Voice processing method, voice processing device, and program
The sound processing method automates the process of adjusting audio volumes for multilingual game development, reducing manual effort by searching for corresponding audio files and displaying their waveforms in parallel, thereby ensuring consistent playback across languages.
Patent Information
- Application Number
- JP2023179015
- Authority / Receiving Office
- JP · JP
- Patent Type
- Applications
- Current Assignee / Owner
- Filing Date
- 2023-10-17
- Publication Date
- 2025-04-30
- Estimated Expiration
- 2043-10-17
AI Technical Summary
In the field of game development, making audio content multilingual requires significant effort to adjust signal levels so that volume variations between languages are minimized.
A sound processing method that automatically searches for corresponding audio files in different languages, displays their waveforms in parallel, and adjusts the volume of the secondary language audio to match the primary language audio, reducing the manual effort required.
This method significantly reduces the effort needed to make audio content multilingual by automating the process of finding corresponding audio files and adjusting their volumes, ensuring consistent playback across languages.
Smart Images

Figure 2025068905000001_ABST
Abstract
Description
[Technical field]
[0001] The present invention relates to a voice processing method, a voice processing device, and a program. [Background technology]
[0002] Equipment and services that provide audio may be required to support multiple languages (see, for example, Patent Document 1). In addition, in services that play various sounds, if there is variation in the playback volume of each audio material, it may be difficult for listeners to hear. For this reason, it is necessary to make the playback volume of each audio material constant. Loudness values are used as a measure to reduce the variation in loudness (see, for example, Patent Document 2). [Prior art documents] [Patent documents]
[0003] [Patent Document 1] JP 2018-060043 A [Patent Document 2] Patent Publication No. 2022-042892 Summary of the Invention [Problem to be solved by the invention]
[0004] For example, in the field of game development, when a game is made multilingual, it is necessary to adjust the signal level so that the volume does not differ significantly between languages. Because a large amount of audio content is installed in a game, it takes a lot of effort to adjust the volume of each audio content so that it does not differ between languages.
[0005] An object of the present invention is to provide an advantageous technique for reducing the effort required to translate audio content into multiple languages. [Means for solving the problem]
[0006] According to one aspect of the present invention, there is provided an audio processing method comprising: an acquisition step of acquiring a first folder path that is a folder path of a first folder and a second folder path that is a folder path of a second folder different from the first folder; a search step of searching, based on a file name, from the second folder identified by the second folder path for a second audio file corresponding to a first audio file stored in the first folder identified by the first folder path; a display control step of displaying a first waveform that is a waveform of the audio of the first audio file and a second waveform that is a waveform of the audio of the second audio file in a display area of a display; and a processing step of processing the audio of the second audio file so that a volume of the audio of the second audio file approaches a volume of the audio of the first audio file, wherein the display area includes a first display area and a second display area arranged in parallel with the first display area in a second direction perpendicular to a first direction parallel to a time axis, and the display control step includes a first step of displaying the first waveform in the first display area, and a second step of displaying the second waveform in the second display area so as not to overlap with the first waveform in the first direction. Effect of the Invention
[0007] According to the present invention, it is possible to provide an advantageous technique for reducing the effort required for multilingualizing audio content. [Brief description of the drawings]
[0008] [Figure 1] 1 is a block diagram showing a configuration of a sound processing device according to an embodiment. [Diagram 2] 1 is a flow chart of an audio processing method. [Diagram 3] 1 is a flow chart of an audio processing method. [Figure 4] FIG. [Diagram 5] FIG. 13 is a diagram showing an example of a display of an audio waveform. [Figure 6] FIG. 13 is a diagram showing an example of a display of an audio waveform. [Figure 7]FIG. 13 is a diagram showing an example of a display of an audio waveform. DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS
[0009] Hereinafter, the embodiments will be described in detail with reference to the accompanying drawings. Note that the following embodiments do not limit the invention according to the claims, and not all combinations of features described in the embodiments are essential to the invention. Two or more features among the multiple features described in the embodiments may be arbitrarily combined. In addition, the same reference numbers are used for the same or similar configurations, and duplicated descriptions are omitted.
[0010] 1 shows a block diagram illustrating a 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 processes such as adjusting the signal level of the audio signal.
[0011] The voice processing device C may be a computer device such as a personal computer or a workstation. The voice 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 voice processing device C also includes an audio interface (I / F) 104. A microphone M and a speaker S may be connected to the audio interface 104. An external storage device 110 is connected to the voice processing device C via an interface (I / F) 105. The external storage device 110 may be, for example, a hard disk drive (HDD), a solid state drive (SSD), or a combination thereof. The external storage device 110 may be configured as a secondary storage device inside the voice processing device C. The network interface 106 is connected to a network N to perform communication. The voice processing device C may be connected to a server A via the network N so as to be able to communicate with the server A.
[0012] An input device 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.
[0013] A boot program for starting up the audio processing device C is stored in the ROM 103. As shown in Fig. 1, an operating system (OS) 111, a signal processing program 112 for performing audio signal processing, and one or more audio files 113 may be installed in the external storage device 110. The audio file 113 may be supplied from an external device such as a server A via a network N, or may be supplied from a medium contained in an external media device F. Alternatively, the audio file 113 may be created from sound picked up by a microphone M.
[0014] The audio file 113 is an audio file in which audio content is recorded. In one example, the file format of the audio file 113 may be a WAVE file format that is generally used in personal computers. The 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. The file format of the audio file 113 is not limited to the WAVE file format. The file format of the audio file 113 may be a format other than the WAVE file format, such as AIFF, MP3, AAC, etc.
[0015] For example, in the field of game development, when a game is made multilingual, it is necessary to adjust the signal level so that the volume does not differ greatly between languages. This embodiment facilitates the adjustment of the volume of each audio content so that it does not differ between languages. In this embodiment, an audio file in which an audio in a second language corresponding to an audio file in which an audio in a first language is recorded is automatically searched for, and audio waveforms of both audio files are displayed in the display area of the display D, and audio processing is performed. The second language is a language different from the first language. The first language may be, for example, Japanese, and the second language may be, for example, English. Of course, the first language and the second language are not limited to a specific language and may be any language. The audio in the second language is a dubbed audio for the audio in the first language.
[0016] 2 shows a flowchart of a voice processing method in the voice processing device C. A program corresponding to this flowchart is included in the signal processing program 112 and is executed by the CPU 101.
[0017] In step S11, CPU 101 acquires a folder path (first folder path) of a first folder in which one or more audio files (first audio files) in which audio in a first language is recorded are stored. The "folder path" refers to information indicating the location of a specific folder (directory) on a file system. The first folder path is specified by the user via a setting screen 40 displayed on display D as shown in FIG. 4. The user can specify the first folder path in a first language folder path specification field 41 on the setting screen 40. In one example, when the user clicks the first language folder path specification field 41 with the mouse, a folder selection dialog is displayed. The user can specify the first folder path via the dialog.
[0018] In step S12, CPU 101 acquires a folder path (second folder path) of a second folder in which one or more audio files (second audio files) in which audio in a second language is recorded are stored. The second folder path is specified by the user via setting screen 40, similar to the specification of the first folder path. The user can specify the second folder path in second language folder path specification field 42 on setting screen 40. In one example, when the user clicks with the mouse on second language folder path specification field 42, a folder selection dialog is displayed. The user can specify the second folder path via the dialog.
[0019] In step S2, CPU 101 searches the second folder specified by the second folder path for a second audio file corresponding to the first audio file stored in the first folder specified by the first folder path. In one example, the search is performed based on the file name. For example, the first audio file and the second audio file, which is a version of the first audio file in another language, are given the same file name. In this case, the second folder is searched for a second audio file having the same file name as the first audio file stored in the first folder.
[0020] In step S3, the CPU 101 performs display control to display a first waveform, which is the waveform of the sound of the first audio file, and a second waveform, which is the waveform of the sound of the second audio file obtained by the search, in the display area of the display D. FIG. 5 shows an example of display of the first and second waveforms. 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. Here, the direction parallel to the time axis is the first direction, and the direction perpendicular to the first direction is the second direction. The display area R of the display D includes a first display area R1 and a second display area R2 arranged in parallel with the first display area R1 in the second direction. Note that the borders of the display area R, the first display area R1, and the second display area R2 may or may not be displayed. The CPU 101 as a display control means displays a first waveform W1, which is the waveform of, for example, the entire section of the sound of the first audio file, in the first display area R1 (first step). Furthermore, the CPU 101 displays in the second display area R2 a second waveform W2, which is, for example, a waveform of the entire section of the audio of the second audio file (second step). At this time, the CPU 101 controls the waveform display on the display D so that the second waveform W2 is displayed in the second display area R2 so as not to overlap with the first waveform W1 in the first direction. For example, as shown in Fig. 5, the second waveform W2 is displayed in the second display area R2 with a predetermined time interval G from the end of the first waveform W1. The predetermined time interval G can be set to a time specified by the user in the waveform interval input field 43 of the setting screen 40 in Fig. 4 (see step S13 in Fig. 3 described later).
[0021] In step S4, the CPU 101 processes the audio of the second audio file so that the volume of the audio of the second audio file approaches the volume of the first audio file (volume adjustment process). The recording environment of the audio of the first language is often not the same as the recording environment of the audio of the second language, which is a dubbed audio. Therefore, in general, the first audio file in which the audio of the first language is recorded and the second audio file in which the audio of the second language is recorded in an environment different from that of the first audio file are provided on different media. The difference in recording environment appears as a difference in the perceived playback volume. Therefore, when the audio of the first audio file and the audio of the second audio file are played back at the same volume, undesirable variations in the perceived playback volume may occur. In the volume adjustment process in this embodiment, the audio of the first language is considered to be the main audio and the audio of the second language is considered to be the secondary audio, and the volume level of the secondary audio is normalized with respect to the volume level of the main audio.
[0022] In this specification, the term "volume" should be taken in a broad sense. For example, RMS (Root Mean Square) may be used as a measure of volume. Alternatively, loudness that takes into account human hearing characteristics may be used as a measure (index) of volume. The loudness value is expressed in units of LUFS (Loudness Units Full Scale) or LKFS (Loudness K-Weighted Full Scale). In this embodiment, it is assumed that loudness is used as a measure of volume. For example, in the field of game development, the volume of audio content is specified to be within a LUFS level defined by a loudness standard or the like. Here, it is assumed that the volume of the audio of the first audio file is adjusted according to such a specification.
[0023] In addition, in the setting screen 40 of FIG. 4, the user can specify a loudness measurement method in a loudness setting field 45. As the loudness measurement method, for example, any one of MaxMomentary, MaxShort-Term, and Integrated can be selected. MaxMomentary refers to performing loudness calculations in each of a plurality of measurement windows (400 msec long) obtained by sliding a predetermined time on the time axis, and adopting the maximum value among them as the loudness value. MaxShort-Term refers to performing loudness calculations in each of a plurality of measurement windows (3 sec long) obtained by sliding a predetermined time on the time axis, and adopting the maximum value among them as the loudness value. Integrated refers to measuring the loudness of the entire sound source. In the example of FIG. 4, MaxMomentary is selected. Furthermore, instead of the specific measurement window length described above, any measurement window length may be specified. In addition, in the setting screen 40 of FIG. 4, the user can specify a target (target) for loudness adjustment in a target loudness setting field 46. The options include the first language audio (audio of the first audio file), a loudness value defined by a loudness standard, etc. In the example of FIG. 4, the first language audio is specified. In this case, in the volume adjustment process (step S4), the loudness of the audio of the second language, which is the secondary audio, is normalized based on the loudness of the audio of the first language, which is the primary audio. Details of loudness measurement are described in, for example, JP 2022-042892 A (Patent Document 2).
[0024] According to the above processing, a second audio file in which audio in a second language is recorded and which corresponds to a first audio file in which audio in a first language is recorded is automatically searched for, a first waveform which is the waveform of the audio of the first audio file and a second waveform which is the waveform of the audio of the second audio file are displayed side by side in the display area, and the loudness value of the audio of the second audio file is adjusted so that it is equivalent to the loudness value of the audio of the first audio file.
[0025] Before loudness adjustment is applied to the audio of the second audio file, dynamic compression may be performed as an option. There may be a large variation in the playback volume between audio files. If the volume of the audio sources is not adjusted, the playback volume of one audio may be too low or too high, making it difficult to hear. Therefore, it is necessary to align the signal levels of each audio source. Dynamic range compression is performed to align the signal levels between such audio sources to a constant level. Dynamic range compression generally involves a process that suppresses parts that include peaks in the signal level and increases parts with low signal levels. However, it is not enough to simply make the signal level constant. In the case of human speech, if there is a certain degree of flatness, the compressed feeling will be strong. Therefore, in dynamic range compression, it is necessary to appropriately set the signal level threshold to determine the compression target.
[0026] Dynamic range compression can also be performed manually by the user by moving any of the multiple adjustment points placed on the envelope (manual compression). However, performing manual compression on all audio is a huge effort. For this reason, it is also possible to perform dynamic range compression automatically on the entire audio file. Automatic dynamic range compression will be referred to here as "auto compression."
[0027] Automatic compression may include, for example, the following process. The audio signal of the target audio file is made up of multiple frames. First, the envelope of the audio signal is obtained. Next, peak values of the envelope for each frame are detected, and an average value (first average value) of the peak values detected for each frame is calculated. Next, peak values higher than the first average value are detected, and an average value (second average value) of these peak values is calculated. Then, the envelope is adjusted so that at least some of the peak values higher than the second average value are suppressed. Note that such an automatic compression processing method is merely one example, and may be realized by other processing methods.
[0028] In this embodiment, the user can specify whether or not to apply automatic compression to the second audio file. The setting screen 40 in FIG. 4 is provided with an automatic compression setting field 44 for instructing the execution of automatic compression on the second audio file. For example, a check box is prepared in the automatic compression setting field 44, and the execution of automatic compression is specified by checking the check box. In the example of FIG. 4, the automatic compression setting field 44 is checked. In this case, in the volume adjustment process in step S4, dynamic range compression is performed on the audio of the second audio file, and then, based on the loudness value of the audio of the first audio file, the loudness value of the audio of the multiple second audio files that have been subjected to dynamic range compression is adjusted.
[0029] The display of the second waveform W2 in the second display area R2 may be updated with the waveform of the audio of the second audio file processed in step S4.
[0030] In the above, in order to simplify the explanation for understanding the outline of the present invention, one first audio file and one corresponding second audio file have been described. However, in reality, a plurality of first audio files may be stored in the first folder, and a plurality of second audio files may be stored in the second folder. In the following, an audio processing method for automatically processing a plurality of first audio files and a plurality of second audio files will be described.
[0031] Fig. 3 shows a flowchart of an audio processing method for processing a plurality of audio files in the audio processing device C. A program corresponding to this flowchart is included in the signal processing program 112, and is executed by the CPU 101. In Fig. 3, the same processing blocks as those in the flowchart in Fig. 2 are given the same reference symbols. Descriptions of those processing contents will be omitted or simplified as appropriate.
[0032] In step S11, CPU 101 acquires a folder path (first folder path) of a first folder in which one or more audio files (first audio files) in which audio in a first language is recorded are stored. The first folder path is specified by the user via a setting screen 40 displayed on display D as shown in Fig. 4. The user can specify the first folder path in a first language folder path specification field 41 on the setting screen 40.
[0033] In step S12, CPU 101 acquires a folder path (second folder path) of a second folder in which one or more audio files (second audio files) in which audio in a second language is recorded are stored. The second folder path is designated by the user via setting screen 40, similar to the designation of the first folder path. The user can designate the second folder path in second language folder path designation field 42 on setting screen 40.
[0034] In step S13, the CPU 101 acquires the time interval (waveform interval) between the first and second waveforms based on the time specified by the user in the waveform interval input field 43 on the setting screen 40 in FIG.
[0035] In step S2, CPU 101 searches for a second audio file corresponding to the first audio file stored in the first folder specified by the first folder path from the second folder specified by the second folder path. In the example of FIG. 3, step S2 includes the processes of steps S21 to S23. In step S21, CPU 101 acquires (selects) one of the multiple first audio files from the first folder. In step S22, CPU 101 searches for a second audio file having the same file name as the acquired first audio file from the second folder. If the second audio file having the same file name as the first audio file exists in the second folder, the process proceeds to step S3. If the second audio file having the same file name as the first audio file does not exist in the second folder, the process proceeds to step S23.
[0036] In step S3, CPU 101 performs display control to display a first waveform, which is the waveform of the audio of the first audio file, and a second waveform, which is the waveform of the audio of the second audio file obtained by the search, in the display area of display D. CPU 101 as display control means displays a first waveform W11, which is the waveform of the audio of the first audio file, in a first display area R1, as shown in Fig. 6. CPU 101 also displays a second waveform W21, which is the waveform of the audio of the second audio file, in a second display area R2. At this time, the second waveform W21 is displayed in the second display area R2 with a time interval from the end of the first waveform W11 according to the waveform interval acquired in S13.
[0037] In step S23, the CPU 101 determines whether there is another first audio file in the first folder. If there is another first audio file in the first folder, the process returns to step S21, and the process is repeated for the other first audio file. As shown in FIG. 6, the CPU 101 displays the first waveform W12, which is the waveform of the audio of the first audio file, in the first display area R1. At this time, the first waveform W12 is displayed in the first display area R1 with a time interval according to the waveform interval acquired in S13 from the end of the second waveform W21. The CPU 101 also displays the second waveform W22, which is the waveform of the audio of the second audio file, in the second display area R2. At this time, the second waveform W22 is displayed in the second display area R2 with a time interval according to the waveform interval acquired in S13 from the end of the first waveform W12. FIG. 6 shows a display example in the case where there are three first audio files in the first folder. In this case, the first waveform W13 is displayed in the first display region R1 at a time interval according to the waveform interval obtained in S13 from the end of the second waveform W22, and the second waveform W23 is displayed in the second display region R2 at a time interval according to the waveform interval obtained in S13 from the end of the first waveform W13.
[0038] If it is determined in step S23 that the search and display control for all the first audio files in the first folder is completed, the process proceeds to step S4. In step S4, CPU 101 executes a volume adjustment process for the audio of each of the displayed second audio files.
[0039] In this way, when a plurality of first audio files are stored in the first folder, search and display control are repeatedly executed for each of the plurality of first audio files. By the repeatedly executed display control step (S3), a plurality of first waveforms (W11, W12, W13) and a plurality of second waveforms (W21, W22, W23) are displayed in the first display region R1 and the second display region R2, respectively, so that the first waveforms and the second waveforms are alternately arranged along the time axis.
[0040] As a result, multiple second audio files corresponding to multiple first audio files are automatically searched for, the audio is displayed in parallel, and loudness adjustment is performed. Conventionally, a user had to search for a second audio file corresponding to a first audio file from a file list, and manually specify and display each of the found second audio files. Then, the user had to subjectively adjust the volume by listening to or comparing the displayed first and second waveforms. According to this embodiment, all of these processes are automated, so that the user's workload can be significantly reduced.
[0041] Before loudness adjustment is performed on the audio of the second audio file, dynamic compression may be performed as an option. If the automatic compression setting field 44 in Fig. 4 is checked, in the volume adjustment process of step S4, dynamic range compression (automatic compression) is performed on the audio of the second audio files, and then the loudness values of the audio of the second audio files that have been subjected to dynamic range compression are adjusted based on the loudness values of the audio of the first audio files.
[0042] FIG. 7 shows a modified example of FIG. 6. In FIG. 7, the display area R further includes a third display area R3 arranged in parallel with the first display area R1 and the second display area R2 in the second direction. In the example shown in FIG. 7, the third display area R3 is arranged below the second display area R2. The CPU 101 displays a plurality of third waveforms W31, W32, and W33 obtained by newly writing out the audio of the plurality of second audio files after performing dynamic range compression (automatic compression) and loudness value adjustment in the third display area R3 so as to overlap with the plurality of second waveforms W21, W22, and W23 in the first direction (third step). Also, as shown in FIG. 7, a plurality of adjustment points P may be displayed on each of the plurality of second waveforms W21, W22, and W23, which 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 the position by, for example, dragging any adjustment point with the mouse. When the signal level is manually adjusted and / or the loudness is adjusted again, the waveform in the third display area R3 may be updated to further reflect the results. Note that the above-described display manner of the waveform is merely an example, and other display manners may be adopted.
[0043] 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.
[0044] The invention is not limited to the above-described embodiment, and various modifications and variations are possible within the scope of the gist of the invention. [Explanation of symbols]
[0045] A: server, C: audio processing device, D: display, K: input device, 101: CPU, 112: signal processing program
Claims
1. an acquiring step of acquiring a first folder path which is a folder path of a first folder and a second folder path which is a folder path of a second folder different from the first folder; a searching step of searching, based on a file name, from the second folder specified by the second folder path for a second audio file corresponding to the first audio file stored in the first folder specified by the first folder path; a display control step of displaying a first waveform, which is a waveform of the audio of the first audio file, and a second waveform, which is a waveform of the audio of the second audio file, in a display area of a display; a processing step of processing the audio of the second audio file so that the volume of the audio of the second audio file approaches the volume of the audio of the first audio file; having the display area includes a first display area and a second display area arranged in parallel with the first display area in a second direction perpendicular to a first direction parallel to a time axis, The display control step includes: a first step of displaying the first waveform in the first display area; a second step of displaying the second waveform in the second display area so as not to overlap the first waveform in the first direction; 13. A method for processing audio, comprising:
2. 2. The audio processing method according to claim 1, wherein the second step includes a step of displaying the second waveform in the second display area at a predetermined time interval from an end of the first waveform.
3. 3. The method according to claim 2, further comprising a setting step of setting the predetermined time interval to a time specified by a user.
4. 2. The method of claim 1, wherein the measure of volume is a loudness value.
5. The processing steps include: performing dynamic range compression of the audio of the second audio file; adjusting a loudness value of the audio of the second audio file subjected to the dynamic range compression based on a loudness value of the audio of the first audio file; 5. The method of claim 4, further comprising:
6. 2. The audio processing method of claim 1, further comprising a step of updating the display of the second waveform in the second display area with the waveform of the audio of the second audio file processed in the processing step.
7. the first audio file is an audio file in which audio in a first language is recorded, the second audio file is an audio file in which audio in a second language different from the first language is recorded, the audio in the second language is a dubbed audio for the audio in the first language; 2. The audio processing method according to claim 1.
8. When a plurality of first voice files are stored in the first folder, the search step and the display control step are repeatedly executed for each of the plurality of first voice files; By the repeatedly executed display control step, a plurality of first waveforms and a plurality of second waveforms are displayed in the first display area and the second display area, respectively, such that the first waveforms and the second waveforms are alternately arranged along a time axis.
2. The audio processing method according to claim 1.
9. 9. The audio processing method according to claim 8, wherein after the search step and the display control step are repeatedly executed, the processing step is executed for a plurality of second audio files corresponding to the plurality of first audio files.
10. The processing steps include: performing dynamic range compression of the audio of the second plurality of audio files; adjusting a loudness value of the audio of the plurality of second audio files subjected to the dynamic range compression based on a loudness value of the audio of the plurality of first audio files; 10. The method of claim 9, further comprising:
11. the display area further includes a third display area arranged in parallel with the first display area and the second display area in the second direction, the display control step further includes a third step of displaying, in the third display area, a third waveform that is a waveform of an audio signal reflecting a result of the dynamic range compression and the adjustment of the loudness value, so as to overlap with the second waveform in the first direction.
11. The audio processing method according to claim 10.
12. The audio processing method according to claim 11, further comprising a step of updating the display of the third waveform in the third display area with audio waveforms of the second audio files having further loudness value adjustments.
13. an acquisition means for acquiring a first folder path that is a folder path of a first folder and a second folder path that is a folder path of a second folder different from the first folder; a search means for searching, based on a file name, from the second folder specified by the second folder path for a second voice file corresponding to the first voice file stored in the first folder specified by the first folder path; a display control means for displaying a first waveform, which is a waveform of the audio of the first audio file, and a second waveform, which is a waveform of the audio of the second audio file, in a display area of a display; a processing means for processing the audio of the second audio file so that the volume of the audio of the second audio file approaches the volume of the audio of the first audio file; having the display area includes a first display area and a second display area arranged in parallel with the first display area in a second direction perpendicular to a first direction parallel to a time axis, the display control means controls waveform display on the display so as to display the first waveform in the first display area and to display the second waveform in the second display area so as not to overlap the first waveform in the first direction.
13. A voice processing device comprising:
14. A program for causing a computer to execute each step of the audio processing method according to any one of claims 1 to 12.
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