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Description  |
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BACKGROUND OF THE INVENTION
The present invention relates to speech coding systems, and more
particularly to a speech coding system used in telephone communication
which is carried out in such a manner that a speech signal is converted
into a compressed digital signal on the transmitting side and is
reproduced from the compressed digital signal on the receiving side, and
suitable for processing a speech signal which is generated in a noisy
environment.
The signal waveform is given by a combination of fundamental waveform
patterns, each of which appears two to ten times in a time interval of,
for example, about 20 msec (hereinafter referred to as a "frame"). In
conventional speech analysis-synthesis systems, the transmitting side
performs a sampling operation for an input speech signal and extracts
transmission parameters indicative of the feature and repetition period
(namely, pitch period) of a fundamental waveform pattern from the sampled
values of the input speech signal at each frame, and the receiving side
reproduces the speech signal on the basis of the transmission parameters.
In the PARCOR (partial auto-correlation) system which is representative of
one of the conventional speech analysis-synthesis systems, it is judged
whether each of the frames formed in analyzing a speech signal is a voiced
frame or unvoiced frame, and a reproducing operation is performed in such
a manner that the output of an excitation source for generating white
noise is used for the unvoiced frame and a single pulse which represents a
fundamental waveform pattern and is generated at an interval equal to the
pitch period thereof indicated by the transmission parameters, is used for
the voiced frame. The PARCOR system, as mentioned above, uses a simple
excitation source, and hence is advantageous in that a speech signal can
be coded at a low bit rate but disadvantageous in that the quality of a
synthesized speech is degraded. The PARCOR system is described in, for
example, an article entitled "An audio response unit based upon partial
auto correlation" (IEEE Transaction Communication, Vol. COM-20, pages 792
to 797, Aug., 1972).
Further, systems for improving the quality of a synthesized speech by
generating a plurality of pulses representative of a fundamental waveform
pattern at an interval equal to the pitch period thereof, are proposed in,
for example, an article entitled "A New Model of LPC Excitation for
Producing Natural-Sounding Speech at Low Bit Rates" by B. S. Atal and J.
R. Remde (Proc. ICASSP 82, Vol. 1, pages 614 to 617, 1982), and an article
entitled "A Speech Coding Method Using Thinned-Out Residual" by A.
Ichikawa et al. (Proc. ICASSP 85, Vol. 3, pages 961 to 964, 1985).
In the above systems, in order to reduce the number of bits necessary for a
coding operation, a pulse train generated at an interval equal to the
pitch period of a fundamental waveform pattern is made identical with a
pulse train generated at an interval equal to the pitch period of another
fundamental waveform pattern, in one frame. In this case, however,
information on the position of each pulse is required, and thus the number
of pulses generated in one pitch period of a fundamental waveform pattern
is limited. Accordingly, the quality of a synthesized speech is not
satisfactory.
In order to further improve the quality of a synthesized speed, a system
has been proposed for synthesizing a fundamental waveform pattern by using
a predetermined number of pulses continuous to each other, in U.S. patent
application Ser. No. 878,434 assigned to the assignee of the present
invention (corresponding to JP-A-61-296398). In this case, information on
the position of each pulse is not required. However, in all of
above-mentioned speech analysis-synthesis systems, no attention is paid to
the influence of a noisy environment on telephone conversations, for
example, the degradation in speech quality of a telephone conversation due
to the environment containing noise, for example, from the fan of an air
conditioner. According to the conventional speech analysis-synthesis
systems, noise which is introduced into the systems through a telephone in
a period when a speech pauses, is processed in the same manner as the
speech. Accordingly, a frame containing only noise is treated as a voiced
frame, and thus transmission parameters extracted from noise are sent to
the receiving side, to form a synthesized speech on the basis of the
transmission parameters. Accordingly, the synthesized speech which is
different from input noise and offensive to the ear of a listener, reaches
the ear of the listener in pause of the speech, and thus the listener
feels strange.
SUMMARY OF THE INVENTION
It is an object of the present invention to provide a speech coding system
capable of eliminating the influence of environmental noise on telephone
communication in a period when a speech pauses.
In order to attain the above object, according to an aspect of the present
invention, a period when a speech continues, is discriminated from a
period when the speech pauses, and transmission parameters are extracted
from an input speech signal at each frame during the period when the
speech continues, to form a synthesized speech on the receiving side on
the basis of the transmission parameters. Further, the period when the
speech pauses, is treated as an unvoiced frame.
In order to discriminate between a period when a speech continues and a
period when the speech pauses, of a telephone conversation made in a noisy
environment, according to another aspect of the present invention, a
speech analysis-synthesis system includes means for calculating the power
(or energy) of an input signal supplied from a telephone or calculating
the integrated value of the power (or energy) for a predetermined time
period, means for attenuating the power or the integrated value thereof at
a first attenuation rate (namely, in a first output-to-input ratio)
indicating a relatively small value, to obtain a first threshold value,
selector means for selecting and outputting a larger one of the first
threshold value and a second threshold value, means for attenuating the
output of the selector means at a second attenuation rate indicating a
relatively large value, to obtain the second threshold value, and
comparator means for comparing the output of the selector means with the
power of the input signal or the integrated value of the power. The output
of the selector means serves as a variable threshold value.
When a speech signal is supplied to the speech analysis-synthesis system,
input power increases abruptly, and the first threshold value increases in
proportion to the input power. Hence, the first threshold value is
selected by the selector means, and is then compared with the input power
or the integrated value thereof. When the first and second attenuation
rates are appropriately set, the input power exceeds the variable
threshold value for a period when a speech continues, and is smaller than
the variable threshold value for a period when the speech pauses. Thus,
the comparator means can deliver a discriminating signal for
discriminating between the period when the speech continues and the period
when the speech pauses. When the second attenuation rate is made small,
the variable threshold value is kept relatively high even in the period
when the speech pauses, and thus the whole input noise less than the
variable threshold value is neglected. Accordingly, when the same signal
processing as performed for an unvoiced frame in the conventional system
is carried out during the output of the comparator means indicates a
period when a speech pauses, a strange synthesized speech corresponding to
input noise is never formed on the receiving side.
When the speech is again started and input power exceeds the variable
threshold value, the output of the comparator means indicates a period
when the speech continues, and ordinary processing for speech analysis and
synthesis is carried out. Further, the variable threshold value is updated
by the above input power. When the telephone conversation is completed,
the variable threshold value decreases gradually to an initial small
value.
The foregoing and other objects, advantages, manner of operation and novel
features of the present invention will be understood from the following
detailed description when read in connection with the accompanying
drawings.
BRIEF DESCRIPTION OF THE DRAWINGS
FIG. 1 is a block diagram showing an embodiment of a speech coding system
according to the present invention.
FIG. 2 is a waveform chart showing a signal which is delivered from a
telephone and contains a speech signal and noise.
FIG. 3 is a waveform chart for explaining the operation of the above
embodiment applied with, for example, the signal of FIG. 2.
DESCRIPTION OF THE PREFERRED EMBODIMENT
FIG. 1 is a block diagram showing an embodiment of a speech coding system
according to the present invention.
Referring to FIG. 1, a digitized speech signal is applied to a speech
analyzer 2 and a power calculator 3 through an input terminal 1. The power
calculator 3 calculates input power at each frame. For example, in a case
where the speech signal of one frame is composed of n sampled data, the
power calculator 3 calculates an average value by dividing the sum of
squares of n data by n. In the present embodiment, in order to stabilize
the circuit operation, the average values at a plurality of frames are
integrated by an integrator 4 with leakage (LPF). The output of the
integrator 4 is applied to a first attenuator 5 having a predetermined
level attenuation rate. The first attenuator 5 is formed of a multiplier
for multiplying the output S'.sub.v of the integrator 4 by, for example,
an coefficient 0.5. Thus, the output level of the integrator 4 is reduced
to one-half thereof. The output of the first attenuator 5 is applied to an
input terminal of a selector 6 for delivering a variable threshold value,
which is to be compared with the output of the integrator 4. The output of
the selector 6 is applied to a delay circuit 7, which is formed of a
buffer memory for storing the output of the selector 6 only for the period
of one frame. That is, the delay circuit 7 delays the output of the
selector 6 by the period of one frame. The output of the delay circuit 7
is applied to a second attenuator 8 having a level attenuation rate, which
is made smaller than the level attenuation rate of the first attenuator 5.
For example, the level attenuation rate of the second attenuator 8 is made
equal to 1/10, that is, the output level of the delay circuit 7 is reduced
by nine tenths thereof. The output TH.sub.5 of the first attenuator 5 and
the output TH.sub.8 of the second attenuator 8 are applied to a comparator
9, so that they are compared with each other. It should be noted here that
the output of the delay circuit 7 may be directly supplied to one of the
input terminals of the comparator 9, without going through the second
attenuator 8, so that the outputs of the delay circuit 7 and the first
attenuator 5 are compared at the comparator 9. The selector 6 selects and
delivers the larger one of the outputs TH.sub.5 or TH.sub.8 on the basis
of the result of the comparison made by the comparator 9. The output of
the selector 6 (that is, a threshold value) and the output S'.sub.v of the
integrator 4 are applied to a comparator 10, to be compared with each
other. For example, the output of the comparator 10 is kept at a level "1"
for a period when the output S'.sub.v of the integrator 4 is not less than
the output of the selector 6, to indicate a period when a speech
continues. Further, the output of the comparator 10 is kept at a level "0"
for a period when the output S'.sub.v of the integrator 4 is less than the
output of the selector 6, to indicate a period when the speech pauses. The
output of the comparator 10 is applied to a coder 11. In the period when
the output of the comparator 10 takes the level "1" (that is, the period
when the speech continues), the coder 11 extracts transmission parameters
such as a pulse indicative of a fundamental waveform pattern and the pitch
period of the pulse, from a residual signal which is delivered from the
speech analyzer 2, to produce a voiced frame. In the period when the
output of the comparator 10 takes the level "0", the coder 11 produces an
unvoiced frame. The voiced and unvoiced frames thus obtained are
successively delivered from a coded data output terminal 12. The speech
analyzer 2 and the coder 11 mayb e the same ones as used in the
conventional systems which are described in the above-referred articles.
Each of the frames delivered from the coder 11 contains a flag for
discriminating between voiced and unvoiced frames. According to the
present embodiment, coded data delivered from the output terminal 12 is
the same as delivered in the conventional systems, except that an input
signal containing only noise whose level is smaller than a variable
threshold value (namely, the output of the selector 6), is treated as an
unvoiced frame. Accordingly, a conventional speech synthesizer can be used
for reproducing a speech signal from a voiced frame. Further, the output
of an excitation source for generating white noise is used for an unvoiced
frame. Alternatively, in order to inform the receiving side of the
background noise on the transmitting side, a coding method for the
unvoiced frame is made different from that for the voiced frame so that a
favorable signal is reproduced from the unvoiced frame.
FIG. 2 shows an example of a waveform of signals supplied to a telephone.
In FIG. 2, reference symbols S.sub.v1 and S.sub.v2 designate voice
signals, and S.sub.n noise.
FIG. 3 shows the level of output signal at various parts of the embodiment
of FIG. 1, for a case where the signal of FIG. 2 is applied to the
embodiment. In FIG. 3, reference simbols S'.sub.v1 and S'.sub.v2 designate
the outputs of the integrator 4 corresponding to the voice signals
S.sub.v1 and S.sub.v2 and S'.sub.n the output of the integrator 4
corresponding to the noise S.sub.n. Further, in FIG. 3, a waveform portion
TH.sub.5 proportional to the outputs S'.sub.v1 and S'.sub.v2 of the
integrator 4 indicates a threshold value delivered from the first
attenuator 5, gradually varying waveform portion TH.sub.8 indicates
another threshold value delivered from the second attenuator 8, and a
waveform which is composed of the waveform portions TH.sub.5 and TH.sub.8
and is expressed by a solid line, indicates a variable threshold value
delivered from the selector 6.
The variable threshold value is equal to a minimum value which is set by
the second attenuator 8, during a period prior to a time the voice signal
S.sub.v1 is applied to the present embodiment. When the voice signal
S.sub.v1 is applied to the embodiment and the output S'hd v1 of the
integrator 4 increases, the output TH.sub.5 of the first attenuator 5
which increases in proportion to the output S'.sub.v1 of the integrator 4,
serves as the variable threshold value. When the output TH.sub.5 becomes
smaller than a peak value, the output TH.sub.8 of the second attenuator 8
serves as the variable threshold value. A period T.sub.1 when the output
S'.sub.v1 or S'.sub.v2 of the integrator 4 is not less than the variable
threshold value, is judged to be a period when a speech continues. A
period other than the period T.sub.1 is judged to be a period T.sub.0 when
the speech pauses. The input speech power is far greater than noise power.
Hence, noise which is introduced into the present embodiment in a period
when a speech pauses, is neglected by comparing the noise with the
variable threshold value. Accordingly, in the period when the speech
continues, the same coding processing as in the conventional systems can
be made for a voiced frame. While, in the period when the speech pauses,
the processing for an unvoiced frame is carried out. Accordingly, in a
speech synthesizing circuit on the receiving side, a sound which is
delivered from an excitation source for generating white noise and is not
offensive to the ear of a listener, is used as a reproduced sound for the
unvoiced frame. Further, in a case where input noise is coded to form an
unvoiced frame, the reproducing operation for the unvoiced frame is made
different from that for the voiced frame so that the input noise is
reproduced on the receiving side as natural background noise.
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