CN107105095B - Sound processing method and mobile terminal - Google Patents
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- CN107105095B CN107105095B CN201710274001.9A CN201710274001A CN107105095B CN 107105095 B CN107105095 B CN 107105095B CN 201710274001 A CN201710274001 A CN 201710274001A CN 107105095 B CN107105095 B CN 107105095B
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M1/00—Substation equipment, e.g. for use by subscribers
- H04M1/72—Mobile telephones; Cordless telephones, i.e. devices for establishing wireless links to base stations without route selection
- H04M1/724—User interfaces specially adapted for cordless or mobile telephones
- H04M1/72448—User interfaces specially adapted for cordless or mobile telephones with means for adapting the functionality of the device according to specific conditions
- H04M1/72454—User interfaces specially adapted for cordless or mobile telephones with means for adapting the functionality of the device according to specific conditions according to context-related or environment-related conditions
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
- G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
- G10L21/0272—Voice signal separating
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L25/00—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00
- G10L25/48—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00 specially adapted for particular use
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Abstract
The invention discloses a sound processing method, which comprises the steps of collecting environmental sounds around an area where a user is located through a sound sensor of an earphone, and separating the collected environmental sounds; identifying and matching the separated various sounds with sample sounds in a database; comprehensively calculating decibels of the successfully matched environmental sounds to obtain an environmental sound comprehensive sound value H1; processing according to the noise reduction value of the earphone and the environment sound comprehensive sound value H1 to obtain the mixed sound value H2; and adjusting the sound channel output of the earphone according to the calculated mixed sound value H2. The invention also discloses a sound processing mobile terminal, which solves the problem that the intelligent volume output can not be carried out according to the environment where the user is located in the related technology, and optimizes the sound channel output of the earphone according to the calculated mixed sound value by collecting and separating the environment sound, so that the user obtains more comfortable earphone sound reception experience, and the user experience is improved.
Description
Technical Field
The present invention relates to the field of mobile communications technologies, and in particular, to a sound processing method and a mobile terminal.
Background
Along with the development of the internet and the popularization of the terminal, the user group of the terminal is larger and larger, and meanwhile, more intelligent and humanized requirements are provided for software.
In the prior art, a real terminal is used as a game machine or a television by a user, possibly a learning machine, possibly a playground of a baby and the like, so that more fun is brought to the life of people. With the upgrading of communication products, mobile terminals (such as mobile phones, Personal Digital Assistants (PDAs), etc.) have become a necessary communication tool for people. Various functions which are convenient for people to live can be realized on the mobile terminal, such as mobile phone television, GPS, mobile payment and the like, and the functions can be realized only by accessing the mobile terminal to the Internet.
In recent years, with the development of technology, a large amount of image and sound information is received, and a signal including the information is detected by a sensor. In the aspect of sensor detection, people always pursue detection of real source signals, and new principles, new methods and new technologies are continuously provided, for example, sensors are arranged to be close to the source signal position as much as possible, and measures such as a plurality of sensors are adopted to detect real source signals. Since the detected signal is a mixed signal, it is more difficult to detect the true source signal in some situations (e.g., public place signals, etc.). For example, a mobile phone inputs three voice signals through a sound sensor, and then simulates a real scene by mixing the three signals, so as to achieve the situation of the voice signals received by a common microphone.
The data received actually is complex and various, wherein the data comprises useful information and useless information, particularly when the number of sensors is large, the calculation is complex, and the observed data needs to be preprocessed, so that the data dimension can be reduced under certain conditions, and the calculation amount of the subsequent processing is reduced.
For the problem that intelligent volume output cannot be performed according to the environment where the user is located in the related art, no solution is proposed at present.
Disclosure of Invention
The invention mainly aims to provide a sound processing method and a mobile terminal, and aims to solve the problem that intelligent volume output cannot be performed according to the environment where a user is located in the related art.
To achieve the above object, an embodiment of the present invention provides a sound processing method, including:
collecting environmental sounds around an area where a user is located through a sound sensor of an earphone, and separating the collected environmental sounds;
identifying and matching the separated various sounds with sample sounds in a database;
comprehensively calculating decibels of the successfully matched environmental sounds to obtain an environmental sound comprehensive sound value H1;
processing according to the noise reduction value of the earphone and the environment sound comprehensive sound value H1 to obtain the mixed sound value H2;
and adjusting the sound channel output of the earphone according to the calculated mixed sound value H2.
Preferably, before the separating process is performed on the collected environmental sound, the method further includes: and carrying out noise reduction processing on the collected mixed voice signal containing the environmental sound, and carrying out whitening processing on the mixed voice signal after the noise reduction processing.
Preferably, after the mixed sound value H2 is obtained by processing the noise reduction value of the headphone and the ambient sound integrated sound value H1, the method further includes:
uploading the H2 to a multimedia sound library.
Preferably, adjusting the channel output of the headphone according to the calculated mixed sound value includes:
adding the H2 to the current pre-output sound value H3;
comparing and analyzing the added result with a preset comfortable value interval;
and adjusting the volume of the sound channel output of the earphone according to the analysis result.
Preferably, adjusting the volume of the sound channel output of the headphone according to the analysis result comprises:
if H2+ H3 > comfort value interval, automatically reducing the sound channel output of the earphone to the comfort value interval;
if H2+ H3 belongs to the comfortable value interval, the sound channel output of the earphone is output according to the sound value H3;
and if H2+ H3 is less than the comfortable value range, automatically increasing the sound channel output of the earphone to the comfortable value range.
According to another aspect of the embodiments of the present invention, there is also provided a sound processing mobile terminal including:
the separation module is used for collecting the environmental sound around the area where the user is located through a sound sensor of the earphone and separating the collected environmental sound;
the identification matching module is used for identifying and matching the separated various sounds with sample sounds in the database;
the comprehensive calculation module is used for performing comprehensive calculation on the successfully matched environmental sounds in decibels to obtain an environmental sound comprehensive sound value H1;
the processing module is used for processing the noise reduction value of the earphone and the environment sound comprehensive sound value H1 to obtain the mixed sound value H2;
and the adjusting module is used for adjusting the sound channel output of the earphone according to the calculated mixed sound value H2.
Preferably, the mobile terminal further includes:
the noise reduction processing module is used for carrying out noise reduction processing on the collected mixed voice signal containing the environmental sound before carrying out separation processing on the collected environmental sound;
and the whitening processing module is used for whitening the mixed voice signal after the noise reduction processing.
Preferably, the mobile terminal further includes:
and the uploading module is used for uploading the H2 to a multimedia sound library after the mixed sound value H2 is obtained by processing the noise reduction value of the earphone and the environmental sound comprehensive sound value H1.
Preferably, the adjusting module comprises:
an adding unit for adding the H2 to the current pre-output sound value H3;
the comparison and analysis unit is used for comparing and analyzing the added result with a preset comfortable value interval;
and the adjusting unit is used for adjusting the volume output by the sound channel of the earphone according to the analysis result.
Preferably, the adjusting unit is also used for
If H2+ H3 > comfort value interval, automatically reducing the sound channel output of the earphone to the comfort value interval;
if H2+ H3 belongs to the comfortable value interval, the sound channel output of the earphone is output according to the sound value H3;
and if H2+ H3 is less than the comfortable value range, automatically increasing the sound channel output of the earphone to the comfortable value range.
According to the invention, the environmental sounds around the area where the user is located are collected through the sound sensor of the earphone, and the collected environmental sounds are separated; identifying and matching the separated various sounds with sample sounds in a database; comprehensively calculating decibels of the successfully matched environmental sounds to obtain an environmental sound comprehensive sound value H1; processing according to the noise reduction value of the earphone and the environment sound comprehensive sound value H1 to obtain the mixed sound value H2; the sound channel output of the earphone is adjusted according to the calculated mixed sound value H2, the problem that intelligent volume output cannot be performed according to the environment where the user is located in the related art is solved, the sound channel output of the earphone is optimized according to the calculated mixed sound value by collecting and separating the environment sound, so that the user obtains more comfortable earphone sound reception experience, and the user experience is improved.
Drawings
Fig. 1 is a schematic diagram of a hardware structure of an optional mobile terminal for implementing various embodiments of the present invention;
FIG. 2 is a diagram of a wireless communication system for the mobile terminal shown in FIG. 1;
FIG. 3 is a flow chart of a sound processing method according to an embodiment of the present invention;
FIG. 4 is a schematic illustration of separating mixed sounds according to an embodiment of the invention;
FIG. 5 is a flow diagram of a hybrid sound process according to an embodiment of the present invention;
FIG. 6 is a flow diagram of a hybrid sound process in accordance with a preferred embodiment of the present invention;
fig. 7 is a block diagram of a sound processing mobile terminal according to an embodiment of the present invention;
fig. 8 is a block diagram of a sound processing mobile terminal according to a preferred embodiment of the present invention.
The implementation, functional features and advantages of the objects of the present invention will be further explained with reference to the accompanying drawings.
Detailed Description
It should be understood that the specific embodiments described herein are merely illustrative of the invention and are not intended to limit the invention.
A mobile terminal implementing various embodiments of the present invention will now be described with reference to the accompanying drawings. In the following description, suffixes such as "module", "component", or "unit" used to denote elements are used only for facilitating the explanation of the present invention, and have no specific meaning in themselves. Thus, "module" and "component" may be used in a mixture.
The mobile terminal may be implemented in various forms. For example, the terminal described in the present invention may include a mobile terminal such as a mobile phone, a smart phone, a notebook computer, a digital broadcast receiver, a PDA (personal digital assistant), a PAD (tablet computer), a PMP (portable multimedia player), a navigation device, and the like, and a stationary terminal such as a digital TV, a desktop computer, and the like. In the following, it is assumed that the terminal is a mobile terminal. However, it will be understood by those skilled in the art 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 moving purposes.
Fig. 1 is a schematic diagram of a hardware structure of an optional mobile terminal for implementing various embodiments of the present invention.
The mobile terminal 100 may include a wireless communication unit 110, an a/V (audio/video) input unit 120, a user input unit 130, a sensing unit 140, an output unit 150, a memory 160, an interface unit 170, a controller 180, and a power supply unit 190, etc.
Fig. 1 illustrates the mobile terminal 100 having various components, but it is to be understood that not all illustrated components are required to be implemented. More or fewer components may alternatively be implemented. The elements of the mobile terminal 100 will be described in detail below.
The wireless communication unit 110 may generally include one or more components that allow radio communication between the mobile terminal 100 and a wireless communication system or network. For example, the wireless communication unit 110 may include at least one of a broadcast receiving module 111, a mobile communication module 112, a wireless internet module 113, a short-range communication module 114, and a location information module 115.
The broadcast receiving module 111 receives a broadcast signal and/or broadcast associated information from an external broadcast management server via a broadcast channel. The broadcast channel may include a satellite channel and/or a terrestrial channel. The broadcast management server may be a server that generates and transmits a broadcast signal and/or broadcast associated information or a server that receives a previously generated broadcast signal and/or broadcast associated information and transmits it to a terminal. The broadcast signal may include a TV broadcast signal, a radio broadcast signal, a data broadcast signal, and the like. Also, the broadcast signal may further include a broadcast signal combined with a TV or radio broadcast signal. The broadcast associated information may also be provided via a mobile communication network, and in this case, the broadcast associated information may be received by the mobile communication module 112. The broadcast signal may exist in various forms, for example, it may exist in the form of an Electronic Program Guide (EPG) of Digital Multimedia Broadcasting (DMB), an Electronic Service Guide (ESG) of digital video broadcasting-handheld (DVB-H), and the like. The broadcast receiving module 111 may receive a signal broadcast by using various types of broadcasting systems. In particular, the broadcast receiving module 111 may receive digital broadcasting by using a digital broadcasting system such as a data broadcasting system of multimedia broadcasting-terrestrial (DMB-T), digital multimedia broadcasting-satellite (DMB-S), digital video broadcasting-handheld (DVB-H), forward link media (MediaFLO @), terrestrial digital broadcasting integrated service (ISDB-T), and the like. The broadcast receiving module 111 may be constructed to be suitable for various broadcasting systems that provide broadcast signals as well as the above-mentioned digital broadcasting systems. The broadcast signal and/or broadcast associated information received via the broadcast receiving module 111 may be stored in the memory 160 (or other type of storage medium).
The mobile communication module 112 transmits and/or receives radio signals to and/or from at least one of a base station (e.g., access point, node B, etc.), an external terminal, and a server. Such radio signals may include voice call signals, video call signals, or various types of data transmitted and/or received according to text and/or multimedia messages.
The wireless internet module 113 supports wireless internet access of the mobile terminal. The module may be internally or externally coupled to the terminal. The wireless internet access technology to which the module relates may include WLAN (wireless LAN) (Wi-Fi), Wibro (wireless broadband), Wimax (worldwide interoperability for microwave access), HSDPA (high speed downlink packet access), and the like.
The short-range communication module 114 is a module for supporting short-range communication. Some examples of short-range communication technologies include bluetooth (TM), Radio Frequency Identification (RFID), infrared data association (IrDA), Ultra Wideband (UWB), zigbee (TM), and the like.
The location information module 115 is a module for checking or acquiring location information of the mobile terminal. A typical example of the location information module 115 is a GPS (global positioning system). According to the current technology, the GPS calculates distance information and accurate time information from three or more satellites and applies triangulation to the calculated information, thereby accurately calculating three-dimensional current location information according to longitude, latitude, and altitude. Currently, a method for calculating position and time information uses three satellites and corrects an error of the calculated position and time information by using another satellite. In addition, the GPS can calculate speed information by continuously calculating current position information in real time.
The a/V input unit 120 is used to receive an audio or video signal. The a/V input unit 120 may include a camera 121 and a microphone 122, and the camera 121 processes image data of still pictures or video obtained by an image capturing apparatus in a video capturing mode or an image capturing mode. The processed image frames may be displayed on the display unit 151. The image frames processed by the cameras 121 may be stored in the memory 160 (or other storage medium) or transmitted via the wireless communication unit 110, and two or more cameras 121 may be provided according to the construction of the mobile terminal 100. The microphone 122 may receive sounds (audio data) via the microphone 122 in a phone call mode, a recording mode, a voice recognition mode, or the like, and is capable of processing such sounds into audio data. The processed audio (voice) data may be converted into a format output transmittable to a mobile communication base station via the mobile communication module 112 in case of a phone call mode. The microphone 122 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 audio signals.
The user input unit 130 may generate key input data to control various operations of the mobile terminal 100 according to a command input by a user. The user input unit 130 allows a user to input various types of information, and may include a keyboard, dome sheet, touch pad (e.g., a touch-sensitive member that detects changes in resistance, pressure, capacitance, and the like due to being touched), scroll wheel, joystick, and the like. In particular, when the touch pad is superimposed on the display unit 151 in the form of a layer, a touch screen may be formed.
The sensing unit 140 detects a current state of the mobile terminal 100 (e.g., an open or closed state of the mobile terminal 100), a position of the mobile terminal 100, presence or absence of contact (i.e., touch input) by a user with the mobile terminal 100, an orientation of the mobile terminal 100, acceleration or deceleration movement and direction of the mobile terminal 100, and the like, and generates a command or signal for controlling an operation of the mobile terminal 100. For example, when the mobile terminal 100 is implemented as a slide-type mobile phone, the sensing unit 140 may sense whether the slide-type phone is opened or closed. In addition, the sensing unit 140 can detect whether the power supply unit 190 supplies power or whether the interface unit 170 is coupled with an external device. The sensing unit 140 may include a proximity sensor 141.
The interface unit 170 serves as an interface through which at least one external device is connected to the mobile terminal 100. For example, the external device may include a wired or wireless headset port, an external power supply (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 identification module may store various information for authenticating a user using the mobile terminal 100 and may include a User Identity Module (UIM), a Subscriber Identity Module (SIM), a Universal Subscriber Identity Module (USIM), and the like. In addition, a device having an identification module (hereinafter, referred to as an "identification device") may take the form of a smart card, and thus, the identification device may be connected with the mobile terminal 100 via a port or other connection means. The interface unit 170 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 the external device.
In addition, when the mobile terminal 100 is connected with an external cradle, the interface unit 170 may serve as a path through which power is supplied from the cradle to the mobile terminal 100 or may serve as a path through which various command signals input from the cradle are transmitted to the mobile terminal 100. Various command signals or power input from the cradle may be used as a signal for identifying whether the mobile terminal 100 is accurately mounted on the cradle. The output unit 150 is configured to provide output signals (e.g., audio signals, video signals, alarm signals, vibration signals, etc.) in a visual, audio, and/or tactile manner. The output unit 150 may include a display unit 151, an audio output module 152, an alarm unit 153, and the like.
The display unit 151 may display information processed in the mobile terminal 100. For example, when the mobile terminal 100 is in a phone call mode, the display unit 151 may display a User Interface (UI) or a Graphical User Interface (GUI) related to a call or other communication (e.g., text messaging, multimedia file downloading, etc.). When the mobile terminal 100 is in a video call mode or an image capturing mode, the display unit 151 may display a captured image and/or a received image, a UI or GUI showing a video or an image and related functions, and the like.
Meanwhile, when the display unit 151 and the touch pad are overlapped with each other in the form of a layer to form a touch screen, the display unit 151 may serve as an input device and an output device. The display unit 151 may include at least one of a Liquid Crystal Display (LCD), a thin film transistor LCD (TFT-LCD), an Organic Light Emitting Diode (OLED) display, a flexible display, a three-dimensional (3D) display, and the like. Some of these displays may be configured to be transparent to allow a user to view from the outside, which may be referred to as transparent displays, and a typical transparent display may be, for example, a TOLED (transparent organic light emitting diode) display or the like. Depending on the particular desired implementation, mobile terminal 100 may include two or more display units (or other display devices), for example, mobile terminal 100 may include an external display unit (not shown) and an internal display unit (not shown). The touch screen may be used to detect a touch input pressure as well as a touch input position and a touch input area.
The audio output module 152 may convert audio data received by the wireless communication unit 110 or stored in the memory 160 into an audio signal and output as sound when the mobile terminal 100 is in a call signal reception mode, a call mode, a recording mode, a voice recognition mode, a broadcast reception mode, or the like. Also, the audio output module 152 may provide audio output related to a specific function performed by the mobile terminal 100 (e.g., a call signal reception sound, a message reception sound, etc.). The audio output module 152 may include a speaker, a buzzer, and the like.
The alarm unit 153 may provide an output to notify the mobile terminal 100 of the occurrence of an event. Typical events may include call reception, message reception, key signal input, touch input, and the like. In addition to audio or video output, the alarm unit 153 may provide output in different ways to notify the occurrence of an event. For example, the alarm unit 153 may provide an output in the form of vibration, and when a call, a message, or some other incoming communication (communicating communication) is received, the alarm unit 153 may provide a tactile output (i.e., vibration) to inform the user thereof. By providing such a tactile output, the user can recognize the occurrence of various events even when the user's mobile phone is in the user's pocket. The alarm unit 153 may also provide an output notifying the occurrence of an event via the display unit 151 or the audio output module 152.
The memory 160 may store software programs and the like for processing and controlling operations performed by the controller 180, or may temporarily store data (e.g., a phonebook, messages, still images, videos, and the like) that has been or will be output. Also, the memory 160 may store data regarding various ways of vibration and audio signals output when a touch is applied to the touch screen.
The memory 160 may include at least one type of storage medium including a flash memory, a hard disk, a multimedia card, a card-type memory (e.g., SD or DX memory, etc.), a Random Access Memory (RAM), a Static Random Access Memory (SRAM), a read-only memory (ROM), an electrically erasable programmable read-only memory (EEPROM), a programmable read-only memory (PROM), a magnetic memory, a magnetic disk, an optical disk, and the like. Also, the mobile terminal 100 may cooperate with a network storage device that performs a storage function of the memory 160 through a network connection.
The controller 180 generally controls the overall operation of the mobile terminal. For example, the controller 180 performs control and processing related to voice calls, data communications, video calls, and the like. In addition, the controller 180 may include a multimedia module 181 for reproducing (or playing back) multimedia data, and the multimedia module 181 may be constructed within the controller 180 or may be constructed separately from the controller 180. The controller 180 may perform a pattern recognition process to recognize a handwriting input or a picture drawing input performed on the touch screen as a character or an image.
The power supply unit 190 receives external power or internal power and provides appropriate power required to operate various elements and components under the control of the controller 180.
The various embodiments described herein may be implemented in a computer-readable medium using, for example, computer software, hardware, or any combination thereof. For a hardware implementation, the embodiments described herein may be implemented using at least one of an Application Specific Integrated Circuit (ASIC), a Digital Signal Processor (DSP), a Digital Signal Processing Device (DSPD), a Programmable Logic Device (PLD), a Field Programmable Gate Array (FPGA), a processor, a controller, a microcontroller, a microprocessor, an electronic unit designed to perform the functions described herein, and in some cases, such embodiments may be implemented in the controller 180. For a software implementation, the implementation such as a process or a function may be implemented with a separate software module that allows performing at least one function or operation. The software codes may be implemented by software applications (or programs) written in any suitable programming language, which may be stored in the memory 160 and executed by the controller 180.
Up to this point, the mobile terminal 100 has been described in terms of its functionality. In addition, the mobile terminal 100 in the embodiment of the present invention may be a mobile terminal such as a folder type, a bar type, a swing type, a slide type, and other various types, and is not limited herein.
The mobile terminal 100 as shown in fig. 1 may be configured to operate with communication systems such as wired and wireless communication systems and satellite-based communication systems that transmit data via frames or packets.
A communication system in which a mobile terminal according to the present invention is operable will now be described with reference to fig. 2.
Such communication systems may use different air interfaces and/or physical layers. For example, the air interface used by the communication system includes, for example, Frequency Division Multiple Access (FDMA), Time Division Multiple Access (TDMA), Code Division Multiple Access (CDMA), and Universal Mobile Telecommunications System (UMTS) (in particular, Long Term Evolution (LTE)), global system for mobile communications (GSM), and the like. By way of non-limiting example, the following description relates to a CDMA communication system, but such teachings are equally applicable to other types of systems.
Referring to fig. 2, a CDMA wireless communication system may include a plurality of intelligent terminals 100, a plurality of Base Stations (BSs) 270, Base Station Controllers (BSCs) 275, and a Mobile Switching Center (MSC) 280. The MSC 280 is configured to interface with a Public Switched Telephone Network (PSTN) 290. The MSC 280 is also configured to interface with a BSC275, which may be coupled to the base station 270 via a backhaul. The backhaul line may be constructed according to any of several known interfaces, which may include, for example, european/american standard high capacity digital lines (E1/T1), Asynchronous Transfer Mode (ATM), network protocol (IP), point-to-point protocol (PPP), frame relay, high-rate digital subscriber line (HDSL), Asymmetric Digital Subscriber Line (ADSL), or various types of digital subscriber lines (xDSL). It will be understood that a system as shown in fig. 2 may include multiple BSCs 275.
Each BS 270 may serve one or more sectors (or regions), each sector covered by a multi-directional antenna or an antenna pointing in a particular direction being radially distant from the BS 270. Alternatively, each partition may be covered by two or more antennas for diversity reception. Each BS 270 may be configured to support multiple frequency allocations, with each frequency allocation having a particular frequency spectrum (e.g., 1.25MHz, 5MHz, etc.).
The intersection of partitions with frequency allocations may be referred to as a CDMA channel. The BS 270 may also be referred to as a Base Transceiver Subsystem (BTS) or other equivalent terminology. In such a case, the term "base station" may be used to generically refer to a single BSC275 and at least one BS 270. The base stations may also be referred to as "cells". Alternatively, each partition of a particular BS 270 may be referred to as a plurality of cell sites.
As shown in fig. 2, a Broadcast Transmitter (BT)295 transmits a broadcast signal to the mobile terminal 100 operating within the system. A broadcast receiving module 111 as shown in fig. 1 is provided at the mobile terminal 100 to receive a broadcast signal transmitted by the BT 295. In fig. 2, several Global Positioning System (GPS) satellites 300 are shown. The satellite 300 assists in locating at least one of the plurality of mobile terminals 100.
In fig. 2, a plurality of satellites 300 are depicted, but it is understood that useful positioning information may be obtained with any number of satellites. The location information module 115 (e.g., GPS) as shown in fig. 1 is generally configured to cooperate with the satellites 300 to obtain desired positioning information. Other techniques that can track the location of the mobile terminal may be used instead of or in addition to GPS tracking techniques. In addition, at least one GPS satellite 300 may selectively or additionally process satellite DMB transmission.
As a typical operation of the wireless communication system, the BS 270 receives reverse link signals from various mobile terminals 100. The mobile terminal 100 is generally engaged in conversations, messaging, and other types of communications. Each reverse link signal received by a particular base station is processed within a particular BS 270. The obtained data is forwarded to the associated BSC 275. The BSC provides call resource allocation and mobility management functions including coordination of soft handoff procedures between BSs 270. The BSCs 275 also route the received data to the MSC 280, which provides additional routing services for interfacing with the PSTN 290. Similarly, the PSTN290 interfaces with the MSC 280, the MSC interfaces with the BSCs 275, and the BSCs 275 accordingly control the BS 270 to transmit forward link signals to the mobile terminal 100.
Based on the above mobile terminal, an embodiment of the present invention provides a sound processing method, and fig. 3 is a flowchart of the sound processing method according to the embodiment of the present invention, as shown in fig. 3, the method includes the following steps:
step S302, collecting the environmental sound around the area where the user is located through a sound sensor of the earphone, and separating the collected environmental sound;
step S304, identifying and matching the separated various sounds with sample sounds in a database;
step S306, comprehensively calculating decibels of the successfully matched environmental sounds to obtain an environmental sound comprehensive sound value H1;
step S308, processing is carried out according to the noise reduction value of the earphone and the environment sound comprehensive sound value H1, and the mixed sound value H2 is obtained;
and step S310, adjusting the sound channel output of the earphone according to the calculated mixed sound value H2.
Through the steps, the ambient sound around the area where the user is located is collected through the sound sensor of the earphone, and the collected ambient sound is separated; identifying and matching the separated various sounds with sample sounds in a database; calculating a mixed sound value according to the successfully matched environmental sound; the sound channel output of the earphone is adjusted according to the calculated mixed sound value, the problem that intelligent sound output cannot be carried out according to the environment where the user is located in the related technology is solved, the environment sound is collected and separated, the sound channel output of the earphone is optimized according to the calculated mixed sound value, the user obtains more comfortable earphone sound reception experience, and the user experience is improved.
In order to reduce the complexity of subsequent separation, before the separation processing is performed on the collected environment sound, the dimension reduction processing is performed on the mixed speech, including the noise reduction processing is performed on the collected mixed speech signal containing the environment sound, and the whitening processing is performed on the mixed speech signal after the noise reduction processing.
Preferably, after the mixed sound value H2 is obtained by processing the noise reduction value of the earphone and the ambient sound integrated sound value H1, the H2 is uploaded to a multimedia sound library.
Adjusting the channel output of the headphones according to the calculated mixed sound value may include: adding the H2 to the current pre-output sound value H3; comparing and analyzing the added result with a preset comfortable value interval; and adjusting the volume of the sound channel output of the earphone according to the analysis result.
Preferably, adjusting the volume of the sound channel output of the headphone according to the analysis result comprises:
if H2+ H3 > comfort value interval, automatically reducing the sound channel output of the earphone to the comfort value interval;
if H2+ H3 belongs to the comfortable value interval, the sound channel output of the earphone is output according to the sound value H3;
and if H2+ H3 is less than the comfortable value range, automatically increasing the sound channel output of the earphone to the comfortable value range.
The embodiment of the invention realizes the separation of the voice signals by using the FastICA algorithm, and the whole process needs the input of signals, the separation of FastICA voice signals and the output of final result voice signals.
The voice signal Separation processing is to process a section of voice signal detected by a microphone by using a Blind Source Separation (BSS). For example, three speech signals are used as input signals, and then mixed and processed. In mixing three voice signals, the three signals are processed into a matrix in a matrix manner. And multiplying the combined signal matrix by a random weight matrix to obtain a processed mixed signal. And pre-processing the mixed observation signals before ICA. Prior to ICA processing, the observed data is usually pre-processed, mainly by signal centering and whitening. Signal centering is the centering of the observed signal X by subtracting its mean value E { X }, such that E { X' } is 0, X ═ E { X }, which is called signal X. After the unmixing, the mean vector is added back to the signal resulting from the unmixing. The whitening process is to apply a linear transformation to the observed signal X to V, i.e., V ═ MX, and let E { VV' } ═ I, where M is the whitening matrix. The transformation simply removes the correlation between the observed signals X. If the effect of reducing the dimension is achieved, the characteristic value decomposition can be carried out on the covariance matrix of the optical fiber, and the effect of reducing the dimension is achieved. And processing the preprocessed data through a FastICA algorithm, and performing characteristic extraction processing on the signals so as to finish the separation of the voice signals.
The separation of the separated mixed speech is explained in detail below. The method mainly comprises the following steps:
first, the pre-processing of the mixed speech signal. Preprocessing the signals before separating the mixed signals is very necessary, and in order to reduce the interference of noise, firstly, noise reduction processing is carried out on the sampled signals; and then, the de-averaging and de-correlation spheroidizing treatment is carried out on the de-noised data, so that the calculation amount can be reduced. The pretreatment process comprises the following steps:
and (3) noise reduction treatment: LMS (least mean square) filtering can quickly track the changing signal and automatically adjust the parameters thereof to achieve the best filtering effect, and the method is adopted to filter each mixed signal and carry out filtering processing on the ith signal mi＝[mi(1),...,mi(N)]The specific LMS filtering process is as follows:
firstly, initializing parameters, setting simulation times g, the length N of mi, the order k of an LMS filter, u being 0.001, and the current simulation times q being 1;
second, input signal miTaking the first k values as the first k values of the output x, initializing i to k +1, and setting the initial tap weight value as a 0 matrix w of a row and k columns;
the third step: m is to beiThe values from (i) to (k) are taken as a column vector XN, then the filter output x (i) w XN of the point (i) is calculated, and the deviation e (i) m (i) -x (i) of each point is calculated;
the matrix w, w +2 u e (i) XN' is updated again.
And (3) repeating the simulation in the step (3), adding 1 to q every time of the simulation until the simulation times reach g, and outputting all x (i) to form a vector x with a row and N columns, wherein x is the output of the filter.
The whitening treatment comprises two steps: the most basic and necessary pre-processing de-averaging and de-correlation. The advantage of de-averaging is that zero-mean data is easy to calculate; decorrelation can reduce correlation, find out and remove signals with smaller eigenvalues, reduce the number of estimated source signals, reduce the amount of computation, and the like. In actual calculation, each path of microphone obtains filtered signal x by adopting arithmetic mean value instead of mathematical expectation, and the ith path of signal is subjected to mean value removal as follows:
decorrelation is a covariance matrix of x0 decomposed by eigenvalues
Wherein D is Rx0A diagonal matrix composed of eigenvalues, Q is a matrix composed of eigenvectors corresponding to the eigenvalues, and the whitening matrix T is obtained as D-1/ 2QTBy transformation z ═ Tx0A whitened signal z is obtained.
Separating mixed signals, namely separating the preprocessed signals z by adopting a step-length-variable natural gradient algorithm, wherein a core separation matrix of the separation algorithm is as follows:
W(k+1)＝W(k)+η(k)[I-f(y)yT]W(k)(2)
the speech signal is a super gaussian signal and the non-linear function is selected from f (y) tan (y).
Multiplying each element of the gradient of the current moment by the corresponding element of the gradient of the previous moment, and then taking
The length of the variable step length is taken as the variable quantity of the step length, and the actual step length adjusting formula is as follows:
when the set iteration times are reached, a separation matrix W can be obtained, and the estimated values of all the original signals are obtained:
y＝Wz
y＝[y1,y2,...,yn]Tthe specific separation process comprises the following specific steps:
the first step is as follows: initializing parameters: simulation times maxits, a signal z to be separated, the number of rows N and columns (the number of samples of the signal N), a separation step ga _ W, an adjustment factor ro, generally set ro equal to 0.01, generally set a separation matrix W to be 0.1 times of an N-dimensional unit matrix, total being the gradient of the current moment, the gradient of the previous moment of total _ old, and initialization of total and total _ old as a zero matrix;
the second step is that: the signal z to be separated is divided into nb blocks, each of which has a length bsize of (2 × N)/(nb + 1);
the third step: within each block the following operations are performed:
calculating a matrix consisting of (k-1) × bsize/2+1 columns to (k +1) × bsize/2 columns, where y ═ W:, (k-1) × bsize/2+1, (k +1) × bsize/2, multiplied by W;
and solving a nonlinear function value of each point of y:
fy＝tanh(y)；
updating the separation matrix W:
if it is the first loop then calculate:
tal＝(I-fy*y'/bsize)*W；
W＝W+ga_W*(I-fy*y'/bsize)*W；
tal_old＝tal；
otherwise, calculating:
tal＝(I-fy*y'/bsize)*W；
ga_W＝ga_W+ro*trace(tal*tal_old')；
W＝W+ga_W*(I-fy*y'/bsize)*W；
tal_old＝tal；
until all nb blocks are calculated; carrying out maxits iteration on the third step cycle to obtain a separation matrix W;
the fourth step: each row vector of y is an estimate of the original speech signal.
Fig. 4 is a schematic diagram of separating mixed sounds according to an embodiment of the present invention, as shown in fig. 4, actually received data is complex and various, and includes useful information and useless information, especially when the number of sensors is large, the calculation is complex, and then the observed data is preprocessed, so that the data dimension can be reduced under certain conditions, and the calculation amount of the subsequent processing can be reduced.
The separating the mixed sound specifically includes: given that the image of the sound file is not intuitive enough, the simulation can be done in different forms by two main functions. And functionalizing each functional module, and completing simulation in a function calling mode. This makes the flow of the ICA algorithm appear clearer in the main program, and enhances the portability of the simulation program. (main _ ica. m mainly completes the separation of the sound file; main _ ica _ fig. m mainly completes the separation of the signal generated by the program).
Fig. 5 is a flowchart of a mixed sound process according to an embodiment of the present invention, as shown in fig. 5, in which sound signals are read, mixed, then unmixed, and output.
The mobile terminal receives the surrounding environment sound of the region where the user is located through the sound tube, transmits the surrounding environment sound to the CPU terminal for analysis and processing, and separates various sounds including but not limited to subway sound, crowd noise, horn sound and the like. Various sounds are identified and matched to sample sounds in a large database. And after matching, performing comprehensive calculation of decibels to obtain a decibel value interval range of the environmental sound, namely the environmental sound comprehensive sound value H1. The mobile terminal identifies the noise reduction value of the inserted earphone to carry out comprehensive addition and subtraction processing with the decibel value of the environmental sound, namely the mixed sound value H2. The mobile terminal center uploads H2 to the multimedia sound library, adds the current pre-output sound value H3 and compares the result with the preset comfortable value interval, if H2+ H3 > the comfortable value interval, the earphone sound channel output will automatically reduce the volume output to the comfortable value interval. If H2+ H3 ∈ comfortable value interval, the output of the headphone channel will be output according to the sound value H3. If H2+ H3 < comfort level range, the volume will be automatically increased at the headphone output to the comfort level range. Fig. 6 is a flow chart of the mixed sound processing according to the preferred embodiment of the present invention, as shown in fig. 6, specifically including the following steps:
step S601, collecting environmental sound through a sound sensor;
step S602, separating the collected environmental sound, and outputting sound values of each sound source, which are respectively marked as M1, M2 and M3 …;
step S603, identifying and matching M1, M2 and M3 … with samples in a large database;
step S604, judging whether the matching is successful, if the judgment result is No, executing the step S605, and if the judgment result is yes, executing the step S606;
step S605, removing small noise, sound break and the like;
step S606, calculating and obtaining a decibel value interval range of the environmental sound, namely the environmental sound comprehensive sound value H1;
step S607, the noise reduction value of the earphone inserted in the mobile phone (i.e. mobile terminal) and the comprehensive sound value H1 of the environment sound are processed by comprehensive operation, and the mixed sound value H2 is output;
step S608, adding H2 and the current pre-output sound value H3, and comparing and judging the result with a preset comfortable value interval;
step S609, if H2+ H3 is larger than the comfortable value interval, the earphone sound channel output will automatically reduce the volume output to the comfortable value interval;
step S610, if H2+ H3 belongs to a comfortable value interval, outputting a sound value H3 at the output end of the earphone sound channel;
in step S611, if H2+ H3 is less than the comfort value range, the headphone channel output will automatically increase the volume to the comfort value range.
The earphone is not only stopped in the aspect of blocking the interference of environmental sounds to the ears, namely noise reduction, but also used for judging and optimizing the output value of the earphone sound channel by calculating a more accurate mixed sound value through collecting and analyzing various sounds of the surrounding environment of the human body, so that a user can obtain more comfortable earphone sound receiving experience. The large database and multi-layer algorithm operation are combined, and the user is provided with the experience that the user can hear comfortable sound output by the earphone under any environment.
The microphone collects the ambient noise signal of the scene where the headset is located. The microphone may be an ECM microphone that collects ambient noise signals at predetermined intervals in the scene in which the headset is located.
And comparing the noise sound pressure value with a preset threshold value after acquiring the noise sound pressure value according to the environment noise signal. The method includes the steps of firstly performing analog-to-digital conversion on an environment noise signal, then analyzing a digital environment noise signal to obtain a noise sound pressure value, and then comparing the noise sound pressure value with a preset threshold value. Wherein the predetermined threshold comprises a first predetermined threshold and a second predetermined threshold, preferably the first predetermined threshold is 65dB, and the second predetermined threshold is any one of 30 to 40 dB.
And loading the noise reduction parameters according to the comparison result to generate an inverse noise signal for counteracting the environment noise signal. The step of loading the noise reduction parameter by the noise reduction module according to the comparison result to generate an inverse noise signal for canceling the ambient noise signal includes: obtaining amplitude response and phase response corresponding to passive noise reduction of the environment noise signal by a noise reduction module; obtaining amplitude response and phase response corresponding to active noise reduction according to the amplitude response and the phase response corresponding to the passive noise reduction by the noise reduction module; obtaining noise reduction parameters by a noise reduction module according to amplitude response and phase response corresponding to active noise reduction; and loading the noise reduction parameters by the noise reduction module according to the comparison result to generate an inverse noise signal for offsetting the environment noise signal.
The passive noise reduction means that the earphone isolates external environment noise from human ears by using physical characteristics, and noise is mainly blocked by a sound insulation material. Active noise reduction is to generate an inverse signal for offsetting an ambient noise signal after capturing the ambient noise outside the earphone through a microphone, and then play the signal through a loudspeaker.
In this embodiment, the phase-inverted noise signal includes a first phase-inverted noise signal and a second phase-inverted noise signal, wherein the first phase-inverted noise signal has a larger noise reduction amplitude than the second phase-inverted noise signal.
When the sound pressure value of the noise is greater than or equal to the first preset threshold value, the scene where the earphone is located is very noisy, and the noise reduction processing needs to be performed on the environmental noise signal to a larger extent. When the noise sound pressure value is smaller than the first preset threshold value and larger than or equal to the second preset threshold value, the scene where the earphone is located is generally noisy, only noise reduction processing with a small amplitude is needed to be carried out on the environmental noise signal, and at the moment, the noise reduction module is loaded to enable the noise reduction module to generate noise reduction parameters of a second opposite phase noise signal. When the noise sound pressure is smaller than the second preset threshold value, the scene where the earphone is located is quite quiet, noise reduction processing on an environment noise signal is not needed, and at the moment, noise reduction parameters enabling the noise reduction module to be in an out-of-operation state are loaded by the noise reduction module.
According to another aspect of the embodiments of the present invention, there is also provided a sound processing mobile terminal, and fig. 7 is a block diagram of the sound processing mobile terminal according to the embodiments of the present invention, as shown in fig. 7, including:
a separation module 72, configured to collect, through a sound sensor of an earphone, ambient sound around an area where a user is located, and separate the collected ambient sound;
the identification matching module 74 is used for identifying and matching the separated various sounds with sample sounds in the database;
the comprehensive calculation module 76 is configured to perform comprehensive calculation on the successfully matched environmental sounds in decibels to obtain an environmental sound comprehensive sound value H1;
a processing module 78, configured to perform processing according to the noise reduction value of the earphone and the ambient sound combination sound value H1, to obtain the mixed sound value H2;
and the adjusting module 710 is configured to adjust the channel output of the headphone according to the calculated mixed sound value H2.
Preferably, the mobile terminal further includes:
the noise reduction processing module is used for carrying out noise reduction processing on the collected mixed voice signal containing the environmental sound before carrying out separation processing on the collected environmental sound;
and the whitening processing module is used for whitening the mixed voice signal after the noise reduction processing.
Preferably, the mobile terminal further includes:
and the uploading module is used for uploading the H2 to a multimedia sound library after the mixed sound value H2 is obtained by processing the noise reduction value of the earphone and the environmental sound comprehensive sound value H1.
Fig. 8 is a block diagram of a sound processing mobile terminal according to a preferred embodiment of the present invention, and as shown in fig. 8, the adjusting module 710 includes:
an adding unit 82 for adding the H2 to the current pre-output sound value H3;
a comparison and analysis unit 84, configured to perform comparison and analysis on the added result and a preset comfort value interval;
and the adjusting unit 86 is used for adjusting the volume of the sound channel output by the earphone according to the analysis result.
Preferably, the adjusting unit 86 is also used for
If H2+ H3 > comfort value interval, automatically reducing the sound channel output of the earphone to the comfort value interval;
if H2+ H3 belongs to the comfortable value interval, the sound channel output of the earphone is output according to the sound value H3;
and if H2+ H3 is less than the comfortable value range, automatically increasing the sound channel output of the earphone to the comfortable value range.
According to the embodiment of the invention, the environmental sounds around the area where the user is located are collected through the sound sensor of the earphone, and the collected environmental sounds are separated; identifying and matching the separated various sounds with sample sounds in a database; calculating a mixed sound value according to the successfully matched environmental sound; the sound channel output of the earphone is adjusted according to the calculated mixed sound value, the problem that intelligent sound output cannot be carried out according to the environment where the user is located in the related technology is solved, the environment sound is collected and separated, the sound channel output of the earphone is optimized according to the calculated mixed sound value, the user obtains more comfortable earphone sound reception experience, and the user experience is improved.
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 an … …" does not exclude the presence of other like elements in a process, method, article, or apparatus that comprises the element.
The above-mentioned serial numbers of the embodiments of the present invention are merely for description and do not represent the merits of the embodiments.
Through the above description of the embodiments, those skilled in the art will clearly understand that the method of the above embodiments can be implemented by software plus a necessary general hardware platform, and certainly can also be implemented by hardware, but in many cases, the former is a better implementation manner. Based on such understanding, the technical solutions of the present invention may be embodied in the form of a software product, which is stored in a storage medium (such as ROM/RAM, magnetic disk, optical disk) and includes instructions for enabling a terminal device (such as a mobile phone, a computer, a server, an air conditioner, or a network device) to execute the method according to the embodiments of the present invention.
It will be apparent to those skilled in the art that the modules or steps of the present invention described above may be implemented by a general purpose computing device, they may be centralized on a single computing device or distributed across a network of multiple computing devices, and alternatively, they may be implemented by program code executable by a computing device, such that they may be stored in a storage device and executed by a computing device, and in some cases, the steps shown or described may be performed in an order different than that described herein, or they may be separately fabricated into individual integrated circuit modules, or multiple ones of them may be fabricated into a single integrated circuit module. Thus, the present invention is not limited to any specific combination of hardware and software.
The above description is only a preferred embodiment of the present invention, and not intended to limit the scope of the present invention, and all modifications of equivalent structures and equivalent processes, which are made by using the contents of the present specification and the accompanying drawings, or directly or indirectly applied to other related technical fields, are included in the scope of the present invention.
Claims (10)
1. A sound processing method, comprising:
collecting environmental sounds around an area where a user is located through a sound sensor of an earphone, separating the collected environmental sounds, and outputting sound values of sound sources;
identifying and matching the sound values of the sound sources with sample sounds in a database;
comprehensively calculating decibels of the successfully matched environmental sounds to obtain an environmental sound comprehensive sound value H1;
processing according to the noise reduction value of the earphone and the environment sound comprehensive sound value H1 to obtain a mixed sound value H2;
and adjusting the sound channel output of the earphone according to the calculated mixed sound value H2.
2. The method of claim 1, wherein prior to the separating the collected environmental sounds, the method further comprises:
carrying out noise reduction processing on the collected mixed voice signal containing the environmental sound;
and carrying out whitening processing on the mixed voice signal after the noise reduction processing.
3. The method according to claim 1, wherein after obtaining the mixed sound value H2 by processing the noise reduction value of the headphone and the ambient sound integrated sound value H1, the method further comprises:
uploading the H2 to a multimedia sound library.
4. The method according to any one of claims 1 to 3, wherein adjusting the channel output of the headphones according to the calculated mixed sound value H2 comprises:
adding the H2 to the current pre-output sound value H3;
comparing and analyzing the added result with a preset comfortable value interval;
and adjusting the volume of the sound channel output of the earphone according to the analysis result.
5. The method of claim 4, wherein adjusting the volume level of the channel output of the headphone according to the analysis comprises:
if H2+ H3 > comfort value interval, automatically reducing the sound channel output of the earphone to the comfort value interval;
if H2+ H3 belongs to the comfortable value interval, the sound channel output of the earphone is output according to the sound value H3;
and if H2+ H3 is less than the comfortable value range, automatically increasing the sound channel output of the earphone to the comfortable value range.
6. A sound processing mobile terminal, comprising:
the separation module is used for collecting the environmental sounds around the area where the user is located through a sound sensor of the earphone, separating the collected environmental sounds and outputting the sound values of the sound sources;
the identification matching module is used for identifying and matching the sound values of the sound sources with sample sounds in a database;
the comprehensive calculation module is used for performing comprehensive calculation on the successfully matched environmental sounds in decibels to obtain an environmental sound comprehensive sound value H1;
the processing module is used for processing the noise reduction value of the earphone and the environment sound comprehensive sound value H1 to obtain a mixed sound value H2;
and the adjusting module is used for adjusting the sound channel output of the earphone according to the calculated mixed sound value H2.
7. The mobile terminal of claim 6, wherein the mobile terminal further comprises:
the noise reduction processing module is used for carrying out noise reduction processing on the collected mixed voice signal containing the environmental sound before carrying out separation processing on the collected environmental sound;
and the whitening processing module is used for whitening the mixed voice signal after the noise reduction processing.
8. The mobile terminal of claim 6, wherein the mobile terminal further comprises:
and the uploading module is used for uploading the H2 to a multimedia sound library after the mixed sound value H2 is obtained by processing the noise reduction value of the earphone and the environmental sound comprehensive sound value H1.
9. The mobile terminal according to any of claims 6 to 8, wherein the adjusting module comprises:
an adding unit for adding the H2 to the current pre-output sound value H3;
the comparison and analysis unit is used for comparing and analyzing the added result with a preset comfortable value interval;
and the adjusting unit is used for adjusting the volume output by the sound channel of the earphone according to the analysis result.
10. The mobile terminal of claim 9, wherein the adjusting unit is further configured to adjust the mobile terminal according to the received signal strength
If H2+ H3 > comfort value interval, automatically reducing the sound channel output of the earphone to the comfort value interval;
if H2+ H3 belongs to the comfortable value interval, the sound channel output of the earphone is output according to the sound value H3;
and if H2+ H3 is less than the comfortable value range, automatically increasing the sound channel output of the earphone to the comfortable value range.
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| CN106254648B (en) * | 2016-08-01 | 2018-09-18 | 珠海格力电器股份有限公司 | Adjust the method, apparatus and mobile device of mobile device volume |
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