Howling suppression method, device, sound system and sound amplification system

Abstract

The embodiments of the present application provide a howling suppression method, device, audio and sound amplification system, and relate to the field of digital signal processing technology. The method includes: pre-processing the audio signal in the sound amplification system, converting the audio signal to the frequency domain; performing the following processing on each frame signal of each frequency point in the converted audio signal: filtering the converted signal played by the loudspeaker twice based on an adaptive filter; determining the signal to be amplified based on the difference between the signal collected by the converted microphone and the signal played by the loudspeaker after the two filtering, and updating the filter coefficient twice during the processing using the signal played by the loudspeaker in the current frame signal as a reference signal for use in the next frame signal; converting the signal to be amplified into the time domain after the amplification processing to obtain the target audio. By updating the filter coefficient twice, the loop gain from the microphone to the loudspeaker and the howling suppression capability of the sound amplification system can be improved.

Description

Technical Field

[0001] The present application relates to the technical field of digital signal processing, and in particular to a howling suppression method, device, speaker and sound amplification system. Background Art

[0002] In the sound reinforcement system, the signal collected by the microphone is transmitted to the speaker for amplification and broadcasting, and the audio signal played by the speaker is picked up again by the microphone. The transmission and feedback of the audio signal between the speaker and the microphone constitute an acoustic loop. During the transmission process, when the volume is high, the feedback loop of the sound forms positive feedback, that is, the acoustic loop gain is greater than 1. The sound is amplified step by step in the continuous feedback, resulting in a harsh howling sound, which seriously affects the user's auditory experience.

[0003] At present, the methods for suppressing howling in sound reinforcement systems include: frequency shifting and phase shifting, notch suppression, adaptive howling suppression, etc. Among them, the frequency shifting and phase shifting method, in the process of sound processing, changes the frequency or phase of the sound in real time to destroy the phase characteristics required for positive feedback. The notch suppression method, for the frequency point where howling occurs, forcibly lowers the acoustic loop gain of the frequency point through a notch filter, but both change the frequency response of the sound signal or system, causing certain distortion to the sound.

[0004] The adaptive howling suppression method uses an adaptive filter to track the feedback path and offset its effect, which can prevent the generation of howling. However, the howling suppression capability is not sufficient and the loop gain improvement of the system is also insufficient. Summary of the invention

[0005] The embodiments of the present application provide a howling suppression method, device, speaker and sound amplification system, aiming to solve at least one technical problem in the prior art.

[0006] According to a first aspect of an embodiment of the present application, a howling suppression method is provided, the method comprising:

[0007] Preprocessing the audio signal in the sound reinforcement system, converting the audio signal into the frequency domain, wherein the audio signal includes: a signal collected by a microphone and a signal played by a loudspeaker;

[0008] The following processing is performed on each frame signal of each frequency point in the converted audio signal:

[0009] The converted signal played by the loudspeaker is filtered twice based on an adaptive filter;

[0010] Based on the difference between the signal collected by the converted microphone and the signal played by the loudspeaker after two filtering, the signal to be amplified is determined, and during the processing, the filter coefficients are updated twice using the signal played by the loudspeaker in the current frame signal as a reference signal for use in the next frame signal processing;

[0011] The signal to be amplified is amplified and then converted into the time domain to obtain the target audio.

[0012] In a possible implementation, the process of updating the filter coefficients twice using the signal played by the speaker in the current frame signal as a reference signal includes:

[0013] Taking the signal played by the loudspeaker in the current frame signal as a reference signal, updating the filter coefficient based on the difference between the signal collected by the microphone in the current frame signal and the reference signal to obtain a first filter coefficient;

[0014] The signal played by the loudspeaker after being filtered once in the current frame signal is used as a reference signal, and the first filter coefficient is updated based on the difference between the signal collected by the microphone in the current frame signal and the reference signal to obtain the second filter coefficient.

[0015] In yet another possible implementation, before processing the initial frame signal at any frequency point, the method further includes:

[0016] Determining initial filter coefficients and initial error signals corresponding to a transfer function of a loudspeaker-to-microphone path in the sound reinforcement system;

[0017] The following processing is performed on the initial frame signal of any frequency point:

[0018] Based on the initial filter coefficient, the initial error signal and the initial frame signal, obtaining an updated first filter coefficient and a first error signal;

[0019] Based on the first filter coefficient, the first error signal and the initial frame signal, obtaining an updated second filter coefficient and a second error signal;

[0020] comparing power spectra corresponding to the first error signal and the second error signal, and determining the error signal corresponding to the smaller power spectrum as the signal to be amplified corresponding to the initial frame signal;

[0021] The initial error signal is determined based on the signal played by the loudspeaker and the signal collected by the microphone in the initial frame signal, as well as the initial filter coefficient.

[0022] In another possible implementation, the process of determining the first error signal includes:

[0023] Determine a corresponding first filter coefficient based on a preset update step size, the initial filter coefficient, the initial error signal, and a signal played by a speaker in the initial frame signal;

[0024] Based on the first filter coefficient, filtering the signal played by the loudspeaker in the initial frame signal to obtain a corresponding first filtered signal;

[0025] The difference between the signal collected by the microphone in the initial frame signal and the first filtered signal is determined as the first error signal.

[0026] In another possible implementation, the process of determining the second error signal includes:

[0027] Obtaining corresponding second filter coefficients based on a preset update step size, the first filter coefficients, the first error signal, and a signal played by a speaker in the initial frame signal;

[0028] Based on the second filter coefficient, filtering the signal played by the loudspeaker in the initial frame signal to obtain a corresponding second filtered signal;

[0029] The difference between the signal collected by the microphone in the initial frame signal and the second filtered signal is determined as the second error signal.

[0030] In another possible implementation, the following processing is performed on any non-initial frame signal at any frequency point:

[0031] Based on the first filter coefficient and the first error signal updated in the previous frame signal processing process, and the current frame signal, obtain the first filter coefficient and the first error signal corresponding to the current frame signal;

[0032] Based on the first filter coefficient and the first error signal corresponding to the current frame signal, and the current frame signal, obtain the second filter coefficient and the second error signal corresponding to the current frame signal;

[0033] The power spectra corresponding to the first error signal and the second error signal corresponding to the current frame signal are compared, and the error signal corresponding to the smaller power spectrum is determined as the signal to be amplified corresponding to the current frame signal.

[0034] In another possible implementation, if the preprocessing is a fast Fourier transform, converting the signal to be amplified into a time domain after amplification processing to obtain a target audio includes:

[0035] The signal to be amplified is amplified and then inverse Fourier transformed back to the time domain to obtain the target audio.

[0036] According to a second aspect of an embodiment of the present application, a howling suppression device is provided, the device comprising: a signal processing module, an adder, a comparator, an adaptive filter, and a fast Fourier transform module and an inverse transform module, wherein the input end of the adaptive filter is connected to a loudspeaker, the output end of the adaptive filter is connected to the input end of the adder, the output end of the adder is connected to the input end of the comparator, the output end of the comparator is connected to the input end of the signal processing module, and the output end of the signal processing module is connected to the loudspeaker;

[0037] The fast Fourier transform module is used to pre-process the audio signal collected by the microphone and convert the audio signal into the frequency domain, wherein the audio signal includes: the signal collected by the microphone and the signal played by the speaker;

[0038] The adaptive filter is used to perform echo suppression and howling suppression processing twice on the converted audio signal and then output it to the adder;

[0039] The adder is used for respectively subtracting the converted audio signal from the signals output twice by the adaptive filter and outputting the subtractions to the comparator;

[0040] The comparator is used to compare the two received signals and output the smaller signal to the signal processing module;

[0041] The signal processing module performs local amplification processing on the received signal and then transmits it to the inverse Fourier transform module;

[0042] The inverse Fourier transform module is used to convert the received signal into the time domain to obtain the target audio and transmit it to the speaker for playback.

[0043] According to a third aspect of the embodiments of the present application, a processor is provided, including: a signal conversion module and a signal processing update module, wherein:

[0044] The signal conversion module is used to pre-process the audio signal in the sound reinforcement system and convert the audio signal into the frequency domain, wherein the audio signal includes: a signal collected by a microphone and a signal played by a speaker;

[0045] The signal processing update module is used to perform the following processing on each frame signal of each frequency point in the converted audio signal: filtering the converted signal played by the loudspeaker twice based on the adaptive filter; determining the signal to be amplified based on the difference between the signal collected by the microphone after the conversion and the signal played by the loudspeaker after the two filtering, and updating the filter coefficient twice during the processing using the signal played by the loudspeaker in the current frame signal as a reference signal for use in the next frame signal processing;

[0046] The signal conversion module is further used to convert the signal to be amplified into the time domain after amplification processing to obtain the target audio.

[0047] According to a fourth aspect of the embodiments of the present application, there is provided a sound system, comprising: a speaker and the howling suppression device according to the embodiment of the second aspect, wherein:

[0048] The speaker is connected to the input end of the adaptive filter in the howling suppression device, the output end of the adaptive filter is connected to the input end of the adder in the howling suppression device, the output end of the adder is connected to the input end of the comparator, the output end of the comparator is connected to the input end of the signal processing module in the howling suppression device, and the output end of the signal processing module is connected to the speaker.

[0049] According to a fifth aspect of the embodiments of the present application, a sound amplification system is provided, comprising: a microphone, a loudspeaker, and the howling suppression device described in the embodiment of the second aspect above, wherein the howling suppression device is arranged between the microphone and the loudspeaker, and the howling suppression device is used to receive the audio signal collected by the microphone and output the generated target audio to the loudspeaker for playback.

[0050] The beneficial effects of the technical solution provided by the embodiment of the present application are:

[0051] Based on the filter coefficients of the adaptive filter, the audio signal in the frequency domain is processed frame by frame and frequency by frequency to obtain the corresponding output signal. During the processing, the filter coefficients are updated twice with the signal played by the speaker in the current frame signal as the reference signal for use in the next frame signal processing. Since the howling suppression reuses the filter structure used in echo cancellation, and the reference signal when the filter coefficient is updated is the signal played by the speaker and is updated twice, the loop gain from the microphone to the speaker can be improved, thereby improving the howling suppression capability of the sound reinforcement system. BRIEF DESCRIPTION OF THE DRAWINGS

[0052] In order to more clearly illustrate the technical solutions in the embodiments of the present application, the drawings required for use in describing the embodiments of the present application are briefly introduced below.

[0053] Figure 1 It is a schematic diagram of the signal transmission process in the related art;

[0054] Figure 2 A schematic diagram of a signal transmission process corresponding to a howling suppression method provided in an embodiment of the present application;

[0055] Figure 3 A flowchart of a howling suppression method provided in an embodiment of the present application;

[0056] Figure 4 A schematic diagram of the structure of a processor provided in an embodiment of the present application.

[0057] Icon: 10 - loudspeaker; 20 - microphone; 30 - local sound reinforcement system; 40 - adder; 50 - adaptive filter; 60 - comparator. DETAILED DESCRIPTION

[0058] The embodiments of the present application are described below in conjunction with the drawings in the present application. It should be understood that the implementation methods described below in conjunction with the drawings are exemplary descriptions for explaining the technical solutions of the embodiments of the present application and do not constitute a limitation on the technical solutions of the embodiments of the present application.

[0059] In order to make the objectives, technical solutions and advantages of the present application clearer, the implementation methods of the present application will be further described in detail below with reference to the accompanying drawings.

[0060] When a sound reinforcement system uses a microphone to pick up sound, the sound signal collected by the microphone is transmitted to the speaker for amplification and playback, and the sound signal played by the speaker is transmitted through space and collected by the microphone again. Since it is impossible to completely isolate the sound pickup area of ​​the microphone from the playback area of ​​the speaker, the sound played by the speaker can easily be transmitted to the microphone through space and cause feedback howling. Figure 1 The figure shows the signal transmission process in the sound reinforcement system, where x is the near-end speech signal, that is, the real speaking sound, u is the audio signal finally played by the loudspeaker, k is the feedback signal after the loop transfer function H, that is, the audio signal played by the loudspeaker is collected by the microphone again after being transmitted through the space, y is the sound signal collected by the microphone, and G is the local sound reinforcement system. It can be seen that an acoustic feedback loop is formed between the microphone and the loudspeaker. When the loop enters positive feedback, the signal is gradually amplified in the continuous feedback, and finally howling is generated.

[0061] The adaptive feedback suppression method uses an adaptive filter to track the feedback path and offset its effect, which can prevent the generation of howling. However, the howling suppression capability is not sufficient and the loop gain improvement of the system is also insufficient.

[0062] In view of the above-mentioned technical problems existing in the prior art, the embodiments of the present application provide a howling suppression method, device, speaker and sound amplification system.

[0063] The following describes several exemplary embodiments to illustrate the technical solutions of the embodiments of the present application and the technical effects produced by the technical solutions of the present application. It should be noted that the following embodiments can refer to, draw on or combine with each other, and the same terms, similar features and similar implementation steps in different embodiments will not be described repeatedly.

[0064] Figure 2 The schematic diagram of the signal transmission process of the first howling suppression method provided by the embodiment of the present application is shown. n, k represent time and frequency respectively. Among them, X(n, k) represents the near-end speech signal, Y(n, k) represents the audio signal collected by the microphone, U(n, k) represents the audio signal played by the speaker, and E(n, k) represents the error signal after the adaptive filter is updated. The filter coefficients of the adaptive filter corresponding to the transmission function H(n, k) of the feedback path from the speaker to the microphone are expressed as and Indicates that G(n, k) represents the processing performed by the signal processing module in the local sound reinforcement system, including automatic gain control, signal amplification, power amplification, etc.

[0065] Specifically, in this embodiment, the following steps are included:

[0066] Step 1: Based on the debugging audio, the initial filter coefficients corresponding to the transfer function of the loudspeaker-to-microphone path in the sound reinforcement system can be estimated.

[0067] Step 2: Initialize the adaptive filter according to the initial filter coefficients.

[0068] Step 3: The audio signal collected by the microphone is converted to the frequency domain through fast Fourier transform, and processed frequency-by-frequency and frame-by-frame in the frequency domain.

[0069] Step 4: During the processing, the signal is vectorized. Specifically, assuming that the number of frequency points is K and the number of filters is M, the vector corresponding to the nth frame of the frequency point k in the audio signal collected by the microphone is:

[0070]

[0071] Similarly, we can obtain X(n,k) and U(n,k) in vector form.

[0072] The vector form of the filter coefficients is:

[0073]

[0074] Similarly, we can get the vector form

[0075] The input signal of the microphone can be expressed as:

[0076]

[0077] in, express The conjugate transpose of , Y(n,k) and U(n,k) are known quantities, and the others are unknown quantities.

[0078] The error signal can be expressed as:

[0079]

[0080] That is to say, after performing echo suppression and howling suppression on a frame of audio signals collected by a microphone, a corresponding error signal can be obtained. That is, the error signal of a frame of audio signals collected by a microphone is equal to the difference between the frame of audio signals and the echo cancellation amount and the howling cancellation amount.

[0081] In the solution of the present application, the adaptive filter needs to be updated twice, specifically, the first filter coefficient updated in the process of processing the previous frame signal is used as the filter coefficient of the filter in the process of processing the current frame signal.

[0082] Based on the minimum mean square error (MMSE) criterion, the first update formula of the adaptive filter can be obtained as follows:

[0083]

[0084] Among them, δ is a small integer, generally 0.0001, to prevent the inner product of the signal U(n,k) from being too small, causing the stability performance to decrease. α is the update step size, which is generally updated in a variable step size manner. For details, please refer to the relevant technology and will not be repeated here.

[0085] The first error signal of the current frame (nth frame) can be expressed as:

[0086]

[0087] The formula for the second update based on the result of the first update is:

[0088]

[0089] The second error signal of the current frame can be expressed as:

[0090]

[0091] Finally, the output signal of the sound reinforcement system is:

[0092] U'(n,k)=C(n,k)*min(E1(n,k),E 2 (n, k)}

[0093] Step 5: Convert the output signal U′(n, k) obtained in step 4 back to the time domain to obtain the target audio.

[0094] The solution proposed in the embodiment of the present application combines a dual filter structure to ensure a better filter effect, and then adds secondary filtering processing, that is, a secondary filtering is performed on the result after the first filtering. Compared with the traditional adaptive howling suppression method, the method of the present application has a more obvious improvement on the loop gain of the system.

[0095] Figure 3 A flowchart of a howling suppression method provided in an embodiment of the present application is shown in FIG. Figure 3 The methods shown include:

[0096] S101, pre-processing the audio signal in the sound reinforcement system, converting the audio signal into the frequency domain. The audio signal includes: the signal collected by the microphone and the signal played by the speaker.

[0097] S102, performing the following processing on each frame signal of each frequency point in the converted audio signal:

[0098] The converted signal played by the loudspeaker is filtered twice based on an adaptive filter;

[0099] Based on the difference between the signal collected by the converted microphone and the signal played by the speaker after two filtering, the signal to be amplified is determined, and during the processing, the filter coefficients are updated twice using the signal played by the speaker in the current frame signal as a reference signal for use in the next frame signal processing.

[0100] S103, converting the signal to be amplified into the time domain after amplification processing to obtain the target audio.

[0101] In this embodiment, the preprocessing in S101 is Fast Fourier Transform (FFT), and the specific implementation process of S101 is: divide the audio signal into frames, generally 10 to 30 ms as one frame, and generally set an overlap rate of 50%. Select a time domain window function (such as Hanning window), move the window function, add a window to the time domain audio signal, and then perform a fast Fourier transform to convert the time domain signal to the frequency domain.

[0102] For example, the frame length is 256 sampling points, the overlap is 0.5, and the Hanning window (win) with a window length of 256 points is used to preprocess the audio signals y(n) and u(n) in the sound reinforcement system. The obtained frequency domain signals can be expressed as:

[0103] Y(n, k) = FFT(y(n)*win);

[0104] U(n,k)=FFT(u(n)*win).

[0105] Correspondingly, in S103, the signal to be amplified can be amplified and then inverse Fourier transform (IFFT) can be performed back to the time domain to obtain the target audio. Specifically, after the output signal is inverse fast Fourier transform, each frame signal is multiplied by a window function, and then overlapped and added to obtain the target audio.

[0106] For example, the target audio signal can be expressed as:

[0107] Output=IFFT(OUT(n, k))*win, wherein OUT(n, k) is the output signal after the signal to be amplified is amplified.

[0108] By adopting the above-mentioned method of the embodiment of the present application, the audio signal in the frequency domain is processed frame by frame and frequency by frequency point based on the filter coefficient of the adaptive filter to obtain the corresponding output signal, and during the processing, the filter coefficient is updated twice with the signal played by the speaker in the current frame signal as the reference signal for use in the next frame signal processing. Since the howling suppression reuses the filter structure adopted by the echo cancellation, and the reference signal when the filter coefficient is updated is the signal played by the speaker and is updated twice, the loop gain from the microphone to the speaker can be improved, thereby improving the howling suppression capability of the sound reinforcement system.

[0109] In an embodiment of the present application, a possible implementation method is provided. In S102, the filter coefficient is updated twice using the signal played by the speaker in the current frame signal as a reference signal to obtain the second filter coefficient, including:

[0110] Taking the signal played by the loudspeaker in the current frame signal as a reference signal, updating the filter coefficient based on the difference between the signal collected by the microphone in the current frame signal and the reference signal to obtain a first filter coefficient;

[0111] The signal played by the loudspeaker after one filtering in the current frame signal is used as a reference signal, and the first filter coefficient is updated based on the difference between the signal collected by the microphone in the current frame signal and the reference signal to obtain the second filter coefficient.

[0112] Specifically, in the embodiment of the present application, a secondary filtering process is added, that is, a second filtering is performed on the result after the first filtering. Compared with the traditional adaptive howling suppression method, the method of the present application improves the loop gain of the system more significantly.

[0113] In an embodiment of the present application, a possible implementation method is provided. Before processing the initial frame signal of any frequency point in S102, the following steps may also be included:

[0114] S100 (not shown in the drawings), determining initial filter coefficients and an initial error signal corresponding to a transfer function from a loudspeaker to a microphone path in a sound reinforcement system.

[0115] The initial error signal is determined based on the signal played by the loudspeaker and the signal collected by the microphone in the initial frame signal, as well as the initial filter coefficient.

[0116] For example, for the nth frame (initial frame) signal at frequency k, the corresponding initial error signal can be determined according to the following formula:

[0117]

[0118] in, is the estimated initial filter coefficient corresponding to the nth frame signal of frequency point k, Y(n, k) is the nth frame of frequency point k in the signal collected by the microphone, and U(n, k) is the nth frame of frequency point k in the signal played by the speaker.

[0119] In S102, the initial frame signal of any frequency point is processed as follows:

[0120] Based on the initial filter coefficients, the initial error signal and the initial frame signal, corresponding first filter coefficients and a first error signal are obtained.

[0121] Based on the first filter coefficient, the first error signal and the initial frame signal, corresponding second filter coefficients and a second error signal are obtained.

[0122] The power spectra corresponding to the first error signal and the second error signal are compared, and the error signal corresponding to the smaller power spectrum is determined as the signal to be amplified corresponding to the initial frame signal.

[0123] The initial frame is determined based on the number of adaptive filters.

[0124] Specifically, in this embodiment, the initial filter coefficients corresponding to the transfer function from the speaker to the microphone path in the sound reinforcement system can be estimated based on the debugging audio. The estimation method can adopt an offline filter coefficient calculation method. For the sake of brevity of description, the specific calculation process will not be repeated here.

[0125] In an embodiment of the present application, the adaptive filter used is a normalized least mean square (NLMS) adaptive filter, which includes a group of filters. The initial frame can be determined based on the number of filters in the group. For example, if the group of filters includes 10, the initial frame of any frequency point is the 10th frame. When processing the 10th frame signal, the 1st to 10th frame signals are required. While processing the 10th frame signal to obtain the corresponding output signal, the coefficients of the initial filter need to be updated twice to obtain the second filter coefficients for use in processing the 11th frame signal. Similarly, when processing the 11th frame signal, the 2nd to 11th frame signals are required. While processing the 11th frame signal to obtain the corresponding output signal, the coefficients of the filter need to be updated twice for use in processing the 12th frame signal, and the audio signal is processed frame by frame and frequency point by frequency point.

[0126] It should be noted that, in this embodiment, each frame signal is processed with reference to the M frames of signal before the frame signal, so that the echo and howling can be eliminated more cleanly, that is, the howling suppression effect is better. Wherein, M is an integer greater than 1, which is the number of filters in the filter structure.

[0127] Specifically, in this embodiment, the initial frame signal of any frequency point can be filtered based on the corresponding initial filter coefficient and the initial error signal to obtain the corresponding first filter coefficient and the first error signal, and then, based on the first filtering result, that is, the first filter coefficient and the first error signal, the initial frame signal is filtered again to obtain the corresponding second filter coefficient and the second error signal. Finally, the power spectra corresponding to the first error signal and the second error signal are compared, and the error signal corresponding to the smaller power spectrum is determined as the signal to be amplified corresponding to the initial frame signal.

[0128] In an embodiment of the present application, a possible implementation manner is provided, and a process of determining a first error signal includes:

[0129] Based on the preset update step size, the initial filter coefficient, the initial error signal, and the signal played by the speaker in the initial frame signal, the corresponding first filter coefficient is determined.

[0130] Based on the first filter coefficient, the signal played by the loudspeaker in the initial frame signal is filtered to obtain a corresponding first filtered signal.

[0131] The difference between the signal collected by the microphone in the initial frame signal and the first filtered signal is determined as a first error signal.

[0132] Specifically, in this embodiment, the filter coefficient corresponding to the nth frame signal of the frequency point k obtained by the first update (the first filter coefficient mentioned above) can be estimated according to the following formula:

[0133]

[0134] in, is the estimated initial filter coefficient corresponding to the n-th frame signal of frequency point k, E(n, k) is the initial error signal corresponding to the n-th frame signal of frequency point k, U(n, k) is the n-th frame signal of frequency point k in the signal played by the speaker, α is the update step size, δ is a small integer, generally 0.0001, to prevent the inner product of signal U(n, k) from being too small, causing the stability performance to decrease.

[0135] In this embodiment, the first error signal corresponding to the nth frame signal at frequency k can be obtained according to the following formula:

[0136]

[0137] Among them, Y(n, k) is the nth frame signal of frequency point k in the signal collected by the microphone, is the first filtered signal corresponding to the nth frame signal of frequency point k.

[0138] In an embodiment of the present application, a possible implementation method is provided, and a process of determining a second error signal includes:

[0139] Based on the preset update step size, the first filter coefficient, the first error signal, and the signal played by the speaker in the initial frame signal, a corresponding second filter coefficient is obtained.

[0140] Based on the second filter coefficient, the signal played by the loudspeaker in the initial frame signal is filtered to obtain a corresponding second filtered signal.

[0141] The difference between the signal collected by the microphone in the initial frame signal and the second filtered signal is determined as the second error signal.

[0142] Specifically, in this embodiment, the second filter coefficient after the first filter coefficient is updated for the second time can be estimated according to the following formula:

[0143]

[0144] in, is the first filter coefficient, E 1 (n, k) is the first error signal corresponding to the nth frame signal of frequency point k, U(n, k) is the nth frame signal of frequency point k in the signal played by the loudspeaker, α is the update step size, δ is a small integer, generally 0.0001, to prevent the inner product of signal U(n, k) from being too small, causing a decrease in stability performance.

[0145] In this embodiment, the second error signal can be obtained according to the following formula:

[0146]

[0147] Among them, Y(n, k) is the nth frame signal of frequency point k in the signal collected by the microphone, It is the second filtered signal corresponding to the nth frame signal of frequency point k.

[0148] A possible implementation method is provided in an embodiment of the present application. The process of processing any non-initial frame signal at any frequency point includes:

[0149] Based on the first filter coefficient and the first error signal updated in the previous frame signal processing process, and the current frame signal, obtain the first filter coefficient and the first error signal corresponding to the current frame signal;

[0150] Based on the first filter coefficient and the first error signal corresponding to the current frame signal, and the current frame signal, obtain the second filter coefficient and the second error signal corresponding to the current frame signal;

[0151] The power spectra corresponding to the first error signal and the second error signal corresponding to the current frame signal are compared, and the error signal corresponding to the smaller power spectrum is determined as the signal to be amplified corresponding to the current frame signal.

[0152] In this embodiment, if the current frame signal (not the initial frame signal) is the 20th frame signal of a certain frequency point, the 20th frame signal is processed based on the first filter coefficient and the first error signal obtained during the processing of the 19th frame signal of the frequency point to obtain a corresponding output signal, and the first filter coefficient is updated twice for use in processing the 21st frame signal.

[0153] It should be understood that in this embodiment, the specific process of processing the non-initial frame signal at any frequency point is similar to the specific process of processing the initial frame signal in the above embodiment.

[0154] It should be noted that, in this embodiment, when processing the current frame signal of any frequency point, the first filter system obtained in the process of processing the previous frame signal and the first error signal obtained after processing the previous frame signal are used.

[0155] For example, the error signal corresponding to the signal of the n+1th frame (any non-initial frame) at frequency k can be determined according to the following formula:

[0156]

[0157] Among them, Y(n+1, k) is the n+1th frame signal of frequency point k in the signal collected by the microphone, To update the initial filter coefficient corresponding to the nth frame signal of frequency k The transpose of the first filter coefficients obtained.

[0158] The first error signal corresponding to the (n+1)th frame signal at frequency point k can be determined according to the following formula:

[0159]

[0160] Among them, Y(n+1, k) is the n+1th frame signal of frequency point k in the signal collected by the microphone, is the second filtered signal corresponding to the n+1th frame signal of frequency point k, To update the initial filter coefficient corresponding to the n+1th frame signal of frequency k The transpose of the first filter coefficients after .

[0161] The first filter coefficient obtained by updating the filter coefficient corresponding to the n+1th frame signal of frequency point k is obtained according to the following formula:

[0162]

[0163] in, is the first filter coefficient obtained by updating the initial filter coefficient corresponding to the nth frame signal of frequency point k, and E(n+1, k) is the initial error signal corresponding to the (n+1)th frame signal of frequency point k.

[0164] The second error signal corresponding to the (n+1)th frame signal at frequency point k can be determined according to the following formula:

[0165]

[0166] Among them, Y(n+1, k) is the n+1th frame signal of frequency point k in the signal collected by the microphone, is the second filtered signal corresponding to the n+1th frame signal of frequency point k, The transpose of the second filter coefficient obtained by updating the first filter coefficient corresponding to the (n+1)th frame signal of frequency point k.

[0167] The second filter coefficient is obtained according to the following formula:

[0168]

[0169] in, is the first filter coefficient obtained by updating the filter coefficient corresponding to the n+1th frame signal of frequency point k, E 1 (n+1, k) is the first error signal corresponding to the (n+1)th frame signal of frequency point k.

[0170] It should be noted that, in this embodiment, during the process of calculating the error signal, if n=1, then U=0, that is, when processing the first frame of input signal, the speaker has no output signal yet.

[0171] On the basis of determining the first error signal and the second error signal, the corresponding output signal can be determined according to the following formula:

[0172] U'(n+1,k)=C(n+1,k)*min(E 1 (n+1, k), E 2 (n+1, k)}

[0173] Wherein, G(n+1, k) represents the first error signal E corresponding to the n+1th frame of frequency point k. 1 (n+1, k) and the second error signal E 2 The smaller one in (n+1,k) is used for signal processing, including automatic gain control, signal amplification, power amplification, etc. U′(n+1,k) represents the output signal after the n+1th frame signal processing at frequency k.

[0174] In summary, the howling suppression method provided in the embodiment of the present application multiplexes the filter structure adopted by the howling suppression echo cancellation. When the filter coefficient is updated, the reference signal is the signal played by the speaker, and it is updated twice. Therefore, the loop gain from the microphone to the speaker can be improved, thereby improving the howling suppression capability of the sound reinforcement system.

[0175] The embodiment of the present application further provides a howling suppression device, including: a signal processing module, an adder, a comparator, an adaptive filter, a fast Fourier transform module and an inverse transform module.

[0176] The input end of the adaptive filter is connected to the speaker, the output end of the adaptive filter is connected to the input end of the adder, the output end of the adder is connected to the input end of the comparator, the output end of the comparator is connected to the input end of the signal processing module, and the output end of the signal processing module is connected to the speaker.

[0177] The fast Fourier transform module is used to pre-process the audio signal collected by the microphone and convert the audio signal into the frequency domain, wherein the audio signal includes: the signal collected by the microphone and the signal played by the speaker. The adaptive filter is used to perform two echo suppression and howling suppression processes on the converted audio signal and then output it to the adder.

[0178] The adder is used to subtract the converted audio signal from the two output signals of the adaptive filter and output them to the comparator. The comparator is used to compare the two received signals and output the smaller signal to the signal processing module. The signal processing module performs local sound amplification processing on the received signal and transmits it to the inverse Fourier transform module. The inverse Fourier transform module is used to convert the received signal into the time domain, obtain the target audio and transmit it to the speaker for playback.

[0179] The embodiment of the present application provides a sound system, comprising: a speaker and the howling suppression device provided in the above embodiment. The speaker is connected to the input end of the adaptive filter in the howling suppression device, the output end of the adaptive filter is connected to the input end of the adder in the howling suppression device, the output end of the adder is connected to the input end of the comparator, the output end of the comparator is connected to the input end of the signal processing module in the howling suppression device, and the output end of the signal processing module is connected to the speaker.

[0180] The audio signal collected by the microphone can be transmitted to the speaker in a wireless or wired manner. For example, the speaker in this embodiment can be a Bluetooth speaker, which transmits the audio signal data to the microphone via Bluetooth, or can be connected to the microphone via WiFi or other local area network access methods. When the microphone collects the audio signal, it is transmitted to the howling suppression device in the speaker. After the howling suppression device performs a howling analysis on the audio signal, the audio signal and the generated reference signal are sent to the speaker for playback.

[0181] An embodiment of the present application provides a sound amplification system, including: a microphone, a speaker, and the howling suppression device provided in the above embodiment, wherein the howling suppression device is arranged between the microphone and the speaker, and is used to receive an audio signal collected by the microphone and output the generated target audio to the speaker for playback.

[0182] Figure 4 A schematic diagram of the structure of a processor provided in an embodiment of the present application. Figure 4 The processor 20 shown includes: a signal conversion module 201 and a signal processing update module 202. Among them,

[0183] The signal conversion module 201 is used to pre-process the audio signal in the sound reinforcement system and convert the audio signal into the frequency domain, wherein the audio signal includes: a signal collected by a microphone and a signal played by a speaker.

[0184] The signal processing update module 202 is used to perform the following processing on each frame signal of each frequency point in the converted audio signal: filtering the converted signal played by the speaker twice based on the adaptive filter; determining the signal to be amplified based on the difference between the signal collected by the converted microphone and the signal played by the speaker after the two filtering, and updating the filter coefficient twice during the processing using the signal played by the speaker in the current frame signal as a reference signal for use in processing the next frame signal.

[0185] The signal conversion module 201 is further used to convert the signal to be amplified into the time domain after amplification processing to obtain the target audio.

[0186] The processor of the embodiment of the present application can execute the howling suppression method provided in the embodiment of the present application, and its implementation principle is similar. The actions performed by each module and unit in the processor in each embodiment of the present application correspond to the steps in the howling suppression method in each embodiment of the present application. For the detailed functional description of each module of the processor, please refer to the description in the corresponding howling suppression method shown in the previous text, which will not be repeated here.

[0187] It should be noted here that the above-mentioned processor provided in the embodiment of the present application can implement all the method steps implemented in the above-mentioned method embodiment, and can achieve the same technical effect. The parts and beneficial effects that are the same as those in the method embodiment will not be described in detail here.

[0188] It should be noted that each embodiment in this specification is described in a progressive manner, and each embodiment focuses on the differences from other embodiments, and the same or similar parts between the embodiments can be referred to each other. For the device embodiment, since it is basically similar to the method embodiment, the description is relatively simple, and the relevant parts can be referred to the partial description of the method embodiment.

[0189] In several embodiments provided in the present application, it should be understood that the disclosed devices and methods may also be implemented in other ways. The device embodiments described above are merely schematic. For example, the flowcharts and block diagrams in the accompanying drawings show the possible architectures, functions and operations of the devices, methods and computer program products according to multiple embodiments of the present application. In this regard, each box in the flowchart or block diagram may represent a module, a program segment or a part of a code, and a module, a program segment or a part of a code includes one or more executable instructions for implementing a specified logical function.

[0190] It should also be noted that in some alternative implementations, the functions marked in the blocks may also occur in a different order than that marked in the accompanying drawings. For example, two consecutive blocks may actually be executed substantially in parallel, and they may sometimes be executed in reverse order, depending on the functions involved. It should also be noted that each block in the block diagram and / or flow chart, and the combination of blocks in the block diagram and / or flow chart, may be implemented with a dedicated hardware-based system that performs the specified functions or actions, or may be implemented with a combination of dedicated hardware and computer instructions.

[0191] In addition, the functional modules in the various embodiments of the present application may be integrated together to form an independent part, or each module may exist separately, or two or more modules may be integrated to form an independent part.

[0192] If the function is implemented in the form of a software function module and sold or used as an independent product, it can be stored in a computer-readable storage medium. Based on this understanding, the technical solution of the present application, or the part that contributes to the prior art or the part of the technical solution, can be embodied in the form of a software product, which is stored in a storage medium and includes several instructions for a computer device (which can be a personal computer, a server, or a network device, etc.) to perform all or part of the steps of the various embodiments of the present application. The aforementioned storage medium includes: U disk, mobile hard disk, read-only memory (ROM, Read-Only Memory), random access memory (RAM, Random Access Memory), disk or optical disk, and other media that can store program codes.

[0193] The above are only embodiments of the present application and are not intended to limit the scope of protection of the present application. For those skilled in the art, the present application may have various changes and variations. Any modifications, equivalent substitutions, improvements, etc. made within the spirit and principles of the present application should be included in the scope of protection of the present application. It should be noted that similar reference numerals and letters represent similar items in the following drawings, so once an item is defined in one drawing, it does not need to be further defined and explained in the subsequent drawings.

[0194] The above are only optional implementation methods for some implementation scenarios of the present application. It should be pointed out that for ordinary technicians in this technical field, without departing from the technical concept of the scheme of the present application, other similar implementation methods based on the technical ideas of the present application are also within the protection scope of the embodiments of the present application.

Claims

1. A howling suppression method, It is characterized in that include: Preprocessing the audio signal in the sound reinforcement system, converting the audio signal into the frequency domain, wherein the audio signal includes: a signal collected by a microphone and a signal played by a loudspeaker; The following processing is performed on each frame signal of each frequency point in the converted audio signal: The converted signal played by the loudspeaker is filtered twice based on an adaptive filter; Based on the difference between the signal collected by the converted microphone and the signal played by the loudspeaker after two filtering, the signal to be amplified is determined, and during the processing, the filter coefficients are updated twice using the signal played by the loudspeaker in the current frame signal as a reference signal for use in the next frame signal processing; The signal to be amplified is processed and then converted into the time domain to obtain the target audio; The process of updating the filter coefficients twice using the signal played by the loudspeaker in the current frame signal as the reference signal includes: Taking the signal played by the loudspeaker in the current frame signal as a reference signal, updating the filter coefficient based on the difference between the signal collected by the microphone in the current frame signal and the reference signal to obtain a first filter coefficient; The signal played by the loudspeaker after being filtered once in the current frame signal is used as a reference signal, and the first filter coefficient is updated based on the difference between the signal collected by the microphone in the current frame signal and the reference signal to obtain the second filter coefficient.

2. The method according to claim 1, It is characterized in that Before processing the initial frame signal of any frequency point, the method further includes: Determining initial filter coefficients and initial error signals corresponding to a transfer function of a loudspeaker-to-microphone path in the sound reinforcement system; The following processing is performed on the initial frame signal of any frequency point: Based on the initial filter coefficient, the initial error signal and the initial frame signal, obtaining an updated first filter coefficient and a first error signal; Based on the first filter coefficient, the first error signal and the initial frame signal, obtaining an updated second filter coefficient and a second error signal; comparing power spectra corresponding to the first error signal and the second error signal, and determining the error signal corresponding to the smaller power spectrum as the signal to be amplified corresponding to the initial frame signal; The initial error signal is determined based on the signal played by the loudspeaker and the signal collected by the microphone in the initial frame signal, as well as the initial filter coefficient.

3. The method according to claim 2, It is characterized in that The process of determining a first error signal comprises: Determine a corresponding first filter coefficient based on a preset update step size, the initial filter coefficient, the initial error signal, and a signal played by a speaker in the initial frame signal; Based on the first filter coefficient, filtering the signal played by the loudspeaker in the initial frame signal to obtain a corresponding first filtered signal; The difference between the signal collected by the microphone in the initial frame signal and the first filtered signal is determined as the first error signal.

4. The method according to claim 2, It is characterized in that The process of determining the second error signal comprises: Obtaining corresponding second filter coefficients based on a preset update step size, the first filter coefficients, the first error signal, and a signal played by a speaker in the initial frame signal; Based on the second filter coefficient, filtering the signal played by the loudspeaker in the initial frame signal to obtain a corresponding second filtered signal; The difference between the signal collected by the microphone in the initial frame signal and the second filtered signal is determined as the second error signal.

5. The method according to any one of claims 2 to 4, It is characterized in that The following processing is performed on any non-initial frame signal at any frequency point: Based on the first filter coefficient and the first error signal updated in the previous frame signal processing process, and the current frame signal, obtain the first filter coefficient and the first error signal corresponding to the current frame signal; Based on the first filter coefficient and the first error signal corresponding to the current frame signal, and the current frame signal, obtain the second filter coefficient and the second error signal corresponding to the current frame signal; The power spectra corresponding to the first error signal and the second error signal corresponding to the current frame signal are compared, and the error signal corresponding to the smaller power spectrum is determined as the signal to be amplified corresponding to the current frame signal.

6. A howling suppression device, It is characterized in that include: Signal processing module, adder, comparator, adaptive filter, fast Fourier transform module and inverse Fourier transform module; The input end of the adaptive filter is connected to the speaker, the output end of the adaptive filter is connected to the input end of the adder, the output end of the adder is connected to the input end of the comparator, the output end of the comparator is connected to the input end of the signal processing module, and the output end of the signal processing module is connected to the speaker; The fast Fourier transform module is used to pre-process the audio signal collected by the microphone and convert the audio signal into the frequency domain, wherein the audio signal includes: the signal collected by the microphone and the signal played by the speaker; The adaptive filter is used to perform echo suppression and howling suppression processing twice on the converted audio signal and then output it to the adder; The adder is used for respectively subtracting the converted audio signal from the signals output twice by the adaptive filter and outputting the subtractions to the comparator; The comparator is used to compare the two received signals and output the smaller signal to the signal processing module; The signal processing module performs local amplification processing on the received signal and then transmits it to the inverse Fourier transform module; The inverse Fourier transform module is used to convert the received signal into the time domain, obtain the target audio and transmit it to the speaker for playback; The process of the adaptive filter performing two echo suppression and howling suppression processes on the converted audio signal includes: The following processing is performed on each frame signal of each frequency point in the converted audio signal: The converted signal played by the loudspeaker is filtered twice based on an adaptive filter; Based on the difference between the signal collected by the converted microphone and the signal played by the loudspeaker after two filtering, the signal to be amplified is determined, and during the processing, the filter coefficients are updated twice using the signal played by the loudspeaker in the current frame signal as a reference signal for use in the next frame signal processing; The process of updating the filter coefficients twice using the signal played by the loudspeaker in the current frame signal as the reference signal includes: Taking the signal played by the loudspeaker in the current frame signal as a reference signal, updating the filter coefficient based on the difference between the signal collected by the microphone in the current frame signal and the reference signal to obtain a first filter coefficient; The signal played by the loudspeaker after being filtered once in the current frame signal is used as a reference signal, and the first filter coefficient is updated based on the difference between the signal collected by the microphone in the current frame signal and the reference signal to obtain the second filter coefficient.

7. A processor, It is characterized in that include: A signal conversion module and a signal processing update module, wherein: The signal conversion module is used to pre-process the audio signal in the sound reinforcement system and convert the audio signal into the frequency domain, wherein the audio signal includes: a signal collected by a microphone and a signal played by a speaker; The signal processing update module is used to perform the following processing on each frame signal of each frequency point in the converted audio signal: filtering the converted signal played by the loudspeaker twice based on the adaptive filter; determining the signal to be amplified based on the difference between the signal collected by the microphone after the conversion and the signal played by the loudspeaker after the two filtering, and updating the filter coefficient twice during the processing using the signal played by the loudspeaker in the current frame signal as a reference signal for use in the next frame signal processing; The signal conversion module is further used to convert the signal to be amplified into the time domain after amplification processing to obtain the target audio; The process of updating the filter coefficients twice using the signal played by the loudspeaker in the current frame signal as the reference signal includes: Taking the signal played by the loudspeaker in the current frame signal as a reference signal, updating the filter coefficient based on the difference between the signal collected by the microphone in the current frame signal and the reference signal to obtain a first filter coefficient; The signal played by the loudspeaker after being filtered once in the current frame signal is used as a reference signal, and the first filter coefficient is updated based on the difference between the signal collected by the microphone in the current frame signal and the reference signal to obtain the second filter coefficient.

8. A sound system, It is characterized in that include: A speaker, and a howling suppression device as claimed in claim 6, wherein: The speaker is connected to the input end of the adaptive filter in the howling suppression device, the output end of the adaptive filter is connected to the input end of the adder in the howling suppression device, the output end of the adder is connected to the input end of the comparator, the output end of the comparator is connected to the input end of the signal processing module in the howling suppression device, and the output end of the signal processing module is connected to the speaker.

9. A sound amplification system, It is characterized in that include: A microphone, a speaker, and a howling suppression device as claimed in claim 6, wherein the howling suppression device is arranged between the microphone and the speaker, and the howling suppression device is used to receive the audio signal collected by the microphone and output the target audio obtained by signal processing to the speaker for playback.

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