. Can you still recognize the sound at 4-bit and 2-bit (even 1-bit)
quantization?
Construct a table of the Signal to Noise Ratio (in decibels) for bit resolutions from 1 bit to 16
bits. (each additional bit is a 6 dB increase of signal to quantization noise ratio)
Keeping in mind that a perceived loudness increase of a factor of two corresponds to
approximately 9 dB increase, comment on your perception of the signal to noise ratio at the
various bit resolution levels. For example, referring to the table, how many bits of resolution
would give a quantization noise that is "twice" as loud as it is at 12 bits? Prepare sound files to
check your perception against the table predictions. Keep in mind that this is subjective.
4. Maximum compression and intelligibility of speech
In the final part of this lab you will investigate the maximum extent that you can compress a
speech sound file and maintain its intelligibility. Note that the total size (number of bytes) of a
sound file is determined by the sampling rate AND the bit resolution. Determine the tradeoff of
sampling rate and bit resolution that best maintains intelligibility, i.e., is it better to use a high
sample rate and small number of bits or a low sample rate and a greater number of bits. What is
the greatest data compression factor compared to 44,100 Hz 16-bit sound that you can
achieve?

Write-up

Your write-up should answer all questions in bold posed above and well as to furnish qualitative
descriptions of what you heard in the various tests.

ECE140

Lab #2

Listing of Quantlevels.m (available for download from the ECE140 website)

%
%

M-file to convert number of quantization levels of a wav file

MB 9/27/2011

clear
close all
Nsec = 3;
bits = 12;

% Number of seconds of the sound file to process

% Number of bits for requantization

[x,R, nbits] = wavread('salinas.wav');

you like

% Read in a wave file, use any file

% .........You do not have to change anything below this line ............

R, nbits
nbits

% Print out the sample rate R and the number of quantization levels

x = x(1:Nsec*R)*2^(nbits-1);
normalize to integer levels
soundsc(x,R)
pause(Nsec + 0.5)

% Just keep the first Nsec of the file and

% Play back the original

xq = round(x/2^(nbits-bits));

% Scale wave file up to +- 2^nbits

soundsc(xq,R)

% listen to the requantized sound

start = 1;
stop = length(x);
len = stop-start;

% Select the part of the file to plot

% If stop = length(x) the entire file is plotted

subplot(211)
plot(x(start:stop), 'k.')
hold on
plot(x(start:stop),'r')
axis([0 len -2^(nbits-1) 2^(nbits-1)])
grid on
title('Original audio')
xlabel('Sample Number')
ylabel('Level')
subplot(212)
plot(xq(start:stop), 'k.')
hold on
plot(xq(start:stop))
axis([0 len -2^(bits-1) 2^(bits-1)])
grid on
title('Re-quantized audio')
xlabel('Sample Number')
ylabel('Level')