Audio acquisition device and method, electronic equipment and computer program product

Abstract

The invention provides an audio acquisition device and method, electronic equipment and a computer program product. The device comprises a plurality of directional microphone units which are used for respectively collecting audio signals corresponding to the directional directions of the directional microphone units; the plurality of filter units are used for eliminating the audio signals in the non-pointing direction collected by each of the plurality of pointing microphone units; wherein the plurality of directional microphone units comprise a first directional microphone unit corresponding to a first directional direction and a second directional microphone unit corresponding to a second directional direction; one of the plurality of filter units is configured to cancel the audio signal in the first directional direction acquired by the second directional microphone unit. According to the invention, by calibrating the coefficients of the plurality of filter units, the extraction of the audio signal in the specific pointing direction is realized while the far-end audio signal is suppressed, so that the acquisition effect is enhanced, and the conference quality is improved.

Description

Technical Field

[0001] The present disclosure relates to the field of audio acquisition, and in particular to an audio acquisition device, method, electronic device and computer program product. Background Art

[0002] At present, online meetings have become the most commonly used form of office meetings, and a variety of smart conference speakers have emerged on the market. In the existing technology, most of them use microphone array beamforming algorithms, using the center of the microphone array as the reference coordinate system to receive all near-end sound signals around the conference speaker, suppress far-end sound signals, and prevent echoes. However, the disadvantage of this solution is that all near-end sound signals will be transmitted to the far end, which will interfere with the conference effect when there is noise interference nearby. Summary of the invention

[0003] The present disclosure is proposed in view of the above problems. The present disclosure provides an audio collection device, method, electronic device and computer program product.

[0004] According to one aspect of the present disclosure, there is provided an audio collection device, comprising: a plurality of directional microphone units, for respectively collecting audio signals corresponding to their directional directions; a plurality of filter units, for eliminating audio signals in non-directional directions collected by each of the plurality of directional microphone units; wherein the plurality of directional microphone units comprise a first directional microphone unit corresponding to a first directional direction and a second directional microphone unit corresponding to a second directional direction, and one of the plurality of filter units is configured to eliminate the audio signal in the first directional direction collected by the second directional microphone unit.

[0005] In addition, according to an aspect of the audio collection device of the present disclosure, it also includes: each of the multiple directional microphone units and each directional microphone unit in a non-directional direction are connected to the same filter unit; and based on the different directional directions of the audio signal, the first filter unit connected to the first directional microphone unit and the second directional microphone unit is different from the second filter unit connected to the second directional microphone unit and the first directional microphone unit.

[0006] In addition, according to an aspect of the audio collection device of the present disclosure, the directional microphone unit includes: a slot subunit, used to connect to the shell unit of the audio collection device; a sealing subunit, used to seal both ends of the slot subunit; and a directional microphone subunit, located inside the slot subunit, for collecting audio signals.

[0007] In addition, according to an aspect of the audio collection device of the present disclosure, it further includes: the slot body subunit is in a strip shape; and the distances between the directional microphone subunit and the sealing subunits at both ends are different.

[0008] According to another aspect of the present disclosure, an audio collection method is provided, the method comprising: a first directional microphone unit and a second directional microphone unit respectively receive an audio signal in a first directional direction, and obtain a first audio signal in the first directional direction and a second audio signal in the first directional direction; based on the first audio signal in the first directional direction and the second audio signal in the first directional direction, a coefficient calibration is performed on a first filter unit connected to the first directional microphone unit and the second directional microphone unit to eliminate the interference of the second audio signal in the first directional direction on the first audio signal in the first directional direction.

[0009] In addition, according to an aspect of the present disclosure, an audio acquisition method, wherein, based on a first audio signal in a first pointing direction and a second audio signal in the first pointing direction, a coefficient calibration is performed on a first filter unit connected to a first directional microphone unit and a second directional microphone unit to eliminate interference of the second audio signal in the first pointing direction with the first audio signal in the first pointing direction, including: initializing the coefficient of the first filter unit; performing frequency domain conversion on the first audio signal in the first pointing direction and the second audio signal in the first pointing direction to generate a first signal frequency domain feature and a second signal frequency domain feature; determining new coefficients of the first filter unit based on the first audio signal in the first pointing direction and the second audio signal in the first pointing direction so that the product of the second signal frequency domain feature and the new coefficients is opposite in phase to the first signal frequency domain feature; performing time domain conversion on the new coefficients to determine the target coefficients of the first filter unit, and the target coefficients of the first filter unit are used to eliminate interference of the second audio signal in the first pointing direction with the first audio signal in the first pointing direction.

[0010] In addition, according to an aspect of the audio collection method of the present disclosure, it also includes: calibrating the coefficients of the filter units connected to each directional microphone unit in the multiple directional microphone units and each directional microphone unit in its non-directional direction, generating target coefficients for each filter unit to eliminate the interference of the audio signals collected by each directional microphone unit in the non-directional direction on the audio signals collected by the directional microphone unit in the directional direction.

[0011] In addition, according to an aspect of the audio acquisition method of the present disclosure, it also includes: denoising the audio signal in each directional direction based on the target coefficient of each filter unit to generate a denoised audio signal; based on the preset path of the directional direction of each directional microphone unit in a plurality of directional microphone units, weighted summing each denoised audio signal to generate a fused audio signal.

[0012] According to another aspect of the present disclosure, an electronic device is provided, including: a memory for storing computer-readable instructions; and a processor for executing the computer-readable instructions so that the electronic device performs the audio acquisition method as described above.

[0013] According to yet another aspect of the present disclosure, a computer program product is provided, including a computer program, wherein when the computer program is executed by a processor, the audio acquisition method as described above is implemented.

[0014] As will be described in detail below, according to the audio collection method of the embodiment of the present disclosure, the present disclosure calibrates multiple filter unit coefficients to suppress far-end audio signals while extracting audio signals in a specific direction, thereby enhancing the collection effect and improving the conference quality.

[0015] It is to be understood that both the foregoing general description and the following detailed description are exemplary, and are intended to provide further explanation of the technology as claimed. BRIEF DESCRIPTION OF THE DRAWINGS

[0016] The above and other purposes, features and advantages of the present disclosure will become more apparent by describing the embodiments of the present disclosure in more detail in conjunction with the accompanying drawings. The accompanying drawings are used to provide a further understanding of the embodiments of the present disclosure and constitute a part of the specification. Together with the embodiments of the present disclosure, they are used to explain the present disclosure and do not constitute a limitation of the present disclosure. In the accompanying drawings, the same reference numerals generally represent the same components or steps.

[0017] Figure 1 It is a schematic diagram illustrating an application scenario of the audio acquisition device according to an embodiment of the present disclosure.

[0018] Figure 2 is a device diagram illustrating an audio acquisition device according to an embodiment of the present disclosure.

[0019] Figure 3 It is a schematic diagram further illustrating the structure of a directional microphone unit according to an embodiment of the present disclosure.

[0020] Figure 4 FIG. 4 is a flowchart illustrating an audio acquisition method according to an embodiment of the present disclosure.

[0021] Figure 5 is a schematic diagram illustrating a test audio signal structure according to an embodiment of the present disclosure.

[0022] Figure 6 is a method flow chart further illustrating a coefficient calibration method according to an embodiment of the present disclosure.

[0023] Figure 7 is a schematic diagram further illustrating a coefficient calibration method according to an embodiment of the present disclosure.

[0024] Figure 8 is a hardware block diagram illustrating an electronic device according to an embodiment of the present disclosure.

[0025] Fig. 9 is a schematic diagram illustrating a computer program product according to an embodiment of the present disclosure. DETAILED DESCRIPTION

[0026] In order to make the purpose, technical solution and advantages of the present disclosure more obvious, the exemplary embodiments according to the present disclosure will be described in detail with reference to the accompanying drawings. Obviously, the described embodiments are only part of the embodiments of the present disclosure, rather than all the embodiments of the present disclosure, and it should be understood that the present disclosure is not limited to the exemplary embodiments described here.

[0027] First, refer to Figure 1 The application scenarios according to the embodiments of the present disclosure are summarized.

[0028] Figure 1 Schematic diagram of an application scenario of an audio acquisition device according to an embodiment of the present disclosure. Figure 1 As shown, the application scenario at least includes: an audio acquisition device 10 and at least one audio signal (for example: audio signal 11, audio signal 12, audio signal 13, audio signal 14).

[0029] Among them, the audio collection device 10 can be a smart speaker for online meetings, which is used to collect the sound emitted by at least one audio signal. At least one audio signal can be understood as a sound emitted by a human, a device or a system (for example: a participant, a computer, a mobile phone, a sound player, etc.). In addition, the audio signal can be arranged according to the pickup direction of the audio collection device 10. It is easy to understand that the present disclosure does not limit the specific type of the audio collection device 10, and the type and number of entities that emit audio signals.

[0030] The audio collection device 10 can be as follows Figure 2 As shown, Figure 2 2 is a diagram illustrating an audio collection device according to an embodiment of the present disclosure. Figure 2 As shown, the audio acquisition device 10 may at least include:

[0031] The housing unit 101 , as an external physical structure of the audio collection device 10 , may be connected to one or more internal components, thereby protecting the internal components from external damage and facilitating sound resonance.

[0032] The playback unit 102 is used to convert the electrical signal into sound, and generate air vibration through the diaphragm in the vibration driving module, thereby producing sound.

[0033] Directional microphone unit 103: The audio collection device 10 may include one or more directional microphone units 103, which are used to collect audio signals corresponding to the direction of the directional microphone units, for example Figure 2 There are four directional microphone units 103, and the audio signals collected by them point in different directions;

[0034] Furthermore, in order to collect audio signals from all directions to the maximum extent, the directional microphone units 103 can be evenly distributed in the housing unit 101 .

[0035] Filter unit 104 ( Figure 2 (not shown) for eliminating the audio signals collected by each of the multiple directional microphone units 103 that are not in the direction in which they are pointed.

[0036] Specifically, each of the multiple directional microphone units 103 is connected to the same filter unit 104 as each of the directional microphone units 103 in the non-directional direction; and based on the different directional directions of the audio signal, the first filter unit 104 connected to the first directional microphone unit 103 and the second directional microphone unit 103 is different from the second filter unit 104 connected to the second directional microphone unit 103 and the first directional microphone unit 103 (specifically, will be described in Figure 6 Detailed description in ).

[0037] In one embodiment of the present disclosure, the audio collection device 10 may be a disc-shaped audio collection device (e.g., a Bluetooth speaker), wherein the housing unit 101 is disc-shaped; the playback unit 102 is located in the center of the audio collection device 10; and the directional microphone units 103 are evenly distributed around the playback unit 102 along the diameter direction of the housing unit 101, that is, the directional directions of the directional microphone units 103 are evenly spaced around the center of the circle along the diameter direction.

[0038] Furthermore, in order to enhance the sound reception effect, a sound reception direction may be marked on the housing unit 101; the audio collection device 10 may be connected to one or more terminal devices via Bluetooth.

[0039] It should be noted that the terminal device can be a device with Bluetooth function, such as a mobile phone, a computer, etc., and there is no specific limitation; the connection method between the audio acquisition device 10 and the terminal device is also not limited.

[0040] Figure 3 is a schematic diagram further illustrating the structure of the directional microphone unit according to an embodiment of the present disclosure. Figure 3 As shown, the directional microphone unit 103 of the embodiment of the present disclosure may include at least:

[0041] The slot subunit 1031 is used to connect to the housing unit 101 of the audio collection device 10;

[0042] The sealing subunit 1032 is used to seal both ends of the tank subunit 1031;

[0043] The directional microphone subunit 1033 is located inside the slot subunit and is used to collect audio signals.

[0044] In one embodiment of the present disclosure, the shape of the slot body subunit 1031 may be a long strip, and the distances between the directional microphone subunit 1033 and the sealing subunits 1032 at both ends in the slot body subunit 1031 may be different.

[0045] In an exemplary embodiment of the present disclosure, the directional microphone subunit 1033 may be a directional electret capacitor microphone (ECM). Preferably, the distance between the directional microphone subunit 1033 and one sealing subunit 1032 in the slot subunit 1031 may be 0.5 cm, and the distance between the directional microphone subunit 1033 and another sealing subunit 1032 may be 1.5 cm. Figure 3 The directional microphone pickup direction shown in the figure can be understood as the front channel length in the slot subunit 1031 is 1.5 cm, and the back channel length is 0.5 cm. The above structure can enhance the suppression effect of the directional microphone subunit 1033 on the back audio signal (about 7dB). Figure 4-Figure 6 How the audio collection device 10 performs audio collection is specifically introduced.

[0046] Specifically, the audio acquisition device 10 may be provided with a calibration mode and a normal mode. Among them, the calibration mode refers to the stage before the audio acquisition device 10 is put into normal use (i.e., normal mode), in which the current environment can be calibrated by testing audio signals, and the coefficients of multiple filter units can be calibrated to achieve clear directionality to near-end sound signals while suppressing far-end sound signals, thereby enhancing the acquisition effect and improving the quality of the meeting; the normal mode refers to the stage in which the audio acquisition device 10 is put into normal use after completing the above calibration (i.e., calibrating the coefficients of multiple filter units). The specific calibration method will refer to Figure 4 Give a description.

[0047] Figure 4 FIG. 1 is a flowchart illustrating an audio acquisition method according to an embodiment of the present disclosure. Figure 4 As shown, the audio acquisition method of the embodiment of the present disclosure may at least include the following steps.

[0048] In step S401, the first directional microphone unit and the second directional microphone unit respectively receive an audio signal in a first directional direction, and obtain a first audio signal in the first directional direction and a second audio signal in the first directional direction. Figure 4 Described is the specific method of the calibration mode.

[0049] In one embodiment of the present disclosure, the audio collection device 10 can be placed in a calibration mode. Further, it can be placed in the center of a support (such as a desk) on which the audio collection device 10 is placed, with the directional directions of its multiple directional microphone units being the sound collection directions, and an external test audio source being placed along the sound collection direction.

[0050] The first directional microphone unit and the second directional microphone unit can be microphones pointing in any direction, and the present disclosure does not make any specific limitations. For the convenience of description, they may be referred to as directional microphone 1 (or Mic1) and directional microphone 2 (or Mic2) below. It can be understood that Mic1 and Mic2 have different pointing directions.

[0051] The audio signal in the first pointing direction can be understood as test audio 1, and its pointing direction is the same or similar to that of Mic1. The audio signal of test audio 1 received by Mic1 is the first audio signal in the first pointing direction (denoted as X1), and the audio signal of test audio 1 received by Mic2 is the second audio signal in the first pointing direction (denoted as X2).

[0052] Specifically, the signal structure of the test audio 1 can be as follows: Figure 5 shown.

[0053] Figure 5 Schematic diagram of a test audio signal structure according to an embodiment of the present disclosure. Figure 5 As shown, the test audio signal may include leading data and valid data, wherein the leading data refers to the beginning part of the test audio signal, which is used to help the receiving end to synchronize and detect, and usually includes a known sequence or pattern, which can be used for time synchronization, frequency synchronization or signal decoding; the valid data refers to the data part of the test audio signal that actually carries information, that is, the signal content after the leading data, and the valid data is the main content transmitted in the communication system.

[0054] In one embodiment of the present disclosure, the leading data may include 10 64-point pseudo-random noise (PN) data; the valid data may be 4096 points of valid data. The PN data is a sequence that appears to be random but is actually generated by a certain algorithm, and can be used for signal extension, coding, and interference suppression.

[0055] In step S402, based on the first audio signal in the first pointing direction and the second audio signal in the first pointing direction, the coefficients of the first filter unit connected to the first directional microphone unit and the second directional microphone unit are calibrated to eliminate the interference of the second audio signal in the first pointing direction on the first audio signal in the first pointing direction.

[0056] As described above, based on X1 and X2 received in step S401, the first filter unit (referred to as filter 21) connected to Mic1 and Mic2 is calibrated for coefficients, wherein filter 21 is used to eliminate the interference of X2 on X1. Specifically, the method of coefficient calibration can be found in Figure 5 Further description is given.

[0057] Figure 6 is a method flow chart further illustrating a coefficient calibration method according to an embodiment of the present disclosure. Figure 6 As shown, the coefficient calibration method of the embodiment of the present disclosure may at least include the following steps.

[0058] In step S601, the coefficients of the first filter unit are initialized. As described above, the coefficients of the filter 21 are initialized. Initialization is usually performed with random values ​​or initial default values.

[0059] In step S602, the first audio signal in the first directional direction and the second audio signal in the first directional direction are frequency-domain converted to generate a first signal frequency-domain feature and a second signal frequency-domain feature. As described above, the X1 and X2 received in step S401 are frequency-domain converted to obtain their respective frequency-domain features, where the first signal frequency-domain feature is recorded as F1 and the second signal frequency-domain feature is recorded as F2.

[0060] In one embodiment of the present disclosure, frequency domain conversion can be performed based on Fourier Transform (FT), Discrete Fourier Transform (DFT), Fast Fourier Transform (FFT), Wavelet Transform, etc. Time domain signal and frequency domain signal are two commonly used concepts in signal processing, which describe the characteristics of signals in different domains (time domain and frequency domain). The purpose of converting X1 into F1 and converting X2 into F2 in this step is that frequency domain signals are more convenient for spectrum analysis and related processing than time domain signals.

[0061] In step S603, based on the first audio signal in the first directional direction and the second audio signal in the first directional direction, a new coefficient of the first filter unit is determined so that the product of the frequency domain characteristic of the second signal and the new coefficient is opposite in phase to the frequency domain characteristic of the first signal. As described above, based on F1 and F2 obtained in step S602, the coefficient of the filter 21 is calibrated so that the calibrated coefficient (i.e., the new coefficient) H×F2 is opposite in phase to F1.

[0062] Among them, the phase represents the offset or position of the signal relative to a reference point in time or frequency. In signal processing, phase opposition means that the phase difference between the two signals is 180 degrees (or π radians). When the phases of two signals are opposite, their frequency components in the frequency domain are exactly the same, but the phase difference is 180 degrees. It can be understood that in this step, H×F2 and F1 are opposite in phase so that they can cancel each other out by addition or subtraction in the subsequent fusion process, thereby achieving the effect of signal interference suppression or noise elimination.

[0063] In one embodiment of the present disclosure, the calculation method of the new coefficient H may be, but is not limited to, the following steps:

[0064] Calculate the amplitude spectra A1 and A2, and the phase spectra P1 and P2 of X1 and X2. The amplitude spectrum represents the amplitude of the signal at different frequencies, and the phase spectrum represents the phase of the signal at different frequencies.

[0065] Since the phases of X2*H and X1 are opposite, there should be a 180-degree difference between the phase spectrum P2 and the phase spectrum P1. Therefore, this can be achieved by calculating the phase rotation angle θ of P2-P1+θ=180 degrees.

[0066] According to the phase rotation angle θ, a frequency response H (ie, new coefficient H) of the filter 21 can be constructed in the frequency domain so that its corresponding phase spectrum is θ.

[0067] It should be noted that when calculating the phase rotation angle θ, a variety of methods may be considered, such as linear interpolation, least squares method, etc., to obtain the optimal filter coefficient, which is not specifically limited in the embodiments of the present disclosure.

[0068] In step S604, the new coefficient is converted into the time domain to determine the target coefficient of the first filter unit, which is used to eliminate the interference of the second audio signal in the first directional direction to the first audio signal in the first directional direction. In order to facilitate the use of the subsequent normal mode, this step converts the frequency response H of the filter 21 into the filter coefficient (i.e., the target coefficient) h (or the target impulse response h) in the time domain.

[0069] In one embodiment of the present disclosure, time domain conversion can be performed based on inverse Fourier transform (IFT), inverse discrete Fourier transform (IDFT), time-domain filtering, convolution, wavelet transform, etc.

[0070] The audio acquisition method of the embodiment of the present disclosure is described above by taking the filter 21 as an example. The audio acquisition device 10 may include multiple filter units, each of which needs to complete the above audio acquisition. Figure 7 Further description is given.

[0071] Figure 7 is a schematic diagram further illustrating a coefficient calibration method according to an embodiment of the present disclosure. Figure 7 As shown, the audio collection device 10 includes four directional microphone units (Mic1 to Mic4). Accordingly, according to the connection method described above, the audio collection device 10 includes 12 filter units (filter 12 to filter 43).

[0072] It should be noted that, as described above, filter 21 is a filter unit connected to Mic1 and Mic2, and its sound receiving direction is the same as or similar to the direction of Mic1 (i.e., the first direction, i.e., the direction of test audio 1), and filter 21 is used to eliminate the interference of X2 to X1;

[0073] By analogy, filter 31 is a filter unit connected to Mic1 and Mic3, and filter 31 is used to eliminate the interference of X3 on X1, wherein X3 is the third audio signal in the first pointing direction;

[0074] The filter 41 is a filter unit connected to Mic1 and Mic4. The filter 41 is used to eliminate the interference of X4 on X1, wherein X4 is the fourth audio signal in the first directional direction.

[0075] Correspondingly, filter 12 is also a filter unit connected to Mic1 and Mic2. The difference is that its sound reception direction is the same as or similar to the direction of Mic2 (i.e., the second direction, i.e., the direction of the test audio 2). Filter 12 is used to eliminate the interference of X1' to X2', where X1' is the first audio signal in the second direction, and X2' is the second audio signal in the second direction.

[0076] By analogy, filter 13 is also a filter unit connected to Mic1 and Mic3. Filter 13 is used to eliminate the interference of X1" to X3", wherein X1" is the first audio signal in the third directional direction, and X3" is the third audio signal in the third directional direction.

[0077] The filter 14 is also a filter unit connected to Mic1 and Mic4. The filter 14 is used to eliminate the interference of X1''' to X4''', wherein X1''' is the first audio signal in the fourth directional direction, and X4''' is the fourth audio signal in the fourth directional direction.

[0078] The same can be said for other filters, which will not be elaborated here.

[0079] For each of the above-mentioned multiple directional microphone units, the filter unit connected to each directional microphone unit in the non-directional direction is calibrated to generate a target coefficient for each filter unit to eliminate the interference of the audio signal collected by each directional microphone unit in the non-directional direction on the audio signal collected by the directional microphone unit in the directional direction. After the target coefficients of all filter units are determined, they can be written into the audio collection device 10 for subsequent use.

[0080] So far, the calibration mode has been introduced. After the calibration is completed, the audio acquisition device 10 can be put into normal use (ie, enter the normal mode).

[0081] Specifically, based on the target coefficient of each filter unit, the audio signal in each directional direction is denoised to generate a denoised audio signal; then based on the preset path of the directional direction of each directional microphone unit in the multiple directional microphone units, each denoised audio signal is weighted and summed to generate a fused audio signal.

[0082] Specifically, a weight can be set for each directional microphone in each preset path, and the weight can be used to describe the contribution of the audio signal received by the directional microphone to the fused audio signal. The weight can be preset and calculated based on factors such as the physical position of the microphone, the direction of the sound source, and the sound wave propagation.

[0083] Then, the denoised sound signal is weighted and summed according to the weight of each directional microphone in each preset path. That is, at each time point, the sound signal on each directional microphone and its path weight need to be multiplied separately, and then accumulated to obtain the fused sound signal.

[0084] Finally, the purpose of denoising and fusing the sound signals picked up by multiple directional microphones can be achieved by outputting the fused sound signal.

[0085] Among them, the preset path can be understood as Figure 7 Specifically, taking a as an example, path a is a first directional path, the sound receiving direction on the path is the first directional direction, and the path includes filter 12, filter 13, and filter 14. The functions of these three filters are to eliminate the interference of X2, X3, and X4 received by Mic2, Mic3, and Mic4 on X1 received by Mic1, so as to suppress the influence of X2, X3, and X4 on X1, so that the quality of the fused audio signal after weighted summation is higher.

[0086] Similarly, Figure 7 Path b in the figure is a second directional path, and the sound receiving direction on the path is the second directional direction. The path includes filters 21, 23, and 24. The functions of these three filters are to eliminate the interference of X1', X3', and X4' received by Mic1, Mic3, and Mic4 on X2' received by Mic2, so as to suppress the influence of X1', X3', and X4' on X2', so that the quality of the fused audio signal after weighted summation is higher.

[0087] The same can be said for other preset paths, which will not be described here.

[0088] It should be noted that the above four paths a, b, c, and d are only exemplary descriptions, and do not impose specific restrictions on the number of directional microphone units, the number of filter units, and the number of preset paths in the embodiments of the present disclosure.

[0089] Figure 8 1 is a hardware block diagram of an electronic device according to an embodiment of the present disclosure. The electronic device according to an embodiment of the present disclosure includes at least a processor; and a memory for storing computer-readable instructions. When the computer-readable instructions are loaded and executed by the processor, the processor executes the audio acquisition method as described above.

[0090] Figure 8 The electronic device 800 shown specifically includes: a central processing unit (CPU) 801, a graphics processing unit (GPU) 802 and a memory 803. These units are interconnected via a bus 804. The central processing unit (CPU) 801 and / or the graphics processing unit (GPU) 802 can be used as the above-mentioned processor, and the memory 803 can be used as the above-mentioned memory for storing computer-readable instructions. In addition, the electronic device 800 may also include a communication unit 805, a storage unit 806, an output unit 807, an input unit 808 and an external device 809, which are also connected to the bus 804.

[0091] Fig. 9 is a schematic diagram illustrating a computer program product according to an embodiment of the present disclosure. Fig. 9As shown, a computer program product 900 according to an embodiment of the present disclosure has a computer program 901 stored thereon. When the computer program 901 is executed by a processor, the audio acquisition method described with reference to the above figures is executed. The computer program product includes, but is not limited to, for example, a volatile memory and / or a non-volatile memory. The volatile memory may, for example, include a random access memory (RAM) and / or a cache memory (cache), etc. The non-volatile memory may, for example, include a read-only memory (ROM), a hard disk, a flash memory, an optical disk, a magnetic disk, etc.

[0092] The audio acquisition method, device, electronic device and computer program product according to the embodiments of the present disclosure are described above with reference to the accompanying drawings. According to the audio acquisition method of the embodiments of the present disclosure, the present disclosure calibrates multiple filter unit coefficients to suppress far-end audio signals while extracting audio signals in a specific direction, thereby enhancing the acquisition effect and improving the quality of the meeting.

[0093] Those of ordinary skill in the art will appreciate that the units and algorithm steps of each example described in conjunction with the embodiments disclosed herein can be implemented in electronic hardware, or a combination of computer software and electronic hardware. Whether these functions are performed in hardware or software depends on the specific application and design constraints of the technical solution. Professional and technical personnel can use different methods to implement the described functions for each specific application, but such implementation should not be considered to be beyond the scope of this disclosure.

[0094] The basic principles of the present disclosure are described above in conjunction with specific embodiments. However, it should be noted that the advantages, strengths, effects, etc. mentioned in the present disclosure are only examples and not limitations, and it cannot be considered that these advantages, strengths, effects, etc. are required by each embodiment of the present disclosure. In addition, the specific details disclosed above are only for the purpose of illustration and ease of understanding, and are not limitations. The above details do not limit the present disclosure to the necessity of adopting the above specific details to be implemented.

[0095] The block diagrams of the devices, apparatuses, equipment, and systems involved in this disclosure are only illustrative examples and are not intended to require or imply that they must be connected, arranged, and configured in the manner shown in the block diagrams. As will be appreciated by those skilled in the art, these devices, apparatuses, equipment, and systems can be connected, arranged, and configured in any manner. Words such as "including," "comprising," "having," and the like are open words, referring to "including but not limited to," and can be used interchangeably therewith. The words "or" and "and" used herein refer to the words "and / or," and can be used interchangeably therewith, unless the context clearly indicates otherwise. The word "such as" used herein refers to the phrase "such as but not limited to," and can be used interchangeably therewith.

[0096] Additionally, as used herein, "or" used in a list of items beginning with "at least one" indicates a separate list, so that, for example, a list of "at least one of A, B, or C" means A or B or C, or AB or AC or BC, or ABC (i.e., A and B and C). Furthermore, the word "exemplary" does not mean that the example described is preferred or better than other examples.

[0097] It should also be noted that in the system and method of the present disclosure, each component or each step can be decomposed and / or recombined. Such decomposition and / or recombination should be regarded as equivalent solutions of the present disclosure.

[0098] Various changes, substitutions, and modifications of the techniques described herein may be made without departing from the teachings defined by the appended claims. Furthermore, the scope of the claims of the present disclosure is not limited to the specific aspects of the processes, machines, manufactures, compositions of events, means, methods, and actions described above. Currently existing or later to be developed processes, machines, manufactures, compositions of events, means, methods, or actions that perform substantially the same functions or achieve substantially the same results as the corresponding aspects described herein may be utilized. Thus, the appended claims include such processes, machines, manufactures, compositions of events, means, methods, or actions within their scope.

[0099] The above description of the disclosed aspects is provided to enable any person skilled in the art to make or use the present disclosure. Various modifications to these aspects will be readily apparent to those skilled in the art, and the general principles defined herein may be applied to other aspects without departing from the scope of the present disclosure. Therefore, the present disclosure is not intended to be limited to the aspects shown herein, but rather to the widest scope consistent with the principles and novel features disclosed herein.

[0100] The above description has been given for the purpose of illustration and description. In addition, this description is not intended to limit the embodiments of the present disclosure to the forms disclosed herein. Although multiple example aspects and embodiments have been discussed above, those skilled in the art will recognize certain variations, modifications, changes, additions and sub-combinations thereof.

Claims

1. An audio acquisition device, It is characterized in that The device comprises: A plurality of directional microphone units, used for respectively collecting audio signals corresponding to their directional directions; A plurality of filter units, used for eliminating the audio signals in non-directional directions collected by each of the plurality of directional microphone units; Among them, the multiple directional microphone units include a first directional microphone unit corresponding to a first directional direction and a second directional microphone unit corresponding to a second directional direction, and one of the multiple filter units is configured to eliminate the audio signal in the first directional direction collected by the second directional microphone unit.

2. The audio acquisition device according to claim 1, It is characterized in that Also includes: Each directional microphone unit among the plurality of directional microphone units and each directional microphone unit in a non-directional direction are connected to the same filter unit; as well as, Based on the different directional directions of the audio signal, the first filter unit connected to the first directional microphone unit and the second directional microphone unit is different from the second filter unit connected to the first directional microphone unit.

3. The audio acquisition device according to claim 1, It is characterized in that The directional microphone unit comprises: A slot subunit, used to connect to the housing unit of the audio collection device; A sealing subunit, used for sealing both ends of the tank subunit; The directional microphone subunit is located inside the slot subunit and is used to collect the audio signal.

4. The audio acquisition device according to claim 3, It is characterized in that Also includes: The tank subunit is in the shape of a long strip; as well as, The distances between the directional microphone subunit and the sealing subunits at both ends are different.

5. An audio acquisition method, It is characterized in that The method comprises: The first directional microphone unit and the second directional microphone unit respectively receive an audio signal in a first directional direction, and obtain a first audio signal in the first directional direction and a second audio signal in the first directional direction; Based on the first audio signal in the first directional direction and the second audio signal in the first directional direction, coefficients of the first filter unit connected to the first directional microphone unit and the second directional microphone unit are calibrated to eliminate interference of the second audio signal in the first directional direction with the first audio signal in the first directional direction.

6. The audio acquisition method according to claim 5, It is characterized in that The method of calibrating coefficients of a first filter unit connected to the first directional microphone unit and the second directional microphone unit based on the first audio signal in the first directional direction and the second audio signal in the first directional direction to eliminate interference of the second audio signal in the first directional direction on the first audio signal in the first directional direction includes: Initializing the coefficients of the first filter unit; Performing frequency domain conversion on a first audio signal in the first directional direction and a second audio signal in the first directional direction to generate a first signal frequency domain feature and a second signal frequency domain feature; Determine new coefficients of the first filter unit based on a first audio signal in the first pointing direction and a second audio signal in the first pointing direction, so that a product of a frequency domain feature of the second signal and the new coefficients has an opposite phase to the frequency domain feature of the first signal; The new coefficients are transformed into a time domain to determine target coefficients of the first filter unit, where the target coefficients of the first filter unit are used to eliminate interference of the second audio signal in the first directional direction with the first audio signal in the first directional direction.

7. The audio acquisition method according to claim 5 or 6, It is characterized in that Also includes: The coefficient calibration is performed on each directional microphone unit among the multiple directional microphone units and the filter unit connected to each directional microphone unit in its non-directional direction, and the target coefficient of each filter unit is generated to eliminate the interference of the audio signal collected by each directional microphone unit in the non-directional direction on the audio signal collected by the directional microphone unit in the directional direction.

8. The audio acquisition method according to claim 7, It is characterized in that Also includes: De-noising the audio signal in each directional direction based on the target coefficient of each filter unit to generate a de-noised audio signal; Based on the preset path of the directional direction of each directional microphone unit in the plurality of directional microphone units, weighted summation is performed on each of the denoised audio signals to generate a fused audio signal.

9. An electronic device, It is characterized in that include: a memory for storing computer readable instructions; as well as A processor is used to run the computer-readable instructions so that the electronic device performs the audio acquisition method as described in any one of claims 5 to 8.

10. A computer program product comprising a computer program, It is characterized in that When the computer program is executed by a processor, the audio collection method according to any one of claims 5 to 8 is implemented.

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