CN112911056A - Audio recording calibration method and device and computer readable storage medium - Google Patents
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Abstract
The invention discloses an audio recording calibration method, equipment and a computer readable storage medium, wherein the method comprises the following steps: acquiring audio data in a preset calibration period according to a preset gain value, and judging whether the gain amplitude of the audio data is smaller than a preset amplitude; if the gain amplitude of the audio data is smaller than the preset amplitude, performing amplification processing on the audio data; then, counting the times of executing the amplification processing operation in the calibration period; and if the times exceed a preset value, performing amplification processing on the preset gain value, and taking the gain value after the amplification processing operation as a fixed gain value for subsequently acquiring the audio data. The humanized audio recording calibration scheme is realized, the operation convenience of audio calibration is improved, a series of problems caused by echo cancellation are avoided, and the user experience is enhanced.
Description
Technical Field
The present invention relates to the field of mobile communications, and in particular, to a method and an apparatus for calibrating audio recording, and a computer-readable storage medium.
Background
In the prior art, with the continuous development of intelligent terminal devices, the use frequency of devices by users is higher and higher, and at present, the gain of recorded sound caused by the difference between the sensitivities of recording microphones of devices such as mobile phones or the loss in the use process of the microphones is reduced. The problem of the recording is not clearly heard or the opposite end of the call can not hear the speaking voice of the opposite end of the call, and the opposite end of the call is mistakenly considered to be silent in the call or failed in the recording, and the like. Meanwhile, considering the imperfection of echo cancellation technology of most mobile phones and other devices, when the gain of the sound recorded by the microphone is smaller than the gain of the echo reference signal, the situation of over-cancellation or incomplete cancellation may occur, so that the sound after noise cancellation processing appears in the situations of interruption, word dropping, and even silence, and user experience is affected.
Disclosure of Invention
In order to solve the technical defects in the prior art, the invention provides an audio recording calibration method, which comprises the following steps:
acquiring audio data in a preset calibration period according to a preset gain value, and judging whether the gain amplitude of the audio data is smaller than a preset amplitude;
if the gain amplitude of the audio data is smaller than the preset amplitude, performing amplification processing on the audio data;
counting the times of executing the amplification processing operation in the calibration period;
and if the times exceed a preset value, performing amplification processing on the preset gain value, and taking the gain value after the amplification processing operation as a fixed gain value for subsequently acquiring the audio data.
Optionally, the acquiring audio data in a preset calibration period according to a preset gain value, and determining whether a gain amplitude of the audio data is smaller than a preset amplitude includes:
determining a preset call duration as a first calibration period in a normal call scene;
and determining a preset test duration as a second calibration period under the scene of calibration setting.
Optionally, the obtaining audio data in a preset calibration period according to a preset gain value, and determining whether a gain amplitude of the audio data is smaller than a preset amplitude further includes:
taking the fixed gain value after the previous calibration as the preset gain value;
and acquiring first audio data of each section recorded under the scene of the normal call in the first calibration period, or acquiring at least one section of second audio data recorded under the scene of the calibration setting in the second calibration period.
Optionally, the obtaining audio data in a preset calibration period according to a preset gain value, and determining whether a gain amplitude of the audio data is smaller than a preset amplitude further includes:
setting a first preset amplitude value under the scene of the normal call, and setting a second preset amplitude value under the scene of the calibration setting;
and acquiring the gain amplitude of the first audio data recorded in each section, and successively comparing the gain amplitude with the first preset amplitude, or acquiring the gain amplitude of the second audio data recorded in at least one section, and comparing the gain amplitude with the second preset amplitude.
Optionally, if the gain amplitude of the audio data is smaller than the preset amplitude, performing amplification processing on the audio data, including:
determining a preset proportion value as a first amplification proportion in a normal conversation scene, and determining a preset proportion value as a second amplification proportion in a calibration setting scene;
and if the gain amplitude of the first audio data of any section is determined to be smaller than the first preset amplitude, performing amplification processing according to a first amplification proportion, or if the gain amplitude of the second audio data of at least one section is determined to be smaller than the second preset amplitude, performing amplification processing according to a second amplification proportion.
Optionally, the counting the number of times of performing the amplification processing operation in the calibration period includes:
accumulating the operations of executing the amplification processing in one first calibration period, and counting the times obtained by accumulation;
or, in a plurality of second calibration periods, accumulating the operations for executing the amplification processing, and counting the times obtained by accumulation.
Optionally, if the number of times exceeds a preset value, performing amplification processing on the preset gain value, and taking the gain value after performing the amplification processing operation as a fixed gain value for subsequently acquiring the audio data, including:
in one first calibration period, if the times exceed a preset value, performing amplification processing on the preset gain value, and taking the gain value after the amplification processing operation as a first fixed gain value of the audio data in the remaining time of the first calibration period;
in a plurality of second calibration periods, if the number of times exceeds a preset value, performing amplification processing on the preset gain value, and taking a gain value after the amplification processing operation as a second fixed gain value of the audio data of a next second calibration period.
Optionally, if the number of times exceeds a preset value, performing amplification processing on the preset gain value, and taking the gain value after performing the amplification processing operation as a fixed gain value for subsequently acquiring the audio data, further comprising:
obtaining the first fixed gain value and the second fixed gain value according to the first calibration period and the second calibration period;
and superposing the first fixed gain value and the second fixed gain value to be used as the calibrated fixed gain value.
The invention also provides an audio recording calibration device, which comprises a memory, a processor and a computer program stored on the memory and capable of running on the processor, wherein the computer program realizes the steps of the audio recording calibration method according to any one of the above items when being executed by the processor.
The invention further provides a computer readable storage medium, which stores an audio recording calibration program, and the audio recording calibration program realizes the steps of the audio recording calibration method when being executed by a processor.
By implementing the audio recording calibration method, the audio recording calibration device and the computer readable storage medium, audio data are obtained in a preset calibration period according to a preset gain value, and whether the gain amplitude of the audio data is smaller than a preset amplitude value or not is judged; if the gain amplitude of the audio data is smaller than the preset amplitude, performing amplification processing on the audio data; then, counting the times of executing the amplification processing operation in the calibration period; and if the times exceed a preset value, performing amplification processing on the preset gain value, and taking the gain value after the amplification processing operation as a fixed gain value for subsequently acquiring the audio data. The humanized audio recording calibration scheme is realized, the operation convenience of audio calibration is improved, a series of problems caused by echo cancellation are avoided, and the user experience is enhanced.
Drawings
The invention will be further described with reference to the accompanying drawings and examples, in which:
fig. 1 is a schematic diagram of a hardware structure of a mobile terminal according to the present invention;
fig. 2 is a communication network system architecture diagram provided by an embodiment of the present invention;
FIG. 3 is a flowchart of a first embodiment of an audio recording calibration method according to the present invention;
FIG. 4 is a flowchart of a second embodiment of the calibration method for audio recording according to the present invention;
FIG. 5 is a flowchart illustrating a third exemplary embodiment of an audio recording calibration method according to the present invention;
FIG. 6 is a flowchart illustrating a fourth exemplary embodiment of an audio recording calibration method according to the present invention;
fig. 7 is a flowchart of a fifth embodiment of the calibration method for audio recording according to the present invention;
FIG. 8 is a flowchart of a sixth embodiment of an audio recording calibration method of the present invention;
fig. 9 is a flowchart of a seventh embodiment of the calibration method for audio recording according to the present invention;
fig. 10 is a flowchart of an eighth embodiment of the calibration method for audio recording according to the present invention;
FIG. 11 is a flowchart illustrating dynamic calibration of an eighth exemplary embodiment of an audio recording calibration method according to the present invention;
fig. 12 is a flowchart illustrating static calibration of an audio recording calibration method according to an eighth embodiment of the invention.
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.
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 itself. Thus, "module", "component" or "unit" may be used mixedly.
The 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 tablet computer, a notebook computer, a palmtop computer, a Personal Digital Assistant (PDA), a Portable Media Player (PMP), a navigation device, a wearable device, a smart band, a pedometer, and the like, and a fixed terminal such as a Digital TV, a desktop computer, and the like.
The following description will be given by way of example of a mobile terminal, and it will be understood by those skilled in the art that the construction according to the embodiment of the present invention can be applied to a fixed type terminal, in addition to elements particularly used for mobile purposes.
Referring to fig. 1, which is a schematic diagram of a hardware structure of a mobile terminal for implementing various embodiments of the present invention, the mobile terminal 100 may include: RF (Radio Frequency) unit 101, WiFi module 102, audio output unit 103, a/V (audio/video) input unit 104, sensor 105, display unit 106, user input unit 107, interface unit 108, memory 109, processor 110, and power supply 111. Those skilled in the art will appreciate that the mobile terminal architecture shown in fig. 1 is not intended to be limiting of mobile terminals, which may include more or fewer components than those shown, or some components may be combined, or a different arrangement of components.
The following describes each component of the mobile terminal in detail with reference to fig. 1:
the radio frequency unit 101 may be configured to receive and transmit signals during information transmission and reception or during a call, and specifically, receive downlink information of a base station and then process the downlink information to the processor 110; in addition, the uplink data is transmitted to the base station. Typically, radio frequency unit 101 includes, but is not limited to, an antenna, at least one amplifier, a transceiver, a coupler, a low noise amplifier, a duplexer, and the like. In addition, the radio frequency unit 101 can also communicate with a network and other devices through wireless communication. The wireless communication may use any communication standard or protocol, including but not limited to GSM (Global System for Mobile communications), GPRS (General Packet Radio Service), CDMA2000(Code Division Multiple Access 2000), WCDMA (Wideband Code Division Multiple Access), TD-SCDMA (Time Division-Synchronous Code Division Multiple Access), FDD-LTE (Frequency Division duplex Long Term Evolution), and TDD-LTE (Time Division duplex Long Term Evolution).
WiFi belongs to short-distance wireless transmission technology, and the mobile terminal can help a user to receive and send e-mails, browse webpages, access streaming media and the like through the WiFi module 102, and provides wireless broadband internet access for the user. Although fig. 1 shows the WiFi module 102, it is understood that it does not belong to the essential constitution of the mobile terminal, and may be omitted entirely as needed within the scope not changing the essence of the invention.
The audio output unit 103 may convert audio data received by the radio frequency unit 101 or the WiFi module 102 or stored in the memory 109 into an audio signal and output as sound when the mobile terminal 100 is in a call signal reception mode, a call mode, a recording mode, a voice recognition mode, a broadcast reception mode, or the like. Also, the audio output unit 103 may also provide audio output 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 unit 103 may include a speaker, a buzzer, and the like.
The a/V input unit 104 is used to receive audio or video signals. The a/V input Unit 104 may include a Graphics Processing Unit (GPU) 1041 and a microphone 1042, the Graphics processor 1041 Processing image data of still pictures or video obtained by an image capturing device (e.g., a camera) in a video capturing mode or an image capturing mode. The processed image frames may be displayed on the display unit 106. The image frames processed by the graphic processor 1041 may be stored in the memory 109 (or other storage medium) or transmitted via the radio frequency unit 101 or the WiFi module 102. The microphone 1042 may receive sounds (audio data) via the microphone 1042 in a phone call mode, a recording mode, a voice recognition mode, or the like, and may be 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 radio frequency unit 101 in case of a phone call mode. The microphone 1042 may implement various types of noise cancellation (or suppression) algorithms to cancel (or suppress) noise or interference generated in the course of receiving and transmitting audio signals.
The mobile terminal 100 also includes at least one sensor 105, such as a light sensor, a motion sensor, and other sensors. Specifically, the light sensor includes an ambient light sensor that can adjust the brightness of the display panel 1061 according to the brightness of ambient light, and a proximity sensor that can turn off the display panel 1061 and/or a backlight when the mobile terminal 100 is moved to the ear. As one of the motion sensors, the accelerometer sensor can detect the magnitude of acceleration in each direction (generally, three axes), can detect the magnitude and direction of gravity when stationary, and can be used for applications of recognizing the posture of a mobile phone (such as horizontal and vertical screen switching, related games, magnetometer posture calibration), vibration recognition related functions (such as pedometer and tapping), and the like; as for other sensors such as a fingerprint sensor, a pressure sensor, an iris sensor, a molecular sensor, a gyroscope, a barometer, a hygrometer, a thermometer, and an infrared sensor, which can be configured on the mobile phone, further description is omitted here.
The display unit 106 is used to display information input by a user or information provided to the user. The Display unit 106 may include a Display panel 1061, and the Display panel 1061 may be configured in the form of a Liquid Crystal Display (LCD), an Organic Light-Emitting Diode (OLED), or the like.
The user input unit 107 may be used to receive input numeric or character information and generate key signal inputs related to user settings and function control of the mobile terminal. Specifically, the user input unit 107 may include a touch panel 1071 and other input devices 1072. The touch panel 1071, also referred to as a touch screen, may collect a touch operation performed by a user on or near the touch panel 1071 (e.g., an operation performed by the user on or near the touch panel 1071 using a finger, a stylus, or any other suitable object or accessory), and drive a corresponding connection device according to a predetermined program. The touch panel 1071 may include two parts of a touch detection device and a touch controller. The touch detection device detects the touch 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 110, and can receive and execute commands sent by the processor 110. In addition, the touch panel 1071 may be implemented in various types, such as a resistive type, a capacitive type, an infrared ray, and a surface acoustic wave. In addition to the touch panel 1071, the user input unit 107 may include other input devices 1072. In particular, other input devices 1072 may include, but are not limited to, one or more of a physical keyboard, function keys (e.g., volume control keys, switch keys, etc.), a trackball, a mouse, a joystick, and the like, and are not limited to these specific examples.
Further, the touch panel 1071 may cover the display panel 1061, and when the touch panel 1071 detects a touch operation thereon or nearby, the touch panel 1071 transmits the touch operation to the processor 110 to determine the type of the touch event, and then the processor 110 provides a corresponding visual output on the display panel 1061 according to the type of the touch event. Although the touch panel 1071 and the display panel 1061 are shown in fig. 1 as two separate components to implement the input and output functions of the mobile terminal, in some embodiments, the touch panel 1071 and the display panel 1061 may be integrated to implement the input and output functions of the mobile terminal, and is not limited herein.
The interface unit 108 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 interface unit 108 may be used to receive input (e.g., data information, power, etc.) from external devices 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 external devices.
The memory 109 may be used to store software programs as well as various data. The memory 109 may mainly include a storage program area and a storage data area, wherein the storage program area may store an operating system, an application program required by at least one function (such as a sound playing function, an image playing function, etc.), and the like; the storage data area may store data (such as audio data, a phonebook, etc.) created according to the use of the cellular phone, and the like. Further, the memory 109 may include high speed random access memory, and may also include non-volatile memory, such as at least one magnetic disk storage device, flash memory device, or other volatile solid state storage device.
The processor 110 is a control center of the mobile terminal, connects various parts of the entire mobile terminal using various interfaces and lines, and performs various functions of the mobile terminal and processes data by operating or executing software programs and/or modules stored in the memory 109 and calling data stored in the memory 109, thereby performing overall monitoring of the mobile terminal. Processor 110 may include one or more processing units; preferably, the processor 110 may integrate an application processor, which mainly handles operating systems, user interfaces, application programs, etc., and a modem processor, which mainly handles wireless communications. It will be appreciated that the modem processor described above may not be integrated into the processor 110.
The mobile terminal 100 may further include a power supply 111 (e.g., a battery) for supplying power to various components, and preferably, the power supply 111 may be logically connected to the processor 110 via a power management system, so as to manage charging, discharging, and power consumption management functions via the power management system.
Although not shown in fig. 1, the mobile terminal 100 may further include a bluetooth module or the like, which is not described in detail herein.
In order to facilitate understanding of the embodiments of the present invention, a communication network system on which the mobile terminal of the present invention is based is described below.
Referring to fig. 2, fig. 2 is an architecture diagram of a communication Network system according to an embodiment of the present invention, where the communication Network system is an LTE system of a universal mobile telecommunications technology, and the LTE system includes a UE (User Equipment) 201, an E-UTRAN (Evolved UMTS Terrestrial Radio Access Network) 202, an EPC (Evolved Packet Core) 203, and an IP service 204 of an operator, which are in communication connection in sequence.
Specifically, the UE201 may be the terminal 100 described above, and is not described herein again.
The E-UTRAN202 includes eNodeB2021 and other eNodeBs 2022, among others. Among them, the eNodeB2021 may be connected with other eNodeB2022 through backhaul (e.g., X2 interface), the eNodeB2021 is connected to the EPC203, and the eNodeB2021 may provide the UE201 access to the EPC 203.
The EPC203 may include an MME (Mobility Management Entity) 2031, an HSS (Home Subscriber Server) 2032, other MMEs 2033, an SGW (Serving gateway) 2034, a PGW (PDN gateway) 2035, and a PCRF (Policy and Charging Rules Function) 2036, and the like. The MME2031 is a control node that handles signaling between the UE201 and the EPC203, and provides bearer and connection management. HSS2032 is used to provide registers to manage functions such as home location register (not shown) and holds subscriber specific information about service characteristics, data rates, etc. All user data may be sent through SGW2034, PGW2035 may provide IP address assignment for UE201 and other functions, and PCRF2036 is a policy and charging control policy decision point for traffic data flow and IP bearer resources, which selects and provides available policy and charging control decisions for a policy and charging enforcement function (not shown).
The IP services 204 may include the internet, intranets, IMS (IP Multimedia Subsystem), or other IP services, among others.
Although the LTE system is described as an example, it should be understood by those skilled in the art that the present invention is not limited to the LTE system, but may also be applied to other wireless communication systems, such as GSM, CDMA2000, WCDMA, TD-SCDMA, and future new network systems.
Based on the above mobile terminal hardware structure and communication network system, the present invention provides various embodiments of the method.
Example one
Fig. 3 is a flowchart of a first embodiment of an audio recording calibration method according to the present invention. A method of calibrating audio recording, the method comprising:
s1, acquiring audio data in a preset calibration period according to a preset gain value, and judging whether the gain amplitude of the audio data is smaller than a preset amplitude;
s2, if the gain amplitude of the audio data is smaller than the preset amplitude, performing amplification processing on the audio data;
s3, counting the times of executing the amplification processing operation in the calibration period;
and S4, if the times exceed a preset value, performing amplification processing on the preset gain value, and taking the gain value after the amplification processing operation as a fixed gain value for subsequently acquiring the audio data.
Alternatively, in the present embodiment, in consideration of the difference between the sensitivities of the recording microphones of the devices such as the mobile phone or the like in the related art, or the gain of the recorded sound introduced by the loss during the use of the microphones becomes small. The problem of the recording is not clearly heard or the opposite end of the call can not hear the speaking voice of the opposite end of the call, and the opposite end of the call is mistakenly considered to be silent in the call or failed in the recording, and the like. Meanwhile, considering the imperfection of echo cancellation technology of most mobile phones and other devices, when the gain of the sound recorded by the microphone is smaller than the gain of the echo reference signal, the situation of over-cancellation or incomplete cancellation may occur, so that the sound after noise cancellation processing appears in the situations of interruption, word dropping, and even no sound. Therefore, the embodiment provides a method for detecting the gain of the terminal sound recording in real time, and when it is found that the acquired sound recorded by the microphone is smaller than the preset gain value, the gain value of the current recording data is increased by using an algorithm or the (fixed analog/digital) gain value adopted by the microphone recording is increased to increase the gain value of the recording data, so as to achieve the recording data with ideal gain.
Optionally, in this embodiment, in a preset calibration period, audio data is obtained, and whether a gain amplitude of the audio data is smaller than a preset amplitude is determined; if the gain amplitude of the audio data is smaller than the preset amplitude, performing amplification processing on the audio data; then, counting the times of executing the amplification processing operation in the calibration period; and if the times exceed a preset value, performing amplification processing on the preset gain value, and taking the gain value after the amplification processing operation as a fixed gain value for subsequently acquiring the audio data. Specifically, the gain calibration may be implemented in two ways. Taking the recorded sound as an example, firstly, a dynamic data gain calibration scheme is adopted, wherein a sound gain monitoring algorithm is started while a sound recording channel is started, and amplitude monitoring is performed on the recorded data acquired from the microphone in real time. In this embodiment, under the condition of keeping the original sound waveform, the amplitude of this section of sound waveform is raised as a whole to meet the requirement of the latest sound amplitude. In order to implement dynamic regulation and determine a trigger condition of the dynamic regulation, in this embodiment, if a single discontinuous data frame is lower than a preset gain value, an algorithm is used to prompt an amplitude of the single data frame, and a number of data frames in which the amplitude needs to be adjusted in a unit time is recorded. If the number of data frames of which the amplitude needs to be adjusted exceeds a certain threshold value, the amplitude of the recorded data is integrally improved by directly setting the analog or digital gain value of the sound recording to be a larger value; optionally, in this embodiment, to implement the above calibration scheme, an algorithm is further required to monitor each currently recorded data frame in real time, and then add a short fixed gain adjustment.
Optionally, in this embodiment, in order to improve the applicable range of calibration, a static data gain calibration scheme is further proposed in a preset calibration period, in which a user manually starts a recording data gain calibration procedure (including a recording procedure and data frame amplitude monitoring) for a period of time (1 minute or 30s), and simultaneously plays a fixed amplitude sound to be recorded from a microphone. And then, calculating and setting the gain which needs to adjust the digital or analog according to the difference value of the recorded sound amplitude and the amplitude of the preset value, namely, the sound recording gain of the subsequent terminal can adjust the data according to the newly set gain value, and the amplitude of the recorded data frame does not need to be monitored in real time.
Optionally, in this embodiment, specifically, audio data is obtained in a preset calibration period according to a preset gain value, and it is determined whether a gain amplitude of the audio data is smaller than a preset amplitude; if the gain amplitude of the audio data is smaller than the preset amplitude, performing amplification processing on the audio data; then, counting the times of executing the amplification processing operation in the calibration period; and if the times exceed a preset value, performing amplification processing on the preset gain value, and taking the gain value after the amplification processing operation as a fixed gain value for subsequently acquiring the audio data.
Optionally, in this embodiment, the above embodiments relate to audio recording, amplitude adjustment, gain adjustment, and audio playing. The audio recording can be understood as the starting and the path configuration of the mobile phone recording or the uplink data of the call, and the data (amplitude) value obtained from the microphone end of the terminal mobile phone or the microphone end of the earphone can be transmitted to the amplitude adjustment normally or normally; amplitude adjustment, which is used for comparing the amplitude of the input recording data frame with a preset amplitude value and then determining whether adjustment is needed or not, adjusting the amplitude value of the data frame, outputting the output data frame to an application, and simultaneously transmitting the number of the recorded data frames processed in unit time to gain adjustment; gain adjustment, which is used for receiving the data frame value processed in unit time and input by the data frame amplitude adjustment module to compare with a preset value, if the input value is greater than the preset value, the analog or digital volume gain value of the recording data channel is newly set to improve the integral recording data amplitude, and meanwhile, the input value is reset and fed back to the amplitude adjustment for recounting; and the audio playing is used for actively providing a sound recording source with fixed amplitude for providing the accuracy of amplitude monitoring when the static data gain calibration function is used.
The embodiment has the advantages that the audio data is obtained in a preset calibration period according to the preset gain value, and whether the gain amplitude of the audio data is smaller than a preset amplitude is judged; if the gain amplitude of the audio data is smaller than the preset amplitude, performing amplification processing on the audio data; then, counting the times of executing the amplification processing operation in the calibration period; and if the times exceed a preset value, performing amplification processing on the preset gain value, and taking the gain value after the amplification processing operation as a fixed gain value for subsequently acquiring the audio data. The humanized audio recording calibration scheme is realized, the operation convenience of audio calibration is improved, a series of problems caused by echo cancellation are avoided, and the user experience is enhanced.
Example two
Fig. 4 is a flowchart of a second embodiment of the calibration method for audio recording according to the present invention, where based on the above embodiments, the acquiring audio data according to a preset gain value in a preset calibration period, and determining whether a gain amplitude of the audio data is smaller than a preset amplitude includes:
s11, determining a preset call duration as a first calibration period in a normal call scene;
and S12, determining a preset test duration as a second calibration period under the calibration setting scene.
Optionally, in this embodiment, for the dynamic calibration, that is, in a normal call scenario, a preset call duration is determined as a first calibration period;
optionally, in this embodiment, the first calibration period may be a time period intercepted during a normal call, for example, a time period of 30 seconds;
optionally, in this embodiment, for the static calibration, a preset test duration is determined as a second calibration period in a calibration setting scenario;
optionally, in this embodiment, the second calibration period may be a preset time of a calibration setting option provided by a device such as a mobile phone, and it can be understood that the preset time may be adjusted according to an actual requirement of a user.
The method has the advantages that a preset call duration is determined as a first calibration period in a normal call scene; then, a preset test duration is determined as a second calibration period in the calibration setting scenario. The method provides a definite basis of the calibration period for realizing a humanized audio recording calibration scheme, improves the operation convenience of audio calibration, avoids a series of problems caused by echo cancellation, and enhances the user experience.
EXAMPLE III
Fig. 5 is a flowchart of a third embodiment of the calibration method for audio recording according to the present invention, where based on the above embodiments, the obtaining audio data in a preset calibration period according to a preset gain value, and determining whether a gain amplitude of the audio data is smaller than a preset amplitude further includes:
s13, taking the fixed gain value after the previous calibration as the preset gain value;
s14, in the first calibration period, obtaining first audio data of each segment recorded under the scene of the normal call, or, in the second calibration period, obtaining at least one segment of second audio data recorded under the scene of the calibration setting.
Optionally, in this embodiment, the audio recording calibration of the present scheme may be triggered in a targeted manner within the normal use time of a device such as a mobile phone;
optionally, in this embodiment, the triggering condition may be that a keyword that is included in the voice of the other party of the call and has a semantic meaning such as inaudible speech, please click a loud sound, or the like is recognized;
optionally, in this embodiment, the fixed gain value after the previous calibration performed according to this scheme is used as the preset gain value of this calibration;
optionally, in this embodiment, a fixed gain value obtained after the dynamic calibration is used as a preset gain value for the next dynamic calibration or static calibration;
optionally, in this embodiment, a fixed gain value obtained after the static calibration is used as a preset gain value for the next dynamic calibration or static calibration.
The embodiment has the advantages that the fixed gain value after the previous calibration is taken as the preset gain value; then, in the first calibration period, first audio data of each segment recorded under the scene of the normal call is acquired, or, in the second calibration period, at least one segment of second audio data recorded under the scene of the calibration setting is acquired. The method provides a determination basis of the preset gain value for realizing a humanized audio recording calibration scheme, improves the operation convenience of audio calibration, avoids a series of problems caused by echo cancellation, and enhances the user experience.
Example four
Fig. 6 is a flowchart of a fourth embodiment of the calibration method for audio recording according to the present invention, where based on the above embodiments, the obtaining audio data in a preset calibration period according to a preset gain value, and determining whether a gain amplitude of the audio data is smaller than a preset amplitude further includes:
s15, setting a first preset amplitude value under the normal call scene, and setting a second preset amplitude value under the calibration setting scene;
and S16, acquiring the gain amplitude of the first audio data recorded in each section, and successively comparing the gain amplitude with the first preset amplitude, or acquiring the gain amplitude of the second audio data recorded in at least one section, and comparing the gain amplitude with the second preset amplitude.
Optionally, in this embodiment, in a call process, a voice segment of the user at the device end is determined, that is, continuous voice of the user each time is taken as a segment of this embodiment, and then a gain amplitude of the first audio data recorded in each segment is obtained in real time and is successively compared with the first preset amplitude;
optionally, in this embodiment, in a normal conversation process, at least one segment of voice of the device end user exists, so that the voice can be used as the first audio data;
optionally, in this embodiment, in the static calibration process, in the second calibration period, the user may speak a test voice that is completed in one segment and is continuous, and may also speak a test voice at one end in segments;
optionally, in this embodiment, the test speech is segmented, and a gain amplitude of at least one segment of recorded second audio data is obtained and compared with the second preset amplitude.
The method has the advantages that a first preset amplitude value is set in the normal call scene, and a second preset amplitude value is set in the calibration setting scene; then, gain amplitude of the first audio data recorded in each section is obtained and is successively compared with the first preset amplitude, or gain amplitude of the second audio data recorded in at least one section is obtained and is compared with the second preset amplitude. The method provides a judgment basis for real-time segmentation comparison for realizing a humanized audio recording calibration scheme, improves the operation convenience of audio calibration, avoids a series of problems caused by echo cancellation, and enhances the user experience.
EXAMPLE five
Fig. 7 is a flowchart of a fifth embodiment of the calibration method for audio recording according to the present invention, where based on the above embodiments, if the gain amplitude of the audio data is smaller than the preset amplitude, the performing amplification processing on the audio data includes:
s21, determining a preset proportion value as a first amplification proportion in a normal conversation scene, and determining a preset proportion value as a second amplification proportion in a calibration setting scene;
and S22, if the gain amplitude of the first audio data of any section is determined to be smaller than the first preset amplitude, performing amplification processing according to a first amplification proportion, or if the gain amplitude of the second audio data of at least one section is determined to be smaller than the second preset amplitude, performing amplification processing according to a second amplification proportion.
Optionally, in this embodiment, the first amplification ratio is obtained according to empirical data, or obtained according to the number of amplification operations, for example, the larger the number of amplification operations is, the higher the ratio is, so as to quickly improve the calibration effect;
optionally, in this embodiment, in the dynamic calibration process, if it is determined that the gain amplitude of the first audio data of any segment is smaller than the first preset amplitude, performing real-time amplification processing according to a first amplification proportion, so that the voice effects of the segment and subsequent segments can be effectively improved;
optionally, in this embodiment, in the static calibration process, if it is determined that the gain amplitude of the at least one segment of the second audio data is smaller than the second preset amplitude, the subsequent amplification processing of any call is performed according to a second amplification ratio, so that the subsequent voice effect can be effectively improved.
The method has the advantages that a preset proportion value is determined as a first amplification proportion in a normal conversation scene, and a preset proportion value is determined as a second amplification proportion in a calibration setting scene; and if the gain amplitude of the first audio data of any section is determined to be smaller than the first preset amplitude, performing amplification processing according to a first amplification proportion, or if the gain amplitude of the second audio data of at least one section is determined to be smaller than the second preset amplitude, performing amplification processing according to a second amplification proportion. The method provides a dynamic and static differentiated proportion regulation and control basis for realizing a humanized audio recording and calibration scheme, improves the operation convenience of audio calibration, avoids a series of problems caused by echo cancellation, and enhances the user experience.
EXAMPLE six
Fig. 8 is a flowchart of a sixth embodiment of the calibration method for audio recording according to the present invention, where based on the above embodiments, the counting the number of times of performing the amplification processing operation in the calibration period includes:
s31, accumulating the operations of performing the amplification processing in one of the first calibration periods, and counting the number of times obtained by the accumulation;
s32, accumulating the operations of performing the amplification processing in a plurality of the second calibration periods, and counting the number of times of accumulation.
Optionally, in this embodiment, the voice of the user at the device end is analyzed in a segmented manner, where the segmentation scheme provided in the above embodiment is a pause recognition of a natural voice of the user, that is, when the user starts listening to the voice of the other party, the voice of the user at the previous segment is taken as one end, and considering that the user may have a short conversation time in a conversation process, or the voice of the user is short and cannot effectively and accurately perform gain determination, the voice of the user can be analyzed in a segmented manner according to a preset time period, for example, each segment is 2s, the time is taken as a sub-period in a first calibration period, the operations of performing the amplification processing are accumulated, and the number of times obtained through accumulation is counted;
optionally, in this embodiment, in the static calibration process, the user may obtain corresponding multiple segments of voices in multiple scenes of simulated environment sounds, for example, in the static calibration process again, a quiet environment, a station noisy environment, a road noisy environment, and the like are sequentially provided;
optionally, in this embodiment, under each of the above environments, a test voice of the user is obtained;
optionally, in this embodiment, the number of times the test speech is amplified is accumulated.
The embodiment has the advantages that the operation of executing the amplification processing is accumulated in one first calibration period, and the times obtained by accumulation are counted; or, in a plurality of second calibration periods, accumulating the operations for executing the amplification processing, and counting the times obtained by accumulation. The method provides a dynamic and static differentiated counting judgment basis for realizing a humanized audio recording calibration scheme, improves the operation convenience of audio calibration, avoids a series of problems caused by echo cancellation, and enhances the user experience.
EXAMPLE seven
Fig. 9 is a flowchart of a seventh embodiment of the calibration method for audio recording according to the present invention, where based on the above embodiments, if the number of times exceeds a preset value, the amplifying processing is performed on the preset gain value, and the gain value after the amplifying processing operation is performed is used as a fixed gain value for subsequently acquiring the audio data, including:
s41, in one first calibration period, if the number of times exceeds a preset value, performing amplification processing on the preset gain value, and taking the gain value after performing the amplification processing operation as a first fixed gain value of the audio data in the remaining time of the first calibration period;
s42, in a plurality of the second calibration periods, if the number of times exceeds a preset value, performing amplification processing on the preset gain value, and taking the gain value after performing the amplification processing operation as a second fixed gain value of the audio data in the next second calibration period.
Optionally, in this embodiment, it is understood that the dynamic gain calibration is to perform gain on the entered data in real time during the use process, perform monitoring comparison, or dynamically raise the amplitude of the data frame, or dynamically adjust the amplitude of the data within a certain time period;
alternatively, in this embodiment, it is understood that the static gain calibration is a gain adjustment for recorded data that is slightly damaged or the microphone pick-up hole is partially blocked to affect the pick-up amplitude, and the adjusted gain is stored for a long time and used as a fixed analog or digital gain on each future recording channel. It is a compensation for damage to the (pick-up microphone) components or structures of the terminal equipment.
The present embodiment has the beneficial effects that, in one first calibration period, if the number of times exceeds a preset value, the preset gain value is subjected to amplification processing, and a gain value after the amplification processing operation is performed is used as a first fixed gain value of the audio data in the remaining time of the first calibration period; then, in a plurality of second calibration periods, if the number of times exceeds a preset value, performing amplification processing on the preset gain value, and taking a gain value after the amplification processing operation as a second fixed gain value of the audio data of a next second calibration period. The method provides a dynamic and static differentiated gain compensation determination basis for realizing a humanized audio recording calibration scheme, improves the operation convenience of audio calibration, avoids a series of problems caused by echo cancellation, and enhances the user experience.
Example eight
Fig. 10 is a flowchart of an eighth embodiment of the calibration method for audio recording according to the present invention, where based on the above embodiments, if the number of times exceeds a preset value, the method performs amplification processing on the preset gain value, and takes the gain value after performing the amplification processing operation as a fixed gain value for subsequently acquiring the audio data, and further includes:
s43, obtaining the first fixed gain value and the second fixed gain value according to the first calibration period and the second calibration period;
and S44, overlapping the first fixed gain value and the second fixed gain value to be used as a calibrated fixed gain value.
Optionally, in this embodiment, refer to a dynamic calibration flowchart of an eighth embodiment of the audio recording calibration method shown in fig. 11. In this process, first, recorded voice data is acquired by a microphone, and then this voice data is compared with a preset voice gain amplitude: if the amplitude of the data frame is larger than the preset value, the data frame is directly stored in a data cache area without any processing, and at the moment, the application can acquire the recording data or transmit the data required by the noise elimination algorithm processing according to the requirement; if the amplitude of the data frame is smaller than the preset value, the data frame is subjected to amplification processing, meanwhile, 1 is added to the count value of the data frame subjected to amplification processing, the counting operation is executed in a circulating mode, whether the count of the data frame subjected to amplification processing in the preset unit time is larger than the preset value or not is judged, if the count is larger than the preset value, a new fixed gain value of the recording data is set according to an actual result, and the count value of the data frame subjected to amplification processing is requested to start to be counted again.
Optionally, in this embodiment, refer to a static calibration flowchart of an eighth embodiment of the audio recording calibration method shown in fig. 12. In the process, firstly, sound with a fixed amplitude is actively played as a recording sound source through the setting or test item of the terminal equipment, then, the sound data is recorded through a recording microphone, and similarly, the sound data is compared with a preset sound gain amplitude: if the amplitude of the data frame is larger than the preset value, the data frame is directly stored in a data cache area without any processing, and at the moment, the application can acquire the recording data or transmit the data required by the noise elimination algorithm processing according to the requirement; if the amplitude of the data frame is smaller than the preset value, setting a fixed gain value of new recording data according to an actual result, wherein the calibration process can be executed in a circulating manner until the calibration time is reached, or the playing of the exemplary recording sound source for setting or testing is finished. Likewise, it will be appreciated that during static calibration, it is also possible to use a segmented version of the set-up or tested exemplary recorded sound source to implement a calibration scheme in which counting triggers a fixed gain value during dynamic calibration.
Optionally, in this embodiment, in order to adapt to a wider microphone sensitivity range, it is ensured that the input sound amplitude satisfies the requirement of the minimum noise reduction processing input amplitude, and the sound is not processed to be silent by the noise reduction algorithm because the sound pickup amplitude is too small. For the condition that the microphone sound pick-up hole is slightly damaged or the microphone sound pick-up hole is partially blocked to influence the sound pick-up amplitude, the user can integrally raise the recorded data amplitude in a static data gain calibration mode. The service life of the mobile phone terminal (microphone) can be prolonged;
optionally, in this embodiment, the dynamic calibration and the static calibration may be performed alternately, and the calibration effects may be superimposed;
optionally, in this embodiment, after detecting that dynamic calibration is performed every preset number of times, the user is prompted to perform static calibration once, so as to timely find and remind the user of a problem about a failure of a microphone hardware.
The present embodiment has the beneficial effects that the first fixed gain value and the second fixed gain value are obtained by respectively obtaining the first fixed gain value and the second fixed gain value according to the first calibration period and implementing the second calibration period; and superposing the first fixed gain value and the second fixed gain value to be used as the calibrated fixed gain value. The method provides a dynamic and static combined gain compensation determination basis for realizing a humanized audio recording calibration scheme, improves the operation convenience of audio calibration, avoids a series of problems caused by echo cancellation, and enhances the user experience.
Example nine
Based on the foregoing embodiments, the present invention further provides an audio recording calibration apparatus, which includes a memory, a processor, and a computer program stored on the memory and executable on the processor, where the computer program, when executed by the processor, implements the steps of the audio recording calibration method according to any one of the above.
It should be noted that the device embodiment and the method embodiment belong to the same concept, and specific implementation processes thereof are detailed in the method embodiment, and technical features in the method embodiment are correspondingly applicable in the device embodiment, which is not described herein again.
Example ten
Based on the foregoing embodiments, the present invention further provides a computer-readable storage medium, where an audio recording calibration program is stored on the computer-readable storage medium, and when being executed by a processor, the audio recording calibration program implements the steps of the audio recording calibration method according to any one of the foregoing embodiments.
It should be noted that the media embodiment and the method embodiment belong to the same concept, and specific implementation processes thereof are detailed in the method embodiment, and technical features in the method embodiment are correspondingly applicable in the media embodiment, which is not described herein again.
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 (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.
While the present invention has been described with reference to the embodiments shown in the drawings, the present invention is not limited to the embodiments, which are illustrative and not restrictive, and it will be apparent to those skilled in the art that various changes and modifications can be made therein without departing from the spirit and scope of the invention as defined in the appended claims.
Claims (10)
1. A method for calibrating audio recording, the method comprising:
acquiring audio data in a preset calibration period according to a preset gain value, and judging whether the gain amplitude of the audio data is smaller than a preset amplitude;
if the gain amplitude of the audio data is smaller than the preset amplitude, performing amplification processing on the audio data;
counting the times of executing the amplification processing operation in the calibration period;
and if the times exceed a preset value, performing amplification processing on the preset gain value, and taking the gain value after the amplification processing operation as a fixed gain value for subsequently acquiring the audio data.
2. The method of claim 1, wherein the obtaining audio data within a predetermined calibration period according to a predetermined gain value and determining whether the gain amplitude of the audio data is smaller than a predetermined amplitude value comprises:
determining a preset call duration as a first calibration period in a normal call scene;
and determining a preset test duration as a second calibration period under the scene of calibration setting.
3. The audio recording calibration method of claim 2, wherein the audio data is obtained within a preset calibration period according to a preset gain value, and whether the gain amplitude of the audio data is smaller than a preset amplitude value is determined, further comprising:
taking the fixed gain value after the previous calibration as the preset gain value;
and acquiring first audio data of each section recorded under the scene of the normal call in the first calibration period, or acquiring at least one section of second audio data recorded under the scene of the calibration setting in the second calibration period.
4. The audio recording calibration method of claim 3, wherein the audio data is obtained within a preset calibration period according to a preset gain value, and whether the gain amplitude of the audio data is smaller than a preset amplitude value is determined, further comprising:
setting a first preset amplitude value under the scene of the normal call, and setting a second preset amplitude value under the scene of the calibration setting;
and acquiring the gain amplitude of the first audio data recorded in each section, and successively comparing the gain amplitude with the first preset amplitude, or acquiring the gain amplitude of the second audio data recorded in at least one section, and comparing the gain amplitude with the second preset amplitude.
5. The method of claim 4, wherein if the gain amplitude of the audio data is smaller than the predetermined amplitude, performing an amplification process on the audio data, includes:
determining a preset proportion value as a first amplification proportion in a normal conversation scene, and determining a preset proportion value as a second amplification proportion in a calibration setting scene;
and if the gain amplitude of the first audio data of any section is determined to be smaller than the first preset amplitude, performing amplification processing according to a first amplification proportion, or if the gain amplitude of the second audio data of at least one section is determined to be smaller than the second preset amplitude, performing amplification processing according to a second amplification proportion.
6. The method of calibrating audio recording according to claim 5, wherein said counting the number of times the amplification process is performed in the calibration period comprises:
accumulating the operations of executing the amplification processing in one first calibration period, and counting the times obtained by accumulation;
or, in a plurality of second calibration periods, accumulating the operations for executing the amplification processing, and counting the times obtained by accumulation.
7. The method of claim 6, wherein if the number of times exceeds a predetermined value, performing amplification processing on the predetermined gain value, and taking the gain value after performing the amplification processing operation as a fixed gain value for subsequently acquiring the audio data, comprises:
in one first calibration period, if the times exceed a preset value, performing amplification processing on the preset gain value, and taking the gain value after the amplification processing operation as a first fixed gain value of the audio data in the remaining time of the first calibration period;
in a plurality of second calibration periods, if the number of times exceeds a preset value, performing amplification processing on the preset gain value, and taking a gain value after the amplification processing operation as a second fixed gain value of the audio data of a next second calibration period.
8. The method of claim 7, wherein if the number of times exceeds a predetermined value, performing amplification processing on the predetermined gain value, and taking the gain value after performing the amplification processing operation as a fixed gain value for subsequently acquiring the audio data, further comprising:
obtaining the first fixed gain value and the second fixed gain value according to the first calibration period and the second calibration period;
and superposing the first fixed gain value and the second fixed gain value to be used as the calibrated fixed gain value.
9. An audio recording calibration device, characterized in that the device comprises a memory, a processor and a computer program stored on the memory and executable on the processor, which computer program, when executed by the processor, carries out the steps of the audio recording calibration method according to any one of claims 1 to 8.
10. A computer-readable storage medium, having an audio recording calibration program stored thereon, which when executed by a processor implements the steps of the audio recording calibration method according to any one of claims 1 to 8.
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