CN111031446B - Sound production system and method for automatically compensating transfer function of audio channel - Google Patents
Sound production system and method for automatically compensating transfer function of audio channel Download PDFInfo
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Abstract
The invention discloses a sound production system and a method for automatically compensating an audio channel transfer function. The invention makes up the vacancy in the market of audio systems, and can reduce the audio channel distortion caused by factors such as sound field environment, sound production unit characteristics, simulation characteristics and the like, thereby improving the sound quality. The invention has simple use and operation, can generate the compensation filter and carry out digital filtering through simple calibration so as to reduce the sound signal distortion caused by factors such as sound field environment, analog circuit characteristics, sounder characteristics and the like, and saves complicated steps of acoustic debugging and balancing while bringing high-fidelity acoustic experience to users, thereby saving a great deal of time and energy.
Description
Technical Field
The invention relates to the field of audio systems, in particular to a sound production system and a method for automatically compensating an audio channel transfer function.
Background
With the improvement of living standard of people, consumers have higher and higher requirements on the tone quality, sound image and fidelity of an audio system, but a general audio system has multilayer distortion from the input of an audio signal to the emission of an acoustic signal and the hearing of human ears, and the distortion is caused by factors such as an analog circuit, the characteristics of a sound production unit, the sound field environment and the like. How to rapidly, conveniently and effectively reduce the audio channel distortion is a problem to be solved urgently in the industry at present.
Disclosure of Invention
It is an object of the present invention to solve the above mentioned problems of the background art by a sound generation system and method that automatically compensates for the transfer function of the audio path.
In order to achieve the purpose, the invention adopts the following technical scheme:
a sound production system for automatically compensating transfer function of an audio channel comprises M monitoring microphones with preamplifiers, M passive sound production units, M input and output power amplification circuits and M input and output compensation controllers; the M monitoring microphones with the preamplifiers are connected with M paths of input and output compensation controllers and used for acquiring acoustic signals in a calibration stage and outputting the acoustic signals to the compensation controllers; the input of the M paths of input and output power amplification circuits is connected with the analog output of the M paths of input and output compensation controllers, and the output of the M paths of input and output power amplification circuits is connected with the M passive sounding units and used for amplifying the analog output signal power of the controllers and driving the passive sounding units; the input of the M passive sounding units is connected with the output of the M paths of input and output power amplification circuits and is used for sounding; the M paths of input and output compensation controllers are used for sending excitation signals, collecting excitation and feedback signals and calculating a transfer function and a compensation filter according to preset parameters in a calibration stage; in the compensation stage, digital signals are input or analog signals input through the audio chip are digitally filtered by using the compensation filter calculated in the calibration stage, and the filtered signals are output to the audio chip and then are sounded through the passive sounding unit through the power amplification circuit.
Particularly, the compensation controller comprises a Digital Signal Processing (DSP) chip, a Bluetooth chip, an M-channel input and output audio chip and a memory chip; the Bluetooth chip is used for brushing parameters set by a user on the intelligent terminal into the DSP chip through Bluetooth wireless connection, and when the user plays audio on the intelligent terminal, the audio can be wirelessly transmitted to the DSP chip through the Bluetooth connection; the memory chip is used for storing a transfer function modeling module, a compensation filter calculation module, a filtering algorithm module, user set parameter data, a built audio channel transfer function coefficient and a compensation filter transfer function coefficient which are called by the DSP chip; the transfer function modeling module is used for carrying out transfer function modeling according to the number of the sound production units and the microphones; the compensation filter calculation module is used for calculating a compensation filter according to parameters set by a user and the established transfer function model after the transfer function modeling is finished, and determining an optimal compensation filter coefficient; and the filtering algorithm module is used for carrying out real-time digital filtering on digital or analog audio input based on the calculated compensation filter and outputting the digital or analog audio input to the audio chip for compensation filtering work.
In particular, the transfer function modeling module is specifically configured to: the method comprises the following steps of carrying out transfer function modeling according to the number of the passive sounding units and the number of the monitoring microphones, evaluating the established transfer function according to an evaluation algorithm, and judging whether the transfer function meets the compensation requirement, wherein the method specifically comprises the following steps:
generating white noise signal X in real time by DSP chipwnAnd real-time outputs to all the channels of the audio chip, and the DSP simultaneously collects X in real timewnAnd storing the data into a DSP chip for storage. Simultaneously with the step, the DSP collects M paths of microphone input signals Y from the audio chip in real timewn,m＝1,2,...,MAnd storing the data into a DSP chip for storage. According to user setting or default setting, when X iswnAnd Ywn,m＝1,2,...,MLength equal to NSStopping real-time output and real-time acquisition;
secondly, outputting the DSP chip to the signal X of the audio chipwnAnd microphone acquisition signal Ywn，m＝1，2，...，MThe coherence analysis was performed, and the formula is as follows:
thirdly, get CXY(f) Represents the coherence value C of the corresponding frequencynAnd calculating the root mean square error of the error, wherein the formula is as follows:
if root mean square error Crms≥0.9, the established transfer function meets the requirement, otherwise, the transfer function modeling module repeats the operation of the first step;
if the root mean square condition in the third step is satisfied, the mth excitation signal X of the signal is calculated according to the following formulawnAnd a feedback signal Ywn，m＝1，2，...，MAudio path transfer function Sm＝1，2，…，M. The transfer function has a length of IsDetermined by user settings or default settings;
wherein
Is a signal XwnThe autocorrelation matrix of bmxIs the signal Ywn，m＝1，2，...，MAnd XwnThe cross-correlation vector of (a).
In particular, the compensation filter calculation module is specifically configured to, after the transfer function modeling is completed, calculate a transfer function S according to parameters set by a user and the established transfer function Sm＝1,2,…,MCalculating a compensation filter, and determining an optimal compensation filter coefficient, wherein the specific process is as follows:
generating a white noise signal WN with N data numbers through a DSP, wherein N is default setting of a system or user setting;
secondly, generating a band-pass filter W with a specific frequency band and a length I according to default settings of a system or user settingsBPIf full band compensation is set, WBPIs an all-pass filter;
thirdly, digital filtering is carried out on the white noise WN generated in the first step, and the filter is a band-pass filter W generated in the second stepBPGenerating a signal X after filteringBPThe filter formula is as follows:
four, calculate a total of MMth compensating filter W in audio pathmThe sub-steps are as follows:
a. for the band-pass signal X obtained in step threeBPFurther performing digital filtering, the digital filter is a transfer function S corresponding to the mth audio channelmFiltering to obtain signal YBP,mThe formula is as follows:
b. according to the user setting or default setting, the band-pass signal X obtained in the step three is subjected toBPPerforming D delay operations and obtaining a signal XBPDThe concrete formula is as follows:
c. calculating a compensation filter W of length C according to user default settingsmThe formula is as follows:
wherein
Is the signal YBP，mThe autocorrelation matrix of bdyIs a signal XBPD(n) and YBP，mThe cross-correlation vector of (a).
IN particular, the filtering algorithm module is specifically configured to perform real-time digital filtering operation on digital audio input or audio input IN converted from analog to digital by an audio chip after the calculation of the compensation filter is completed, and output OUT after filteringmInputting to the mth audio chip; the digital filter is a compensation filter W calculated by a compensation filter calculation modulemThe concrete formula is as follows:
particularly, the Bluetooth chip in the compensation controller is used for realizing wireless connection between the compensation controller and the intelligent terminal; user setting compensation controller through APP installed on intelligent terminal includes but is not limited to calibration phase data length NSLength of transfer function ISLength N of white noise data in compensation stage, band pass filter WBPFrequency band, delay parameter D, compensation filter length C; in the compensation stage, the user can make the sound production unit play high-fidelity bluetooth audio frequency through the bluetooth audio frequency of intelligent terminal and compensation controller wireless connection.
Based on the sound production system for automatically compensating the transfer function of the audio channel, the invention also provides a sound production method for automatically compensating the transfer function of the audio channel, which comprises the following steps:
s101, arranging and connecting all parts of a sound production system for automatically compensating transfer functions of an audio channel, wherein the parts comprise M monitoring microphones with preamplifiers, M passive sound production units, M input and output power amplification circuits and M input and output compensation controllers; the M monitoring microphones with the preamplifiers are connected with M paths of input and output compensation controllers, M paths of input and output power amplification circuits are connected with the compensation controllers, and M passive sound production units are connected with the M paths of input and output power amplification circuits; wherein, M monitoring microphones with preamplifiers correspond to M passive sound production units one by one;
s102, electrifying the compensation controller, if a user needs to change the parameters of the compensation controller, starting an APP (application program) on the intelligent terminal, connecting the compensation controller through Bluetooth and modifying the parameters;
s103, the compensation controller enters a calibration stage after parameter modification is completed, a transfer function modeling module is called, after a signal coherence requirement is passed, transfer functions of M paths of audio channels are calculated and stored, and otherwise, the transfer functions are rebuilt;
s104, after the transfer function of the audio channel is established, the compensation controller calls a compensation filter calculation module to calculate and store the compensation filters of the M audio channels;
and S105, connecting audio equipment (including an intelligent terminal) to the compensation controller through an analog circuit or connecting the compensation controller through Bluetooth audio transmission of the intelligent terminal by a user according to the requirement of the user, and calling a real-time filtering algorithm module by the compensation controller to perform real-time audio channel compensation.
And S106, after the user finishes using the sound production system of the automatic compensation audio channel transfer function, closing the sound production system of the automatic compensation audio channel transfer function and powering off.
The sound production system and the method for automatically compensating the transfer function of the audio channel make up for the vacancy in the market of the audio system, and can reduce the distortion of the audio channel caused by factors such as sound field environment, sound production unit characteristics, simulation characteristics and the like so as to improve the sound quality. The invention has simple use and operation, can generate the compensation filter and carry out digital filtering through simple calibration so as to reduce the sound signal distortion caused by factors such as sound field environment, analog circuit characteristics, sounder characteristics and the like, and saves complicated steps of acoustic debugging and balancing while bringing high-fidelity acoustic experience to users, thereby saving a great deal of time and energy.
Drawings
FIG. 1 is a schematic diagram of a sound system for automatically compensating an audio channel transfer function according to an embodiment of the present invention;
fig. 2 is a flowchart of a method for automatically compensating an audio channel transfer function according to an embodiment of the present invention.
Detailed Description
To facilitate an understanding of the invention, the invention will now be described more fully with reference to the accompanying drawings. Preferred embodiments of the present invention are shown in the drawings. This invention may, however, be embodied in many different forms and should not be construed as limited to the embodiments set forth herein. Rather, these embodiments are provided so that this disclosure will be thorough and complete. It will be understood that when an element is referred to as being "connected" to another element, it can be directly connected to the other element or intervening elements may also be present. Unless defined otherwise, all technical and scientific terms used herein have the same meaning as commonly understood by one of ordinary skill in the art to which this invention belongs. The terminology used in the description of the invention herein is for the purpose of describing particular embodiments only and is not intended to be limiting of the invention. As used herein, the term "and/or" includes any and all combinations of one or more of the associated listed items.
Referring to fig. 1, fig. 1 is a schematic diagram of a sounding system for automatically compensating an audio channel transfer function according to an embodiment of the present invention, where 101 is M microphones with front-end monitoring, i.e., hereinafter monitoring microphones, 102 is M passive sounding units, 103 is M power amplification circuits, i.e., hereinafter power amplification circuits, 104 is a complementary controller, 1041 is a digital signal processor, i.e., hereinafter DSP chip, 1042 is a bluetooth chip, 1043 is an audio chip, and 1044 is a memory chip.
The sound production system for automatically compensating the transfer function of the audio channel in the embodiment specifically comprises M monitoring microphones 101 with preamplifiers, M passive sound production units 102, M input and output power amplification circuits 103 and M input and output compensation controllers 104; the M monitoring microphones 101 with preamplifiers are connected with M input and output compensation controllers 104, and are used for acquiring acoustic signals in a calibration stage and outputting the acoustic signals to the compensation controllers 104; the input of the M input and output power amplification circuits 103 is connected with the analog output of the M input and output compensation controller 104, and the output of the M input and output power amplification circuits 103 is connected with the M passive sound generating units 102, and is used for amplifying the analog output signal power of the controller and driving the passive sound generating units 102; the inputs of the M passive sound generating units 102 are connected to the outputs of the M input and output power amplifying circuits 103 for sound generation; the M input and output compensation controllers 104 are used for sending excitation signals, collecting excitation and feedback signals and calculating a transfer function and a compensation filter according to preset parameters in a calibration stage; in the compensation stage, digital signals are input or analog signals input through the audio chip 1043 are digitally filtered by using the compensation filter calculated in the calibration stage, and the filtered signals are output to the audio chip 1043 and then are sounded through the passive sounding unit 102 through the power amplification circuit 103.
Specifically, in this embodiment, the compensation controller 104 includes a DSP chip 1041, a bluetooth chip 1042, an M-channel input/output audio chip 1043, and a storage chip 1044; the bluetooth chip 1042 brushes the parameters set by the user on the intelligent terminal 106 into the DSP chip 1041 through bluetooth wireless connection, and when the user plays audio at the intelligent terminal 106, the audio can be wirelessly transmitted to the DSP chip 1041 through bluetooth connection; the storage chip 1044 is configured to store a transfer function modeling module, a compensation filter calculation module, a filtering algorithm module, user-set parameter data, a transfer function coefficient of a built audio channel, and a transfer function coefficient of a compensation filter, which are called by the DSP chip 1041; the transfer function modeling module is used for carrying out transfer function modeling according to the number of the sound production units 102 and the number of the microphones; the compensation filter calculation module is used for calculating a compensation filter according to parameters set by a user and the established transfer function model after the transfer function modeling is finished, and determining an optimal compensation filter coefficient; the filtering algorithm module is configured to perform real-time digital filtering on digital or analog audio input based on the calculated compensation filter and output the digital or analog audio input to the audio chip 1043 for compensation filtering.
Specifically, in this embodiment, the transfer function modeling module is specifically configured to: the method includes the steps of performing transfer function modeling according to the number of the passive sound generating units 102 and the number of the monitoring microphones 101, evaluating the built transfer function according to an evaluation algorithm, and judging whether the transfer function meets the compensation requirement, and specifically includes the following steps:
second, the signal X for outputting the DSP chip 1041 to the audio chip 1043wnAnd microphone acquisition signal Ywn，m＝1，2，...，MThe coherence analysis was performed, and the formula is as follows:
thirdly, get CXY(f) Represents the coherence value C of the corresponding frequencynAnd calculating the root mean square error of the error, wherein the formula is as follows:
if root mean square error CrmsIf the value is more than or equal to 0.9, the established transfer function meets the requirement, otherwise, the transfer function modeling module repeats the operation of the first step;
if the root mean square condition in the third step is satisfied, the mth excitation signal X of the signal is calculated according to the following formulawnAnd a feedback signal Ywn，m＝1，2，...，MAudio path transfer function Sm＝1，2，...，M. The transfer function has a length of IsDetermined by user settings or default settings;
wherein
Is a signal XwnThe autocorrelation matrix of bmxIs the signal Ywn，m＝1，2，...，MAnd XwnThe cross-correlation vector of (a).
Specifically, in this embodiment, the compensation filter calculation module is specifically configured to, after the transfer function modeling is completed, calculate a transfer function S according to parameters set by a user and the established transfer function Sm＝1,2,...,MCalculating a compensation filter, and determining an optimal compensation filter coefficient, wherein the specific process is as follows:
generating a white noise signal WN with N data numbers through a DSP, wherein N is default setting of a system or user setting;
secondly, generating a band-pass filter W with a specific frequency band and a length I according to default settings of a system or user settingsBPIf full band compensation is set, WBPIs an all-pass filter;
thirdly, digital filtering is carried out on the white noise WN generated in the first step, and the filter is a band-pass filter W generated in the second stepBPGenerating a signal X after filteringBPThe filter formula is as follows:
fourthly, calculating the mth compensation filter W in the M audio channelsmThe sub-steps are as follows:
a. for the band-pass signal X obtained in step threeBPFurther performing digital filtering, the digital filter is a transfer function S corresponding to the mth audio channelmFiltering to obtain signal YBP,mThe formula is as follows:
b. according to the user setting or default setting, the band-pass signal X obtained in the step three is subjected toBPPerforming D delay operations and obtaining a signal XBPDThe concrete formula is as follows:
c. calculating a compensation filter W of length C according to user default settingsmThe formula is as follows:
wherein
Is the signal YBP，mThe autocorrelation matrix of bdyIs a signal XBPD(n) and YBP，mThe cross-correlation vector of (a).
Specifically, IN this embodiment, the filtering algorithm module is specifically configured to perform real-time digital filtering operation on digital audio input or audio input IN after analog-to-digital conversion by the audio chip 1043 after calculation of the compensation filter is completed, and output OUT after filteringmInputting to the mth audio chip 1043; the digital filter is a compensation filter W calculated by a compensation filter calculation modulemThe concrete formula is as follows:
specifically, in this embodiment, the bluetooth chip 1042 in the compensation controller 104 is used for implementing wireless connection between the compensation controller 104 and the intelligent terminal 106; user setting of compensation controller 104 through APP application installed on smart terminal 106 includes, but is not limited to, calibration phase data length NSLength of transfer function ISLength N of white noise data in compensation stage, band pass filter WBPFrequency band, delay parameter D, compensation filter length C; in the compensation phase, the user can wirelessly connect with the compensation controller 104 through the bluetooth audio of the intelligent terminal 106, so that the sound generating unit 102 plays the high-fidelity bluetooth audio.
Based on the sound production system for automatically compensating the transfer function of the audio path in the foregoing embodiment, as shown in fig. 2, the present embodiment provides a sound production method for automatically compensating the transfer function of the audio path, including the following steps:
s1, arranging and connecting the parts of the sound production system for automatically compensating the transfer function of the audio channel in the embodiment, wherein the parts comprise M monitoring microphones 101 with preamplifiers, M passive sound production units 102, M input and output power amplification circuits 103 and M input and output compensation controllers 104; the M monitoring microphones 101 with preamplifiers are connected with M input and output compensation controllers 104, M input and output power amplification circuits 103 are connected with the compensation controllers 104, and M passive sound production units 102 are connected with the M input and output power amplification circuits 103; wherein, M monitoring microphones 101 with preamplifiers correspond to M passive sound units 102 one by one, and they should be placed at positions with close distances;
s2, powering on the compensation controller 104, if the user wants to change the controller parameters, starting an APP on the intelligent terminal 106, connecting the compensation controller 104 through Bluetooth and modifying the parameters;
s3, the compensation controller 104 enters a calibration stage after the parameters are modified, a transfer function modeling module is called, the transfer functions of the M paths of audio channels are calculated and stored after the signal coherence requirement is passed, and otherwise, the transfer functions are rebuilt;
s4, after the transfer function of the audio channel is established, the compensation controller 104 calls a compensation filter calculation module to calculate and store the compensation filters of the M audio channels;
s5, the user connects the audio device 105 (including the intelligent terminal 106) to the compensation controller 104 through the analog circuit or connects the compensation controller 104 through the bluetooth audio transmission of the intelligent terminal 106 according to the requirement, and the compensation controller 104 calls the real-time filtering algorithm module to perform real-time audio path compensation.
And S6, after the user finishes using the sound production system of the automatic compensation audio channel transfer function, closing the sound production system of the automatic compensation audio channel transfer function and powering off.
The technical scheme of the invention makes up the vacancy in the market of the audio system, and can reduce the audio channel distortion caused by factors such as sound field environment, sound production unit characteristics, simulation characteristics and the like, thereby improving the sound quality. The invention has simple use and operation, can generate the compensation filter and carry out digital filtering through simple calibration so as to reduce the sound signal distortion caused by factors such as sound field environment, analog circuit characteristics, sounder characteristics and the like, and saves complicated steps of acoustic debugging and balancing while bringing high-fidelity acoustic experience to users, thereby saving a great deal of time and energy.
It will be understood by those skilled in the art that all or part of the above embodiments may be implemented by the computer program to instruct the relevant hardware, and the program may be stored in a computer readable storage medium, and when executed, may include the procedures of the embodiments of the methods as described above. The storage medium may be a magnetic disk, an optical disk, a read-only memory or a random access memory.
It is to be noted that the foregoing is only illustrative of the preferred embodiments of the present invention and the technical principles employed. It will be understood by those skilled in the art that the present invention is not limited to the particular embodiments described herein, but is capable of various obvious changes, rearrangements and substitutions as will now become apparent to those skilled in the art without departing from the scope of the invention. Therefore, although the present invention has been described in greater detail by the above embodiments, the present invention is not limited to the above embodiments, and may include other equivalent embodiments without departing from the spirit of the present invention, and the scope of the present invention is determined by the scope of the appended claims.
Claims (2)
1. A sound production system for automatically compensating transfer function of an audio channel is characterized by comprising M monitoring microphones with preamplifiers, M passive sound production units, M input and output power amplification circuits and M input and output compensation controllers; the M monitoring microphones with the preamplifiers are connected with M paths of input and output compensation controllers and used for acquiring acoustic signals in a calibration stage and outputting the acoustic signals to the compensation controllers; the input of the M paths of input and output power amplification circuits is connected with the analog output of the M paths of input and output compensation controllers, and the output of the M paths of input and output power amplification circuits is connected with the M passive sounding units and used for amplifying the analog output signal power of the controllers and driving the passive sounding units; the input of the M passive sounding units is connected with the output of the M paths of input and output power amplification circuits and is used for sounding; the M paths of input and output compensation controllers are used for sending excitation signals, collecting excitation and feedback signals and calculating a transfer function and a compensation filter according to preset parameters in a calibration stage; in the compensation stage, digital signals are input or analog signals input through an audio chip are digitally filtered by using a compensation filter calculated in the calibration stage, and the filtered signals are output to the audio chip and then are sounded through a passive sounding unit through a power amplification circuit; the compensation controller comprises a DSP chip, a Bluetooth chip, an M-channel input and output audio chip and a memory chip; the Bluetooth chip is used for brushing parameters set by a user on the intelligent terminal into the DSP chip through Bluetooth wireless connection, and when the user plays audio on the intelligent terminal, the audio can be wirelessly transmitted to the DSP chip through the Bluetooth connection; the memory chip is used for storing a transfer function modeling module, a compensation filter calculation module, a filtering algorithm module, user set parameter data, a built audio channel transfer function coefficient and a compensation filter transfer function coefficient which are called by the DSP chip; the transfer function modeling module is used for carrying out transfer function modeling according to the number of the sound production units and the microphones; the compensation filter calculation module is used for calculating a compensation filter according to parameters set by a user and the established transfer function model after the transfer function modeling is finished, and determining an optimal compensation filter coefficient; the filtering algorithm module is used for carrying out real-time digital filtering on digital or analog audio input based on the calculated compensation filter and outputting the digital or analog audio input to an audio chip for compensation filtering work; the transfer function modeling module is specifically configured to: the method comprises the following steps of carrying out transfer function modeling according to the number of the passive sounding units and the number of the monitoring microphones, evaluating the established transfer function according to an evaluation algorithm, and judging whether the transfer function meets the compensation requirement, wherein the method specifically comprises the following steps:
generating white noise signal X in real time by DSP chipwnAnd real-time outputs to all the channels of the audio chip, and the DSP simultaneously collects X in real timewnAnd storing the data into a DSP chip, and simultaneously, the DSP collects M paths of microphone input from an audio chip in real timeSignal Ywn,m＝1,2,…,MAnd storing the data in a DSP chip for storage, and according to user setting or default setting, when X iswnAnd Ywn,m＝1,2,…,MLength equal to NSStopping real-time output and real-time acquisition;
secondly, outputting the DSP chip to the signal X of the audio chipwnAnd microphone acquisition signal Ywn，m＝1，2，...，MThe coherence analysis was performed, and the formula is as follows:
thirdly, get CXY(f) Represents the coherence value C of the corresponding frequencynAnd calculating the root mean square error of the error, wherein the formula is as follows:
if root mean square error CrmsIf the value is more than or equal to 0.9, the established transfer function meets the requirement, otherwise, the transfer function modeling module repeats the operation of the first step;
if the root mean square condition in the third step is satisfied, the mth excitation signal X of the signal is calculated according to the following formulawnAnd a feedback signal Ywn，m＝1，2，...，MAudio path transfer function Sm＝1，2，...，MThe transfer function has a length of IsDetermined by user settings or default settings;
wherein
Is a signal XwnThe autocorrelation matrix of bmxIs the signal Ywn，m＝1，2，...，MAnd XwnThe cross-correlation vector of (a); the compensation filterThe calculation module is specifically used for establishing the transfer function S according to the parameters set by the user after the transfer function modeling is finishedm＝1,2,…,MCalculating a compensation filter, and determining an optimal compensation filter coefficient, wherein the specific process is as follows:
generating a white noise signal WN with N data numbers through a DSP, wherein N is default setting of a system or user setting;
secondly, generating a band-pass filter W with a specific frequency band and a length I according to default settings of a system or user settingsBPIf full band compensation is set, WBPIs an all-pass filter;
thirdly, digital filtering is carried out on the white noise WN generated in the first step, and the filter is a band-pass filter W generated in the second stepBPGenerating a signal X after filteringBPThe filter formula is as follows:
fourthly, calculating the mth compensation filter W in the M audio channelsmThe sub-steps are as follows:
a. for the band-pass signal X obtained in step threeBPFurther performing digital filtering, the digital filter is a transfer function S corresponding to the mth audio channelmFiltering to obtain signal YBP,mThe formula is as follows:
b. according to the user setting or default setting, the band-pass signal X obtained in the step three is subjected toBPPerforming D delay operations and obtaining a signal XBPDThe concrete formula is as follows:
c. according to user default settingsConstant, calculated length C compensation filter WmThe formula is as follows:
wherein
Is the signal YBP，mThe autocorrelation matrix of bdyIs a signal XBPD(n) and YBP，mThe cross-correlation vector of (a) is,
IN particular, the filtering algorithm module is specifically configured to perform real-time digital filtering operation on digital audio input or audio input IN converted from analog to digital by an audio chip after the calculation of the compensation filter is completed, and output OUT after filteringmInputting to the mth audio chip; the digital filter is a compensation filter W calculated by a compensation filter calculation modulemThe concrete formula is as follows:
2. A method of automatically compensating an audio path transfer function for an audio system based on the automatically compensating audio path transfer function of claim 1, the method comprising the steps of:
s101, arranging and connecting all parts of a sound production system for automatically compensating transfer functions of an audio channel, wherein the parts comprise M monitoring microphones with preamplifiers, M passive sound production units, M input and output power amplification circuits and M input and output compensation controllers; the M monitoring microphones with the preamplifiers are connected with M paths of input and output compensation controllers, M paths of input and output power amplification circuits are connected with the compensation controllers, and M passive sound production units are connected with the M paths of input and output power amplification circuits; wherein, M monitoring microphones with preamplifiers correspond to M passive sound production units one by one;
s102, electrifying the compensation controller, if a user needs to change the parameters of the compensation controller, starting an APP (application program) on the intelligent terminal, connecting the compensation controller through Bluetooth and modifying the parameters;
s103, the compensation controller enters a calibration stage after parameter modification is completed, a transfer function modeling module is called, after a signal coherence requirement is passed, transfer functions of M paths of audio channels are calculated and stored, and otherwise, the transfer functions are rebuilt;
s104, after the transfer function of the audio channel is established, the compensation controller calls a compensation filter calculation module to calculate and store the compensation filters of the M audio channels;
s105, connecting audio equipment to a compensation controller through an analog circuit or connecting the audio equipment to the compensation controller through Bluetooth audio transmission of an intelligent terminal by a user according to the requirement of the user, and calling a real-time filtering algorithm module by the compensation controller to perform real-time audio channel compensation;
and S106, after the user finishes using the sound production system of the automatic compensation audio channel transfer function, closing the sound production system of the automatic compensation audio channel transfer function and powering off.
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