JPH0564275A - Digital multiplex device - Google Patents

Abstract

PURPOSE:To provide the multiplex device of voice telephone signals, which is inexpensive in terms of a cost with simple constitution. CONSTITUTION:The device is constituted in such a way that a transmitting part 100 is provided with band filters 110, a transmission timing generating circuit 120, gate switches 130, an encoding circuit 140 and a multiplex circuit 15 and a receiving part 200 is provided with a separating circuit 210, a decoding circuit 220, a reception timing generating circuit 230, the gate switches 240 and low-pass filters 250. It is also constituted in such a way that the plural voice telephone signals are picked-up, taken-out, multiplexed and transmitted by mutually opening the gates of the gate switches 130 at a timing generated by the transmission timing circuit 120 and, at a reception side, the gates of the gate switches 240 are mutually opened so that the plural and multiplexed voice telephone signals are respectively separated into the plural voice telephone signals.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a multiplexer for multiplexing a plurality of analog voice telephone signals into one channel digital signal.

In digital communication, a voice signal is converted into 6
Communication is performed by PCM encoding at 4 kbps. This is to sample a voice telephone signal in the 4 kHz band at 8 kHz and perform 8-bit quantization, and it is possible to reproduce almost the original voice signal.

However, from the viewpoint of effective use of the line, there is a demand for a digital multiplexer capable of efficiently compressing a digital voice signal and making a call efficiently without impairing the clarity.

[0004]

2. Description of the Related Art FIG. 6 is a diagram for explaining a conventional example. In the figure, 10 is a subtractor, 20 is an adaptive quantization circuit, 21 is a quantization step adaptation circuit, 30 is an adaptive dequantization circuit, 40 is an adder, and 50 is an adaptive prediction circuit.

In the figure, the subtractor 10 calculates the difference between the present input signal "X" and the prediction signal "X _A " generated based on the past input signals to obtain the prediction error signal "X-X _A ". Then, the adaptive quantization circuit 20 performs quantization.

This adaptive quantization is based on the prediction error signal "X-X".
_In order to obtain a quick response to a rapid change in _A ', by not performing uniform quantization, the quantization step size is increased or decreased by the quantization step adaptation circuit 21 according to the quantization level of the immediately preceding sample. We are trying to improve quality.

On the other hand, the adaptively quantized output is adaptively inversely quantized.
The circuit 30 allows the analog quantity "XX _AAfter returning to
The prediction signal “X_AWill be added and the next entry
Prediction signal of force signal "X_A”Adaptive prediction circuit to predict
Stored in 50.

The above-mentioned conventional example is an example of the adaptive differential coding system which is also adopted as the CCITT standard. When the voice telephone signal is ADPCM coded at 32 kbps, the voice is comparable to the coded at 64 kbps. Signals can be obtained and are widely used.

[0009]

In the above-mentioned conventional example, various arithmetic processes are required for calculation of the predicted value, setting of the step size of adaptive quantization, etc., resulting in a complicated circuit configuration and high cost. Become.

The present invention is intended to realize a voice telephone signal multiplexer having a simple structure and being inexpensive.

[0011]

FIG. 1 is a block diagram for explaining the principle of the present invention. Reference numeral 100 in the figure is a transmission unit, 110 in the transmission unit 100 is a bandpass filter that band-limits an input analog voice telephone signal to a telephone band signal, and 120 is a plurality of analog voice telephone signals in one channel. Reference numeral 130 denotes a transmission timing generation circuit that generates timing for multiplexing.
20 is a gate switch that switches a plurality of analog voice telephone signals in a predetermined time width according to the timing of generation and sequentially extracts the signals.
Reference numeral 150 is a coding circuit for coding a signal in which a plurality of analog voice telephone signals output from 0 are multiplexed into a digital signal, and 150 is a multiplexing circuit for multiplexing the digital signal output from the coding circuit 140 with other digital signals. Circuit.

Reference numeral 200 denotes a receiving unit, and the receiving unit 20
Reference numeral 210 in 0 is a separation circuit for separating the multiplexed digital signals for each channel, and 220 is a decoding circuit for decoding the digital signal in which the plurality of voice telephone signals output from the separation circuit 210 are multiplexed. Yes, 230 is a reception timing generation circuit that generates a timing for separating signals multiplexed with a plurality of analog voice telephones output from the decoding circuit 220 into respective voice telephone signals, and 240 is a reception timing generation circuit 230. Is a gate switch for sequentially extracting a plurality of multiplexed analog voice telephone signals with a predetermined time width according to the timing of occurrence of a low-pass filter 250 for removing a high frequency component of the analog voice telephone signal output from the gate switch 240. It is a vessel.

[0013]

Normally, in a digital line, one line is formed in units of 64 kbps, and a voice telephone signal is encoded into a digital signal of 64 kbps with a sampling frequency of 8 kHz and a quantization bit number of 8 bits.

The present invention multiplexes a plurality of voice telephone signals on one channel of 64 kbps. First, only the component of 300 to 3400 Hz in the input voice telephone signal is passed by the bandpass filter 110.

The plurality of voice telephone signals are sequentially extracted at the timing generated by the transmission timing circuit 120.
For example, when two telephone channels are multiplexed, the gate switch 130 alternately opens the gate every 20 mS to alternately extract the two-channel voice telephone signal with a width of 20 mS.

Therefore, one channel 300 to 34
Two-channel voice telephone signal is to be multiplexed in the analog signal of 00Hz.
After being converted into a PCM signal of kbps, it is multiplexed with another digital signal and transmitted.

In the receiving section 200, the demultiplexing circuit 210 decodes the channel in which the voice telephone signal is multiplexed into the decoding circuit 220.
After being decoded into an analog signal, the original voice telephone signal is taken out by alternately extracting the multiplexed voice telephone signal at the timing generated by the reception timing circuit 230 that generates the same timing as the transmission unit 100. be able to.

[0018]

FIG. 2 is a diagram for explaining an embodiment of the present invention.
In the figure, 110 is a bandpass filter, 120 is a transmission timing circuit, 130 is a gate switch (shown as a gate SW in the figure), 141 is a sampling circuit, 142 is a quantization circuit, 14
3 is an encoding circuit, 150 is a multiplexing circuit, 64 kb
The operation will be described with an example in which two voice telephone signals are multiplexed on the ps1 channel.

First, when analog voice telephone signals 1 and 2 (the signal waveform of the voice telephone signal is input) are input, the band-pass filter 110 extracts only the component of 300 to 3400 Hz (the signal waveform is shown). In the transmission timing circuit 130, a timing of 20 mS width is generated, the gate switch 130 is opened with a width of 20 mS at that timing, and the voice telephone signals 1 and 2 are alternately taken out (the signal waveform is shown by).

In the sampling circuit 141, the signal in which two voice telephone signals are multiplexed is sampled at 8 kHz, and the quantization circuit 142 converts it into a discrete amplitude, and the encoding circuit 14
3 is converted to an 8-bit binary code (signal waveform is shown in).

Here, two voice telephone signals of 64 kb are transmitted.
1.54 after multiplexing to ps1 channel and further to another 64 kbps channel and 24 channel
It is sent to the transmission line as a 4 Mbps signal (shown in the signal waveform).

FIG. 3 is a diagram for explaining the characteristics of the audio signal. A shows the voice spectrum distribution, and B shows the contribution to the intelligibility. As is clear from the figure, as the frequency becomes higher, the spectrum distribution also becomes smaller and the contribution to the intelligibility becomes smaller. Therefore, the required intelligibility is secured by using 300 to 3400 Hz as the band of the voice telephone signal. be able to.

Considering the speed of ordinary conversation, an announcer of ordinary television and radio news broadcasts speaks at a speed of about 400 characters per minute.
Therefore, about 1 character utters about 7 characters,
The time required for one character is about 140 to 160 mS.

The voice telephone signal is 300 to 3400H.
z, and one long cycle is about 3 which is the reciprocal of 300 Hz.
mS. Therefore, one character is about 140 to 160 ms
Then, one vocalized sound is recognized as one vocalized sound with a cycle of about 50 cycles.

FIG. 4 is a diagram for explaining the division processing of the voice telephone signal. 400 Hz, 800 Hz, 1600 Hz, 3
The number of waves that enter when a signal of each frequency of 200 Hz is divided by 15 mS and 150 mS is shown.

In this embodiment, the division time of the voice telephone signal is set to 15 to 20 mS under the following conditions.
An integer multiple of one voice frequency.

The vocalization time per character is about 150 m
S. Time of discontinuity due to split playback. Time difference between the time of one vocal sound and the reproduced one vocal sound.

FIG. 5 is a diagram for explaining division reproduction of a voice telephone signal according to the embodiment of the present invention. Here, as described above, the division time is 21 ms, which is an integral multiple of 12 multiframes of 1.5 mS. (The description is performed assuming that one cycle is 20 mS.) The generated sound is shown. Here, "A",
Say “a”, “u”, and “d” to indicate that the durations were 140 mS, 150 mS, and 120 mS, respectively.

The division time is set to 20 mS, and the point A is "a".
In this example, “U” continues for 10 ms each. 20m
The audio signal transmitted by dividing by S is shown. Voice telephone signals of other channels are inserted between the voice signals, but they are omitted in the figure.

The received voice signal is shown. The audio signal reproduced through the low-pass filter 250 is shown. For example,
At point A in the figure, since the utterance has changed in the middle of the divided 20 mS, "a" and "u" are captured and transmitted by 10 mS each. As a reproduction signal, a signal of 20 mS is extended to 40 mS in order to fill a blank portion, so that "a" and "c" of 10 mS each are 2 of A1 and A2.
Repeated times. As shown in the figure, the first 10 mS "a" is valid, but the next 10 mS each "u",
“A” is invalid, and the last 10 ms “U” is the next 20
It is effective because it will lead to mS “u”.

Therefore, the durations of "a", "a", and "u" were 140 mS, 150 mS, and 120 mS, but 160 mS, 130 mS, and 100 mS, respectively.
Is reproduced as a voice signal of, and an unrecognizable portion as voice occurs in the middle.

A discontinuity occurs every 20 mS, but the discontinuity is smoothed by the low-pass filter 250. Furthermore, the discontinuity has a width of 20 mS and the noise component is about 50 Hz.
Since the 0 Hz component cannot be reproduced normally, it is not perceived as noise.

Further, by using a frame synchronizing signal as the synchronizing signal of the transmitting unit 100 and the receiving unit 200, accurate synchronization can be achieved. The above embodiment is 2
The operation has been described in which the voice telephone signal of 2 channels is multiplexed by division with 0 mS, but the voice telephone signal of 3 channels can be multiplexed with division with 15 mS.

Further, since it becomes possible to multiplex a plurality of voice telephone signals on one digital channel of 64 kbps, more voice telephone lines can be accommodated in the same speed digital line to improve the line efficiency of the digital line. be able to.

[0035]

According to the present invention, a plurality of voice telephone signals are alternately switched and extracted by a gate switch, multiplexed into one analog signal, and then converted into a normal digital signal of 64 kbps. It is possible to realize a simple and inexpensive multiplexer.

[Brief description of drawings]

FIG. 1 is a block diagram illustrating the principle of the present invention.

FIG. 2 is a diagram illustrating an embodiment of the present invention.

FIG. 3 is a diagram for explaining characteristics of an audio signal.

FIG. 4 is a diagram illustrating a division process of a voice telephone signal.

FIG. 5 is a diagram illustrating divided reproduction of a voice telephone signal according to the embodiment of this invention.

FIG. 6 is a diagram illustrating a conventional example.

[Explanation of symbols]

100 transmitter 200 receiver 110 bandpass filter 210 separation circuit 120 transmission timing circuit 220 decoding circuit 130, 240 gate switch 230 reception timing circuit 140, 143 encoding circuit 250 low-pass filter 141 sampling circuit 142 quantization circuit 150 Multiplexing circuit 10 Subtractor 20 Adaptive quantization circuit 21 Quantization step adaptation circuit 30 Adaptive dequantization circuit 40 Adder 50 Adaptive prediction circuit

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Claims (3)

[Claims] 1. A device for multiplexing a plurality of analog voice telephone signals into one-channel digital signals, wherein a transmitting section (100) band-pass filters the input analog voice telephone signals to a telephone band signal. (110), a transmission timing generation circuit (120) for generating timing for multiplexing a plurality of analog voice telephone signals into one channel, and a plurality of analog voices according to the timing generated by the transmission timing generation circuit (120). A gate switch (130) for sequentially extracting a telephone signal with a predetermined time width, and an encoding circuit (140) for encoding a signal in which a plurality of analog voice telephone signals output from the gate switch (130) are multiplexed into a digital signal. ) And the digital signal output from the encoding circuit (140) is multiplexed with another digital signal. Multiplexing circuit (150)
The receiving unit (200) includes a demultiplexing circuit (210) for demultiplexing a plurality of multiplexed digital signals for each channel, and a plurality of voice telephone signals output from the demultiplexing circuit (210). A decoding circuit (220) for decoding a digital signal and a reception timing for generating a timing for separating a signal multiplexed with a plurality of analog voice telephones output from the decoding circuit (220) into respective voice telephone signals. A gate switch (240) for sequentially extracting a plurality of multiplexed analog voice telephone signals with a predetermined time width according to the timing generated by the generation circuit (230) and the timing generation circuit (230).
And a low-pass filter (25) for removing high frequency components of the analog voice telephone signal output from the gate switch (240).
0) provided with a digital multiplexer. 2. The transmission timing generation circuit (120) and the reception timing generation circuit (230) are each 20 m long.
2. The digital multiplexer according to claim 1, wherein a timing pulse of s width is generated, and two voice telephone signals are alternately switched, extracted and multiplexed. 3. The transmission timing generation circuit (120) and the reception timing generation circuit (230) are 15 m long.
2. The digital multiplexer according to claim 1, wherein a timing pulse of s width is generated, and three voice telephone signals are alternately switched to be extracted and multiplexed.