KR20010048948A - Method for multiplexing using low bit rate voice compression - Google Patents

Abstract

PURPOSE: A multiplexing method using a low bit rate voice compressing technique is provided which allocates a plurality of channels to one time slot in order to improve band width efficiency. CONSTITUTION: A voice channel is compressed using the first voice compressing technique. A voice-grade data channel is compressed using the second voice compressing technique. Four voice channels and one voice-grade data channel are placed in one time slot. The transmission period of the voice channels and voice-grade data channel is set in accordance with the transmission period of a voice compressing technique among the voice compressing techniques that has longer period. A signal transmission mode according to the transmission of the voice channel and voice-grade data channel is set in accordance with the number of the entire channels. The first voice compressing technique is CS-ACELP mode compression technique with the transmission rate of 8kbps. The second voice compressing technique is adaptive differential pulse code modulation mode compressing technique with the transmission rate of 32kbps.

Description

Multiplexing method using low bit rate speech compression technology {METHOD FOR MULTIPLEXING USING LOW BIT RATE VOICE COMPRESSION}

The present invention relates to a multiplexing method using a low bit rate speech compression technique and a computer readable recording medium recording a program for realizing the method. A method of multiplexing and transmitting at a bit rate and a computer-readable recording medium having recorded thereon a program for realizing the method.

Multiplexing is used to improve the efficiency of transmission paths and communication systems. Multiplexing is a method of allowing multiple signals to use the same transmission line or system, and superimposing a low bit rate signal into a high bit rate signal in the frequency domain or the time domain. In particular, the method of multiplexing in the time domain is called time division multiplexing (TDM). The time occupied by each signal in the TDM is called a time slot, and n timeslots included in one period are called a frame.

In the 1st TDM method using Pulse Code Modulation (PCM: Pulse Code Modulation, 64kbit per sec) signal channel, 24 channels of North America and 32 channels of Europe (hereinafter referred to as E1) are the International Telecommunication Union-Communication Part (ITU). -T is recommended for use by the Telecommunication part of International Telecommunication Union. In the 32-channel European scheme, 30 timeslots are used for the PCM channel, the first timeslot is the synchronization channel, and the 17th timeslot is used for signaling. In addition, 16 frames are bundled to form a multiframe, and the 17th timeslot of the first frame is used for multiframe synchronization.

TDM method based on 64kbps PCM signal channel used in general telephone network guarantees bandwidth for time sensitive traffic such as voice, and has the advantage of easy configuration, but it takes about 60% of the time in most voice conversations. In terms of silence, there is a problem of inefficient bandwidth usage.

In contrast, Internet telephones, mobile phones, and personal communication systems (PCSs) using the Internet use voice compression technology to reduce the bandwidth of voice transmission and use statistical packet multiplexing (SPM). By using it efficiently, it is possible to carry more channels on the same transmission path as compared to a general telephone network using TDM, thereby achieving economical benefits.

However, even though statistical packet multiplexing has the advantages of increased bandwidth efficiency and flexible bandwidth allocation, the system configuration is very complicated, increases transmission delay, and makes it impossible to predict the arrival time of transmitted packets. In comparison with TDM, there is a problem.

This is a multiplexing method that increases bandwidth efficiency while retaining the advantages of TDM, and is a digital line multiplier widely used for international communication, a TDM device using voice compression technology and digital signal interpolation (DSI). (DCME), which is difficult to increase the length of the transmission path, does not have a lot of traffic, and is effective in using only voice.

SUMMARY OF THE INVENTION The present invention has been made to solve the above problems, and the multiplexing method using a low bit rate speech compression technique which allocates several channels to one timeslot using a speech compression technique to improve bandwidth efficiency. An object of the present invention is to provide a computer-readable recording medium having recorded thereon a program for realizing this.

1 is an exemplary configuration diagram of a communication system to which the present invention is applied.

2 is a diagram illustrating an embodiment of a multiplexed frame used in a multiplexing method using a low bit rate speech compression technique according to the present invention.

3 is a diagram illustrating an embodiment of a time slot configuration of a frame in a multiplexing method using a low bit rate speech compression technique according to the present invention;

4 is a flow diagram of an embodiment of a multiplexing method using a low bit rate speech compression technique in accordance with the present invention.

* Explanation of symbols for the main parts of the drawings

3: transmitter multiplexing apparatus 4: E1 transmission path

5: receiving side multiplexing device

A method of the present invention for achieving the above object comprises: a multiplexing method using a low bit rate speech compression technique applied to a communication system, comprising: a first step of compressing a voice channel using a first speech compression technique; Compressing the voice grade data channel using a second speech compression technique; Positioning four said voice channels and one said voice grade data channel in one timeslot; A fourth step of setting transmission periods of the voice channel and the voice data channel in accordance with the transmission period of the voice compression technology having a longer period among the voice compression technologies; And a fifth step of setting and multiplexing a signal transmission method according to transmission of the voice channel and the voice data channel according to the total number of channels according to the transmission period.

In addition, another method of the present invention is a multiplexing method using a low bit rate speech compression technique applied to a communication system, the pulse code modulation (PCM) having a transmission rate of 64 kilobits per second (kbps: kilobit per sec) A first step of compressing a voice channel of a Pulse Code Modulation) using a first voice compression technique having a transmission rate of 8 kbps; A second step of compressing a voice code data channel of a pulse code modulation (PCM) method having a transmission rate of 64 kbps using a second speech compression technique having a transmission rate of 32 kbps; Positioning the four compressed voice channels and the one voice level data channel in a timeslot of a multiplex frame having a transmission rate of 2048 kbps; A fourth step of forming one multiframe with the 16 multiplexed frames and one transmission period with five multiframes; And a fifth step of setting and multiplexing a signal transmission method according to transmission of the voice channel and the voice data channel according to the total number of channels according to the transmission period.

The present invention also provides a communication system having a large capacity processor, comprising: a first function of compressing a voice channel using a first voice compression technique; A second function of compressing a voice quality data channel using a second voice compression technique; A third function of locating the four voice channels and one voice grade data channel in one timeslot; A fourth function of setting transmission periods of the voice channel and the voice data channel according to transmission periods of the voice compression technology having a longer period among the voice compression technologies; And a computer readable recording medium having recorded thereon a program for realizing a fifth function of setting and multiplexing a signal transmission method according to transmission of the voice channel and the voice data channel according to the total number of channels according to the transmission period. to provide.

In addition, the present invention, 8kbps pulse code modulation (PCM) voice channel having a transmission rate of 64 kilobits per second (kbps :) in a communication system having a large capacity processor. A first function of compressing using a first speech compression technique having a transmission rate of? A second function of compressing a voice code data channel of a pulse code modulation (PCM) scheme having a transmission rate of 64 kbps using a second speech compression technique having a transmission rate of 32 kbps; A third function of placing the four compressed voice channels and the one voice level data channel in a timeslot of a multiplex frame having a transmission rate of 2048 kbps; A fourth function of forming one multiframe with the sixteen multiplexed frames and one transmission period with five multiframes; And a computer readable recording medium having recorded thereon a program for realizing a fifth function of setting and multiplexing a signal transmission method according to transmission of the voice channel and the voice data channel according to the total number of channels according to the transmission period. to provide.

Looking at the PCM method used in the existing E1 multiplexing and the speech compression technology used in the present invention are as follows.

The PCM encoding method is recommended in ITU-T G.711. It has a transmission rate of 64 kbps and the frame size, which is the basic unit to be encoded, is 125 ㎲. That is, every 125 ms of speech data is encoded after sampling and quantization to generate an 8-bit digital signal.

PCM is widely used to convert analog signals to digital signals because it has the advantage of encoding not only voice but also fax and modem data.

Adaptive Differential Pulse Code Modulation (ADPCM) is recommended in ITU-T G.726. It has a transmission rate of 32 kbps and the frame size is 125 ㎲ like PCM. Less four bits are encoded. In addition, like the PCM, since the fax and modem data can be encoded and transmitted in a transparent manner, the present invention is used.

CS-ACELP is recommended in ITU-T G.729. It has toll-class phone quality and 8kbps transmission speed, and has a transmission bandwidth of 8: 1 than the conventional PCM method. There is an advantage to reduce. In addition, the frame size is 10 ms and generates 80 bits of digital signal in one period. Compared with other voice compression techniques, it is used in the present invention because it has toll-class phone quality at low bit rate and short transmission delay, which is suitable for real-time service of voice.

The above objects, features and advantages will become more apparent from the following detailed description taken in conjunction with the accompanying drawings. Hereinafter, exemplary embodiments of the present invention will be described in detail with reference to the accompanying drawings.

1 is an exemplary configuration diagram of a communication system to which the present invention is applied, and is composed of a transmission-side multiplexing apparatus 3, an E1 transmission path 4, and a receiving-side multiplexing apparatus 5. The PCM voice channels 1 and 6 are connected to the PCM data channels 2 and 7.

As shown in the figure, in the schematic configuration diagram of a system to which the present invention is applied, 120 PCM voice channels 1 and 30 PCM data channels 2 having 64 kbps bandwidth are connected to the input of the multiplexing device 3.

In the multiplexing apparatus 1, the voice channel information is compressed by CS-ACELP compression, and generates 80 bits every 10 ms period.

In addition, the data channel information is compressed by the ADPCM compression scheme, and generates 4 bits every 125 ms. The compressed information is multiplexed and then transmitted to the receiving multiplexer 5 through one E1 transmission path 4.

The electrical and physical characteristics of the E1 transmission path are 2048 kbps according to ITU-T G.703, and the line code applies the HDB3 (High Density Bipolar of order 3) code. In addition, the basic frame structure is in accordance with ITU-T G.704. The data transmitted to the receiving side multiplexing device 5 is demultiplexed and then restored according to the channel information and output to 120 PCM voice channels 6 and 30 PCM data channels 7.

2 is a diagram illustrating an embodiment of a multiplexed frame used in a multiplexing method using a low bit rate speech compression technique according to the present invention.

2 is a diagram illustrating a frame structure for multiplexing, i.e., multiplexing 120 PCM voice channels and 30 PCM data channels into an E1 frame according to the present invention.

Time slot 11 of the E1 frame is for frame synchronization and is in accordance with ITU-T G.704 Table B. Time slots 1 to 15 and 17 to 31 are used for voice and data channels, and time slot 16 is used for signal transmission. The voice channel 12 and the data channel 13 shown in the figure show how channels are allocated to one timeslot.

The multiplexing of the voice channel 12 is performed by allocating one voice channel for each bit. One bit of each channel corresponds to one bit of 80 bits, which is the result of compressing a 64 Kbps PCM voice channel by 8 kbps CS-ACELP compression. By allocating 1 bit for each E1 frame, 80 bits can be filled for 10 ms. That is, if the E1 frame structure in which 16 frames are composed of multiframes at 2 ms periods is extended to a structure in which five multiframes constitute one cycle, 80-bit CS-ACELP voice information may be carried in the E1 frame.

Multiplexing of the data channel 13 is achieved by allocating one channel to 4 bits. 4 bits corresponds to 4 bits, which is the result of compressing 8-bit PCM data channel by ADPCM compression. That is, the data allocation such as a modem and a fax can be carried in an E1 frame through 125 bits, which is the same period as that of the PCM method, and the number of bits per sample is 4 bits, which is half the value of the PCM.

With the multiplexing method as described above, as shown in the voice channel 12 and the data channel 13 shown in the figure, the first four bits carry four voice channels in one time slot and the other four bits carry one data channel. By multiplexing, five channels per timeslot can be delivered. That is, 150 voice and data channels can be contained and transmitted in an E1 frame structure having 30 existing time slots.

3 is a diagram illustrating an embodiment of a time slot configuration of a frame in a multiplexing method using a low bit rate speech compression technique according to the present invention.

FIG. 3 is a diagram illustrating a bit allocation method of time slot 16 for accommodating individual channel signaling (CAS) signals in an E1 frame structure in which 16 frames are composed of multiple frames at a period of 2 ms.

The 16 timeslot in the E1 frame structure is used for signaling, and can accommodate both Common Channel Signaling (CCS) and Channel Associated Signaling (CAS). When the signaling method is CCS, the signal information is transmitted through the entire time slot 16 having a transmission speed of 64kbps, and the signaling method is CAS as shown in FIG. 3.

Since the CAS method in Korea uses the a and b signaling methods, the present invention follows this, and other signaling methods (a, b, c, and d) are excluded from consideration. The 16 timeslot 21 of frame 0 is used for multiframe synchronization consisting of 16 frames and repeated every multiframe, which is the same as specified in ITU-T G.704.

The signal transmission timeslot 22 for the voice channel uses the first 4 bits of the 16 timeslot. 2 bits are allocated to each channel for the a and b signaling methods as described above, and thus signal information for 30 channels can be loaded in one multiframe. That is, up to voice channels 1-30 in the first multiframe, voice channels 31-60 in the second multiframe, voice channels 61-90 in the third multiframe, and voice channels 90 ~ 90 in the fourth multiframe. Up to 120 signal information can be loaded. Finally, the first 4 bits of the 16th time slot of the fifth multiframe are not used. In this case, the transmission speed of the signal information is 200bps.

The timeslot 23 for signal transmission for the data channel uses bits 5 to 8 of the 16 timeslot. Like voice channels, 2 bits are allocated to each channel for a and b signaling, and thus signal information for 30 channels can be loaded in one multiframe. That is, the transmission speed of the signal information is 1kbps.

4 is a flowchart illustrating an embodiment of a multiplexing method using a low bit rate speech compression technique according to the present invention.

A compression is performed using a 8 kbps CS-ACELP voice compression technique for a 64 kbp PCM voice channel (401). At this time, one period of the CS-ACELP is 10ms, and generates 80-bit digital signal for one period.

A 64 kbps PCM speech data channel is compressed using a 32 kbps ADPCM speech compression technique (402). At this time, the ADPCM method has a cycle of 125 ms as in the PCM method.

Four compressed audio channels and one data channel are located in one timeslot of the E1 frame (403).

By using the CS-ACELP method for speech compression, the E1 frame structure in which 16 frames are composed of multiframes every 2 ms is extended to a structure in which five multiframes constitute one period (404).

In operation 405, it is determined whether one of the common channel signaling method (CCS) and the individual channel signaling method is used as the signal transmission method.

As a result of determining the signal transmission method to be used, if the individual channel signaling method (CAS) is used, time slot 16 of frame 0 for one multiframe is used for multiframe synchronization (406). At this time, time slot 16 is a time slot used for signal transmission in each frame according to the format defined in ITU-T.

There are a, b, c, and d methods in the individual channel signal system (CAS), and there are a, b, and a, b, c, and d methods, which are used in certain countries. , b will be described.

In order to transmit a voice channel, 2 bits are allocated to each of the 16 timeslots of the remaining frames except for the 0 frame of the multiframe for each of the a and b signaling schemes (407).

Only up to a fourth multiframe out of five multiframes constituting one period is allocated signal information of a voice channel to transmit a signal (408).

In order to transmit the signal of the data channel, 2 bits are allocated to each of the 16 timeslots of the remaining frames except the 0 frame of the multiframe according to the a and b signaling schemes to transmit the signal (409). At this time, it is arranged in time slot 16 so as to be distinguished from the voice channel. In the present embodiment, the first four bits are used for signal transmission of the voice channel, and the rear four bits are used for signal transmission of the data channel.

In this case, the signal transmission rate of the individual channel signaling method is 200bps for the voice channel and 1kbps for the data channel.

As a result of determining the signal transmission method to be used, when the common channel signaling method (CCS) is used, signal information is transmitted through the entire time slot 16 having a transmission rate of 64 kbps (410).

The present invention described above is not limited to the above-described embodiments and the accompanying drawings, and various substitutions, modifications, and changes can be made in the art without departing from the technical spirit of the present invention. It will be apparent to those of ordinary knowledge.

As described above, the present invention substitutes 120 voice channels and 30 data channels through a single E1 transmission path using 8 kbps voice channel compression transmission method and 32 kbps data channel compression transmission method instead of 64 kbps voice and voice level frame transmission. It can be transmitted by multiplexing, it has the advantage of five times the transmission capacity increase compared to the conventional E1 multiplexing method, it is effective to accommodate not only voice but also fax and modem data at the same time.

Claims (8)

A multiplexing method using a low bit rate speech compression technique applied to a communication system, Compressing the voice channel using a first voice compression technique; Compressing the voice grade data channel using a second speech compression technique; Positioning four said voice channels and one said voice grade data channel in one timeslot; A fourth step of setting transmission periods of the voice channel and the voice data channel in accordance with the transmission period of the voice compression technology having a longer period among the voice compression technologies; And A fifth step of setting and multiplexing a signal transmission method according to transmission of the voice channel and the voice data channel according to the total number of channels according to the transmission period; Multiplexing method using a low bit rate speech compression technique comprising a. The method of claim 1, The voice channel and the voice data channel, Using a low bit rate speech compression technique characterized in that the voice channel and the voice data channel of the pulse code modulation (PCM) method having a transmission rate of 64 kilobits per second (kbps) Multiplexing method. A multiplexing method using a low bit rate speech compression technique applied to a communication system, Pulse code modulation (PCM) speech channel having a transmission rate of 64 kilobits per second (kbps) is compressed using a first speech compression technique having a transmission rate of 8 kbps. First step; A second step of compressing a voice code data channel of a pulse code modulation (PCM) method having a transmission rate of 64 kbps using a second speech compression technique having a transmission rate of 32 kbps; Positioning the four compressed voice channels and the one voice level data channel in a timeslot of a multiplex frame having a transmission rate of 2048 kbps; A fourth step of forming one multiframe with the 16 multiplexed frames and one transmission period with five multiframes; And A fifth step of setting and multiplexing a signal transmission method according to transmission of the voice channel and the voice data channel according to the total number of channels according to the transmission period; Multiplexing method using a low bit rate speech compression technique comprising a. The method according to any one of claims 1 to 3, The first voice compression technology, A multiplexing method using a low bit rate speech compression technique, characterized by a CS-ACELP speech compression technique having a transmission rate of 8 kbps. The method of claim 4, wherein The second voice compression method, A multiplexing method using a low bit rate speech compression technique, characterized in that it is an adaptive differential pulse code modulation (ADPCM) speech compression technique having a transmission rate of 32 kbps. The method according to any one of claims 1 to 3, The fifth step, A sixth step of selecting and confirming a signal transmission method for multiplexing the voice channel and the voice data channel; A seventh step of transmitting a signal through a signal transmission time slot having a transmission rate of 64 kbps as a result of the checking in the sixth step; And As a result of the checking in the sixth step, if the individual channel signal type, the signal transmission time slot of one frame is used for synchronization for one multi-frame, and the voice channel and the voice data are included in the signal transmission time slot of the remaining frames. Eighth step of transmitting signals by allocating 2 bits for the channel Multiplexing method using a low bit rate speech compression technique comprising a. In a communication system having a large capacity processor, A first function of compressing a voice channel using a first voice compression technique; A second function of compressing a voice quality data channel using a second voice compression technique; A third function of locating the four voice channels and one voice grade data channel in one timeslot; A fourth function of setting transmission periods of the voice channel and the voice data channel according to transmission periods of the voice compression technology having a longer period among the voice compression technologies; And A fifth function of setting and multiplexing a signal transmission method according to transmission of the voice channel and the voice data channel according to the total number of channels according to the transmission period A computer-readable recording medium having recorded thereon a program for realizing this. In a communication system having a large capacity processor, Pulse code modulation (PCM) speech channel having a transmission rate of 64 kilobits per second (kbps) is compressed using a first speech compression technique having a transmission rate of 8 kbps. First function; A second function of compressing a voice code data channel of a pulse code modulation (PCM) scheme having a transmission rate of 64 kbps using a second speech compression technique having a transmission rate of 32 kbps; A third function of placing the four compressed voice channels and the one voice level data channel in a timeslot of a multiplex frame having a transmission rate of 2048 kbps; A fourth function of forming one multiframe with the sixteen multiplexed frames and one transmission period with five multiframes; And A fifth function of setting and multiplexing a signal transmission method according to transmission of the voice channel and the voice data channel according to the total number of channels according to the transmission period A computer-readable recording medium having recorded thereon a program for realizing this.