Volume adjusting method and system and intelligent terminal

Abstract

The invention provides a volume adjusting method and system and an intelligent terminal. The volume adjusting method comprises the following steps: in response to a preset instruction, releasing a PWM signal with a preset frequency; then receiving a microphone signal, and carrying out sound mixing processing based on the PWM signal and the microphone signal to obtain a sound mixing signal; further dividing the sound mixing signal into two paths to carry out filtering processing so as to obtain a bass signal and extract an original microphone signal; and adjusting the output volume of the terminal based on the bass signal. According to the invention, the bass signal and the original microphone signal are separated, so that the microphone signal cannot be distorted or the tone quality cannot be degraded in the volume adjusting process; the voice recognition rate of the voice recognition chip is remarkably improved on the premise that the tone quality of the microphone is guaranteed.

Description

Technical Field

[0001] The present application relates to the field of audio technology, and in particular to a volume adjustment method, system and intelligent terminal. Background Art

[0002] Voice recognition chips are widely used in smart terminal devices. However, the sound of the audio system itself will seriously affect the accuracy of the voice recognition chip.

[0003] The existing audio systems do not lower the volume of voice recognition, making it difficult and costly to improve the recognition rate and providing a poor user experience. Summary of the invention

[0004] In order to solve the above-mentioned technical problems, the present application provides a volume adjustment method, system and intelligent terminal that can improve the voice recognition rate.

[0005] Specifically, the present application provides a volume adjustment method, comprising the following steps: S100: Responding to a preset instruction, releasing a PWM signal of a preset frequency.

[0006] S200: Receive a microphone signal, and perform mixing processing based on the PWM signal and the microphone signal to obtain a mixed signal.

[0007] S300: Divide the mixed signal into two paths for filtering to obtain a bass signal and extract an original microphone signal.

[0008] And, S400: adjusting the output volume of the terminal based on the bass signal.

[0009] In the above technical solution, by separating the bass signal and the original microphone signal, it is ensured that the microphone signal will not be distorted or the sound quality will not be degraded during the volume adjustment process; the dynamic adjustment mechanism based on the bass signal can reduce the interference components picked up by the voice recognition chip, thereby significantly improving the voice recognition rate of the voice recognition chip.

[0010] Furthermore, before executing step S100, the following steps are included: The preset action is recognized in real time by the gyroscope, so that when the preset action is recognized, a preset instruction is issued.

[0011] In the above technical solution, the gyroscope recognizes preset actions, providing users with an intuitive and convenient way to adjust the volume. Users can start the volume adjustment function through simple gestures or actions without complicated operations; the real-time detection capability of the gyroscope ensures the rapid response of the volume adjustment function, and users can complete the volume adjustment in an instant.

[0012] In addition, users can customize preset actions according to their habits and preferences, enhancing the personalization and flexibility of the volume adjustment function.

[0013] Furthermore, before executing step S100, the method further includes: Presetting the preset frequency; wherein, comprising: The recording range of the current microphone is obtained to set the preset frequency based on the recording range.

[0014] In the above technical solution, by obtaining the recording range of the current microphone, the system can automatically adjust the frequency of the PWM signal to adapt to different recording environments. This environmental adaptability ensures the best volume adjustment effect in various environments; the preset frequency set based on the recording range can better balance the sound quality loss in the volume adjustment process and reduce unnecessary power consumption, which is especially important for portable devices and smart terminals and can extend battery life.

[0015] In addition, users do not need to manually adjust settings. The system can automatically adapt to environmental changes and provide a consistent and high-quality volume adjustment experience, enhancing user satisfaction and product competitiveness.

[0016] Furthermore, the current microphone includes at least a first channel and a second channel; after releasing the PWM signal in step S100, the following is further included: The PWM signal is low-pass filtered to obtain a low-frequency signal for transmission to the first path.

[0017] In the above technical solution, the low-frequency component in the PWM signal is separated through low-pass filtering and transmitted to the first channel of the microphone. This signal separation method helps to better control the processing of low-frequency signals during volume adjustment, reduce distortion and sound quality degradation; low-pass filtering can remove high-frequency noise and interference in the PWM signal, thereby improving the purity of the low-frequency signal.

[0018] In addition, by transmitting the low-frequency signal to the first path and isolating it from other signal paths, cross-interference between different signal paths can be reduced and the overall performance and stability of the system can be improved.

[0019] Furthermore, the step S200 includes: The microphone signal is received through the second path, and the low-frequency signal and the microphone signal are output to the mixer through the first path and the second path respectively, so as to obtain a mixed signal through the mixer.

[0020] In the above technical solution, the signal separation and integration are achieved by transmitting the low-frequency signal and the original microphone signal to the mixer through different paths respectively. This separation process helps to retain the details of the original signal, while introducing the low-frequency signal for volume adjustment to ensure the integrity of the sound quality; the introduction of the low-frequency signal makes the processing of the low-frequency signal more precise during the volume adjustment process, avoiding the sound quality degradation caused by directly amplifying or compressing the original signal. Through mixing processing, the volume can be dynamically adjusted while retaining the sound quality.

[0021] In addition, the use of a mixer provides a more flexible mixing control method, which can adjust the balance between different signal paths and adapt to different environmental changes. This flexibility enhances the adaptability of the system. This signal processing method is compatible with a variety of audio processing systems and is easy to integrate into existing audio processing equipment and smart terminals. Through simple hardware and software configuration, efficient volume adjustment function can be achieved, enhancing the market competitiveness of the product.

[0022] Furthermore, the step S300 includes: A low-pass filter process is performed based on the mixed signal to obtain the bass signal; and a high-pass filter process is performed based on the mixed signal to extract the original microphone signal.

[0023] In the above technical solution, the separation of low-frequency and high-frequency components in the mixed signal is achieved through low-pass and high-pass filtering. This separation processing helps to process the bass and treble signals separately during the volume adjustment process, thereby achieving more precise volume control and sound quality optimization; low-pass filtering can effectively remove high-frequency noise and interference in the mixed signal, and improve the purity and clarity of the bass signal. At the same time, high-pass filtering can retain the high-frequency details of the original microphone signal, ensuring that the sound quality of the treble part is not affected.

[0024] In addition, by processing bass and treble signals separately, the system can more flexibly adjust the volume dynamically to adapt to different audio content. This dynamic adjustment mechanism further improves the accuracy of volume adjustment.

[0025] Furthermore, before executing step S400, the following steps are included: The current output volume of the terminal is obtained, and the speech recognition rate is obtained based on the microphone signal.

[0026] In the above technical solution, by obtaining the current output volume of the terminal, the system can monitor and adjust the volume in real time to adapt to different environments. This dynamic volume adjustment mechanism ensures the best speech recognition rate in all situations; obtaining the speech recognition rate helps the system evaluate the recognizability of the current speech signal. Based on this, the system can automatically adjust the audio processing parameters to improve the accuracy and efficiency of speech recognition.

[0027] In addition, by combining real-time monitoring of volume and speech recognition rate, the system can better understand environmental changes, thereby providing more intelligent audio processing and interactive experience; for example, when the speech recognition rate is low, the system can automatically increase the volume or perform noise reduction to enhance the recognition effect.

[0028] Furthermore, the step S400 includes: Determine whether the bass signal is greater than a preset threshold, and if so, adjust the output volume based on the speech recognition rate; otherwise, return to step S100.

[0029] In the above technical solution, by judging whether the bass signal exceeds the preset threshold, the system can automatically identify and adjust the volume of the terminal output. This automatic adjustment mechanism ensures that the volume will not be too loud when the bass signal is strong, thereby avoiding sound distortion. By combining the voice recognition rate to adjust the volume, the system can dynamically adjust the volume according to the clarity and recognizability of the voice signal. This adjustment method effectively improves the accuracy of voice recognition, especially when the voice recognition rate is low. The system can enhance the clarity of the signal by increasing the volume, thereby improving the voice recognition rate.

[0030] In addition, automatic volume adjustment and voice recognition-driven optimization make voice recognition more intelligent and personalized, and the system can automatically adjust the volume according to actual usage.

[0031] Further, based on the same concept, the present application also provides a system using the volume adjustment method, including: The microphone module is used to release a PWM signal of a preset frequency in response to a preset instruction; and receive a microphone signal, and perform mixing processing based on the PWM signal and the microphone signal to obtain a mixed signal to output to the power amplifier module.

[0032] The power amplifier module is used to filter the mixed signal into two paths to obtain a bass signal and extract the original microphone signal; and to adjust the output volume of the terminal based on the bass signal.

[0033] In the above technical solution, the microphone module is responsible for signal input and mixing processing, and the power amplifier module is responsible for filtering processing and volume adjustment. The two work together to achieve efficient volume control. Through mixing processing, the system can effectively integrate PWM signals and microphone signals to ensure signal integrity and consistency. The filtering processing of the power amplifier module further improves the separation of bass and treble signals, ensuring the purity of sound quality. Based on the strength judgment of the bass signal, the system can monitor and adjust the output volume in real time to avoid distortion caused by excessive microphone signals. This dynamic adjustment mechanism ensures the best voice recognition rate under various audio content.

[0034] In addition, through automatic volume adjustment and voice recognition-driven optimization, the system can automatically adjust the volume according to actual usage to ensure the best listening experience in various environments; users do not need to manually adjust the volume, and get a more convenient and intelligent audio experience.

[0035] Furthermore, based on the same concept, the present application also provides an intelligent terminal, which includes a volume adjustment system, and the volume adjustment system uses the volume adjustment method to achieve volume control.

[0036] Compared with the prior art, the beneficial effects of this application are: The present application first responds to a preset instruction and releases a PWM signal of a preset frequency; then receives a microphone signal, and performs mixing processing based on the PWM signal and the microphone signal to obtain a mixed signal; further divides the mixed signal into two paths for filtering processing to obtain a bass signal and extract the original microphone signal; and then adjusts the output volume of the terminal based on the bass signal. The present application ensures that the microphone signal will not be distorted or the sound quality will not be degraded during the volume adjustment process by separating the bass signal and the original microphone signal; the present application significantly improves the speech recognition rate of the speech recognition chip while ensuring the sound quality of the microphone. BRIEF DESCRIPTION OF THE DRAWINGS

[0037] Figure 1 This is a flow chart of the volume adjustment method described in this application.

[0038] Figure 2 To adopt Figure 1 A system framework diagram of the volume adjustment method. DETAILED DESCRIPTION

[0039] The following is a further detailed description of a volume adjustment method, system and intelligent terminal of the present application in conjunction with specific embodiments and drawings.

[0040] Embodiment 1: See also Figure 1 , the present application provides a volume adjustment method, comprising the following steps: S100: Responding to a preset instruction, releasing a PWM signal of a preset frequency.

[0041] Furthermore, before executing step S100, the following steps are included: The preset action is recognized in real time by the gyroscope, so that when the preset action is recognized, a preset instruction is issued.

[0042] In some embodiments, the preset action may be a simple physical action, such as shaking, that is, the user triggers the volume adjustment function by shaking the device.

[0043] Furthermore, the gyroscope at the microphone end obtains the coordinate axis transformation of the device in space, for example, detects whether the coordinate axis is rapidly transformed back and forth a preset number of times (such as 3 times) or more, to determine whether the preset action exists.

[0044] It should be noted that technicians in this field can set a variety of preset actions, not just limited to shaking, thereby providing users with more choices and flexibility; the number of times the coordinate axis is transformed can also be set by technicians in this field according to actual application requirements, and is not limited to this.

[0045] In the above technical solution, the gyroscope recognizes preset actions, providing users with an intuitive and convenient way to adjust the volume. Users can start the volume adjustment function through simple gestures or actions without complicated operations; the real-time detection capability of the gyroscope ensures the rapid response of the volume adjustment function, and users can complete the volume adjustment in an instant, ensuring the voice recognition rate of the voice recognition chip.

[0046] In addition, users can customize preset actions according to their habits and preferences, enhancing the personalization and flexibility of the volume adjustment function.

[0047] Furthermore, before executing step S100, the method further includes: Presetting the preset frequency; wherein, comprising: The recording range of the current microphone is obtained to set the preset frequency based on the recording range.

[0048] In some embodiments, the preset frequency is, for example, 100 Hz. Those skilled in the art can adjust the preset frequency according to actual needs and recording range, which has strong flexibility and diversity.

[0049] In the above technical solution, by obtaining the recording range of the current microphone, the system can automatically adjust the frequency of the PWM signal to adapt to different recording environments. This environmental adaptability ensures the best volume adjustment effect in various environments; the preset frequency set based on the recording range can better balance the sound quality loss in the volume adjustment process and reduce unnecessary power consumption, which is especially important for portable devices and smart terminals and can extend battery life.

[0050] In addition, users do not need to manually adjust settings. The system can automatically adapt to environmental changes and provide a consistent and high-quality volume adjustment experience, enhancing user satisfaction and product competitiveness.

[0051] Furthermore, the current microphone includes at least a first channel and a second channel; after releasing the PWM signal in step S100, the following is further included: The PWM signal is low-pass filtered to obtain a low-frequency signal for transmission to the first path.

[0052] In some embodiments, the microphone device includes at least two signal channels, namely a first channel and a second channel, which can be regarded as channel A and channel B.

[0053] According to the preset frequency and after identifying the preset action, the system generates and releases a PWM signal of the preset frequency, performs low-pass filtering on the PWM signal, and extracts the low-frequency signal through the low-pass filter. The low-pass filter is set to a cutoff frequency of, for example, 100 Hz to ensure that the extracted low-frequency signal contains the main bass component.

[0054] Furthermore, the low-frequency signal processed by low-pass filtering is transmitted to channel A. This step ensures that the first received signal is mainly low-frequency components, providing a basis for subsequent bass enhancement and mixing processing.

[0055] In the above technical solution, the low-frequency component in the PWM signal is separated through low-pass filtering and transmitted to the first channel of the microphone. This signal separation method helps to better control the processing of low-frequency signals during volume adjustment, reduce distortion and sound quality degradation; low-pass filtering can remove high-frequency noise and interference in the PWM signal, thereby improving the purity of the low-frequency signal.

[0056] In addition, by transmitting the low-frequency signal to the first path and isolating it from other signal paths, cross-interference between different signal paths can be reduced and the overall performance and stability of the system can be improved.

[0057] S200: Receive a microphone signal, and perform mixing processing based on the PWM signal and the microphone signal to obtain a mixed signal.

[0058] Furthermore, the step S200 includes: The microphone signal is received through the second path, and the low-frequency signal and the microphone signal are output to the mixer through the first path and the second path respectively, so as to obtain a mixed signal through the mixer.

[0059] In some embodiments, the microphone device receives an original microphone signal through a second channel (channel B), and the original microphone signal includes high-frequency and intermediate-frequency components for subsequent mixing processing.

[0060] Further, channel A outputs the low-frequency signal processed by low-pass filtering to the mixer, while channel B outputs the received original microphone signal to the mixer.

[0061] Furthermore, the mixer receives the signals from the first channel and the second channel, performs mixing processing, and outputs the mixed signal (ie, the mixed signal) to the power amplifier module.

[0062] It should be noted that during the mixing process, the system can also make dynamic adjustments based on the recording range and preset frequency to ensure the integrity and consistency of the signal.

[0063] In the above technical solution, the signal separation and integration are achieved by transmitting the low-frequency signal and the original microphone signal to the mixer through different paths respectively. This separation process helps to retain the details of the original signal, while introducing the low-frequency signal for volume adjustment to ensure the integrity of the sound quality; the introduction of the low-frequency signal makes the processing of the low-frequency signal more precise during the volume adjustment process, avoiding the sound quality degradation caused by directly amplifying or compressing the original signal. Through mixing processing, the volume can be dynamically adjusted while retaining the sound quality.

[0064] In addition, the use of a mixer provides a more flexible mixing control method, which can adjust the balance between different signal paths and adapt to different environmental changes. This flexibility enhances the adaptability of the system. This signal processing method is compatible with a variety of audio processing systems and is easy to integrate into existing audio processing equipment and smart terminals. Through simple hardware and software configuration, efficient volume adjustment function can be achieved, enhancing the market competitiveness of the product.

[0065] S300: Divide the mixed signal into two paths for filtering to obtain a bass signal and extract an original microphone signal.

[0066] Furthermore, the step S300 includes: A low-pass filter process is performed based on the mixed signal to obtain the bass signal; and a high-pass filter process is performed based on the mixed signal to extract the original microphone signal.

[0067] In some embodiments, the mixed signal output by the mixer is input to a low-pass filter. After passing through the low-pass filter, the bass signal in the mixed signal is extracted for subsequent bass enhancement and audio processing; wherein the low-pass filter is set to a cutoff frequency of, for example, 100 Hz, to ensure that the extracted signal is mainly a low-frequency component.

[0068] The mixed signal output by the mixer is simultaneously input to a high-pass filter. After passing through the high-pass filter, the high-frequency signal in the mixed signal, that is, the original microphone signal, is extracted for subsequent high-frequency signal processing and audio output; wherein the high-pass filter is set to a cutoff frequency of, for example, 150 Hz to ensure that the extracted signal is mainly a high-frequency component.

[0069] In the above technical solution, the separation of low-frequency and high-frequency components in the mixed signal is achieved through low-pass and high-pass filtering. This separation processing helps to process the bass and treble signals separately during the volume adjustment process, thereby achieving more precise volume control and sound quality optimization; low-pass filtering can effectively remove high-frequency noise and interference in the mixed signal, and improve the purity and clarity of the bass signal. At the same time, high-pass filtering can retain the high-frequency details of the original microphone signal, ensuring that the sound quality of the treble part is not affected.

[0070] In addition, by processing bass and treble signals separately, the system can more flexibly adjust the volume dynamically to adapt to different audio content. This dynamic adjustment mechanism further improves the accuracy of volume adjustment.

[0071] And, S400: adjusting the output volume of the terminal based on the bass signal.

[0072] Furthermore, before executing step S400, the following steps are included: The current output volume of the terminal is obtained, and the speech recognition rate is obtained based on the microphone signal.

[0073] In some embodiments, the current output volume is obtained through the current music player, and the current output volume is stored for subsequent volume adjustment judgment.

[0074] The microphone signal is processed by the voice recognition chip to obtain the current voice recognition rate, which is stored and used as a basis for judging voice quality and volume adjustment.

[0075] In the above technical solution, by obtaining the current output volume of the terminal, the system can monitor and adjust the volume in real time to adapt to different environments and personal preferences. This dynamic volume adjustment mechanism ensures the best auditory experience in all situations; obtaining the speech recognition rate helps the system evaluate the recognizability of the current speech signal. Based on this, the system can automatically adjust the audio processing parameters to improve the accuracy and efficiency of speech recognition.

[0076] In addition, by combining real-time monitoring of volume and speech recognition rate, the system can better understand environmental changes, thereby providing more intelligent audio processing and interactive experience; for example, when the speech recognition rate is low, the system can automatically increase the volume or perform noise reduction to enhance the recognition effect.

[0077] Furthermore, the step S400 includes: Determine whether the bass signal is greater than a preset threshold, and if so, adjust the output volume based on the speech recognition rate; otherwise, return to step S100.

[0078] In some embodiments, it is detected whether the intensity of the bass signal is greater than a preset threshold (for example, the threshold is set to 100 mV); if so, the volume adjustment process is entered; otherwise, the process returns to step S100 and restarts the volume adjustment process.

[0079] Adjust output volume based on speech recognition rate, including: Analyze the current speech recognition rate and determine the speech quality. If the speech recognition rate is high (for example, the speech recognition rate is greater than 85%), gradually increase the output volume to improve the clarity and loudness of the audio output. If the speech recognition rate is low (for example, the speech recognition rate is less than 70%), gradually reduce the output volume to reduce noise and background interference.

[0080] In the above technical solution, by judging whether the bass signal exceeds the preset threshold, the system can automatically identify and adjust the volume of the audio output. This automatic adjustment mechanism ensures that the volume will not be too loud when the bass signal is strong, thereby avoiding sound distortion. By combining the voice recognition rate to adjust the volume, the system can dynamically adjust the volume according to the clarity and recognizability of the voice signal. This adjustment method effectively improves the accuracy of voice recognition, especially when the voice recognition rate is low. The system can enhance the clarity of the signal by increasing the volume, thereby enhancing the voice recognition rate of the voice recognition chip.

[0081] In addition, automatic volume adjustment and voice recognition-driven optimization make voice recognition more intelligent and personalized, and the system can automatically adjust the volume according to actual usage.

[0082] Embodiment 2: See also Figure 2 The present application also provides a system using the volume adjustment method described in Example 1, including: The microphone module is used to release a PWM signal of a preset frequency in response to a preset instruction; and receive a microphone signal, and perform mixing processing based on the PWM signal and the microphone signal to obtain a mixed signal to output to the power amplifier module.

[0083] In some embodiments, a preset action is recognized in real time by a gyroscope, so that when the preset action is recognized, a preset instruction is issued. The preset action can be a simple physical action, such as shaking, that is, the user triggers the volume adjustment function by shaking the device. The gyroscope of the microphone module obtains the coordinate axis transformation of the device in space, for example, detects whether the coordinate axis is quickly transformed back and forth for a preset number of times (such as 3 times) or more, to determine whether the preset action exists.

[0084] It should be noted that those skilled in the art can set a variety of preset actions, not just limited to shaking; the number of times the coordinate axis is transformed can also be set by those skilled in the art according to actual application requirements, and is not limited to this.

[0085] Furthermore, it is necessary to perform low-pass filtering on the PWM signal to obtain a low-frequency signal, and then further perform mixing processing based on the low-frequency signal and the microphone signal to obtain a mixed signal.

[0086] The power amplifier module is used to filter the mixed signal into two paths to obtain a bass signal and extract the original microphone signal; and to adjust the output volume of the terminal based on the bass signal.

[0087] In some embodiments, it is necessary to determine whether the bass signal is greater than a preset threshold. If so, the output volume is adjusted based on the speech recognition rate; otherwise, the signal is returned to the microphone module.

[0088] Wherein, the speech recognition rate is obtained based on a microphone signal.

[0089] In the above technical solution, the microphone module is responsible for signal input and mixing processing, and the power amplifier module is responsible for filtering processing and volume adjustment. The two work together to achieve efficient volume control. Through mixing processing, the system can effectively integrate PWM signals and microphone signals to ensure signal integrity and consistency. The filtering processing of the power amplifier module further improves the separation of bass and treble signals, ensuring the purity of sound quality. Based on the strength judgment of the bass signal, the system can monitor and adjust the output volume in real time to avoid distortion caused by excessive microphone signals. This dynamic adjustment mechanism ensures the best voice recognition rate under various audio contents.

[0090] In addition, through automatic volume adjustment and voice recognition-driven optimization, the system can automatically adjust the volume according to actual usage to ensure the best listening experience in various environments; users do not need to manually adjust the volume, and get a more convenient and intelligent audio experience.

[0091] Embodiment three: The present application also provides an intelligent terminal, which includes a volume adjustment system. The volume adjustment system uses the volume adjustment method described in Example 1 to achieve volume control.

[0092] In summary, the present application provides a volume adjustment method, system and intelligent terminal; first, in response to a preset instruction, a PWM signal of a preset frequency is released; then a microphone signal is received, and mixing processing is performed based on the PWM signal and the microphone signal to obtain a mixed signal; the mixed signal is further divided into two paths for filtering processing to obtain a bass signal and extract the original microphone signal; and the output volume of the terminal is adjusted based on the bass signal. The present application separates the bass signal and the original microphone signal to ensure that the microphone signal will not be distorted or the sound quality will not be degraded during the volume adjustment process; the present application significantly improves the speech recognition rate of the speech recognition chip under the premise of ensuring the sound quality of the microphone.

[0093] Although example embodiments have been described herein with reference to the accompanying drawings, it should be understood that the above example embodiments are merely exemplary and are not intended to limit the scope of the present application to this. Those of ordinary skill in the art may make various changes and modifications therein without departing from the scope and spirit of the present application. All these changes and modifications are intended to be included within the scope of the present application as required by the appended claims.

[0094] Those of ordinary skill in the art will appreciate that the units and algorithm steps of each example described in conjunction with the embodiments disclosed herein can be implemented in electronic hardware, or a combination of computer software and electronic hardware. Whether these functions are performed in hardware or software depends on the specific application and design constraints of the technical solution. Professional and technical personnel can use different methods to implement the described functions for each specific application, but such implementation should not be considered to be beyond the scope of this application.

[0095] In the several embodiments provided in this application, it should be understood that the disclosed devices and methods can be implemented in other ways. For example, the device embodiments described above are only schematic, for example, the division of the units is only a logical function division, and there may be other division methods in actual implementation, such as multiple units or components can be combined or integrated into another device, or some features can be ignored or not executed.

[0096] The various component embodiments of the present application can be implemented in hardware, or in software modules running on one or more processors, or in a combination thereof. It should be understood by those skilled in the art that a microprocessor or digital signal processor (DSP) can be used in practice to implement some or all functions of some modules according to the embodiments of the present application. The application can also be implemented as a device program (e.g., computer program and computer program product) for executing a part or all of the methods described herein. Such a program implementing the present application can be stored on a computer-readable medium, or can have the form of one or more signals. Such a signal can be downloaded from an Internet website, or provided on a carrier signal, or provided in any other form.

[0097] It should be noted that, in this article, relational terms such as first and second, etc. are only used to distinguish one entity or operation from another entity or operation, and do not necessarily require or imply any such actual relationship or order between these entities or operations. Moreover, the terms "include", "comprise" or any other variants thereof are intended to cover non-exclusive inclusion, so that a process, method, article or device including a series of elements includes not only those elements, but also other elements not explicitly listed, or also includes elements inherent to such process, method, article or device. In the absence of further restrictions, the elements defined by the sentence "comprise a ..." do not exclude the existence of other identical elements in the process, method, article or device including the elements.

[0098] Although the description of the present application is carried out in conjunction with the above specific embodiments, it is obvious that those skilled in the art can make many substitutions, modifications and changes based on the above content. Therefore, all such substitutions, improvements and changes are included in the spirit and scope of the appended claims.

Claims

1. A volume adjustment method, characterized in that: The following steps are involved: S100: Responding to a preset instruction, releasing a PWM signal of a preset frequency; S200: receiving a microphone signal, and performing mixing processing based on the PWM signal and the microphone signal to obtain a mixed signal; S300: dividing the mixed signal into two paths for filtering to obtain a bass signal and extract an original microphone signal; And, S400: adjusting the output volume of the terminal based on the bass signal.

2. The volume adjustment method according to claim 1, characterized in that: Before executing step S100, the process includes: The preset action is recognized in real time by the gyroscope, so that when the preset action is recognized, a preset instruction is issued.

3. The volume adjustment method according to claim 1, characterized in that: Before executing step S100, the method further includes: Presetting the preset frequency; Among them, include: The recording range of the current microphone is obtained to set the preset frequency based on the recording range.

4. The volume adjustment method according to claim 1, characterized in that: The current microphone includes at least a first channel and a second channel; after releasing the PWM signal in step S100, the following is further included: The PWM signal is low-pass filtered to obtain a low-frequency signal for transmission to the first path.

5. The volume adjustment method according to claim 4, characterized in that: The step S200 includes: The microphone signal is received through the second path, and the low-frequency signal and the microphone signal are output to the mixer through the first path and the second path respectively, so as to obtain a mixed signal through the mixer.

6. The volume adjustment method according to claim 5, characterized in that: The step S300 includes: Performing low-pass filtering based on the mixed signal to obtain the bass signal; At the same time, high-pass filtering is performed based on the mixed signal to extract the original microphone signal.

7. The volume adjustment method according to claim 6, characterized in that: Before executing step S400, the process includes: The current output volume of the terminal is obtained, and the speech recognition rate is obtained based on the microphone signal.

8. The volume adjustment method according to claim 7, characterized in that: The step S400 includes: Determine whether the bass signal is greater than a preset threshold, and if so, adjust the output volume based on the speech recognition rate; otherwise, return to step S100.

9. A system using the volume adjustment method according to any one of claims 1 to 8, characterized in that: include: The microphone module is used to release a PWM signal of a preset frequency in response to a preset instruction; and receiving a microphone signal, and performing mixing processing based on the PWM signal and the microphone signal to obtain a mixed signal to output to a power amplifier module; The power amplifier module is used to filter the mixed signal into two paths to obtain a bass signal and extract the original microphone signal; and to adjust the output volume of the terminal based on the bass signal.

10. An intelligent terminal, characterized in that: The intelligent terminal includes a volume adjustment system, and the volume adjustment system implements volume control by using the volume adjustment method according to any one of claims 1 to 8.

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