CN110634462B - Sound adjusting system and adjusting method - Google Patents
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Classifications
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10H—ELECTROPHONIC 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
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10H—ELECTROPHONIC 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
- G10H1/00—Details of electrophonic musical instruments
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- G10H1/00—Details of electrophonic musical instruments
- G10H1/36—Accompaniment arrangements
- G10H1/361—Recording/reproducing of accompaniment for use with an external source, e.g. karaoke systems
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10H—ELECTROPHONIC 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
- G10H1/00—Details of electrophonic musical instruments
- G10H1/46—Volume control
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 output 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 the sound adjustment module; and the sound signal output module is used for receiving the adjustment sound data and converting the adjustment sound data into sound signals to be played. The invention has the beneficial effects that: people who sing songs and run off easily can adjust the sound according to the requirements, and the sound of running the adjustment is played after being corrected.
Description
Technical Field
The present invention relates to the field of audio processing, and in particular, to a sound adjusting system and an adjusting method.
Background
Along with the improvement of the living standard of people, the pursuit of people on cultural entertainment life is also improved, and the music function has become a necessary application in terminals such as computers or mobile phones due to the high-speed development of communication and information technology. Music products with KTV function are increasingly appeared in the terminals of users, mostly according to the music playing accompaniment selected by the users and displaying the subtitles along with the music accompaniment, the users prompt the singing time corresponding to the lyrics according to font color prompt or other marks on the displayed subtitles until the whole song is completed. In recent years, K songs are becoming a recreation way for people, but because different people have uneven perceptibility of music feel and melody, the situation that the running tone or the rhythm cannot keep pace occurs, and the mood of singing songs is seriously affected. The existing method for solving the problem of running tone or not keeping pace with the rhythm can only find a professional recording studio to carry out later sound trimming treatment so as to correct the running tone. Although the recording studio has higher professional sound repairing performance, under the limitation of long manufacturing period, low manufacturing cost and other conditions, the common people are hardly accepted, and the interest degree of K songs of people is reduced.
Meanwhile, the beauty function of live broadcast software is deeply favored by the public, but the public is served only by the beauty function without the function of 'beauty sound', and when many host songs are played in live broadcast, singing voice is defective and is a great pain point, and the problem is solved by 'beauty sound' synchronous through the access of third party software, so that a user is required to install and register two different software and debug, and the process is complex and complicated.
Disclosure of Invention
This section is intended to outline some aspects of embodiments of the invention and to briefly introduce some preferred embodiments. Some simplifications or omissions may be made in this section as well as in the description summary and in the title of the application, to avoid obscuring the purpose of this section, the description summary and the title of the invention, which should not be used to limit the scope of the invention.
The present invention has been made in view of the above-described problems occurring in the prior art.
Therefore, one technical problem solved by the present invention is: the sound adjusting system is used for adjusting the sound of people during singing, and solves the problem that the tone is frequently easy to run.
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 output module and a sound output 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 the sound adjustment module; and the sound signal output module is used for receiving the adjustment sound data and converting the adjustment sound data into sound signals to be played.
As a preferable embodiment of the sound adjustment system according to the present invention, wherein: the signal processing module further 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 adjustment sound data output by the sound adjustment module and processing the adjustment sound data into the sound signal output module.
As a preferable embodiment of the sound adjustment system according to the present invention, wherein: the original sound data acquired by the sound signal input module further comprises original loudness, original tone color of input sound and original time parameters corresponding to the original loudness, the original tone and the original tone color.
As a preferable embodiment of the sound adjustment system according to the present invention, wherein: the signal processing module further includes an adjustment reference standard module for referencing specified standard sound data including a standard loudness, a standard tone, a standard timbre, and a standard time parameter corresponding to the standard loudness, the standard tone, and the standard timbre.
As a preferable embodiment of the sound adjustment system according to 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 tone in the original sound data to be the standard loudness, the standard tone and the standard tone corresponding to the original loudness, the standard tone and the standard tone, and the sound adjusting module is used for positioning according to the one-to-one correspondence between the original time parameter and the standard time parameter.
As a preferable embodiment of the sound adjustment system according to 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 embodiment of the sound adjustment system according to the present invention, wherein: the signal processing module further comprises a storage module further comprising means for storing the standard sound data waiting for reference call, and further comprising means for receiving the adjusted sound data and storing as new sound data, which is fed to the input module as the original sound data.
The invention solves the other technical problem that: the sound adjusting method can be realized by 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 and performs pre-adjustment; the sound adjusting module is combined with the adjusting reference standard module and the intelligent voice matching module to position, match and adjust the preset sound data into adjusting sound data; the output module receives the adjustment sound data and integrates the adjustment 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 for playing.
As a preferable embodiment of the sound adjustment method according to the present invention, wherein: the output module receives the adjustment sound data to integrate and process the adjustment 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 next cycle adjustment is executed.
As a preferable embodiment of the sound adjustment method according to the present invention, wherein: the sound adjusting module adjusts the original sound data to include a self-defined adjusting quantity range and also self-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 tone is adjusted, simultaneously referring to a plurality of specified different standard tone colors, and performing mixed adjustment.
The invention has the beneficial effects that: people who sing songs and run off easily can adjust the sound according to the requirements, and the sound of running the adjustment is played after being corrected.
Drawings
In order to more clearly illustrate the technical solutions of the embodiments of the present invention, the drawings that are needed in the description of the embodiments will be briefly described below, it being obvious that the drawings in the following description are only some embodiments of the present invention, and that other drawings may be obtained according to these drawings without inventive effort for a person skilled in the art. Wherein:
FIG. 1 is a schematic diagram of the overall principle of a sound adjustment system according to a first embodiment of the present invention;
fig. 2 is a schematic block diagram of a sound adjustment system according to a first embodiment of the present invention;
FIG. 3 is a schematic overall flow chart of a sound adjusting method according to a second embodiment of the invention;
FIG. 4 is a flowchart of an optimized output of an original sound signal according to a second embodiment of the present invention;
FIG. 5 is a schematic representation of a sampled and quantized image of an original sound signal 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 invention;
Fig. 7 is a schematic overall flow chart of a method for adjusting adaptive volume according to a third embodiment of the present invention;
FIG. 8 is a comparative schematic diagram of actual test results according to a third embodiment of the present invention;
FIG. 9 is a comparison diagram of actual test result errors according to a third embodiment of the present invention.
Description of the embodiments
So that the manner in which the above recited objects, features and advantages of the present invention can be understood in detail, a more particular description of the invention, briefly summarized above, may be had by reference to the embodiments, some of which are illustrated in the appended drawings. All other embodiments, which can be made by one of ordinary skill in the art based on the embodiments of the present invention without making any inventive effort, shall fall within the 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 other than those described herein, and persons skilled in the art will readily appreciate that the present invention is not limited to the specific embodiments disclosed below.
Further, reference herein to "one embodiment" or "an embodiment" means that a particular feature, structure, or characteristic can be 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.
While the embodiments of the present invention have been illustrated and described in detail in the drawings, the cross-sectional view of the device structure is not to scale in the general sense for ease of illustration, and the drawings are merely exemplary and should not be construed as limiting the scope of the invention. In addition, the three-dimensional dimensions of length, width and depth should be included in actual fabrication.
Also in the description of the present invention, it should be noted that the orientation or positional relationship indicated by the terms "upper, lower, inner and outer", etc. are based on the orientation or positional relationship shown in the drawings, are merely for convenience of describing the present invention and simplifying the description, and do not indicate or imply that the apparatus or elements referred to must have a specific orientation, be constructed and operated in a specific orientation, and thus should not 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 coupled" should be construed broadly in this disclosure unless otherwise specifically indicated and defined, such as: can be fixed connection, detachable connection or integral connection; it may also be a mechanical connection, an electrical connection, or a direct connection, or may be indirectly connected through an intermediate medium, or may be a communication between two elements. The specific meaning of the above terms in the present invention will be understood in specific cases by those of ordinary skill in the art.
Examples
Referring to fig. 1-2, a sound adjusting system is shown in which K songs are a recreation way, but because different people have uneven perceptibility of feelings and melodies, the situation that the running tone or the rhythm cannot keep up occurs, and the moods of singing and listeners are seriously affected. Therefore, the present embodiment proposes a sound adjusting system for adjusting the sound of running tone, the system includes a sound signal input module, a signal processing module 200 and a sound signal output module 300, wherein the output of the sound signal input module is connected with the input of the signal processing module 200, and the output of the signal processing module 200 is connected with the input of the sound signal output module 300. Specifically, the sound signal input module collects the sound signal of the original human voice which is not subjected to any post-processing and processing, so in this embodiment, the sound signal input module 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, i.e. an electrical signal, and a specific form of the original sound data is, for example, an mp3 converted signal. Further, the microphone is called a microphone, and is an energy conversion device for converting a sound signal into an electric signal, and is also called a microphone or a microphone. In the twentieth century, microphones were developed from the initial development of resistive-to-acoustic-to-electric conversion to inductive-to-capacitive conversion, and a number of new microphone technologies were developed, including aluminum ribbon, moving coil, and other microphones, as well as currently widely used capacitive microphones and electret microphones, in which sound vibrations were transmitted to the diaphragm of the microphone to push the magnet inside to form a varying current, and the varying current was sent to a subsequent sound processing circuit for amplification processing, to obtain the desired sound data. The signal processing module 200 receives the original sound data and outputs the adjusted sound data after adjusting the original sound data with reference to the standard sound. The sound signal output module 300 receives and plays the adjusted sound data, and in this embodiment, the sound signal output module 300 uses sound equipment or equipment with a speaker.
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 200, the signal processing module 200 is configured to receive the original sound data transmitted by the sound signal acquisition module, and adjust the original sound data to adjusted sound data through the sound adjustment module 201; 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 simultaneously receive one or more original sound data generated by the sound signal acquisition module for pre-adjustment; the output module 203 is configured to receive the adjusted sound data output by the sound adjustment module 201 and process the adjusted sound data into a sound signal output module 300. The original sound data acquired by the sound signal input module further comprises original loudness, original tone, original timbre of the input sound and original time parameters corresponding to the original loudness, the original tone and the original timbre. The signal processing module 200 further includes an adjustment reference standard module 204, where the adjustment reference standard module 204 is configured to reference specified standard sound data, where the standard sound data includes standard loudness, standard tone, standard timbre, and standard time parameters corresponding to the standard loudness, standard tone, and standard timbre.
The sound adjustment 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 a standard loudness, a standard pitch, and a standard timbre corresponding to the original loudness, the original pitch, and the original timbre, and the sound adjustment module 201 performs positioning according to the 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, where the intelligent voice matching module 205 is configured to analyze voice content in the original sound data and the standard sound data, perform intelligent matching, and position adjustment of original loudness, original pitch, and original timbre in the original sound data according to a matching result. The signal processing module 200 further comprises a storage module 206, the storage module 206 further comprising means for storing standard sound data waiting for reference calls, and further comprising means for receiving the adjusted sound data and storing as new sound data, which is fed as raw sound data to the input module 202.
It should be further noted that, the signal processing module 200 is an integrated circuit board device with embedded algorithm program, and the sound adjusting module 201, the input module 202, the output module 203, the adjusting reference standard module 204 and the storage module 206 are integrated circuit modules on the integrated circuit board by circuit integration, and the algorithm program is written into the integrated circuit board by the existing program writing method, so that a plurality of integrated circuits, such as a microcontroller and a microprocessor in the integrated circuit, are also commonly called as a single chip microcomputer. The algorithm program is an algorithm for adjusting sound data, the writing of the algorithm program is that, for example, a downloader is connected into a computer, then the writing port of the downloader is aligned with the writing port of a circuit board, the circuit board is connected to supply power, the downloader software is opened in the computer after the 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 to fill in the input information, and the writing program is clicked. A circuit board device with a sound adjustment algorithm program is obtained, i.e. the present embodiment proposes a hardware device of the signal processing module 200.
Examples
Referring to the illustrations of fig. 3-5, the present embodiment provides a sound adjusting method, applied to the above system, for correcting the tone running sound, comprising the following steps,
s1: the voice signal input module obtains the original voice data, in this step, the singer provides the original data or adjusts the audio frequency 'batch' made by the singer in the hard disk, then the storage provides the original data in the case of adjusting the batch audio frequency, and the step further includes the step of splitting the original voice data into the digital signals of the original loudness, the original tone and the original tone.
Specifically, the digital processing of the sound signal includes sampling, quantization and encoding, where sampling refers to the process of converting a time-continuous analog signal into a digital signal, where the amplitude and time of the converted digital signal are discrete. The sampling process comprises two important physical quantities of a sampling period and a sampling frequency, wherein the sampling period is a sampling time interval of each time, and the reciprocal of the sampling period is the sampling frequency. And the higher the sampling frequency is, the more data is 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 is at least twice the highest frequency of the sound signal. The quantization of the sound signal means that each sample of the sample is represented by a discrete value, which is called analog-to-digital conversion, and the sample is quantized and presented in a binary number mode, wherein the binary number is the quantization precision. It is apparent that quantization has higher accuracy, sound has higher quality, but the more storage space is required instead. In this embodiment, the image of the original sound signal after being sampled and quantized may refer to the schematic diagram of fig. 5.
Further, the audio signal is quantized and represented in digital form, but it is encoded and compressed according to a certain rule for convenient transmission, processing and storage. The sound signal is encoded to obtain digital audio. The volume of sound information is increasing, and the space for storing is also increasing, and assuming that analog sound is digitized and then not compressed in bytes, the storage volume can be calculated by the following formula:
memory = sampling frequency x number of quantization bits/8 x number of channels x time.
For example, the sampling frequency is 44.1KHz, the recording of 1s stereo is performed using 16 bits of quantization bit, and the storage space required for generating the waveform=44100×16/8×2×1= 176400 bytes. Therefore, the problem of storage space is also considered during sound processing, and the integrity of sound processing information storage is ensured.
Wherein the tone is mainly related to the frequency of the waveform, and the frequency of the sound wave is high, so is the tone; the loudness is related to the amplitude of waveform vibration, and the greater the vibration amplitude is, the greater the loudness is; timbre is a characteristic by which one distinguishes two sounds of the same loudness, the same pitch, and is related to the shape of the vibration waveform of the sound wave, or to the spectral structure of the sound. The waveform signal generated by the sound can encode and de-read the signals of the original loudness, the original tone and the original timbre 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-conditioning, which includes using digital signal processing algorithms to make the sound softer, more magnetic, and free of noise, etc., and digital signal processing techniques are techniques for converting analog information (e.g., sound of the present embodiment) into digital information. Usually refers to chips or processors that perform these functions, or they may also be used to process this information and then output it as an analog electrical signal, obviously in this embodiment a pre-conditioning of the digital signal. This step therefore includes a process of performing noise reduction processing on the sound signal.
S3: the voice adjustment module 201 combines the adjustment reference standard module 204 and the intelligent voice matching module 205 to position, match and adjust the pre-adjusted voice data into adjusted voice data. In this step, the sound adjusting module 201 adjusts one or more of the original loudness, the original tone and the original timbre in the original sound data to the standard loudness, the standard tone and the standard timbre corresponding to the original loudness, the original tone and the original timbre, 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 one song, for the same time parameter of the original sound data and the standard sound data, the original loudness corresponds to the standard loudness, the original tone corresponds to the standard tone, and the original timbre corresponds to the standard timbre. The adjustment reference standard module 204 stores standard sound data in advance, and calls it when necessary.
The intelligent voice matching module 205 analyzes voice contents in the original voice data and the standard voice data, performs intelligent matching, and positions the adjustment of the original loudness, the original tone and the original tone in the original voice data according to the matching result. The positioning of the intelligent voice matching module 205 and the voice adjusting module 201 are two independent positioning modes, which can be selected by the user, i.e. the matching positioning can be performed according to the time parameter or the matching positioning can be performed according to the lyrics.
In this embodiment, the time location is for example 2 minutes 06 seconds (location) of the song, the original singing voice is so (reference standard data), and the singer sings out voice is mi (original data), and the system adjusts the singer singing mi to so at the time node of 2 minutes 06 seconds.
The intelligent voice matching module 205 performs recognition according to the voice content of the original voice data, and matches the voice content in the standard voice data, for example, matches the lyric content sung by the singer with the original lyrics. And in this embodiment, the adjustment of the original sound data by the sound adjustment module 201 includes a custom adjustment range, and further includes,
when the original loudness is adjusted, the size of 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;
when the original tone color is adjusted, the mixing adjustment is performed by referring to a plurality of specified different standard tone colors at the same time.
In this embodiment, for example, when the user sings, the original singing is the tone of the singer Liu Mou, and the singer Liu Mou or other reference standard tone can completely cover the original tone of the user through customization, so that the singed song is a new singer Liu Mou singing version; at the same time, the user may choose to blend the timbre of 30% of the singer Liu Mou into the original timbre, leaving the user's voice to sound with the sensation of 30% of the singer Liu Mou. The user can actively select the scheme to be adjusted through the operation interface, for example, in ktv, a singer can manually select what effect to listen to in the system before singing, or can directly use the effect in mobile phone software.
S4: the output module 203 receives the adjusted sound data and performs integration processing to output signals, specifically, integrates the adjusted sound into digital signals (electric signals) and outputs the digital signals, wherein the integration processing comprises batch packing and sorting of the digital signals, and performs song names or singer labels on the sound signals packed by the digital signals, so that the user can conveniently perform secondary or more adjustment processing operations on the singing sounds stored by the user;
S5: the sound signal output module 300 receives the output electric signal of the output module 203, converts the output electric signal into voice content, plays the voice content, obtains the "beautified" sound signal in this step, and then transmits the "beautified" sound signal to the sound, or directly stores the "beautified" sound signal in the storage module 206, inputs the stored digital signal (electric signal) into the sound adjustment module 201 again through the storage module 206, and outputs the "beautified" sound signal after readjustment, and if the user does not satisfy the adjustment of this time, the output module 203 may execute the cyclic adjustment from the storage module 206 to the sound adjustment module 201 until the user is satisfied with the sound data.
The storage module 206 in this embodiment further includes the following steps,
the output module 203 receives the adjusted sound data for integration processing to output signals;
the original sound data, the adjusted sound data and the output signal are transmitted to the storage module 206 for storage;
the raw sound data stored 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 raw sound data for the next adjustment, and the next cycle adjustment is performed.
In this embodiment, the storage module 206 is compared with the storage of the standard sound data stored in the adjustment reference standard module 204, and the storage module 206 additionally stores more original sound data (for example, the record of singing before me can be stored therein, and the user can adjust the sound data at any time), and the storage module can store the adjusted sound data therein (for example, when the system is used, the user can beautify the user's voice during singing, and can process and store the record of singing before the user in batch).
The method can process a plurality of adjustment tasks simultaneously, and the standard sound data can also be directly used as original sound data and adjustment sound data.
In the practical use process, people often run and tune easily when singing, so the method of the embodiment is mainly used for singing, live broadcasting and the like. By analyzing the original song, the singer tone (intonation) is adjusted according to the original tone; and (3) carrying out tone adjustment according to the original song time axis (accompaniment) or intelligent voice (lyrics) matching positioning music position. Beautifying the singer's voice, and mixing the original tone into the singer's voice to different degrees; and is suitable for solo singing, chorus or singing, and a single person can have singing effects, such as:
singers (KTV, live broadcast) can beautify the singer through the system due to insufficient tone and poor tone quality when singing; when a person sings, the system acquires the original singing tone to modify the singer tone, and optionally, the system can acquire the original singing tone to modify the singer tone to a certain extent; when a person sings, optionally, the person can choose to sing the chorus (chorus) ", when the original sings are re-singing, the person can also choose to sing one sound part, and the person can sing other sound parts, so as to carry out re-singing; when a person sings, optionally, when the original singing is re-singing or chorus, the own voice can be adjusted and output into a plurality of voice parts to simulate re-singing or chorus; when a plurality of persons sings, a plurality of sounds can be mixed together for optimization, and the sounds are adjusted and optimized as singing sounds; when a plurality of persons sings, the combination of singing and chorus can be arbitrarily selected.
Examples
Because in the process of recording or singing actual sound, the distance between the microphone and the sound source can influence the recording performance of the input sound signal input module, when a user inputs a fixed sound volume value at a client, the volume of the actual output is not stably output on the fixed sound volume value but continuously changes along with the difference of the distance between the microphone and the sound source, and the problem that the volume of the follow-up output sound is large in hours can occur, so that the effect of sound adjustment is very influenced. In the prior art, aiming at the problems, the volume of the subsequent output is regulated according to the volume fed back after the actual output of the sound, and the mode has a certain delay defect due to the need of feeding back, so that the synchronization of the volume of the output at the same time can not be ensured.
Referring to fig. 7, the present embodiment also considers the problem that the position distance between the sound source and the microphone cannot stabilize the actual output volume, and proposes an adaptive volume adjustment method based on the sound adjustment module 201, unlike the prior art, the method detects in advance at the front end of the sound input device, and the adjustment occurs at the front end, thereby avoiding the problem of asynchronism during adjustment, and having higher stability precision and instantaneity. Compared with distance detection, as the distance is the root cause of the volume difference, different distances need to calculate and match corresponding volumes, when the actual distance is identified, the corresponding volumes are output, and the sounds sent by different users at the same distance are definitely different, so that the volume of the sound source at the distance cannot be ensured at all, the adjustment precision is influenced, and the method for directly detecting the volume of the sound source obviously has higher adjustment precision relative to the detection distance.
Further, the sound adjusting module 201 in this embodiment specifically further includes the following adjusting steps:
the sound detection module 201a detects an actual sound 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 to calculate the difference value with the built-in preset volume threshold value;
inputting the calculated difference value into the compensation suppression module 201c for difference compensation and outputting a compensation sound value;
the actual volume value and the compensation volume value are synchronously overlapped 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 after the original sound is output.
In this embodiment, the volume is also called loudness and intensity, and refers to subjective feeling of the ear on the intensity of the sound, and the objective evaluation scale is the amplitude of the sound, and this feeling is derived from the pressure generated when the object vibrates, i.e. sound pressure. The object vibrates through different media, which conducts its vibrational energy away. In order to quantify the perception of sound into a monitorable index, sound pressure is divided into "levels", i.e. sound pressure levels, so that the intensity of sound can be objectively represented, and the unit is called "decibel" (dB).
Further, the sound detection module 201a is disposed at the front end of the microphone circuit of the microphone, and the output of the sound detection module 201a is connected with the input of the microphone, where the sound detection module 201a includes a microphone, an amplifier, an attenuator, a weighting network, a detector and an indicator, and the environmental sound is converted into an electrical signal by matching the transformation impedance of the preamplifier with the output impedance of the microphone, and finally the electrical signal is input to the effective value detector to output a sound level value. More specifically, in this embodiment, the 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 microphone output voltage and the saturation current is:
R L ＝u max /I DSS in the formula u max Is the maximum peak value of microphone output voltage, I DSS For saturation current, typically 480 μA, the maximum peak of the microphone output voltage is calculated to determine the load size.
Defining the ratio of microphone output voltage U to input sound pressure P: s=u/P;
according to the relation between the sound pressure P and the sound pressure level L: l=20 lg (P/P) 0 ) Wherein P is square root sound pressure, P 0 For reference sound pressure (2X 10) -5 Pa), the effective value of the microphone output voltage is: u=s×p 0 ×10 L/20 The maximum peak of the microphone output voltage:
U max ＝K 1 ×S×p 0 ×10L max/20 k in the formula 1 Is the peak factor of the microphone output voltage signal, L max Is the maximum sound pressure level of the microphone input.
Referring to the microphone impedance matching circuit schematic of fig. 6, according to the impedance matching circuit of the sensor which is finally determined to be satisfactory, the load size is determined to be 19.8kΩ, and the output characteristics of the microphone are as follows: voltage range: -9.5V- +9.5v, the minimum value of the voltage change is 0.366 μv.
The sound is converted into an electric signal through a microphone, the output signal of the microphone is formed by an amplifier (an operational amplifier LM386 is used for amplifying a trace alternating voltage signal and full-wave rectifying the alternating voltage signal), so that the output amplitude-frequency characteristic meets the voltage signal required by measurement, the microphone is an energy conversion device for converting an acoustic wave signal into the electric signal, firstly, the amplifier is used for adding the output signal into a weighting network, externally connecting a filter to the signal, then the signal is amplified to a certain amplitude value through an attenuator and the amplifier, and is sent to an effective value detector for detection, and finally, an LED nixie tube is driven to display on the indicator.
In this embodiment, the detector converts the ac signal into the 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, squaring, time weighting, logarithmic conversion and display or output, wherein the time weighting is exponential averaging of a square regulated time constant of instantaneous sound pressure, and a digital signal processor converts a sound pressure signal into a corresponding decibel value and outputs the corresponding decibel value, and the conversion relation is that: x is X dB ＝10lg(P/p 0 ) Wherein X is dB Is the decibel value actually displayed.
Further, the present embodiment defines a preset volume threshold S dB For 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 present embodiment inputs the decibel value (actual volume value) output by the above-mentioned sound detection module 201a into the feedback processing module 201b to perform the difference calculation: d, d dB =S dB －X dB . The resulting difference is input into compensation suppression module 201c to be converted into output voltage U for compensation or suppression, which is the inverse derivation process, according to d dB ＝10lg(P/p 0 ) The actual sound pressure P is calculated, and the ratio S of the output voltage U to the input sound pressure P defined above, that is, s=u/P, is converted into u=s·p, so that the output voltage U value actually compensated or suppressed can be obtained. According to the calculated output voltage U value, converting the voltage value to be compensated into the corresponding required input voltage to be compensated by the driving power supply, outputting the voltage to the sound detection module 201a by using the D/A conversion interface for output, inputting the compensated or suppressed decibel value into the feedback processing module 201b for secondary difference calculation, inputting the voltage to the sound adjustment module 201 for sound adjustment when ddB is smaller than 0.1, and finally outputting the voltage by the sound signal output module 300. It will be appreciated that the suppression of the voltage is relative to the provision of the drive power supply, e.g. the provision of a suppression capacitor The voltage compensation and suppression of the present embodiment can be achieved by employing both the synchronous voltage suppressor and the voltage compensator. When the calculated difference is positive, the voltage device is turned on to discharge, and when the difference is negative, the voltage device is turned on to charge, and compensation and suppression are finally realized, which is not described in detail.
In order to verify the adjustment accuracy of the method, the error of the actual output volume value relative to the preset value is detected at different distances, the sound source position is given, the positions which are 10mm, 20mm, 40mm, 60mm and 90mm away from the sound source position are respectively provided with a microphone No. 1, a microphone No. 2 which is not provided with the method and is used for detecting the traditional distance, the sound source is selected to be the audio with the sound source with the stepwise fluctuation in the test, the volume of the corresponding sound source is 10dB, 20dB, 30dB, 40dB and 50dB when the sound source is detected at the 5 detection positions, the volume of the preset microphone is 30dB, the sound detector is used for detecting the volume of the three microphone outputs in real time when the sound source is detected at the sound generation positions at the different distances, the actual test result is as shown in fig. 8-9, and as the sound source 1 has no compensation and suppression effects, the actual output volume is 10dB, 20dB, 30dB, 40dB and 50dB, and the actual output volume is reduced when the sound source is detected at the different distances. And the microphone No. 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 No. 3, namely the self-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 along with the increase of the distance, and the output error of the microphone 2 is as high as 2 along with the increase of the distance by utilizing 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 mode.
It should be appreciated that embodiments of the invention may be implemented or realized 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 a computer program, where the storage medium so configured causes a computer to operate in a specific and predefined manner in accordance with 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.
Furthermore, the operations of the processes described herein may be performed in any suitable order unless otherwise indicated herein or otherwise clearly contradicted by context. The processes (or variations and/or combinations thereof) described herein may be performed under 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), by hardware, or combinations thereof, collectively executing on one or more processors. 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 computing platform, including, but not limited to, a personal computer, mini-computer, mainframe, workstation, network or distributed computing environment, separate or integrated computer platform, or in communication with a charged particle tool or other imaging device, and so forth. Aspects of the invention may be implemented 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, optical read and/or write storage medium, RAM, ROM, etc., such that it is readable by a programmable computer, which when read by a computer, is operable to configure and operate the computer to perform the processes described herein. Further, the machine readable code, or portions thereof, may be transmitted over a wired or wireless network. When such media includes instructions or programs that, in conjunction with a microprocessor or other data processor, implement the steps described above, the invention described herein includes these and other different types of non-transitory computer-readable storage media. The invention also includes the computer itself when programmed according to the methods and techniques of the present invention. The computer program can be applied to the input data to perform the functions described herein, thereby converting the input data to generate output data that is stored to the 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 specific 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, the components may be, but are not limited to: 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 may reside within a process and/or thread of execution and a component may be localized on one computer and/or distributed between two or more computers. Furthermore, 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 embodiments are only for illustrating the technical solution of the present invention and not for limiting the same, 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 the technical solution of the present invention may be modified or substituted without departing from the spirit and scope of the technical solution of the present invention, which is intended to be covered in the scope of the claims of the present invention.
Claims (2)
1. A sound adjustment system, characterized by: comprising the steps of (a) a step of,
the sound signal input module is used for sensing and collecting sound signals and converting the sound signals into original sound data;
a signal processing module (200), wherein the signal processing module (200) 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 (201);
the sound signal output module (300) is used for receiving the adjustment sound data and converting the adjustment sound data into sound signals to be played;
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 pieces of original sound data generated by the sound signal acquisition module for pre-adjustment;
The output module (203) is configured to receive the adjusted sound data output by the sound adjustment module (201) and process the adjusted sound data into an input signal of the sound signal output module (300);
the original sound data acquired by the sound signal input module further comprises original loudness, original tone color of input sound and original time parameters corresponding to the original loudness, the original tone and the original tone color;
the signal processing module (200) further comprises an adjustment reference standard module (204), the adjustment reference standard module (204) being configured to reference specified standard sound data, the standard sound data comprising a standard loudness, a standard pitch, a standard timbre, and a standard time parameter corresponding to the standard loudness, the standard pitch, the standard timbre;
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, 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;
Inputting the decibel value output by the sound detection module (201 a) into the feedback processing module (201 b) for difference calculation: d, d dB ＝S dB －X dB The method comprises the steps of carrying out a first treatment on the surface of the The obtained difference is input into a compensation suppression module (201 c) to be converted into an output voltage U for compensation or suppression, which is the process of reverse derivation and is based on d dB ＝10lg(P/p 0 ) Calculating the actual sound pressure P, and converting the ratio S of the defined output voltage U to the input sound pressure P, namely S=U/P, into U=S.P to obtain the output voltage U value of actual compensation or suppression; according to the calculated output voltage U value, converting the voltage value to be compensated into the corresponding required input voltage to be compensated by the driving power supply, outputting the voltage to the sound detection module (201 a) by using the D/A conversion interface for output, inputting the compensated or suppressed decibel value into the feedback processing module (201 b) for secondary difference value calculation, and when D dB When the input sound is smaller 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);
the signal processing module (200) further comprises an intelligent voice matching module (205), wherein the intelligent voice matching module (205) is used for analyzing voice contents in the original voice data and the standard voice data and performing intelligent matching, and positioning the adjustment of the original loudness, the original tone and the original tone in the original voice data according to a matching result;
The signal processing module (200) further comprises a storage module (206), the storage module (206) further comprising means for storing the standard sound data waiting for reference call, and further comprising means for receiving the adjusted sound data and storing as new sound data, which new sound data is fed as the original sound data to the input module (202);
the sound adjusting module (201) further comprises the steps of calculating and matching corresponding sound volumes according to different distances, directly detecting the sound source size, and outputting the corresponding sound volume size after the actual distance is identified;
according to the relation between the sound pressure P and the sound pressure level L: l=20 lg (P/P) 0 )
Wherein P is square root sound pressure, P 0 Is the reference sound pressure of 2 multiplied by 10 -5 Pa, the effective value of the microphone output voltage is:
U＝S×p 0 ×10 L/20
maximum peak of microphone output voltage:
Umax＝K 1 ×S×p 0 ×10 Lmax/20
k in the formula 1 Is the peak factor of the microphone output voltage signal, L max Is the maximum sound pressure level of the microphone input.
2. A sound conditioning method using the system of claim 1, characterized by: comprises the steps of,
the sound signal input module acquires original sound data;
an input module (202) receives the original sound data for pre-adjustment;
the sound adjusting module (201) is combined with the adjusting reference standard module (204) and the intelligent voice matching module (205) to position and match the preset sound data and adjust the sound data into adjusting sound data;
An output module (203) receives the adjustment sound data and performs integration processing to obtain 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;
the output module (203) receives the adjustment sound data and performs integration processing to obtain an output signal;
the original sound data, the adjustment sound data and the output signal are transmitted to a storage module (206) for storage;
selecting to call out the stored original sound data from the storage module (206) for batch processing, or selecting to input the output signal to the input module (202) as the original sound data for next adjustment, and executing next cycle adjustment;
the adjustment of the original sound data by the sound adjustment module (201) comprises a custom adjustment range, and further comprises,
when the original loudness is adjusted, the size of 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;
when the original tone color is adjusted, the mixing adjustment is performed by referring to a plurality of specified different standard tone colors at the same time.
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| CN108801443A (en) * | 2018-06-20 | 2018-11-13 | 核工业理化工程研究院 | The noise high precision measuring device and measurement method of high rotation speed servo motor |
| CN113641858A (en) * | 2021-08-27 | 2021-11-12 | 傅静芬 | Management analysis system for acquiring big data based on video sound |
| CN113873387A (en) * | 2021-10-25 | 2021-12-31 | 深圳千岸科技股份有限公司 | Volume adjusting method and system based on earphone terminal |
| CN116092458A (en) * | 2023-02-10 | 2023-05-09 | 牡丹江师范学院 | Vocal music adjusting system and method based on big data |
Citations (10)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| US5641927A (en) * | 1995-04-18 | 1997-06-24 | Texas Instruments Incorporated | Autokeying for musical accompaniment playing apparatus |
| CN1684509A (en) * | 2004-04-13 | 2005-10-19 | 乐金电子（沈阳）有限公司 | Audio frequency processing device and method for digital TV set |
| CN1885871A (en) * | 2005-06-24 | 2006-12-27 | 乐金电子(中国)研究开发中心有限公司 | Apparatus and method for adjusting voice volume of mobile communication terminal |
| CN101277412A (en) * | 2008-03-31 | 2008-10-01 | 深圳创维数字技术股份有限公司 | Method and system for balancing set-top box output volume |
| CN101783656A (en) * | 2010-03-17 | 2010-07-21 | 北京爱德发科技有限公司 | Loudness control method, module and device of stereo system |
| CN101860774A (en) * | 2010-05-31 | 2010-10-13 | 中山大学 | Voice equipment and method capable of automatically repairing sound |
| CN104282316A (en) * | 2013-07-01 | 2015-01-14 | 亿览在线网络技术（北京）有限公司 | Karaoke scoring method based on voice matching, and device thereof |
| CN104538011A (en) * | 2014-10-30 | 2015-04-22 | 华为技术有限公司 | Tone adjusting method and device and terminal device |
| CN105592394A (en) * | 2016-03-04 | 2016-05-18 | 南京迈宸科技有限公司 | High-sensitivity pickup and directional sound amplifying device and method thereof |
| CN107103915A (en) * | 2016-02-18 | 2017-08-29 | 广州酷狗计算机科技有限公司 | A kind of audio data processing method and device |
Family Cites Families (5)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| US20070195963A1 (en) * | 2006-02-21 | 2007-08-23 | Nokia Corporation | Measuring ear biometrics for sound optimization |
| CN108074557B (en) * | 2017-12-11 | 2021-11-23 | 深圳Tcl新技术有限公司 | Tone adjusting method, device and storage medium |
| CN108307273A (en) * | 2018-02-05 | 2018-07-20 | 广东小天才科技有限公司 | A kind of microphone |
| CN108683975A (en) * | 2018-05-14 | 2018-10-19 | 维沃移动通信有限公司 | A kind of audio frequency apparatus |
| CN109448735B (en) * | 2018-12-21 | 2022-05-20 | 深圳创维－Rgb电子有限公司 | Method and device for adjusting video parameters based on voiceprint recognition and read storage medium |
-
2019
- 2019-09-30 CN CN201910942540.4A patent/CN110634462B/en active Active
Patent Citations (10)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| US5641927A (en) * | 1995-04-18 | 1997-06-24 | Texas Instruments Incorporated | Autokeying for musical accompaniment playing apparatus |
| CN1684509A (en) * | 2004-04-13 | 2005-10-19 | 乐金电子（沈阳）有限公司 | Audio frequency processing device and method for digital TV set |
| CN1885871A (en) * | 2005-06-24 | 2006-12-27 | 乐金电子(中国)研究开发中心有限公司 | Apparatus and method for adjusting voice volume of mobile communication terminal |
| CN101277412A (en) * | 2008-03-31 | 2008-10-01 | 深圳创维数字技术股份有限公司 | Method and system for balancing set-top box output volume |
| CN101783656A (en) * | 2010-03-17 | 2010-07-21 | 北京爱德发科技有限公司 | Loudness control method, module and device of stereo system |
| CN101860774A (en) * | 2010-05-31 | 2010-10-13 | 中山大学 | Voice equipment and method capable of automatically repairing sound |
| CN104282316A (en) * | 2013-07-01 | 2015-01-14 | 亿览在线网络技术（北京）有限公司 | Karaoke scoring method based on voice matching, and device thereof |
| CN104538011A (en) * | 2014-10-30 | 2015-04-22 | 华为技术有限公司 | Tone adjusting method and device and terminal device |
| CN107103915A (en) * | 2016-02-18 | 2017-08-29 | 广州酷狗计算机科技有限公司 | A kind of audio data processing method and device |
| CN105592394A (en) * | 2016-03-04 | 2016-05-18 | 南京迈宸科技有限公司 | High-sensitivity pickup and directional sound amplifying device and method thereof |
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|---|---|
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