Note: Descriptions are shown in the official language in which they were submitted.
CA 02600284 2007-09-25
Specification
Title of the Invention
Speech Signal Decoding Method and Apparatus
Background of the Invention
The present invention relates to encoding and
decoding apparatuses for transmitting a speech signal at
a low bit rate and, more particularly, to a speech
signal decoding method and apparatus for improving the
quality of unvoiced speech.
As a popular method of encoding a speech
signal at low and middle bit rates with high efficiency,
a speech signal is divided into a signal for a linear
predictive filter and its driving sound source signal
(sound source signal). One of the typical methods is
CELP (Code Excited Linear Prediction) . CELP obtains a
synthesized speech signal (reconstructed signal) by
driving a linear prediction filter having a linear
prediction coefficient representing the frequency
characteristics of input speech by an excitation signal
given by the sum of a pitch signal representing the
pitch period of speech and a sound source signal made up
of a random number and a pulse. CELP is described in M..
Schroeder et al., "Code-excited linear prediction:
High-quality speech at very low bit rates", Proc. of
IEEE Int. Conf. on Acoust., Speech and Signal Processing,
pp. 937 - 940, 1985 (reference 1).
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Mobile communications such as portable phones
require high speech communication quality in noise
environments represented by a crowded street of a city
and a driving automobile. Speech coding based on the
above-mentioned CELP suffers deterioration in the
quality of speech (background noise speech) on which
noise is superposed. To improve the encoding quality of
background noise speech, the gain of a sound source
signal is smoothed in the decoder.
A method of smoothing the gain of a sound
source signal is described in "Digital Cellular
Telecommunication System; Adaptive Multi-Rate Speech
Transcoding", ETSI Technical Report, GSM 06.90 version
2Ø0, January 1999 (reference 2).
Fig. 4 shows an example of a conventional
speech signal decoding apparatus for improving the
coding quality of background noise speech by smoothing
the gain of a sound source signal. A bit stream is
input at a period (frame) of Tfr msec (e. g. , 20 msec) ,
and a reconstructed vector is calculated at a period
(subframe) of Tfr/Nsfr msec (e.g., 5 msec) for an integer
Nsfi (e. g. , 4). The frame length is given by Lfr samples
(e.g., 320 samples), and the subframe length is given by
L~fr samples (e.g., 80 samples). These numbers of
samples are determined by the sampling frequency (e.g.,
16 kHz) of an input signal. Each block will be
described.
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The code of a bi.t stream is input from an
input terminal 10. A code input circuit 1010 segments
the code of the bit stream input from the input terminal
into several segments, and converts them into indices
5 corresponding to a plurality of decoding parameters.
The code input circuit 1010 outputs an index
corresponding to LSP (Linear Spectrum Pair) representing
the frequency characteristics of the input signal to an
LSP decoding circuit 1020. The circuit 1010 outputs an
10 index corresponding to a delay Lpd representing the pitch
period of the input signal to a pitch signal decoding
circuit 1210, and an index corresponding to a sound
source vector made up of a random number and a pulse to
a sound source signal decoding circuit 1110. The
circuit 1010 outputs an index corresponding to the first
gain to a first gain decoding circuit 1220, and an index
corresponding to the second gain to a second gain
decoding circuit 1120.
The LSP decoding circuit 1020 has a table
which stores a plurality of sets of LSPs. The LSP
decoding circuit 1020 receives the index output from the
code input circuit 1010, reads an LSP corresponding to
the index from the table, and sets the LSP as 'LSPq~ sfr)(n),
j = 1, A, Np in the Nsfrth subframe of the current frame
(nth frame). Np is a linear prediction order. The LSPs
of the first to (Nsfr-1) th subframes are obtained by
linearly interpolating qMSgr)(n) and q(Nsfr) (n - 1) . LSPq(m)(n),
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j= 1, A, Np, m= l, A, Nsfr are output to a linear
prediction coefficient conversion circuit 1030 and
smoothing coefficient calculation circuit 1310.
The linear prediction coefficient conversion
circuit 1030 receives LSPq~'")(n) , j = 1, A, Np, m= 1, /1, Nsfr
output from the LSP decoding circuit 1020. The linear
prediction coefficient conversion circuit 1030 converts
the received q~m)(n) into a linear prediction coefficient
&"n)(n), j = 1, A, Np, m= 1, A, Nsfri and outputs (X.(m'(n) to a
synthesis filter 1040. Conversion of the LSP into the
linear prediction coefficient can adopt a known method,
e.g., a method described in Section 5.2.4 of reference 2.
The sound source signal decoding circuit 1110
has a table which stores a plurality of sound source
vectors. The sound source signal decoding circuit 1110
receives the index output from the code input circuit
1010, reads a sound source vector corresponding to the
index from the table, and outputs the vector to a second
gain circuit 1130.
The second gain decoding circuit 1120 has a
table which stores a plurality of gains. The second
gain decoding circuit 1120 receives the index output
from the code input circuit 1010, reads a second gain
corresponding to the index from the table, and outputs
the second gain to a smoothing circuit 1320.
The second gain circuit 1130 receives the
first sound source vector output from the sound source
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signal decoding circuit 1110 and the second gain output
from the smoothing circuit 1320, multiplies the first
sound source vector and the second gain to decode a
second sound source vector, and outputs the decoded
second sound source vector to an adder 1050.
A storage circuit 1240 receives and holds an
excitation vector from the adder 1050. The storage
circuit 1240 outputs an excitation vector which was
input and has been held to the pitch signal decoding
circuit 1210.
The pitch signal decoding circuit 1210
receives the past excitation vector held by the storage
circuit 1240 and the index output from the code input
circuit 1010. The index designates the delay Lpd. The
pitch signal decoding circuit 1210 extracts a vector for
Lsf= samples corresponding to the vector length from the
start point of the current frame to a past point by Lpd
samples in the past excitation vector. Then, the
circuit 1210 decodes a first pitch signal (vector) For
Lpd < Lsfl, the circuit 1210 extracts a vector for Lpd
samples, and repetitively couples the extracted Lpd
samples to decode the first pitch vector having a vector
length of Lsfr samples. The pitch signal decoding
circuit 1210 outputs the first pitch vector to a first
gain circuit 1230.
The first gain decoding circuit 1220 has a
table which stores a plurality of gains. The first gain
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decoding circuit 1220 receives the index output from the
code input circuit 1010, reads a first gain
corresponding to the index, and outputs the first gain
to the first gain circuit 1230.
The first gain circuit 1230 receives the first
pitch vector output from the pitch signal decoding
circuit 1210 and the first gain output from the first
gain decoding circuit 1220, multiplies the first pitch
vector and the first gain to generate a second pitch
vector, and outputs the generated second pitch vector to
the adder 1050.
The adder 1050 receives the second pitch
vector output from the first gain circuit 1230 and the
second sound source vector output from the second gain
circuit 1130, adds them, and outputs the sum as an
excitation vector to the synthesis filter 1040.
The smoothing coefficient calculation circuit
1310 receives LSPq~m)(n) output from the LSP decoding
circuit 1020, and calculates an average LSPqo,(n):
qp~(n) = 0. 84 = qpl (n - 1) + 0. 16 = q~ StT) (n)
The smoothing coefficient calculation circuit
1310 calculates an LSP variation amount do(m) for each
subframe m:
Np Iq.7(n)
do(m) - _
j=1 q0j (n)
The smoothing coefficient calculation circuit 1310
calculates a smoothing coefficient ko(m) of the subframe
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m:
ko (m) = min ( 0 . 25, max ( 0, do (m) -0 . 4 ) ) / 0 . 25
where min(x,y) is a function using a smaller one of x
and y, and max(x,y) is a function using a larger one of
x and y. The smoothing coefficient calculation circuit
1310 outputs the smoothing coefficient ko(m) to the
smoothing circuit 1320.
The smoothing circuit 1320 receives the
smoothing coefficient ko(m) output from the smoothing
coefficient calculation circuit 1310 and the second gain
output from the second gain decoding circuit 1120. The
smoothing circuit 1320 calculates an average gain go(m)
from a second gain go(m) of the subframe m by
4
1
go(m) - > o(m - i)
5 j=0
The second gain go(m) is replaced by
go(m) = go(m) = ko(m) + go(m) = (1 - ko(m) )
The smoothing circuit 1320 outputs the second
gain go(m) to the second gain circuit 1130.
The synthesis filter 1040 receives the
excitation vector output from the adder 1050 and a
linear prediction coefficient a;, i = 1,A,Np output from
the linear prediction coefficient conversion circuit
1030. The synthesis filter 1040 calculates a
reconstructed vector by driving the synthesis filter
1/A(z) in which the linear prediction coefficient is set,
by the excitation vector. Then, the synthesis filter
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1040 outputs the reconstructed vector from an output
terminal 20. Letting a;, i = 1, A, Np be the linear
prediction coefficient, the transfer function 1/A(z) of
the synthesis filter is given by
Np
1 / (A)z = 1 / (1 - ~ aizl )
Fig. 5 shows the arrangement of a speech
signal encoding apparatus in a conventional speech
signal encoding/decoding apparatus. A first gain
circuit 1230, second gain circuit 1130, adder 1050, and
storage circuit 1240 are the same as the blocks
described in the conventional speech signal decoding
apparatus in Fig. 4, and a description thereof will be
omitted.
An input signal (input vector) generated by
sampling a speech signal and combining a plurality of
samples as one frame into one vector is input from an
input terminal 30. A linear prediction coefficient
calculation circuit 5510 receives the input vector from
the input terminal 30. The linear prediction
coefficient calculation circuit 5510 performs linear
prediction analysis for the input vector to obtain a
linear prediction coefficient. Linear prediction
analysis is described in Chapter 8 "Linear Predictive
Coding of Speech" of reference 4.
The linear prediction coefficient calculation
circuit 5510 outputs the linear prediction coefficient
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to an LSP conversion/quantization circuit 5520,
weighting filter 5050, and weighting synthesis filter
5040.
The LSP conversion/quantization circuit 5520
receives the linear prediction coefficient output from
the linear prediction coefficient calculation circuit
5510, converts the linear prediction coefficient into
LSP, and quantizes the LSP to attain the quantized LSP.
Conversion of the linear prediction coefficient into the
LSP can adopt a known method, e.g., a method described
in Section 5.2.4 of reference 2.
Quantization of the LSP can adopt a method
described in Section 5.2.5 of reference 2. As described
in the LSP decoding circuit of Fig. 4 (prior art), the
quantized LSP is the quantized LSPq~ Sfrl(n) , j = 1, A, Np in
the Nsfr subframe of the current frame (nth frame). The
quantized LSPs of the first to (Nsfr-1)th subframes are
obtained by linearly interpolating Sfr)(n) and
~ sfr)(n - 1). The LSP is LSPq~ sfr)(n) , j = 1, !L, Np in the
Nsfr subframe of the current frame (nth frame). The LSPs
of the first to (Nsfr-1)th subframes are obtained by
linearly interpolating q~ sfr)(n) and q~Nsfr)(n - 1) .
The LSP conversion/quantization circuit 5520
outputs the LSPq(7)(n), j = 1, A, NP, m= 1, A, Nsfr, and the
quantized LSPq('"'(n), j = 1, A, Np, m = 1, A, NSfr to a linear
prediction coefficient conversion circuit 5030, and an
index corresponding to the quantized LSPq(Nsfr)(n) j
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1,A,NP to a code output circuit 6010.
The linear prediction coefficient conversion
circuit 5030 receives the LSPq~ '(n) , j 1, A, NP, m=
1, A, Nsfr, and the quantized LSPq('")(n), j 1, A, Np, m=
1,A,Nsfr output from the LSP conversion/quantization
circuit 5520. The circuit 5030 converts q~'"~(n) into a
linear prediction coefficient a~ )(n), j = 1, A, NP, m=
1, A, Nsfri and q(' )(n) into a quantized linear prediction
coefficient a(m)(n), j = 1, A, Np, m = 1, A, Nsfr. The linear
prediction coefficient conversion circuit 5030 outputs
the a~~(n) to the weighting filter 5050 and weighting
synthesis filter 5040, and a~ )(n) to the weighting
synthesis filter 5040. Conversion of the LSP into the
linear prediction coefficient and conversion of the
quantized LSP into the quantized linear prediction
coefficient can adopt a known method, e.g., a method
described in Section 5.2.4 of reference 2.
The weighting filter 5050 receives the input
vector from the input terminal 30 and the linear
prediction coefficient output from the linear prediction
coefficient conversion circuit 5030, and generates a
weighting filter W(z) corresponding to the human sense
of hearing using the linear prediction coefficient. The
weighting filter is driven by the input vector to obtain
a weighted input vector. The weighting filter 5050
outputs the weighted input vector to a subtractor 5060.
The transfer function W(z) of the weighting filter 5050
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is given by W(z) = Q(z/y1)/Q(z/yZ) .
Np
Note that Q = 1 - ~ Y I Z zl and Q( z/ y 2)
Np
= 1-~ aim)~y2z1 where y 1 and y 2 are constants, e. g., y 1=
0.9 and y2= 0.6. Details of the weighting filter are
described in reference 1.
The weighting synthesis filter 5040 receives
the excitation vector output from the adder 1050, and
the linear prediction coefficient (x~ )(n), j = 1,A,Np, m=
1,11,Nsfrf and the quantized linear prediction coefficient
&(m)(n), j = 1, t1, NP, m= 1, A, Nsfr that are output from the
linear prediction coefficient conversion circuit 5030.
A weighting synthesis filter H(z)W(z) = Q(z/y
1) /[A ( z) Q( z/ y 2)] having a~ )(n) and &~ '(n) is driven by the
excitation vector to obtain a weighted reconstructed
vector. The transfer function H(z) = 1/A(z) of the
Np
synthesis filter is given by 1/A(z) = 1/ (1- y&(m)zi )
The subtractor 5060 receives the weighted
input vector output from the weighting filter 5050 and
the weighted reconstructed vector output from the
weighting synthesis filter 5040, calculates their
difference, and outputs it as a difference vector to a
minimizing circuit 5070.
The minimizing circuit 5070 sequentially
outputs all indices corresponding to sound source
vectors stored in a sound source signal generation
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circuit 5110 to the sound source signal generation
circuit 5110. The minimizing circuit 5070 sequentially
outputs indices corresponding to all delays Lpd within a
range defined by a pitch signal generation circuit 5210
to the pitch signal generation circuit 5210. The
minimizing circuit 5070 sequentially outputs indices
corresponding to all first gains stored in a first gain
generation circuit 6220 to the first gain generation
circuit 6220, and indices corresponding to all second
gains stored in a second gain generation circuit 6120 to
the second gain generation circuit 6120.
The minimizing circuit 5070 sequentially
receives difference vectors output from the subtractor
5060, calculates their norms, selects a sound source
vector, delay Lpd, and first and second gains that
minimize the norm, and outputs corresponding indices to
the code output circuit 6010. The pitch signal
generation circuit 5210, sound source signal generation
circuit 5110, first gain generation circuit 6220, and
second gain generation circuit 6120 sequentially receive
indices output from the minimizing circuit 5070.
The pitch signal generation circuit 5210,
sound source signal generation circuit 5110, first gain
generation circuit 6220, and second gain generation
circuit 6120 are the same as the pitch signal decoding
circuit 1210, sound source signal decoding circuit 1110,
first gain decoding circuit 1220, and second gain
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decoding circuit 1120 in Fig. 4 except for input/output
connections, and a detailed description of these blocks
will be omitted.
The code output circuit 6010 receives an index
corresponding to the quantized LSP output from the LSP
conversion/quantization circuit 5520, and indices
corresponding to the sound source vector, delay Lpd, and
first and second gains that are output from the
minimizing circuit 5070. The code output circuit 6010
converts these indices into a bit stream code, and
outputs it via an output terminal 40.
The first problem is that sound different from
normal voiced speech is generated in short unvoiced
speech intermittently contained in the voiced speech or
part of the voiced speech. As a result, discontinuous
sound is generated in the voiced speech. This is
because the LSP variation amount do(m) decreases in the
short unvoiced speech to increase the smoothing
coefficient. Since do(m) greatly varies over time, do(m)
exhibits a large value to a certain degree in part of
the voiced speech, but the smoothing coefficient does
not become 0.
The second problem is that the smoothing
coefficient abruptly changes in unvoiced speech. As a
result, discontinuous sound is generated in the unvoiced
speech. This is because the smoothing coefficient is
determined using do(m) which greatly varies over time.
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The third problem is that proper smoothing
processing corresponding to the type of background noise
cannot be selected. As a result, the decoding quality
degrades. This is because the decoding parameter is
smoothed based on a single algorithm using only
different set parameters.
Summary of the Invention
It is an object of the present invention to
provide a speech signal decoding method and apparatus
for improving the quality of reconstructed speech
against background noise speech.
To achieve the above object, according to the
present invention, there is provided a speech signal
decoding method comprising the steps of decoding
information containing at least a sound source signal, a
gain, and filter coefficients from a received bit stream,
identifying voiced speech and unvoiced speech of a
speech signal using the decoded information, performing
smoothing processing based on the decoded information
for at least either one of the decoded gain and the
decoded filter coefficients in the unvoiced speech, and
decoding the speech signal by driving a filter having
the decoded filter coefficients by an excitation signal
obtained by multiplying the decoded sound source signal
by the decoded, gain using a result of the smoothing
processing.
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Br;Pf Description of the Drawings
Fig. 1 is a block diagram showing a speech
signal decoding apparatus according to the first
embodiment of the present invention;
Fig. 2 is a block diagram showing a speech
signal decoding apparatus according to the second
embodiment of the present invention;
Fig. 3 is a block diagram showing a speech
signal encoding apparatus used in the present invention;
Fig. 4 is a block diagram showing a
conventional speech signal decoding apparatus; and
Fig. 5 is a block diagram showing a
conventional speech signal encoding apparatus.
Description of the Preferred Embodiments
The present invention will be described in
detail below with reference to the accompanying drawings.
Fig. 1 shows a speech signal decoding
apparatus according to the first embodiment of the
present invention. An input terminal 10, output
terminal 20, LSP decoding circuit 1020, linear
prediction coefficient conversion circuit 1030, sound
source signal decoding circuit 1110, storage circuit
1240, pitch signal decoding circuit 1210, first gain
circuit 1230, second gain circuit 1130, adder 1050, and
synthesis filter 1040 are the same as the blocks
described in the prior art of Fig. 4, and a description
thereof will be omitted.
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A code input circuit 1010, voiced/unvoiced
identification circuit 2020, noise classification
circuit 2030, first switching circuit 2110, second
switching circuit 2210, first filter 2150, second filter
2160, third filter 2170, fourth filter 2250, fifth
filter 2260, sixth filter 2270, first gain decoding
circuit 2220, and second gain decoding circuit 2120 will
be described.
A bit stream is input at a period (frame) of
Tfr msec (e.g., 20 msec), and a reconstructed vector is
calculated at a period (subframe) of Tfr/Nsfr msec (e.g.,
5 msec) for an integer Nsfr (e.g., 4). The frame length
is given by Lfr samples (e.g., 320 samples), and the
subframe length is given by Lsfr samples (e . g. , 80
samples). These numbers of samples are determined by
the sampling frequency (e.g., 16 kHz) of an input signal.
Each block will be described.
The code input circuit 1010 segments the code
of a bit stream input from an input terminal 10 into
several segments, and converts them into indices
corresponding to a plurality of decoding parameters.
The code input circuit 1010 outputs an index
corresponding to LSP to the LSP decoding circuit 1020.
The circuit 1010 outputs an index corresponding to a
speech mode to a speech mode decoding circuit 2050, an
index corresponding to a frame energy to a frame power
decoding circuit 2040, an index corresponding to a delay
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LPd to the pitch signal decoding circuit 1210, and an
index corresponding to a sound source vector to the
sound source signal decoding circuit 1110. The circuit
1010 outputs an index corresponding to the first gain to
the first gain decoding circuit 2220, and an index
corresponding to the second gain to the second gain
decoding circuit 2120.
The speech mode decoding circuit 2050 receives
the index corresponding to the speech mode that is
output from the code input circuit 1010, and sets a
speech mode Smode corresponding to the index. The speech
mode is determined by threshold processing for an
intra-frame average GoP(n) of an open-loop pitch
prediction gain Gop(m) calculated using a perceptually
weighted input signal in a speech encoder. The speech
mode is transmitted to the decoder. In this case,n
represents the frame number; and m, the subframe number.
Determination of the speech mode is described in K.
Ozawa et al., "M-LCELP Speech Coding at 4 kb/s with
Multi-Mode and Multi-Codebook", IEICE Trans. On Commun.,
Vol. E77-B, No. 9, pp. 1114 - 1121, September 1994
(reference 3).
The speech mode decoding circuit 2050 outputs
the speech mode Smode to the voiced/unvoiced
identification circuit 2020, first gain decoding circuit
2220, and second gain decoding circuit 2120.
The frame power decoding circuit 2040 has a
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table 2040a which stores a plurality of frame energies.
The frame power decoding circuit 2040 receives the index
corresponding to the frame power that is output from the
code input circuit 1010, and reads a frame power Erms
corresponding to the index from the table 2040a. The
frame power is attained by quantizing the power of an
input signal in the speech encoder, and an index
corresponding to the quantized value is transmitted to
the decoder. The frame power decoding circuit 2040
outputs the frame power Er,,,s to the voiced/unvoiced
identification circuit 2020, first gain decoding circuit
2220, and second gain decoding circuit 2120.
The voiced/unvoiced identification circuit
2020 receives LSPq~)(n) output from the LSP decoding
circuit 1020, the speech mode Smode output from the
speech mode decoding circuit 2050, and the frame power
EMS output from the frame power decoding circuit 2040.
The sequence of obtaining the variation amount of a
spectral parameter will be explained.
As the spectral parameter, LSPq( m)(n) is used.
In the nth frame, a long-term average qj(n) of the LSP is
calculated by
qj (n) _(3o qj (n - 1) +(1 -(30) = q(Nsfr)(n) ~ j l. A, Np
where ~ o = 0. 9.
A variation amount dn(n) of the LSP in the nth
frame is defined by
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N D(m) (n )
dq(n) g'j
71 m=1 qj(n)
where Dq,~(n) corresponds to the distance between qj(n) and
(n) For example,
Da,, (n) = (qj (nj - (n) )2
or
Dq,j(n) = Iqj(n) q(~')(n)j
In this case, Dq j(n) = Iqj(n) - q~m)(n)I is employed.
A section where the variation amount dq(n) is
large substantially corresponds to voiced speech,
whereas a section where the variation amount dq(n) is
small substantially corresponds to unvoiced speech.
However, the variation amount dq(n) greatly varies over
time, and the range of dq(n) in voiced speech and that
in unvoiced speech overlap each other. Thus, a
threshold for identifying voiced speech and unvoiced
speech is difficult to set.
For this reason, the long-term average of
dq(n) is used to identify voiced speech and unvoiced
speech. A long-term average dq,(n) of dq(n) is calculated
using a linear or non-linear filter. As dql(n), the
average, median, or mode of dq(n) can be applied. In
this case,
dqi (n) dq,(n - 1) + (1 - (3! ) dq(n)
is used where al = 0.9.
Threshold processing for dqi(n) determines an
identification flag S15:
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if (dql (n) ? Cti,l) then S,s = 1
else Sv5 = 0
where Ct,,l is a given constant ( e. g., 2. 2), SVS = 1
corresponds to voiced speech, and SõS = 0 corresponds to
unvoiced speech.
Even voiced speech may be mistaken for
unvoiced speech in a section where steadiness is high
because dq(n) is small. To avoid this, a section where
the frame power and pitch prediction gain are large is
regarded as voiced speech. For SV5 = 0, Sõs is corrected
by the following additional determination:
if (Erms Crms and Smode ~ 2) then Sõ5 = 1
else Sõs = 0
where Crms is a given constant (e. g. , 10, 000) , and Smode ~
2 corresponds to an intra-frame average GoP(n) of 3.5 dB
or more for the pitch prediction gain.
This is defined by the encoder.
The voiced/unvoiced identification circuit
2020 outputs Sõ5 to the noise classification circuit 2030,
first switching circuit 2110, and second switching
circuit 2210, and dq,(n) to the noise classification
circuit 2030.
The noise classification circuit 2030 receives
dql(n) and S,S that are output from the voiced/unvoiced
identification circuit 2020. In unvoiced speech (noise),
a value dg2(n) which reflects the average behavior of
dq,(n) is obtained using a linear or non-linear filter.
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For Sõ5 = 0,
dq2 (n) = (j2 = dq2 (n - 1) + (1 - R2 ) = dq, (n)
is calculated for az = 0.94.
Threshold processing for dq2(n) classifies
noise to determine a classification flag SõZ:
if (dq2(n) Cthz) then SnZ = 1
else SnZ = 0
where Cth2 is a given constant (e.g., 1.7) , SnZ = 1
corresponds to noise whose frequency characteristics
unsteadily change over time, and S,Z = 0 corresponds to
noise whose frequency characteristics steadily change
over time. The noise classification circuit 2030
outputs SRZ to the first and second switching circuits
2110 and 2210.
The first switching circuit 2110 receives
LSP4~)(n) output from the LSP decoding circuit 1020, the
identification flag Sõ5 output from the voiced/unvoiced
identification circuit 2020, and the classification flag
SnZ output from the noise classification circuit 2030.
The first switching circuit 2110 is switched in
accordance with the identification and classification
flag values to output LSPq~ )(n) to the first filter 2150
for Sõs = 0 and SõZ = 0, to the second filter 2160 for Sõs
= 0 and Sõ_ = 1, and to the third filter 2170 for Sõs = 1.
The first filter 2150 receives LSPq(m)(n) output
from the first switching circuit 2110, smoothes it using
a linear or non-linear filter, and outputs it as a first
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smoothed LSPq~ ~)(n) to the linear prediction coefficient
conversion circuit 1030. In this case, the first filter
2150 uses a filter given by
qi ~)(n) = Y~ ' qi m-u(n) +(1 - yl)= q(m)(n) , 7= 1, A, Np
where q~ ~(n) = qi ~sfr) (n - 1) , and y,= 0.5.
The second filter 2160 receives LSPq~m~(n)
output from the first switching circuit 2110, smoothes
it using a linear or non-linear filter, and outputs it
as a second smoothed LSPq2~(n) to the linear prediction
coefficient conversion circuit 1030. In this case, the
second filter 2160 uses a filter given by
qi j(n) = Yz qi ~ (n) +(1 y2) q~ '(n) , 7= 1, A, NP
where q2 ~(n) = qN~fr)(n - 1), and y 0Ø
The third filter 2170 receives LSPq(P)(n) output
from the first switching circuit 2110, smoothes it using
a linear or non-linear filter, and outputs it as a third
smoothed LSPq~~(n) to the linear prediction coefficient
conversion circuit 1030. In this case, q3 ~(n) = qj (n) .
The second switching circuit 2210 receives the
second gain g2)(n) output from the second gain decoding
circuit 2120, the identification flag Sõ5 output from the
voiced/unvoiced identification circuit 2020, and the
classification flag Sr,z output from the noise
classification circuit 2030. The second switching
circuit 2210 is switched in accordance with the
identification and classification flag values to output
the second gain g;m)(n) to the fourth filter 2250 for Sõ5 =
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CA 02600284 2007-09-25
0 and SõZ = 0, to the fifth filter 2260 for S,, = 0 and
S.Z = 1, and to the sixth filter 2270 for S,, = 1.
The fourth filter 2250 receives the second
gain g?3(n) output from the second switching circuit 2210,
smoothes it using a linear or non-linear filter, and
outputs it as a first smoothed gain gZm)(n) to the second
gain circuit 1130. In this case, the fourth filter 2250
uses a filter given by
gim)(n) - 72 ' g(2m-1)(n) + (1 - 72) = g2 )(n)
where g2 ~(n) = g2NSfr)(n and y Z = 0. 9.
The fifth filter 2260 receives the second gain
g2)(n) output from the second switching circuit 2210,
smoothes it using a linear or non-linear filter, and
outputs it as a second smoothed gain g22(n) to the second
gain circuit 1130. In this case, the fifth filter 2260
uses a filter given by
922(n) = 72 " gzm "(n) + (1 - 72) gi )(n)
where g2 2(n) = g2 2 fr~(n - 1) , and 'y Z= 0. 9.
The sixth filter 2270 receives the second gain
42)(n) output from the second switching circuit 2210,
smoothes it using a linear or non-linear filter, and
outputs it as a third smoothed gain g23(n) to the second
gain circuit 1130. In this case, g23(n) = g2 ~(n) .
The first gain decoding circuit 2220 has a
table 2220a which stores a plurality of gains. The
first gain decoding circuit 2220 receives an index
corresponding to the third gain output from the code
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CA 02600284 2007-09-25
input circuit 1010, the speech mode Smode output from the
speech mode decoding circuit 2050, the frame power Eõs
output from the frame power decoding circuit 2040, the
linear prediction coefficient a~"(n), j = 1, A, NP of the
mth subframe of the nth frame output from the linear
prediction coefficient conversion circuit 1030, and a
pitch vector ca,,( i), i= 1, A, Lsfr output from the pitch
signal decoding circuit 1210.
The first gain decoding circuit 2220
calculates a k parameter k~ )(n) , j= 1, A, Np (to be simply
represented as k,) from the linear prediction
coefficient &~m)(n) . This is calculated by a known method,
e.g., a method described in Section 8.3.2 in L.R.
Rabiner et al., "Digital Processing of Speech Signals",
Prentice-Hall, 1978 (reference 4) Then, the first gain
decoding circuit 2220 calculates an estimated residual
power Eres using ki :
Eres Erms IIJ?,(I - k~ )
The first gain decoding circuit 2220 reads a
third gain Ygac corresponding to the index from the
table 2220a switched by the speech mode Smode, and
calculates a first gain gac:
Eres
ga~ = Yga~
L f< ' <
i i=0 Cac()
The first gain decoding circuit 2220 outputs
the first gain ga, to the first gain circuit 1230. The
second gain decoding circuit 2120 has a table 2120a
- 24 -
CA 02600284 2007-09-25
which stores a plurality of gains.
The second gain decoding circuit 2120 receives
an index corresponding to the fourth gain output from
the code input circuit 1010, the speech mode Smode output
from the speech mode decoding circuit 2050, the frame
power Ers output from the frame power decoding circuit
2040, the linear prediction coefficient &~m'(n), j =
1,A,Np of the mth subframe of the nth frame output from
the linear prediction coefficient conversion circuit
1030, and a sound source vector ceC ( i), i = l, A, Lsfr
output from the sound source signal decoding circuit
1110.
The second gain decoding circuit 2120
calculates a k parameter k(')(n), j = 1, A, Np (to be simply
represented as k,) from the linear prediction
coefficient &(m'(n). This is calculated by the same known
method as described for the first gain decoding circuit
2220. Then, the second gain decoding circuit 2120
calculates an estimated residual power Eres using kj:
2 0 Eres ' Erms V n~ pi (1 - k~ )
The second gain decoding circuit 2120 reads a fourth
gain YgeC corresponding to the index from the table
2120a switched by the speech mode Smode, and calculates a
second gain gP~ :
Y 9ec Eres
~Ja~ =
Lsrr-I
i-0 C
VE ec(~ )
The second gain decoding circuit 2120 outputs
- 25 -
CA 02600284 2007-09-25
the second gain geC to the second switching circuit 2210.
Fig. 2 shows a speech signal decoding
apparatus according to the second embodiment of the
present invention.
This speech signal decoding apparatus of the
present invention is implemented by replacing the frame
power decoding circuit 2040 in the first embodiment with
a power calculation circuit 3040, the speech mode
decoding circuit 2050 with a speech mode determination
circuit 3050, the first gain decoding circuit 2220 with
a first gain decoding circuit 1220, and the second gain
decoding circuit 2120 with second gain decoding circuit
1120. In this arrangement, the frame power and speech
mode are not encoded and transmitted in the encoder, and
the frame power (power) and speech mode are obtained
using parameters used in the decoder.
The first and second gain decoding circuits
1220 and 1120 are the same as the blocks described in
the prior art of Fig. 4, and a description thereof will
be omitted.
The power calculation circuit 3040 receives a
reconstructed vector output from a synthesis filter 1040,
calculates a power from the sum of squares of the
reconstructed vectors, and outputs the power to a
voiced/unvoiced identification circuit 2020. In this
case, the power is calculated for each subframe.
Calculation of the power in the mth subframe uses a
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CA 02600284 2007-09-25
reconstructed signal output from the synthesis filter
1040 in the (m-1)th subframe. For a reconstructed
signal Ssyn ( i), i = 0, A, Lgfr, the power Erms is calculated
by, e.g., RMS (Root Mean Square):
Lsfr-1
Erms - ~ Ssyn(~-)
i=0
The speech mode determination circuit 3050
receives a past excitation vector emem ( i), i= 0, A, Lmem-l
held by a storage circuit 1240, and the index output
from the code input circuit 1010. The index designates
a delay Lpa = Lmem is a constant determined by the maximum
value of Lpd.
In the mth subframe, a pitch prediction gain
Ge111em (m) , m = 1, A, Nsf= is calculated from the past
excitation vector emem(i) and delay Lpd:
C'emem (m) = 10 = loglo ( gemem (m) )
where
gemem(m) - E~(m)
1 -
Eal (m)EaZ(m)
Lsf 1 2
Eal(m) - E'mem(~)
i=0
Lsfr
Ea2 (m) - 7, emem (1. - Lpd)
i=0
Lsfr-I
E'(m) _ E emem(1)emem(i - Lpd)
i=0
The pitch prediction gain Gemem (m) or the
intra-frame average Gemem(n) in the nth frame of Gemem(m)
undergoes the iollowing threshold processing to set a
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CA 02600284 2007-09-25
speech mode Smode :
lf (C'emem(n) 3. 5) then Smode 2
else Smode - 0
The speech mode determination circuit 3050 outputs the
speech mode Smode to the voiced/unvoiced identification
circuit 2020.
Fig. 3 shows a speech signal encoding
apparatus used in the present invention.
The speech signal encoding apparatus in Fig. 3
is implemented by adding a frame power calculation
circuit 5540 and speech mode determination circuit 5550
in the prior art of Fig. 5, replacing the first and
second gain generation circuits 6220 and 6120 with first
and second gain generation circuits 5220 and 5120, and
replacing the code output circuit 6010 with a code
output circuit 5010. The first and second gain
generation circuits 5220 and 5120, an adder 1050, and a
storage circuit 1240 are the same as the blocks
described in the prior art of Fig. 5, and a description
thereof will be omitted.
The frame power calculation circuit 5540 has a
table 5540a which stores a plurality of frame energies.
The frame power calculation circuit 5540 receives an
input vector from an input terminal 30, calculates the
RMS (Root Mean Square) of the input vector, and
quantizes the RMS using the table to attain a quantized
frame power Erms For an input vector si ( i), i = 0, A, L
sfr'
- 28 -
CA 02600284 2007-09-25
a power Eirms is given by
Lsfr-~
Eirms
i=0
The frame power calculation circuit 5540
outputs the quantized frame power Erms to the first and
second gain generation circuits 5220 and 5120, and an
index corresponding to Erms to the code output circuit
5010.
The speech mode determination circuit 5550
receives a weighted input vector output from a weighting
filter 5050.
The speech mode Smode is determined by
executing threshold processing for the intra-frame
average Gop(n) of an open-loop pitch prediction gain
Gop(m) calculated using the weighted input vector. In
this case, n represents the frame number; and m, the
subframe number.
In the mth subframe, the following two
equations are calculated from a weighted input vector
s(i) and the delay Ltmp, and Ltmp which maximizes
Esctmp(m) / Esa2tmp is obtained and set as Lop:
Lsf 1
Esctmp (m) ' , Swi (i)sWi (i - Ltmp)
i=0
Lsfr-I
Esa2tmp (m) - s 2 Wi(]. - Ltmp)
i=0
From the weighted input vector s,,i(i) and the
delay Lop, the pitch prediction gain Gop (m) , m = 1, A, Nsfr
is calculated:
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CA 02600284 2007-09-25
GoP (m) = 10=logio ( goP (m)
where
where
gap(Ill) = 2
E(m)
s
1-
Esal (m)Esa2 (m)
Lsfr-1
wi(1)
Esa](m) s2
i=0
Lsfr-1 2
Esa2(m) - y SWi(1 - LoP)
i=0
Lsfr-1
E5C(m) _ Y, sWi(i)sWi(i - Lop)
i=0
The pitch prediction gain GoP(m) or the intra-frame
average Gop(n) in the nth frame of Gop (m) undergoes the
following threshold processing to set the speech mode
Smode =
if (Gop(n) 3. 5) then Smode = 2
e l s e S mode 0
Determination of the speech mode is described
in K. Ozawa et al., "M-LCELP Speech Coding at 4 kb/s
with Multi-Mode and Multi-Codebook", IEICE Trans. On
Commun., Vol. E77-B, No. 9, pp. 1114 - 1121, 1994
(reference 3).
The speech mode determination circuit 5550
outputs the speech mode Smode to the first and second
gain generation circuits 5220 and 5120, and an index
corresponding to the speech mode Smode to the code output
circuit 5010.
A pitch signal generation circuit 5210, a
- 30 -
CA 02600284 2007-09-25
sound source signal generation circuit 5110, and the
first and second gain generation circuits 5220 and 5120
sequentially receive indices output from a minimizing
circuit 5070. The pitch signal generation circuit 5210,
sound source signal generation circuit 5110, first gain
generation circuit 5220, and second gain generation
circuit 5120 are the same as the pitch signal decoding
circuit 1210, sound source signal decoding circuit 1110,
first gain decoding circuit 2220, and second gain
decoding circuit 2120 in Fig. 1 except for input/output
connections, and a detailed description of these blocks
will be omitted.
The code output circuit 5'010 receives an index
corresponding to the quantized LSP output from the LSP
conversion/quantization circuit 5520, an index
corresponding to the quantized frame power output from
the frame power calculation circuit 5540, an index
corresponding to the speech mode output from the speech
mode determination circuit 5550, and indices
corresponding to the sound source vector, delay Lpd, and
first and second gains that are output from the
minimizing circuit 5070. The code output circuit 5010
converts these indices into a bit stream code, and
outputs it via an output terminal 40.
The arrangement of a speech signal encoding
apparatus in a speech signal encoding/decoding apparatus
according to the fourth embodiment of the present
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CA 02600284 2007-09-25
invention is the same as that of the speech signal
encoding apparatus in the conventional speech signal
encoding/decoding apparatus, and a description thereof
will be omitted.
In the above-described embodiments, the
long-term average of do(m) varies over time more
gradually than do(m), and does not intermittently
decrease in voiced speech. If the smoothing coefficient
is determined in accordance with this average,
discontinuous sound generated in short unvoiced speech
intermittently contained in voiced speech can be reduced.
By performing identification of voiced or unvoiced
speech using the average, the smoothing coefficient of
the decoding parameter can be completely set to 0 in
voiced speech.
Also for unvoiced speech, using the long-term
average of do(m) can prevent the smoothing coefficient
from abruptly changing.
The present invention smoothes the decoding
parameter in unvoiced speech not by using single
processing, but by selectively using a plurality of
processing methods prepared in consideration of the
characteristics of an input signal. These methods
include moving average processing of calculating the
decoding parameter from past decoding parameters within
a limited section, auto-regressive processing capable of
considering long-term past influence, and non-linear
- 32 -
CA 02600284 2007-09-25
processing of limiting a preset value by an upper or
lower limit after average calculation.
According to the first effect of the present
invention, sound different from normal voiced speech
that is generated in short unvoiced speech
intermittently contained in voiced speech or part of the
voiced speech can be reduced to reduce discontinuous
sound in the voiced speech. This is because the
long-term average of do(m) which hardly varies over time
is used in the short unvoiced speech, and because voiced
speech and unvoiced speech are identified and the
smoothing coefficient is set to 0 in the voiced speech.
According to the second effect of the present
invention, abrupt changes in smoothing coefficient in
unvoiced speech are reduced to reduce discontinuous
sound in the unvoiced speech. This is because the
smoothing coefficient is determined using the long-term
average of do(m) which hardly varies over time.
According to the third effect of the present
invention, smoothing processing can be selected in
accordance with the type of background noise to improve
the decoding quality. This is because the decoding
parameter is smoothed selectively using a plurality of
processing methods in accordance with the
characteristics of an input signal.
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