CN109495649B - Volume adjusting method, system and storage medium - Google Patents
Volume adjusting method, system and storage medium Download PDFInfo
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- CN109495649B CN109495649B CN201811533623.XA CN201811533623A CN109495649B CN 109495649 B CN109495649 B CN 109495649B CN 201811533623 A CN201811533623 A CN 201811533623A CN 109495649 B CN109495649 B CN 109495649B
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
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- H04M1/00—Substation equipment, e.g. for use by subscribers
- H04M1/72—Mobile telephones; Cordless telephones, i.e. devices for establishing wireless links to base stations without route selection
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- H04M1/72448—User interfaces specially adapted for cordless or mobile telephones with means for adapting the functionality of the device according to specific conditions
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Abstract
The invention discloses a volume adjusting method, a system and a storage medium, wherein the method comprises the following steps: when a specific sound signal is received, acquiring the distance between a sound source corresponding to the specific sound signal and the sound control component; adjusting the volume of the output audio to the first volume corresponding to the distance between the sound source and the sound control component according to a preset volume table; the preset volume table at least comprises the corresponding relation between the distance between a sound source and the sound control assembly and the first volume. The invention carries out the volume adjustment operation by receiving the specific sound signal so as to avoid incorrect volume adjustment caused by other sound source factors and make the volume adjustment more pertinent; the distance between the sound source and the sound control assembly is calculated, so that the volume is adjusted to the volume suitable for the hearing of the user; the whole process is intelligently controlled by sound, and the volume is adjusted, so that the intelligent life requirement is met.
Description
Technical Field
The present invention relates to the field of electronic communication technologies, and in particular, to a volume adjustment method, system, and storage medium.
Background
With the development of science and technology, the functions of electronic products are more and more diversified and widely applied by people. The sound playing function is one of important functions, and users can generally use electronic products such as mobile phones and smart speakers to play media sounds, such as music and news, so that the entertainment life of the users is greatly enriched. In the use, people need adjust the volume of electronic product in order to adapt to own current environment, along with living more and more intelligent, the volume control mode of current electronic product is mostly manual regulation, can't satisfy user's demand.
Disclosure of Invention
The invention mainly aims to provide a volume adjusting method, a system and a storage medium, which can realize intelligent volume adjustment so as to meet the requirements of users.
The invention provides a volume adjusting method, which comprises the following steps:
when a specific sound signal is received, acquiring the distance between a sound source corresponding to the specific sound signal and a sound control component;
adjusting the volume of the output audio to a first volume corresponding to the distance between the sound source and the sound control component according to a preset volume table; the preset volume table at least comprises the corresponding relation between the distance between the sound source and the sound control assembly and the first volume.
Further, the sound control assembly comprises a microphone array, and the step of acquiring the distance between the sound source corresponding to the specific sound signal and the sound control assembly when the specific sound signal is received comprises the steps of:
when a specific sound signal is received by the microphone array, recording the time point of each microphone in the microphone array receiving the specific sound signal;
calculating the time difference of receiving a specific sound signal between the microphones in the microphone array according to each time point;
and acquiring the distance between the sound source corresponding to the specific sound signal and the sound control component according to the time difference.
Further, the step of obtaining the distance between the sound source corresponding to the specific sound signal and the sound control component according to the time difference includes:
reading a preset geometric relational expression of positions among the microphones, the sound source and the sound control assembly;
and calculating the distance between the sound source and the sound control component according to the geometrical relation and the time difference and a preset formula.
Further, according to the preset volume table, after the step of adjusting the volume of the output audio to the first volume corresponding to the distance between the sound source and the sound control component, the method includes:
detecting environmental noise to obtain a noise value in the environment;
calculating according to the noise value and a preset rule to obtain a second volume;
the volume of the output audio is adjusted to a second volume.
Further, according to the noise value, calculating according to a preset rule to obtain a second volume, including:
acquiring a weighting coefficient corresponding to the noise value according to a preset weighting coefficient table; the preset weighting coefficient table comprises a corresponding relation between a noise value and a weighting coefficient;
calculating weighted volume according to the noise value and the weighting coefficient; wherein, the weighted volume is a noise value and a weighting coefficient;
calculating according to the weighted volume to obtain the second volume; wherein the second volume is the first volume + the weighted volume.
Further, according to the preset volume table, after the step of adjusting the volume of the output audio to the first volume corresponding to the distance between the sound source and the sound control component, the method further comprises the following steps:
acquiring an audio signal of an output audio through an analog-to-digital converter and acquiring the volume of the audio signal;
judging whether the volume of the audio signal is within a preset range;
if the output audio is not in the preset range, the volume of the output audio is adjusted to be in the preset range by adjusting the amplification factor of the power amplifier.
Further, if the output audio is not within the preset range, adjusting the volume of the output audio to be within the preset range by adjusting the amplification factor of the power amplifier, including:
if the volume of the output audio is larger than the preset range, the amplification factor of the power amplifier is reduced to reduce the volume of the output audio to be within the preset range;
if the volume of the output audio is smaller than the preset range, the amplification factor of the power amplifier is increased to increase the volume of the output audio to be within the preset range.
The present invention also provides a volume adjustment system, comprising:
the acquisition module is used for acquiring the distance from a sound source corresponding to a specific sound signal to the sound control component when the specific sound signal is received;
the first adjusting module is used for adjusting the volume of the output audio to a first volume corresponding to the distance between the sound source and the sound control component according to a preset volume table; the preset volume table at least comprises the corresponding relation between the distance from the sound source to the sound control assembly and the first volume.
Further, the sound control assembly comprises a microphone array, and the acquisition module comprises:
the recording unit is used for recording the time point of each microphone in the microphone array receiving the specific sound signal when the specific sound signal is received by the microphone array;
a first calculation unit for calculating a time difference of receiving a specific sound signal between each microphone in the microphone array according to each time point;
and the first acquisition unit is used for acquiring the distance between the sound source corresponding to the specific sound signal and the sound control component according to the time difference.
The present invention also provides a computer-readable storage medium having stored thereon a computer program which, when executed by a processor, performs the steps of the volume adjustment method described above.
The invention carries out the volume adjustment operation by receiving the specific sound signal so as to avoid incorrect volume adjustment caused by other sound source factors and make the volume adjustment more pertinent; the method comprises the steps that the distance between a sound source and a sound control assembly is obtained, and a preset volume table is combined, so that the volume suitable for the hearing of a user is obtained, and the volume is adjusted to the volume suitable for the hearing of the user; the whole process is intelligently controlled by sound, and the volume is adjusted, so that the intelligent life requirement of a user is met.
Drawings
FIG. 1 is a schematic diagram illustrating steps of a volume adjusting method according to an embodiment of the present invention;
FIG. 2 is a diagram illustrating mathematical modeling of an embodiment of a volume adjustment method according to the present invention;
FIG. 3 is a schematic view illustrating the operation flow of the interior of the sound control module according to an embodiment of the volume adjustment method of the present invention;
FIG. 4 is a schematic diagram of a volume adjustment system according to an embodiment of the present invention;
FIG. 5 is a schematic structural diagram of an acquisition module in an embodiment of a volume adjustment system according to the present invention;
fig. 6 is a schematic structural diagram of a first obtaining unit in an embodiment of a volume adjusting system according to the present invention;
FIG. 7 is a schematic structural diagram of a volume adjustment system according to another embodiment of the present invention;
FIG. 8 is a schematic structural diagram of a computing module in an embodiment of a volume adjustment system according to the invention;
FIG. 9 is a schematic structural diagram of a volume adjustment system according to another embodiment of the present invention;
FIG. 10 is a schematic structural diagram of an adjustment module in an embodiment of a volume adjustment system according to the invention;
fig. 11 is a reference diagram of a distance range between an intelligent terminal and a sound source according to an embodiment of the volume adjustment method of the present invention.
The implementation, functional features and advantages of the objects of the present invention will be further explained with reference to the accompanying drawings.
Detailed Description
It should be understood that the specific embodiments described herein are merely illustrative of the invention and are not intended to limit the invention.
Referring to fig. 1, an embodiment of a volume adjustment method according to the present invention includes:
s1, when receiving the specific sound signal, obtaining the distance between the sound source corresponding to the specific sound signal and the sound control component;
s2, adjusting the volume of the output audio to a first volume corresponding to the distance between the sound source and the sound control component according to a preset volume table; the preset volume table at least comprises the corresponding relation between the distance between the sound source and the sound control assembly and the first volume.
In the above step S1, the present invention can be applied to a smart terminal, but is not limited to a smart terminal; the sound control component can be integrated on an intelligent terminal, and the intelligent terminal comprises any terminal equipment such as a mobile phone, a tablet Personal computer, a notebook computer, a Personal Digital Assistant (PDA), a Point of Sales (POS), a vehicle-mounted computer, an intelligent sound box, a multifunctional telephone and the like; the specific sound signal may be an instruction issued by the user, and the instruction may include a keyword, such as "wow", "Hi, Alexa", "too loud", "volume up point", or the like; the keywords may also be keywords suitable for the scene according to the change of the scene, for example, when the user connects to the smart speaker through the mobile terminal to perform a mobile call, and when the smart speaker determines that the call is connected, the user says "feed, you! When keywords are equal, the intelligent sound box can automatically adjust the volume; the sound signal is received by a microphone array in the sound control assembly, and the microphone array at least comprises two microphones; when each microphone of the microphone array receives a sound signal, recording the time when each microphone respectively receives the sound signal; the sound source is a place for emitting a specific sound signal; the distance is the distance from the sound emitting point to the center of the sound control assembly, and preferably, the center of the sound control assembly is the same as the center of the microphone array; and calculating the time difference of sound signals received by each microphone, and determining the position of a sound source corresponding to the specific sound signal from the sound control component by adopting a time delay sound source positioning technology.
Referring to fig. 11, in step S2, the distance between the sound source and the sound control unit in the preset volume scale may be a distance range, the first volume in the preset volume scale may be a volume suitable for a normal hearing range of a person, and the preset volume scale, the volume output table including a relationship between the distance range and the first volume, for example: establishing a volume output reference data table for the output volumes corresponding to the equidistant ranges of 0-0.5m, 0.5-1m, 1-1.5m, 1.5-2m, 2-2.5m and 2.5-3 m; after the distance between the sound source and the sound control assembly is obtained through calculation, a first volume corresponding to the distance range of the distance is obtained through table lookup, and then the sound control assembly outputs audio frequency in the first volume; as shown in fig. 11, the distance range is a circular range, for example: and if the distance between the calculated sound source and the intelligent terminal is 0.8m, the distance range is 0.5-1m, and the first volume is a volume value corresponding to 0.5-1 m.
The invention carries out the volume adjustment operation by receiving the specific sound signal so as to avoid incorrect volume adjustment caused by other sound source factors and make the volume adjustment more pertinent; the method comprises the steps that the distance between a sound source and a sound control assembly is obtained, and a preset volume table is combined, so that the volume suitable for the hearing of a user is obtained, and the volume is adjusted to the volume suitable for the hearing of the user; the whole process is intelligently controlled by sound, and the volume is adjusted, so that the intelligent life requirement of a user is met.
In an embodiment of the present invention, the step S1 of acquiring the distance between the sound source corresponding to the specific sound signal and the sound control module when the specific sound signal is received includes:
s11, when a specific sound signal is received by the microphone array, recording the time point of each microphone in the microphone array receiving the specific sound signal;
s12, calculating a time difference between each microphone in the microphone array when receiving a specific sound signal according to each time point;
and S13, acquiring the distance between the sound source corresponding to the specific sound signal and the sound control component according to the time difference.
In step S11, since each microphone in the microphone array has a different position from the sound source, the time point of receiving a specific sound signal is different, and thus the distance between the sound source and the sound control module is obtained by using the time-delay sound source localization technique according to the time point of receiving the sound signal by each microphone.
In step S12, the time difference is a time difference between two microphones of the microphone array receiving the sound signals, and taking the three microphones as an example, the time points at which the first microphone, the second microphone and the third microphone receive the sound signals are a, b and c, respectively, so that the time difference between the first microphone and the second microphone receiving the sound signals is an absolute value of a-b, i.e. a positive number, and the time differences between the other microphones are calculated in the same manner.
In step S13, the sound propagation speed may be obtained by looking up a table to obtain different propagation speeds corresponding to different propagation media, and under the condition of known speed and time difference, the distance difference from the sound source to different microphones may be obtained; the distance from each microphone to the center of the sound control assembly can be used for installing a microphone array according to a preset distance when the sound control assembly is produced; according to the geometric relation that forms between each microphone and the sound source again, the distance difference can calculate the distance that obtains the sound source to the sound control subassembly, and then realizes the volume of intelligent regulation sound control subassembly according to predetermineeing the sound scale to do not need the manual regulation volume of user, easy operation accords with user's demand more, promotes user experience.
In an embodiment of the present invention, the step S13 of obtaining the distance between the sound source corresponding to the specific sound signal and the smart speaker according to the time difference includes:
s131, reading a preset geometric relational expression of positions among the microphones, the sound source and the sound control assembly;
and S132, calculating the distance between the sound source and the sound control component according to the geometrical relational expression and the time difference and a preset formula.
Referring to fig. 2, in step S131, the microphone array is a four-microphone circular array (it is understood that other shapes and numbers of microphone arrays are also possible), the microphones A, B, C, D are defined on a circle, the center O of the circle is the center of the sound control assembly, and a rectangular coordinate system with an X axis and a Y axis is established by taking a circle center O as an origin, the radius of the circle is r, the distance from the sound source Q to the circle center is s, the included angle between the straight line where the circle center O and the sound source Q are positioned and the X axis of the coordinate axis is theta, the distances from the sound source Q to the microphone A, B, C are s1, s2 and s3 respectively, wherein r is the distance from each microphone to the center of the sound control assembly, and is a preset radius when the sound control assembly is produced, that is, the known quantity, in the non-circular microphone array, the distances from the microphones to the center of the sound control assembly may not be equal, but are the known quantity preset during production and manufacturing; according to the cosine theorem in trigonometric functions, the following can be obtained:
further, a geometric relation of the distance between the microphone a and the sound source Q is obtained:
geometrical relation of distance between the microphone B and the sound source Q:
and the geometrical relation of the distance of the microphone B from the sound source Q:
and (3) prestoring the geometric relational expression in a sound control assembly, reading the geometric relational expression when necessary, and calculating by using a geometric algorithm.
In step S132, the speed of sound propagation is defined as v, t, by taking the four-microphone circular array as an exampleBA、tBCThe time difference of the microphone B and the microphone A receiving the sound signals and the time difference of the microphone B and the microphone C receiving the sound signals are respectively; according to the plane geometry, the following can be known: vtBA＝s2-s1、vtBC＝s2-s3The geometrical relation according to the distances between the microphones and the sound source is obtained as follows:
wherein v, tBA, tBC, r are known quantities, so the distance s between the sound source Q and the center O of the sound control assembly can be obtained; the problem is simplified by converting the geometric problem into an algebraic problem, a specific numerical value of the distance s between the sound source Q and the center O of the sound control assembly is obtained through algebraic operation, the position relation becomes visual, the numerical value can be recognized by the sound control assembly, the volume is adjusted to be the volume within the hearing range of the user, and the user experience is improved.
In an embodiment of the present invention, after the step S2 of adjusting the volume of the output audio to the first volume corresponding to the distance between the sound source and the sound control assembly according to the preset volume table, the method includes:
s3, detecting environmental noise to obtain a noise value in the environment;
s4, calculating according to the noise value and a preset rule to obtain a second volume;
s5, adjusting the volume of the output audio to a second volume.
In step S3, the environmental noise is a background noise of the current environment of the user; the environmental noise has certain influence on the volume, particularly in the daytime, the environmental noise in the daytime generally floats at 45dB when the environment is indoors, and the environmental noise at night generally floats at 20 dB; in some embodiments, the ambient noise may be detected by a microphone array and an analog signal of the detected ambient noise is converted to a digital signal by an analog-to-digital converter (ADC), resulting in a noise value.
In the above-described steps S4 and S5, the volume of the output audio is adjusted to the second volume with respect to the measured noise value; the preset rule is that a certain volume value is added according to a noise value on the basis of the first volume of the current output audio, and the certain volume value can be the noise value or can be obtained by converting the noise value. The influence of the environmental noise is counteracted by increasing a certain volume value on the basis of the first volume, so that the audio volume output by the sound control assembly is the volume which is most suitable for the hearing range of a user in the current environment, the intelligent control and volume adjustment in various environments are realized, and the user experience is further improved. In an embodiment of the present invention, the step S4 of calculating the second volume according to the noise value and the preset rule, and adjusting the volume of the output audio to the second volume includes:
s41, acquiring a weighting coefficient corresponding to the noise value according to a preset weighting coefficient table; the preset weighting coefficient table comprises a corresponding relation between a noise value and a weighting coefficient;
s42, calculating weighted volume according to the noise value and the weighting coefficient; wherein, the weighted volume is a noise value and a weighting coefficient;
s43, calculating a second volume according to the weighted volume; wherein the second volume is the first volume + the weighted volume.
In the step S41, the preset weighting factor table is obtained by the developer through experiments, and the measurement rule may be that according to a statistical analysis method, that is, a plurality of background noise values in different environments are measured, and according to the principle of normal distribution in mathematical statistics, the hearing ability of a normal person (that is, the influence of noise is effectively reduced) that how much weight the volume with which the current volume is most suitable for being added in different noise environments is obtained, so as to determine the weighting factor.
In step S42, the weighted volume is a product of the noise value and the weighting factor.
In the above step S43, the volume is increased by a certain weight (i.e., the product of the current noise value and the weighting coefficient) based on the first volume, so that the volume is increased to reduce the influence of noise on the output audio. How much the specific influence of the environmental noise on the output audio volume is a fuzzy problem, but the embodiment measures the volume which is most suitable for the hearing range of normal people under different noise environments through experiments, reasonably concreties the fuzzy problem, and provides a specific solution of the fuzzy problem, namely, the volume with certain weight is increased on the basis of the first volume, thereby offsetting the influence of the environmental noise on the output audio volume, the volume of the product of different weighting coefficients and noise values is increased aiming at different noise values, so as to avoid the excessive volume from being increased under the low noise environment, or avoid the too little volume from being increased under the high noise environment, so that the adjusted volume more conforms to the hearing range of users.
In an embodiment of the present invention, after the step S2 of adjusting the volume of the output audio to the first volume corresponding to the distance between the sound source and the sound control assembly according to the preset volume table, the method further includes:
s6, acquiring an audio signal of an output audio through an analog-to-digital converter and acquiring the volume of the audio signal;
s7, judging whether the volume of the audio signal is in a preset range;
and S8, if the output audio is not in the preset range, adjusting the volume of the output audio to be in the preset range by adjusting the amplification factor of the power amplifier.
Referring to fig. 3, in step S6, the audio signal of the output audio is amplified to the volume that needs to be adjusted by the current sound control module through the power amplifier, and the audio is output by the speaker, and simultaneously passes through the ADC and is collected and converted into a digital signal by the ADC, and the digital signal is transmitted to the processor.
In the step S7, the ADC converts the audio signal into a digital signal and transmits the digital signal to the processor, and the processor determines whether the volume of the digital signal is within a preset range; the preset range is obtained according to a preset error range of the volume to be adjusted of the current sound control assembly, namely, the volume +/-error value is required to be adjusted currently, for example, the volume is required to be adjusted to a first volume by the intelligent terminal, the audio with the first volume is output, and when the actual output volume is different from the first volume due to the loss of the line power and is out of the allowable error range, the volume of the audio is adjusted again to the volume within the allowable error range.
In step S8, since the line power loss causes the volume corresponding to the actually output audio signal to be out of the preset volume range that needs to be adjusted, the amplification factor of the power amplifier is adjusted to adjust the volume of the output audio to the preset range, so that the volume of the output audio is the volume that is most suitable for the hearing range of the user.
In an embodiment of the invention, if the output audio is not within the preset range, the step S8 of adjusting the amplification factor of the power amplifier to adjust the volume of the output audio to be within the preset range includes:
s81, if the volume of the output audio is larger than the preset range, the volume of the output audio is reduced by reducing the amplification factor of the power amplifier;
s82, if the volume of the output audio is smaller than the predetermined range, the volume of the output audio is increased by increasing the amplification factor of the power amplifier.
In the above step S81, the power of the output audio is too large so that the volume of the output is larger than the preset range, so that the amplification factor of the power of the output audio is reduced when the output audio passes through the power amplifier, thereby reducing the volume of the output audio to be within the preset range.
In the above step S82, since the power of the output audio is too small, the output volume is smaller than the preset range, and the amplification factor of the power of the output audio is increased when the output audio passes through the power amplifier, thereby increasing the volume of the output audio to be within the preset range.
Referring to fig. 4, an embodiment of the volume adjustment system of the present invention includes:
the system comprises an acquisition module 1, a sound control component and a processing module, wherein the acquisition module is used for acquiring the distance from a sound source corresponding to a specific sound signal to the sound control component when the specific sound signal is received;
the first adjusting module 2 is used for adjusting the volume of the output audio to a first volume corresponding to the distance between the sound source and the sound control component according to a preset volume table; the preset volume table at least comprises the corresponding relation between the distance from the sound source to the sound control assembly and the first volume.
In the acquisition module 1, the sound control component may be integrated in the terminal, and the terminal includes any terminal devices such as a mobile phone, a tablet computer, a notebook computer, a PDA (Personal Digital Assistant), a POS (Point of Sales), a vehicle-mounted computer, an intelligent sound box, and a multifunctional telephone; the specific sound signal may be an instruction issued by the user, and the instruction may include a keyword, such as "wow", "Hi, Alexa", "too loud", "volume up point", or the like; the keywords may also be keywords suitable for the scene according to the change of the scene, for example, when the user connects to the smart speaker through the mobile terminal to perform a mobile call, and when the smart speaker determines that the call is connected, the user says "feed, you! When the key words are equal, the intelligent sound box can automatically adjust the volume; the sound signal is received by the microphone array of the sound control assembly, and the microphone array at least comprises two microphones; recording the time of reception when each microphone of the microphone array receives a sound signal; the sound source is a place for emitting a specific sound signal; the distance is the distance from the sound emitting point to the center of the sound control assembly, and preferably, the center of the sound control assembly is the same as the center of the microphone array; and calculating the time difference of sound signals received by each microphone, and determining the position of a sound source corresponding to the specific sound signal from the sound control component by adopting a time delay sound source positioning technology.
In the first adjusting module 2, a distance between the sound source and the sound control component in the preset volume meter may be a distance range, a first volume in the preset volume meter may be a volume suitable for a hearing range of a normal person, and the preset volume meter is a volume output meter including a corresponding relationship between the distance range and the first volume; after the distance between the sound source and the sound control assembly is calculated, the first adjusting module 2 obtains a first volume corresponding to the distance range of the distance through table lookup, and then the sound control assembly outputs audio at the first volume.
Referring to fig. 5, in an embodiment of the present invention, the obtaining module 1 includes:
a recording unit 11, configured to record, when a specific sound signal is received by a microphone array on the sound control assembly, a time point at which each microphone in the microphone array receives the specific sound signal;
a first calculation unit 12 for calculating a time difference of receiving a specific sound signal between each microphone in the microphone array according to each time point;
the first obtaining unit 13 is configured to obtain a distance between a sound source corresponding to the specific sound signal and the sound control component according to the time difference.
In the above-described recording unit 11, since the microphones in the microphone array are different in position from the sound source, the time point at which a specific sound signal is received is also different.
In the first calculating unit 12, the time difference is a time difference between two microphones of the microphone array receiving the sound signals, taking three microphones as an example, the time points of receiving the sound signals by the first microphone, the second microphone and the third microphone are respectively a, b and c, and the time difference between receiving the sound signals by the first microphone and the second microphone is an absolute value of a-b, i.e. a positive number, and the calculation methods of the time differences between other microphones are the same.
In the first obtaining unit 13, the propagation speed of the sound may be obtained by looking up a table to obtain different propagation speeds corresponding to different propagation media, and under the condition of known speed and time difference, the distance difference from the sound source to different microphones may be obtained; the distance from each microphone to the center of the sound control assembly can be obtained when the microphone array is installed; and then the distance from the sound source to the sound control component can be calculated according to the geometric relationship and the distance difference formed between each microphone and the sound source.
Referring to fig. 6, in an embodiment of the present invention, the first obtaining unit 13 includes:
a reading subunit 131, configured to read a preset geometric relation among the microphones, the sound source, and the sound control assembly;
the obtaining subunit 132 is configured to calculate, according to the geometric relationship and the time difference, a distance between the sound source and the sound control component according to a preset formula.
In the reading subunit 131, the microphone array is exemplified by a four-microphone circular array (it is understood that other shapes and numbers of microphone arrays are also possible), the microphone A, B, C, D is defined on a circle, the center O is the center of the sound control assembly, and a plane rectangular coordinate system is established by taking the circle center O as the origin, the radius of the circle is r, the distance from the sound source Q to the circle center is s, the included angle between the straight line of the circle center O and the sound source Q and the axis X of the coordinate axis is theta, the distances from the sound source Q to the microphone A, B, C are s1, s2 and s3 respectively, wherein r is the distance from each microphone to the center of the sound control assembly, and is a preset radius when the sound control assembly is produced, that is, the known quantity, in the non-circular microphone array, the distances from the microphones to the center of the sound control assembly may not be equal, but are the known quantity preset during production and manufacturing; according to the cosine theorem in trigonometric functions, the following can be obtained:
further, a geometric relation of the distance between the microphone a and the sound source Q is obtained:
geometrical relation of distance between the microphone B and the sound source Q:
and the geometrical relation of the distance of the microphone B from the sound source Q:
the geometric relational expression is pre-stored in the sound control component, and the reading subunit 131 reads the geometric relational expression as needed, and performs calculation by using a geometric algorithm.
In the calculation subunit 132, the propagation velocity of sound is defined as v, t, by taking the above-mentioned four-wheat circular array as an exampleBA、tBCThe time difference of the microphone B and the microphone A receiving the sound signals and the time difference of the microphone B and the microphone C receiving the sound signals are respectively; according to the plane geometry, the following can be known: vtBA＝s2-s1、vtBC＝s2-s3The geometrical relation according to the distances between the microphones and the sound source is obtained as follows:
wherein v, tBA、tBCAnd r are known quantities, the distance s between the sound source Q and the center O of the sound control unit can be obtained.
Referring to fig. 7, in an embodiment of the present invention, the system further includes:
the detection module 3 is used for detecting environmental noise to obtain a noise value in the environment;
the calculating module 4 is used for calculating according to the noise value and a preset rule to obtain a second volume;
and the second adjusting module 5 is used for adjusting the volume of the output audio to a second volume.
In the detection module 3, the environmental noise is a background noise of the current environment of the user; the environmental noise has certain influence on the volume, particularly in the daytime, the environmental noise in the daytime generally floats at 45dB when the environment is indoors, and the environmental noise at night generally floats at 20 dB; in some embodiments, the detection module 3 may be a microphone, and the ambient noise is detected by a microphone array, and an analog signal of the detected ambient noise is converted into a digital signal by an analog-to-digital converter (ADC), so as to obtain a noise value.
In the calculating module 4 and the second adjusting module 5, the calculating module 4 calculates a second volume according to the noise value measured by the detecting module 3, and the second adjusting module 5 adjusts the volume of the output audio to the second volume; the preset rule is that a certain volume value is added according to the noise value on the basis of the first volume of the current output audio, and the certain volume value can be the noise value or can be obtained by converting the noise value.
Referring to fig. 8, in an embodiment of the present invention, the calculating module 4 includes:
a second obtaining unit 41, configured to obtain a weighting coefficient corresponding to the noise value according to a preset weighting coefficient table; the preset weighting coefficient table comprises a corresponding relation between a noise value and a weighting coefficient;
a second calculation unit 42 for calculating a weighted volume based on the noise value and the weighting coefficient; wherein, the weighted volume is a noise value and a weighting coefficient;
an adjusting unit 43 for adjusting the volume of the output audio to a second volume according to the weighted volume; wherein the second volume is the first volume + the weighted volume.
In the second obtaining unit 41, the preset weighting coefficient table is obtained through experiments by developers, and the measurement rule may be according to a statistical analysis method, that is, measuring a plurality of background noise values in different environments, and using a principle of normal distribution in mathematical statistics, obtaining how much weighted volume is to be added to the hearing ability of a normal person (that is, effectively reducing the influence of noise) on the current volume in different noise environments, so as to determine the weighting coefficient.
In the second calculation unit 42, the weighted sound volume is a product of a noise value and a weighting coefficient.
In the above-mentioned adjusting unit 43, the adjusting unit 43 increases the volume with a certain weight (i.e. the product of the current noise value and the weighting coefficient) on the basis of the first volume, thereby increasing the volume to achieve the effect of reducing the noise on the output audio.
Referring to fig. 9, in an embodiment of the present invention, the system further includes:
the acquisition module 6 is used for acquiring an audio signal of an output audio through the analog-to-digital converter and acquiring the volume of the audio signal;
the judging module 7 is used for judging whether the volume of the audio signal is within a preset range;
and the adjusting module 8 is used for adjusting the volume of the output audio to be within a preset range by adjusting the amplification factor of the power amplifier if the output audio is not within the preset range.
In the acquisition module 6, the audio signal of the output audio is amplified to the volume required to be adjusted by the current sound control component through the power amplifier, the audio is output by the loudspeaker, and meanwhile, the audio signal passes through the ADC, is acquired and converted into a digital signal by the ADC, and the digital signal is transmitted to the processor.
In the judging module 7, the acquisition module 6 converts the audio signal into a digital signal through the ADC and transmits the digital signal to the judging module 7, and the judging module 7 judges whether the volume of the digital signal meets a preset range; the preset range is obtained according to a preset error range of the volume to be adjusted of the current sound control assembly, namely, the volume +/-error value is required to be adjusted currently, for example, the volume is required to be adjusted to a first volume by the intelligent terminal, the audio with the first volume is output, and when the actual output volume is different from the first volume due to the loss of the line power and is out of the allowable error range, the volume of the audio is adjusted again to the volume within the allowable error range.
In the adjusting module 8, due to the power loss of the line, the volume corresponding to the actually output audio signal is not in the preset volume range that needs to be adjusted, so the volume of the output audio is adjusted to the preset range by adjusting the amplification factor of the power amplifier.
Referring to fig. 10, in an embodiment of the present invention, the adjusting module 8 includes:
the reducing unit 81 is configured to reduce the volume of the output audio to a preset range by reducing the amplification factor of the power amplifier if the volume of the output audio is greater than the preset range;
the increasing unit 82 increases the volume of the output audio to a preset range by increasing the amplification factor of the power amplifier if the volume of the output audio is smaller than the preset range.
In the above-mentioned reducing unit 81, the power of the output audio is too large so that the volume of the output is larger than the preset range, so that the amplification factor of the power of the output audio is reduced when the output audio passes through the power amplifier, thereby reducing the volume of the output audio to be within the preset range.
In the above-mentioned increasing unit 82, the power of the output audio is too small so that the volume of the output is smaller than the preset range, so that the amplification factor of the power of the output audio is increased when the output audio passes through the power amplifier, thereby increasing the volume of the output audio to be within the preset range.
The embodiment of the intelligent terminal comprises a memory and a processor 8;
the memory stores a computer program, and the processor executes the computer program to realize the steps of the volume adjusting method.
In an embodiment of the present invention, the intelligent terminal further includes a sound control component coupled to the processor, where the sound control component includes a microphone array, a power amplifier, and an analog-to-digital converter;
the microphone array is used for receiving a sound signal and detecting environmental noise;
the power amplifier is used for adjusting the volume of the output audio;
the analog-to-digital converter is used for collecting the volume of output audio.
In an embodiment of the present invention, the intelligent terminal further includes a WiFi module coupled to the processor, and configured to be in communication connection with the mobile terminal.
An embodiment of the present invention provides a computer-readable storage medium, on which a computer program is stored, where the computer program, when executed by a processor, implements a volume adjustment method, including: when the processor receives a specific sound signal, the distance between a sound source corresponding to the specific sound signal and the sound control component is acquired; adjusting the volume of the output audio to a first volume corresponding to the distance between the sound source and the sound control component according to a preset volume table; the preset volume table at least comprises the corresponding relation between the distance between the sound source and the sound control assembly and the first volume.
According to the volume adjusting method, the volume adjusting operation is executed by receiving the specific sound signal, so that incorrect volume adjustment caused by other sound source factors is avoided, and the volume adjustment is more targeted; the method comprises the steps that the distance between a sound source and a sound control assembly is obtained, and a preset volume table is combined, so that the volume suitable for the hearing of a user is obtained, and the volume is adjusted to the volume suitable for the hearing of the user; the whole process is intelligently controlled by sound, and the volume is adjusted, so that the intelligent life requirement is met.
It will be understood by those skilled in the art that all or part of the processes of the methods of the embodiments described above can be implemented by hardware instructions of a computer program, which can be stored in a non-volatile computer-readable storage medium, and when executed, can include the processes of the embodiments of the methods described above. Any reference to memory, storage, database, or other medium provided herein and used in the examples may include non-volatile and/or volatile memory. Non-volatile memory can include read-only memory (ROM), Programmable ROM (PROM), Electrically Programmable ROM (EPROM), Electrically Erasable Programmable ROM (EEPROM), or flash memory. Volatile memory can include Random Access Memory (RAM) or external cache memory. By way of illustration and not limitation, RAM is available in a variety of forms such as Static RAM (SRAM), Dynamic RAM (DRAM), Synchronous DRAM (SDRAM), double-rate SDRAM (SSRSDRAM), Enhanced SDRAM (ESDRAM), synchronous link (Synchlink) DRAM (SLDRAM), Rambus Direct RAM (RDRAM), direct bus dynamic RAM (DRDRAM), and bus dynamic RAM (RDRAM).
It should be noted that, in this document, the terms "comprises," "comprising," or any other variation thereof, are intended to cover a non-exclusive inclusion, such that a process, apparatus, article, or method that comprises a list of elements does not include only those elements but may include other elements not expressly listed or inherent to such process, apparatus, article, or method. Without further limitation, an element defined by the phrase "comprising an … …" does not exclude the presence of other like elements in a process, apparatus, article, or method that includes the element.
The above description is only a preferred embodiment of the present invention, and not intended to limit the scope of the present invention, and all modifications of equivalent structures and equivalent processes, which are made by using the contents of the present specification and the accompanying drawings, or directly or indirectly applied to other related technical fields, are included in the scope of the present invention.
Claims (5)
1. A method of adjusting volume, comprising:
when a specific sound signal is received, acquiring the distance between a sound source corresponding to the specific sound signal and a sound control component;
adjusting the volume of the output audio to a first volume corresponding to the distance between the sound source and the sound control component according to a preset volume table; the preset volume table at least comprises the corresponding relation between the distance between a sound source and the sound control component and the first volume;
acquiring an audio signal of the output audio through an analog-to-digital converter and acquiring the volume of the audio signal;
judging whether the volume of the audio signal is within a preset range;
if the output audio is not in the preset range, adjusting the volume of the output audio to be in the preset range by adjusting the amplification factor of the power amplifier;
the distance between a sound source in the preset sound meter and the sound control assembly is a distance range;
if the output audio is not in the preset range, the step of adjusting the volume of the output audio to be in the preset range by adjusting the amplification factor of the power amplifier comprises the following steps:
if the volume of the output audio is larger than a preset range, reducing the volume of the output audio to be within the preset range by reducing the amplification factor of the power amplifier;
if the volume of the output audio is smaller than the preset range, increasing the amplification factor of the power amplifier to increase the volume of the output audio to be within the preset range;
the sound control assembly comprises a microphone array, and the step of acquiring the distance between a sound source corresponding to a specific sound signal and the sound control assembly when the specific sound signal is received comprises the following steps:
when a specific sound signal is received by the microphone array, recording the time point of each microphone in the microphone array receiving the specific sound signal;
calculating the time difference of receiving the specific sound signal among the microphones in the microphone array according to each time point;
acquiring the distance between a sound source corresponding to the specific sound signal and the sound control component according to the time difference;
the step of obtaining the distance between the sound source corresponding to the specific sound signal and the sound control component according to the time difference includes:
reading a preset geometric relational expression of positions among the microphones, the sound source and the sound control assembly;
calculating the distance between the sound source and the sound control component according to a preset formula and the geometric relational expression and the time difference;
the microphone array is a four-microphone circular array, the microphones comprise a microphone A, a microphone B and a microphone C, and the step of calculating the distance between the sound source and the sound control component according to the geometric relation and the time difference and a preset formula comprises the following steps:
the distance between the sound source and the center of the sound control assembly is calculated by the following formula,
wherein, tBAFor the microphones B and A to receive the soundThe time difference of the signals; t is tBCThe time difference between the microphone B and the microphone C receiving the sound signals; v is the propagation velocity of sound; and s is the distance between the sound source and the center of the sound control assembly.
2. The volume adjustment method according to claim 1, wherein the step of adjusting the volume of the output audio to a first volume corresponding to the distance between the sound source and the sound control assembly according to a preset volume table comprises:
detecting environmental noise to obtain a noise value in the environment;
calculating according to the noise value and a preset rule to obtain a second volume;
adjusting the volume of the output audio to a second volume.
3. The volume adjusting method according to claim 2, wherein the step of calculating the second volume according to the noise value and a preset rule comprises:
acquiring a weighting coefficient corresponding to the noise value according to a preset weighting coefficient table; wherein the preset weighting coefficient table comprises a corresponding relation between the noise value and the weighting coefficient;
calculating weighted volume according to the noise value and the weighting coefficient; wherein the weighted volume is a noise value and a weighting coefficient;
calculating the second volume according to the weighted volume; wherein the second volume is the first volume + the weighted volume.
4. A volume adjustment system, comprising:
the system comprises an acquisition module, a sound control module and a processing module, wherein the acquisition module is used for acquiring the distance from a sound source corresponding to a specific sound signal to a sound control component when the specific sound signal is received;
the first adjusting module is used for adjusting the volume of the output audio to a first volume corresponding to the distance between the sound source and the sound control assembly according to a preset volume table; the preset volume table at least comprises the corresponding relation between the distance from a sound source to the sound control component and the first volume;
the acquisition module is used for acquiring an audio signal of an output audio through the analog-to-digital converter and acquiring the volume of the audio signal;
the judging module is used for judging whether the volume of the audio signal is within a preset range;
the adjusting module is used for adjusting the volume of the output audio to be within a preset range by adjusting the amplification factor of the power amplifier if the output audio is not within the preset range;
the distance between a sound source in the preset sound meter and the sound control assembly is a distance range;
the adjustment module includes:
the reducing unit is used for reducing the volume of the output audio to be within a preset range by reducing the amplification factor of the power amplifier if the volume of the output audio is larger than the preset range;
the increasing unit increases the volume of the output audio to be within a preset range by increasing the amplification factor of the power amplifier if the volume of the output audio is smaller than the preset range; the sound control assembly includes a microphone array, the acquisition module includes:
the recording unit is used for recording time points of receiving specific sound signals by each microphone in the microphone array when the specific sound signals are received by the microphone array;
a first calculation unit configured to calculate a time difference between the microphones of the microphone array at which the specific sound signal is received, according to each of the time points;
the first acquisition unit is used for reading a preset geometric relational expression of positions among the microphones, the sound source and the sound control assembly; calculating the distance between the sound source and the sound control component according to a preset formula and the geometric relational expression and the time difference;
the microphone array is a four-microphone circular array, the microphones comprise a microphone A, a microphone B and a microphone C, and the step of calculating the distance between the sound source and the sound control component according to the geometric relational expression and the time difference and a preset formula comprises the following steps:
the distance between the sound source and the center of the sound control assembly is calculated by the following formula,
wherein, tBAThe time difference between the microphone B and the microphone A when the sound signals are received; t is tBCThe time difference between the microphone B and the microphone C receiving the sound signals; v is the propagation velocity of sound; and s is the distance between the sound source and the center of the sound control assembly.
5. A computer-readable storage medium, on which a computer program is stored, which, when being executed by a processor, carries out the steps of the method according to any one of claims 1-3.
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| CN111190566A (en) * | 2019-12-27 | 2020-05-22 | 拉克诺德(深圳)科技有限公司 | Volume control method and device, computer equipment and storage medium |
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| CN111510905A (en) * | 2020-05-04 | 2020-08-07 | 宋彦震 | Method for adaptively adjusting sound size of player based on user distance |
| CN111930336A (en) * | 2020-07-29 | 2020-11-13 | 歌尔科技有限公司 | Volume adjusting method and device of audio device and storage medium |
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