Telephonic communications – Audio message storage – retrieval – or synthesis – Voice activation or recognition
Reexamination Certificate
2000-07-10
2004-07-20
Foster, Roland (Department: 2645)
Telephonic communications
Audio message storage, retrieval, or synthesis
Voice activation or recognition
C704S231000, C704S275000, C704S501000, C704S503000
Reexamination Certificate
active
06765995
ABSTRACT:
BACKGROUND OF THE INVENTION
The present invention relates to a telephone system, and especially, to a telephone system having a voice recognition function which is used under transmission of a compressed voice signal.
CTI (Computer Telephony Integration) is being used, which is a system in which insufficient functions are mutually supplemented by integrating a system of a telephone (a PBX and so forth) and a system of information processing (a computer, a LAN and so forth) that were conventionally constructed separately. It is applied to reception work of a telecommunication sale company, telephone banking in a financial agency and so forth, for example, and saving of work is promoted by substituting conventional reception with a customer by an operator for a voice recognition device and a voice response device.
On the other hand, in such CTI, conversation by telephone via a conventional analog circuit is being changed to conversation by telephone (an internet phone and so forth) via a LAN, an ISDN, an internet and so forth. Accordingly, voice transmission by means of a conventional analog signal is shifting to voice transmission by means of a digital signal, and further, in order to suppress increase of transmission volume, a compression technology of voice is being used.
However, in a system in which a voice compression technology is applied to a voice recognition device, various tasks occur. Accordingly, such conventional tasks will be explained.
With regard to a technology of voice transmission by means of a digital signal, a unified standard is advised by a telecommunication standardization department (ITU-T) of an International Telecommunication Union (ITU). Presently, there are some standards, such as G.711 (PCM (:Pulse Code Modulation), 64 kbits/sec), G726 (ADPCM (:Adaptive Differential PCM), 32 kbits/sec), G.728 (LD-CELP (:Low Delay Code Excited liner Prediction), 16 kbits/sec), G.729 (CS-ACELP (:Conjugate Structure Algebraic CELP), 8 kbits/sec), and G.723.1 (MP-MLQ/ACELP, 6.3 k/5.3 k bits/sec). Out of them, a hybrid coding method, such as the G728, G.729 and G.723.1, has a higher compression ratio than that of a waveform coding method such as the G.711, and is expected to be a promising coding method in future.
FIGS. 6A
an
6
B are an explanation view for explaining a waveform coding method and a hybrid coding method. As shown in
FIG. 6A
, the waveform coding method is a method of conducting coding by sampling and quantizing a voice waveform. Accordingly, although, if a bit rate more than a certain value exists, voice of high quality can be obtained, there are tasks that a compression ratio is lowered by maintaining a high bit rate, and also, that voice quality is remarkably deteriorated when the bit rate is lowered.
On the other hand, as shown in
FIG. 6B
, the hybrid coding method is a method in which two kinds of information are complexly used, which are normalized information that is a previously prepared basic waveform pattern, and sound source information that is a difference between a waveform made of this normalized information and an original voice waveform. The normalized information is information in which for example a bit sequence of three bits is associated with the basic waveform pattern, and is stored in code books that are set on a transmission side and a reception side. Also, the sound source information is information provided by coding a difference by means of PCM between the original voice waveform and a waveform in which a plurality of basic waveform patterns are superimposed and rough shape of a voice waveform is reproduced, and is a signal including specific information of voice of a speaker, a background noise and so forth. Accordingly, since in the hybrid coding method most of parts of a voice waveform are represented by normalized information of about three bits, a compression rate of the hybrid coding method is higher than that of the waveform coding method. Also, by adding the sound source information that is a difference from the original voice waveform, there is an advantage that characteristic of voice of a speaker can be exactly reproduced, and voice of high quality can be produced.
As a telephone system in which such a hybrid coding method is adopted, there is a system which is defined in H.323. The H.323 is a standard of a conference system that is associated with a packet exchange network such as a LAN and an internet, based on H.320 that is international standard advice by ITU-T. It is mainly used for a personal computer conference system and so forth, and regards real-time characteristic as important. The G.723.1 and G.729 are corresponding to the coding of voice, and H.261 and H.263 are corresponding to the coding of an image.
FIG. 7
is a block diagram showing a conventional telephone system based on the H.323. As shown in the figure, the conventional telephone system is constructed of a telephone set
200
that is set on a side of a person who utilizes service, and a telephone set
201
for providing automatic response service of voice. The telephone set
200
and the telephone set
201
are connected to each other via a network
202
such as an internet. Also, a gate keeper
202
a
for conducting call control between the telephone sets, address conversion, bandwidth control and so forth is connected to the network
202
.
The telephone set
200
is constructed of a microphone
203
, an A/D converter
204
, an encoder
205
, a packeting device
206
, a network interface card (referred to as an NIC, hereinafter)
207
, a receiving buffer
208
, a depacketing device
209
, a decoder
210
, a D/A converter
211
, a speaker
212
, and a call controller
213
.
The telephone set
201
is a telephone set having an automatic voice response function, and is constructed of an NIC
214
, a receiving buffer
215
, a depacketing device
216
, a decoder
217
, a D/A converter
218
, a speaker
219
, a voice recognition and response device
220
, an encoder
221
, a packeting device
222
, and a call controller
223
.
The operation of such a conventional telephone set is as follows:
First, when the telephone set
201
is phoned from the telephone set
200
, call control is conducted by the call controllers
213
and
223
, and setting of a call, and so forth are conducted. Thereafter, information notification in relation to a terminal function is mutually conducted between the telephone set
200
and the telephone set
201
, and a channel in relation to voice is set.
When voice is input to the microphone
203
, the telephone set
200
on a dialing side converts it into an analog electric signal, and thereafter, supplies it to the A/D converter
204
. The A/D converter
204
converts the supplied analog electric signal into a digital signal, and thereafter, supplies it to the encoder
205
. The encoder
205
encodes the supplied signal, and thereafter, supplies it to the packeting device
206
. The packeting device
206
packets the supplied signal, and thereafter, supplies a packet signal to the NIC
207
. The NIC
207
transmits the supplied packet signal to the telephone set
201
via the network
202
.
When receiving the packet signal from the telephone set
200
, the NIC
214
of the telephone set
201
successively stores it in the receiving buffer
215
. The depacketing device
216
reads out the packet signal stored in the receiving buffer
215
, and converts it into a signal prior to being packeted and supplies it to the decoder
217
. The decoder
217
decodes the supplied signal, and supplies it to the D/A converter
218
or the voice recognition and response device
220
. In case that the signal is supplied to the speaker
219
via the D/A converter
218
, the voice sent from the telephone set
200
can be heard also on a side of the telephone set
201
. Also, the voice can be sent via a microphone
225
and an A/D converter
224
.
The voice recognition and response device
220
conducts voice recognition of the signal supplied from the decoder
217
, and makes a predetermined response. The voice recognition and response device
Foster Roland
NEC Infrontia Corporation
Young & Thompson
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