JP7461090B1 - Audio processing device, audio processing method, and program - Google Patents

Abstract

[Problem] To provide an advantageous technique for automatic volume adjustment for multiple audio files. [Solution] An audio processing device has a storage unit that stores middleware, which is software for processing audio, and a digital audio workstation (DAW), which is software different from the middleware for processing audio, and a processor that executes the middleware and the DAW, and the processor obtains the amount of change in the volume value of the audio on the middleware based on routing information that indicates the path taken by the audio recorded in the audio file set by the middleware on the DAW until it reaches the output. [Selected Figure] Figure 1

Description

The present invention relates to an audio processing device, an audio processing method, and a program.

In applications that handle multiple audio files, it is often desirable that the volume of each file be adjusted to a designated volume. For example, in a game, if the volume of the action sound (for example, walking sound) of the same character varies greatly depending on the scene, it may give the user a sense of discomfort. Therefore, developers spend a lot of effort adjusting the volume of multiple audio files installed in a game.

Conventionally, volume adjustment for a plurality of audio files has been performed, for example, using the following procedure.
(a) A plurality of delivered audio files are stored in a storage device.
(b) Listen to and compare the reference audio file and one audio file selected from a plurality of audio files.
(c) Adjust the signal level of the audio file so that the perceptual volume is the same.
(d) Repeat (b) and (c) for unprocessed audio files among the plurality of audio files.

Note that the signal level adjustment performed in step (c) above is not limited to changing the audio data itself. For example, Patent Document 1 describes that an automatic volume adjustment element is stored in association with audio data, and the volume is adjusted using the automatic volume adjustment element when reproducing the audio data. Patent Document 2 describes adding a playback control identifier regarding playback volume to the file name of a music file, and adjusting the sound volume using the playback control identifier when playing the music file.

JP 2003-243952 A JP 2011-197664 A

However, for example, the number of audio files used in a game may reach tens of thousands. Adjusting the volume of such a large number of audio files one by one would require a huge amount of work. Therefore, it is desired to reduce the labor required for volume adjustment work by automating the volume adjustment for multiple audio files. In addition, in game development, two software programs are used to create and adjust audio files: middleware (audio middleware) and digital audio workstation (DAW). However, on the DAW, it is not possible to grasp how the volume of each of the multiple audio files has been adjusted by the middleware, and it is not possible to adjust the volume taking into account the volume adjustment results of the middleware.
The present invention provides advantageous techniques for automatic volume adjustment for multiple audio files.

According to one aspect of the present invention, there is provided an audio processing device for processing audio, the audio processing device comprising: a storage unit for storing middleware, which is software for processing audio, and a digital audio workstation (DAW), which is software different from the middleware for processing audio; and a processor for executing the middleware and the DAW, the processor acquiring the amount of change in the volume value of the audio on the middleware based on routing information indicating the path taken by the audio recorded in the audio file set by the middleware on the DAW to reach the output.

The present invention provides an advantageous technique for automatic volume adjustment for multiple audio files.

FIG. 1 is a block diagram showing the configuration of an audio processing device according to an embodiment. FIG. 4 is a diagram showing an example of the structure of a loudness table. FIG. 4 is a diagram illustrating a loudness value setting screen. FIG. 1 is a conceptual diagram illustrating the hierarchical structure of audio on middleware. FIG. 3 is a diagram illustrating a setting screen for acquiring the amount of change in volume value by middleware. Flowchart of the audio processing method. FIG. 3 is a diagram illustrating a display example of audio waveforms.

Hereinafter, embodiments will be described in detail with reference to the accompanying drawings. Note that the following embodiments do not limit the claimed invention, and not all combinations of features described in the embodiments are essential to the invention. Two or more features among the plurality of features described in the embodiments may be arbitrarily combined. In addition, the same or similar configurations are given the same reference numerals, and duplicate explanations will be omitted.

FIG. 1 shows a block diagram showing the configuration of an audio processing device C according to an embodiment. The audio processing device C is a device that displays an audio signal recorded in a file and performs various processing on the audio signal, such as adjusting the signal level. As used herein, the term "sound" should be understood in a broad sense. "Sound" may include not only sounds made by humans or animals, but also musical sounds, computer-generated sound effects, etc. That is, as used herein, the term "voice" is intended to include "speech," "sound," and "audio."

The audio processing device C may be a computer device such as a personal computer or a workstation. The audio processing device C includes a CPU (central processing unit) 101 that controls the entire device, a RAM 102 that functions as a main storage device and provides a work area for the CPU 101, and a ROM 103 that stores fixed data and programs. The audio processing device C also includes an audio interface (I/F) 104. A microphone M and a speaker S can be connected to the audio interface 104. A storage device (secondary storage device) 110 (storage unit) is connected to the audio processing device C via an interface (I/F) 105. Storage device 110 may be, for example, a hard disk drive (HDD), a solid state drive (SSD), or a combination thereof. Note that the storage device 110 may be configured inside the audio processing device C, or may be configured externally. Network interface 106 connects to network N and performs communication. The audio processing device C can be communicatively connected to the server A via a network N, for example.

An input device K such as a keyboard or a mouse can be connected to the audio processing device C via an interface 107. An external media device F such as a CD-ROM drive or a DVD drive can be connected to the audio processing device C via an interface 108. Furthermore, the audio processing device C includes a video controller 109. The video controller 109 controls image display by a display device (display) D. A touch panel screen may be configured by integrating the input device K and the display D.

A boot program for starting the audio processing device C is stored in the ROM 103. Further, as shown in FIG. 1, an operating system (OS) 111 and one or more audio files 113 may be installed in the storage device 110. The audio file 113 may be supplied from an external device such as the server A via the network N, or may be supplied from a medium stored in the external media device F. Alternatively, the audio file 113 may be created from the sound picked up by the microphone M. The storage device 110 also stores a loudness table 114, which will be described later.

Audio file 113 is an audio file in which audio content is recorded. In one example, the file format of audio file 113 may be a WAVE file format that is commonly used in personal computers. A WAVE file may include a header and audio signal data. The header may include information such as monaural/stereo type, sampling frequency, and quantization bit rate. Note that the file format of audio file 113 is not limited to the WAVE file format. The file format of audio file 113 may be a format other than the WAVE file format, such as AIFF, MP3, AAC, etc.

As an example, consider that the audio processing device C is used for game development. To implement audio in game development, a sound creator creates an audio file, and a programmer uses a game engine to program the created audio so that it is played in accordance with user operations. In the production of audio files, two major tools have come to be used as game development becomes larger and more complex. One is a DAW for creating a variety of audio files, and the other is middleware (audio middleware) that saves the labor of incorporating audio into a game engine. DAW is an abbreviation for Digital Audio Workstation, and is software that allows recording/editing of audio for the purpose of audio production. Middleware is software that plays, processes, and manages audio to be passed to a game engine, and can create audio data to be played by a DAW. An example of such middleware is Wwise manufactured by Audiokinetic. Therefore, the middleware 115 and the DAW 112 are also installed in the storage device 110 of the audio processing device C. CPU 101 can function as a processor that executes middleware and DAW. The DAW and middleware are configured to be able to perform cooperative processing such as passing audio files between them. For example, it is possible to create audio in a DAW, write the audio file, move the audio file from the DAW to middleware, and use the middleware to implement the audio file in a game engine. Furthermore, if it is necessary to adjust the audio that was moved from the DAW to the middleware, the audio file can be moved from the middleware to the DAW again, and the audio file can be adjusted using the DAW.

The number of audio files implemented in a game can reach tens of thousands. Because there is variation in the volume of the initial audio files delivered, it is necessary to adjust the volume (signal level adjustment) for each audio file. However, adjusting the volume of that many audio files one by one would require an enormous amount of work.

The sounds used in games can include a wide variety of sounds, such as character dialogue, situation explanations (success, failure, etc.), sound effects, footsteps, explosions, environmental sounds, background music, etc. The inventors have noted that there is a relationship between the content of such sounds and appropriate volume values. In this embodiment, the volume value is determined according to the content of the sound in the audio file.

In the field of game development, each audio file is generally named so that the attributes of the audio can be determined to some extent. An "attribute" is something that can identify the content of the audio, such as a character name, scene name, action name, or dialogue. A file name may contain multiple attribute information, such as "character name + action name." In game development, naming rules for audio files are usually established so that they are not significantly changed during development. Therefore, it is possible to identify the content of the audio from the file name of the audio file, and to determine the volume value according to the identified content of the audio.

In this embodiment, when adjusting the volume of the sound recorded in each audio file on the DAW side, a volume table in which a target volume value is described is referenced. Here, the volume value is described. In this embodiment, a loudness value that takes into account the human hearing characteristics is used as a measure (index) of the volume value. The loudness value is expressed in units of, for example, LUFS (Loudness Units Full Scale) or LKFS (Loudness K-Weighted Full Scale). Therefore, in this embodiment, when adjusting the loudness of the sound recorded in each audio file on the DAW side, the loudness table 114 in which the target loudness value is described is referenced as the volume table. The loudness table 114 is a lookup table in which the correspondence between a character string that can be a part of the file name of the audio file and a loudness value (target loudness value) that is a volume value is described. The loudness table may be called a "loudness list". FIG. 2 shows an example of the structure of the loudness table 114. The loudness table 114 includes a pair of a character string (registered character string) and a loudness value (target loudness value) as one record. The registered character string described in each record is a character string that can become part of the file name of an audio file. Note that the present invention is not limited to using loudness values as a scale for volume values. A scale other than loudness values (e.g., RMS) may be used as a scale for volume values.

3 shows an example of a loudness value setting screen 30 displayed on the display D when the CPU 101 executes the DAW 112. The CPU 101 as a display control unit displays each record of the loudness table in an editable manner on the setting screen 30 on the display D. A user can add and register a record to the loudness table 114 via this setting screen 30. The number of records registered in the loudness table 114 is displayed in a record number display window 31. A record can be added in response to pressing the add button 32 (clicking with a mouse, tapping via a touch panel). The contents of each registered record are displayed in the list 35. Each record in the list 35 has a "Search" and "Value" column. The "Search" column displays the registered character string to be searched, and the "Value" column displays the loudness value corresponding to the registered character string. If all records cannot be displayed in the display area of the list 35, they can be scrolled using a scroll bar 36.

The user can specify a loudness measurement method in the loudness setting field 33. Examples of loudness measurement methods include MaxMomentary, MaxShort-Term, and Integrated. In the loudness setting field 33, one of these can be selected. MaxMomentary refers to a method in which loudness is calculated in each of multiple measurement windows (400 msec length) obtained by sliding the audio waveform on the time axis for a predetermined period of time, and the maximum value among them is adopted as the loudness value. MaxShort-Term refers to a method in which loudness is calculated in each of multiple measurement windows (3 seconds long) obtained by sliding a predetermined time on the time axis, and the maximum value among them is adopted as the loudness value. Integrated refers to something that measures the loudness of the entire sound source (the entire audio of one audio file). In the example of FIG. 3, MaxMomentary is selected. Furthermore, instead of the specific measurement window length described above, it may be possible to specify an arbitrary measurement window length.

Dynamic range compression may optionally be performed before loudness adjustment is performed on the audio of the audio file. There may be large variations in playback volume between audio files. If the volume of the sound source is not adjusted, the reproduction volume of certain audio may be too low or too loud, making it difficult to hear. Therefore, it is necessary to equalize the signal levels of each sound source. Dynamic range compression is performed to equalize signal levels between such sounds. Dynamic range compression generally includes processing that suppresses portions of signal levels that include peaks and increases portions of low signal levels. However, it is not sufficient to simply keep the signal level constant. In the case of human speech, if there is no intonation to some extent, the sound will feel compressed. Therefore, in dynamic range compression, it is necessary to appropriately set a signal level threshold for determining the compression target.

Dynamic range compression can also be performed manually by the user by moving any adjustment point among a plurality of adjustment points arranged on the envelope (manual compression). However, it takes a lot of effort to manually compress all audio. Therefore, it is also possible to automatically perform dynamic range compression on the entire audio file. Automatically performing dynamic range compression is herein referred to as "auto compression."

Automatic compression may include, for example, the following process. The audio signal of the target audio file is composed of multiple frames. First, the envelope of the audio signal is obtained. Next, the peak value of the envelope for each frame is detected, and the average value (first average value) of the detected peak values for each frame is calculated. Next, peak values higher than the first average value are detected, and their average value (second average value) is calculated. Then, the envelope is adjusted so that at least some of the peak values higher than the second average value are suppressed. For example, peak values higher than the second average value are further detected, and their average value (third average value) is calculated. Furthermore, peak values higher than the third average value are detected, and adjustments are made so that they approach the third average value. Note that such an automatic compression processing method is merely one example, and may be realized by other processing methods.

In this embodiment, the user can specify whether or not to apply automatic compression to all audio files stored in the working folder of the storage device 110. The setting screen 30 is provided with an automatic compression setting field 34 for instructing the execution of automatic compression. The automatic compression setting field 34 may include, for example, a radio button or a check box, and the execution of automatic compression is specified by selecting the field (ON). In the example of FIG. 3, the automatic compression setting field 34 is set ON by a radio button. In this case, loudness adjustment is performed after dynamic range compression is performed on the audio of the audio file.

The setting screen 30 further includes a file name display column 37 that displays the file names of one or more audio files that are stored in the work folder of the storage device 110 and are subject to loudness adjustment.

Next, management of audio files on the middleware 115 will be explained. An individual audio file may contain one or more audio materials (portions of recorded sound). One audio material is also called a "track". In middleware, audio (tracks) are classified in a hierarchical structure. FIG. 4 is a conceptual diagram showing an example of a hierarchical structure H of audio managed on middleware. Tracks are input to the input terminal, all tracks are collected into a master track MS, and output from the output terminal. In other words, all the sounds played in the game ultimately pass through the master track MS and are output. Effect E and/or volume adjustment V is applied to tracks IN1, IN2, and IN3 input to the input terminals individually. The hierarchical structure H may include bus tracks. A bus truck is a truck that is a collection of one or more trucks. In FIG. 4, bus track B1 combines track IN1 and track IN2 into one track and outputs it to master track MS. By using a bus, you can apply effects and/or volume adjustments to multiple tracks at once. Furthermore, the hierarchical structure H may include auxiliary tracks. An auxiliary track is a copy of a certain track that is sent. In FIG. 4, the auxiliary track AUX inputs the track IN3 and outputs it to the bus track B2. The bus track B2 inputs the track IN3 as well as the auxiliary track AUX. The auxiliary track is used, for example, when a track to which an effect (reverb, delay, etc.) has been applied and a track to which no effect has been applied coexist.

A user can design a hierarchical structure via a hierarchical structure setting screen (not shown), and determine which input terminal in the hierarchical structure to place for any track of an audio file registered on the middleware. This determines routing indicating the path from the input to the output in the hierarchical structure for each track. In this way, a volume adjustment unit is provided for each level of the hierarchical structure H, and the track can be subjected to volume adjustment every time it passes through each volume adjustment unit. A user can determine the effect to be applied for each level of the designed hierarchical structure and the volume value of the volume adjustment unit. The total change in the volume value of the audio on the middleware is calculated by summing up the volume values of the volume adjustment units on the path. For example, as shown in FIG. 4, it is assumed that the loudness value of the master track is set to -6 dB, the loudness value of the bass track B1 is set to +4 dB, the loudness value of the track IN1 is set to +2 dB, and the loudness value of the track IN2 is set to -4 dB. In this case, the total change in the loudness value of the track IN1 on the middleware is
(-6dB) + (+4dB) + (+2dB) = 0dB
The total change in loudness value of track IN2 on the middleware is
(-6dB) + (+4dB) + (-4dB) = -6dB
It becomes.

By designing a hierarchical structure in the middleware as described above, routing information is created that indicates the path along which the audio recorded in the audio file reaches the output. The routing information may include information on the path along which the audio recorded in the audio file reaches the output (hereinafter referred to as "routing"), and information on the volume value at each volume adjustment unit along the path.
In the above explanation, an example of one type of hierarchical structure H has been shown, but a configuration may be adopted in which a plurality of types of hierarchical structures can be constructed on the middleware.

As described above, loudness adjustment on the DAW is performed using a loudness value determined according to the file name of the audio file with reference to the loudness table. However, conventionally, the DAW side has not been able to grasp the volume adjustment history of each audio on the middleware. Therefore, on the DAW side, loudness adjustment was performed without considering the volume adjustment history in middleware.

In this embodiment, the CPU 101 acquires the amount of change in the volume value in the routing of the audio on the middleware based on the routing information created by the middleware on the DAW. Then, the CPU 101 adjusts the volume on the DAW based on the amount of change acquired. Fig. 5 shows an example of a setting screen 40 for acquiring the amount of change in the volume value on the middleware, which is displayed on the display D when the CPU 101 executes the DAW 112.
On the setting screen 40, a search check box 41 is a box for setting a middleware connection.
As described above, multiple types of hierarchical structures may be built on the middleware. The search path setting field 44 is a field for setting which hierarchical structure is to be searched when multiple types of hierarchical structures are built on the middleware.
The detection path setting field 45 is a field for setting a layer to be given priority when a plurality of audio files with the same file name are found in the search path.
The excluded routing setting field 47 is a field for setting routing (excluded routing) to be excluded from the calculation when determining the total change in volume value on the middleware, and the user can specify information identifying the excluded routing in the routing specification field 48.
The additive routing setting field 49 is a field for setting a routing (additive routing) in a hierarchical structure other than the hierarchical structure set in the search path setting field 44, which should be added to the calculation target when calculating the total change amount of the volume value on the middleware, and the user can specify information for specifying the additive routing in the routing specification field 50. In addition to volume adjustment within the routing, there are several exceptions, such as volume adjustment using an effector such as upmix (e.g., 2.0ch to 4.0ch), downmix (e.g., 4.0ch to 2.0ch), gain, limiter, and comp. The additive routing setting field 49 is prepared for such exceptional cases.
If the number of routing specification fields 48, 50 increases and they cannot all be displayed in the display area, a scroll bar 51 can be used to scroll through them.

FIG. 6 shows a flowchart of the audio processing method in the audio processing device C. A program corresponding to this flowchart is included in the DAW 112 and is executed by the CPU 101 while the DAW 112 is being executed.

In step S11, the CPU 101 obtains the audio file registered in the middleware and stores it in a specified working folder in the storage device 110.

In step S12, the CPU 101 performs automatic compression on the acquired audio file. However, this step S12 is an option when the automatic compression setting field 34 on the setting screen 30 shown in FIG. 3 is selected. If the automatic compression setting field 34 is not selected, step S12 is skipped.

In step S13, the CPU 101 searches the loudness table 114 for a record having a registered character string that partially matches the file name for the acquired audio file. The CPU 101 obtains the loudness value R described in the record obtained by this search.

In step S14, the CPU 101 determines whether there is an audio file below the search path set in the search path setting field 44 (that is, in at least one of the routings in the hierarchical structure to be searched). If there is an audio file, the process advances to step S15. In step S15, the CPU 101 determines whether there are multiple audio files under the search path. If there are multiple audio files, the process proceeds to step S16, and if there is only one audio file, the process proceeds to step S18.

In step S16, the CPU 101 determines whether or not there is an audio file in the priority hierarchy set in the detection path setting field 45. If there is an audio file in the priority hierarchy set in the detection path setting field 45, the process proceeds to step S18; if not, the process proceeds to step S17. In step S17, the CPU 101 obtains routing information for the path of the first audio file found among the multiple audio files identified in step S15.

In step S18, the CPU 101 acquires all routing information on the middleware, including the routing information acquired in step S17. As described above, the routing information may include information on the path along which the audio recorded in the audio file set in the middleware is output, and information on the volume value at each volume adjustment unit on the path.

In step S19, the CPU 101 excludes the non-target routings set in the non-target routing setting field 47 from the calculations made when calculating the total change in loudness value in step S21 described below.

Only one routing should be added to the calculation target when calculating the total amount of change in the volume value on the middleware. For example, when applying effects, it is possible to set multiple routings. Figure 4 shows the routing in which track IN3 passes through the auxiliary track AUX and is input to bus B2 with an effect applied, and the track IN3 is input to bus B2 in the original (dry) state. An example where routing coexists is described. Of these two routings, the one in which the volume value should be obtained is the one in which the track IN3 is input to the bus B2 in the original sound (dry) state. Therefore, in this case, the routing in which the track IN3 is input to the bus B2 via the auxiliary track AUX should be set to non-target routing. However, there may be cases where the user forgets to set such a route to a non-targeted route. Therefore, in step S20, the CPU 101 determines whether there is only one route to be searched. If there is not one route to be searched, the process advances to step S26, where the CPU 101 outputs an error and ends the process. If there is only one route to be searched, the process advances to step S21. In step S21, the CPU 101 calculates the total amount of change T in the loudness value on the middleware. The total amount of change T is the amount of change in loudness value in the routing set in the search path setting field 44, plus the amount of change in the loudness value in the routing set in the routing setting field 49, to the amount of change in the loudness value in the routing where the audio file is located among the search paths set in the search path setting field 44. Obtained by adding. However, if in step S14 there is no audio file below the search path set in the search path setting field 44, the amount of change in the loudness value of the search path is set to 0 in step S25.

In step S22, the CPU 101 calculates a final loudness value FR (final volume value). The final loudness value FR is obtained by calculating the difference between the loudness value R obtained from the loudness table in step S13 and the total amount of change T in the loudness values obtained in step S21.

In step S23, the CPU 101 adjusts the loudness of the audio recorded in the target audio file using the final loudness value FR calculated in step S22. The loudness adjustment is performed, for example, by measuring the loudness value of the audio of the target audio file (if step S12 is executed, the audio of the target audio file after automatic compression has been executed) according to the loudness measurement method specified in the loudness setting field 33, and adjusting the gain value of the audio based on the measurement result so that the loudness value becomes the target loudness value.

In step S24, the CPU 101 as a display control unit displays a first waveform (waveform before loudness adjustment), which is the waveform of the audio of the audio file acquired in step S11 or the audio that has been automatically compressed in step S12, and a second waveform, which is the waveform of the audio after loudness adjustment, in the display area of the display D. Examples of waveform display will be described later.

If there are other unprocessed audio files registered in the middleware, the process returns to step S11, and the process is repeated for the next audio file. Therefore, if there are multiple audio files registered in the middleware, the search in step S13 through loudness adjustment in step S23 are performed sequentially for each of the multiple audio files.

Figure 7 shows an example of the waveform display in step S24. Here, an example of the waveform display when three audio files are processed is shown. The displayed waveform is a time domain waveform. Therefore, the horizontal axis of the waveform is the time axis, and the vertical axis indicates the signal level. In Figure 7, in the upper part of the display area, a waveform W11 after automatic compression (before loudness adjustment) of the audio of the first audio file, a waveform W12 after automatic compression (before loudness adjustment) of the audio of the second audio file, and a waveform W13 after automatic compression (before loudness adjustment) of the audio of the third audio file are arranged side by side along the time axis direction. Each of the waveforms W11, W12, and W13 may display a plurality of adjustment points P that are discretely arranged on the envelope obtained in the automatic compression in order to adjust the signal level. The user can manually adjust the signal level at any position by, for example, dragging an arbitrary adjustment point with the mouse.

In FIG. 7, the lower part of the display area shows a waveform W21 of the first audio file after loudness adjustment, a waveform W22 of the second audio file after loudness adjustment, and a waveform W23 of the third audio file after loudness adjustment, arranged side by side along the time axis. Each waveform after loudness adjustment is obtained by writing the audio that has been loudness adjusted in step S23 to a new file.

Also, information on the loudness value of the audio file that is the target of waveform display can be displayed here. For example, the file name of the audio file of each waveform, the total amount of change T in the loudness value on the middleware, the loudness value R obtained by searching the loudness table, and the final loudness value FR are displayed. This allows the user to understand how much the volume of each audio file has been adjusted in the middleware and DAW. Further, according to the present embodiment, the volume can be adjusted on the DAW in consideration of the volume adjustment history in the middleware. Note that the above-described display manner of the information on the waveform and loudness value is merely an example, and other display manners may be adopted.

(Other examples)
As shown in FIG. 2, the plurality of records in the loudness table 114 are grouped based on the commonality of the prefixes of the registered character strings. The prefix can be a string of characters that represents an attribute of the audio as defined by a naming convention. In that case, the fact that the prefixes are common means that the audio attributes are common. For example, the prefix "vo" represents the character's voice, the prefix "atk" represents the shout during an attack, and so on. In the example in Figure 2, the multiple records include group 1 with the prefix "vo_", group 2 with the prefix "vo_atk", group 3 with the prefix "vo_dmg", and group 3 with the prefix "vo_move". It is classified into Group 4, which has the prefix "vo_cmm", and Group 5, which has the prefix "vo_cmm".

In step S13, the CPU 101 searches the loudness table 114 for a prefix that partially matches the file name of the target audio file, identifies a group, and selects a registered character string that partially matches the file name from among the identified groups. search for. In the example of FIG. 2, each group includes a representative record in which a pair of a character string consisting only of a prefix and a loudness value is described. A representative record exists in the first row of each group.

In step S13, if the search result shows that no registered character string that partially matches the file name is found in the specified group other than the representative record, in step S23, loudness adjustment is performed using the loudness value described in the representative record. A specific example is described below. In step S13, first, a prefix that partially matches the file name of the target audio file is searched from the representative record in the first row of each group. For example, assume that the prefix "vo_atk" partially matches the file name of the target audio file. In this case, the group to be searched is limited to group 2. Then, a registered character string that partially matches the file name is searched for in group 2. In addition to the representative record, group 2 includes records with registered character strings such as "vo_atk_charge" and "vo_atk_s". If no registered character string that partially matches the file name is found in group 2 other than the representative record, in step S23, loudness adjustment is performed using the loudness value "-21" corresponding to the representative record (registered character string "vo_atk").

According to the above processing, the search range of the loudness table can be limited, so the search speed is improved.

Note that in the example of FIG. 2, group 1 having a prefix of "vo_" is positioned as a layer above other groups having a prefix of "vo_" and another character string following it. If the only prefix that partially matches the file name of the target audio file is "vo_", the loudness value will be "-23", which corresponds to "vo_" in group 1.

According to the embodiment described above, a search is performed for a record having a registered character string that partially matches the file name of the audio file from a volume table (loudness table) that contains a pair of a character string and a volume value (loudness value) as one record. Then, the volume of the audio recorded in the audio file is adjusted by the volume value described in the record obtained by the search. If the volume table is created in advance, there is no need to perform separate settings for volume adjustment. In addition, when multiple audio files are processed, the above search and volume adjustment are performed sequentially for each audio file. In this way, volume adjustment is performed automatically for multiple audio files. In addition, the number of records included in the volume table is significantly smaller than the number of multiple audio files. Therefore, according to this embodiment, the user's work hours are significantly reduced compared to the conventional technology in which the audio of each of the multiple audio files was adjusted one by one. Furthermore, according to this embodiment, as described above, the user can grasp how much volume adjustment has been performed on each audio file in the middleware and DAW. In addition, according to this embodiment, volume adjustment can be performed on the DAW taking into account the volume adjustment history in the middleware.

It is not essential that the loudness table 114 is stored in the storage device 110. For example, the loudness table 114 may be stored in an external device (e.g., server A in FIG. 1) connected via a network N, and the audio processing device C may refer to the loudness table 114 stored in the external device via the network N.

The present invention can also be implemented by causing a computer to execute a program for executing each step of the audio processing method described in the above embodiment.

The invention is not limited to the above-described embodiments, and various modifications and changes can be made within the scope of the invention.

A: Server, C: Audio processing device, D: Display, K: Input device, 101: CPU, 112: DAW, 114: Loudness table, 115: Middleware

Claims (9)

An audio processing device that processes audio,
a storage unit that stores middleware that is software for processing audio and a digital audio workstation (DAW) that is software different from the middleware for processing audio;
a processor that executes the middleware and the DAW;
has
The processor, on the DAW,
obtaining the amount of change in the volume value of the audio on the middleware based on routing information indicating the path of the audio recorded in the audio file set on the middleware until it reaches the output;
A voice processing device characterized by: The processor, on the DAW, further
adjusting the volume of the audio based on the obtained amount of change;
The audio processing device according to claim 1, characterized in that: In the middleware, the audio is classified in a hierarchical structure, and a volume adjustment section is provided for each layer,
The routing information includes information on the path of the audio and information on volume values at each volume adjustment unit on the path,
The processor, on the DAW,
calculating the total amount of change in the volume value of the audio on the middleware by summing the volume values in each volume adjustment unit on the path;
adjusting the volume of the audio based on the calculated total amount of change;
The audio processing device according to claim 2, characterized in that: The processor, on the DAW,
Searching for a record having a character string that partially matches the file name of the audio file as a registered character string from a volume table that includes a pair of a character string and a volume value as one record;
adjusting the volume of the audio recorded in the audio file using a final volume value that is the difference between the volume value described in the record obtained by the search and the total amount of change;
The audio processing device according to claim 3, characterized in that: 5. The audio processing device according to claim 4, further comprising a setting means for setting a non-target routing, which is a routing to be excluded from calculation targets when calculating the total amount of change. The audio according to claim 4, further comprising a setting means for setting an addition routing that is a routing in a hierarchical structure different from the hierarchical structure, which is to be added to the calculation target when calculating the total amount of change. Processing equipment. The audio processing device according to claim 1, characterized in that the volume value scale is a loudness value. An audio processing method executed by an audio processing device having a storage unit that stores middleware, which is software for processing audio, and a digital audio workstation (DAW), which is software different from the middleware and is used for processing audio, and a processor that executes the middleware and the DAW, comprising:
The processor, during execution of the DAW,
obtaining routing information indicating a path along which the audio recorded in the audio file set by the middleware reaches an output;
acquiring a change in a volume value of the audio on the middleware;
13. A method for processing audio, comprising: A program for causing a computer to execute each step of the audio processing method described in claim 8.

Images