KR100652719B1 - Method for selecting the quality of sound in push-to-talk terminal - Google Patents

Abstract

The present invention relates to a method for selecting audio quality of a push-to-talk terminal. In the related art, a PTT terminal allocates a buffer having an appropriate size in consideration of network delay to prevent audio dropping, but does not allocate an appropriate buffer size to the network delay. As a result, there was a problem that audio disconnection occurred due to a change in communication environment of the network. In consideration of such a problem, the present invention includes the steps of determining a user's sound quality selection in a PTT terminal and setting a buffer size corresponding to the selected sound quality mode; Determining a buffer size that can be allocated and allocating a buffer corresponding to a sound quality mode; By buffering the voice packet received in the allocated buffer and decoding and outputting the voice data in real time, the user can directly select the voice quality to hear the other party's voice of the desired sound quality.

Description

TECHNICAL FOR SELECTING THE QUALITY OF SOUND IN PUSH-TO-TALK TERMINAL}

1 is a flowchart illustrating an audio sound quality selection method of a push-to-talk terminal according to an exemplary embodiment of the present invention.

2 is a flowchart illustrating an audio sound quality selection method of a push-to-talk terminal according to another exemplary embodiment of the present invention.

The present invention relates to a method for selecting an audio quality of a push-to-talk terminal, and more particularly, to a method for selecting an audio quality of a push-to-talk terminal, which can provide the best sound quality by overcoming the degradation of sound quality due to network delay and bandwidth limitation. It is about.

Push-To-Talk (PTT) service selects the group ID or user ID of a colleague or friend displayed on the terminal while the user is logged in, presses the PTT key and starts talking. When finished, release the PTT key. The PTT service uses RTP for real-time voice packet transmission using SIP for signaling transmission. Session Initiation Protocol (SIP) is a signaling protocol that exchanges session information required before starting a call, removes session information that was in progress when the call ended, and Real Time Transport Protocol (RTP) provides voice packets in real time when a call session is established. Provide a PTT service by sending.

Assuming a mobile communication terminal providing a PTT service in a mobile communication system as a PTT terminal, the PTT terminal selects a group ID or a user ID displayed while logged in and presses the PTT key to start a group call with another PTT terminal. The group call repeats the operation of transmitting the current speaker's voice to multiple users at the same time by using a dedicated channel, and a user who is not subscribed to the dedicated channel cannot hear the contents of the group call.

In a mobile communication system, a PTT terminal establishes a call session with a PTT server and a SIP protocol, transmits a voice packet through an RTP protocol, provides a PTT service, and outputs the current speaker as text to indicate the user's identity. The user of the PTT terminal uses the PTT service while looking at the current speaker outputted as text to confirm the identity of the speaker and to determine when to speak.

The PTT terminal digitizes the speaker's voice, converts it into digital data, encodes the digital data, outputs the voice data, and transmits the voice data in a voice packet in real time. The PTT terminal transmits the speaker's voice in digital form instead of in analog form to maintain the call quality at a predetermined level.

The PTT terminal attempts to prevent audio interruption by allocating a buffer having an appropriate size in consideration of network delay, but there is a problem in that audio interruption occurs due to a change in the communication environment of the network because an appropriate buffer size is not allocated to the network delay.

In addition, the PTT terminal maintains the sound quality at a predetermined level by using an encoding method according to the allocated bandwidth, but a low encoding rate has a problem in that the sound quality is degraded and the sound quality is degraded.

Accordingly, the present invention has been made in view of the above-mentioned problems, and the push-to-talk terminal is configured to overcome the degradation of sound quality due to network delay and bandwidth limitation by selecting different sound quality levels and different buffer sizes and encoding methods. The purpose of the present invention is to provide a method for selecting audio quality.

The present invention for achieving the above object comprises the steps of setting the buffer size corresponding to the selected sound quality mode by determining the user's sound quality selection in the PTT terminal; Determining a buffer size that can be allocated and allocating a buffer corresponding to a sound quality mode; And buffering the voice packet received in the allocated buffer and decoding and outputting the voice data in real time.

Hereinafter, exemplary embodiments of the present invention will be described in detail with reference to the accompanying drawings.

1 is a flowchart illustrating a method for selecting an audio quality of a push-to-talk terminal according to an exemplary embodiment of the present invention. As shown in FIG. 1, a PTT terminal determines a sound quality of a user and determines a buffer size corresponding to a selected sound quality mode. Setting up; Determining a buffer size that can be allocated and allocating a buffer corresponding to a sound quality mode; If a reception miss of the voice packet occurs more than a predetermined number of times, a step of outputting a message indicating the instability of the communication environment.

The PTT server relays the voice packet received from another PTT terminal to the PTT terminal, and the PTT terminal receives the voice packet from the PTT server. An error occurs in the transmission and reception of voice packets due to the difference in the communication environment depending on the location of each terminal between the PTT terminal and the PTT server, and this phenomenon causes a network delay.

The PTT terminal calculates the buffer size in consideration of the network delay and allocates a buffer of a predetermined size to prevent the exhaustion of the voice packet due to the network delay and to maintain the buffered voice packet at a predetermined level or more.

The PTT terminal determines the sound quality selection of the user and sets the buffer size corresponding to the selected sound quality mode among the highest sound quality, high sound quality, and general sound quality to provide an optimal sound quality. The PTT terminal determines the buffer size by calculating the network delay in normal sound quality, and sets the buffer size to be large as the highest sound quality mode is used to prepare for missed reception of voice packets due to sudden network delay.

The PTT terminal determines the buffer size that can be allocated and allocates a buffer corresponding to the sound quality mode. If the reception loss of voice packets occurs more than a predetermined number of times in the highest sound quality mode, the PTT terminal cannot provide the best sound quality due to instability of the communication environment. Output When the PTT terminal cannot provide the best sound quality, the PTT terminal recovers a predetermined size from the allocated buffer to manage the memory, or switches from the highest sound quality mode to the high sound quality mode to adapt to the missed reception of the packet more than the predetermined number of times.

FIG. 2 is a flowchart illustrating an audio sound quality selection method of a push-to-talk terminal according to another embodiment of the present invention. As shown in FIG. Adjusting; Determining the bandwidth allocated in the communication channel and determining an applicable encoding method to indicate whether a sound quality mode is provided according to a sound quality selection of a user; Periodically determining the bandwidth of the communication channel to determine and apply the applicable encoding method.

The PTT terminal transmits and receives voice data through a communication channel of the PTT service, and the communication channel has a bandwidth of a predetermined size. The PTT terminal requests and allocates the bandwidth required for the PTT service to the PTT server and transmits and receives voice data through a communication channel having the allocated bandwidth.

The PTT terminal adjusts the size of voice data by applying an encoding method differently according to a sound quality mode. The larger the voice data size, the better the sound quality but the more bandwidth is used.

The PTT terminal determines the sound quality selection of the user and provides an optimum sound quality by adjusting the encoding method according to the sound quality mode selected among the highest sound quality, high sound quality and general sound quality. The PTT terminal applies an encoding method having a low data rate in the general sound quality mode, and applies an encoding method having a higher data rate toward the highest sound quality mode. A high data rate improves voice quality due to the large size of voice data and low voice loss.

The PTT terminal determines the bandwidth allocated in the communication channel to be used and determines the applicable encoding method to indicate whether the sound quality mode is provided according to the sound quality selection of the user. The PTT terminal outputs a message that it cannot provide the best sound quality when the allocated bandwidth is small in the best sound quality mode and a high encoding method cannot be applied.

The PTT terminal determines the applicable encoding method by periodically determining the bandwidth of the communication channel in the highest quality mode and applies an encoding method corresponding to a changing communication environment. When the communication channel bandwidth is small, the encoding method with a low data rate is applied. When the communication channel is large, an encoding method with a high data rate is applied.

In another embodiment, the PTT terminal applies a buffer size and an encoding method in combination with the selected sound quality mode, adjusts and allocates a buffer size according to a communication environment, and selects and applies an encoding method according to a bandwidth. The PTT terminal adjusts the buffer size and encoding method differently according to the sound quality mode to prepare for a rapidly changing communication environment in real time.

As described in detail above, the present invention selects a sound quality level and sets a different buffer size and encoding method to overcome sound degradation due to network delay and bandwidth limitation, so that the user directly selects the voice quality to clean the other party's voice with the desired sound quality. Listening is effective.

Claims (6)

Determining a sound quality selection of the user by the PTT terminal and setting a buffer size corresponding to the selected sound quality mode; Determining a buffer size that can be allocated and allocating a buffer corresponding to a sound quality mode; And buffering the voice packet received in the allocated buffer and decoding and outputting the voice data in real time. The buffer setting of claim 1, wherein the buffer setting corresponding to the sound quality mode is configured to calculate the network delay in the general sound quality mode to determine the buffer size and to forcely set the buffer size toward the highest sound quality mode. -How to select the audio quality of the talk terminal. The method of claim 1, wherein the buffering of the voice packet determines the occurrence of missing reception, and notifies the instability of the communication environment when more than a predetermined number of times occurs. Determining a sound quality selection of the user by the PTT terminal and adjusting an encoding method according to the selected sound quality mode; Determining the bandwidth allocated in the communication channel and determining an applicable encoding method to indicate whether a sound quality mode is provided according to a sound quality selection of a user; The audio quality selection method of the push-to-talk terminal, characterized in that for determining the applicable encoding method by periodically determining the bandwidth of the communication channel. 5. The method of claim 4, wherein the adjustment of the encoding method according to the sound quality mode is performed by applying an encoding method having a low data rate in the general sound quality mode and applying an encoding method having a higher data rate toward the highest sound quality mode. How to select audio quality of two-talk terminal. The push-to-band encoding method according to claim 4, wherein the encoding method according to the bandwidth is configured to apply an encoding method having a low data rate when the bandwidth of the communication channel is small and an encoding method having a high data rate when the communication channel is large. -How to select the audio quality of the talk terminal.

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