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Automatically Estimating the Input Parameters of Formant-Based Speech Synthesizers

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Automatically Estimating the Input Parameters of Formant-Based Speech Synthesizers

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The paper presents preliminary results of a new framework for automatically extracting the input parameters of a class of synthe-sizers. The framework allows to speed up the process of utterance copy (or speech imitation), where one has to find the model parameters that lead to a synthesized speech sounding close enough to the natural target speech. The results confirm that the error surface is non-convex with many local minimal, making the task a hard-to-solve inverse problem. Therefore, the framework is based on genetic algorithms, which is a ro-bust non-linear optimization technique. The work also discusses the use of speech analysis toolkits, such as Praat and Snack, to improve conver-gence.
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Automatically Estimating the Input Parameters
of Formant-Based Speech Synthesizers
Aline Figueiredo1, Tales Imbiriba1, Edward Bruckert2and Aldebaro Klautau1
1Signal Processing Laboratory (LaPS) - Universidade Federal do Par´a
DEEC, UFPA, 66075-110, Belem, Para, Brazil. http://www.laps.ufpa.br
2Fonix Corporation
180 W. Election Road, Suite 200, Draper, UT 84020, USA. http://www.fonix.com
Abstract. The paper presents preliminary results of a new framework
for automatically extracting the input parameters of a class of synthe-
sizers. The framework allows to speed up the process of utterance copy
(or speech imitation), where one has to find the model parameters that
lead to a synthesized speech sounding close enough to the natural target
speech. The results confirm that the error surface is non-convex with
many local minimal, making the task a hard-to-solve inverse problem.
Therefore, the framework is based on genetic algorithms, which is a ro-
bust non-linear optimization technique. The work also discusses the use
of speech analysis toolkits, such as Praat and Snack, to improve conver-
gence.
1 Introduction
This paper presents preliminary experimental results of a framework based on
evolutionary computing for estimating the input parameters of the so-called for-
mant-based speech synthesizers. More especifically, two synthesizers are studied:
Klatt’s [1, 2] and HLsyn [3, 4].
The framework aims to speedup the process of utterance copy, where one
has to find the model parameters that lead to a synthesized speech sounding
close enough to the natural target (or reference) speech. The proposed solution
is centered in evolutionary computing; more specifically, in genetic algorithms
(GA) [5]. The task can be cast as a hard inverse problem, because it is not an
easy task to extract the desired parameters automatically (see, e.g., [6]). Because
of that, in spite of recent efforts [7–9], most studies using parametric synthesizers
adopt a relatively time-consuming process (see, e.g., [2]) for utterance copy and
end up using short speech segments (words or short sentences). The possibility of
automatically analyzing speech corpora is very important to increase the knowl-
edge about phonetic and phonological aspects of specific dialects, endangered
languages, spontaneous speech, etc.
The work is organized as follows. Section 2 gives information about the two
adopted synthesizers. In Section 3 we describe the proposed approach while
Section 4 presents preliminary simulation results, followed by the conclusions in
Section 5.
Proceedings of the International Joint Conference IBERAMIA/SBIA/SBRN 2006 - 4th Workshop in
Information and Human Language Technology (TIL’2006), Ribeir˜ao Preto, Brazil, October 23–28, 2006.
CD-ROM. ISBN 85-87837-11-7
1
2 The Synthesizers
The speech synthesizer is the back end of text-to-speech (TTS) systems [10–13].
For example, in [14, 15], Klatt-based synthesizers were used in TTS systems for
the Romanian and Brazilian Portuguese languages, respectively. Synthesizers
are also useful in speech analysis, such as in experiments about perception and
production [16]. For example, in [17], an analysis-by-synthesis approach using
Klatt’s was adopted to accurately synthesize electrolarynx speech.
The Klatt’s synthesizer is called a formant synthesizer because some of its
most important parameters are the formant frequencies (the resonance frequen-
cies of the vocal tract, see, e.g., [18]). HLsyn is another parametric synthesizer,
which runs on top of Klatt’s, and is an hybrid of the formant and articula-
tory approaches. It should be noticed that these two were chosen due to their
widespread use, but the framework is valid for other parametric synthesizers as
well (e.g., the ones described in [19, 12]).
Parametric synthesis has been adopted in commercial products such as the
DECTalk (see [20] for demos). However, nowadays, most commercial TTS sys-
tems adopt concatenative synthesis [13], which relies on algorithms such as
PSOLA [21] and MBROLA [22] that modify and concatenate previously stored
segments of speech to generate the synthesized speech.
When speech technologists shifted from formant-based to concatenative syn-
thesis, a gap was created in some areas of speech sciences, especially linguistics.
In spite of not being competitive with concatenative techniques for develop-
ing some commercial applications, formant-based synthesis is very important in
many speech studies. Most parameters of a formant synthesizer are closely re-
lated to physical parameters and have a high degree of interpretability. This is
essential, for example, in studies of the acoustic correlates of voice quality, such
as male/female voice conversion, and the simulation of breathiness, roughness,
and vocal fry [19]. Unfortunately, after loosing most of the economical appeal
to concatenative techniques, the number of research efforts on developing auto-
matic tools for dealing with formant synthesizers is currently very limited. This
paper aims to reduce this gap.
2.1 Klatt’s
There are many versions of the Klatt’s synthesizer. The one called KLSYN [2]
had its source code published in [1] (in Fortran). Later [2], Dennis Klatt and
his daughter presented an improved version called KLSYN88, which is currently
commercialized by Sensimetrics (www.sens.com). In the early Nineties, a C ver-
sion of KLSYN was posted in the comp.speech USENET group. Jon Iles rewrote
it in C++ and called it Object Formant Synthesizer (OFS). Some of the dif-
ferences between KLSYN and KLSYN88 are discussed in [2]. With respect to
source-filter modeling of speech production, KLSYN88 has three “sources”: im-
pulsive, “Klatt’s natural” and Liljencrants-Fant.
Basically, the Klatt synthesizer works as follows. For each frame (its duration
is set by the user, often in the range from 5 to 10 milliseconds), a new set of
parameters drives the synthesizer. Some parameters are used to generate the
excitation signal (mimicking the influence of the air flow), while others are used
to set the filters that shape the speech spectrum (mimicking the action of the
vocal tract organs).
Table 1 lists the KLSYN88 parameters used in this work. For their complete
description, the reader is referred to [1, 2]). The parameters that do not vary
over time are not listed here. There are three filter-banks in KLSYN88: one
in which the resonators are in series (cascade) and two in which they are in
parallel. As conventionally done, the voicing-excited parallel bank was not used
(the amplitude parameters ANV, ATV, A1V, A2V,..., A6V were assumed to
be zero). The number of cascaded resonators was NF=5. The glottal source was
the KLGOTT88 (SQ was not used).
Table 1. KLSYN88 parameters used in this work.
Description ID Range
Fundamental frequency (tenths of Hz) F0 [0, 5000]
Amplitude of voicing (dB) AV [0, 80]
Open quotient OQ, KOPEN [10, 99]
Extra tilt of voicing spectrum TL, TILT [0, 41]
Flutter - random fluctuation in F0 (%) FL [0, 100]
Diplophonia (%) DI [0, 100]
Amplitude of aspiration AH, ASP [0, 80]
Amplitude of frication AF [0, 80]
Formant F1 F1 [180, 1300]
F1 bandwidth B1 [30, 1000]
F1 change - open portion of a period DF1, df [0, 100]
B1 change - open portion of a period DB1, db [0, 400]
Formant F2 F2 [550, 3000]
F2 bandwidth B2 [40, 1000]
Formant F3 F3 [1200, 4800]
F3 bandwidth B3 [60, 1000]
Formant F4 F4 [2400, 4990]
F4 bandwidth B4 [100, 1000]
Formant F5 F5 [3000, 4990]
F5 bandwidth B5 [100,1500]
Formant F6 F6 [3000, 4990]
F6 bandwidth B6 [100,4000]
Nasal pole frequency FNP, fp [180, 500]
Nasal pole bandwidth BNP, bp [40, 1000]
Nasal zero frequency FNZ, fz [180, 800]
Nasal zero bandwidth BNZ, bz [40, 1000]
Frequency of the tracheal pole FTP [300, 3000]
Bandwidth of the tracheal pole BTP [40, 1000]
Frequency of the tracheal zero FTZ [300, 3000]
Bandwidth of the tracheal zero BTZ [40, 2000]
Amplitude of frication-excited parallel formants (x=2 to 6) AxF [0, 80]
Bypass path amplitude AB [0, 80]
Bandwidth of frication-excited parallel formants B2F [40, 1000]
Bandwidth of frication-excited parallel formants B3F [60, 1000]
Bandwidth of frication-excited parallel formants B4F [100, 1000]
Bandwidth of frication-excited parallel formants B5F [100, 1500]
Bandwidth of frication-excited parallel formants B6F [100, 4000]
2.2 HLsyn
HLsyn incorporates specializard knowledge about acoustic and articulatory pho-
netics. Its purpose is to work as a wrapper (or an upper layer) to the Klatt
synthesizer and achieve a reduction on the number of input parameters. Hence,
for synthesizing each frame of speech, HLsyn requires 13 parameters, which are
then converted to the 48 parameters used by Klatt’s. The Klatt’s synthesizer is
invoked and actually generates the speech samples. Besides reducing the num-
ber of parameters, HLsyn imposes restrictions that help avoiding non-feasible
solutions. That is, a given set of 48 Klatt’s parameters can eventually match the
desired sound, but they may be physically unfeasible [3]. Table 2 lists the HLsyn
parameters used in this work and their range.
Table 2. List of HLsyn parameters and the range adopted in the simulations.
ID Description Unity Range
f1 First natural (formant) frequency Hz [180, 1300]
f2 Second natural (formant) frequency Hz [550, 3000]
f3 Third natural (formant) frequency Hz [1200, 4800]
f4 Fourth natural (formant) frequency Hz [2400, 4990]
f0 Fundamental frequency (known as “pitch”) Hz [0, 500]
ag Average area of glottal (membranous portion) mm20.01 ×[0, 4000]
ap Area of the posterior glottal opening mm20.01 ×[0, 1000]
ps Subglottal pressure cm H2O 0.01 ×[0, 1000]
al Cross-sectional area of constriction at the lips mm20.1 ×[0, 1000]
ab Cross-sectional area of tongue-blade constriction mm20.1 ×[0, 1000]
an Cross-sectional area of velopharyngeal port mm20.1 ×[0, 1000]
ue Rate of increase of vocal-tract volume cm3/s [0,1000]
dc Change in vocal-fold or wall compliances % [-150, 150]
3 Automatically Learning the Input Parameters
The approach described in this section tries to solve the following problem: given
an utterance to be synthesized, find for each frame a sensible set of parameters
to drive the synthesizers. The number of parameters and their dynamic range
make an exhaustive search unfeasible. GA [5] was adopted as the main learning
strategy. One key point when posing a new optimization problem is to choose
the objective (or fitness) function. The following subsections discuss the options
considered in this work.
3.1 Figures of merit - Fitness functions
One of the fitness functions used in this work is the mean square error (MSE),
calculated between the target h(n) and synthesized s(n) waveforms. The main
reason for considering MSE is its simplicity. On the other hand, it is well-known
that MSE does not reflect the perceived quality of speech signals.
Another adopted fitness function was the spectral distortion (SD) between
the target spectrum H(f) and the synthesized spectrum S(f), which is given by
SD =s1
f2f1Zf2
f120 log10
|H(f)|
|S(f)|2
df
and calculated through a fast Fourier transform (FFT) routine.
The spectral distortion is widely adopted in speech coding for quantizing
the filter in the source-filter model (also known as LPC filter, after the linear
prediction algorithm used for its estimation). The related concept of transparent
coding [23], which means that H(f) and S(f) are close enough to be perceptually
undistinguishable, is based on the following statistics of SD: a) the average SD
(among all frames) is less or equal than 1 dB, b) there should be no frames for
which SD > 4 dB, and c) the number of frames for which 2 S D 4 dB is less
than 2%. This empirical performance goal was modified (to be less restringing)
and used as part of the GA termination procedure.
3.2 Architecture
Among many possible ways of using GA for solving the posed problem, three
architectures were studied:
Intraframe: for each speech frame, we setup a conventional GA problem. For
example, if the target utterance is 1 second long and the frame duration is 10
milliseconds (ms), 100 GA problems are solved in a completely independent
way from each other, with the populations randomly initialized for each
frame. The random numbers are uniformly distributed on the range specified
for each parameter.
Interframe: a fraction Fof the initial population for frame tis obtained by
copying some of the best individuals from the previous frame t1. The
remaining 1 Ffraction is randomly (and uniformly) initialized. This archi-
tecture takes in account that speech varies smoothly, specially for stationary
sounds such as vowels.
Knowledge-based : the initial population for frame tdepends not only on
previous frames, but also on parameter values estimated through speech
analysis algorithms such as the ones adopted in Praat and Snack.
Note the knowledge-based architecture has many degrees of freedom. Some
possibilities, not implemented yet, are briefly discussed here. For example, one
can take in account the kind of sound to be synthesized in each frame. Such
information can be obtained from an algorithm for phonetic segmentation (see,
e.g., [24]) and used both when doing the speech analysis and GA optimization.
While it is quite difficult to get good accuracy with a large vocabulary continuous
speech recognition as the one used in [24], much better phonetic segmentations
can be obtained when the orthographic transcription is known a priori (by us-
ing the so-called forced alignement). It is fair to say that this is the case with
utterance copy experiments and, consequentely, one can assume a reasonably
accurate phonetic transcription could be obtained (assuming the speech signal
is not too noisy).
4 Results
The first stage of the work was to evaluate the error surface for the optimization
problem. The motivation was to get insight about the difficulty an algorithm
would face. This section also presents results for synthesizing vowels with GA
using the intraframe architecture. These results are followed by figures that
illustrate how speech analysis can be used in the knowledge-based architecture.
The fitness function used in these experiments was the spectral distortion, which
outperformed the MSE.
4.1 Studying the error surface
The experiment used artificial vowels that were synthesed by HLsyn itself. Be-
cause the “right” parameters were known, the experiment can be easily con-
trolled. First, a set of 13 HLsyn parameters were used to produce few frames of
a stationary sound. Later, new versions of this sound were obtained by varying
a couple of parameters. The task of the GA algorithm was to find the correct
pair of values corresponding to the modified parameters. This allows to easily
visualize the error surface as shown in Figure 1, which describes results of four
different pairs of parameters.
Figure 1(a) shows the error surface when the correct values of F0 and F1 are
1200 and 500, respectively. These values correspond to a fundamental frequency
F0 of 120 Hz and a first formant F1 of 500 Hz. One can see that the curve has
many local minima. The other three pairs also show a similar situation. From
these four graphs, one can note that the sensitivity to each parameter also varies
considerably.
4.2 Synthetic Vowels
This subsection shows some results when using GA to obtain the parameters of
vowels generated by the Klatt synthesizer. The motivation was to study the GA
convergence and tune its parameters, such as the mutation probability. Figure 2
illustrates the convergence when the studied pair of parameters were the first
two formants F1 and F2 with correct values of 500 and 1000 Hz, respectively.
The GA population was 50 individuals. The figures indicate that, in this case,
around 50 iterations were sufficient for having individuals close to the optimal
values. After 100 iterations, all individuals were very close to the optimum point.
The next subsection discusses more ellaborated experiments with whole sen-
tences.
300 400 500 600 700 800 900 1000
600
800
1000
1200
1400
1600
1.4
1.6
1.8
2
2.2
2.4
2.6
2.8
F1
FO
log10(distortion)+0.1
(a) F0 vs. F1
0200 400 600 800 1000
0
500
1000
1500
−1
−0.5
0
0.5
1
1.5
2
2.5
AN
AB
log10(distortion)+0.1
(b) AB vs. AN
0100 200 300 400 500
0
200
400
600
800
1000
−1
−0.5
0
0.5
1
1.5
2
2.5
3
DC
PS
log10(distortion)+0.1
(c) PS vs. DC
0
500
1000
1500
0
200
400
600
800
1000
−1
−0.5
0
0.5
1
1.5
2
2.5
AP
UE
log10(distortion)+0.1
(d) UE vs. AP
Fig. 1. Error surfaces and contours of four pairs of parameters. The curves illustrate
the local minima and varied sensitivity of parameters.
4.3 Synthetic Sentences
Before studying the synthesis of natural (human-generated) sounds, the problem
to be circumvented is the computational cost of the optimization procedure. It
takes too long for the synthesis of a whole sentence. Each frame is a GA problem
to be solved, and even using the interframe architecture, the time is still too long.
The approach adopted was to invest on the knowledge-based architecture,
using speech analysis algorithms to estimate parameters. For extracting these
speech parameters two widely known speech analysis toolkits were used: Praat
(www.praat.org) and Snack (www.speech.kth.se/snack). Both provide support
to scripts. We used these tools to estimate formants (central frequencies and
bandwidths) and F0 (fundamental frequency). Given the estimated parameters
for a specific frame, these parameters become the means of Gaussians that are
used to randomly initialize part of the GA population (the other part comes
from a percentage of the population from the previous GA problem, as in the
interframe architecture). The variances of these Gaussians were empirically ob-
tained.
(a) After 1 iteration
500 1000 1500 2000
480
500
520
540
560
580
600
620
640
660
680
F2
F1
(b) 10 iterations
500 1000 1500 2000
480
500
520
540
560
580
600
620
640
660
680
F2
F1
(c) 50 iterations
500 1000 1500 2000
480
500
520
540
560
580
600
620
640
660
680
F2
F1
(d) 100 iterations
Fig. 2. Error contour and location of the GA individuals when finding the optimum
F1 (y-axis) and F2 (x-axis) values for a synthetic vowel. Panel (a) has different axes
range. The optimal values F1=500 and F2=1000 Hz are indicated by a blue ).
The use of speech analysis was quite effective to speed up convergence when
dealing with F0 and the formants. Figure 3 shows that Praat and Snack achieve
a reasonable result and basically agree in their estimations. The utterance was
“five women played basketball”, generated by HLsyn for a female voice (available
at www.sens.com/hlsyn overview.htm).
The next stage, after tuning the system with synthetic sentences, is to con-
duct a formal evaluation using the TIMIT corpus, which is the most popular
among the corpora distributed by the LDC (www.ldc.upenn.edu). Currently, the
main difficulty is that there are no ready-to-use routines for estimating the other
parameters of the synthesizers and the synthesis of a long utterance still takes
considerable time.
5 CONCLUSIONS
A new framework for automatically extracting the input parameters of the Klatt
and HLsyn synthesizers was presented. A formal evaluation is still required, but
(a) Praat result
0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6
−1
−0.8
−0.6
−0.4
−0.2
0
0.2
0.4
0.6
0.8
1
(b) Time domain only
0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6
0
20
40
60
80
100
120
140
160
Time (s)
F0 (Hz)
Praat
Snack
(c) F0 tracks by Praat and Snack
0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6
0
500
1000
1500
2000
2500
3000
3500
4000
Time (s)
F1−F3 (Hz)
(d) Formant tracks by Praat and Snack
Fig. 3. Whole sentence analyzed by Praat and Snack.
some preliminary results with synthetic sounds illustrated the problem charac-
teristics (e.g., error surfaces) and possibile solutions. Currently, new algorithms
for estimating parameters through speech analysis are under development. The
idea is to split the task into two: speech analysis to obtain key parameters, fol-
lowed by tuning of these and the rest of the parameters through evolutionary
computing (GA, particle swarm, etc.). The final product of this research will be
a tool for speech analysis, helpful to speech therapists, phoneticians and other
professionals in related areas.
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This paper describes a system for synthesis of speech with a terminal analog synthesizer like the Klatt synthesizer. The control of the synthesizer is accomplished with a small number (about 10) of high‐level (HL) parameters that capture directly certain attributes of the state of the vocal tract during speech production. The multiplicity of low‐level parameters in the synthesizer are derived from the HL parameters through a set of mapping relations. The HL parameters consist of (1) the fundamental frequency; (2) a specification of the vocal tract in terms of the first three or four natural frequencies of the tract assuming no opening to the glottis or the nose; (3) the areas of three major orifices in the vocal tract: the glottal opening, the velopharyngeal opening, and (for consonants) the principal constriction in the vocal tract; (4) a measure of active expansion or contraction of the vocal‐tract volume for obstruents; (5) a measure of the efficiency of turbulencenoise generation for obstruents. The mapping relations for obstruents involve the calculation of airflows, pressures, wall displacements, glottal waveform characteristics, vocal‐tract losses, and noise due to turbulence. Some examples of the mapping relations for both obstruents and sonorants are given. [Research supported by the National Institutes of Health.]
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The ability to accurately synthesize electrolarynx (EL) speech may provide a basis for better understanding the acoustic deficits that contribute to its poor quality. Such information could also lead to the development of acoustic enhancement methods that would improve EL speech quality. This effort was initiated with an analysis-by-synthesis approach that used the Klatt formant synthesizer to study vowels at the end of utterances spoken by the same subjects both before and after laryngectomy (normal versus EL speech). The temporal and spectral features of the original speech waveforms were analyzed and the results were used to guide synthesis and to identify parameters for modification. EL speech consistently displayed utterance-final fixed (mono) pitch and normal-like falling amplitude across different vowels and subjects. Subsequent experiments demonstrated that it was possible to closely match the acoustic characteristics and perceptual quality of both normal and EL speech with synthesized replicas. It was also shown that the perceived quality of the synthesized EL speech could be improved by modification of pitch parameters to more closely resemble normal speech. Some potential approaches for modifying the pitch of EL speech in real time will be discussed. [Funded by NIH Grant R01 DC006449.]
Article
A softwareformantsynthesizer is described that can generate synthetic speech using a laboratory digital computer. A flexible synthesizer configuration permits the synthesis of sonorants by either a cascade or parallel connection of digital resonators, but frication spectra must be synthesized by a set of resonators connected in parallel. A control program lets the user specify variable control parameter data, such as formant frequencies as a function of time, as a sequence of 〈time, value〉 points. The synthesizer design is described and motivated in Secs. I–III, and fortran listings for the synthesizer and control program are provided in an appendix. Computer requirements and necessary support software are described in Sec. IV. Strategies for the imitation of any speech utterance are described in Sec. V, and suggested values of control parameters for the synthesis of many English sounds are presented in tabular form.
Conference Paper
This paper reports the implementation of high-quality synthesis of speech with varying speaking styles using the Klatt (1980) synthesizer. This research is based on previously-reported research that determined that the glottal waveforms of various styles of speech are significantly and identifiably different. Given the parameter tracks that control the synthesis of a normal version of an utterance, those parameters that control known acoustic correlates of speaking style are varied appropriately, relative to normal, to synthesize styled speech. In addition to varying the parameters that control the glottal waveshape, phoneme duration, phoneme intensity, and pitch contour are also varied appropriately. Listening tests that demonstrate that the synthetic speech is perceptibly and appropriately styled, and that the synthetic speech is natural-sounding, were performed, and the results are presented