Audio signal adjustment method and audio signal adjustment system

Abstract

The application provides an audio signal adjusting method and system, which is applied to a microphone device and comprises the following steps: obtaining a data adjusting instruction and an audio data stream converted from an analog audio signal generated by a target sound source, wherein the data adjusting instruction carries current sensitivity level information, the current sensitivity level information is related to a current distance between the microphone device and the target sound source; then, adjusting the processed audio data stream according to the current sensitivity level information to obtain a target audio signal; finally, outputting the target audio signal. In the application, the processed audio data stream is adjusted according to the current sensitivity level information related to the current distance between the microphone device and the target sound source to obtain the target audio signal, so that the received environmental noise signal can be filtered out to the maximum extent while the analog audio signal generated by the target sound source is not lost.

Description

Technical Field

[0001] This invention relates generally to the field of communication technology, and more specifically to an audio signal adjustment method and an audio signal adjustment system. Background Technology

[0002] With the continuous development of electronic technology, many applications in life involve the processing of audio signals. Therefore, how to ensure the flexibility of audio signal processing is a problem that needs to be solved. Summary of the Invention

[0003] To address the above-mentioned or other problems, the present invention provides the following technical solutions.

[0004] In a first aspect, the present invention provides an audio signal adjustment method applied to a microphone device, the audio signal adjustment method comprising:

[0005] The system acquires data adjustment instructions and an audio data stream converted from an analog audio signal generated by a target sound source, wherein the data adjustment instructions carry current sensitivity level information, and the current sensitivity level information is related to the current distance between the microphone device and the target sound source;

[0006] The processed audio data stream is adjusted using the current sensitivity level information to obtain the target audio signal;

[0007] Output the target audio signal.

[0008] According to an embodiment of the audio signal adjustment method of the present invention, the step of adjusting the processed audio data stream with the current sensitivity level information to obtain a target audio signal specifically includes:

[0009] Configure the current data right shift bit number based on the current sensitivity level information;

[0010] The audio data to be adjusted in the processed audio data stream is adjusted by shifting the current data right by a certain number of bits to obtain the target audio signal.

[0011] According to an embodiment of the audio signal adjustment method of the present invention, the current distance is negatively correlated with the current sensitivity level information and the number of bits shifted right from the current data.

[0012] According to an embodiment of the audio signal adjustment method of the present invention, after the step of outputting the target audio signal, the method further includes:

[0013] Calculate the signal threshold range corresponding to the number of bits shifted right of the current data, and monitor the maximum and minimum values ​​of the target audio signal;

[0014] Obtain an update instruction generated based on the signal threshold range, the maximum value, and the minimum value, wherein the update instruction carries sensitivity feedback information;

[0015] The current sensitivity level information is updated based on the sensitivity feedback information.

[0016] According to an embodiment of the audio signal adjustment method of the present invention, the signal threshold range has a maximum value and a minimum value, wherein:

[0017] If the minimum value is less than the minimum value, the sensitivity feedback information indicates that the current sensitivity level information needs to be reduced to improve sensitivity;

[0018] If the maximum value is greater than the maximum value, then the sensitivity feedback information indicates that the current sensitivity level information needs to be increased to reduce the sensitivity; and,

[0019] If the minimum value is greater than the minimum value and the maximum value is less than the maximum value, then the sensitivity feedback information indicates that there is no need to change the current sensitivity level information.

[0020] An audio signal adjustment method according to an embodiment of the present invention further includes, before the step of outputting the target audio signal:

[0021] Obtain a delay instruction, wherein the delay instruction carries a beamforming time, the beamforming time being determined by the difference between the current distance and other distances between other microphone devices and the target sound source, wherein the microphone devices and the other microphone devices constitute a microphone array;

[0022] The pulse frequency hopping time is determined based on the beamforming time.

[0023] The audio data stream is frequency-hopped according to the pulse hopping time so that the target audio signal is phase-aligned with other audio signals output by the other microphone devices.

[0024] An audio signal adjustment method according to an embodiment of the present invention further includes, before the step of adjusting the processed audio data stream with the current sensitivity level information:

[0025] The audio data stream is digitally filtered to generate the processed audio data stream.

[0026] According to an embodiment of the audio signal adjustment method of the present invention, the current sensitivity level information is calculated in real time based on the current distance between the microphone device and the target sound source.

[0027] In a second aspect, the present invention provides an audio signal adjustment system, comprising:

[0028] The main rangefinder is positioned on the target sound source; and,

[0029] A microphone array, comprising multiple microphone devices, wherein the microphone devices include:

[0030] A microphone sensor is used to convert the analog audio signal generated by the target sound source into an audio data stream;

[0031] The current distance between the microphone device and the target sound source is measured from the rangefinder, which, together with the main rangefinder, determines this distance.

[0032] A processor, coupled to the microphone sensor and the rangefinder, is used to perform the audio signal adjustment method described in any of the preceding claims.

[0033] According to an embodiment of the audio signal adjustment system of the present invention, the microphone device further includes a sensitivity modulator for generating the data adjustment command, wherein the sensitivity modulator calculates the current sensitivity level information in real time based on the current distance.

[0034] According to an embodiment of the audio signal adjustment system of the present invention, the sensitivity modulator stores a mapping relationship between multiple distances and multiple sensitivity levels.

[0035] According to an embodiment of the present invention, an audio signal adjustment system is provided, wherein the master rangefinder and the slave rangefinder determine the current distance based on the received signal strength.

[0036] According to an embodiment of the audio signal adjustment system of the present invention, the processor further includes a filtering unit configured to perform digital filtering processing on the audio data stream to generate the processed audio data stream.

[0037] The beneficial effects of this invention are as follows: This invention provides an audio signal adjustment method and an audio signal adjustment system. The audio signal adjustment method is applied to a microphone device and includes: acquiring a data adjustment instruction and an audio data stream converted from an analog audio signal generated by a target sound source. The data adjustment instruction carries current sensitivity level information, which is related to the current distance between the microphone device and the target sound source. Then, the processed audio data stream is adjusted using the current sensitivity level information to obtain a target audio signal. Finally, the target audio signal is output. Because the audio signal adjustment method provided by this invention adjusts the processed audio data stream using current sensitivity level information related to the current distance between the microphone device and the target sound source to obtain the target audio signal, it ensures that the analog audio signal generated by the target sound source is not lost, while also filtering out received environmental noise signals to the greatest extent possible. Attached Figure Description

[0038] To more clearly illustrate the technical solutions of the present invention, the drawings used in the description of the various embodiments made according to the present invention will be briefly introduced below. Obviously, the drawings described below are only some embodiments of the present invention. For those skilled in the art, other drawings can be obtained from these drawings without creative effort.

[0039] Figure 1 This is a schematic flowchart of an audio signal adjustment method provided by an embodiment of the present invention.

[0040] Figure 2 This is a further flowchart illustrating the audio signal adjustment method provided by an embodiment of the present invention.

[0041] Figure 3 This is a schematic diagram of the structure of an audio signal adjustment system provided in an embodiment of the present invention.

[0042] Figure 4 This is a schematic diagram of the structure of a microphone device in an audio signal adjustment system provided according to an embodiment of the present invention. Detailed Implementation

[0043] The technical solutions of the embodiments of the present invention will be clearly and completely described below with reference to the accompanying drawings. Obviously, the described embodiments are only a part of the embodiments of the present invention, and not all of them. All other embodiments obtained by those skilled in the art based on the embodiments of the present invention without creative effort are within the scope of protection of the present invention.

[0044] In the description of this invention, it should be understood that the terms "first" and "second" are used for descriptive purposes only and should not be construed as indicating or implying relative importance or implicitly specifying the number of indicated technical features. Therefore, features defined as "first" and "second" may explicitly or implicitly include one or more of the stated features. In the description of this invention, "a plurality of" means two or more, unless otherwise explicitly specified.

[0045] In the description of this invention, it should be noted that, unless otherwise explicitly specified and limited, the terms "installation," "connection," and "linking" should be interpreted broadly. For example, they can refer to a fixed connection, a detachable connection, or an integral connection; they can refer to a mechanical connection, an electrical connection, or a connection that allows for communication; they can refer to a direct connection or an indirect connection through an intermediate medium; they can refer to the internal communication of two components or the interaction between two components. Those skilled in the art can understand the specific meaning of the above terms in this invention according to the specific circumstances.

[0046] The following disclosure provides many different embodiments or examples to implement different structures of the invention. To simplify the disclosure, specific examples of components and arrangements are described below. Of course, these are merely examples and are not intended to limit the invention. Furthermore, reference numerals and / or letters may be repeated in different examples; such repetition is for simplification and clarity and does not in itself indicate a relationship between the various embodiments and / or arrangements discussed. In addition, examples of various specific processes and materials are provided in this invention, but those skilled in the art will recognize the application of other processes and / or the use of other materials.

[0047] Please see Figure 1 and Figure 4 , Figure 1 and Figure 4 The following diagrams illustrate a flowchart of an audio signal adjustment method provided by an embodiment of the present invention and a structural schematic diagram of a microphone device 100 in an audio signal adjustment system 200. Figure 1 and Figure 4 As shown, the audio signal adjustment method is applied to the microphone device 100 described above, and specifically includes the following steps:

[0048] First acquisition step S101: Acquire data adjustment instructions and audio data stream converted from analog audio signals generated by the target sound source, wherein the data adjustment instructions carry current sensitivity level information, and the current sensitivity level information is related to the current distance between the microphone device 100 and the target sound source;

[0049] Adjustment step S102: Adjust the processed audio data stream according to the current sensitivity level information to obtain the target audio signal;

[0050] Output step S103: Output the target audio signal.

[0051] It should be noted that a microphone device is an electronic device that can receive, process, and output audio signals. However, some microphone device processors cannot adjust the amplitude of their adjustment of the received audio signal according to the distance between them and the target sound source. Furthermore, in some microphone arrays composed of multiple microphone devices, although the processors of each microphone device can perform simple beamforming processing on the received audio signal, there are problems in these embodiments, such as the need to manually set various parameters of the processor, and the inability of the processor to adjust these parameters accordingly when the position of the target sound source changes. This results in a lack of flexibility in the process of processing audio signals using the aforementioned microphone devices.

[0052] For further information, please refer to [link / reference]. Figure 1 and Figure 4 In an embodiment of the present invention, the microphone device 100 includes a processor 130 configured to perform the first acquisition step S101 to the output step S103 described above, and different sensitivity level information represents different degrees of adjustment required to the processed audio data stream. That is, the processor 130 in the present invention can control the degree of adjustment of the processed audio data stream in real time based on its current distance from the target sound source, thus ensuring flexibility when processing the analog audio signal generated by the target sound source using the microphone device 100 with the applied audio signal adjustment method.

[0053] It should be noted that during the process of the microphone device 100 receiving the analog audio signal generated by the target sound source, it will also receive some environmental noise signals with small signal amplitudes, and convert these environmental noise signals together with the analog audio signal into an audio data stream. In this embodiment, the current sensitivity level information related to the current distance can indicate the maximum amplitude range of environmental noise signals that the processor 130 can filter out at the current distance without losing the analog audio signal generated by the target sound source. In other words, when the processor 130 adjusts the processed audio data stream with the above-mentioned current sensitivity level information, it can ensure that the received environmental noise signals are filtered out to the greatest extent possible without affecting the analog audio signal generated by the target sound source.

[0054] Further, please refer to Figure 2The illustrated diagram shows a further flow chart of the audio signal adjustment method provided by an embodiment of the present invention, as shown below. Figure 2 As shown, in this embodiment, the above adjustment step S102 specifically includes:

[0055] Configuration sub-step S1021: Configure the number of bits to right shift the current data according to the current sensitivity level information;

[0056] Adjustment sub-step S1022: Adjust the audio data to be adjusted in the processed audio data stream by shifting the current data right by a certain number of bits to obtain the target audio signal.

[0057] It should be noted that the data to be adjusted has high-order bits and low-order bits, and the signals corresponding to the low-order bits have smaller amplitudes compared to the high-order bits. Furthermore, in this embodiment, the number of right shifts of the current data indicates the number of low-order bits in the audio data to be adjusted that need to be discarded at the current distance, and the discarding of low-order bits starts from the least significant bit. Therefore, the process by which the processor 130 adjusts the audio data to be adjusted in the processed audio data stream by the number of right shifts of the current data is actually filtering out the ambient noise signal received by the microphone device 100.

[0058] For further information, please refer to [link / reference]. Figure 4 In this embodiment, the processor 130 further includes a data unit 134 configured to execute the aforementioned adjustment sub-step S1022, and the data unit 134 is coupled to the output port of the processor 130. Specifically, during the execution of the aforementioned configuration sub-step S1021 by the processor 130, the value of the corresponding register in the processor 130 is set according to the aforementioned current sensitivity level information. This value indicates the number of low-order data bits that the data unit 134 needs to discard in the data to be adjusted. Then, during the execution of the aforementioned adjustment sub-step S1022 by the data unit 134, it only needs to obtain the value of the aforementioned register and perform the corresponding adjustment operation according to the value.

[0059] The processor 130 can be any module capable of digitally filtering the data stream. For example, the processor 130 can be a microcontroller, microprocessor, or digital signal processing (DSP) module with digital filtering capabilities. This invention is not limited thereto.

[0060] Specifically, in this embodiment, the current distance is negatively correlated with the current sensitivity level information and the number of bits shifted right from the current data. For example, when the microphone device 100 is far from the target sound source, in order to ensure that the processor 130 does not lose signals during the adjustment of the processed audio data stream, the processor 130 needs to ensure the integrity of the processed audio data stream to a certain extent. Therefore, the maximum amplitude range of the environmental noise signal that it can filter out is smaller, that is, the number of low-order data bits that the processor 130's data unit 134 needs to discard in the data to be adjusted is smaller.

[0061] Specifically, the number of bits of the audio data to be adjusted in the processed audio data stream input to the data unit 134 can be up to 32 bits, and the number of bits of the data in the target audio signal output from the data unit 134 can be up to 24 bits.

[0062] Please continue reading. Figure 4 In this embodiment, the microphone device 100 further includes a sensitivity modulator 140, which is coupled to the data unit 134 of the processor 130. Specifically, the sensitivity modulator 140 is a coprocessor independent of the processor 130 and capable of communicating with the processor 130. It is configured to calculate the current sensitivity level information in real time based on the current distance between the microphone device 100 and the target sound source, and is configured to generate a data adjustment instruction carrying the aforementioned current sensitivity level information. The data adjustment instruction can be acquired by the processor 130 in the first acquisition step S101.

[0063] Furthermore, in some other embodiments of the present invention, certain modules disposed in the processor 130 may also calculate the current sensitivity level information in real time based on the current distance between the microphone device and the target sound source, and generate data adjustment instructions carrying the current sensitivity level information. That is to say, in these embodiments, it is not necessary to add a coprocessor independent of the processor 130 in the microphone device to implement the sensitivity modulator 140.

[0064] Furthermore, in a specific embodiment according to the present invention, the sensitivity modulator 140 may pre-store multiple mapping relationships between distances and multiple sensitivity level information. Specifically, the mapping relationship between the multiple distances and multiple sensitivity level information can be a piecewise, step-like mathematical model, for example:

[0065]

[0066] It should be noted that in the above formula, L represents distance, and the unit of distance is M (meters), and D represents sensitivity level information (where the smaller the number of sensitivity level information D, the higher the corresponding sensitivity), that is, the number of low-order data in the data to be adjusted that data unit 134 needs to discard. Further, in this embodiment, after the sensitivity modulator 140 obtains the current distance between the microphone device and the target sound source, it can directly obtain the current sensitivity level information corresponding to the current distance based on its pre-stored mathematical model. In other words, since the sensitivity modulator 140 has pre-stored multiple mapping relationships between distances and multiple sensitivity level information, in this embodiment, the sensitivity modulator 140 does not need to calculate the current sensitivity level information in real time based on the above current distance.

[0067] Specifically, in this embodiment, the method for establishing a mathematical model of sensitivity level information varying with distance in the sensitivity modulator 140 can be as follows: First, the target sound source is positioned 0.5 meters away from the microphone device and emits sound at a preset volume. Simultaneously, the number of low-order data bits in the audio data to be adjusted that can be discarded by the data unit while ensuring that the target sound source can be fully received is calculated. Then, the distance between the target sound source and the microphone device is increased by 0.5 meters successively, and the calculation continues until the different numbers of low-order data bits that can be discarded corresponding to different distances are obtained. Furthermore, in the process of establishing the above mathematical model, multiple experiments can be conducted and the average value can be obtained to ensure the accuracy of the established mathematical model. Furthermore, in some other embodiments of the present invention, the mapping relationship between the above multiple distances and multiple sensitivity level information can also be represented by other mathematical models, and the present invention does not limit this.

[0068] Furthermore, considering that in some situations, there may be a certain error between the current distance between the microphone device 100 and the target sound source acquired by the processor 130 and the actual distance, or that due to environmental factors, the processor 130 may be unable to achieve the goal of filtering out environmental noise signals to the greatest extent possible without losing the analog audio signal generated by the target sound source when adjusting the audio data to be adjusted by shifting the current data right by a certain number of bits, therefore, after the processor 130 first acquires the current sensitivity level information in the first acquisition step S101, it can update the current sensitivity level information as needed to compensate for the error caused by ranging and the influence of environmental factors, thereby ensuring the reliability of the target audio signal output by the processor 130. For example, please refer to... Figure 2 ,like Figure 2 As shown, after the output step S103 described above, the steps executed by the processor 130 further include:

[0069] Calculation and monitoring step S104: Calculate the signal threshold range corresponding to the right shift of the current data, and monitor the maximum and minimum values ​​of the target audio signal;

[0070] The third acquisition step S105: Acquire the update instruction generated based on the signal threshold range, maximum value and minimum value, wherein the update instruction carries sensitivity feedback information;

[0071] Update step S106: Update the current sensitivity level information based on the sensitivity feedback information.

[0072] For details, please continue reading. Figure 4 In this embodiment, the processor 130 further includes a threshold monitor 135 and an extreme value monitor 136. The threshold monitor 135 is coupled to the data unit 134 and can calculate the theoretically expected signal threshold range of the target audio signal output from the output port of the processor 130 based on the configured right shift bits of the current data (in other embodiments, this signal threshold range of the threshold monitor 135 can also be implemented by a user configuration register; of course, the configured value should be smaller than the theoretically expected signal threshold range). The extreme value monitor 136 is coupled to the output port and can collect the maximum and minimum values ​​of the target audio signal in real time. Therefore, the threshold monitor 135 is configured to perform the "calculation of the signal threshold range corresponding to the right shift bits of the current data" in the above calculation and monitoring step S104, and the extreme value monitor 136 is configured to perform the "monitoring of the maximum and minimum values ​​of the target audio signal" in the above calculation and monitoring step S104.

[0073] It should be noted that the extreme value monitor 136 stores the maximum and minimum values ​​of the target audio signal it acquires in real time in corresponding registers. When the maximum value acquired by the extreme value monitor 136 at the current moment is greater than the maximum value acquired and stored in the register at the previous moment, the value in the register will be updated to the maximum value acquired at the current moment. Similarly, when the minimum value acquired by the extreme value monitor 136 at the current moment is less than the minimum value acquired and stored in the register at the previous moment, the value in the register will be updated to the minimum value acquired at the current moment.

[0074] Furthermore, in this embodiment, the sensitivity modulator 140 described above is also coupled to the threshold monitor 135 and the extreme value monitor 136, and is further configured to generate the update instruction in the third acquisition step S105 described above.

[0075] It should be noted that in this embodiment, when the maximum and minimum values ​​of the target audio signal acquired in real time by the extreme value monitor 136 exceed the signal threshold range of the target audio signal, the sensitivity feedback information will indicate that the current sensitivity level needs to be adjusted. Specifically, the signal threshold range calculated by the threshold monitor 135 has a maximum and a minimum value. When comparing the maximum and minimum values ​​with the above-mentioned maximum and minimum values, the following three situations may occur:

[0076] (1) If the minimum value of the target audio signal acquired in real time is less than the minimum value it should theoretically have, it means that during the process of the processor 130 adjusting the audio data to be adjusted in the processed audio data stream by shifting the current data right by bits, the analog audio signal generated by the target sound source may be lost. Therefore, the sensitivity feedback information will indicate that the current sensitivity level information needs to be reduced to improve the sensitivity, that is, to reduce the number of low-bit data in the audio data to be adjusted that need to be discarded.

[0077] (2) If the maximum value of the target audio signal acquired in real time is greater than its theoretical maximum value, it means that during the process of the processor 130 adjusting the audio data to be adjusted in the processed audio data stream by shifting the current data right by bits, it is possible that while ensuring that the analog audio signal generated by the target sound source is not lost, the received environmental noise signal has not been filtered out to the greatest extent. Therefore, the sensitivity feedback information will indicate that the current sensitivity level information needs to be increased to reduce the sensitivity, that is, to increase the number of low-bit data in the audio data to be adjusted that need to be discarded.

[0078] (3) If the minimum value of the target audio signal acquired in real time is greater than its theoretical minimum value and the maximum value of the target audio signal acquired in real time is less than its theoretical maximum value, it means that in the process of the processor 130 adjusting the audio data to be adjusted in the processed audio data stream by shifting the current data right by bits, it can ensure that the analog audio signal generated by the target sound source is not lost and that the received environmental noise signal is filtered out to the greatest extent. Therefore, the sensitivity feedback information will indicate that there is no need to change the current sensitivity level information, that is, there is no need to change the number of low-bit data to be discarded in the audio data to be adjusted.

[0079] Specifically, in this embodiment, the adjustment range of the current sensitivity level information can be selected based on the degree to which the maximum or minimum value exceeds the signal threshold range. For example, in one case, if the minimum value is less than the minimum value, and the difference between the minimum and the minimum value is not less than 10% of the value corresponding to the signal threshold range, then the current sensitivity level information can be reduced by one level, that is, the number of low-order data in the audio data to be adjusted is reduced by one. In another case, if the minimum value is less than the minimum value, and the difference between the minimum and the minimum value is greater than 10% of the value corresponding to the signal threshold range, then the current sensitivity level information can be reduced by N levels, that is, the number of low-order data in the audio data to be adjusted is reduced by N, where N>1.

[0080] Please continue reading. Figure 4 In a further embodiment, the microphone device 100 further includes a microphone sensor 110. In a further embodiment, the processor 130 further includes a filtering unit 132, wherein the microphone sensor 110 is coupled to the processor 130, and the filtering unit 132 is coupled between the microphone sensor 110 and the data unit 134.

[0081] Specifically, when the analog audio signal generated by the target sound source enters the microphone device 100, the microphone sensor 110 converts the analog audio signal into an audio data stream before inputting it to the processor 130. This audio data stream is a 1-bit data stream modulated using Pulse Density Modulation (PDM). Further, before the audio data stream enters the data unit 134 of the processor 130, it is further converted into a processed audio data stream by the filtering unit 132. This processed audio data stream is a data stream modulated using Pulse Code Modulation (PCM). Specifically, the steps performed by the filtering unit 132 are as follows: Figure 2 The steps shown are executed before the configuration sub-step S1021 described above:

[0082] Processing step S107: Perform digital filtering on the audio data stream to generate a processed audio data stream.

[0083] For further information, please refer to [link / reference]. Figure 4The threshold monitor 135 described above is also coupled to the filtering unit 132. Specifically, the threshold monitor 135 calculates, based on the parameters of the filtering unit 132 and the number of right shifts of the current data described above, the theoretically required signal threshold range of the target audio signal obtained by adjusting the processed audio data stream with the current sensitivity level information.

[0084] Further, please refer to Figure 3 as well as Figure 4 , Figure 3 A schematic diagram of the structure of an audio signal adjustment system 200 provided according to an embodiment of the present invention is shown. Figure 3 as well as Figure 4 As shown, in this embodiment, a master rangefinder 210 is provided on the target sound source, and a microphone device 100 has a slave rangefinder 120. Multiple microphone devices 100 constitute a microphone array 220, wherein the master rangefinder 210 and the microphone array 220 together constitute an audio signal adjustment system 200.

[0085] Specifically, in the microphone device 100, the rangefinder 120 is coupled to the processor 130 and the sensitivity modulator 140, respectively. The rangefinder 120 is configured to jointly measure the current distance between the microphone device 100 and the target sound source with the master rangefinder 210 carried by the target sound source. The sensitivity modulator 140 can communicate with the rangefinder 120 to obtain or receive the aforementioned current distance.

[0086] Furthermore, in this embodiment, the master rangefinder 210 and the slave rangefinder 120 determine the current distance based on the received signal strength (RSS). However, in some other embodiments according to the present invention, the master rangefinder and the slave rangefinder may also determine the current distance via Bluetooth or other means, and the present invention does not limit this to such methods.

[0087] It should be noted that in this embodiment, all wireless rangefinders (including master rangefinder 210 and slave rangefinder 120) can communicate with each other. Furthermore, the master rangefinder 210 will synchronize the time of all the wireless rangefinders and initiate the task of measuring the current distance between the corresponding microphone device 100 and the target sound source to all slave rangefinders 120 at the same time.

[0088] For further information, please refer to [link / reference]. Figure 3The audio signal adjustment system 200 also includes an adder 230 and a speaker 240. The adder 230 is coupled to the output port of each microphone device 100, and the speaker 240 is coupled to the output port of the adder 230. Specifically, the target audio signals output by each microphone device 100 are superimposed into one signal by the adder 230 and then output by the speaker 240.

[0089] Furthermore, considering that the current distances between each microphone device 100 and the target sound source are not exactly the same, there will be phase differences between the target audio signals output by each microphone device 100, which will lead to problems affecting the quality of the final output signal of the speaker 240. Therefore, please continue to refer to... Figure 2 ,like Figure 2 As shown, prior to the above-described processing step S107, the steps performed by the processor 130 further include:

[0090] Second acquisition step S108: Acquire delay instruction, wherein the delay instruction carries beamforming time, the beamforming time is determined by the difference between the current distance between the microphone device 100 and the target sound source and other distances between the other microphone devices 100 and the target sound source, wherein the microphone device 100 and other microphone devices 100 constitute a microphone array 220.

[0091] Determine step S109: Determine the pulse frequency hopping time based on the beamforming time;

[0092] Frequency hopping step S110: Frequency hopping processing of the audio data stream is performed according to the pulse frequency hopping time.

[0093] Please continue reading. Figure 3 and Figure 4 In this embodiment, the master rangefinder 210 can acquire the current distance measured by each slave rangefinder 120 and determine a reference value (e.g., the current distance with the smallest value). After each slave rangefinder 120 receives the reference value, it can calculate the difference between the value of its measured current distance and the reference value, and further calculate the beamforming time based on the difference and the speed of sound propagation, that is, the time required to delay the target audio signal corresponding to it.

[0094] For details, please continue reading. Figure 4The processor 130 also includes a data channel 131, which is coupled between the microphone sensor 110 and the filtering unit 132, and is also coupled to a corresponding rangefinder 120. The data channel 131 is configured to perform the aforementioned frequency hopping step S110 to align the phase of the target audio signal output from the microphone device 100 with that of other audio signals output from other microphone devices 100. Furthermore, the frequency hopping step S110 is performed before the aforementioned processing step S107; that is, the audio data stream output from the microphone sensor 110, after entering the processor 130, will first pass through the data channel 131 before being sent to the filtering unit 132.

[0095] Specifically, in the audio signal adjustment system 200 provided in this embodiment, the operation of measuring the current distance between each microphone device 100 and the target sound source, as well as the operation of updating the current sensitivity level information, are repeated at a period of milliseconds.

[0096] Specifically, in one embodiment, the processor 130 described above may be a GD32. TM The series of microcontroller units (MCUs) are described, and in this embodiment, the processor 130 may include a high-performance digital filter (HPDF) module, in which the data channel, filtering unit, data unit, threshold monitor and extreme value monitor described above are disposed.

[0097] According to the foregoing embodiments, the present invention can ensure flexibility when processing the analog audio signal generated by the target sound source using the microphone device 100 with the audio signal adjustment method described herein. Specifically, it can ensure that the analog audio signal generated by the target sound source is not lost, while filtering out the received environmental noise signal to the greatest extent. Furthermore, in the process of processing the analog audio signal generated by the target sound source, it can compensate for the error caused by ranging and the influence of environmental factors.

[0098] In addition to the embodiments described above, the present invention may have other implementations. All technical solutions formed by equivalent substitutions or equivalent replacements fall within the protection scope claimed by the present invention.

[0099] In summary, although the preferred embodiments of the present invention have been disclosed above, the above preferred embodiments are not intended to limit the present invention. Those skilled in the art can make various modifications and refinements without departing from the spirit and scope of the present invention. Therefore, the scope of protection of the present invention shall be determined by the scope defined in the claims.

Claims

1. An audio signal adjustment method, characterized in that, The audio signal adjustment method, applied to a microphone device, includes: The system acquires data adjustment instructions and an audio data stream converted from an analog audio signal generated by a target sound source, wherein the data adjustment instructions carry current sensitivity level information, and the current sensitivity level information is related to the current distance between the microphone device and the target sound source; Configure the current data right shift bit number based on the current sensitivity level information; The audio data to be adjusted in the processed audio data stream is adjusted by shifting the current data right by the specified number of bits to obtain the target audio signal; Output the target audio signal; Calculate the signal threshold range corresponding to the number of bits shifted right of the current data, and monitor the maximum and minimum values ​​of the target audio signal; Obtain an update instruction generated based on the signal threshold range, the maximum value, and the minimum value, wherein the update instruction carries sensitivity feedback information; The current sensitivity level information is updated based on the sensitivity feedback information.

2. The audio signal adjustment method according to claim 1, characterized in that, The current distance is negatively correlated with the current sensitivity level information and the number of times the current data is shifted right.

3. The audio signal adjustment method according to claim 1, characterized in that, The signal threshold range has a maximum value and a minimum value, wherein: If the minimum value is less than the minimum value, the sensitivity feedback information indicates that the current sensitivity level information needs to be reduced to improve sensitivity; If the maximum value is greater than the maximum value, then the sensitivity feedback information indicates that the current sensitivity level information needs to be increased to reduce the sensitivity; and, If the minimum value is greater than the minimum value and the maximum value is less than the maximum value, then the sensitivity feedback information indicates that there is no need to change the current sensitivity level information.

4. The audio signal adjustment method according to claim 1, characterized in that, Before the step of outputting the target audio signal, the method further includes: Obtain a delay instruction, wherein the delay instruction carries a beamforming time, the beamforming time being determined by the difference between the current distance and other distances between other microphone devices and the target sound source, wherein the microphone devices and the other microphone devices constitute a microphone array; The pulse frequency hopping time is determined based on the beamforming time. The audio data stream is frequency-hopped according to the pulse hopping time so that the target audio signal is phase-aligned with other audio signals output by the other microphone devices.

5. The audio signal adjustment method according to claim 1, characterized in that, Before the step of adjusting the audio data to be adjusted in the processed audio data stream by the current data right shift by the specified number of bits, the method further includes: The audio data stream is digitally filtered to produce the processed audio data stream.

6. The audio signal adjustment method according to claim 1, characterized in that, The current sensitivity level information is calculated in real time based on the current distance between the microphone device and the target sound source.

7. An audio signal adjustment system, characterized in that, include: The main rangefinder is positioned on the target sound source; and, A microphone array, comprising multiple microphone devices, wherein the microphone devices include: A microphone sensor is used to convert the analog audio signal generated by the target sound source into an audio data stream; The current distance between the microphone device and the target sound source is measured from the rangefinder, which, together with the main rangefinder, determines this distance. A processor, coupled to the microphone sensor and the rangefinder, is configured to perform the audio signal adjustment method as described in any one of claims 1-6.

8. The audio signal adjustment system according to claim 7, characterized in that, The microphone device further includes a sensitivity modulator for generating the data adjustment command, wherein the sensitivity modulator calculates the current sensitivity level information in real time based on the current distance.

9. The audio signal adjustment system according to claim 8, characterized in that, The sensitivity modulator stores a mapping relationship between multiple distances and multiple sensitivity levels.

10. The audio signal adjustment system according to claim 7, characterized in that, The master rangefinder and the slave rangefinder determine the current distance based on the received signal strength.

11. The audio signal adjustment system according to claim 7, characterized in that, The processor further includes a filtering unit configured to perform digital filtering on the audio data stream to produce a processed audio data stream.

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