WO2019033987A1

WO2019033987A1 - Prompting method and apparatus, storage medium, and terminal

Info

Publication number: WO2019033987A1
Application number: PCT/CN2018/099662
Authority: WO
Inventors: 许钊铵; 严锋贵; 甘高亭; 郑志勇; 杨海
Original assignee: Oppo广东移动通信有限公司
Priority date: 2017-08-18
Filing date: 2018-08-09
Publication date: 2019-02-21
Also published as: CN107580113B; CN107580113A

Abstract

A prompting method and apparatus, a storage medium, and a terminal. The method comprises: when it is detected that a pre-set call event or a pre-set voice recording event is triggered, controlling a microphone to acquire an audio signal; analyzing the sound loudness corresponding to the audio signal; and prompting, according to the analysis result, the user to adjust the call status or the voice recording status.

Description

Prompt method, device, storage medium and terminal

The present application claims the priority of the Chinese Patent Application, filed on Jan.

Technical field

The embodiments of the present application relate to the field of terminal technologies, for example, to a prompting method, device, storage medium, and terminal.

Background technique

In order to meet the needs of users for voice calls, video calls, and recordings in their lives and work, most terminals have call or recording functions, such as mobile phones, tablets, music players, and voice recorders.

Generally, the terminal implements a call or recording function through an integrated microphone (microphone, mic), also known as a microphone. A microphone is an energy conversion device that converts a sound signal into an electrical signal. When the vibration of the sound is transmitted to the diaphragm of the microphone, the magnet inside changes a current signal, and the current signal is processed by the sound processing circuit. Transfer to the opposite end of the call or store it for call or recording.

Summary of the invention

The embodiment of the present application provides a prompting method, device, storage medium, and terminal, which can automatically prompt the user to adjust the call or recording state during a call or recording process.

The embodiment of the present application provides a prompting method, including:

Controlling the microphone to collect audio signals when a preset call event is detected or a preset recording event is triggered;

Performing an analysis of the acoustic sound level corresponding to the audio signal;

According to the analysis result, the user is prompted to adjust the call state or the recording state.

The embodiment of the present application provides a prompting apparatus, including:

The audio signal collecting module is configured to control the microphone to collect the audio signal when the preset call event is detected or the preset recording event is triggered;

An acoustic sound analysis module configured to analyze an acoustic sound level corresponding to the audio signal;

The prompt module is set to prompt the user to adjust the call state or the recording state according to the analysis result.

The embodiment of the present application provides a computer readable storage medium, where the computer program stores a computer program, and when the program is executed by the processor, the prompting method as described in the embodiment of the present application is implemented.

The embodiment of the present application provides a terminal, including a microphone, a memory, a processor, and a computer program stored in the memory and executable by the processor, and the processor executes the computer program to implement the embodiment of the present application. Prompt method.

The embodiment of the present application further provides a computer program product, the computer program product comprising a computer program stored on a non-transitory computer readable storage medium, the computer program comprising program instructions, when the program instructions are executed by a computer The computer is caused to perform any of the above methods.

The prompting solution provided in the embodiment of the present application analyzes the sound and sound level corresponding to the audio signal collected by the microphone during the call or recording process, and prompts the user to adjust the call state or the recording state according to the analysis result. By adopting the above technical solution, during the call or recording process, the user can promptly and accurately prompt the user to adjust the call state or the recording state, and can maintain a good call effect or recording effect.

DRAWINGS

FIG. 1 is a schematic flow chart of a prompting method provided by an embodiment.

FIG. 2 is a schematic structural diagram of an audio processing hardware system of a smart phone according to an embodiment.

FIG. 3 is a block diagram of an audio system architecture provided by an embodiment.

FIG. 4 is a schematic flow chart of another prompting method provided by an embodiment.

FIG. 5 is a schematic flow chart of another prompting method provided by an embodiment.

FIG. 6 is a schematic flow chart of another prompting method provided by an embodiment.

FIG. 7 is a structural block diagram of a prompting apparatus according to an embodiment.

FIG. 8 is a schematic structural diagram of a terminal according to an embodiment.

Detailed ways

The technical solutions of the present application will be described below with reference to the accompanying drawings and specific embodiments. It is understood that the specific embodiments described herein are merely illustrative of the application and are not intended to be limiting. In addition, it should be noted that, for the convenience of description, only some but not all of the structures related to the present application are shown in the drawings.

Before discussing the exemplary embodiments in more detail, it should be noted that some exemplary embodiments are described as a process or method depicted as a flowchart. Although the flowchart depicts multiple steps as a sequential process, many of the steps can be implemented in parallel, concurrently, or concurrently. In addition, the order of each step can be rearranged. The process may be terminated when its operation is completed, but may also have additional steps not included in the figures. The processing may correspond to methods, functions, procedures, subroutines, subroutines, and the like.

FIG. 1 is a schematic flowchart diagram of a schematic diagram of a prompting method according to an embodiment of the present disclosure. The method may be performed by a prompting apparatus, where the apparatus may be implemented by software and/or hardware, and generally Integrated in the terminal. As shown in Figure 1, the method includes:

Step 110: When detecting a preset call event or a preset recording event is triggered, the control microphone collects an audio signal.

For example, the terminal in the embodiment of the present application may include a device configured with a microphone, such as a mobile phone, a tablet computer, a music player, and a voice recorder. The microphone can be either built-in or external. A microphone (referred to as a microphone, also known as a microphone or a microphone) is an energy conversion device that converts a sound signal into an electrical signal. When the vibration of the sound is transmitted to the diaphragm of the microphone, the magnet inside changes a current signal. The current signal is processed by the sound processing circuit and transmitted to the opposite end of the call or stored, thereby implementing a call or recording. The embodiment of the present application does not limit the type, the number, and the location of the microphone. For example, for a mobile phone, it may be at least one electret microphone disposed on the lower side of the mobile phone.

For example, the preset call event may be a call event having the prompt function in the embodiment of the present application; the preset recording event may be a recording event having the prompt function in the embodiment of the present application. In an embodiment, when a call event is detected when the prompt function is in an open state, it may be determined that a preset call event is detected or a recording event is detected when the prompt function is turned on, and it may be determined that the preset recording event is detected. The call event is, for example, a phone call or a voice chat call; the recording event is, for example, to start recording.

In order to facilitate the understanding of the embodiments of the present application, a smart phone is taken as an example to briefly introduce the audio processing hardware system and system architecture.

2 is a schematic structural diagram of an audio processing hardware system of a smart phone according to an embodiment of the present application. The audio processing circuit is generally in the main control board. The location of the audio processing circuit may be different due to the different designs of different mobile phones. The audio processing circuit of the smartphone includes an audio signal processing circuit, a baseband signal processing circuit, an audio power amplifier, a headphone signal amplifier, an earpiece, a speaker, a microphone, and a headphone interface. Among them, the audio signal processing circuit is the core of the entire audio processing circuit. The audio processing circuit is composed of a receiving audio circuit, a sending circuit, a headphone talking circuit, and the like, including analog/digital (A/D) conversion, digital/analog (A/D) conversion, digital voice signal processing, and analog audio amplification of analog audio. Circuits, etc.

During the call, the local microphone is first called to convert the sound acoustic signal of the sound into an analog audio signal, amplified by an analog audio amplifier circuit, and subjected to A/D conversion by an internal multimode converter to obtain a digital audio signal; The digital audio signal is sent to the baseband processor for speech encoding, channel encoding, etc.; a series of processing such as encryption and interleaving is performed again; finally, the digital narrowband modulation module sent to the baseband processor is modulated to generate the transmitted baseband signal. The RF circuit is modulated into a transmitting intermediate frequency and sent to the other party.

When recording, similar to the above process, the mechanical sound wave signal of the sound is first converted into an analog audio signal by a microphone, amplified by an analog audio amplifying circuit, and digital audio signal is obtained after A/D conversion, according to a preset audio format. Coding and storage.

The audio signal collected by the microphone in the embodiment of the present application may be the above-mentioned analog audio signal converted from the mechanical sound wave signal, or may be an amplified analog audio signal, or may be an A/D converted digital audio signal. The embodiment of the present application is not limited.

As shown in FIG. 3, the audio system architecture provided in this embodiment includes a user space, a kernel space, and a hardware system. The user space includes an application layer, an application framework layer, and a hardware abstraction layer (HAL), and the kernel space includes a driver layer. The application layer is the top layer of the audio system. You can write an application to perform corresponding logical operations, such as detecting an application that triggers a recording event, presetting standard audio conditions, and issuing audio playback commands. The application framework layer includes an audio control interface and a standardized plug-in module to provide an audio playback form control interface, as well as a speaker volume control interface. The application framework layer provides two classes, AudioTrack and AudioRecorder, as well as the AudioManager, AudioService, and AudioSystem classes. A system runtime layer (Libraries) is also included between the application framework layer and the hardware abstraction layer. We know that many classes in the framework layer are actually just "mediations" for applications that use Android library files. Because upper-level applications are generally written in Java, they require the most direct support of the Java interface, which is one of the meanings of the framework layer. As an "intermediary", they don't really implement specific functions, or only implement some of them, but focus on the library. For example, the above AudioTrack, AudioRecorder, MediaPlayer and MediaRecorder can find the corresponding class in the library. This part of the code is placed in the frameworks/av/media/libmedia of the project, mostly written in C++. The hardware abstraction layer of audio is mainly divided into two parts, namely AudioFlinger and AudioPolicyService. In fact, the latter is not a real device, just a virtual device to allow manufacturers to easily customize their own strategies. Depending on the product, audio devices vary widely. In the audio architecture of Android Android, these problems are solved by the audio.primary of the HAL layer, etc., without the need to modify the upper layer implementation on a large scale. The hardware abstraction layer is the transition from the application framework layer to the driver layer to achieve compatibility with the underlying hardware. The driver layer controls the audio codec according to the characteristics of the audio codec to ensure that the audio codec can work normally, and the audio data obtained by the audio codec is provided to the system layer. In the embodiment of the present application, when the audio signal is collected, the main class involved is the above-mentioned AudioRecorder class.

Step 120: Analyze the acoustic sound level corresponding to the audio signal.

The three main properties of the sound are volume, pitch, and tone. Among them, the volume is also called loudness or sound intensity, which refers to the subjective feeling of the human ear on the strength of the sound heard. The objective evaluation scale is the amplitude of the sound. Amplitude refers to the maximum distance from the original position during the vibration of the object. The loudness of the sound heard by the human ear is related to the amplitude of the sound source. Generally, the louder the louder the stronger the amplitude.

Exemplarily, taking the sound source as the user's mouth as an example, although the user can hear the sound of his own voice during the call, but cannot know whether the volume of the current speech is loud or small for the person at the opposite end of the call, if the sound is louder than Small, the other party may not be able to hear, if the sound is loud, it may make the other party feel deafening; the user can also hear the sound of his own voice when recording, but can not know the size of his own voice recorded by the terminal, if the sound is small It may not be audible when playing a recording. If the sound is loud, the listener may feel uncomfortable while playing the recording. In the embodiment of the present application, the sound level corresponding to the audio signal is analyzed to know whether the sound level corresponding to the audio signal collected by the terminal is appropriate.

For example, the preset time period may be an analysis unit, and the sound sound level in each preset time period may be analyzed, and the sound sound level corresponding to the current preset time period may be recorded as the current sound sound level. The preset time period may be a preset time length that is timed forward from the current time. In order to ensure real-time performance, the preset time length corresponding to the preset time period may be set shorter, for example, within 0.5 seconds of the current time as the starting point.

Step 130: prompt the user to adjust the call state or the recording state according to the analysis result.

Exemplarily, there are many factors affecting the sound level corresponding to the audio signal collected by the terminal microphone, wherein the factors related to the call state or the recording state can generally be determined by the user. For example, factors related to the state of the call or the state of the recording may include the size of the voice of the user's own voice or the intensity of the voice of other sound sources that the user wishes to record, and may also include a sound source (such as a user's mouth or other sound source) and a microphone. The relative distance may also include the orientation of the microphone and the like.

Exemplarily, the target sound level information of the audio signal that satisfies the requirement may be determined by the system by default or according to actual conditions, and the target sound level of the audio signal that satisfies the requirement may be determined by the user in advance according to the demand. After analyzing the sound intensity of the currently collected audio signal, the current sound level is compared with the target sound level, and then the user is prompted to make corresponding adjustments. For example, during the recording process, it is judged that the current sound level is smaller than the target sound level, and the user may be prompted to increase the intensity of the voice, or to bring the microphone to the mouth and the like.

In an embodiment, when the user is prompted according to the analysis result, the user may be reminded in the form of a text, a voice, or an indicator. For example, the description text of the adjustment mode may be displayed on the display screen; the prompt voice of the adjustment mode may be played, and the prompt voice may be filtered out in the audio signal collected by the microphone; different adjustment modes may be represented by different colors of the indicator light, Or different flashing frequencies with different indicators to indicate different adjustment methods, and so on.

The prompting method provided by the embodiment of the present invention analyzes the sound and sound degree corresponding to the audio signal collected by the microphone during the call or recording process, and prompts the user to adjust the call state or the recording state according to the analysis result. By adopting the above technical solution, during the call or recording process, the user can promptly and accurately prompt the user to adjust the call state or the recording state, and can maintain a good call effect or recording effect.

In some embodiments, the prompting the user to adjust the call state or the recording state according to the analysis result comprises: prompting the user to adjust the relative distance between the microphone and the sound source and/or the sounding state of the sound source according to the analysis result. The advantage of this setting is that the adjustment method is simple and fast, and the user can quickly respond according to the prompts, so that the adjustment can be quickly adjusted to the required state when the adjustment is needed, and the high requirements for the real-time performance of the call and the recording are satisfied.

In some embodiments, the analyzing the acoustic sound level corresponding to the audio signal may include: analyzing amplitude information of the audio signal to obtain an acoustic sound analysis result. As mentioned above, the objective evaluation scale of the acoustic sound is the amplitude of the sound, and the audio signal is converted from the mechanical acoustic signal of the sound, so the amplitude information can be used to analyze the acoustic sound of the audio signal. The larger the amplitude, the louder the sound, that is, the higher the sound energy value. For example, the preset time period may be an analysis unit, and the amplitude of the audio signal is collected at a preset sampling frequency within a preset time period to obtain a plurality of amplitude values, and the average value of the amplitude in the preset time period is used as the current sound level. Optimization here converts the analysis of acoustic sound into analysis of amplitude information, which simplifies the analysis process and speeds up the analysis.

In some embodiments, analyzing the amplitude information of the audio signal to obtain an acoustic sound analysis result, further comprising: extracting amplitude information of the corresponding human voice in the audio signal; and performing amplitude information of the corresponding human voice Analyze to obtain the results of the acoustic sound analysis. Illustratively, during a call, the vocal is the subject in the audio signal, and other ambient sounds can be regarded as interference sounds; in addition, during recording, the user can select the vocal recording mode, in which case the vocal is also an audio signal. In the main body, other environmental sounds can be regarded as interference sounds. In these cases, the sound source, that is, the user's mouth, after extracting the amplitude information corresponding to the human voice in the audio signal, can greatly reduce the data to be analyzed, thereby improving the analysis speed, and also prompting the user in a more timely manner, which is helpful. Improve the timeliness of adjustments for call status or recording status.

In some embodiments, prompting the user to adjust the relative distance between the microphone and the sound source according to the analysis result may include: prompting the user when the current sound intensity analysis result includes the current sound sound level being less than the first preset loudness threshold. The relative distance between the microphone and the sound source is reduced; the method further includes: prompting the user to compare the relative distance between the microphone and the sound source when the current sound intensity analysis result includes the current sound sound level being greater than the second preset loudness threshold turn up. Of course, the above two methods can also be included at the same time. The first preset loudness threshold and the second preset loudness threshold may be the same or different, and the value may be a preset fixed value or a dynamically adjusted change value according to actual conditions.

Exemplarily, when the current acoustic sound is less than the first preset loudness threshold, it may be stated that the current sound level in the audio signal is small, the opposite end of the call may not be able to hear the words spoken by the local user, or the recorded file may be caused during subsequent playback. Others cannot hear the sound of the recording at this time. Therefore, the user can be prompted to reduce the relative distance between the microphone and the sound source to improve the sound level in the audio signal collected next. Exemplarily, when the current sound level is greater than or equal to the first preset loudness threshold, the prompt may not be performed. The advantage of this setting is that it can be applied to an application scenario where the upper limit of the acoustic sound is not required, and the prompting efficiency is ensured.

Exemplarily, when the current sound level is greater than the second preset loudness threshold, the current sound level in the audio signal may be large, and the call to the opposite end may cause the other party to feel deafening, or the recorded file may cause subsequent playback. Excessive sound affects listening, so the user can be prompted to increase the relative distance between the microphone and the sound source to reduce the sound level in the audio signal collected next. Illustratively, when the current acoustic level is less than or equal to the second preset loudness threshold, no prompt may be made. The advantage of this setting is that it can be applied to an application scenario where the lower limit of the acoustic sound is not required, and the prompting efficiency is ensured.

For example, the above two prompt modes may be combined, that is, when the current sound level is small and when the sound is large, the reminder is performed, and the current sound level is greater than or equal to the first preset loudness threshold and less than or equal to the second preset. When the loudness threshold is used, it indicates that the sound level in the audio signal is appropriate, and no prompt is given. The advantage of this setting is that it can be applied to the application scenarios that require the upper and lower limits of the acoustic sound level to ensure the accuracy of the prompts.

In some embodiments, prompting the user to adjust the sounding condition of the sound source according to the analysis result may include: prompting the user to sound source when the sound sound intensity analysis result includes the current sound sound level being less than the third preset loudness threshold. The vocal intensity is increased; the method further includes: prompting the user to reduce the vocal intensity of the sound source when the current sound intensity analysis result includes the current sound loudness being greater than the fourth preset loudness threshold. The third preset loudness threshold and the fourth preset loudness threshold may be the same or different. The third preset loudness threshold and the first preset loudness threshold may be the same or different. The fourth preset loudness threshold and the second preset loudness threshold may be the same or different. The value of the third preset loudness threshold and the fourth preset loudness threshold may be a preset fixed value, or may be a dynamically adjusted change value according to actual conditions.

Exemplarily, when the current sound level is less than the third preset loudness threshold, it may be stated that the current sound level in the audio signal is small, the opposite end of the call may not be able to hear the words spoken by the local user, or the recorded file may be caused during subsequent playback. Others cannot hear the sound of the recording at this time. Therefore, the user can be prompted to reduce the relative distance between the microphone and the sound source to improve the sound level in the audio signal collected next. Exemplarily, when the current sound level is greater than or equal to the third preset loudness threshold, no prompt may be made. The advantage of this setting is that it can be applied to an application scenario where the upper limit of the acoustic sound is not required, and the prompting efficiency is ensured.

Exemplarily, when the current sound level is greater than the fourth preset loudness threshold, it may indicate that the current sound level in the audio signal is large, and the call to the opposite end may cause the other party to feel deafening, or the recording file may cause subsequent playback. Excessive sound affects listening, so the user can be prompted to increase the relative distance between the microphone and the sound source to reduce the sound level in the audio signal collected next. Illustratively, when the current acoustic level is less than or equal to the fourth preset loudness threshold, no prompt may be made. The advantage of this setting is that it can be applied to an application scenario where the lower limit of the acoustic sound is not required, and the prompting efficiency is ensured.

For example, the above two adjustment modes may be combined, that is, when the current sound level is small and when the sound is large, the reminder is performed, and the current sound level is greater than or equal to the third preset loudness threshold and less than or equal to the fourth preset. When the loudness threshold is used, it indicates that the sound level in the audio signal is appropriate, and no prompt is given. The advantage of this setting is that it can be applied to the application scenarios that require the upper and lower limits of the acoustic sound level to ensure the accuracy of the prompts.

In an embodiment, the user may also be prompted to adjust the relative distance between the microphone and the sound source and the sounding of the sound source. Illustratively, when the current acoustic sound analysis result includes the current acoustic soundness being less than the fifth preset loudness threshold, prompting the user to reduce the relative distance between the microphone and the sound source and increase the sound intensity of the sound source; and/or And when the current acoustic sound analysis result includes the current sound loudness greater than the sixth preset loudness threshold, prompting the user to increase the relative distance between the microphone and the sound source and to reduce the sound intensity of the sound source. The advantage of this setting is that it can quickly adjust the call state or recording state to the ideal state.

In some embodiments, the user may be more specifically prompted based on the difference (eg, the difference) between the current acoustic level and the corresponding preset loudness threshold. Exemplarily, different difference values correspond to different adjustment values, and the correspondence relationship may be determined in advance according to a simulation or the like. For example, the difference between the current acoustic sound level and the first preset loudness threshold is a, and the corresponding relative distance adjustment value is 5 cm, prompting the user to reduce the relative distance between the microphone and the sound source by 5 cm. The advantage of this setting is that the prompts to the user are more clear, and the user can quickly adjust the call state or the recording state to the ideal state.

In some embodiments, when the triggered event is a preset call event, the prompting the user to adjust the relative distance between the microphone and the sound source according to the analysis result may further include: acquiring an attribute of the call peer contact Information mode and/or profile mode information of the call end; determining a corresponding first preset loudness threshold and/or second preset loudness threshold according to the attribute information and/or the scene mode information. The advantage of this setting is that the first preset loudness threshold and/or the second preset loudness threshold can be determined more accurately according to actual conditions, thereby improving the call effect.

Illustratively, the attribute information may include age (or age group), or include whether it is an elderly person or a child or the like. In an embodiment, obtaining the attribute information of the call peer contact may include: obtaining the note information of the call peer contact in the address book, extracting the attribute information from the note information, and may also include contacting the call end. The human voice performs speech recognition, and the corresponding attribute information is determined according to the recognition result. When the other party is an elderly person, the hearing may be poor, and the required sound level should be larger, so the first preset loudness threshold and/or the second preset loudness threshold may be set higher.

Exemplarily, the scene mode information of the call end may include mode information such as a silent mode, a conference mode, a normal mode, and an outdoor mode. Users usually set the corresponding scene according to their environment. For example, in a relatively quiet environment such as class or meeting, you may choose silent mode or conference mode. In a noisy environment outside, you may choose outdoor mode. When answering calls in these environments, there may be different demands on the sound level. For example, in the conference mode, since the environment is relatively quiet, the sound level does not have to be large enough to hear the other party's voice; while in the outdoor mode, because the environment is noisy, a relatively large sound level is required to hear the other party's voice. Based on the above reasons, the corresponding relationship between the different preset modes and the first preset loudness threshold and/or the second preset loudness threshold may be preset, and in the actual call, the acquired context mode information may be determined. A corresponding first preset loudness threshold and/or second preset loudness threshold. In an embodiment, the acquiring the scenario mode information of the call peer may include: sending a scenario mode information acquisition request to the call peer end, and receiving the scenario mode information that the call peer end requests the feedback according to the scenario mode information acquisition request.

In some embodiments, when the triggered event is a preset call event, the prompting the user to adjust the utterance of the sound source according to the analysis result, further comprising: obtaining attribute information and/or a call of the call peer contact The context mode information of the peer end; determining a corresponding third preset loudness threshold and/or fourth preset loudness threshold according to the attribute information and/or the scene mode information. For details of implementation, reference may be made to the above for determining relevant content of the corresponding first preset loudness threshold and/or second preset loudness threshold according to the attribute information and/or the scene mode information. The advantage of this setting is that the third preset loudness threshold and/or the fourth preset loudness threshold can be determined more accurately according to actual conditions, thereby improving the call effect.

FIG. 4 is a schematic flowchart of another prompting method according to an embodiment of the present application, where the method is applicable to a recording scenario, including:

Step 401: When it is detected that the preset recording event is triggered, the control microphone collects the audio signal.

Step 402: Perform real-time analysis on amplitude information of the collected audio signal to obtain an acoustic sound analysis result.

Step 403: Determine whether the current sound level is less than the preset loudness threshold A. If the current sound level is less than the preset loudness threshold A, perform step 404; if the current sound level is greater than or equal to the preset loudness threshold A, perform step 405.

It can be understood that as the recording progresses, the terminal analyzes the amplitude information of the audio signal collected by the microphone in real time, and the current acoustic sound level also changes continuously as the analysis progresses.

Step 404, prompting the user to reduce the relative distance between the microphone and the sound source or prompting the user to increase the sound intensity of the sound source.

In step 405, it is determined whether the current sound level is greater than the preset loudness threshold B. If the current sound level is greater than the preset loudness threshold B, step 406 is performed; if the current sound level is less than or equal to the preset loudness threshold B, step 407 is performed.

Step 406, prompting the user to increase the relative distance between the microphone and the sound source or prompting the user to reduce the sound intensity of the sound source.

Step 407: Determine whether a recording pause or recording stop command is received, and if the recording pause or recording stop command is received, the process ends; if the recording pause or recording stop command is not received, the process returns to step 403.

The prompting method provided by the embodiment of the present application can accurately prompt the user to adjust the relative distance between the microphone and the sound source or the sounding intensity of the sound source during the recording process, and can maintain a good recording effect.

FIG. 5 is a schematic flowchart of another prompting method according to an embodiment of the present disclosure. The method is applicable to a call scenario, and includes:

Step 501: When it is detected that the preset call event is triggered, the control microphone collects the audio signal.

Step 502: Analyze amplitude information of the collected audio signal in real time to obtain an acoustic sound analysis result.

In step 503, it is determined whether the current sound level is less than the preset loudness threshold C. If the current sound level is less than the preset loudness threshold C, step 504 is performed; if the current sound level is greater than or equal to the preset loudness threshold C, step 505 is performed.

It can be understood that as the call progresses, the terminal analyzes the amplitude information of the audio signal collected by the microphone in real time, and the current acoustic sound level also changes continuously as the analysis progresses.

Step 504, prompting the user to reduce the relative distance between the microphone and the sound source or prompting the user to increase the sound intensity of the sound source.

Generally, the sound source during the call is the user's mouth, and the adjustment of the sound intensity of the sound source means that the user adjusts the size of his own voice.

In an embodiment, when the call is a call made through a mobile phone, the user can be prompted in a voice form, which can be more easily perceived by the user. In an embodiment, the voice prompt is played through the earpiece to prompt the user to reduce the relative distance between the microphone and the sound source or to prompt the user to increase the sound intensity of the sound source. For example, play "Please be closer to the microphone" or "Please increase the volume of the voice" through the handset.

Step 505: Determine whether the current sound level is greater than the preset loudness threshold D. If the current sound level is greater than the preset loudness threshold D, perform step 506; if the current sound level is less than or equal to the preset loudness threshold D, perform step 507.

Step 506, prompting the user to increase the relative distance between the microphone and the sound source or prompting the user to reduce the sound intensity of the sound source.

Step 507: Determine whether a call end command is received. If the call end command is received, the process ends. If the call end command is not received, the process returns to step 503.

The prompting method provided by the embodiment of the present application can accurately prompt the user to adjust the relative distance of the microphone and the mouth or the size of the speaking voice during the call, and can maintain a good call effect.

FIG. 6 is a schematic flowchart of another prompting method according to an embodiment of the present disclosure. The method is applicable to a call scenario, and includes:

Step 601: When it is detected that the preset call event is triggered, the control microphone collects the audio signal.

Step 602: Extract amplitude information of the corresponding human voice in the audio signal in real time, and analyze amplitude information of the corresponding human voice to obtain an acoustic sound analysis result.

Step 603: Acquire context mode information of the opposite end of the call, and determine a corresponding preset loudness threshold E and a preset loudness threshold F according to the scene mode information.

In an embodiment, in order to accurately determine the preset loudness threshold E and the preset loudness threshold F, the attribute information of the opposite end of the call may also be determined. For example, when the scene mode is the outdoor mode, the attribute information is young, the preset loudness threshold is determined as E ₁ ; when the scene mode is the outdoor mode, and the attribute information is the elderly, the preset loudness threshold is determined as E ₂ ; _{2 is} greater than E ₁ .

In step 604, it is determined whether the current sound level is less than the preset loudness threshold E. If the current sound level is less than the preset loudness threshold E, step 605 is performed; if the current sound level is greater than or equal to the preset loudness threshold E, step 606 is performed.

It can be understood that the current sound level corresponds to the loudness of the user's speech sound.

Step 605, prompting the user to reduce the relative distance between the microphone and the sound source or prompting the user to increase the sound intensity of the sound source.

Step 606: Determine whether the current sound level is greater than the preset loudness threshold F. If the current sound level is greater than the preset loudness threshold F, perform step 607; if the current sound level is less than or equal to the preset loudness threshold F, perform step 608.

Step 607: prompt the user to increase the relative distance between the microphone and the sound source or prompt the user to reduce the sound intensity of the sound source.

Step 608: Determine whether a call end command is received. If the call end command is received, the process ends; if the call end command is not received, the process returns to step 604.

The prompting method provided by the embodiment of the present invention can promptly prompt the user to adjust the relative distance between the microphone and the mouth or the size of the speaking voice according to the scene mode information of the calling party during the call, and can improve the call effect more specifically.

FIG. 7 is a structural block diagram of a prompting apparatus according to an embodiment of the present disclosure. The apparatus may be implemented by software and/or hardware, and is generally integrated in a terminal, and may prompt the user by executing a prompting method. As shown in Figure 7, the device includes:

The audio signal collection module 701 is configured to control the microphone to collect an audio signal when a preset call event is detected or a preset recording event is triggered;

The acoustic sound analysis module 702 is configured to analyze the sound intensity corresponding to the audio signal;

The prompting module 703 is configured to prompt the user to adjust the call state or the recording state according to the analysis result.

The prompting device provided by the embodiment of the present invention can promptly and promptly prompt the user to adjust the call state or the recording state during the call or recording process, and can maintain a good call effect or a recording effect.

In an embodiment, the prompting module 703 is configured to:

When the triggered event is a preset call event or a preset recording event, the user is prompted to adjust the relative distance of the microphone and the sound source and/or the sounding of the sound source according to the analysis result.

In an embodiment, the acoustic sound analysis module 702 is configured to:

The amplitude information of the audio signal is analyzed to obtain an acoustic sound analysis result.

In an embodiment, the acoustic sound analysis module 702 is configured to:

Extracting amplitude information of the corresponding human voice in the audio signal;

The amplitude information of the corresponding human voice is analyzed to obtain an acoustic sound analysis result.

In an embodiment, the prompting module 703 is configured to include:

When the sound intensity analysis result includes that the current sound level is less than the first preset loudness threshold, prompting the user to reduce the relative distance between the microphone and the sound source; and/or,

When the sound sound analysis result includes the current sound loudness greater than the second preset loudness threshold, the user is prompted to increase the relative distance between the microphone and the sound source.

In an embodiment, the prompting module 703 further includes:

The first obtaining submodule is configured to obtain attribute information of the call peer contact and/or context mode information of the call peer end;

The first determining submodule is configured to determine a corresponding first preset loudness threshold and/or second preset loudness threshold according to the attribute information and/or the scene mode information.

In an embodiment, the prompting module 703 is configured to:

When the sound intensity analysis result includes that the current sound level is less than the third preset loudness threshold, prompting the user to increase the sound intensity of the sound source; and/or,

When the sound sound analysis result includes the current sound loudness greater than the fourth preset loudness threshold, the user is prompted to reduce the sound intensity of the sound source.

In an embodiment, the prompting module 703 further includes:

The second obtaining submodule is configured to obtain attribute information of the call peer contact and/or context mode information of the call peer end;

The second determining submodule is configured to determine a corresponding third preset loudness threshold and/or fourth preset loudness threshold according to the attribute information and/or the scene mode information.

The embodiment of the present application further provides a storage medium including computer executable instructions for executing a prompting method when executed by a computer processor, the method comprising:

And analyzing an acoustic sound level corresponding to the audio signal;

Storage media - any different type of storage device or storage device. The term "storage medium" is intended to include: a mounting medium such as a CD-ROM, a floppy disk or a tape device; a computer system memory or a random access memory (RAM) such as a dynamic random access memory (dynamic random access memory). DRAM), display data random access memory (DDR RAM), static random access memory (SRAM), extended data output random access memory (EDO RAM) , Rambus RAM, etc.; non-volatile memory, such as flash memory, magnetic media (such as hard disk or optical storage); registers or other similar types of memory components, and the like. The storage medium may also include other types of memory or a combination thereof. Additionally, the storage medium may be located in a first computer system in which the program is executed, or may be located in a different second computer system, the second computer system being coupled to the first computer system via a network, such as the Internet. The second computer system can provide program instructions to the first computer for execution. The term "storage medium" can include at least two storage mediums that can reside in different locations (eg, in different computer systems connected through a network). The storage medium may store program instructions (eg, implemented as a computer program) executable by the at least one processor.

Of course, the storage medium containing the computer executable instructions provided by the embodiment of the present application is not limited to the prompt operation as described above, and may also be related to the prompting method provided by any embodiment of the present application. operating.

The embodiment provides a terminal in which the prompting device provided in this embodiment can be integrated. FIG. 8 is a schematic structural diagram of a terminal according to an embodiment of the present disclosure. The terminal may be, for example, a mobile terminal. As shown in FIG. 8, the terminal may include a casing (not shown), a memory 801, and a central processing unit (CPU) 802 (also referred to as a processor, below). Referred to as CPU), a circuit board (not shown), a power supply circuit (not shown), and a microphone 813. The circuit board is disposed inside a space enclosed by the casing; the CPU 802 and the memory 801 are disposed on the circuit board; and the power circuit is configured to supply power to a plurality of circuits or devices of the terminal The memory 801 is configured to store executable program code; the CPU 802 runs a computer program corresponding to the executable program code by reading executable program code stored in the memory 801 to implement the following steps:

And analyzing an acoustic sound level corresponding to the audio signal;

The terminal further includes: a peripheral interface 803, a radio frequency (RF) circuit 805, an audio circuit 806, a speaker 811, a power management chip 808, other input/output (I/O) subsystem input/control devices, and a touch screen. 812, other input/control devices 810 and external port 804, these components communicate via at least one communication bus or signal line 807.

It should be understood that the illustrated terminal 800 is merely one example of a terminal, and that the terminal 800 may have more or fewer components than those shown in the figures, may combine at least two components, or may have different Component configuration. The various components shown in the figures can be implemented in hardware, software, or a combination of hardware and software, including at least one signal processing and/or application specific integrated circuit.

The following is a detailed description of the terminal for prompting the user provided in this embodiment. The terminal uses a mobile phone as an example.

The memory 801 can be accessed by the CPU 802, the peripheral interface 803, etc., and the memory 801 can include a high speed random access memory, and can also include a non-volatile memory, such as one or more magnetic disk storage devices, flash memory devices. Or other volatile solid-state storage devices.

Peripheral interface 803, which can connect the input and output peripherals of the device to CPU 802 and memory 801.

I/O subsystem 809, which can connect input and output peripherals on the device, such as touch screen 812 and other input/control devices 810, to peripheral interface 803. The I/O subsystem 809 can include a display controller 8091 and at least one input controller 8092 configured to control other input/control devices 810. Wherein, at least one input controller 8092 receives electrical signals from other input/control devices 810 or transmits electrical signals to other input/control devices 810, and other input/control devices 810 may include physical buttons (press buttons, rocker buttons, etc.), Dial, slide switch, joystick, click wheel. It is worth noting that the input controller 8092 can be connected to any of the following: a keyboard, an infrared port, a USB interface, and a pointing device such as a mouse.

The touch screen 812 is an input interface and an output interface between the user terminal and the user, and displays the visual output to the user. The visual output may include graphics, text, icons, videos, and the like.

Display controller 8091 in I/O subsystem 809 receives an electrical signal from touch screen 812 or an electrical signal to touch screen 812. The touch screen 812 detects the contact on the touch screen, and the display controller 8091 converts the detected contact into an interaction with the user interface object displayed on the touch screen 812, that is, realizes human-computer interaction, and the user interface object displayed on the touch screen 812 may be running. The icon of the game, the icon of the network to the corresponding network, and the like. It is worth noting that the device may also include a light mouse, which is a touch sensitive surface that does not display a visual output, or an extension of a touch sensitive surface formed by the touch screen.

The RF circuit 805 is configured to establish communication between the mobile phone and the wireless network (ie, the network side) to implement data reception and transmission between the mobile phone and the wireless network. For example, sending and receiving short messages, emails, and the like. The RF circuit 805 receives and transmits an RF signal, also referred to as an electromagnetic signal, and the RF circuit 805 converts the electrical signal into an electromagnetic signal or converts the electromagnetic signal into an electrical signal and communicates with the communication network and other devices through the electromagnetic signal. The RF circuit 805 can include known circuitry configured to perform these functions including, but not limited to, an antenna system, an RF transceiver, at least one amplifier, a tuner, at least one oscillator, a digital signal processor, a codec (COder- DECoder, CODEC) Chipset, Subscriber Identity Module (SIM), etc.

The audio circuit 806 is arranged to receive audio data from the peripheral interface 803, convert the audio data into an electrical signal, and transmit the electrical signal to the speaker 811.

The speaker 811 is arranged to restore the voice signal received by the mobile phone from the wireless network through the RF circuit 805 to sound and play the sound to the user.

The power management chip 808 is configured to provide power and power management for the hardware connected to the CPU 802, the I/O subsystem, and the peripheral interface.

The terminal provided by the embodiment of the present invention can promptly and promptly prompt the user to adjust the call state or the recording state during the call or recording process, and can maintain a good call effect or a recording effect.

The embodiment of the present application further provides a computer program product, the computer program product comprising a computer program stored on a non-transitory computer readable storage medium, the computer program comprising program instructions, when the program instructions are executed by a computer The computer is caused to perform any of the above-described prompting methods.

The prompting device, the storage medium and the terminal provided in the above embodiments can perform the prompting method provided by any embodiment of the present application, and have the corresponding functional modules and beneficial effects of executing the method. For the technical details that are not described in detail in the above embodiments, reference may be made to the prompting method provided by any embodiment of the present application.

Claims

A prompting method, including:

Controlling the microphone to collect audio signals when a preset call event is detected or a preset recording event is triggered;

Performing an analysis of the acoustic sound level corresponding to the audio signal;

According to the analysis result, the user is prompted to adjust the call state or the recording state.
The method according to claim 1, wherein the prompting the user to adjust the call state or the recording state according to the analysis result comprises:

According to the analysis result, the user is prompted to adjust at least one of the following: the relative distance between the microphone and the sound source and the sounding of the sound source.
The method of claim 2, wherein the analyzing the acoustic sound level corresponding to the audio signal comprises:

The amplitude information of the audio signal is analyzed to obtain an acoustic sound analysis result.
The method according to claim 3, wherein analyzing the amplitude information of the audio signal to obtain an acoustic sound analysis result comprises:

Extracting amplitude information of the corresponding human voice in the audio signal;

The amplitude information of the corresponding human voice is analyzed to obtain an acoustic sound analysis result.
The method according to claim 3 or 4, wherein the user is prompted to adjust the relative distance between the microphone and the sound source according to the analysis result, including at least one of the following:

When the acoustic sound analysis result includes that the current sound sensitivity is less than the first preset loudness threshold, prompting the user to reduce the relative distance between the microphone and the sound source; and

When the acoustic sound analysis result includes the current sound loudness being greater than the second preset loudness threshold, the user is prompted to increase the relative distance between the microphone and the sound source.
The method of claim 5, wherein when the triggered event is a preset call event, the prompting the user to adjust the relative distance between the microphone and the sound source according to the analysis result further includes:

Obtaining at least one of the following information: attribute information of the call peer contact and profile mode information of the call peer;

Determining at least one of the following thresholds based on the acquired information: a first preset loudness threshold and a second preset loudness threshold.
The method according to claim 3 or 4, wherein the user is prompted to adjust the utterance of the sound source according to the analysis result, including at least one of the following:

When the sound intensity analysis result includes that the current sound level is less than the third preset loudness threshold, prompting the user to increase the sound intensity of the sound source; and

When the sound intensity analysis result includes that the current sound level is greater than the fourth preset loudness threshold, the user is prompted to reduce the sound intensity of the sound source.
The method of claim 7, wherein when the triggered event is a preset call event, the prompting the user to adjust the sounding of the sound source according to the analysis result further includes:

Obtaining at least one of the following information: attribute information of the call peer contact and profile mode information of the call peer;

Determining at least one of the following thresholds based on the acquired information: a third preset loudness threshold and a fourth preset loudness threshold.
The method according to claim 1, wherein when the triggered event is a preset call event, the user is prompted to adjust the call state according to the analysis result, including:

When the triggered event is a preset call event, the user is prompted to adjust the call state according to the analysis result in the form of text, voice, or indicator light.
The method according to claim 1, wherein when the triggered event is a preset recording event, the user is prompted to adjust the recording state according to the analysis result, including:

When the triggered event is a preset recording event, the user is prompted to adjust the recording state according to the analysis result in the form of text, voice, or indicator light.
A prompting device comprising:

The audio signal collecting module is configured to control the microphone to collect the audio signal when the preset call event is detected or the preset recording event is triggered;

An acoustic sound analysis module configured to analyze an acoustic sound level corresponding to the audio signal;

The prompt module is set to prompt the user to adjust the call state or the recording state according to the analysis result.
The apparatus of claim 11 wherein said prompting module is configured to:

When the triggered event is a preset call event or a preset recording event, the user is prompted according to the analysis result to adjust at least one of the following: the relative distance between the microphone and the sound source and the sounding condition of the sound source.
The apparatus of claim 12 wherein said acoustical sound analysis module is configured to:

The amplitude information of the audio signal is analyzed to obtain an acoustic sound analysis result.
The apparatus of claim 13 wherein said acoustical sound analysis module is configured to:

Extracting amplitude information of the corresponding human voice in the audio signal;

The amplitude information of the corresponding human voice is analyzed to obtain an acoustic sound analysis result.
The apparatus according to claim 13 or 14, wherein said prompting module is arranged to perform at least one of the following:

When the acoustic sound analysis result includes that the current sound sensitivity is less than the first preset loudness threshold, prompting the user to reduce the relative distance between the microphone and the sound source; and

When the acoustic sound analysis result includes the current sound loudness being greater than the second preset loudness threshold, the user is prompted to increase the relative distance between the microphone and the sound source.
The apparatus according to claim 15, wherein the prompting module further comprises:

The first obtaining submodule is configured to obtain at least one of the following information: attribute information of the call peer contact and context mode information of the call peer end;

The first determining submodule is configured to determine at least one of the following thresholds according to the acquired information: a first preset loudness threshold and a second preset loudness threshold.
The apparatus according to claim 13 or 14, wherein said prompting module is arranged to perform at least one of the following:

When the sound intensity analysis result includes that the current sound level is less than the third preset loudness threshold, prompting the user to increase the sound intensity of the sound source; and

When the sound intensity analysis result includes that the current sound level is greater than the fourth preset loudness threshold, the user is prompted to reduce the sound intensity of the sound source.
The apparatus according to claim 17, wherein the prompting module further comprises:

The second obtaining submodule is configured to obtain at least one of the following information: attribute information of the call peer contact and context mode information of the call peer end;

The second determining submodule is configured to determine at least one of the following according to the acquired information: a third preset loudness threshold and a fourth preset loudness threshold.
A computer readable storage medium storing a computer program, the program being executed by a processor to implement the prompting method according to any one of claims 1-10.
A terminal comprising a microphone, a memory, a processor, and a computer program stored on the memory and operable by the processor, the processor executing the computer program to implement the prompting method according to any one of claims 1-10 .