CN111343540B - Piano audio processing method and electronic equipment - Google Patents
Piano audio processing method and electronic equipment Download PDFInfo
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- CN111343540B CN111343540B CN202010145978.2A CN202010145978A CN111343540B CN 111343540 B CN111343540 B CN 111343540B CN 202010145978 A CN202010145978 A CN 202010145978A CN 111343540 B CN111343540 B CN 111343540B
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Abstract
The invention provides a processing method of piano audio and electronic equipment, wherein the method comprises the following steps: acquiring audio data; determining whether the audio data is piano audio according to the narrowband characteristic information of the audio data and the loudspeaker diaphragm speed corresponding to the audio data; and under the condition that the audio data is piano audio, compressing the audio data by dynamically adjusting the parameters of the equalizer EQ. The method and the device can reduce the possibility of misjudgment when judging whether the audio is the piano audio, thereby improving the accuracy of piano audio judgment.
Description
Technical Field
The invention relates to the technical field of audio processing, in particular to a piano audio processing method and electronic equipment.
Background
With the development of electronic equipment, the micro loudspeaker is more widely applied to portable electronic equipment such as smart phones and internet of things terminals; because the loudspeaker adopts the design of a small cavity and the short-time input power is limited, when a large-volume audio signal is played, noise is easy to appear. When a piano is played, a speaker monomer generates a large amount of distortion under low-frequency and large-amplitude impact, so that sound is impure, or airflow noise is caused by airflow friction sound outlet holes in a speaker cavity. Therefore, the noise phenomenon is more obvious when the piano solo is played.
In order to eliminate the noise when playing the piano audio, the current processing method generally comprises: whether the piano audio is detected is judged by judging the energy difference between the high frequency and the middle and low frequency of the piano signal. In the processing mode, the high-frequency characteristics of other narrow-band signals are the same as those of a piano when the narrow-band signals are played, so that misjudgment is easy to occur when high-frequency energy is calculated, and the sound quality of other musical instruments or human voice can be influenced.
Disclosure of Invention
The invention provides a processing method of piano audio and electronic equipment, which aim to solve the problem that misjudgment is easy to occur when judging whether the piano audio is in the current mode of eliminating piano noise.
In order to solve the technical problem, the invention is realized as follows:
in a first aspect, an embodiment of the present invention provides a processing method for piano audio, which is applied to an electronic device, and the method includes:
acquiring audio data;
determining whether the audio data is piano audio according to the narrowband characteristic information of the audio data and the loudspeaker diaphragm speed corresponding to the audio data;
in the case where the audio data is piano audio, the audio data is subjected to compression processing by dynamically adjusting parameters of an Equalizer (EQ).
In a second aspect, an embodiment of the present invention further provides an electronic device, including:
the acquisition module is used for acquiring audio data;
the first determining module is used for determining whether the audio data is piano audio according to the narrow-band characteristic information of the audio data and the loudspeaker diaphragm speed corresponding to the audio data;
and the processing module is used for compressing the audio data by dynamically adjusting the parameters of the EQ under the condition that the audio data is the piano audio.
In a third aspect, an embodiment of the present invention further provides an electronic device, which includes a processor, a memory, and a computer program stored on the memory and executable on the processor, and when the computer program is executed by the processor, the steps of the processing method for piano audio described above are implemented.
In a fourth aspect, the embodiment of the present invention further provides a computer-readable storage medium, on which a computer program is stored, and the computer program, when executed by a processor, implements the steps of the processing method for piano audio as described above.
In the embodiment of the invention, under the condition that the audio data is determined to be piano audio according to the narrow-band characteristic information of the audio data and the loudspeaker diaphragm speed corresponding to the audio data, the audio data is compressed by dynamically adjusting the parameters of EQ. Therefore, the possibility of misjudgment when the audio is judged to be the piano audio can be reduced, and the accuracy of piano audio judgment is improved.
Drawings
In order to more clearly illustrate the technical solutions of the embodiments of the present invention, the drawings needed to be used in the description of the embodiments of the present invention will be briefly introduced below, and it is obvious that the drawings in the following description are only some embodiments of the present invention, and it is obvious for those skilled in the art that other drawings can be obtained according to these drawings without inventive labor.
FIG. 1 shows a flow chart of a piano audio processing method of an embodiment of the invention;
FIG. 2 is a flowchart illustrating a process of calculating a slope of a narrowband feature frequency point according to an embodiment of the present invention;
FIG. 3 is a flow chart illustrating a process for calculating a velocity of a diaphragm of a loudspeaker according to an embodiment of the present invention;
FIG. 4 shows a flow chart for determining the resonant frequency and quality factor of a loudspeaker according to an embodiment of the invention;
FIG. 5 shows a schematic diagram of a loudspeaker amplitude model of an embodiment of the invention;
FIG. 6 is a flow chart showing a piano audio playing method according to an embodiment of the present invention;
fig. 7 is a schematic structural diagram of a micro-speaker according to an embodiment of the present invention;
FIG. 8 shows a block diagram of an electronic device of an embodiment of the invention;
fig. 9 is a schematic diagram showing a hardware configuration of an electronic device according to an embodiment of the present invention.
Detailed Description
The technical solutions in the embodiments of the present invention will be clearly and completely described below with reference to the drawings in the embodiments of the present invention, and it is obvious that the described embodiments are some, not all, embodiments of the present invention. All other embodiments, which can be derived by a person skilled in the art from the embodiments given herein without making any creative effort, shall fall within the protection scope of the present invention.
As shown in fig. 1, an embodiment of the present invention provides a processing method for piano audio, which is applied to an electronic device, and the method includes:
step 11: audio data is acquired.
Alternatively, the audio data may be any audio data to be played through a speaker.
Step 12: and determining whether the audio data is piano audio according to the narrowband characteristic information of the audio data and the loudspeaker diaphragm speed corresponding to the audio data.
Alternatively, the narrowband characteristic information may be spectral characteristic information smaller than a first predetermined frequency range, such as the first predetermined frequency may be 4 kHz. The loudspeaker diaphragm velocity may be a predicted or estimated diaphragm velocity corresponding to the audio data if played by the loudspeaker.
Step 13: and under the condition that the audio data is piano audio, compressing the audio data by dynamically adjusting parameters of EQ.
In the above scheme, when the audio data is determined to be piano audio according to the narrowband characteristic information of the audio data and the loudspeaker diaphragm speed corresponding to the audio data, the audio data is compressed by dynamically adjusting the parameters of the EQ, so as to reduce the possibility of erroneous judgment when judging whether the audio is piano audio, thereby improving the accuracy of piano audio judgment.
Optionally, step 12 may specifically include:
judging whether the narrowband characteristic information of the audio data meets a first preset condition or not;
judging whether the speed of the loudspeaker diaphragm corresponding to the audio data meets a second preset condition or not;
and if the narrow-band characteristic information meets the first preset condition and the speed of the loudspeaker diaphragm meets the second preset condition, determining that the audio data is the piano audio.
In this embodiment, the narrowband characteristic information of the audio data is used as a basis for judging whether the audio data is the piano audio, so that the possibility of misjudgment when the audio is judged to be the piano audio can be reduced, and the accuracy of piano audio judgment is improved; in addition, when the narrowband characteristic information is judged to meet the first preset condition and the speed of the loudspeaker diaphragm meets the second preset condition, the audio data is determined to be the piano audio, and the accuracy of piano audio judgment is further improved.
Optionally, the step of determining whether the narrowband feature information of the audio data meets a first preset condition may specifically include:
processing the audio data to obtain a frequency spectrum envelope;
determining the narrow-band characteristic information according to a region with the frequency smaller than the target frequency in the spectrum envelope;
calculating to obtain a narrow-band characteristic frequency point slope according to the narrow-band characteristic information;
and if the slope of the narrow-band characteristic frequency point exceeds a preset slope threshold, determining that the narrow-band characteristic information meets the first preset condition.
The following will specifically describe the steps of calculating the slope of the narrowband characteristic frequency point with reference to fig. 2:
step 201: acquiring a target frame signal, wherein the target frame signal is a frame signal of input audio data;
step 206, performing fourier transform on the windowed target frame signal to convert the target frame signal from a time domain to a frequency domain, thereby obtaining a spectrum envelope;
specifically, a region with the frequency less than or equal to 4KHz is selected from the spectrum envelope to serve as narrow-band characteristic information; selecting a first frequency point and a second frequency point (the specific frequency of the frequency point can be adjusted) in the narrow-band characteristic information; and smoothing the first frequency point and the second frequency point by a smoothing factor so as to prevent the influence of sudden change of signals.
208, determining a straight line according to the first frequency point and the second frequency point after the smoothing treatment; and determining the slope of the straight line as the slope of the narrow-band characteristic frequency point.
Optionally, the step of determining whether the speed of the loudspeaker diaphragm corresponding to the audio data meets a second preset condition may specifically include:
according to the loudspeaker modeling parameters, calculating to obtain the loudspeaker diaphragm speed corresponding to the audio data;
and if the speed of the loudspeaker diaphragm exceeds a preset speed threshold value, determining that the speed of the loudspeaker diaphragm meets the second preset condition.
Optionally, as shown in fig. 3, the step of calculating, according to the speaker modeling parameter, a diaphragm velocity of the speaker corresponding to the audio data may specifically include:
step 301: determining a loudspeaker impedance model filter coefficient according to the loudspeaker modeling parameter;
the loudspeaker modeling parameters, also called TS parameters, are centralized parameters representing a loudspeaker model, and comprise mass Mms, damping coefficient Rms, stiffness coefficient Kms, force compliance Cms, force factor Bl, direct current resistance Re, inductance Le and the like;
step 302: determining a loudspeaker admittance model filter coefficient according to the loudspeaker impedance model filter coefficient;
step 303: taking the input voltage corresponding to the audio data as an input signal of a loudspeaker admittance model filter to obtain the output current predicted by the loudspeaker admittance model filter;
step 304: and calculating to obtain the speed of the loudspeaker diaphragm according to the input voltage, the output current, and the force factor and the direct current resistance in the loudspeaker modeling parameters.
Specifically, according to the electrical formula, the calculation formula for the diaphragm velocity can be derived as follows:
wherein v is the diaphragm velocity; bl is a force factor; re is the direct flow resistance; u is the input voltage; i is the output current.
Optionally, step 13 may specifically include: and compressing the audio data by dynamically adjusting the center frequency, the suppression bandwidth and the gain of the EQ.
Thus, under the condition that the audio data are determined to be piano audio, the center frequency, the suppression bandwidth and the gain of the EQ can be dynamically adjusted according to the characteristics of the piano audio, only the resonance frequency F0 of the loudspeaker is compressed, and the signals of the rest frequency bands are kept unchanged, so that the output loudness is ensured.
Specifically, if the audio data is determined to be piano audio (such as audio in a piano solo state) according to the narrowband characteristic information of the audio data and the loudspeaker diaphragm speed corresponding to the audio data, calculating the dynamic gain of the EQ; if the audio data is determined not to be piano audio according to the narrowband characteristic information of the audio data and the loudspeaker diaphragm speed corresponding to the audio data, setting a gain multiple to be 1, and converting the gain into decibels to be used as parameters of EQ; if the audio data is in the no-piano state, the gain may be set to 0 dB. In particular, the embodiment of the present invention may also set the gain of the EQ to a fixed value according to the setting selection of the user.
Optionally, after step 11, the method may further include:
calculating the resonance frequency and the quality factor of the loudspeaker according to the audio data;
determining a center frequency and a rejection bandwidth of the EQ according to a resonant frequency and a quality factor of the speaker; specifically, the suppression bandwidth of the EQ may be determined according to the figure of merit; determining a center frequency of the EQ according to a resonant frequency of the speaker;
and calculating the gain of the EQ according to the diaphragm speed.
Optionally, after the step 11 and before the step 12, calculating a resonant frequency and a quality factor of the speaker according to the audio data; and determining the center frequency and rejection bandwidth of the EQ according to the resonance frequency and quality factor of the loudspeaker; after the step 12, a step of calculating the gain of the EQ according to the diaphragm velocity is performed.
Of course, the resonant frequency and the quality factor of the loudspeaker are calculated according to the audio data; determining a center frequency and a rejection bandwidth of the EQ according to a resonant frequency and a quality factor of the speaker; and calculating the gain of the EQ according to the diaphragm velocity, which may be performed after step 12.
Optionally, the step of calculating the resonant frequency and the quality factor of the speaker according to the audio data may specifically include:
acquiring a frame signal of the audio data as an input frame;
calculating a target maximum amplitude corresponding to the input frame;
if the target maximum amplitude is smaller than or equal to the first displacement, determining that a first resonant frequency corresponding to the first loudspeaker amplitude model is the resonant frequency of the loudspeaker, and determining that a first quality factor corresponding to the first loudspeaker amplitude model is the quality factor of the loudspeaker.
Optionally, the step of calculating the resonant frequency and the quality factor of the speaker according to the audio data may further include:
acquiring a frame signal of the audio data as an input frame;
calculating a target maximum amplitude corresponding to the input frame;
if the target maximum amplitude is larger than the first displacement corresponding to the first loudspeaker amplitude model, acquiring a first displacement, a first resonant frequency and a first quality factor corresponding to the first loudspeaker amplitude model, and acquiring a second displacement, a second resonant frequency and a second quality factor corresponding to the second loudspeaker amplitude model;
calculating to obtain the resonant frequency of the loudspeaker according to the first displacement, the first resonant frequency, the second displacement, the second resonant frequency and the target maximum amplitude, and calculating to obtain the quality factor of the loudspeaker according to the first displacement, the first quality factor, the second displacement, the second quality factor and the target maximum amplitude.
Specifically, as shown in fig. 4, a flow chart for determining the resonant frequency and quality factor of a speaker is given; the method specifically comprises the following steps:
step 401: acquiring a frame signal of the audio data as an input frame, and determining a loudspeaker amplitude model; alternatively, the loudspeaker amplitude model may be a first loudspeaker model (alternatively referred to as a small signal model with a resonant frequency of F1 and a quality factor of Q1) or a second loudspeaker model (alternatively referred to as a large signal model with a resonant frequency of F2 and a quality factor of Q2) as shown in fig. 5;
step 402: calculating the amplitude corresponding to the input frame signal;
step 403: calculating to obtain a target maximum amplitude Xmax according to the amplitude corresponding to the input frame signal;
step 404: judging whether the target maximum amplitude Xmax is larger than a first displacement X1 corresponding to the first loudspeaker model;
step 405: if the target maximum amplitude Xmax is less than or equal to the first displacement X1 corresponding to the first speaker model, it is determined that the currently input audio signal is a small signal or a high frequency, and no or little suppression of the EQ parameters is required, F1 may be determined as the resonant frequency of the speaker, and Q1 may be determined as the quality factor of the speaker;
step 406: calculating the resonance frequency and the quality factor of the loudspeaker according to the audio data if the target maximum amplitude Xmax is larger than the first displacement X1 corresponding to the first loudspeaker model;
the step of calculating the resonant frequency and the quality factor of the speaker according to the audio data may specifically include:
acquiring a first displacement, a first resonant frequency and a first quality factor corresponding to a first loudspeaker amplitude model; optionally, according to the speaker model 1, i.e. the small signal model, calculating a first displacement X1, a first resonant frequency F1 and a first quality factor Q1 corresponding to the small signal model;
acquiring a second displacement, a second resonant frequency and a second quality factor corresponding to the second loudspeaker amplitude model; optionally, according to the speaker model 2, that is, the large signal model, calculating a second displacement X2, a second resonant frequency F2 and a second quality factor Q2 corresponding to the large signal model;
calculating to obtain the resonant frequency of the loudspeaker according to the displacement, the first resonant frequency, the second displacement, the second resonant frequency and the target maximum amplitude, and calculating to obtain the quality factor of the loudspeaker according to the displacement, the first quality factor, the second displacement, the second quality factor and the target maximum amplitude.
Optionally, calculating the resonant frequency of the loudspeaker by using the following formula;
wherein F0 is the resonant frequency of the speaker, F1 is the first resonant frequency corresponding to the first speaker amplitude model, and X1 is the first displacement corresponding to the first speaker amplitude model; f2 is a second resonance frequency corresponding to the second speaker amplitude model, X2 is a second displacement corresponding to the second speaker amplitude model, and Q2 is a second quality factor corresponding to the second speaker amplitude model; xmax is the target maximum amplitude for the input signal.
Optionally, the quality factor of the speaker may also be calculated in the same manner, for example, F1 in the above formula is replaced by a first quality factor Q1 corresponding to the first speaker amplitude model, and F2 is replaced by a second quality factor Q2 corresponding to the second speaker amplitude model, so as to obtain the corresponding quality factor Q of the speaker.
Alternatively, in the case where the target maximum amplitude Xmax is greater than X1, if Xmax is greater than X2, i.e., around the resonance frequency, Xmax is likely to exceed X2, where Xmax is limited to within X2, and the resonance frequency and the quality factor of the speaker are calculated in the above manner.
Optionally, the step of calculating the gain of the EQ according to the diaphragm speed may specifically include:
calculating the gain of EQ by the following formula;
Z=1-v/vmax
where Z is the gain of EQ, v is the diaphragm velocity, and vmax is the maximum velocity of the diaphragm (e.g., the maximum velocity threshold for determining piano audio).
As shown in fig. 6, an embodiment of the present invention provides a flowchart of a piano audio playing method, which specifically includes:
step 601: the digital audio signal is input, i.e., the audio data is input.
Step 602: the method comprises the following steps of carrying out time delay buffering on an input digital audio signal, such as: the frame length is N milliseconds (N is more than or equal to 1).
Step 603: speaker diaphragm velocity prediction, i.e. estimating the velocity of the speaker diaphragm in case of said audio data as signal input; because the loudspeaker has speed resonance at the resonance frequency F0, the higher the diaphragm speed is, the more the air flow of the diaphragm compressed air is, and the more the cavity is prone to generate noise due to air flow friction; therefore, the speed of the diaphragm can be predicted according to the input signal, and the possibility of piano noise is judged according to the speed of the diaphragm; for a specific method for calculating the diaphragm speed, reference may be made to the above embodiments, and details are not repeated herein.
Step 604: calculating the frequency point slope of the narrow-band characteristics of the audio data; such as: windowing and Fourier transforming each frame of input signals to a frequency domain, acquiring narrow-band characteristic information (the frequency range is less than or equal to 4KHz) from a spectrum envelope, taking two or more frequency points, and calculating the slope of any two points; because the medium-high frequency energy of the human voice is stronger, whether the audio data is the piano audio is judged according to the narrow-band characteristic information of the audio data, so that misjudgment on the human voice singing can be reduced; for a specific calculation method of the narrow-band characteristic frequency point slope, reference may be made to the above embodiments, and details are not described here.
Step 605: calculating the maximum amplitude, the resonant frequency and the quality factor of the loudspeaker, wherein the amplitude response of the loudspeaker under different excitation signals is shown in FIG. 5, wherein a small signal model corresponds to a first resonant frequency F1, a first quality factor Q1 and a first displacement X1, and a large signal model corresponds to a second resonant frequency F2, a second quality factor Q2 and a second displacement X2; it can be seen from fig. 5 that when a large signal is input, the resonant frequency of the speaker shifts, and F2 is smaller than F1, it can be understood that the speaker parameters change when a large signal is input, and since the piano audio is suppressed to adjust the EQ parameters, the shift of the speaker resonant frequency must be accurately tracked to adjust the center frequency of the EQ; the suppression bandwidth of EQ can also be dynamically set following the transformation of the loudspeaker quality factor.
Of course, the scheme can also customize the F0 and Q value according to the input of the user, and a fixed F0 or Q value can also be set in the scheme. Specifically, the method for calculating the target maximum amplitude, the resonant frequency and the quality factor of the speaker is as described in the above embodiments, and will not be described herein again.
Step 606: under the conditions that the diaphragm speed in the step 603 is determined as the piano audio and the frequency point slope in the step 604 is determined as the piano audio, calculating the dynamic gain of EQ;
specifically, the maximum speed threshold of the vibrating diaphragm can be calculated through the second loudspeaker model 2, namely the large signal model, and the vibrating diaphragm speed corresponding to the piano audio frequency is calculated according to the multi-piano solo audio frequency; judging whether the piano solo is played according to the maximum speed threshold and the diaphragm speed (if the diaphragm speed is greater than the maximum speed threshold, the piano solo is determined); judging whether the piano solo is played according to the calculated frequency point slope and the piano frequency point slope threshold (if the calculated frequency point slope is greater than the piano frequency point slope threshold, the piano solo is determined); when both described methods judge as the piano, it is determined that the final judgment result is yes; if the piano solo is adopted, calculating the gain of the EQ, if not, setting the gain multiple to be 1, converting the gain into decibels and setting the decibels to the parameter EQ, and in the state without the piano, setting the gain to be 0 dB; the embodiment of the invention can also set the gain of the EQ as a fixed value according to the setting selection of a user.
Step 607: dynamically adjusting parameters of the EQ, such as: parameters that the EQ can set include filter type, center frequency, gain, and bandwidth; when the audio data is determined to be the piano audio, the piano audio is suppressed, the filter type can be set as a peak filter (peak filter), the resonance frequency F0 of the loudspeaker is used as the suppressed center frequency, the gain is negative or 0dB, and the bandwidth is fixed or adjustable, so that the piano audio is suppressed by dynamically adjusting the parameters of EQ to remove noise;
step 608: performing digital-to-analog conversion (DAC) on the compressed audio data, converting the digital signal into an analog signal, and transmitting the analog signal to a power amplifier;
In the above scheme of the embodiment of the invention, by judging the narrow-band characteristic information, the misjudgment generated when other narrow-band signals are played and the misjudgment generated aiming at the vocal singing are reduced, so that the misjudgment rate aiming at the piano audio is reduced; and when parameters of EQ are dynamically adjusted, only the resonance frequency F0 of the loudspeaker is compressed, other frequency range signals are kept unchanged, output loudness is guaranteed, parameters of EQ, such as dynamic gain, dynamic resonance frequency F0 and dynamic quality factor Q, are calculated through various methods, and piano noise can be accurately restrained.
As shown in fig. 8, an embodiment of the present invention further provides an electronic device 800, including:
an obtaining module 810, configured to obtain audio data;
a first determining module 820, configured to determine whether the audio data is piano audio according to the narrowband feature information of the audio data and the loudspeaker diaphragm speed corresponding to the audio data;
and the processing module 830 is configured to, when the audio data is piano audio, perform compression processing on the audio data by dynamically adjusting parameters of the EQ.
Optionally, the first determining module 820 includes:
the first judgment submodule is used for judging whether the narrowband characteristic information of the audio data meets a first preset condition or not;
the second judgment submodule is used for judging whether the speed of the loudspeaker diaphragm corresponding to the audio data meets a second preset condition or not;
and the determining submodule is used for determining that the audio data is the piano audio if the narrow-band characteristic information meets the first preset condition and the speed of the loudspeaker diaphragm meets the second preset condition.
Optionally, the first determining sub-module includes:
the processing unit is used for processing the audio data to obtain a frequency spectrum envelope;
a first determining unit, configured to determine the narrowband feature information according to a region in the spectral envelope where a frequency is smaller than a target frequency;
the first calculation unit is used for calculating the slope of the narrow-band characteristic frequency point according to the narrow-band characteristic information;
and the second determining unit is used for determining that the narrowband characteristic information meets the first preset condition if the slope of the narrowband characteristic frequency point exceeds a preset slope threshold.
Optionally, the first computing unit includes:
a selecting subunit, configured to select a first frequency point and a second frequency point in the narrowband feature information;
a first processing subunit, configured to perform smoothing processing on the first frequency point and the second frequency point through a smoothing factor;
the first determining subunit is used for determining a straight line according to the first frequency point and the second frequency point after the smoothing processing;
and the second determining subunit is used for determining the slope of the straight line as the slope of the narrow-band characteristic frequency point.
Optionally, the second determining sub-module includes:
the second calculation unit is used for calculating the loudspeaker diaphragm speed corresponding to the audio data according to the loudspeaker modeling parameters;
and the third determining unit is used for determining that the loudspeaker diaphragm speed meets the second preset condition if the loudspeaker diaphragm speed exceeds a preset speed threshold value.
Optionally, the second calculating unit includes:
the third determining subunit is used for determining the filter coefficient of the loudspeaker impedance model according to the loudspeaker modeling parameter;
the fourth determining subunit is configured to determine a loudspeaker admittance model filter coefficient according to the loudspeaker impedance model filter coefficient;
the second processing subunit is used for taking the input voltage corresponding to the audio data as an input signal of a loudspeaker admittance model filter to obtain the output current predicted by the loudspeaker admittance model filter;
and the calculating subunit is used for calculating the speed of the loudspeaker diaphragm according to the input voltage, the output current, and the force factor and the direct current resistance in the loudspeaker modeling parameters.
Optionally, the processing module 830 includes:
and the processing submodule is used for compressing the audio data by dynamically adjusting the center frequency, the suppression bandwidth and the gain of the EQ.
Optionally, the electronic device 800 further includes:
the first calculation module is used for calculating the resonant frequency and the quality factor of the loudspeaker according to the audio data;
a second determining module, configured to determine a center frequency and a rejection bandwidth of the EQ according to a resonant frequency and a quality factor of the speaker; and the second calculation module is used for calculating the gain of the EQ according to the diaphragm speed.
Optionally, the first computing module includes:
a first obtaining sub-module for obtaining a frame signal of the audio data as an input frame;
and the second determining submodule is used for determining that the first resonant frequency corresponding to the first loudspeaker amplitude model is the resonant frequency of the loudspeaker and determining that the first quality factor corresponding to the first loudspeaker amplitude model is the quality factor of the loudspeaker if the target maximum amplitude is smaller than or equal to the first displacement.
Optionally, the first computing module includes:
a second obtaining sub-module, configured to obtain a frame signal of the audio data as an input frame;
a second calculation submodule, configured to calculate a target maximum amplitude corresponding to the input frame;
the third obtaining submodule is used for obtaining a first displacement, a first resonant frequency and a first quality factor corresponding to the first loudspeaker amplitude model and obtaining a second displacement, a second resonant frequency and a second quality factor corresponding to the second loudspeaker amplitude model if the target maximum amplitude is larger than the first displacement corresponding to the first loudspeaker amplitude model;
and the third calculation submodule is used for calculating the resonant frequency of the loudspeaker according to the first displacement, the first resonant frequency, the second displacement, the second resonant frequency and the target maximum amplitude, and calculating the quality factor of the loudspeaker according to the first displacement, the first quality factor, the second displacement, the second quality factor and the target maximum amplitude.
The electronic device provided in the embodiment of the present invention can implement each process implemented by the electronic device in the method embodiments of fig. 1 to fig. 7, and is not described herein again to avoid repetition.
In the electronic device 800 in the embodiment of the present invention, when it is determined that the audio data is a piano audio according to the narrowband characteristic information of the audio data and the loudspeaker diaphragm speed corresponding to the audio data, the audio data is compressed by dynamically adjusting the parameters of the EQ, so as to obtain target audio data. Thus, the possibility of misjudgment when judging whether the audio is the piano audio can be reduced, and the accuracy of piano audio judgment is improved.
Fig. 9 is a schematic diagram of a hardware structure of an electronic device implementing various embodiments of the present invention.
The electronic device 900 includes, but is not limited to: a radio frequency unit 901, a network module 902, an audio output unit 903, an input unit 904, a sensor 905, a display unit 906, a user input unit 907, an interface unit 908, a memory 909, a processor 910, and a power supply 911. Those skilled in the art will appreciate that the electronic device configuration shown in fig. 9 does not constitute a limitation of the electronic device, and that the electronic device may include more or fewer components than shown, or some components may be combined, or a different arrangement of components. In the embodiment of the present invention, the electronic device includes, but is not limited to, a mobile phone, a tablet computer, a notebook computer, a palm computer, a vehicle-mounted terminal, a wearable device, a pedometer, and the like.
The input unit 904 may be configured to obtain audio data;
the processor 910 may be configured to determine whether the audio data is piano audio according to the narrowband feature information of the audio data and the loudspeaker diaphragm speed corresponding to the audio data; and under the condition that the audio data is piano audio, compressing the audio data by dynamically adjusting parameters of EQ.
In the electronic device 900 in the embodiment of the present invention, when it is determined that the audio data is a piano audio according to the narrowband characteristic information of the audio data and the loudspeaker diaphragm speed corresponding to the audio data, the audio data is compressed by dynamically adjusting the parameters of the EQ, so as to obtain target audio data. Thus, the possibility of misjudgment when judging whether the audio is the piano audio can be reduced, and the accuracy of piano audio judgment is improved.
It should be understood that, in the embodiment of the present invention, the radio frequency unit 901 may be used for receiving and sending signals during a message transmission and reception process or a call process, and specifically, after receiving downlink data from a base station, the downlink data is processed by the processor 910; in addition, the uplink data is transmitted to the base station. Generally, the radio frequency unit 901 includes, but is not limited to, an antenna, at least one amplifier, a transceiver, a coupler, a low noise amplifier, a duplexer, and the like. In addition, the radio frequency unit 901 can also communicate with a network and other devices through a wireless communication system.
The electronic device provides wireless broadband internet access to the user via the network module 902, such as assisting the user in sending and receiving e-mails, browsing web pages, and accessing streaming media.
The audio output unit 903 may convert audio data received by the radio frequency unit 901 or the network module 902 or stored in the memory 909 into an audio signal and output as sound. Also, the audio output unit 903 may provide audio output related to a specific function performed by the electronic device 900 (e.g., a call signal reception sound, a message reception sound, etc.). The audio output unit 903 includes a speaker, a buzzer, a receiver, and the like.
The input unit 904 is used to receive audio or video signals. The input Unit 904 may include a Graphics Processing Unit (GPU) 9041 and a microphone 9042, and the Graphics processor 9041 processes image data of a still picture or video obtained by an image capturing device (such as a camera) in a video capture mode or an image capture mode. The processed image frames may be displayed on the display unit 906. The image frames processed by the graphic processor 9041 may be stored in the memory 909 (or other storage medium) or transmitted via the radio frequency unit 901 or the network module 902. The microphone 9042 can receive sounds and can process such sounds into audio data. The processed audio data may be converted into a format output transmittable to a mobile communication base station via the radio frequency unit 901 in case of the phone call mode.
The electronic device 900 also includes at least one sensor 905, such as light sensors, motion sensors, and other sensors. Specifically, the light sensor includes an ambient light sensor and a proximity sensor, wherein the ambient light sensor may adjust the brightness of the display panel 9061 according to the brightness of ambient light, and the proximity sensor may turn off the display panel 9061 and/or the backlight when the electronic device 900 is moved to the ear. As one type of motion sensor, an accelerometer sensor can detect the magnitude of acceleration in each direction (generally three axes), detect the magnitude and direction of gravity when stationary, and can be used to identify the posture of an electronic device (such as horizontal and vertical screen switching, related games, magnetometer posture calibration), and vibration identification related functions (such as pedometer, tapping); the sensors 905 may also include a fingerprint sensor, a pressure sensor, an iris sensor, a molecular sensor, a gyroscope, a barometer, a hygrometer, a thermometer, an infrared sensor, etc., which are not described in detail herein.
The display unit 906 is used to display information input by the user or information provided to the user. The Display unit 906 may include a Display panel 9061, and the Display panel 9061 may be configured in the form of a Liquid Crystal Display (LCD), an Organic Light-Emitting Diode (OLED), or the like.
The user input unit 907 may be used to receive input numeric or character information and generate key signal inputs related to user settings and function control of the electronic device. Specifically, the user input unit 907 includes a touch panel 9071 and other input devices 9072. The touch panel 9071, also referred to as a touch screen, may collect touch operations by a user on or near the touch panel 9071 (e.g., operations by a user on or near the touch panel 9071 using a finger, a stylus, or any other suitable object or accessory). The touch panel 9071 may include two parts, a touch detection device and a touch controller. The touch detection device detects the touch direction of a user, detects a signal brought by touch operation and transmits the signal to the touch controller; the touch controller receives touch information from the touch sensing device, converts the touch information into touch point coordinates, sends the touch point coordinates to the processor 910, receives a command from the processor 910, and executes the command. In addition, the touch panel 9071 may be implemented by using various types such as a resistive type, a capacitive type, an infrared ray, and a surface acoustic wave. The user input unit 907 may include other input devices 9072 in addition to the touch panel 9071. Specifically, the other input devices 9072 may include, but are not limited to, a physical keyboard, function keys (such as a volume control key, a switch key, and the like), a track ball, a mouse, and a joystick, which are not described herein again.
Further, the touch panel 9071 may be overlaid on the display panel 9061, and when the touch panel 9071 detects a touch operation on or near the touch panel 9071, the touch panel is transmitted to the processor 910 to determine the type of the touch event, and then the processor 910 provides a corresponding visual output on the display panel 9061 according to the type of the touch event. Although in fig. 9, the touch panel 9071 and the display panel 9061 are two independent components to implement the input and output functions of the electronic device, in some embodiments, the touch panel 9071 and the display panel 9061 may be integrated to implement the input and output functions of the electronic device, which is not limited herein.
The interface unit 908 is an interface for connecting an external device to the electronic apparatus 900. For example, the external device may include a wired or wireless headset port, an external power supply (or battery charger) port, a wired or wireless data port, a memory card port, a port for connecting a device having an identification module, an audio input/output (I/O) port, a video I/O port, an earphone port, and the like. The interface unit 908 may be used to receive input from external devices (e.g., data information, power, etc.) and transmit the received input to one or more elements within the electronic device 900 or may be used to transmit data between the electronic device 900 and external devices.
The memory 909 may be used to store software programs as well as various data. The memory 909 may mainly include a storage program area and a storage data area, wherein the storage program area may store an operating system, an application program required for at least one function (such as a sound playing function, an image playing function, etc.), and the like; the storage data area may store data (such as audio data, a phonebook, etc.) created according to the use of the cellular phone, and the like. Further, the memory 909 may include high-speed random access memory, and may also include non-volatile memory, such as at least one magnetic disk storage device, flash memory device, or other volatile solid-state storage device.
The processor 910 is a control center of the electronic device, connects various parts of the entire electronic device using various interfaces and lines, and performs various functions of the electronic device and processes data by running or executing software programs and/or modules stored in the memory 909 and calling data stored in the memory 909, thereby performing overall monitoring of the electronic device. Processor 910 may include one or more processing units; preferably, the processor 910 may integrate an application processor, which mainly handles operating systems, user interfaces, application programs, etc., and a modem processor, which mainly handles wireless communications. It is to be appreciated that the modem processor described above may not be integrated into processor 910.
The electronic device 900 may further include a power supply 911 (e.g., a battery) for supplying power to various components, and preferably, the power supply 911 may be logically connected to the processor 910 through a power management system, so as to manage charging, discharging, and power consumption management functions through the power management system.
In addition, the electronic device 900 includes some functional modules that are not shown, and thus are not described in detail herein.
Preferably, an embodiment of the present invention further provides an electronic device, which includes a processor 910, a memory 909, and a computer program stored in the memory 909 and capable of running on the processor 910, and when the computer program is executed by the processor 910, the processes of the above embodiment of the processing method for piano audio are implemented, and the same technical effect can be achieved, and details are not repeated here to avoid repetition.
The embodiment of the present invention further provides a computer-readable storage medium, where a computer program is stored on the computer-readable storage medium, and when the computer program is executed by a processor, the computer program implements each process of the above-mentioned piano audio processing method embodiment, and can achieve the same technical effect, and in order to avoid repetition, details are not repeated here. The computer-readable storage medium may be a Read-Only Memory (ROM), a Random Access Memory (RAM), a magnetic disk or an optical disk.
It should be noted that, in this document, the terms "comprises," "comprising," or any other variation thereof, are intended to cover a non-exclusive inclusion, such that a process, method, article, or apparatus that comprises a list of elements does not include only those elements but may include other elements not expressly listed or inherent to such process, method, article, or apparatus. Without further limitation, an element defined by the phrase "comprising an … …" does not exclude the presence of other like elements in a process, method, article, or apparatus that comprises the element.
Through the above description of the embodiments, those skilled in the art will clearly understand that the method of the above embodiments can be implemented by software plus a necessary general hardware platform, and certainly can also be implemented by hardware, but in many cases, the former is a better implementation manner. Based on such understanding, the technical solutions of the present invention may be embodied in the form of a software product, which is stored in a storage medium (such as ROM/RAM, magnetic disk, optical disk) and includes instructions for enabling a terminal (such as a mobile phone, a computer, a server, an air conditioner, or a network device) to execute the method according to the embodiments of the present invention.
While the present invention has been described with reference to the embodiments shown in the drawings, the present invention is not limited to the embodiments, which are illustrative and not restrictive, and it will be apparent to those skilled in the art that various changes and modifications can be made therein without departing from the spirit and scope of the invention as defined in the appended claims.
Claims (18)
1. A processing method of piano audio is applied to electronic equipment, and is characterized by comprising the following steps:
acquiring audio data;
determining whether the audio data is piano audio according to the narrowband characteristic information of the audio data and the loudspeaker diaphragm speed corresponding to the audio data;
under the condition that the audio data is piano audio, compressing the audio data by dynamically adjusting parameters of an equalizer EQ;
the determining whether the audio data is piano audio according to the narrowband characteristic information of the audio data and the loudspeaker diaphragm speed corresponding to the audio data includes:
judging whether the narrowband characteristic information of the audio data meets a first preset condition or not;
judging whether the speed of the loudspeaker diaphragm corresponding to the audio data meets a second preset condition or not;
if the narrowband characteristic information meets the first preset condition and the speed of the loudspeaker diaphragm meets the second preset condition, determining that the audio data is piano audio;
the judging whether the narrowband feature information of the audio data meets a first preset condition includes:
processing the audio data to obtain a frequency spectrum envelope;
determining the narrow-band characteristic information according to a region with the frequency smaller than the target frequency in the spectrum envelope;
calculating to obtain a narrow-band characteristic frequency point slope according to the narrow-band characteristic information;
and if the slope of the narrow-band characteristic frequency point exceeds a preset slope threshold, determining that the narrow-band characteristic information meets the first preset condition.
2. The processing method of piano audio according to claim 1, wherein the calculating a narrow-band characteristic frequency point slope according to the narrow-band characteristic information includes:
selecting a first frequency point and a second frequency point in the narrow-band characteristic information;
smoothing the first frequency point and the second frequency point by a smoothing factor;
determining a straight line according to the first frequency point and the second frequency point after smoothing treatment;
and determining the slope of the straight line as the slope of the narrow-band characteristic frequency point.
3. The processing method of the piano audio according to claim 1, wherein the judging whether the speed of the loudspeaker diaphragm corresponding to the audio data meets a second preset condition comprises:
according to the loudspeaker modeling parameters, calculating to obtain the loudspeaker diaphragm speed corresponding to the audio data;
and if the speed of the loudspeaker diaphragm exceeds a preset speed threshold value, determining that the speed of the loudspeaker diaphragm meets the second preset condition.
4. The processing method of piano audio of claim 3, wherein the calculating loudspeaker diaphragm velocity corresponding to the audio data according to loudspeaker modeling parameters comprises:
determining a loudspeaker impedance model filter coefficient according to the loudspeaker modeling parameter;
determining a loudspeaker admittance model filter coefficient according to the loudspeaker impedance model filter coefficient;
taking the input voltage corresponding to the audio data as an input signal of a loudspeaker admittance model filter to obtain the output current predicted by the loudspeaker admittance model filter;
and calculating to obtain the speed of the loudspeaker diaphragm according to the input voltage, the output current, and the force factor and the direct current resistance in the loudspeaker modeling parameters.
5. The method of claim 1, wherein the compressing the audio data by dynamically adjusting the parameters of the equalizer EQ comprises:
and compressing the audio data by dynamically adjusting the center frequency, the suppression bandwidth and the gain of the EQ.
6. The processing method of piano audio according to claim 5, wherein after said acquiring of said audio data, further comprising:
calculating the resonance frequency and the quality factor of the loudspeaker according to the audio data;
determining a center frequency and a rejection bandwidth of the EQ according to a resonant frequency and a quality factor of the speaker;
and calculating the gain of the EQ according to the diaphragm speed.
7. The method of claim 6, wherein said calculating the resonant frequency and quality factor of the speaker according to the audio data comprises:
acquiring a frame signal of the audio data as an input frame;
calculating a target maximum amplitude corresponding to the input frame;
if the target maximum amplitude is smaller than or equal to a first displacement corresponding to a first loudspeaker amplitude model, determining a first resonant frequency corresponding to the first loudspeaker amplitude model as a resonant frequency of the loudspeaker, and determining a first quality factor corresponding to the first loudspeaker amplitude model as a quality factor of the loudspeaker.
8. The method of claim 6, wherein said calculating the resonant frequency and quality factor of the speaker according to the audio data comprises:
acquiring a frame signal of the audio data as an input frame;
calculating a target maximum amplitude corresponding to the input frame;
if the target maximum amplitude is larger than a first displacement corresponding to a first loudspeaker amplitude model, acquiring a first displacement, a first resonance frequency and a first quality factor corresponding to the first loudspeaker amplitude model, and acquiring a second displacement, a second resonance frequency and a second quality factor corresponding to a second loudspeaker amplitude model;
calculating to obtain the resonant frequency of the loudspeaker according to the first displacement, the first resonant frequency, the second displacement, the second resonant frequency and the target maximum amplitude, and calculating to obtain the quality factor of the loudspeaker according to the first displacement, the first quality factor, the second displacement, the second quality factor and the target maximum amplitude.
9. An electronic device, comprising:
the acquisition module is used for acquiring audio data;
the first determining module is used for determining whether the audio data is piano audio according to the narrow-band characteristic information of the audio data and the loudspeaker diaphragm speed corresponding to the audio data;
the processing module is used for compressing the audio data by dynamically adjusting the parameters of the equalizer EQ under the condition that the audio data is piano audio;
the first determining module includes:
the first judgment submodule is used for judging whether the narrowband characteristic information of the audio data meets a first preset condition or not;
the second judgment submodule is used for judging whether the speed of the loudspeaker diaphragm corresponding to the audio data meets a second preset condition or not;
the first determining submodule is used for determining the audio data as piano audio if the narrow-band characteristic information meets the first preset condition and the speed of the loudspeaker diaphragm meets the second preset condition;
the first judgment sub-module includes:
the processing unit is used for processing the audio data to obtain a frequency spectrum envelope;
a first determining unit, configured to determine the narrowband feature information according to a region in the spectral envelope where a frequency is smaller than a target frequency;
the first calculation unit is used for calculating the slope of the narrow-band characteristic frequency point according to the narrow-band characteristic information;
and the second determining unit is used for determining that the narrowband characteristic information meets the first preset condition if the slope of the narrowband characteristic frequency point exceeds a preset slope threshold.
10. The electronic device according to claim 9, wherein the first calculation unit includes:
a selecting subunit, configured to select a first frequency point and a second frequency point in the narrowband feature information;
a first processing subunit, configured to perform smoothing processing on the first frequency point and the second frequency point through a smoothing factor;
the first determining subunit is used for determining a straight line according to the first frequency point and the second frequency point after the smoothing processing;
and the second determining subunit is used for determining the slope of the straight line as the slope of the narrow-band characteristic frequency point.
11. The electronic device of claim 9, wherein the second determination submodule comprises:
the second calculation unit is used for calculating the loudspeaker diaphragm speed corresponding to the audio data according to the loudspeaker modeling parameters;
and the third determining unit is used for determining that the loudspeaker diaphragm speed meets the second preset condition if the loudspeaker diaphragm speed exceeds a preset speed threshold value.
12. The electronic device according to claim 11, wherein the second calculation unit includes:
the third determining subunit is used for determining the filter coefficient of the loudspeaker impedance model according to the loudspeaker modeling parameter;
the fourth determining subunit is configured to determine a loudspeaker admittance model filter coefficient according to the loudspeaker impedance model filter coefficient;
the second processing subunit is used for taking the input voltage corresponding to the audio data as an input signal of a loudspeaker admittance model filter to obtain the output current predicted by the loudspeaker admittance model filter;
and the calculating subunit is used for calculating the speed of the loudspeaker diaphragm according to the input voltage, the output current, and the force factor and the direct current resistance in the loudspeaker modeling parameters.
13. The electronic device of claim 9, wherein the processing module comprises:
and the processing submodule is used for compressing the audio data by dynamically adjusting the center frequency, the suppression bandwidth and the gain of the EQ.
14. The electronic device of claim 13, further comprising:
the first calculation module is used for calculating the resonant frequency and the quality factor of the loudspeaker according to the audio data;
a second determining module, configured to determine a center frequency and a rejection bandwidth of the EQ according to a resonant frequency and a quality factor of the speaker;
and the second calculation module is used for calculating the gain of the EQ according to the diaphragm speed.
15. The electronic device of claim 14, wherein the first computing module comprises:
a first obtaining sub-module for obtaining a frame signal of the audio data as an input frame;
a first calculation submodule, configured to calculate a target maximum amplitude corresponding to the input frame;
the second determining submodule is used for determining that the first resonant frequency corresponding to the first loudspeaker amplitude model is the resonant frequency of the loudspeaker and determining that the first quality factor corresponding to the first loudspeaker amplitude model is the quality factor of the loudspeaker if the target maximum amplitude is smaller than or equal to the first displacement corresponding to the first loudspeaker amplitude model.
16. The electronic device of claim 15, wherein the first computing module comprises:
a second obtaining sub-module, configured to obtain a frame signal of the audio data as an input frame;
a second calculation submodule, configured to calculate a target maximum amplitude corresponding to the input frame;
a third obtaining submodule, configured to obtain a first displacement, a first resonant frequency, and a first quality factor corresponding to a first speaker amplitude model, and obtain a second displacement, a second resonant frequency, and a second quality factor corresponding to a second speaker amplitude model, if the target maximum amplitude is greater than a first displacement corresponding to the first speaker amplitude model;
and the third calculation submodule is used for calculating the resonant frequency of the loudspeaker according to the first displacement, the first resonant frequency, the second displacement, the second resonant frequency and the target maximum amplitude, and calculating the quality factor of the loudspeaker according to the first displacement, the first quality factor, the second displacement, the second quality factor and the target maximum amplitude.
17. An electronic device comprising a processor, a memory and a computer program stored on the memory and executable on the processor, the computer program, when executed by the processor, implementing the steps of the piano audio processing method of any of claims 1 to 8.
18. A computer-readable storage medium, characterized in that the computer-readable storage medium has stored thereon a computer program which, when executed by a processor, implements the steps of the processing method of piano audio according to any one of claims 1 to 8.
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CN112163117A (en) * | 2020-09-18 | 2021-01-01 | 维沃移动通信有限公司 | Noise detection method and device and electronic equipment |
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