JP2005151104A - VoIP TERMINAL - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a VoIP terminal capable of preventing the interruption or missing of a conversation or a rapidly and considerably distorted conversation during a speech and attaining a sufficiently practical voice speech via an IP network. <P>SOLUTION: The VoIP terminal is connected to an IP network 6 to which a plurality of nodes are connected, includes an IP network interface section 7, makes communication of voice data 2 with an optional node, and includes: a reception buffer 3 for sequentially storing the voice data 2 transmitted from the IP network 2 via a prescribed data processing and transmitting the received voice data 2 to a voice reproduction section 5 for reproducing the voice data 2 as voice; a voice data detection section 8 for detecting the voice data 2 transmitted from the reception buffer 3; and a voice data copy instruction section 9 for copying the voice data 2 stored in the reception buffer 3 under the instruction of the voice data detection section 8. <P>COPYRIGHT: (C)2005,JPO&NCIPI

Description

  The present invention relates to a VoIP terminal that is connected to an IP network to which a plurality of nodes are connected, has an IP network interface unit, and communicates voice data with an arbitrary node.

  FIG. 22 is a block diagram showing a conventional IP network. The IP network in this specification refers to a network as shown in FIG. 22 typified by the Internet network, and IP (Internet Protocol) as the main body of the network technology is being renovated to a packet system. In this packet system, a group of information (bit string) is attached with a header in which the address of the transmission destination is written, and the packet exchange sees the address and sends it to an appropriate route. Each packet has its own address, so even if multiple communication packets are mixed on a single line, they can be sorted by destination on the receiving exchange and can be used effectively. .

  VoIP is Voice over IP. Conventionally, transmission and reception of images and document data has been mainly performed on the Internet. Recently, however, a system that puts voice on an IP network, such as an IP telephone, has appeared. An increase in traffic in the IP network causes delay and noise, and the sound quality deteriorates. Therefore, a fluctuation absorbing buffer is often provided on the receiving side. As shown in FIG. 1 to be described later, fluctuation absorption is performed by storing voice packets in a buffer 3 such as a bucket and continuously taking out data from a hole provided at the bottom thereof at a constant speed so that the sound at the time of reproduction is reduced. Interruption can be prevented. The buffer is controlled by software using the memory on the substrate. Also, audio playback requires real-time characteristics, but audio data arrives early or late (fluctuation of delay time) due to variable delay factors on the network, and if it is played as it is, it sounds like a rattling sound. I am interrupted. Therefore, once the audio data is stored in the buffer, even if the delay time fluctuates, the collected audio data can only be increased or decreased, and the extraction can be performed in a fixed time. The deeper the depth, the larger the delay variation can be absorbed. However, the delay for storing the buffer increases, and it is the same as when reporting overseas using satellites, etc. It becomes audible with a delay, the real-time property is impaired, and the user feels uncomfortable.

  A conventional VoIP terminal (information communication terminal) prevents sound interruption by a method in which a data amount control unit manages the amount of data stored in a transmission buffer while observing the amount of data transmitted to a network (for example, (See Patent Document 1) and (Non-Patent Document 1). There are the following methods for managing the amount of data.

  (1) When a larger amount of audio data than the amount assumed in the transmission buffer is accumulated, the data amount control unit is generated by operating the quantization step of the quantization unit (compresses data). Reduce the amount of data to be saved.

  (2) If the transmission buffer is still full, the voice synthesizing unit notifies that the call is stopped, and stops generating data when the transmission buffer is full.

(3) When the transmission buffer is close to empty, it is not necessary to reduce the amount of data stored in the transmission buffer, so the data is transmitted with fine quantization steps.
Japanese Patent Laid-Open No. 11-308373 Tsutsuhiro Matsuda, “How to build an integrated corporate data / voice network”, Nikkei Business Publications 2000

  The above (1) to (3) have the following problems.

  In the method of operating the quantization step of (1), for example, if data having different compression rates are mixed in the data, the transmission speed on the network is not constant, and it takes time to demodulate on the receiving side. When the transmission buffer of (2) becomes full, it is naturally impossible to make a call by stopping the data transmission from the transmission buffer. From the user's point of view, it is very inconvenient. The method of reducing the quantization step of (3), increasing the amount of data, and bringing the amount of transmission data from the transmission buffer close to normal is similar to (1), in which data with different compression rates is mixed in the data. The transmission speed on the IP network is not constant, and it takes time to demodulate on the receiving side. Since the delay factor such as the primary increase in the number of accesses to the Internet home page on the IP network is not taken into consideration, the conversation is interrupted during the call.

  This VoIP terminal is required to be able to prevent a conversation from being interrupted, dropped out, or suddenly significantly distorted during a call, and to perform a voice call through a sufficiently practical IP network. ing.

  In order to satisfy this requirement, the present invention can prevent a conversation from being interrupted, dropped, or suddenly significantly distorted during a call, and can perform a voice call through a sufficiently practical IP network. An object of the present invention is to provide a VoIP terminal that can be used.

  In order to solve the above problems, a VoIP terminal of the present invention is a VoIP terminal that is connected to an IP network to which a plurality of nodes are connected, has an IP network interface unit, and communicates voice data with an arbitrary node. A receiving side buffer for sequentially storing voice data transmitted from the IP network through predetermined data processing and sending the transmitted voice data as voice; and voice data sent from the receiving buffer. And a voice data duplication instruction unit that duplicates the voice data that is instructed by the voice data detection unit and accumulated in the reception buffer.

  As a result, it is possible to prevent the conversation from being interrupted, lost, or suddenly significantly distorted during a call, and a VoIP terminal capable of performing a voice call through a sufficiently practical IP network is obtained. .

The present invention is a VoIP terminal that is connected to an IP network to which a plurality of nodes are connected, has an IP network interface unit, and communicates voice data with an arbitrary node. The received audio data is sequentially stored, and the received audio data is transmitted to the audio reproduction unit that reproduces the transmitted audio data as audio, the audio data detection unit that detects the audio data sent from the reception buffer, and the audio A voice data detection unit for voice data transmitted from the reception buffer of voice data transmitted from the IP network by having a voice data duplication instruction unit for copying the voice data instructed by the data detection unit and stored in the reception buffer By detecting missing voice data at The audio data duplication unit can duplicate the previous audio data or the audio data two or three times before the missing audio data to interpolate the missing audio data, so that the audio data is sent to the audio reproduction unit without interruption. By doing so, it is possible to prevent interruption of the sound, and it is possible to prevent the conversation from being interrupted, lost, or suddenly severely distorted during the call, and a voice call through a sufficiently practical IP network. The advantageous effect that it can be performed is obtained.

  Furthermore, when the audio data detection unit is instructed by the audio data detection unit that the audio data sent from the reception buffer is missing, the audio data duplication instruction unit outputs the audio data immediately before the missing audio data or the missing audio data. Alternatively, by duplicating and interpolating the previous three audio data, the audio data can be sent to the audio playback unit without interruption, so that the conversation is interrupted or lost during a call, or is significantly distorted rapidly. Thus, an advantageous effect is obtained that a voice call can be made through a sufficiently practical IP network.

  Further, the voice data duplication instruction unit, when instructed by the voice data detection unit that the voice data sent from the reception buffer is missing, is the two of the voice data immediately after the missing voice data or the missing voice data. Or, by copying and interpolating the next three voice data, the voice data can be sent to the voice playback unit without interruption, so the conversation is interrupted or lost during a call, or it is distorted rapidly. Thus, an advantageous effect is obtained that a voice call can be made through a sufficiently practical IP network.

  Furthermore, when the audio data detection unit is instructed that the audio data sent from the reception buffer is missing, the audio data duplication instruction unit duplicates the audio data obtained by averaging the audio data before and after the missing audio data. By interpolating, the voice data can be sent to the voice playback unit without interruption, so that it is possible to prevent the conversation from being interrupted, lost, or suddenly significantly distorted during a call, There is an advantageous effect that a voice call can be made through a sufficiently practical IP network.

  Furthermore, when the voice data duplication instruction unit is instructed by the voice data detection unit that the voice data transmitted from the reception buffer is missing, the first half of the missing voice data is immediately before or missing from the missing voice data. Duplicate the audio data two or three times before the audio data, and for the second half of the missing audio data, duplicate the audio data immediately after the missing audio data or two or three after the missing audio data and interpolate By doing so, the audio data can be sent to the audio playback unit without interruption, so that it is possible to prevent the conversation from being interrupted, lost, or suddenly severely distorted during a call. Advantageous effect that a voice call through a typical IP network can be performed.

  Furthermore, a white noise control unit that generates white noise when instructed by the audio data detection unit is instructed by the audio data detection unit that audio data sent from the reception buffer is missing. In some cases, by interpolating by generating white noise, the audio data can be sent to the audio playback unit without interruption, so that the conversation is interrupted or lost during a call, or is significantly distorted rapidly. Therefore, an advantageous effect is obtained that a voice call can be made through a sufficiently practical IP network.

In addition, a peripheral noise measuring unit that measures and digitally converts ambient noise during a call when instructed by the voice data detecting unit and records it, the peripheral noise measuring unit lacks voice data transmitted from the reception buffer. When the voice data detection unit instructs to read out the recorded ambient noise and interpolate it, the voice data can be sent to the voice playback unit without interruption, so the conversation is interrupted during a call, It is possible to prevent missing or abruptly distorting, and it is possible to obtain an advantageous effect that a voice call can be performed through a sufficiently practical IP network.

  In addition, when the transmitted audio data is likely to overflow from the reception buffer, the audio data detection unit is instructed to adjust the amount of audio data in the reception buffer, and the reception buffer control unit By compressing the audio data so that it does not exceed the processing capacity of the assumed reception buffer, the audio data can be sent to the audio playback unit without interruption, so the conversation is interrupted or lost during the call, It is possible to prevent abrupt and significant distortion and to obtain an advantageous effect that a voice call can be made through a sufficiently practical IP network.

  Further, the reception buffer control unit, when instructed by the audio data detection unit that the transmitted audio data is likely to overflow in the reception buffer, displays the header information for its own address so that it can be recognized as the data addressed to itself. It is assigned and returned to the IP network, temporarily stored on the IP network, and the voice data detection unit determines whether it is for itself by referring to whether the voice data in the IP network interface unit is addressed to itself. When it is determined that the audio data in the reception buffer does not overflow, the audio data detection unit instructs the reception buffer control unit, removes the header information addressed to itself, and returns the data to the reception buffer again. It is possible to prevent missing due to overflowing, so that the conversation is interrupted or dropped during a call, It is possible to prevent the distorted significantly, is advantageous effect that it is possible to perform voice communication via a sufficiently practical IP network obtained.

  Furthermore, the reception buffer control unit extracts the transmitted audio data in the reception buffer at regular intervals when instructed by the audio data detection unit that the transmitted audio data is likely to overflow in the reception buffer, Since the audio data immediately before the extracted audio data or the audio data before or after the extracted audio data is duplicated and interpolated, the audio data can be sent to the audio reproduction unit without interruption. It is possible to prevent the conversation from being interrupted, lost, or suddenly severely distorted during a call, and an advantageous effect is obtained that a voice call can be made through a sufficiently practical IP network.

  Furthermore, the reception buffer control unit extracts the transmitted audio data in the reception buffer at regular intervals when instructed by the audio data detection unit that the transmitted audio data is likely to overflow in the reception buffer, When audio data is extracted at regular intervals to prevent overflow of the reception buffer by duplicating and interpolating the audio data immediately after the extracted audio data or the audio data after 2 or 3 of the extracted audio data In addition, since the audio data can be sent to the audio playback unit without interruption, it is possible to prevent the conversation from being interrupted, dropped, or suddenly significantly distorted during a call. An advantageous effect is obtained that a voice call can be made through a simple IP network.

  Furthermore, the reception buffer control unit extracts the transmitted audio data in the reception buffer at regular intervals when instructed by the audio data detection unit that the transmitted audio data is likely to overflow in the reception buffer, By copying and interpolating audio data averaged with audio data before and after the extracted audio data, even if audio data is extracted at regular intervals to prevent overflow of the reception buffer, the audio data is not interrupted Since it can be sent to the audio playback unit, it is possible to prevent the conversation from being interrupted, lost, or suddenly severely distorted during a call, and a voice call through a sufficiently practical IP network can be performed. The advantageous effect of being able to be obtained is obtained.

Furthermore, the reception buffer control unit extracts the transmitted audio data in the reception buffer at regular intervals when instructed by the audio data detection unit that the transmitted audio data is likely to overflow in the reception buffer, For the first half of the extracted audio data, duplicate the audio data just before the extracted audio data or two or three times before the extracted audio data, and for the second half of the extracted audio data, immediately after or without the extracted audio data By reproducing and interpolating the audio data after two or three of the audio data, the audio reproduction unit can be used without interruption even when the audio data is extracted at regular intervals to prevent the reception buffer from overflowing. So that the conversation can be prevented from being interrupted, dropped, or suddenly significantly distorted during a call. Advantageous effect that it is possible to perform voice communication via the use specific IP network can be obtained.

  In addition, a white noise control unit that generates white noise when instructed by the audio data detection unit, the reception buffer control unit indicates that the transmitted audio data is likely to overflow in the reception buffer. When it is instructed, the transmitted audio data in the reception buffer is extracted at regular intervals, and the extracted audio data is interpolated by generating white noise in the white noise control unit to prevent overflow of the reception buffer. Therefore, even if audio data is extracted at regular intervals, the audio data can be sent to the audio playback unit without interruption, so that the conversation is interrupted or lost during a call, or it is significantly distorted. And the advantageous effect of being able to make a voice call over a sufficiently practical IP network It is.

  In addition, it has an ambient noise measurement unit that measures ambient noise during a call when instructed by the audio data detection unit, and converts and records the noise, and the reception buffer control unit is likely to overflow the transmitted audio data in the reception buffer. When the voice data detection unit instructs that it has become, the transmitted voice data in the reception buffer is extracted at regular intervals, and for the extracted voice data, the ambient noise is read from the ambient noise measurement unit and interpolated. Therefore, even if audio data is extracted at regular intervals to prevent overflow of the reception buffer, the audio data can be sent to the audio playback unit without interruption, so that the conversation is interrupted or lost during a call. This is advantageous in that it can be prevented from being distorted sharply and can carry out a voice call through a sufficiently practical IP network. Results can be obtained.

  In addition, an audio data sending unit for controlling the sending amount of the audio data is provided at the exit of the receiving buffer, and the receiving buffer control unit is short of transmitted audio data or is likely to overflow the transmitted audio data. Is received from the audio data detection unit, the audio data transmission unit adjusts the audio data transmission timing to adjust the audio data in the reception buffer. The advantageous effect of being able to be adjusted and prevented at the audio data transmission timing in is obtained.

  Further, an absolute time management unit for calculating the absolute time is provided, and the reception buffer control unit is instructed by the audio data detection unit that the transmitted audio data is insufficient or the transmitted audio data is likely to overflow. In order to adjust the audio data in the reception buffer, the overflow in the reception buffer is controlled by controlling the transmission timing in the audio data transmission unit of the audio data transmitted from the absolute time of the absolute time management unit. An advantageous effect is obtained that it can be adjusted and prevented at the audio data transmission timing in the data transmission unit.

  The present invention prevents the conversation from being interrupted, dropped, or suddenly significantly distorted during a call, and has the purpose of performing a voice call over a sufficiently practical IP network at the exit of the reception buffer. The voice data detection unit detects the missing voice data, duplicates the voice data immediately before the missing voice data, or two or three voice data before the missing voice data, and interpolates the missing voice data. It was possible to prevent the sound from being interrupted by transmitting the sound without interruption.

  A first invention made to solve the above problem is a VoIP terminal that is connected to an IP network to which a plurality of nodes are connected, has an IP network interface unit, and communicates voice data with an arbitrary node. Then, the audio data transmitted from the IP network through the predetermined data processing is sequentially stored, and the reception side buffer for transmitting to the audio reproduction unit for reproducing the transmitted audio data as audio, and the audio transmitted from the reception buffer An audio data detection unit that detects data, and an audio data duplication instruction unit that replicates audio data that is instructed by the audio data detection unit and accumulated in the reception buffer, and that has been transmitted from the IP network For the audio data sent from the data reception buffer, the audio data detector For example, the voice data immediately before the missing voice data or the voice data two or three times before the missing voice data can be duplicated in the voice data duplication instruction section, and the missing voice data can be interpolated. By sending audio data to the audio playback unit without interruption, it is possible to prevent interruption of the sound, and it is possible to prevent the conversation from being interrupted, lost, or suddenly significantly distorted during the call, There is an effect that a voice call can be made through a sufficiently practical IP network.

  According to a second aspect of the present invention for solving the above-described problem, the audio data duplication instruction unit is configured such that when the audio data detection unit indicates that the audio data sent from the reception buffer is missing, the missing audio data The audio data immediately before or the audio data before or after the missing audio data is duplicated and interpolated, and the audio data can be sent to the audio reproduction unit without interruption. It is possible to prevent the conversation from being interrupted, lost, or suddenly severely distorted during a call, and has the effect of being able to make a voice call over a sufficiently practical IP network.

  According to a third aspect of the present invention for solving the above-described problems, the audio data duplication instruction unit is configured such that when the audio data detection unit indicates that the audio data sent from the reception buffer is missing, the missing audio data The audio data immediately after or the audio data after two or three of the missing audio data is copied and interpolated, and the audio data can be sent to the audio reproduction unit without interruption. It is possible to prevent the conversation from being interrupted, lost, or suddenly severely distorted during a call, and has the effect of being able to make a voice call over a sufficiently practical IP network.

  According to a fourth aspect of the present invention for solving the above-described problems, the audio data duplication instruction unit is configured such that when the audio data detection unit indicates that the audio data sent from the reception buffer is missing, the missing audio data The voice data that averaged the voice data before and after is copied and interpolated, and the voice data can be sent to the voice playback unit without interruption, so the conversation is interrupted or missing during the call And a sharp and significant distortion can be prevented, and a voice call via a sufficiently practical IP network can be performed.

  According to a fifth aspect of the present invention for solving the above problems, the audio data duplication instruction unit, when instructed by the audio data detection unit that the audio data sent from the reception buffer is missing, For the first half, the voice data two or three times before the missing voice data or the missing voice data is duplicated, and for the second half of the missing voice data, immediately after the missing voice data or the missing voice data 2 The second or third subsequent voice data is duplicated and interpolated, and the voice data can be sent to the voice playback unit without interruption, so the conversation is interrupted or lost during the call, It is possible to prevent sudden and significant distortion and to perform a voice call over a sufficiently practical IP network. It has an effect.

  A sixth invention made to solve the above-described problem includes a white noise control unit that generates white noise when instructed by the audio data detection unit, and the white noise control unit is configured to output audio transmitted from the reception buffer. When the voice data detection unit indicates that data has been lost, white noise is generated and interpolated, and the voice data can be sent to the voice playback unit without interruption. It is possible to prevent the conversation from being interrupted, dropped out, or suddenly significantly distorted, and it is possible to perform a voice call over a sufficiently practical IP network.

  A seventh invention made to solve the above-mentioned problem comprises an ambient noise measuring unit that measures ambient noise during a call when it is instructed by the voice data detecting unit, converts it into a digital signal, and records it. If the audio data detection unit indicates that the audio data sent from the reception buffer is missing, the recorded ambient noise is read and interpolated, and the audio data is reproduced without interruption. Since the conversation can be prevented from being interrupted, dropped out, or suddenly significantly distorted during a call, it is possible to perform a voice call via a sufficiently practical IP network. Has the effect of being able to.

  In an eighth aspect of the invention made to solve the above-described problem, a reception buffer control for adjusting an amount of audio data in a reception buffer instructed by an audio data detection unit when transmitted audio data is likely to overflow from the reception buffer. The reception buffer control unit compresses the transmitted audio data so as not to exceed the assumed processing capacity of the reception buffer, and sends the audio data to the audio reproduction unit without interruption. It is possible to prevent the conversation from being interrupted, lost, or suddenly severely distorted during a call, and to perform a voice call over a sufficiently practical IP network. Has an effect.

  According to a ninth aspect of the present invention for solving the above-mentioned problems, the reception buffer control unit, when instructed by the audio data detection unit that the transmitted audio data is likely to overflow in the reception buffer, The header information for the user is added to the IP network so that it can be recognized as the addressed data, returned to the IP network, temporarily stored on the IP network, and the voice data detection unit in the voice data in the IP network interface unit. If it is determined that it is intended for itself and the audio data in the reception buffer does not overflow, the audio data detection unit instructs the reception buffer control unit and removes the header information addressed to itself. The data is returned to the reception buffer again, and it is possible to prevent audio data from being lost due to overflow in the reception buffer. Therefore, it is possible to prevent the conversation from being interrupted, lost, or suddenly significantly distorted during a call, and to have a function and effect that a voice call can be made through a sufficiently practical IP network. .

  According to a tenth aspect of the present invention, there is provided a reception buffer control unit, wherein the reception buffer control unit receives the reception buffer when instructed by the audio data detection unit that the transmitted audio data is likely to overflow in the reception buffer. The extracted audio data is extracted at regular intervals, and the audio data immediately before the extracted audio data or the audio data two or three before the extracted audio data is duplicated and interpolated. Since the voice data can be sent to the voice playback unit without interruption, it is possible to prevent the conversation from being interrupted, lost, or suddenly severely distorted during a call, and a sufficiently practical IP network. It has the effect of being able to make a voice call via.

According to an eleventh aspect of the invention for solving the above-mentioned problem, the reception buffer control unit receives the reception buffer when the audio data detection unit instructs that the transmitted audio data is likely to overflow in the reception buffer. The transmitted voice data is extracted at regular intervals, and the voice data immediately after the extracted voice data or two or three voice data after the extracted voice data is duplicated and interpolated. Even when audio data is removed at regular intervals to prevent overflow of the reception buffer, the audio data can be sent to the audio playback unit without interruption, so that the conversation is interrupted or lost during a call. And the like can be prevented from being significantly distorted, and a voice call through a sufficiently practical IP network can be performed.

  According to a twelfth aspect of the present invention, in order to solve the above problem, the reception buffer control unit receives the reception buffer when the audio data detection unit instructs that the transmitted audio data is likely to overflow in the reception buffer. The extracted audio data is extracted at regular intervals, and the audio data averaged with the audio data before and after the extracted audio data is copied and interpolated. Even if the data is extracted at regular intervals, the audio data can be sent to the audio playback unit without interruption, preventing the conversation from being interrupted, dropped out, or suddenly severely distorted during a call. It is possible to perform a voice call through a sufficiently practical IP network.

  In a thirteenth aspect of the present invention for solving the above-mentioned problem, the reception buffer control unit receives the reception buffer when the audio data detection unit instructs that the transmitted audio data is likely to overflow in the reception buffer. The transmitted audio data is extracted at regular intervals, and for the first half of the extracted audio data, the audio data immediately before the extracted audio data or the audio data two or three before the extracted audio data is duplicated and the extracted audio data is extracted. In the second half, the audio data is copied and interpolated immediately after the extracted audio data or two or three audio data after the extracted audio data, and the audio data is kept constant to prevent overflow of the reception buffer. Even if it is pulled out at intervals, the voice data can be sent to the voice playback unit without interruption, so the conversation is interrupted or lost during a call, It is possible to prevent the distorted significantly, with an action and effect that it is possible to sufficiently voice calls over a practical IP network.

  A fourteenth invention made to solve the above problems comprises a white noise control unit that generates white noise when instructed by the audio data detection unit, and the reception buffer control unit receives the transmitted audio data. When the audio data detection unit indicates that the buffer is about to overflow, the transmitted audio data in the reception buffer is extracted at regular intervals, and white noise is generated by the white noise control unit for the extracted audio data. In order to prevent overflow of the reception buffer, even when audio data is extracted at regular intervals, the audio data can be sent to the audio playback unit without interruption. It is possible to prevent the conversation from being interrupted, lost or suddenly severely distorted. Performing the voice call that has the action and effect that it is.

  A fifteenth aspect of the invention made to solve the above-mentioned problems includes an ambient noise measuring unit that measures ambient noise during a call when it is instructed by the voice data detecting unit, converts it into a digital signal, and records it. When the voice data detection unit indicates that the transmitted voice data is likely to overflow in the reception buffer, the transmitted voice data in the reception buffer is extracted at regular intervals. The ambient noise is read from the ambient noise measurement unit and interpolated. Even if audio data is extracted at regular intervals to prevent overflow of the reception buffer, the audio data is sent to the audio playback unit without interruption. Therefore, it is possible to prevent the conversation from being interrupted, lost, or suddenly severely distorted during a call, and a sufficiently practical IP network. It has operation and effect that it is possible to perform voice communication via the click.

  A sixteenth aspect of the invention made to solve the above-mentioned problem is provided with an audio data sending unit for controlling the sending amount of the audio data at the exit of the receiving buffer, and the receiving buffer control unit checks whether the transmitted audio data is insufficient. Or, when the audio data detection unit instructs that the transmitted audio data is likely to overflow, the audio data transmission unit adjusts the audio data transmission timing to adjust the audio data in the reception buffer. Therefore, it is possible to prevent the overflow in the reception buffer by adjusting the audio data transmission timing in the audio data transmission unit.

  A seventeenth aspect of the invention made to solve the above-described problem is provided with an absolute time management unit for calculating an absolute time, and the reception buffer control unit has insufficient transmitted audio data or overflowed transmitted audio data. When instructed by the voice data detection unit to do so, in the voice data sending unit, the voice data transmitted from the absolute time of the absolute time management unit is sent to adjust the voice data in the reception buffer. The timing is controlled, and it has an operation and effect that it is possible to prevent overflow in the reception buffer by adjusting the audio data transmission timing in the audio data transmission unit.

(Embodiment 1)
FIG. 1 is a block diagram showing a receiving-side voice transmission apparatus as a VoIP terminal according to Embodiment 1 of the present invention.

  In FIG. 1, reference numeral 1 denotes a receiving-side audio transmission device that is a receiving side of an audio transmission device such as an IP telephone that transmits and receives packetized audio data and image data, and reference numeral 2 denotes an analog audio signal in order to reduce the amount of data. Audio data that has been sampled, digitally converted, packetized, quantized, and quantized audio data within a specified time, 3 is in the reception-side audio transmission device 1, and a reception buffer that accumulates a sufficient amount of audio data 2, Reference numeral 4 is in the reception buffer 3, an audio data transmission unit for transmitting the audio data 2 in the reception buffer 3, and 5 is a reverse packetization of the audio data 2 transmitted from the audio data transmission unit 4 to convert from digital to analog and audio A voice playback unit 6 for playing back, for example, a transmission side voice transmission device (not shown) such as PBX (private branch exchange) and a reception side voice transmission A router (not shown) including a communication device such as a hub, which is present in the middle of an IP network that connects the device 1 and transmits packetized voice data 2, and 7 is a receiving side voice transmission device 1, an IP network I / F (interface) unit 8 serving as a window for voice data 2 sent from the IP network 6 is used to detect the presence or absence of the voice data 2 sent from the voice data sending unit 4. An audio data detection unit that confirms header information (not shown) given to the audio data 2 and detects the data amount of the audio data 2 in the reception buffer 3, and 9 is an audio that detects that the audio data 2 is missing An audio data duplication instruction unit 10 that is instructed by the data detection unit 8 and instructs a recording unit 10 (to be described later) to duplicate the audio data 2 is in the reception buffer 3. Temporarily records the audio data 2 of a further recording unit for sending the voice data 2 missing in instruction speech data copy instructing unit 9 in the receiving buffer 3. The recording unit 10 temporarily records the audio data 2, and erases the audio data 2 when a reference value such as a certain time or a certain recording amount is exceeded. In FIG. 1, the recording unit 10 is provided in the reception buffer 3, but one or more recording units 10 may be provided outside the reception buffer 3.

  The operation of the receiving-side voice transmission apparatus as the VoIP terminal configured as described above will be described with reference to FIG. FIG. 2 is an explanatory diagram showing a case where voice interpolation is performed using voice data 112 before missing voice data (missing voice data) 102 not transmitted from the IP network 6. In FIG. 2, the receiving-side audio transmission device 1, audio data 2, reception buffer 3, audio data sending unit 4, audio reproducing unit 5, IP network interface unit 7, audio data detecting unit 8, audio data duplication instruction unit 9, recording Since the part 10 is the same as that of FIG. 1, it attaches | subjects the same code | symbol and abbreviate | omits description.

  After detecting that the audio data 2 is missing (that is, detecting the missing audio data 102), the audio data detection unit 8 instructs the audio data duplication instruction unit 9 to interpolate the missing audio data 102. In response to this instruction, the audio data duplication instruction unit 9 instructs the recording unit 10 to transmit the previously transmitted audio data 112 from the recording unit 10. The previous voice data 112 is preferably just before the missing voice data 102 or in the vicinity of two or three before the missing voice data 102 because there is a sense of incongruity when the sound is suddenly changed by the person listening on the receiver side. Basically, the missing voice data 102 is lost when the traffic (the degree of congestion on the network) fluctuates due to voice data sent from other communication devices in the IP network 6 and the traffic volume increases. Therefore, there is a missing voice packet 102 that cannot be received by the receiving side voice transmission apparatus 1 in a state where the voice packet is missing. The voice data detection unit 8 may be provided near the IP network I / F unit 7 or at the entrance of the reception buffer 3.

  As described above, according to the first embodiment, the audio data transmitted from the IP network 6 through the predetermined data processing is sequentially stored and transmitted to the audio reproducing unit that reproduces the transmitted audio data as audio. The reception side buffer 3, the audio data detection unit 8 that detects the audio data sent from the reception buffer 3, and the audio data 2 that is instructed by the audio data detection unit 8 and accumulated in the reception buffer 3 are duplicated (precisely recorded) The voice data duplication unit 9 (which sends the corresponding voice data from the unit 10) has the voice data detection unit 8 for the voice data 2 sent from the reception buffer 3 of the voice data transmitted from the IP network 6. Detecting missing voice data, for example, two of the voice data 2 immediately before the missing voice data 102 or the missing voice data 102 In other words, since the audio data duplication instruction unit 9 can duplicate the three previous audio data 2 and interpolate the missing audio data 102, the audio data 2 can be output to the audio reproduction unit 5 without interruption. The interruption can be prevented, the conversation can be prevented from being interrupted, dropped out, or suddenly significantly distorted during the call, and the voice call through the sufficiently practical IP network 6 can be performed. it can.

  When the audio data detection unit 8 instructs that the audio data sent from the reception buffer 3 is missing, the audio data duplication instruction unit 9 or the audio data immediately before the missing audio data or the missing audio data The second or third previous voice data is copied and interpolated, and the voice data can be sent to the voice playback unit 5 without interruption, so that the conversation is interrupted or missing during the call. Or a sharp and significant distortion can be prevented, and a voice call through the sufficiently practical IP network 6 can be performed.

(Embodiment 2)
The configuration of the receiving-side voice transmission apparatus as the VoIP terminal according to the second embodiment of the present invention is the configuration of FIG.

  The operation of the receiving-side audio transmission apparatus configured as described above will be described with reference to FIG. FIG. 3 is an explanatory diagram showing a case where voice interpolation is performed using the voice data 122 after the missing voice data 102. In FIG. 3, the receiving side audio transmission device 1, audio data 2, reception buffer 3, audio data sending unit 4, audio reproducing unit 5, IP network interface unit 7, audio data detecting unit 8, audio data duplication instruction unit 9, recording Since the part 10 is the same as that of FIG. 1, it attaches | subjects the same code | symbol and abbreviate | omits description.

As in the case of Embodiment 1, after detecting that the audio data 2 is missing, the audio data detection unit 8 instructs the audio data duplication instruction unit 9 to interpolate the missing audio data 102. Upon receiving this instruction, the audio data duplication instruction unit 9 instructs the recording unit 10 to transmit the audio data 122 scheduled to be transmitted later from the recording unit 10. As the later voice data 122, it is preferable that the voice data is immediately after the missing voice data 102 or in the vicinity of two or three after the missing voice data 102 because the person who is listening on the receiving side suddenly changes the sound.

  As described above, according to the second embodiment, when the audio data detection unit 8 instructs that the audio data output from the reception buffer 3 is lost, the audio data duplication instruction unit 9 Since the audio data 2 immediately after the data or the audio data 2 or 3 after the missing audio data 2 is duplicated and interpolated, the audio data 2 can be sent to the audio reproducing unit 5 without interruption. Thus, it is possible to prevent the conversation from being interrupted, dropped out, or suddenly significantly distorted during a call, and a voice call through the sufficiently practical IP network 6 can be performed.

(Embodiment 3)
The configuration of the receiving-side voice transmission apparatus as the VoIP terminal according to the third embodiment of the present invention is the configuration of FIG.

  The operation of the reception-side audio transmission apparatus configured as described above will be described with reference to FIG. FIG. 4 is an explanatory diagram showing, in an enlarged manner, the main parts when speech interpolation is performed using the speech data 112 and 122 before and after the missing speech data 102. In FIG. 4, the receiving side audio transmission device 1, the audio data 2, the reception buffer 3, the audio data transmission unit 4, the IP network interface unit 7, the audio data detection unit 8, the audio data duplication instruction unit 9, and the recording unit 10 are shown in FIG. Therefore, the same reference numerals are given and description thereof is omitted.

  As in the first and second embodiments, after detecting that the audio data 2 is missing, the audio data detection unit 8 instructs the audio data duplication instruction unit 9 to interpolate the missing audio data 102. In response to this instruction, the audio data duplication instruction unit 9 instructs the recording unit 10. Here, one of the voice data 112 immediately before the missing voice data 102 temporarily stored in the recording unit 10 or two or three previous voice data 112, the voice data 122 immediately after the missing voice data 102, or two or three thereof. One of the subsequent audio data 122 is arbitrarily selected in the recording unit 10, and average audio data 132 obtained by averaging the selected audio data is transmitted. Here, one of the voice data 112 immediately before the missing voice data 102 or the second and third previous voice data 112 and one of the voice data 122 immediately after the missing voice data 102 or the second and third subsequent voice data 122 are counted. Although the average of the two is selected, there is no problem if a plurality are selected and averaged so as to have the same size capacity (k bytes or the like) as the missing voice data 102. The use of the sound data in the vicinity of the missing sound data 102 is to prevent the person listening on the receiving side from feeling uncomfortable when the sound suddenly changes.

  As described above, according to the third embodiment, when the audio data detection unit 8 instructs that the audio data output from the reception buffer 3 is missing, the audio data duplication instruction unit 9 By duplicating and interpolating the voice data obtained by averaging the voice data before and after the data, the voice data can be sent to the voice reproduction unit 5 without interruption, so that the conversation is interrupted or lost during the call. Therefore, it is possible to prevent a sharp distortion, and a voice call through the sufficiently practical IP network 6 can be performed.

(Embodiment 4)
The configuration of the receiving side voice transmission apparatus as the VoIP terminal according to the fourth embodiment of the present invention is the same as that of the first embodiment as shown in FIG.

The operation of the receiving-side audio transmission apparatus configured as described above will be described with reference to FIG. FIG. 5 is an explanatory diagram showing, in an enlarged manner, a main part of the case where voice interpolation is performed using half of the voice data 112 and 122 before and after the missing voice data 102. In FIG. 5, the receiving side audio transmission device 1, audio data 2, reception buffer 3, audio data sending unit 4, IP network interface unit 7, audio data detection unit 8, audio data duplication instruction unit 9, and recording unit 10 are shown in FIG. Therefore, the same reference numerals are given and description thereof is omitted.

  Similarly to the first and second embodiments, after detecting that the voice data 2 is missing, the voice data detection unit 8 instructs the voice data duplication instruction unit 9 to interpolate the missing voice data 102. In response to this instruction, the audio data duplication instruction unit 9 instructs the recording unit 10. Here, one of the voice data 112 immediately before the missing voice data 102 temporarily stored in the recording unit 10 or two or three previous voice data 112, the voice data 122 immediately after the missing voice data 102, or two or three thereof. One of the following audio data 122 is arbitrarily selected in the recording unit 10, and the audio data obtained by halving the selected audio data, that is, the previous 1/2 audio data 212 and the subsequent 1/2 audio data 222, The average audio data 232 that is a combination of is sent out. Here, one of the voice data 112 immediately before the missing voice data 102 or two or three previous voice data 112, one of the voice data 122 immediately after the missing voice data 102, or two or three of the voice data 122 after that. However, there is no problem if it is distributed so that it has the same size capacity (k bytes or the like) as the missing voice data 102 without being limited to ½. For example, for the first half of the missing voice data, duplicate the voice data immediately before the missing voice data or two or three times before the missing voice data, and for the second half of the missing voice data, immediately after the missing voice data or The audio data two or three after the missing audio data may be duplicated. The use of the sound data in the vicinity of the missing sound data 102 is to prevent the person listening on the receiving side from feeling uncomfortable when the sound suddenly changes.

  As described above, according to the fourth embodiment, when the audio data detection unit 8 instructs that the audio data output from the reception buffer 3 is lost, the audio data duplication instruction unit 9 For the first half of the data, duplicate the voice data two or three times before the missing voice data or the missing voice data, and for the second half of the missing voice data, immediately after the missing voice data or the missing voice data The second or third subsequent voice data is copied and interpolated, and the voice data can be sent to the voice reproduction unit 5 without interruption, so that the conversation is interrupted or lost during the call. Or a sharp distortion can be prevented, and a voice call through the sufficiently practical IP network 6 can be performed.

(Embodiment 5)
The configuration of the receiving side voice transmission apparatus as a VoIP terminal according to the fifth embodiment of the present invention is basically the configuration of FIG.

  The operation of the reception-side audio transmission apparatus configured as described above will be described with reference to FIG. FIG. 6 is an explanatory diagram showing, in an enlarged manner, main parts when speech interpolation is performed using white noise. In FIG. 6, the receiving side audio transmission device 1, the receiving buffer 3, the audio data sending unit 4, the IP network interface unit 7, and the audio data detecting unit 8 are the same as those in FIG. To do. A white noise control unit 11 generates white noise 12 in accordance with an instruction from the audio data detection unit 8. That is, the configuration of the receiving side audio transmission apparatus according to the present embodiment is obtained by adding a white noise control unit 11 to the configuration of FIG.

After detecting that the audio data 2 is missing, the audio data detection unit 8 instructs the white noise control unit 11 to generate white noise 12 in order to interpolate the missing audio data 102. The white noise control unit 11 sends the white noise 12 to the missing voice data 102. Even when a normal analog telephone is used, the receiver side picks up a sound like background noise. The white noise 12 is a sound close to the background noise. By using a sound that is intentionally heard normally, the missing voice data 102 prevents the sound from being cut off during reproduction and is given to the person who is listening on the receiver side. It suppresses a sense of incongruity.

  As described above, according to the fifth embodiment, the white noise control unit 11 that generates the white noise 12 when instructed by the audio data detection unit 8 is provided, and the white noise control unit 11 sends out from the reception buffer 3. When the audio data detection unit 8 instructs that the audio data to be lost is generated, white noise 12 is generated from the white noise control unit 11 and interpolated to the audio reproduction unit 5 without interruption. Since it can be transmitted, it is possible to prevent the conversation from being interrupted, dropped out, or suddenly significantly distorted during a call, and a voice call via a sufficiently practical IP network 6 can be performed. .

(Embodiment 6)
The configuration of the receiving side voice transmission apparatus as a VoIP terminal according to the sixth embodiment of the present invention is basically the configuration of FIG.

  The operation of the receiving-side audio transmission apparatus configured as described above will be described with reference to FIGS. FIG. 7 is an explanatory diagram showing, in an enlarged manner, the main part of the case where speech interpolation is performed using white noise (background noise), and FIG. 8 is a waveform diagram of white noise. In FIG. 7, the receiving side audio transmission device 1, the receiving buffer 3, the audio data sending unit 4, the IP network interface unit 7, the audio data detecting unit 8, and the white noise control unit 11 are the same as those in FIG. The description is omitted. Reference numeral 13 denotes an ambient noise measuring unit that measures ambient noise. That is, the configuration of the receiving side audio transmission apparatus according to the sixth embodiment is obtained by adding a white noise control unit 11 and an ambient noise measurement unit 13 to the configuration of FIG.

  After detecting that the voice data 2 is missing, the voice data detection unit 8 instructs the ambient noise measurement unit 13 to interpolate the missing voice data 102. The ambient noise measurement unit 13 that mainly measures ambient noise (white noise 512) during a call measures the white noise 512, samples the frequently occurring analog waveform of the white noise 512, and performs digital conversion as shown in FIG. Then, the recorded data is sent to the white noise control unit 11 according to the instruction of the audio data detection unit 8, and the white noise 612 is used instead of the missing audio data 102. Ambient noise is a sound that is normally heard. By intentionally using such a sound that is usually heard, sound interruption during reproduction due to missing voice data 102 is prevented and given to a person who is listening on the receiver side. It suppresses a sense of incongruity.

  As described above, according to the sixth embodiment, the ambient noise measuring unit 13 is provided which measures the ambient noise during a call when instructed by the voice data detecting unit 8 and converts it into a digital signal and records it. When the audio data detection unit 8 instructs that the audio data sent from the reception buffer 3 is missing, the audio reproduction unit 13 reads out the recorded ambient noise and interpolates the audio data without interruption. Therefore, it is possible to prevent the conversation from being interrupted, dropped out, or suddenly severely distorted during a call, and a voice call through a sufficiently practical IP network can be performed. it can.

(Embodiment 7)
The configuration of the receiving-side voice transmission apparatus as a VoIP terminal according to the seventh embodiment of the present invention is basically the configuration of FIG.

The operation of the receiving-side audio transmission apparatus configured as described above will be described with reference to FIGS. FIG. 9 is an explanatory diagram showing a case where audio data is compressed and controlled by the reception buffer control unit so that the reception buffer does not overflow with audio data, and FIG. 10 is an enlarged view of the main part of FIG. 9 and 10, the receiving side audio transmission device 1, the audio data 2, the reception buffer 3, the audio data sending unit 4, the audio reproducing unit 5, the IP network interface unit 7, and the audio data detecting unit 8 are the same as those in FIG. Since these are the same, the same reference numerals are given and the description is omitted. Reference numeral 14 denotes an empty space in the reception buffer 3, and reference numeral 15 denotes a reception buffer control unit that compresses and controls audio data so that the reception buffer 3 does not overflow with audio data. That is, the configuration of the receiving side audio transmission apparatus according to the seventh embodiment is obtained by adding a reception buffer control unit 15 to the configuration of FIG.

  When the audio data 2 is likely to overflow from the reception buffer 3 beyond the capacity of the reception buffer 3, the reception buffer control unit 15 which is instructed by the audio data detection unit 8 and adjusts the capacity of the audio data 2 in the reception buffer 3, The A part of the audio data 2 is compressed into compressed audio data 142 so that the processing capacity of the reception buffer 3 assumed for the audio data 2 is not exceeded, and an empty space 14 is provided in the reception buffer 3 so that the audio data 2 Smoothing and recording are performed, and the compressed audio data 142 is sent from the reception buffer 3 to the audio reproducing unit 5. The audio reproducing unit 5 restores the compressed A part, and then reverse-packets it to convert it from digital to analog to obtain audio. Prevents sound interruption during playback. The detection of whether or not it is likely to overflow will be described. When the audio data detection unit 8 detects the audio data in the reception buffer 3 and the audio data 2 is likely to overflow from the reception buffer 3 beyond the capacity of the reception buffer 3, the audio data is detected. Instructed by the data detection unit 8.

  As described above, according to the seventh embodiment, when the transmitted audio data is likely to overflow from the reception buffer 3, the reception buffer is instructed by the audio data detection unit 8 to adjust the amount of audio data in the reception buffer 3. The reception buffer control unit 15 includes a control unit 15 and sends the audio data to the audio reproduction unit 5 without interruption by compressing the transmitted audio data so as not to exceed the processing capacity of the assumed reception buffer. Therefore, it is possible to prevent the conversation from being interrupted, dropped out, or suddenly significantly distorted during the call, and a voice call through the sufficiently practical IP network 6 can be performed.

(Embodiment 8)
FIG. 11 is a block diagram showing a receiving-side voice transmission apparatus as a VoIP terminal according to the eighth embodiment of the present invention.

  11, the receiving side audio transmission device 1, audio data 2, reception buffer 3, audio reproduction unit 5, IP network 6, IP network interface unit 7, audio data detection unit 8, and reception buffer control unit 15 are shown in FIG. 9 are the same as those in FIG. Reference numeral 16 denotes an IP network storage unit that temporarily stores voice data.

  The operation of the reception-side audio transmission apparatus configured as described above will be described with reference to FIG. FIG. 12 is a data diagram showing the structure of audio data.

  When the audio data 2 is likely to overflow from the reception buffer 3 beyond the capacity of the reception buffer 3, the reception buffer control unit 15 which is instructed by the audio data detection unit 8 and adjusts the capacity of the audio data 2 in the reception buffer 3, As shown in FIG. 12, the header information 62 addressed to itself is added to the header information 52 describing the destination of the reception-side audio transmission apparatus 1 as the original transmission destination in the audio data 2 so that it can be understood as the data addressed to itself. Then, the IP network 6 is returned to the IP network 6 and temporarily stored in the IP network storage unit 16 on the IP network 6 constructed by the company that temporarily provides the IP network service. The audio data 2 in the / F unit 7 refers to the header information addressed to itself, determines that it is addressed to itself, and When it is determined that the voice data 2 does not overflow, the voice data detection unit 8 instructs the reception buffer control unit 15 to remove the header information 62 addressed to itself and return to the reception buffer 3 again to prevent sound interruption during reproduction. .

As described above, according to the eighth embodiment, the reception buffer control unit 15 is informed when the audio data detection unit 8 indicates that the transmitted audio data is likely to overflow in the reception buffer. In the voice data 2 in which the voice data detection unit 8 is provided in the IP network interface unit 7, the header information for the user is attached to the IP network so that it can be recognized as the destination data, and is returned to the IP network and temporarily stored on the IP network. If it is determined that the header information is addressed to itself and that it is addressed to itself and that the audio data 2 in the reception buffer 3 does not overflow, the audio data detection unit 8 instructs the reception buffer control unit 15; By removing the header information 62 addressed to itself and returning it to the reception buffer 3 again, it is possible to prevent audio data from being lost due to overflow in the reception buffer 3. Since it is, or interrupted conversation during a call, missing or, it is possible to prevent the distorted abruptly significantly, it is possible to sufficiently voice calls over a practical IP network 6.

(Embodiment 9)
FIG. 13 is a block diagram showing a receiving-side voice transmission apparatus as a VoIP terminal according to Embodiment 9 of the present invention, and FIG. 14 is an enlarged view of a main part of FIG.

  13 and 14, the receiving side audio transmission device 1, audio data 2, reception buffer 3, audio reproduction unit 5, IP network 6, IP network interface unit 7, audio data detection unit 8, audio data duplication instruction unit 9, Since the recording unit 10 and the reception buffer control unit 15 are the same as those shown in FIGS. 1 and 9, the same reference numerals are given and description thereof is omitted.

  The operation of the reception-side audio transmission apparatus configured as described above will be described.

  When the audio data 2 is likely to overflow from the reception buffer 3 beyond the capacity of the reception buffer 3, the reception buffer control unit 15 which is instructed by the audio data detection unit 8 and adjusts the capacity of the audio data 2 in the reception buffer 3, The audio data 2 in the reception buffer 3 is extracted at regular intervals. The extracted audio data becomes thinned audio data 152. If this is the case, the sound is cut off during reproduction, and therefore it is necessary to interpolate the thinned audio data 152. Therefore, as in the first embodiment, in order to interpolate the thinned audio data 152, the audio data detection unit 8 instructs the audio data duplication instruction unit 9, and the audio data duplication instruction unit 9 further instructs the recording unit 10. The recorded audio data 162 is sent from the recording unit 10. As the previous audio data 162, since the user who is listening on the receiving side suddenly changes the sound, the audio data immediately before the thinned audio data 152 or the audio data in the vicinity up to a few times before is used. . Alternatively, the previous audio data 162 may be transmitted by directly instructing the recording unit 10 using the reception buffer control unit 15.

  As described above, according to the ninth embodiment, the reception buffer control unit 15 receives when the audio data detection unit 8 instructs that the transmitted audio data is likely to overflow in the reception buffer 3. The audio data transmitted in the buffer 3 is extracted at regular intervals, and the audio data immediately before the extracted audio data or the audio data two or three before the extracted audio data is duplicated and interpolated. Can be transmitted to the audio reproduction unit 5 without interruption, so that it is possible to prevent the conversation from being interrupted, lost, or suddenly significantly distorted during a call, and a sufficiently practical IP network 6 can be prevented. Voice calls can be made via

(Embodiment 10)
FIG. 15 is a block diagram showing a receiving side voice transmission apparatus as a VoIP terminal according to the tenth embodiment of the present invention. The receiving buffer control unit 15 prevents the receiving buffer 3 shown in FIG. 9 from overflowing with voice data. The case of control is shown by.

In FIG. 15, the audio data 2, the reception buffer 3, the audio reproduction unit 5, the audio data detection unit 8, the audio data duplication instruction unit 9, the recording unit 10, and the reception buffer control unit 15 are the same as those in FIGS. The same reference numerals are given and the description is omitted.

  The operation of the reception-side audio transmission apparatus configured as described above will be described.

  When the audio data 2 is likely to overflow from the reception buffer 3 beyond the capacity of the reception buffer 3, the reception buffer control unit 15 which is instructed by the audio data detection unit 8 and adjusts the capacity of the audio data 2 in the reception buffer 3, The audio data 2 in the reception buffer 3 is extracted at regular intervals. The extracted audio data becomes thinned audio data 152. If this is the case, the sound is cut off during reproduction, and therefore it is necessary to interpolate the thinned audio data 152. Therefore, as in the second embodiment, the audio data detection unit 8 instructs the audio data duplication instruction unit 9 and the audio data duplication instruction unit 9 instructs the recording unit 10 to interpolate the thinned audio data 152. Then, the audio data 172 sent later is sent from the recording unit 10. As the later voice data 172, since the person who is listening on the receiving side has a strange feeling when the sound suddenly changes, the voice data 172 is used immediately after the thinned voice data 152 or in the vicinity of the second to third. Alternatively, the recording unit 10 may be directly instructed using the reception buffer control unit 15 and the subsequent audio data 172 may be transmitted.

  As described above, according to the tenth embodiment, the reception buffer control unit 15 receives the reception buffer when the audio data detection unit 8 indicates that the transmitted audio data is likely to overflow in the reception buffer. 3 is extracted at regular intervals and the reception buffer 3 is copied and interpolated by duplicating and interpolating the audio data immediately after the extracted audio data or the audio data two or three after the extracted audio data. Even when audio data is removed at regular intervals to prevent overflow, the audio data can be sent to the audio playback unit 5 without interruption, so that the conversation is interrupted or lost during a call, or suddenly It can be prevented from being distorted significantly, and a voice call through the sufficiently practical IP network 6 can be performed.

(Embodiment 11)
FIG. 16 is a block diagram showing a receiving side voice transmission apparatus as a VoIP terminal according to the eleventh embodiment of the present invention. The receiving buffer control unit 15 prevents the receiving buffer 3 shown in FIG. 9 from overflowing with voice data. The case of control is shown by.

  In FIG. 16, the receiving side audio transmission device 1, the audio data 2, the reception buffer 3, the audio data sending unit 4, the audio reproducing unit 5, the IP network interface unit 7, the audio data detecting unit 8, the audio data duplicating instruction unit 9, and the recording Since the unit 10 and the reception buffer control unit 15 are the same as those shown in FIGS. 1 and 9, the same reference numerals are given and the description thereof is omitted.

  The operation of the reception-side audio transmission apparatus configured as described above will be described.

When the audio data 2 is likely to overflow from the reception buffer 3 beyond the capacity of the reception buffer 3, the reception buffer control unit 15 which is instructed by the audio data detection unit 8 and adjusts the capacity of the audio data 2 in the reception buffer 3, The audio data 2 in the reception buffer 3 is extracted at regular intervals. The extracted audio data becomes thinned audio data 152. If this is the case, the sound is cut off during reproduction, and therefore it is necessary to interpolate the thinned audio data 152. Therefore, as in the third embodiment, the audio data duplication instruction unit 9 is instructed to interpolate the thinned audio data 152, and the audio data duplication instruction unit 9 further instructs the recording unit 10. Here, one from the audio data 162 immediately before or two or three times before the thinned audio data 152 temporarily stored in the recording unit 10, one from the audio data 172 immediately after or two or three after the thinned audio data 152. Are arbitrarily selected in the recording unit 10 and averaged average audio data 242 is transmitted. In this example, one of the audio data 162 immediately before or after the thinned audio data 152, one from the audio data 162 before the audio thinned out, and one after the audio data 172 after the audio thinned out 152, or the audio data 172 after the average, are averaged. There is no problem if a plurality of them are selected and averaged so as to have the same size capacity (k bytes or the like) as the thinned audio data 152. The use of the sound data in the vicinity of the thinned sound data 152 is to prevent the person listening on the receiver side from feeling uncomfortable when the sound suddenly changes.

  As described above, according to the eleventh embodiment, the reception buffer control unit 15 receives data when the audio data detection unit 8 instructs that the transmitted audio data is likely to overflow in the reception buffer 3. The audio data transmitted in the buffer 3 is extracted at regular intervals, and the audio data averaged with the audio data before and after the extracted audio data is duplicated and interpolated to prevent the reception buffer 3 from overflowing. Since the voice data can be sent to the voice reproduction unit 5 without interruption even when the is removed at regular intervals, it is possible to prevent the conversation from being interrupted, lost, or suddenly severely distorted during a call. Therefore, it is possible to make a voice call through the sufficiently practical IP network 6.

(Embodiment 12)
FIG. 17 is a block diagram showing a receiving side voice transmission apparatus as a VoIP terminal according to the twelfth embodiment of the present invention. The receiving buffer control unit 15 prevents the receiving buffer 3 shown in FIG. 9 from overflowing with voice data. The case of control is shown by.

  In FIG. 17, the receiving side voice transmission device 1, voice data 2, reception buffer 3, voice data sending unit 4, IP network interface unit 7, voice data detection unit 8, voice data duplication instruction unit 9, recording unit 10, reception buffer Since the control part 15 is the same as that of FIG. 1, FIG. 9, the same code | symbol is attached | subjected and description is abbreviate | omitted.

  The operation of the reception-side audio transmission apparatus configured as described above will be described.

  When the audio data 2 is likely to overflow from the reception buffer 3 beyond the capacity of the reception buffer 3, the reception buffer control unit 15 which is instructed by the audio data detection unit 8 and adjusts the capacity of the audio data 2 in the reception buffer 3, The audio data 2 in the reception buffer 3 is extracted at regular intervals. The extracted audio data becomes thinned audio data 152. If this is the case, the sound is cut off during reproduction, and therefore it is necessary to interpolate the thinned audio data 152. Therefore, as in the fourth embodiment, the audio data duplication instruction unit 9 is instructed, and the audio data duplication instruction unit 9 further instructs the recording unit 10. Here, one from the audio data 162 immediately before or two or three times before the thinned audio data 152 temporarily stored in the recording unit 10, one from the audio data 172 immediately after or two or three after the thinned audio data 152. Is arbitrarily selected in the recording unit 10, and the audio data obtained by halving the selected audio data, that is, the average audio data 272 obtained by combining the previous 1/2 audio data 252 and the subsequent 1/2 audio data 262 is obtained. Send it out. Here, one is selected from the audio data 162 immediately before or after the thinned audio data 152, and one from the audio data 172 immediately after or after the audio data 152 after the thinned audio data 152. Although two are combined, it is not limited to ½, and there is no problem if it is distributed so as to have the same size capacity (k bytes or the like) as the thinned audio data 152. For example, in the first half of the extracted audio data 152, the audio data immediately before the extracted audio data or two or three audio data before the extracted audio data is duplicated, and the extracted audio data is extracted in the second half of the extracted audio data 152. The audio data two or three after the extracted audio data is copied and interpolated. The use of the sound data in the vicinity of the thinned sound data 152 is to prevent the person listening on the receiver side from feeling uncomfortable when the sound suddenly changes.

As described above, according to the twelfth embodiment, the reception buffer control unit 15 receives data when the audio data detection unit 8 instructs that the transmitted audio data is likely to overflow in the reception buffer 3. The transmitted audio data in the buffer 3 is extracted at regular intervals. For the first half of the extracted audio data, the audio data immediately before the extracted audio data or the audio data two or three before the extracted audio data is duplicated and extracted. In the second half of the audio data, the audio data is spaced at regular intervals to prevent overflow of the reception buffer 3 by duplicating and interpolating the audio data immediately after the extracted audio data or two or three after the extracted audio data. Even if it is pulled out, voice data can be sent to the voice playback unit 5 without interruption, so that the conversation is interrupted or lost during a call, It is possible to prevent the distorted, it is possible to perform a voice call sufficiently through practical IP network 6.

(Embodiment 13)
FIG. 18 is a block diagram showing a receiving side voice transmission apparatus as a VoIP terminal according to the thirteenth embodiment of the present invention. The receiving buffer control unit 15 prevents the receiving buffer 3 shown in FIG. 9 from overflowing with voice data. The case of control is shown by.

  In FIG. 18, the receiving side audio transmission device 1, audio data 2, reception buffer 3, audio data sending unit 4, IP network interface unit 7, audio data detection unit 8, recording unit 10, white noise control unit 11, reception buffer control Since the part 15 is the same as that of FIG.1, FIG.6, FIG.9, the same code | symbol is attached | subjected and description is abbreviate | omitted.

  The operation of the reception-side audio transmission apparatus configured as described above will be described.

  When the audio data 2 is likely to overflow from the reception buffer 3 beyond the capacity of the reception buffer 3, the reception buffer control unit 15 which is instructed by the audio data detection unit 8 and adjusts the capacity of the audio data 2 in the reception buffer 3, The audio data 2 in the reception buffer 3 is extracted at regular intervals. The extracted audio data becomes thinned audio data 152. In this state, the sound is cut off during reproduction, so it is necessary to interpolate the thinned audio data 152. Therefore, as in the fifth embodiment, the audio data detection unit 8 instructs the white noise control unit 11 to generate white noise 12. Further, the white noise control unit 11 sends the white noise 512 to the thinned audio data 152. Even when a normal analog telephone is used, the receiver side picks up a sound like background noise. This background noise is a sound usually heard. By using a sound that is intentionally heard normally, the skipped sound data 152 prevents sound interruption during reproduction and suppresses the uncomfortable feeling given to the person listening on the receiver side.

  As described above, according to the thirteenth embodiment, the white noise control unit 11 that generates white noise when instructed by the audio data detection unit 8 is provided, and the reception buffer control unit 15 receives the transmitted audio data. When the audio data detection unit 8 instructs that the reception buffer is about to overflow, the transmitted audio data in the reception buffer 3 is extracted at regular intervals, and the extracted audio data is processed by the white noise control unit 11. By interpolating by generating white noise, the audio data can be sent to the audio reproduction unit 5 without interruption even when the audio data is extracted at a constant interval in order to prevent the reception buffer 3 from overflowing. A sufficiently practical IP network that can prevent conversations from being interrupted, lost, or suddenly severely distorted during a call. It is possible to perform the voice communication via the.

(Embodiment 14)
FIG. 19 is a block diagram showing a receiving-side voice transmission apparatus as a VoIP terminal according to the fourteenth embodiment of the present invention. The receiving buffer control unit 15 prevents the receiving buffer 3 shown in FIG. 9 from overflowing with voice data. The case of control is shown by.

  In FIG. 19, the receiving side audio transmission device 1, audio data 2, reception buffer 3, audio data sending unit 4, IP network interface unit 7, audio data detection unit 8, recording unit 10, white noise control unit 11, ambient noise measurement Since the unit 13 and the reception buffer control unit 15 are the same as those shown in FIGS. 1, 6, 7, and 9, the same reference numerals are given and the description thereof is omitted.

  The operation of the reception-side audio transmission apparatus configured as described above will be described.

  When the audio data 2 is likely to overflow from the reception buffer 3 beyond the capacity of the reception buffer 3, the reception buffer control unit 15 which is instructed by the audio data detection unit 8 and adjusts the capacity of the audio data 2 in the reception buffer 3, The audio data 2 in the reception buffer 3 is extracted at regular intervals. The extracted audio data becomes thinned audio data 152. If this is the case, the sound is cut off during reproduction, and therefore it is necessary to interpolate the thinned audio data 152. Therefore, as in the sixth embodiment, the voice data detection unit 8 is a peripheral noise measurement unit 13 such as a microphone that mainly measures the peripheral noise (white noise 512) during a call in order to interpolate the thinned voice data 152. White noise 512 is measured. As shown in FIG. 8, the ambient noise measurement unit 13 samples an analog waveform that frequently appears as white noise 512, and sends the recorded signal after digital conversion to the white noise control unit 11. Is used instead of the thinned audio data 152. White noise is a sound that is normally heard, and by intentionally using a sound that is normally heard, sound interruption during reproduction by the thinned-out sound data 152 is prevented, and a sense of incongruity given to a person listening on the receiver side is suppressed. .

  As described above, according to the fourteenth embodiment, the ambient noise measuring unit 13 that measures ambient noise during a call when it is instructed by the voice data detecting unit 8, digitally converts and records the received noise, is provided. 15, when the voice data detection unit 8 indicates that the transmitted voice data is likely to overflow in the reception buffer 3, the transmitted voice data in the reception buffer 3 is extracted and removed at regular intervals. For audio data, the ambient noise is read from the ambient noise measurement unit 13 and interpolated, so that the audio reproduction unit can be used without interruption even when the audio data is extracted at regular intervals to prevent the reception buffer 3 from overflowing. Therefore, it is possible to prevent the conversation from being interrupted, lost, or suddenly severely distorted during a call. It is possible to perform voice calls through the workpiece 6.

(Embodiment 15)
20 is a block diagram showing a receiving-side voice transmission apparatus as a VoIP terminal according to a fifteenth embodiment of the present invention, in which the receiving buffer 3 shown in FIG. 1 or FIG. The case where it controls by the part 15 is shown.

  In FIG. 20, the receiving side audio transmission device 1, audio data 2, reception buffer 3, audio data sending unit 4, audio reproduction unit 5, IP network 6, IP network interface unit 7, audio data detection unit 8, reception buffer control unit 15 is the same as FIG. 1 and FIG.

  The operation of the reception-side audio transmission apparatus configured as described above will be described.

  If the audio data 2 is missing and insufficient, or if the audio data 2 is likely to overflow from the reception buffer 3 beyond the capacity of the reception buffer 3, it is received by the reception buffer control unit 15 instructed by the audio data detection unit 8. In order to adjust the audio data 2 in the buffer 3, the audio data sending unit 4 at the exit of the receiving buffer 3 adjusts the sending amount by adjusting the sending timing of the audio data 2, thereby preventing sound interruption during reproduction. This suppresses the uncomfortable feeling given to the person listening on the receiver side.

As described above, according to the fifteenth embodiment, the audio data transmission unit 4 that controls the transmission amount of audio data is provided at the exit of the reception buffer 3, and the reception buffer control unit 15 has insufficient transmitted audio data. If the audio data detection unit 8 instructs that the transmitted audio data is likely to overflow, the audio data sending unit 4 adjusts the audio data in the reception buffer 3 to adjust the audio data. By adjusting the transmission timing, overflow in the reception buffer 3 can be adjusted and prevented at the audio data transmission timing in the audio data transmission unit 4.

(Embodiment 16)
FIG. 21 is a block diagram showing a receiving-side voice transmission apparatus as a VoIP terminal according to the sixteenth embodiment of the present invention. The receiving buffer 3 shown in FIG. 1 or FIG. The case where it controls by the part 15 is shown.

  In FIG. 21, the receiving side audio transmission device 1, audio data 2, reception buffer 3, audio data sending unit 4, audio reproduction unit 5, IP network 6, IP network interface unit 7, audio data detection unit 8, reception buffer control unit 15 is the same as FIG. 1 and FIG.

  The operation of the reception-side audio transmission apparatus configured as described above will be described.

  If the audio data 2 is missing and insufficient, or if the audio data 2 is likely to overflow from the reception buffer 3 beyond the capacity of the reception buffer 3, it is received by the reception buffer control unit 15 instructed by the audio data detection unit 8. In order to adjust the audio data 2 in the buffer 3, the audio data sending unit 4 at the exit of the receiving buffer 3 uses the absolute time calculated by the absolute time management unit 17 such as a radio clock or GPS in the receiving buffer 3. By controlling the transmission timing of the audio data 2, sound interruption during playback is prevented, and the uncomfortable feeling given to the person listening on the receiver side is suppressed.

  As described above, according to the sixteenth embodiment, the absolute time management unit 17 that calculates the absolute time is provided, and the reception buffer control unit 15 has insufficient transmitted audio data or overflowed transmitted audio data. When the audio data detection unit 8 instructs that this is the case, the audio data of the transmitted audio data is obtained from the absolute time of the absolute time management unit 17 in order to adjust the audio data in the reception buffer 3. By controlling the transmission timing in the transmission unit 4, overflow in the reception buffer 3 can be adjusted and prevented by the audio data transmission timing in the audio data transmission unit 4.

  The present invention relates to a VoIP terminal that is connected to an IP network to which a plurality of nodes are connected, has an IP network interface unit, and communicates voice data with an arbitrary node, and conversations are interrupted or lost during a call. Therefore, it is possible to prevent a sharp distortion and a voice call through a sufficiently practical IP network.

1 is a block diagram showing a receiving side voice transmission apparatus as a VoIP terminal according to Embodiment 1 of the present invention; Explanatory drawing which shows the case where voice interpolation is performed using voice data before missing voice data (missing voice data) that has not been transmitted from the IP network Explanatory drawing showing the case of voice interpolation using voice data after missing voice data Explanatory drawing showing an enlarged main part when voice interpolation is performed using voice data before and after missing voice data Explanatory drawing showing an enlarged main part when voice interpolation is performed using half of the audio data before and after the missing voice data Explanatory drawing showing the main part enlarged when audio interpolation is performed using white noise Explanatory drawing showing the main part enlarged when audio interpolation is performed using white noise (background noise) Waveform diagram of white noise Explanatory drawing showing a case where audio data is compressed and controlled by the reception buffer controller so that the reception buffer does not overflow with audio data The main part enlarged view of FIG. The block diagram which shows the receiving side voice transmission apparatus as a VoIP terminal by Embodiment 8 of this invention Data diagram showing the structure of audio data The block diagram which shows the receiving side voice transmission apparatus as a VoIP terminal by Embodiment 9 of this invention 13 is an enlarged view of the main part of FIG. FIG. 11 is a block diagram showing a receiving side voice transmission apparatus as a VoIP terminal according to a tenth embodiment of the present invention. FIG. 11 is a block diagram showing a receiving-side voice transmission device as a VoIP terminal according to an eleventh embodiment of the present invention. Block diagram showing a receiving side voice transmission apparatus as a VoIP terminal according to a twelfth embodiment of the present invention. A block diagram showing a receiving side voice transmission apparatus as a VoIP terminal according to a thirteenth embodiment of the present invention. A block diagram showing a receiving side voice transmission apparatus as a VoIP terminal according to a fourteenth embodiment of the present invention. Block diagram showing a receiving-side voice transmission apparatus as a VoIP terminal according to a fifteenth embodiment of the present invention. Block diagram showing a receiving side voice transmission apparatus as a VoIP terminal according to a sixteenth embodiment of the present invention. Configuration diagram showing a conventional IP network

Explanation of symbols

DESCRIPTION OF SYMBOLS 1 Reception side audio | voice transmission apparatus 2 Audio | voice data 3 Reception buffer 4 Audio | voice data transmission part 5 Audio | voice reproduction | regeneration part 6 IP network 7 IP network I / F part (IP network interface part)
8 Audio data detection unit 9 Audio data duplication instruction unit 10 Recording unit 11 White noise control unit 12 White noise 13 Ambient noise measurement unit 14 Free space 15 Reception buffer control unit 16 IP network storage unit 17 Absolute time management unit

Claims (17)

A VoIP terminal connected to an IP network to which a plurality of nodes are connected, having an IP network interface unit, and performing voice data communication with any of the nodes,
Audio data transmitted from the IP network through predetermined data processing is sequentially stored, and a reception-side buffer that transmits the transmitted audio data to the audio reproduction unit that reproduces the audio as audio, and is transmitted from the reception buffer A VoIP terminal comprising: an audio data detection unit that detects audio data; and an audio data duplication instruction unit that duplicates audio data that is instructed by the audio data detection unit and accumulated in the reception buffer. The voice data duplication instruction unit, when instructed by the voice data detection unit that voice data transmitted from the reception buffer is missing, the voice data immediately before the missing voice data or the missing voice data The VoIP terminal according to claim 1, wherein the two or three previous voice data are copied and interpolated. The voice data duplication instruction unit, when instructed by the voice data detection unit that voice data sent from the reception buffer is missing, the voice data immediately after the missing voice data or the missing voice data 2. The VoIP terminal according to claim 1, wherein the second or third voice data is copied and interpolated. The audio data duplication instruction unit, when instructed by the audio data detection unit that audio data sent from the reception buffer is missing, is obtained by averaging audio data before and after the missing audio data The VoIP terminal according to claim 1, wherein the VoIP terminal is copied and interpolated. The audio data duplication instruction unit, when instructed by the audio data detection unit that audio data sent from the reception buffer is missing, for the first half of the missing audio data, immediately before the missing audio data Alternatively, the audio data two or three before the missing audio data is duplicated, and the second half of the missing audio data is immediately after the missing audio data or two or three after the missing audio data. The VoIP terminal according to claim 1, wherein the voice data is copied and interpolated. A white noise control unit that generates white noise when instructed by the audio data detection unit, the white noise control unit instructing from the audio data detection unit that audio data sent from the reception buffer is missing The VoIP terminal according to claim 1, wherein in the case of being performed, white noise is generated and interpolated. A peripheral noise measurement unit that measures and converts digital noise during a call when instructed by the voice data detection unit and records the noise is recorded, and the peripheral noise measurement unit lacks voice data transmitted from the reception buffer. The VoIP terminal according to claim 1, wherein when the voice data detection unit instructs to read out, the recorded ambient noise is read out and interpolated. When the transmitted audio data is likely to overflow from the reception buffer, the audio data detection unit is instructed to adjust the audio data amount in the reception buffer, and the reception buffer control unit The VoIP terminal according to claim 1, wherein the transmitted voice data is compressed so as not to exceed a processing capacity of the reception buffer assumed. The reception buffer control unit, when instructed by the audio data detection unit that the transmitted audio data is likely to overflow in the reception buffer, so that it can be recognized as data addressed to itself Information is added and returned to the IP network again to be temporarily stored on the IP network, and the voice data detection unit refers to the header data addressed to itself in the voice data in the IP network interface unit. The voice data detection unit instructs the reception buffer control unit and removes the header information for the user. The VoIP terminal according to claim 8, wherein the VoIP terminal is returned to the reception buffer again. The reception buffer control unit keeps the transmitted audio data in the reception buffer constant when instructed by the audio data detection unit that the transmitted audio data is likely to overflow in the reception buffer. 9. The VoIP terminal according to claim 8, wherein the VoIP terminal is extracted at intervals and duplicated and interpolated with audio data immediately before the extracted audio data or audio data two or three before the extracted audio data. The reception buffer control unit keeps the transmitted audio data in the reception buffer constant when instructed by the audio data detection unit that the transmitted audio data is likely to overflow in the reception buffer. 9. The VoIP terminal according to claim 8, wherein the VoIP terminal is extracted at intervals, and the voice data immediately after the extracted voice data or two or three voice data after the extracted voice data is duplicated and interpolated. The reception buffer control unit keeps the transmitted audio data in the reception buffer constant when instructed by the audio data detection unit that the transmitted audio data is likely to overflow in the reception buffer. The VoIP terminal according to claim 8, wherein voice data averaged with voice data before and after the extracted voice data is copied and interpolated by extracting at intervals. The reception buffer control unit keeps the transmitted audio data in the reception buffer constant when instructed by the audio data detection unit that the transmitted audio data is likely to overflow in the reception buffer. For the first half of the extracted audio data, the audio data immediately before the extracted audio data or the audio data two or three times before the extracted audio data is duplicated, and the second half of the extracted audio data is 2. The VoIP terminal according to claim 1, wherein voice data immediately after the extracted voice data or two or three voice data after the extracted voice data is copied and interpolated. A white noise control unit that generates white noise when instructed by the audio data detection unit, wherein the reception buffer control unit indicates that the transmitted audio data is likely to overflow in the reception buffer; When instructed by the data detection unit, the transmitted audio data in the reception buffer is extracted at regular intervals, and the extracted audio data is interpolated by generating white noise in the white noise control unit. The VoIP terminal according to claim 8, wherein A peripheral noise measuring unit for measuring and converting digitally recorded ambient noise during a call when instructed by the voice data detecting unit; and the reception buffer control unit is configured to receive the transmitted voice data in the reception buffer. When the voice data detection unit indicates that it is about to overflow, the transmitted voice data in the reception buffer is extracted at regular intervals. The VoIP terminal according to claim 8, wherein noise is read and interpolated. An audio data sending unit for controlling the sending amount of audio data is provided at an outlet of the receiving buffer, and the receiving buffer control unit is short of the transmitted audio data or the transmitted audio data is likely to overflow. When the audio data detection unit instructs that this, the audio data transmission unit adjusts the audio data transmission timing in order to adjust the audio data in the reception buffer. Item 9. The VoIP terminal according to Item 8. An absolute time management unit for calculating an absolute time, wherein the reception buffer control unit detects from the audio data detection unit that the transmitted audio data is insufficient or that the transmitted audio data is likely to overflow. When instructed, in order to adjust the audio data in the reception buffer, the transmission timing of the transmitted audio data in the audio data transmission unit is controlled from the absolute time of the absolute time management unit. The VoIP terminal according to claim 16, characterized in that:

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