BACKGROUND OF THE INVENTION
1. Field of the Invention
[0001] The present invention relates to a coding of a mobile communications terminal, and
particularly; to a voice coding apparatus and method using a Perceptual Linear Prediction
(PLP).
2. Background of the Related Art
[0002] As mobile communication techniques are developed, mobile communications terminals
have provided data communications using numbers, characters, symbols, and the like,
and multimedia communications including various image signals as well as voice communications.
A plurality of terminal users receive radio channels allocated thereto from a system
and transmit and receive required data using radio resources. However, the radio channels
have limited bandwidths in order for the plurality of users to use the radio channels
at the same time, and accordingly a data bit rate of each user is deservedly limited.
[0003] Therefore, a coding technique has been proposed for transmitting a greater amount
of data using above limited data bit rate. Various methods exist as the related art
voice coding technique, each of which has several advantages at a certain bit rate.
[0004] For instance, a speech coding using a generic audio coding, a Pulse Code Modulation
(PCM), and an Adaptive Delta Pulse Code Modulation(ADPCM) are effectively used at
a high-bit rate over 16Kbps, and a Code Excited Linear Prediction (CELP) and other
various variations are effectively used at a medium-bit rate at a range of 2.4Kbps
to 16Kbps. In particular, a coding method using LD-CELP, CS-ACELP, VSELP and MELP
and a wideband speech coding can be used at the medium-bit rate. Also, a Linear Predictive
Coding (LPC), Residual Excited Linear Predictive (RELP), formants vocoder and Cepstral
vocoder have many advantages at a low-bit rate at a range of 75bps to 2.4Kbps.
[0005] Thus, in the related art and the present invention, a method for improving the LPC
among coding methods used at the low-bit rate will now be explained.
[0006] Fig. 1 illustrates a structure of the related art LPC encoder.
[0007] As illustrated in the drawing, the related art LPC encoder includes: a correlator
10 for calculating an autocorrelation value r
x[n] of an input signal x[n]; an LP coefficient calculator 11for calculating an LP
coefficient a
L and a gain G by processing the autocorrelation value r
x[n]; a V/UV determining unit 12 for determining whether the input signal x[n] is a
voiced V signal or a unvoiced UV signal; a pitch calculator 13 for calculating a pitch
P of the corresponding signal when the input signal x[n] is the voice V signal; a
parameter coding unit 14 for outputting a bit stream by coding the LP coefficient
an, the gain G and the pitch P received from the LP coefficient calculator 11 and
the pitch calculator 13 according to a V/UV indication bit outputted from the V/UV
determining unit 12.
[0008] An operation of the related art LPC encoder having such construction will now be
explained.
[0009] First, the correlator 10 autocorrelates an input signal x[n]. The LP coefficient
calculator 11 processes an autocorrelation value r
x[n] calculated by the correlator 10 so as to calculate an LP coefficient a
n and a gain G. At this time, the V/UV determining unit 12 determines whether the input
signal x[n] is a voiced V signal or a unvoiced UV signal to output a V/UV indication
bit, and then outputs only the voiced V signal. The pitch calculator 13 calculates
a pitch P of the voiced V signal which is outputted from the V/UV determining unit
12.
[0010] Accordingly, when the V/UV indication bit indicates the voiced V signal, the parameter
coding unit 14 outputs a bit stream by coding (encoding by a low-bit rate) the LP
coefficient a
n, the gain G, and the pitch P received from the LP coefficient calculator 11 and the
pitch calculator 13. Afterwards, a controller (not shown) processes the bit stream
to thusly output it to a radio (wireless) unit (not shown). The radio unit converts
the signal outputted from the control unit into a radio (wireless) signal and transmits
the converted radio signal.
[0011] Thus, in the related art, a mobile communications terminal performs the LPC coding
to transmit an audio signal by a low-bit rate. However, in the related art LPC coding,
a linear predication coefficient is generally used, which does not consider human
auditory sensing features. Therefore, for the related art LPC coding operated using
the low-bit rate, a compression efficiency is not very high (i.e., 1200Kbps to 2400Kbps)
and good sound quality can not be obtained.
SUMMARY OF THE INVENTION
[0012] Therefore, an object of the present invention is to provide a voice coding apparatus
and method of a mobile communications terminal capable of improving compression efficiency
and sound quality by performing an LPC coding using a PLP coefficient.
[0013] To achieve these and other advantages and in accordance with the purpose of the present
invention, as embodied and broadly described herein, there is provided a Linear Predictive
Coding (LPC) encoder of a mobile communications terminal comprising: a Perceptual
Linear Prediction (PLP) coefficient calculator for calculating a PLP coefficient and
a gain by processing an input signal; a V/UV determining unit for determining whether
the input signal is a voiced signal or a unvoiced signal, and thusly outputting the
determination signal and the voiced signal when the input signal is the voiced signal;
a pitch calculator for calculating a pitch of the input signal outputted from the
V/UV determining unit; and a parameter coding unit for performing a low-bit rate coding
using the PLP coefficient, the gain, and the pitch on the basis of the determination
signal.
[0014] To achieve these and other advantages and in accordance with the purpose of the present
invention, as embodied and broadly described herein, there is provided a low-bit rate
voice coding method of a mobile communications terminal comprising: calculating a
Perceptual Linear Prediction (PLP) coefficient and a gain by processing an input signal;
determining whether the input signal is a voiced signal and a unvoiced signal, and
thereby outputting a determination bit value and the voiced signal when the input
signal is determined as the voiced signal; calculating a pitch of the input signal
outputted from a V/UV determining unit; and performing a low-bit rate coding using
the PLP coefficient, the gain and the pitch on the basis of the determination bit
value.
[0015] Preferably, the voiced signal is a speech signal.
[0016] Preferably, the PLP coefficient has about a 7
th degree for a 8 kHz sampling rate.
[0017] The foregoing and other objects, features, aspects and advantages of the present
invention will become more apparent from the following detailed description of the
present invention when taken in conjunction with the accompanying drawings.
BRIEF DESCRIPTION OF THE DRAWINGS
[0018] The accompanying drawings, which are included to provide a further understanding
of the invention and are incorporated in and constitute a part of this specification,
illustrate embodiments of the invention and together with the description serve to
explain the principles of the invention.
[0019] In the drawings:
Fig. 1 illustrates a structure of a related art LPC encoder using an LP coefficient;
Fig. 2 illustrates an LPC encoder using a PLP coefficient according to the present
invention; and
Fig. 3 illustrates sequential steps, in detail, of calculating a PLP coefficient in
Fig. 2.
DETAILED DESCRIPTION OF THE INVENTION
[0020] Reference will now be made in detail to the preferred embodiments of the present
invention, examples of which are illustrated in the accompanying drawings.
[0021] The present invention provides a low-bit rate voice coding using a Perceptual Linear
Prediction (PLP) capable of performing a coding of a degree (an order) lower than
that of a Linear Predictive Coding (LPC) in order to perform a voice coding having
high compressibility.
[0022] First, a difference between the PLP and the LP will now be explained.
[0023] The LP is classically well-known, so that a detailed derived formula therefor will
not be described. The LP basically refers to obtaining a LP coefficient a
k so that a Mean Squared Error (MSE), namely, a value of e[n] can be a minimum value
according to Formula (1) as follows.
[0024] The obtained LP coefficient
ak has about 8
th to 12
th degrees (orders) for a 8 kHz sampling rate. Therefore, the obtained LP coefficient
ak is used for various coding methods (e.g., LPC, CELP, MELP, RELP, etc) using a Linear
Prediction (LP), which is disclosed in more detail in Speech coding and synthesis,
Amsterdam, the Netherlands: Elsevier, 1995.
[0025] The PLP was introduced on a paper of Hermansky in 1990 for the first time. The PLP
uses human auditory sensing features similar to the existing Mel-Frequency Cepstral
Coefficient (MFCC). Therefore, the present invention performs a low-bit rate voice
coding using the PLP coefficient in stead of using the LP coefficient upon performing
the LPC for a low-bit rate.
[0026] That is, the present invention obtains spectrum using the PLP coefficient. The PLP
coefficient reflects a human auditory effect. Accordingly, in aspect of the MSE, a
greater error may occur in the spectrum using the PLP coefficient than using the LP.
However, the spectrum using the PLP coefficient may have a less error when considering
the auditory effect. Also, for coefficient transmissions, in case of LPC, for a typical
8kHz sampling rate, transmissions of about a 10
th degree (order) are used, but for PLP, transmissions of about a 7
th degree (order) are used, thus the bit rate can be lowered.
[0027] Fig. 2 illustrates a construction of an LPC encoder using the PLP coefficient according
to the present invention.
[0028] Referring to the Fig. 2, an LPC encoder using the PLP coefficient is constructed
as same as the related art LPC encoder shown in Fig. 1, except of which the correlator
10 is not included and a PLP coefficient calculator 20 replaces the LP coefficient
calculator 11.
[0029] The PLP coefficient calculator 20 processes a speech signal S[n] to calculate a PLP
coefficient a
P and a gain G in which the auditory effect is considered.
[0030] An operation of the LPC encoder using the PLP coefficient having such construction
according to the present invention will now be explained with reference to the accompanying
drawing.
[0031] First, the PLP coefficient calculator 20 receives the speech signal S[n], so as to
calculate the PLP coefficient a
P and the gain G by sequentially performing operations shown in Fig. 3.
[0032] That is, the PLP coefficient calculator 20 performs a fast Fourier transform (FFT)
of the input signal, namely, the speech signal S[n]. A critical-bank integration and
resampling processing is performed for the Fourier-transformed speech signal to thusly
remove noise components from the speech signal S[n] by a frequency unit.
[0033] Once removing the noise components, the PLP coefficient calculator 20 performs equalizing
and loudness processing of the Fourier-transformed speech signal into sound components
having magnitudes appropriate for human auditory sensing, and then the speech signal
is matched with an output power to allow listening by humans.
[0034] When the power matching is completed, the PLP coefficient calculator 20 performs
an inverse discrete Fourier transform of the corresponding speech signal to thereafter
obtain a set of Linear equations from the corresponding speech signal. Therefore,
the PLP coefficient calculator 20 performs a Cepstral Recursion processing for the
set of Linear equations, and thus outputs Cepstral Coefficients of a PLP model, namely,
the PLP coefficients ap. In other words, the PLP coefficient calculator 20 outputs
to the parameter coding unit 23 a low degree (order) of the PLP coefficients a
P and a gain G reflecting the human auditory sensing features as parameter values.
[0035] At this time, the V/UV determining unit 21 outputs a V/UV Indication bit and transfers
the speech signal S[n] to the pitch calculator 22. The pitch calculator 22 calculates
a pitch P of the speech signal S[n].
[0036] Accordingly, the parameter coding unit 23 outputs a bit stream by coding (encoding
by a low-bit rate) the V/UV Indication bit value, the PLP coefficient a
P, the gain G and the pitch P received from the PLP coefficient calculator 20 and the
pitch calculator 22. Preferably, a degree of the transmitted PLP coefficient a
P is about a 7
th degree for a 8 kHz sampling rate. Afterwards, a controller (not shown) processes
the bit stream and then outputs the processed bit stream to a radio (wireless) unit
(not shown). The radio unit converts the signal outputted from the controller into
a radio signal (wireless signal) and transmits it.
[0037] As described above, in the present invention, the LPC is performed by using the PLP
coefficient, and thus a compressibility can be improved and voice-grade signal can
be transmitted by a more efficient low-bit rate.
[0038] In addition, in the present invention, a higher compressibility can be realized and
a quality of signal with high sound quality can be expected by using the PLP coefficient
as a parameter rather than using the existing LP coefficient.
[0039] Therefore, the voice coding apparatus and method according to the present invention
can be used for coding and decoding voice using a low-bit rate, or be used for a device
which takes up a small area and performs a voice synthesis using PLP parameters.
[0040] Furthermore, the voice coding apparatus and method according to the present invention
can be used for a speech coding for an application as much as a voice itself is not
very important but enough to hear. Also, an effective voice conversation can be performed
on the Internet which stores data by a high compressibility or requires a low-bit
rate in an embedded system with a limited memory.
[0041] As the present invention may be embodied in several forms without departing from
the spirit or essential characteristics thereof, it should also be understood that
the above-described embodiments are not limited by any of the details of the foregoing
description, unless otherwise specified, but rather should be construed broadly within
its spirit and scope as defined in the appended claims, and therefore all changes
and modifications that fall within the metes and bounds of the claims, or equivalence
of such metes and bounds are therefore intended to be embraced by the appended claims.
1. A voice coding apparatus in a mobile communications terminal comprising:
a Perceptual Linear Prediction (PLP) coefficient calculator for calculating a PLP
coefficient and a gain by processing an input signal;
a V/UV determining unit for determining whether the input signal is a voiced signal
or a unvoiced signal, and thus outputting a determination results and the voiced signal
when the input signal is the voiced signal;
a pitch calculator for calculating a pitch of the input signal outputted from the
V/UV determining unit; and
a parameter coding unit for performing a low-bit rate coding using the PLP coefficient,
the gain, and the pitch on the basis of the determination results.
2. The apparatus of claim 1, wherein the voiced signal is a speech signal.
3. The apparatus of claim 1, wherein the determination results denotes a bit value for
whether the input signal is the voiced signal or the unvoiced signal.
4. The apparatus of claim 1, wherein a degree of the PLP coefficient is about a 7th degree for a 8 kHz sampling rate.
5. A voice coding method of a mobile communications terminal comprising:
calculating a Perceptual Linear Prediction (PLP) coefficient and a gain by processing
an input signal;
determining whether the input signal is a voiced signal and a unvoiced signal, and
thereby outputting the determination signal and the voiced signal when the input signal
is determined as the voiced signal;
calculating a pitch of the input signal outputted from a V/UV determining unit; and
performing a low-bit rate coding using the PLP coefficient, the gain and the pitch
on the basis of the determination signal.
6. The method of claim 5, wherein the voiced signal is a speech signal.
7. The method of claim 5, wherein the step of calculating the PLP coefficient and the
gain comprises:
performing a fast Fourier transform (FFT) for the input signal;
performing a critical-bank integration and resampling of the Fourier transformed speech
signal to thus remove noise components by a frequency unit;
performs equalizing and loudness processing of the Fourier-transformed speech signal
into sound components having magnitudes appropriate for human auditory sensing, and
then matching the speech signal with an appropriate output power;
performing an inverse discrete Fourier transform of the speech signal matched with
the output power, and thereby obtaining a set of linear equations; and
performing a ceptstral recursion processing for the set of linear equations, and
thereby obtaining a PLP coefficient and a gain.
8. The method of claim 5, wherein a degree of the PLP coefficient is about a 7th degree for a 8 kHz sampling rate.