Note: Descriptions are shown in the official language in which they were submitted.
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SPEECH ENCODER WITH FEATURES EXTRACTED
FROM CURRENT AND PREVIOUS FRAMES
BACKGROUND OF THE INVENTION:
This invention relates to a speech encoder device
for encoding a speech or voice signal at a short frame
period into encoder output codes having a high code
quality.
A speech encoder device of this type is described
as a speech codec in a paper contributed by Kazunori
Ozawa and five others including the present sole inventor
to the IEICE Trans. Commun. Volume E77-B, No. 9
(September 1994), pages 1114 to 1121, under the title of
"M-LCELP Speech Coding at 4 kb/s with Multi-Mode and
Multi-Codebook". According to this Ozawa et al paper, an
input speech signal is encoded as follows.
The input speech signal is segmented or divided
into original speech frames, each typically having a
frame period or length of 40 ms. By LPC (linear
predictive coding), extracted from the speech frames are
spectral parameters representative of spectral
characteristics of the speech signal. Before so
calculating feature or characteristic quantities, it is
preferred to convert the original speech frames to
weighted speech frames by using a perceptual or auditory
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2
weight. The feature quantities are used in deciding
modes of segments, such as vowel and consonant segments,
to produce decided mode results indicative of the modes.
In an encoding part of this Ozawa et al encoder
device, each original frame is subdivided into original
subframe signals, each being typically 8 ms long. Such
speech subframes are used in deciding excitation signals.
In accordance with the modes, adaptive parameters (delay
parameters corresponding to pitch periods and gain
parameters) are extracted from an adaptive codebook for
each current speech subframe based on a previous
excitation signal. In this manner, the adaptive codebook
is used in extracting pitches of the speech subframes
with prediction. For a,residual signal obtained by pitch
prediction, an optimal excitation code vector is selected
from a speech codebook (vector quantization codebook)
composed of noise signals of a predetermined kind. The
excitation signals are quantized by calculating an
optimal gain.
The excitation code vector is selected so as to
minimize an error power between the residual signal and a
signal composed of selected noise signal. Either for
transmission to a speech decoder device or storage in a
recording device for later reproduction, a multiplexer is
used to produce an encoder device output signal into
which multiplexed are the mode results and indexes
indicative of the adaptive parameters including the gain
parameters and the kind of optimal excitation code
21 b7552
3
vectors.
In a conventional speech encoder device of Ozawa
et al, it is necessary on reducing a processing delay to
use a short frame period for the original or the weighted
speech frames. The feature quantities are subjected to
considerable fluctuations with time when the frame period
is 5 ms or shorter. The fluctuations give rise to
unstable and erroneous interswitching of the modes and
therefore in a deteriorated code quality.
Moreover, selected modes, predicted pitches, and
extracted levels are subjected to appreciable fluctua-
tions when the frame period is 5 ms or shorter. The
appreciable fluctuations give rise, not.only to the
unstable and erroneous interswitching, but also to
unstable and erroneous pitch extraction and level
extraction and accordingly to a deteriorated code
quality.
When the levels of the input speech signal are
used on encoding the input speech signal, indexes
indicative of the levels are additionally used in the
encoder device output signal. When the pitches are used,
the encoder device output signal need not include the
indexes indicative of the pitches.
SUMMARY OF THE INVENTION:
In view of the foregoing, it is an object of the
present invention to provide a speech encoder device
operable with a short processing delay even when an input
speech signal is segmented into original speech frames of
CA 02167552 2000-02-04
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4
a short frame period, such as 5 to 10 ms long or shorter.
It is another object of this invention to provide a
speech encoder device which is of the type described and which
can prevent feature quantities from being subjected to
appreciable fluctuations with time.
It is still another object of this invention to
provide a speech encoder device which is of the type described
and which can exactly decide modes for the original frames or
for weighted frames.
It is yet another object of this invention to provide
a speech encoder device which is of the type described and
which can exactly extract pitches from speech subframes.
It is further object of this invention to provide a
speech encoder device which is of the type described to produce
encoder output codes of a high code quality.
Other objects of this invention will become clear as
the description proceeds.
CA 02167552 2000-02-04
64768-331
In accordance with an aspect of this invention, there
is provided a speech signal encoder device comprising
segmenting means for segmenting an input speech signal into
original speech frames at a predetermined frame period
5 including a current speech frame and a previous speech frame at
each current frame and at least one frame period prior to said
each current frame, weighting means for perceptually weighting
said original speech frames into weighted speech frames,
deciding means for using said weighted speech frames in
deciding a predetermined number of modes of said weighted
speech frames to produce decided mode results, and encoding
means, for encoding said input speech signal into codes at said
frame period and in response to said modes to produce said
decided mode results and said codes as an encoder device output
signal, wherein said deciding means makes use, in deciding a
current mode of said modes for said current speech frame, of
rates of variation with time in feature quantities of at least
one kind extracted from said current speech frame and from said
previous speech frame.
In accordance with another aspect of this invention,
there is provided a speech signal encoder device comprising (a)
segmenting means for segmenting an input speech signal into
original speech frames at a predetermined frame period (b)
deciding means for using the original speech frames in deciding
a predetermined number of modes of the original speech frames
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6
to produce decided mode results, (c) extracting means for
extracting pitches from the input speech signal, and (d)
encoding means for encoding the input speech signal into codes
at the frame period and in response to the modes to produce the
decided mode results and the codes as an encoder device output
signal, wherein: (A) the extracting means comprises: (A1)
feature quantity extracting means for extracting feature
quantities by using at least each current speech frame
segmented from the input speech
21 b7552
signal at the frame period; and (A2) feature quantity
adjusting means for using the feature quantities as the
pitches to adjust the pitches into adjusted pitches in
response to each current mode decided for the current
speech frame and a previous mode decided at least one
frame period prior to the current mode; (B) the encoding
means encoding the input speech signal into the codes in
response further to the adjusted pitches.
In accordance with still another different aspect
of this invention, there is provided a speech signal
encoder device comprising (a) segmenting means for
segmenting an input speech signal into original speech
frames at a predetermined frame period, (b) deciding
means for using the original speech frames in deciding a
predetermined number of modes of the original speech
frames to produce decided mode results, (c) extracting
means for extracting levels from the input speech signal,
and (d) encoding means for encoding the input speech
signal into codes at the frame period and in response to
the modes to produce the decided mode results and the
codes as an encoder device output signal, wherein: (A)
the extracting means comprises: (A1) feature quantity
extracting means for extracting feature quantities by
using at least each current speech frame segmented from
the input speech signal at the frame period; and (A2)
feature quantity adjusting means for using the feature
quantities as the levels to adjust the levels into
adjusted levels in response to each current mode decided
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8
for the current speech frame and a previous mode decided
at least one frame period prior to the current mode; (B)
the encoding means encoding the input speech signal into
the codes in response further to the adjusted levels.
BRIEF DESCRIPTION OF THE DRAWING:
Fig. 1 is a block diagram of a speech signal
encoder device according to a first embodiment of the
instant invention;
Fig. 2 is a block diagram of a mode decision
circuit used in the speech signal encoder device
illustrated in Fig. 1;
Fig. 3 is a block diagram of another mode
decision circuit for use in a speech signal encoder
device according to a second embodiment of this
invention;
Fig. 4 is a block diagram of a pitch extracting
circuit for use in a speech encoder device according to a
third embodiment of this invention;
Fig. 5 is a block diagram of a speech signal
encoder device according to a fourth embodiment of this
invention;
Fig. 6 is a block diagram of a speech signal
encoder device according to a fifth embodiment of this
invention;
Fig. 7 is a block diagram of a mode decision
circuit used in the speech signal encoder device
illustrated in Fig. 6;
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9
Fig. 8 is a block diagram of another mode
decision circuit for use in the speech signal encoder
device shown in Fig. 6;
Fig. 9 shows in blocks a feature quantity
calculator used in the mode decision circuit depicted in
Fig. 8;
Fig. 10 shows in blocks another feature quantity
calculator used in the mode decision circuit depicted in
Fig. 8;
Fig. 11 shows in blocks a different feature
quantity calculator for use in place of the feature
quantity calculator illustrated in Fig. 10;
Fig. 12 is a block diagram of still another mode
decision circuit for use in the speech signal encoder
device shown in Fig. 6;
Fig. 13 shows a feature quantity calculator used
in the mode decision circuit depicted in Fig. 12;
Fig. 14 shows in blocks a different feature
quantity calculator for use in place of the feature
quantity calculator illustrated in Fig. 12;
Fig. 15 is a block diagram of yet another mode
decision circuit for use in the speech encoder device
shown in Fig. 6;
Fig. 16 is a block diagram of a speech signal
encoder device according to a sixth embodiment of this
invention;
Fig. 17 is a block diagram of a pitch extracting
circuit used in the speech signal encoder device
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1~
illustrated in Fig. 16;
Fig. 18 shows in blocks an additional feature
quantity calculator used in the pitch extracting circuit
depicted in Fig. 17;
Fig. 19 is a block diagram of another pitch
extracting circuit for use in the speech signal encoder
device illustrated in Fig. 16;
Fig. 20 shows in blocks another additional
feature quantity calculator for use in the pitch
extracting circuit depicted in Fig. 17;
Fig. 21 is a block diagram of still another pitch
extracting circuit for use in the speech signal encoder
device illustrated in Fig. 16;
Fig. 22 shows in blocks an additional feature
quantity calculator used in the pitch extracting circuit
depicted in Fig. 21;
Fig. 23 is a block diagram of yet another pitch
extracting circuit for use in the speech signal encoder
device illustrated in Fig. 16;
Fig. 24 shows in blocks an additional feature
quantity calculator used in the pitch extracting circuit
depicted in Fig. 23;
Fig. 25 is a block diagram of a speech signal
encoder device according to a seventh embodiment of this
invention;
Fig. 26 is a block diagram of an RMS extracting
circuit used in the speech signal encoder device
illustrated in Fig. 25;
21 b7 552
Fig. 27 is a block diagram of another RMS
extracting circuit for use in the speech signal encoder
device illustrated in Fig. 25;
Fig. 28 is a block diagram of still another RMS
extracting circuit for use in the speech signal encoder
device illustrated in Fig. 25;
Fig. 29 is a block diagram of yet another RMS
extracting circuit for use in the speech signal encoder
device illustrated in Fig. 25; and
Fig. 30 is a block diagram of a further RMS
extracting circuit for use in the speech signal encoder
device illustrated in Fig. 25.
DESCRIPTION OF THE PREFERRED EMBODIMENTS:
Referring to Fig. 1, a speech signal encoder
device is according to a first preferred embodiment of
the present invention. An input speech or voice signal
is supplied to the speech signal encoder device through a
device input terminal 31. The speech signal encoder
device comprises a multiplexes (MUX) 33 for delivering an
encoder output signal to a device output terminal 35.
Delivered through the device input terminal 31,
the input speech signal is segmented or divided by a
frame dividing circuit 37 into original speech frames at
a frame period which is typically 5 ms long. A subframe
dividing circuit 39 further divides each original speech
frame into original speech subframes, each having a
subframe period of, for example, 2.5 ms.
,. _
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12
Although connected in Fig. 1 to the frame
dividing circuit 37, a spectral parameter calculator 41
calculates spectral parameters of the input speech signal
up to a predetermined order, such as up to a tenth order
(P = 10) by applying a window of a window length of
typically 24 ms to at least one each of the speech
subframes. In the example being illustrated; the
spectral parameter calculator 41 calculates the spectral
parameters according to Burg analysis described in a book
written by Nakamizo and published 1988 by Korona-Sya
under the title of, as transliterated according to ISO 1
3602, "Singo Kaiseki to Sisutemu Dotei" (Signal Analysis
and System Identification), pages 82 to 87. It is
possible to use an LPC analyzer or a like as the spectral
parameter calculator 41.
Besides calculating linear prediction coeffici-
ents a (i) by the Burg analysis for i = 1, 2, ..., and
10, the spectral parameter calculator 41 converts the
linear prediction coefficients to LSP (linear spectral
pair) parameters which are suitable to guantization and
interpolation. In the spectral parameter calculator 4I
being illustrated, the linear prediction coefficients are
converted to the LSP parameters according to a paper
contributed by Sugamura and another to the Transactions
of the Institute of Electronics and Communication
Engineers of Japan, J64-A (1981), pages 599 to 606, under
the title of "Sen-supekutoru Tui Onsei Bunseki Gosei
Hosiki ni yoru Onsei Zyoho Assyuku" (Speech Data
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13
Compression by LSP Speech Analysis-Synthesis Technique,
as translated by the contributors).
More particularly, each speech frame consists of
first and second subframes in the example being
described. The linear prediction coefficients are
calculated and converted to the LSP parameters for the
second subframe. For the first subframe, the LSP
parameters are calculated by linear interpolation of the
LSP parameters of second subframes and are inverse
converted to the linear prediction coefficients. In this
manner, the spectral parameter calculator 41 produces LSP
parameters and linear prediction coefficients a (i, p)
for the first and the second subframes where p = 1, 2,
..., and 5.
Supplied from the spectral parameter calculator
41 with the LSP parameters of each predetermined
subframe, such as the second subframe, a spectral
parameter quantizer 43 converts the linear prediction
coefficients to converted prediction coefficients
a '(i, p) for each subframe. Furthermore, the spectral
parameter quantizer 43 vector quantizes the linear
prediction coefficients.
To speak of this vector quantization first, it is
possible to use various known methods. An example is
described in a paper contributed by Toshiyuki Hamada and
three others to the Proc. Mobile Multimedia Communica-
tions, pages B.2.5-1 to B.2.5-4 (1933), under the title
of "LSP Coding Using VQ-SVQ with Interpolation in 4.075
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14
kbps M-LCELP Speech Coder". Other examples are disclosed
in Japanese Patent Prepublication (A) Nos. 171,500
of 1992, 363,000 of 1992, and 6,199 of 1993. In the
example being illustrated, use is made of an LSP codebook
45.
As for conversion into the converted prediction
coefficients, the spectral parameter quantizer 43 first
reproduces the LSP parameters for the first and the
second subframes from the LSP parameters quantized in
connection with each second subframe. In practice, the
LSP parameters are reproduced by linear interpolation
between the quantized prediction coefficients of a
current one of the second subframes and those of a
previous one of the second subframes that is one frame
period prior to the current one of the second subframes.
More in detail, the spectral parameter quantizer
43 is operable as follows. First, a code vector is
selected so as to minimize an error power between the LSP
parameters before and after quantization and then
reproduces by linear interpolation the LSP parameters for
the first and the second subframes. In order to achieve
a high quantization efficiency, it is possible to
preselect a plurality of code vector candidates for
minimization of the error power, to calculate cumulative
distortions in connection with the candidates, and to
select one of combinations of interpolated LSP parameters
that minimizes the cumulative distortions.
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Alternatively, it is possible instead of the
linear interpolation to prepare interpolation LSP
patterns for a predetermined number of bits, such as two
bits, and to select one of combinations of the inter-
polation LSP patterns that minimizes the cumulative
distortions as regards the first and the second
subframes. This results in an increase in an amount of
output information although this makes it possible to
more exactly follow variations of the LSP parameters in
each speech frame.
It is possible either to prepare the interpola-
tion LSP patterns by learning of LSP data for training or
to store predetermined patterns. For storage, the
patterns may be those described in a paper contributed by
Tomohiko Taniguchi and three others to the Proc. ICSLP
(1992), pages 41 to 44, under the title of "Improved CELP
Speech Coding at 4 kbit/s and below". Alternatively, it
is possible for further improved performance to preselect
the interpolation LSP patterns, to calculate an error
signal between actual values of the LSP parameters and
interpolated LSP values, and to quantize the error signal
with reference to an error codebook (not shown).
The spectral parameter quantizer 43 produces the
converted prediction coefficients for the subframes. In
addition, the spectral parameter quantizer 43 supplies
the multiplexer 33 with indexes indicative of the code
vectors selected for quantized prediction coefficients in
connection with the second subf rames.
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16
Connected to the subframe dividing circuit 39 and
to the spectral parameter calculator and quantizer 41 and
43, a perceptual weighting circuit 47 gives perceptual or
auditory weights 7 i to respective samples of the speech
subframes to produce a perceptually weighted signal
x[w](n), where n represents sample identifiers of the
respective speech samples in_each frame. The weights are
decided primarily by the linear prediction coefficients.
Supplied with the perceptually weighted signal
frame by frame, a mode decision circuit 49 extracts
feature quantities from the perceptually weighted signal.
Furthermore, the mode decision circuit 49 uses the
feature quantities in deciding modes as regards frames of
the perceptually weighted signal to produce decided mode
results indicative of the modes.
Turning temporarily to Fig. 2 with Fig. 1 continu-
ously referred to, the mode decision circuit 49 is
operable as follows in the speech encoder device being
illustrated. The mode decision circuit 49 has mode
decision circuit input and output terminals 49(I) and
49(O) supplied with the perceptually weighted signal and
producing the decided mode results.
Supplied through the mode decision circuit input
terminal 49(I) with the perceptually weighted signal
frame by frame, a feature quantity calculator 51 calcu-
laces in this example a pitch prediction gain G. A frame
delay (D) 53 is for giving one frame delay to the pitch
prediction gain to produce a one-frame delayed gain. A
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17
weighted sum calculator 55 calculates a weighted sum Gav
of the pitch prediction gain and the one-frame delayed
gain according to:
2
Gav = ~ v (i)G(i),
i=1
where v (i) represents gain weights for i-th subframes.
The feature quantities are given typically by
such weighted sums in connection with each current frame
and a previous frame which is one frame period prior to
the current frame. Supplied with the feature quantities,
a mode decision unit 57 selects one of the modes for each
current frame and delivers the decided mode results in
successive frame periods to the mode decision circuit
output terminal 49(0).
The mode decision unit 57 has a plurality of
predetermined threshold values, for example, three in
number. In this event, the modes are four in number.
The decided mode results are delivered to the multiplexer
33.
In Fig. 1, the spectral parameter calculator and
quantizer 41 and 43 supply a response signal calculator
59 with the linear prediction coefficients subframe by
subframe and with the converted prediction coefficients
also subframe by subframe. The response signal
calculator 59 keeps filter memory values for respective
subframes. In response to a response calculator input
signal d(n) which will presently become clear, the
response signal calculator 59 calculates a response
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signal x[z](n) for each subframe according to:
x[z] (n) - d(n)
a (i)d(n - i)
i=1
10
+ ~ a (i) 7 i y(n - i)
i=1
10
+ ~ a '(i)7 i x[z](n - i),
i=1
where:
y(n) - d(n) - ~ a (i)d(n - i)
i=1
10
+ ~ a (i) y i y(n - i) .
i=1
Connected to the perceptual weighting circuit 47
and to the response signal calculator 59, a speech
subframe subtracter 61 subtracts the response signal from
the perceptually weighted signal to produce a subframe
difference signal according to:
x [wl ' (n) - x f w] (n) - x [ z 1 (n) .
Connected to the spectral parameter quantizer 45, an
impulse response calculator 63 calculates, at a predeter-
mined number L of points, impulse responses h[w](n) of a
weighted filter of the z-transform which is represented
as:
10 _
H[w] (z) - (1 - ~ a (i)z i)
i=1
10 _
. (1 - ~ a ' (i) 7 i z i)2 .
i=1
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19
Controlled by the modes decided by the mode
decision circuit 49 and by the impulse responses
calculated by the impulse response calculator 63, an
adaptive codebook circuit 65 is connected to the subframe
subtracter 61 and to a pattern accumulating circuit 67.
Depending on the modes, the adaptive codebook circuit 65
calculates pitch parameters and supplies the multiplexer
33 with a prediction difference signal defined by:
z(n) - x(w]'(n) - b(n),
where b(n) represents a pitch prediction signal given by:
b (n) - ~ v (n - T) * h (w] (n) ,
where, in turn, ~ represents the gain of the adaptive
codebook circuit 65, v(n) representing here an adaptive
code vector, and T representing a delay. The asterisk
mark represents convolution.
Controlled by the modes decided by the mode
decision circuit 49 and by the impulse responses
calculated by the impulse response calculator 63, an
excitation quantizer 69 is supplied with the prediction
difference signal from the adaptive codebook circuit 65
and refers to a sparse excitation codebook 71. Being of
a non-regular pulse type, the sparse excitation codebook
71 keeps excitation code vectors, each of which is
composed of non-zero vector components of an individual
non-zero number or count. The excitation quantizer 69
produces, as optimal excitation code vectors c(j](n),
either a part or all of the excitation code vectors to
minimize j-th differences defined by:
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D(j) - ~ fz(n) - 7 (j)cIj) (n)hlwl (n))2.
n
Controlled by the impulse responses calculated by
the impulse response calculator 63 and supplied with the
prediction difference signal from the adaptive codebook
circuit 65 and with the excitation code vectors selected
by the excitation quantizer 69, a gain quantizer 73
refers to a gain codebook 75 of gain code vectors.
Reading the gain code vectors, the gain quantizer 73
selects combinations of the excitation code vectors and
the gain code vectors so as to minimize (j,k)-th
differences defined by:
D(j,k) - ~ fxlw] (n)
n
- ~3 ' (k)v(n - T)kfw] (n)
- y ' (k) c f j 1 (n) h fwl (n) ) 2.
where ~ '(k) and y '(k) represent a k-th two-dimensional
code vector of the gain code vectors. Selecting the
combinations, the gain quantizer 73 supplies the
multiplexer 33 with the indexes indicative of the
excitation and the gain code vectors of such selected
combinations.
In the Ozawa et al paper cited heretobefore, the
excitation quantizer 69 selects at least two kinds, such
as for an unvoiced and a voiced mode, of optimal exci-
tation code vectors. In the example being illustrated,
the gain quantizer 73 selects the optimal code vectors
produced by the excitation quantizer 69 under control by
the modes. It is possible upon selection by the gain
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21
quantizer 73 to specify the optimal excitation code
vectors of a single kind. Alternatively, it is possible
on applying the above-described equation for the j-th
differences D(j) only to a part of the excitation code
vectors to preliminarily select excitation code vector
candidates for application of the equation in question to
the excitation code vector candidates, to select the
optimal code vectors of only one kind from the excitation
code vector candidates.
Connected to the spectral parameter calculator
and quantizer 41 and 43 and to the gain quantizer 73, a
weighting signal calculator 77 reads the excitation and
the gain code vectors with reference to their indexes
and calculates a drive excitation signal v(n) according
to:
v(n) - ~ ' (n)v(n - T) + y ' (k)c[ j1 (n) .
Subsequently, the weighting signal calculator 77
calculates a weighting signal s[wl(n) for delivery to the
response signal calculator 59 according to:
s [w] (n) - v (n)
10
- ~ a (i)v(n - i)
i=1
10
+ ~ a (i) 7 i p(n - i)
i=1
10
+ ~ a ' (i) 7 i s [wl (n - i) ,
i=1
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22
where:
p(n) - v(n) - ~ . a (i)v(n - i)
i=1
+ E a (i) 7 i p(n - i)
i=1
It is now understood in connection with the
example being illustrated that the modes are decided
either for each original speech frame or for each
weighted speech frame by the feature quantities extracted
from the input speech signal for a longer period which
is longer than one frame period. Even if the frame
period is only S ms long or shorter and if the feature
quantities may be erroneous when extracted from the
current speech frame alone, the previous speech frame
would give correct and precise feature quantities when
the previous speech frame is at least one frame period
prior to the current speech frame. As a consequence, it
is possible for unstable and erroneous interswitching of
the modes to prevent the code quality from deteriorating.
Referring to Fig. 3 with Figs. 1 and 2 continu-
ously referred to, another mode decision circuit is for
use in a speech signal encoder device according to a
second preferred embodiment of this invention.
Throughout the following, similar parts are designated by
like reference numerals and are similarly operable with
likewise named signals unless specifically otherwise
mentioned. This mode decision circuit is therefore
designated by the reference numeral 49. Except for the
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mode decision circuit 49 which will be described in the
following, the speech signal encoder device is not
different from that illustrated with reference to Fig. 1.
In the mode decision circuit 49 being illust-
rated, the frame delay 53 is connected directly to the
mode decision circuit input terminal 49(I). Supplied
from the perceptual weighting circuit 47 with the
perceptually weighted signal through the mode decision
circuit input terminal 49(I), the frame delay 53 produces
a delayed weighted signal with a one-frame delay.
Connected to the frame delay 53 and to the mode
decision circuit input terminal 49(I), the feature
quantity calculator 51 calculates a pitch prediction gain
G for each speech frame as the feature quantities. The
pitch prediction gain is calculated according to:
G = 10 1og10(P/E),
where:
N-1
P = ~ x[w] 2 (n)
n=-N+1
N-1
and E = P - [ ~ x[w] (n)x[w] (n - T) ]2
n=-N+1
N-1
. [ ~ x[w12(n - T)1.
n=-N+1
where, in turn, T represents here an optimal delay that
maximizes such prediction delays, N representing a total
number of speech samples in each frame.
Connected to the feature quantity calculator 51,
the mode decision unit 57 compares the pitch prediction
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24
gain with predetermined threshold values to decide modes
of the input speech signal from frame to frame. The
modes are delivered as decided mode results through the
mode decision circuit output terminal 49(0) to the
multiplexer 33, the adaptive codebook circuit 65, and the
excitation quantizer 69.
In the speech signal encoder device including the
mode decision circuit 49 being illustrated, mode
information is produced as an average for more than one
frame period. This makes it possible to suppress
deterioration which would otherwise take place in the
code quality.
Further turning to Fig. 4 with Figs. 1 and 2
continuously referred to, a pitch extracting circuit is
for use in a speech signal encoder device according to a
third preferred embodiment of this invention. The pitch
extracting circuit is used in place of the mode deciding
circuit 49 and is therefore designated by a similar
reference symbol 49(A). In other respects, the speech
signal encoder device is not much different from that
illustrated with reference to Fig. 1 except for the
adaptive codebook circuit 65 which is now operable as
will shortly be described.
In Fig. 4, pitch extracting circuit input and
output terminals correspond to the mode decision circuit
input and output terminals 49(I) and 49(O) described in
conjunction with Fig. 2 and are consequently designated
by the reference symbols 49(I) and 49(0). The pitch
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extracting circuit 49(A) comprises the frame delay 53
connected directly to the pitch extracting circuit input
terminal 49(I) as in the mode decision circuit 49
described with reference to Fig. 3.
Connected to the frame delay 53 and to the pitch
extracting circuit input terminal 49(I) is a pitch
calculator 79. Supplied from the perceptual weighting
circuit 47 through the pitch extracting circuit input
terminal 49(I) with the perceptually weighted signal as
an undelayed weighted signal and from the frame delay 53
with the delayed weighted signal, the pitch calculator 79
calculates pitches T (the same reference symbol being
used) which maximizes a novel error power E(T) defined
by-
N-1
E (T) - ~ x (wl 2 (n)
n=-N+1
N-1
- ( ~ x (wl (n) x (wl (n - T) 1 2
n=-N+1
N-1
. [ ~ xfwl2(n - T)l.
n=-N+1
Extracting the pitches T from the input speech
signal in this manner, the pitch extracting circuit 49(A)
delivers the pitches to the adaptive codebook circuit 65.
Although connections are depicted in Fig. 1 between the
mode deciding circuit 49 and the multiplexer 33 and
between the mode deciding circuit 49 and the excitation
quantizer 69, it is unnecessary for the pitch extracting
circuit 49(A) to deliver the pitches to the multiplexer
2167552
26
33 and to the excitation quantizer 69.
Supplied from the pitch extracting circuit 49(A)
with the pitches, the adaptive codebook unit 65 closed-
loop searches for lag parameters near the pitches in the
subframes of the subframe difference signal. Further-
more, the adaptive codebook circuit 65 carries out pitch
prediction to produce the prediction difference signal
z(n) described before.
It has been confirmed that the pitch extracting
circuit 49(A) is excellently operable. In the Ozawa et
al paper cited before, the pitches T are calculated so as
to minimize a conventional error power defined by:
N-1 N-1
E (T) - ~ x (wl 2 (n) - f ~ x Iwl (n) x (wl (n - T) 1 2
n=0 n=0
N-1
. [ ~ x(wl2(n - T) 1 .
n=0
In contrast, the pitch extracting circuit 49(A)
calculates for each original or weighted speech frame an
averaged pitch over two or more frame periods. This
avoids extraction of unstable and erroneous pitches and
prevents the code quality from being inadvertently
deteriorated.
Referring afresh to Fig. 5, a speech signal
encoder device is similar, according to a fourth
preferred embodiment of this invention, to that
illustrated with reference to Figs. 1 and 4.
Between the perceptual weighting unit 47 and the
mode decision unit 57 which is described in connection
2167552
27
with Fig. 3, use is made of a pitch and pitch prediction
gain (T & G) extracting circuit 49(B) connected to the
adaptive codebook circuit 65. Instead of the sparse
excitation codebook 71, first through N-th sparse
excitation codebooks 71(1) through 71(N) are connected to
the excitation quantizer 69.
It is possible to understand that Fig. 4 shows
also the pitch and pitch prediction gain extracting
circuit 49(B). A pitch and predicted pitch gain
extracting circuit input terminal is connected to the
perceptual weighting circuit 47 to correspond to the mode
decision or the pitch extracting circuit input terminal
and is designated by the reference symbol 49(I). A pitch
and pitch prediction gain calculator 79(A) is connected
to the frame delay 53 like the pitch gain calculator 79
and calculates the pitches T to maximize the novel error
power defined before and the pitch prediction gain G by
using the equation which is given before and in which E
is clearly equal to the novel error power. In the manner
understood from Fig. 5, the pitch and pitch prediction
gain extracting unit 49(B) has two pitch and pitch
prediction gain extracting circuit output terminals
connected to the pitch and pitch prediction gain
calculator 79(A) instead of only one pitch extracting
circuit output terminal 49(O).
One of these two output terminals is for the
pitches T and is connected to the adaptive codebook
circuit 65. The other is for the pitch prediction gain G
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28
and is connected to the mode decision circuit 49, which
uses such pitch prediction gains as the feature
quantities.
The adaptive codebook circuit 65 is controlled by
the modes and is operable to closed-loop search for the
lag parameters in the manner described above. The
excitation quantizer 69 uses either a part or all of the
excitation code vectors stored in the first through the
N-th excitation codebooks 71(1) to 71(N).
Referring now to Fig. 6, the description will
proceed to a speech signal encoder device according to a
fifth preferred embodiment of this invention. This
speech signal encoder device is similar to that
illustrated with reference to Fig. 1 except for the
following. That is, the mode decision circuit 49 is
supplied from the spectral parameter calculator 41 with
the spectral parameters a (i, p) for the first and the
second subframes besides supplied from the perceptual
weighing circuit 47 with the weighted speech subframes
x[w1(n) at the frame period.
Turning to Fig. 7 with Fig. 6 continuously
referred to, the mode decision circuit 49 has first and
second circuit input terminals 49(1) and 49(2) connected
to the perceptual weighting circuit 47 and to the
spectral parameter calculator 41, respectively.
Corresponding to the mode decision circuit output
terminal described in connection with Fig. 2, a sole
circuit output terminal is designated by the reference
2161552
29
symbol 49(0) and connected to the multiplexer 33 and to
the adaptive codebook circuit 65 and the excitation
quantizer 69.
Connected to the first circuit input terminal
49(1), a first feature quantity calculator 81 calculates
primary feature quantities, such as the pitch prediction
gains which are described before and will hereafter be
indicated by PG. Connected to the first and the second
circuit input terminals 49(1) and 49(2), a second feature
quantity calculator 83 calculates secondary feature
quantities which may be short-period or short-term
predicted gains SG.
Supplied with the primary and the secondary
feature quantities and with delayed mode information
through a frame delay 85, a mode decision unit 87 selects
one of the modes for each current frame as output mode
information like the mode decision unit 57 described in
conjunction with Fig. 2 by comparing a combination of the
primary and the secondary feature quantities and the
delayed mode information with the predetermined threshold
values of the type described before. The output mode
information is delivered to the sole circuit output
terminal 49(0) and to the frame delay 85, which gives a
delay of one frame period to supply the delayed mode
information back to the mode decision unit 87. It is
preferred that the combination of the delayed mode
information and the primary and the secondary feature
quantities should be a weighted combination of the type
Z1 b~~552
of the weighted sum Gav described in connection with
Fig. 2.
In other respects, operation of this speech
signal encoder device is not different from that
described in conjunction with Fig. 1. It is possible
with the mode decision circuit 49 described with
reference to Fig. 7 to achieve the above-pointed out
technical merits.
Referring to Fig. 8, another mode decision
circuit is for use in the speech signal encoder device
described in the foregoing and is designated again by the
reference numeral 49.
As illustrated with reference to Fig. 7, this
mode decision circuit 49 has the first and the second
circuit input terminals 49(1) and 49(2) and the sole
circuit output terminal 49(O) and comprises the first and
the second feature quantity calculators 81 and 83, the
frame delay 85, and the mode decision unit 87. Operable
in the manner described in conjunction with Fig. 7, the
first feature quantity calculator 81 delivers the pitch
prediction gains PG to the mode decision unit 87. In the
example being illustrated, the second feature quantity
calculator 83 is supplied only with the weighted speech
subframes and calculates; for supply to the mode decision
unit 87, RMS ratios RR as the secondary feature
quantities in the manner which will presently be
described. Connected to the first and the second circuit
input terminals 49(1) and 49(2) and being operable as
21 b7 552
31
will shortly be described, a third feature quantity
calculator 89 calculates, for delivery to the mode
decision unit 87, the short-period predicted gains SG and
short-period predicted gain ratios SGR collectively as
ternary feature quantities. The frame delay 85 and the
mode decision unit 87 are operable in the manner
described above.
Turning to Fig. 9 and Figs. 6 and 8 again
referred to, the second feature quantity calculator 83
comprises an RMS calculator 91 supplied with the weighted
speech subframes frame by frame through the first circuit
input terminal 49(1) to calculate RMS values R which are
used in the Ozawa et al paper. Connected to the RMS
calculator 91, a frame delay (D) 93 gives a delay of one
frame period to the RMS values to produce delayed values.
Supplied with the RMS values and the delayed values, an
RMS ratio calculator 95 calculates the RMS ratios for
delivery to the mode decision unit 87. Each RMS ratio is
a rate of variation of the RMS values with respect to a
time axis scaled by the frame period.
Further turning to Fig. 10 with Figs. 6 and 8
continuously referred to, the third feature quantity
calculator 89 comprises a short-period predicted gain
(SG) calculator 97 connected to the first and the second
circuit input terminals 49(1) and 49(2) to calculate the
short-period predicted gains for supply to the mode
decision unit 87. Although separated from the frame
delay described in conjunction with Fig. 9, a frame delay
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32
(D) is indicated by the reference numeral 93 merely for
convenience of illustration and is similarly operable to
produce delayed prediction gains which are related to the
previous frame described before. Responsive to the short-
period prediction gains and to the delayed prediction
gains, a short-period prediction gain ratio (SGR)
calculator 99 calculates the short-period predicted gain
ratios for delivery to the mode decision unit 87.
Still further turning to Fig. 11 with Figs. 6 and
8 continuously referred to, the third feature quantity
calculator 89 comprises first and second frame delays
93(1) and 93(2) in place of the frame delay 93 depicted
in Fig. 9. As a consequence, the third feature quantity
calculator 89 supplies the mode decision unit 87 with the
short-period predicted gains which are calculated by
comparing the predetermined threshold values with a sum,
preferably a weighted sum, calculated in each frame by a
short-period predicted gain and a delayed predicted gain
delivered from the first and the second frame delays
93(1) and 93(2) with a total delay of two frame periods
given to the short-period predicted gain.
Referring to Fig. 12 with Fig. 6 continuously
referred to, the mode decision circuit 49 is similar
partly to that described in connection with Fig. 8 and
partly to that of Fig. 9. More particularly, the second
feature quantity calculator 83 supplies the mode decision
unit 87 with the RMS values R in addition to the RMS
ratios RR. The first and the third feature quantity
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33
calculators 81 and 89, the frame delay 85, and the mode
decision unit 87 are operable in the manner described
before.
Turning to Fig. 13 with Fig. 12 continuously
referred to, the second feature quantity calculator 83 is
similar to that illustrated with reference to Fig. 9.
The RMS calculator 91 delivers, however, the RMS values
directly to the mode decision unit 87. In addition, the
RMS calculator 91 delivers the RMS values to the RMS
ratio calculator 95 directly and through a series
connection of first and second frame delays (D) which are
separate from those described in connection with Fig. 11
and nevertheless are designated by the reference numerals
93(1) and 93(2). It is now understood that the RMS ratio
calculator 95 calculates the RMS ratio of each current
RMS value to a previous RMS value which is two frame
periods prior to the current RMS value.
Further turning to Fig. 14 with Figs. 6 and 12
again referred to, the second feature vector calculator
83 is similar to that described with reference to Fig. 9.
The RMS calculator 91 delivers, however, the RMS values
directly to the mode decision unit 87 besides to the
frame delay 93 and to the RMS ratio calculator 95.
Referring to Fig. 15 with Fig. 6 continuously
referred to, the mode decision circuit 49 is similar to
that described with reference to Fig. 12. The second
feature quantity calculator 83 delivers, however, only
the RMS values R to the mode decision unit 87.
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34
Referring now to Fig. 16, attention will be
directed to a speech signal encoder device according to a
sixth preferred embodiment of this invention. In this
speech signal encoder device, the mode decision circuit
49 is supplied only from the perceptual weighting circuit
47 with the weighted speech subframes at the frame
period, calculates the pitch prediction gains as the
feature quantities like the first feature quantity
calculator 81 described in conjunction with Fig. 7, 8,
12, or 15, and decides the mode information of each
original speech frame for delivery to the multiplexer 33,
the adaptive codebook circuit 65, and the excitation
quantizer 69. In the example being illustrated, the mode
information is additionally used in the manner which will
be described in the following.
Connected to the perceptual weighting circuit 47,
supplied from the mode decision circuit 49 with the mode
information at the frame period, and accompanied by a
partial feedback loop 101, a pitch extracting circuit 103
calculates corrected pitches CPP in each frame period for
supply to the adaptive codebook circuit 65 as follows.
Turning to Fig. 17 with Fig. 16 continuously
referred to, the pitch extracting circuit 103 has a first
extracting circuit input terminal 103(1) connected to the
mode decision circuit 49, a second extracting circuit
input terminal 103(2) connected to the perceptual
weighting circuit 47, and a third extracting circuit
input terminal 103(3) connected to the partial feedback
2167552
loop 101. An extracting circuit output terminal 103(0)
is connected to the adaptive codebook circuit 65.
In the manner which will presently be described,
the partial feedback loop 101 feeds a current pitch CP of
each current frame to the third extracting circuit input
terminal 103(3). An additional feature quantity
calculator 105 calculates such current pitches, previous
pitches PP, and pitch ratios DR in response to the
current pitches and to the weighted speech subframes
supplied thereto at the frame period. The previous
pitches have a common delay of one frame period relative
to the current pitches. Each pitch ratio represents a
rate of variation in the current pitches in each frame
period.
Connected to the first extracting circuit input
terminal 103(1), a frame delay (D) 107 gives a delay of
one frame period to produce delayed information.
Supplied from the first extracting circuit input terminal
103(1) with the mode information, from the frame delay
107 with the delayed information, and from the additional
feature quantity calculator 105 with the current pitches,
the previous pitches, and the pitch ratios collectively
as feature quantities, a feature quantity adjusting unit
109 compares the pitch ratios with a predetermined
additional threshold value with reference to the mode and
the delayed information to adjust or correct the current
pitches by the previous pitches and the pitch ratios into
adjusted pitches CPP for delivery to the extracting
CA 02167552 2000-02-04
36
circuit output terminal I03(O).
Further turning to Fig. 18 with Figs. 16 and 17
continuously referred to, the additional feature quantity
calculator I05 comprises a pitch calculator 111 connected
to the first extracting circuit input terminal 103(2) to
f
receive the perceptually weighted speech subframes at the
frame period and to calculate the current pitches CP for
delivery to the partial feedback loop 101 and to the
feature quantity adjusting unit 109. Supplied with the
current pitches through the third extracting circuit
input terminal 103(3), a frame delay (D) 113 produces the
previous pitches PP for supply to the feature quantity
adjusting unit 109. Supplied with the current and the
previous pitches, a pitch ratio calculator 115 calculates
the pitch ratios DR for supply to the feature quantity
adjusting unit 109.
In Fig. 16, the adaptive codebook circuit 65 is
operable similar to that described in conjunction with
the speech signal encoder device comprising the pitch
calculator 79 illustrated with reference to Fig. 4. More
specifically, the adaptive codebook circuit 65 closed-
loop searches for the pitches in each previous subframe
of the subframe difference signal near the adjusted
pitches CPP rather than the lag parameters near the
pitches calculated by the pitch calculator 79.
In other respects, the speech signal encoder
device of Fig. 16 is similar to that illustrated with
reference to Fig. 6.
CA 02167552 2000-02-04
37
Referring to Fig. 19 with Fig. 16 additionally
referred to, another pitch extracting circuit is for use
in the speech signal encoder device under consideration.
This pitch extracting circuit corresponds to that
illustrated with reference to Fig. 17 and will be
designated by the reference numeral 103.
The pitch extracting circuit 103 has_only the
first and the second extracting circuit input terminals
103(1) and 103(2) and the extracting circuit output
terminal 103(0). In other words, the pitch extracting
circuit 103 is not accompanied by the partial feedback
loop 101 described in connection with Fig. 16.
Supplied from the perceptual weighting circuit 47
with the weighted speech subframes frame by frame, the
additional feature quantity calculator 105 calculates the
current pitches CP as the feature quantities. Responsive
to the mode information supplied from the mode decision
circuit 49 frame by frame and to the delayed information
produced by the frame delay 107, the feature quantity
adjusting unit 109 adjusts the current pulses into the
adjusted pitches CPP for use in the adaptive codebook
circuit 65.
Referring to Fig. 20 with Figs. 16 and 17
additionally referred to, another additional feature
quantity calculator is for use in the pitch extracting
circuit 103 accompanied by the partial feedback loop 101
and is designated by the reference numeral 105. This
additional feature quantity calculator 105 is similar to
2167552
38
that illustrated with reference to Fig. 18. In the
additional feature quantity calculator 105 being
illustrated, the frame delay 113 of Fig. 18 is afresh
referred to as a first frame delay 113(1) and delivers
the previous pitches PD to the feature quantity adjusting
unit 109.
Supplied through the second extracting circuit
input terminal 103(2) with the perceptually weighted
speech subframes at the frame period, the pitch
calculator 111 calculates the current pitches CP for
supply to the feature quantity calculating unit 109 and
to the partial feedback loop 101 and thence to the third
extracting circuit input terminal 103(3) depicted in Fig.
18. Connected in series to the first frame delay 113(1),
a second delay 113(2) gives a delay of one frame period
to the previous pitches to produce past previous pitches
PPP which have a long delay of two frame periods relative
to the current pitches. So as to deliver the pitch
ratios DR to the feature quantity adjusting unit 109, the
pitch ratio calculator 115 is operable identically with
that described in connection with Fig. 18.
Referring to Fig. 21 with Fig. 16 continuously
referred to, the pitch extracting circuit 103 is for use
in combination with the partial feedback loop 101.
Supplied with the mode information frame by frame through
the first extracting circuit input terminal 103(1), with
the perceptually weighted speech subframes frame by frame
through the second extracting circuit input terminal
CA 02167552 2000-02-04
.- ~ :;,
r
39
103(2), and with the current pitches CP through the third
extracting circuit input terminal 103(3), this pitch
extracting circuit 103 delivers the adjusted pitches CPP
to the adaptive codebook circuit 65 through the
extracting circuit output terminal 103(0).
Connected to the second and the third extracting
circuit input terminals 103(2) and 103.(3), an. additional
feature quantity calculator is similar to that described
with reference to any one of Figs. 17 through 20 and is
consequently designated again by the reference numeral
105. Responsive to the perceptually weighted speech
subframes of each frame and to the current pitches, this
additional feature quantity calculator 105 calculates the
pitch ratios DR for delivery together with the current
pitches to the feature quantity adjusting unit 109
collectively as the feature quantities. Responsive to
the mode and the delayed information, the feature
quantity adjusting unit 109 compares the pitch ratios
with the additional threshold value to adjust the current
pitches now only by the pitch ratios into the adjusted
pitches.
Turning to Fig. 22 with Figs. 16 and 21 continu-
ously referred to, the additional feature quantity
calculator 105 is similar to that illustrated with
reference to Fig. 18 or 20. The previous pitches are,
however, not supplied to the feature quantity adjusting
unit 109.
CA 02167552 2000-02-04
Referring again to Fig. 22 with Figs. 16 and 21
additionally referred to, the additional feature
calculator 105 may comprise, instead of the first and the
second frame delays 113(1) and 113(2), singly the frame
delay 113 between the third extracting circuit input
terminal 103(3) and the pitch ratio calculator 115 as in
Fig. 18 and without supply of the previous pitches to the
feature quantity adjusting unit 109.
Referring anew to Fig. 23 with Fig. 16 continu-
ously referred to, the pitch extracting circuit 103 is
not different from that of Fig. 21 insofar as depicted in
blocks. The additional feature quantity calculator 105
is, however, a little different from that described in
conjunction with Fig. 21. Accordingly, the feature
quantity adjusting unit 109 is somewhat differently
operable.
Turning to. Fig. 24 with Figs. 16 and 23 continu-
ously referred to, the additional feature quantity
calculator 105 comprises the pitch calculator lIl
supplied through the second extracting circuit input
terminal 103(2) with the perceptually weighted speech
subframes at the frame period to deliver the current
pitches CP to the partial feedback loop 101 and to the
feature quantity adjusting unit 109. The frame delay 113
is supplied with the current pitches CP through the third
extracting circuit input terminal 103(3) to supply the
previous pitches PP to the feature quantity adjusting
unit 109.
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41
Turning back to Fig. 23, the feature quantity
adjusting unit 109 is operable as follows. In response
to the mode and the delayed information supplied through
the first extracting circuit input terminal 103(1)
directly and additionally through the frame delay 107,
the feature quantity adjusting unit 109 compares the
previous pitches with predetermined further additional
threshold values to adjust the current pitches by the
previous pitches into the adjusted pitches CPP.
Referring afresh to Fig. 25, the description will
proceed to a speech signal encoder device according to a
seventh preferred embodiment of this invention. This
speech signal encoder device is different as follows from
that illustrated with reference to Fig. 5.
In the manner described referring to Figs. 6 and
7, 8, 12, or 15, the mode decision circuit 49 calculates
the pitch prediction gains at the frame period and
decides the mode information. In the manner described in
the Ozawa et al paper, an RMS extracting circuit 121 is
connected to the frame dividing circuit 37 and is
accompanied by an RMS codebook 123 keeping a plurality of
RMS code vectors. Controlled by the mode information
specifying one of the predetermined modes for each of the
original speech frames into which the input speech signal
is segmented, the RMS extracting circuit 121 selects one
of the RMS code vectors as a selected RMS vector for
delivery to the multiplexer 33 and therefrom to the
device output terminal 35. The RMS extracting circuit
21b7552
42
121 serves as a level extracting arrangement.
Turning to Fig. 26 with Fig. 25 continuously
referred to, the RMS extracting circuit 121 has a first
extracting circuit input terminal 121(1) supplied from
the mode decision circuit 49 with the mode information as
current mode information at the frame period. Connected
to the frame dividing circuit 37, a second extracting
circuit input terminal 121(2) is supplied with the
original speech frames. A third extracting circuit
121(3) is for referring to the RMS codebook 123. An
extracting circuit output terminal 123(0) is for
delivering the selected RMS vector to the multiplexer 33.
Connected to the second extracting circuit input
terminal 121(2), an RMS calculator 125 calculates the RMS
values R like the RMS calculator 91 described in
conjunction with Fig. 9, 13, or 14. Responsive to the
current mode information and to previous mode information
supplied from the first extracting circuit input terminal
121(1) directly and through a frame delay (D) 127, an RMS
adjusting unit 129 compares the RMS values fed from the
RMS calculator 125 as original RMS values with a predeter-
mined still further additional threshold value to adjust
the original RMS values into adjusted RMS values IR.
Connected to the RMS adjusting unit 129 and to the third
extracting circuit input terminal 121(3), an RMS
quantization vector selector 131 selects one of the RMS
code vectors that is most similar to the adjusted RMS
values at each frame period as the selected RMS vector
2167552
43
for delivery to the extracting circuit output terminal
121(0).
Further turning to Fig. 27 with Fig. 25 continu-
ously referred to, the RMS extracting circuit 121
additionally comprises an additional frame delay 133
supplied from the RMS adjusting unit 129 with the
adjusted RMS values as current adjusted values to supply
previous adjusted values back to the RMS adjusting unit
129. Responsive to the current and the previous mode
information and to the previous adjusted values, the RMS
adjusting unit 129 adjusts the original RMS values into
the adjusted RMS values.
Still further turning to Fig. 28 with Fig. 25
continuously referred to, the RMS extracting circuit 121
is different from that illustrated with reference.to Fig.
27 in that the previous adjusted values are not fed back
to the RMS adjusting unit 129. Instead, the additional
frame delay 133 delivers the previous adjusted values to
an RMS ratio calculator 135 which is supplied from the
RMS calculator 125 with the original RMS values to
calculate RMS ratios RR for feed back to the RMS
adjusting unit 129. In connection with the RMS ratios,
it should be noted that the previous adjusted values are
produced by the additional frame delay 133 concurrently
with previous RMS values which are the original RMS
values delivered one frame period earlier from the RMS
calculator 125 to the RMS adjusting unit 129 than the
previous adjusted values under consideration. Each RMS
2167552
44
ratio is a ratio of each original RMS value to one of the
previous adjusted values that is produced by the
additional frame delay 133 concurrently with the previous
RMS value one frame period earlier than the above-
mentioned each original RMS value.
The RMS adjusting unit 129 is now operable like
the feature quantity adjusting unit 109 described by
again referring to Fig. 22. More in detail, the RMS
adjusting unit 129 produces the RMS adjusted values IR by
comparing the original RMS values R with the still
further additional threshold value in response to the
current and the previous mode information and the RMS
ratios.
Referring to Fig. 29 with Fig. 25 continuously
referred to, the RMS extracting circuit 121 comprises the
RMS adjusting unit 129 which is additionally supplied
from the additional frame delay 133 with the previous
adjusted values besides the original RMS values and the
RMS ratios. The RMS adjusting unit 129 is consequently
operable like the feature quantity adjusting unit 109
described in conjunction with Figs. 17 and 18. More
particularly, the RMS adjusting unit 129 produces the RMS
adjusted values IR by comparing the original RMS values
with the still further additional threshold value to
adjust the current RMS values by the previous adjusted
values in response to the current and the previous mode
information and the RMS ratios.
2167 552
Turning to Fig. 30 with Fig. 25 continuously
referred to, the RMS extracting circuit 121 is different
from that illustrated with reference to Fig. 28 in that
the additional frame delay 133 of Fig. 28 is changed to a
series connection of first and second frame delays 133(1)
and 133(2). The RMS ratio calculator 135 calculates RMS
ratios of the current RMS values to past previous RMS
adjusted values produced by the RMS adjusting unit 129 in
response to RMS values which are two frame periods prior
to the current RMS values. The RMS adjusting unit 129 is
operable in the manner described as regards the RMS
extracting circuit 121 illustrated with reference to Fig.
28. It should be noted in this connection that the RMS
ratios are different between the RMS adjusting units
described in conjunction with Figs. 28 and 30.
Referring once more to Figs. 29 and 30 with Fig.
25 continuously referred to, the RMS extracting circuit
121 may comprise the first and the second additional
frame delays 133(1) and 133(2) and a signal line between
the first additional frame delay 133(1) and the RMS
adjusting unit 129 in the manner depicted in Fig. 29.
The RMS ratio calculator 135 is operable as described in
connection with Fig. 30. The RMS adjusting unit 129 is
operable as described in conjunction with Fig. 29.
