KR20020069718A - Reduction method for voice cut caused by rate renegotiation sequence - Google Patents

Abstract

PURPOSE: A method for reducing voice discontinuance generated due to a rate re-negotiation sequence is provided to detect, record and store noises in a normal traffic state in picture communication and to output the stored noises, instead of silence, if a rate re-negotiation sequence is generated. CONSTITUTION: A microcomputer in a picture phone judges whether a rate re-negotiation sequence is generated. In case that the rate re-negotiation sequence is generated, the microcomputer outputs the voice stored in a noise buffer to the speaker. If the rate re-negotiation sequence is not generated, the microcomputer outputs voice data to the speaker. Then the microcomputer judges whether voice data exist. In case that voice data exist, the microcomputer initializes a temporary buffer. If no voice data exist, the microcomputer stores the voice data outputted to the speaker in the temporary buffer and increases the noise count. If it is judged that the increased noise count exceeds a preset time, the microcomputer moves the contents stored in the temporary buffer to the noise buffer.

Description

REDUCTION METHOD FOR VOICE CUT CAUSED BY RATE RENEGOTIATION SEQUENCE}

According to the present invention, when a video communication device is used, a noise of a normal call state is detected and recorded for a silent period (hereinafter referred to as a silent period), and when the silent period occurs, the receiver detects the noise. The present invention relates to a method for reducing voice interruption by generating a silent section for reducing a voice interruption generated when a silent section is generated by exporting through.

1 is a block diagram showing the configuration of a conventional telephone. As shown in FIG. 1, a telephone line 11 connected to a telephone main body via an RJ-11 connector, a user picks up the handset, a switch is closed, and the user is in a call state. When the handset is put down, the hook switch 12 opens when the handset is disconnected, and the DeaE (DAA: Data Access Arrangement or less) (13) which appropriately controls the impedance or current of the telephone set required by the telephone company. And a hybrid 14 which circuitly cuts a telephone line operating at a high voltage and a low voltage, and outputs a signal having an appropriate magnitude, converts an analog signal received from the hybrid 14 into a digital signal, A data codec unit 15 for converting a digital signal received from the data pumping unit 16 into an analog signal, and the data The data pumping unit 16 demodulates the data inputted from the codec unit 15 and modulates the data received from the following microcomputer 17, and processes the data received from the data pumping unit 16, A microcomputer 17 for controlling each function, and a voice codec unit for converting digital voice data input from the microcomputer 17 into an analog voice signal and converting an analog voice signal received from a handset into digital voice data ( 18), a memory unit 19 for storing various data and connection parameters, and a receiver 20 for receiving voice, the operation of the conventional apparatus configured as described above is as follows.

First, when the user lifts the handset of the telephone main body connected through the telephone line 11, the hook switch 12 is closed and the call is in a call state. When the user puts down the handset, the hook switch 12 is opened and the call is blocked.

Thereafter, the DS 13 appropriately controls the impedance or current of the telephone set required by the telephone company, and the hybrid 14 circuit-blocks the telephone line operated at a high voltage and the telephone operated at a low voltage, and a signal of an appropriate size. Is transmitted to the data codec unit 15.

Then, the data codec unit 15 converts the analog signal received from the hybrid 14 into a digital signal and transmits the digital signal to the data pumping unit 16, and then converts the digital signal received from the data pumping unit 16 to the analog signal. Is converted to the telephone line 11 via the hybrid 14.

Next, the data pumping unit 16 demodulates the data received from the data codec unit 15 and transmits the demodulated data to the microcomputer 17, modulates the data received from the microcomputer 17, and then modulates the data codec unit ( To 15).

The microcomputer 17 processes the data received from the data pumping unit 16, controls each function of the modem (image telephone), and transmits data to the counterpart modem (image telephone) to the data pumping unit 16. .

Accordingly, the voice codec unit 18 converts the digital voice data received from the microcomputer 17 into an analog voice signal and transmits the analog voice signal to the handset 20, and transmits the analog voice signal received from the handset 20 to the digital voice. The data is converted into data and transmitted to the microcomputer 17.

Finally, the memory unit 19 stores various data necessary for communication, and voice signals such as noise.

In general, when a caller makes a sound during a call, the call is heard through the handset 20. When the caller does not make a sound during the call, noise by the noise generated in the vicinity is heard through the handset 20.

However, when a silent section occurs during a video call, not only the above-mentioned call sound or noise is heard but also no sound can be heard through the receiver 20 at all.

In other words, during a video call using a video phone, a silent section occurs according to a change in communication status or a transmission rate. At this time, a voice interruption in a completely silent state occurs without any sound from the receiver for about 1 second.

For reference, when two different people make a phone call, the ratio of time that voice is present and time when it is not is about 3: 8 is known. In other words, when one person speaks, the voice and the silence are heard in the ratio of 3: 8 when the other person does not speak.

However, in the conventional apparatus operating as described above, the voice dropping phenomenon occurring during the silent section may cause the caller who is continuously listening to a certain sound to be sufficiently uncomfortable, i.e., a misjudgment that the call with the other party is disconnected. It can be unpleasant, and if you are making an important call, you may feel suspicious of eavesdropping. However, there is a problem in that there is no provision for reducing the voice interruption occurring during the silent period.

Therefore, the present invention was created in order to solve the above-mentioned conventional problems, and when only a noise is detected in a normal call state during a video call using a video phone, it is recorded as a silent section time and stored in a memory, and a silent section occurs. Accordingly, an object of the present invention is to provide a method for reducing voice interruption by generating a silent section in which a substitute voice stored in a memory is output to a receiver instead of silence.

1 is a block diagram showing the configuration of a conventional telephone.

Figure 2 is an exemplary view showing a silent state during a call of the present invention.

Figure 3 is a flow chart showing the sequence of the present invention speech interruption reduction method.

In order to achieve the above object, the present invention provides a first step of determining whether a silent section is generated when an image is connected, and when a silent section occurs as a result of the determination of the first step, the voice stored in the noise buffer is transferred to the speaker. A second step of outputting, and a third step of determining whether voice data exists after outputting voice data to a speaker when a silent section is not generated as a result of the first step, and as a result of the determination of the third step, If there is data, initialize the temporary buffer (temp buffer) .If the voice data does not exist, store the audio data output from the speaker in the temporary buffer (temp buffer), and determine whether it is larger than the set time while increasing the noise count. In the fourth step and the determination result of the fourth step, if the generated noise count is larger than the set time, the stored temporary buffer (temp buffer) And a fifth step of transferring the dragon to the noise buffer.

Hereinafter, with reference to the accompanying drawings, a method for reducing voice interruption according to the present invention will be described in detail.

First, a video connection is attempted using a video telephone, and when a video connection is made, it is determined whether a silent section is generated.

As a result of the above determination, when a silent section occurs, the noise stored in the noise buffer, that is, the stored noise when there is no voice data during normal call, is output through the speaker. As a result, after the audio data is output to the speaker, it is determined whether the voice data exists.

As a result of the above determination, if the voice data is present, the noise count is zero, that is, the temporary buffer (temp buffer) is initialized. If the voice data does not exist, the output voice data is stored in the temporary buffer (temp buffer). Save and increase the noise count to see if it is greater than the set time (here 2 seconds).

If the generated noise is larger than the set time, the voice data contents stored in the temporary buffer (temp buffer) are transferred to the noise buffer.

Accordingly, when the silent section occurs, the noise of the stored noise buffer is passed through the above steps, and when the silent section occurs, the noise is sent to the receiver instead of the complete silence.

2 is a diagram illustrating a silent state occurring during a call. As shown in FIG. 2, a voice call in a communication connection state through a modem is different from an analog voice call state (a call state of a general telephone). Digital processing eliminates the need to consider echo output signal components.

Therefore, the caller can detect the silent state of the pure counterpart from the person who does not speak.

For reference, the microcomputer detects the silent section from the same voice signal and records the signal at this time in the memory. If the silent section occurs for some reason thereafter, the microcomputer immediately outputs the silent signal recorded in the memory to the voice codec, This allows the caller to continue the natural call without audible interruptions.

As described above, in the present invention, unlike the analog voice call state, the voice call in the communication connection state through the modem does not need to consider the echo signal component because all signals go through digital processing.

In addition, when a silent section occurs, by outputting a silent signal recorded in the memory to the voice codec unit, the caller can continue a natural call without sudden sound interruption.

Claims (6)

A first step of determining whether a silent section occurs during image connection; a second step of outputting a voice stored in a noise buffer to a speaker when a silent section occurs as a result of the determination of the first step; and the first step As a result, if the silent section does not occur, and outputting the voice data to the speaker after the third step of determining the presence of the voice data, and as a result of the determination of the third step, if the voice data is present, initialize the temporary buffer, If the voice data does not exist, the fourth step of determining whether the voice data output from the speaker is stored in the temporary buffer, the noise count is greater than the set time while increasing the noise count, and as a result of the determination of the fourth step, The silent phrase, characterized in that the fifth step of moving the contents of the stored temporary buffer to the noise buffer when the time is larger than the set time How to reduce the voice interruption due to the occurrence. 2. The method of claim 1, wherein the silent section of the first step is a voice interruption in a completely silent state in which no sound is heard from the receiver for about 1 second. The voice of claim 1, wherein the noise buffer of the second step outputs the stored noise to the speaker when the voice data is larger than the set noise count time when the voice data does not exist in a normal communication state. How to reduce stuttering. The method of claim 1, wherein the third step is to determine whether voice data is present in order to detect noise in a normal call state without generating a silent section. The temporary buffer of claim 4, wherein the temporary buffer of the fourth step is initialized to a temporary buffer with a noise count of zero when voice data is present, and outputted to a speaker when a silent section occurs when voice data is not present. A method for reducing speech interruption caused by silent section generation, characterized in that the temporarily stored sound data. The silent section as claimed in claim 1, wherein the fifth step transfers the contents of the stored temporary buffer to the noise buffer when the generated noise count is greater than the set noise count. How to reduce voice interruption caused by generation.