JP2006139190A - Speech data processor and speech data processing method - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a high-grade speech processing and broadcasting with a simple and inexpensive configuration. <P>SOLUTION: AES/EBU format speech data is received through a D/D converter 7 from each of a plurality of sound sources and is converted to PCM format speech data. The PCM format speech data is accumulated in an input buffer 1a and an input message queue 1b, and is processed and composited in a speech data processing/compositing section 1c according to output speech edition information, and is then outputted through an output section 1f. <P>COPYRIGHT: (C)2006,JPO&NCIPI

Description

  The present invention relates to an audio data processing apparatus used for control and processing of digital audio data in a broadcasting station such as a radio broadcasting station or a television broadcasting station.

  In recent years, master facilities in broadcasting stations have been digitized from the viewpoint of transmission characteristics, management efficiency, and the like. In such a tendency, the audio data processing device in the master facility is being improved in order to perform efficient processing of digital audio signal control.

  FIG. 12 shows an overall conceptual diagram of radio master equipment as an audio data processing device. The radio master facility is separated into respective device units, ie, an audio switching unit and a reproduction output unit. Further, FIG. 13 conceptually shows a processing procedure of audio data in the radio master facility of FIG.

  Generally, audio data in accordance with PCM (Pulse Code Modulation), which is one of digital formats, is input to the reproduction output unit from various internal or external sound sources. On the other hand, the transmission standard of digital audio signals for broadcasting is unified to the AES / EBU format defined by AES (Audio Engineering Society) and EBU (European Broadcast Union). For this reason, the radio master facility is generally configured to support input / output data in the AES / EBU format. Therefore, the digital data format used in the voice switching unit is also the AES / EBU format. Therefore, as shown in FIGS. 12 and 13, a converter for converting the PCM format data of the reproduction output unit into AES / EBU format data is used. For conversion to AES / EBU format data and processing (for example, volume control) of AES / EBU format data, a device that performs high-speed digital signal processing such as a DSP (Digital Signal Processor) is used.

  Further, as shown in FIG. 14, the program switcher in the voice switching unit uses a matrix switcher that switches input voice data acquired from various sound sources. In the matrix structure, input audio data to be output is selected by controlling on / off the audio data output to the output side, fader control of each input audio data is performed by the mixing circuit, and the input data is synthesized Thus, the final output sound can be obtained.

JP-A-5-048551 JP-A-9-181968

  As described above, in the conventional radio master equipment, the matrix switcher corresponding to the input audio data switching circuit and the output audio mixing circuit are completely separated as separate hardware. In such a configuration, there is a need for the matrix switcher and the mixing circuit to cooperate to prevent noise generated when switching the input audio data. That is, it is necessary to consider the consistency of control timing between the voice switching control in the matrix switcher and the fader control (volume control) in the mixing circuit. For example, when a new sound source is newly input, it is necessary to turn on the sound source in the matrix switcher after the sound volume is reduced in advance by the mixing circuit. If the timing consistency is not good, noise may occur. In addition, such consideration causes an increase in input system construction work and cost.

  Also, since the audio data in the playback output unit is in PCM format, the standby command is sent from the voice switching unit before being sent to the voice switching unit for the data, and then the playback command is output from the voice switching unit to the playback output unit for playback. There is a problem that it is necessary to convert the PCM format data to the AES / EBU format and output it at the output unit, and it becomes difficult to select and switch the sound source smoothly (seamless sound source switching cannot be performed).

  Further, as described above, in the conventional radio master equipment, the digital data format used in the audio switching unit is the AES / EBU format, and the digital data format used in the reproduction output unit is PCM. These two different formats cannot be processed by the same processing system. Therefore, as shown in FIG. 13, it is necessary to use a converter for converting the AES / EBU format to PCM, which leads to an increase in system construction work and cost.

  The present invention provides an audio data processing apparatus and an audio data processing method that efficiently and smoothly process digital signal control with a simple configuration.

  An audio data processing device according to the present invention includes an audio data receiving unit that receives audio data with additional information including additional information from each of a plurality of sound sources, and the received audio data with additional information for processing that does not include additional information. A first audio data conversion unit for converting into audio data; a first accumulation unit for accumulating processing audio data; and processing audio data corresponding to each sound source accumulated in the first accumulation unit, An audio data processing / synthesizing unit that processes and synthesizes based on the output audio editing information, and an output unit that outputs the processed and synthesized processing audio data.

  The audio data processing device can be configured such that the first accumulation unit includes a plurality of virtual tracks corresponding to each sound source, and the sound data for processing of each sound source is developed on the corresponding track. . Each track can be composed of cues. Furthermore, the first storage unit may be provided with a buffer for temporarily storing the processing sound data before the processing sound data is developed on the track.

  Further, a second storage unit that stores the processed and synthesized processing voice data and then outputs the processed speech data to the output unit may be further provided.

  In addition, a second audio data conversion unit that converts the processing audio data output from the output unit into audio data with additional information may be further provided.

  Further, it is possible to further provide a reproduction output unit for accumulating and reproducing the processing audio data independently of the audio data receiving unit and the first audio data conversion unit.

  The output sound editing information can include multiplication information for processing sound data of each sound source and addition information of processing sound data for all sound sources.

  In the above description, when the audio data with additional information is AES / EBU format audio data and the processing audio data is PCM format audio data, the audio data processing device of the present invention can be applied.

  Furthermore, the audio data processing method of the present invention receives audio data with additional information including additional information from each of a plurality of sound sources, and converts the received audio data with additional information into processing audio data that does not include additional information. Then, the processing voice data is stored, the processing voice data corresponding to each stored sound source is processed and synthesized based on the output voice editing information, and the processed and synthesized processing voice data is output. A program for causing a computer to execute these steps is also included in the present invention.

  According to the audio data processing apparatus and the audio data processing method of the present invention, high-quality audio processing and broadcasting can be realized with a simple and inexpensive configuration.

  Hereinafter, embodiments of the present invention will be described with reference to the drawings. FIG. 1 is a block diagram showing a configuration of an audio data processing apparatus of the present invention and a connection relationship between the apparatus and other devices and equipment.

  The audio data processing apparatus 100 is installed in a radio broadcasting station, and is a facility that provides various programs according to time zones and audio elements (music, talk, etc.) constituting the programs. The audio data processing device 100 includes a digital switcher 1, a reproduction output unit 3, and an audio switching control unit 5. Furthermore, the audio data processing device 100 also includes a D / D converter 7, an A / D converter 11, a D / D converter 15, and a D / A converter 21 connected to the digital switcher 1.

  The audio data processing apparatus 100 receives various digital sound sources (audio material) such as DAF (digital audio file) 9a, digital audio tape recorder (DAT) 9b, MO (magneto-optical disk) 9c, and MO9d via the D / D converter 7. )It is connected to the. The digital audio data in AES / EBU format from these digital sound sources is sent to the audio data processing apparatus 100 after being converted from the AES / EBU format to the PCM format along with a predetermined frequency conversion by the D / D converter 7. It is done. The D / D converter 7 constitutes an audio data receiving unit that receives AES / EBU format audio data, and a first audio data conversion unit that converts the received AES / EBU format audio data into PCM format audio data. . Of course, the audio data receiving function can be separated from the D / D converter 7 and a separate audio data receiving unit can be provided.

  The audio data processing apparatus 100 is connected via the A / D converter 11 to various analog sound sources (audio materials) such as a studio 13a, a studio 13b, a tape recorder 13c, and a tape recorder 13d. Analog audio data from these analog sound sources is converted into predetermined digital audio data having a PCM format by the A / D converter 11 and then sent to the audio data processing apparatus 100. Each of these sound sources provides the above-described program and audio elements.

  As described above, the digital audio data input to the audio data processing apparatus 100 is subjected to processing described later, and then sent to the D / D converter 15 and the D / A converter 21. The digital audio data output to the D / D converter 15 is converted from the PCM format to the AES / EBU format together with a predetermined frequency conversion, and then output to the transmitter 17 that is normally used. It is transmitted to other receiving stations. The D / D converter 15 constitutes a second audio data converter that converts PCM format audio data into AES / EBU format audio data. The digital audio data output to the D / A converter 21 is converted into analog audio data and sent to other studios 23a, 23b, and 23c.

  The type and number of sound sources on the input side connected to the audio data processing device 100 are not limited. Also, the type and number of output destinations such as broadcast stations and studios on the output side connected to the audio data processing apparatus 100 are not limited.

  Next, each of the digital switcher 1, the reproduction output unit 3, and the audio switching control unit 5 constituting the audio data processing device 100 will be described.

  The digital switcher 1 determines a sound source to be sent to the output side by selecting an appropriate sound source from a plurality of input sound sources (audio materials) such as the above-described studio sound source and tape recorder as described above according to the time zone. is there. The selection and determination of the sound source can be automatically performed continuously for 24 hours, but various modes are conceivable depending on the broadcast. The digital switcher 1 plays the role of a radio broadcast automatic program transmission device and automatic operation device, and is sometimes referred to as an APC (Auto Program Controller), APS (Auto Program System), or the like. The function of the digital switcher 1 will be described in detail later.

  The reproduction output unit 3 can be regarded as one of sound sources, and accumulates audio data in the PCM format format. In short, the reproduction output unit 3 accumulates and reproduces PCM format audio data independently of the audio data receiving unit and the D / D converter 7 which is the first audio data conversion unit. However, PCM format audio data input via the D / D converter 7 and the A / D converter 11 can also be stored. The reproduction output unit 3 is sometimes called DAF (Digital Audio File).

  In FIG. 1, the reproduction output unit 3 is described as a part of the digital switcher 1. However, the present invention is not limited to such a configuration, and the digital switcher 1 and the reproduction output unit 3 are configured separately. It is also possible. A combination of the digital switcher 1 and the reproduction output unit 3 may be called an ACC (Audio Control Center).

  The audio switching control unit 5 controls the digital switcher 1 and the reproduction output unit 3 in response to user settings from an external device such as a console in the studio or a program stored in advance. The voice switching control unit 5 can be composed of a general-purpose CPU that is normally used. The voice switching control unit 5 may be called a DCC (Data Control Center). The function of the voice switching control unit 5 will also be described in detail later.

  FIG. 2 conceptually shows the entire processing procedure of digital audio data under the present invention. In the audio data processing apparatus 100 of the present invention, all input audio data is unified in the PCM format, and then switching processing and fader control are performed. As described above, audio data input from a broadcasting device or the like generally has a broadcasting audio data format (AES / EBU in this description) standardized in the broadcasting industry. Input audio data having such a format is converted into PCM format audio data using a D / D converter or an A / D converter, and is always (streamwise) input to the program switcher (digital switcher 1). Also, the audio material (reproduction output unit 3) separately stored inside is stored in the PCM format and is always input (in a stream) to the program switcher. A plurality of PCM format audio data input to the program switcher is synthesized with the volume (fader) being changed and synthesized by the program switcher under software control by the CPU (audio switching control unit 5). Is output.

  On the other hand, under the conventional processing shown in FIGS. 12 to 14, audio data having a broadcast audio data format (AES / EBU) is input from an external broadcasting device or the like and input to a program switcher (matrix switcher) as it is. The Also, the audio material separately stored in the PCM format is converted to the same format (AES / EBU) and input to the program switcher. Then, a plurality of AES / EBU format audio data input to the program switcher is selected by selecting a sound source by turning on / off the matrix switch as hardware processing as shown in FIG. 12 under the control of the DSP. Are synthesized and output to the outside.

  Data in a broadcast audio data format such as AES / EBU format audio data is audio data with additional information including various additional information other than substantial audio data. The additional information may include all information other than audio data substantially corresponding to audio, such as information representing the nature of substantive audio data and information required at the time of data transmission (eg, synchronization preamble, Critical, parity flag, user data, channel status, parity, etc.). That is, the audio data to be processed includes not only pure audio data but also other data, and is not configured as a series of continuous audio data. It is impossible to process such data in software, and the prior art requires a matrix switcher that is hardware. With the introduction of the matrix switcher, problems such as the aforementioned noise generation and difficulty in realizing seamless sound source switching have occurred.

  However, in the audio data processing apparatus of the present invention, as described above, data processing is performed in a state where the audio data is unified into PCM format data. The PCM format data is processing audio data that does not include additional information such as data in AES / EBU format, and is composed of only substantial audio data. In other words, in the present invention, data processing is performed after processing for removing additional information from AES / EBU format data is performed. For such format data, it is possible to perform sound source switching processing and fader adjustment in software, eliminating the need to use a separate matrix switch. As a result, it is possible to prevent noise inevitably associated with the on / off of the mechanical matrix switch and to perform seamless (seamless) input voice switching. Further, by performing the sound source switching process and the fader adjustment by a single voice switching control unit 5 (CPU), it becomes possible to shorten the control delay time. In the conventional DSP, the switching of plural voices cannot be processed by the DSP.

  Next, a specific data processing method inside the apparatus of the present invention will be described. In the present invention, input audio data in the PCM format is processed using a virtual track that is defined in correspondence with each sound source, which is called a virtual track selection method. FIG. 3 conceptually shows a state in which input audio data in the PCM format from the D / D converter 7, the A / D converter 11, and the reproduction output unit 3 is developed on a virtual track. After all input audio data corresponding to each sound source is expanded to each virtual track by software processing, the input audio data is selected by selecting the virtual track according to the control signal of the audio switching control unit 5 To do. In the example of FIG. 3, audio data is always input to eight tracks 1 to 8, but only the audio data of tracks 1 and 4 is selected in the selection control.

  FIG. 4 conceptually shows a state in which input audio data developed on a virtual track is synthesized. In the present invention, PCM format audio data always flows through each track for each block DB (data segment) of a predetermined unit, as shown in the track image in the figure. In such a virtual track selection method, since voice switching control and fader adjustment are realized simultaneously, input voice data can be switched instantaneously. Further, it is not necessary to consider consistency at the control timing of the voice switching control and the fader control for noise countermeasures.

  In the example of FIG. 4, the audio data of the selected virtual tracks 1 and 2 are respectively processed (equivalent to fader control and volume control) and synthesized. On the other hand, the audio data of the virtual track 3 that has not been selected is discarded (fader coefficient described later is 0). The data of each track is sequentially expanded for each block DB in a predetermined unit according to the order of transmission. Here, as will be described later, FIFO (First In First Out) processing using a queue is performed on the audio data.

  An example of the configuration of the digital switcher 1 for realizing the processing as described above is shown in FIG. The digital switcher 1 includes an input buffer 1a, an input message queue 1b, an audio data processing / synthesis unit 1c, an output message queue 1d, an output buffer 1e, and an output unit 1f. The input buffer 1a and the input message queue 1b constitute a first accumulation unit, and the output message queue 1d and the output buffer 1e constitute a second accumulation unit, but the output message queue 1d and the output buffer 1e are omitted. Is possible. Each component is realized by various commonly used processors, memories, and the like, and is controlled according to a predetermined program by various commands and information (including output audio editing information described later) from the audio switching control unit 5.

  The operation of the digital switcher 1 controlled by the voice switching control unit 5 will be described with reference to FIGS. 5 and 6 to 8 described above. FIG. 6 is a process flow showing the procedure of the audio data interrupt process performed when the audio data interrupt is input. The audio data interruption process is performed in the order of input audio data acquisition process (step S100), audio data processing and synthesis process (step S200), and output audio data setting process (step S300).

  FIG. 7 is a processing flow showing a detailed processing procedure of the input audio data acquisition processing (FIG. 6; step S100) (corresponding to the first half of the processing for developing the track image in FIGS. 3 and 4). First, audio data is written to the input buffer 1a from the D / D converter 7, the A / D converter 11, and the reproduction output unit 3, and the audio data is temporarily stored in the input buffer. Using this accumulation as an interrupt input for audio data, the input audio channel index is initialized (step S110), and it is then determined whether or not acquisition of audio data for all input audio channels has been completed (step S110). S120). If it is determined that initialization of all input audio channels has not been completed (step S120; No), audio data is acquired from the input buffer 1a (step S130), and the audio data is associated with the input message queue 1b. The track is developed and stored (step S140). Here, “audio channel” refers to a physical input source or output destination. On the other hand, the above-mentioned “track” refers to an accumulation image of audio data. Therefore, the audio data is processed in the flow such as the input channel, the track, and the output channel.

  Then, the input audio channel index is updated (step S150), and the process returns to step S120 to determine again whether or not the initialization of all the input audio channels has been completed (step S120). If it is determined that initialization of all input audio channels has been completed (step S120; Yes), the input audio data acquisition process ends. If it is determined that the initialization of all input audio channels has not been completed (step S120; No), the processes of steps S130 to S150 are performed again.

  FIG. 8 is a processing flow showing a detailed processing procedure of the audio data processing and synthesis processing (FIG. 6; step S200) (corresponding to the middle stage of the processing for developing the track image in FIGS. First, when it is detected that audio data is stored in the input message queue 1b, the output audio channel index is initialized (step S210), and then audio data acquisition for all output audio channels is completed. It is determined whether or not (step S220). If it is determined that initialization of all output audio channels has not been completed (step S220; No), the audio data processing / synthesizing unit 1c sends output audio editing information (in the studio) from the audio switching control unit 5. User setting input by an installed console or the like, information generated based on preprogrammed settings, etc.) is acquired (step S230), and voice data is acquired from the input message queue 1b (step S240). . Then, the input voice data is processed / synthesized (step S250), and the processed / synthesized voice data is stored in the output message queue 1d (step S260).

  In the above steps, fader control (volume control) of each sound source is performed with reference to the volume information of each sound source included in the above-described output audio editing information. Processing is realized by multiplying audio data by a predetermined coefficient, and synthesis is realized by adding all audio data (audio data from all sound sources). When the predetermined coefficient described above corresponds to the fader (volume) of each sound source, the audio data is multiplied by the fader coefficient. When data of a certain sound source is not selected, the fader coefficient of the sound data of the sound source becomes 0 and is discarded as shown in the virtual track 3 of FIG. The addition of all audio data includes the case where such audio data with fader coefficient 0 exists. After all, setting the fader coefficient to 0 is equivalent to executing the voice switching control, and the voice switching control and the fader control are performed simultaneously.

  Then, the output audio channel index is updated (step S270), and the process returns to step S220 to determine again whether or not the initialization of all output audio channels has been completed (step S220). If it is determined that the initialization of all output audio channels has been completed (step S220; Yes), the audio data processing and synthesis processing are terminated. If it is determined that initialization of all output audio channels has not been completed (step S220; No), the processing from step S230 to step S270 is performed again.

  FIG. 9 is a process flow showing a detailed processing procedure of the output audio data setting process (FIG. 6; step S300) (corresponding to the latter half of the process of developing the track image in FIGS. 3 and 4). First, when storage of audio data in the output message queue 1d is detected, an output audio channel index is initialized (step S310), and whether or not audio data acquisition for all output audio channels is completed. Is determined (step S320). If it is determined that initialization of all output audio channels has not been completed (step S320; No), audio data is acquired from the output message queue 1d (step S330), and the audio data is set in the output buffer. (Step S340).

  Then, the output audio channel index is updated (step S350), and the process returns to step S320 to determine again whether or not the initialization of all output audio channels has been completed (step S320). If it is determined that initialization of all output audio channels has been completed (step S320; Yes), the output audio data setting process is terminated. If it is determined that initialization of all output audio channels has not been completed (step S320; No), the processing from step S330 to step S350 is performed again.

  The output audio data set in the output buffer 1e in this way is sent to the output unit 1f at a predetermined timing, and final output audio data is output.

  The PCM format audio data output from the output unit 1f is converted to AES / EBU format or analog by the D / D converter 15 and the D / A converter 21, and transmitted to the output destination.

  Further, in the present embodiment, the output unit 1f adopts the MADI (Multichannel Audio Digital Interface) standard, and adopts a configuration for outputting PCM format audio data by the MADI standard method. The MADI standard is a general term for MADI (AES10-1991) with maximum 56 channel input / output defined by AES in 1991 and MADI-X (MADI-Extended) format with maximum 64 channel input / output defined in 2001. With one coaxial cable (coaxial cable) or optical cable (optical cable), it is possible to transmit broadcast information and audio information exceeding 50 channels at a time.

  In particular, in the above-described MADI-X, it is possible to expand the output string to the maximum monaural 64 outputs (stereo 32 outputs) for each monaural 8 outputs (stereo 4 outputs) with respect to the output unit. As a result, it is possible to increase the channels of output audio without physically changing the size of the audio processing device.

  It is also possible to assign a converter type for each monaural 8 output (stereo 4 output). Therefore, the number of converters mounted can be reduced.

  FIG. 10 schematically shows transmission of audio data at the output unit 1f under the MADI standard. This figure shows a state in which four audio channels 1 to 4 are output from the audio data processing device 100 through one serial line 40 by adopting the MADI standard. By adopting the MADI standard, data for four channels can be sent by the serial line 40. In this case, the output unit 1f includes a MADI interface incorporated on the main body side of the audio data processing apparatus 100 and a MADI unit 30 (MADI output unit) provided separately from the main body. However, the MADI interface and the MADI unit 30 are not necessarily configured as separate units, and may be configured as a single unit.

  FIG. 11 shows an outline of the rear configuration of the MADI unit 30. The MADI unit 30 is connected to the output unit 1 f via the serial line 40. Alternatively, it can also be said that the MADI unit 30 constitutes a part of the output unit 1f. The MADI unit 30 is further connected to the D / D converter 15 and the D / A converter 21 of FIG. An output side port 30 a is provided on the back surface of the MADI unit 30, and each one is connected to the serial line 40. The input side port 30b is disposed between the D / D converter 7 or A / D converter 11 on the input side and the digital switcher 1 and is used when data is transmitted and received in the same manner as the MADI standard. However, it is not essential in the present invention.

  As described above, in the present invention, the audio data to be processed in the apparatus is format data (PCM) without additional information. Therefore, it is possible to process the audio data using software. Specifically, after the input audio data is developed into a virtual track by software control, the virtual track can be selected by fader control and audio can be mixed.

  In such a virtual track selection method, it is considered that voice switching control and fader control are realized simultaneously. Therefore, the input voice can be switched instantaneously (with blank time 0). Further, it is not necessary to consider the consistency of the timings of the voice switching control and the fader control for noise countermeasures. In addition, it is possible to give variations to the switching timing of the input voice.

  Further, since the matrix switcher used in the voice switching unit of the prior art is not necessary, the apparatus configuration can be simplified and the cost can be reduced.

  The audio data processing apparatus of the present invention is preferably used in a digital master system (radio master system in the case of radio broadcasting) used for digital signal control in broadcasting stations such as radio broadcasting stations and television broadcasting stations. The application is not limited. Further, the data with additional information includes “ADPCM”, “MPEG2 / AAC”, “MP3”, etc., but the present invention is also applied to processing of these audio data and compressed data.

  Furthermore, the present invention includes a method for processing audio data. The method includes: 1) receiving audio data with additional information including additional information from each of a plurality of sound sources; and 2) converting the received audio data with additional information into processing audio data not including additional information. 3) a step of storing processing voice data, 4) a step of processing and synthesizing the stored processing voice data corresponding to each sound source based on the output voice editing information, and 5) processing and synthesis. Outputting the processed audio data for processing.

  A program for causing a computer to execute the above steps is also included in the present invention. This program can be incorporated in or outside the device in various formats. For example, the program can be recorded in a predetermined memory in the voice switching control unit 5. It is also possible to record the program in an information recording device such as a hard disk, or an information recording medium such as a CD-ROM, DVD-ROM, or memory card, and load it into an audio data processing device.

  Although various embodiments of the present invention have been described above, the present invention is not limited to the matters shown in the above-described embodiments, and those skilled in the art can make modifications and applications based on the description and well-known techniques. This is also the scope of the present invention, and is included in the scope of seeking protection.

  According to the audio data processing apparatus and the audio data processing method of the present invention, high-quality audio processing and broadcasting can be realized with a simple and inexpensive configuration.

The block diagram which shows the outline | summary of the connection relation with the audio | voice data processing apparatus of this invention, another apparatus, and installation. The conceptual diagram which shows the process sequence of the audio | voice data in the audio | voice data processing apparatus of this invention. The conceptual diagram of expansion | deployment to the virtual track | truck of input audio | voice data. The conceptual diagram which shows the selection method of the input audio | voice data expand | deployed by the virtual track | truck. The block diagram which shows the structure of a digital switcher. The processing flow which shows the procedure of the audio | voice data interruption process implemented at the time of audio | voice data interruption input. 7 is a processing flow showing a processing procedure of input voice data acquisition processing in FIG. FIG. 7 is a processing flow showing a processing procedure of voice data processing and synthesis processing in FIG. 6. 7 is a processing flow showing a processing procedure of output audio data setting processing in FIG. The schematic diagram which shows transmission of the audio | speech data in an output part under MADI specification. Schematic of the back of the MADI unit. The conceptual diagram of the conventional radio master equipment. The conceptual diagram which shows the process sequence of the audio | voice data in the conventional radio master equipment. The conceptual diagram of the matrix switcher used in the conventional radio master equipment.

Explanation of symbols

DESCRIPTION OF SYMBOLS 1 Digital switcher 3 Reproduction | regeneration output part 5 Audio | voice switching control part 7 D / D converter 11 A / D converter 15 D / D converter 21 D / A converter 30 MADI unit 40 Serial line 100 Audio | voice data processing apparatus

Claims (11)

An audio data receiving unit for receiving audio data with additional information including additional information from each of a plurality of sound sources;
A first audio data converter that converts the received audio data with additional information into audio data for processing not including additional information;
A first storage unit for storing the processing voice data;
An audio data processing / synthesizing unit that processes and synthesizes the processing audio data corresponding to each sound source stored in the first storage unit based on output audio editing information;
An audio data processing apparatus comprising: an output unit that outputs the processed and synthesized audio data for processing. The audio data processing device according to claim 1,
The first storage unit includes a plurality of virtual tracks corresponding to the sound sources,
An audio data processing device in which audio data for processing of each sound source is developed on the corresponding track. The audio data processing device according to claim 2,
An audio data processing apparatus, wherein each track is composed of cues. The voice data processing device according to claim 2 or 3,
The audio data processing device, wherein the first storage unit includes a buffer that temporarily stores the processing audio data before the processing audio data is expanded to the track. The voice data processing device according to any one of claims 1 to 4,
An audio data processing apparatus further comprising a second storage unit that stores the processed and synthesized processing audio data and then outputs the processed audio data to the output unit The voice data processing device according to any one of claims 1 to 5,
An audio data processing apparatus, further comprising: a second audio data conversion unit that converts the processing audio data output from the output unit into audio data with additional information. The voice data processing device according to any one of claims 1 to 6,
An audio data processing apparatus further comprising a reproduction output unit for accumulating and reproducing the audio data for processing independently of the audio data receiving unit and the first audio data conversion unit. The voice data processing device according to any one of claims 1 to 7,
The output audio editing information is an audio data processing device including multiplication information for processing sound data of each sound source and addition information of processing sound data of all sound sources. The voice data processing device according to any one of claims 1 to 8,
The audio data with additional information is AES / EBU format audio data,
An audio data processing apparatus, wherein the processing audio data is PCM format audio data. Receive audio data with additional information including additional information from each of a plurality of sound sources,
The received voice data with additional information is converted into processing voice data not including additional information,
Storing the processing voice data;
Processing and synthesizing the processing audio data corresponding to each of the accumulated sound sources based on the output audio editing information,
An audio data processing method for outputting the processed and synthesized audio data for processing.   The program for making a computer perform each step of Claim 10.

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