UNIVERSITY OF HERTFORDSHIRE

School of Electronic, Communication and Electrical Engineering

Biometric Systems and Speech Processing

Speech Processing
Laboratory Experiments
Object: To investigate methods for the enhancement and analysis of speech signals.
Software tools to be used: Matlab-based adaptive filter and Matlab-based speech
feature extraction engine.
In this laboratory exercise, you will conduct two sets of experiments. The first set is
concerned with the reduction of noise in speech using adaptive filtering. The second
set of experiments is about the processes involved in extracting speech features. In
connection with each set of experiments you are asked to answer some questions or
comment on your findings. You will need to record these in the lab answer sheet
provided. The completed answer sheets must be submitted to the academic staff in
charge before leaving the laboratory.
1. Reduction of noise in speech
The automatic cancellation of background noise in speech is considered important in
a variety of applications. For example, the presence of noise in speech can reduce the
accuracy of an automatic speech recognition system. Background noise can also
reduce the intelligibility of speech in telephony. It is therefore desired to adopt
methods which can effectively reduce the noise content in speech signals.
A main approach to the cancellation of noise in speech is Adaptive Filtering.
The process of adaptive filtering for the reduction of ambient noise in speech is
illustrated in Figure 1. As observed, the system has two inputs of primary and
reference. The contaminated speech is applied to the primary input. A reference noise
signal captured from the environment is applied to the system reference input. The
reference noise signal does not need to be exactly the same as the noise contained in
speech, but it must be correlated with it. In other words, the two noise signals (i.e. the
one in speech and the one used as the reference) must have been produced by the
same source.
In Figure 1, S is the Clean speech signal, N is noise, S+N is the contaminated
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speech, and N is the reference noise signal which is correlated with N. The adaptive
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digital filter attempts to produce an estimate of N (noise in speech) by processing N .
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The result, N , is subtracted from the noisy speech to produce an estimate of the clean
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speech. This estimate is denoted by S in Figure 1. If N is equal to N, then the output

is minimised and S is equal to S (pure speech). However, to start with, N is not a

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good estimate of N. As shown in Figure 1, S is used to adjust the parameters of the
adaptive filter so that it produces a better estimate of N. This process continues until
the system converges and produces the best estimate of noise that it can. In this case,
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S is very close to S. However, it should be noted that in practice S can never be
exactly the same as S. If, after system convergence, the noise characteristics change,
the system will start adapting again until it produces the best estimate of the clean
speech.

S + N: Contaminated Speech

Output: S

Primary Input

Adaptive Digital Filter

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N

Reference

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N : Reference Noise

Figure 1. Adaptive reduction of noise in speech

1.1. Noise reduction experiment
Launch Matlab, and then change its active directory to the folder containing the
relevant software tool for noise reduction experiments. You will be advised about the
name and location of this folder.
Type "start" (case sensitive) in the command window.
This will bring up a screen displaying the clean speech signal to be used for the
experiments.
Click on the Listen button to play back the clean speech signal.
In the box marked Input SNR choose 10 dB, and then click on the Next button.
This will reduce the signal-to-noise ratio of speech to 10 dB by contaminating it with
additive noise. By clicking on the Listen button, playback the resultant signal and
confirm that it has been contaminated with noise.
Now, Click on the Next button. This will apply the contaminated speech signal to
the primary input of an adaptive noise cancellation system as shown in Figure 1. A
reference noise will also be applied to the reference input of the adaptive filter.
You will now observe the speech signal enhanced as a result of adaptive filtering on
the screen together with its SNR value. Play back this signal and make a note of its
SNR value.
Now, provide answers to the following questions in the corresponding boxes in the
answer sheet.

i. Briefly describe the relationship between the noise signal used for
contaminating speech, and the noise signal applied to the reference input of the
adaptive system.
ii. Determine the factor by which the noise energy in the contaminated speech
has been reduced through adaptive filtering.
iii. Based on listening to the noisy speech applied to the adaptive system and the
speech at the output of it, briefly describe how this adaptive system operates.
1.2. Speech Features
Change the active directory of Matlab to the folder containing the relevant software
tool for speech feature extraction experiments. You will be advised about the name
and location of this folder.
Type "FrontEndprocessing (case sensitive) in the command window.
This will bring up the main screen for the front-end experiments. Read the guide notes
on the left side of the screen.
In the middle of the screen, you observe a window displaying the speech signal used
for the experiments.
By clicking on the next button, you can go through various stages of speech feature
extraction. This will involve applying the relevant processes to the successive
segments of the speech signal. You should refer to your lecture notes for more details
about the operations involved for this purpose. Two types of speech features are
considered in these experiments. These are linear predictive coding (LPC) parameters
and LPC-based cepstrum (LPCC).
Based on your observation of the experimental processes and results, provide answers
to the following questions in the corresponding boxes in the answer sheet.
iv. Comment on the differences between the speech spectra obtained using
parametric features and directly from the speech data.
v. With reference to the spectra and covariance matrices obtained for LPC and
LPCC parameters, compare and contrast these two types of speech features.

Aladdin Ariyaeeinia & Amit Malegaonkar