JP2000244339A - Call signal transmission method - Google Patents

Abstract

(57) [Summary] [PROBLEMS] To provide a communication signal transmission system in which voice loss does not occur at both a transmission start time and a transmission end time while maintaining a natural conversation state. In A of press-to-talk type wireless systems, the transmit button pressing time t ₀ after the terminal was a voice signal A _C compressed by a time during the first part of the predetermined time of the input come audio signal A T In the above, transmission is performed with a predetermined delay time τ delayed. [Effect] Even if the transmission is started immediately after the transmission button is pressed t ₀ , the head is not dropped, and the possibility that the communication quality is deteriorated due to the delay of the communication can be prevented.

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a radio communication system including a plurality of communication stations, and more particularly to a communication signal transmission system suitable for a radio communication system in which a communication is performed via a line control device for establishing a communication channel.

[0002]

2. Description of the Related Art In a simplex wireless device such as a transceiver, the transmission / reception switching operation is often performed by a press-talk system. In this press talk system, a button for press talk, that is, a push button switch (transmit button) for transmission is provided on the housing of the wireless device, and when the transmit button is pressed, the own station wireless device performs a transmission operation. Switching is performed, and it is promised that the user presses the transmission button at the start of transmission and keeps the transmission button pressed during transmission.

As one type of communication system using such a press-talk system, as shown in FIG. 6, a plurality of terminals (wireless devices), for example, terminals X, Y and Z communicate via a base station M. There is a relay-type wireless system that performs a communication with a terminal X on the transmission start side and a terminal X or a terminal Z on the other side when a transmission button is pressed. A request to form a communication channel between them is issued to the line controller of the base station M.

In the case of such a relay system, after the transmission button is pressed, the line controller starts control, and the line connection between the transmission terminal and the reception terminal is completed and a communication channel is formed. A delay occurs, and this delay time can be as long as several hundred milliseconds depending on the configuration of the system.

[0005] As a result, as shown in FIG.
If the transmission is started until a certain period of time elapses after the transmission button is pressed, the voice of the first part of the transmission is not transmitted, and a phenomenon called so-called communication dropping occurs. However, the call quality is reduced.

In FIG. 7, for an audio signal A,
S is the time required for the depression of the transmission button until the line connection process is completed, i.e. when the a line connection time, from the time t ₀ the transmission button is pressed, the time of until the line connection end t ₁ t If the transmission is started at _S , the voice uttered between the time point t _S and the time point t ₁ will be lost as shown in the figure.

Therefore, in the prior art, the transmission side terminal delays the voice signal for a predetermined time so that the time at which the voice of the transmission start portion actually reaches the base station becomes equal to or later than the line connection end time. I was trying not to drop my head.

[0008]

In the above-mentioned prior art, no consideration is given to the delay of the entire conversation due to the delay of the voice signal, and when the conversation is performed until immediately before the transmission button is released, the last part of the conversation is missing. There was a problem that would occur.

That is, in the prior art, as shown in FIG. 8, a delay time D is given to the audio signal A to delay the audio signal A _D
By transmitting this from the terminal on the transmitting side, it is possible to eliminate head omission due to the existence of the line connection time T, but on the other hand, transmission and reception always have a delay of time D, and conversation like satellite communication is not performed. In addition, the sound becomes unnatural, and as shown in the figure, the sound is left even after the transmission button release time t _E , so that the remaining part is lost, and a so-called final part of the sound is lost. .

[0010] It is an object of the present invention to provide a communication signal transmission system in which a speech loss is not generated at both a transmission start time and a transmission end time while maintaining a natural conversation state.

[0011]

SUMMARY OF THE INVENTION It is an object of the present invention to provide a communication system including a plurality of communication stations, in which a transmitting station time-compresses an audio signal for a predetermined period from the time of a transmission switching operation, and starts transmitting the time-compressed audio signal. This is achieved by delaying the time by a predetermined time from the transmission switching operation time.

[0012]

BRIEF DESCRIPTION OF THE DRAWINGS FIG. 1 is a block diagram showing a communication signal transmission system according to the present invention.
FIG. 1 shows an example of a terminal to which the first embodiment of the present invention is applied. This terminal is used, for example, as terminals X, Y, and Z in the communication system of FIG. FIG. 1 shows only the components necessary for one embodiment of the communication signal transmission system according to the present invention.

In this embodiment, as shown in FIG. 1, an A / D (analog / digital) for converting an analog audio signal output from a microphone 1 and a transmission switch 2 for transmitting voice and a microphone 1 into a digital signal. A) a converter 3, an audio storage device 4 for storing a digital audio signal A output from the A / D converter 3, a writing control device 5 for controlling writing of a digital audio signal to the audio storage device 4, and a storage device. A read control device 6 controls reading of digital audio signals from the digital audio signal 4, and various paths 11 to 16 connecting between them and an output path 17.

Here, the transmission switch 2 corresponds to the transmission button, and the point in time when the transmission button is pressed is the point in time of the transmission switching operation.

A write counter is incorporated in the write control device 5 and a read counter is incorporated in the read control device 6, and when the transmission switch 2 is turned on,
That is, when the transmission button is pressed, a transmission switching operation is given, and these counters are activated.

[0016] Then, the digital audio signal A _C is read out to the output path 17, transmitted by a modulation circuit (not shown) transmitting circuit, entrained in a carrier wave of a predetermined frequency is transmitted from the antenna as radio waves to the base station It is supposed to be.

Next, the operation of the first embodiment will be described. An analog audio signal output from the microphone 1 is transmitted to an A / D converter 3 via an analog audio path 11.
, Where it is converted into a digital signal having a fixed bit length every predetermined sampling period, and is supplied to the audio storage device 4.

As shown in the figure, the audio storage device 4 includes a plurality of entries (memory areas), and a set of sampled fixed bit data is stored for each entry. Therefore, each entry has a unique address, and writing and reading of data to and from the voice storage device 4 are performed in entry units by designating this address.

The transmission switch 2 is connected to the write control device 5 and the read control device 6 via the write control path 13 and the read control path 14, and therefore, when the transmission switch 2 is turned on, Thus, the write control device 5 and the read control device 6 start operating.

Then, the write control device 5 resets a built-in write counter by turning on the transmission switch 2, starts a count operation, and generates a write signal. Since the count output of the write control device 5 is the write address of the voice storage device 4, when the transmission switch 2 is turned on, a write signal is input to the voice storage device 4 and the digital voice signal is stored. Is started, and digital audio signals are written sequentially from entry 0.

The read control device 6 also resets the built-in read counter when the transmission switch 2 is turned on, and starts a counting operation after a lapse of a predetermined time. The output of the read counter is an audio storage device. As a result, the digital audio signal is read from the entry 0 after the transmission switch 2 is turned on and a predetermined time has elapsed, and the digital audio signal is sequentially transmitted to the transmission output path 17. Is read out.

Therefore, the write counter of the write controller 5 and the read counter of the read controller 6 are reset here.
If (update) is performed in the same cycle, and writing and reading of the digital audio signal are wrapped around from the last entry N to the first entry 0 of the audio storage device 4, the result becomes the same as that of the prior art. .

That is, in this case, the transmission output path 1
7, a voice which is always delayed from the voice input from the microphone 1 by a predetermined time is output as a digital signal, and a conversation delay occurs as in the prior art. In the worst case, the voice The lack of the final part occurs.

Therefore, in this embodiment, the read counter of the read control device 6 is characterized in that the count operation is not counted up one by one, but is performed by a jump operation as appropriate. Hereinafter, the operation of this embodiment will be described with reference to FIG.
This will be described below. In FIG. 2, the numbers on the time axis are the numbers of the times representing the sampling times.
It is also the count value of the counter.

First, the write counter of the write control device 5
As shown in FIG. 2 (a), when activated by pressing the time t ₀ of the send button, and are configured to Yuku it is sequentially counted up immediately from zero.

On the other hand, the read counter of the read control device 6 also
Also activated by pressing the send button, but
In this case, adapted to be activated as shown in FIG. 2 (b), rather than immediately started in pressing time t _0, then a certain time, i.e., at time t ₁ the delay time τ has elapsed I have. The delay time τ is determined to be the same time or a slightly longer time depending on the time required for the line connection at the base station. In this embodiment, the delay time τ is determined to be three sampling periods. .

[0027] Then, further the read counter, the time t ₁ later, instead of being sequentially one count by one count up from the sampling numbers 0, as shown,
Until a predetermined time T elapses, jump counting is performed. As a result, the data in the sampling period of the skipped count value is not read out from the audio storage device 4 and is skipped.

In this embodiment, the number of sampling periods that are skipped, that is, the number of sampling periods that are decimated, is three, which is the same as the number of sampling periods that are between the delay times τ. They are numbered 2 and 5, and number 8.

In this case, as shown in the figure, when a predetermined time, that is, the compression time T has elapsed, the count value of the write counter matches the count value of the read counter. Will be continued.

FIG. 3 shows the state of the audio signal at this time. Assuming that the audio signal A has a waveform as shown in FIG. From time t ₀ , sampled audio data is stored sequentially from No. 0 in each sampling period according to the count value of the write counter.

Then, the stored audio data is read out in accordance with the count value of the read counter and output as output audio data A _C. At this time, the output audio data A _C is read first. The time at which
As shown in (b), it is time t ₁ , and after that, 3
Since there is no count value in the fifth, fifth, and eighth sampling periods, the reading of the audio data is thinned out.

[0032] Consequently, as shown, until the elapse of time T, a portion of the sampled data is audio data A _C of decimated state output from the later time T, the input speech data The audio data A _{C in the} same state as A is output.

Since the storage operation and the read operation are actually delayed, the sampling time shown in FIG.
Shift occurs at the sampling time in the audio data A _C
Although the sampling time is slightly delayed from the sampling time of the audio data A, this delay is ignored in FIG. However, this delay can be on the order of a few milliseconds, so practically it can be almost ignored,
There is no danger of giving the conversation an unnatural feeling.

[0034] Therefore, according to this first embodiment, as shown in FIG. 4 (a), pressing time t ₀ after the transmission switch 2, the compression time T has started to be inputted from the microphone 1 to until passage As shown in FIG. 2B, the sound is first delayed until the line connection time t ₁ , and from this time t _1, the sound starts to be transmitted as time-compressed sound by thinning out the sampling data. After the passage of time, it is transmitted as normal audio that is not time-compressed.

As a result, according to the first embodiment, even if the transmission starts immediately after the transmission switch 1 is pressed, as shown in FIG. Only the sound whose sound range is shifted to the high frequency side due to the compression can be heard from the beginning, so that the head drop can be reliably eliminated.

In this case as well, after the elapse of the compression time T, the sound is in a normal sound range, and the delay between the input sound and the output sound can be neglected. There is very little, and the level of normal conversation can be easily maintained.

The shift of the voice range to the high frequency side at the beginning of the transmission starts with the delay time τ, the compression time T,
Since it can be changed depending on the frequency of the jump count or the like, it is possible to select an arbitrary range according to the operating conditions so as to be shifted.

Next, a second embodiment of the present invention will be described. In the first embodiment, the operation shown in FIG. 4 is obtained by controlling the reading of audio data from the audio storage device 4, but the same operation can also be obtained by controlling the writing of audio data. be able to. Therefore, in the second embodiment, the present invention is implemented using the audio data write control.

As shown in FIG. 5, in this second embodiment, as shown in FIG. 5 (a), and starts the count-up of the write counter of the write control unit 5 at the transmission button pressing time t ₀ The points are the same as in the embodiment of FIG. 2, except that the count-up is skipped three times during the compression time T, and
At this time, the writing signal is not generated.

On the other hand, the read counter of the read control device 6
FIG 5 (b) as shown in the, rather than being immediately activated by the transmission button pressing time t _0, then a certain time, that there be configured to be activated at the time t ₁ the delay time τ has elapsed After that, unlike the embodiment of FIG. 2, the counter is configured to count up one by one in order from the count value 0.

As a result, the operation of storing the audio signal in the audio storage device 4 by the writing control device 5 is as follows. Submit button pressing time after t _0, the sampling data at the time number 0 point of the voice signal input from the microphone 1 is stored in the entry 0 of the voice storage device 4, the sampling data at the time number 1 point stored in the entry 1 Is done.

However, the sampling data at the time number 2 is not stored, the sampling data at the time number 3 is stored in the entry 2, and the sampling data at the time number 4 is stored in the entry 3. You.

The sampling data at the time number 5 is not stored, and the sampling data at the time number 6 is stored in the entry 4 and the sampling data at the time number 7 is stored in the entry 5 sequentially. However, storage of the sampling data at time point 8 is not performed.

At each time after time number 9, all sampling data is stored in each entry.
As a result, after this point in time, the sampling data is sequentially stored in each entry at all time numbers, with the time number and the entry number being shifted by three.

As a result, the audio signal is stored in the audio storage device 4 in a state where the sampling of the audio signal at three points in the time numbers during the compression time T is skipped and thinned out. Will be done.

Meanwhile, the reading operation of the audio signal from the audio memory device 4 according to the read controller 6, only the beginning is the time t ₁ the delay time τ has elapsed from the transmission button pressing time t _0, is subsequently, The operation is to read sampling data from each entry in the order of numbers from the first entry 0.

As a result, during the period of the compression time T, of the nine sampling data from time numbers 0 to 8, three data are skipped, and the thinned out six sampling data are the time t _1. Is read out within the period from the end of the compression time T to the end of the compression time T. Therefore, the operation shown in FIG. 4 is obtained as in the first embodiment.

Therefore, in the second embodiment, as in the first embodiment, even if the transmission is started immediately after the transmission switch 1 is pressed, as shown in FIG. In this case, all sounds can be heard from the beginning only by shifting the sound range to the high frequency side, so that the risk of head dropping can be reliably eliminated.

In this case as well, after the elapse of the compression time T, the sound is in the normal sound range, and the delay between the input sound and the output sound is negligible. There is very little, and the level of normal conversation can be easily maintained.

It should be noted that also in the second embodiment, the shift of the sound range of the voice at the first part at the start of transmission is caused by the delay time τ,
Since it can be changed depending on the compression time T, the frequency of the jump count, and the like, it goes without saying that it is possible to select any shift in the sound range according to the operating conditions.

In the above embodiment, the jump frequency and the jump width are all fixed values, but they may be variable. Normally, immediately after the transmission button is pressed, the probability of starting the transmission is low, and the probability increases with the passage of time from this point.

Thus, for example, within the compression time after the transmission button is pressed, the jump frequency and width are set to be large in the first direction, and are gradually reduced from time to time so that they become zero at the end of the compression time. , So first,
The compression rate of the voice may be large, and may be gradually reduced. In this way, the probability that the voice is shifted to a high frequency range at the start of transmission is suppressed, so that a more natural compression rate is obtained. Call can be obtained.

In the above embodiment, the radio system of the press-talk system (simplex system) shown in FIG. 6 has been described as an example. However, the present invention is not limited to this.
The present invention can be applied to a duplex wireless communication system and further to a communication system using a wired line.

[0054]

According to the present invention, even if the transmission is started immediately after the transmission button is depressed, there is no possibility that the head of the call will drop. Can be suppressed.

Further, according to the present invention, there is almost no possibility that a delay will occur in a call, and there is no possibility of voice loss at the final part of the transmission. Can be suppressed.

[Brief description of the drawings]

FIG. 1 is a block diagram illustrating an example of a terminal of a wireless system to which an embodiment of a call signal transmission method according to the present invention is applied.

FIG. 2 is a timing chart for explaining the operation of the first embodiment of the present invention.

FIG. 3 is a waveform chart for explaining the operation of the first embodiment of the present invention.

FIG. 4 is a timing chart for explaining a transmission operation according to the communication signal transmission method of the present invention.

FIG. 5 is a timing chart for explaining the operation of the second embodiment of the present invention.

FIG. 6 is an explanatory diagram showing an example of a wireless system to which the present invention is applied.

FIG. 7 is an explanatory diagram of head drop in a press-talk type communication system.

FIG. 8 is a timing chart for explaining an operation in a wireless system according to the related art.

[Explanation of symbols]

 Reference Signs List 1 microphone 2 transmission switch (transmission button) 3 A / D converter 4 audio storage device 5 write control device 6 read control device 11 analog audio path 12 digital audio path 13 write control path 14 read control path 15 audio write path 16 Audio readout path 17 Audio output path

Claims (2)

[Claims] In a communication system including a plurality of communication stations, an audio signal is time-compressed for a predetermined period from a transmission switching operation time at a transmission station, and the transmission start time of the time-compressed audio signal is changed from the transmission switching operation time. A call signal transmission method characterized by being configured to delay by a predetermined time. 2. The invention according to claim 1, wherein the time compression is given by skipping sampling of the audio signal, and the delay of the predetermined time is a writing time and a reading time of the audio signal with respect to a storage device. A communication signal transmission system characterized by being provided by the difference between