Communication terminal device and buffer control method

The present invention relates to a packet communication system, and more particularly to a communication terminal device and a buffer control method suitable for use in real-time communication.

Conventionally, a packet communication system that performs real-time communication using a packet network such as an IP (Internet Protocol) network is known. For example, there are IP telephone systems and video conferencing systems. In a packet network, the delay time required for packet transmission fluctuates, so that the time required for a packet to arrive from a transmitting terminal to a receiving terminal fluctuates. Further, the same time fluctuation occurs due to the error of the clock frequency between the transmitting terminal and the receiving terminal. In real-time communication, the packet arrival fluctuation (jitter) is one of the factors that deteriorate the communication quality. For this reason, as a measure against jitter , a buffer for absorbing jitter (hereinafter referred to as a jitter buffer) is provided in the receiving terminal . Then, the receiving terminal reads the received packet from the jitter buffer after accumulating a certain amount (initial accumulated amount) of the received packet in the jitter buffer, and reproduces the voice or the like. Further, as a method of adjusting the initial storage amount of the jitter buffer, a method of increasing or decreasing the initial storage amount according to the jitter amount at the time of arrival of the immediately preceding packet is known.<patcit num="1"><text>Japanese Patent No. 3075246</text></patcit>

<p> However, in the above-mentioned conventional technique, since the initial accumulation amount of the jitter buffer is adjusted according to the amount of jitter at the time of the arrival of the immediately preceding packet, there is a possibility that the adjustment is limited to the momentary adjustment corresponding to the current state of the packet network. There is a problem that proper adjustment is not always made. For example, if the packet arrival delay increases at a certain moment, the initial accumulation amount increases in accordance with the delay fluctuation, even though the delay increase is instantaneous. As a result, a redundant amount of packets stays in the jitter buffer, and an unnecessary fixed delay is added to the received packets.</p><p> Further, when a plurality of packets whose arrival is delayed due to network congestion or the like arrive together, the accumulated amount of the jitter buffer sharply increases. However, in the conventional technology, since the increase is left to the natural elimination, it takes time to eliminate the increased retention amount, and the fixed delay added to the received packet continues for a long time. Deterioration of quality of real-time communication becomes a problem.</p><p> The present invention has been made in consideration of such circumstances, and an object of the present invention is to appropriately adjust the initial accumulation amount of the jitter buffer and to reduce the fixed delay added to the received packet by the jitter buffer. It is an object of the present invention to provide a communication terminal device and a buffer control method that can be controlled so as to be properly maintained.</p>

<p> In order to solve the above problems, the communication terminal device according to the present invention is a communication terminal device provided with a jitter buffer that connects to a packet network to perform packet communication and temporarily stores received packets for a certain period of time in the past. Jitter amount calculating means for calculating the jitter amount in the period from the received packet, initial storage amount setting means for updating the initial storage amount of the jitter buffer based on the calculated jitter amount, and storage amount of the jitter buffer. It is characterized in that it is provided with a speed control means for changing the packet output speed from the jitter buffer based on the above.</p><p> The communication terminal device according to the present invention is characterized by including a threshold value setting means for setting a threshold value for determining a change in the packet output speed based on the calculated jitter amount.</p><p> The communication terminal device according to the present invention includes a reproduction means for reproducing the received data included in the received packet, and the speed control means adjusts the reproduction speed by the reproduction means in accordance with the change in the packet output speed. It is characterized by changing.</p><p> The buffer control method according to the present invention is a buffer control method in a communication terminal device provided with a jitter buffer that connects to a packet network to perform packet communication and temporarily stores received packets, and is received in a certain period in the past. A process of calculating the amount of jitter in the period from the packet, a process of updating the initial accumulation amount of the jitter buffer based on the calculated jitter amount, and a packet from the jitter buffer based on the accumulation amount of the jitter buffer. It is characterized by including the process of changing the output speed.</p><p> The buffer control method according to the present invention is further characterized by including a process of setting a threshold value for determining a change in the packet output speed based on the calculated jitter amount.</p><p> The buffer control method according to the present invention is characterized by further including a process of changing the speed of reproducing the received data included in the received packet in accordance with the change of the packet output speed.</p>

<p> According to the present invention, the amount of jitter in the period is calculated from the packets received in the past fixed period, and the initial accumulation amount is updated based on this jitter amount. Therefore, the initial accumulation amount of the jitter buffer is appropriately adjusted. be able to.</p><p> Further, since the packet output speed from the jitter buffer is changed based on the accumulated amount of the jitter buffer, it is possible to control so as to keep the fixed delay added to the received packet by the jitter buffer appropriately.</p>

Hereinafter, an embodiment of the present invention will be described with reference to the drawings. In the present embodiment, an IP telephone system using voice packets will be described as a specific example of the packet communication system. The present invention is applicable not only to voice packets but also to various packets such as image packets, and is particularly suitable for use in real-time communication. For example, it can be applied to a video conferencing system and the like.

FIG. 1 is a block diagram showing a schematic configuration of an IP telephone system according to an embodiment of the present invention. In FIG. 1, packet telephones 1a and 1b (communication terminal devices), which are IP telephone terminals, are connected to the packet network 100, respectively. The packet telephones 1a and 1b send and receive voice packets to and from each other via the packet network 100. Call voice data is stored in the voice packet. Voice packet communication using this voice packet enables voice communication between packet telephones 1a and 1b.

FIG. 2 is a block diagram showing a configuration of packet telephones 1a and 1b (hereinafter, referred to as packet telephone 1 unless otherwise specified) shown in FIG. In FIG. 2, the packet telephone 1 has a network connection interface unit 11, a voice packet transmission unit 12, a voice packet reception unit 13, a jitter buffer 14, a voice coding unit 15, a voice decoding unit 16, a microphone 17, a speaker 18, and a buffer threshold. It includes a setting unit 20 and a buffer control unit 21. Hereinafter, the configuration and operation of the packet telephone 1 will be described in detail with reference to FIG.

First, the network connection interface unit 11 has a connection interface function with the packet network 100, and connects to the packet network 100 to send and receive packets.

Next, the configuration and operation related to voice packet transmission will be described. The voice signal input by the microphone 17 is input to the voice coding unit 15. The voice coding unit 15 digitizes and encodes this input voice and outputs it to the voice packet transmission unit 12. The voice packet transmission unit 12 packetizes this voice-encoded data and outputs it to the network connection interface unit 11. The network connection interface unit 11 transmits this voice packet to the packet network 100.

Next, the configuration and operation related to voice packet reception will be described. When the network connection interface unit 11 receives a voice packet from the packet network 100, the network connection interface unit 11 outputs the voice packet to the voice packet receiving unit 13. The voice packet receiving unit 13 stores the received packet in the jitter buffer 14.

The jitter buffer 14 temporarily stores received packets. The initial storage amount of the jitter buffer 14 is variable. In the jitter buffer 14, the accumulated received packets are not output until the accumulated amount of the received packets reaches the initial accumulated amount after the accumulation of the received packets is started. Then, after the received packets corresponding to the initial accumulated amount are accumulated, the output of the accumulated received packets is started. This packet output order follows the accumulated order. Further, the packet output speed from the jitter buffer 14 is variable. It is also possible to pause and resume packet output.

The voice decoding unit 16 decodes and analogizes the voice-coded data in the received packet output from the jitter buffer 14 and outputs the voice-coded data to the speaker 18. Then, the speaker 18 reproduces the input audio signal. In the audio decoding unit 16, the audio reproduction speed is variable.

Next, the configuration and operation related to the adjustment of the accumulated amount of the jitter buffer 14 described above will be described. The buffer threshold setting unit 20 sets various buffer thresholds related to the adjustment of the accumulated amount of the jitter buffer 14. The threshold value to be set is a value corresponding to the amount of jitter obtained by calculating the amount of jitter in the period from the packets received in the past fixed period.

FIG. 3 is a block diagram showing the configuration of the buffer threshold value setting unit 20. FIG. 4 is an explanatory diagram for explaining the jitter amount calculation method according to the present embodiment. In FIG. 3, the fluctuation calculation unit 202 calculates the jitter amount of the packet received in the fixed reference period immediately before the switching time point at each fixed switching interval. The timing generation unit 201 generates the switching timing and outputs it to the fluctuation calculation unit 202. FIG. 4 shows the times t1, t2, t3, ..., T4, t5, and t6 for each switching interval. Then, the fluctuation calculation unit 202 calculates the jitter amount of the packet received in the reference period from the past time t3 to t6 at the time t6, for example. The reference period is set so as to include not only the immediately preceding arrival packet but also a plurality of past arrival packets.

The fluctuation calculation unit 202 receives the reception time for each packet received from the voice packet reception unit 13, and also receives the transmission time included in the received packet. The transmission time is the time when the packet telephone 1 of the communication partner transmits the packet. Then, the difference between the reception time and the transmission time is calculated, and the amount of jitter in the reference period is obtained based on this time difference. For example, in the reference period, the standard deviation of the fluctuation amount of the time difference between the reception time and the transmission time is calculated. Alternatively, the amount of jitter specified in the IETF (Internet Engineering Task Force) RFC 3550 is calculated. The fluctuation calculation unit 202 outputs the amount of jitter in the current reference period to the threshold value calculation units 203, 204, 205, and 206 for each switching interval.

Each threshold value calculation unit 203, 204, 205, 206 calculates each threshold value according to the amount of jitter from the fluctuation calculation unit 202 and sets it in the buffer control unit 21. FIG. 5 is an explanatory diagram for explaining the setting of various buffer threshold values related to the adjustment of the accumulated amount of the jitter buffer 14. As shown in FIG. 5, the high-speed reproduction threshold value, the reproduction start threshold value, the low-speed reproduction threshold value, and the reproduction stop threshold value are set in descending order of the threshold values.

The reproduction start threshold value calculation unit 203 calculates the reproduction start threshold value according to the amount of jitter. The reproduction start threshold value is a threshold value that defines the initial storage amount of the jitter buffer 14. The high-speed reproduction threshold value calculation unit 204 calculates the high-speed reproduction threshold value according to the amount of jitter. The high-speed reproduction threshold value is a threshold value that defines the amount of accumulation for determining whether or not the packet output speed from the jitter buffer 14 and the voice reproduction speed are higher than the normal speed. The low-speed reproduction threshold calculation unit 205 calculates the low-speed reproduction threshold according to the amount of jitter. The low-speed reproduction threshold value is a threshold value that defines the amount of accumulation for determining whether or not the packet output speed from the jitter buffer 14 and the audio reproduction speed are lower than the normal speed. The reproduction stop threshold value calculation unit 206 calculates the reproduction stop threshold value according to the amount of jitter. The reproduction stop threshold value is a threshold value that defines the amount of accumulation for determining whether or not to stop the packet output from the jitter buffer 14 and stop the audio reproduction.

The normal speed related to the packet output speed is a predetermined packetization speed at the time of packet transmission by the packet phone 1 on the transmitting side, and the packet phone 1 on the receiving side usually has a jitter buffer at the same speed as the packetization speed. Output the received packet from 14. The normal speed related to the voice reproduction speed is the input speed of the speaker voice in the packet telephone 1 on the transmitting side, and the packet telephone 1 on the receiving side usually reproduces the voice at the same speed as the input speed.

The buffer control unit 21 monitors the accumulated amount of the jitter buffer 14, and based on this accumulated amount and the above-mentioned high-speed reproduction threshold value, reproduction start threshold value, low-speed reproduction threshold value, and reproduction stop threshold value, packet output from the jitter buffer 14 and Controls the voice reproduction of the voice decoding unit 16. FIG. 6 is a transition diagram of the control state related to the buffer control unit 21.

In FIG. 6, first, when the accumulation of the received packet in the jitter buffer 14 is started, the accumulation is continued without outputting the packet from the jitter buffer 14 until the accumulated amount reaches the reproduction start threshold value. The audio decoding unit 16 does not reproduce the audio (step S1).

In the accumulation state of step S1, when the accumulated amount reaches the reproduction start threshold value, the output of the received packet is started from the jitter buffer 14, and the audio decoding unit 16 performs audio reproduction at a normal speed (step S2). In this normal playback state, the packet output speed from the jitter buffer 14 is the normal speed.

In the normal reproduction in step S2, when the accumulated amount increases to the high-speed reproduction threshold value, the packet output speed from the jitter buffer 14 is increased and the audio reproduction speed in the audio decoding unit 16 is increased (step S3). The amount of increase in the packet output speed and the audio reproduction speed correspond to each other. Due to the high-speed playback state in step S3, the amount of retention of received packets temporarily increased in the jitter buffer 14 can be quickly eliminated.

In the high-speed playback state of step S3, when the accumulated amount falls below the high-speed playback threshold value, the packet output speed from the jitter buffer 14 is returned to the normal speed, and the voice playback speed in the voice decoding unit 16 is also returned to the normal speed.

In the normal reproduction in step S2, when the accumulated amount decreases to the low-speed reproduction threshold value, the packet output speed from the jitter buffer 14 is decreased, and the audio reproduction speed in the audio decoding unit 16 is decreased (step S4). The amount of decrease in the packet output speed and the audio reproduction speed correspond to each other. Due to the low-speed playback state in step S4, the amount of retention of the received packets temporarily reduced in the jitter buffer 14 can be quickly returned to the normal level, and the playback voice can be prevented from being interrupted due to the shortage of received packets.

In the low-speed playback state of step S4, when the accumulated amount exceeds the low-speed playback threshold value, the packet output speed from the jitter buffer 14 is returned to the normal speed, and the voice playback speed in the voice decoding unit 16 is also returned to the normal speed.

In the low-speed reproduction state of step S4, when the accumulated amount further decreases and the reproduction stop threshold is reached, the packet output from the jitter buffer 14 is stopped, and the audio reproduction by the audio decoding unit 16 is also stopped. Then, it transitions to the accumulation state of the above step S1.

7 to 9 are graphs for explaining the method of changing the audio reproduction speed described above. In the method of changing the audio reproduction speed shown in FIG. 7, the reproduction speed is changed with a constant increase value in the high-speed reproduction state and with a constant decrease value in the high-speed reproduction state with respect to the reproduction speed in the normal reproduction state. To change. This change method is expressed by Eq. (1). h = n + k (1) However, h is the reproduction speed, n is the normal reproduction speed, and k is a constant (a positive value in the high-speed reproduction state and a negative value in the low-speed reproduction state).

In the method of changing the audio reproduction speed shown in FIG. 8, the reproduction speed is changed at a rate of increase proportional to the increase in the accumulated amount from the high-speed reproduction threshold value in the high-speed reproduction state with respect to the reproduction speed in the normal reproduction state. In the reproduction state, the reproduction speed is changed with a decrease rate proportional to the decrease in the accumulated amount. This change method is expressed by Eq. (2). h = n + ad (2) However, h is the playback speed, n is the normal playback speed, and d is the difference in the amount of accumulation from the threshold value (in the high-speed playback state, the absolute value of the difference from the high-speed playback threshold value, low speed). In the playback state, the absolute value of the difference from the low-speed playback threshold value), and a is the speed change coefficient (positive value in the high-speed playback state, negative value in the low-speed playback state).

In the method of changing the audio reproduction speed shown in FIG. 9, the reproduction speed is changed at a rate of increase proportional to the square of the increase in the accumulated amount from the high-speed reproduction threshold in the high-speed reproduction state with respect to the reproduction speed in the normal reproduction state. In the low-speed reproduction state, the reproduction speed is changed with a decrease rate proportional to the square of the decrease in the accumulated amount. This change method is expressed by Eq. (3). h = n + ad² (3) However, h is the playback speed, n is the normal playback speed, and d is the difference in the amount of accumulation from the threshold value (difference from the high-speed playback threshold value in the high-speed playback state, and from the low-speed playback threshold value in the low-speed playback state). Difference), a is the speed change coefficient (positive value in the high-speed playback state, negative value in the low-speed playback state).

The above-mentioned change in the sound reproduction speed may be a simple change in the reproduction frequency, or may be a change in maintaining the pitch (pitch) of the reproduced sound at a constant level.

According to the above-described embodiment, the amount of jitter in the period is calculated from the packets received in the past fixed period, and the initial accumulation amount is updated based on this jitter amount. Therefore, it is appropriate to adjust the initial accumulation amount of the jitter buffer. Can be done.

Further, since the control for increasing the packet output speed from the jitter buffer is performed by using the high-speed reproduction threshold value, the retention amount of the received packet temporarily increased in the jitter buffer can be quickly eliminated.

In addition, since the low-speed playback threshold is used to control the packet output speed from the jitter buffer, the amount of accumulated received packets that has temporarily decreased in the jitter buffer is quickly returned to the normal level, and the playback audio due to the shortage of received packets Can be prevented from being interrupted.

As described above, according to the present embodiment, it is possible to control so as to appropriately maintain the fixed delay added to the received packet by the jitter buffer.

Further, since the reproduction speed is changed according to the change of the packet output speed from the jitter buffer described above, the quality of the reproduced voice can be kept good.

As described above, according to the embodiment of the present invention, an appropriate buffer amount is provided for time fluctuations caused by a state such as network congestion or an error in the clock frequency between the transmitting terminal and the receiving terminal. It becomes possible to control to maintain.

Although the embodiments of the present invention have been described in detail with reference to the drawings, the specific configuration is not limited to this embodiment, and design changes and the like within a range not deviating from the gist of the present invention are also included.

<figref num="1">It is a block diagram which shows the structure of the IP telephone system by one Embodiment of this invention.</figref><figref num="2">It is a block diagram which shows the structure of the packet telephone 1 (communication terminal apparatus) shown in FIG.</figref><figref num="3">It is a block diagram which shows the structure of the buffer threshold value setting part 20 shown in FIG.</figref><figref num="4">It is explanatory drawing for demonstrating the jitter amount calculation method which concerns on embodiment of this invention.</figref><figref num="5">It is explanatory drawing for demonstrating the setting of various buffer thresholds concerning the adjustment of the accumulation amount of the jitter buffer which concerns on embodiment of this invention.</figref><figref num="6">It is a transition diagram of the control state related to the buffer control unit 21 shown in FIG.</figref><figref num="7">It is a graph for demonstrating the method of changing the audio reproduction speed which concerns on embodiment of this invention.</figref><figref num="8">It is a graph for demonstrating the method of changing the audio reproduction speed which concerns on embodiment of this invention.</figref><figref num="9">It is a graph for demonstrating the method of changing the audio reproduction speed which concerns on embodiment of this invention.</figref>

Code description

1,1a, 1b ... Packet telephone (communication terminal device), 13 ... Voice packet receiver, 14 ... Jitter buffer, 16 ... Voice decoding unit, 20 ... Buffer threshold setting unit, 21 ... buffer control unit, 100 ... packet network, 201 ... timing generation unit, 202 ... fluctuation calculation unit, 203 ... playback start threshold (initial accumulation amount) calculation unit, 204 ... high speed Playback threshold calculation unit, 205 ... Low-speed playback threshold calculation unit, 206 ... Playback stop threshold calculation unit.