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Voice Signal Processing For Speech Synthesis

Conference Paper · June 2006

DOI: 10.1109/AQTR.2006.254660 · Source: IEEE Xplore

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 Voice Signal Processing For Speech Synthesis
Ovidiu Buza, Gavril Toderean, Alina Nica
Department of Telecommunications, Technical University of Cluj, Romania
26 – 28 G. Baritiu Str., 400027, Cluj-Napoca, Romania
email: Ovidiu.Buza@cs.utcluj.ro, Gavril.Toderean@com.utcluj.ro
Abstract - Our goal was to build a software application capable
to manipulate and analyse speech signal, extract characteristic
parameters needed for speech synthesis and to enhance the
speech quality. This article presents speech signal parameters
used in speech synthesis, the realized application and its
facilities, and main experimental results obtained.
I. INTRODUCTION
Today exist some applications dedicated to signal
processing, like MatLab, LabView, or to audio signal
processing, like CoolEditor or Goldwave [4]. Although they
provide many facilities that can be used in signal processing,
like digital filters or determining characteristic coefficients,
these applications can be used in general purpose analysis
(CoolEditor, Goldwave) or providing built-in API for signal
processing applications (MatLab, LabView). Specific
analysis of speech signal and adaptive filtering must be done
in own applications by the developers.
So in our work we have realized an application dedicated
to vocal signal analysis and processing, application named
SPE – Speech Processing and Enhancement Application.
SPE application offers facilities for processing and analysing
speech signal and other operations specifically needed in
speech synthesis. All these facilities and experimental results
are presented in this paper.
II. SPEECH SIGNAL MAIN PARAMETERS
Although speech signal is a continuous, non-stationary
signal, it can be assumed stationary on short segments of time
(about 20 ms). On these segments speech signal main
parameters vary in small amounts and can be determined by
different methods.
Speech represents the output signal of the phonatory system
modelled as a linear system having glottal signal as excitation
input. Speech signal s(t) is calculated as convolution of the
impulse response of the system h(t) and excitation signal e(t)
[2]:
t
s(t )   e( )h(t   )d (1)

s ( t )  h ( t ) * e( t ) (2)
Translating (2) in frequency domain, we obtain the spectrum
of the speech signal:
S ( )  H ( ) E ( ) (3)
The spectrum characterizes the signal in frequency domain,
giving information about spectral components. Speech signal
spectrum is calculated as the product of transfer function of
the phonatory system H(ω) and spectrum of glottal excitation
E(ω). | H(ω) | is called spectral winding and gives the mode as
spectral components vary within frequency.
Spectral peaks of high energy correspond to the maximums
of spectral winding and are characteristics for each phoneme in
speech. These peaks are called formants and are useful
parameters in analysis and synthesis of speech signal.
Excitation signal is calculated in two ways, depending of the
type of sound: for vocals the excitation is a uniform impulse
train, and for the consonants – a white noise. In the case of
vocals is important to specify excitation signal frequency, or
fundamental frequency. This parameter is connected with
speech prosody, giving different ways of pronunciation and
subjective emotional states.
Speech Signal Acoustical Parameters
Perception of speech signal is influenced by three factors :
volume, pitch and timbre. Volume is a measure of sound
intensity and corresponds with the amplitude of the signal.
Pitch is given by the fundamental frequency of the speech
signal and represents a measure of how a specific subject is
perceiving the sound. Timbre is determined by the harmonics
of the sound and corresponds with frequency components of
the signal spectrum.
III. SPEECH SIGNAL ANALYSIS
Many methods for speech signal analysis start with
Fourier transform, that gives complete description of signal
components in frequency domain. Expression for Discrete
Fourier Transform is [1]:
N 1
X (k )   x (n )e
n0
 j 2  nk / N
(4)
k = 1,2,...,N ,
where: X(k) is the value of spectral component k,
x(n) represents the signal sample n,
N is number of samples.
 The calculus is made in complex domain, so each
component X(k) is characterized by two real values:
amplitude Ak and phase φk
Ak= |X(k)| , φk=arg(X(k)). (5)
Signal spectrum leads to the detection of specific
parameters of voice signal, like spectral envelope and the
formants.
IV. THE SPEECH PROCESSING APPLICATION
Graphical interface of SPE application is presented in
figure 1. Main application window allows simultaneous
display of wave, amplitudes and phases of the signal. SPE
facilities are organized in two main categories : general wave
facilities, analyse and processing facilities.
SPE allows working with standard wave files : creating,
opening, saving and editing speech recordings in WAVE
format files, which have following structure [5]:
struct WAVEFORMATEX{
WORD wFormatTag;
WORD nChannels;
DWORD nSamplesPerSec;
DWORD nAvgBytesPerSec;
WORD nBlockAlign;
WORD wBitsPerSample;
WORD cbSize;
};
Wave structure parameters have following meaning:
wFormatTag is type of Waveform-audio format,
nChannels specifies number of channels,
nSamplesPerSec represents sampling rate,
nAvgBytesPerSec is data rate,
nBlockAlign is data block alignment,
wBitsPerSample gives sampling bit resolution,
Figure 1. SPE Main Application Window
cbSize represents the size of extra wave information.
SPE Application offers following facilities for vocal signal
analysis:
- manipulating speech recorded files (wave files);
- speech signal visualisation;
- pitch synchronous time window analysis;
- phase and amplitude spectrum analysis;
- automate detection of fundamental frequency;
- automate detection of speech formants;
- selective frequency filtering;
- signal harmonics editing for signal analysis and
enhancement.
Main SPE facilities are available through the application
toolbars. Corresponding commands are :
- wave file manipulating : NEW, OPEN, SAVE for creating,
opening and saving WAVE format files ;
- wave signal, amplitude and phase spectra visualisation:
VIEW WAVE, VIEW FFT, VIEW PHASE allow displaying
of wave, FFT amplitude spectrum and phase windows ;
- wave editing: SELECT/UNSELECT allows selecting
speech signal windows or segments; CUT, COPY, PASTE
allows editing of loaded signal;
- finding zero crossing: ZERO CROSS LEFT/RIGHT will
move left or right border of the selection zone to zero
crossing neighbours points;
- wave and spectrum information: VIEW FFT INFO allows
interactive visualisation of information associated with
amplitude spectrum (FFT) or phase spectrum (PHASE). User
can display current frequency and spectrum value interractive
way on mouse moving in spectral displaying area ;
- changing wave displaying resolution : ZOOM IN –
increases displaying resolution of current window (applies for
wave, amplitude and phase windows). Command reduces
window dimension in the benefit of analyse precision ;
ZOOM OUT is the complementary command, decreasing
displaying resolution of current window ; zoom out increases
amount of information displaying in the window, but reduces
analyse precision. SAMPLE ZOOM OUT allows controlled
reducing of displaying resolution. It is used for progressive
increasing of analyse dimension by adding samples one by
one in analyse window. SAMPLE ZOOM IN –
complementary command, allows progressive reducing of
analyse dimension by removing samples from analyse
window and so improving analyse precision. ZOOM ALL –
allows entire display of wave signal or spectra.
- sound playing : PLAY and PLAY LOOP – for playing
wave file or selected wave ; PAUSE, STOP to interrupt
playing.
- selective filtering : FILTER allows selective frequence
filtering and signal spectrum editing for analysis and signal
enhancement. This command provides interactive filtering of
frequencies and graphical editing of speech signal formants
and harmonics.
 Working with SPE application
First of all user will open a speech signal recording from a
WAVE file. Speech wave will be displayed in the upper part
of the screen, and FFT global spectra of amplitudes and
phases – in the bottom. Calculus of spectra is limited to 16k
samples, and this means that spectra don’t correspond with
entire wave but only to the first 16k samples of the wave. In
the most cases this value is quite enough for spectral analysis
of a windowed signal.
Magnitude of current window will be displayed in the
status bar. Tw parameter indicates dimension in time units
(seconds, milliseconds and microseconds), and Nw parameter
shows sampling dimension. Sampling frequency will be also
shown in FFT displaying zone, by Fs parameter. Sampling
frequency doubles spectral window magnitude because of
FFT mirroring effect: spectrum values are symmetrical from
a value that equals middle of sampling frequency. So spectral
window is calculated to display only first half of spectrum
values. FFT displaying zone will show also spectral
resolution, that is frequency distance between two
consecutive values of spectrum.
For analysing specific segments of speech signal, SPE
application allows windowed analysis. User can select
interactive way an analyse window and automatically adjust
window borders to zero crossing values. This fact gives a
maximum precision in computing the FFT spectrum, and so
Gaussian windowing method is not necessary. Rules for
computing FFT spectra are the following:
(1) if no selection exists, then spectra are calculated on
current displayed window;
(2) if user has selected a distinct analyse zone, spectra are
calculated for selected zone.
As user wants to make a specific analyse, he can select
displaying of wave form, amplitude spectrum or phase
spectrum. Application will display as default all the three
analyse windows.
In spectral analyse windows some useful information is
available. By moving mouse pointer on the amplitude or
phase spectrum, user gets the precise values of frequencies
and magnitude of spectrum, amplitude or phase.
Once a WAVE file has been loaded and speech signal was
displayed, application will display both amplitude and phase
signal spectra. Into speech analysing, more important is
amplitude spectrum, that gives information about the quality
of speech. By analysing this amplitude spectrum, one can get
important parameters of speech: fundamental frequency
corresponding to the tonality, formants or speech timbre,
spectrum magnitude corresponding to speech volume.
Application automatically detects formants or local
maximums of spectrum envelope, maximums that have to go
above a prior defined threshold. Magnitude, central frequence
and bandwidth of each formant are computed, that are
important parameters for speech synthesis. Central frequence
of the formant with higher amplitude (in most cases first or
second formant) is assumed to give the fundamental
frequency. This rule applies only for voiced segments of
speech, because consonants or vocal noises have no
fundamental frequency.
An example of automated formant detecting is shown in
figure 2. Here is illustrated a voiced segment of speech where
corresponding formants are detected at 204 Hz, 414 Hz, 613
Hz, 824 Hz . Fundamental frequency is given in this case by
the frequence of the first formant: 204 Hz.
Figure 2. Formants and fundamental frequency of a voiced segment of
speech
As previously said, SPE application allows selective
frequence filtering and spectrum editing, very important for
speech analysis and signal enhancement. The FILTER
command provides interactive filtering of frequencies and
graphical editing of speech signal formants and harmonics.
By dragging the mouse in FFT spectrum zone, user can easily
remove bands of frequencies for noise suppression or
increase other bands for rising signal energy in some specific
zones.
User can also modify spectrum formants or harmonics in
order to modify sound timbre and will be able for immediate
listening of sound acoustics modification. The experiments
that have been done show that a good quality voice own a
rich set of harmonics. Especially high order harmonics are
determinative for quality of speech. This is useful inside the
process of creating a vocal database used in speech synthesis,
where some vocal segments could be enriched by adding
higher harmonics.
Figure 3 shows the difference between two sounds (/A/
vowel) before and after enrichment with higher frequencies.
In the second case, the perception of sound is better.
Harmonics are always inserted at multiples of the first
formant frequence.
 Figure 3. Vowel /A/ enriched with higher harmonics
V. EXPERIMENTS MADE ON SPEECH SIGNAL
Using SPE application, many experiments and spectral
analyses have been done. The goal of these experiments was
to determine specific characteristics of speech signal
corresponding with pronunciation of various phonemes in
Romanian by distinct speakers.
Vocal signal was recorded by the mean of an unidirectional
dynamic microphone SM-500, then signal was sampled and
stored in PCM WAVE format files using a Creative Labs
Soundblaster. There have been also analysed audio signal
samples generated through a Creative Labs Synthetizer.
Experiments were focused on the idea to determine the
characteristics of speech signal that give a higher sound
quality. Thus, we have accomplished following analyses:
spectral analysis of vowels pronounced by different speakers,
spectral analysis of consonants, spectral analysis of
multitonal and modulated sounds ; perceptive sound
analyses : perceptive analysis of sounds emitted at different
phases, relationship between timbre of the sound and auditive
perception.
We have studied with SPE application main factors that are
significant for a good quality of speech recordings:
normalization of signal energy across speech segments,
constant rhythm of speech (that is given by a constant
fundamental frequency for each type of sound), rich timbre
with high harmonics, and not less important, professional
conditions of speech recording.
We have also done vowel sound analyses specific for
Romanian speech synthesis. In this way we have studied the
characteristics of Romanian vowel sounds in different
phonetic and prosodic contexts. Diagrams 1 and 2 show a
comparative study of vowel durations and frequencies made
on three characteristic segments: attack , middle and final.
These segments are illustrated in figure 4.
VI. CONCLUSIONS
We have described in this paper an original application
named Signal Processing and Enhancement Application -
SPE, designed for analysing and processing of speech signal.
We have presented main facilities of the application and also
how to work with application. Experiments accomplished by
working with SPE were presented and only first results were
shown here. In future articles we will present the whole set of
experiments used for speech synthesys.
REFERENCES
[1] A. Mateescu, Semnale şi sisteme- Aplicaţii în filtrarea semnalelor, Ed.
Teora, 2001
[2] E. Lupu, P. Pop, Prelucrarea numerică a semnalului vocal, vol.1, Ed.
Risoprint, 2004.
[3] T. Dutoit, “High-quality text-to-speech synthesis: an overview”,
http://tcts.fpms.ac.be/ synthesis/introtts.html
[4] O. Buza, “Stadiul actual în analiza şi prelucrarea semnalului vocal",
doctoral paper, Electronics and Tele-communications Faculty, Technical
University of Cluj-Napoca, 2005,unpublished
[5] O. Buza, “Contribuţii în analiza şi sinteza semnalului vocal”, doctoral
paper, Electronics and Telecommunications Faculty, Technical
University of Cluj-Napoca, 2005, unpublished
Atack Median Final
Figure 4. The three segments of vowel pronunciation
 DIAGRAM 1
DURATION OF ROMANIAN VOWEL SOUNDS – STATISTICS

Atack Median Final

(%) (%) (%) Total (ms)
(ms) (ms) (ms)
A 42.2 25 89.6 53 37.5 22 169.3
E 45.4 23 87.7 44 66 33 199.1
I 51.6 22 122 52 60.3 26 233.9
O 41.1 27 73.2 48 38.5 25 152.8
U 34.3 27 51.6 41 40.7 32 126.6

DIAGRAM 2
FREQUENCES OF ROMANIAN VOWEL SOUNDS – STATISTICS
Percents are calculated relative to median segment

Atack Median Final

(%) (%) (%)
(Hz) (Hz) (Hz)
A 120 110 109 100 124 114
E 112 106 106 100 117 110
I 130 115 113 100 123 109
O 118 109 108 100 119 110
U 116 117 99 100 99 100

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