Abstract:
An audio playback apparatus includes a playback section for playing audio back, a buffer section ( 13 ) for temporarily storing voice data to be played back, a switch ( 14 ) for on/off switching transmission of the voice data from the buffer section ( 13 ) to the playback section, at least one threshold judgment section ( 18 ) for judging whether an amount of voice data buffered in the buffer section ( 13 ) is equal to or smaller than at least one threshold value, the at least one threshold judgment section ( 18 ) providing a true output when the buffered voice data amount is equal to or smaller than the at least one threshold value, a timer ( 19 ) triggered by an output from the at least one threshold judgment section to output a trigger pulse with a predetermined duration of time, and a first control section ( 191 ) for controlling on/off of the switch ( 14 ) to prevent transmission of the voice data in the buffer section ( 13 ) to the playback section for the predetermined duration of time of the trigger pulse when the true output is provided from the at least one threshold judgment section ( 18 ).

Description:
PRIORITY CLAIM 
   This application claims priority from Japanese patent application No.2004-301611, filed on Oct. 15, 2004, which is incorporated herein by reference. 
   BACKGROUND OF THE INVENTION 
   1. Field of the Invention 
   The present invention relates to an audio playback apparatus and method for controlling pause and resume of audio. Particularly, the present invention relates to an audio playback apparatus used for conversation in an IP (Internet Protocol) phone and in an IP video telephony. 
   2. Description of the Related Art 
     FIG. 1  illustrates a functional configuration of a preceded digital audio playback apparatus that is not known in public. 
   The preceded digital audio playback apparatus  1  shown in  FIG. 1  is configured to receive packets containing compressed audio data and play back audio in the packets. This apparatus  1  has an audio packet receiver section  11 , an audio decoder  12 , a buffer  13 , a switch  14 , a D/A converter section  15 , an amplifier  16  and an initial buffering judgment section  17 . 
   The audio packet receiver section  11  receives packets containing compressed audio data from the network, and transmits the compressed audio data to the audio decoder  12 . The audio decoder  12  decodes the compressed audio data into non-compressed PCM (pulse code modulation) data and outputs the PCM data to the buffer  13 . The buffer  13  temporarily stores the PCM data and outputs them to the D/A converter section  15  through the switch  14 . The initial buffering judgment section  17  monitors the amount of data stored or buffered in the buffer  13  and controls based upon the buffered data amount on/off of the switch  14 . The D/A converter section  15  converts the PCM data input through the switch  14  into an analog signal and outputs the converted analog signal to the amplifier  16 . The analog audio signal output from the amplifier  16  is provided to the speaker  2  to play back the audio. 
   The initial buffering judgment section  17  is provided with a NOR gate  171 , a comparator  172 , an RS flip-flop  173  and an initial buffering value storage  174 . The NOR gate  171  outputs an “H” level signal to the flip-flop  173  when the buffered data amount becomes zero. The comparator  172  compares the buffered data amount in the buffer  13  with an initial buffering value and outputs a “H” level signal to the flip-flop  173  when the buffered data amount becomes equal to or higher than the initial buffering value stored in the storage  174 . The flip-flop  173  controls turning on/off of the switch  14  depending upon whether it is set or reset. 
   Hereinafter, operation of this audio playback apparatus will be described. 
   Under initial conditions, no PCM data is stored in the buffer  13 . Therefore, “0” is input to the NOR gate  171  and thus its output becomes “H” level signal. As a result, the flip-flop  173  is set to turn the switch  14  off, so that PCM data output from the audio decoder  12  are stored in the buffer  13  without being output to the D/A converter section  15 . 
   When the buffered amount of the PCM data in the buffer  13  becomes equal to or higher than the initial buffering value, the comparator  172  outputs an “H” level signal. Thus, the flip-flop  173  is reset and the switch  14  turns on, so that the PCM data stored in the buffer  13  is provided to the D/A converter section  15  and that the analog audio signal is provided to the speaker  2  through the amplifier  16  to play the audio back. 
   In such audio playback apparatus where packets containing audio data are received and audio in the packets is played back while receiving another packets, if receiving timings of the packets vary due to changes in the transmission rate through the network, it may occur that no packet can be received for a time period longer than the time equivalent to the buffered amount. In such case, all the buffered data may be extracted from the buffer to fall into an underflow state. 
   When the buffered amount of the PCM data in the buffer  13  becomes zero or underflows, the output of the NOR gate  171  becomes the “H” level signal, the flip-flop  173  is set and thus the switch  14  turns off. Thus, the PCM data output from the audio decoder  12  are not fed to the D/A converter section  15  but stored in the buffer  13 . Then, when the buffered amount of the PCM data in the buffer  13  becomes equal to or higher than the initial buffering value, the switch  14  turns on. 
   During a period where the switch  14  is off state, playback of audio is paused and therefore break or interruption of voice occurs. This operation of the buffer during the off state of the switch is called as a re-buffering operation. 
   If the audio packets arrive without delay, the buffered amount of the data will not lower than the initial buffering value. However if delay in arrival of the audio packets occurs, the buffered level goes downward. If the delay continues, it will cause underflow. Thus break or interruption of voice will occur due to the re-buffering operation. Then when the delay in arrival of the audio packets is over and the delayed packets arrive at a time, the buffered data amount will abruptly increase. 
   The initial buffering operation and the re-buffering operation should be carried out for a somewhat long time so that the buffering data amount never underflows again. In case of real time applications such as Voice over IP (VoIP) or IP video telephony, it is necessary to perform the re-buffering operation for a period of one hundred milliseconds to several hundreds milliseconds in consideration of tradeoff between the resiliency against delay variation. Whereas in case of non-real time applications such as video streaming, a period of the re-buffering operation is in general set to several seconds in order to give a particular importance to stability. 
   However, in case of applications for voice communication such as VoIP or IP video telephony, break or interruption of voice for a period longer than one hundred milliseconds will be clearly recognized and deteriorate quality of audio communications. Therefore, in order to improve the audio quality in the audio playback system for receiving packets containing audio data and playing audio in the packets back, it will be necessary to shorten the period of break or interruption of voice due to the re-buffering. 
   As for known technique of voice buffering in voice information communication, International Publication No. WO 01/01614 A1 discloses a system for changing the delay on a communication link by adjusting relative positions of read and write pointers of a buffer during silent periods. However, this known technique cannot shorten the period of break or interruption of voice due to the re-buffering. 
   SUMMARY OF THE INVENTION 
   It is therefore an object of the present invention to provide an audio playback apparatus, whereby a period of break or interruption of voice due to the re-buffering can be effectively shortened. 
   According to the present invention, an audio playback apparatus includes a playback section for playing audio back, a buffer section for temporarily storing voice data to be played back, a switch for on/off switching transmission of the voice data from the buffer section to the playback section, at least one threshold judgment section for judging whether an amount of voice data buffered in the buffer section is equal to or smaller than at least one threshold value, the at least one threshold judgment section providing a true output when the buffered voice data amount is equal to or smaller than the at least one threshold value, a timer triggered by an output from the at least one threshold judgment section to output a trigger pulse with a predetermined duration of time, and a first control section for controlling on/off of the switch to prevent transmission of the voice data in the buffer section to the playback section for the predetermined duration of time of the trigger pulse when the true output is provided from the at least one threshold judgment section. 
   It is preferred that the at least one threshold judgment section includes a threshold judgment section, storing a threshold value, for judging whether an old buffered data amount that represents the amount of the buffered data predetermined period earlier is larger than the threshold value, and the threshold value is larger than a buffered data amount at the present time. 
   It is also preferred that the at least one threshold judgment section includes a plurality of threshold judgment section for judging whether the buffered voice data amount is equal to or smaller than the respective threshold values, each of the plurality of threshold judgment section providing a true output when the buffered voice data amount is equal to or smaller than the threshold value, and that the at least one threshold judgment section provides a true output when the true output is provided from any one of the plurality of threshold judgment section. 
   It is further preferred that the at least one threshold value includes an initial buffering value required for audio playback at initial stage. 
   It is preferred that the trigger pulse from the timer has a duration corresponding to one frame in audio coding. 
   It is also preferred that the at least one threshold judgment section provides a true output of a “H” level when the buffered voice data amount is equal to or smaller than the at least one threshold value, and a false output of a “L” level when the buffered voice data amount is larger than the at least one threshold value, that the timer provides a “H” level output with a predetermined duration of time, and that the first control section includes an AND gate receiving the “H” level or “L” level output from the at least one threshold judgment section and the “H” level output from the timer, an output from the AND gate controlling on/off of the switch. 
   It is preferred that the at least one threshold judgment section provides a true output of a “H” level when the buffered voice data amount is equal to or smaller than the at least one threshold value, and a false output of a “L” level when the buffered voice data amount is larger than the at least one threshold value, that the timer provides a “H” level output with a predetermined duration of time, that the first control section includes an AND gate receiving the “H” level or “L” level output from the at least one threshold judgment section and the “H” level output from the timer, and that the apparatus further includes an initial buffering judgment section for monitoring the buffered voice data amount and providing a “H” level output when the buffered voice data amount is equal to or smaller than an initial buffering value, and a second control section having an OR gate receiving an output from the first control section and an output from the initial buffering judgment section, an output from the OR gate controlling on/off of the switch. 
   According to the present invention, also, an audio playback method executed by a computer includes a step of temporarily storing voice data to be played back into a buffer section, a step of on/off switching transmission of the voice data from the buffer section for playing audio back, a step of judging whether an amount of voice data stored in the buffer section is equal to or smaller than at least one threshold value to provide a true output when the buffered voice data amount is equal to or smaller than the at least one threshold value, a step of outputting a trigger pulse with a predetermined duration of time in response to the provided output, and a step of controlling the on/off switching step to prevent transmission of the voice data in the buffer section for the predetermined duration of time of the trigger pulse when the true output is provided. 
   It is preferred that the judging step includes a step of storing a threshold value, and a step of judging whether an old buffered data amount that represents the amount of the buffered data predetermined period earlier is larger than the threshold value, and the threshold value is larger than a buffered data amount at the present time. 
   It is also preferred that the judging step includes a plurality of steps of judging whether the buffered voice data amount is equal to or smaller than a plurality of threshold values, each providing a true output when the buffered voice data amount is equal to or smaller than the threshold value, and a step of providing a true output when the true output is provided from any one of the plurality of steps of judging. 
   It is further preferred that the at least one threshold value includes an initial buffering value required for audio playback at initial stage. 
   It is preferred that the trigger pulse has a duration corresponding to one frame in audio coding. 
   It is also preferred that the judging step includes a step of providing a true output of a “H” level when the buffered voice data amount is equal to or smaller than the at least one threshold value, and a false output of a “L” level when the buffered voice data amount is larger than the at least one threshold value, that the outputting step includes a step of providing a “H” level trigger pulse with a predetermined duration of time, and that the controlling step includes a step of calculating logical AND of the “H” level or “L” level output provided by the judging step and the “H” level trigger pulse, a calculated logical AND output being used in the on/off controlling step. 
   It is further preferred that the judging step includes a step of providing a true output of a “H” level when the buffered voice data amount is equal to or smaller than the at least one threshold value, and a false output of a “L” level when the buffered voice data amount is larger than the at least one threshold value, that the outputting step includes a step of providing a “H” level trigger pulse with a predetermined duration of time, that the controlling step includes a step of calculating logical AND of the “H” level or “L” level output provided by the judging step and the “H” level trigger pulse, and that the method further includes a step of monitoring the buffered voice data amount, a step of providing an initial buffering judged “H” level output when the buffered voice data amount is equal to or smaller than an initial buffering value, and a step of calculating logical OR of the calculated logical AND output and the initial buffering judged “H” level output, a calculated logical OR output being used in the on/off controlling step. 
   According to the present invention, extremely short breaks or interruptions of voice that are not audible as interruptions are intentionally inserted so as to keep the buffered data amount near the initial buffering value, which avoids falling into the underflow state. Also, because the interruption period of voice is extremely short, no disruption in conversation will occur. During the interruption period of voice, no voice data is extracted from the buffer and therefore it is expected that the buffered data amount increase over the threshold value. Therefore, a frequency of re-buffering at the underflow state of the buffer can be extremely reduced and thus quality of conversation will be maintained even if there are packet delay variations on the network. 
   Further objects and advantages of the present invention will be apparent from the following description of the preferred embodiments of the invention as illustrated in the accompanying drawings. 

   
     BRIEF DESCRIPTION OF THE DRAWINGS 
       FIG. 1  already described is a block diagram illustrating a functional configuration of a preceded digital audio playback apparatus; 
       FIG. 2  is a block diagram illustrating a functional configuration of a digital audio playback apparatus in a preferred embodiment according to the present invention; and 
       FIG. 3  is a block diagram illustrating a functional configuration of a digital audio playback apparatus in another embodiment according to the present invention. 
   

   DESCRIPTION OF THE PREFERRED EMBODIMENTS 
     FIG. 2  illustrates a functional configuration of a digital audio playback apparatus in a preferred embodiment according to the present invention. The functional configuration of the audio playback apparatus may be implemented by a digital computer with a program. 
   The digital audio playback apparatus  1  shown in  FIG. 2  is configured to receive packets containing compressed audio data and play back audio in the packets. This apparatus  1  has an audio packet receiver section  11 , an audio decoder  12 , a buffer  13 , a switch  14 , a D/A converter section  15 , an amplifier  16 , a threshold judgment section  18 , a timer  19  and an AND gate  191  for the timer. 
   The audio packet receiver section  11  receives packets containing compressed audio data from the network, and transmits the compressed audio data to the audio decoder  12 . The audio decoder  12  decodes the compressed audio data into non-compressed PCM (pulse code modulation) data and outputs the PCM data to the buffer  13 . The buffer  13  temporarily stores the PCM data and outputs them to the D/A converter section  15  through the switch  14 . The D/A converter section  15  converts the PCM data input through the switch  14  into an analog signal and outputs the converted analog signal to the amplifier  16 . The analog audio signal output from the amplifier  16  is provided to the speaker  2  to play back the audio. 
   The threshold judgment section  18  is provided with a threshold storage  187  for storing a threshold value and a comparator  180 . The comparator  180  compares the buffered data amount in the buffer  13  with the threshold value and outputs a “H” level signal that represents “true” when the buffered data amount becomes equal to or lower than the threshold value for example, otherwise outputs a “L” level signal that represents “false”. The timer  19  repeatedly outputs 20 millisecond-pulsive signal that corresponds to one typical audio encoding frame when the input signal level is “H”, which is coming from the comparator  180  shown in  FIG. 2 . The AND gate  191  receives a judgment signal (“H” level signal/“L” level signal) from the threshold judged section  18  and the trigger pulse signal (“H” level signal/“L” level signal) from the timer  19 . This AND gate  191  outputs a “H” level signal only when both the input signals are “H” level and outputs a “L” level signal otherwise. The switch  14  turns off in response to the “H” level signal, and turns on in response to the “L” level signal. 
   Hereinafter, operation of this audio playback apparatus will be described. 
   Under initial conditions, no PCM data is stored in the buffer  13 . Therefore, the buffered data amount is lower than the threshold value and the comparator  180  outputs a “true” or “H” level signal. Thus, the timer  19  outputs the “H” level trigger pulse signal with duration of 20 milliseconds, and therefore the AND gate  191  outputs an “H” level signal for 20 milliseconds. As a result, the switch  14  turns off in response to the “H” level signal and keeps off state for a very short period of 20 milliseconds of the “H” level signal. During the off-state of the switch  14 , PCM data output from the audio decoder  12  are stored in the buffer  13  without being output to the D/A converter section  15 . In 20 milliseconds, the trigger pulse signal from the timer  19  turns to “L” level and thus output from the AND gate  191  changes into “L” level to turn on the switch  14  for 20 milliseconds. When the switch  14  turns on and keeps on-state, the PCM data stored in the buffer  13  is provided to the D/A converter section  15  and that the analog audio signal is provided to the speaker  2  through the amplifier  16  to play the audio back. After another 20 milliseconds, the pulse signal from the timer  19  turns to “H” level to turn off the switch  14  again. Thus, the switch  14  periodically turns on and off in every 20 milliseconds until the buffer level exceeds the threshold. 
   Thus, initially, break or interruption of voice may occur. However, because this interruption period of voice is very short as 20 milliseconds, it will be heard as a momentary noise and therefore no disruption in conversation will occur. During the interruption period of voice, no PCM data is extracted from the buffer  13  and therefore it is expected that the buffered data amount increase over the threshold value. 
   If the buffered data amount becomes equal to or higher than the threshold value, the comparator  180  outputs a “false” or “L” level signal. In this case, the AND gate  191  always outputs an “L” level signal irrespective of the output level of the timer  19 . As a result, the switch  14  keeps on state and therefore the PCM data stored in the buffer  13  is provided to the D/A converter section  15  so that the analog audio signal is provided to the speaker  2  through the amplifier  16  to play the audio back. 
   Thereafter, if the buffered data lowers than the threshold value due to delay in the arrival of packets from the network that causes decrease of the level of the buffer  13 , the comparator  180  outputs a “true” or “H” level signal. Thus, the timer  19  outputs the “H” level trigger pulse signal with duration of 20 milliseconds, and therefore the AND gate  191  outputs an “H” level signal for 20 milliseconds. As a result, the switch  14  turns off in response to the “H” level signal and keeps off state for a very short period of 20 milliseconds of the “H” level signal. During the off-state of the switch  14 , PCM data output from the audio decoder  12  are stored in the buffer  13  without being output to the D/A converter section  15 . In 20 milliseconds, the trigger pulse signal from the timer  19  turns to “L” level and thus output from the AND gate  191  changes into “L” level to turn on the switch  14 . When the switch  14  turns on and keeps on-state, the PCM data stored in the buffer  13  is provided to the D/A converter section  15  and the analog audio signal is provided to the speaker  2  through the amplifier  16  to resume playback of audio. 
   In this case, as similar to the initial state, break or interruption of voice may occur. However, because the interruption period of voice is extremely short as 20 milliseconds, it will not be audible as interruption but will be heard as a momentary noise, and therefore no disruption in conversation will occur. During the interruption period of voice, no PCM data is extracted from the buffer  13  and therefore it is expected that the buffered data amount increase over the threshold value. That is, once the buffered data amount lowers than the threshold value that will be larger than zero, turning off of the switch  14  occurs and the buffered data amount starts to increase. Therefore, a frequency of re-buffering at the underflow state of the buffer can be extremely reduced and thus quality of conversation will be maintained even if there are packet delay variations on the network. 
     FIG. 3  illustrates a functional configuration of a digital audio playback apparatus in another embodiment according to the present invention. The functional configuration of the audio playback apparatus may be implemented by a digital computer with a program. 
   The audio playback apparatus shown in  FIG. 3  differs from that shown in  FIG. 2  in the following points. In the apparatus of  FIG. 3 , an initial buffering judgment section  17  and an OR gate  190  for the switch are additionally provided. Also, in the apparatus of  FIG. 3 , the threshold judgment section  18  has a plurality of comparators. Thus, judgment of the underflow state can be implemented as the audio playback apparatus shown in  FIG. 1 . Furthermore, because a plurality of threshold values are stepwise provided down to the underflow state, the number of breaks or interruptions of voice can be controlled depending upon the buffered data amount. 
   As shown in  FIG. 3 , the threshold judgment section  18  has four comparators  180 - 183 , AND gates  184  and  185 , an OR gate  186 , first and second threshold storages  187  and  188 , and a delay circuit  189 . The delay circuit  189  can provide an old buffered data amount that represents the amount of the buffered data predetermined period earlier (buffered data amount at the past time). If it is designed that the duration of break or interruption of voice, that is, the duration of trigger pulse signal from the timer  19  is 20 milliseconds, the delay time may be determined about 30 milliseconds. The first and second threshold storages  187  and  188  store different first and second threshold values as threshold values of the buffered data amount in previous stage down to the underflow state. The first threshold value may be equal to or smaller than the initial buffering value, and the second threshold value may be equal to one-half the first threshold value or one-half the initial buffering value. 
   In case of one-way distribution of voice data in which a large delay in end-to-end communication is allowed, a large value of the initial buffering value will be selected so that change in delay in network communication can be absorbed. However, in case of communication for conversation such as VoIP or IP video telephony, required is a small delay and thus a large initial buffering value is not allowed in such communication so that the communication has to execute under insufficient buffered data amount. 
   In this embodiment, two threshold values are prepared and therefore two pairs of comparators  180  and  181 , and  1812  and  183  are provided in the threshold judgment section  18 . The number of pairs of comparators is not limited to two as this embodiment but optionally determined to any number corresponding to the number of the threshold values. 
   The comparator  180  judges whether the old buffered data amount or the buffered data amount at the past time is larger than the first threshold value, the comparator  181  judges whether the first threshold value is larger than the buffered data amount at the present time, the comparator  182  judges whether the buffered data amount at the past time is larger than the second threshold value, and the comparator  183  judges whether the second threshold value is larger than the buffered data amount at the present time. 
   The AND gate  184  outputs a “H” level signal only when both the comparators  180  and  181  output “true” or “H” level signals, and the AND gate  185  outputs a “H” level signal only when both the comparators  182  and  183  output “true” or “H” level signals. Thus, the AND gate  184  outputs the “H” level signal only when the buffered data amount lowers through the first threshold value during the predetermined period from the past time to the present time. Also, the AND gate  185  outputs the “H” level signal only when the buffered data amount lowers through the second threshold value during the predetermined period from the past time to the present time. 
   The OR gate  186  outputs an “H” level signal when the output from the AND gate  184  or  185  becomes “H” level. This “H” level signal from the OR gate  186  or the threshold judgment section  18  is provided to the timer  19  to trigger it and is directly applied to the AND gate  191  for timer. The timer  19  repeatedly outputs 20 millisecond-pulsive signal that corresponds to one typical audio encoding frame when the input signal level is “H”, which is coming from the comparator  180  shown in  FIG. 2 . The AND gate  191  receives the judgment signal from the threshold judged section  18  and the trigger pulse signal from the timer  19 . This AND gate  191  outputs a “H” level signal only when both the input signals are “H” level and outputs a “L” level signal otherwise to the OR gate  190  for switch. 
   The initial buffering judgment section  17  has a NOR gate  171 , a comparator  172 , an RS flip-flop  173  and an initial buffering value storage  174 . The NOR gate  171  outputs an “H” level signal to the flip-flop  173  when the buffered data amount becomes zero. The comparator  172  compares the buffered data amount in the buffer  13  with an initial buffering value and outputs a “H” level signal to the flip-flop  173  when the buffered data amount becomes equal to or higher than the initial buffering value stored in the storage  174 . Thus, the flip-flop  173  outputs an “H” level signal when the buffered data amount is under the underflow state and increases up to the initial buffering value. Whereas, the flip-flop  173  outputs a “L” level signal when the buffered data amount is equal to or larger than the initial buffering value and when the buffered data amount decreases down to zero but not equal to zero. 
   The OR gate  190  receives both the output from the threshold judgment section  18  and the output from the initial buffering judgment section  17 . Therefore, this OR gate  190  outputs a “H” level signal when the initial buffering judgment section  17  judges that the buffered data amount increases from zero but not reaches the initial buffering value yet. This OR gate  190  also outputs a “H” level signal for 20 milliseconds when the buffered data amount lowers through the first or second threshold value during the predetermined period from the past time to the present time. The switch  14  turns off in response to the “H” level signal to pause the output of the audio data from the buffer  13  to the D/A converter section  15 . 
   According to this embodiment, extremely short breaks or interruptions of voice that are not audible as interruptions are intentionally inserted so as to keep the buffered data amount near the initial buffering value, which avoids falling into the underflow state. Also, because the interruption period of voice is extremely short as 20 milliseconds, no trouble for conversation will occur. During the interruption period of voice, no PCM data is extracted from the buffer  13  and therefore it is expected that the buffered data amount increase over the threshold value. Therefore, a frequency of re-buffering at the underflow state of the buffer can be extremely reduced and thus quality of conversation will be maintained even if there are packet delay variations on the network. 
   Hereinafter, a concrete example of buffered data amount control will be described. 
   Suppose, for example, output data from the voice decoder  12  has parameters of 8000 Hz sampling frequency, 16 bit quantization, and a monophonic channel. In this case, if the initial delay is 100 milliseconds (0.1 seconds), the initial buffering value for starting playback becomes 1600 bytes from the equation of:
 
1600 bytes=8000 Hz×16 bits/8 bits×0.1 seconds.
 
   If the input rate to and the output rate from the buffer  13  are equal to each other, the buffered data amount will stay near the initial buffering value (1600 bytes). However, if delay in arrival of the voice data occurs, the buffered data amount will decrease. Contrary to this, if the delay is resolved and the delayed data are arrived at once, the buffered data amount will abruptly increase. 
   In case of the initial buffering value of 1600 bytes, two threshold values, namely a first threshold value of 1200 bytes and a second threshold value of 800 bytes may be used. In modifications, buffered data amount control with three threshold values, in which a third threshold value of 400 bytes for an additional pair of comparators is used may be executed. 
   The buffered data amount B(t) in normal state is expressed as:
 
 B ( t )= B in( t )− B out( t−t   0 )
 
where Bin(t) is a total amount of data input into the buffer until a time t, Bout(t) is a total amount of data output from the buffer for a time period t, which is represented as in case of voice data Bout(t)=8000 Hz×16 bits/8 bits×t, and t 0  is a time period of the initial buffering during which the playback of audio is paused.
 
   After the communication, when the buffered data amount kept near 1600 bytes decreases down to the first threshold value of 1200 bytes or less due to the delay in receive of packets, an extremely short break in voice for 20 milliseconds is intentionally produced. For this 20 milliseconds, extraction of the voice data from the buffer is paused and thus the buffered data amount increases, in other words decrease in the buffered data amount is prevented. The buffered data amount B(t) at this time is expressed as B(t)=Bin(t)−Bout(t−0.1 seconds−0.02 seconds). 
   When the buffered data amount still decreases down to the second threshold value of 800 bytes or less due to further delay in receive of packets, an extremely short break in voice for 20 milliseconds is further intentionally produced. For this 20 milliseconds, extraction of the voice data from the buffer is paused and thus the buffered data amount increases, in other words decrease in the buffered data amount is prevented. The buffered data amount B(t) at this time is expressed as B(t)=Bin(t)−Bout(t−0.1 seconds−0.02 seconds−0.02 seconds). 
   In the aforementioned embodiments, turning off/on the switch controls pause and resume of audio playback. However, according to the present invention, pause and resume of audio playback can be controlled by any method for temporarily preventing extraction of the data from the buffer, such as for example a method for stopping the conversion operation of the D/A converter section. 
   Many widely different embodiments of the present invention may be constructed without departing from the spirit and scope of the present invention. It should be understood that the present invention is not limited to the specific embodiments described in the specification, except as defined in the appended claims.