U.S. patent application number 10/799503 was filed with the patent office on 2004-09-16 for voicing index controls for celp speech coding.
This patent application is currently assigned to Mindspeed Technologies, Inc.. Invention is credited to Gao, Yang.
Application Number | 20040181411 10/799503 |
Document ID | / |
Family ID | 33029999 |
Filed Date | 2004-09-16 |
United States Patent
Application |
20040181411 |
Kind Code |
A1 |
Gao, Yang |
September 16, 2004 |
Voicing index controls for CELP speech coding
Abstract
An approach for improving quality of speech synthesized using
analysis-by-synthesis (ABS) coders is presented. An unstable
perceptual quality in analysis-by-synthesis type speech coding
(e.g. CELP) may occur because the periodicity degree in a voiced
speech signal may vary significantly for different segments of the
voiced speech. Thus, the present invention uses a voicing index,
which may indicate the periodicity degree of the speech signal, to
control and improve ABS type speech coding. The voicing index may
be used to improve the quality stability by controlling encoder
and/or decoder in: fixed-codebook short-term enhancement including
the spectrum tilt; perceptual weighting filter; sub-fixed codebook
determination; LPC interpolation; fixed-codebook pitch enhancement;
post-pitch enhancement; noise injection into the high-frequency
band at decoder; LTP Sinc window; signal decomposition, etc.
Inventors: |
Gao, Yang; (Mission Viejo,
CA) |
Correspondence
Address: |
FARJAMI & FARJAMI LLP
Suite 360
26522 La Alameda Avenue
Mission Viejo
CA
92691
US
|
Assignee: |
Mindspeed Technologies,
Inc.
|
Family ID: |
33029999 |
Appl. No.: |
10/799503 |
Filed: |
March 11, 2004 |
Related U.S. Patent Documents
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Application
Number |
Filing Date |
Patent Number |
|
|
60455435 |
Mar 15, 2003 |
|
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Current U.S.
Class: |
704/262 ;
704/E19.028; 704/E19.035; 704/E19.046; 704/E21.011 |
Current CPC
Class: |
G10L 19/12 20130101;
G10L 19/09 20130101; G10L 19/087 20130101; G10L 19/005 20130101;
G10L 21/0232 20130101; G10L 21/038 20130101; G10L 21/0208 20130101;
G10L 19/20 20130101; G10L 19/265 20130101; G10L 25/90 20130101 |
Class at
Publication: |
704/262 |
International
Class: |
G10L 019/04 |
Claims
What is claimed is:
1. A method of improving synthesized speech quality comprising:
obtaining an input speech signal; coding said input speech using a
Code Excited Linear Prediction coder to generate code parameters
for synthesis of said input speech; and using a voicing index
representing a characteristic of said input speech in enhancing
said synthesis of said input speech.
2. The method of claim 1, wherein said characteristic of said input
speech is periodicity of said input speech.
3. The method of claim 1, wherein said enhancing said synthesis of
said input speech is by controlling an adaptive highpass filter
with said voicing index to enhance high frequency region during
said coding.
4. The method of claim 1, wherein said enhancing said synthesis of
said input speech is by controlling an adaptive perceptual
weighting filter in said Code Excited Linear Prediction coder with
said voicing index.
5. The method of claim 1, wherein said enhancing said synthesis of
said input speech is by controlling an adaptive Sinc window used in
said Code Excited Linear Prediction coder for pitch contribution
with said voicing index.
6. The method of claim 1, wherein said enhancing said synthesis of
said input speech is by controlling spectrum tilt of said input
speech by short-term enhancement of a fixed-codebook of said Code
Excited Linear Prediction coder with said voicing index.
7. The method of claim 1, wherein said enhancing said synthesis of
said input speech is by controlling a perceptual weighting filter
of said Code Excited Linear Prediction coder with said voicing
index.
8. The method of claim 1, wherein said enhancing said synthesis of
said input speech is by controlling a linear prediction coder of
said Code Excited Linear Prediction coder with said voicing
index.
9. The method of claim 1, wherein said enhancing said synthesis of
said input speech is by controlling a pitch enhancement
fixed-codebook of said Code Excited Linear Prediction coder with
said voicing index.
10. The method of claim 1, wherein said enhancing said synthesis of
said input speech is by controlling post pitch enhancement of said
Code Excited Linear Prediction coder with said voicing index.
11. The method of claim 1, wherein said voicing index selects at
least one sub-codebook from a plurality of sub-codebooks of said
Code Excited Linear Prediction coder based on said characteristic
of said input speech signal.
12. A method of improving synthesized speech quality comprising:
obtaining code parameters of an input speech signal; obtaining a
voicing index for use in enhancing synthesis of said input speech
signal from said code parameters; and processing said code
parameters through a Code Excited Linear Prediction coder using
information provided by said voicing index to generate a
synthesized version of said input speech signal.
13. The method of claim 12, wherein said voicing index provides
periodicity of said input speech signal.
14. The method of claim 12, wherein said voicing index provides
characteristics of an adaptive highpass filter used to enhance high
frequency region of said excitation during generation of said code
parameters for said input speech.
15. The method of claim 12, wherein said voicing index provides
characteristics of an adaptive perceptual weighting filter used to
enhance perceptual quality of said input speech during generation
of said code parameters for said input speech.
16. The method of claim 12, wherein said voicing index provides
characteristics of an adaptive Sinc window for pitch contribution
used to enhance perceptual quality of said input speech during
generation of said code parameters for said input speech.
17. The method of claim 12, wherein said enhancing synthesis of
said input speech is by controlling spectrum tilt of said input
speech by short-term enhancement of a fixed-codebook of said Code
Excited Linear Prediction coder with said voicing index.
18. The method of claim 12, wherein said enhancing of said
synthesis of said input speech is by controlling a linear
prediction coder filter of said Code Excited Linear Prediction
coder with said voicing index.
19. The method of claim 12, wherein said enhancing of said
synthesis of said input speech is by controlling a pitch
enhancement fixed-codebook of said Code Excited Linear Prediction
coder with said voicing index.
20. The method of claim 12, wherein said enhancing said synthesis
of said input speech is by controlling post pitch enhancement of
said Code Excited Linear Prediction coder with said voicing
index.
21. The method of claim 12, wherein said voicing index selects at
least one sub-codebook from a plurality of sub-codebooks of said
Code Excited Linear Prediction coder based on said characteristic
of said input speech signal.
22. An apparatus for improving synthesized speech quality
comprising: an input speech signal; a Code Excited Linear
Prediction coder for coding said input speech signal to generate
code parameters for synthesis of said input speech; and a voicing
index having a characteristic of said input speech for use in
enhancing said synthesis of said input speech.
23. The apparatus of claim 22, wherein said characteristic of said
input speech is periodicity of said input speech.
24. The apparatus of claim 22, wherein said characteristic of said
input speech is a characteristic of an adaptive highpass filter
used to enhance high frequency region of said excitation during
said coding.
25. The apparatus of claim 22, wherein said characteristic of said
input speech is a characteristic of an adaptive perceptual
weighting filter used in said Code Excited Linear Prediction
coder.
26. The apparatus of claim 22, wherein said characteristic of said
input speech is a characteristic of an adaptive Sinc window used in
said Code Excited Linear Prediction coder.
27. The apparatus of claim 22, wherein said voicing index selects
at least one sub-codebook from a plurality of sub-codebooks of said
Code Excited Linear Prediction coder based on said characteristic
of said input speech signal.
28. An apparatus for improving synthesized speech quality
comprising: a set of code parameters of an input speech signal; a
voicing index for use in enhancing synthesis of said input speech
signal from said code parameters; and a Code Excited Linear
Prediction coder using said code parameters and information
provided by said voicing index to generate a synthesized version of
said input speech signal.
29. The apparatus of claim 28, wherein said voicing index provides
periodicity of said input speech signal.
30. The apparatus of claim 28, wherein said voicing index provides
characteristics of a highpass filter used to enhance high frequency
region of said excitation during generation of said code parameters
for said input speech.
31. The apparatus of claim 28, wherein said voicing index provides
characteristics of an adaptive perceptual weighting filter used to
enhance perceptual quality of said input speech during generation
of said code parameters for said input speech.
32. The apparatus of claim 28, wherein said voicing index provides
characteristics of an adaptive Sinc window used to enhance
perceptual quality of said input speech during generation of said
code parameters for said input speech.
33. The apparatus of claim 28, wherein said voicing index selects
at least one sub-codebook from a plurality of sub-codebooks of said
Code Excited Linear Prediction coder based on characteristics of
said input speech signal.
34. A method of improving synthesized speech quality comprising:
generating a plurality of frames from an input speech signal;
coding each frame of said plurality of frames using a Code Excited
Linear Prediction coder to generate code parameters for synthesis
of said each frame of said input speech; and transmitting a voicing
index having a plurality of bits indicative of a classification of
said each frame of said input speech.
35. The method of claim 34, wherein said plurality of bits are
three bits.
36. The method of claim 34, wherein said classification is
indicative of periodicity of said input speech signal.
37. The method of claim 34, wherein said classification is
indicative of an irregular voiced speech signal.
38. The method of claim 34, wherein said classification is
indicative of a periodic index.
39. The method of claim 38, wherein said periodic index ranges from
low periodic index to high periodic index.
40. A method of improving synthesized speech quality comprising:
receiving a frame of an input speech signal, said frame having a
plurality of code parameters and a voicing index, wherein said
voicing index comprises a plurality of bits; determining a
classification for said frame of said input speech signal from said
plurality of bits of said voicing index; and decoding said frame
using a Code Excited Linear Prediction coder based on said
classification to synthesize said input speech.
41. The method of claim 40, wherein said plurality of bits are
three bits.
42. The method of claim 40, wherein said classification is
indicative of a noisy speech signal.
43. The method of claim 40, wherein said classification is
indicative of an irregular voiced speech signal.
44. The method of claim 40, wherein said classification is
indicative of a periodic index.
45. The method of claim 44, wherein said periodic index ranges from
low periodic index to high periodic index.
Description
RELATED APPLICATIONS
[0001] The present application claims the benefit of U.S.
provisional application serial No. 60/455,435, filed Mar. 15, 2003,
which is hereby fully incorporated by reference in the present
application.
[0002] The following co-pending and commonly assigned U.S. patent
applications have been filed on the same day as this application,
and are incorporated by reference in their entirety:
[0003] U.S. patent application Ser. No. ______, "SIGNAL
DECOMPOSITION OF VOICED SPEECH FOR CELP SPEECH CODING," Attorney
Docket Number: 0160112.
[0004] U.S. patent application Ser. No. ______, "SIMPLE NOISE
SUPPRESSION MODEL," Attorney Docket Number: 0160114.
[0005] U.S. patent application Ser. No. ______, "ADAPTIVE
CORRELATION WINDOW FOR OPEN-LOOP PITCH," Attorney Docket Number:
0160115.
[0006] U.S. patent application Ser. No. ______, "RECOVERING AN
ERASED VOICE FRAME WITH TIME WARPING," Attorney Docket Number:
0160116.
BACKGROUND OF THE INVENTION
[0007] 1. Field of the Invention
[0008] The present invention relates generally to speech coding
and, more particularly, to Code Excited Linear Prediction (CELP)
speech coding.
[0009] 2. Related Art
[0010] Generally, a speech signal can be band-limited to about 10
kHz without affecting its perception. However, in
telecommunications, the speech signal bandwidth is usually limited
much more severely. It is known that the telephone network limits
the bandwidth of the speech signal to between 300 Hz to 3400 Hz,
which is known as the "narrowband". Such band-limitation results in
the characteristic sound of telephone speech. Both the lower limit
at 300 Hz and the upper limit at 3400 Hz affect the speech
quality.
[0011] In most digital speech coders, the speech signal is sampled
at 8 kHz, resulting in a maximum signal bandwidth of 4 kHz. In
practice, however, the signal is usually band-limited to about 3600
Hz at the high-end. At the low-end, the cut-off frequency is
usually between 50 Hz and 200 Hz. The narrowband speech signal,
which requires a sampling frequency of 8 kb/s, provides a speech
quality referred to as toll quality. Although this toll quality is
sufficient for telephone communications, for emerging applications
such as teleconferencing, multimedia services and high-definition
television, an improved quality is necessary.
[0012] The communications quality can be improved for such
applications by increasing the bandwidth. For example, by
increasing the sampling frequency to 16 kHz, a wider bandwidth,
ranging from 50 Hz to about 7000 Hz can be accommodated, which is
referred to as the "wideband". Extending the lower frequency range
to 50 Hz increases naturalness, presence and comfort. At the other
end of the spectrum, extending the higher frequency range to 7000
Hz increases intelligibility and makes it easier to differentiate
between fricative sounds.
[0013] Digitally, speech is synthesized by a well-known approach
known as Analysis-By-Synthesis (ABS). Analysis-By-Synthesis is also
referred to as closed-loop approach or waveform-matching approach.
It offers relatively better speech coding quality than other
approaches for medium to high bit rates. A known ABS approach is
the so-called Code Excited Linear Prediction (CELP). In CELP
coding, speech is synthesized by using encoded excitation
information to excite a linear predictive coding (LPC) filter. The
output of the LPC filter is compared against the voiced speech and
used to adjust the filter parameters in a closed loop sense until
the best parameters based upon the least error is found. One of the
facts influencing CELP coding is that voicing degree can
significantly vary for different voiced speech segments thus
causing an unstable perceptual quality in the speech coding.
[0014] The present invention addresses the above
analysis-by-synthesis voiced speech issue.
SUMMARY OF THE INVENTION
[0015] In accordance with the purpose of the present invention as
broadly described herein, there is provided systems and methods for
improving quality of synthesized speech by using a voicing index to
control the speech coding process.
[0016] According to one embodiment of the present invention, a
voicing index is used to control and improve ABS type speech
coding, which indicates the periodicity degree of the speech
signal. The periodicity degree can significantly vary for different
voiced speech segments, and this variation causes an unstable
perceptual quality in analysis-by-synthesis type speech coding,
such as CELP.
[0017] The voicing index can be used to improve the quality
stability by controlling encoder and/or decoder, for example, in
the following areas: (a) fixed-codebook short-term enhancement
including the spectrum tilt, (b) perceptual weighting filter, (c)
sub-fixed codebook determination, (d) LPC interpolation, (e)
fixed-codebook pitch enhancement, (f) post-pitch enhancement, (g)
noise injection into the high-frequency band at decoder, (h) LTP
Sinc window, (i) signal decomposition, etc. In one embodiment for
CELP speech coding, the voicing index may be based on a normalized
pitch correlation.
[0018] These and other aspects of the present invention will become
apparent with further reference to the drawings and specification,
which follow. It is intended that all such additional systems,
methods, features and advantages be included within this
description, be within the scope of the present invention, and be
protected by the accompanying claims.
BRIEF DESCRIPTION OF DRAWINGS
[0019] FIG. 1 is an illustration of the frequency domain
characteristics of a sample speech signal.
[0020] FIG. 2 is an illustration of a voicing index classification
available to both the encoder and the decoder.
[0021] FIG. 3 is an illustration of a basic CELP coding block
diagram.
[0022] FIG. 4 is an illustration of a CELP coding process with an
additional adaptive weighting filter for speech enhancement in
accordance with an embodiment of the present invention.
[0023] FIG. 5 is an illustration of a decoder implementation with
post filter configuration in accordance with an embodiment of the
present invention.
[0024] FIG. 6 is an illustration of a CELP coding block diagram
with several sub-codebooks.
[0025] FIG. 7A is an illustration of sampling for creation of a
Sinc window.
[0026] FIG. 7B is an illustration of a Sinc window.
DETAILED DESCRIPTION
[0027] The present application may be described herein in terms of
functional block components and various processing steps. It should
be appreciated that such functional blocks may be realized by any
number of hardware components and/or software components configured
to perform the specified functions. For example, the present
application may employ various integrated circuit components, e.g.,
memory elements, digital signal processing elements, transmitters,
receivers, tone detectors, tone generators, logic elements, and the
like, which may carry out a variety of functions under the control
of one or more microprocessors or other control devices. Further,
it should be noted that the present application may employ any
number of conventional techniques for data transmission, signaling,
signal processing and conditioning, tone generation and detection
and the like. Such general techniques that may be known to those
skilled in the art are not described in detail herein.
[0028] Voicing index is traditionally one of the important indexes
sent to the decoder for Harmonic speech coding. The voicing index
generally represents the degree of periodicity and/or periodic
harmonic band boundary of voiced speech. Voicing index is
traditionally not used in CELP coding systems. However, embodiments
of the present invention use the voicing index to provide control
and improve the quality of synthesized speech in a CELP or other
analysis-by-synthesis type coder.
[0029] FIG. 1 is an illustration of the frequency domain
characteristics of a sample speech signal. In this illustration,
the spectrum domain in the wideband extends from slightly above 0
Hz to around 7 kHz. Although the highest possible frequency in the
spectrum ends at 8 kHz (i.e. Nyquist folding frequency) for a
speech signal sampled at 16 kHz, this illustration shows that the
energy is almost zero in the area between 7.0 kHz to 8 kHz. It
should be apparent to those of skill in the arts that the ranges of
signals used herein are for illustration purposes only and that the
principles expressed herein are applicable to other signal
bands.
[0030] As illustrated in FIG. 1, the speech signal is quite
harmonic at lower frequencies, but at higher frequencies the speech
signal does not remain as harmonic because the probability of
having noisy speech signal increases as the frequency increases.
For instance, in this illustration the speech signal exhibits
traits of becoming noisy at the higher frequencies, e.g., above 5.0
kHz. This noisy signal makes waveform matching at higher
frequencies very difficult. Thus, techniques like ABS coding (e.g.
CELP) becomes unreliable if high quality speech is desired. For
example, in a CELP coder, the synthesizer is designed to match the
original speech signal by minimizing the error between the original
speech and the synthesized speech. A noisy signal is unpredictable
thus making error minimization very difficult.
[0031] Given the above problem, embodiments of the present
invention use a voicing index which is sent to the decoder, from
the encoder, to improve the quality of speech synthesized by an ABS
type speech coder, e.g., CELP coder.
[0032] The voicing index, which is transmitted by the encoder to
the decoder, may represent the periodicity of the voiced speech or
the harmonic structure of the signal. In another example
embodiment, the voicing index may be represented by three bits thus
providing up to eight classes of speech signal. For instance, FIG.
2 is an illustration of a voicing index classification available to
both the encoder and the decoder. In this illustration, index 0
(i.e. "000") may indicate background noise, index 1 (i.e. "001")
may indicate noise-like or unvoiced speech signal, index 2 (i.e.
"010") may indicate irregular voiced signal such as voiced signal
during onset, and indices 3-7 (i.e. "011" to "111") could each
indicate the periodicity of the speech signals. For instance, index
3 ("011") may represent the least periodic signal and index 7
("111") may indicate the most periodic signal.
[0033] The voicing index information can be transmitted by the
encoder as part of each encoded frame. In other words, each frame
may include the voicing index bits (e.g. three bits), which
indicate the periodicity degree of that particular frame. In one
embodiment, the voicing index for CELP may be based on a normalized
pitch correlation parameter, Rp, and may be derived from the
following equation: 10 log(1-Rp).sup.2, where
-1.0<Rp<1.0.
[0034] In one example, the voicing index may be used for fixed
codebook short-term enhancement, including the spectrum tilt. FIG.
3 is an illustration of a basic CELP coding block diagram. As
illustrated, the CELP coding block 300 comprises the Fixed Codebook
301, gain block 302, Pitch filter block 303, and LPC filter 304.
CELP coding block 300 further comprises comparison block 306,
Weighting Filter block 320, and Mean Squared Error (MSE)
computation block 308.
[0035] The basic idea behind CELP coding is that Input Speech 307
is compared against the synthesized output 305 to generate error
309, which is the mean squared error. The computation continues in
a closed loop sense with selection of a new coding parameters until
error 309 is minimal.
[0036] On the receiving side, the decoder synthesizes the speech
using similar blocks 301-304 (see FIG. 5). Thus, the encoder passes
information to the decoder as needed to select the proper codebook
entry, gain, and filters, . . . , etc . . .
[0037] In a CELP speech coding system, when the speech signal is
more periodic, the pitch filter (e.g. 303) contribution is heavier
than the fixed codebook (e.g. 301) contribution. As a result, an
embodiment of the present invention may use the voicing index to
place more focus in the high frequency region by implementing an
adaptive high pass filter, which is controlled by the value of the
voicing index. An architecture such as the one shown in FIG. 4 may
be implemented. For instance, Adaptive Filter 310 could be an
adaptive filter emphasizing the power in the high frequency region.
In the illustration, the weighting filter 420 may also be an
adaptive filter for improving the CELP coding process.
[0038] On the decoder side, the voicing index may be used to select
the appropriate Post Filter 520 parameters. FIG. 5 is an
illustration of the decoder implementation with post filter
configuration. In one or more embodiments, Post Filter 520 may have
several configurations saved in a table, which may be selectable
using information in the voicing index.
[0039] In another example, the voicing index may be used in
conjunction with the perceptual weighting filter of CELP. The
perceptual weighting filter may be represented by Adaptive filter
420 of FIG. 4, for example. As is well known, waveform matching
minimizes the error in the most important portion (i.e. the high
energy portion) of the speech signal and ignores low energy area by
performing a mean squared error minimization. Embodiments of the
present invention use an adaptive weighting process to enhance the
low energy area. For instance, the voicing index may be used to
define the aggressiveness of the weighting filter 420 depending on
the periodicity degree of the frame.
[0040] In yet another embodiment, as illustrated in FIG. 6, the
voicing index may be used to determine the sub-fixed codebook.
There are possibly several sub-codebooks for the fixed codebook,
for example, one sub-codebook 601 with less pulses but higher
position resolution, one sub-codebook 602 with more pulses but
lower position resolutions, and a noise sub-codebook 603.
Therefore, if the voicing index indicates a noisy signal, then the
sub-codebook 602 or noisy sub-codebook 603 can be used; if the
voicing index does not indicate a noisy signal, then one of the
sub-codebooks (e.g. 601 or 602) may be used depending on the degree
of periodicity of the given frame. Note that the gain block
(codebook) 302 may also be applied individually to each
sub-codebook in one or more embodiments.
[0041] Further, the voicing index may be used in conjunction with
the LPC interpolation. For example, during linear interpolation,
the previous LPC is equally important as the current LPC if the
location of the interpolated LPC is at the middle between the
previous one and the current one. Thus, if the voicing index, for
example, indicates that the previous frame was unvoiced and the
present frame is voiced, then during the LPC interpolation, the LPC
interpolation algorithm may favor the current frame more than the
previous
[0042] The voicing index may also be used for fixed codebook pitch
enhancement. Typically, the previous pitch gain is used to perform
pitch enhancement. However, the voicing index provides information
relating to the current frame and, thus, could be a better
indicator than the previous pitch gain information. The magnitude
of the pitch enhancement may be determined based on the voicing
index. In other words, the more periodic the frame (based on the
voicing index value), the higher the magnitude of the enhancement.
For example, the voicing index may be used in conjunction with the
U.S. patent application Ser. No. 09/365,444, filed Aug. 2, 1999,
specification of which is incorporated herein by reference, to
determine the magnitude of the enhancements in the bi-directional
pitch enhancement system defined therein.
[0043] As a further example, the voicing index may be used in place
of pitch gain for post pitch enhancement. This is advantageous,
since, as discussed above, the voicing index may be derived from a
normalized pitch correlation value, i.e. Rp, which is typically
between 0.0 and 1.0; however, pitch gain may exceed 1.0 and can
adversely affect the post pitch enhancement process.
[0044] As another example, the voicing index may also be used to
determine the amount of noise that should be injected in the high
frequency band at the decoder side. This embodiment may be used
when the input speech is decomposed into a voiced portion and a
noise portion as discussed in pending U.S. patent application Ser.
No. ______, filed concurrently herewith, entitled "SIGNAL
DECOMPOSITION OF VOICED SPEECH FOR CELP SPEECH CODING",
specification of which is incorporated herein by reference.
[0045] The voicing index may also be used to control modification
of the Sinc window. The Sinc window is used to generate an adaptive
codebook contribution vector, i.e. LTP excitation vector, with
fractional pitch lag for CELP coding. In wideband speech coding, it
is known that strong harmonics appear in the low frequency area of
the band and the noisy signals appear in the high frequency
area.
[0046] Long-term prediction or LTP produces the harmonics by taking
a previous excitation and copying it to a current subframe
according to the pitch period. It should be noted that if a pure
copy of the previous excitation is made, then the harmonic is
replicated all the way to the end spectrum in the frequency domain.
However, that would not be an accurate representation of a true
voice signal and especially not in wideband speech coding.
[0047] In one embodiment, for wideband speech signal when the
previous signal is used to represent the current signal, an
adaptive low pass filter is applied to the Sinc interpolation
window, since there is a high probability of noise in high
frequency area.
[0048] In CELP coding, the fixed codebook contributes to coding of
the noisy or irregular portion of the speech signal, and a pitch
adaptive codebook contributes to the voice or regular portion of
the speech signal. The adaptive codebook contribution is generated
using a Sinc window, which is used due to the fact that the pitch
lag can be fractional. If the pitch lag were an integer, one
excitation signal could be copied to the next; however, because the
pitch lag is fractional, straight copying of the previous
excitation signal would not work. After the Sinc window is
modified, the straight copying would not work even for integer
pitch lag. In order to generate pitch contribution, several samples
are taken, as shown in FIG. 7A, which are weighted and then added
together, where the weights for the samples is called the Sinc
window, which originally has a symmetric shape, as shown in FIG.
7B. The shape in practice depends on the fractional portion of the
pitch lag and the adaptive lowpass filter applied to the Sinc
window. Application of the Sinc window is similar to convolution or
filtering, but the Sinc window is a non-causal filter. In the
representation shown below, a window signal w(n) is convoluted with
the signal s(n) in the time domain, which is an equivalent
representation to spectrum of the window W(w) multiplied by the
spectrum of the signal S(w) in the frequency domain:
U.sub.ACB(n)=w(n)*s(n).rarw..fwdarw.W(w)S(w).
[0049] According to the above representation, low passing of the
Sinc window is equivalent to low passing the final adaptive
codebook contribution (UACB (n)) or excitation signal; however, low
passing of the Sinc window is advantageous due to the fact that the
Sinc window is shorter than the excitation. Thus, it is easier to
modify the Sinc window than the excitation; further more, the
filtering of the Sinc window can be pre-calculated and
memorized.
[0050] In one embodiment of the present invention, the voicing
index may be used to provide information to control modification of
the low pass filter for the Sinc window. For instance, the voicing
index may provide information as to whether the harmonic structure
is strong or weak. If the harmonic structure is strong, then a weak
low pass filter is applied to the Sinc window, and if the harmonic
structure is weak, then a strong low pass filter is applied to the
Sinc window.
[0051] Although the above embodiments of the present application
are described with reference to wideband speech signals, the
present invention is equally applicable to narrowband speech
signals.
[0052] The methods and systems presented above may reside in
software, hardware, or firmware on the device, which can be
implemented on a microprocessor, digital signal processor,
application specific IC, or field programmable gate array ("FPGA"),
or any combination thereof, without departing from the spirit of
the invention. Furthermore, the present invention may be embodied
in other specific forms without departing from its spirit or
essential characteristics. The described embodiments are to be
considered in all respects only as illustrative and not
restrictive.
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