CN109151702B - Sound effect adjusting method of audio equipment, audio equipment and readable storage medium - Google Patents
Sound effect adjusting method of audio equipment, audio equipment and readable storage medium Download PDFInfo
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- CN109151702B CN109151702B CN201811099820.5A CN201811099820A CN109151702B CN 109151702 B CN109151702 B CN 109151702B CN 201811099820 A CN201811099820 A CN 201811099820A CN 109151702 B CN109151702 B CN 109151702B
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Abstract
The invention discloses a sound effect adjusting method of audio equipment, which comprises the following steps: acquiring initial reverberation time of a detection signal measurement environment space; applying a first target model to the detection signal according to the initial reverberation time to obtain a first test signal generated based on the first target model and a test reverberation time of a corresponding measurement environment space of the first test signal; when the preset target reverberation time is received, adjusting the test reverberation time to obtain a second target model and a second test signal generated based on the second target model; acquiring the current reverberation time of a second test signal measurement environment space; and outputting a second target model to play the sound effect when the difference value between the current reverberation time and the target reverberation time meets a preset range. The invention also discloses audio equipment and a computer readable storage medium. The invention can pertinently improve the reverberation effect according to the reverberation state of the listening environment where the user is located, so as to improve the listening experience of the listener.
Description
Technical Field
The present invention relates to the field of audio technologies, and in particular, to a sound effect adjusting method for an audio device, and a computer-readable storage medium.
Background
Reverberation exists really in the real world, but the recording of a song is often completed in a sound-proof or sound-absorbing sound-proof studio, so that sound reflection in the real environment is lacked, and if the recording is directly heard, the sound-proof and sound-absorbing recording studio lacks presence. Therefore, in order to enhance the subjective listening feeling of a song, the existing audio equipment can add reverberation effect to the song, and the space feeling can be increased by reprocessing the relatively dry original sound, so that the fullness of the sound is improved.
Although the existing audio equipment can increase the reverberation effect of songs, the various listening environments of users are not considered, and the listening environments have larger reverberation, at the moment, if the reverberation is added to the playing effect, the overall reverberation is too much, and the listening feeling of listeners is reduced.
Disclosure of Invention
The present invention mainly aims to provide a sound effect adjusting method for an audio device, an audio device and a computer readable storage medium, which aim to improve the reverberation effect in a targeted manner according to the reverberation state of the listening environment where a user is located, so as to improve the listening experience of a listener.
In order to achieve the above object, the present invention provides a sound effect adjusting method of an audio device, the sound effect adjusting method of the audio device comprising the steps of:
acquiring initial reverberation time of a detection signal measurement environment space;
applying a first target model to the detection signal according to the initial reverberation time to obtain a first test signal generated based on the first target model and obtain a test reverberation time of a corresponding measurement environment space of the first test signal;
when a preset target reverberation time is received, adjusting the test reverberation time to obtain a second target model and a second test signal generated based on the second target model;
acquiring the current reverberation time of the second test signal measurement environment space;
and outputting the second target model to play the sound effect when the difference value between the current reverberation time and the target reverberation time meets a preset range.
Preferably, the step of adjusting the test reverberation time when the target reverberation time is received comprises:
obtaining the type of the first target model when target reverberation time is received;
when the type of the first target model is a basic reverberation unit model, adjusting delay factor and/or attenuation factor parameters;
when the type of the first target model is an open interface reverberation model, adjusting early reflection time and/or dry-to-wet ratio parameters;
and obtaining a second target model corresponding to the adjusted test reverberation time according to the adjusted test reverberation time.
Preferably, the audio device comprises a loudspeaker and a microphone, and the step of acquiring the detection signal to measure an initial reverberation time of the ambient space comprises:
obtaining an input function based on the detection signal played by the loudspeaker and an output function collected to the microphone after reverberation is generated by the environment space;
calculating according to the input function and the output function to obtain a transfer function;
performing inverse Laplace transform on the transfer function to obtain an impulse response function;
and calculating the initial reverberation time according to the impulse response function.
Preferably, the step of calculating the initial reverberation time according to the impulse response function comprises:
obtaining an energy attenuation curve according to the impulse response function;
acquiring the corresponding relation between a predetermined number of attenuation decibels and attenuation time within a predetermined decibel range of the attenuation of the detection signal;
calculating the slope of the energy attenuation curve according to the corresponding relation between the attenuation decibels of the preset number and the attenuation time;
and calculating attenuation time corresponding to 60 decibels of attenuation of the detection signal according to the slope to obtain the initial reverberation time.
Preferably, the step of acquiring the second test signal to measure the current reverberation time of the ambient space further comprises:
and when the difference value between the current reverberation time and the target reverberation time does not meet a preset range, re-executing the step of adjusting the test reverberation time.
Preferably, the preset range is: 25ms to 25 ms.
Preferably, the detection signal is any one of a frequency sweep signal, a steady-state noise signal or a pulse signal.
In order to achieve the above object, the present invention further provides an audio device, where the audio device includes a processor and an audio effect adjusting program of the audio device stored in the processor and capable of running on the processor, where the audio effect adjusting program of the audio device, when executed by the processor, implements the steps of the audio effect adjusting method of the audio device as described above.
Preferably, the audio device comprises a speaker and a microphone.
The invention provides a sound effect adjusting method of audio equipment, the audio equipment and a computer readable storage medium, which are characterized in that the initial reverberation time of an environment space is measured by obtaining a detection signal, a first target model is applied to the detection signal according to the initial reverberation time to obtain a first test signal generated based on the first target model and obtain the test reverberation time of the corresponding measurement environment space of the first test signal, the test reverberation time is adjusted when a preset target reverberation time is received to obtain a second target model, a second test signal generated based on the second target model is further obtained to obtain the current reverberation time of the second test signal measurement environment space, and finally the second target model is output when the difference value between the current reverberation time and the target reverberation time meets a preset range, so as to play sound effect. Therefore, the corresponding initial reverberation time is obtained by testing the current listening room of the user, and then the parameter is optimized, so that the reverberation effect can be improved in a targeted manner, and the listening experience of the listener is improved.
Drawings
Fig. 1 is a functional block diagram of an embodiment of an audio device according to an embodiment of the present invention;
FIG. 2 is a flowchart illustrating a first embodiment of a sound effect adjusting method of an audio device according to the present invention;
FIG. 3 is a schematic diagram of room reverberation generated by an audio device such as a smart sound box;
FIG. 4 is a detailed flowchart of step S3 in FIG. 2;
FIG. 5 is a detailed flowchart of step S1 in FIG. 2;
fig. 6 is a detailed flowchart of step S14 in fig. 5.
The implementation, functional features and advantages of the objects of the present invention will be further explained with reference to the accompanying drawings.
Detailed Description
It should be understood that the specific embodiments described herein are merely illustrative of the invention and are not intended to limit the invention.
As shown in fig. 1, the audio apparatus of the present invention includes: a processor 1001, such as a CPU, a user interface 1002, memory 1003, a communication bus 1004, a speaker 1005, a microphone 1006, and a bluetooth module 1007. Wherein a communication bus 1004 is used to enable connective communication between these components. The user interface 1002 may include an input unit. The speaker 1005 is used to play the detection signal, and the microphone 1006 collects the reverberation signal generated by the environment space.
The memory 1003 may be a high-speed RAM memory or a non-volatile memory (e.g., a disk memory). The memory 1003 may alternatively be a storage device separate from the processor 1001. Used for storing environment information of various occasions.
Those skilled in the art will appreciate that the audio device shown in fig. 1 does not constitute a limitation of the audio device and may include more or fewer components than those shown, or some components may be combined, or a different arrangement of components.
The memory 1003, which is a kind of computer storage medium, may include therein an operating system, a network communication module, a user interface module, and a sound effect adjustment program of an audio device.
The user interface 1002 is mainly used for receiving a user instruction triggered by a user input instruction through an input unit; the processor 1001 is configured to call the sound effect adjustment program of the audio device stored in the memory 1003, and perform the following operations:
acquiring initial reverberation time of a detection signal measurement environment space;
applying a first target model to the detection signal according to the initial reverberation time to obtain a first test signal generated based on the first target model and obtain a test reverberation time of a corresponding measurement environment space of the first test signal;
when a preset target reverberation time is received, adjusting the test reverberation time to obtain a second target model and a second test signal generated based on the second target model;
acquiring the current reverberation time of the second test signal measurement environment space;
and outputting the second target model to play the sound effect when the difference value between the current reverberation time and the target reverberation time meets a preset range.
Further, the processor 1001 may call the sound effect adjustment program of the audio apparatus stored in the memory 1003, and also perform the following operations:
obtaining the type of the first target model when target reverberation time is received;
when the type of the first target model is a basic reverberation unit model, adjusting delay factor and/or attenuation factor parameters;
when the type of the first target model is an open interface reverberation model, adjusting early reflection time and/or dry-to-wet ratio parameters;
and obtaining a second target model corresponding to the adjusted test reverberation time according to the adjusted test reverberation time.
Further, the processor 1001 may call the sound effect adjustment program of the audio apparatus stored in the memory 1003, and also perform the following operations:
obtaining an input function based on the detection signal played by the loudspeaker and an output function collected to the microphone after reverberation is generated by the environment space;
calculating according to the input function and the output function to obtain a transfer function;
performing inverse Laplace transform on the transfer function to obtain an impulse response function;
and calculating the initial reverberation time according to the impulse response function.
Further, the processor 1001 may call the sound effect adjustment program of the audio apparatus stored in the memory 1003, and also perform the following operations:
obtaining an energy attenuation curve according to the impulse response function;
acquiring the corresponding relation between a predetermined number of attenuation decibels and attenuation time within a predetermined decibel range of the attenuation of the detection signal;
calculating the slope of the energy attenuation curve according to the corresponding relation between the attenuation decibels of the preset number and the attenuation time;
and calculating attenuation time corresponding to 60 decibels of attenuation of the detection signal according to the slope to obtain the initial reverberation time.
Further, the processor 1001 may call the sound effect adjustment program of the audio apparatus stored in the memory 1003, and also perform the following operations:
and when the difference value between the current reverberation time and the target reverberation time does not meet a preset range, re-executing the step of adjusting the test reverberation time.
Referring to fig. 2, in a first embodiment, the present invention provides a sound effect adjusting method of an audio device, including the following steps:
step S1, obtaining the initial reverberation time of the detection signal measurement environment space;
in this embodiment, the audio device includes a speaker and a microphone. Fig. 3 is a schematic diagram of room reverberation generated when an audio device such as a smart speaker is played, wherein:
d1 is the first sound played by the loudspeaker received by the microphone, i.e. the direct sound, which determines the clarity of the sound and transmits the direction information such as the content, direction and position of the sound source; d2 and d3 are the following sounds which are obviously separated and relatively separated and received by the microphone, namely 'early reflected sound', the early reflected sound within 50ms can effectively improve the strength and definition of the direct sound, but the sound image localization is influenced by too strong sound; the sounds of d4 and d5 and the following lingering sounds are called 'late reverberation', which can make the sound field uniform and have full sound quality, can help people to distinguish the closed space characteristics of the environmental space, such as the room volume and the space height, but does not contain information quantity, can generate masking effect on the direct sound, reduces the definition of the sound, and in addition, the reverberation exceeding 100ms can become echo.
Wherein, the detection signal can be any one of a frequency sweep signal, a steady-state noise signal or a pulse signal. When the original sound source is an impulse signal, the reverberation of the detection signal is the multiple weighted superposition of the original sound source, which is equivalent to the convolution of the original sound source and the impulse response.
The environment space is a current listening room of a user, corresponding initial reverberation time is obtained by testing the current listening room of the user, and then the parameters are optimized, so that the playing effect of a specific space can be optimized, and the user experience is improved.
Step S2, applying a first target model to the detection signal according to the initial reverberation time to obtain a first test signal generated based on the first target model, and obtaining a test reverberation time of a corresponding measurement environment space of the first test signal;
in this embodiment, a corresponding relationship between the initial reverberation time and a first target model may be preset, for example, when the initial reverberation time is 0.3s, the corresponding first target model is a language studio; when the initial reverberation time is 0.4s, the corresponding first target model is a strong-suction sound recording room; when the initial reverberation time is 1s, the corresponding first object model is a cinema, and so on. Of course, the corresponding first target model may also be determined in combination with parameters such as the room size or the room type.
The first object model may be specifically divided into: a stadium, a concert hall, an opera house, a multi-function hall, a theater, a movie theater, a stereo movie, a music recording studio (natural mixing), a strong suction recording studio (multi-channel division recording), a language recording studio, etc., as specified in the following table one:
watch 1
When a first target model is applied to the detection signal according to the initial reverberation time, a first test signal generated based on the first target model may be obtained.
When measuring the ambient space with the original detection signal, an initial reverberation time T is obtained0After applying the first target model to the detection signal, a new reverberation measurement sound source is generated and used as a new testTesting the test reverberation time T of the ambient space by an audio source, i.e. a first test signal1At this time, since two reverberations (ambient spatial reverberation and reverberation in the reverberation model) are superimposed, T is1＞T0。
Step S3, when a preset target reverberation time is received, adjusting the test reverberation time to obtain a second target model and a second test signal generated based on the second target model;
in this embodiment, the target reverberation time may be a preset empirical reference value, as shown in table one above; and a large amount of subjective experience tests are required before the reverberation time is defined by the disc-jockey in advance.
And when the target reverberation time is inconsistent with the test reverberation time, adjusting the test reverberation time, and obtaining a corresponding second target model through the adjusted test reverberation time. Specifically, when a first target model such as a voice recording studio is applied to the detection signal according to the initial reverberation time, the test reverberation time corresponding to the first target model is obtained to be 0.3-0.4 s, and the target reverberation time is 0.6s, so that the test reverberation time needs to be adjusted to 0.6s, and at this time, the corresponding second target model is a strong-suction recording studio. The second target model is then applied to the detection signal to generate a new reverberant test sound source, i.e. a second test signal.
Step S4, acquiring the current reverberation time of the second test signal measurement environment space;
and step S5, outputting the second target model to play sound effect when the difference value between the current reverberation time and the target reverberation time meets a preset range.
In this embodiment, a second test signal is used to measure an environment space, so as to obtain a current reverberation time of the environment space, and the current reverberation time is compared with the target reverberation time, so as to determine whether an error range between the current reverberation time and the target reverberation time meets a preset range, specifically, a difference between the current reverberation time and the target reverberation time is calculated, and when the difference meets the preset range, such as-25 ms to 25ms, a second target model is output, so that the audio device performs sound effect playing; and when the difference does not meet the preset range, such as-25 ms to 25ms, re-executing the step of adjusting the test reverberation time in the step S2.
The sound effect adjusting method of the audio equipment provided by the invention measures the initial reverberation time of the environment space by obtaining the detection signal, then, a first target model is applied to the detection signal according to the initial reverberation time to obtain a first test signal generated based on the first target model and obtain a test reverberation time of a corresponding measurement environment space of the first test signal, and when a preset target reverberation time is received, adjusting the test reverberation time to obtain a second target model, further obtaining a second test signal based on the second test signal generated by the second target model to measure the current reverberation time of the environment space, and finally outputting the second target model to play the sound effect when the difference value between the current reverberation time and the target reverberation time meets a preset range. Therefore, the corresponding initial reverberation time is obtained by testing the current listening room of the user, and then the parameter is optimized, so that the reverberation effect can be improved in a targeted manner, and the listening experience of the listener is improved.
Referring to fig. 4, in the second embodiment, based on the first embodiment, the step of adjusting the test reverberation time when the target reverberation time is received in the step S3 includes:
step S31, when receiving the target reverberation time, obtaining the type of the first target model;
step S32, when the type of the first target model is the basic reverberation unit model, adjusting delay factor and/or attenuation factor parameters;
in this embodiment, the types of the first target model include a basic reverberation unit model and an open interface reverberation model, where the basic reverberation unit model is the most basic unit constituting each model in the first table, and the open interface reverberation model refers to a "human-computer interaction" interface provided on software or an application program, and is used for a user to self-adjust some key parameters.
And when the target reverberation time is received, acquiring the type of the first target model, and if the type of the first target model is a basic reverberation unit model, adjusting the delay factor, or adjusting an attenuation factor, or adjusting the delay factor and the attenuation factor. It is understood that the parameters for specific adjustment are not limited to these two types, and can be set reasonably according to actual conditions.
Step S33, when the type of the first target model is an open interface reverberation model, adjusting early-reflection time and/or dry-wet ratio parameters;
if the type of the first target model is an open interface reverberation model, the early reflection time may be adjusted, or the dry-to-wet ratio parameter may be adjusted, or the early reflection time and the dry-to-wet ratio parameter may be adjusted. It is understood that the parameters for specific adjustment are not limited to these two types, and can be set reasonably according to actual conditions.
It is understood that the retardation factor and the attenuation factor are internal parameters, the surface of which is reflected in the dry-wet ratio and the like. Therefore, the adjustment of the dry-wet ratio parameter in the open interface reverberation model can facilitate the user to simply operate according to the actual needs of the user.
Referring to fig. 5, in the third embodiment, based on the first embodiment, the step S1 includes:
step S11, obtaining an input function based on the detection signal played by the loudspeaker and an output function collected to the microphone after reverberation is generated by the environment space;
in this embodiment, the environment space may be used as a system, the detection signal played by the loudspeaker is equivalent to an input signal of the environment space, and the input signal may be used as an input function xi(s) of the system; the detection signal generates room reverberation in the ambient space, that is, the detection signal is processed by the system and then collected by the microphone, so as to obtain an output signal processed by the system, and the output signal can be used as an output function xo(s) of the system.
Step S12, calculating according to the input function and the output function to obtain a transfer function;
step S13, performing inverse Laplace transform on the transfer function to obtain an impulse response function;
in this embodiment, the transfer function g(s) is calculated according to the input function xi(s) and the output function xo(s):
g(s) ═ xi (s)/xo(s); where S represents a complex number, and is an argument of laplace transform.
And when the transfer function is obtained through calculation, performing inverse Laplace transform on the transfer function to obtain the impulse response function.
And step S14, calculating the initial reverberation time according to the impulse response function.
In this embodiment, when the impulse response function is obtained through calculation, a corresponding energy decay curve may be obtained, and the initial reverberation time may be calculated according to the energy decay curve.
In the embodiment, the current listening room of the user is tested to obtain the corresponding initial reverberation time, and then the parameters are optimized, so that the playing effect of a specific space can be optimized, and the listening experience of the user is improved.
Referring to fig. 6, in the fourth embodiment, based on the third embodiment, the step S14 includes:
step S141, obtaining an energy attenuation curve according to the impulse response function;
step S142, acquiring the corresponding relation between a predetermined number of attenuation decibels and attenuation time within a predetermined decibel range of the attenuation of the detection signal;
in this embodiment, when an energy attenuation curve is obtained according to the impulse response function, a corresponding relationship between a plurality of decibels and attenuation time within a predetermined decibel range of attenuation of the detection signal, for example, from-5 dB to-25 dB, can be obtained. Specifically, when the detection signal is attenuated by 5dB, the corresponding attenuation time is 10 ms; when the detection signal is attenuated by 10dB, the corresponding attenuation time is 40 ms; when the detection signal is attenuated by 15dB, the corresponding attenuation time is 100 ms; when the detected signal is attenuated by 25dB, the corresponding attenuation time is 260ms, and so on.
Step S143, calculating the slope of the energy attenuation curve according to the corresponding relation between the predetermined number of attenuation decibels and the attenuation time;
in this embodiment, the predetermined number may be 2 or more, and the specific number may be reasonably selected according to the actual form of the curve, or may be reasonably set according to a custom standard. And calculating to obtain the slope of the energy attenuation curve to be-0.07 according to the corresponding relation between the three attenuation decibels and the attenuation time.
And step S144, calculating attenuation time corresponding to 60 dB attenuation of the detection signal according to the slope to obtain the initial reverberation time.
In this embodiment, since the calculated slope is 0.5, the attenuation time corresponding to the detection signal attenuation of 60 db may be calculated to be 780ms, that is, the initial reverberation time is obtained.
In the embodiment, the slope is calculated by only selecting the corresponding relation between 2 or a small number of attenuation decibels and the attenuation time, and the attenuation time corresponding to the attenuation 60 decibels of the detection signal can be obtained according to the slope, so that the calculated amount is small, and the distance between an actual listener and a sound source such as a singer and the approximate situation of the surrounding environment can be quickly reflected.
The invention also provides an audio device, which comprises a processor and an audio device sound effect adjusting program which is stored in the processor and can run on the processor, wherein the audio device sound effect adjusting program realizes the steps of the audio device sound effect adjusting method when being executed by the processor.
Further, the audio device may be a smart speaker, and specifically includes a speaker and a microphone. The loudspeaker is used for playing the detection signal, and the microphone is used for collecting a reverberation signal generated by the environment space.
The invention also provides a computer readable storage medium, which is characterized in that the computer readable storage medium stores a sound effect adjusting program of the audio equipment, and the sound effect adjusting program of the audio equipment is executed by a processor to realize the steps of the sound effect adjusting method of the audio equipment.
It should be noted that, in this document, the terms "comprises," "comprising," or any other variation thereof, are intended to cover a non-exclusive inclusion, such that a process, method, article, or system that comprises a list of elements does not include only those elements but may include other elements not expressly listed or inherent to such process, method, article, or system. Without further limitation, an element defined by the phrase "comprising an … …" does not exclude the presence of other like elements in a process, method, article, or system that comprises the element.
The above-mentioned serial numbers of the embodiments of the present invention are merely for description and do not represent the merits of the embodiments.
Through the above description of the embodiments, those skilled in the art will clearly understand that the method of the above embodiments can be implemented by software plus a necessary general hardware platform, and certainly can also be implemented by hardware, but in many cases, the former is a better implementation manner. Based on such understanding, the technical solutions of the present invention may be embodied in the form of a software product, which is stored in a storage medium (e.g., ROM/RAM, magnetic disk, optical disk) as described above and includes several instructions for enabling a terminal device (e.g., a mobile phone, a computer, a server, a controlled terminal, or a network device) to execute the method according to the embodiments of the present invention.
Claims (8)
1. A sound effect adjusting method of audio equipment is characterized in that the audio equipment comprises a loudspeaker and a microphone, and the sound effect adjusting method of the audio equipment comprises the following steps:
acquiring an input function based on a detection signal played by the loudspeaker and an output function collected to the microphone after reverberation is generated in an environment space;
calculating according to the input function and the output function to obtain a transfer function;
performing inverse Laplace transform on the transfer function to obtain an impulse response function;
calculating the initial reverberation time according to the impulse response function;
applying a first target model to the detection signal according to the initial reverberation time to obtain a first test signal generated based on the first target model and obtain a test reverberation time of a corresponding measurement environment space of the first test signal;
when a preset target reverberation time is received, adjusting the test reverberation time to obtain a second target model and a second test signal generated based on the second target model;
acquiring the current reverberation time of the second test signal measurement environment space;
and outputting the second target model to play the sound effect when the difference value between the current reverberation time and the target reverberation time meets a preset range.
2. The method for adjusting sound-effect of audio equipment according to claim 1, wherein the step of adjusting the test reverberation time upon receiving the target reverberation time comprises:
obtaining the type of the first target model when target reverberation time is received;
when the type of the first target model is a basic reverberation unit model, adjusting delay factor and/or attenuation factor parameters;
when the type of the first target model is an open interface reverberation model, adjusting early reflection time and/or dry-to-wet ratio parameters;
and obtaining a second target model corresponding to the adjusted test reverberation time according to the adjusted test reverberation time.
3. The method for adjusting sound effect of audio equipment according to claim 1, wherein the step of calculating the initial reverberation time according to the impulse response function comprises:
obtaining an energy attenuation curve according to the impulse response function;
acquiring the corresponding relation between a predetermined number of attenuation decibels and attenuation time within a predetermined decibel range of the attenuation of the detection signal;
calculating the slope of the energy attenuation curve according to the corresponding relation between the attenuation decibels of the preset number and the attenuation time;
and calculating attenuation time corresponding to 60 decibels of attenuation of the detection signal according to the slope to obtain the initial reverberation time.
4. The sound-effect adjustment method of an audio device according to claim 1, wherein the step of obtaining the second test signal to measure the current reverberation time of the ambient space further comprises:
and when the difference value between the current reverberation time and the target reverberation time does not meet a preset range, re-executing the step of adjusting the test reverberation time.
5. The sound effect adjustment method of an audio device according to claim 1, wherein the preset range is: 25ms to 25 ms.
6. The method for adjusting the sound effect of an audio device according to any one of claims 1 to 5, wherein the detection signal is any one of a frequency sweep signal, a stationary noise signal, or an impulse signal.
7. An audio device comprising a processor and an audio device acoustics adjustment program stored in and executable on the processor, wherein the audio device acoustics adjustment program when executed by the processor implements the steps of the audio device acoustics adjustment method of any of claims 1 to 6.
8. A computer-readable storage medium, characterized in that the computer-readable storage medium has stored thereon an effect adjustment program of an audio apparatus, the effect adjustment program of the audio apparatus being executed by a processor to implement the steps of the effect adjustment method of the audio apparatus according to any one of claims 1 to 6.
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