Note: Descriptions are shown in the official language in which they were submitted.
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SPEECH CODER
The present invention relates to an audio or speech synthesiser for use with
compressed digitally encoded audio or speech signals. In particular, to a post-
processor for processing signals derived from an excitation code book and
adaptive
code book of a LPC type speech decoder.
In digital radia telephone systems the information, i.e. speech, is digitally
encoded
prior to being transmitted over the air. The encoded speech is then decoded at
the
receiver. First, an analogue speech signal is digitally encoded using Pulse
Code
Modulation (PCM) for example. Then speech coding and decoding of the PCM
speech
(or original speech) is implemented by speech coders and decoders. Due to the
increase in use of radio telephone systems the radio spectrum available for
such
systems is becoming crowded. In order to make the best possible use of the
available
radio spectrum, radio telephone systems utilise speech coding techniques which
require low numbers of bits to encode the speech in order to reduce the
bandwidth
required for the transmission. Efforts are continually being made to reduce
the
number of bits required for speech coding to further reduce the bandwidth
required for
speech transmission.
A known speech coding/decoding method is based on linear predictive coding
(LPC)
techniques, and utilises analysis-by-synthesis excitation coding. In an
encoder utilising
such a method, a speech sample is first analysed to derive parameters which
represent characteristics such as wave form information (LPC) of the speech
sample.
These parameters are used as inputs to short-term synthesis filter. The short-
term
synthesis filter is excited by signals which are derived from a code book of
signals.
The excitation signals may be random, e.g. a stochastic code book, or may be
adaptive or specifically optimised for use in speech coding. Typically, the
code book
comprises two parts, a fixed code book and the adaptive code book. The
excitation
' outputs of respective code books are combined and the total excitation input
to the
short term synthesis filter. Each total excitation signal is filtered and the
result
compared with the original speech sample (PCM coded) to derive an "error" or
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difference between the synthesised speech sample and the original speech
sample.
The total excitation which results in the lowest error is selected as the
excitation for
representing the speech sample. The code book indices, or addresses, of the
location
of respective partial optimal excitation signals in the fixed and adaptive
code book are
transmitted to a receiver, together with the LPC parameters or coefficients. A
composite code book identical to that at the transmitter is also located at
the receiver,
and the transmitted code book indices and parameters are used to generate the
appropriate total excitation signal from the receiver's code book. This total
excitation
signal is then fed to a short-term synthesis filter identical to that in the
transmitter, and
having the transmitted LPC coefficients as respective inputs. The output from
the
short-term synthesis filter is a synthesised speech frame which is the same as
that
generated in the transmitter by the analysis-by-synthesis method.
Due to the nature of digital coding, although the synthesised speech is
objectively
accurate it sounds artificial. Also, degradations, distortions and artifacts
are
introduced into the synthesised speech due to quantisation effects and other
anomalies due to the electronic processing. Such artifacts particularly occur
in low bit-
rate coding since there is insufficient information to reproduce the original
speech
signal exactly. Hence there have been attempts to improve the perceptual
quality of
synthesised speech. This has been attempted by the use of post-filters which
operate
on the synthesised speech sample to enhance its perceived quality. Known post-
filters are located at the output of the decoder and process the synthesised
speech
signal to emphasise or attenuate what are generally considered to be the most
important frequency regions in speech. The importance of respective regions of
speech frequencies has been analysed primarily using subjective tests on the
quality
of the resulting speech signal to the human ear. Speech can be split into two
basic
parts, the spectral envelope (formant structure) or the spectral harmonic
structure (line
structure), and typically post-filtering emphasises one or other, or both of
these parts
of a speech signal. The filter coefficients of the post-filter are adapted
depending on
the characteristics of the speech signal to match the speech sounds. A filter
emphasising or attenuating the harmonic structure is typically referred to as
a long-
term, cr pitch or long delay post filter. and a fitter emphasising the
spectral envelope
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structure is typically referred to as a short delay post filter or short-term
post
filter.
A further known filtering technique for improving the perceptual quality of
synthesised speech is disclosed in International Patent Application WO
91!06091. A pitch prefilter is disclosed in WO 91/06091 comprising a pitch
enhancement filter, normally disposed at a position after a speech synthesis
or LPC filter, moved to a position before the speech synthesis or LPC filter
where it filters pitch information contained in the excitation signals input
to the
speech synthesis or LPC filter.
However, there is still a desire to produce synthesised speech which has even
better perceptual quality.
According to a first aspect of the present invention there is provided an LPC-
type speech synthesiser, comprising a post-processing means for operating
on a first signal (ex(n)) including speech periodicity information and derived
from an excitation signal source,
wherein the excitation signal source comprises a fixed code book and
an adaptive code book, and means for obtaining the first signal by combining
first and second partial excitation signals originating from the fixed and
adaptive code books, respectively,
wherein the post-processing means is adapted to modify the speech
periodicity information content of the first signal in accordance with a
second
signal generated from the excitation signal source by comprising gain control
means for scaling the second signal in accordance with a first scaling factor
(p) derived from pitch information associated with the first signal and means
for combining the second signal with the first signal.
An advantage of the present invention is that the first signal is modified by
a
second signal originating from the same source as the first signal, and thus
no
additional sources of distortion or artifacts such as extra filters are
introduced.
- r..Nn~~~<.f ~,I,~.~,..~I 11
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Only the signals generated in the excitation source are utilised. The relative
contributions of the signals inherent to the excitation generator in a speech
synthesiser are being modified, with
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no artificial added signals, to re-scale the synthesiser signals.
Good speech enhancement may be obtained if post-processing of the excitation
is
based on modifying the relative contributions of the excitation components
derived
within the excitation generator of the speech synthesiser itself.
Processing the excitation by filtering the total excitation ex(n) without
considering or
modifying the relative contributions of the signals inherent to the excitation
generator,
i.e. v(n) and c,(n) typically does not give the best possible enhancement.
Modifying the first signal in accordance with the second signal from the same
excitation source increases waveform continuity within the excitation and in
the
resulting synthesised speech signal, thereby improving its perceptual quality.
In a preferred embodiment the excitation source comprises a fixed code book
and an
adaptive code book, the first signal being derivable from a combination of
first and
second partial excitation signals respectively selectable from the fixed and
adaptive
code books, which is a particularly convenient excitation source for a speech
synthesiser.
Preferably, there is a gain element for scaling the second signal in
accordance with
a scaling factor (p) derivable from pitch information associated with the
first signal from
the excitation source, which has the advantage that the first signal speech
periodicity
information content is modified which has greater effect on perceived speech
quality
than other modifications.
Suitably, the scaling factor (p) is derivable from an adaptive code book
scaling factor
(b), and the scaling factor (p) is derivable in accordance with the following
equation,
b < TH,~.;, then p = 0.0
TH,~~~. <_ b < TH, then p = a2"",f,(b)
THZ <_ b < TH3 Then p = a2",,,f2(bJ
if
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THN_r <_ b <_ THuPper then p = aenhN-r f rv-r(b)
b > TH~aPer then p = ae"hN f nr (b)
where TH represents threshold values, b is the adaptive code book gain factor,
p is
the post-processor means scale factor, ae",, is a linear sealer and f(b) is a
function of
gain b
In a specific embodiment the scaling factor (p) is derivable in accordance
with
b < TH,oW , then p = 0.0
if TH,oW <_ b <_ THuPp~r then p = ae",,b2
b > TH"Pper then p = aennb
where ae",, is a constant that controls the strength of the enhancement
operation, b is
adaptive code book gain, TH are threshold values and p is the post-processor
scale
factor which utilises the insight that speech enhancement is most effective
for voiced
speech where b typically has a high value, whereas for unvoiced sounds where b
has
a low value a not so strong enhancement is required.
The second signal may originate from the adaptive code book, and may also be
substantially the same as the second partial excitation signal. Alternatively,
the
second signal may originate from the fixed code book, and may also be
substantially
the same as the first partial excitation signal.
For the second signal originating from the fixed code book, the gain control
means is
adapted to scale the second signal in accordance with a second scaling factor
(p')
where,
Pr=_ RP
(P +b)
and g is a fixed code book scaling factor. b is an adaptive code book scaling
factor
I a , 11~ ~Ildk~d' ~~' ~. 11 ,.
CA 02224688 2002-07-17
6
and p is the first scaling factor.
The first signal may be a first excitation signal suitable for inputting to a
speech synthesis filter, and the second signal may be a second excitation
signal suitable for inputting to a speech synthesis filter. The second
excitation
signal may be substantially the same as the second partial excitation signal.
Optionally, the first signal may be a first synthesised speech signal output
from a first speech synthesis filter and derivable from the first excitation
signal, and the second signal may be the output from a second speech
synthesis filter and derivable from the second excitation signal. An advantage
of this is that speech enhancement is carried out on the actual synthesised
speech and thus there are less electronic components to introduce distortion
to the signal before it is rendered audible.
Advantageously, there is provided an adaptive energy control means adapted
to scale a modified first signal in accordance with the following
relationship,
N-1
8~~1t)
k= n~0
N-1
ew r2(n)
n~0
where N is a suitably chosen adaption period, ex(n) is first signal, ew'(n) is
modified first signal and k is an energy scale factor, which normalises the
resulting enhanced signal to the power input to the speech synthesiser.
In a second aspect of the invention there is provided a post-processing
method for enhancing LPC-synthesised speech, comprising the steps of
deriving a first signal including speech periodicity information from an
excitation signal source, wherein the excitation signal source comprises a
fixed code book and an adaptive code book,
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obtaining the first signal by combining first and second partial excitation
signals originating from the fixed and adaptive code books, respectively,
modifying the speech periodicity information content of the first signal in
accordance with a second signal generated from the excitation signal source
by scaling the second signal in accordance with a first scaling factor derived
from pitch information associated with the first signal and combining the
second signal with the first signal.
In a third aspect of the invention there is provided an LPC-type speech
synthesiser, comprising
an adaptive code book and a fixed code book for generating first and
second partial excitation signals respectively,
scaling unit means for scaling the first and second partial excitation
signals with received adaptive and fixed codebook scaling factors
respectively, modifying means for modifying a first excitation signal in
accordance with a further scaling factor, the scaling factor being a function
of
pitch information associated with the first excitation signal, and
means for combining the second partial excitation signal with the modified
first
partial excitation signal.
In a fourth aspect of the invention there is provided an LPC-type speech
synthesiser, comprising
an adaptive code book and a fixed codebook for generating first and
second partial excitation signals respectively,
scaling unit means for scaling the first and second partial excitation
signals with received adaptive and fixed codebook scaling factors
respectively, modifying means for modifying a second excitation signal in
accordance with a further scaling factor, the scaling factor being a function
of
pitch information associated with the first excitation signal, and
means for combining the modified second partial excitation signal with
the first partial excitation signal.
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According to a fifth aspect of the invention there is provided a method for
LPC-type speech synthesis, comprising
deriving a first partial excitation signal from an adaptive code book and
a second partial excitation signal from a fixed code book,
scaling the first and second partial excitation signals with received
adaptive and fixed codebook scaling factors respectively,
modifying the first partial excitation signal in accordance with a further
scaling
factor, the scaling factor being a function of pitch information associated
with
the first partial excitation signal, and combining the second partial
excitation
signal with the modified first partial excitation signal.
According to a sixth aspect of the invention there is provided a method for
LPC- type speech synthesis, comprising
deriving a first partial excitation signal from an adaptive code book and
a second partial excitation signal from a fixed code book,
scaling the first and second partial excitation signals with received
adaptive and fixed codebook scaling factors respectively,
modifying the second partial excitation signal in accordance with a
further scaling factor, the scaling factor being a function of pitch
information
associated with the first partial excitation signal, and combining the
modified
second partial excitation signal with the first partial excitation signal.
Embodiments in accordance with the invention will now be described by way
of example only, and with reference to the accompanying drawings in which:
Figure 1 shows a schematic diagram of a known Code Excitation Linear
Prediction (CELP) encoder;
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Figure 2 shows a schematic diagram of a known CELP decoder;
Figure 3 shows a schematic diagram of a CELP decoder in accordance with a
first
embodiment of the invention;
Figure 4 shows a second embodiment in accordance with the invention;
Figure 5 shows a third embodiment in accordance with the invention;
Figure 6 shows a fourth embodiment in accordance with the invention; and
Figure 7 shows a fifth embodiment in accordance with the invention.
A known CELP encoder 100 is shown in Figure 1. Original speech signals are
input
to the encoder at 102 and Long Term Prediction (LTP) coefficients T,b are
determined
using adaptive code book 104. The LTP prediction coefficients are determined
for
segments of speech typically comprising 40 samples and are 5 ms in length. The
LTP
coefficients relate to periodic characteristics of the original speech. This
includes any
periodicity in the original speech and not just to periodicity which
corresponds to the
pitch of the original speech due to vibrations in the vocal cords of a person
uttering
the original speech. '
Long Term Prediction is performed using adaptive code book 104 and gain
element
114, which comprise a part of excitation signal (ex(n)) generator 126 shown
dotted in
Figure 1. Previous excitation signals ex(n) are stored in the adaptive code
book 104
by virtue of feedback loop 122. During the LTP process the adaptive code book
is
searched by varying an address T, known as a delay or lag, pointing to
previous
excitation signals ex(n). These signals are sequentially output and amplified
at gain
element 114 with a scaling factor b to form signals v(n) prior to being added
at 118
to an excitation signal c,(n) derived from the fixed code book 112 and scaled
by a
factor g at gain element 116. Linear Prediction Coefficients (LPC) for the
speech
sample are calculated at 106. The LPC coefficients are then quantised at 108.
The
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quantised LPC coefficients are then available for transmission over the air
and to be
input to short term filter 110. The LPC coefficients (r(i), i=1..., m where m
is prediction
order) are calculated for segments of speech comprising 160 samples over 20
ms.
" All further processing is typically performed in segments of 40 samples,
that is to say
an excitation frame length of 5 ms. The LPC coefficients relate to the
spectral
envelope of the original speech signal.
Excitation generator 126 effectively comprises a composite code book 104, 112
comprising sets of codes for exciting short term synthesis filter 110. The
codes
comprise sequences of voltage amplitudes, each corresponding to a speech
sample
in the speech frame.
Each total excitation signal ex(n) is input to short term or LPC synthesis
filter 110 to
form a synthesised speech sample s(n). The synthesised speech sample s(n) is
input
to a negative input of adder 120, having an original speech sample as a
positive input.
The adder 120 outputs the difference between the original speech sample and
the
synthesised speech sample, this difference being known as an objective error.
The
objective error is input to a best excitation selection element 124, which
selects the
total excitation ex(n) resulting in a synthesised speech frame s(n) having the
least
objective error. During the selection the objective error is typically further
spectrally
weighted to emphasise those spectral regions of the speech signal important
for
human perception. The respective adaptive and fixed code book parameters (gain
6
and delay T, and gain g and index i) giving the best excitation signal ex(n)
are then
transmitted, together with the LPC filter coefficients r(i), to a receiver to
be used in
synthesising the speech frame to reconstruct the original speech signal.
A decoder suitable for decoding speech parameters generated by an encoder as
described with reference to Figure 1 is shown in Figure 2. Radio frequency
unit 201
receives a coded speech signal via an antenna 212. The received radio
frequency
signal is down converted to a baseband frequency and demodulated in the RF
unit
201 to recover speech information. Generally, coded speech is further encoded
prior
to being transmitted to comprise channel coding and error correction coding.
This
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channel coding and error correction coding has to be decoded at the receiver
before
the speech coding can be accessed or recovered. Speech coding parameters are
recovered by parameter decoder 202.
The speech coding parameters in LPC speech coding are the set of LPC synthesis
filter coefficients r(iJ; i = 1,...,m, (where m is the order of the
prediction), fixed code
book index i and gain g. The adaptive code book speech coding parameters delay
T and gain b are also recovered.
The speech decoder 200 utilises the above mentioned speech coding parameters
to
create from the excitation generator 211 an excitation signal ex(n) for
inputting to the
LPC synthesis filter 208 which provides a synthesised speech frame signal s(n)
at its
output as a response to the excitation signal ex(n). The synthesised speech
frame
signal s(n) is further processed in audio processing unit 209 and rendered
audible
through an appropriate audio transducer 210.
In typical linear predictive speech decoders, the excitation signal ex(n) for
the LPC
synthesis filter 208 is formed in excitation generator 211 comprising a fixed
code book
203 generating excitation sequence c,(n) and adaptive code book 204. The
location
of the code book excitation sequence ex(n) in the respective code books 203.
204 is
indicated by the speech coding parameter i and delay T. The fixed code book
excitation sequence c,(n) partially used to form the excitation signal ex (n)
is taken from
the fixed excitation code book 203 from a location indicated by index l and is
then
suitably scaled by the transmitted gain factor g in the scaling unit 205.
Similarly, the
adaptive code book excitation sequence v(n) also partially used to form
excitation
signal ex(n) is taken from the adaptive code book 204 from a location
indicated by
delay T using selection logic inherent to the adaptive code book and is then
suitably
scaled by the transmitted gain factor b in scaling unit 206.
The adaptive code book 204 operates on the fixed code book excitation sequence
c,(n) by adding a second partial excitation component v(n) to the code book
excitation
sequence g c,(n). The second component is derived from past excitation signals
in a
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manner already described with reference to Figure 1, and is selected from the
adaptive code book 204 using selection logic suitably included in the adaptive
code
book. The component v(n) is suitably scaled in the scaling unit 206 by the
transmitted
adaptive code book gain b and then added to g c,(n) in the adder 207 to form
the total
excitation signal ex(n), where
ex(n) = g c,(n) + b v(n). (1 )
The adaptive code book 204 is then updated by using the total excitation
signal ex(n).
The location of the second partial excitation component v(n) in the adaptive
code
book 204 is indicated by the speech coding parameter T. The adaptive
excitation
component is selected from the adaptive code book using speech coding
parameter
T and selection logic included in the adaptive code book.
An LPC speech synthesis decoder 300 in accordance with the invention is shown
in
Figure 3. The operation of speech synthesis according to Figure 3 is the same
as for
Figure 2 except that the total excitation signal ex(n) is, prior to being used
as the
excitation for the LPC synthesis filter 208, processed in excitation post-
processing unit
317. The operation of circuit elements 201 to 212 in Figure 3 are similar to
those in
Figure 2 with the same numerals.
In accordance with an aspect of the invention, a post-processing unit 317 for
the total
excitation ex(n) is used in the speech decoder 300. The post-processing unit
317
comprises an adder 313 for adding a third component to the total excitation
ex(n). A
gain unit 315 then appropriately scales the resulting signal ew~(n) to form
signal ew(n)
which is then used to excite the LPC synthesis filter 208 to produce
synthesised
speech signal szW(n). The speech synthesised according to the invention has
improved perceptual quality compared to the speech signal s(n) synthesised by
the
prior art speech synthesis decoder shown in Figure 2.
The post-processing unit 317 has the total excitation ex(n) input to it, and
outputs a
perceptually enhanced total excitation ew(n). The post-processing unit 317
also has
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the adaptive code book gain b, and an unsealed partial excitation component
v(n)
taken from the adaptive code book 204 at a location indicated by the speech
coding
parameters as further inputs. Partial excitation component v(n) is suitably
the same ,
component which is employed inside the excitation generator 211 to form the
second
excitation component bv(n) which is added to the scaled code book excitation
gc;(n) .
to form the total excitation ex(n). By using an excitation sequence which is
derived
from the adaptive code book 204, no further sources of artifacts are added to
the
speech processing electronics, as is the case with the known post or pre-
filter
techniques which use extra filters. The excitation post-processing unit 317
also
comprises scaling unit 314 which scales the partial excitation component
v(n)by a
scale factor p, and the scaled component pv(n) is added by adder 313 to the
total
excitation component ex(n). The output of adder 313 is an intermediate total
excitation signal ew'(n). It is of the form,
ew'(n) = gc,(n) + bv(n) + pv(n) (2)
= 9c~(n) + (b + p) v(n).
The scaling factor p for scaling unit 314 is determined in the perceptual
enhancement
gain control unit 312 using the adaptive code book gain b. The scaling factor
p re-
scales the contribution of the two excitation components from the fixed and
adaptive
code book, c,(n) and v(n), respectively. The scaling factor p is adjusted so
that during
synthesised speech frame samples that have high adaptive code book gain value
b
the scale factor p is increased, and during speech that has low adaptive code
book
gain value b the scaling factor p is reduced. Furthermore, when b is less than
a
threshold value (b < TH,oW) the scaling factor p is set to zero. The
perceptual
enhancement gain control unit 314 operates in accordance with equation (3)
given
below,
b < TH,oW Then p = 0.0 -
if TH,o,~. ~ b _ TH"aPZ, then p = a~n~b2
b > TH~PPe~ then p = ar""b ,
(3)
where a"", is a constant that controls the strength of the enhancement
operation. The
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applicant has found that a good value for ae"h is 0.25, and good values for
TH,oW and
TH"ppe, are 0.5 and 1.0, respectively.
Equation 3 can be of a more general form, and a general formulation of the
- enhancement function is shown below in equation (4). In the general case,
there
could be more than 2 thresholds for the enhancement gain b. Also, the gain
could be
defined as a more general function of b.
b < TH,oW then p = 0.0
TH,oW <_ b < TH2 then p = a~"",f,(b)
TH2 ~ b < TH3 then p = a~"h2f2(b)
if
(4)
THN-, _ b <_ TH"Pp2~ then P - aenhN-i f rv-r(b)
b > TH~QPer then P = aenhN f N (b)
In the preferred embodiment previously described N=2, TH,oW = 0.5, THZ = 7.0,
TH3
- ~, aenhl- 025, and az",,2 = 0.25, f, (b)=b2, and f2(b)=b.
The threshold values (TH), enhancement values (az",~ and the gain functions
(f(b)) are
arrived at empirically. Since the only realistic measure of perceptual speech
quality
can be obtained by human beings listening to the speech and giving their
subjective
opinions on the speech quality, the values used in equations (3) and (4) are
determined experimentally. Various values for the enhancement thresholds and
gain
functions are tried, and those resulting in the best sounding speech are
selected. The
applicant has utilised the insight that the enhancement to the speech quality
using this
method is particularly effective for voiced speech where b typically has a
high value,
whereas for less voiced sounds which have a lower value of b not so strong an
enhancement is required. Thus, gain value p is controlled such that for voiced
sounds, where the distortions are most audible, the effect is strong and for
unvoiced
sounds the effect is weaker or not used at all. Thus, as a general rule, the
gain
functions (f") should be chosen so that there is a greater effected for higher
values of
b, than for lower values of b. This increases the difference between the pitch
components of the speech and the other components.
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In the preferred embodiment, operating in accordance with equation (3), the
functions
operating on gain value b are a squared dependency for mid-range values of b
and
a linear dependency for high-range values of b. It is the applicant's present
understanding that this gives good speech quality since for high values of b,
i.e. highly
voiced speech, there is greater effect and for lower values of b there is less
effect.
This is because b typically lies in the range -7<b<9 and therefore b2 < b.
To ensure unity power gain between the input signal ex(n), and the output
signal ew(n)
of the excitation post-processing unit 317, a scale factor is computed and is
used to
scale the intermediate excitation signal ew'(n) in the scaling unit 315 to
form the post-
processed excitation signal ew(n). The scale factor k is given as
N-1
~ ~(n)
n=0
N-1
ew ~~(n)
n=0
(5)
where N is a suitably chosen adaption period. Typically, N is set equal to the
excitation frame length of the LPC speech codec.
In the adaptive code book of the encoder, for values of T which are less than
the
frame length or excitation length a part of the excitation sequence is
unknown. For
these unknown portions a replacement sequence is locally generated within the
adaptive code book by using suitable selection logic. Several adaptive code
book
techniques to generate this replacement sequence are known from the state of
the art.
Typically, a copy of a portion of the known excitation is copied to where the
unknown
portion is located thereby creating a complete excitation sequence. The copied
portion may be adapted in some manner to improve the quality of the resulting
speech .
signal. When doing such copying, the delay value T is not used since it would
point
to the unknown portion. Instead. a particular selection logic resulting in a
modified
value for T is used (for example. using T multiplied by an integer factor so
that it
always points to the known signal portion). 5o that the decoder is
synchronised with
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the encoder, similar modifications are employed in the adaptive code book of
the
decoder. By using such a selection logic to generate a replacement sequence
within
the adaptive code book, the adaptive code book is able to adapt for high pitch
voices
such as female and child voices resulting in efficient excitation generation
and
improved speech quality for these voices.
For obtaining good perceptual enhancement, all modifications inherent to the
adaptive
code book e.g. for values of T less than the frame length are taken into
account in the
enhancement post-processing. This is obtained in accordance with the invention
by
the use of the partial excitation sequence from the adaptive code book v(n)
and the
re-scaling of the excitation components, inherent to the excitation generator
of the
speech synthesiser.
In summary, the method enhances the perceptual quality of the synthesised
speech
and reduces audible artifacts by adaptively scaling the contribution of the
partial
excitation components taken from the code book 203 and from the adaptive code
book
204, in accordance with equations (2), (3), (4) and (5).
Figure 4 shows a second embodiment in accordance with the invention, wherein
the
excitation post-processing unit 417 is located after the LPC synthesis filter
208 as
illustrated. In this embodiment an additional LPC synthesis filter 408 is
required for
the third excitation component derived from the adaptive code book 204. In
Figure
4, elements which have the same function as in Figures 2 and 3, also have the
same
reference numerals.
In the second embodiment shown in Figure 4, the LPC synthesised speech is
perceptually enhanced by post-processor 417. The total excitation signal ex(n)
derived
from the code book 203 and adaptive code book 204 is input to LPC synthesis
filter
208 and processed in a conventional manner in accordance with the LPC
coefficients
> r(i). The additional or third partial excitation component v(n) derived from
the adaptive
code book 204 in the manner described in relation to Figure 3 is input
unsealed to a
second LPC synthesis filter 408 and processed in accordance with the LPC
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16
coefficients r(i). The outputs s(n) and s~(n) of respective LPC filters 208,
408 are input
to post-processor 417 and added together in adder 413. Prior to being input to
adder
413, signal sv(n) is scaled by scale factor p. As described with reference to
Figure 3,
the values for processing scale factor or gain p can be arrived at
empirically.
Additionally, the third partial excitation component may be derived from the
fixed code
book 203 and the scaled speech signal p' s~(n) subtracted from speech signal
s(n).
The resulting perceptually enhanced output sW(n) is then input to the audio
processing
unit 209.
Optionally, a further modification of the enhancement system can be formed by
moving the scaling unit 414 of Figure 4 to be in front of the LPC synthesis
filter 408.
Locating the post-processor 417 after the LPC or short term synthesis filters
208, 408
can give better control of the emphasis of the speech signal since it is
carried out
directly on the speech signal, not on the excitation signal. Thus, less
distortions are
likely to occur.
Optionally, enhancement can be achieved by modifying the embodiments described
with reference to Figures 3 and 4 respectively, such that the additional
(third)
excitation component is derived from the fixed code book 203 instead of the
adaptive
code book 204. Then, a negative scaling factor should be used instead of the
original
positive gain factor p, to decrease the gain for excitation sequence c;{n)
from the fixed
code book. This results in a similar modification of the relative
contributions of the
partial excitation signals c;(n) and v(n), to speech synthesis as achieved
with the
embodiments of Figures 3 and 4.
Figure 5 shows an embodiment in accordance with the invention in which the
same
result as obtained by using scaling factor p and the additional excitation
component .
from the adaptive code book may be achieved. In this embodiment, the fixed
code
book excitation sequence c;(n) is input to scaling unit 314 which operates in
accordance with scale factor p' output from perceptual enhancement gain
control 2
512. The scaled fixed code book excitation, p' c;(n), output from scaling unit
314 is
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17
input to adder 313 where it is added to total excitation sequence ex(n)
comprising
components c;(n) and v(n) from the fixed code book 203 and adaptive code book
204
respectively.
When increasing the gain for the excitation sequence signal v(n) from the
adaptive
code book 204 the total excitation (before adaptive energy control 316) is
given by
equation (2), viz.
ew' (n) = g c;(n) + (b + p) v(n) (2)
When decreasing the gain for an excitation sequence c;(n) from the fixed code
book
203, the total excitation (before adaptive energy control 316) is given as
ew' (n) _ (9 + P') c~(n) + bv(n) (6),
where p' is the scaling factor derived by perceptual enhancement gain control
2 512
shown in Figure 5. Taking equation (2) and reformulating it into a form
similar to
equation (6) gives:
ew' (n) = g c~(n) + (b + P) v(n)
pbb~(P ~)c'(n)+bv(n),
pbbL(g P b)c;(n)+bv(n)
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Thus, selecting p~=-
(p +b)
in the embodiment of Figure 5 a similar enhancement as obtained with the
embodiment of Figure 3 will be achieved. When the intermediate total
excitation
signal ew'(n) is scaled by adaptive energy control 316 to the same energy
content as
ex(n), then both embodiments, Figure 3 and Figure 5, result in the same total
excitation signal ew(n).
Perceptual enhancement gain control 2 512 can therefore utilise the same
processing
as employed in relation to the embodiments of Figures 3 and 4 to generate "p",
and
then utilise equation (8) to get p'.
The intermediate total excitation signal ew'(n) output from adder 313 is
scaled in
scaling unit 315 under control of adaptive energy control 316 in a similar
manner as
described above in relation to the first and second embodiments.
Referring now to Figure 4, LPC synthesised speech may be perceptually enhanced
by post-processor 417 by synthesised speech derived from additional excitation
signals from the fixed code book.
The dotted line 420 in Figure 4 shows an embodiment wherein the fixed code
book
excitation signals c;(n) are coupled to LPC synthesis filter 408. The output
of the LPC
synthesis filter 408 (sc;(n)) is then scaled in unit 414 in accordance with
scaling factor
p' derived from perceptual enhancement gain control 512, and added to the
synthesised signal s(n) in adder 413 to produce intermediate synthesis signal
s'W(n).
After normalisation in scaling unit 415 the resulting synthesis signal sv,(n)
is forwarded
to the audio processing unit 209.
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The foregoing embodiments comprise adding a component derived from the
adaptive
code book 204 or fixed code book 203 to an excitation ex(n) or synthesised
s(n), to
form an intermediate excitation ew'(n) or synthesised signal s'W(n).
Optionally, post-processing may be dispensed with and the adaptive code book
v(n)
or fixed code book c;(n) excitation signals may be scaled and directly
combined
together. Thereby obviating the addition of components to unsealed combined
fixed
and adaptive code book signals.
Figure 6 shows an embodiment in accordance with an aspect of the invention
having
the adaptive code book excitation signals v(n) scaled and then combined with
the fixed
code book excitation signals c;(n) to directly form an intermediate signal
ew'(n).
Perceptual enhancement gain control 612 outputs parameter "a" to control
scaling unit
614. Scaling unit 614 operates on adaptive code book excitation signal v(n) to
scale-
up or amplify excitation signal v(n) over the gain factor b used to get the
normal
excitation. Normal excitation ex(n) is also formed and coupled to the adaptive
code
book 204 and adaptive energy control 316. Adder 613 combines up-scaled
excitation
signal av(n) and fixed code book excitation c;(n) to form an intermediate
signal;
ew'(n) = g c;(n) + av(n) (9)
If a = b+p, then the same processing as given by equation (2) may be achieved.
Figure 7 shows an embodiment operable in a manner similar to that shown in
Figure
6, but down-scaling or attenuating the fixed code book excitation signal
c;(n). For this
embodiment the intermediate excitation sign ew'(n) is given by:
ew'(n) _ (g + p') c;(n) + bv(n)
= a'c;(n) + bv(n) (10) ,
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where
a~_g_ SP = gb
P+b P+b
(11 ).
Perceptual enhancement gain control 712 outputs a control signal a' in
accordance
with equation (11 ), to obtain a similar result as obtained with equation (6)
in
accordance with equation (8). The down-scaled fixed code book excitation
signal
a'c;(n) is combined with adaptive code book excitation signal v(n) in adder
713 to form
intermediate excitation signal ew'(n). The remaining processing is carried out
as
described before, to normalise the excitation signal and formed synthesised
signal
Sew(n)'
The embodiments described with reference to Figures 6 and 7 perform scaling of
the
excitation signals within the excitation generator, and directly from the code
books.
The determination of scaling factor "p" for the embodiments described with
reference
to Figures 5, 6 and 7 may be made in accordance with equations (3) or (4)
described
above .
Various methods of control of the enhancement level (a2",~ may be employed. In
addition to the adaptive code book gain b, the amount of enhancement could be
a
function of the lag or delay value Tfor the adaptive code book 204. For
example, the
post processing could be turned on (or emphasised) when operating in a high
pitch
range orwhen the adaptive code book parameter Tis shorter than the excitation
block
length (virtual lag range). As a result, female and child voices, for which
the invention
is most beneficial, would be highly post processed.
The post processing control could also be based on voiced/unvoiced speech ,
decisions. For example, the enhancement could be stronger for voiced speech,
and
it could be totally turned off when the speech is classified as unvoiced. This
can be
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21
derived from the adaptive code book gain value b which is itself a simple
measure of
voicedlunvoiced speech, that is to say the higher b, the -more voiced speech
present
in the original speech signal.
Embodiments in accordance with the present invention may be modified, such
that the
third partial excitation sequence is not the same partial excitation sequence
derived
from the adaptive code book or fixed code book in accordance with conventional
speech synthesis, but is selectable via selection logic typically included in
respective
code books to choose another third partial excitation sequence. The third
partial
excitation sequence may be chosen to be the immediately previously used
excitation
sequence or to be always a same excitation sequence stored in the fixed code
book.
This would act to reduce the difference between speech frames and thereby
enhance
the continuity of the speech. Optionally, b and/or T can be recalculated in
the
decoder from the synthesised speech and used to derive a third partial
excitation
sequence. Further, a fixed gain p and/or fixed excitation sequence can be
added or
subtracted as appropriate to the total excitation sequence ex(n) or speech
signal
s(n) depending on the location of the post-processor.
In view of the foregoing description it will be evident to a person skilled in
the art that
various modifications may be made within the scope of the invention. For
example,
variable-frame-rate coding, fast code book searching, reversal of the order of
pitch
prediction and LPC prediction may be utilised in the codec. Additionally, post-
processing in accordance with the present invention could also be included in
the
encoder, not just the decoder. Furthermore, aspects of respective embodiments
described with reference to the drawings rnay be combined to provide further
embodiments in accordance with the invention.
The scope of the present disclosure includes any novel feature or combination
of
features disclosed therein either explicitly or implicitly or any
generalisation thereof
irrespective of whether or not it relates to the claimed invention or
mitigates any or all
of the problems addressed by the present invention. The applicant hereby gives
notice that new claims may be formulated to such features during prosecution
of this
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application or of any such further application derived therefrom.
