Afin de réduire le débit binaire de la transmission de la parole sans perte de qualité de celle-ci, nous développons un vocodeur qui utilise des méthodes differentes pour le codage des trames voisées et non voisées. Nous présentons ici une nouvelle idée de décrire des phonèmes fricatifs (sifflantes) et plosifs avec seulement 20 bit par trame de t = 20ms. Nous montrons que ces phonèmes peuvent être représentés par des coefficients de la prédiction linéaire combinés avec un signal résiduel extrait d'un autre phonème prononcé par une personne differente connue à la station réceptrice du système de codage (voir figure 1). La présente contribution décrit aussi des algorithmes qui garantissent des transitions douces dans d'autres catégories de phonèmes. En appliquant cette technique on peut considérablement réduire le débit de transmission (jusqu'à 1 kbit/seconde) pour les trames non voisées. Nous obtenons de meilleurs résultats qu'en utilisant des variantes de CELP (prédiction linéaire excitée par une table de codage) à 4 kbit/seconde. La combinaison de ce codage avec des méthodes de codage harmonique (par exemple le MBE: 'Multiband Excitation') pour les trames voisées resulte en un débit binaire variable de moins de 3 kbit/seconde

EHNERT, Wolfram

HEUTE, Ulrich

I-Revues

SEIZIÈME COLLOQUE GRETSI — 15-19 SEPTEMBRE 1997 — GRENOBLE 993
Variable-rate speech coding: Replacing unvoiced excitations by linear
prediction residues of different phonemes
Wolfram Ehnert and Ulrich Heute
Institute for Network & System Theory, University of Kiel, Germany
Tel: +49 431 77572 401, Fax: +49 431 77572 403,
E-Mail: weh@techfak.uni-kiel.d400.de and uh@techfak.uni-kiel.d400.de
RÉSUMÉ
Afin de réduire le débit binaire de la transmission de la parole
sans perte de qualité de celle-ci, nous développons un vocodeur qui
utilise des méthodes differentes pour le codage des trames voisées
et non voisées. Nous présentons ici une nouvelle idée de décrire des
phonèmes fricatifs (sifflantes) et plosifs avec seulement 20 bit par
trame de t D 20ms.
Nous montrons que ces phonèmes peuvent être représentés par des
coefficients de la prédiction linéaire combinés avec un signal résiduel
extrait d’un autre phonème prononcé par une personne differente
connue à la station réceptrice du système de codage (voir figure 1).
La présente contribution décrit aussi des algorithmes qui garantissent
des transitions douces dans d’autres catégories de phonèmes.
En appliquant cette technique on peut considérablement réduire le
débit de transmission (jusqu’à 1 kbit/seconde) pour les trames non
voisées. Nous obtenons de meilleurs résultats qu’en utilisant des
variantes de CELP (prédiction linéaire excitée par une table de
codage) à 4 kbit/seconde. La combinaison de ce codage avec des
méthodes de codage harmonique (par exemple le MBE: ‘Multiband
Excitation’) pour les trames voisées resulte en un débit binaire
variable de moins de 3 kbit/seconde.
ABSTRACT
In order to reduce the bit rate of speech transmission while maintain-
ing the speech quality we are developing a vocoder which uses dif-
ferent methods for coding voiced and unvoiced frames. Within this
framework we present an idea for expressing fricative and plosive
phonemes with only 20 bits per frame (t D 20ms).
We show that they can be represented by LP(Linear Prediction)
coefficients and a residual signal where this residue is always taken
from a fixed phoneme of a test speaker known at the receiver station
of the coding system (see figure 1). Algorithms ensuring smooth
transitions to other speech-frame categories are also described below.
Using this technique the transmission rate of unvoiced frames can
be considerably reduced (down to 1 kbit/s) getting better listening
results than using CELP (Code Excited Linear Prediction) variants at
4 kbit/s instead. The resulting ‘Multi-Class Vocoder’ (voiced frames
are coded by Harmonic Coding at 4 kbit/s) has a variable rate of less
than 3 kbit/s on the average.
1 Introduction
Generally, speech can be subdivided into voiced, unvoiced
and mixed (voiced/unvoiced) segments [1]. At bit rates of
approximately 4 kbit/s, voiced speech can preferably be coded
with Harmonic or Sinusoidal Coding (e.g. [2] and [3]) while
the other speech categories miss a clear pitch frequency
needed for the harmonic reconstruction of the short-time
Fourier spectrum (STFS). Former approaches [4] [5] tried
to code these frames (plosives, fricatives and zero frames)
by Code-Excited Linear Prediction (CELP). The erroneous
assumption [5] that the LPC-residues of plosives and fricatives
are nearly white noise and can be coded only with stochastic
codebooks resulted in rather poor speech quality for a bit rate
of 4 kbit/s. The introduction of adaptive codebooks, however,
requires a higher bitrate. Furthermore, CELP coding realizes
a concatenation of artificial residual vectors each one being
chosen according to criteria based on the comparison of the
synthetic result to the original frame. It is evident that a non-
split excitation of a real human voice represents the physical
speech production more accurate, thus leading to better coding
results for unvoiced frames (at this bit rate). In the following
two paragraphs we show how such residues can be used
instead of the well-known residual-vector codebooks of CELP.
In the succeeding section we discuss possible realizations of a
variable-rate coder based on this idea.
2 Replacing mechanism, main idea
The original samples s.k/ (k D 1::S, S = number of samples
per frame) of each speech frame can be represented by the
LPC coefficients ai (i D 1::p, p = filter order of the linear
predictor) and the residual samples r.k/ according to
s.k/ D r.k/Ä
pX
iD1
ai s.k Ä i/: (1)
In an ideal coding system both r.k/ and ai will be calculated
at the transmitter station and transmitted totally to the receiver
where the original signal s.k/ can be recovered. For lack of
channel capacity in real applications, the transmitted data must
be reduced as much as possible. The LPC coefficients might
be coded as Line-Spectral Pair (LSP) frequencies with non-
linear quantizers or vector codebooks. Following the idea of
994
replacing r.k/ by residues of different phonemes we now can
cancel r.k/ totally (see figure 1).
TRANSMITTER
RECEIVER
s.k/
ai r.k/
HHHHj

Oai rnew.k/
Os.k/
@@R   	
external phoneme: snew.k/
ai;new
@
@R

PP
PPi





CHANNEL
Figure 1: Coding scheme of fricatives and plosives
At the receiver station we obtain a synthesized signal Os.k/
by filtering the speaker-independent residue rnew.k/ with the
decoded LPC coefficients Oai . As the experimental results show,
there is no need to provide different rnew.k/ for each kind of
phoneme. It is sufficient to distinguish between fricatives and
plosives. The residue rnew= f r ic.k/ of the phoneme ‘sh’ filtered
with the received coefficients Oai of any other fricative phoneme
(e.g. ‘s’, ‘f’ etc.) turns into the fricative sound corresponding
to Oai . This corresponds to the well-known fact that a sound is
mainly described by its spectral envelope rather than the fine
structure. In the same way it is possible to generate any kind
of unvoiced plosives (e.g. ‘k’, ‘t’ etc.; note that voiced plosives
like ‘b’ should be processed with harmonic coding) out of the
corresponding LPC coefficients Oai and the residue rnew=plos.k/
of a plosive phoneme of a different speaker.
According to the experiments, in both cases rnew= f r ic.k/
and rnew=plos.k/, this external speaker should have preferably
a female voice with a higher pitch frequency (because the
spectrum of the residue of a female voice is supposed to be
more constant at frequencies close to f D 4k H z). If possible,
the fricative substitution residue should be longer than the
fricative phoneme to be coded. Thus, the LPC coefficients
of each speech frame can be passed through the channel in
order to filter always the next S residual samples rnew= f r ic.k/
until the original phoneme has finished. The transition to
the following zero frame or voiced speech segment can be
realized with adequate windowing. Transitions from fricatives
to plosives do not exist because all plosives are preceded by
zeros [6]. This is one of the reasons why the synthesis of
plosive frames can be realized in a different and even more
efficient way as shown in the next section.
The residues rnew= f r ic.k/ and rnew=plos.k/ are normalized
vectors. Therefore, we have to transmit the energy factor (gain
factor) as well. However, the real advantage of the residue
replacement still consists in not transmitting the residues.
3 Replacing plosive residues
Plosive phonemes (unvoiced stops) are always produced by
an interruption of the airstream through the vocal tract (occlu-
sion) followed by an air burst [6]. Thus, the energy containing
samples of a plosive are always preceded by zeros.
0 100 200 300 400 500 600 700
−4000
−2000
0
2000
4000
time/samples −>
o
ri
gi
na
l s
ig
na
l
0 100 200 300 400 500 600 700
−4000
−2000
0
2000
4000
time/samples −>
s
yn
th
es
iz
ed
 s
ig
na
l
Figure 2: Original and synthesized phoneme ‘k’ of ‘cases’
This zero segment can be extended easily by adding new
zero samples. Therefore, we can define the beginning of
the air burst by ourselves. The actual realization of the
replacement algorithm turns each plosive phoneme of any
length into a phoneme (out of rnew=plos.k/ and Oai ) with
the fixed length of exactly two frames. Figure 2 compares
exemplarily the synthesized signal to the original samples (the
listening impressions of this result will be discussed in the
following sections).
If we permit a delay of two frames (40ms) at both transmit-
ter and receiver, then it is now even possible to generate the
plosive phoneme in the same way as described for the frica-
tive phoneme but with a trick:
Figure 3 shows a possible sequence of frame classifications
for the original speech: Three plosive frames (representing
one plosive phoneme) are followed by a voiced frame. The
synthesis procedure always knows two frame classifications
in advance.
At the moment T D 0 the decoding system already received
the decision of point T D 2 to be a plosive frame. As the
plosive which replaces the original one is only two frames
long, it is now obvious that it cannot start at T D 0. As
mentioned above, plosives are always preceded by zeros.
Therefore, the frame at T D 0 can be filled up with zeros.
At T D 1 the receiver will be informed about the frame of
T D 3 which is voiced. Now the substitution algorithm can
start using the plosive residue of the examplary phoneme:
The synthetic signal will be constructed backwards by cal-
culating the first values of the voiced frame using its harmonic
information (spectral magnitudes, pitch frequency etc.). These
samples (marked with an arrow in figure 3) serve (after being
995
flipped in time from forward to backward) as initial values for
the Linear Prediction filter with the coefficients received for
the respective plosive frame. rnew=plos.k/ must now be passed
through this filter starting with k D 320 down to k D 1. Thus,
we get a smooth transition between the plosive and the follow-
ing voiced frame. Transitions between plosives and fricatives
are handled in the normal way using the last plosive samples
as initial values for the succeeding fricative LP filtering.
0 50 100 150 200 250 300 350
−800
−600
−400
−200
0
200
400
600
800
0 50 100 150 200 250 300 350
−800
−600
−400
−200
0
200
400
600
800
0 50 100 150 200 250 300 350 400 450 500
−4000
−3000
−2000
−1000
0
1000
2000
3000
4000
5000
Endergebnis
0 20 40 60 80 100 120 140 160
−4000
−3000
−2000
−1000
0
1000
2000
3000
4000
5000
(plosive)(plosive)(plosive)
z
 1
z
 1
z
 1
(plosive) (voiced)(plosive)(plosive)
+ + +
(voiced)
T = 0 T = 2
+
a^
p
a^
2
a^
1
T = 1 T = 3
resulting
initial values
s^(k)
initial values
s^(k)
(signal backwards)
r
new=plos
(1::160) r
new=plos
(161::320)
(ipped again in time)
-
Figure 3: Reconstruction of plosive phonemes
rnew=plos.k/ is the residue of a natural plosive phoneme.
This phoneme has once been cut at the transition to a voiced
frame. Therefore, it was necessary to find a new transition
algorithm. The transition from the preceding zero samples
(interruption of the airstream) to the plosive, however, has
been included in the phoneme. Therefore, this transition is
expressed by the residue rnew=plos.k/ itself and needs no
correction (smoothing algorithm) anymore.
4 A coder based on phonetic categories
The resulting coder using the residue-replacing algorithms
presented above is a variable-rate coder because voiced and
unvoiced speech frames are coded with different methods and
rates. The resulting ‘Multi-Class Vocoder’ needs to distinguish
between voiced frames, plosives, fricatives, and zero frames.
If we already allow a delay of 40ms we know the original
samples of the two succeeding frames. Then it is possible to
obtain a quite acceptable (but never perfect) classification of
non-disturbed speech. Mainly three errors can occur:
– A voiced frame is interpreted as a fricative or plosive:
In this (worst) case the periodic signal will be replaced
by the LP-filtered fricative or plosive residue. Since
the residue does not contain any pitch frequency, the
synthesized frame is aperiodic as well. Furthermore,
it is impossible to perform an acceptable transition to
previous or next frames which might be voiced and
correctly recognized as such. These cases as well as the
next one fortunately happen rarely.
– An unvoiced frame is interpreted as a voiced segment:
The effect is known from the first harmonic-coding
approaches. The synthesized signal sounds metallic
because of the missing random components of unvoiced
speech segments. The lack of naturalness could be
reduced using a Multiband-Excitation (MBE) vocoder
[2] instead of Harmonic Coding for coding voiced
frames (see following section).
– Fricatives are coded as plosives and vice versa: This
confusion can be explained by the fact that some speech
segments are phonetically hard to define: A ‘t’ can be
spoken in different ways and might sound more frica-
tive than plosive. In this special case the plosive com-
ponent of the fricative-sounding ‘t’ will be generated
automatically by the transition from the preceding zero
frame to the synthesized fricative. The synthesized ‘t’
is phonetically closer to the original speech when using
rnew= f r ic.k/. It results in an advantage that the classifi-
cator attaches more importance to the sound than to the
spelling of words.
Figure 4 shows the structure of the introduced vocoder. The
results are far better than those using a CELP coder (at 4 kbit/s)
for unvoiced frames [5]. The bit rate, however, is going down
to less than 3 kbit/s if we code the fricatives and plosives with
20 bits per frame (20ms). Tests have even shown that the LPC
coefficients of fricatives can be updated for only each second
frame without changing the speech quality.
Till now we achieved the best results combining the 4.15
kbit/s - IMBE vocoder (Inmarsat-M standard) with the new
substitution algorithm for unvoiced phonemes as presented
above. In some cases the fricatives and plosives coded by the
IMBE vocoder itself (applied for voiced and unvoiced frames)
resulted to be closer to the original phoneme. In other cases
especially the fricatives sounded reverberant in the IMBE
version and could be highly improved by the substitution
algorithm. In total, improvements and deteriorations of the
listening results of the new ‘Multi-Class’ (MC) vocoder seem
to keep the balance in comparison to the former IMBE
realization of Griffin and Lim [2] which did not distinguish
methodically between voiced and unvoiced frames. However,
at the moment the actual ‘Multi-Class’ version including the
recognition of the mentioned phoneme categories is far away
from being realized in real time. Furthermore, the delay of
td D 40ms at both transmitter and receiver may exceed
the demands and standards of a telefonic system. There are
ideas to reduce the delay of the receiver to only one frame
(t D 20ms) using a different algorithm for the transitions from
plosives to voiced frames. In some cases, however, this time
saving might result in a worse phoneme detection.
996
speechframe: s.k/
voiced frame zero frame plosive/
fricative
Harmonic Coding Linear Prediction
80 bits 3 bits 20 bits
RECEIVER
CHANNEL
?
?
?
?
?
??
?
?

Q
Q
QQs
?
recognition
?
synthesized signal: Os.k/
Figure 4: Proposal for a variable-rate coder
5 Applications
As described in the section above it is still not possible to use
the new substitution algorithm for a real time realization of a
coding system. TTS (Text-to-Speech) systems, however, need
at best the synthesis algorithm to be realized in real time (all
other delays are not relevant). The coding will be applied only
once (before application) in order to get a memory-efficient
database of phonemes, diphones or syllables. If a phoneme of
the original speech data has not been recognized correctly, the
classification can be rectified ‘by hand’. As the new ‘Multi-
Class’ version is already concatenating voiced and unvoiced
speech parts it might be useful for a concatenation-based TTS
system. At present, investigations are being carried out on this
topic.
6 Conclusion
Residues of fricatives and plosives do not represent white
noise. Nevertheless, they can be replaced by different residues
without changing the individual characteristics of a speaker,
thus getting better listening results than by using the residues
of a stochastic codebook. For the coding application unvoiced
frames must be subdivided into plosives, fricatives and zero
frames, but a confusion of plosives and fricatives does not
result in a wrong phoneme at the receiver station. Unvoiced
frames are represented by only one set of LPC coefficients
and the gain. Therefore, we get a highly reduced variable rate
(below 3 kbit/s on the average) for the Multi-Class Vocoder
which is, however, not realizable in real time up to now.
References
[1] A. Das and A. Gersho, “Variable dimension spectral cod-
ing of speech at 2400bps and below with phonetic classifi-
cation,” in Proceedings of the International Conference on
Acoustics, Speech and Signal Processing (ICASSP), vol. 1,
pp. 492–495, May 1995.
[2] D. Griffin and J. Lim, “Multiband excitation vocoder,”
IEEE Transactions on Acoustics, Speech, and Signal
Processing, vol. 36, pp. 1223–1235, August 1988.
[3] R. McAulay and T. Quatieri, “Speech analysis-synthesis
based on a sinusoidal representation,” IEEE Transactions
on Acoustics, Speech, and Signal Processing, vol. 34,
pp. 744–754, 1986.
[4] I. Trancoso, L. Almeida, and J. Tribolet, “A study on the
relationships between stochastic and harmonic coding,” in
Proceedings of the International Conference on Acoustics,
Speech and Signal Processing (ICASSP), 1986.
[5] S. Otte, Mehrklassen-Sprachkodierung: Verarbeitung pe-
riodischer Rahmen. Diploma, LNS/CAU Kiel, 1996.
[6] M. Torres and P. Iparraguirre, “Acoustic parameters for
place of articulation identification and classification of
spanish unvoiced stops,” Speech Communication, vol. 18,
pp. 369–379, 1996.


Variable-rate speech coding: Replacing unvoiced excitations by linear prediction residues of different phonemes

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