CN110634462A - Sound adjusting system and adjusting method - Google Patents

Sound adjusting system and adjusting method Download PDF

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Publication number
CN110634462A
CN110634462A CN201910942540.4A CN201910942540A CN110634462A CN 110634462 A CN110634462 A CN 110634462A CN 201910942540 A CN201910942540 A CN 201910942540A CN 110634462 A CN110634462 A CN 110634462A
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sound
module
original
sound data
standard
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CN110634462B (en
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孙任飞
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Shenzhen Weipu Medical Technology Co ltd
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Shenzhen Tongshihai Precision Machinery Co Ltd
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    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10HELECTROPHONIC MUSICAL INSTRUMENTS; INSTRUMENTS IN WHICH THE TONES ARE GENERATED BY ELECTROMECHANICAL MEANS OR ELECTRONIC GENERATORS, OR IN WHICH THE TONES ARE SYNTHESISED FROM A DATA STORE
    • G10H1/00Details of electrophonic musical instruments
    • G10H1/0033Recording/reproducing or transmission of music for electrophonic musical instruments
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10HELECTROPHONIC MUSICAL INSTRUMENTS; INSTRUMENTS IN WHICH THE TONES ARE GENERATED BY ELECTROMECHANICAL MEANS OR ELECTRONIC GENERATORS, OR IN WHICH THE TONES ARE SYNTHESISED FROM A DATA STORE
    • G10H1/00Details of electrophonic musical instruments
    • G10H1/0091Means for obtaining special acoustic effects
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10HELECTROPHONIC MUSICAL INSTRUMENTS; INSTRUMENTS IN WHICH THE TONES ARE GENERATED BY ELECTROMECHANICAL MEANS OR ELECTRONIC GENERATORS, OR IN WHICH THE TONES ARE SYNTHESISED FROM A DATA STORE
    • G10H1/00Details of electrophonic musical instruments
    • G10H1/36Accompaniment arrangements
    • G10H1/361Recording/reproducing of accompaniment for use with an external source, e.g. karaoke systems
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10HELECTROPHONIC MUSICAL INSTRUMENTS; INSTRUMENTS IN WHICH THE TONES ARE GENERATED BY ELECTROMECHANICAL MEANS OR ELECTRONIC GENERATORS, OR IN WHICH THE TONES ARE SYNTHESISED FROM A DATA STORE
    • G10H1/00Details of electrophonic musical instruments
    • G10H1/46Volume control

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  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Circuit For Audible Band Transducer (AREA)

Abstract

The invention discloses a sound adjusting system and a sound adjusting method, wherein the sound adjusting system comprises a sound signal input module, a sound signal output module and a sound signal processing module, wherein the sound signal input module is used for perceiving and collecting sound signals and converting the sound signals into original sound data; the signal processing module is used for receiving the original sound data transmitted by the sound signal acquisition module and adjusting the original sound data into adjusted sound data through a sound adjustment module; and the sound signal output module is used for receiving the adjusted sound data and converting the adjusted sound data into a sound signal to be played. The invention has the beneficial effects that: the method can adjust the sound of the person singing the song and easily running off according to the requirement, and play the running sound after being corrected.

Description

Sound adjusting system and adjusting method
Technical Field
The present invention relates to the field of audio processing technologies, and in particular, to a sound adjusting system and method.
Background
With the improvement of living standard of people, the pursuit of people for cultural entertainment life is also improved, and the music function has become a necessary application in terminals such as computers or mobile phones and the like due to the high-speed development of communication and information technology. More and more music products with the KTV function appear in a terminal of a user, the accompaniment is played according to music selected by the user and the caption is displayed along with the music accompaniment, and the user prompts the singing time of the corresponding lyric according to font color prompt or other marks on the displayed caption until the whole song is finished. In recent years, the singing K gradually becomes a recreational way for people, but the mood of singing is seriously influenced because different people have different feelings of difference in the perception abilities of the pleasure and the melody and the conditions of running or untimely rhythm occur. At present, the method for solving the problem that the running tune cannot follow the rhythm is only to find a professional recording studio to carry out post-period tone correction processing so as to correct the running tune. Although the speciality of sound repair of a recording studio is high, under the conditions of long production period, high production cost and the like, the common people are difficult to accept, and the interest of people in K songs is reduced.
Meanwhile, the beautifying function of live broadcast software is deeply loved by the public, but the live broadcast software only has the beautifying function and does not have the function of 'beautifying sound' to serve the public, the defect of singing sound when a plurality of masters play songs in live broadcast is a great pain point, and the problem is usually solved by 'beautifying sound' synchronized by accessing third-party software, so that a user is required to install and register two different types of software and debug the process is complex and tedious.
Disclosure of Invention
This section is for the purpose of summarizing some aspects of embodiments of the invention and to briefly introduce some preferred embodiments. In this section, as well as in the abstract and the title of the invention of this application, simplifications or omissions may be made to avoid obscuring the purpose of the section, the abstract and the title, and such simplifications or omissions are not intended to limit the scope of the invention.
The present invention has been made in view of the above-mentioned conventional problems.
Therefore, one technical problem solved by the present invention is: a sound adjusting system is provided, which can adjust the sound of people when singing and solve the problem of frequent easy running and adjustment.
In order to solve the technical problems, the invention provides the following technical scheme: a sound adjusting system comprises a sound signal input module, a sound signal processing module and a sound adjusting module, wherein the sound signal input module is used for sensing and collecting sound signals and converting the sound signals into original sound data; the signal processing module is used for receiving the original sound data transmitted by the sound signal acquisition module and adjusting the original sound data into adjusted sound data through a sound adjustment module; and the sound signal output module is used for receiving the adjusted sound data and converting the adjusted sound data into a sound signal to be played.
As a preferable aspect of the sound adjusting system of the present invention, wherein: the signal processing module also comprises an input module and an output module; the input module is used for simultaneously receiving one or more original sound data generated by the sound signal acquisition module for pre-adjustment; the output module is used for receiving the adjusted sound data output by the sound adjusting module and processing the adjusted sound data into the sound signal output module.
As a preferable aspect of the sound adjusting system of the present invention, wherein: the original sound data acquired by the sound signal input module further includes an original loudness, an original tone, and an original time parameter corresponding to the original loudness, the original tone, and the original tone of the input sound.
As a preferable aspect of the sound adjusting system of the present invention, wherein: the signal processing module further comprises an adjustment reference standard module for referring to specified standard sound data including a standard loudness, a standard tone color, and a standard time parameter corresponding to the standard loudness, the standard tone, and the standard tone color.
As a preferable aspect of the sound adjusting system of the present invention, wherein: the sound adjusting module is used for adjusting one or more of the original loudness, the original tone and the original timbre in the original sound data into the standard loudness, the standard tone and the standard timbre corresponding to the original loudness, the standard tone and the standard timbre, and the sound adjusting module carries out positioning according to the one-to-one correspondence of the original time parameter and the standard time parameter.
As a preferable aspect of the sound adjusting system of the present invention, wherein: the signal processing module further comprises an intelligent voice matching module, wherein the intelligent voice matching module is used for analyzing voice contents in the original sound data and the standard sound data, performing intelligent matching, and positioning the adjustment of the original loudness, the original tone and the original tone in the original sound data according to a matching result.
As a preferable aspect of the sound adjusting system of the present invention, wherein: the signal processing module further comprises a storage module, the storage module is further used for storing the standard sound data to wait for reference call, and the storage module is further used for receiving the adjusted sound data and storing the adjusted sound data as new sound data, and the new sound data is fed to the input module as the original sound data.
The invention solves another technical problem that: a sound adjusting method is provided, and the sound adjusting method can be realized by relying on the system.
In order to solve the technical problems, the invention provides the following technical scheme: a sound adjusting method comprises the following steps that a sound signal input module acquires original sound data; the input module receives the original sound data for pre-adjustment; the sound adjusting module combines the adjusting reference standard module and the intelligent voice matching module to position, match and adjust the pre-adjusted sound data into adjusted sound data; the output module receives the adjusted sound data and integrates the adjusted sound data into an output signal; the sound signal output module receives the output signal of the output module and converts the output signal into a voice signal to play.
As a preferable aspect of the sound adjusting method of the present invention, wherein: the output module receives the adjusted sound data and integrates the adjusted sound data into an output signal; the storage module receives the output signal for storage and inputs the output signal to the input module as the original sound data, and executes the next cycle adjustment.
As a preferable aspect of the sound adjusting method of the present invention, wherein: the adjustment of the original sound data by the sound adjustment module comprises a user-defined adjustment amount range and also comprises the user-defined loudness on the basis of the standard loudness when the original loudness is adjusted; when the original tone is adjusted, the tone height is customized on the basis of the standard tone; and when the original timbre is adjusted, the appointed different standard timbres are simultaneously referred for mixing adjustment.
The invention has the beneficial effects that: the method can adjust the sound of the person singing the song and easily running off according to the requirement, and play the running sound after being corrected.
Drawings
In order to more clearly illustrate the technical solutions of the embodiments of the present invention, the drawings needed to be used in the description of the embodiments will be briefly introduced below, and it is obvious that the drawings in the following description are only some embodiments of the present invention, and it is obvious for those skilled in the art to obtain other drawings based on these drawings without inventive exercise. Wherein:
fig. 1 is a schematic structural diagram of an overall sound adjusting system according to a first embodiment of the present invention;
FIG. 2 is a schematic structural diagram of modules of a sound adjustment system according to a first embodiment of the present invention;
FIG. 3 is a flowchart illustrating an overall sound adjustment method according to a second embodiment of the present invention;
FIG. 4 is a flowchart illustrating an optimized output of an original sound signal according to a second embodiment of the present invention;
FIG. 5 is a diagram illustrating an image of an original sound signal after sampling and quantization according to a second embodiment of the present invention;
FIG. 6 is a schematic diagram of an impedance matching circuit of a microphone according to a third embodiment of the present invention;
fig. 7 is a schematic overall flow chart of a method for adaptive volume adjustment according to a third embodiment of the present invention;
FIG. 8 is a comparison of actual test results according to a third embodiment of the present invention;
FIG. 9 is a diagram illustrating an error comparison of actual test results according to a third embodiment of the present invention.
Detailed Description
In order to make the aforementioned objects, features and advantages of the present invention comprehensible, specific embodiments accompanied with figures are described in detail below, and it is apparent that the described embodiments are a part of the embodiments of the present invention, not all of the embodiments. All other embodiments, which can be obtained by a person skilled in the art without making creative efforts based on the embodiments of the present invention, shall fall within the protection scope of the present invention.
In the following description, numerous specific details are set forth in order to provide a thorough understanding of the present invention, but the present invention may be practiced in other ways than those specifically described and will be readily apparent to those of ordinary skill in the art without departing from the spirit of the present invention, and therefore the present invention is not limited to the specific embodiments disclosed below.
Furthermore, reference herein to "one embodiment" or "an embodiment" means that a particular feature, structure, or characteristic described in connection with the embodiment is included in at least one implementation of the invention. The appearances of the phrase "in one embodiment" in various places in the specification are not necessarily all referring to the same embodiment, nor are separate or alternative embodiments mutually exclusive of other embodiments.
The present invention will be described in detail with reference to the drawings, wherein the cross-sectional views illustrating the structure of the device are not enlarged partially in general scale for convenience of illustration, and the drawings are only exemplary and should not be construed as limiting the scope of the present invention. In addition, the three-dimensional dimensions of length, width and depth should be included in the actual fabrication.
Meanwhile, in the description of the present invention, it should be noted that the terms "upper, lower, inner and outer" and the like indicate orientations or positional relationships based on the orientations or positional relationships shown in the drawings, and are only for convenience of describing the present invention and simplifying the description, but do not indicate or imply that the referred device or element must have a specific orientation, be constructed in a specific orientation and operate, and thus, cannot be construed as limiting the present invention. Furthermore, the terms first, second, or third are used for descriptive purposes only and are not to be construed as indicating or implying relative importance.
The terms "mounted, connected and connected" in the present invention are to be understood broadly, unless otherwise explicitly specified or limited, for example: can be fixedly connected, detachably connected or integrally connected; they may be mechanically, electrically, or directly connected, or indirectly connected through intervening media, or may be interconnected between two elements. The specific meanings of the above terms in the present invention can be understood in specific cases to those skilled in the art.
Example 1
Referring to the illustrations of fig. 1-2, the sound adjusting system provided in this embodiment is illustrated in a way that sings K are enjoyed by people, but because different people have different perceptions of joy and melody, and the conditions of running tune or rhythm not following up occur, the singing and the mood of the listener are seriously affected. Therefore, the present embodiment proposes a sound adjusting system for adjusting the sound of a pitch, the system includes a sound signal input module 100, a signal processing module 200, and a sound signal output module 300, an output of the sound signal input module 100 is connected to an input of the signal processing module 200, and an output of the signal processing module 200 is connected to an input of the sound signal output module 300. Specifically, the sound signal input module 100 collects a sound signal of an original human voice without any post-processing and processing, so in this embodiment, the sound signal input module 100 adopts a microphone or a mobile device (such as a mobile phone or a computer) with a recording function, and converts the sound signal into original sound data, where the original sound data in this embodiment is a digital signal, that is, an electrical signal, and its specific form is, for example, a signal converted by mp 3. Further, the microphone is called a microphone, and is an energy conversion device for converting a sound signal into an electric signal, which is also called a microphone or a microphone. In the twentieth century, microphones were developed from the initial resistance-to-sound-electricity conversion to the inductance-to-capacitance conversion, and a great number of new microphone technologies were developed, including aluminum strip microphones, moving coil microphones, etc., and condenser microphones and electret microphones which are widely used at present, wherein the microphone is a diaphragm of the microphone, and the vibration of sound is transmitted to the diaphragm, and a magnet at the inner side is pushed to form a varying current, and the varying current is sent to a following sound processing circuit for amplification processing, so as to obtain the required sound data. The signal processing module 200 receives the original sound data, adjusts the original sound data with reference to a standard sound, and outputs the adjusted sound data. The sound signal output module 300 receives and plays the adjusted sound data, and in this embodiment, the sound signal output module 300 is an audio device or a device with a speaker.
More specifically, the sound signal input module 100 is configured to sense and collect a sound signal, and convert the sound signal into original sound data; the signal processing module 200, the signal processing module 200 is configured to receive the original sound data transmitted by the sound signal collecting module 100, and adjust the original sound data into adjusted sound data through the sound adjusting module 201; and the sound signal output module 300, the sound signal output module 300 is configured to receive the adjusted sound data and convert the adjusted sound data into a sound signal for playing.
Wherein the signal processing module 200 further comprises an input module 202 and an output module 203; the input module 202 is configured to receive one or more original sound data generated by the sound signal collecting module 100 for pre-adjustment; the output module 203 is used for receiving the adjusted sound data output by the sound adjusting module 201 and processing the adjusted sound data into the sound signal output module 300. The original sound data acquired by the sound signal input module 100 further includes an original loudness, an original pitch, an original tone, and an original time parameter corresponding to the original loudness, the original pitch, and the original tone of the input sound. The signal processing module 200 further includes an adjustment reference standard module 204, and the adjustment reference standard module 204 is configured to refer to specified standard sound data, which includes standard loudness, standard tone color, and standard time parameter corresponding to the standard loudness, the standard tone, and the standard tone color.
The sound adjusting module 201 is configured to adjust one or more of an original loudness, an original pitch, and an original timbre in the original sound data to a standard loudness, a standard pitch, and a standard timbre corresponding thereto, and the sound adjusting module 201 performs positioning according to a one-to-one correspondence between the original time parameter and the standard time parameter.
The signal processing module 200 further includes an intelligent voice matching module 205, and the intelligent voice matching module 205 is configured to analyze voice contents in the original sound data and the standard sound data, perform intelligent matching, and position adjustment of original loudness, original tone, and original timbre in the original sound data according to a matching result. The signal processing module 200 further comprises a storage module 206, wherein the storage module 206 further comprises a module for storing standard sound data waiting for reference calls, and a module for receiving the adjusted sound data and storing the adjusted sound data as new sound data, and the new sound data is fed to the input module 202 as original sound data.
It should be further noted that the signal processing module 200 is an integrated circuit board device embedded with an algorithm program, and the sound adjusting module 201, the input module 202, the output module 203, the adjustment reference standard module 204 and the storage module 206 are all integrated circuit modules on the integrated circuit board in a circuit integration manner, and the algorithm program is written into the integrated circuit board in a conventional program burning manner, and the integrated circuit has various types, for example, a microcontroller and a microprocessor in the integrated circuit are also commonly called a single chip microcomputer. The related algorithm program is an algorithm for adjusting sound data, for example, a downloader is connected into a computer after being written, then the writing port of the downloader is aligned to the writing port of the circuit board, the circuit board is powered on, downloader software is opened in the computer after connection is completed, program information setting in a program menu is opened, a corresponding chip model and a program file are selected, finally, the chip model is checked, input information is filled in, and the writing program is clicked. A circuit board device with a sound adjustment algorithm program is obtained, i.e., the hardware device of the signal processing module 200 is proposed in the present embodiment.
Example 2
Referring to fig. 3-5, the present embodiment provides a sound adjusting method applied in the above system for correcting the pitch sound, including the following steps,
s1: the sound signal input module 100 obtains original sound data, in this step, a singer provides the original data or adjusts audio made by the singer in a hard disk in batch, and in the case of batch audio adjustment, a storage provides the original data.
Specifically, the digital processing of the sound signal includes sampling, quantization and encoding, the sampling refers to a process of converting a time-continuous analog signal into a digital signal, and the amplitude and time of the converted digital signal are discrete. The sampling process comprises two important physical quantities, namely a sampling period and a sampling frequency, wherein the sampling period is each sampling time interval, and the reciprocal of the sampling period is the sampling frequency. The higher the sampling frequency is, the more data are obtained in unit time, the better fidelity effect can be achieved, and in order to ensure the sound sampling quality and avoid distortion, the sampling frequency at least reaches twice of the highest frequency of the sound signal. The sound signal quantization means that each sample of the sampling is represented by a discrete value, and the process is also called analog-to-digital conversion, and the samples are represented in a binary number mode after being quantized, wherein the number of the binary number is quantization precision. It is clear that the higher the accuracy with which the quantization is performed, the higher the quality of the sound, but the larger the required storage space. In this embodiment, the image of the original sound signal after sampling and quantization can be referred to the schematic diagram of fig. 5.
Furthermore, the audio signal is already represented in a digital form after being quantized, but needs to be encoded and compressed according to a certain rule for convenience of transmission, processing and storage. The sound signal is encoded to obtain digital audio. The amount of sound information increases and the space for storage increases, and assuming that analog sound is digitized, in bytes and not compressed, the storage amount can be calculated by the following formula:
the storage amount is the sampling frequency × the number of quantization bits/8 × the number of channels × time.
For example, the sampling frequency is 44.1KHz, 1s stereo sound is recorded by using 16-bit quantization bits, and the storage space required for generating the waveform is 44100 × 16/8 × 2 × 1 or 176400 bytes. Therefore, the problem of storage space is also considered during sound processing, and the integrity of sound processing information storage is ensured.
The tone is mainly related to the frequency of the waveform, and the frequency of the sound wave is high, so the tone is also high; the loudness is related to the amplitude of waveform vibration, and the loudness is larger when the vibration amplitude is larger; timbre is the characteristic of a person that distinguishes two sounds having the same loudness and the same pitch from each other, and is related to the shape of the vibration waveform of a sound wave or the spectral structure of the sound. Therefore, the waveform signal generated by the sound can encode and interpret the signals of the original loudness, the original tone and the original tone of the original sound, and finally obtain the digital signal of the digital audio.
S2: the input module 202 receives the original sound data for pre-adjustment, where the pre-adjustment includes applying a digital signal processing algorithm to make the sound softer, more magnetic, and without noise, and the digital signal processing technology is a technology for converting analog information (such as the sound of this embodiment) into digital information. Generally referred to as a chip or processor that performs these functions, or they may also be used to process this information and output it as an analog electrical signal, obviously a pre-conditioning of a digital signal in this embodiment. Therefore, this step includes a process of performing noise reduction processing on the sound signal.
S3: the sound adjusting module 201 combines the adjusting reference module 204 and the intelligent voice matching module 205 to position, match and adjust the pre-adjusted sound data into the adjusted sound data. In this step, the sound adjusting module 201 adjusts one or more of the original loudness, the original pitch, and the original timbre in the original sound data to the standard loudness, the standard pitch, and the standard timbre corresponding thereto, and the sound adjusting module 201 performs positioning according to the one-to-one correspondence between the original time parameter and the standard time parameter. For example, in a song, the original loudness corresponds to the standard loudness, the original pitch corresponds to the standard pitch, and the original timbre corresponds to the standard timbre for the same time parameter of the original sound data and the standard sound data. The adjustment reference module 204 stores standard voice data in advance, and calls the standard voice data when necessary.
The intelligent voice matching module 205 analyzes the voice content in the original voice data and the standard voice data, performs intelligent matching, and locates the adjustment of the original loudness, the original tone, and the original tone in the original voice data according to the matching result. The intelligent voice matching module 205 and the sound adjusting module 201 are positioned in two independent positioning modes, which can be selected by a user, i.e. the matching positioning can be performed according to time parameters or the matching positioning can be performed according to lyrics.
In the embodiment where the time-positioning is, for example, 2 minutes and 06 seconds (positioning) of the song, the original vocal sound is so (reference standard data), and the vocal sound sung by the singer is mi (original data), the system adjusts mi sung by the singer to so at the time node of 2 minutes and 06 seconds.
The intelligent voice matching module 205 performs a recognition process based on the voice content of the original voice data, and matches the voice content in the standard voice data, for example, matches the content of the lyrics sung by the singer with the original lyrics. In this embodiment, the adjustment of the original sound data by the sound adjustment module 201 includes a range of customized adjustment amount, and further includes,
when the original loudness is adjusted, the loudness is customized on the basis of the standard loudness;
when the original tone is adjusted, the height of the tone is customized on the basis of the standard tone;
when the original tone is adjusted, the mixing adjustment is carried out by simultaneously referring to a plurality of specified different standard tones.
In this embodiment, for example, when the user sings a song, the user's own tone, the original singing is the tone of liu deluxe, and the original tone of the user can be completely covered by the liu deluxe or other reference standard tones through customization, and the singed song is a new liu deluxe singing version; meanwhile, the user can also choose to blend 30% of the Liu De Hua timbre into the original timbre, so that the user can hear the sound of 30% Liu De Hua. In the process, a user can actively select a scheme to be adjusted through an operation interface, for example, in ktv, a singer can manually select what effect to listen to in a system before singing, or can directly use the scheme in mobile phone software.
S4: the output module 203 receives and integrates the adjusted voice data into an output signal, specifically integrates the adjusted voice into a digital signal (electric signal) for output, and the integration processing comprises batch packaging and sorting of the digital signal, and labeling of a song name or a singer on the packaged voice signal, so that a user can conveniently adjust and process the stored singing voice for two or more times;
s5: the sound signal output module 300 receives the output electric signal of the output module 203 and converts the output electric signal into a voice content to play, after a 'beautified' sound signal is obtained in the step, the sound signal can be transmitted to a sound device, or can be directly stored in the storage module 206, the stored digital signal (electric signal) is input into the sound adjusting module 201 again through the storage module 206 to be adjusted again and then output, if the user does not satisfy the adjustment of the time, the circulation adjustment from the output module 203 to the storage module 206 and then to the sound adjusting module 201 can be executed until the user is satisfied with the sound data.
The storage-based module 206 in this embodiment further includes the following steps,
the output module 203 receives the adjusted sound data and integrates the adjusted sound data into an output signal;
the original sound data, the adjusted sound data and the output signal are transmitted to the storage module 206 for storage;
the stored original sound data is selected to be called out for batch processing by the storage module 206, or the output signal is selected to be input to the input module 202 as the original sound data for next adjustment, and the next loop adjustment is performed.
In this embodiment, compared to the storage of the standard sound data stored in the storage module 206 in advance in the adjustment reference standard module 204, the storage module 206 may further store more original sound data (for example, a recording that me sings before may be stored therein, and may be adjusted as desired by him or her at any time).
The method can simultaneously process a plurality of adjustment tasks, and the standard sound data can also be directly used as the original sound data and the adjustment sound data.
In the actual use process, people often easily run and tune when singing, so the method of the embodiment is mainly used for K singing, live broadcasting and the like. Adjusting the tone (intonation) of the singer according to the original singing tone by analyzing the original song of the song; and matching and positioning the music position according to the original music time axis (accompaniment) or intelligent voice (lyric) to adjust the tone. Beautifies the voice of the singer, and blends the original singing tone into the voice of the singer to different degrees; and is suitable for solo, chorus or chong, and a single person can also have chong effects, such as:
the singer (KTV, live broadcast) cannot accurately sing the song, the tone is not good, and the system can beautify the song; when one sings, the system acquires the tone standard of the original singing to modify the tone standard of the singer, and optionally, the system can acquire the tone of the original singing to modify the tone of the singer to a certain degree; when one sings, optionally, the user can select ' chorus ' (chorus) ' with the original singing, and when the original singing is the chorus, the user can also select one vocal part to sing by himself, and the original singing can select other vocal parts to perform ' chorus '; when a person sings, optionally, when the original singing is the singing and the chorus, the voice of the person can be adjusted and output into a plurality of vocal parts to be simulated as the singing and the chorus; when multiple persons sing songs, multiple sounds can be mixed together for optimization, and the sounds as solo are adjusted and optimized; when multiple persons sing songs, the combination of singing and sing can be selected at will.
Example 3
Because the distance between the microphone and the sound source affects the recording performance of the input sound signal input module 100 during the actual sound recording or singing process, when a user inputs a fixed volume value at a client, the actually output volume value is not stably output at the fixed volume value but continuously changes along with the difference of the distance position of the microphone, and the problem that the volume value of the subsequently tracked output sound is large in time is caused, so that the sound adjustment effect is greatly affected. In the prior art, the volume of subsequent output is adjusted according to the fed back volume after the actual output of the sound, and the mode has a certain delay defect due to the fact that the feedback is needed, so that the synchronization of the volume output at the same time cannot be ensured.
Referring to the illustration of fig. 7, therefore, the present embodiment also considers the problem that the distance between the sound source and the microphone cannot be stabilized with respect to the actual output volume, and proposes a self-adaptive volume adjustment method based on the sound adjustment module 201. Compared with distance detection, the distance is the root cause of volume difference, different distances need to be calculated and matched with corresponding volumes, corresponding volumes are output after the actual distance is identified, and sounds emitted by different users are definitely different under the same distance, so that the volume emitted by a sound source under the distance cannot be ensured at all, and the adjustment precision is influenced.
Further, in this embodiment, the sound adjusting module 201 specifically further includes the following adjusting steps:
the sound detection module 201a detects an actual volume value of a sound source;
the user inputs a preset volume threshold value into the feedback processing module 201b through the provided interactive interface;
the actual volume value is input into the feedback processing module 201b and is subjected to difference calculation with a built-in preset volume threshold;
inputting the calculated difference value into the compensation suppression module 201c for difference compensation and then outputting a compensation volume value;
the actual volume value and the compensation volume value are synchronously superposed to be used as a common output value;
the common output value is used as a final volume control value of the sound adjusting module 201 to control and adjust the actual volume of the original sound after being output.
In this embodiment, the sound volume is also called loudness and intensity, and refers to subjective feeling of the human ear on the intensity of the heard sound, and the objective evaluation scale is the amplitude of the sound, and the feeling is derived from the pressure generated when the object vibrates, i.e., the sound pressure. The object vibrates through different media, conducting its vibrational energy away. In order to quantify the perception of sound as a monitorable indicator, one divides the sound pressure into "levels" -sound pressure levels, so as to objectively represent the intensity of the sound, which is called "decibels" (dB).
Further, the sound detection module 201a is disposed at the front end of a microphone circuit of the microphone, and an output of the sound detection module 201a is connected to an input of the microphone, wherein the sound detection module 201a includes a microphone, an amplifier, an attenuator, a weighting network, a detector and an indicator, and the sound detection module converts an ambient sound into an electrical signal by matching a pre-amplifier transformation impedance with an output impedance of the microphone, and finally inputs the electrical signal to the effective value detector to output a sound level value. More specifically, in this embodiment, a variable resistor is used as a load to perform impedance matching, and the relationship between the load size and the maximum peak value of the output voltage of the microphone and the saturation current is as follows:
RL=umax/IDSSin the formula umaxIs the maximum peak value of the output voltage of the microphone, IDSSFor saturation current, typically 480 μ A, the maximum peak microphone output voltage is calculated to determine the load size.
Defining the ratio of the microphone output voltage U to the input sound pressure P: s is U/P;
according to the relation between the sound pressure P and the sound pressure level L: l ═ 20lg (P/P)0) Where P is the root mean square sound pressure, P0As reference sound pressure (2X 10)-5Pa), the effective value of the microphone output voltage is known as: u ═ S × p0×10L/20So the maximum peak value of the microphone output voltage:
Umax=K1×S×p0×10Lmax/20in the formula K1Is the peak factor, L, of the output voltage signal of the microphonemaxIs the maximum sound pressure level of the microphone input.
Referring to the schematic of the microphone impedance matching circuit of fig. 6, the load size is determined to be 19.8k Ω according to the above-described impedance matching circuit of the finally determined satisfactory sensor, and the output characteristic of the microphone is: voltage range: -9.5V to +9.5V, and the minimum value of the voltage change is 0.366 μ V.
The sound is converted into electric signal by microphone, the output signal of microphone is passed through amplifier (formed from operational amplifier LM386, and can be used for amplifying trace quantity of A.C. voltage signal, at the same time can be used for full-wave rectification of A.C. signal), so that the amplitude-frequency characteristics of the output voltage signal can meet the measurement requirements, the microphone is an energy conversion device for converting sound wave signal into electric signal, firstly, the amplifier can be used for adding output signal into weighting network, and can be used for making external filter for signal, then the signal can be amplified to a certain amplitude value by means of attenuator and amplifier, and can be fed into effective value detector for detection, and finally the indication head can give out display for.
In this embodiment, the detector converts the ac signal into a dc signal, and sends the dc signal to the indicator to display the measurement result, and the converted dc signal may correspond to the average value, peak value or effective value (RMS) of the ac signal, so there is an average value, peak value or effective value detector. The method comprises the steps of A/D conversion, square, time weighting, logarithmic conversion and display or output, wherein the time weighting is exponential average of a time constant specified for the square of instantaneous sound pressure, a digital signal processor converts a sound pressure signal into a corresponding decibel value and outputs the decibel value, and the conversion relation is as follows: xdB=10lg(P/p0) Wherein X isdBIs the actual displayed decibel value.
Further, the embodiment defines a preset volume threshold SdBFor the target volume value set by the user through the software interface, since the distance of the sound source causes the deviation between the actually output volume and the target volume, the decibel value (actual volume value) output by the sound detection module 201a is input into the feedback processing module 201b for difference calculation: ddB=SdB-XdB. Compensating for rejection based on the resulting difference inputThe internal rotation of the module 201c into the output voltage U is compensated or suppressed, which is a reverse derivative process according to ddB=10lg(P/p0) The actual sound pressure P is calculated, and the ratio S of the output voltage U to the input sound pressure P defined above, i.e., S ═ U/P, is converted into U ═ S · P, and the value of the output voltage U actually compensated or suppressed is known. Converting the voltage value to be compensated into the required human output voltage to be compensated by the driving power supply according to the calculated output voltage U value, outputting the voltage to the sound detection module 201a by using the D/A conversion interface, inputting the compensated or suppressed decibel value to the feedback processing module 201b again for secondary difference calculation, and when D is the calculated output voltage U valuedBWhen the sound signal is less than 0.1, the input sound adjusting module 201 adjusts the sound, and finally the sound signal is output by the sound signal output module 300. It is understood that the suppression of the voltage is implemented by using a synchronous voltage suppressor and a voltage compensator, for example, as compared with the driving power supply provided with a suppression capacitor. And when the calculated difference is a positive number, turning on the voltage device for discharging, and when the difference is a negative number, turning on the voltage device for charging, and finally realizing compensation and inhibition, which is not described in detail.
In this embodiment, to verify the adjustment accuracy of the method, the error of the actually output volume value with respect to the preset value is detected at different distances, the sound source position is given, and the microphone 1 which is not equipped with the method, the microphone 2 which is used for traditional distance detection and the microphone 3 which is used for sound source detection are respectively placed at positions 10mm, 20mm, 40mm, 60mm and 90mm away from the sound source position, in the test, the sound source is selected to be the audio frequency with the stepwise fluctuation of the volume, when the above 5 detection positions are detected, the volume of the corresponding sound source is 10dB, 20dB, 30dB, 40dB and 50dB, the preset microphone output volume is 30dB, the sound volume output by the three microphones is detected in real time by using the sound decibel detector at the sound production positions at the above different distances, the actual test result is as shown in fig. 8-9, as can be known from fig. 8, the microphone 1 has no compensation and inhibition effects, the volume level of the actual output is set to correspond to the volume emission level of the sound source of 10dB, 20dB, 30dB, 40dB and 50dB, and the volume level of the actual output is decreased as the distance increases. The microphone 2 has an adjusting function, and when the position is far and the sound is large, the actual output effect is obviously lower than that of the microphone 3, namely the adaptive adjusting method provided by the embodiment. The error of fig. 8 shows that the error of the method can be stabilized between 0 and 0.2 with the increase of the distance, and the output error of the microphone 2 is as high as 2 with the increase of the distance by using the distance detection, and the output precision is obviously lower than that of the method, so that the adjusting method provided by the embodiment has obvious advantages compared with the traditional detection distance adjusting method.
It should be recognized that embodiments of the present invention can be realized and implemented by computer hardware, a combination of hardware and software, or by computer instructions stored in a non-transitory computer readable memory. The methods may be implemented in a computer program using standard programming techniques, including a non-transitory computer readable storage medium configured with the computer program, where the storage medium so configured causes a computer to operate in a specific and predefined manner according to the methods and figures described in the detailed description. Each program may be implemented in a high level procedural or object oriented programming language to communicate with a computer system. However, the program(s) can be implemented in assembly or machine language, if desired. In any case, the language may be a compiled or interpreted language. Furthermore, the program can be run on a programmed application specific integrated circuit for this purpose.
Further, the operations of processes described herein can be performed in any suitable order unless otherwise indicated herein or otherwise clearly contradicted by context. The processes described herein (or variations and/or combinations thereof) may be performed under the control of one or more computer systems configured with executable instructions, and may be implemented as code (e.g., executable instructions, one or more computer programs, or one or more applications) collectively executed on one or more processors, by hardware, or combinations thereof. The computer program includes a plurality of instructions executable by one or more processors.
Further, the method may be implemented in any type of computing platform operatively connected to a suitable interface, including but not limited to a personal computer, mini computer, mainframe, workstation, networked or distributed computing environment, separate or integrated computer platform, or in communication with a charged particle tool or other imaging device, and the like. Aspects of the invention may be embodied in machine-readable code stored on a non-transitory storage medium or device, whether removable or integrated into a computing platform, such as a hard disk, optically read and/or write storage medium, RAM, ROM, or the like, such that it may be read by a programmable computer, which when read by the storage medium or device, is operative to configure and operate the computer to perform the procedures described herein. Further, the machine-readable code, or portions thereof, may be transmitted over a wired or wireless network. The invention described herein includes these and other different types of non-transitory computer-readable storage media when such media include instructions or programs that implement the steps described above in conjunction with a microprocessor or other data processor. The invention also includes the computer itself when programmed according to the methods and techniques described herein. A computer program can be applied to input data to perform the functions described herein to transform the input data to generate output data that is stored to non-volatile memory. The output information may also be applied to one or more output devices, such as a display. In a preferred embodiment of the invention, the transformed data represents physical and tangible objects, including particular visual depictions of physical and tangible objects produced on a display.
As used in this application, the terms "component," "module," "system," and the like are intended to refer to a computer-related entity, either hardware, firmware, a combination of hardware and software, or software in execution. For example, a component may be, but is not limited to being: a process running on a processor, an object, an executable, a thread of execution, a program, and/or a computer. By way of example, both an application running on a computing device and the computing device can be a component. One or more components can reside within a process and/or thread of execution and a component can be localized on one computer and/or distributed between two or more computers. In addition, these components can execute from various computer readable media having various data structures thereon. The components may communicate by way of local and/or remote processes such as in accordance with a signal having one or more data packets (e.g., data from one component interacting with another component in a local system, distributed system, and/or across a network such as the internet with other systems by way of the signal).
It should be noted that the above-mentioned embodiments are only for illustrating the technical solutions of the present invention and not for limiting, and although the present invention has been described in detail with reference to the preferred embodiments, it should be understood by those skilled in the art that modifications or equivalent substitutions may be made on the technical solutions of the present invention without departing from the spirit and scope of the technical solutions of the present invention, which should be covered by the claims of the present invention.

Claims (10)

1. A sound adjustment system, characterized by: comprises the steps of (a) preparing a mixture of a plurality of raw materials,
the voice signal input module (100), the voice signal input module (100) is used for perceiving and collecting voice signals, and converting the voice signals into original voice data;
the signal processing module (200), the signal processing module (200) is used for receiving the original sound data transmitted by the sound signal acquisition module (100), and the original sound data is adjusted into adjusted sound data through a sound adjustment module (201);
and the sound signal output module (300) is used for receiving the adjusted sound data and converting the adjusted sound data into a sound signal to play.
2. The sound adjustment system of claim 1, wherein: the signal processing module (200) further comprises an input module (202) and an output module (203);
the input module (202) is used for simultaneously receiving one or more original sound data generated by the sound signal acquisition module (100) for pre-adjustment;
the output module (203) is used for receiving the adjusted sound data output by the sound adjusting module (201) and processing the adjusted sound data into an input signal of the sound signal output module (300).
3. A sound adjustment system according to claim 1 or 2, characterized by: the original sound data acquired by the sound signal input module (100) further includes an original loudness, an original pitch, an original tone, and an original time parameter corresponding to the original loudness, the original pitch, and the original tone of the input sound.
4. The sound adjustment system of claim 3, wherein: the signal processing module (200) further comprises an adjustment reference standard module (204), wherein the adjustment reference standard module (204) is used for referring to specified standard sound data, and the standard sound data comprises standard loudness, standard tone and standard time parameters corresponding to the standard loudness, the standard tone and the standard tone.
5. The sound adjustment system of claim 4, wherein: the sound adjusting module (201) is configured to adjust one or more of the original loudness, the original pitch, and the original timbre in the original sound data to the standard loudness, standard pitch, and standard timbre corresponding thereto, and the sound adjusting module (201) performs positioning according to a one-to-one correspondence between the original time parameter and the standard time parameter.
6. Sound adjustment system according to claim 4 or 5, characterized in that: the signal processing module (200) further comprises an intelligent voice matching module (205), and the intelligent voice matching module (205) is configured to analyze voice contents in the original sound data and the standard sound data, perform intelligent matching, and position adjustment of the original loudness, the original tone, and the original tone in the original sound data according to a matching result.
7. The sound adjustment system of claim 6, wherein: the signal processing module (200) further comprises a storage module (206), the storage module (206) further comprises a wait reference call module for storing the standard sound data, and a wait reference call module for receiving the adjusted sound data and storing the adjusted sound data as new sound data, and the new sound data is fed to the input module (202) as the original sound data.
8. A method of adjusting sound, comprising: comprises the following steps of (a) carrying out,
the method comprises the steps that a sound signal input module (100) obtains original sound data;
an input module (202) receives the original sound data for pre-adjustment;
the sound adjusting module (201) combines the adjusting reference standard module (204) and the intelligent voice matching module (205) to position and match the pre-adjusted sound data, and adjusts the pre-adjusted sound data into adjusted sound data;
the output module (203) receives the adjusted sound data and integrates the adjusted sound data into an output signal;
the sound signal output module (300) receives the output signal of the output module (203) and converts the output signal into a voice signal for playing.
9. The sound adjusting method according to claim 8, wherein: the method also comprises the following steps of,
the output module (203) receives the adjusted sound data and integrates the adjusted sound data into an output signal;
the original sound data, the adjusted sound data and the output signal are transmitted to a storage module (206) for storage;
selecting to call the stored raw sound data out of the storage module (206) for batch processing, or selecting to input the output signal to the input module (202) as the raw sound data for next adjustment, and performing the next loop adjustment.
10. A sound adjustment method according to claim 8 or 9, characterized by: the adjustment of the original sound data by the sound adjustment module (201) comprises a self-defined adjustment amount range and further comprises,
when the original loudness is adjusted, the loudness is customized on the basis of the standard loudness;
when the original tone is adjusted, the tone height is customized on the basis of the standard tone;
and when the original timbre is adjusted, the appointed different standard timbres are simultaneously referred for mixing adjustment.
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