JPH0888607A - Digital telephone - Google Patents

Abstract

(57) [Abstract] [Purpose] An object of the present invention is to provide a digital telephone applicable to a digital communication network, which is capable of high-quality voice transmission. In a digital telephone equipped with a μ-PCM coding circuit 71 for performing μ-law PCM coding conversion of a voice signal to be transmitted / received, a frequency band of a transmission / reception voice signal is set to 2 in parallel with the μ-PCM coding circuit 71. S that doubles and sends and receives
A B-ADPCM encoding circuit 72 is provided, and a selection signal is detected from a digital signal transmitted / received via the line interface unit 60 according to an instruction from the ten-key controller 50.
According to the selection signal, the μ-PCM encoding circuit 71 and SB
-A control unit 40 for selecting the coding algorithm of the ADPCM coding circuit 72 is provided.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a digital telephone applied to a digital communication network, and more particularly to a digital telephone having a signal processing system for improving voice quality.

In recent years, integrated service digital network (ISD)
N) and the development of VLSI technology have increased the demand for economically realizing a realistic acoustic environment even between geographically distant points, and wideband acoustic signal coding has been attracting attention.

The present invention is intended to realize a high-quality digital telephone by applying these techniques to a telephone.

[0004]

2. Description of the Related Art The prior art will be described below with reference to FIG. FIG. 3 is a diagram showing the configuration of a conventional telephone set according to an embodiment. In FIG. 3, 101 is transmission (T) and reception (R)
And 102 is a speaker having a hands-free function. In addition, 103 and 104 are hybrid transformers (HT), and 105 and 106 are amplification circuits for transmission and reception.

Further, 110 is a μ-law pulse code modulation (μ-
PCM) encoding system is applied to the μ-PCM encoding circuit. The μ-PCM encoding circuit 110 has 8 bits and 8 kH.
z sampling analog / digital conversion circuit (8
b ADC) 111 and a digital / analog conversion circuit (8b DAC) 112.

Further, 121 is a ten-key circuit, and 122
Is an interface circuit having the functions of ISDN layers 2 and 3, and 123 is a line interface unit.
Reference numeral 131 is a power supply circuit at the time of power failure, and 132 is a commercial power supply circuit that receives AC power from the AC adapter 133.

The operation of the conventional circuit shown in FIG. 3 will be briefly described below. [For sending] Send (T) from handset 101 0.2
An analog voice signal of .about.3.4 kHz passes through the routes of HT103, HT104 and the amplifier circuit 105 and is input to the 8b ADC111 in the .mu.-PCM encoding circuit 110.

The 8b ADC 111 performs sampling of 8 bits and 8 kHz by applying the μ-PCM coding method to the input analog voice signal and converts it into a digital voice signal having a bit rate of 64 kbps. Then, the digital audio signal resulting from this conversion is transferred to the line interface unit.
Enter in 123.

Further, the ten-key circuit 121 sends out a code signal created by inputting the telephone number of the other party, the interface circuit 122 converts this code signal into a bit rate of 8 to 16 kbps, and the conversion result The code signal is input to the line interface unit 123.

In the line interface section 123, 8bA
The digital voice signal (64 kbps) from DC111 and the code signal (8 to 16 kbps) from the interface circuit 122 are multiplexed and converted into a digital line signal with a bit rate of 144 kbps, and this 144 kbps digital line signal is modular J
To the digital line terminating device (DSU). [In the case of reception] The digital line signal having a bit rate of 144 kbps transmitted from the DSU is received by the line interface unit 123 via the modular J.

In the line interface section 123, 64 kbps
The digital audio signal of the bit rate of is extracted and the extracted 64 kbps digital audio signal is
To enter.

In the 8bDAC112, the digital voice signal having a bit rate of 64 kbps is converted into an analog voice signal (0.2 to
It demodulates to 3.4 kHz, and the demodulation result is divided into two and output.
On the other hand, the analog voice signal resulting from the demodulation passes through the route of the amplifier circuit 106 and the HT104 and HT103, and then the handset 101
Then, the handset 101 receives (S).

The other analog audio signal resulting from the demodulation is monitor-received by the hands-free speaker 102. And the sidetone when making a call using the handset 101 is
The handset 101 sends it in the form of an analog signal from the transmission (T) side to the reception (R) side via the HT 103, and receives it without processing. [Other Operations] The normal power supply is made by supplying AC power from the AC adapter 133 to the commercial power supply circuit 132. The power supply at the time of power failure is from the modular J to the power supply circuit at power failure 13
It is made from the digital line signal supplied to 1.

As described above, the .mu.-PCM coding method is applied to the digitization of the voice signal, and the voice signal is transmitted from the line interface section 123 at a bit rate of 144 kbps using a digital network such as ISDN network. To do.

Further, the generation of the sidetone in the handset 101 is processed in an analog manner simply by using the signal passing through the HT 103. In terms of voice quality, since the μ-PCM coding method is applied to digitize the voice signal, the frequency band of the voice signal of 20 kHz is 4 kHz.
The quality is limited to the following, the quality is kept to the extent that it does not hinder the call, and the signal processing of music signals is not considered.

Recently, due to the construction of a network such as an ISDN network, general telephones, which have conventionally communicated in an analog amount, are being digitized. However, when a digital telephone is connected to an ISDN network, it is basically limited to a digital amount with a bit rate of 64 kbps. Here, the μ-PCM coding system is generally used as a coding system for converting an audio signal into a digital amount having a bit rate of 64 kbps.

However, when the μ-PCM coding method is applied, the presence of voice quality is lost because the frequency band is limited to 4 kHz or less, and a signal having a wide frequency band such as an audio signal is transmitted. Becomes impossible,
The voice quality becomes extremely poor.

[0018]

Therefore, in the conventional digital telephone, since the frequency band is limited to 4 kHz or less, there is a problem that high-quality transmission of audio signals and the like becomes impossible.

An object of the present invention is to provide a digital telephone capable of high quality voice transmission.

[0020]

In order to achieve the above object, as shown in FIGS. 1 and 2, analog / digital conversion and 16 kHz of a voice signal transmitted from a handset 11 are performed.
16b ADC31 and handset for sampling
Digital-to-analog conversion of received voice signal to 11 and 16 k
ADC with 16b DAC32 for sampling Hz
/ DAC unit 30, a μ-PCM encoding circuit 71 for performing μ-law PCM encoding conversion of a voice signal transmitted / received by the ADC / DAC unit 30, and a numeric keypad control unit for outputting a code signal corresponding to the telephone number of the other party. 50 and the μ-PCM encoding circuit described above
In a digital telephone equipped with a line interface unit 60 that multiplexes both signals output by the 71 and the numeric keypad control unit 50 and transmits them to a digital line, or decodes a received signal from the digital line, the frequency band of transmitted and received voice signals Bandwidth division type adaptive difference P for transmitting and receiving with twice the
A CM encoding circuit (SB-ADPCM encoding circuit) 72 is provided in parallel with the μ-PCM encoding circuit 71, and digital transmission / reception is performed via the line interface unit 60 according to an instruction from the ten-key controller 50. A control unit 40 is provided which detects a selection signal from the signal and selects the coding algorithm of the μ-PCM coding circuit 71 and the SB-ADPCM coding circuit 72 by the selection signal.

The digital telephone is provided with an audio system processing section 80 for sampling and converting an input digital audio signal, and a voice signal transmitted from the handset 11 and a digital signal after sampling conversion are provided.
The audio signals are added in the control unit 40, and the addition result is transmitted to the digital line through the SB-ADPCM encoding circuit 72.

Further, in the digital telephone, the transmission voice signal from the handset 11 and the reception voice signal from the SB-ADPCM encoding circuit 72 are added to the control unit 40 to digitally generate sidetone. Adder to
40 and a digital
The attenuator 42 is provided.

Further, in the digital telephone,
An MD or DCC interface circuit 83 for interfacing with MD or DCC is provided in the audio system processing unit, the input audio signal is recorded in the MD or DCC for storage, and processed by the SB-ADPCM encoding circuit 72 during a call. Configured to allow recording of audio signals.

Furthermore, in the digital telephone, by controlling the signal route of the digital audio signal input through the audio system processing section 80 by the control section 40 to form the received voice, the speaker can be used even when the telephone is not in use. Configured to allow monitoring of the audio signal via.

[0025]

In the present invention, as shown in FIGS. 1 and 2, a μ-PCM encoding conversion of a voice signal transmitted / received by the handset 11, the ADC / DAC unit 30 and the ADC / DAC unit 30 is performed.
The PCM encoding circuit 71, the ten-key controller 50 for outputting a code signal corresponding to the telephone number of the other party, and the above μ
-In a digital telephone having a line interface unit 60 for multiplexing both signals output from the PCM encoding circuit 71 and the ten-key control unit 50 or decoding a received signal,
An SB-ADPCM coding circuit 72 is provided in parallel with the μ-PCM coding circuit 71 so that the frequency band of the transmitted / received audio signal is 2
Further, it is possible to double the transmission and reception, and further, by providing a control unit 40, a selection signal is detected from a digital signal transmitted and received through the line interface unit 60 in response to an instruction from the ten-key control unit 50, and this selection signal is detected. The above μ-PCM
The coding algorithm of the coding circuit 71 and the SB-ADPCM coding circuit 72 can be selected.

Further, by sampling-converting the digital audio signal input by the audio system processing unit 80, the transmission voice signal from the handset 11 and the digital audio signal after sampling conversion are added in the control unit 40. Enable the result of this addition to the SB-ADP
The data is transmitted to the digital line through the CM encoding circuit 72.

Further, an adder circuit 43 is provided in the control unit 40, and the transmission voice signal from the handset 11 and the S signal are transmitted.
The received voice signals from the B-ADPCM encoding circuit 72 are added to digitally add the side tone generation, and the digital attenuator 42 is provided so that the side tone generation amount can be adjusted. There is.

Furthermore, in the audio system processing section, M
MD or DCC to interface with D or DCC
An interface circuit 83 is provided to record the input audio signal in an MD or a DCC for storage, and SB
-Enable recording of a voice signal during a call processed by the ADPCM encoding circuit 72, or switch a signal route of a digital audio signal input via the audio processing unit 80 by the control unit 40. By using the received voice, the audio signal can be monitored through the speaker even when the telephone is not used.

[0029]

EXAMPLE An example of the present invention will be described below with reference to FIG. FIG. 2 is a diagram showing the configuration of a telephone according to an embodiment of the present invention. In FIG. 2, 11 is a handset having the functions of transmission (T) and reception (R), 12 is a speaker having a hands-free function, 21 is an amplifier circuit for transmission,
22 is an amplifier circuit for reception.

Reference numeral 31 is a 16-bit analog / digital conversion circuit (16b ADC). This 16b ADC 31 converts the analog (A) signal into a digital (D) signal.
It is a circuit that performs sampling of 16 kHz in bits.

Reference numeral 32 denotes a 16-bit digital / analog conversion circuit (16bDAC). This 16bDAC 32 is a 16-bit converter for converting a digital (D) signal into an analog (A) signal.
It is a circuit that performs sampling of 16 kHz in bits.

A circuit composed of 31 16b ADCs and 32 16b DACs is a circuit shown as ADC / DAC 30 in FIG. Reference numeral 40 is a control unit that performs selection processing of an encoding algorithm and switching of signal routes. The control unit 40 includes
A digital mixing circuit 41, a digital attenuator 42, an adding circuit 43, a level control circuit 44 and a switch 45 are provided.

Reference numeral 50 is a ten-key controller. The ten-key controller 50 is provided with a ten-key circuit 51 for inputting the telephone number of the other party and an interface circuit 52 having the functions of layers 2 and 3 of ISDN. Reference numeral 60 denotes a line interface section that interfaces with the ISDN network via the modular J.

Reference numeral 71 is a μ-to which the μ-PCM coding system is applied.
Reference numeral 72 is a PCM coding circuit, and 72 is an SB-ADPCM coding circuit (band division adaptive differential PCM coding circuit).
Further, reference numeral 80 is an audio processing unit that processes the original music on hold and background music (BGM). The audio processing unit 80 includes a sampling conversion circuit 81, a storage type compression encoding circuit 82, a mini disk or a digital compact cassette (MDorDCC) interface unit 83.
Is provided.

Reference numeral 84 is an audio device interface circuit for interfacing with audio devices. Further, 91 is a power supply circuit at the time of power failure, and 92 is a commercial power supply circuit that receives AC power from the AC adapter 93.

The operation of the circuit of the present invention shown in FIG. 2 will be described below. In addition, the handset 11 described in the present invention
And the speaker 12, the numeric keypad control unit 50 including the numeric keypad circuit 51 and the interface circuit 52, and the configuration and function of each circuit of the power supply circuit 91 at the time of power failure and the power supply circuit at the time of commercial power supply 92, the conventional circuits 101, 102, the circuits 121, 122 and the circuit parts. 131,13
It is the same as 2 and duplicate explanation shall be avoided.

In the embodiment of the present invention, as shown in FIG.
By adding the B-ADPCM encoding circuit 72,
-To be able to double the frequency band with the same digital quantity converted by the PCM coding scheme.

As a decoding method for audio signals and the like, a storage compression encoding method (for example, Moving Picture Expert Group / Audio: MPEG / Audi established by ISO / IEC JTC1 / WG11) is used.
World standard encoding method called o) is applied to enable processing of audio signals.

Further, by providing a sampling conversion circuit 81 for mutually exchanging the SB-ADPCM coding circuit 72 and the storage type compression coding circuit 82, communication of audio signals is enabled.

In addition, the side tone generation is digitally processed by the control unit 40, so that the side tone can be generated while the frequency band is widened without paying attention to analog noise. Hereinafter, the configuration of the present invention will be described in detail. [Transmission operation] 0.2 to 8 transmitted (T) from the handset 11
An analog audio signal of kHz (twice the bandwidth of conventional) is
It is input to the 16b ADC 31 through the amplifier circuit 21.

In the 16b ADC31, the input analog voice signal is converted into a digital signal and 16-bit, 16 kH
Sampling of z (twice the number of bits as in the prior art) is performed, and the digital audio signal resulting from this conversion is added to the digital mixing circuit 41 of the control unit 40.

Normal voice signal (frequency band: 0.2-4 k)
(Hz) is transmitted, the digital mixing circuit 41 does not digitally mix the music signal with the digital audio signal, and passes the digital audio signal as it is to the μ-PCM encoding circuit 71.

In this case, in the μ-PCM encoding circuit 71,
The input digital voice signal is processed by the μ-PCM coding method and the bit rate is 64 kbps (the same speed as before).
The digital audio signal of the conversion result is input to the line interface unit 60.

On the other hand, in the case of transmitting a high-quality voice signal and music signal (frequency band: 0.2 to 8 kHz), the digital mixing circuit 41 digitally mixes the music signal with the digital sound signal. Then, the digital signal resulting from this mixing is input to the SB-ADPCM encoding circuit 72.

In this case, the SB-ADPCM encoding circuit 72
Then, the input digital audio signal is divided into two bands (low band: 50
-4 kHz, high frequency band: 4-8 kHz), each band is ADPCM encoded and converted into a digital voice signal with a bit rate of 64 kbps (the same speed as in the past), and then the high quality digital result of this conversion is obtained. The audio signal is added to the line interface unit 60.

In the line interface section 60, the μ-PC
The digital audio signal (64 kbps) from the M coding circuit 71 or the SB-ADPCM coding circuit 72 and the code signal (8 to 16 kbps) from the ten-key controller 50 are multiplexed and converted into a digital line signal having a bit rate of 144 kbps. The 144 kbps digital line signal resulting from the conversion is transmitted to the DSU via the modular J.

It should be noted that the encoding system at the time of transmission is μ-PCM.
The selection process of the encoding algorithm such as selecting either the encoding circuit 71 or the SB-ADPCM encoding circuit 72 and the switching of the signal route are performed by the line interface unit 60 judging the code signal output from the ten-key controller 50. , Control unit
Determined by the selection signal sent to 40. [In case of receiving only] The digital line signal having a bit rate of 144 kbps transmitted from the DSU is received by the line interface unit 60 via the modular J.

In the line interface unit 60, 144 kbps
A digital voice signal having a bit rate of 64 kbps and a selection signal for deciding an encoding method used at the time of receiving are extracted from the digital line signal having a bit rate of.

Then, the extracted digital audio signal is added to the μ-PCM coding circuit 71 or the SB-ADPCM coding circuit 72 to perform a predetermined process. A selection signal which is another extraction result is added to the control unit 40, and the μ-PCM coding circuit 71 or the SB-ADPC is used in the control unit 40 as in the case of transmission.
Selection processing of any one of the M coding circuits 72 and switching of the signal route are performed.

In the following, processing is carried out by a route completely opposite to the case of transmission to demodulate into an analog signal, and the handset 11 receives (R)
At the same time, the speaker 102 receives the monitor. In addition,
The side tone of the call using the handset 11 is digitally processed by the route of 16b ADC circuit 31 → digital mixing circuit 41 → digital attenuator 42 → adding circuit 43 → contact a of switch 45 → 16b DAC circuit 32.

As described above, in the transmission / reception by the telephone of the present invention, the analog / digital conversion or the digital / analog conversion of the voice signal is performed by the conventional 8-bit 2-bit conversion.
Double 16 bits, and the converted signal is subjected to modulation processing and demodulation processing by μ-PCM coding method or SB-ADPCM coding method, respectively, to create a digital line signal (64 kbps) with the same bit rate as the conventional one, and ISDN. Send and receive via the network. [Music Signal Addition Operation] First, the operation of the control unit 40 will be described. In the control unit 40, the music signal processed by the audio system processing unit 80 is received from the sampling conversion circuit 81, and the digital signal converted by the 16b ADC circuit 31 and the music signal are digitally mixed by the digital mixing circuit 41. BGM for a call is added.

Digi BG in digital mixing circuit 41
The volume of M is adjusted by the level control circuit 44 so that the presence or absence of BGM can be selected. The digital signal, which has been digitally mixed by the digital mixing circuit 41, is returned to the adding circuit 43 via the digital attenuator 42 and is added by the adding circuit 43 to the μ-PCM coding circuit 71.
Alternatively, the signal is added to the received signal from the SB-ADPCM coding circuit 72, and the addition result is added to the signal route switch 45.
The side contact is generated digitally by activating the contact a of the above and adding it to the 16b DAC32.

Generally, the magnitude of the sidetone signal in the analog system is generally 6 dB or more smaller than the generated sound. Therefore, the amount of sidetone generated is determined by the digital attenuator.
Adjust with 42.

Further, the initial setting value of the analog audio signal adjusted by the digital attenuator 22 is 6 dB, so that it can be arbitrarily adjusted. Further, the music signal processed by the audio system processing unit 80 is disconnected from the normal call route (contact point a) so that the contact point b of the switch 45 for changeover is made effective, so that the music signal is monitored by the broadband speaker 12. You can also

Next, the function of the audio processing unit 80 will be described. The MPEG / Audio digital audio signal in the frequency band of 20 kHz, which is input from the audio equipment interface circuit 84, is stored in the storage type compression encoding circuit 82.
Then, the signal is compressed and recorded via the MD / DCC interface circuit 83 in the MD or DCC as a storage medium as an original hold sound.

The voice signal decoded by the SB-ADPCM coding circuit 72 is sampled by the sampling conversion circuit 81 at the sampling rate (40 kbps) of the audio signal and further compressed by the storage type compression coding circuit 82. With this, voice during a call can be recorded.

Further, the original hold tone signal already recorded in the MD or DCC is added to the storage type compression encoding circuit 82 via the MDorDCC interface circuit 83 and converted into a 16 kbps digital voice signal and then to the switch 45. By sending the signal and switching the signal route so that the contact b becomes effective and sending it to the 16b DAC 32, it is possible to use it for adding BGM at the time of receiving a voice and for monitoring by the speaker 12.

Furthermore, the original hold tone signal already recorded in MDorDCC is added to the storage type compression encoding circuit 82 via the MDorDCC interface circuit 83,
Alternatively, the digital audio signal in the frequency band of 20 kHz input from the audio equipment interface circuit 84 is added to the storage type compression encoding circuit 82.

Then, the storage type compression encoding circuit 82 converts the input original hold tone signal or digital audio signal into a digital signal having a bit rate of 40 kbps, and the sampling conversion circuit 81 outputs SB-ADPC.
The M-encoding circuit 72 performs sampling conversion to the sampling rate (64 kbps), and the digital mixing circuit 41 digitally mixes the digital audio signal to be transmitted.

By this operation, the line interface section
It is also possible to transmit high-quality music signals such as music signals to the other party via 60. As mentioned above,
By digitally mixing a high-quality music signal having a bandwidth of 8 kHz with normal transmission / reception voice, high-quality transmission / reception of a voice signal becomes possible.

[0061]

As is apparent from the above description, according to the present invention, it is possible to apply the μ-PCM or SB-ADPCM coding method to a voice signal which has been conventionally limited to a band of 4 kHz or less. With this, it is possible to reproduce a sound signal with a realistic sensation, and it is effective in that it greatly contributes to the improvement of call quality.

Further, by extending the voice band up to 8 kHz, it is possible to pass an audio signal, and there is an effect that it greatly contributes to high functionality and multimedia of the digital telephone.

[Brief description of drawings]

FIG. 1 is a diagram showing the basic configuration of a telephone according to the present invention.

FIG. 2 is a diagram showing a configuration of a telephone according to an embodiment of the present invention.

FIG. 3 is a diagram showing a configuration of a conventional telephone according to an embodiment.

[Explanation of symbols]

11 Handset 12 Speaker 20 Amplifier 30 Analog / digital converter and digital / analog converter (ADC / DAC part 31 16-bit analog / digital converter (16bAD
C) 32 16-bit digital / analog converter (16bDA
C) 40 Control unit 41 Digital mixing circuit 42 Digital attenuator 43 Addition circuit 44 Level control circuit 45 Switch 50 Numeric keypad control unit 51 Numeric keypad circuit 52 Interface circuit 60 Line interface unit 71 μ-law PCM coding circuit (μ-PCM coding circuit ) 72 Band Division Adaptive Differential PCM Coding Circuit (SB-AD
PCM encoding circuit) 80 Audio processing unit 81 Sampling conversion circuit 82 Storage type compression encoding circuit 83 MDorDCC interface circuit

Claims (5)

[Claims] 1. Performing analog / digital conversion and 16 kHz sampling of a voice signal transmitted from a handset.
Performs digital / analog conversion and 16 kHz sampling of received voice signals to 16b ADC and handset
ADC / DAC unit with 16b DAC and the ADC / D
A μ-PCM coding circuit that performs μ-law PCM coding conversion of a voice signal transmitted and received by the AC unit, a ten-key control unit that outputs a code signal corresponding to the telephone number of the other party, and the aforementioned μ
-In a digital telephone equipped with a line interface unit that multiplexes both signals output from the PCM encoding circuit and the numeric keypad control unit and transmits them to a digital line, or decodes a received signal from the digital line. The band-division type adaptive differential PCM encoding circuit for transmitting and receiving with the frequency band doubled is the μ-PC.
It is provided in parallel with the M encoding circuit, and further detects a selection signal from a digital signal transmitted and received through the line interface section according to an instruction from the ten-key control section, and uses the selection signal to detect the μ-PC.
A digital telephone characterized by comprising a control unit for selecting an encoding algorithm of an M encoding circuit and a band division type adaptive difference PCM encoding circuit. 2. The digital telephone according to claim 1, further comprising an audio system processing section for sampling and converting an input digital audio signal, wherein the transmission voice signal from the handset and the digital audio signal after the sampling conversion are provided. 2. The digital telephone according to claim 1, wherein the addition is performed in the control unit, and the addition result is transmitted to the digital line through the band division type adaptive differential PCM encoding circuit. 3. The digital telephone according to claim 1, wherein the control unit adds a transmission voice signal from the handset and a reception voice signal from the band division type adaptive differential PCM encoding circuit to generate a side tone. 2. The digital telephone according to claim 1, further comprising an adding section for digitally generating the sound and a digital attenuator for adjusting the amount of the side sound generated. 4. The digital telephone according to claim 1, wherein an MD or DCC interface circuit for interfacing with an MD or a DCC is provided in the audio processing unit, and the input audio signal is recorded in the MD or the DCC for storage, and 2. The digital telephone according to claim 1, wherein a voice signal during a call processed by a band division type adaptive differential PCM encoding circuit can be recorded. 5. The digital telephone according to claim 1, wherein the digital telephone is inputted through an audio processing unit.
2. The digital telephone according to claim 1, wherein the audio signal can be monitored through the speaker even when the telephone is not in use by switching the signal route of the audio signal by the control unit to form the received voice.