JP2005340973A - Voice relay broadcast system and voice relay broadcast method - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a voice relay broadcast system and a voice relay broadcast method capable of carrying out relaying with high voice quality sustainable and appropriate to broadcasting without causing unnaturalness to a conversation between a reporter at a site and a radio personality in a studio. <P>SOLUTION: The voice relay broadcast system is provided with: a voice packet transmission means for transmitting voice at the site to the studio via a packet channel network; a voice transmission means for transmitting the voice in the studio to the side via an ordinary channel network; a studio side voice packet reception means for receiving the voice at the site via the packet channel network; a site side ordinary voice reception means for receiving the voice in the studio via the ordinary channel network; and a broadcast means for broadcasting the voice produced from the studio and the voice received by the studio side voice packet reception means. <P>COPYRIGHT: (C)2006,JPO&NCIPI

Description

  The present invention relates to a technique for relaying audio from the field to the studio side in radio broadcasting, and more specifically, there is no unnatural conversation between the reporter at the field and the personality of the studio, and it can withstand broadcasting. The present invention relates to an audio relay broadcast system and an audio relay broadcast method that can perform high-quality sound relay.

  In radio broadcasts and the like, in order to provide highly fresh information and a realistic program, conversations between the reporter at the site and the personality of the studio are often incorporated in live broadcasting.

By the way, radio broadcasts generally require high sound quality, so in order to relay the sound from the field in real time to the studio side in a high sound quality state, a wide bandwidth dedicated line (wired, wireless, satellite), etc. Will be used. In addition, a temporary base station or the like may be provided on the site side for relaying.
JP-A-8-288919 JP-A-10-145636

  As described above, in the conventional radio broadcasting or the like, in order to relay the sound from the site with high sound quality, use of a dedicated line with a wide bandwidth and facilities are necessary, and there is a problem that the cost of the relay is high.

  On the other hand, mobile phones, which have been popular recently, have low usage fees and excellent mobility, but they cannot withstand broadcasts in terms of sound quality, and simple conversations that do not bother the poor sound quality. There was a problem that it was necessary to use it limitedly or to use the sound quality poorly.

  In addition, the use of VoIP (Voice over Internet Protocol) using a wireless packet line is also being studied, but the data compression rate cannot be increased so much in order to perform high sound quality relay, and the amount of data increases. There was a problem that a delay that occurred on the network caused an “in-between” that could not be ignored in the conversation between the reporter in the field and the personality of the studio, giving an unnatural impression to listeners such as radio broadcasting.

  In addition, although techniques for eliminating delays such as relay voice from the field in broadcasting have been disclosed (see, for example, Patent Documents 1 and 2), these prior art documents target delays in satellite circuits and the like. It is not intended for delay due to a packet line.

  The present invention has been proposed in view of the above-described conventional problems, and the object of the present invention is that there is no unnatural conversation between the reporter at the site and the personality of the studio, and it can withstand broadcasting. An object of the present invention is to provide an audio relay broadcast system and an audio relay broadcast method capable of relaying sound quality.

  In order to solve the above problems, according to the present invention, as described in claim 1, voice packet transmission means for transmitting voice on the site side to the studio side via a packet line network, and the studio side Voice transmitting means for transmitting the voice of the site to the site side via the normal line network, studio side voice packet receiving means for receiving the site side voice via the packet line network, and via the normal line network A field-side normal sound receiving means for receiving the studio-side sound; and a broadcasting means for broadcasting the sound generated on the studio side and the sound received by the studio-side sound packet receiving means.

  As a result, the sound from the reporter on site to the personality of the studio is performed with high-quality sound, and the sound from the personality of the studio to the reporter on-site is performed on a normal line with low sound quality but no delay. Since the audio from the studio personality to the reporter on site is broadcast using the audio collected in the studio, it becomes possible to perform high-quality broadcasting while minimizing the influence of delay due to the packet line.

  In addition, as described in claim 2, since broadcasting is performed by selecting a voice to be used for broadcasting based on a delay amount generated by the packet line network, if the delay amount is increased, the voice quality is reduced but the delay is increased. It is possible to broadcast audio transmitted on a normal line without any problems, and it is possible to avoid a more serious failure such as interruption of the relay.

  According to a third aspect of the present invention, the delay amount is calculated by comparing the voice received on the packet line with the voice received on the normal line. As a result, even in an environment where the delay amount fluctuates, a highly accurate delay measurement is possible.

  According to a fourth aspect of the present invention, the voice on the site side is converted into a voice packet and transmitted to the studio side via the packet line network, and the voice on the studio side is sent to the field side via the normal line network. And broadcasting the audio on the studio side and the audio received via the packet line network. This makes it possible to perform high-quality sound broadcasting while minimizing the influence of delay due to the packet line.

  According to the present invention, by using a packet communication line for which high-quality sound transmission can be expected and a voice communication line for which real-time transmission can be expected, in the case of voice using a packet line network, the delay amount is half of the conventional amount. It is possible to relay high-quality sound that can withstand broadcasting while suppressing. In addition, since the sound used for broadcasting is selected according to the delay amount generated in the packet communication line, it is possible to perform broadcasting with the optimum sound even when the communication status or the like changes every moment. In addition, when the same delay amount as that of the prior art can be tolerated, the transmission data can have a large amount of data with a low compression rate, and further higher sound quality can be achieved.

  Hereinafter, embodiments of the present invention will be described with reference to the drawings.

  FIG. 1 is a configuration diagram of a conversational voice relay system according to an embodiment of the present invention.

  In FIG. 1, the reporter's voice is collected by the microphone 11 at the site 1 and transmitted to the studio 3 side with high quality via the packet line network 21, and the reporter's voice is sent to the microphone 13. And a normal audio transmission / reception unit 15 for transmitting / receiving audio signals to / from the studio via the normal line network 22 in order to transmit the personality sound of the studio 3 side to the reporter via the speaker 14. ing.

  The packet line network 21 is a data communication line that can be accessed wirelessly, and the normal line network 22 is a mobile phone line. These are not expensive leased lines but publicly available public lines and can be used at low communication costs. As the network 2 including both services of the packet line network 21 and the normal line network 22, for example, “FOMA” (trademark) of NTT DoCoMo, Inc. can be used, and this is used for each of the packet line network 21 and the normal line network 22. I will do it.

  The internal configuration of the voice packet transmission unit 12 and the normal voice transmission / reception unit 15 will be described later. As a specific image, the voice packet transmission unit 12 includes a personal computer and a mobile phone capable of transmitting packet data. The voice is converted into a packet and transmitted to the packet line network 21. The normal voice transmitting / receiving unit 15 can be configured by a mobile phone itself. Therefore, the reporter can be gathered as a shoulder back that is worn on the shoulder and can be excellent in mobility.

  On the other hand, on the studio 3 side, the voice packet receiving unit 31 that receives the voice coming via the packet line network 21 and transmits the voice of the reporter on the site 1 side to the personality by the speaker 34, and the voice of the personality by the microphone A normal voice transmission / reception unit 32 that transmits and receives audio signals to and from the site 1 via the normal line network 22 in order to collect and transmit to the site 1 side and receive the voice of the reporter on the site 1 side; The microphone 36 and the amplifier 37 that collect the voice of the personality conversation, and the received voice of the normal voice transmitting / receiving unit 32 indicate how much the voice arriving from the packet line network 21 via the voice packet receiver 31 is delayed. The amount of delay measurement unit 35 that measures by comparison with the generated delay amount is within an allowable value. Is selected from the audio obtained from the audio packet receiving unit 31 and the audio obtained from the normal audio transmitting / receiving unit 32, and synthesized with the output audio from the amplifying unit 37 to the broadcasting apparatus. An audio output device 39 is provided.

  When the above-described “FOMA” line is used as the packet line network 21 or the normal line network 22, the normal line network 22 can be used while using the packet line network 21. The packet line network 21 can also be used while using. At this time, on the site 1 side, the same voice is transmitted via the normal line network 22 in parallel with transmitting the voice packet of the reporter on the site 1 side via the packet line network 21. The delay amount measuring unit 35 measures the amount of delay occurring in the packet line network 21 by comparing the arrival times of the same voice using this.

  Although it is preferable that the allowable value of the delay amount for performing the delay determination is determined in advance, it may be determined as appropriate depending on the convenience of the line.

  FIG. 2 shows the configuration of the voice packet transmission unit 12 provided on the site 1 side. The voice I / F 121 that digitizes and imports the voice signal collected by the microphone 11, and compresses voice data and assembles the packet. High-quality sound data communication software 122 and a packet line I / F 127 for connecting to the packet line network 21. The high sound quality data communication software 122 includes an audio CODEC 123 that compresses audio using an algorithm such as AAC (Advanced Audio Coding), and a packet processor 124 that assembles packets. The packet processor 124 is an RTP (RTP). An RTP processing unit 125 that performs packetization according to Real-time Transport Protocol) and a UDP / IP processing unit 126 that performs packetization according to UDP / IP (User Datagram Protocol / Internet Protocol).

  FIG. 3 shows the configuration of the voice packet receiver 31 provided on the studio 3 side. The packet line I / F 311 for connecting to the packet line network 21 and the high level for extracting and decompressing voice data from the packet are shown. The audio quality data communication software 312 and an audio I / F 317 for restoring an audio signal from digital data are configured. The high sound quality data communication software 312 includes a packet processing unit 313 that extracts audio data from a packet, and an audio CODEC 316 that decompresses audio data using an algorithm such as AAC. The packet processing unit 313 includes a UDP / IP. A UDP / IP processing unit 314 that extracts a voice data packet from a packet that complies with RTP, and an RTP processing unit 315 that extracts voice data from a packet according to RTP.

  FIG. 4 shows the configuration of the normal audio transmission / reception unit 15 provided on the site 1 side and the normal audio transmission / reception unit 32 provided on the studio 3 side, and the connection with the microphone 13 (33) and the speaker 14 (34) is shown. A voice I / F 151 to be performed, a normal voice telephone software 152 including a voice CODEC 153 that compresses and decompresses voice by an algorithm such as AMR (Advanced Multi Rate), and a normal line I / F 154 to connect the normal line network 22 It is composed of

  FIG. 5 shows a voice packet transmission unit and a normal voice transmission / reception unit using a multi-access function in which the voice packet transmission unit 12 and the normal voice transmission / reception unit 15 on the site 1 side are integrated, and both the packet line and the normal line can be used in parallel. 19 shows the configuration. Thereby, the equipment carried by the reporter on site 1 can be configured more compactly.

  In FIG. 5, a voice packet transmitting unit / normal voice transmitting / receiving unit 19 is a personal computer 1902 including a voice I / F 1901 for connecting to microphones 16, 17 and a speaker 18, high-quality data communication software 1903, and normal voice telephone software 1908. And a mobile terminal 1910 including a packet line I / F 1911 for connection to the packet line network 21 and a normal line I / F 1912 for connection to the normal line network 22. Note that the microphone 17 can be omitted by using it as the microphone 16.

  The high sound quality data communication software 1903 includes an audio CODEC 1904 that compresses audio using an algorithm such as AAC, and a packet processing unit 1905 that assembles packets. The packet processing unit 1905 performs packetization according to RTP. It comprises an RTP processing unit 1906 that performs and a UDP / IP processing unit 1907 that performs packetization according to UDP / IP. The normal voice telephone software 1908 is composed of a voice CODEC 1909 that compresses and decompresses voice by an algorithm such as AMR.

  FIG. 6 is a signal waveform diagram of each part showing the operation state of the above embodiment.

  Now, when the reporter utters “this site ...” in the field 1, this voice is received from the voice packet transmitting unit 12 by the voice packet receiving unit 31 on the studio 3 side via the packet line network 21. . The voice uttered by the reporter at the work site 1 is represented by a signal a, and the voice received by the voice packet receiving unit 31 on the studio 3 side is represented by a signal b. Although the signal b is output after being delayed by the delay Δt generated in the packet line network 21, relatively high-quality sound can be used for broadcasting. This delay is increased by suppressing the data compression rate in order to improve sound quality.

  Correspondingly, the voice of the utterance that the personality is “here studio...” In the studio 3 to the microphones 33 and 36 is the signal c. Are output to the speaker 14.

  A signal d is a combination of the voice of the reporter on the site 1 side received by the voice packet receiving unit 31 and the voice of the personality on the studio 3 side collected by the microphone 36 by the output device 39, and can withstand broadcast quality. Sound quality can be obtained.

  Since the sound (signal b) from the site 1 to the studio 3 has a delay Δt, the response from the studio 3 to the site 1 is delayed by the delay Δt. However, the reporter at the site 1 does not know how late the voice he / she has reached the studio 3. Even when there is a delay Δt, it seems that the personality of the studio 3 is intentionally “open”. This conversational voice is delayed by Δt from the actual conversation, but there is no problem in broadcasting because the timing of cueing is shifted. From the viewpoint of viewing the broadcast, since the audio collected on the studio 3 side is being broadcast, even if there is a delay Δt, it feels as if the reporter at the site 1 is intentionally spaced. Degree. Compared with the conventional method, the delay that has occurred in the round trip between the site 1 and the studio 3 occurs only in one way from the site 1 to the studio 3, and it is possible to broadcast higher-quality audio data. Therefore, since the restriction due to the delay amount of the packet line network 21 is relaxed, the transmission data can have a lower compression ratio and a larger amount of data, and further higher sound quality can be achieved.

  However, if the delay Δt becomes too large, a conversation that can withstand broadcasting becomes impossible. Therefore, the delay amount measuring unit 35 measures the delay amount continuously or every predetermined time. When the delay determination unit 38 determines that the delay amount measured by the delay amount measurement unit 35 exceeds the predetermined upper limit of the delay amount, it is not voice communication via the packet line network 21 but the normal line network 22. Switch to voice communication via. In this case, since voice communication via the normal line network 22 is performed in parallel with voice packet communication via the packet line network 21, it is used as an output signal of the voice packet receiving unit 31 or a normal voice transmission / reception unit 32. The output can be smoothly switched by simply switching the output with the audio output device 39. However, it is desirable to switch the audio so that the switching is not noticeable, for example, during commercial broadcasting.

  Further, by performing voice communication via the normal line network 22 in parallel with voice packet communication via the packet line network 21, not only when the delay increases, but also the communication state of one line In the case of worsening, it is possible to perform relay using only a better line.

  The present invention has been described above by the preferred embodiments of the present invention. While the invention has been described with reference to specific embodiments, various modifications and changes may be made to the embodiments without departing from the broad spirit and scope of the invention as defined in the claims. Obviously you can. In other words, the present invention should not be construed as being limited by the details of the specific examples and the accompanying drawings.

1 is a configuration diagram of an audio relay broadcast system according to an embodiment of the present invention. It is a block diagram of a voice packet transmission unit. It is a block diagram of a voice packet receiver. It is a block diagram of a normal voice transmission / reception part. It is a block diagram of a voice packet transmitting unit and a normal voice transmitting / receiving unit. It is a signal waveform diagram of each part.

Explanation of symbols

DESCRIPTION OF SYMBOLS 1 On-site 11 Microphone 12 Voice packet transmission part 121 Voice I / F
122 High-quality sound data communication software 123 Voice CODEC
124 packet processing unit 125 RTP processing unit 126 UDP / IP processing unit 127 packet line I / F
13 Microphone 14 Speaker 15 Normal audio transmission / reception unit 151 Audio I / F
152 Normal voice telephone software 153 Voice CODEC
154 Normal line I / F
16 microphone 17 microphone 18 speaker 19 voice packet transmitter / normal voice transmitter / receiver 1901 voice I / F
1902 Personal Computer 1903 High Quality Data Communication Software 1904 Voice CODEC
1905 Packet processing unit 1906 RTP processing unit 1907 UDP / IP processing unit 1908 Normal voice telephone software 1909 Voice CODEC
1910 Mobile terminal 1911 Packet line I / F
1912 Normal line I / F
2 network 21 packet line network 22 normal line network 3 studio 31 voice packet receiving unit 32 normal voice transmitting / receiving unit 33 microphone 34 speaker 35 delay amount measuring unit 36 microphone 37 amplifying unit 38 delay determining unit 39 output device 311 packet line I / F
312 High-quality sound data communication software 313 Packet processing unit 314 UDP / IP processing unit 315 RTP processing unit 316 Voice CODEC
317 Voice I / F

Claims (4)

Voice packet transmission means for transmitting the voice on the site side to the studio side via the packet line network;
Audio transmitting means for transmitting the studio side audio to the site side via a normal line network;
Studio-side voice packet receiving means for receiving the field-side voice via the packet line network;
Field-side normal audio receiving means for receiving the studio-side audio via the normal line network;
Broadcasting means for broadcasting the audio generated at the studio side and the audio received by the studio side audio packet receiving means,
A voice relay broadcasting system characterized by comprising:   Delay measuring means for measuring a delay amount of the on-site voice received via the packet line network; and a voice selecting means for selecting a sound to be broadcast by the broadcasting means based on the delay amount measured by the delay measuring means. The audio relay broadcast system according to claim 1, further comprising: The delay measuring means includes
3. The delay amount is calculated by comparing the on-site voice received via the packet line network and the on-site voice received via the normal line network. Voice relay broadcasting system. The voice on the site side is converted into a voice packet and sent to the studio side via the packet line network.
Send the studio side audio to the site side via a normal network,
An audio relay broadcasting method, wherein the audio on the studio side and the audio received via the packet line network are broadcast.

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