Patent Description:
An intelligent device may use a microphone (MIC) array for receiving sound. A MIC beamforming technology may be used to improve voice signal processing quality to increase a voice recognition rate in a real environment. However, a multi-MIC beamforming technology may be sensitive to a MIC position error, thereby affecting performance. In addition, increase of the number of MICs may increase product cost of the device.

Therefore, more and more intelligent devices are provided with only two MICs. A blind source separation technology completely different from the multi-MIC beamforming technology may be used for the two MICs for voice enhancement. How to improve the processing efficiency of blind source separation and reduce the latency is a problem to be solved in the blind source separation technology.

The present disclosure provides an audio signal processing method and device.

According to an aspect of the embodiments of the present disclosure, a non-transitory computer-readable storage medium is provided, which may have stored computer-executable instructions that, when executed by a processor, implement the audio signal processing method of any of the above. The technical solutions provided by embodiments of the present disclosure may have the following beneficial effects. In the embodiments of the present disclosure, audio signals may be processed by windowing, so that the audio signal of each frame can get stronger and then weaker. There is an overlapping area between every two adjacent frames, that is, a frame shift, so that the separated signal can maintain continuity. Meanwhile, in the embodiments of the present disclosure, an asymmetric window is used to window the audio signals, so that the length of a frame shift can be set according to actual needs. If a smaller frame shift is set, less system latency can be achieved, which in turn improves the processing efficiency and the timeliness of separated audio signals.

<FIG> is a flowchart of an audio signal processing method according to an exemplary embodiment. As shown in <FIG>, the method includes the following operations.

In S101, audio signals sent by at least two sound sources respectively are acquired through at least two MICs to obtain respective original noisy signals of the at least two MICs in a time domain.

In S102, for each frame in the time domain, a first asymmetric window is used to perform a windowing operation on the respective original noisy signals of the at least two MICs to acquire windowed noisy signals.

In S103, time-frequency conversion is performed on the windowed noisy signals to acquire respective frequency-domain noisy signals of the at least two sound sources.

In S104, frequency-domain estimated signals of the at least two sound sources are acquired according to the frequency-domain noisy signals.

In S105, audio signals produced respectively by the at least two sound sources are obtained according to the frequency-domain estimated signals.

The method may be applied to a terminal. The terminal may be an electronic device integrated with two or more than two MICs. For example, the terminal may be a vehicle terminal, a computer or a server.

In an implementation, the terminal may be an electronic device connected with a predetermined device integrated with two or more than two MICs. The electronic device may receive an audio signal acquired by the predetermined device based on this connection and send the processed audio signal to the predetermined device based on the connection. For example, the predetermined device may be a speaker.

In a practical application, the terminal may include at least two MICs. The at least two MICs may simultaneously detect the audio signals respectively sent by the at least two sound sources to obtain the respective original noisy signals of the at least two MICs. Herein, it can be understood that, in the embodiment, the at least two MICs synchronously may detect the audio signals sent by the two sound sources.

Audio signals of audio frames in a predetermined time can be separated only after original noisy signals of the audio frames in the predetermined time are completely acquired.

There may be two or more than two MICs, and there may be two or more than two sound sources.

The original noisy signal is a mixed signal including sounds produced by at least two sound sources. For example, there may be two MICs, i.e., a MIC <NUM> and a MIC <NUM> respectively, and there may be two sound sources, i.e., a sound source <NUM> and a sound source <NUM> respectively. In such case, the original noisy signal of the MIC <NUM> may include audio signals of the sound source <NUM> and the sound source <NUM>, and the original noisy signal of the MIC <NUM> also may include the audio signals of both the sound source <NUM> and the sound source <NUM>.

In an example, there may be three MICs, i.e., a MIC <NUM>, a MIC <NUM> and a MIC <NUM> respectively, and there may be three sound sources, i.e., a sound source <NUM>, a sound source <NUM> and a sound source <NUM> respectively. In such case, the original noisy signal of the MIC <NUM> may include the audio signals of the sound source <NUM>, the sound source <NUM> and the sound source <NUM>, and the original noisy signals of the MIC <NUM> and the MIC <NUM> also may include the audio signals of all the sound source <NUM>, the sound source <NUM> and the sound source <NUM>.

It can be understood that, if a signal generated in a MIC based on a sound produced by a sound source is an audio signal, a signal generated by another sound source in the MIC is a noise signal. The sounds produced by the at least two sound sources need to be recovered from the at least two MICs. The number of sound sources is the same as the number of MICs.

It can be understood that, when a MIC acquires an audio signal of a sound produced by a sound source, an audio signal of at least one audio frame is acquired and the acquired audio signal is an original noisy signal of each MIC. The original noisy signal is a time-domain signal. The time-domain signal is converted into a frequency-domain signal based on time-frequency conversion.

Time-frequency conversion may be mutual conversion between a time-domain signal and a frequency-domain signal. Frequency-domain transformation may be performed on a time-domain signal based on Fast Fourier Transform (FFT). Or, frequency-domain transformation may be performed on a time-domain signal based on Short-Time Fourier Transform (STFT). Or, frequency-domain transformation may also be performed on a time-domain signal based on other Fourier transform.

In an implementation, when a n th frame of time-domain signal of the p th MIC is <MAT>, the n th frame of time-domain signal is converted into a frequency-domain signal, and a n th frame of original noisy signal may be determined to be <MAT>, where m is the number of discrete time points of the n th frame of time-domain signal, and k is a frequency point. Therefore, according to the embodiments, each frame of original noisy signal is obtained by change from the time domain to the frequency domain. Each frame of original noisy signal may also be obtained based on another FFT formula. There are no limits made herein.

In the embodiments, an asymmetric analysis window is used to perform a windowing operation on an original noisy signal in the time domain, and a signal segment of each frame is intercepted through a first asymmetric window to obtain a windowed noisy signal of each frame. Since voice data and video data are different, there is no concept of frames. However, in order to transmit and store data and to process programs in batches, data may be segmented according to a specified time period or based on the number of discrete time points, thereby forming audio frames in the time domain. However, direct segmentation to form audio frames may destroy the continuity of audio signals. In order to ensure the continuity of audio signals, part of overlapping data need to be retained in different frames. That is, there is a frame shift. The part where two adjacent frames overlap is the frame shift.

The asymmetric window means that a graph formed by a function waveform of a window function is an asymmetric graph. For example, function waveforms on both sides with the peak as the axis may be asymmetric.

In the embodiments, the window function is used to process each frame of audio signal, so that the signal can change from the minimum to the maximum and then to the minimum. In this way, the overlapping parts of two adjacent frames may not cause distortion after being superimposed.

When an audio signal is processed based on a symmetric window function, a frame shift may be half of a frame length, which may cause a large system latency, thereby reducing the separation efficiency and degrading the real-time interactive experience. Therefore, in the embodiments of the present disclosure, the asymmetric window is adopted to perform windowing processing on an audio signal, so that after each frame of audio signal is subjected to windowing, a higher intensity signal can be in the first half or the second half. Therefore, the overlapping parts between two adjacent frames of signals can be concentrated in a shorter interval, thereby reducing the latency and improving the separation efficiency.

In some embodiments, a definition domain of the first asymmetric window hA (m) may be greater than or equal to <NUM> and less than or equal to N, a peak may be hA (m<NUM>) = <NUM>, m<NUM> may be less than N and greater than <NUM>. 5N, and N may be a frame length of the audio signal.

In the embodiments of the present disclosure, the first asymmetric window hA (m) may be used as an analysis window to perform windowing processing on the original noisy signal of each frame. The frame length of the system is N, and the window length is also N, that is, each frame of signal has audio signal samples at N discrete time points.

The windowing processing performed according to the first asymmetric window refers to multiplying a sample value at each time point of a frame of audio signal by a function value at a corresponding time point of the function hA (m) , so that each frame of audio signal subjected to windowing can gradually get larger from <NUM> and then gradually get smaller. At the time point m<NUM> of the peak of the first asymmetric window, the windowed audio signal is the same as the original audio signal.

In the embodiments of the present disclosure, the time point m<NUM> where the peak of the first asymmetric window is may be less than N and greater than <NUM>. 5N, that is, after the center point. In such case, an overlap between two adjacent frames can be reduced, that is, the frame shift is reduced, thereby reducing the system latency and improving the efficiency of signal processing.

In some embodiments, the first asymmetric window hA(m) may include formula (<NUM>):
<MAT>
where HK(x) is a Hanning window with a window length of K, and M is a frame shift.

In the embodiments of the present disclosure, the first asymmetric window shown in formula (<NUM>) is provided. When the value of the time point m is less than N-M, the function of the first asymmetric window is represented by <MAT>, where H<NUM>(N-M)(m) is a Hanning window with a window length of <NUM>(N-M). The Hanning window is a type of cosine window, which may be represented by formula (<NUM>):
<MAT>.

When the value of the time point m is greater than N-M, the function of the first asymmetric window is represented by <MAT>, where H<NUM> (m - (N - <NUM>M)) is a Hanning window with a window length of <NUM>.

Therefore, the peak value of the first asymmetric window is at m=N-M. In order to reduce the latency, the frame shift M may be set smaller, for example, M=N/<NUM> or M=N/<NUM>, etc. In this way, the total latency of the system is only <NUM>, but less than N, so that the latency can be reduced.

The operation that audio signals produced respectively by the at least two sound sources are obtained according to the frequency-domain estimated signals includes that:.

In the embodiments of the present disclosure, an original noisy signal may be converted into a frequency-domain noisy signal after windowing processing and video conversion. Based on the frequency-domain noisy signal, separation processing may be performed to obtain frequency-domain signals of at least two sound sources after separation. In order to restore the audio signals of at least two sound sources, the obtained frequency-domain signal need to be converted back to the time domain after time-frequency conversion.

Time-domain conversion may be performed on the frequency-domain signal based on Inverse Fast Fourier Transform (IFFT). Or, the frequency-domain signal may be converted into a time-domain signal based on Inverse Short-Time Fourier Transform (ISTFT). Or, time-domain transform may also be performed on the frequency-domain signal based on other Fourier transform.

The separation signal back to the time domain is a time-domain separation signal in which each sound source is divided into different frames. In order to obtain a continuous audio signal from the sound source, windowing may be performed again to remove unnecessary duplicate parts. Then, continuous audio signals may be obtained by synthesis, and the respective audio signals from the sound sources are restored.

In this way, the noise in the restored audio signal can be reduced and the signal quality can be improved.

In some embodiments, the operation that a windowing operation is performed on the respective time-domain separation signals of the at least two sound sources using a second asymmetric window to acquire windowed separation signals may include that:
a windowing operation is performed on the time-domain separation signal of the nth frame using a second asymmetric window hS(m) to acquire an nth-frame windowed separation signal.

The operation that audio signals produced respectively by the at least two sound sources are acquired according to windowed separation signals may include that:
the audio signal of the (n-<NUM>)th frame is superimposed according to the nth-frame windowed separation signal to obtain the audio signal of the nth frame, where n is an integer greater than <NUM>.

In the embodiments of the present disclosure, a second asymmetric window may be used as a synthesis window to perform windowing processing on the above time-domain separation signal to obtain windowed separation signals. Then, the windowed separation signal of each frame may be added to a time-domain overlapping part of a preceding frame to obtain a time-domain separation signal of a current frame. In this way, a restored audio signal can maintain continuity and can be closer to the audio signal from the original sound source, and the quality of the restored audio signal can be improved.

In some embodiments, a definition domain of the second asymmetric window hS(m) may be greater than or equal to <NUM> and less than or equal to N, a peak may be hS(m<NUM>) = <NUM>, m<NUM> may be equal to N-M, N may be a frame length of each of the audio signals, and M may be a frame shift.

In the embodiments of the present disclosure, the second asymmetric window may be used as a synthesis window to perform windowing processing on each frame of separation audio signal. The second asymmetric window may take values only within twice the length of the frame shift, intercept the last <NUM> audio segments of each frame, and then add them to the overlapping part between a preceding frame and the current frame, that is, the frame shift part, to obtain the time-domain separation signal of the current frame. In this way, an audio signal from an original sound source can be restored based on consecutive processed each frame.

In some embodiments, the second asymmetric window hS(m) may include: <MAT>
where Hx(x) is a Hanning window with a window length of K.

In the embodiments of the present disclosure, the second asymmetric window shown in formula (<NUM>) is provided. When the value of the time point m is less than N-M and greater than N-<NUM>+<NUM>, the function of the first asymmetric window is represented by <MAT>, where H<NUM>(N-M) (m) is a Hanning window with a window length of <NUM>(N-M), and H<NUM> (m - (N - <NUM>M)) is a Hanning window with a window length of <NUM>.

When the value of the time point m is greater than N-M, the function of the second asymmetric window is represented by <MAT>, where H<NUM>M (m - (N - <NUM>M)) is a Hanning window with a window length of <NUM>. In this way, the peak value of the second asymmetric window is also located at m=N-M.

The operation that frequency-domain estimated signals of the at least two sound sources are acquired according to the frequency-domain noisy signals includes that:.

According to an initialized separation matrix consisting of a separation matrix of a preceding frame, a frequency-domain noisy signal is preliminarily separated to obtain a priori estimated signal, and then the separation matrix is updated according to the priori estimated signal. Finally, the frequency-domain noisy signal is separated according to the separation matrix to obtain a separated frequency-domain estimated signal, that is, a frequency-domain posterior estimated signal.

For example, the above separation matrix may be determined based on an eigenvalue solved by a covariance matrix. The covariance matrix Vp(k,n) may satisfy the following relationship <MAT>, where β is a smoothing coefficient, Vp(k,n-<NUM>) is the covariance matrix of the preceding frame, and Xp(k,n) is the original noisy signal of the current frame, that is, the frequency-domain noisy signal. <MAT> is a conjugate transpose matrix of the original noisy signal of the current frame. <MAT> is a weighting factor, where <MAT> is an auxiliary variable. G(Yp (n)) = -log p(Yp (n)) is a contrast function. Herein, p(Yp (n)) represents a multi-dimensional super-Gaussian prior probability density distribution model based on the entire frequency band of the p th sound source, which is the above-mentioned distribution function. Yp(n) is a conjugate matrix of Yp (n), Yp (n) is the frequency-domain estimated signal of the pth sound source in the nth frame, and Yp (k,n) represents the frequency-domain estimated signal of the pth sound source at the kth frequency point of the nth frame, that is, the frequency-domain priori estimated signal.

By updating the separation matrix according to the above method, a more accurate frequency domain estimation signal can be obtained with higher separation performance. After time-frequency conversion, the audio signal from the sound source may be restored.

The embodiments of the present disclosure also provide the following examples.

<FIG> is a schematic diagram of an application scenario of an audio signal processing method according to an exemplary embodiment. <FIG> is a flowchart of an audio signal processing method according to an exemplary embodiment. Referring to <FIG> and <FIG>, in the audio signal processing method, sound sources include a sound source <NUM> and a sound source <NUM>, and MICs include a MIC <NUM> and a MIC <NUM>. Based on the audio signal processing method, the sound source <NUM> and the sound source <NUM> are recovered from signals of the MIC <NUM> and the MIC <NUM>. As shown in <FIG>, the method includes the following operations.

In operation S301, W(k) and Vp (k) are initialized.

Initialization may include the following operations.

It is supposed that a system frame length is Nfft, and a frequency point is K=Nfft/<NUM>+<NUM>.

<MAT>, where <MAT> is a zero matrix, p represents a MIC, and p = <NUM>,<NUM>.

In operation S302, an n th frame of original noisy signal of the p th MIC is obtained. <MAT> represents a frame of time-domain signal of the p th MIC. m = <NUM>,. Nfft represents the system frame length and the length of FFT, and M represents a frame shift.

An asymmetric analysis window is added to <MAT> for performing FFT to obtain:
<MAT>
where m is the number of points selected for Fourier transform, FFT is fast Fourier transform, and <MAT> is an n th frame of time-domain signal of the p th MIC. The time-domain signal is an original noisy signal. hA (m) is the asymmetric analysis window.

A measured signal of Xp(k,n) is X(k,n)=[X<NUM>(k,n),X<NUM>(k,n)]T, where [X<NUM>(k,n),X<NUM>(k,n)]T is a transposed matrix.

STFT refers to multiplying a time-domain signal of a current frame by an analysis window and performing FFT to obtain time-frequency data. A separation matrix may be estimated through an algorithm to obtain time-frequency data of a separated signal, IFFT may be performed to convert the time-frequency data to the time domain, and then the converted signal may be multiplied with a synthesis window and added to a time-domain overlapping part output from a preceding frame to obtain a reconstructed separated time-domain signal. This is called an overlap-add technology.

Existing windowing algorithms generally apply a symmetry based Hanning window or Hamming window or other window functions. For example, a root period Hanning window may be used:
<MAT>
where the frame shift is <MAT>, and the window length is N = Nfft. The system latency is Nfft points. Since Nfft is generally <NUM> or greater, the latency may be <NUM> or greater when a system sampling rate is fs = <NUM>kHz.

In the embodiments of the present disclosure, an asymmetric analysis window and a synthesis window may be adopted, a window length may be N=Nfft, and a frame shift may be M. In order to obtain a low latency, M generally is small. For example, it may be set to <MAT>, or other values.

For example, the asymmetric analysis window may apply the following function:
<MAT>.

The asymmetric synthesis window may apply the following function:
<MAT>.

When N=<NUM> and M=<NUM>, the function curve of the asymmetric analysis window is as shown in <FIG>, and the function curve of the asymmetric synthesis window is as shown in <FIG>.

In operation S303, a priori frequency-domain estimate of signals of the two sound sources is obtained by use of W(k) of a preceding frame.

It may be set that the priori frequency-domain estimate of the signals of the two sound sources is <MAT>, where <MAT> are estimated values of the sound source <NUM> and the sound source <NUM> at a frequency-frequency point (k, n) respectively.

A measured matrix X(k, n) may be separated through the separation matrix W(k) to obtain Y(k,n)=W(k)'X(k,n) , where W'(k) is a separation matrix ofa preceding frame (i.e., a last frame prior to a current frame).

Then, a priori frequency-domain estimate of the n th sound source in the p th frame is: <MAT>.

In operation S304, a weighted covariance matrix Vp(k,n) is updated.

The updated weighted covariance matrix may be calculated by: <MAT>, where β is a smoothing coefficient, β being <NUM> in an example; Vp(k,n-<NUM>) is a weighted covariance matrix of the preceding frame; <MAT> is a conjugate transpose of Xp(k,n); <MAT> is a weighting coefficient, <MAT> being an auxiliary variable; and G(Yp (n)) = -log p(Yp (n)) is a contrast function.

p(Yp (n)) represents a whole-band-based multidimensional super-Gaussian priori probability density function of the p th sound source. In an example, <MAT>. In such case, if <MAT>, then <MAT>.

In operation S305, an eigenproblem is solved to obtain an eigenvector ep(k,n).

Herein, ep(k,n) is an eigenvector corresponding to the p th MIC.

The eigenproblem V<NUM>(k,n)ep(k,n)=λp(k,n)V<NUM>(k,n)ep(k,n) is solved to obtain:
<MAT>
<MAT>
<MAT>and
<MAT>
where <MAT>, tr(A) is a trace function and refers to making a sum of elements on a main diagonal of a matrix A; det(A) refers to calculating a determinant of the matrix A; and λ<NUM>, λ<NUM>, e<NUM>, and e<NUM> are eigenvalues.

In operation S306, an updated separation matrix W(k) of each frequency point is obtained.

The updated separation matrix <MAT> of the current frame is obtained based on the eigenvector of the eigenproblem.

In operation S307, a posteriori frequency-domain estimate of the signals of the two sound sources is obtained by use of W(k) of the current frame.

The original noisy signal is separated by use of W(k) of the current frame to obtain the posteriori frequency-domain estimate Y(k,n) = [Y<NUM>(k,n),Y<NUM>(k,n)]T=W(k)X(k,n) of the signals of the two sound sources.

In operation S308, time-frequency conversion is performed based on the posteriori frequency-domain estimate to obtain a separated time-domain signal.

IFFT may be performed, a synthesis window may be added, the time-domain overlapping part of a current frame may be added to the time-domain overlapping part of a preceding frame to obtain the separated time-domain signal yp(m) of the current frame, and p=<NUM>,<NUM>. <MAT><MAT><MAT><MAT><MAT> is a signal after windowing the time-domain signal of the current frame, <MAT> is the time-domain overlapping part of each frame preceding the current frame, and <MAT> is the time-domain overlapping part of the current frame. <MAT> is updated for use of overlapping addition of the next frame.

ISTFT and overlapping-addition may be performed on Yp(n) = [Yp (<NUM>, n),. Yp (K,n)]Tk = <NUM>,. , K respectively to obtain a separated time-domain sound source signal <MAT>, that is, <MAT>, where m=<NUM>,. ,Nfft, and p=<NUM>,<NUM>.

After the above processing by the analysis window and the synthesis window, the system latency can be <NUM> points and the latency can be <NUM> / fs ms (millisecond). When the number of FFT points is changed, the system latency that meets actual needs can be obtained by controlling the size of M, and the contradiction between the system latency and the performance of the algorithm is solved.

<FIG> is a block diagram of an audio signal processing device according to an exemplary embodiment. Referring to <FIG>, the device <NUM> includes a first acquisition module <NUM>, a first windowing module <NUM>, a first conversion module <NUM>, a second acquisition module <NUM>, and a third acquisition module <NUM>. Each of these modules may be implemented as software, or hardware, or a combination of software and hardware.

The first acquisition module <NUM> is configured to acquire audio signals from at least two sound sources respectively through at least two MICs to obtain respective original noisy signals of the at least two MICs in a time domain.

The first windowing module <NUM> is configured to perform, for each frame in the time domain, a windowing operation on the respective original noisy signals of the at least two MICs using a first asymmetric window to acquire windowed noisy signals.

The first conversion module <NUM> is configured to perform time-frequency conversion on the windowed noisy signals to acquire respective frequency-domain noisy signals of the at least two sound sources.

The second acquisition module <NUM> is configured to acquire frequency-domain estimated signals of the at least two sound sources according to the frequency-domain noisy signals.

The third acquisition module <NUM> is configured to obtain audio signals produced respectively by the at least two sound sources according to the frequency-domain estimated signals.

In some embodiments, a definition domain of the first asymmetric window hA (m) may be greater than or equal to <NUM> and less than or equal to N, a peak may be hA (m<NUM>) = <NUM>, m<NUM> may be less than N and greater than <NUM>. 5N, and N may be a frame length of each of the audio signals.

In some embodiments, the first asymmetric window hA (m) may include:
<MAT>
where HK(x) is a Hanning window with a window length of K, and M is a frame shift.

The third acquisition module <NUM> includes:.

In some embodiments, the second windowing module is specifically configured to:
perform a windowing operation on a time-domain separation signal of a nth frame using the second asymmetric window hs (m) to acquire an nth-frame windowed separation signal.

The first acquisition sub-module is specifically configured to:
superimpose an audio signal of a (n-<NUM>)th frame according to the nth-frame windowed separation signal to obtain an audio signal of the nth frame, where n is an integer greater than <NUM>.

In some embodiments, a definition domain of the second asymmetric window hs (m) may be greater than or equal to <NUM> and less than or equal to N, a peak may be hs (m<NUM>) = <NUM>, m<NUM> may be equal to N-M, N may be a frame length of each of the audio signals, and M is a frame shift.

In some embodiments, the second asymmetric window hs (m) may include:
<MAT>
where HK(x) is a Hanning window with a window length of K.

In some embodiments, the second acquisition module may include:.

With respect to the device in the above embodiment, the specific manners for performing operations by individual modules therein have been described in detail in the embodiment regarding the method, which will not be repeated herein.

<FIG> is a block diagram of a physical structure of a device <NUM> for audio signal processing according to an exemplary embodiment. For example, the device <NUM> may be a mobile phone, a computer, a digital broadcast terminal, a messaging device, a gaming console, a tablet, a medical device, exercise equipment, a personal digital assistant and the like.

The processing component <NUM> may include one or more processors <NUM> to execute instructions to perform all or part of the operations in the abovementioned method. Moreover, the processing component <NUM> may include one or more modules which facilitate interaction between the processing component <NUM> and the other components. For instance, the processing component <NUM> may include a multimedia module to facilitate interaction between the multimedia component <NUM> and the processing component <NUM>.

Examples of such data include instructions for any application programs or methods operated on the device <NUM>, contact data, phonebook data, messages, pictures, video, etc. The memory <NUM> may be implemented by any type of volatile or non-volatile memory devices, or a combination thereof, such as an Static Random Access Memory (SRAM), an Electrically Erasable Programmable Read-Only Memory (EEPROM), an Erasable Programmable Read-Only Memory (EPROM), a Programmable Read-Only Memory (PROM), a Read-Only Memory (ROM), a magnetic memory, a flash memory, and a magnetic or optical disk.

The multimedia component <NUM> includes a screen providing an output interface between the device <NUM> and a user. In some embodiments, the screen may include a Liquid Crystal Display (LCD) and a Touch Panel (TP). If the screen includes the TP, the screen may be implemented as a touch screen to receive an input signal from the user. The TP includes one or more touch sensors to sense touches, swipes and gestures on the TP. The touch sensors may not only sense a boundary of a touch or swipe action but also detect a duration and pressure associated with the touch or swipe action. The front camera and/or the rear camera may receive external multimedia data when the device <NUM> is in an operation mode, such as a photographing mode or a video mode. Each of the front camera and the rear camera may be a fixed optical lens system or have focusing and optical zooming capabilities.

The audio component <NUM> is configured to output and/or input an audio signal. For example, the audio component <NUM> includes a MIC, and the MIC is configured to receive an external audio signal when the device <NUM> is in the operation mode, such as a call mode, a recording mode and a voice recognition mode. The received audio signal may further be stored in the memory <NUM> or sent through the communication component <NUM>. In some embodiments, the audio component <NUM> further includes a speaker configured to output the audio signal.

The communication component <NUM> is configured to facilitate wired or wireless communication between the device <NUM> and another device. The device <NUM> may access a communication-standard-based wireless network, such as a Wireless Fidelity (WiFi) network, a 2nd-Generation (<NUM>) or 3rd-Generation (<NUM>) network or a combination thereof. In an exemplary embodiment, the communication component <NUM> receives a broadcast signal or broadcast associated information from an external broadcast management system through a broadcast channel. In an exemplary embodiment, the communication component <NUM> further includes a Near Field Communication (NFC) module to facilitate short-range communication. For example, the NFC module may be implemented based on a Radio Frequency Identification (RFID) technology, an Infrared Data Association (IrDA) technology, an Ultra-Wide Band (UWB) technology, a Bluetooth (BT) technology and another technology.

In an exemplary embodiment, the device <NUM> may be implemented by one or more Application Specific Integrated Circuits (ASICs), Digital Signal Processors (DSPs), Digital Signal Processing Devices (DSPDs), Programmable Logic Devices (PLDs), Field Programmable Gate Arrays (FPGAs), controllers, micro-controllers, microprocessors or other electronic components, and is configured to execute the abovementioned method.

In an exemplary embodiment, there is also provided a non-transitory computer-readable storage medium including an instruction, such as the memory <NUM> including instructions, and the instructions may be executed by the processor <NUM> of the device <NUM> to implement the abovementioned method. For example, the non-transitory computer-readable storage medium may be a ROM, a Random Access Memory (RAM), a Compact Disc Read-Only Memory (CD-ROM), a magnetic tape, a floppy disc, an optical data storage device and the like.

A non-transitory computer-readable storage medium is provided. When instructions in the storage medium are executed by a processor of a mobile terminal, the mobile terminal can implement any of the methods provided in the above embodiment.

In the description of the present disclosure, the terms "one embodiment," "some embodiments," "example," "specific example," or "some examples" and the like can indicate a specific feature described in connection with the embodiment or example, a structure, a material or feature included in at least one embodiment or example. In the present disclosure, the schematic representation of the above terms is not necessarily directed to the same embodiment or example.

In some embodiments, the control and/or interface software or app can be provided in a form of a non-transitory computer-readable storage medium having instructions stored thereon is further provided. For example, the non-transitory computer-readable storage medium can be a ROM, a CD-ROM, a magnetic tape, a floppy disk, optical data storage equipment, a flash drive such as a USB drive or an SD card, and the like.

Other implementation solutions of the present disclosure will be apparent to those skilled in the art from consideration of the specification and practice of the present disclosure. It is intended that the specification and examples be considered as exemplary only, with a true scope of the present disclosure being indicated by the following claims.

Claim 1:
A method for audio signal processing, comprising:
acquiring (S101) audio signals from at least two sound sources respectively through at least two microphones, MICs, to obtain respective original noisy signals of the at least two MICs in a time domain, wherein the number of the at least two sound sources is same as the number of the at least two MICs, the original noisy signal of each of the at least two MICs is a mixed signal including sounds produced by the at least two sound sources, and comprises an audio signal from one of the at least two sound sources and an audio signal(s) which is/are taken as a noise signal(s) from other one(s) of the at least two sound sources;
for each frame in the time domain, performing (S102) a windowing operation on the respective original noisy signals of the at least two MICs using a first asymmetric window to acquire respective windowed noisy signals of the at least two MICs;
performing (S103) time-frequency conversion on the respective windowed noisy signals of the at least two MICs to acquire respective frequency-domain noisy signals of the at least two sound sources;
acquiring (S104) respective frequency-domain estimated signals of the at least two sound sources according to the respective frequency-domain noisy signals of the at least two sound sources; and
obtaining (S105) audio signals produced respectively by the at least two sound sources according to the respective frequency-domain estimated signals of the at least two sound sources,
characterized in that,
acquiring (S104) the respective frequency-domain estimated signals of the at least two sound sources according to the respective frequency-domain noisy signals of the at least two sound sources comprises:
separating the respective frequency-domain noisy signals according to a separation matrix of a preceding frame of a current frame, to obtain frequency-domain priori estimated signals of the at least two sound sources;
updating the separation matrix according to the frequency-domain priori estimated signals of the at least two sound sources, to obtain an updated separation matrix as a separation matrix of the current frame; and
separating the respective frequency-domain noisy signals according to the separation matrix of the current frame, to obtain separated frequency-domain estimated signals as the frequency-domain estimated signals of the at least two sound sources.