CN106686249B - Voice communication method and mobile terminal - Google Patents
Voice communication method and mobile terminal Download PDFInfo
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- CN106686249B CN106686249B CN201710031871.3A CN201710031871A CN106686249B CN 106686249 B CN106686249 B CN 106686249B CN 201710031871 A CN201710031871 A CN 201710031871A CN 106686249 B CN106686249 B CN 106686249B
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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
- H04M1/72403—User interfaces specially adapted for cordless or mobile telephones with means for local support of applications that increase the functionality
- H04M1/7243—User interfaces specially adapted for cordless or mobile telephones with means for local support of applications that increase the functionality with interactive means for internal management of messages
- H04M1/72433—User interfaces specially adapted for cordless or mobile telephones with means for local support of applications that increase the functionality with interactive means for internal management of messages for voice messaging, e.g. dictaphones
Abstract
The invention provides a voice call method and a mobile terminal, wherein the method comprises the following steps: after a microphone in the mobile terminal is started, judging the current call state of the mobile terminal; if the current call state is a voice answering state, adjusting an uplink amplification gain value of the voice signal to a first preset gain value; if the current call state is a voice sending state, adjusting an uplink amplification gain value of the voice signal to a second preset gain value, wherein the first preset gain value is smaller than the second preset gain value. The voice call scheme provided by the invention can provide high-quality voice call service for users.
Description
Technical Field
The present invention relates to the field of communications technologies, and in particular, to a voice call method and a mobile terminal.
Background
With the rapid development of mobile internet and the dependence of young people pursuing fashion on mobile terminals, mobile terminals are becoming more and more popular, and manufacturers of large mobile terminals are also continuously improving and beautifying functions of their products. In order to improve the voice call quality, most mobile terminals are provided with a call voice drying mechanism, so that when the mobile terminals are in a noisy city and a noisy environment for calling, both parties can hear the voice of the other party clearly, and the voice call quality is improved.
Currently, the most commonly used call voice noise cancellation methods for mobile terminals mainly include a single-microphone spectrum noise reduction and cancellation method and a multi-microphone noise cancellation method. The single-microphone spectrum noise reduction and elimination method can only eliminate steady-state noise generally, and has no obvious inhibition effect on unsteady-state noise (such as other noise except some other non-conversation voices in shopping malls, supermarkets, restaurants and the like). The method can eliminate unsteady noise, so that the multi-microphone noise elimination method basically replaces a single-microphone spectrum noise reduction and elimination method.
When the mobile terminal is developed and designed, a design developer sets an uplink amplification gain value of a sound signal of the mobile terminal to a relatively high fixed value, and the setting causes the following problems of the multi-microphone noise cancellation method:
firstly, noise carried in a circuit board of the mobile terminal, such as noise of bottom noise, current sound and the like, is transmitted to sound signals collected by a microphone in a conduction or coupling mode, and the sound signals become noise after being amplified according to an uplink amplification gain value and are transmitted to opposite-end equipment;
secondly, in a noisy environment, such as a vegetable market, a supermarket, a restaurant and the like, when the mobile terminal is in a voice state of answering the voice sent by the opposite terminal equipment, the mobile terminal can send ambient environmental noise to the opposite terminal equipment through amplification of a fixed uplink amplification gain value in the mobile terminal, and therefore answering experience of a user of the opposite terminal equipment is affected.
Therefore, the existing multi-microphone noise elimination method has poor effect and cannot provide high-quality voice call service for users.
Disclosure of Invention
The embodiment of the invention provides a voice call method and a mobile terminal, and aims to solve the problems that the noise elimination effect in the prior art is poor and high-quality voice call service cannot be provided for a user.
According to an aspect of the present invention, there is provided a voice call method, wherein the method includes: after a microphone in the mobile terminal is started, judging the current call state of the mobile terminal; if the current call state is a voice answering state, adjusting an uplink amplification gain value of the voice signal to a first preset gain value; if the current call state is a voice sending state, adjusting an uplink amplification gain value of the voice signal to a second preset gain value, wherein the first preset gain value is smaller than the second preset gain value.
According to another aspect of the present invention, there is provided a mobile terminal, wherein the mobile terminal comprises: the mobile terminal comprises a call state judgment module, a call state judgment module and a call state judgment module, wherein the call state judgment module is used for judging the current call state of the mobile terminal after a microphone in the mobile terminal is started; the first adjusting module is used for adjusting the uplink amplification gain value of the sound signal to a first preset gain value if the current call state is a voice answering state; and the second adjusting module is used for adjusting the uplink amplification gain value of the sound signal to a second preset gain value if the current call state is a voice sending state, wherein the first preset gain value is smaller than the second preset gain value.
Compared with the prior art, the invention has the following advantages:
according to the voice call method and the mobile terminal provided by the invention, the current call state of the mobile terminal is determined to be a voice receiving state or a voice sending state in real time in the call process of the mobile terminal, and the uplink amplification gain value of the voice signal is adaptively adjusted in real time according to the call state. Specifically, when the mobile terminal is in a voice emitting state, the uplink amplification gain value of the voice signal is adjusted to a second preset gain value, so that the user of the opposite terminal device can clearly hear the voice emitted by the mobile terminal. When the mobile terminal is in a voice answering state, the uplink amplification gain value of the voice signal is adjusted to be the first preset gain value, and the uplink amplification gain value of the voice signal is reduced, so that even if the mobile terminal is currently in a noise environment, the mobile terminal can reduce the noise volume sent to opposite-end equipment, the noise interference caused to an opposite-end equipment user is relieved, and high-quality voice call service is provided for the user.
The foregoing description is only an overview of the technical solutions of the present invention, and the embodiments of the present invention are described below in order to make the technical means of the present invention more clearly understood and to make the above and other objects, features, and advantages of the present invention more clearly understandable.
Drawings
Various advantages and benefits will become apparent to those of ordinary skill in the art upon reading the following detailed description of the preferred embodiments. The drawings are only for purposes of illustrating the preferred embodiments and are not to be construed as limiting the invention. Also, like reference numerals are used to refer to like parts throughout the drawings. In the drawings:
fig. 1 is a flowchart illustrating steps of a voice call method according to a first embodiment of the present invention;
fig. 2 is a flowchart illustrating steps of a voice call method according to a second embodiment of the present invention;
fig. 3 is a block diagram of a mobile terminal according to a third embodiment of the present invention;
fig. 4 is a block diagram of a mobile terminal according to a fourth embodiment of the present invention;
fig. 5 is a block diagram of a mobile terminal according to a fifth embodiment of the present invention;
fig. 6 is a block diagram of a mobile terminal according to a sixth embodiment of the present invention.
Detailed Description
Exemplary embodiments of the present disclosure will be described in more detail below with reference to the accompanying drawings. While exemplary embodiments of the present disclosure are shown in the drawings, it should be understood that the present disclosure may be embodied in various forms and should not be limited to the embodiments set forth herein. Rather, these embodiments are provided so that this disclosure will be thorough and complete, and will fully convey the scope of the disclosure to those skilled in the art.
Example one
Referring to fig. 1, a flowchart illustrating steps of a voice call method according to a first embodiment of the present invention is shown.
The voice call method of the embodiment of the invention comprises the following steps:
step 101: and after a microphone in the mobile terminal is started, judging the current call state of the mobile terminal.
The call state comprises a voice answering state and a voice sending state.
The voice answering state means that the user of the opposite terminal equipment inputs voice, and the user of the equipment answers the voice of the opposite terminal. The voice sending state means that the user of the device is inputting voice, and the user of the opposite terminal device is answering the voice of the user of the device.
Step 102: if the current call state is a voice answering state, the uplink amplification gain value of the voice signal is adjusted to a first preset gain value.
The first preset gain value and the second preset gain value may be set by a user in advance according to actual requirements, or may be default values of the system. Wherein the first preset gain value is smaller than the second preset gain value.
Because the mobile terminal is in a voice listening state, there is no need to send a voice signal to the peer device, so in order to reduce interference of noise carried in the surrounding environment or the circuit board of the mobile terminal to the peer device user, in the embodiment of the present invention, the uplink amplification gain value of the voice signal is adjusted to a smaller gain value, that is, a first preset gain value. Thus, even if the sound signal corresponding to the noise is transmitted to the opposite terminal device, the interference caused to the opposite terminal device user is not obvious. The sound signal comprises a voice signal and a sound signal corresponding to noise.
Step 103: and if the current call state is the voice sending state, adjusting the uplink amplification gain value of the voice signal to a second preset gain value.
Because the mobile terminal is in a voice sending state, there is a need to send a voice signal to the opposite terminal device, and it is necessary to provide clear voice to the user of the opposite terminal device as much as possible, in the embodiment of the present invention, the uplink amplification gain value of the voice signal is adjusted to a larger gain value, that is, a second preset gain value. Therefore, when the sound signal is sent to the opposite terminal equipment, the opposite terminal equipment user can hear clear voice.
By the voice call method provided by the embodiment of the invention, the current call state of the mobile terminal is determined to be a voice receiving state or a voice sending state in real time in the call process of the mobile terminal, and the uplink amplification gain value of the voice signal is adaptively adjusted in real time according to the call state. Specifically, when the mobile terminal is in a voice emitting state, the uplink amplification gain value of the voice signal is adjusted to a second preset gain value, so that the user of the opposite terminal device can clearly hear the voice emitted by the mobile terminal. When the mobile terminal is in a voice answering state, the uplink amplification gain value of the voice signal is adjusted to be the first preset gain value, and the uplink amplification gain value of the voice signal is reduced, so that even if the mobile terminal is currently in a noise environment, the mobile terminal can reduce the noise volume sent to opposite-end equipment, the noise interference caused to an opposite-end equipment user is relieved, and high-quality voice call service is provided for the user.
Example two
Referring to fig. 2, a flowchart illustrating steps of a voice call method according to a second embodiment of the present invention is shown.
The voice communication method provided by the embodiment of the invention is suitable for the mobile terminal comprising two or two upper microphones, and specifically comprises the following steps:
step 201: when the setting operation of the user on the first preset gain value and the second preset gain value is monitored, the first preset gain value and the second preset gain value are determined according to the setting operation.
In a specific implementation process, a setting interface for presetting a gain value can be provided for a user in the mobile terminal, and the user only needs to set the preset gain value in the setting interface. The first preset gain value is an uplink amplification gain value of the sound signal when the mobile terminal is in a voice answering state in the conversation process; the second preset gain value is an uplink amplification gain value of the sound signal when the mobile terminal is in a voice sending state in the call process. The first predetermined gain value is less than the second predetermined gain value.
In the specific implementation process, the user can set the first preset gain value and the second preset gain value according to actual requirements, and it is only required to ensure that the first preset gain value is smaller than the second preset gain value.
It should be noted that, if the first preset gain value and the second preset gain value are preset in the mobile terminal, step 201 does not need to be executed again.
Step 202: after a microphone in the mobile terminal is started, determining a first signal intensity corresponding to a sound signal received by a main microphone.
According to the distance between the sound source and each microphone of the mobile terminal, the sound field model can be divided into two types: the system comprises a near-field model and a far-field model, wherein the near-field model regards sound waves as spherical waves and considers the signal intensity difference among sound signals received by different microphones; the far-field model treats sound waves as plane waves, which ignore the signal strength differences between the sound signals received by the microphones. In the embodiment of the present invention, the current call state of the mobile terminal is determined based on the near field model, and the specific determination manner is as shown in step 202 to step 205.
The mobile terminal in the embodiment of the invention mainly comprises a near-field voice detection module consisting of a main microphone, auxiliary microphones and a voice signal processing circuit, wherein the near-field voice detection module is used for receiving voice signals received by the main microphone and voice signals received by each auxiliary microphone and judging whether a user is in a speaking state (which is equivalent to that the mobile terminal is in a voice sending state) or an answering state (which is equivalent to that the mobile terminal is in a voice answering state) according to a signal intensity value between the two types of signals.
The specific detection principle is as follows: the mobile terminal obtains external sound signals through the main microphone and the auxiliary microphone, the main microphone is installed at the bottom end of the mobile terminal and close to the position of the mouth of a person, the signal intensity of the received sound signals is high, the auxiliary microphone is installed at the top end of the mobile terminal and far away from the position of the mouth of the person, the intensity of the received sound signals is low, the signal intensity of the sound signals received by the two microphones is close to the environmental noise in an external far-field model, and therefore the fact that a user of the mobile terminal is in a speaking state or an answering state can be judged by comparing the signal intensity difference of the sound signals received by the main microphone and the auxiliary microphone.
Step 203: and determining a second signal strength corresponding to the sound signal received by the secondary microphone in the mobile terminal.
When the mobile terminal includes a sub microphone, the signal strength of the sound signal received by the sub microphone is determined as the second signal strength.
When the mobile terminal comprises at least two auxiliary microphones, respectively determining the signal intensity of the sound signal received by each auxiliary microphone, calculating the average value of the signal intensity, and determining the calculated average value as the second signal intensity corresponding to the sound signal received by the auxiliary microphone.
Step 204: a difference between the first signal strength and the second signal strength is calculated.
For example: and the first signal intensity is A, the second signal intensity is B, and the difference value of A and B is calculated to be C.
Step 205: judging whether the difference value is larger than a preset threshold value or not; if not, go to step 206, and if yes, go to step 207.
It should be noted that, in the specific implementation process, the preset threshold may be set by a person skilled in the art according to actual needs, and this is not specifically limited in the embodiment of the present invention. If the difference is larger than the preset threshold, determining that the mobile terminal user is in a speaking state, namely the mobile terminal is in a voice sending state; otherwise, the mobile terminal user is determined to be in the answering state, namely the mobile terminal is in the voice answering state.
Step 206: and if the difference value is smaller than or equal to the preset threshold value, determining that the current call state of the mobile terminal is a voice answering state, and adjusting the uplink amplification gain value of the voice signal to a first preset gain value.
Because the mobile terminal is in a voice listening state, there is no need to send a voice signal to the peer device, so in order to reduce interference of noise carried in the surrounding environment or the circuit board of the mobile terminal to the peer device user, in the embodiment of the present invention, the uplink amplification gain value of the voice signal is adjusted to a smaller gain value, that is, a first preset gain value. Therefore, even if the sound signal corresponding to the noise is sent to the opposite terminal equipment, the interference caused by the opposite terminal equipment user is not obvious, and better conversation experience is brought to both parties of conversation.
Step 207: if the difference value is larger than the preset threshold value, the current call state of the mobile terminal is determined to be a voice sending state, and the uplink amplification gain value of the voice signal is adjusted to a second preset gain value.
Because the mobile terminal is in a voice sending state, there is a need to send a voice signal to the opposite terminal device, and it is necessary to provide clear voice to the user of the opposite terminal device as much as possible, in the embodiment of the present invention, the uplink amplification gain value of the voice signal is adjusted to a larger gain value, that is, a second preset gain value. Therefore, when the sound signal is sent to the opposite terminal equipment, the opposite terminal equipment user can hear clear voice.
By the voice call method provided by the embodiment of the invention, the current call state of the mobile terminal is determined to be a voice receiving state or a voice sending state in real time in the call process of the mobile terminal, and the uplink amplification gain value of the voice signal is adaptively adjusted in real time according to the call state. Specifically, when the mobile terminal is in a voice emitting state, the uplink amplification gain value of the voice signal is adjusted to a second preset gain value, so that the user of the opposite terminal device can clearly hear the voice emitted by the mobile terminal. When the mobile terminal is in a voice answering state, the uplink amplification gain value of the voice signal is adjusted to be the first preset gain value, and the uplink amplification gain value of the voice signal is reduced, so that even if the mobile terminal is currently in a noise environment, the mobile terminal can reduce the noise volume sent to opposite-end equipment, the noise interference caused to an opposite-end equipment user is relieved, and high-quality voice call service is provided for the user.
EXAMPLE III
Referring to fig. 3, a block diagram of a mobile terminal according to a third embodiment of the present invention is shown.
The mobile terminal of the embodiment of the invention comprises: a call state judgment module 301, configured to judge a current call state of the mobile terminal after a microphone in the mobile terminal is turned on; a first adjusting module 302, configured to adjust an uplink amplification gain value of the voice signal to a first preset gain value if the current call state is a voice answering state; the second adjusting module 303 is configured to adjust an uplink amplification gain value of the sound signal to a second preset gain value if the current call state is a voice sending state, where the first preset gain value is smaller than the second preset gain value.
The mobile terminal provided by the embodiment of the invention judges whether the current call state of the mobile terminal is a voice answering state or a voice sending state in real time in the call process, and adaptively adjusts the uplink amplification gain value of the voice signal in real time according to the call state. Specifically, when the mobile terminal is in a voice emitting state, the uplink amplification gain value of the voice signal is adjusted to a second preset gain value, so that the user of the opposite terminal device can clearly hear the voice emitted by the mobile terminal. When the mobile terminal is in a voice answering state, the uplink amplification gain value of the voice signal is adjusted to be the first preset gain value, and the uplink amplification gain value of the voice signal is reduced, so that even if the mobile terminal is currently in a noise environment, the mobile terminal can reduce the noise volume sent to opposite-end equipment, the noise interference caused to an opposite-end equipment user is relieved, and high-quality voice call service is provided for the user.
Example four
Referring to fig. 4, a block diagram of a mobile terminal according to a fourth embodiment of the present invention is shown.
The mobile terminal of the embodiment of the present invention is further optimized for the mobile terminal of the third embodiment, and the optimized mobile terminal includes: a call state judgment module 401, configured to judge a current call state of the mobile terminal after a microphone in the mobile terminal is turned on; a first adjusting module 402, configured to adjust an uplink amplification gain value of the voice signal to a first preset gain value if the current call state is a voice answering state; the second adjusting module 403 is configured to adjust an uplink amplification gain value of the sound signal to a second preset gain value if the current call state is a voice sending state, where the first preset gain value is smaller than the second preset gain value.
Preferably, the call state determining module 401 includes: a first signal strength determining submodule 4011, configured to determine a first signal strength corresponding to a sound signal received by a primary microphone in the mobile terminal; the second signal strength determining submodule 4012 is configured to determine a second signal strength corresponding to a sound signal received by a secondary microphone in the mobile terminal; a difference calculation submodule 4013, configured to calculate a difference between the first signal strength and the second signal strength; the judgment sub-module 4014 is configured to judge whether the difference is greater than a preset threshold; the state determining submodule 4015 is configured to determine that the current call state of the mobile terminal is a voice sending state if the determination result of the determining submodule is yes; and if the judgment result of the judgment submodule is negative, determining that the current call state of the mobile terminal is a voice answering state.
Preferably, the second signal strength determining sub-module 4012 is specifically configured to: and when the mobile terminal comprises a secondary microphone, determining the signal intensity of the sound signal received by the secondary microphone as a second signal intensity value.
Preferably, the second signal strength determining sub-module 4012 includes: and the determining unit is used for respectively determining the signal intensity of the sound signal received by each secondary microphone when the mobile terminal comprises at least two secondary microphones, calculating the average value of the signal intensity, and determining the average value as a second signal intensity value corresponding to the sound signal received by the secondary microphone.
Preferably, the mobile terminal of the embodiment of the present invention further includes: the monitoring module 404 is configured to determine a first preset gain value and a second preset gain value according to a setting operation when the setting operation of a user on the first preset gain value and the second preset gain value is monitored.
The mobile terminal of the embodiment of the present invention is used to implement the voice call methods in the first and second embodiments, and has the corresponding beneficial effects as in the method embodiments, which are not described herein again.
EXAMPLE five
Referring to fig. 5, a block diagram of a mobile terminal according to an embodiment of the present invention is shown.
The mobile terminal 700 of the embodiment of the present invention includes: at least one processor 701, memory 702, at least one network interface 704, and other user interfaces 703. The various components in the mobile terminal 700 are coupled together by a bus system 705. It is understood that the bus system 705 is used to enable communications among the components. The bus system 705 includes a power bus, a control bus, and a status signal bus in addition to a data bus. But for clarity of illustration the various busses are labeled in figure 5 as the bus system 705.
The user interface 703 may include, among other things, a display, a keyboard, or a pointing device (e.g., a mouse, track ball, touch pad, or touch screen, etc.).
It is to be understood that the memory 702 in embodiments of the present invention may be either volatile memory or nonvolatile memory, or may include both volatile and nonvolatile memory. The non-volatile Memory may be a Read-Only Memory (ROM), a Programmable ROM (PROM), an Erasable PROM (EPROM), an Electrically Erasable PROM (EEPROM), or a flash Memory. Volatile Memory can be Random Access Memory (RAM), which acts as external cache Memory. By way of example, but not limitation, many forms of RAM are available, such as Static random access memory (Static RAM, SRAM), Dynamic Random Access Memory (DRAM), Synchronous Dynamic random access memory (Synchronous DRAM, SDRAM), Double Data Rate Synchronous Dynamic Random Access Memory (DDRSDRAM), Enhanced Synchronous SDRAM (ESDRAM), Synchronous link SDRAM (SLDRAM), and direct memory bus DRAM (DRRAM). The memory 702 of the systems and methods described in this embodiment of the invention is intended to comprise, without being limited to, these and any other suitable types of memory.
In some embodiments, memory 702 stores the following elements, executable modules or data structures, or a subset thereof, or an expanded set thereof: an operating system 7021 and application programs 7022.
The operating system 7021 includes various system programs, such as a framework layer, a core library layer, a driver layer, and the like, for implementing various basic services and processing hardware-based tasks. The application 7022 includes various applications, such as a Media Player (Media Player), a Browser (Browser), and the like, for implementing various application services. Programs that implement methods in accordance with embodiments of the present invention can be included within application program 7022.
In the embodiment of the present invention, by calling a program or an instruction stored in the memory 702, specifically, a program or an instruction stored in the application program 7022, the processor 701 is configured to determine a current call state of the mobile terminal after a microphone in the mobile terminal is turned on; if the current call state is a voice answering state, adjusting an uplink amplification gain value of the voice signal to a first preset gain value; if the current call state is a voice sending state, adjusting an uplink amplification gain value of the voice signal to a second preset gain value, wherein the first preset gain value is smaller than the second preset gain value.
The method disclosed in the above embodiments of the present invention may be applied to the processor 701, or implemented by the processor 701. The processor 701 may be an integrated circuit chip having signal processing capabilities. In implementation, the steps of the above method may be implemented by integrated logic circuits of hardware or instructions in the form of software in the processor 701. The Processor 701 may be a general-purpose Processor, a Digital Signal Processor (DSP), an Application Specific Integrated Circuit (ASIC), an off-the-shelf programmable Gate Array (FPGA) or other programmable logic device, discrete Gate or transistor logic device, or discrete hardware components. The various methods, steps and logic blocks disclosed in the embodiments of the present invention may be implemented or performed. A general purpose processor may be a microprocessor or the processor may be any conventional processor or the like. The steps of the method disclosed in connection with the embodiments of the present invention may be directly implemented by a hardware decoding processor, or implemented by a combination of hardware and software modules in the decoding processor. The software module may be located in ram, flash memory, rom, prom, or eprom, registers, etc. storage media as is well known in the art. The storage medium is located in the memory 702, and the processor 701 reads the information in the memory 702 and performs the steps of the above method in combination with the hardware thereof.
It is to be understood that the embodiments described herein may be implemented in hardware, software, firmware, middleware, microcode, or any combination thereof. For a hardware implementation, the Processing units may be implemented within one or more Application Specific Integrated Circuits (ASICs), Digital Signal Processors (DSPs), Digital Signal Processing Devices (DSPDs), Programmable Logic Devices (PLDs), Field Programmable Gate Arrays (FPGAs), general purpose processors, controllers, micro-controllers, microprocessors, other electronic units configured to perform the functions described herein, or a combination thereof.
For a software implementation, the techniques described in this embodiment of the invention may be implemented with modules (e.g., procedures, functions, and so on) that perform the functions described in this embodiment of the invention. The software codes may be stored in a memory and executed by a processor. The memory may be implemented within the processor or external to the processor.
Optionally, when the processor 701 determines the current call state of the mobile terminal, the processor is specifically configured to: determining a first signal intensity corresponding to a sound signal received by a main microphone in the mobile terminal; determining a second signal intensity corresponding to a sound signal received by a secondary microphone in the mobile terminal; calculating a difference between the first signal strength and the second signal strength; judging whether the difference value is larger than a preset threshold value or not; if so, determining that the current call state of the mobile terminal is a voice sending state; if not, determining that the current call state of the mobile terminal is a voice answering state.
Optionally, when the mobile terminal includes a secondary microphone, and the processor 701 determines a second signal strength corresponding to a sound signal received by the secondary microphone, the processor is specifically configured to: and determining the signal intensity of the sound signal received by the auxiliary microphone as a second signal intensity.
Optionally, when the mobile terminal includes at least two secondary microphones, the processor 701 is specifically configured to, when determining a second signal strength corresponding to a sound signal received by the secondary microphone: and respectively determining the signal intensity of the sound signal received by each secondary microphone, calculating the average value of the signal intensity, and determining the average value as the second signal intensity corresponding to the sound signal received by the secondary microphone.
Optionally, the processor 701 is further configured to: when the setting operation of a user on the first preset gain value and the second preset gain value is detected, the first preset gain value and the second preset gain value are determined according to the setting operation.
The mobile terminal 700 can implement the processes implemented by the mobile terminal in the foregoing embodiments, and details are not repeated here to avoid repetition.
The mobile terminal provided by the embodiment of the invention judges whether the current call state of the mobile terminal is a voice answering state or a voice sending state in real time in the call process, and adaptively adjusts the uplink amplification gain value of the voice signal in real time according to the call state. Specifically, when the mobile terminal is in a voice emitting state, the uplink amplification gain value of the voice signal is adjusted to a second preset gain value, so that the user of the opposite terminal device can clearly hear the voice emitted by the mobile terminal. When the mobile terminal is in a voice answering state, the uplink amplification gain value of the voice signal is adjusted to be the first preset gain value, and the uplink amplification gain value of the voice signal is reduced, so that even if the mobile terminal is currently in a noise environment, the mobile terminal can reduce the noise volume sent to opposite-end equipment, the noise interference caused to an opposite-end equipment user is relieved, and high-quality voice call service is provided for the user.
EXAMPLE six
Referring to fig. 6, a block diagram of a mobile terminal according to an embodiment of the present invention is shown.
The mobile terminal in the embodiment of the present invention may be a mobile phone, a tablet computer, a Personal Digital Assistant (PDA), or a vehicle-mounted computer.
The mobile terminal in fig. 6 includes a Radio Frequency (RF) circuit 810, a memory 820, an input unit 830, a display unit 840, a processor 860, an audio circuit 870, a wifi (wireless fidelity) module 880, and a power supply 890.
The input unit 830 may be used, among other things, to receive numeric or character information input by a user and to generate signal inputs related to user settings and function control of the mobile terminal. Specifically, in the embodiment of the present invention, the input unit 830 may include a touch panel 831. The touch panel 831, also referred to as a touch screen, can collect touch operations performed by a user on or near the touch panel 831 (e.g., operations performed by the user on the touch panel 831 using a finger, a stylus, or any other suitable object or accessory), and drive the corresponding connection device according to a preset program. Alternatively, the touch panel 831 may include two portions, i.e., a touch detection device and a touch controller. The touch detection device detects the touch direction 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 sensing device, converts the touch information into touch point coordinates, sends the touch point coordinates to the processor 860, and can receive and execute commands sent by the processor 860. In addition, the touch panel 831 may be implemented by various types such as a resistive type, a capacitive type, an infrared ray, and a surface acoustic wave. In addition to the touch panel 831, the input unit 830 may include other input devices 832, and the other input devices 832 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, a mouse, a joystick, and the like.
Among them, the display unit 840 may be used to display information input by a user or information provided to the user and various menu interfaces of the mobile terminal. The display unit 840 may include a display panel 841, and the display panel 841 may be alternatively configured in the form of an LCD or an Organic Light-Emitting Diode (OLED), or the like.
It should be noted that the touch panel 831 can overlay the display panel 841 to form a touch display screen, which, when it detects a touch operation thereon or nearby, is passed to the processor 860 to determine the type of touch event, and then the processor 860 provides a corresponding visual output on the touch display screen according to the type of touch event.
The touch display screen comprises an application program interface display area and a common control display area. The arrangement modes of the application program interface display area and the common control display area are not limited, and can be an arrangement mode which can distinguish two display areas, such as vertical arrangement, left-right arrangement and the like. The application interface display area may be used to display an interface of an application. Each interface may contain at least one interface element such as an icon and/or widget desktop control for an application. The application interface display area may also be an empty interface that does not contain any content. The common control display area is used for displaying controls with high utilization rate, such as application icons like setting buttons, interface numbers, scroll bars, phone book icons and the like.
The processor 860 is a control center of the mobile terminal, connects various parts of the entire mobile phone using various interfaces and lines, and performs various functions of the mobile terminal and processes data by operating or executing software programs and/or modules stored in the first memory 821 and calling data stored in the second memory 822, thereby performing overall monitoring of the mobile terminal. Optionally, processor 860 may include one or more processing units.
In the embodiment of the present invention, the processor 860 is configured to determine a current call state of the mobile terminal after a microphone in the mobile terminal is turned on by calling a software program and/or a module stored in the first memory 821 and/or data stored in the second memory 822; if the current call state is a voice answering state, adjusting an uplink amplification gain value of the voice signal to a first preset gain value; if the current call state is a voice sending state, adjusting an uplink amplification gain value of the voice signal to a second preset gain value, wherein the first preset gain value is smaller than the second preset gain value.
Optionally, when the processor 860 determines the current call state of the mobile terminal, it is specifically configured to: determining a first signal intensity corresponding to a sound signal received by a main microphone in the mobile terminal; determining a second signal intensity corresponding to a sound signal received by a secondary microphone in the mobile terminal; calculating a difference between the first signal strength and the second signal strength; judging whether the difference value is larger than a preset threshold value or not; if so, determining that the current call state of the mobile terminal is a voice sending state; if not, determining that the current call state of the mobile terminal is a voice answering state.
Optionally, when the mobile terminal includes a secondary microphone, the processor 860 is specifically configured to, when determining the second signal strength corresponding to the sound signal received by the secondary microphone: and determining the signal intensity of the sound signal received by the auxiliary microphone as a second signal intensity.
Optionally, when the mobile terminal includes at least two secondary microphones, the processor 860 is specifically configured to, when determining the second signal strength corresponding to the sound signal received by the secondary microphone: and respectively determining the signal intensity of the sound signal received by each secondary microphone, calculating the average value of the signal intensity, and determining the average value as the second signal intensity corresponding to the sound signal received by the secondary microphone.
Optionally, the processor 860 is further configured to: when the setting operation of a user on the first preset gain value and the second preset gain value is detected, the first preset gain value and the second preset gain value are determined according to the setting operation.
The mobile terminal provided by the embodiment of the invention judges whether the current call state of the mobile terminal is a voice answering state or a voice sending state in real time in the call process, and adaptively adjusts the uplink amplification gain value of the voice signal in real time according to the call state. Specifically, when the mobile terminal is in a voice emitting state, the uplink amplification gain value of the voice signal is adjusted to a second preset gain value, so that the user of the opposite terminal device can clearly hear the voice emitted by the mobile terminal. When the mobile terminal is in a voice answering state, the uplink amplification gain value of the voice signal is adjusted to be the first preset gain value, and the uplink amplification gain value of the voice signal is reduced, so that even if the mobile terminal is currently in a noise environment, the mobile terminal can reduce the noise volume sent to opposite-end equipment, the noise interference caused to an opposite-end equipment user is relieved, and high-quality voice call service is provided for the user.
For the device embodiment, since it is basically similar to the method embodiment, the description is simple, and for the relevant points, refer to the partial description of the method embodiment.
The voice call scheme provided herein is not inherently related to any particular computer, virtual system, or other apparatus. Various general purpose systems may also be used with the teachings herein. The structure required to construct a system incorporating aspects of the present invention will be apparent from the description above. Moreover, the present invention is not directed to any particular programming language. It is appreciated that a variety of programming languages may be used to implement the teachings of the present invention as described herein, and any descriptions of specific languages are provided above to disclose the best mode of the invention.
In the description provided herein, numerous specific details are set forth. It is understood, however, that embodiments of the invention may be practiced without these specific details. In some instances, well-known methods, structures and techniques have not been shown in detail in order not to obscure an understanding of this description.
Similarly, it should be appreciated that in the foregoing description of exemplary embodiments of the invention, various features of the invention are sometimes grouped together in a single embodiment, figure, or description thereof for the purpose of streamlining the disclosure and aiding in the understanding of one or more of the various inventive aspects. However, the disclosed method should not be interpreted as reflecting an intention that: that the invention as claimed requires more features than are expressly recited in each claim. Rather, as the following claims reflect, inventive aspects lie in less than all features of a single foregoing disclosed embodiment. Thus, the claims following the detailed description are hereby expressly incorporated into this detailed description, with each claim standing on its own as a separate embodiment of this invention.
Those skilled in the art will appreciate that the modules in the device in an embodiment may be adaptively changed and disposed in one or more devices different from the embodiment. The modules or units or components of the embodiments may be combined into one module or unit or component, and furthermore they may be divided into a plurality of sub-modules or sub-units or sub-components. All of the features disclosed in this specification (including any accompanying claims, abstract and drawings), and all of the processes or elements of any method or apparatus so disclosed, may be combined in any combination, except combinations where at least some of such features and/or processes or elements are mutually exclusive. Each feature disclosed in this specification (including any accompanying claims, abstract and drawings) may be replaced by alternative features serving the same, equivalent or similar purpose, unless expressly stated otherwise.
Furthermore, those skilled in the art will appreciate that while some embodiments described herein include some features included in other embodiments, rather than other features, combinations of features of different embodiments are meant to be within the scope of the invention and form different embodiments. For example, in the claims, any of the claimed embodiments may be used in any combination.
The various component embodiments of the invention may be implemented in hardware, or in software modules running on one or more processors, or in a combination thereof. Those skilled in the art will appreciate that a microprocessor or Digital Signal Processor (DSP) may be used in practice to implement some or all of the functions of some or all of the components of a voice call scheme according to embodiments of the present invention. The present invention may also be embodied as apparatus or device programs (e.g., computer programs and computer program products) for performing a portion or all of the methods described herein. Such programs implementing the present invention may be stored on computer-readable media or may be in the form of one or more signals. Such a signal may be downloaded from an internet website or provided on a carrier signal or in any other form.
It should be noted that the above-mentioned embodiments illustrate rather than limit the invention, and that those skilled in the art will be able to design alternative embodiments without departing from the scope of the appended claims. In the claims, any reference signs placed between parentheses shall not be construed as limiting the claim. The word "comprising" does not exclude the presence of elements or steps not listed in a claim. The word "a" or "an" preceding an element does not exclude the presence of a plurality of such elements. The invention may be implemented by means of hardware comprising several distinct elements, and by means of a suitably programmed computer. In the unit claims enumerating several means, several of these means may be embodied by one and the same item of hardware. The usage of the words first, second and third, etcetera do not indicate any ordering. These words may be interpreted as names.
Claims (8)
1. A voice call method, comprising:
after a microphone in the mobile terminal is started, judging the current call state of the mobile terminal;
if the current call state is a voice answering state, adjusting an uplink amplification gain value of the voice signal to a first preset gain value;
if the current call state is a voice sending state, adjusting an uplink amplification gain value of a voice signal to a second preset gain value, wherein the first preset gain value is smaller than the second preset gain value;
the first preset gain value and the second preset gain value are preset by a user according to actual requirements or are default values of a system; the sound signal comprises a voice signal and a sound signal corresponding to noise;
wherein the method further comprises:
when the setting operation of the first preset gain value and the second preset gain value by a user is detected, the first preset gain value and the second preset gain value are determined according to the setting operation.
2. The method according to claim 1, wherein the step of determining the current call state of the mobile terminal comprises:
determining a first signal intensity corresponding to a sound signal received by a main microphone in the mobile terminal;
determining a second signal intensity corresponding to a sound signal received by a secondary microphone in the mobile terminal;
calculating a difference between the first signal strength and the second signal strength;
judging whether the difference value is larger than a preset threshold value or not;
if so, determining that the current call state of the mobile terminal is a voice sending state;
if not, determining that the current call state of the mobile terminal is a voice answering state.
3. The method of claim 2, wherein when the mobile terminal includes a secondary microphone, the step of determining the second signal strength corresponding to the sound signal received by the secondary microphone comprises:
and determining the signal intensity of the sound signal received by the auxiliary microphone as a second signal intensity.
4. The method according to claim 2, wherein when the mobile terminal includes at least two secondary microphones, the step of determining the second signal strength corresponding to the sound signal received by the secondary microphones comprises:
and respectively determining the signal intensity of the sound signal received by each secondary microphone, calculating the average value of the signal intensity, and determining the average value as the second signal intensity corresponding to the sound signal received by the secondary microphone.
5. A mobile terminal, characterized in that the mobile terminal comprises:
the mobile terminal comprises a call state judgment module, a call state judgment module and a call state judgment module, wherein the call state judgment module is used for judging the current call state of the mobile terminal after a microphone in the mobile terminal is started;
the first adjusting module is used for adjusting the uplink amplification gain value of the sound signal to a first preset gain value if the current call state is a voice answering state;
the second adjusting module is used for adjusting the uplink amplification gain value of the sound signal to a second preset gain value if the current call state is a voice sending state, wherein the first preset gain value is smaller than the second preset gain value;
the first preset gain value and the second preset gain value are preset by a user according to actual requirements or are default values of a system; the sound signal comprises a voice signal and a sound signal corresponding to noise;
wherein the mobile terminal further comprises:
and the monitoring module is used for determining the first preset gain value and the second preset gain value according to the setting operation when the setting operation of the first preset gain value and the second preset gain value by a user is monitored.
6. The mobile terminal according to claim 5, wherein the call state determining module comprises:
the first signal strength determining submodule is used for determining first signal strength corresponding to a sound signal received by a main microphone in the mobile terminal;
the second signal strength determining submodule is used for determining second signal strength corresponding to the sound signal received by the secondary microphone in the mobile terminal;
a difference calculation submodule for calculating a difference between the first signal strength and the second signal strength;
the judgment submodule is used for judging whether the difference value is larger than a preset threshold value or not;
the state determining submodule is used for determining that the current call state of the mobile terminal is a voice sending state if the judgment result of the judging submodule is yes; and if the judgment result of the judgment submodule is negative, determining that the current call state of the mobile terminal is a voice answering state.
7. The mobile terminal of claim 6, wherein the second signal strength determination submodule is specifically configured to:
and when the mobile terminal comprises a secondary microphone, determining the signal intensity of the sound signal received by the secondary microphone as a second signal intensity value.
8. The mobile terminal of claim 6, wherein the second signal strength determination submodule comprises:
and the determining unit is used for respectively determining the signal intensity of the sound signal received by each secondary microphone when the mobile terminal comprises at least two secondary microphones, calculating the average value of the signal intensity, and determining the average value as a second signal intensity value corresponding to the sound signal received by the secondary microphone.
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| Publication number | Priority date | Publication date | Assignee | Title |
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| CN108564961A (en) * | 2017-11-29 | 2018-09-21 | 华北计算技术研究所（中国电子科技集团公司第十五研究所） | A kind of voice de-noising method of mobile communication equipment |
| CN110428806B (en) * | 2019-06-03 | 2023-02-24 | 交互未来(北京)科技有限公司 | Microphone signal based voice interaction wake-up electronic device, method, and medium |
| CN110971769B (en) * | 2019-11-19 | 2022-05-03 | 百度在线网络技术（北京）有限公司 | Call signal processing method and device, electronic equipment and storage medium |
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