Note: Descriptions are shown in the official language in which they were submitted.
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MULTI-PULSE TYPE CODING SYSTEM
BACKGROUND OF THE INVENTION
The present invention relates to a multi-pulse type
coding system and, more particularly, to a multi-pulse
type coding system for coding a speech signal at a low
bit rate (a low transmission rate).
Efficient coding of an input speech signal is
classified into two large methods. One is a spectral
coding method which codes a spectral structure of the
speech signal, and the other is a waveform coding method
which codes a waveform of the speech signal itself. The
spectral coding method is capable of transforming a speech
signal at a remarkably low bit rate, e.g., 4.8 Kb/s, but
degrates the quality of a replica speech waveform. On
the other hand, the waveform coding method is capable of
realizing a replica speech signal of relatively higher
quality. However, the coding bit rate according to the
waveform coding method is generally higher than that by
the spectral coding method.
In the waveform coding method, an input speech signal
is whitized so as to improve coding efficiency. This
whitizing operation performs flattening a spectral
structure of the speech signal. Information on the
spectral speech structure is, otherwise, required for
reproducing the speech signal. In the waveform coding
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method, generally speaking, the spectral structure of the
speech signal is transmitted by utilizing the spectral
coding method.
In the waveform coding method, when a whitized speech
signal is coded, an amount of information after coding
depènds upon a degree of whitizing. For the higher degree
of whitizing, more specifically, the amount of information
necessary for coding the whitized speech signal can be
reduced the more.
Multi-pulse type coding is known as one of more
efficient waveform coding methods. In the multi-pulse
type coding, the spectral structure of the speech signal
is expressed by a set of LPC parameters. On the other
hand, the whitized speech signal is additionally expressed
by a plurality of excitation pulses (multi-pulses) featured
by their amplitudes and their position during a frame
period. Such multi-pulse type coding is disclosed in U.S.
Patents No. 4,282,405; No. 4,472,832 and No. 4,701,954;
for example.
One subject in the multi-pulse type coding is to
reduce an arithmetic amount necessary for searching the
multi-pulses. As a solution for this subject, there is
known a method of searching the multi-pulses through
correlation calculation. In this method, the search of
the multi-pulses is performed by considering correlations
between a filtered impulse response waveform derived from
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the LPC parameters and the speech signal. Therefore,
it is necessary to determine LPC parameters in a period
sufficiently exceeding a duration time of an impulse
response. Accordingly, the LPC parameters have been
conventionally updated every 20 msecs, for example.
In order to precisely express the spectral structure
of a speech signal, it is empirically known that a shorter
period, e.g., about 5 msecs is preferable for updating the
LPC parameters. However, for the aforementioned reason,
the updating period of the LPC parameters has to be set
at about 20 msecs in the multi-pulse type coding, causing
limitation of expressiveness of the spectral structure.
As a result, the coding efficiency is limited to a coding
bit rate of about 8 Kb/s to maintain the coding quality.
Namely,when the multi-pulse type coding of a coding rate
less than 8 Kb/s is applied, the coding quality cannot be
retained but may be inferior to that by the spectral coding
method.
SUMMARY OF THE INVENTION
An object of the present invention is to provide a
multi-pulse type coding sys.em which is capable of keeping
a practically sufficient quality even when a lower bit
rate, e.g., a bit rate less than 8 Kb/s is applied for
coding.
According to the present invention, there is provided
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a multi-pulse type coding system for coding a speech signal into a
plurality of pulse signals, comprisingS means for producing one
set of parameters indicative of a spectral envelope of said speech
signal for each search frame period; means, coupled to said
parameter producing means and responsive to a plurality of sets of
parameters, each of said sets of parameters belonging to adjacent
search frame periods, for producing a plurality of sets of
interpolated parameters during each search frame period; means for
extracting a segmented speech signal from said speech signal and
for delivering said segmented speech signal in backward time
sequence, said segmented speech signal having a period
corresponding to said each search frame period; means for
filtering said segmented speech signal in accordance with a
filtering characteristic defined by said set of parameters and
said plurality of sets of interpolated parameters during said each
search frame period and for directly producing a cross-correlation
signal representative of a transition of cross-correlation between
said segmented speech signal and an impulse response defined by
said set of parameters and said sets of interpolated parameters
during said each search frame period, said filtering
characteristic being varied during said each search frame period
in accordance with a backward time sequence of said set of
parameters and said plurality of interpolated parameters, said
segmented speech signal being received in backward time sequence;
and means for generating said plurality of pulse signals in
response to said cross-correlation signal.
Accor*ing to another aspect of the invention there is
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provided a multi-pulse type coding system for coding a speech
signal into a plurality of pulse signals, comprising: LPC analyzer
means for producing one set of LPC parameters indicative of a
spectral envelope of the speech signal for each search frame
period; interpolation means responsive to said one set of LPC
parameters and another set of LPC parameters for an adjacent
search frame period for producing a plurality of interpolated LPC
parameters during said search frame period; means for receiving
said speech signal in backward time sequence; filter means for
backwardly filtering said speech signal under control of said LPC
parameters and said interpolated LPC parameters to produce cross-
correlation between said speech signal and an impulse response
defined by said LPC parameters of said interpolated LPC
parameters, said cross-correlation being representative of a
correlative transition during said each search frame period, a
filtering characteristic of said filtering means being varied
during said each search frame period in accordance with a backward
time sequence of said LPC parameters and said interpolated LPC
parameters, said speech signal being received in backward time
sequence; and search means for searching said plurality of pulse
signals in response to said cross-correlation.
BRIEF DESCRIPTION OF THE DRAWINGS
Fig. 1 is a block diagram æhowing one embodiment of a
multi-pulse type coding system according to the present invention;
Fig. 2 is a diagram for explaining a frame extraction
operation in the coding system of the present invention;
Figs. 3~a) and 3(b) are explanatory diagrams showing a
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66446-452
backward filtering processor according to the present invention;
and
Figs. 4(a) to 4(f) are diagrams for explaining the
operations of an impulse response unit in the embodiment of the
present invention;
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DESCRIPTION OF THE PREFERRED EMBODIMENTS
First of all, the summary of the present invention
will be described in the following.
According to the present invention, in a multi-pulse
type coding at a relatively low bit rate, an updating
period of a spectral envelope parameter is set to be
shorter than a frame period for searching the multi-pulses
in order to enhance the expressiveness of the spectral
envelope.
Accordingly, respective spectral envelope parameters
can be obtained for individual plural blocks in one frame
period so that the spectral envelope information can be
expressed more reliably. In other words, a low bit rate
coding for a narrow frequency band can be accomplished
while applying the multi-pulse type coding.
The embodiment of the present invention will be
described with reference to Figs. 1 to 4, hereinafter.
As shown in Fig. 1, the multi-pulse type coding
system of the presënt invention comprises an LPC analyzing
unit 1, a backward processing unit 2, a waveform coding
unit 3, a waveform decoding unit 4 and an LPC synthesizer 5O
These individual circuit components will be described in
detail in the following.
A degitized input speech signal 100 is supplied to
the LPC analyzing unit 1 and the backward processing unit 2.
The LPC analyzing unit 1 comprises a first waveform
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extractor 11 and an LPC analyzer 12. In this unit 1,
a frame period for producing LPC parameters to be
transmitted and for searching multi-pulses is set to
20 msecs. However, as shown in Fig. 2, the first waveform
extractor 11 segments the input speech signal into a period
of 30 msecs in an overlaped manner, for example, and
supplies the segmented speech signal to the LPC analyzer 120
As a result, the LPC analyzer 12 produces LPC parameters A
associated with each frame period of 20 msecs, as shown in
Fig. 2, and delivers an LPC parameter signal 101 indicative
of the LPC parameters A to a K-quantization decoder 13 in
the backward processing unit 2.
The backward processing unit 2 comprises the
K-quantization decoder 13, a K-interpolator 14, a
K- a -converter 15, a temporary memory 16, a second
waveform extractor 17 and a backward filtering processor 18.
In this unit 2, the LPC parameter signal 101 is supplied
to the K-quantization decoder 13 and the quantized LPC
parameter signal 104 delivered from the decoder 13 is
outputted to a multiplexer 23 in the waveform coder 3 to
be transmitted. Moreover, the LPC parameter signal thus
quantized and decoded is supplied from the decoder 13 to
the K-interpolator 14.
The K-interpolator 14 produces a plurality of
interpolated LPC parameters during each frame period of
20 msec on the basis of two successively supplied LPC
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parameters A. In this interpolator 14, as shown in Fig. 2,
three interpolated LPC parameters B, C and D are produced
from two adjacent LPC parameters of two adjacent frame
periods and, thus, four respective LPC parameters A, B, C
and D are obtained during each frame period. Generally,
the LPC parameters include a plurality of coefficient data
associated with respective orders and, therefore, the
interpolating calculation using a linear interpolation
method, for example, is performed for the respective
coefficient data in practice. Also, other various
interpolation methods can be applied to the interpolating
calculator and, further, it is possible to produce
interpolated LPC parameters from more than two LPC
parameters, i.e., from more than two frame periods.
The plurality of LPC parameters A, B, C and D from the
K-interpolator 14 are supplied through the K-~-converter
15 to the temporary memory 16.
Next, the backward filtering processor 18 for
equivalently producing the correlation between the speech
signal and an impulse response associated with LPC
parameters will be described, hereinafter.
The first step of the multi-pulse search is to
determined the correlation between an impulse response
of a LPC synthesizing filter, which is based upon the
result of the LPC analysis of the input speech signal, and
the input~speech signal. For this first step, there are
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calculated products between the value of a certain time
of the input speech signal and the values of the individual
points (i.e., slots split from a predetermined block) of
the predetermined block of the impulse response signal of
the filter, which are constructed on the basis of the
result of the LPC analysis of the input speech signal.
For each of these products, a sum is calculated of the
predetermined block. This sum is the correlation signal
between the input speech signal and the impulse response.
Conventionally, the aforementioned calculation requires a
great deal of an amount of arithmatic operation. Moreover,
if the coefficient of the LPC synthesizing filter is
frequently updated during the impulse response, the
calculation to obtain the impulse response should be done
at 160 sample points to compute the correlation during one
frame period, in a case the sampling frequency of 8 KHz
and the frame period of 20 msecs are applied. Therefore,
the arithmetic amount further increases. This increase
of the arithmetic amount is a cause for disabling the
search period of the LPC parameters to become shorter
than the frame period in the prior art. This problem is
solved in the present invention by using a filtering
operation instead of using the impulse response to compute
the correlation.
It is assumed that the impulse response of a LPC
synthesizing filter is indicated by Ii (i = 0, 1, 2, ---),
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the output at a time point j corresponding to the filter
input "1" at a time points j-k is expressed by Ik, and
the output corresponding to a filter input Sk is expressed
by Ik- Sk. When the filter inputs So, Sl, S2, ---, Sk/
and so on are applied at time points j, j-l, j-2, ---, j-k,
and so on, the filter output Bj at the time point j is
expressed by the following formula (1):
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Bj = ~ I~-S~ ..., ,,. (1)
This formula (1) implies that the correlation between
the speech waveform samples S0, Sl~-S2~ ---, Sk, and
so on and the filtered im ulse response Ii can be
determined as an output of a IIR filter. In this case,
the input order of the speech waveform samples to the
filter is directed backward, i.e., from a future sample
to a past sample. Further, it is quite apparent according
to this method that the filter output Bj_l at the time
point j-l is outputted continuously as a filter output
after the output Bj and that the arithmetic amount does
not increase even if filter coefficients are updated
midway.
Referring back to Fig. 1, the temporary memory 16
stores the LPC parameters including the interpolated
parameters. The LPC parameters 103 for each frame period
are read out in the reverse sequence order, as shown in
Fig. 3(a), from the memory 6 and supplied to the backward
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filtering processor 18 and to an impulse response
arithmetic circuit 24 and an autocorrelation arithmetic
circuit 25 in the waveform coding unit 3.
In response to the input digitized speech signal 100,
on the other hand, a second waveform extractor 17 extracts
each segmented signal of the prame period of 20 msecs,
as shown in Fig. 2, in synchronism with the operation of
the first waveform extractor 11. In this case, the
segmented speech signal is delivered from the extractor 17
to the backward filtering processor 18 in the reverse time
direction in synchronism with the operation of the
processor 18.
The backward filtering processor 18 is constructed,
as shown in Fig. 3, of an LPC synthesizing filter which is
controlled by the LPC parameters 103 for each frame period.
As described above, the LPC parameters 103 are inputted in
the backward manner (i.e., while having the leading and
- trailing ends of the signal reversed). On the other hand,
the input speech s~gnal for each frame period delivered
from the second waveform extractor 17 is inputted in the
backward manner to the backward filtering processor 18.
Here, the relation between the LPC parameters A, B, C and D
during one frame period and the input speech signal of one
frame period are shown in Fig. 3(a). In this way, a
correlation signal 102 representative of the correlation
between the impulse response of the LPC synthesizing filter
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and the input speech signal is obtained for each frame
period and supplied to a temporary memory 19 in the
waveform coding unit 3.
Next, this waveform coding unit 3 will be described
in the following. This coding unit 3 is composed of the
temporary member 19, a maximum value searching circuit 20,
an amplitude normalizer 21, a pulse quantizer 22, the
- multiplexer 23, the impuse response arithmetic circuit 24,
the autocorrelation arithmetic circuit 25 and a
compensator 26.
When the correlation signal 102 of one frame is
stored in the temporary memory 19, as shown in Fig. 4(a),
it is supplied to the maximum value searching circuit 20,
in which the amplitude and the position in the frame period
associated with the maximum value of the correlation signal
is searched, as shown in Fig. 4(b). As a result, a
position signal 117 is supplied to the impulse response
arithmetic circuit 24, the autocorrelation arithmetic
circuit 25 and the'compensator 26, and an amplitude
signal 116 is supplied to the amplitude normalizer 21.
The impulse response arithmetic circuit 24 receives
the LPC parameters 103 shown in Fig. 4(d), and the
position signal 117, in the normal (forward) order, as
indicated by an arrow in Fig. 4(e), so that the impulse
response of the corresponding LPC synthesizing filter is
calculated. The autocorrelation arithmetic circuit 25
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receives the LPC parameters-103 shown in Fig. 4(d), the
impulse response signal obtained by the impulse response
arithmetic circuit 24 and shown in Fig. 4(c), and the
position signal 117 in the backward order as shown in
Fig. 4(f), and the autocorrelation is calculated under
backward processing of the autocorrelation filter and
the position signal 117, so that the autocorrelation
signal is obtained and it is supplied to the amplitude
normalizer 21 and the compensator 26.
On the other hand, the amplitude signal 116 and
the autocorrelation signal are supplled to the amplitude
normalizer 21. In the amplitude normalizer 21, the
amplitude signal 116 is normalized such that the maximum
value of the autocorrelation signal becomes equal to the
quantized and decoded amplitude of the amplitude signal
116, and supplied to the pulse quantizer 22 and the
compensator 26. The normalized amplitude signal and
the position signal 117 are quantized in the pulse
quantizer 22. Moreover, the multi-pulse signal 111 which
shows the maximum pulse position and its amplitude is
supplied to the multiplexer 23.
The autocorrelation signal delivered from the
autocorrelation arithmetic circuit 25, the quantized and
decoded amplitude signal delivered from the amplitude
normalizer 21, and the position signal 117 are supplied
to the compensator 26. As a result, this corrector 26
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generates an autocorrelation signal in which the maximum
amplitude and the position on the frame period are
determined upon the reception of those signals.
On the other hand, the correlation signal stored in
the temporary memory 19 is read out to the compensator 26,
and the aforementioned autocorrelation having the same
maximum amplitude and the same position is subtracted from
that correlation signal and the result is returned to the
temporary memory 19. Next, the correlation signal stored
in the temporary memory 19 is read out and supplied to the
maximum value searching circuit 20 so that the multi-pulse
signal having the second maximum amplitude is obtained
from the maximum value search circuit 20. This procedure
is continued until the number of the multi-pulses reaches
15- a predetermined value or until an amplitude of a detected
pulse becomes smaller than a predetermined amplitude, so
that the multi-pulse signal lll indicative of a plurality
of multi-pulses is completely inputted to the
- multiplexer 23.
The multiplexer 23 recelves the LPC parameter signal
104 and the multi-pulse signal 111 and multiplexes them.
The resultant multiplexed signal 105 is outputted from the
multiplexer 23 and transmitted to the waveform decoding
unit 4 through a transmission line.
Next, the waveform decoding unit 4 will be described
in the following. The waveform decoding unit 4 is composed
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of a demultiplexer 31, a pulse decoder 32, a K-decoder 33,
a K-lnterpolator 34, and a K-~-converter 35. When a
multiplexed signal 105 is inputted to the demultiplexer 31
from the waveform decoding unit 3, the demultiplexer 31
outputs both an LPC parameter signal 114 corresponding to
the LPC `parameter signal 104 and a multi-pulse signal 121
- corresponding to the multi-pulse si,gnal 111.
The LPC parameter signal 114 delivered from the
demultiplexer 31 is decoded by the K-decoder so that the
decoded signal is inputted to the K-interpolator 34. This
K-interpolator 34 interpolates the LPC parameter signal of
one frame like the aforementioned K-interpolator 14 so
that the representative LPC parameter signal is converted
by the K-~-converter into a converted LPC parameter signal ~-
107 and supplied to the LPC synthesizer 5. On the other
hand, the multi-pulse signal 121 delivered from the
demultiplexer 31 is decoded by the pulse decoder 32 into
a decoded multi-pulse signal 116, which is then outputted
to the LPC synthesizer 5.
The multi-pulse signal 106 is inputted to the LPC
synthesizer 5 and controlled in accordance with the LPC
parameter signal 107 so that a decoded outputdigitieed speech
signal 108 is outputted.
In the embodiment, a plurality of the LPC parameters
during one frame period are produced by interpolating the
two LPC parameters of adjacent frame periods so as to
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enhance the expressiveness of the spectral envelope of
the input speech signal.
Otherwise, it is also possible to obtain a plurality
- of the LPC parameters during one frame period by
accomplishing a plurality of LPC analyses in one frame
period. In this case, high speed arithmetic operation
is required in circuit components such as the waveform
extractor and the LPC analyzer of Fig. 1. Therefore,
when this alternative method is applied, in the block
diagram of Fig. 1 showing the structure of the embodiment,
the K-interpolators 13 and 34 can be omitted and their
input and output terminals are connected directly.
However, the LPC analyzing unit 1 has to accomplish the
LPC analysis once during each of segmented portions
provided by dividing one frame period to obtain a plurality
of the LPC parameters for one frame period. Further, the
operations are similar to those of the embodiment except
that the LPC parameter signal to pass through the
K-quantization decoder 13, the multiplexer 23, the
demultiplexer 31 and .he K-decoder 33 experiences several
updation for the one frame period.
As has been described in detail hereinbefore,
according to the present invention,when the speech signal
is to be coded into the multi-pulses, in order to attain
accurate spectral envelope information, a plurality of
the LPC parameters are produced for one frame period.
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As a result, a low bit rate coding is realized while
keeping practically efficient coding quality.