CN112911056B - Audio recording calibration method, device and computer readable storage medium - Google Patents

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 or not; if the gain amplitude of the audio data is smaller than the preset amplitude, amplifying the audio data; then, counting the times of executing the amplification processing operation in the calibration period; if the times exceeds a preset value, amplifying the preset gain value, and taking the gain value after the amplifying operation is executed as a fixed gain value for obtaining the audio data later. 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

Audio recording calibration method, device and computer readable storage medium

Technical Field

The present invention relates to the field of mobile communications, and in particular, to an audio recording calibration method, apparatus, and computer readable storage medium.

Background

In the prior art, with the continuous development of intelligent terminal equipment, the use frequency of the equipment by users is also higher and higher, and at present, the gain of recorded sound introduced by the loss in the use process of the microphone is smaller due to the difference between the sensitivity of the recording microphone of the equipment such as a mobile phone and the like. The sound recording problem is not clear, or the opposite terminal can not hear the speaking sound of the user, and the user can mistakenly consider the situations of silence or recording failure, etc. Meanwhile, considering the imperfection of the echo cancellation technology of most mobile phones and other devices, when the gain of the recorded sound of the microphone is smaller than that of the echo reference signal, the situation of over cancellation or unclean cancellation may occur, so that the sound after the noise cancellation processing is intermittent, word dropping and even soundless occur, and the user experience is affected.

Disclosure of Invention

In order to solve the above technical drawbacks in the prior art, the present invention provides an audio recording calibration method, which includes:

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 or not;

if the gain amplitude of the audio data is smaller than the preset amplitude, amplifying the audio data;

Counting the times of executing the amplification processing operation in the calibration period;

if the times exceeds a preset value, amplifying the preset gain value, and taking the gain value after the amplifying operation is executed as a fixed gain value for obtaining the audio data later.

Optionally, the obtaining the audio data according to the preset gain value in a preset calibration period, and determining whether the gain amplitude of the audio data is smaller than a preset amplitude includes:

determining a preset call duration as a first calibration period under a normal call scene;

and determining a preset test duration as a second calibration period in the calibration setting scene.

Optionally, the obtaining the audio data according to the preset gain value in a preset calibration period, and determining whether the 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 in the first calibration period, acquiring first audio data of each section recorded in the scene of the normal call, or in the second calibration period, acquiring second audio data of at least one section recorded in the scene of the calibration setting.

Optionally, the obtaining the audio data according to the preset gain value in a preset calibration period, and determining whether the gain amplitude of the audio data is smaller than a preset amplitude, further includes:

setting a first preset amplitude value in the normal conversation scene, and setting a second preset amplitude value in the calibration setting scene;

gain amplitude values of the first audio data of each recorded section are obtained and are compared with the first preset amplitude values successively, or gain amplitude values of the second audio data of at least one recorded section are obtained and are compared with the second preset amplitude values.

Optionally, if the gain amplitude of the audio data is smaller than the preset amplitude, the amplifying processing is performed 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;

if the gain amplitude of the first audio data of any section is 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 smaller than the second preset amplitude, performing amplification processing according to a second amplification proportion.

Optionally, the counting the number of times the amplification processing operation is performed in the calibration period includes:

accumulating the operations for executing the amplification processing in one first calibration period, and counting the accumulated times;

or, in a plurality of the second calibration periods, accumulating the operations of performing the amplification processing, and counting the accumulated times.

Optionally, if the number of times exceeds a preset value, performing amplification processing on the preset gain value, and taking the gain value after the amplification processing operation is performed as a fixed gain value for subsequently acquiring the audio data, including:

in the 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 the amplification processing operation is performed as a first fixed gain value of the audio data in the residual time of the first calibration period;

and in a plurality of second calibration periods, 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 is performed as a second fixed gain value of the audio data in the next second calibration period.

Optionally, if the number of times exceeds a preset value, amplifying the preset gain value, and taking the gain value after the amplifying operation is performed as a fixed gain value for subsequently acquiring the audio data, and further including:

obtaining the first fixed gain value and the second fixed gain value according to the first calibration period and the implementation second calibration period respectively;

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 proposes an audio recording calibration apparatus comprising a memory, a processor and a computer program stored on said memory and executable on said processor, said computer program implementing the steps of the audio recording calibration method as defined in any one of the preceding claims when executed by said processor.

The invention also proposes a computer readable storage medium having stored thereon an audio recording calibration program which, when executed by a processor, implements the steps of the audio recording calibration method as described in any of the preceding claims.

According to the audio recording calibration method, the audio recording calibration equipment and the computer readable storage medium, whether the gain amplitude of the audio data is smaller than a preset amplitude is judged by acquiring the audio data according to the preset gain value in a preset calibration period; if the gain amplitude of the audio data is smaller than the preset amplitude, amplifying the audio data; then, counting the times of executing the amplification processing operation in the calibration period; if the times exceeds a preset value, amplifying the preset gain value, and taking the gain value after the amplifying operation is executed as a fixed gain value for obtaining the audio data later. 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 schematic diagram of a communication network system according to an embodiment of the present invention;

FIG. 3 is a flowchart of a first embodiment of an audio recording calibration method of the present invention;

FIG. 4 is a flowchart of a second embodiment of an audio recording calibration method of the present invention;

FIG. 5 is a flowchart of a third embodiment of an audio recording calibration method of the present invention;

FIG. 6 is a flowchart of a fourth embodiment of an audio recording calibration method of the present invention;

FIG. 7 is a flowchart of a fifth embodiment of an audio recording calibration method of the present invention;

FIG. 8 is a flowchart of a sixth embodiment of an audio recording calibration method according to the present invention;

FIG. 9 is a flowchart of a seventh embodiment of an audio recording calibration method of the present invention;

FIG. 10 is a flowchart of an eighth embodiment of an audio recording calibration method according to the present invention;

FIG. 11 is a dynamic calibration flowchart of an eighth embodiment of an audio recording calibration method of the present invention;

fig. 12 is a static calibration flowchart of an eighth embodiment of the audio recording calibration method of the present invention.

Detailed Description

It should be understood that the specific embodiments described herein are for purposes of illustration only and are not intended to limit the scope of the invention.

In the following description, suffixes such as "module", "component", or "unit" for representing elements are used only for facilitating the description of the present invention, and have no specific meaning per se. Thus, "module," "component," or "unit" may be used in combination.

The terminal may be implemented in various forms. For example, the terminals described in the present invention may include mobile terminals such as cell phones, tablet computers, notebook computers, palm computers, personal digital assistants (Personal Digital Assistant, PDA), portable media players (Portable Media Player, PMP), navigation devices, wearable devices, smart bracelets, pedometers, and fixed terminals such as digital TVs, desktop computers, and the like.

The following description will be given taking a mobile terminal as an example, and those skilled in the art will understand that the configuration according to the embodiment of the present invention can be applied to a fixed type terminal in addition to elements particularly used for a moving purpose.

Referring to fig. 1, which is a schematic diagram of a hardware structure of a mobile terminal implementing various embodiments of the present invention, the mobile terminal 100 may include: an RF (Radio Frequency) unit 101, a WiFi module 102, an audio output unit 103, an a/V (audio/video) input unit 104, a sensor 105, a display unit 106, a user input unit 107, an interface unit 108, a memory 109, a processor 110, and a power supply 111. Those skilled in the art will appreciate that the mobile terminal structure shown in fig. 1 is not limiting of the mobile terminal and that the mobile terminal may include more or fewer components than shown, or may combine certain components, or a different arrangement of components.

The following describes the components of the mobile terminal in detail with reference to fig. 1:

the radio frequency unit 101 may be used for receiving and transmitting signals during the information receiving or communication process, specifically, after receiving downlink information of the base station, processing the downlink information by the processor 110; and, the uplink data is transmitted to the base station. Typically, the radio frequency unit 101 includes, but is not limited to, an antenna, at least one amplifier, a transceiver, a coupler, a low noise amplifier, a duplexer, and the like. In addition, the radio frequency unit 101 may also communicate with networks and other devices via wireless communications. The wireless communication may use any communication standard or protocol, including but not limited to GSM (Global System of Mobile communication, global System for Mobile communications), GPRS (General Packet Radio Service ), CDMA2000 (Code Division Multiple Access, CDMA 2000), WCDMA (Wideband Code Division Multiple Access ), TD-SCDMA (Time Division-Synchronous Code Division Multiple Access, time Division synchronous code Division multiple Access), FDD-LTE (Frequency Division Duplexing-Long Term Evolution, frequency Division Duplex Long term evolution), and TDD-LTE (Time Division Duplexing-Long Term Evolution, time Division Duplex Long term evolution), etc.

WiFi belongs to a short-distance wireless transmission technology, and a mobile terminal can help a user to send and receive e-mails, browse web pages, access streaming media and the like through the WiFi module 102, so that wireless broadband Internet access is provided for the user. Although fig. 1 shows a WiFi module 102, it is understood that it does not belong to the necessary constitution of a mobile terminal, and can be omitted entirely as required within a range that does not change the essence of the invention.

The audio output unit 103 may convert audio data received by the radio frequency unit 101 or the WiFi module 102 or stored in the memory 109 into an audio signal and output as sound when the mobile terminal 100 is in a call signal reception mode, a talk mode, a recording mode, a voice recognition mode, a broadcast reception mode, or the like. Also, the audio output unit 103 may also provide audio output (e.g., a call signal reception sound, a message reception sound, etc.) related to a specific function performed by the mobile terminal 100. The audio output unit 103 may include a speaker, a buzzer, and the like.

The a/V input unit 104 is used to receive an audio or video signal. The a/V input unit 104 may include a graphics processor (Graphics Processing Unit, GPU) 1041 and a microphone 1042, the graphics processor 1041 processing image data of still pictures or video obtained by an image capturing device (e.g., a camera) in a video capturing mode or an image capturing mode. The processed image frames may be displayed on the display unit 106. The image frames processed by the graphics processor 1041 may be stored in the memory 109 (or other storage medium) or transmitted via the radio frequency unit 101 or the WiFi module 102. The microphone 1042 can receive sound (audio data) via the microphone 1042 in a phone call mode, a recording mode, a voice recognition mode, and the like, and can process such sound into audio data. The processed audio (voice) data may be converted into a format output that can be transmitted to the mobile communication base station via the radio frequency unit 101 in the case of a telephone call mode. The microphone 1042 may implement various types of noise cancellation (or suppression) algorithms to cancel (or suppress) noise or interference generated in the course of receiving and transmitting the audio signal.

The mobile terminal 100 also includes at least one sensor 105, such as a light sensor, a motion sensor, and other sensors. Specifically, the light sensor includes an ambient light sensor and a proximity sensor, wherein the ambient light sensor can adjust the brightness of the display panel 1061 according to the brightness of ambient light, and the proximity sensor can turn off the display panel 1061 and/or the backlight when the mobile terminal 100 moves to the ear. As one of the motion sensors, the accelerometer sensor can detect the acceleration in all directions (generally three axes), and can detect the gravity and direction when stationary, and can be used for applications of recognizing the gesture of a mobile phone (such as horizontal and vertical screen switching, related games, magnetometer gesture calibration), vibration recognition related functions (such as pedometer and knocking), and the like; as for other sensors such as fingerprint sensors, pressure sensors, iris sensors, molecular sensors, gyroscopes, barometers, hygrometers, thermometers, infrared sensors, etc. that may also be configured in the mobile phone, the detailed description thereof will be omitted.

The display unit 106 is used to display information input by a user or information provided to the user. The display unit 106 may include a display panel 1061, and the display panel 1061 may be configured in the form of a liquid crystal display (Liquid Crystal Display, LCD), an Organic Light-Emitting Diode (OLED), or the like.

The user input unit 107 may be used to receive input numeric or character information and to generate key signal inputs related to user settings and function control of the mobile terminal. In particular, the user input unit 107 may include a touch panel 1071 and other input devices 1072. The touch panel 1071, also referred to as a touch screen, may collect touch operations thereon or thereabout by a user (e.g., operations of the user on the touch panel 1071 or thereabout by using any suitable object or accessory such as a finger, a stylus, etc.) and drive the corresponding connection device according to a predetermined program. The touch panel 1071 may include two parts of a touch detection device and a touch controller. The touch detection device detects the touch azimuth of a user, detects a signal brought by touch operation and transmits the signal to the touch controller; the touch controller receives touch information from the touch detection device, converts it into touch point coordinates, and sends the touch point coordinates to the processor 110, and can receive and execute commands sent from the processor 110. Further, the touch panel 1071 may be implemented in various types such as resistive, capacitive, infrared, and surface acoustic wave. The user input unit 107 may include other input devices 1072 in addition to the touch panel 1071. In particular, other input devices 1072 may include, but are not limited to, one or more of a physical keyboard, function keys (e.g., volume control keys, switch keys, etc.), a trackball, mouse, joystick, etc., as specifically not limited herein.

Further, the touch panel 1071 may overlay the display panel 1061, and when the touch panel 1071 detects a touch operation thereon or thereabout, the touch panel 1071 is transferred to the processor 110 to determine the type of touch event, and then the processor 110 provides a corresponding visual output on the display panel 1061 according to the type of touch event. Although in fig. 1, the touch panel 1071 and the display panel 1061 are two independent components for implementing the input and output functions of the mobile terminal, in some embodiments, the touch panel 1071 may be integrated with the display panel 1061 to implement the input and output functions of the mobile terminal, which is not limited herein.

The interface unit 108 serves as an interface through which at least one external device can be connected with the mobile terminal 100. For example, the external devices may include a wired or wireless headset port, an external power (or battery charger) port, a wired or wireless data port, a memory card port, a port for connecting a device having an identification module, an audio input/output (I/O) port, a video I/O port, an earphone port, and the like. The interface unit 108 may be used to receive input (e.g., data information, power, etc.) from an external device and transmit the received input to one or more elements within the mobile terminal 100 or may be used to transmit data between the mobile terminal 100 and an external device.

Memory 109 may be used to store software programs as well as various data. The memory 109 may mainly include a storage program area that may store an operating system, application programs required for at least one function (such as a sound playing function, an image playing function, etc.), and a storage data area; the storage data area may store data (such as audio data, phonebook, etc.) created according to the use of the handset, etc. In addition, memory 109 may include high-speed random access memory, and may also include non-volatile memory, such as at least one magnetic disk storage device, flash memory device, or other volatile solid-state storage device.

The processor 110 is a control center of the mobile terminal, connects various parts of the entire mobile terminal using various interfaces and lines, and performs various functions of the mobile terminal and processes data by running or executing software programs and/or modules stored in the memory 109 and calling data stored in the memory 109, thereby performing overall monitoring of the mobile terminal. Processor 110 may include one or more processing units; preferably, the processor 110 may integrate an application processor that primarily handles operating systems, user interfaces, applications, etc., with a modem processor that primarily handles wireless communications. It will be appreciated that the modem processor described above may not be integrated into the processor 110.

The mobile terminal 100 may further include a power source 111 (e.g., a battery) for supplying power to the respective components, and preferably, the power source 111 may be logically connected to the processor 110 through a power management system, so as to perform functions of managing charging, discharging, and power consumption management through the power management system.

Although not shown in fig. 1, the mobile terminal 100 may further include a bluetooth module or the like, which is not described herein.

In order to facilitate understanding of the embodiments of the present invention, a communication network system on which the mobile terminal of the present invention is based will be described below.

Referring to fig. 2, fig. 2 is a schematic diagram of a communication network system according to an embodiment of the present invention, where the communication network system is an LTE system of a general mobile communication technology, and the LTE system includes a UE (User Equipment) 201, an e-UTRAN (Evolved UMTS Terrestrial Radio Access Network ) 202, an epc (Evolved Packet Core, evolved packet core) 203, and an IP service 204 of an operator that are sequentially connected in communication.

Specifically, the UE201 may be the terminal 100 described above, and will not be described herein.

The E-UTRAN202 includes eNodeB2021 and other eNodeB2022, etc. The eNodeB2021 may be connected with other eNodeB2022 by a backhaul (e.g., an X2 interface), the eNodeB2021 is connected to the EPC203, and the eNodeB2021 may provide access from the UE201 to the EPC 203.

EPC203 may include MME (Mobility Management Entity ) 2031, hss (Home Subscriber Server, home subscriber server) 2032, other MMEs 2033, SGW (Serving Gate Way) 2034, pgw (PDN Gate Way) 2035 and PCRF (Policy and Charging Rules Function, policy and tariff function entity) 2036, and so on. The MME2031 is a control node that handles signaling between the UE201 and EPC203, providing bearer and connection management. HSS2032 is used to provide registers to manage functions such as home location registers (not shown) and to hold user specific information about service characteristics, data rates, etc. All user data may be sent through SGW2034 and PGW2035 may provide IP address allocation and other functions for UE201, PCRF2036 is a policy and charging control policy decision point for traffic data flows and IP bearer resources, which selects and provides available policy and charging control decisions for a policy and charging enforcement function (not shown).

IP services 204 may include the internet, intranets, IMS (IP Multimedia Subsystem ), or other IP services, etc.

Although the LTE system is described above as an example, it should be understood by those skilled in the art that the present invention is not limited to LTE systems, but may be applied to other wireless communication systems, such as GSM, CDMA2000, WCDMA, TD-SCDMA, and future new network systems.

Based on the above mobile terminal hardware structure and the communication network system, various embodiments of the method of the present invention are provided.

Example 1

Fig. 3 is a flowchart of a first embodiment of an audio recording calibration method of the present invention. An audio recording calibration method, 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 or not;

s2, if the gain amplitude of the audio data is smaller than the preset amplitude, amplifying 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, carrying out amplification processing on the preset gain value, and taking the gain value after the amplification processing operation is carried out as a fixed gain value for subsequently acquiring the audio data.

Alternatively, in this embodiment, the difference between the sensitivities of the recording microphones of devices such as mobile phones and the like in the prior art is considered, or the gain of the recorded sound due to loss during use of the microphones becomes smaller. The sound recording problem is not clear, or the opposite terminal can not hear the speaking sound of the user, and the user can mistakenly consider the situations of silence or recording failure, etc. Meanwhile, considering the imperfection of the echo cancellation technology of most mobile phones and other devices, when the gain of the recorded sound of the microphone is smaller than that of the echo reference signal, the situation of over cancellation or unclean cancellation may occur, so that the sound after the noise cancellation processing is intermittent, word dropping and even silence occur. Therefore, the embodiment proposes a method for detecting the gain of the sound recording of the terminal in real time, and when the obtained sound recorded by the microphone is found to be smaller than the preset gain value, the gain value of the current recording data is increased by the algorithm or the gain value (fixed analog/digital) adopted by the microphone recording is increased, so that the recording data with ideal gain is achieved.

Optionally, in this embodiment, in a preset calibration period, audio data is acquired, and whether the 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, amplifying the audio data; then, counting the times of executing the amplification processing operation in the calibration period; if the times exceeds a preset value, amplifying the preset gain value, and taking the gain value after the amplifying operation is executed as a fixed gain value for obtaining the audio data later. In particular, the gain calibration may be achieved in two ways. Taking 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 carried out on recorded data obtained from a microphone in real time. In this embodiment, the amplitude of the original sound waveform is raised integrally under the condition of maintaining the original sound waveform, so as to meet the requirement of the latest sound amplitude. In this embodiment, if a single discontinuous data frame appears to be lower than a preset gain value, an algorithm is adopted to prompt the amplitude of the single data frame, and the number of data frames of which the amplitude needs to be adjusted in a unit time is recorded. If the number of data frames of which the amplitude is required to be adjusted exceeds a certain threshold value, directly setting an analog or digital gain value of sound recording to be a larger value so as to integrally improve the amplitude of recorded data; optionally, in this embodiment, to implement the calibration scheme described above, an algorithm is also required to monitor each data frame currently recorded in real time, plus a short fixed gain adjustment.

Optionally, in this embodiment, in order to improve the application range of calibration, a static data gain calibration scheme is also proposed in a preset calibration period, where the user manually starts a recording data gain calibration procedure (including recording procedure and data frame amplitude monitoring) for a period of time (1 minute or 30 seconds), and simultaneously plays a sound with a fixed amplitude for recording from the microphone. Then, according to the difference value between the recorded sound amplitude and the amplitude of the preset value, the gain which needs to be adjusted in digital or analog mode is calculated and set, namely, the sound recording gain of the subsequent terminal can adjust 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 whether the 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, amplifying the audio data; then, counting the times of executing the amplification processing operation in the calibration period; if the times exceeds a preset value, amplifying the preset gain value, and taking the gain value after the amplifying operation is executed as a fixed gain value for obtaining the audio data later.

Optionally, in this embodiment, the above embodiment relates to audio recording, amplitude adjustment, gain adjustment, and audio playing. The audio recording can be understood as the starting and the channel configuration of mobile phone recording or call uplink data, and can normally or data (amplitude) values obtained from a mobile phone microphone or an earphone microphone end of a terminal can be transmitted to the amplitude adjustment; amplitude adjustment, which is used to compare the amplitude of the input recording data frame with the preset amplitude value and then determine whether to adjust, how to adjust the amplitude value of the segment of data frame, and output the output data frame to the application, and at the same time, transmit the data frame processed in the unit time to gain adjustment; gain adjustment, which is used for receiving the data frame value processed in unit time input by the data frame amplitude adjustment module, comparing with a preset value, if the input value is larger than the preset value, setting the analog or digital volume gain value of the recording data channel to improve the whole recording data amplitude, and resetting the input value and feeding back to the amplitude adjustment re-counting process; and the audio playing is used for actively providing a recording sound source with fixed amplitude for providing the accuracy of amplitude monitoring when the static data gain calibration function is used.

The method has the advantages that whether the gain amplitude of the audio data is smaller than a preset amplitude is judged by acquiring the audio data in a preset calibration period according to the preset gain value; if the gain amplitude of the audio data is smaller than the preset amplitude, amplifying the audio data; then, counting the times of executing the amplification processing operation in the calibration period; if the times exceeds a preset value, amplifying the preset gain value, and taking the gain value after the amplifying operation is executed as a fixed gain value for obtaining the audio data later. 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 an audio recording calibration method according to the present invention, based on the above embodiment, the step of obtaining audio data according to a preset gain value in a preset calibration period, and determining whether the gain amplitude of the audio data is smaller than a preset amplitude includes:

s11, determining a preset call duration as a first calibration period under a normal call scene;

S12, determining a preset test duration as a second calibration period in a calibration setting scene.

Optionally, in this embodiment, for the dynamic calibration, that is, determining a preset call duration as the first calibration period in a normal call scenario;

alternatively, in this embodiment, the first calibration period may be a period of time intercepted during a normal call, for example, may be a period of 30 seconds;

optionally, in this embodiment, for the static calibration, a preset test duration is determined as the second calibration period in the calibration setting scenario;

optionally, in this embodiment, the second calibration period may be a preset time of calibration setting options provided by a device such as a mobile phone, and it is understood that the preset time may be adjusted according to an actual requirement of a user.

The method has the advantages that the preset call duration is determined as the first calibration period under the normal call scene; then, a preset test duration is determined as a second calibration period in the calibration setting scenario. The method provides a determination basis of a 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 user experience.

Example III

Fig. 5 is a flowchart of a third embodiment of an audio recording calibration method according to the present invention, based on the above embodiment, the audio data is obtained according to a preset gain value in a preset calibration period, and the method further includes:

s13, taking the fixed gain value after the previous calibration as the preset gain value;

s14, in the first calibration period, acquiring first audio data of each section recorded in a scene of the normal call, or in the second calibration period, acquiring at least one section of second audio data recorded in a scene of the calibration setting.

Optionally, in this embodiment, during a normal use time of a device such as a mobile phone, the audio recording calibration of the present solution may be triggered in a targeted manner;

alternatively, in this embodiment, the triggering condition may be that a keyword that recognizes that the voice of the opposite party includes semantics such as inaudible, loud voice, etc.;

optionally, in this embodiment, the fixed gain value after the previous calibration performed according to this scheme is used as the preset gain value for the present calibration;

optionally, in this embodiment, the fixed gain value obtained after the dynamic calibration is used as the preset gain value of the next dynamic calibration or static calibration;

Optionally, in this embodiment, the fixed gain value obtained after static calibration is used as the preset gain value for the next dynamic calibration or static calibration.

The embodiment has the beneficial effects that the fixed gain value after the previous calibration is used as the preset gain value; and then, in the first calibration period, acquiring first audio data of each section recorded in the scene of the normal call, or in the second calibration period, acquiring second audio data of at least one section recorded in the scene of the calibration setting. The method provides a determination basis of a 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 user experience.

Example IV

Fig. 6 is a flowchart of a fourth embodiment of an audio recording calibration method according to the present invention, based on the above embodiment, the audio data is obtained according to a preset gain value in a preset calibration period, and the method further includes:

s15, setting a first preset amplitude value in the normal conversation scene, and setting a second preset amplitude value in the calibration setting scene;

S16, acquiring the gain amplitude of each recorded section of the first audio data, and comparing the gain amplitude with the first preset amplitude successively, or acquiring the gain amplitude of at least one recorded section of the second audio data, and comparing the gain amplitude with the second preset amplitude.

Optionally, in this embodiment, during a call, a voice segment of the user at the device end is determined, that is, a continuous voice of each time of the user is taken as a segment of this embodiment, and then a gain amplitude of the first audio data of each segment is acquired in real time and is compared with the first preset amplitude successively;

optionally, in this embodiment, during a normal call, at least one section of voice of the user at the device end exists, so that the voice can be used as the first audio data;

optionally, in this embodiment, during the static calibration process, during the second calibration period, the user may speak a segment of the completed and continuous test voice, and may speak a segment of the test voice at one end as well;

optionally, in this embodiment, the test speech is segmented, and gain amplitude values of at least one segment of the second audio data recorded are obtained and compared with the second preset amplitude values.

The beneficial effects of the embodiment are that the first preset amplitude value is set in the normal conversation scene, and the second preset amplitude value is set in the calibration setting scene; then, gain amplitude values of the first audio data of each recorded section are obtained and are compared with the first preset amplitude values successively, or gain amplitude values of the second audio data of at least one recorded section are obtained and are compared with the second preset amplitude values. A real-time segmentation comparison judgment basis is provided for realizing a humanized audio recording calibration scheme, 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 five

Fig. 7 is a flowchart of a fifth embodiment of an audio recording calibration method according to the present invention, based on the above embodiment, the step of performing an amplification process on the audio data if the gain amplitude of the audio data is smaller than the preset amplitude 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;

S22, if the gain amplitude of the first audio data of any section is 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 smaller than the second preset amplitude, performing amplification processing according to a second amplification proportion.

Alternatively, in the present embodiment, the first amplification ratio is obtained according to empirical data, or according to the number of amplification operations, for example, the greater the number of times, the higher the ratio, thereby rapidly improving the effect of calibration;

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, real-time amplification processing is performed according to a first amplification ratio, so that the effects of the voices of the segment and the subsequent segment can be effectively improved;

optionally, in this embodiment, in the static calibration process, if it is determined that the gain amplitude of at least one section 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 effect of the subsequent voice can be effectively improved.

The method has the advantages that a preset proportion value is determined to serve as a first amplification proportion in a normal conversation scene, and a preset proportion value is determined to serve as a second amplification proportion in a calibration setting scene; if the gain amplitude of the first audio data of any section is 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 smaller than the second preset amplitude, performing amplification processing according to a second amplification proportion. The dynamic and static differentiated proportional control foundation is provided for realizing a humanized audio recording calibration scheme, 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 six

Fig. 8 is a flowchart of a sixth embodiment of an audio recording calibration method according to the present invention, based on the above embodiment, the counting of the number of times the amplification processing operation is performed in the calibration period includes:

s31, accumulating the operation of executing the amplification processing in one first calibration period, and counting the accumulated times;

S32, accumulating the operations for executing the amplification processing in a plurality of second calibration periods, and counting the accumulated times.

Optionally, in this embodiment, the speech of the user of the device side is subjected to the segmentation analysis, where the segmentation scheme given in the foregoing embodiment is pause recognition of natural speech of the user, that is, when the user begins to hear the voice of the other party, the voice of the user of the previous section is taken as one end, and in consideration that the user may have a shorter conversation time in the course of a conversation, or when the speech of the user is short and the gain determination cannot be effectively and accurately performed, the speech of the user can be subjected to segmentation analysis according to a preset time period, for example, each time period is 2s, the time period is taken as a sub-period in a first calibration period, the operations for executing the amplification processing are accumulated, and the accumulated times are counted;

optionally, in this embodiment, in the static calibration process, the user may obtain corresponding multiple segments of speech under multiple simulated environmental sound scenes, 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;

alternatively, in the present embodiment, the number of times the test voice is subjected to the amplification processing is accumulated.

The method has the advantages that the operations for executing the amplification processing are accumulated in one first calibration period, and the accumulated times are counted; or, in a plurality of the second calibration periods, accumulating the operations of performing the amplification processing, and counting the accumulated times. The dynamic and static differentiated counting judgment basis is provided for realizing a humanized audio recording calibration scheme, 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 seven

Fig. 9 is a flowchart of a seventh embodiment of an audio recording calibration method according to the present invention, based on the above embodiment, the step of performing an amplification process on the preset gain value if the number of times exceeds a preset value, and taking the gain value after the amplification process operation as a fixed gain value for subsequently obtaining the audio data includes:

S41, in a first calibration period, if the times exceed a preset value, amplifying the preset gain value, and taking the gain value after the amplifying operation as a first fixed gain value of the audio data in the residual time of the first calibration period;

s42, in a plurality of second calibration periods, if the times exceed a preset value, amplifying the preset gain value, and taking the gain value after the amplifying operation as a second fixed gain value of the audio data in the next second calibration period.

Optionally, in this embodiment, it may be understood that the dynamic gain calibration is to perform gain and monitor comparison on the recorded data in real time during the use process, or dynamically raise the amplitude of the data frame, or dynamically adjust the data amplitude within a certain period of time;

alternatively, in this embodiment, it can be understood that the static gain calibration is to adjust the gain by using recorded data, which is slightly damaged or the pick-up hole of the microphone is partially blocked to affect the pick-up amplitude, and the adjusted gain is stored for a long period of time and is used as a fixed-mode analog or digital gain on each recording path in the future. Is a compensation for damage to the (microphone) components or structures of the terminal equipment.

The method has the advantages that in one first calibration period, if the times exceed a preset value, the preset gain value is subjected to amplification processing, and the gain value after the amplification processing operation is executed is used as a first fixed gain value of the audio data in the residual 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 the gain value after the amplification processing operation is performed as a second fixed gain value of the audio data in the next second calibration period. The dynamic and static differentiated gain compensation determination basis is provided for realizing a humanized audio recording calibration scheme, 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 eight

Fig. 10 is a flowchart of an eighth embodiment of an audio recording calibration method according to the present invention, based on the above embodiment, the step of performing an amplification process on the preset gain value if the number of times exceeds a preset value, and taking the gain value after the amplification process operation as a fixed gain value for subsequently obtaining the audio data, further includes:

S43, obtaining the first fixed gain value and the second fixed gain value according to the first calibration period and the implementation second calibration period respectively;

s44, superposing the first fixed gain value and the second fixed gain value to serve as a calibrated fixed gain value.

Optionally, in this embodiment, referring to fig. 11, a dynamic calibration flowchart of an eighth embodiment of the audio recording calibration method of the present invention is shown. In this procedure, firstly, recorded sound data is acquired through a microphone, and then, the sound data is compared with a preset sound gain amplitude: if the amplitude of the data frame is larger than the preset value, directly storing the data frame into a data buffer area without any processing, and at the moment, obtaining recording data or transmitting data required by noise elimination algorithm processing by the application 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, the count value of the data frame subjected to the amplification processing is increased by 1, the counting operation is circularly executed, whether the count of the data frame subjected to the amplification processing in the preset unit time is larger than the preset value is judged, if so, a fixed gain value of new recording data is set according to an actual result, and the count value of the data frame subjected to the amplification processing is requested to be counted again.

Alternatively, in this embodiment, reference is made to a static calibration flowchart of an eighth embodiment of the audio recording calibration method of the present invention shown in fig. 12. In the process, firstly, a sound with a fixed amplitude is actively played as a recording sound source through a setting or test item of the terminal equipment, then, the sound data is recorded through a recorded 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, directly storing the data frame into a data buffer area without any processing, and at the moment, obtaining recording data or transmitting data required by noise elimination algorithm processing by the application according to the requirement; if the data frame amplitude is smaller than the preset value, setting a fixed gain value of new recording data according to the actual result, wherein the calibration process can be circularly executed 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, the exemplary recorded sound source set or tested may likewise be in segmented form to achieve a calibration scheme in which the counts during dynamic calibration trigger a fixed gain value.

Optionally, in this embodiment, in order to be able to adapt to a wider microphone sensitivity range, it is ensured that the input sound amplitude meets the requirement of minimum noise reduction processing input amplitude, so that the input sound amplitude is not processed to be silent by the noise cancellation algorithm due to too small pick-up amplitude. For the situation that the pick-up amplitude is affected by slight damage or partial blockage of the pick-up hole of the microphone, a user can integrally lift the recorded data amplitude in a static data gain calibration mode. The service life of a mobile phone terminal (microphone) can be prolonged;

alternatively, in the present embodiment, dynamic calibration and static calibration may be alternately performed, and the effects of the calibration may be superimposed;

optionally, in this embodiment, after detecting that each preset number of dynamic calibrations is performed, the user is prompted to perform a static calibration, so that the user is timely found and reminded of a Guan Maike wind hardware failure problem.

The beneficial effects of this embodiment are that the first fixed gain value and the second fixed gain value are obtained by respectively 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 dynamic and static combined gain compensation determination basis is provided for realizing a humanized audio recording calibration scheme, 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 nine

Based on the above embodiments, the present invention also proposes an audio recording calibration apparatus, comprising a memory, a processor and a computer program stored on the memory and executable on the processor, the computer program implementing the steps of the audio recording calibration method as defined in any one of the above when being executed by the processor.

It should be noted that the above device embodiments and method embodiments belong to the same concept, the specific implementation process of the device embodiments is detailed in the method embodiments, and technical features in the method embodiments are correspondingly applicable to the device embodiments, which are not repeated herein.

Examples ten

Based on the above embodiments, the present invention further proposes a computer readable storage medium having stored thereon an audio recording calibration program, which when executed by a processor implements the steps of the audio recording calibration method as described in any of the above.

It should be noted that the medium embodiment and the method embodiment belong to the same concept, the specific implementation process of the medium embodiment and the method embodiment are detailed, and technical features in the method embodiment are correspondingly applicable in the medium embodiment, which is not repeated herein.

It should be noted that, in this document, the terms "comprises," "comprising," or any other variation thereof, are intended to cover a non-exclusive inclusion, such that a process, method, article, or apparatus that comprises a list of elements does not include only those elements but may include other elements not expressly listed or inherent to such process, method, article, or apparatus. Without further limitation, an element defined by the phrase "comprising one … …" does not exclude the presence of other like elements in a process, method, article, or apparatus that comprises the element.

The foregoing embodiment numbers of the present invention are merely for the purpose of description, and do not represent the advantages or disadvantages of the embodiments.

From the above description of the embodiments, it will be clear to those skilled in the art that the above-described embodiment method may be implemented by means of software plus a necessary general hardware platform, but of course may also be implemented by means of hardware, but in many cases the former is a preferred embodiment. Based on such understanding, the technical solution of the present invention may be embodied essentially or in a part contributing to the prior art in the form of a software product stored in a storage medium (e.g. ROM/RAM, magnetic disk, optical disk) comprising instructions for causing a terminal (which may be a mobile phone, a computer, a server, an air conditioner, or a network device, etc.) to perform the method according to the embodiments of the present invention.

The embodiments of the present invention have been described above with reference to the accompanying drawings, but the present invention is not limited to the above-described embodiments, which are merely illustrative and not restrictive, and many forms may be made by those having ordinary skill in the art without departing from the spirit of the present invention and the scope of the claims, which are to be protected by the present invention.

Claims (3)

1. A method of audio recording calibration, 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 or not;

if the gain amplitude of the audio data is smaller than the preset amplitude, amplifying the audio data;

counting the times of executing the amplification processing operation in the calibration period;

if the times exceeds a preset value, amplifying the preset gain value, and taking the gain value after the amplifying operation is executed as a fixed gain value for obtaining the audio data later;

the step of obtaining 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 or not comprises the following steps:

Determining a preset call time length as a first calibration period of dynamic calibration under a normal call scene;

determining a preset test duration as a second calibration period of static calibration in a calibration setting scene;

taking the fixed gain value after the previous calibration as the preset gain value;

acquiring first audio data of each section recorded in the scene of the normal call in the first calibration period, or acquiring second audio data of at least one section recorded in the scene of the calibration setting in the second calibration period;

setting a first preset amplitude value in the normal conversation scene, and setting a second preset amplitude value in the calibration setting scene;

gain amplitude values of the first audio data of each recorded section are obtained and are compared with the first preset amplitude values successively, or gain amplitude values of the second audio data of at least one recorded section are obtained and are compared with the second preset amplitude values;

if the gain amplitude of the audio data is smaller than the preset amplitude, the audio data is amplified, which comprises the following steps:

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;

If the gain amplitude of the first audio data of any section is 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 smaller than the second preset amplitude, performing amplification processing according to a second amplification proportion;

the counting the number of times of executing the amplification processing operation in the calibration period includes:

accumulating the operations for executing the amplification processing in one first calibration period, and counting the accumulated times;

or, in a plurality of second calibration periods, accumulating the operations for executing the amplification processing, and counting the accumulated times;

if the number of times exceeds a preset value, performing amplification processing on the preset gain value, and taking the gain value after the amplification processing operation is performed as a fixed gain value for subsequently acquiring the audio data, wherein the steps include:

in the 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 the amplification processing operation is performed as a first fixed gain value of the audio data in the residual time of the first calibration period;

In a plurality of second calibration periods, 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 is performed as a second fixed gain value of the audio data in the next second calibration period;

obtaining the first fixed gain value and the second fixed gain value according to the first calibration period and the implementation second calibration period respectively;

superposing the first fixed gain value and the second fixed gain value to be used as calibrated fixed gain values;

wherein,,

alternately performing the dynamic calibration and the static calibration, and superposing the calibration results of the dynamic calibration and the static calibration;

and prompting a user to perform the static calibration once after detecting the dynamic calibration performed for preset times.

2. An audio recording calibration apparatus comprising a memory, a processor and a computer program stored on the memory and executable on the processor, which when executed by the processor performs the steps of the audio recording calibration method of claim 1.

3. A computer readable storage medium, wherein an audio recording calibration program is stored on the computer readable storage medium, which when executed by a processor, implements the steps of the audio recording calibration method of claim 1.