JPH03157027A - Multiplex transmission system - Google Patents

Abstract

PURPOSE:To attain efficient transmission without causing a nonuse bit in a transmission format by converting a multi-frame into a transmission format having the time slot of a bit number corresponding to coding speed and transmission speed of each time slot. CONSTITUTION:A multiplex section 36 multiplexes plural coded voice signals with different coding speed and plural low speed system data with different transmission speed as multi-frame constitution allocated to separate time slots, and converts the multi-frame into a transmission format having the time slot of the bit number corresponding to each time slot coding speed and transmission speed. Moreover, a packet data is multiplexed for a unvoiced period of voice signal and for an invalid period in which no low speed system data not processed as a packet exists and the result is sent to a transmission line 50. Thus, no non-use bit exists in the transmission format, efficient transmission is attained and flexible network buildup is attained.

Description

DETAILED DESCRIPTION OF THE INVENTION [Field of Industrial Application] The present invention relates to a multiplex transmission system, and pertains to a multiplex transmission system in which packet data is multiplexed and transmitted in a silent section of voice and an invalid section of low-speed data.

Data in the format shown is sent to the transmission line 14, and this data is separated into the first to fourth channels and packet data by the separation unit, and the first to fourth channels and packet data are separated into the first to fourth channels and packet data. The second channel is decoded by the audio interface section 16, and the third channel is decoded by the audio interface section 16. The fourth channel is output through the low-speed interface section 17.

[Conventional technology]

Conventionally, there is a multiplexing method in which packet data is inserted into each silent section of audio or an invalid section where no low-speed data that has not been packetized exists and is sent out to a transmission path for highly efficient multiplexing.

As shown in Fig. 5, this method uses the voice interface section 1
1 encoded with 1. The audio data of the second channel and the third data input to the low-speed data interface unit 12. The low-speed data of the fourth channel is multiplexed with the packet data by the multiplexer 13. At this time, the first. The silent section of the second channel and the third channel. Packet data is inserted into the invalid section of the fourth channel. The multiplexing unit 13 is shown in FIG.
Problems to be Solved by the Invention] The encoding speed of audio signals, that is, the speed at which audio can be detected, is only one class H, and even in low-speed data, the unit for detecting the presence or absence of data, that is, the band of the channel in which packet data is inserted. was fixed at a certain speed.

On the other hand, audio compression technology currently allows audio signals to be compressed to 32K.
BOS, 16 kbps, 8 kbps 8(
1') "It has become possible to transmit data at 1M degrees, and the corresponding voice detection technology has been established, so a more efficient transmission method is required.

However, since conventional methods only support one type of encoding speed, there are problems in that it is not possible to mix various audio encoding speeds, and even for low-speed data, packets can only be inserted in fixed speed units. When transmitting data below the speed, unused bits appear and become inefficient.
Furthermore, there is a problem in that it is not possible to construct a flexible network that freely multiplexes and transmits various audio speeds and various low-speed data speeds, and it is not possible to perform efficient transmission.

The present invention has been made in view of the above points, and it is an object of the present invention to provide a multiplex transmission system that allows efficient transmission without unused bits and enables flexible network construction. .

(Means for solving problem a1) FIG. 1 shows a diagram of the principle of the present invention.

In the figure, an audio interface section 32 of a transmitting section 31 obtains a plurality of encoded voices with different encoding speeds and supplies them to a multiplexing section 36, and a low-speed data interface section 39 receives a plurality of low-speed data with different transmission speeds. is supplied to the multiplexing section 36. The multiplexing unit 36 multiplexes a plurality of encoded voices with different encoding speeds and a plurality of low-speed data with different transmission speeds into a multi-frame structure in which each of them is assigned to separate time slots of a certain number of bits, and the multi-frame is ? The transmission format is converted into a transmission format having time slots with a number of bits corresponding to the encoding speed and transmission speed of each time slot of the multi-frame. Further, packet data is multiplexed into a silent section of audio and an invalid section where low-speed data that is not packetized exists and is sent to the transmission path 50.

In the receiving section 52, a separating section 54 separates the packet data, converts the transmission format into a multi-frame configuration, and sends each time slot to the fame interface section 6.
2. The audio interface unit 62 decodes and outputs each of the plurality of voices, and the low-speed data interface unit 63 decodes each of the plurality of low-speed data having different transmission speeds at 1!7. Output with .

[Effect]

In the present invention, since a multiframe is converted into a transmission format that has time slots with a number of bits corresponding to the encoding rate and transmission rate of each time slot, the transmission format does not generate unused bits and can achieve efficient transmission. It is possible to select from a plurality of voice encoding speeds and low-speed data transmission speeds, making it possible to construct a flexible network.

〔Example〕

FIG. 2 shows a block diagram of one embodiment of the inventive system.

In the figure, voice signals from telephones 30a, 30b, and 30c are transmitted to voice detecting units 33a, 33b, and 33c and encoding units 34a and 3 in the voice interface unit 32 in the transmitting unit 31.
4b and 34c, respectively. Audio detection section 33a, 3
3b and 33c each detect a voice signal/silence detection bit ID1. ID2. Generate ID3, encoder 3
For example, each of 4a, 34b, and 34c has a rating of 8k.
bps, 16 kbps, and 32 kbos, and time slot TS1. TS2. T
It is supplied to the multiplex synchronization section 37 in the multiplexing section 36 as S3.

For example, the bit rates from the terminals 38a, 38b, and 38c are 2.4 kbps, 4.8 kbps, and 9
．． Each unpacketized low-speed data of 6k bps is supplied to the low-speed data interface unit 39, and when the data detection units 40a, 40b, and 40c detect the respective low-speed data, they are assigned a data presence/absence detection bit ID4. ID5. ID6 is generated and added to time slot TS4. TS5. It is supplied to the multiplex m section 37 as TS6.

Furthermore, packet data from the packet terminals 418 to 41n is multiplexed by the packet switch 42 and supplied to the packet insertion section 43 in the multiplexing section 36.

The multiplex synchronization unit 37 receives the time slot TS from the audio interface unit 32 and the low-speed data interface unit 39.
1 to TS6 are synchronized to create a 1-room 8K system as shown in Figure 3.
The configuration is assumed to be 20 multi-frames of l-1z. 1 in Figure 3
The frame is divided into two time slots, and each time slot corresponds one-to-one to a telephone and a low-speed data terminal.

Each time slot has 8 bits, a synchronization pattern F for establishing synchronization is inserted in bit b1, and signaling 5IGI to 3 for switching equipment control etc. is inserted in bit b8 of voice time slots TSI to TS3, and low-speed data Transmission request control signals R84-6 are inserted into bits b8 of time slots rS4-TS6. time slot TS
In I to TS6, detection bits 101 to ID6 are set to bit b3 of the first frame. The 3 kbpS encoded voice is inserted into the shaded bit b3 of the time slot TS1. With this as the basic middle level, 16k bps encoded audio is bit b3 shown by diagonal lines in time slot TS2.
．． b4, and the 32k bps encoded audio is inserted into bit b3. b4. b5
．． It is inserted into b6. Similarly, low-speed data of 2.4 kbps is inserted into bits b2 to b7 of the second frame shown by diagonal lines in time slot TS4. With this as the basic medium, low-speed data of 4.8 kbps is the 2nd speed data of time slot TS5. The low-speed data of 9.13 kbps is inserted into bits b2 to b7 of the third frame, and the low-speed data of 9.13 kbps is inserted into bits 2, 3, 4, and 5 of time slot TS6.

The multi-frames described above are supplied to the frame converting section 46, where they are converted into the transmission format shown in FIG. Convert. ID1-[)6.5IG1-3 in FIG. R84-6 correspond to those in FIG. Time slot f in Figure 4
v1 to ■3 of S1 to S3 are the converted encoded voices in the shaded areas of the corresponding slots in Figure 3, and d1 to d3 of time slots S4 to TS6 in Figure 4 correspond to those in Figure 3. This is the converted low-speed data in the shaded portion of the time slot.

! The D separation unit 48 separates the detection data ID1 to ID106 of each time slot of the multi-frame and sends it to the packet insertion unit 4.
Perform the switching of 3. The packet insertion unit selects encoded voices v1 to v3 and low-speed data d1 to d3 in the sound section of the voice and the valid section of the low-speed data in each time slot of the transmission format shown in FIG. Then, the packet data PKT is selected and transmitted from the line interface unit 49 to the transmission line (high-speed digital line) 50 at the transmission fort 7 in FIG.

The fourth signal received by the line interface 53 of the receiving unit 52
Data in the transmission format shown in the figure is supplied to the demultiplexing section 54. The I[) separating unit 54 causes the packet separating unit 55 to separate this packet when the detection bits [)1 to 106 instruct the insertion of packet data PK, and the separated packet is sent to the packet terminal 578 via the packet switch 56. ~
57n respectively.

On the other hand, the frame converter 60 converts the data in the transmission format shown in FIG. 4 into the multi-frame format shown in FIG. supply to,
The time slots S4 to TS6 are supplied to the low-speed data interface section 63.

Decoding units 64a to 64c in the audio interface unit 62
The encoded audio of time slots TS1 to TS3 is encoded at a bit rate of 8 kbps and 16 kbps, respectively.

Decode at 32k bps and selectors 65a to 65C
supply to. The selectors 65a to 65c are the detection bits 1D1 to ID3 separated by the ID separation units 668 to 66c, respectively.
Accordingly, the decoded audio signal is selected in the sound section, and the white noise from the noise generators 67a to 67G is selected in the silent section, and is supplied to the telephones 68a to 68c, respectively.

Time slot 1-r in the low-speed interface section 63
s4 to [S6 bit rate 2.4 kbps.

The low-speed data of 4.8 kbps and 9.6 kbps are supplied to selectors 708 to 70c. The selectors 70a to 70C select low-speed data in the valid section according to the detection bits ID4 to 106 separated by the ID separators 71a to 71c, and select the 2.4 kbps data from the fixed pattern generators 728 to 72C in the invalid section, respectively. .8k
bps and 9.13 kbps are selected and supplied to the terminals 73a to 73c, respectively.

In this way, since the multi-frame shown in Fig. 3 is converted to the transmission format shown in Fig. 4, which has time slots with the number of bits corresponding to the encoding speed and transmission speed of each time slot, unused bits occur in the transmission format. It is possible to perform efficient transmission without any noise, and to select from a wide range of audio encoding speeds and low-speed data transmission speeds, making it possible to construct a flexible network.

[Effect beam of invention]

As described above, according to the multiplexing method of the present invention, there are no unused bits in the transmission format, so efficient transmission can be performed, and flexible network construction is possible, which is extremely useful in practice.

[Brief explanation of the drawing]

Figure 1 is a diagram of the principle of the present invention, Figure 2 is a block diagram of an embodiment of the method of the present invention, Figure 3 is a diagram showing the multi-frame format of the method of the present invention, and Figure 4 is the transmission of the method of the present invention. FIG. 5 is a block diagram of an example of the conventional method, and FIG. 6 is a diagram showing the transmission frame format of the conventional method. In the figure, 31 is a transmitter, and 32 and 62 are audio interfaces. 36 is a multiplexing unit, 39.63 is a low-speed data interface unit, 50 is a transmission path, 52 is a receiving unit, and 54 is a demultiplexing unit. Figure 8 showing the 7 frame format

Claims (1)

[Claims] In a multiplex transmission method in which packet data is multiplexed with coded audio obtained by encoding the audio and the low-speed data in a silent interval of audio and an invalid interval in which low-speed data that is not packetized is present, and then transmitted. , a plurality of encoded voices with different encoding speeds and a plurality of low-speed data with different transmission speeds are multiplexed as a multi-frame format in which each of the plurality of encoded voices with different transmission speeds is allocated to separate time slots of a fixed number of bits, and the multi-frame is A multiplex transmission method characterized by converting to a transmission format having time slots with a number of bits corresponding to the encoding speed and transmission speed of the time slots and transmitting the data.