KR970006951B1 - Comprising transfer apparatus of voice signal at adpcm - Google Patents

Abstract

The ADPCM voice signal compression transmitter is comprised of: an SLIC unit directly connecting voice signals input through a few subscriber lines to respective channels; a CODEC provided in respective channels to digital-convert and decode analog signals from an SLIC of respective channels, a signal processing unit provided in each channel to control a hook state of SLIC unit subscriber signal; a band converting unit provided in each channel, and band-compressing pulse modulating signals received from the CODEC and decoding signals from the subscriber; a channel selecting unit selectively outputting signals to be transmitted/received to each band converting unit of each channel; a signal controlling unit inputting the hook state from the signal processing unit of each channel to control the selection of the channel selecting unit.

Description

ADP PMM compression signal transmission device

1 is a configuration diagram of a conventional NAS (NAS) signaling system.

2 is a block diagram of a conventional CEPT signaling method.

3 is a block diagram of the present invention the PD C signal transmission apparatus.

4 is a configuration diagram of a channel multiplexer for explaining FIG.

5 is a frame structure diagram of the present invention.

* Explanation of symbols for main parts of the drawings

10: first channel 11: first SLC unit

12: first codec unit 13: signal processing unit

40: band converter 41: encoder

42: decoder 70: channel selector

71: multiplexer 72: demultiplexer

80: signal control unit

The present invention compresses an existing 64 Kbps (bit per sec) band of voice channels into 24Kbps or 16Kbps to integrate two or three voice subscribers into one 64Kbps channel without transmitting a separate signal path when transmitting the same. The present invention relates to an apparatus for transmitting signals of integrated channels in an in-band, and more particularly, to provide a signal transmission apparatus having self-synchronization and an error detection function irrespective of a conventional PC frame structure.

The voice subscriber's signal transmission method currently used in digital communication is the NAS (NAS) method, in order to transmit all the signal bits A (A), B (B), C (C), D (D) 4 bits, every 6 A (A) bit in the first frame, B (B) bit in the 12th frame, C (C) bit in the I8th frame, and D (D) bit in the 24th frame. Do and receive.

In order to transmit all four bits in the NAS method, it is required to have an extended super frame (hereinafter referred to as ESF) structure, and a conventionally used frame structure uses only half of the ESF. As a superframe structure of FIG. 4, only 2 bits of A and B are used among the related 4-bit signal bits.

As shown in FIG. 2, the CEPT method transmits four signal bits of A, B, C, and D using the 16th channel of every frame, as shown in FIG.

The above method is a normal signal method when one voice is used at 64 Kbps, and ATT is used to compress and transmit the existing 64 Kbps band to 32 Kbsp, 24 Kbps, 16 Kbps, etc. using ADP PMM's voice compression method. M44 method or No. of the International Telegraph and Telephone Advisory Committee (CCITT). 7 Send using COMS (COMMON CHANNEL SIGNALLING).

Referring to FIG. 1 and FIG. 2 attached to the operation of the speech compression transmission device configured as described above is as follows.

If you configure and transmit 64Kbps voice subscriber of one channel to 2 or more channels with ADP, you can only send signals for 24 subscribers in NAS method and 30 subscribers in CEPT method. In the existing method, a signal path must be created and transmitted on a specific channel such as AT T's M44, or a complex protocol such as CCS can be used to transmit a signal.

Here, AT T's M44 method is used only for 32K ADPCM compression, and is possible only in NAS frame structure. It consists of 24 channels in 4 bundles to accommodate 6 channel rate 11 subscribers per bundle, and the rest. Control the signal of 11 subscribers with 4 bits.

However, the conventional AT T M44 method supports only 32K ADPCM in NAS method, and it is difficult to apply when only one or two voice subscribers operate using ADPMC because 6 channels are bundled in one bundle. That's the way.

In addition, CCITT NO. When controlling the signal using the 7 CCS method, the subscriber cannot use the CCS channel of the 64Kbps band once, and has a disadvantage of having a complicated configuration for operating the protocol.

Therefore, the present invention is to provide a signal transmission apparatus for solving the above problems.

The object of the present invention is to provide a subscriber line interface (SLIC) unit for directly connecting a voice signal input through a few subscriber lines to each channel, and digital conversion and decoding of analog signals from the respective channel SLIC units. A codec unit for each channel, a signal processing unit for each channel controlling the overall state of the SLIC unit subscriber signal, and a band for each channel for band-compressing the pulse modulated signal received from the codec unit and decoding data to the subscriber side. A conversion unit, a channel selection unit for selectively outputting data to be transmitted / received by the channel conversion unit, and a signal control unit which receives a corresponding signal from the signal processing unit of each channel and controls the selection of the channel selection unit. The band converter comprises an encoder unit for compressing a 64 kilobit signal per second into 16 kilobits and 24 kilobits of data; And a decoder that decodes the data transmitted to the subscriber side into original 64 kilobits of data, and the frame generated by the signal controller comprises seven sync bits, four signal bits for each channel, and five Redundancy can also be achieved by constructing check bits.

3 is a schematic diagram of a signal transmission apparatus of the present invention, which includes a SLIC unit 11, 21, and 31 directly connected to a subscriber telephone line, and a codec unit 12 for digital and analog conversion of analog and digital voice signals. 22, 32, three presser channels 10, 20, 30, each of which comprises a signal processor 13, 23, 33 for controlling signals such as on-hook, off-hook, and the like; A band converter 40, 50, 60 for band-compressing 64 K PCM data of a channel into an ADPCM, and selecting one of the data compressed by the first to third band converters or multiplying an input signal; A channel selector 70 for transmitting to each band converter decoder 42 and a signal controller 80 for processing the signals of each channel according to the frame structure as shown in FIG. Configure.

Each connected ADPCM data is combined with two or three subscribers within the 64K band of the channel selector 70, and at the same time, signals of each channel are processed by the signal controller 80 to the channel selector 70. Is sent.

The operation of the signal transmission apparatus of the present invention configured as described above will be described with reference to FIGS. 3 to 5 as follows.

The 64Kbps pulse code modulation (PCM) data digitally converted in the existing subscriber channel connector (12, 22, 32) is banded to 16K or 24K bits through the encoders (41, 51, 61) of the band converter (40, 50, 60). As shown in Fig. 4, the original 8-bit pulse coded modulation data is compressed into 2-bit or 3-bit data and multiplexed into bit 8 to 3 positions as shown in (a) and (b) to a single 64K. Configure the audio unit.

In the above configuration, bit 1 is always fixed to 1, which is prepared in case that the network currently configured and operated does not support 64K clear channel.

In the above configuration, the bit 2 position is a bit position for transmitting a signal, and this signal constitutes one frame as shown in FIG. 5 in the band converter 40 and the signal controller 80 by outputting signals from the three channel units. By controlling the signal of each channel.

In the frame structure of FIG. 5, the upper 7 bits generated by the signal controller 80 are used to maintain the start synchronization of the frame, and 8 to l9 bits are the A, B, C, and D signal bits of each channel. 5 bits of bits to 24 bits are Cyclic Redundancy Check (CRC) values of the frame, and the receiving end may detect the seed value to detect whether there is an error in the frame.

The operation of the frame is a configuration method when 16K ADPCM is configured. In case of 24K ADPCM, only two channels are accepted as shown in (b) of FIG. 4 in 64K band. Set 4 bits of 3A, 3B, 3C, and 3D to 1 to operate.

As described above, the present invention performs the band compression of the speech code modulation data with ADPCM to improve the efficiency of the channel and to transmit the signal of the related channel in the same band as the voice channel and to transmit the existing signal. Since there is no separate redundancy band, the usable capacity of the channel can be increased. Especially, the signal of each multiplexed channel can be increased by using its own framing sync bit regardless of which network has a separate frame structure. Synchronization can be maintained and the malfunction of the received signal data can be detected by using the redundancy check (CRC-5) bit to prevent malfunction.

Claims (3)

SLIC unit for directly connecting voice signals input through a few subscriber lines to respective channels, a codec unit for each channel for digitally converting and decoding analog signals from the respective channel SLIC units, and the SLIC unit subscriber signals A signal processing section for each channel controlling the overall state of the channel, a band converting section for each channel for band-compressing the pulse modulated signal received from the codec section, and decoding the data to the subscriber side, and transmitting each channel band converting section. To be received And a channel selector which selects and outputs data, and a signal controller which receives a signal from the signal processor of each channel and controls the selection of the channel selector. 2. The apparatus of claim 1, wherein the band conversion unit encodes a 64 kilobit signal per second into 16 kilobits and 24 kilobits of data, and the compressed and transmitted data to the subscriber is converted into original 64 kilobits of data. And a decoder that decodes the ADP MPM audio signal compression transmission device. The apparatus of claim 1, wherein the frame of the band converter comprises seven sync bits, four signal bits for each channel, and five redundancy check bits.