Analysis and Synthesis of Speech Using Matlab

International Journal of Advancements in Research & Technology, Volume 2, Issue 5, M ay-2013
ISSN 2278-7763
373
ANALYSIS AND SYNTHESIS OF SPEECH USING MATLAB

Vishv Mohan (State Topper Himachal Pradesh 2008, 2009, 2010)
B.E(Hons.) Electrical and Electronics Engineering; President-NCSTU
University: Birla Institute Of Technology & Science, Pilani -333031(Rajasthan)-India
E-mail: vishv.mohan.1@gmail.com
ABSTRACT
waves on a medium such as a phonograph.
The interval of each sound wave has

different frequency in its sub-sections. This
paper has made an analysis of two matlab
functions namely GenerateSpectrogram.m
and MatrixToSound.m , in order to analyze
and synthesis the speech signals. The first
Matlab
code
section
GenerateSpectrogram.m record the user
input sound for user (more precisely from
the source) defined duration and asks
required parameters for computation of
spectrogram and returns a matrix with
frequency as rows and time as column and
corresponding matrix element as amplitude
of that frequency. MatrixToSound.m uses
the method of additive synthesis of sound
to generate sound from the user defined
matrix with frequencies as its rows and time
as its columns. Sound recording is an
electrical
or
mechanical
inscription
of sound waves, such as spoken voice,
singing, instrumental music, or sound
effects. The two main classes of sound
recording
technology
are analog
recording and digital recording. Acoustic
analog recording is achieved by a
small microphone diaphragm
that
can
detect changes in atmospheric pressure
(acoustic sound waves) and record them as
a graphic representation of the sound
Digital recording converts the analog sound

signal picked up by the microphone to a digital
form by a process of digitization, allowing it to
be stored and transmitted by a wider variety of
media. Digital recording stores audio as a series
of binary numbers representing samples of
the amplitude of the audio signal at equal time
intervals, at a sample rate high enough to
convey all sounds capable of being heard. The
feature of analysis and synthesis of sound, is
applied to create the speech with the help of
matrix of elements as frequency or time domain
analyzed parameters with specific amplitude.
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Copyright 2013 SciResPub.
Keywords : spectrum, synthesis, simulation,

frequency, sound-waves, amplitude, wave
sequence.
INTRODUCTION
The speech is an acoustic signal, hence, it is
a mechanical wave that is an oscillation of
pressure transmitted through solid liquid or
gas and it is composed of frequencies
within hearing range. Sound is a sequence
of waves of pressure that propagates
through compressible media such as air or
water. Audible range of sound is 20 Hz to
20KHz, at standard temperature and
pressure. During propagation, waves can
be reflected, refracted,
or attenuated by
the medium.
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Recording of Sound
Sound recording is an electrical or
mechanical inscription of sound waves,
such as spoken voice, singing, instrumental
music, or sound effects. The two main
classes of sound recording technology
are analog recording and digital recording.
Acoustic analog recording is achieved by a
small microphone diaphragm
that
can
detect changes in atmospheric pressure
(acoustic sound waves) and record them as
a graphic representation of the sound
waves
on
a
medium
such
as
a phonograph (in which a stylus senses
grooves on a record). In magnetic
tape recording, the sound waves vibrate the
microphone diaphragm and are converted
into a varying electric current, which is then
converted to a varying magnetic field by
an electromagnet,
which
makes
a
representation of the sound as magnetized
areas on a plastic tape with a magnetic
coating on it.
374
fidelity (wider frequency

response or
dynamic range), but because the digital
format can prevent much loss of quality
found in analog recording due to noise
and electromagnetic
interference in
playback, and mechanical deterioration or
damage to the storage medium. A digital
audio signal must be reconverted to analog
form during playback before it is applied to
a loudspeaker or earphones.
Analysis of Sound Signal
The long-term frequency analysis of speech
signals yields good information about the
overall frequency spectrum of the signal, but
no information about the temporal location
of those frequencies. Since speech is a very
dynamic signal with a time-varying
spectrum, it is often insightful to look at
frequency spectra of short sections of the
speech signal.
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Digital recording converts the analog sound

signal picked up by the microphone to a
digital form by a process of digitization,
allowing it to be stored and transmitted by
a wider variety of media. Digital recording
stores audio as a series of binary numbers
representing samples of the amplitude of
the audio signal at equal time intervals, at
a sample rate high enough to convey all
sounds capable of being heard. Digital
recordings are considered higher quality
than analog recordings not necessarily
because
they
have higher
Long-term frequency analysis

The frequency response of a system is
defined as the discrete-time Fourier
transform (DTFT) of the system's impulse
response h[n]:
Similarly, for a sequence x[n], its long-term

frequency spectrum is defined as the DTFT
of the Sequence
Theoretically, we must know the sequence

x[n] for all values of n (from n=- until
n=) in order to compute its frequency
spectrum. Fortunately, all terms where x[n]
= 0 do not matter in the sum, and therefore
an equivalent expression for the sequence's
spectrum is
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Here we've assumed that the sequence

starts at 0 and is N samples long. This tells
us that we can apply the DTFT only to all of
the sequence, that is, over only part of the
non-zero samples of the sequence?
Window sequence
375
the non-zero samples of x[n], and still

obtain the sequence's true spectrum X ().
But what is the correct mathematical
expression to compute the spectrum over a
short
section
of
Then we compute the spectrum of the

windowed sequence x w [n] as usual
It turns out that the mathematically correct

way to do that is to multiply the sequence
x[n] by a window sequence w[n] that is
non-zero only for n=0 L-1, where L, the
length of the window, is smaller than the
length N of the sequence x[n]:
The following figure illustrates how a

window sequence w[n] is applied to the
sequence x[n]:
As the figure shows, the windowed

sequence is shorter in length than the
original sequence. So we can further
truncate the DTFT of the windowed
sequence:
Effect of the window

To answer that question, we need to
introduce an important property of the
Fourier transform. The diagram below
illustrates the property graphically:
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I. Implementation of an LTI system in the
time domain.
Using this windowing technique, we can

select any section of arbitrary length of the
input sequence x[n] by choosing the length
and location of the window accordingly. The
window sequence w[n] affect the shortterm frequency spectrum.
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II. Equivalent
implementation of an LTI system in
the frequency domain.
And since the time domain and the

frequency domain are each others dual in
the Fourier transform, it is also true that
multiplication in the time domain =
convolution in the frequency domain:
376
The two implementations of an LTI system

are equivalent: they will give the same
output for the same input. Hence,
convolution in the time domain =
multiplication in the frequency domain:
This shows that multiplying the sequence
x[n] with the window sequence w[n] in the
time domain is equivalent to convolving the
spectrum of the sequence X (), with the
spectrum of the window W(). The result of
the convolution of the spectra in the
frequency domain is that the spectrum of
the sequence is smeared by the spectrum
of the window. This is best illustrated by the
example in the figure below:
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a) Choice of window
Because the window determines the

spectrum of the windowed sequence to a
great extent, the choice of the window is
important. Matlab supports a number of

common windows, each with their own
strengths and weaknesses. Some common
choices of windows are shown below.
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All windows share the same characteristics.

Their spectrum has a peak, called the main
lobe, and ripples to the left and right of the
main lobe called the side lobes. The width
of the main lobe and the relative height of
the side lobes are different for each
window. The main lobe width determines
how accurate a window is able to resolve
different frequencies: wider is less accurate.
The side lobe height determines how much
spectral leakage the window has. An
important thing to realize is that we can't
have short-term frequency analysis without
a window. Even if we don't explicitly use a
window, we are implicitly using a
rectangular window.
b) Parameters
of the short-term
frequency spectrum
Besides the type of window rectangular,
hamming, etc. there are two other
factors in Matlab that control the shortCopyright 2013 SciResPub.
term frequency spectrum: window length

and the number of frequency sample
points.
The
window
length controls
the
fundamental trade-off between time
resolution and frequency resolution of the
short-term spectrum, irrespective of the
window's shape. A long window gives poor
time resolution, but good frequency
resolution. Conversely, a short window
gives good time resolution, but poor
frequency resolution. For example, a 250
millisecond long window can, roughly
speaking, resolve frequency components
when they are 4 Hz or more apart (1/0.250
= 4), but it can't tell where in those 250
millisecond those frequency components
occurred. On the other hand, a 10
millisecond window can only resolve
frequency components when they are 100
Hz or more apart (1/0.010= 100), but the
uncertainty in time about the location of
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those frequencies is only 10 millisecond.

The result of short-term spectral analysis
using a long window is referred to as a
narrowband spectrum (because a long
window has a narrow main lobe), and the
result of short-term spectral analysis using a
short window is called a wideband
spectrum. In short-term spectral analysis of
speech, the window length is often chosen
with respect to the fundamental period of
the speech signal, i.e., the duration of one

period of the fundamental frequency. A
common choice for the window length is
either less than 1 times the fundamental
period, or greater than 2-3 times the
fundamental period.
Examples of narrowband and wideband
short-term spectral analysis of speech are
given in the figures below:
The other factor controlling the short-term

spectrum in Matlab is the number of points
at which the frequency spectrum H () is
evaluated. The number of points is usually
equal to the length of the window.
Sometimes a greater number of points is
chosen to obtain a smoother looking
spectrum. Evaluating H () at fewer points
than the window length is possible, but very
rare.
c) Time-frequency
domain:
Spectrogram
An important use of short-term spectral

analysis is the short-time Fourier
transform or spectrogram of a signal.
The spectrogram of a sequence is
constructed by computing the short term
spectrum of a windowed version of the
sequence, then shifting the window over
to a new location and repeating this
process until the entire sequence has
been analyzed. The whole process is
illustrated in the figure below:
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Together, these short-term spectra (bottom

row) make up the spectrogram, and are
typically shown in a two-dimensional plot,
where the horizontal axis is time, the vertical

axis is frequency, and magnitude is the color
or intensity of the plot. For example:
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The appearance of the spectrogram is

controlled by a third parameter: window
overlap. Window overlap determines how
much the window is shifted between
repeated computations of the short term
spectrum. Common choices for window
overlap are 50% or 75% of the window
length. For example, if the window length is
200 samples and window overlap is 50%,
the window would be shifted over 100
samples
between
each
short-term
spectrum. In the case that the overlap was
75%, the window would be shifted over 50
samples. The choice of window overlap
depends on the application. When a
temporally
smooth
spectrogram
is
desirable, window overlap should be 75% or
more. When computation should be at a
minimum, no overlap or 50% overlap are
good choices. If computation is not an issue,
you could even compute a new short-term
spectrum for every sample of the sequence.
In that case, window overlap = window
length 1, and the window would only shift
379
1 sample between the spectra. But doing so

is wasteful when analyzing speech signals,
because the spectrum of speech does not
change at such a high rate. It is more
practical to compute a new spectrum every
20-50 millisecond, since that is the rate at
which the speech spectrum changes.
In a wideband spectrogram (i.e., using a
window shorter than the fundamental
period), the fundamental frequency of the
speech signal resolves in time. That means
that you can't really tell what the
fundamental frequency is by looking at the
frequency axis, but you can see energy
fluctuations at the rate of the fundamental
frequency along the time axis. In a
narrowband Spectrogram (i.e., using a
window 2-3 times the fundamental period),
the fundamental frequency resolves in
frequency, i.e., you can see it as an energy
peak
along
the
frequency
axis.
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GenerateTimeVsFreq.m
1) Duration=input('Enter the time in seconds for which you want to

record:');
2) samplingRate=input('Enter what sampling rate is required of audio 8000
or
22050: ');
3) timeResolution=input('Enter the time resolution desired in millisecond:
');
4) frequencyResolution=input('enter the frequency resolution required: ');
5) usedWindowLength
=ceil(samplingRate/frequencyResolution);
6) recObj = audiorecorder(samplingRate,8,1);
7) disp('Start speaking.')
8) recordblocking(recObj,Duration);
9) disp('End of Recording.');
10)
% Play back the recording.
11)
play(recObj);
12)
% Store data in double-precision array.
13)
myRecordingData = getaudiodata(recObj);
14)
figure(1)
15)
plot (myRecordingData);
16)
% No of Data points= samplingRate*Duration;
17)
% No of columns in spectrogram=(duration*1000)/timeResolution;
18)
% =duration*frequencyResolution;
19)
actualWindowLength= ceil((samplingRate*timeResolution)/1000);
20)
overlapLength= usedWindowLength -actualWindowLength +4;
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21)
% Plot the spectrogram
22)
S
=spectrogram(myRecordingData,usedWindowLength,overlapLength,samplingRat
e-1,samplingRate,'yaxis');
23)
[ar ac]=size(S);
24)
S1=imresize(S,[ar (Duration*1000)/timeResolution]);
25)
AbsoluteMagnitude=abs(S1);
26)
figure(2)
27)
spectrogram(myRecordingData,256,200,256,samplingRate-1,'yaxis');
28)
TimeInterval=input('Enter the time interval in terms of multiple
of time resolution to see the frequencies present at that moment:');
29)
figure(3)
30)
plot(AbsoluteMagnitude(:,timeInterval));
Synthesis of Sound
There are many methods of sound
synthesis. Jeff Pressing in "Synthesizer
Performance and Real-Time Techniques"
gives this list of approaches to sound
synthesis, namely Additive synthesis,
Subtractive
synthesis,
frequency
modulation synthesis ,sampling ,composite
synthesis ,phase distortion , wave shaping
,Re-synthesis ,granular synthesis ,linear
predictive coding ,direct digital synthesis
,wave sequencing ,vector synthesis ,physical
modeling.
Additive synthesis generates sound by

adding the output of multiple sine wave
generators. Harmonic additive synthesis is
closely related to the concept of a Fourier
series which is a way of expressing a
periodic function as the sum of sinusoidal
functions with frequencies equal to integer
multiples of a common fundamental
frequency. These sinusoids are called
harmonics, overtones, or generally, partials.
In general, a Fourier series contains an
infinite number of sinusoidal components,
with no upper limit to the frequency of the
sinusoidal functions and includes a DC
component (one with frequency of 0 Hz).
Frequencies outside of the human audible
range can be omitted in additive synthesis.
As a result only a finite number of
sinusoidal terms with frequencies that lie
within the audible range are modeled in
additive synthesis.
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We are using additive synthesis to

synthesize the sound from matrix having
rows as different frequencies and columns
as time intervals.
a) Additive Synthesis
Additive synthesis is a sound synthesis

technique that creates timbre by adding
sine waves together. In music, timbre also
known as tone color or tone quality from
psychoacoustics(i.e. scientific study of
sound perception) , is the quality of a
musical note or sound or tone that
distinguishes different types of sound
production, such as voices and musical
instruments, string instruments, wind
instruments, and percussion instruments
b) Harmonic form
The simplest harmonic additive synthesis
can be mathematically expressed as:
where ,y(t) is the synthesis output, ,

,
and
are the amplitude, frequency, and
the phase offset of the th harmonic partial
of a total of
harmonic partials, and is
the fundamental
frequency of
the
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waveform and the frequency of the musical

note.
381
a function of time, , in which case

the synthesis output is
c) Time-dependent amplitudes
More generally, the amplitude of
each harmonic can be prescribed as
d) Matlab Code
MatrixToSound.m
1)
2)
3)
4)
5)
% FUNCTION TO PLAY SOUND FROM THE MATRIX

samplingRate=
input('please enter the sampling rate used: ');
timeResolution= input('Please enter the time resolution in milliseconds: ');
matrix=
input('please enter the matrix for conversion to sound');
lowerThreshold= input('Please enter the lower threshold value below which the
matrix element should be neglected( a number between 0 and 255: ');
6) time=0:1/samplingRate:(timeResolution/1000);
7) [mrows mcolumn]= size(matrix);
8) count=0;
9) [timerow NoOfComponents]= size(time);
10)
SineVector=zeros(1,NoOfComponents);
11)
InitialSoundMatrix=zeros(NoOfComponents,mcolumn);
12)
for j=1:mcolumn
13)
for i=1:mrows
14)
if(matrix(i,j)>lowerThreshold)
15)
t=matrix(i,j)*sin(2*pi*time*i);
16)
count=count+1;
17)
SineVector=SineVector+t;
18)
end
19)
end
20)
InitialSoundMatrix(:,j)=(SineVector)
21)
end
22)
SoundMatrix=InitialSoundMatrix./(255*count);
23)
[SMRow SMColumn]=size(SoundMatrix);
24)
SoundColumn=reshape(SoundMatrix,SMRow*SMColumn,1);
25)
soundsc(SoundColumn,samplingRate);
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Conclusion
The spectra of the sound corresponding to
time can be computed using the
GenerateTimeVsFrequency.m matlab file
and its result matches approximately with
that of specgramdemo function of the
matlab. Additive synthesis of sound can be
simulated with the help of the matlab file
created MatrixToSound.m. It approximates
the actual sound.
Acknowledgement
My research paper is dedicated to my
parents Sh. Vasu Dev Sharma, Lecturer
Biology at Government Senior Secondary
School Bilaspur Himachal Pradesh(India)
and
Smt.
Bandna
Sharma;
T.G.T
Mathematics at Sarswati Vidya Mandir
Bilaspur Himachal Pradesh(India) whose
blessing and wishes made me capable to
complete this paper more effectively and
efficiently.
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References
(a) Textbooks
[1] Oppenheim, A.V., and R.W.

Schafer, Discrete-Time Signal Processing,
Prentice-Hall, Englewood Cliffs, NJ, 1989,
pp.713-718.
[2] Rabiner, L.R., and R.W. Schafer, Digital
Processing of Speech Signals, Prentice-Hall,
Englewood Cliffs, NJ, 1978.
(b) Websources
1)
http://www.mathworks.in/matlabcentr
al/fileexchange/index?utf8=%E2%9C%9
3&term=spectrogram
2)
http://en.wikipedia.org/wiki/Additive_s
ynthesis#Time-dependent_amplitudes
3)
http://isdl.ee.washington.edu/people/s
tevenschimmel/sphsc503/
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http://hyperphysics.phyastr.gsu.edu/hbase/audio/synth.html
By: Er. Vishv Mohan, s/o Sh. Vasu Dev Sharma

State Topper Himachal Pradesh 2008, 2009, 2010.
B.E(Hons.) Electrical & Electronics Engineering,
BITS-Pilani_(Rajasthan)-333031_India.
vasuvishv@gmail.com
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Analysis and Synthesis of Speech Using Matlab

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International Journal of Advancements in Research & Technology, Volume 2, Issue 5, M ay-2013