CN112492450B - Sound parameter regulation and control method, device and computer readable storage medium - Google Patents
Sound parameter regulation and control method, device and computer readable storage medium Download PDFInfo
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- 230000003993 interaction Effects 0.000 claims description 34
- 239000000428 dust Substances 0.000 claims description 29
- 230000001276 controlling effect Effects 0.000 claims description 24
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- 230000002452 interceptive effect Effects 0.000 claims description 7
- 238000004590 computer program Methods 0.000 claims description 6
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R1/00—Details of transducers, loudspeakers or microphones
- H04R1/12—Sanitary or hygienic devices for mouthpieces or earpieces, e.g. for protecting against infection
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
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Abstract
The invention discloses a sound parameter regulation and control method, equipment and a computer readable storage medium, wherein the method comprises the following steps: sending out preset dynamic frequency sweeping audio through a loudspeaker of the terminal equipment; then, synchronously recording the dynamic frequency-sweeping audio through a microphone of the terminal equipment to obtain recorded audio corresponding to the dynamic frequency-sweeping audio; then, analyzing the dynamic frequency sweeping audio and the recorded audio to obtain frequency response parameters and loudness parameters; and finally, respectively comparing the frequency response parameter and the loudness parameter with preset reference parameters to obtain a parameter difference value, and obtaining a first compensation parameter of the loudspeaker and a second compensation parameter of the microphone according to the parameter difference value. The humanized sound parameter regulation and control scheme is realized, the phenomenon that the subjective feeling sound of a user is reduced or distorted due to blockage of a loudspeaker or a microphone is avoided, the audio effect of the loudspeaker or the microphone is improved, and the user experience is enhanced.
Description
Technical Field
The present invention relates to the field of mobile communications, and in particular, to a method and apparatus for adjusting and controlling sound parameters, and a computer readable storage medium.
Background
In the prior art, with the continuous development of intelligent terminal devices, the requirements of users for sound recording and sound playing of the terminal devices are also higher and higher. For example, the audio parameters of the current mobile phone products and electronic products are stored in the device in the form of parameter files, and when in use, the current parameters and corresponding channels are directly called, or the playing and the size control of the sound are carried out through related application programs. However, during the daily use of the user, as the use time increases, or in a poor environment when used for a long time, the sound recording and playing of the terminal device may not meet the use requirement of the user, for example, some terminal devices may cause the sound inlet holes of some microphones to be blocked, or the dust-proof screen of some speakers to be blocked, etc. due to the long-term use, so that the user subjectively experiences the sound to be smaller or distorted. Therefore, a solution for solving the poor sound recording and playing effects of the speaker and microphone is needed.
Disclosure of Invention
In order to solve the technical defects in the prior art, the invention provides a sound parameter regulating and controlling method, which comprises the following steps:
sending out preset dynamic frequency sweeping audio through a loudspeaker of the terminal equipment;
synchronously recording the dynamic frequency-sweeping audio through a microphone of the terminal equipment to obtain recorded audio corresponding to the dynamic frequency-sweeping audio;
analyzing the dynamic frequency sweeping audio and the recorded audio to obtain frequency response parameters and loudness parameters;
and respectively comparing the frequency response parameter and the loudness parameter with preset reference parameters to obtain a parameter difference value, and obtaining a first compensation parameter of the loudspeaker and a second compensation parameter of the microphone according to the parameter difference value.
Optionally, before the preset dynamic frequency sweeping audio is sent out through the speaker of the terminal device, the method includes:
presetting a dust removal period of the terminal equipment according to the service time of the loudspeaker and/or the microphone;
and carrying out ultrasonic dust removal on the loudspeaker and/or the microphone according to the dust removal period.
Optionally, the sending, by the speaker of the terminal device, a preset dynamic frequency-sweeping audio includes:
Presetting audio parameters and audio playing time of the dynamic frequency sweeping audio according to the using state and the using environment of the loudspeaker and/or the microphone;
and playing the dynamic frequency-sweeping audio composed of the audio parameters according to the audio playing time through the loudspeaker.
Optionally, the recording the dynamic frequency-sweeping audio synchronously through the microphone of the terminal device, to obtain a recorded audio corresponding to the dynamic frequency-sweeping audio, includes:
selecting one loudspeaker from a plurality of loudspeakers to play the dynamic frequency sweeping audio, and simultaneously determining one microphone corresponding to the one loudspeaker;
and synchronously recording the dynamic frequency sweeping audio through one microphone.
Optionally, the analyzing the dynamic frequency-sweeping audio and the recorded audio to obtain a frequency response parameter and a loudness parameter includes:
acquiring a first preset parameter of the loudspeaker, and simultaneously acquiring a second preset parameter of the microphone;
and analyzing the dynamic frequency sweeping audio and the recorded audio by combining the first preset parameter and the second preset parameter to obtain the frequency response parameter and the loudness parameter.
Optionally, the comparing the frequency response parameter and the loudness parameter with preset reference parameters to obtain parameter differences, and obtaining a first compensation parameter of the speaker and a second compensation parameter of the microphone according to the parameter differences, including:
Analyzing the frequency response parameters by combining the first preset parameters, the second preset parameters and the reference parameters to obtain a first difference value of the frequency response parameters;
and analyzing the loudness parameter by combining the first preset parameter, the second preset parameter and the reference parameter to obtain a second difference value of the loudness parameter.
Optionally, the comparing the frequency response parameter and the loudness parameter with preset reference parameters to obtain parameter differences, and obtaining a first compensation parameter of the speaker and a second compensation parameter of the microphone according to the parameter differences, and further includes:
presetting an interactive interface for regulating and controlling sound parameter compensation;
and presetting a first interaction option corresponding to the first difference value in the interaction interface, and presetting a second interaction option corresponding to the second difference value in the interaction interface.
Optionally, the comparing the frequency response parameter and the loudness parameter with preset reference parameters to obtain parameter differences, and obtaining a first compensation parameter of the speaker and a second compensation parameter of the microphone according to the parameter differences, and further includes:
Acquiring a first adjustment proportion of the first interaction option and/or acquiring a second adjustment proportion of the second interaction option;
the first compensation parameter is determined according to the current parameter of the loudspeaker, the first adjustment proportion and/or the second adjustment proportion, and the second compensation parameter is determined according to the current parameter of the microphone, the first adjustment proportion and/or the second adjustment proportion.
The invention also proposes a sound parameter regulating device comprising a memory, a processor and a computer program stored on the memory and executable on the processor, which computer program when executed by the processor implements the steps of the sound parameter regulating method as defined in any one of the above.
The present invention also proposes a computer-readable storage medium having stored thereon a sound parameter controlling program which, when executed by a processor, implements the steps of the sound parameter controlling method as set forth in any one of the above.
According to the sound parameter regulation and control method, the sound parameter regulation and control equipment and the computer-readable storage medium, a preset dynamic frequency sweeping audio is sent out through a loudspeaker of the terminal equipment; then, synchronously recording the dynamic frequency-sweeping audio through a microphone of the terminal equipment to obtain recorded audio corresponding to the dynamic frequency-sweeping audio; then, analyzing the dynamic frequency sweeping audio and the recorded audio to obtain frequency response parameters and loudness parameters; and finally, respectively comparing the frequency response parameter and the loudness parameter with preset reference parameters to obtain a parameter difference value, and obtaining a first compensation parameter of the loudspeaker and a second compensation parameter of the microphone according to the parameter difference value. The humanized sound parameter regulation and control scheme is realized, the phenomenon that the subjective feeling sound of a user is reduced or distorted due to blockage of a loudspeaker or a microphone is avoided, the audio effect of the loudspeaker or the microphone is improved, and the user experience is enhanced.
Drawings
The invention will be further described with reference to the accompanying drawings and examples, in which:
fig. 1 is a schematic diagram of a hardware structure of a mobile terminal according to the present invention;
fig. 2 is a schematic diagram of a communication network system according to an embodiment of the present invention;
FIG. 3 is a flowchart of a first embodiment of the sound parameter regulating method of the present invention;
FIG. 4 is a flowchart of a second embodiment of the sound parameter regulating method of the present invention;
FIG. 5 is a flowchart of a third embodiment of the sound parameter controlling method of the present invention;
FIG. 6 is a flowchart of a fourth embodiment of the sound parameter controlling method of the present invention;
FIG. 7 is a flowchart of a fifth embodiment of a sound parameter controlling method of the present invention;
FIG. 8 is a flowchart of a sixth embodiment of a sound parameter controlling method of the present invention;
FIG. 9 is a flowchart of a seventh embodiment of a sound parameter controlling method of the present invention;
fig. 10 is a flowchart of an eighth embodiment of the sound parameter adjusting method of the present invention.
Detailed Description
It should be understood that the specific embodiments described herein are for purposes of illustration only and are not intended to limit the scope of the invention.
In the following description, suffixes such as "module", "component", or "unit" for representing elements are used only for facilitating the description of the present invention, and have no specific meaning per se. Thus, "module," "component," or "unit" may be used in combination.
The terminal may be implemented in various forms. For example, the terminals described in the present invention may include mobile terminals such as cell phones, tablet computers, notebook computers, palm computers, personal digital assistants (Personal Digital Assistant, PDA), portable media players (Portable Media Player, PMP), navigation devices, wearable devices, smart bracelets, pedometers, and fixed terminals such as digital TVs, desktop computers, and the like.
The following description will be given taking a mobile terminal as an example, and those skilled in the art will understand that the configuration according to the embodiment of the present invention can be applied to a fixed type terminal in addition to elements particularly used for a moving purpose.
Referring to fig. 1, which is a schematic diagram of a hardware structure of a mobile terminal implementing various embodiments of the present invention, the mobile terminal 100 may include: an RF (Radio Frequency) unit 101, a WiFi module 102, an audio output unit 103, an a/V (audio/video) input unit 104, a sensor 105, a display unit 106, a user input unit 107, an interface unit 108, a memory 109, a processor 110, and a power supply 111. Those skilled in the art will appreciate that the mobile terminal structure shown in fig. 1 is not limiting of the mobile terminal and that the mobile terminal may include more or fewer components than shown, or may combine certain components, or a different arrangement of components.
The following describes the components of the mobile terminal in detail with reference to fig. 1:
the radio frequency unit 101 may be used for receiving and transmitting signals during the information receiving or communication process, specifically, after receiving downlink information of the base station, processing the downlink information by the processor 110; and, the uplink data is transmitted to the base station. Typically, the radio frequency unit 101 includes, but is not limited to, an antenna, at least one amplifier, a transceiver, a coupler, a low noise amplifier, a duplexer, and the like. In addition, the radio frequency unit 101 may also communicate with networks and other devices via wireless communications. The wireless communication may use any communication standard or protocol, including but not limited to GSM (Global System of Mobile communication, global System for Mobile communications), GPRS (General Packet Radio Service ), CDMA2000 (Code Division Multiple Access, CDMA 2000), WCDMA (Wideband Code Division Multiple Access ), TD-SCDMA (Time Division-Synchronous Code Division Multiple Access, time Division synchronous code Division multiple Access), FDD-LTE (Frequency Division Duplexing-Long Term Evolution, frequency Division Duplex Long term evolution), and TDD-LTE (Time Division Duplexing-Long Term Evolution, time Division Duplex Long term evolution), etc.
WiFi belongs to a short-distance wireless transmission technology, and a mobile terminal can help a user to send and receive e-mails, browse web pages, access streaming media and the like through the WiFi module 102, so that wireless broadband Internet access is provided for the user. Although fig. 1 shows a WiFi module 102, it is understood that it does not belong to the necessary constitution of a mobile terminal, and can be omitted entirely as required within a range that does not change the essence of the invention.
The audio output unit 103 may convert audio data received by the radio frequency unit 101 or the WiFi module 102 or stored in the memory 109 into an audio signal and output as sound when the mobile terminal 100 is in a call signal reception mode, a talk mode, a recording mode, a voice recognition mode, a broadcast reception mode, or the like. Also, the audio output unit 103 may also provide audio output (e.g., a call signal reception sound, a message reception sound, etc.) related to a specific function performed by the mobile terminal 100. The audio output unit 103 may include a speaker, a buzzer, and the like.
The a/V input unit 104 is used to receive an audio or video signal. The a/V input unit 104 may include a graphics processor (Graphics Processing Unit, GPU) 1041 and a microphone 1042, the graphics processor 1041 processing image data of still pictures or video obtained by an image capturing device (e.g., a camera) in a video capturing mode or an image capturing mode. The processed image frames may be displayed on the display unit 106. The image frames processed by the graphics processor 1041 may be stored in the memory 109 (or other storage medium) or transmitted via the radio frequency unit 101 or the WiFi module 102. The microphone 1042 can receive sound (audio data) via the microphone 1042 in a phone call mode, a recording mode, a voice recognition mode, and the like, and can process such sound into audio data. The processed audio (voice) data may be converted into a format output that can be transmitted to the mobile communication base station via the radio frequency unit 101 in the case of a telephone call mode. The microphone 1042 may implement various types of noise cancellation (or suppression) algorithms to cancel (or suppress) noise or interference generated in the course of receiving and transmitting the audio signal.
The mobile terminal 100 also includes at least one sensor 105, such as a light sensor, a motion sensor, and other sensors. Specifically, the light sensor includes an ambient light sensor and a proximity sensor, wherein the ambient light sensor can adjust the brightness of the display panel 1061 according to the brightness of ambient light, and the proximity sensor can turn off the display panel 1061 and/or the backlight when the mobile terminal 100 moves to the ear. As one of the motion sensors, the accelerometer sensor can detect the acceleration in all directions (generally three axes), and can detect the gravity and direction when stationary, and can be used for applications of recognizing the gesture of a mobile phone (such as horizontal and vertical screen switching, related games, magnetometer gesture calibration), vibration recognition related functions (such as pedometer and knocking), and the like; as for other sensors such as fingerprint sensors, pressure sensors, iris sensors, molecular sensors, gyroscopes, barometers, hygrometers, thermometers, infrared sensors, etc. that may also be configured in the mobile phone, the detailed description thereof will be omitted.
The display unit 106 is used to display information input by a user or information provided to the user. The display unit 106 may include a display panel 1061, and the display panel 1061 may be configured in the form of a liquid crystal display (Liquid Crystal Display, LCD), an Organic Light-Emitting Diode (OLED), or the like.
The user input unit 107 may be used to receive input numeric or character information and to generate key signal inputs related to user settings and function control of the mobile terminal. In particular, the user input unit 107 may include a touch panel 1071 and other input devices 1072. The touch panel 1071, also referred to as a touch screen, may collect touch operations thereon or thereabout by a user (e.g., operations of the user on the touch panel 1071 or thereabout by using any suitable object or accessory such as a finger, a stylus, etc.) and drive the corresponding connection device according to a predetermined program. The touch panel 1071 may include two parts of a touch detection device and a touch controller. The touch detection device detects the touch azimuth of a user, detects a signal brought by touch operation and transmits the signal to the touch controller; the touch controller receives touch information from the touch detection device, converts it into touch point coordinates, and sends the touch point coordinates to the processor 110, and can receive and execute commands sent from the processor 110. Further, the touch panel 1071 may be implemented in various types such as resistive, capacitive, infrared, and surface acoustic wave. The user input unit 107 may include other input devices 1072 in addition to the touch panel 1071. In particular, other input devices 1072 may include, but are not limited to, one or more of a physical keyboard, function keys (e.g., volume control keys, switch keys, etc.), a trackball, mouse, joystick, etc., as specifically not limited herein.
Further, the touch panel 1071 may overlay the display panel 1061, and when the touch panel 1071 detects a touch operation thereon or thereabout, the touch panel 1071 is transferred to the processor 110 to determine the type of touch event, and then the processor 110 provides a corresponding visual output on the display panel 1061 according to the type of touch event. Although in fig. 1, the touch panel 1071 and the display panel 1061 are two independent components for implementing the input and output functions of the mobile terminal, in some embodiments, the touch panel 1071 may be integrated with the display panel 1061 to implement the input and output functions of the mobile terminal, which is not limited herein.
The interface unit 108 serves as an interface through which at least one external device can be connected with the mobile terminal 100. For example, the external devices may include a wired or wireless headset port, an external power (or battery charger) port, a wired or wireless data port, a memory card port, a port for connecting a device having an identification module, an audio input/output (I/O) port, a video I/O port, an earphone port, and the like. The interface unit 108 may be used to receive input (e.g., data information, power, etc.) from an external device and transmit the received input to one or more elements within the mobile terminal 100 or may be used to transmit data between the mobile terminal 100 and an external device.
Memory 109 may be used to store software programs as well as various data. The memory 109 may mainly include a storage program area that may store an operating system, application programs required for at least one function (such as a sound playing function, an image playing function, etc.), and a storage data area; the storage data area may store data (such as audio data, phonebook, etc.) created according to the use of the handset, etc. In addition, memory 109 may include high-speed random access memory, and may also include non-volatile memory, such as at least one magnetic disk storage device, flash memory device, or other volatile solid-state storage device.
The processor 110 is a control center of the mobile terminal, connects various parts of the entire mobile terminal using various interfaces and lines, and performs various functions of the mobile terminal and processes data by running or executing software programs and/or modules stored in the memory 109 and calling data stored in the memory 109, thereby performing overall monitoring of the mobile terminal. Processor 110 may include one or more processing units; preferably, the processor 110 may integrate an application processor that primarily handles operating systems, user interfaces, applications, etc., with a modem processor that primarily handles wireless communications. It will be appreciated that the modem processor described above may not be integrated into the processor 110.
The mobile terminal 100 may further include a power source 111 (e.g., a battery) for supplying power to the respective components, and preferably, the power source 111 may be logically connected to the processor 110 through a power management system, so as to perform functions of managing charging, discharging, and power consumption management through the power management system.
Although not shown in fig. 1, the mobile terminal 100 may further include a bluetooth module or the like, which is not described herein.
In order to facilitate understanding of the embodiments of the present invention, a communication network system on which the mobile terminal of the present invention is based will be described below.
Referring to fig. 2, fig. 2 is a schematic diagram of a communication network system according to an embodiment of the present invention, where the communication network system is an LTE system of a general mobile communication technology, and the LTE system includes a UE (User Equipment) 201, an e-UTRAN (Evolved UMTS Terrestrial Radio Access Network ) 202, an epc (Evolved Packet Core, evolved packet core) 203, and an IP service 204 of an operator that are sequentially connected in communication.
Specifically, the UE201 may be the terminal 100 described above, and will not be described herein.
The E-UTRAN202 includes eNodeB2021 and other eNodeB2022, etc. The eNodeB2021 may be connected with other eNodeB2022 by a backhaul (e.g., an X2 interface), the eNodeB2021 is connected to the EPC203, and the eNodeB2021 may provide access from the UE201 to the EPC 203.
EPC203 may include MME (Mobility Management Entity ) 2031, hss (Home Subscriber Server, home subscriber server) 2032, other MMEs 2033, SGW (Serving Gate Way) 2034, pgw (PDN Gate Way) 2035 and PCRF (Policy and Charging Rules Function, policy and tariff function entity) 2036, and so on. The MME2031 is a control node that handles signaling between the UE201 and EPC203, providing bearer and connection management. HSS2032 is used to provide registers to manage functions such as home location registers (not shown) and to hold user specific information about service characteristics, data rates, etc. All user data may be sent through SGW2034 and PGW2035 may provide IP address allocation and other functions for UE201, PCRF2036 is a policy and charging control policy decision point for traffic data flows and IP bearer resources, which selects and provides available policy and charging control decisions for a policy and charging enforcement function (not shown).
IP services 204 may include the internet, intranets, IMS (IP Multimedia Subsystem ), or other IP services, etc.
Although the LTE system is described above as an example, it should be understood by those skilled in the art that the present invention is not limited to LTE systems, but may be applied to other wireless communication systems, such as GSM, CDMA2000, WCDMA, TD-SCDMA, and future new network systems.
Based on the above mobile terminal hardware structure and the communication network system, various embodiments of the method of the present invention are provided.
Example 1
Fig. 3 is a flowchart of a first embodiment of the sound parameter adjusting method of the present invention. A method of sound parameter regulation, the method comprising:
s1, sending out preset dynamic frequency sweeping audio through a loudspeaker of terminal equipment;
s2, synchronously recording the dynamic frequency-sweeping audio through a microphone of the terminal equipment to obtain recorded audio corresponding to the dynamic frequency-sweeping audio;
s3, analyzing the dynamic frequency sweeping audio and the recorded audio to obtain frequency response parameters and loudness parameters;
s4, respectively comparing the frequency response parameter and the loudness parameter with preset reference parameters to obtain a parameter difference value, and obtaining a first compensation parameter of the loudspeaker and a second compensation parameter of the microphone according to the parameter difference value.
In this embodiment, first, a preset dynamic frequency-sweeping audio is sent out through a speaker of a terminal device; then, synchronously recording the dynamic frequency-sweeping audio through a microphone of the terminal equipment to obtain recorded audio corresponding to the dynamic frequency-sweeping audio; then, analyzing the dynamic frequency sweeping audio and the recorded audio to obtain frequency response parameters and loudness parameters; and finally, respectively comparing the frequency response parameter and the loudness parameter with preset reference parameters to obtain a parameter difference value, and obtaining a first compensation parameter of the loudspeaker and a second compensation parameter of the microphone according to the parameter difference value.
Specifically, taking a mobile phone as an example, providing an option of audio parameter self-calibration in an interactive interface of the mobile phone, clicking to perform audio parameter self-calibration, sending a dynamic sweep frequency signal by the mobile phone through a microphone, recording the sweep frequency signal by the microphone, and comparing and analyzing the recorded parameter by the microphone with an initial default parameter by frequency response analysis and loudness analysis to obtain a parameter difference value; then, the compensation parameters are determined in a self-adaptive manner through the lever principle, or the user selects the parameter difference value to be compensated, or the parameters are divided in proportion according to the requirement of the user, and the parameters are dynamically added on the working parameters of the microphone and the loudspeaker, namely, the working parameters of the follow-up microphone and the loudspeaker can ensure that the subjective sound requirements of the user on the microphone and the loudspeaker are met.
The method has the beneficial effects that the preset dynamic frequency sweeping audio is sent out through the loudspeaker of the terminal equipment; then, synchronously recording the dynamic frequency-sweeping audio through a microphone of the terminal equipment to obtain recorded audio corresponding to the dynamic frequency-sweeping audio; then, analyzing the dynamic frequency sweeping audio and the recorded audio to obtain frequency response parameters and loudness parameters; and finally, respectively comparing the frequency response parameter and the loudness parameter with preset reference parameters to obtain a parameter difference value, and obtaining a first compensation parameter of the loudspeaker and a second compensation parameter of the microphone according to the parameter difference value. The humanized sound parameter regulation and control scheme is realized, the phenomenon that the subjective feeling sound of a user is reduced or distorted due to blockage of a loudspeaker or a microphone is avoided, the audio effect of the loudspeaker or the microphone is improved, and the user experience is enhanced.
Example two
Fig. 4 is a flowchart of a second embodiment of the sound parameter adjusting method according to the present invention, based on the above embodiment, before the preset dynamic frequency sweep audio is sent out through the speaker of the terminal device, the method includes:
s01, presetting a dust removal period of the terminal equipment according to the service time of the loudspeaker and/or the microphone;
S02, carrying out ultrasonic dust removal on the loudspeaker and/or the microphone according to the dust removal period.
In this embodiment, first, a dust removal period of the terminal device is preset according to a use time of the speaker and/or the microphone; and then, carrying out ultrasonic dust removal on the loudspeaker and/or the microphone according to the dust removal period.
Optionally, in this embodiment, a first dust removal period of the speaker of the terminal device is preset according to a first use time of the speaker;
optionally, in this embodiment, a second dust removal period of the microphone of the terminal device is preset according to a second use time of the microphone;
optionally, in this embodiment, a third dust removal period of the speaker and the microphone of the terminal device is preset according to a third use time of the terminal device;
optionally, in this embodiment, a fourth dust removal period of one speaker of the terminal device is preset according to a fourth use time of the one speaker of the plurality of speakers;
optionally, in this embodiment, a fifth dust removal period of one microphone of the plurality of microphones is preset according to a fifth use time of the one microphone.
The method has the beneficial effects that the dust removal period of the terminal equipment is preset through the use time of the loudspeaker and/or the microphone; and then, carrying out ultrasonic dust removal on the loudspeaker and/or the microphone according to the dust removal period. The periodic ultrasonic dust removal scheme is provided for realizing a humanized sound parameter regulation scheme, so that the phenomenon that the subjective feeling of a user is reduced or distorted due to blockage of a loudspeaker or a microphone is avoided, the audio effect of the loudspeaker or the microphone is improved, and the user experience is enhanced.
Example III
Fig. 5 is a flowchart of a third embodiment of the sound parameter adjusting method according to the present invention, based on the above embodiment, the sending of the preset dynamic frequency sweep audio through the speaker of the terminal device includes:
s11, presetting audio parameters and audio playing time of the dynamic sweep frequency audio according to the using state and the using environment of the loudspeaker and/or the microphone;
s12, playing the dynamic frequency-sweeping audio composed of the audio parameters according to the audio playing time through the loudspeaker.
In this embodiment, first, according to the use state and the use environment of the speaker and/or the microphone, the audio parameters and the audio playing time of the dynamic frequency-sweeping audio are preset; and playing the dynamic frequency-sweeping audio composed of the audio parameters according to the audio playing time through the loudspeaker.
Optionally, in this embodiment, the audio parameters and the audio playing time of the dynamic frequency-sweeping audio are preset according to the use states and the use environments of the speaker and/or the microphone, where the use states include the number and the frequency of the speaker and/or the microphone used in the plurality of speakers and/or the microphone, the use environments include the environmental dust density of the environment where the terminal device is located, and the density value can be obtained by acquiring the information under the current location through the network;
optionally, in this embodiment, it is determined that one speaker among the plurality of speakers plays the dynamic swept-frequency audio composed of the audio parameters according to the audio playing time;
optionally, in this embodiment, if it is determined that one microphone is to be corrected, one speaker corresponding to the microphone position is determined among the plurality of speakers, and dynamic frequency-swept audio composed of the audio parameters is played according to the audio playing time.
The method has the beneficial effects that the audio parameters and the audio playing time of the dynamic sweep audio are preset through the use states and the use environments of the loudspeaker and/or the microphone; and playing the dynamic frequency-sweeping audio composed of the audio parameters according to the audio playing time through the loudspeaker. The dynamic frequency sweeping audio playing scheme is provided for realizing a humanized audio parameter regulating scheme, so that the situation that the subjective feeling of a user is reduced or distorted due to blockage of a loudspeaker or a microphone is avoided, the audio effect of the loudspeaker or the microphone is improved, and the user experience is enhanced.
Example IV
Fig. 6 is a flowchart of a fourth embodiment of the sound parameter adjusting method according to the present invention, based on the above embodiment, the recording the dynamic frequency-sweeping audio synchronously by the microphone of the terminal device, to obtain a recorded audio corresponding to the dynamic frequency-sweeping audio, includes:
s21, selecting one loudspeaker from a plurality of loudspeakers to play the dynamic sweep frequency audio, and determining one microphone corresponding to the one loudspeaker;
s22, synchronously recording the dynamic frequency sweeping audio through one microphone.
In this embodiment, first, one of the plurality of speakers is selected to play the dynamic swept frequency audio, and at the same time, one of the microphones corresponding to the one speaker is determined; the dynamic swept audio is then synchronously recorded by one of the microphones.
Optionally, in this embodiment, if it is determined that one speaker is to be corrected, one speaker to be corrected is selected from a plurality of speakers to play the dynamic frequency-sweeping audio, and at the same time, one microphone corresponding to the position of one speaker is determined, so that more accurate correction can be performed subsequently;
Optionally, in this embodiment, if it is determined that a plurality of speakers are to be corrected, one speaker to be corrected is selected one by one from the plurality of speakers to play the dynamic frequency-sweeping audio, and at the same time, one microphone corresponding to the one speaker is determined one by one, and playing and correcting are performed one by one, so as to improve accuracy of correction.
The method has the advantages that the dynamic frequency sweeping audio is played by selecting one loudspeaker from a plurality of loudspeakers, and meanwhile, one microphone corresponding to one loudspeaker is determined; the dynamic swept audio is then synchronously recorded by one of the microphones. The method provides a selected scheme of the loudspeaker and the microphone for realizing a humanized sound parameter regulation scheme, avoids the situation that the subjective feeling of the user is reduced or distorted due to the blockage of the loudspeaker or the microphone, improves the audio effect of the loudspeaker or the microphone, and enhances the user experience.
Example five
Fig. 7 is a flowchart of a fifth embodiment of the sound parameter adjusting method according to the present invention, based on the above embodiment, the analyzing the dynamic frequency-sweeping audio and the recorded audio to obtain frequency response parameters and loudness parameters includes:
S31, acquiring a first preset parameter of the loudspeaker, and simultaneously, acquiring a second preset parameter of the microphone;
s32, analyzing the dynamic frequency sweeping audio and the recorded audio by combining the first preset parameter and the second preset parameter to obtain the frequency response parameter and the loudness parameter.
In this embodiment, first, a first preset parameter of the speaker is obtained, and at the same time, a second preset parameter of the microphone is obtained; and then, analyzing the dynamic frequency sweeping audio and the recorded audio by combining the first preset parameter and the second preset parameter to obtain the frequency response parameter and the loudness parameter.
Optionally, in this embodiment, the dynamic frequency-sweeping audio and the recorded audio are analyzed by combining the first preset parameter and the second preset parameter to obtain the frequency response parameter of a speaker or the frequency response parameter of a microphone;
optionally, in this embodiment, the dynamic frequency-sweeping audio and the recorded audio are analyzed in combination with the first preset parameter and the second preset parameter to obtain the loudness parameter of a speaker or the loudness parameter of a microphone.
The method has the advantages that the first preset parameters of the loudspeaker are obtained, and meanwhile, the second preset parameters of the microphone are obtained; and then, analyzing the dynamic frequency sweeping audio and the recorded audio by combining the first preset parameter and the second preset parameter to obtain the frequency response parameter and the loudness parameter. The method provides a determination scheme of frequency response parameters and loudness parameters for realizing a humanized sound parameter regulation scheme, avoids the situation that the subjective feeling sound of a user is reduced or distorted due to blockage of a loudspeaker or a microphone, improves the audio effect of the loudspeaker or the microphone, and enhances the user experience.
Example six
Fig. 8 is a flowchart of a sixth embodiment of a sound parameter adjusting method according to the present invention, based on the above embodiment, the comparing the frequency response parameter and the loudness parameter with preset reference parameters to obtain parameter differences, and obtaining a first compensation parameter of the speaker and a second compensation parameter of the microphone according to the parameter differences, where the comparing includes:
s41, analyzing the frequency response parameters by combining the first preset parameters, the second preset parameters and the reference parameters to obtain a first difference value of the frequency response parameters;
s42, analyzing the loudness parameter by combining the first preset parameter, the second preset parameter and the reference parameter to obtain a second difference value of the loudness parameter.
In this embodiment, first, the first preset parameter, the second preset parameter, and the reference parameter are combined, and the frequency response parameter is analyzed to obtain a first difference value of the frequency response parameter; and then, analyzing the loudness parameter by combining the first preset parameter, the second preset parameter and the reference parameter to obtain a second difference value of the loudness parameter.
Optionally, in this embodiment, the first preset parameter, the second preset parameter, and the reference parameter are combined, and the frequency response parameter is analyzed to obtain a first difference value of the frequency response parameter corresponding to one speaker and one microphone adapted to the speaker;
Optionally, in this embodiment, the first preset parameter, the second preset parameter, and the reference parameter are combined, and the frequency response parameter is analyzed to obtain a first difference value of the loudness parameter corresponding to one speaker and one microphone adapted by the speaker.
The method has the beneficial effects that the first difference value of the frequency response parameters is obtained by analyzing the frequency response parameters by combining the first preset parameters, the second preset parameters and the reference parameters; and then, analyzing the loudness parameter by combining the first preset parameter, the second preset parameter and the reference parameter to obtain a second difference value of the loudness parameter. A pair of difference value determining schemes of a loudspeaker and a microphone are provided for realizing a humanized sound parameter regulating and controlling scheme, so that the situation that the subjective feeling sound of a user is reduced or distorted due to blockage of the loudspeaker or the microphone is avoided, the audio effect of the loudspeaker or the microphone is improved, and the user experience is enhanced.
Example seven
Fig. 9 is a flowchart of a seventh embodiment of a sound parameter adjusting method according to the present invention, based on the above embodiment, the comparing and analyzing the frequency response parameter and the loudness parameter with preset reference parameters to obtain parameter differences, and obtaining a first compensation parameter of the speaker and a second compensation parameter of the microphone according to the parameter differences, further includes:
S43, presetting an interactive interface for regulating and controlling sound parameter compensation;
s44, presetting a first interaction option corresponding to the first difference value in the interaction interface, and presetting a second interaction option corresponding to the second difference value in the interaction interface.
In this embodiment, firstly, an interactive interface for adjusting and controlling sound parameter compensation is preset; and then, presetting a first interaction option corresponding to the first difference value in the interaction interface, and presetting a second interaction option corresponding to the second difference value in the interaction interface.
Optionally, in this embodiment, a first interaction option corresponding to the first difference is preset in the interaction interface, typical values of multiple levels of the difference are provided, and an adjustment amplitude of each typical value and an option of starting pre-listening are marked for a user to select and pre-listen for reference;
optionally, in this embodiment, a second interaction option corresponding to the second difference is preset in the interaction interface, typical values of multiple levels of the difference are provided, and an adjustment range of each typical value and an option of starting pre-listening are marked for a user to select and pre-listen for reference.
The method has the beneficial effects that the interactive interface for regulating and controlling the sound parameter compensation is preset; and then, presetting a first interaction option corresponding to the first difference value in the interaction interface, and presetting a second interaction option corresponding to the second difference value in the interaction interface. The interactive option regulation and control scheme is provided for realizing a humanized sound parameter regulation and control scheme, so that the phenomenon that the subjective feeling sound of a user is reduced or distorted due to blockage of a loudspeaker or a microphone is avoided, the audio effect of the loudspeaker or the microphone is improved, and the user experience is enhanced.
Example eight
Fig. 10 is a flowchart of an eighth embodiment of a sound parameter adjusting method according to the present invention, based on the above embodiment, the comparing and analyzing the frequency response parameter and the loudness parameter with preset reference parameters to obtain parameter differences, and obtaining a first compensation parameter of the speaker and a second compensation parameter of the microphone according to the parameter differences, further including:
s45, acquiring a first adjustment proportion of the first interaction option and/or acquiring a second adjustment proportion of the second interaction option;
s46, determining the first compensation parameter according to the current parameter of the loudspeaker, the first adjustment proportion and/or the second adjustment proportion, and determining the second compensation parameter according to the current parameter of the microphone, the first adjustment proportion and/or the second adjustment proportion.
In this embodiment, first, a first adjustment proportion of the first interaction option is obtained, and/or a second adjustment proportion of the second interaction option is obtained; then, the first compensation parameter is determined according to the current parameter of the loudspeaker, the first adjustment proportion and/or the second adjustment proportion, and the second compensation parameter is determined according to the current parameter of the microphone, the first adjustment proportion and/or the second adjustment proportion.
Optionally, in this embodiment, the first compensation parameter is determined according to the current parameter of the speaker, the first adjustment proportion and/or the second adjustment proportion, and the second compensation parameter is adaptively adjusted after the first compensation parameter is determined;
optionally, in this embodiment, after determining the second compensation parameter according to the current parameter of the microphone, the first adjustment proportion and/or the second adjustment proportion, the first compensation parameter is adaptively adjusted, so as to avoid excessive adjustment caused by repeated adjustment of the user.
The method has the advantages that the first adjustment proportion of the first interaction option is obtained, and/or the second adjustment proportion of the second interaction option is obtained; then, the first compensation parameter is determined according to the current parameter of the loudspeaker, the first adjustment proportion and/or the second adjustment proportion, and the second compensation parameter is determined according to the current parameter of the microphone, the first adjustment proportion and/or the second adjustment proportion. The method provides a compensation parameter regulation scheme for realizing a humanized sound parameter regulation scheme, avoids the situation that the subjective feeling of a user of a loudspeaker or a microphone is reduced or distorted due to blockage, improves the audio effect of the loudspeaker or the microphone, and enhances the user experience.
Example nine
Based on the above embodiments, the present invention also proposes a sound parameter controlling apparatus, the apparatus comprising a memory, a processor and a computer program stored on the memory and executable on the processor, the computer program implementing the steps of the sound parameter controlling method as defined in any one of the above when being executed by the processor.
It should be noted that the above device embodiments and method embodiments belong to the same concept, the specific implementation process of the device embodiments is detailed in the method embodiments, and technical features in the method embodiments are correspondingly applicable to the device embodiments, which are not repeated herein.
Examples ten
Based on the above embodiments, the present invention also proposes a computer-readable storage medium, on which a sound parameter controlling program is stored, which when executed by a processor implements the steps of the sound parameter controlling method as set forth in any one of the above.
It should be noted that the medium embodiment and the method embodiment belong to the same concept, the specific implementation process of the medium embodiment and the method embodiment are detailed, and technical features in the method embodiment are correspondingly applicable in the medium embodiment, which is not repeated herein.
It should be noted that, in this document, the terms "comprises," "comprising," or any other variation thereof, are intended to cover a non-exclusive inclusion, such that a process, method, article, or apparatus that comprises a list of elements does not include only those elements but may include other elements not expressly listed or inherent to such process, method, article, or apparatus. Without further limitation, an element defined by the phrase "comprising one … …" does not exclude the presence of other like elements in a process, method, article, or apparatus that comprises the element.
The foregoing embodiment numbers of the present invention are merely for the purpose of description, and do not represent the advantages or disadvantages of the embodiments.
From the above description of the embodiments, it will be clear to those skilled in the art that the above-described embodiment method may be implemented by means of software plus a necessary general hardware platform, but of course may also be implemented by means of hardware, but in many cases the former is a preferred embodiment. Based on such understanding, the technical solution of the present invention may be embodied essentially or in a part contributing to the prior art in the form of a software product stored in a storage medium (e.g. ROM/RAM, magnetic disk, optical disk) comprising instructions for causing a terminal (which may be a mobile phone, a computer, a server, an air conditioner, or a network device, etc.) to perform the method according to the embodiments of the present invention.
The embodiments of the present invention have been described above with reference to the accompanying drawings, but the present invention is not limited to the above-described embodiments, which are merely illustrative and not restrictive, and many forms may be made by those having ordinary skill in the art without departing from the spirit of the present invention and the scope of the claims, which are to be protected by the present invention.
Claims (7)
1. A method for regulating and controlling sound parameters, the method comprising:
sending out preset dynamic frequency sweeping audio through a loudspeaker of the terminal equipment;
synchronously recording the dynamic frequency-sweeping audio through a microphone of the terminal equipment to obtain recorded audio corresponding to the dynamic frequency-sweeping audio;
analyzing the dynamic frequency sweeping audio and the recorded audio to obtain frequency response parameters and loudness parameters;
respectively comparing the frequency response parameter and the loudness parameter with preset reference parameters to obtain a parameter difference value, and obtaining a first compensation parameter of the loudspeaker and a second compensation parameter of the microphone according to the parameter difference value;
before the preset dynamic frequency sweeping audio is sent out by the loudspeaker of the terminal equipment, the method comprises the following steps:
Presetting a dust removal period of the terminal equipment according to the service time of the loudspeaker and/or the microphone;
ultrasonic dust removal is carried out on the loudspeaker and/or the microphone according to the dust removal period;
wherein,,
presetting a first dust removal period of the loudspeaker according to a first use time of the loudspeaker, and presetting a second dust removal period of the microphone according to a second use time of the microphone;
or presetting a third dust removal period of the loudspeaker and the microphone according to a third using time of the terminal equipment;
or presetting a fourth dust removal period of one loudspeaker according to fourth service time of the one loudspeaker, and presetting a fifth dust removal period of the one microphone according to fifth service time of the one microphone;
the method for sending out the preset dynamic frequency sweeping audio through the loudspeaker of the terminal equipment comprises the following steps:
presetting audio parameters and audio playing time of the dynamic frequency sweeping audio according to the using state and the using environment of the loudspeaker and/or the microphone;
playing dynamic sweep frequency audio composed of the audio parameters according to the audio playing time through the loudspeaker;
Wherein,,
the usage status includes the number and frequency of the speakers and/or microphones used among the plurality of speakers and/or the plurality of microphones;
the using environment comprises the environment dust density of the environment where the terminal equipment is located, and the information under the current positioning is obtained through a network to obtain the density value of the environment dust density;
determining one loudspeaker from a plurality of loudspeakers to play the dynamic frequency-sweeping audio composed of the audio parameters according to the audio play time, and if one microphone is determined to be corrected, determining one loudspeaker from the plurality of loudspeakers to play the dynamic frequency-sweeping audio composed of the audio parameters according to the audio play time, wherein the loudspeaker corresponds to the position of the one microphone;
the step of synchronously recording the dynamic frequency-sweeping audio through the microphone of the terminal equipment to obtain recorded audio corresponding to the dynamic frequency-sweeping audio comprises the following steps:
selecting one loudspeaker from a plurality of loudspeakers to play the dynamic frequency sweeping audio, and simultaneously determining one microphone corresponding to the one loudspeaker;
synchronously recording the dynamic frequency sweeping audio through one microphone;
Wherein,,
if one loudspeaker is determined to be corrected, selecting the loudspeaker to be corrected from a plurality of loudspeakers to play the dynamic frequency sweeping audio, and determining one microphone at a position corresponding to the loudspeaker to play and correct;
and if the plurality of loudspeakers are determined to be corrected, one loudspeaker to be corrected is selected one by one from the plurality of loudspeakers to play the dynamic frequency sweeping audio, and meanwhile, one microphone corresponding to the position of the one loudspeaker is determined one by one to play and correct one by one.
2. The sound parameter tuning method of claim 1, wherein the analyzing the dynamic frequency sweep audio and the recorded audio to obtain the frequency response parameter and the loudness parameter comprises:
acquiring a first preset parameter of the loudspeaker, and simultaneously acquiring a second preset parameter of the microphone;
and analyzing the dynamic frequency sweeping audio and the recorded audio by combining the first preset parameter and the second preset parameter to obtain the frequency response parameter and the loudness parameter.
3. The sound parameter adjusting and controlling method according to claim 2, wherein the comparing the frequency response parameter and the loudness parameter with preset reference parameters respectively to obtain parameter differences, and obtaining the first compensation parameter of the speaker and the second compensation parameter of the microphone according to the parameter differences includes:
Analyzing the frequency response parameters by combining the first preset parameters, the second preset parameters and the reference parameters to obtain a first difference value of the frequency response parameters;
and analyzing the loudness parameter by combining the first preset parameter, the second preset parameter and the reference parameter to obtain a second difference value of the loudness parameter.
4. A sound parameter adjusting and controlling method as defined in claim 3, wherein the comparing the frequency response parameter and the loudness parameter with preset reference parameters respectively to obtain parameter differences, and obtaining the first compensation parameter of the speaker and the second compensation parameter of the microphone according to the parameter differences, further comprises:
presetting an interactive interface for regulating and controlling sound parameter compensation;
and presetting a first interaction option corresponding to the first difference value in the interaction interface, and presetting a second interaction option corresponding to the second difference value in the interaction interface.
5. The method of claim 4, wherein the comparing the frequency response parameter and the loudness parameter with preset reference parameters respectively to obtain parameter differences, and obtaining the first compensation parameter of the speaker and the second compensation parameter of the microphone according to the parameter differences, further comprises:
Acquiring a first adjustment proportion of the first interaction option and/or acquiring a second adjustment proportion of the second interaction option;
the first compensation parameter is determined according to the current parameter of the loudspeaker, the first adjustment proportion and/or the second adjustment proportion, and the second compensation parameter is determined according to the current parameter of the microphone, the first adjustment proportion and/or the second adjustment proportion.
6. A sound parameter regulating device, characterized in that the device comprises a memory, a processor and a computer program stored on the memory and executable on the processor, which computer program, when being executed by the processor, implements the steps of the sound parameter regulating method according to any one of claims 1 to 5.
7. A computer-readable storage medium, wherein a sound parameter controlling program is stored on the computer-readable storage medium, which when executed by a processor, implements the steps of the sound parameter controlling method according to any one of claims 1 to 5.
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