JPH02233034A - Signaling information transmission system - Google Patents

Abstract

PURPOSE:To prevent deterioration in the audio quality by inserting a signaling bit data at the sender side into an invalid bit of a digital audio signal and extracting the signaling bit data at the receiver side. CONSTITUTION:An introduced analog audio signal is converted into a digital audio signal with a prescribed length at a 1st conversion means 111. A signaling bit data is inserted to an ineffective bit of the digital audio signal at an insertion means 115 and the resulting data is sent to a transmission line from a transmission means 117. The data sent to the transmission line is received by a reception means 150, fed to an extraction means 151, in which the signaling bit data is extracted. Thus, even when the digital audio signal and the signaling information are multiplexed, no missing exists in the digital audio signal and the deterioration in the audio quality is prevented.

Description

[Detailed Description of the Invention] [Table of Contents] Overview Industrial Field of Application Conventional Technology Problems to be Solved by the Invention Examples of Means and Actions for Solving Problems ■. Correspondence between the example and FIG. 1■. Overall configuration and operation of the embodiment■. Configuration and operation of multiplex circuit■. Configuration and operation of extraction circuitV. Summary of Examples■. Variations of the invention Effects of the invention [1 Already required] Introduced for the purpose of preventing deterioration of voice quality regarding a signaling information transmission system that transmits signaling information necessary for switching processing in digital telephone networks, etc. a first converting means for converting an analog audio signal into a digital audio signal of a predetermined length including invalid bits; a signaling bit generating means for outputting signaling bit data for controlling an exchange process of the analog audio signal; inserting means for inserting data into invalid bits of the digital audio signal output from the first converting means; sending means for sending the data output from the inserting means to a transmission path;
receiving means for receiving data output from the sending means;
and extracting means for extracting signaling bit data included in the data received by the receiving means.

[Industrial application field]

The present invention relates to a signaling information transmission system for transmitting signaling information necessary for switching processing in a digital telephone network or the like. [Prior Art] As the amount of information increases, transmission using digital signals that can multiplex signals and transmit them is increasing. This digital transmission is also used in telephone networks.

In a digital telephone network, on the transmitting side, analog voice signals output from telephones are converted into digital voice signals, frames are created and multiplexed, and on the receiving side, the received frames are separated and converted from digital voice signals. It converts it into an analog voice signal and supplies it to the destination telephone.

FIG. 8 shows the configuration of a digital signal converter and frame transmitter on the transmitting side and a frame receiver and analog signal converter on the receiver side in a conventional digital telephone network.

In the figure, n telephones 211+, 211z, 2 1
1! ,..., 211n are connected to n channels 8101.810g, 810, . . . via an exchange not shown.
..., connected to 810A. n channels 81
0. , 810. , 8103, . . . , 810,, are connected to a multiplex circuit (MUX) 8 1 3.

The telephone 2111 outputs an analog voice signal (V) and starts.
Signaling information (S) for controlling responses etc. is supplied,
It is converted into a digital audio signal by channel 8'IO+ and supplied to MUX8 1 3. Here, each of the channels 810 on the transmitting side includes a sampling circuit 213,
Anaguchi digital (A/D) converter 215. It includes a multiplexing section 811 and a code converter 217.

The sampling circuit 213 converts the signaling information supplied from the telephone 211I into a digital signal and outputs the digital signal. Further, the A/D converter 215 converts the analog audio signal supplied from the telephone 2111 into a logarithm P CM (Log
PCM) method is used to convert the audio signal into a digital audio signal.

The signal output from the sampling circuit 213? The switching information is inserted into the least significant bit of the digital audio signal output from the A/D converter 215 at a constant cycle in the multiplexing section 811. The digital audio signal output from this multiplexing section 81l is supplied to the code converter 217 and converted into a straight line P C
It is converted into a digital audio signal of M (Lin PCM) system and is supplied to MUX813. MUX81
3, in this way, the telephones 211+, 211■, 21
13, . . . , 211, are inserted into a plurality of time slots to create a frame, and then sent to the transmission path.

In this way, A/D conversion in each channel 810 is performed by first converting an analog audio signal into a digital audio signal using the logarithmic PCM method, and then converting the analog audio signal into a digital audio signal using the linear
This is to convert it into a CM format digital audio signal.

On the receiving side, the separation circuit (DMUX) 8 1 5 is m
channels 820. ,820z. 8203,...
・,820■. m channels 820
1.820■,820:l,...,820. teeth? m telephones 231, 231■,
2313,...,231. It is connected to.

The frame sent from MUX813 is DMUX815
are separated into digital audio signals in units of time slots through m channels 820 . ,820■,
8203,...,820. supplied to either.

For example, signaling information is extracted from the least significant bit at regular intervals from a digital audio signal supplied to channel 820, and is output to an exchange (not shown).

Frames sent from MUX8 1 3 are DMUX8
It was also received on 17th. The frame is separated into time slot units via the DMUX 817, converted into an analog audio signal using the linear PCM method, and output to the speaker 229. When outputting an analog audio signal to an audio device such as the speaker 229, data with good sound quality was obtained by directly converting the linear PCM digital audio signal into an analog audio signal.

[Problem to be solved by the invention]

By the way, in the above-described conventional signaling information transmission system, since signaling information is periodically inserted into the least significant bit of a digital audio signal on the transmitting side, omissions occur in the audio signal and errors occur. Furthermore, the digital audio signal into which signaling information has been inserted is converted into a linear PCM digital audio signal, and during this conversion, the influence of errors caused by the signaling information may be magnified. Therefore, there is a problem in that the audio quality on the receiving side deteriorates due to these errors occurring on the transmitting side.

The present invention was created in view of these points, and an object of the present invention is to provide a signaling information transmission method that prevents deterioration of voice quality due to the transmission of signaling information.

[Means to solve the problem]

FIG. 1 is a principle block diagram of the signaling information transmission system of the present invention.

In the figure, the first conversion means 111 converts the introduced analog audio signal into a digital audio signal of a predetermined length including invalid bits.

The signaling bit creation means 113 outputs signaling bit data for controlling the exchange process of analog audio signals.

The inserting means 115 inserts the signaling bit data into the invalid bits of the digital audio signal output from the first converting means 111.

The sending means 117 sends the data output from the inserting means 115 to the transmission path.

The receiving means 150 receives the data output from the sending means 117.

The extracting means 151 extracts the signaling bit data included in the data received by the receiving means 150. Therefore, the overall configuration is such that the signaling pit data is inserted into the invalid bits of the digital audio signal and transmitted.

[For production]

The introduced analog audio signal is converted into a digital audio signal of a predetermined length by the first conversion means 111. The inserting means 115 inserts signaling bit data into the invalid bits of this digital audio signal, and the sending means 11
7 to the transmission path.

The data sent out to the transmission path is received by the receiving means 150, and is supplied to the extracting means 151, where signaling bit data is extracted.

In the present invention, the transmitting side inserts the signaling bit data into the invalid bits of the digital audio signal, and the receiving side extracts this signaling bit data, so even if the digital audio signal and the signaling information are multiplexed, the digital audio signal There are no omissions, and it is possible to prevent deterioration of voice quality.

〔Example〕

Hereinafter, embodiments of the present invention will be described in detail based on the drawings.

FIG. 2 shows the configuration of a signaling information transmission system in an embodiment of the present invention.

Here, the correspondence relationship between the embodiment of the present invention and FIG. 1 will be shown.

The first conversion means 111 corresponds to an A/D converter 215 and a code converter 217.

The signaling bit creation means 113 corresponds to the sampling circuit 213.

The insertion means 115 corresponds to the multiplexing section 219.

The sending means 117 corresponds to the MUX 251. The receiving means 150 corresponds to the DMUX 251.

The extraction means 151 corresponds to the extraction circuit 223.

Examples of the present invention will be described below assuming that the correspondence relationship as described above exists.

■ In Figure 2, the digital telephone network consists of a telephone 211 that outputs an analog voice signal, a channel 210 that converts the signal format on the transmitting side of a digital transmission path, a MUX 251 that creates frames, and a received signal. The system includes a DMUX 252 that separates frames in units of time slots, a channel 220 that converts signal formats on the receiving side of a digital transmission path, and a telephone 231 that is supplied with analog voice signals.

Here, both the transmitting side and the receiving side have a plurality of channels, but one channel is shown for each in FIG. 2. Further, in FIG. 2, the same reference numerals as in FIG. 8 indicate the same circuit parts.

The transmission side channel 210 includes a sampling circuit 213,
A/D converter 2l5, code converter 217, multiplexer 21
9, and is connected to the telephone 211 via an exchange (not shown). Further, the channel 220 on the receiving side includes a code converter 221, an extraction circuit 223 . D/A converter 225
．． It includes a D/A converter 227 and is connected to a telephone 231 and a speaker 229 via an exchange (not shown). In the figure, the analog voice signal from the telephone 211 is A
The signal is supplied to the /D converter 215 and converted into an 8-bit digital audio signal using the logarithmic PCM method.

This digital audio signal is converted into a 16-bit linear PCM digital audio signal by a code converter 217.

The telephone 211 outputs analog voice signals as well as signaling information for controlling activation, response, etc., and is supplied to the sampling circuit 213. sampling circuit 213
converts the signaling information into a digital signal and outputs it, and also outputs a status signal indicating the presence or absence of signaling information to be inserted into the time slot.

FIG. 3(a) shows the configuration of the 8-bit digital audio signal input to the code converter 217 and the configuration of the 16-bit digital audio signal output. In the figure, the most significant bit of the 8 bits in the logarithmic PCM method is the sign bit S
For example, if the polarity of the analog audio signal represented by alternating current is positive, "0" is inserted, and if it is negative, "1" is inserted.

The other 7 bits are data that encodes the absolute value of an analog audio signal represented by alternating current.

When 8-bit data in the logarithmic PCM method is converted to the linear PCM method, the most significant bit is used as the sign bit S, as shown in Figure 3 (a), and this sign bit S and 2 bits (shaded area ), a total of 14 bits, including the lower 13 bits placed in
(See figure (b)). The same data as the sign bit is usually (conventionally) inserted into the two bits (hatched portion) that are not used as the digital audio signal.

Multiplexer 219 multiplexes the digital audio signal, signaling information, and status signal into 16-bit data.

Here, the signaling information and the status signal are inserted into the bits (hatched portions) in which the digital audio signal is not inserted among the 16 bits (see FIG. 3(a)) output by the code converter 217. The MUX 251 inserts the 16-bit data thus created and the 16-bit data supplied from another channel (not shown) into each time slot to create a frame, and sends this frame to the transmission path.

On the receiving side, the frame sent out by the MUX 251 is received by the DMUX 252 and separated into 16-bit units (time slots). One of the separated 16-bit data is provided to channel 220. channel 22
The 16-bit data supplied to code converter 22
1 and the extraction circuit 223 or the D/A converter 227 and the extraction circuit 223.

The extraction circuit 223 extracts signaling information from the 14th and 15th bits of the input data (a 16-bit digital audio signal based on the linear PCM method). Based on this signaling information, the exchange receives a call to the telephone 231 and performs billing.

The code converter 221 converts the input data into an 8-bit logarithm P
Convert to digital audio signal by CM Fou2. This 8-bit digital audio signal is converted into an analog audio signal by the logarithmic PCM method via the D/A converter 225, and is supplied to the telephone 231.

Further, the D/A converter 227 converts the input data into linear PCM
The analog audio signal is converted into an analog audio signal and supplied to the speaker 229.

4.Cardinal L of the mouth J Figure 4 shows the configuration of the multiplexing section 219 used in this embodiment. In the figure, the multiplexing section 219 includes a 16-bit shift register 413 and an AND circuit 411.

The digital audio signal output from the code converter 217 is supplied in parallel to the 1st to 13th bits and the 16th bit of the shift register 413. The 14th and 15th bits of the digital audio signal output by the code converter 217 are not supplied to the shift register 413. Further, the code converter 217 supplies the shift register 413 with a load signal (described later) and a clock signal (described later).

A status signal is supplied from the sampling circuit 213 to the 14th bit of the shift register 413. The sampling circuit 21 is connected to the two human terminals of the AND circuit 411.
3 provide status signals and signaling information, respectively.

When the AND circuit 411 is supplied with a logic level “1” as a status signal, it outputs the logic level supplied as signaling information to the 15th bit of the shift register 413.

FIG. 5 shows the timing of the operation of the multiplexer 219. In the figure, the "clock signal" is the clock signal by which the shift register 413 performs the shift operation, and the "sampling clock signal" is the clock signal with a frequency of 8 (kHz) for digitizing the signaling information used in the sampling circuit 213.
) and "load signal" indicate a signal for loading the signal into the shift register 413, respectively.

Furthermore, the signaling information shown in FIG. 5(d) is shown in FIG.
C) The signaling information (signaling information supplied by the telephone 211) shown in the sampling clock signal (the fifth
(See Figure (b)).

First, a 16-bit parallel digital audio signal is output from the code converter 217, and a load signal is output in accordance with this output operation.

In response to the fall of the load signal, the digital audio signal is taken into the 1st bit to the 13th bit and the 16th bit of the shift register 413.

Further, the status signal (see FIG. 5(g)) is taken into the 14th bit of the shift register 413 in synchronization with the fall of the load signal. Furthermore, the signaling information output from the AND circuit 411 is taken into the 15th bit of the shift register 413 in synchronization with the fall of the load signal.

The parallel data taken into the shift register 413 is converted into serial data and output (see FIG. 5(f)) in synchronization with a clock signal (see FIG. 5(a)).

Thus, in the 16-bit data, a digital audio signal is inserted into the first to 13th bits and the 16th bit, and a status signal and signaling information are inserted into the 14th and 15th bits.

(2) and FIG. 6 show the configuration of the extraction circuit 223 used in this embodiment. In the figure, the extraction circuit 223 includes a 16-bit shift register 611, an AND circuit 613 . It includes 615 D-type flip-flops (D-FF).

The shift register 611 is supplied with a clock signal and a latch signal (described later) from the DMUX 252, and data sent from the MUX 251. The latch signal is also supplied to one input terminal of the AND circuit 613. The 14th bit output of the shift register 611 is sent to the AND circuit 6
The 15th bit output is input to the input terminal D of the D-FF 615. The output of the AND circuit 613 is input to the clock input terminal C of the D-FF 615. The output signal of the extraction circuit 223 is D-
It is output from the output terminal Q of FF615.

FIG. 7 shows the timing of the operation of the extraction circuit 223.

In the figure, a "latch signal" is a signal for loading data into the shift register 611.

The 16-bit serial data supplied from the DMUX 252 is sequentially taken into the shift register 611 in response to a clock signal (see FIG. 7(a)). A latch signal is output in response to the completion of 16-bit capture (Fig. 7(b)
)reference).

If the data (status signal) of the 14th bit of the shift register 611 is at the logic level "1°" (Fig. 7(c)
), the output of the AND circuit 613 becomes logic level "1" in response to the rise of the latch signal. The output of this AND circuit 613 falls in response to the fall of the latch signal. Further, even if the latch signal rises, if the data of the 14th bit (status signal) is at the logic level "O", the output of the AND circuit 613 does not rise.

The output of the AND circuit 613 obtained in this way is D-FF6
The read signal is supplied to the clock input terminal C of No. 15.

The D-FF 615 takes in and outputs the 15th bit data (signaling information) (see FIG. 7(d)) in synchronization with the falling edge of the read signal supplied to the clock input terminal C (see FIG. 7(d)). (see (e)).

Therefore, a read signal is output based on the status signal of the 14th bit, and signaling information of the 15th bit is output from the D-FF6 15 in synchronization with this read signal.

In this way, signaling information is extracted based on the 14th and 15th bits.

■ Stop When a digital audio signal based on the logarithmic PCM method is converted into a digital audio signal based on the linear PCM method in this way, the 14th. Signaling information and a status signal are inserted into the 15th bit and transmitted to the receiving side. Therefore, since signaling information is not inserted into the least significant bit of the digital audio signal to be transmitted as in the conventional method, there is no loss of audio signals, and deterioration of audio quality can be prevented.

Furthermore, in the past, since signaling information was inserted periodically, it was affected by frame phase delays, but according to this embodiment, there is no need to be conscious of timing adjustments due to delays.

Furthermore, in order to periodically insert signaling information, a circuit for counting the phase and extracting the signaling information was required, but this circuit can be omitted.
The circuit scale can be reduced.

■ Note that in the embodiments described above, a status signal indicating the presence or absence of signaling information was inserted, but the signaling information is inserted periodically into unused bits in the time slot. Also good.

Also, rl. In “Correspondence between Examples and FIG. 1”,
Although the correspondence relationship between the present invention and the embodiments has been described, those skilled in the art will easily guess that the present invention is not limited to this and that there are various modifications.

〔Effect of the invention〕

As described above, according to the present invention, the sending side inserts signaling bit data into the invalid bits of the digital audio signal, and the receiving side extracts this signaling bit data, so the digital audio signal and the signaling information are multiplexed. Also, there is no dropout in the digital audio signal, making it possible to prevent deterioration of audio quality, which is extremely useful in practice.

[Brief explanation of the drawing]

FIG. 1 is a principle block diagram of the signaling information transmission method of the present invention, FIG. 2 is a configuration block diagram of the signaling information transmission method according to an embodiment of the present invention, FIG. 3 is an explanatory diagram of transmission data, and FIG. 4 is The configuration diagram of the multiplex circuit, Figure 5 is the timing diagram of the operation of the multiplex circuit, Figure 6 is the configuration diagram of the extraction circuit, Figure 7 is the timing diagram of the operation of the extraction circuit, and Figure 8 is the configuration diagram of the conventional example. be. In the figure, 111 is a first conversion means, 113 is a signaling bit creation means, 115 is an insertion means, 117 is a sending means, 150 is a receiving means, 151 is an extraction means, 210, 220, 810, 211. 231 is a telephone, 213 is a sampling circuit, 820 is a channel, 5 is an A/D converter, 7.221 is a code converter, 9,811 is a multiplexer, 3 is an extraction circuit, 5,227 is a D/A converter, 9 is a speaker, 1,813 is a MUX, 2,815.817 is a DMUX, 1.613 is an AND circuit, 3,611 is a shift register, 5 is a D-FF, and 3 is a separation section. (a) 14th floor, 7th floor (b) Road Figure 5: Kazuya Soda East episode with 7 payouts

Claims (1)

[Claims] (1) First conversion means (1
11); a signaling bit generating means (113) for outputting signaling bit data for controlling the exchange process of the analog audio signal;
11), an insertion means (115) for inserting into invalid bits of the digital audio signal output from the
) to send out data output from the transmission path (
117), receiving means (150) for receiving data output from the sending means (117), and extracting means (15) for extracting the signaling bit data included in the data received by the receiving means (150).
1) A signaling information transmission system characterized by being configured to include the following.