CN115631764A - Audio synchronization method and device for distributed microphones and storage medium - Google Patents
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Abstract
The invention provides an audio synchronization method, an audio synchronization device and a storage medium of a distributed microphone, wherein time correction is carried out on audio signals of a plurality of microphones at different positions based on the time difference from the time when the microphone sends the audio signals to the time when a host receives the audio signals, so that the audio signals tend to be synchronized in time, and then the audio signals are normalized to the audio signal containing the largest volume characteristic according to the characteristic information of the audio signals after the time correction, so that the final audio signal is obtained; the audio signals of the distributed microphones can be synchronously output, and the sound receiving effect is good.
Description
Technical Field
The present invention relates to the field of microphone technologies, and in particular, to an audio synchronization method and apparatus for a distributed microphone, and a storage medium.
Background
To some meeting rooms slightly bigger, place many radio sets and can be better gather participant's sound, when the meeting room has a plurality of microphones, because the position of a plurality of microphones is different, lead to there being the time delay between the sound that different microphones gathered, if directly play these sounds, can have very big interference, the auditory effect of sound can be relatively poor.
Disclosure of Invention
The invention provides an audio synchronization method, an audio synchronization device and a storage medium of a distributed microphone, and aims to improve the sound receiving effect of the distributed microphone and enable the sound of a plurality of microphones to be output synchronously.
In a first aspect, the present invention provides an audio synchronization method for a distributed microphone, including:
acquiring audio signals of a plurality of microphones;
calculating a time difference between audio signals of the plurality of microphones and a host receiving the audio signals;
time correction is carried out on the audio signals of the plurality of microphones according to the time difference;
acquiring characteristic information of a plurality of audio signals after time correction;
and adjusting the plurality of audio signals according to the characteristic information, so that the plurality of audio signals are normalized to the audio signal containing the largest volume characteristic to obtain a final audio signal.
In one embodiment, the adjusting the plurality of audio signals according to the feature information so that the modified plurality of audio signals are normalized to the audio signal containing the largest volume feature to obtain a final audio signal includes:
aligning the plurality of audio signals according to the feature information so that peaks and troughs of the plurality of audio signals are at the same time point;
and performing gain adjustment on other audio signals according to the audio signal containing the largest volume characteristic in the characteristic information, so that the plurality of audio signals are normalized to the audio signal containing the largest volume characteristic, and a final audio signal is obtained.
In one embodiment, the feature information includes feature points, lateral distances between the feature points, and a variation trend between the feature points;
wherein the characteristic points include peaks and valleys.
In one embodiment, the adjusting the modified audio signals according to the feature information so that the modified audio signals are normalized to the audio signal containing the largest volume feature further includes:
filtering a plurality of audio signals normalized to an audio signal containing the most loud volume features;
and fusing the plurality of audio signals after the filtering processing.
In one embodiment, the adjusting the plurality of audio signals according to the feature information so that after the plurality of audio signals are normalized to the audio signal containing the largest volume feature and before the obtaining the final audio signal further includes:
normalizing the plurality of audio signals to an audio signal containing the largest volume characteristic to obtain a normalized audio signal, wherein the normalized audio signal contains an overlapping area and a non-overlapping area, the overlapping area is an area with consistent change trend of the audio signal, and the non-overlapping area is an area with divergent change trend of the audio signal;
determining a start region signal of the normalized audio signal according to the non-overlapping region;
if the amplitude of the ending signal of the audio signal obtained in the previous sampling time period is continuous with the amplitude of the starting signal of the overlapping area of the current sampling time period, discarding the starting area signal; and if the amplitude of the ending signal of the audio signal obtained in the previous sampling time period and the amplitude of the starting signal of the overlapping area of the current sampling time period are not continuous, correcting the starting signal of the overlapping area of the current sampling time period according to the starting area signal so as to enable the corrected starting signal of the overlapping area of the current sampling time period and the ending signal of the audio signal obtained in the previous sampling time period to be continuous.
In one embodiment, the modifying the start signal of the overlap region of the current sampling time period according to the start region signal includes:
if the amplitude of the start area signal is closer to the end signal of the audio signal obtained in the last sampling time period than the amplitude of the start signal of the overlapping area of the current sampling time period, supplementing the start area signal into the audio signal of the overlapping area of the current sampling time period so that the modified start signal of the overlapping area of the current sampling time period is continuous with the end signal of the audio signal obtained in the last sampling time period;
and if the amplitude of the start area signal is farther away from the end signal of the audio signal obtained in the last sampling time period than the amplitude of the start signal of the overlapping area of the current sampling time period, deleting part of the start signal of the overlapping area of the current sampling time period so that the start signal of the overlapping area of the current sampling time period after modification is continuous with the end signal of the audio signal obtained in the last sampling time period.
In one embodiment, before the modifying the start area signal obtained in the current sampling time period based on the end signal of the audio signal obtained in the previous sampling time period, the method further includes:
and fusing the plurality of change trends of the start area signal to obtain a fused audio signal.
In one embodiment, the fusing the plurality of trend changes of the start area signal includes,
and taking one of the intermediate trend, the average trend and the main trend of the plurality of change trends as the final trend of the plurality of change trends.
In a second aspect, the present invention further provides an audio synchronization apparatus for a distributed microphone, including:
an audio signal acquisition unit for acquiring audio signals of a plurality of microphones;
the time delay calculation unit is used for calculating the time difference between the audio signals of the microphones and a host computer receiving the audio signals;
a time correction unit for performing time correction on the audio signals of the plurality of microphones according to the time difference;
a feature acquisition unit configured to acquire feature information of the plurality of audio signals after time correction;
and the signal adjusting unit is used for adjusting the plurality of audio signals according to the characteristic information, so that the plurality of audio signals are normalized to the audio signal containing the largest volume characteristic, and a final audio signal is obtained.
In a third aspect, the present invention also provides a computer storage medium, where a computer program is stored, and when the computer program is executed, the method for audio synchronization of distributed microphones in any one of the above-mentioned aspects is implemented.
The invention relates to an audio synchronization method, an audio synchronization device and a storage medium of a distributed microphone, which are characterized in that time correction is carried out on audio signals of a plurality of microphones at different positions based on the time difference from the audio signal sent by the microphone to the audio signal received by a host computer, so that the audio signals tend to be synchronized in time, and then the audio signals are normalized to the audio signal containing the largest volume characteristic according to the characteristic information of the audio signals after the time correction, so as to obtain the final audio signal; the audio signals of the distributed microphones can be synchronously output, and the sound receiving effect is good.
Drawings
In order to more clearly illustrate the embodiments of the present invention or the technical solutions in the prior art, the drawings required to be used in the embodiments are briefly introduced below, the drawings in the following description are only corresponding to some embodiments of the present invention, and it is obvious for a person skilled in the art to obtain drawings of other embodiments according to these drawings without creative efforts.
FIG. 1 is a flow chart of a method for audio synchronization of distributed microphones in accordance with one embodiment of the present invention;
FIG. 2 is a diagram illustrating feature information of a plurality of audio signals after time correction according to an embodiment of the present invention;
FIG. 3 is a diagram illustrating alignment of multiple audio signals according to one embodiment of the present invention;
FIG. 4 is a schematic diagram of gain adjustment of a plurality of audio signals according to one embodiment of the present invention;
FIG. 5 is a flowchart illustrating a method for audio synchronization of distributed microphones according to another embodiment of the present invention;
FIG. 6 is a diagram illustrating the fusion of multiple audio signals according to one embodiment of the present invention;
fig. 7 is a schematic structural diagram of an audio synchronization apparatus for distributed microphones according to an embodiment of the present invention.
Detailed Description
The technical solutions in the embodiments of the present invention will be clearly and completely described below with reference to the drawings in the embodiments of the present invention, and it is obvious that the described embodiments are only a part of the embodiments of the present invention, and not all of the embodiments. All other embodiments, which can be derived by a person skilled in the art from the embodiments given herein without making any creative effort, shall fall within the protection scope of the present invention.
Unless defined otherwise, all technical and scientific terms used herein have the same meaning as commonly understood by one of ordinary skill in the art to which this invention belongs.
The audio synchronization method and the audio synchronization device for the distributed microphones can be used for electronic equipment with a plurality of distributed microphones. The electronic devices include, but are not limited to, wearable devices, head-worn devices, medical health platforms, personal computers, server computers, hand-held or laptop devices, mobile devices (such as mobile phones, personal Digital Assistants (PDAs), media players, and the like), multiprocessor systems, consumer electronics, minicomputers, mainframe computers, distributed computing environments that include any of the above systems or devices, and the like. The electronic equipment is preferably an audio and video conference system with a plurality of distributed microphones so as to improve the audio output effect of the audio and video conference system.
Referring to fig. 1, the audio synchronization method of a distributed microphone of the present invention, in one embodiment, includes:
in step S101, audio signals of a plurality of microphones are acquired.
The microphones are distributed microphones or radio equipment comprising a plurality of single microphones and are placed at different positions of a conference room, so that voice information of participants can be conveniently collected, and audio signals are collected once at intervals of a certain sampling time period in the conference process.
Step S102, calculating a time difference between the audio signals of the plurality of microphones and a host receiving the audio signals.
In step S103, time correction is performed on the audio signals of the plurality of microphones according to the time difference.
Because the microphones are used for collecting sound, when a participant speaks, certain time delay exists between the voice information collected by the microphones at different positions, and the time of the voice information of the microphones can be adjusted to be consistent based on the time difference between the audio signals of the microphones and the host computer for receiving the audio signals.
Specifically, if at a certain sampling time, the time when one of the microphones sends out the audio signal to the host is T1, and the time when the host receives the audio signal is T2, the time difference between the time when the microphone sends out the audio signal to the host and the time when the host receives the audio signal is T = T2-T1. And delaying the audio signal sent by each microphone by time T based on the time difference so as to ensure that the time of the audio signals of the microphones are consistent.
Step S104, acquiring the characteristic information of the plurality of audio signals after the time correction.
Step S105, adjusting the plurality of audio signals according to the feature information, so that the plurality of audio signals are normalized to the audio signal containing the largest volume feature, and a final audio signal is obtained.
After the audio signals sent by the microphones are time-corrected, the obtained audio signals still have differences in phase and amplitude, and if the audio signals are played at this time, the noise is relatively large.
The audio signals containing the maximum volume characteristics can be deduced to be the microphone closest to a speaker, the acquired signal is the microphone with the clearest volume and the maximum volume, the acquired signal is normalized by taking the acquired signal as a reference to eliminate delay interference brought by the distributed microphone to the maximum extent, and the audio effect of the distributed microphone is optimized.
Here, the normalization of the plurality of audio signals to the audio signal containing the maximum volume characteristic may be understood as performing signal processing on the other audio signals so that the other audio signals approach the audio signal containing the maximum volume characteristic, and finally obtaining the audio signal with less interference.
In one embodiment, step S105 specifically includes:
aligning the plurality of audio signals according to the characteristic information so that peaks and troughs of the plurality of audio signals are at the same time point;
and performing gain adjustment on other audio signals according to the audio signal with the largest volume characteristic in the characteristic information, so that the plurality of audio signals are normalized to the audio signal with the largest volume characteristic to obtain a final audio signal.
Specifically, referring to fig. 2 to 4, the feature information includes feature points, lateral distances between the feature points, and a variation trend between the feature points; wherein the characteristic points comprise peaks and valleys.
Aligning the plurality of audio signals obtained after time correction based on the peaks, the troughs, the transverse distances between the peaks and the troughs and the variation trend between the peaks and the troughs so that the peaks and the troughs of the plurality of audio signals are at the same time point; and taking the characteristic point with the maximum peak or trough amplitude value at the same time point as a large volume characteristic, and taking the audio signal containing the maximum large volume characteristic as a reference signal to perform gain adjustment on other audio signals, so that the plurality of audio signals are normalized to the audio signal containing the maximum large volume characteristic to obtain a final audio signal.
According to the audio synchronization method of the distributed microphone, based on the time difference between the audio signals sent by the microphone and the audio signals received by the host, time correction is performed on the audio signals of the multiple microphones at different positions, so that the multiple audio signals tend to be synchronized in time, then according to the characteristic information of the multiple audio signals after the time correction, the multiple audio signals are normalized to the audio signal containing the largest volume characteristic, and a final audio signal is obtained, specifically, the multiple audio signals are aligned according to the characteristic information, and then amplitude adjustment is performed to normalize the multiple audio signals to the audio signal containing the largest volume characteristic; the audio signals of the distributed microphones can be synchronously output, the interference caused by the distributed microphones is reduced, and the sound receiving effect is good.
Referring to fig. 5, in one embodiment, an audio synchronization method for distributed microphones of the present application includes:
in step S201, audio signals of a plurality of microphones are acquired.
In step S202, time differences between the audio signals of the plurality of microphones and the host receiving the audio signals are calculated.
In step S203, the audio signals of the plurality of microphones are time-corrected according to the time difference.
In step S204, feature information of the plurality of audio signals after the time correction is acquired.
Step S205, aligning the plurality of audio signals according to the characteristic information, so that the wave crests and the wave troughs of the plurality of audio signals are at the same time point.
Step S206, performing gain adjustment on other audio signals according to the audio signal with the largest volume characteristic included in the characteristic information, so that the plurality of audio signals are normalized to the audio signal with the largest volume characteristic.
Step S207, filtering the plurality of audio signals normalized to the audio signal containing the most loud volume feature.
And step S208, fusing the plurality of audio signals after the filtering processing to obtain a final audio signal.
Referring to fig. 6, the plurality of audio signals normalized to the audio signal including the maximum volume characteristic are filtered to remove noise, and the plurality of audio signals after filtering are further fused, so that the plurality of audios are fused into one audio signal, and interference caused by the distributed microphone is completely removed.
After the alignment and gain adjustment are performed on the multiple audio signals based on the characteristic information, there may still be multiple audio signals that do not overlap, and such audio signals still have noise, so that the multiple audio signals may be further filtered and fused, and the interference may be further ablated.
In one embodiment, referring to fig. 4, adjusting the plurality of audio signals according to the feature information so that after the plurality of audio signals are normalized to the audio signal containing the largest volume feature and before the final audio signal is obtained, further includes:
normalizing the plurality of audio signals to obtain a normalized audio signal after the audio signal containing the maximum volume characteristic is normalized, wherein the normalized audio signal contains an overlapping area and a non-overlapping area, the overlapping area is an area with consistent change trend of the audio signal, and the non-overlapping area is an area with divergent change trend of the audio signal;
determining a start region signal of the normalized audio signal according to the non-overlapping region;
if the amplitude of the ending signal of the audio signal obtained in the previous sampling time period is continuous with the amplitude of the starting signal of the overlapping area of the current sampling time period, discarding the starting area signal; and if the amplitude of the ending signal of the audio signal obtained in the previous sampling time period is not continuous with the amplitude of the starting signal of the overlapping area of the current sampling time period, correcting the starting signal of the overlapping area of the current sampling time period according to the starting area signal so as to enable the corrected starting signal of the overlapping area of the current sampling time period to be continuous with the ending signal of the audio signal obtained in the previous sampling time period.
It can be seen that, due to the sequence of the sampling time of the distributed microphone, the normalized audio signals obtained after alignment and gain adjustment are highly overlapped in the DE region, the DE region is used as an overlapping region, the region before the point D is used as a start region signal, the signal after the point E is used as an end region signal, and the change trends of the start region signal and the end region signal diverge. In order to ensure the consistency between the audio signal obtained in the current sampling time period and the audio signal obtained in the previous sampling time period and the audio signal to be obtained in the next sampling time period, the start signal of the overlapping region of the current sampling time period needs to be modified, so that the modified start signal of the overlapping region of the current sampling time period is continuous with the end signal of the audio signal obtained in the previous sampling time period.
Further, the process of correcting the start signal of the overlap region of the current sampling time period according to the start region signal includes:
if the amplitude of the start area signal is closer to the end signal of the audio signal obtained in the last sampling time period than the amplitude of the start signal of the overlapping area of the current sampling time period, supplementing the start area signal into the audio signal of the overlapping area of the current sampling time period, so that the start signal of the overlapping area of the current sampling time period after being modified is continuous with the end signal of the audio signal obtained in the last sampling time period;
and if the amplitude of the start area signal is farther away from the end signal of the audio signal obtained in the previous sampling time period than the amplitude of the start signal of the overlapping area of the current sampling time period, deleting part of the start signal of the overlapping area of the current sampling time period so that the modified start signal of the overlapping area of the current sampling time period is continuous with the end signal of the audio signal obtained in the previous sampling time period.
Specifically, if the amplitude of the end signal of the audio signal obtained in the previous sampling time period is 0, the amplitude interval of the start area signal of the normalized audio signal obtained in the current sampling time period is [0,1], and the amplitude of the start signal of the overlap area is 1, the start area signal is supplemented to the audio signal of the overlap area of the current sampling time period, so that the start signal of the overlap area of the modified current sampling time period is 0, and if the amplitude interval of the start area signal of the normalized audio signal obtained in the current sampling time period is [ -1, -0.2], and the amplitude of the start signal of the overlap area is-0.2, the start signal of the overlap area having the amplitude of [ -0.2,0] is deleted, so that the start signal of the overlap area is continuous with the end signal of the audio signal obtained in the previous sampling time period.
In one embodiment, the method further comprises discarding the end region signal of the normalized audio signal.
In one embodiment, before the modifying the start area signal obtained in the current sampling time period based on the end signal of the audio signal obtained in the last sampling time period, the method further includes:
and fusing the plurality of change trends of the starting area signals to obtain fused audio signals.
Specifically, the merging of the plurality of trends of the audio signal of the start area includes,
and taking one of the intermediate trend, the average trend and the main trend of the multiple changing trends as the final trend of the multiple changing trends. Or other fusion means, and the embodiment is not limited thereto.
When the amplitude of the start area signal is calculated, trend fusion can be performed on the start area signal, the amplitude interval of the start area signal is calculated after the change trend of the start area signal is determined, and then the start area signal is compared with the end signal of the last sampling time period. The specific trend fusion mode comprises the step of taking one of the intermediate trend, the average trend and the main trend of a plurality of change trends as the final trend of the plurality of change trends, and the accuracy of the signal amplitude calculation of the starting area is improved.
In specific implementation, after alignment, gain adjustment and filtering processing are performed on a plurality of audio signals, different fusion processing can be performed on a highly coincident DE region signal and a start region signal and an end region signal, where variation trends diverge, respectively, so as to optimize signals of an effective region maximally, where the effective region corresponds to the DE region, and the start region signal is judged and cut, and the end region signal is discarded, thereby ensuring integrity and continuity between audio signals obtained in different time periods, and optimizing quality of the audio signals.
Referring to fig. 7, an embodiment of the present application further provides an audio synchronization apparatus for a distributed microphone, including:
an audio signal acquisition unit 10 for acquiring audio signals of a plurality of microphones;
a delay time calculation unit 20 for calculating a time difference between the audio signals of the plurality of microphones and a host receiving the audio signals;
a time correction unit 30 for time-correcting the audio signals of the plurality of microphones according to the time difference;
a feature acquisition unit 40 configured to acquire feature information of the plurality of audio signals after time correction;
the signal adjusting unit 50 is configured to adjust the plurality of audio signals according to the feature information, so that the plurality of audio signals are normalized to an audio signal containing the largest volume feature, and a final audio signal is obtained.
In one embodiment, the signal conditioning unit 50 is specifically configured to,
aligning the plurality of audio signals according to the characteristic information so that peaks and troughs of the plurality of audio signals are at the same time point;
and performing gain adjustment on other audio signals according to the audio signal with the largest volume characteristic in the characteristic information, so that the plurality of audio signals are normalized to the audio signal with the largest volume characteristic to obtain a final audio signal.
In one embodiment, the feature information includes feature points, lateral distances between the feature points, and a variation tendency between the feature points;
wherein the characteristic points comprise peaks and valleys.
In one embodiment, the audio synchronization apparatus for distributed microphones further comprises:
a fusion unit for filtering a plurality of audio signals normalized to an audio signal containing the most loud volume features;
and fusing the plurality of audio signals after the filtering processing.
In one embodiment, the audio synchronization apparatus for distributed microphones further comprises:
the starting signal processing unit is used for normalizing the plurality of audio signals to an audio signal containing the largest volume characteristic to obtain a normalized audio signal, wherein the normalized audio signal comprises an overlapping area and a non-overlapping area, the overlapping area is an area with consistent change trend of the audio signals, and the non-overlapping area is an area with divergent change trend of the audio signals;
determining a start region signal of the normalized audio signal according to the non-overlapping region;
if the amplitude of the ending signal of the audio signal obtained in the previous sampling time period is continuous with the amplitude of the starting signal of the overlapping area of the current sampling time period, discarding the starting area signal; and if the amplitude of the ending signal of the audio signal obtained in the previous sampling time period is not continuous with the amplitude of the starting signal of the overlapping area of the current sampling time period, correcting the starting signal of the overlapping area of the current sampling time period according to the starting area signal so as to enable the corrected starting signal of the overlapping area of the current sampling time period to be continuous with the ending signal of the audio signal obtained in the previous sampling time period.
In one embodiment, the process of modifying the start signal of the overlap region of the current sampling time period according to the start region signal includes:
if the amplitude of the start area signal is closer to the end signal of the audio signal obtained in the last sampling time period than the amplitude of the start signal of the overlapping area of the current sampling time period, supplementing the start area signal into the audio signal of the overlapping area of the current sampling time period so as to enable the start signal of the overlapping area of the current sampling time period after modification to be continuous with the end signal of the audio signal obtained in the last sampling time period;
and if the amplitude of the start area signal is farther away from the end signal of the audio signal obtained in the previous sampling time period than the amplitude of the start signal of the overlapping area of the current sampling time period, deleting part of the start signal of the overlapping area of the current sampling time period so that the modified start signal of the overlapping area of the current sampling time period is continuous with the end signal of the audio signal obtained in the previous sampling time period.
In one embodiment, before the modifying the start area signal obtained in the current sampling time period based on the end signal of the audio signal obtained in the last sampling time period, the method further includes:
and fusing the plurality of change trends of the starting area signal to obtain a fused audio signal.
In one embodiment, the fusing the plurality of trends of change of the audio signal of the start area includes,
and taking one of the intermediate trend, the average trend and the main trend of the multiple changing trends as the final trend of the multiple changing trends.
The specific processes of the units for executing the corresponding steps have been described in detail in the above method embodiments, and are not described herein again for brevity.
An embodiment of the present invention further provides a computer device, which includes a processor and a memory, where a computer program is stored in the memory, and when the computer program is loaded and executed by the controller, the computer program implements the method steps described in any of the above method embodiments.
An embodiment of the present invention further provides a computer storage medium, where a computer program is stored, and when the computer program is executed, the method steps described in any of the above method embodiments are implemented.
In the above embodiments provided by the present invention, it should be understood that the disclosed system, apparatus and method may be implemented in other ways. For example, the above-described apparatus embodiments are merely illustrative, and for example, the division of the units is only one logical division, and other divisions may be realized in practice, for example, a plurality of units or components may be combined or integrated into another system, or some features may be omitted, or not executed. In addition, the shown or discussed mutual coupling or direct coupling or communication connection may be an indirect coupling or communication connection through some interfaces, devices or units, and may be in an electrical, mechanical or other form.
The units described as separate parts may or may not be physically separate, and parts displayed as units may or may not be physical units, may be located in one place, or may be distributed on a plurality of network units. Some or all of the units can be selected according to actual needs to achieve the purpose of the solution of the embodiment.
In addition, functional units in the embodiments of the present application may be integrated into one processing unit, or each unit may exist alone physically, or two or more units are integrated into one unit. The integrated unit can be realized in a form of hardware, and can also be realized in a form of a software functional unit. The integrated unit, if implemented in the form of a software functional unit and sold or used as a stand-alone product, may be stored in a computer readable storage medium.
Based on such understanding, the technical solutions of the present application may be embodied in the form of a software product, which is stored in a storage medium and includes several instructions to enable a computer device (which may be a mobile terminal, a personal computer, a server, or a network device) to execute all or part of the steps of the method according to the embodiments of the present application. And the aforementioned storage medium includes: a U-disk, a removable hard disk, a Read-Only Memory (ROM), a Random Access Memory (RAM), a magnetic disk or an optical disk, and other various media capable of storing program codes.
In summary, although the present invention has been disclosed in terms of the preferred embodiments, the scope of the present invention is not limited thereto, and any person skilled in the art can substitute or change the technical solution of the present invention within the scope of the present invention.
All possible combinations of the technical features of the above embodiments may not be described for the sake of brevity, but should be considered as within the scope of the present disclosure as long as there is no contradiction between the combinations of the technical features.
Claims (10)
1. An audio synchronization method for a distributed microphone, comprising:
acquiring audio signals of a plurality of microphones;
calculating a time difference between audio signals of the plurality of microphones and a host receiving the audio signals;
time correction is carried out on the audio signals of the plurality of microphones according to the time difference;
acquiring characteristic information of a plurality of audio signals after time correction;
and adjusting the plurality of audio signals according to the characteristic information, so that the plurality of audio signals are normalized to the audio signal containing the largest volume characteristic, and a final audio signal is obtained.
2. The method of claim 1, wherein the adjusting the plurality of audio signals according to the feature information so that the modified plurality of audio signals are normalized to an audio signal having a largest volume feature to obtain a final audio signal comprises:
aligning the plurality of audio signals according to the characteristic information so that peaks and troughs of the plurality of audio signals are at the same time point;
and performing gain adjustment on other audio signals according to the audio signal containing the largest volume characteristic in the characteristic information, so that the plurality of audio signals are normalized to the audio signal containing the largest volume characteristic, and a final audio signal is obtained.
3. The audio synchronization method of a distributed microphone according to claim 1, wherein the feature information includes feature points, lateral distances between feature points, and a variation tendency between feature points;
wherein the characteristic points include peaks and valleys.
4. The method of claim 1, wherein the adjusting the modified audio signals according to the feature information so that the modified audio signals are normalized to the audio signal having the largest volume feature further comprises:
filtering a plurality of audio signals normalized to an audio signal containing the most loud volume features;
and fusing the plurality of audio signals after the filtering processing.
5. The method for audio synchronization of distributed microphones as claimed in claim 1, wherein said adjusting said plurality of audio signals according to said feature information, after said plurality of audio signals are normalized to the audio signal containing the largest volume feature and before said obtaining the final audio signal, further comprises:
normalizing the plurality of audio signals to an audio signal containing the largest volume characteristic to obtain a normalized audio signal, wherein the normalized audio signal contains an overlapping area and a non-overlapping area, the overlapping area is an area with consistent change trend of the audio signal, and the non-overlapping area is an area with divergent change trend of the audio signal;
determining a start region signal of the normalized audio signal according to the non-overlapping region;
if the amplitude of the ending signal of the audio signal obtained in the previous sampling time period is continuous with the amplitude of the starting signal of the overlapping area of the current sampling time period, discarding the starting area signal; and if the amplitude of the ending signal of the audio signal obtained in the previous sampling time period and the amplitude of the starting signal of the overlapping area of the current sampling time period are not continuous, correcting the starting signal of the overlapping area of the current sampling time period according to the starting area signal so as to enable the corrected starting signal of the overlapping area of the current sampling time period and the ending signal of the audio signal obtained in the previous sampling time period to be continuous.
6. The audio synchronization method of the distributed microphones as claimed in claim 5, wherein the step of modifying the start signal of the overlapping region of the current sampling time period according to the start region signal comprises:
if the amplitude of the start area signal is closer to the end signal of the audio signal obtained in the last sampling time period than the amplitude of the start signal of the overlapping area of the current sampling time period, supplementing the start area signal into the audio signal of the overlapping area of the current sampling time period so that the modified start signal of the overlapping area of the current sampling time period is continuous with the end signal of the audio signal obtained in the last sampling time period;
and if the amplitude of the start area signal is farther away from the end signal of the audio signal obtained in the last sampling time period than the amplitude of the start signal of the overlapping area of the current sampling time period, deleting part of the start signal of the overlapping area of the current sampling time period so that the start signal of the overlapping area of the current sampling time period after modification is continuous with the end signal of the audio signal obtained in the last sampling time period.
7. The audio synchronization method of the distributed microphones as claimed in claim 5, wherein before the end signal of the audio signal obtained based on the last sampling time period is modified from the start area signal obtained by the current sampling time period, further comprising:
and fusing the plurality of change trends of the start area signal to obtain a fused audio signal.
8. The audio synchronization method of distributed microphones of claim 7, wherein said fusing the plurality of trends of change of the start area signal comprises,
and taking one of the intermediate trend, the average trend and the main trend of the plurality of change trends as the final trend of the plurality of change trends.
9. An audio synchronization apparatus for a distributed microphone, comprising:
an audio signal acquisition unit for acquiring audio signals of a plurality of microphones;
the time delay calculation unit is used for calculating the time difference between the audio signals of the microphones and a host computer receiving the audio signals;
a time correction unit for performing time correction on the audio signals of the plurality of microphones according to the time difference;
a feature acquisition unit configured to acquire feature information of the plurality of audio signals after the time correction;
and the signal adjusting unit is used for adjusting the plurality of audio signals according to the characteristic information, so that the plurality of audio signals are normalized to the audio signal containing the largest volume characteristic, and a final audio signal is obtained.
10. A computer storage medium, in which a computer program is stored which, when executed, implements a method of audio synchronization of a distributed microphone according to any one of claims 1 to 8.
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| CN202211296909.7A CN115631764A (en) | 2022-10-21 | 2022-10-21 | Audio synchronization method and device for distributed microphones and storage medium |
| PCT/CN2022/134396 WO2024082378A1 (en) | 2022-10-21 | 2022-11-25 | Audio synchronization method and apparatus for distributed microphones, and storage medium |
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| CN202211296909.7A CN115631764A (en) | 2022-10-21 | 2022-10-21 | Audio synchronization method and device for distributed microphones and storage medium |
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| JP4163294B2 (en) * | 1998-07-31 | 2008-10-08 | 株式会社東芝 | Noise suppression processing apparatus and noise suppression processing method |
| CN107592600B (en) * | 2016-07-06 | 2024-04-02 | 深圳市三诺声智联股份有限公司 | Pickup screening method and pickup device based on distributed microphones |
| CN106710601B (en) * | 2016-11-23 | 2020-10-13 | 合肥美的智能科技有限公司 | Noise-reduction and pickup processing method and device for voice signals and refrigerator |
| CN110782912A (en) * | 2019-10-10 | 2020-02-11 | 安克创新科技股份有限公司 | Sound source control method and speaker device |
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