WO2019033438A1 - 音频信号调节方法、装置、存储介质及终端 - Google Patents

音频信号调节方法、装置、存储介质及终端 Download PDF

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Publication number
WO2019033438A1
WO2019033438A1 PCT/CN2017/098173 CN2017098173W WO2019033438A1 WO 2019033438 A1 WO2019033438 A1 WO 2019033438A1 CN 2017098173 W CN2017098173 W CN 2017098173W WO 2019033438 A1 WO2019033438 A1 WO 2019033438A1
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WIPO (PCT)
Prior art keywords
preset
audio signal
sound
current
amplitude
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PCT/CN2017/098173
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English (en)
French (fr)
Inventor
许钊铵
严锋贵
甘高亭
郑志勇
杨海
Original Assignee
广东欧珀移动通信有限公司
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Application filed by 广东欧珀移动通信有限公司 filed Critical 广东欧珀移动通信有限公司
Priority to CN201780092232.6A priority Critical patent/CN110870201B/zh
Priority to EP17922059.5A priority patent/EP3664291A4/en
Priority to PCT/CN2017/098173 priority patent/WO2019033438A1/zh
Publication of WO2019033438A1 publication Critical patent/WO2019033438A1/zh
Priority to US16/790,991 priority patent/US11251763B2/en

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    • H—ELECTRICITY
    • H03—ELECTRONIC CIRCUITRY
    • H03G—CONTROL OF AMPLIFICATION
    • H03G3/00—Gain control in amplifiers or frequency changers
    • H03G3/20—Automatic control
    • H03G3/30—Automatic control in amplifiers having semiconductor devices
    • H03G3/3005—Automatic control in amplifiers having semiconductor devices in amplifiers suitable for low-frequencies, e.g. audio amplifiers
    • G—PHYSICS
    • G06—COMPUTING; CALCULATING OR COUNTING
    • G06F—ELECTRIC DIGITAL DATA PROCESSING
    • G06F3/00—Input arrangements for transferring data to be processed into a form capable of being handled by the computer; Output arrangements for transferring data from processing unit to output unit, e.g. interface arrangements
    • G06F3/16—Sound input; Sound output
    • G06F3/165—Management of the audio stream, e.g. setting of volume, audio stream path
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L25/00—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00
    • G10L25/48—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00 specially adapted for particular use
    • G10L25/51—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00 specially adapted for particular use for comparison or discrimination
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R29/00—Monitoring arrangements; Testing arrangements
    • H04R29/001—Monitoring arrangements; Testing arrangements for loudspeakers
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00—Circuits for transducers, loudspeakers or microphones
    • H—ELECTRICITY
    • H03—ELECTRONIC CIRCUITRY
    • H03G—CONTROL OF AMPLIFICATION
    • H03G2201/00—Indexing scheme relating to subclass H03G
    • H03G2201/10—Gain control characterised by the type of controlled element
    • H03G2201/103—Gain control characterised by the type of controlled element being an amplifying element
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2430/00—Signal processing covered by H04R, not provided for in its groups
    • H04R2430/01—Aspects of volume control, not necessarily automatic, in sound systems

Definitions

  • Embodiments of the present invention relate to the field of audio processing technologies, and in particular, to an audio signal adjustment method, apparatus, storage medium, and terminal.
  • call or recording functions such as mobile phones, tablets, music players, and voice recorders.
  • the terminal implements a call or recording function through an integrated microphone (referred to as a 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.
  • Embodiments of the present invention provide an audio signal adjustment method, apparatus, storage medium, and terminal, which can automatically adjust an input audio signal during a call or recording process.
  • an embodiment of the present invention provides an audio signal adjustment method, including:
  • the control microphone collects an audio signal, and the preset event includes a preset call event and/or a preset recording event;
  • an embodiment of the present invention provides an audio signal adjusting apparatus, including:
  • the audio signal collecting module is configured to control the microphone to collect an audio signal when the preset event is detected, where the preset event includes a preset call event and/or a preset recording event;
  • a loudness analysis module configured to analyze an acoustic sound level corresponding to the audio signal
  • a loudness adjustment module configured to correspond to the audio signal according to the analysis result and a preset adjustment policy The sound level is dynamically adjusted
  • a signal output module configured to perform corresponding output processing on the adjusted audio signal according to the type of the preset event.
  • an embodiment of the present invention provides a computer readable storage medium, on which a computer program is stored, which is implemented by a processor to implement audio signal adjustment according to an embodiment of the present invention.
  • an embodiment of the present invention provides a mobile terminal, including a microphone, a memory, a processor, and a computer program stored in the memory and executable by the processor, where the processor implements the computer program The audio signal adjustment method described in the embodiment of the invention.
  • FIG. 1 is a schematic flowchart of an audio signal adjustment method according to an embodiment of the present invention.
  • FIG. 2 is a schematic structural diagram of an audio processing hardware system of a smart phone according to an embodiment of the present invention
  • FIG. 3 is a block diagram of an audio system architecture according to an embodiment of the present invention.
  • FIG. 4 is a schematic flowchart of still another method for adjusting an audio signal according to an embodiment of the present invention.
  • FIG. 5 is a schematic flowchart diagram of still another method for adjusting an audio signal according to an embodiment of the present disclosure
  • FIG. 6 is a schematic flowchart diagram of still another method for adjusting an audio signal according to an embodiment of the present disclosure
  • FIG. 7 is a structural block diagram of an audio signal adjusting apparatus according to an embodiment of the present invention.
  • FIG. 8 is a schematic structural diagram of a terminal according to an embodiment of the present invention.
  • An embodiment of the present invention provides an audio adjustment method, where the method includes: when detecting that a preset event is triggered, controlling a microphone to collect an audio signal, where the preset event includes a preset call event and/or a preset recording event; The sound level corresponding to the signal is analyzed; the sound level corresponding to the audio signal is dynamically adjusted according to the analysis result and the preset adjustment strategy; and the adjusted audio signal is subjected to corresponding output processing according to the type of the preset event.
  • the performing including: when the preset event includes a preset call event, sending the adjusted audio signal to the call The opposite end; when the preset event includes a preset recording event, the adjusted audio signal is stored as a recording signal.
  • the analyzing the sound level corresponding to the audio signal comprises: analyzing amplitude information of the audio signal to obtain an acoustic sound analysis result.
  • analyzing amplitude information of the audio signal to obtain an acoustic sound analysis result comprising: extracting amplitude information of a corresponding human voice in the audio signal; and analyzing amplitude information of the corresponding human voice to The result of the acoustic sound analysis is obtained.
  • the dynamically adjusting the sound level corresponding to the audio signal according to the analysis result and the preset adjustment policy including: when the current sound level analysis result includes the current sound level being less than the first preset loudness threshold, And adjusting the current acoustic sound level; and/or, when the current sound intensity analysis result includes the current sound loudness being greater than the second preset loudness threshold, performing the downward adjustment on the current sound sound level.
  • the current acoustic sound level is raised and adjusted, including: when the current amplitude value in the audio signal When less than the first predetermined amplitude threshold, adjusting the current amplitude value to a corresponding first target amplitude, wherein the first target amplitude is greater than or equal to the first predetermined amplitude threshold;
  • the current acoustic level is adjusted to decrease, comprising: adjusting the current amplitude value to a corresponding second target amplitude when the current amplitude value in the audio signal is greater than a second predetermined amplitude threshold, wherein the second target The amplitude is less than or equal to the second predetermined amplitude threshold.
  • the adjusting the current amplitude value to the corresponding first target amplitude comprises: determining a corresponding first target amplitude according to the current amplitude value and the first preset correspondence; according to the first target Amplitude and the current amplitude value determine a corresponding first gain adjustment parameter value; performing signal gain adjustment on a position of the audio signal corresponding to the current amplitude value by using the first gain adjustment parameter value; Adjusting the current acoustic sound to the corresponding second target loudness, comprising: determining a corresponding second target amplitude according to the current amplitude value and the second preset correspondence; determining corresponding according to the current amplitude value and the second target amplitude a second gain adjustment parameter value; performing signal gain adjustment on a position of the audio signal corresponding to the second amplitude value by using the second gain adjustment parameter value.
  • the method before the determining the corresponding first target amplitude according to the current amplitude value and the first preset correspondence, the method further includes: selecting, according to the first preset reference factor, the multiple candidate first preset correspondences Before the determining the corresponding second target amplitude according to the current amplitude value and the second preset correspondence, the method further includes: selecting, according to the second preset reference factor, the plurality of candidate second The second preset correspondence is selected in the preset correspondence.
  • the first preset correspondence is selected from the multiple candidate first preset correspondences according to the first preset reference factor, including: acquiring a call The attribute information of the peer contact and/or the profile mode information of the call peer; the first preset correspondence is selected from the plurality of candidate first preset correspondences according to the attribute information and/or the scenario mode information; And selecting, according to the second preset reference factor, the second preset correspondence relationship from the multiple candidate second preset correspondences, including: acquiring attribute information of the call peer contact and/or context mode information of the call peer; The attribute information and/or the scene mode information selects a second preset correspondence from the plurality of candidate second preset correspondences.
  • the current sound sound level is adjusted and adjusted, including: acquiring attribute information of the call peer contact and/or The scenario mode information of the opposite end of the call; determining a corresponding first preset loudness threshold according to the attribute information and/or the scenario mode information, where the current sound intensity is included in the sound sound analysis result
  • the first preset loudness threshold is used, the current sound level is raised and adjusted.
  • adjusting the current amplitude value to the corresponding first target amplitude includes: acquiring attribute information of the call peer contact and And the context mode information of the call peer; determining a corresponding first preset amplitude threshold according to the attribute information and/or the scene mode information, when the current amplitude value in the audio signal is less than the first preset amplitude threshold, The current amplitude value is adjusted to a corresponding first target amplitude.
  • the obtaining 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 obtains the request feedback according to the scenario mode information.
  • FIG. 1 is a schematic flowchart diagram of an audio signal adjustment method according to an embodiment of the present invention.
  • the method may be implemented by an audio signal adjustment apparatus, where the apparatus may be implemented by software and/or hardware, and may be integrated into a terminal.
  • the method includes:
  • Step 101 When it is detected that the preset event is triggered, the control microphone collects the audio signal.
  • the preset event includes a preset call event and/or a preset recording event.
  • the terminal in the embodiment of the present invention 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 invention does not limit the specific type, number, and location of the microphone. For example, for a mobile phone, it may be one or more electret microphones disposed on the lower side of the mobile phone.
  • the preset call event may be a call event with the audio signal adjustment function in the embodiment of the present invention
  • the preset recording event may be a recording event with the audio signal adjustment function in the embodiment of the present invention.
  • a call event or a recording event is detected when the audio signal adjustment function is turned on, it may be determined that a preset call event or a 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.
  • the preset event includes a preset call event and a preset recording event, it can be understood that the call recording function is enabled, that is, in the right The call content is recorded during the call.
  • a smart phone is taken as an example to briefly introduce the audio processing hardware system and system architecture.
  • the audio processing circuit is generally in the main control board. Due to the different designs of different mobile phones, the specific location of the audio processing circuit may also be different.
  • the audio processing circuit of the smart phone mainly 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 mainly 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 of analog audio. Amplifying circuit, etc.
  • 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.
  • 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 invention 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 invention is not limited.
  • the audio system architecture provided by this embodiment mainly includes user space, kernel space, and 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 the corresponding logical operations, such as detecting the application that triggers the recording event, setting the standard audio condition in advance, and issuing the audio playback finger. Order and so on.
  • 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.
  • libraries Many classes in the framework layer are actually just “mediations” for applications that use Android library files.
  • 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.
  • 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, 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 invention, when the audio signal is collected, the main class involved is the above-mentioned AudioRecorder class.
  • Step 102 Analyze the acoustic sound level corresponding to the audio signal.
  • the three main properties of the sound are volume, pitch, and tone.
  • 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.
  • the user can hear the sound of his own voice during the call, he 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 voice is small, the other party may not be able to hear the voice. The sound is louder, which may make the other party feel deafening; the user can also hear the sound of his own voice when recording, but he cannot 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 addition, there are many reasons why the sound collected by the microphone may be affected by the speaker itself or the state of the call and the recording state.
  • the sound level corresponding to the audio signal is analyzed to know in real time whether the sound level corresponding to the audio signal collected by the terminal is appropriate.
  • 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 103 Dynamically adjust the sound level corresponding to the audio signal according to the analysis result and the preset adjustment strategy.
  • An exemplary preset adjustment strategy may be determined based on a predetermined change in acoustic sound level. For example, when the preset sound sound degree change rule is uniform in sound sound level change, when the sound sound level is large, the sound sound level corresponding to the audio signal can be reduced and adjusted, for example, the amplification of the analog audio signal can be reduced.
  • the multiple is to reduce the gain value; when the sound is small, the sound level corresponding to the audio signal can be raised and adjusted, for example, the amplification factor for amplifying the analog audio signal can be increased, that is, the gain value is increased.
  • Step 104 Perform corresponding output processing on the adjusted audio signal according to the type of the preset event.
  • the adjusted audio signal is sent to the opposite end of the call, and optionally, the voice coding and channel coding processing as described above may also be performed before the sending,
  • the embodiments of the present invention are not limited.
  • the preset event includes a preset recording event
  • the adjusted audio signal is stored as a recording signal.
  • the encoding may be performed according to the preset audio format as described above.
  • the embodiments of the invention are not limited.
  • the audio signal adjustment method provided by the embodiment of the present invention can analyze the sound and sound degree corresponding to the audio signal collected by the microphone during the call or recording process, and perform dynamic adjustment according to the analysis result, and then perform corresponding output processing without a call pair.
  • the end adjusts the volume by itself or does not need to repeatedly adjust the volume by the listener during playback of the recording, and the audio signal can be outputted with a preset volume change rule.
  • 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.
  • 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.
  • 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 (absolute values), and the average value of the amplitudes in the preset time period is taken as 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.
  • analyzing amplitude information of the audio signal to obtain an acoustic sound analysis result may further include: extracting amplitude information of a corresponding human voice in the audio signal; and amplitude information of the corresponding human voice Analyze to obtain the results of the acoustic sound analysis.
  • 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.
  • the data to be analyzed can be greatly reduced, thereby improving the analysis speed, and adjusting the audio signal more timely, which helps to further improve the call state. Or the timeliness of the adjustment of the recording state.
  • dynamically adjusting the sound level corresponding to the audio signal according to the analysis result and the preset adjustment strategy may include: when the current sound level analysis result includes the current sound level being less than the first preset loudness threshold. And adjusting the current sound level; and the method further includes: when the sound sound analysis result includes the current sound level being greater than the second preset loudness threshold, The current sound level is reduced and adjusted.
  • the first preset loudness threshold and the second preset loudness threshold may be the same or different, and the specific value may be a preset fixed value or a dynamically adjusted change value according to actual conditions.
  • the current acoustic sound 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. The sound of the current sound can not be heard by others, and therefore, the current sound level is raised and adjusted so that the sound level in the output audio signal is improved.
  • the current acoustic level when the current acoustic level is greater than or equal to the first preset loudness threshold, no adjustment 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 adjustment efficiency is ensured.
  • 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.
  • the excessive sound affects the listening, and therefore, the current sound level is lowered and adjusted so that the sound level in the outputted audio signal is lowered.
  • the current acoustic level is less than or equal to the second preset loudness threshold, no adjustment 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 adjustment efficiency is ensured.
  • the two adjustment modes may be combined, that is, when the current sound level is small and when the sound is large, the current sound level is greater than or equal to the first preset loudness threshold and less than or equal to the second preset.
  • the loudness threshold When the loudness threshold is used, it indicates that the sound level in the audio signal is appropriate, and adjustment is not possible.
  • the advantage of this setting is that it can be applied to application scenarios that require the upper and lower limits of the sound level to ensure the adjustment effect.
  • the current acoustic sound level is raised and adjusted, including: when the current audio signal is current When the amplitude value is less than the first preset amplitude threshold, the current amplitude value is adjusted to a corresponding first target amplitude, wherein the first target amplitude is greater than or equal to the first predetermined amplitude threshold.
  • the first preset amplitude threshold may be a fixed value or a dynamic change value determined according to actual conditions. The advantage of this setting is that the sound level can be adjusted by adjusting the amplitude value, the adjustment speed is fast, the efficiency is high, and the real-time performance of the audio signal output can be further ensured.
  • the sound sound intensity analysis result includes a current sound level greater than When the second preset loudness threshold is used, the current sound level is reduced and adjusted, and when the current amplitude value in the audio signal is greater than the second preset amplitude threshold, the current amplitude value is adjusted to a corresponding a second target amplitude, wherein the second target amplitude is less than or equal to the second predetermined amplitude threshold.
  • the advantage of this setting is that the sound level can be adjusted by adjusting the amplitude value, the adjustment speed is fast, the efficiency is high, and the real-time performance of the audio signal output can be further ensured.
  • adjusting the current amplitude value to the corresponding first target amplitude may include: determining a corresponding first target amplitude according to the current amplitude value and the first preset correspondence; according to the first The target amplitude and the current amplitude value determine a corresponding first gain adjustment parameter value; and the first gain adjustment parameter value is used to perform signal gain adjustment on a position of the audio signal corresponding to the current amplitude value.
  • the first preset correspondence may exist, for example, in the form of a mapping table, and may be determined by experiments or simulations.
  • the corresponding first gain adjustment coefficient may be determined according to the quotient of the first target amplitude and the current amplitude value, and then the position of the current amplitude value according to the first gain adjustment coefficient.
  • Perform signal gain adjustment For example, if the first gain adjustment coefficient is K1 (greater than 1) and the original gain value is G, then the current gain value can be K1*G. If the amplitude values of other positions do not need to be adjusted, then the amplitude of the other position is simulated by the analog audio amplifier circuit.
  • the gain value is G, and when the current amplitude value is amplified, the gain value is adjusted to K1*G, so that the sound level of the corresponding position of the current amplitude value is improved.
  • the adjusting the current sound level to the corresponding second target loudness may include: determining a corresponding second target amplitude according to the current amplitude value and the second preset correspondence; according to the current amplitude value And determining, by the second target amplitude, a second gain adjustment parameter value; and using the second gain adjustment parameter value to perform signal gain adjustment on a position of the audio signal corresponding to the second amplitude value.
  • the advantage of the setting is that the second target amplitude corresponding to the different amplitude values can be preset in the second preset correspondence, and the second target amplitude corresponding to the current amplitude value can be quickly determined, thereby improving the adjustment efficiency.
  • the second preset correspondence may exist, for example, in the form of a mapping table, and may be determined by experiments or simulations. After determining the second target amplitude corresponding to the current amplitude value, determining a corresponding second gain adjustment coefficient according to the quotient of the second target amplitude and the current amplitude value, and then according to the second The gain adjustment factor performs signal gain adjustment on the position of the current amplitude value. For example, if the second gain adjustment coefficient is K2 (less than 1) and the original gain value is G, then the current gain value may be K2*G. If the amplitude values of other positions do not need to be adjusted, the amplitude of the other position is simulated by the analog audio amplification circuit. When the value is amplified, the gain value is G, and when the current amplitude value is amplified, the gain value is adjusted to K2*G, so that the sound level of the corresponding position of the current amplitude value is lowered.
  • the method before determining the corresponding first target amplitude according to the current amplitude value and the first preset correspondence, the method further includes: corresponding to the multiple candidate first presets according to the first preset reference factor The first preset correspondence is selected in the relationship.
  • the advantage of this setting is that a plurality of first preset correspondences can be set in advance, and the first preset correspondence relationship is dynamically selected according to the actual situation of the current call or recording, so that the adjustment of the sound sound level is more targeted.
  • the required first preset correspondence may be selected according to the actual requirement of the voice end of the call; when the preset event is a preset recording event, Select the desired first preset correspondence according to the actual recording requirements or recording environment, such as the recording distance, the type or intensity of the ambient noise, and the recording mode.
  • the method may further include: selecting, according to the second preset reference factor, the plurality of candidate second preset correspondences The second preset correspondence is selected.
  • the first preset correspondence is selected from the plurality of candidate first preset correspondences according to the first preset reference factor, including: Acquiring the attribute information of the call peer contact and/or the scenario mode information of the call peer; and selecting the first preset correspondence from the plurality of candidate first preset correspondences according to the attribute information and/or the scenario mode information.
  • the attribute information may include age (or age group), or include whether it is an elderly person or a child or the like.
  • 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, or including the call to the call end contact The voice performs voice recognition, and the corresponding attribute information is determined according to the recognition result. when 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 correspondence in which the first target amplitude is set higher may be selected.
  • 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.
  • 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.
  • the mapping relationship between the different context patterns and the different first preset correspondences may be preset, and in the actual call process, the corresponding first preset corresponding information may be determined according to the acquired scene mode information. relationship.
  • the obtaining 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 obtains the request feedback according to the scenario mode information.
  • the selecting the second preset correspondence from the plurality of candidate second preset correspondences according to the second preset reference factor may include: acquiring attribute information of the call peer contact and/or the call peer end.
  • the scenario mode information is selected from the plurality of candidate second preset correspondences according to the attribute information and/or the scenario mode information.
  • the first preset loudness threshold and/or the second preset loudness threshold, and also the first preset amplitude threshold and/or the second preset amplitude threshold may also be based on the attribute information and/or Profile mode information to determine. For example, when the current acoustic sound analysis result includes the current sound loudness is less than the first preset loudness threshold, the current sound sound level is adjusted and adjusted, including: acquiring attribute information and/or a call pair of the call peer contact. And determining, according to the attribute information and/or the scene mode information, a corresponding first preset loudness threshold, and when the sound sound analysis result includes the current sound loudness being less than the first preset loudness threshold, The current sound level is adjusted upwards.
  • the determination condition of the audio signal adjustment needs to be determined according to the attribute information or the scene mode information of the call end, so that the audio signal adjustment is more accurate and targeted.
  • the current amplitude value in the audio signal is less than the first preset amplitude threshold
  • the current amplitude value is adjusted to Corresponding first target amplitude, comprising: obtaining attribute information of the call peer contact and/or context mode information of the call peer; determining a corresponding first preset amplitude threshold according to the attribute information and/or the scenario mode information, when When the current amplitude value in the audio signal is less than the first preset amplitude threshold, the current amplitude value is adjusted to a corresponding first target amplitude.
  • FIG. 4 is a schematic flowchart of still another method for adjusting an audio signal according to an embodiment of the present invention.
  • the method is applicable to a recording scenario, and specifically includes:
  • 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 yes, execute step 404; otherwise, perform step 405.
  • 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 Perform an increase adjustment on the current sound level, and store the adjusted audio signal as a recording signal.
  • Step 405 Determine that the current sound level is greater than the preset loudness threshold B. If yes, execute step 406; otherwise, perform step 407.
  • Step 406 Perform a reduction adjustment on the current sound level, and store the adjusted audio signal as a recording signal.
  • Step 407 Determine whether a recording pause or recording stop instruction is received, and if yes, end the flow; otherwise, return to step 403.
  • the audio signal adjusting method provided by the embodiment of the invention can dynamically adjust the sound sound level corresponding to the collected audio signal during the recording process, automatically improve the sound sound level when the sound sound level is too small, and automatically reduce the sound sound level when the sound sound level is too large.
  • the recording effect can be automatically maintained, and the user does not need to manually adjust the sound when listening to the recording.
  • the beneficial effect of the embodiment of the present invention is particularly obvious when the recorded sound is large and small.
  • FIG. 5 is a schematic flowchart of still another method for adjusting an audio signal according to an embodiment of the present invention.
  • the method is applicable to a call scenario, and specifically 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.
  • Step 503 Determine whether the current sound level is less than the preset loudness threshold C. If yes, execute step 504; otherwise, execute step 505.
  • 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 Perform an increase adjustment on the current sound level, and send the adjusted audio signal to the opposite end of the call.
  • Step 505 Determine that the current sound level is greater than the preset loudness threshold D. If yes, execute step 506; otherwise, perform step 507.
  • Step 506 Perform a lower adjustment on the current sound level, and send the adjusted audio signal to the opposite end of the call.
  • Step 507 Determine whether a call end instruction is received, and if yes, end the process; otherwise, return to step 503.
  • the audio signal adjustment method provided by the embodiment of the invention can dynamically adjust the sound and sound degree corresponding to the collected audio signal during the call, automatically improve the sound level when the sound level is too small, and automatically reduce the sound level when the sound level is too large.
  • the user can automatically maintain a good call effect without manual adjustment by the user at the opposite end of the call.
  • the beneficial effects of the embodiment of the present invention are particularly obvious when the local user speaks loudly.
  • FIG. 6 is a schematic flowchart of still another method for adjusting an audio signal according to an embodiment of the present invention.
  • the method is applicable to a call scenario, and specifically 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 the scenario mode information of the opposite end of the call, and determine a corresponding preset amplitude threshold E, a preset amplitude threshold F, a first preset correspondence, and a second preset correspondence according to the scenario mode information.
  • Step 604 Determine whether the current amplitude value is less than a preset amplitude threshold E. If yes, perform the step. 604; otherwise, step 607 is performed.
  • Step 605 Determine a corresponding first target amplitude according to the current amplitude value and the first preset correspondence, and determine a corresponding first gain adjustment parameter value according to the first target amplitude and the current amplitude value.
  • Step 606 Perform signal gain adjustment on the position of the audio signal corresponding to the current amplitude value by using the first gain adjustment parameter value, and send the adjusted audio signal to the opposite end of the call, and perform step 610.
  • Step 607 Determine whether the current amplitude value is greater than the preset amplitude threshold F. If yes, execute step 608; otherwise, perform step 610.
  • Step 608 Determine a corresponding second target amplitude according to the current amplitude value and the second preset correspondence, and determine a corresponding second gain adjustment parameter value according to the second target amplitude and the current amplitude value.
  • Step 609 Perform signal gain adjustment on a position corresponding to the current amplitude value in the audio signal by using the second gain adjustment parameter value, and send the adjusted audio signal to the opposite end of the call.
  • Step 610 Determine whether a call end instruction is received, and if yes, end the process; otherwise, return to step 604.
  • the audio signal adjustment method provided by the embodiment of the invention can more accurately adjust the sound and sound level corresponding to the audio signal according to the scene mode information of the other party during the call, and can improve the call effect more specifically.
  • FIG. 7 is a structural block diagram of an audio signal adjusting apparatus according to an embodiment of the present invention.
  • the apparatus may be implemented by software and/or hardware, and is generally integrated in a terminal, and the audio signal may be adjusted by performing an audio signal adjusting method.
  • the device includes:
  • the audio signal collecting module 701 is configured to: when detecting that the preset event is triggered, control the microphone to collect an audio signal, where the preset event includes a preset call event and/or a preset recording event;
  • the loudness analysis module 702 is configured to analyze the sound intensity corresponding to the audio signal
  • the loudness adjustment module 703 is configured to dynamically adjust the sound level corresponding to the audio signal according to the analysis result and the preset adjustment policy;
  • the signal output module 704 is configured to perform corresponding output processing on the adjusted audio signal according to the type of the preset event.
  • the audio signal adjusting apparatus provided by the embodiment of the invention can analyze the sound and sound degree corresponding to the audio signal collected by the microphone during the call or recording process, and dynamically adjust according to the analysis result. Then, the corresponding output processing is performed, and the voice is not required to be adjusted by the opposite end of the call or the volume is repeatedly adjusted by the listener during playback of the recording, and the audio signal can be outputted with a preset volume change rule.
  • the signal output module is configured to:
  • the preset event includes a preset call event, sending the adjusted audio signal to the opposite end of the call;
  • the adjusted audio signal is stored as a recording signal.
  • the pair of loudness analysis module is used to:
  • the amplitude information of the audio signal is analyzed to obtain an acoustic sound analysis result.
  • the pair of loudness analysis module is used to:
  • the amplitude information of the corresponding human voice is analyzed to obtain an acoustic sound analysis result.
  • the loudness adjustment module is configured to:
  • the current sound sound level is raised and adjusted; and/or,
  • the current sound sound level is reduced and adjusted.
  • the current sound sound level is raised and adjusted, including:
  • the current sound sound level is reduced and adjusted, including:
  • the adjusting the current amplitude value to the corresponding first target amplitude comprises:
  • the adjusting the current sound level to the corresponding second target loudness comprises:
  • the loudness adjustment module is further configured to:
  • the preset event is a preset call event
  • the first preset correspondence is selected from the multiple candidate first preset correspondences according to the first preset reference factor, including:
  • Embodiments of the present invention also provide a storage medium including computer executable instructions, the computer
  • the executable instructions when executed by a computer processor, are for performing an audio signal conditioning method, the method comprising:
  • the control microphone collects an audio signal, and the preset event includes a preset call event and/or a preset recording event;
  • Storage media any of a variety of types of memory devices or storage devices.
  • 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 such as DRAM, DDR RAM, SRAM, 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 elements, and the like.
  • the storage medium may also include other types of memory or a combination thereof.
  • 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 two or more storage media that can reside in different locations (eg, in different computer systems connected through a network).
  • a storage medium may store program instructions (eg, embodied as a computer program) executable by one or more processors.
  • the computer executable instructions are not limited to the audio signal adjustment operation as described above, and may also perform audio signal adjustment provided by any embodiment of the present invention. Related operations in the method.
  • FIG. 8 is a schematic structural diagram of a terminal according to an embodiment of the present invention.
  • the terminal may be, for example, a mobile terminal.
  • 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).
  • CPU central processing unit
  • 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).
  • CPU central processing unit
  • circuit board not shown
  • a power supply circuit not shown
  • microphone 813 Referred to as a microphone 813.
  • the circuit board is disposed inside a space enclosed by the casing; the CPU 802 and the memory 801 are provided Placed on the circuit board; the power supply circuit is configured to supply power to each circuit or device of the terminal; the memory 801 is configured to store executable program code; and the CPU 802 reads the memory 801 by reading The stored executable program code runs a computer program corresponding to the executable program code to implement the following steps:
  • the control microphone collects an audio signal, and the preset event includes a preset call event and/or a preset recording event;
  • the terminal further includes: a peripheral interface 803, an RF (Radio Frequency) 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 are communicated via one or more communication buses or signal lines 807.
  • RF Radio Frequency
  • 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, two or more components may be combined, or may have Different component configurations.
  • the various components shown in the figures can be implemented in hardware, software, or a combination of hardware and software, including one or more signal processing and/or application specific integrated circuits.
  • the terminal for adjusting an audio signal provided in this embodiment is described in detail below.
  • 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.
  • 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.
  • the 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 one or more input controls for controlling other input/control devices 810 Controller 8092.
  • one or more input controllers 8092 receive electrical signals from other input/control devices 810 or transmit 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.
  • 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.
  • 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 mainly used for establishing communication between the mobile phone and the wireless network (ie, the network side), and realizing data reception and transmission between the mobile phone and the wireless network. For example, sending and receiving short messages, emails, and the like. Specifically, the RF circuit 805 receives and transmits an RF signal, which is 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 through the electromagnetic signal and communication network and other devices Communicate.
  • an RF signal which is also referred to as an electromagnetic signal
  • RF circuitry 805 may include known circuitry for performing these functions including, but not limited to, an antenna system, an RF transceiver, one or more amplifiers, a tuner, one or more oscillators, a digital signal processor, a CODEC ( COder-DECoder, codec) Chipset, Subscriber Identity Module (SIM), etc.
  • an antenna system an RF transceiver, one or more amplifiers, a tuner, one or more oscillators, a digital signal processor, a CODEC ( COder-DECoder, codec) Chipset, Subscriber Identity Module (SIM), etc.
  • CODEC COder-DECoder, codec
  • SIM Subscriber Identity Module
  • the audio circuit 806 is mainly used 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 configured 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 used for power supply and power management of 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 collect audio from the microphone during a call or recording process.
  • the sound intensity corresponding to the signal is analyzed, and the corresponding output processing is performed according to the analysis result, and the corresponding output processing is performed without the need of the call peer to adjust the volume or the speaker repeatedly adjusts the volume when playing the recording, and the audio signal can be preset.
  • the determined volume change law is output.
  • the audio signal adjusting device, the storage medium and the terminal provided in the above embodiments may perform the audio signal adjusting method provided by any embodiment of the present invention, and have corresponding functional modules and beneficial effects for performing the method.
  • the audio signal adjustment method provided by any embodiment of the present invention.

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Abstract

音频信号调节方法、装置、存储介质及终端。该方法包括：检测到预设事件被触发时，控制麦克风采集音频信号（101），预设事件包括预设通话事件和/或预设录音事件；对音频信号对应的声音响度进行分析（102）；根据分析结果以及预设调节策略对音频信号对应的声音响度进行动态调节（103）；根据预设事件的类型对调节后的音频信号进行相应的输出处理。无需通话对端自行调节音量或者无需在播放录音时由收听者反复调节音量，可保持音频信号以预先设定好的音量变化规律被输出。

Description

音频信号调节方法、装置、存储介质及终端 技术领域
本发明实施例涉及音频处理技术领域，尤其涉及音频信号调节方法、装置、存储介质及终端。
背景技术
为了满足用户在生活以及工作中的语音通话、视频通话以及录音等需求，多数终端都具备通话或录音功能，如手机、平板电脑、音乐播放器以及录音笔等。
一般的，终端通过集成麦克风(简称mic，又称传声器)来实现通话或录音功能。麦克风是一种将声音信号转换为电信号的能量转换器件，当声音的振动传到麦克风的振动膜上后，会导致里面的磁铁形成变化的电流信号，由声音处理电路对电流信号进行处理后传输到通话对端或进行存储，进而实现通话或录音。
发明内容
本发明实施例提供音频信号调节方法、装置、存储介质及终端，可以在通话或录音过程中对录入的音频信号进行自动调节。
第一方面，本发明实施例提供了一种音频信号调节方法，包括：
检测到预设事件被触发时，控制麦克风采集音频信号，所述预设事件包括预设通话事件和/或预设录音事件；
对所述音频信号对应的声音响度进行分析；
根据分析结果以及预设调节策略对所述音频信号对应的声音响度进行动态调节；
根据所述预设事件的类型对调节后的音频信号进行相应的输出处理。
第二方面，本发明实施例提供了一种音频信号调节装置，包括：
音频信号采集模块，用于在检测到预设事件被触发时，控制麦克风采集音频信号，所述预设事件包括预设通话事件和/或预设录音事件；
响度分析模块，用于对所述音频信号对应的声音响度进行分析；
响度调节模块，用于根据分析结果以及预设调节策略对所述音频信号对应 的声音响度进行动态调节；
信号输出模块，用于根据所述预设事件的类型对调节后的音频信号进行相应的输出处理。
第三方面，本发明实施例提供了一种计算机可读存储介质，其上存储有计算机程序，该程序被处理器执行时实现如本发明实施例所述的音频信号调节。
第四方面，本发明实施例提供了一种移动终端，包括麦克风，存储器，处理器及存储在存储器上并可在处理器运行的计算机程序，所述处理器执行所述计算机程序时实现如本发明实施例所述的音频信号调节方法。
附图说明
为了更清楚地说明本发明实施例中的技术方案，下面将对实施例描述中所需要使用的附图作简单地介绍。显而易见地，下面描述中的附图仅仅是本发明的一些实施例，对于本领域技术人员来讲，在不付出创造性劳动的前提下，还可以根据这些附图获得其他的附图。
图1为本发明实施例提供的一种音频信号调节方法的流程示意图；
图2为本发明实施例提供的一种智能手机的音频处理硬件系统结构示意图；
图3为本发明实施例提供的一种音频系统架构框图；
图4为本发明实施例提供的又一种音频信号调节方法的流程示意图；
图5为本发明实施例提供的又一种音频信号调节方法的流程示意图；
图6为本发明实施例提供的又一种音频信号调节方法的流程示意图；
图7为本发明实施例提供的一种音频信号调节装置的结构框图；
图8为本发明实施例提供的一种终端的结构示意图。
具体实施方式
下面结合附图并通过具体实施方式来进一步说明本发明的技术方案。可以理解的是，此处所描述的具体实施例仅仅用于解释本发明，而非对本发明的限定。另外还需要说明的是，为了便于描述，附图中仅示出了与本发明相关的部分而非全部结构。
在更加详细地讨论示例性实施例之前应当提到的是，一些示例性实施例被描述成作为流程图描绘的处理或方法。虽然流程图将各步骤描述成顺序的处理， 但是其中的许多步骤可以被并行地、并发地或者同时实施。此外，各步骤的顺序可以被重新安排。当其操作完成时所述处理可以被终止，但是还可以具有未包括在附图中的附加步骤。所述处理可以对应于方法、函数、规程、子例程、子程序等等。
本发明实施例提供音频调节方法，该方法包括：检测到预设事件被触发时，控制麦克风采集音频信号，所述预设事件包括预设通话事件和/或预设录音事件；对所述音频信号对应的声音响度进行分析；根据分析结果以及预设调节策略对所述音频信号对应的声音响度进行动态调节；根据所述预设事件的类型对调节后的音频信号进行相应的输出处理。
可选的，所述根据所述预设事件的类型对调节后的音频信号进行相应的输出处理，包括：当所述预设事件包括预设通话事件时，将调节后的音频信号发送至通话对端；当所述预设事件包括预设录音事件时，将调节后的音频信号作为录音信号进行存储。
可选的，所述对所述音频信号对应的声音响度进行分析，包括：对所述音频信号的振幅信息进行分析，以得到声音响度分析结果。
可选的，对所述音频信号的振幅信息进行分析，以得到声音响度分析结果，包括：提取所述音频信号中对应人声的振幅信息；对所述对应人声的振幅信息进行分析，以得到声音响度分析结果。
可选的，所述根据分析结果以及预设调节策略对所述音频信号对应的声音响度进行动态调节，包括：当所述声音响度分析结果中包括当前声音响度小于第一预设响度阈值时，对所述当前声音响度进行升高调节；和/或，当所述声音响度分析结果中包括当前声音响度大于第二预设响度阈值时，对所述当前声音响度进行降低调节。
可选的，所述当所述声音响度分析结果中包括当前声音响度小于第一预设响度阈值时，对所述当前声音响度进行升高调节，包括：当所述音频信号中的当前振幅值小于第一预设振幅阈值时，将所述当前振幅值调节至对应的第一目标振幅，其中，所述第一目标振幅大于或等于所述第一预设振幅阈值；所述当所述声音响度分析结果中包括当前声音响度大于第二预设响度阈值时，对所述 当前声音响度进行降低调节，包括：当所述音频信号中的当前振幅值大于第二预设振幅阈值时，将所述当前振幅值调节至对应的第二目标振幅，其中，所述第二目标振幅小于或等于所述第二预设振幅阈值。
可选的，所述将所述当前振幅值调节至对应的第一目标振幅，包括：根据所述当前振幅值和第一预设对应关系确定相应的第一目标振幅；根据所述第一目标振幅和所述当前振幅值确定对应的第一增益调节参数值；利用所述第一增益调节参数值对所述音频信号中对应所述当前振幅值的位置进行信号增益调节；所述将所述当前声音响度调节至对应的第二目标响度，包括：根据所述当前振幅值和第二预设对应关系确定相应的第二目标振幅；根据所述当前振幅值和所述第二目标振幅确定对应的第二增益调节参数值；利用所述第二增益调节参数值对所述音频信号中对应所述第二振幅值的位置进行信号增益调节。
可选的，在所述根据所述当前振幅值和第一预设对应关系确定相应的第一目标振幅之前，还包括：根据第一预设参考因素从多个候选第一预设对应关系中挑选出第一预设对应关系；在所述根据所述当前振幅值和第二预设对应关系确定相应的第二目标振幅之前，还包括：根据第二预设参考因素从多个候选第二预设对应关系中挑选出第二预设对应关系。
可选的，当所述预设事件为预设通话事件时，所述根据第一预设参考因素从多个候选第一预设对应关系中挑选出第一预设对应关系，包括：获取通话对端联系人的属性信息和/或通话对端的情景模式信息；根据所述属性信息和/或情景模式信息从多个候选第一预设对应关系中挑选出第一预设对应关系；所述根据第二预设参考因素从多个候选第二预设对应关系中挑选出第二预设对应关系，包括：获取通话对端联系人的属性信息和/或通话对端的情景模式信息；根据所述属性信息和/或情景模式信息从多个候选第二预设对应关系中挑选出第二预设对应关系。
可选的，当所述声音响度分析结果中包括当前声音响度小于第一预设响度阈值时，对所述当前声音响度进行升高调节，包括：获取通话对端联系人的属性信息和/或通话对端的情景模式信息；根据所述属性信息和/或情景模式信息确定对应的第一预设响度阈值，当所述声音响度分析结果中包括当前声音响度小 于第一预设响度阈值时，对所述当前声音响度进行升高调节。
可选的，当所述音频信号中的当前振幅值小于第一预设振幅阈值时，将所述当前振幅值调节至对应的第一目标振幅，包括：获取通话对端联系人的属性信息和/或通话对端的情景模式信息；根据所述属性信息和/或情景模式信息确定对应的第一预设振幅阈值，当所述音频信号中的当前振幅值小于第一预设振幅阈值时，将所述当前振幅值调节至对应的第一目标振幅。
可选的，获取通话对端的情景模式信息具体可包括：向通话对端发送情景模式信息获取请求，接收通话对端根据所述情景模式信息获取请求反馈的情景模式信息。
图1为本发明实施例提供的一种音频信号调节方法的流程示意图，该方法可以由音频信号调节装置执行，其中该装置可由软件和/或硬件实现，一般可集成在终端中。如图1所示，该方法包括：
步骤101、检测到预设事件被触发时，控制麦克风采集音频信号。
其中，所述预设事件包括预设通话事件和/或预设录音事件。
示例性的，本发明实施例中的终端可包括手机、平板电脑、音乐播放器以及录音笔等配置有麦克风的设备。麦克风可以是内置的，也可以是外置的。麦克风(简称mic，又称传声器或话筒)是一种将声音信号转换为电信号的能量转换器件，当声音的振动传到麦克风的振动膜上后，会导致里面的磁铁形成变化的电流信号，由声音处理电路对电流信号进行处理后传输到通话对端或进行存储，进而实现通话或录音。本发明实施例对麦克风的具体类型、数量及所在位置不做限定，例如，对于手机来说，可以是设置在手机下侧面的一个或多个驻极体式麦克风。
示例性的，预设通话事件可以是具备本发明实施例中的音频信号调节功能的通话事件；预设录音事件可以是具备本发明实施例中的音频信号调节功能的录音事件。可选的，在音频信号调节功能处于开启状态下检测到通话事件或录音事件时，可确定检测到预设通话事件或预设录音事件。通话事件例如是电话接通或是语音聊天接通等；录音事件例如是启动录音。当所述预设事件同时包括预设通话事件和预设录音事件时，可理解为开启了通话录音功能，即在与对 端进行通话的过程中对通话内容进行录音。
为了便于理解本发明实施例，下面以智能手机为例对音频处理硬件系统及系统架构进行简单的介绍。
图2为本发明实施例提供的一种智能手机的音频处理硬件系统结构示意图。音频处理电路一般处于主控电路板中，由于不同手机的设计不同，音频处理电路的具体位置也可能不同。智能手机的音频处理电路主要包括音频信号处理电路、基带信号处理电路、音频功率放大器、耳机信号放大器、听筒、扬声器、麦克风及耳机接口等。其中，音频信号处理电路是整个音频处理电路的核心。音频处理电路主要由接收音频电路、送话电路、耳机通话电路等组成，包括模拟音频的模拟/数字(A/D)转换、数字/模拟(A/D)转换、数字语音信号处理及模拟音频放大电路等。
在通话时，首先通话本地端麦克风把声音的机械声波信号转化为模拟音频信号，通过模拟音频放大电路进行放大，经内部的多模转换器进行A/D转换，得到数字音频信号；其次把此数字音频信号送到基带处理器，进行语音编码、信道编码等处理；再次进行加密、交织等一系列处理；最后送到基带处理器中的数字窄带制调模块进行调制，产生发射基带信号送入射频电路调制成发射中频，发送给通话对方。
在录音时，与上述过程类似，首先由麦克风把声音的机械声波信号转化为模拟音频信号，通过模拟音频放大电路进行放大，经过A/D转换后得到数字音频信号，按照预设的音频格式进行编码及存储。
本发明实施例中麦克风采集的音频信号可以是上述的由机械声波信号转换而来的模拟音频信号，也可以是经过放大后的模拟音频信号，还可以是经过A/D转换后的数字音频信号等，本发明实施例不作限定。
如图3所示，本实施例提供的音频系统架构主要包括用户空间、内核空间和硬件系统。用户空间包括应用(Application)层、应用框架(Framework)层和硬件抽象层(Hardware Abstraction Layer，HAL)，内核空间包括驱动(Driver)层，。应用层是音频体系的最上层，可通过编写一个应用程序来执行对应的逻辑操作，例如检测触发录音事件的应用程序，预先设置标准音频条件，下发音频播放指 令等。应用框架层包括音频控制接口和标准化插件模块负责提供音频播放形式控制接口，以及扬声器音量大小控制接口等。应用框架层提供了AudioTrack和AudioRecorder两个类，以及AudioManager、AudioService及AudioSystem类。在应用框架层和硬件抽象层之间还包括一个系统运行库(Libraries)层。我们知道，framework层的很多类，实际上只是应用程序使用Android库文件的“中介”而已。因为上层应用一般采用java语言编写，它们需要最直接的java接口的支持，这就是framework层存在的意义之一。而作为“中介”，它们并不会真正去实现具体的功能，或者只实现其中的一部分功能，而把主要重心放在库中来完成。比如上面的AudioTrack、AudioRecorder、MediaPlayer和MediaRecorder等等在库中都能找到相对应的类。这一部分代码集中放置在工程的frameworks/av/media/libmedia中，多数是C++语言编写的。音频方面的硬件抽象层主要分为两部分，即AudioFlinger和AudioPolicyService。实际上后者并不是一个真实的设备，只是采用虚拟设备的方式来让厂商可以方便地定制出自己的策略。根据产品的不同，音频设备存在很大差异，在Android的音频架构中，这些问题都是由HAL层的audio.primary等来解决的，而不需要大规模地修改上层实现。硬件抽象层是应用框架层到驱动层的过渡，以实现底层硬件的兼容。驱动层按照音频编解码器的特性对其进行控制，确保音频编解码器可以正常工作，将音频编解码器获取到的音频数据提供给系统层。本发明实施例中在进行音频信号采集时，主要涉及的类为上述的AudioRecorder类。
步骤102、对所述音频信号对应的声音响度进行分析。
声音的三个主要属性为音量、音调和音色。其中，音量又称响度或音强，是指人耳对所听到的声音大小强弱的主观感受，其客观评价尺度是声音的振幅大小。振幅指物体震动过程中偏离原来位置的最大距离，人耳听到的声音的响度与音源的振幅相关，一般振幅越大响度越强。
示例性的，用户在通话时虽然能够听到自己说话的声音大小，但是无法知道当前说话的音量对于通话对端的人来说声音是大还是小，若声音较小，对方可能听不清，若声音较大，可能让对方感到震耳；用户在录音时也能够听到自己说话的声音大小，但是无法知道终端记录的自己的声音的大小，若声音较小， 播放录音时可能听不清，若声音较大，播放录音时可能让收听者感到不适。此外，还有很多原因可能导致麦克风采集的声音受到说话人本身或者通话状态及录音状态的影响。例如不同人说话习惯不同，有人说话声音比较轻柔，而有人说话声音比较洪亮；又如，说话过程中用户难以保持声源(如用户的嘴巴)与麦克风之间的距离不变，如用户处于运动状态或处于颠簸的车辆中等等，都可能导致麦克风采集的声音时大时小。上述这些情况都会对通话对方或者收听录音的人产生不好的影响。本发明实施例中，对音频信号对应的声音响度进行分析，以实时获知终端采集的音频信号对应的声音响度是否合适。
示例性的，可以预设时段为分析单位，分析每个预设时段内的声音响度的大小，可将当前预设时段对应的声音响度记为当前声音响度。预设时段可以是以当前时刻为起点向前计时的预设时间长度。为了保证实时性，可以将预设时段对应的预设时间长度设置的短一些，例如是以当前时刻为起点向前计时的0.5秒内。
步骤103、根据分析结果以及预设调节策略对所述音频信号对应的声音响度进行动态调节。
示例性的预设调节策略可以是根据预先设定好的声音响度变化规律来确定。例如，当预先设定好的声音响度变化规律为声音响度变化均匀，那么当声音响度较大时，可对音频信号对应的声音响度进行降低调节，如可减小对模拟音频信号进行放大的放大倍数，即降低增益值；当声音响度较小时，可对音频信号对应的声音响度进行升高调节，如可增大对模拟音频信号进行放大的放大倍数，即提高增益值。
步骤104、根据所述预设事件的类型对调节后的音频信号进行相应的输出处理。
示例性的，当所述预设事件包括预设通话事件时，将调节后的音频信号发送至通话对端，可选的，在发送之前还可进行如上述的语音编码及信道编码等处理，本发明实施例不做限定。当所述预设事件包括预设录音事件时，将调节后的音频信号作为录音信号进行存储，可选的，在发送之前还可进行如上述的按照预设的音频格式进行编码等处理，本发明实施例不做限定。当所述预设事 件同时包括预设通话事件和预设录音事件时，即开启了通话录音功能，则将调节后的音频信号发送至通话对端的同时将调节后的音频信号作为录音信号进行存储。
本发明实施例提供的音频信号调节方法，在通话或录音过程中，可对麦克风采集的音频信号对应的声音响度进行分析，并根据分析结果进行动态调节后再进行相应的输出处理，无需通话对端自行调节音量或者无需在播放录音时由收听者反复调节音量，可保持音频信号以预先设定好的音量变化规律被输出。
在一些实施例中，所述对所述音频信号对应的声音响度进行分析，可包括：对所述音频信号的振幅信息进行分析，以得到声音响度分析结果。如前所述，声音响度的客观评价尺度是声音的振幅大小，而音频信号由声音的机械声波信号转换而来，因此可采用振幅信息对音频信号的声音响度进行分析。振幅越大，声音响度越大，也即声音能量值越高。示例性的，可以预设时段为分析单位，在预设时段内以预设采样频率对音频信号的振幅进行采集，获取多个振幅值(绝对值)，将预设时段内的振幅平均值作为当前声音响度。此处优化将对声音响度的分析转化为对振幅信息的分析，可简化分析过程，提高分析速度。
在一些实施例中，对所述音频信号的振幅信息进行分析，以得到声音响度分析结果，可进一步包括：提取所述音频信号中对应人声的振幅信息；对所述对应人声的振幅信息进行分析，以得到声音响度分析结果。示例性的，在通话过程中，人声是音频信号中的主体，其他环境声音可视为是干扰声音；此外，在录音时，用户可以选择人声录音模式，这时，人声也是音频信号中的主体，其他环境声音可视为是干扰声音。在这些情况下，在提取音频信号中对应人声的振幅信息后，可大幅减少需要分析的数据，进而提高分析速度，也能够更及时地对音频信号进行调节，有助于进一步提高针对通话状态或录音状态的调整时效性。
在一些实施例中，根据分析结果以及预设调节策略对所述音频信号对应的声音响度进行动态调节，可包括：当所述声音响度分析结果中包括当前声音响度小于第一预设响度阈值时，对所述当前声音响度进行升高调节；也可包括：当所述声音响度分析结果中包括当前声音响度大于第二预设响度阈值时，对所 述当前声音响度进行降低调节。当然，也可同时包括上述两种方式。第一预设响度阈值和第二预设响度阈值可以相同，也可以不同，具体数值可以是预先设置的固定值，也可以是根据实际情况进行动态调整的变化值。
示例性的，在当前声音响度小于第一预设响度阈值时，可说明音频信号中当前声音响度较小，通话对端可能无法听清本端用户说的话，或者录音文件在后续播放时可能导致他人听不清此时录取的声音，因此，对所述当前声音响度进行升高调节，以使输出的音频信号中的声音响度有所提高。示例性的，在当前声音响度大于或等于第一预设响度阈值时，可不进行调节。这样设置的好处在于，可适用于对声音响度上限没有要求的应用场景，保证调节效率。
示例性的，在当前声音响度大于第二预设响度阈值时，可说明音频信号中当前声音响度较大，传到通话对端可能会使对方感到震耳，或者录音文件在后续播放时可能导致声音过大影响收听，因此，对所述当前声音响度进行降低调节，以使输出的音频信号中的声音响度有所降低。示例性的，在当前声音响度小于或等于第二预设响度阈值时，可不进行调节。这样设置的好处在于，可适用于对声音响度下限没有要求的应用场景，保证调节效率。
示例性的，可将上述两种调节方式进行结合，即当前声音响度较小时和较大时均会进行调节，而当前声音响度大于或等于第一预设响度阈值且小于或等于第二预设响度阈值时，说明此时音频信号中的声音响度比较合适，可不进行调节。这样设置的好处在于，可适用于对声音响度上限和下限均有要求的应用场景，保证调节效果。
在一些实施例中，所述当所述声音响度分析结果中包括当前声音响度小于第一预设响度阈值时，对所述当前声音响度进行升高调节，包括：当所述音频信号中的当前振幅值小于第一预设振幅阈值时，将所述当前振幅值调节至对应的第一目标振幅，其中，所述第一目标振幅大于或等于所述第一预设振幅阈值。第一预设振幅阈值可以是一个固定值，也可以是根据实际情况确定的动态的变化值。这样设置的好处在于，通过调节振幅值即可实现声音响度的调节，调节速度快，效率高，能够进一步保证音频信号输出的实时性。
在一些实施例中，所述当所述声音响度分析结果中包括当前声音响度大于 第二预设响度阈值时，对所述当前声音响度进行降低调节，包括：当所述音频信号中的当前振幅值大于第二预设振幅阈值时，将所述当前振幅值调节至对应的第二目标振幅，其中，所述第二目标振幅小于或等于所述第二预设振幅阈值。这样设置的好处在于，通过调节振幅值即可实现声音响度的调节，调节速度快，效率高，能够进一步保证音频信号输出的实时性。
在一些实施例中，将所述当前振幅值调节至对应的第一目标振幅，可包括：根据所述当前振幅值和第一预设对应关系确定相应的第一目标振幅；根据所述第一目标振幅和所述当前振幅值确定对应的第一增益调节参数值；利用所述第一增益调节参数值对所述音频信号中对应所述当前振幅值的位置进行信号增益调节。这样设置的好处在于，可在第一预设对应关系中预先设定好不同的振幅值所对应的第一目标振幅，能够快速确定当前振幅值对应的第一目标振幅，从而提高调节效率。第一预设对应关系例如可以映射表的形式存在，可通过实验或仿真等方式来确定。在确定好当前振幅值对应的第一目标振幅后，可根据第一目标振幅与当前振幅值的商来确定对应的第一增益调节系数，然后再根据第一增益调节系数对当前振幅值的位置进行信号增益调节。例如，假设第一增益调节系数为K1(大于1)，原增益值为G，那么当前增益值可以是K1*G，若其他位置振幅值不需要调整，那么通过模拟音频放大电路对其他位置振幅值进行放大时增益值为G，而对当前振幅值进行放大时增益值调整为K1*G，使得当前振幅值对应位置的声音响度提高。
同理，所述将所述当前声音响度调节至对应的第二目标响度，可包括：根据所述当前振幅值和第二预设对应关系确定相应的第二目标振幅；根据所述当前振幅值和所述第二目标振幅确定对应的第二增益调节参数值；利用所述第二增益调节参数值对所述音频信号中对应所述第二振幅值的位置进行信号增益调节。这样设置的好处在于，可在第二预设对应关系中预先设定好不同的振幅值所对应的第二目标振幅，能够快速确定当前振幅值对应的第二目标振幅，从而提高调节效率。第二预设对应关系例如可以映射表的形式存在，可通过实验或仿真等方式来确定。在确定好当前振幅值对应的第二目标振幅后，可根据第二目标振幅与当前振幅值的商来确定对应的第二增益调节系数，然后再根据第二 增益调节系数对当前振幅值的位置进行信号增益调节。例如，假设第二增益调节系数为K2(小于1)，原增益值为G，那么当前增益值可以是K2*G，若其他位置振幅值不需要调整，那么通过模拟音频放大电路对其他位置振幅值进行放大时增益值为G，而对当前振幅值进行放大时增益值调整为K2*G，使得当前振幅值对应位置的声音响度降低。
在一些实施例中，在所述根据所述当前振幅值和第一预设对应关系确定相应的第一目标振幅之前，还包括：根据第一预设参考因素从多个候选第一预设对应关系中挑选出第一预设对应关系。这样设置的好处在于，可预先设置多个第一预设对应关系，根据当前通话或录音的实际情况来动态的选取第一预设对应关系，使对声音响度的调节更具有针对性。示例性的，当预设事件为预设通话事件时，可根据通话对端对声音响度的实际需求来选择所需的第一预设对应关系；当预设事件为预设录音事件时，可根据实际的录音需求或录音环境选择所需的第一预设对应关系，如录音距离、环境噪声的类型或强度以及录音模式等等。同理，在所述根据所述当前振幅值和第二预设对应关系确定相应的第二目标振幅之前，还可包括：根据第二预设参考因素从多个候选第二预设对应关系中挑选出第二预设对应关系。这样设置的好处在于，可预先设置多个第二预设对应关系，根据当前通话或录音的实际情况来动态的选取第二预设对应关系，使对声音响度的调节更具有针对性。
在一些实施例中，当所述预设事件为预设通话事件时，所述根据第一预设参考因素从多个候选第一预设对应关系中挑选出第一预设对应关系，包括：获取通话对端联系人的属性信息和/或通话对端的情景模式信息；根据所述属性信息和/或情景模式信息从多个候选第一预设对应关系中挑选出第一预设对应关系。这样设置的好处在于，能够根据通话对端的实际情况更加准确地确定第一预设对应关系，进一步提升通话效果。
示例性的，属性信息可以包括年龄(或年龄段)，或者包括是否为老人或是否为孩子等。可选的，获取通话对端联系人的属性信息可包括：获取通话对端联系人在通讯录中的备注信息，从所述备注信息中提取属性信息；也可包括对通话对端联系人的声音进行语音识别，根据识别结果确定相应的属性信息。当 对方为老人时，可能听力较差，所需的声音响度应大一些，因此可挑选第一目标振幅被设置得较高的第一预设对应关系。
示例性的，通话对端的情景模式信息可包括静音模式、会议模式、正常模式和户外模式等等模式信息。用户通常会根据自己所处环境设置对应的情景模式，例如在上课或者开会等相对比较安静的环境中，可能会选择静音模式或会议模式，在外面比较吵的环境中可能会选择户外模式，在这些环境中接听电话时，可能对声音响度有不同的需求。例如，会议模式时，由于环境比较安静，声音响度不必很大即可听清对方声音；而户外模式时，由于环境比较吵闹，需要比较大的声音响度才能听清对方声音。基于以上原因，可预先设定好不同的情景模式与不同的第一预设对应关系的映射关系，在实际的通话过程中，能够根据获取到的情景模式信息来确定相应的第一预设对应关系。可选的，获取通话对端的情景模式信息具体可包括：向通话对端发送情景模式信息获取请求，接收通话对端根据所述情景模式信息获取请求反馈的情景模式信息。
同理，所述根据第二预设参考因素从多个候选第二预设对应关系中挑选出第二预设对应关系，可包括：获取通话对端联系人的属性信息和/或通话对端的情景模式信息；根据所述属性信息和/或情景模式信息从多个候选第二预设对应关系中挑选出第二预设对应关系。此处优化的有益效果与前述的针对多个第一预设对应关系的优化类似，此处不再赘述。
在一些实施例中，第一预设响度阈值和/或第二预设响度阈值，还有第一预设振幅阈值和/或第二预设振幅阈值，也可以根据所述属性信息和/或情景模式信息来确定。例如，当所述声音响度分析结果中包括当前声音响度小于第一预设响度阈值时，对所述当前声音响度进行升高调节，包括：获取通话对端联系人的属性信息和/或通话对端的情景模式信息；根据所述属性信息和/或情景模式信息确定对应的第一预设响度阈值，当所述声音响度分析结果中包括当前声音响度小于第一预设响度阈值时，对所述当前声音响度进行升高调节。这样设置的好处在于，可根据通话对端的属性信息或情景模式信息来确定是否需要进行音频信号调节的判定条件，使音频信号调节更加准确、有针对性。又如，当所述音频信号中的当前振幅值小于第一预设振幅阈值时，将所述当前振幅值调节至 对应的第一目标振幅，包括：获取通话对端联系人的属性信息和/或通话对端的情景模式信息；根据所述属性信息和/或情景模式信息确定对应的第一预设振幅阈值，当所述音频信号中的当前振幅值小于第一预设振幅阈值时，将所述当前振幅值调节至对应的第一目标振幅。
图4为本发明实施例提供的又一种音频信号调节方法的流程示意图，该方法适用于录音场景，具体包括：
步骤401、检测到预设录音事件被触发时，控制麦克风采集音频信号。
步骤402、实时对所采集的音频信号的振幅信息进行分析，以得到声音响度分析结果。
步骤403、判断当前声音响度是否小于预设响度阈值A，若是，则执行步骤404；否则，执行步骤405。
可以理解的是，随着录音的进行，终端会实时地对麦克风采集的音频信号的振幅信息进行分析，而当前声音响度也是随着分析的进行而不断发生变化的。
步骤404、对所述当前声音响度进行升高调节，并将调节后的音频信号作为录音信号进行存储。
步骤405、判断当前声音响度大于预设响度阈值B，若是，则执行步骤406；否则，执行步骤407。
步骤406、对所述当前声音响度进行降低调节，并将调节后的音频信号作为录音信号进行存储。
步骤407、判断是否接收到录音暂停或录音停止指令，若是，则结束流程；否则，返回执行步骤403。
本发明实施例提供的音频信号调节方法，能够在录音过程中动态地调整所采集的音频信号对应的声音响度，当声音响度过小时自动提高声音响度，当声音响度过大时自动降低声音响度，可以自动保持良好的录音效果，在收听录音时无需用户手动调节，对于所录取的声音时大时小的情况本发明实施例的有益效果尤为明显。
图5为本发明实施例提供的又一种音频信号调节方法的流程示意图，该方法适用于通话场景，具体包括：
步骤501、检测到预设通话事件被触发时，控制麦克风采集音频信号。
步骤502、实时对所采集的音频信号的振幅信息进行分析，以得到声音响度分析结果。
步骤503、判断当前声音响度是否小于预设响度阈值C，若是，则执行步骤504；否则，执行步骤505。
可以理解的是，随着通话的进行，终端会实时地对麦克风采集的音频信号的振幅信息进行分析，而当前声音响度也是随着分析的进行而不断发生变化的。
步骤504、对所述当前声音响度进行升高调节，并将调节后的音频信号发送至通话对端。
步骤505、判断当前声音响度大于预设响度阈值D，若是，则执行步骤506；否则，执行步骤507。
步骤506、对所述当前声音响度进行降低调节，并将调节后的音频信号发送至通话对端。
步骤507、判断是否接收到通话结束指令，若是，则结束流程；否则，返回执行步骤503。
本发明实施例提供的音频信号调节方法，能够在通话过程中动态地调整所采集的音频信号对应的声音响度，当声音响度过小时自动提高声音响度，当声音响度过大时自动降低声音响度，可以自动保持良好的通话效果，无需通话对端的用户手动调节，对于本端用户说话声音时大时小的情况本发明实施例的有益效果尤为明显。
图6为本发明实施例提供的又一种音频信号调节方法的流程示意图，该方法适用于通话场景，具体包括：
步骤601、检测到预设通话事件被触发时，控制麦克风采集音频信号。
步骤602、实时提取所述音频信号中对应人声的振幅信息，对所述对应人声的振幅信息进行分析，以得到声音响度分析结果。
步骤603、获取通话对端的情景模式信息，并根据情景模式信息确定相应的预设振幅阈值E、预设振幅阈值F、第一预设对应关系以及第二预设对应关系。
步骤604、判断当前振幅值是否小于预设振幅阈值E，若是，则执行步骤 604；否则，执行步骤607。
步骤605、根据当前振幅值和第一预设对应关系确定相应的第一目标振幅，根据第一目标振幅和当前振幅值确定对应的第一增益调节参数值。
步骤606、利用第一增益调节参数值对音频信号中对应当前振幅值的位置进行信号增益调节，并将调节后的音频信号发送至通话对端，执行步骤610。
步骤607、判断当前振幅值是否大于预设振幅阈值F，若是，则执行步骤608；否则，执行步骤610。
步骤608、根据当前振幅值和第二预设对应关系确定相应的第二目标振幅，根据第二目标振幅和当前振幅值确定对应的第二增益调节参数值。
步骤609、利用第二增益调节参数值对音频信号中对应当前振幅值的位置进行信号增益调节，并将调节后的音频信号发送至通话对端。
步骤610、判断是否接收到通话结束指令，若是，则结束流程；否则，返回执行步骤604。
本发明实施例提供的音频信号调节方法，能够在通话过程中根据通话对方的情景模式信息更加准确地调节音频信号对应的声音响度，可以更有针对性的提升通话效果。
图7为本发明实施例提供的一种音频信号调节装置的结构框图，该装置可由软件和/或硬件实现，一般集成在终端中，可通过执行音频信号调节方法来对音频信号进行调节。如图7所示，该装置包括：
音频信号采集模块701，用于在检测到预设事件被触发时，控制麦克风采集音频信号，所述预设事件包括预设通话事件和/或预设录音事件；
响度分析模块702，用于对所述音频信号对应的声音响度进行分析；
响度调节模块703，用于根据分析结果以及预设调节策略对所述音频信号对应的声音响度进行动态调节；
信号输出模块704，用于根据所述预设事件的类型对调节后的音频信号进行相应的输出处理。
本发明实施例提供的音频信号调节装置，在通话或录音过程中，可对麦克风采集的音频信号对应的声音响度进行分析，并根据分析结果进行动态调节后 再进行相应的输出处理，无需通话对端自行调节音量或者无需在播放录音时由收听者反复调节音量，可保持音频信号以预先设定好的音量变化规律被输出。
可选的，所述信号输出模块用于：
当所述预设事件包括预设通话事件时，将调节后的音频信号发送至通话对端；
当所述预设事件包括预设录音事件时，将调节后的音频信号作为录音信号进行存储。
可选的，所述对响度分析模块用于：
对所述音频信号的振幅信息进行分析，以得到声音响度分析结果。
可选的，所述对响度分析模块用于：
提取所述音频信号中对应人声的振幅信息；
对所述对应人声的振幅信息进行分析，以得到声音响度分析结果。
可选的，所述响度调节模块用于：
当所述声音响度分析结果中包括当前声音响度小于第一预设响度阈值时，对所述当前声音响度进行升高调节；和/或，
当所述声音响度分析结果中包括当前声音响度大于第二预设响度阈值时，对所述当前声音响度进行降低调节。
可选的，所述当所述声音响度分析结果中包括当前声音响度小于第一预设响度阈值时，对所述当前声音响度进行升高调节，包括：
当所述音频信号中的当前振幅值小于第一预设振幅阈值时，将所述当前振幅值调节至对应的第一目标振幅，其中，所述第一目标振幅大于或等于所述第一预设振幅阈值；
所述当所述声音响度分析结果中包括当前声音响度大于第二预设响度阈值时，对所述当前声音响度进行降低调节，包括：
当所述音频信号中的当前振幅值大于第二预设振幅阈值时，将所述当前振幅值调节至对应的第二目标振幅，其中，所述第二目标振幅小于或等于所述第二预设振幅阈值。
可选的，所述将所述当前振幅值调节至对应的第一目标振幅，包括：
根据所述当前振幅值和第一预设对应关系确定相应的第一目标振幅；
根据所述第一目标振幅和所述当前振幅值确定对应的第一增益调节参数值；
利用所述第一增益调节参数值对所述音频信号中对应所述当前振幅值的位置进行信号增益调节；
所述将所述当前声音响度调节至对应的第二目标响度，包括：
根据所述当前振幅值和第二预设对应关系确定相应的第二目标振幅；
根据所述当前振幅值和所述第二目标振幅确定对应的第二增益调节参数值；
利用所述第二增益调节参数值对所述音频信号中对应所述第二振幅值的位置进行信号增益调节。
可选的，所述响度调节模块还用于：
在所述根据所述当前振幅值和第一预设对应关系确定相应的第一目标振幅之前，根据第一预设参考因素从多个候选第一预设对应关系中挑选出第一预设对应关系；和/或，
在所述根据所述当前振幅值和第二预设对应关系确定相应的第二目标振幅之前，根据第二预设参考因素从多个候选第二预设对应关系中挑选出第二预设对应关系。
可选的，当所述预设事件为预设通话事件时，
所述根据第一预设参考因素从多个候选第一预设对应关系中挑选出第一预设对应关系，包括：
获取通话对端联系人的属性信息和/或通话对端的情景模式信息；
根据所述属性信息和/或情景模式信息从多个候选第一预设对应关系中挑选出第一预设对应关系；
所述根据第二预设参考因素从多个候选第二预设对应关系中挑选出第二预设对应关系，包括：
获取通话对端联系人的属性信息和/或通话对端的情景模式信息；
根据所述属性信息和/或情景模式信息从多个候选第二预设对应关系中挑选出第二预设对应关系。
本发明实施例还提供一种包含计算机可执行指令的存储介质，所述计算机 可执行指令在由计算机处理器执行时用于执行一种音频信号调节方法，该方法包括：
检测到预设事件被触发时，控制麦克风采集音频信号，所述预设事件包括预设通话事件和/或预设录音事件；
对所述音频信号对应的声音响度进行分析；
根据分析结果以及预设调节策略对所述音频信号对应的声音响度进行动态调节；
根据所述预设事件的类型对调节后的音频信号进行相应的输出处理。
存储介质——任何的各种类型的存储器设备或存储设备。术语“存储介质”旨在包括：安装介质，例如CD-ROM、软盘或磁带装置；计算机系统存储器或随机存取存储器，诸如DRAM、DDR RAM、SRAM、EDO RAM，兰巴斯(Rambus)RAM等；非易失性存储器，诸如闪存、磁介质(例如硬盘或光存储)；寄存器或其它相似类型的存储器元件等。存储介质可以还包括其它类型的存储器或其组合。另外，存储介质可以位于程序在其中被执行的第一计算机系统中，或者可以位于不同的第二计算机系统中，第二计算机系统通过网络(诸如因特网)连接到第一计算机系统。第二计算机系统可以提供程序指令给第一计算机用于执行。术语“存储介质”可以包括可以驻留在不同位置中(例如在通过网络连接的不同计算机系统中)的两个或更多存储介质。存储介质可以存储可由一个或多个处理器执行的程序指令(例如具体实现为计算机程序)。
当然，本发明实施例所提供的一种包含计算机可执行指令的存储介质，其计算机可执行指令不限于如上所述的音频信号调节操作，还可以执行本发明任意实施例所提供的音频信号调节方法中的相关操作。
本发明实施例提供了一种终端，该终端中可集成本发明实施例提供的音频信号调节装置。图8为本发明实施例提供的一种终端的结构示意图。该终端例如可以是移动终端，如图8所示，该终端可以包括：壳体(图中未示出)、存储器801、中央处理器(Central Processing Unit，CPU)802(又称处理器，以下简称CPU)、电路板(图中未示出)、电源电路(图中未示出)和麦克风813。所述电路板安置在所述壳体围成的空间内部；所述CPU802和所述存储器801设 置在所述电路板上；所述电源电路，用于为所述终端的各个电路或器件供电；所述存储器801，用于存储可执行程序代码；所述CPU802通过读取所述存储器801中存储的可执行程序代码来运行与所述可执行程序代码对应的计算机程序，以实现以下步骤：
检测到预设事件被触发时，控制麦克风采集音频信号，所述预设事件包括预设通话事件和/或预设录音事件；
对所述音频信号对应的声音响度进行分析；
根据分析结果以及预设调节策略对所述音频信号对应的声音响度进行动态调节；
根据所述预设事件的类型对调节后的音频信号进行相应的输出处理。
所述终端还包括：外设接口803、RF(Radio Frequency，射频)电路805、音频电路806、扬声器811、电源管理芯片808、输入/输出(I/O)子系统其他输入/控制设备、触摸屏812、其他输入/控制设备810以及外部端口804，这些部件通过一个或多个通信总线或信号线807来通信。
应该理解的是，图示终端800仅仅是终端的一个范例，并且终端800可以具有比图中所示出的更多的或者更少的部件，可以组合两个或更多的部件，或者可以具有不同的部件配置。图中所示出的各种部件可以在包括一个或多个信号处理和/或专用集成电路在内的硬件、软件、或硬件和软件的组合中实现。
下面就本实施例提供的用于调节音频信号的终端进行详细的描述，该终端以手机为例。
存储器801，所述存储器801可以被CPU802、外设接口803等访问，所述存储器801可以包括高速随机存取存储器，还可以包括非易失性存储器，例如一个或多个磁盘存储器件、闪存器件、或其他易失性固态存储器件。
外设接口803，所述外设接口803可以将设备的输入和输出外设连接到CPU802和存储器801。
I/O子系统809，所述I/O子系统809可以将设备上的输入输出外设，例如触摸屏812和其他输入/控制设备810，连接到外设接口803。I/O子系统809可以包括显示控制器8091和用于控制其他输入/控制设备810的一个或多个输入控 制器8092。其中，一个或多个输入控制器8092从其他输入/控制设备810接收电信号或者向其他输入/控制设备810发送电信号，其他输入/控制设备810可以包括物理按钮(按压按钮、摇臂按钮等)、拨号盘、滑动开关、操纵杆、点击滚轮。值得说明的是，输入控制器8092可以与以下任一个连接：键盘、红外端口、USB接口以及诸如鼠标的指示设备。
触摸屏812，所述触摸屏812是用户终端与用户之间的输入接口和输出接口，将可视输出显示给用户，可视输出可以包括图形、文本、图标、视频等。
I/O子系统809中的显示控制器8091从触摸屏812接收电信号或者向触摸屏812发送电信号。触摸屏812检测触摸屏上的接触，显示控制器8091将检测到的接触转换为与显示在触摸屏812上的用户界面对象的交互，即实现人机交互，显示在触摸屏812上的用户界面对象可以是运行游戏的图标、联网到相应网络的图标等。值得说明的是，设备还可以包括光鼠，光鼠是不显示可视输出的触摸敏感表面，或者是由触摸屏形成的触摸敏感表面的延伸。
RF电路805，主要用于建立手机与无线网络(即网络侧)的通信，实现手机与无线网络的数据接收和发送。例如收发短信息、电子邮件等。具体地，RF电路805接收并发送RF信号，RF信号也称为电磁信号，RF电路805将电信号转换为电磁信号或将电磁信号转换为电信号，并且通过该电磁信号与通信网络以及其他设备进行通信。RF电路805可以包括用于执行这些功能的已知电路，其包括但不限于天线系统、RF收发机、一个或多个放大器、调谐器、一个或多个振荡器、数字信号处理器、CODEC(COder-DECoder，编译码器)芯片组、用户标识模块(Subscriber Identity Module，SIM)等等。
音频电路806，主要用于从外设接口803接收音频数据，将该音频数据转换为电信号，并且将该电信号发送给扬声器811。
扬声器811，用于将手机通过RF电路805从无线网络接收的语音信号，还原为声音并向用户播放该声音。
电源管理芯片808，用于为CPU802、I/O子系统及外设接口所连接的硬件进行供电及电源管理。
本发明实施例提供的终端，在通话或录音过程中，可对麦克风采集的音频 信号对应的声音响度进行分析，并根据分析结果进行动态调节后再进行相应的输出处理，无需通话对端自行调节音量或者无需在播放录音时由收听者反复调节音量，可保持音频信号以预先设定好的音量变化规律被输出。
上述实施例中提供的音频信号调节装置、存储介质及终端可执行本发明任意实施例所提供的音频信号调节方法，具备执行该方法相应的功能模块和有益效果。未在上述实施例中详尽描述的技术细节，可参见本发明任意实施例所提供的音频信号调节方法。
注意，上述仅为本发明的较佳实施例及所运用技术原理。本领域技术人员会理解，本发明不限于这里所述的特定实施例，对本领域技术人员来说能够进行各种明显的变化、重新调整和替代而不会脱离本发明的保护范围。因此，虽然通过以上实施例对本发明进行了较为详细的说明，但是本发明不仅仅限于以上实施例，在不脱离本发明构思的情况下，还可以包括更多其他等效实施例，而本发明的范围由所附的权利要求范围决定。

Claims (20)

  1. 一种音频信号调节方法，其特征在于，包括：
    检测到预设事件被触发时，控制麦克风采集音频信号，所述预设事件包括预设通话事件和/或预设录音事件；
    对所述音频信号对应的声音响度进行分析；
    根据分析结果以及预设调节策略对所述音频信号对应的声音响度进行动态调节；
    根据所述预设事件的类型对调节后的音频信号进行相应的输出处理。
  2. 根据权利要求1所述的方法，其特征在于，所述根据所述预设事件的类型对调节后的音频信号进行相应的输出处理，包括：
    当所述预设事件包括预设通话事件时，将调节后的音频信号发送至通话对端；
    当所述预设事件包括预设录音事件时，将调节后的音频信号作为录音信号进行存储。
  3. 根据权利要求2所述的方法，其特征在于，所述对所述音频信号对应的声音响度进行分析，包括：
    对所述音频信号的振幅信息进行分析，以得到声音响度分析结果。
  4. 根据权利要求3所述的方法，其特征在于，对所述音频信号的振幅信息进行分析，以得到声音响度分析结果，包括：
    提取所述音频信号中对应人声的振幅信息；
    对所述对应人声的振幅信息进行分析，以得到声音响度分析结果。
  5. 根据权利要求3或4所述的方法，其特征在于，所述根据分析结果以及预设调节策略对所述音频信号对应的声音响度进行动态调节，包括：
    当所述声音响度分析结果中包括当前声音响度小于第一预设响度阈值时，对所述当前声音响度进行升高调节；和/或，
    当所述声音响度分析结果中包括当前声音响度大于第二预设响度阈值时，对所述当前声音响度进行降低调节。
  6. 根据权利要求5所述的方法，其特征在于，
    所述当所述声音响度分析结果中包括当前声音响度小于第一预设响度阈值时，对所述当前声音响度进行升高调节，包括：
    当所述音频信号中的当前振幅值小于第一预设振幅阈值时，将所述当前振幅值调节至对应的第一目标振幅，其中，所述第一目标振幅大于或等于所述第 一预设振幅阈值；
    所述当所述声音响度分析结果中包括当前声音响度大于第二预设响度阈值时，对所述当前声音响度进行降低调节，包括：
    当所述音频信号中的当前振幅值大于第二预设振幅阈值时，将所述当前振幅值调节至对应的第二目标振幅，其中，所述第二目标振幅小于或等于所述第二预设振幅阈值。
  7. 根据权利要求6所述的方法，其特征在于，
    所述将所述当前振幅值调节至对应的第一目标振幅，包括：
    根据所述当前振幅值和第一预设对应关系确定相应的第一目标振幅；
    根据所述第一目标振幅和所述当前振幅值确定对应的第一增益调节参数值；
    利用所述第一增益调节参数值对所述音频信号中对应所述当前振幅值的位置进行信号增益调节；
    所述将所述当前声音响度调节至对应的第二目标响度，包括：
    根据所述当前振幅值和第二预设对应关系确定相应的第二目标振幅；
    根据所述当前振幅值和所述第二目标振幅确定对应的第二增益调节参数值；
    利用所述第二增益调节参数值对所述音频信号中对应所述第二振幅值的位置进行信号增益调节。
  8. 根据权利要求7所述的方法，其特征在于，
    在所述根据所述当前振幅值和第一预设对应关系确定相应的第一目标振幅之前，还包括：
    根据第一预设参考因素从多个候选第一预设对应关系中挑选出第一预设对应关系；
    在所述根据所述当前振幅值和第二预设对应关系确定相应的第二目标振幅之前，还包括：
    根据第二预设参考因素从多个候选第二预设对应关系中挑选出第二预设对应关系。
  9. 根据权利要求8所述的方法，其特征在于，当所述预设事件为预设通话事件时，
    所述根据第一预设参考因素从多个候选第一预设对应关系中挑选出第一预设对应关系，包括：
    获取通话对端联系人的属性信息和/或通话对端的情景模式信息；
    根据所述属性信息和/或情景模式信息从多个候选第一预设对应关系中挑选出第一预设对应关系；
    所述根据第二预设参考因素从多个候选第二预设对应关系中挑选出第二预设对应关系，包括：
    获取通话对端联系人的属性信息和/或通话对端的情景模式信息；
    根据所述属性信息和/或情景模式信息从多个候选第二预设对应关系中挑选出第二预设对应关系。
  10. 一种音频信号调节装置，其特征在于，包括：
    音频信号采集模块，用于在检测到预设事件被触发时，控制麦克风采集音频信号，所述预设事件包括预设通话事件和/或预设录音事件；
    响度分析模块，用于对所述音频信号对应的声音响度进行分析；
    响度调节模块，用于根据分析结果以及预设调节策略对所述音频信号对应的声音响度进行动态调节；
    信号输出模块，用于根据所述预设事件的类型对调节后的音频信号进行相应的输出处理。
  11. 根据权利要求10所述的装置，其特征在于，所述信号输出模块用于：
    当所述预设事件包括预设通话事件时，将调节后的音频信号发送至通话对端；
    当所述预设事件包括预设录音事件时，将调节后的音频信号作为录音信号进行存储。
  12. 根据权利要求11所述的装置，其特征在于，所述对响度分析模块用于：
    对所述音频信号的振幅信息进行分析，以得到声音响度分析结果。
  13. 根据权利要求12所述的装置，其特征在于，所述对响度分析模块用于：
    提取所述音频信号中对应人声的振幅信息；
    对所述对应人声的振幅信息进行分析，以得到声音响度分析结果。
  14. 根据权利要求11或12所述的装置，其特征在于，所述响度调节模块用于：
    当所述声音响度分析结果中包括当前声音响度小于第一预设响度阈值时，对所述当前声音响度进行升高调节；和/或，
    当所述声音响度分析结果中包括当前声音响度大于第二预设响度阈值时，对所述当前声音响度进行降低调节。
  15. 根据权利要求14所述的装置，其特征在于，
    所述当所述声音响度分析结果中包括当前声音响度小于第一预设响度阈值时，对所述当前声音响度进行升高调节，包括：
    当所述音频信号中的当前振幅值小于第一预设振幅阈值时，将所述当前振幅值调节至对应的第一目标振幅，其中，所述第一目标振幅大于或等于所述第一预设振幅阈值；
    所述当所述声音响度分析结果中包括当前声音响度大于第二预设响度阈值时，对所述当前声音响度进行降低调节，包括：
    当所述音频信号中的当前振幅值大于第二预设振幅阈值时，将所述当前振幅值调节至对应的第二目标振幅，其中，所述第二目标振幅小于或等于所述第二预设振幅阈值。
  16. 根据权利要求15所述的装置，其特征在于，
    所述将所述当前振幅值调节至对应的第一目标振幅，包括：
    根据所述当前振幅值和第一预设对应关系确定相应的第一目标振幅；
    根据所述第一目标振幅和所述当前振幅值确定对应的第一增益调节参数值；
    利用所述第一增益调节参数值对所述音频信号中对应所述当前振幅值的位置进行信号增益调节；
    所述将所述当前声音响度调节至对应的第二目标响度，包括：
    根据所述当前振幅值和第二预设对应关系确定相应的第二目标振幅；
    根据所述当前振幅值和所述第二目标振幅确定对应的第二增益调节参数值；
    利用所述第二增益调节参数值对所述音频信号中对应所述第二振幅值的位置进行信号增益调节。
  17. 根据权利要求16所述的装置，其特征在于，所述响度调节模块还用于：
    在所述根据所述当前振幅值和第一预设对应关系确定相应的第一目标振幅之前，根据第一预设参考因素从多个候选第一预设对应关系中挑选出第一预设对应关系；和/或，
    在所述根据所述当前振幅值和第二预设对应关系确定相应的第二目标振幅之前，根据第二预设参考因素从多个候选第二预设对应关系中挑选出第二预设对应关系。
  18. 根据权利要求17所述的装置，其特征在于，当所述预设事件为预设通话事件时，
    所述根据第一预设参考因素从多个候选第一预设对应关系中挑选出第一预设对应关系，包括：
    获取通话对端联系人的属性信息和/或通话对端的情景模式信息；
    根据所述属性信息和/或情景模式信息从多个候选第一预设对应关系中挑选出第一预设对应关系；
    所述根据第二预设参考因素从多个候选第二预设对应关系中挑选出第二预设对应关系，包括：
    获取通话对端联系人的属性信息和/或通话对端的情景模式信息；
    根据所述属性信息和/或情景模式信息从多个候选第二预设对应关系中挑选出第二预设对应关系。
  19. 一种计算机可读存储介质，其上存储有计算机程序，其特征在于，该程序被处理器执行时实现如权利要求1-9中任一所述的音频信号调节方法。
  20. 一种终端，其特征在于，包括麦克风，存储器，处理器及存储在存储器上并可在处理器运行的计算机程序，其特征在于，所述处理器执行所述计算机程序时实现如权利要求1-7任一所述的音频信号调节方法。
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