Data processing: speech signal processing – linguistics – language – Speech signal processing – Synthesis
Reexamination Certificate
2000-04-17
2003-02-04
McFadden, Susan (Department: 2654)
Data processing: speech signal processing, linguistics, language
Speech signal processing
Synthesis
C704S270000
Reexamination Certificate
active
06516298
ABSTRACT:
BACKGROUND OF THE INVENTION
1. Technical Field of the Invention
The present invention relates to a method for carrying out information transmission by using speech sounds on a portable telephone, Internet or the like.
2. Description of the Related Art
Speech sound communication systems are constructed by connecting transmitters and receivers via wire communication paths such as coaxial cables or radio communication paths such as electromagnetic waves. Though, in the past analog communications were the mainstream where acoustic signals are propagated directly or by being modulated into carrier waves on those communication paths, digital communications have been becoming mainstream where acoustic signals are propagated after being coded once for the purpose of increasing communication-quality with respect to anti-noise properties or distortion and increasing the number of communication channels.
Recent communications systems, such as portable telephones, use the CELP (Schroeder M. R. and Atal B. S.: “Code-Excited Linear Prediction (CELP): High-Quality Speech at Very Low Bit Rates,” Pros. IEEE ICASSP '85, 25.1.1, (April 1985)) system to correct the deficiencies of transmission radio wave bands caused by the rapid spread of such communications systems.
FIG. 7
shows an exemplary configuration example of the CELP speech coding and decoding system.
The processing on the coding end, that is, on the transmission terminals end is as follows. Speech sound signals are processed by partition into frames of, for example, 10 ms or the like. The inputted speech sounds undergo LPC (Linear Prediction Coding) analysis at the LPC analysis part
200
to be converted to a LPC coefficient &agr;
i
representing a vocal tract transmission function.
The LPC coefficient &agr;
i
is converted and quantized to a LSP (Line Spectrum Pair) coefficient &agr;
qi
at an LSP parameter quantization part
201
. &agr;
qi
is given to a synthesizing filter
202
to synthesize a speech sound wave form by a voicing wave form source read out from an adaptive code book
203
corresponding to a code number c
a
. The speech sound wave form is inputted as a periodic wave form in accordance with a pitch period T
0
calculated out by using an auto-correlation method or the like in parallel with the previous processing.
The synthesized speech sound wave form is subtracted from the inputted speech sound to be inputted into a distortion calculation part
207
via an auditory weighting filter
206
. The distortion calculation part
207
calculates out the energy of the difference between the synthetic wave form and the inputted wave form repetitively while changing the code number c
a
for the adaptive code book
203
and determines the code number c
a
that makes the energy value the minimum.
Then the voicing source wave form read out under the determined c
a
and the noise source wave form read out according to the code number c
r
from the noise code book
204
are added to determine the code number c
r
that makes the distortion minimum following similar processing. The gain values are also determined which are to be added to both voicing source and noise source wave forms through-the previously accomplished processing so that the most suitable gain vector corresponding to them is selected from the gain code book to determine the code number c
g
.
The LSP coefficient &agr;
qi
, the pitch period T
0
, the adaptive code number c
a
, the noise code number c
r
, the gain code number c
g
which have been determined as described above are collected into one data series to be transmitted on the communication path.
On the other hand, the processing on the decoding end, that is, on the reception terminal end, is as follows.
The data series received from the communication path is again divided into the LSP coefficient &agr;
qi
, the pitch period T
0
, the adaptive code number c
a
, the noise code number c
r
, and the gain code number c
g
. The periodic voicing source is read out from the adaptive code book
208
in accordance with the pitch period T
0
and the adaptive code number c
a
, and the noise source wave form is read out from the noise code book
209
in accordance with the noise code number c
r
.
Each voicing source receives an amplitude adjustment by the gain represented by the gain vector read out from the gain code book
210
in accordance with the gain code number c
g
to be inputted into the synthesizing filter
211
. The synthesizing filter
211
synthesizes speech sound in accordance with the LSP coefficient &agr;
qi
.
The speech sound communication system as described above has the main purpose of propagating speech sound efficiently with a limited communication path capacitance by compression coding inputted speech sound. That is to say the communication object is solely speech sound emitted by human beings.
Today's communications services, however, are not limited to only speech sound communications between human beings in distant locations but services such as e-mail or short messages are becoming widely used where data are transmitted to a remote reception terminal by inputting text utilizing transmission terminals. And it has become important to provide speech sound from apparatuses to human beings such as those supplying a variety of information by speech sound represented by the CTI (Computer Telephony Integration) or providing operating methods of the apparatuses in speech sound. Moreover, by using the speech sound rule synthesizing technology which converts text information into speech sound it has become possible to listen to the contents of e-mails, news or the like on the phone, which has been attracting attention recently.
In this way it has been required to have a communication service form to convert text information into speech sound. The following two forms are considered as methods to implement those services.
One is a method for transmitting speech sound synthesized on the. service supplying end to the users by using normal speech sound transmissions. In the case of this method the terminal apparatuses on the reception end only receive and reproduce the speech sound signals in the same way as the prior art and common hardware can be used.
Vocalizing a large amount of text, however, means to keep speech sounds flowing for a long period of time into the communication path and in the case of using communication systems such as portable telephones it becomes necessary to maintain the connection for a long period of time. Accordingly, there is the problem that communication charges becomes too expensive.
The other is a method for letting the users hear the speech sound converted by a speech sound synthesizing apparatus of the reception terminals after the information is transmitted on the communication path in the form of text. In the case of this method the information transmission amount is an extremely small amount such as one several hundredths of a speech sound which makes it possible to be transmitted in a very short period of time. Accordingly, the communication charges are held low and it becomes possible for the user to listen to the information by conversion into speech sounds whenever desired if the text is stored in the reception terminal. There is also an advantage that different types of voices such as male or female, speech rates, high pitch or low pitch or the like can be selected at the time of conversion to speech sounds.
The speech sound synthesizing apparatus to be installed as a terminal apparatus on the reception end, however, has different circuits from that used as an ordinary reception terminal such as a portable telephone, therefore, new circuits for synthesizing speech sounds should be mounted, which leads to the problem that the circuit scale is increased and the cost for the terminal apparatus is increased.
SUMMARY OF THE INVENTION
Considering such a conventional problem of the communication method, it is the purpose of the present invention to provide a speech sound communication system which has a smaller communication burden and has a simpler speech synthesizin
Kamai Takahiro
Matsui Kenji
Weizhong Zhu
Matsushita Electric - Industrial Co., Ltd.
McFadden Susan
RatnerPrestia
LandOfFree
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