Audio processing system and method suitable for sound field control in different functional spaces

Abstract

The present invention discloses an audio processing system and method suitable for sound field control of different functional spaces, and relates to the technical field of audio processing, wherein the audio to be processed is obtained, information is extracted from the audio to be processed to obtain multiple audio lines, the same number of audio windows are respectively set for the multiple audio lines, adjacent audio windows in a single audio line are mutually associated, the audio windows on the multiple audio lines are corresponded based on the correspondence between the multiple audio lines, the audio lines are processed one by one and transmitted to the upper network through the audio windows to obtain the target audio line. The present invention has good audio processing and transmission functions, can understand the information of other audio lines based on the information of one audio line, and can ensure the safe storage of audio and the accuracy of the original audio.

Description

Technical Field

[0001] The present invention relates to the technical field of audio processing, and in particular to an audio processing system and method suitable for controlling sound fields in different functional spaces. Background Art

[0002] With the development of society, audio is an important way of information communication in society. Audio collection and processing need to be implemented in sound field control in various occasions, and the different functions of the occasions will have uncertain effects on audio, such as outdoor and indoor environments. Therefore, corresponding processing is required for audio in multiple environments. Audio processing is necessary in the use of audio, which can improve the quality of audio and the need for information acquisition. In the process of audio processing, it is difficult to associate the processed audio with the original audio. Audio processing mostly involves multiple departments. Therefore, audio transmission is required for collaborative audio processing in the process of processing. In the audio processing process, the original audio is the basis for audio processing and is very important basic data. After the processed audio is transmitted to others, the original audio also needs to be transmitted. The process of communicating audio is relatively complicated and it is difficult to ensure the information integrity of the original audio during transmission. Summary of the invention

[0003] The purpose of the present invention is to provide an audio processing system and method suitable for sound field control in different functional spaces, so as to solve the shortcomings of the background technology.

[0004] In order to achieve the above object, the present invention provides the following technical solution: an audio processing method suitable for controlling sound fields in different functional spaces, comprising the following steps:

[0005] Acquire the audio to be processed, extract information from the audio to be processed to obtain multiple audio lines, and establish a corresponding relationship between the multiple audio lines as a lower layer network;

[0006] The same number of audio windows are respectively set corresponding to the multiple audio lines, and the adjacent audio windows in a single audio line are associated with each other, and the audio windows are communicatively connected with the corresponding cloud server based on the upper layer network, wherein the cloud server serves as the upper layer network;

[0007] Based on the correspondence between the multiple audio lines, the audio windows on the multiple audio lines are corresponded to each other, and the cloud servers corresponding to the audio windows on the corresponding multiple audio lines are associated with each other;

[0008] Process the audio lines one by one and transmit them to the upper network through the audio window to obtain the target audio line;

[0009] Combine the target audio lines to obtain the target audio.

[0010] In a preferred embodiment, the step of establishing the correspondence between the plurality of audio lines as the lower network includes:

[0011] Setting an audio frame, wherein the audio frame includes a main audio frame and a plurality of sub-audio frames, and setting a plurality of follow columns based on the audio frame, wherein the follow columns are respectively configured in the sub-audio frames, wherein the follow columns are a collection of a plurality of follow points, and there is a one-to-one correspondence between the follow columns and the sub-audio frames;

[0012] Determine the audio to be processed, put the audio to be processed into the audio frame, and extract the type of sound in the audio to obtain human voice audio and environmental audio;

[0013] The follow point marks the human voice audio and the ambient audio respectively according to the timbre, the follow points in the follow column mark the audio with the same timbre, the follow points in the same follow column are associated, and the human voice audio and the ambient audio are split based on the follow column to obtain multiple audio lines;

[0014] Based on the follow-up point, the multiple audio lines are matched according to the audio playing time to obtain the corresponding relationship between the multiple audio lines as the lower layer network.

[0015] In a preferred embodiment, the follow points in the follow column mark the audio with the same timbre, and the step of associating the follow points in the same follow column includes:

[0016] Play human voice audio and environmental audio, and identify the sounds in the human voice audio and environmental audio based on the timbre to obtain multiple timbre;

[0017] Make a one-to-one correspondence between the follow column and the sounds of multiple timbres in the audio, continue to play the human voice audio and the ambient audio, and mark the follow point in the follow column starting from the sound start point of the corresponding timbre based on the marking condition, and use the marked position on the sound of the corresponding timbre as the marked position, and bind the marked position to the follow point;

[0018] By recording the playing time of the follow point at the marked position, the sound content between adjacent follow points is identified to obtain a single segment of information and stored together in the adjacent follow points;

[0019] The single-segment information is encoded and assigned to the corresponding follow-up point to obtain a follow-up point label, and the follow-up points are sorted and associated according to the follow-up point label.

[0020] In a preferred embodiment, the steps of respectively setting the same number of audio windows for the corresponding multiple audio lines, associating adjacent audio windows in a single audio line with each other, and communicating and connecting the audio windows with the corresponding cloud server based on an upper layer network include:

[0021] Based on the playback time of the audio to be processed, multiple audio lines are virtually extended to correspond to each other, the same number of division windows are set on the multiple virtually extended audio lines, and the division windows on the single virtually extended audio line are in a connected state;

[0022] Acquire the position of the divided window and the follow point tag inside the divided window, and establish the position information between the divided window and the follow point inside the divided window based on the follow point tag, wherein the position of the divided window is the playback time segment of the audio line defined by the divided window, and the position information is the time position relationship between the playback time of the follow point inside the divided window and the playback time segment;

[0023] Using the position information corresponding to the divided window and the position of the divided window as the audio window;

[0024] Adjacent audio windows in a single audio line are associated with each other, and the audio windows are communicatively connected with the corresponding cloud server based on the upper layer network.

[0025] In a preferred embodiment, the step of corresponding the audio windows on the multiple audio lines based on the corresponding relationship between the multiple audio lines, and associating the cloud servers corresponding to the audio windows on the corresponding multiple audio lines, includes:

[0026] The cloud servers corresponding to the audio windows in a single audio line are connected through a protocol to obtain a first-level service chain;

[0027] The audio windows at the same divided window locations on multiple audio lines are matched as chained audio windows, and the cloud servers corresponding to the chained audio windows are connected and associated through a protocol to obtain a secondary service chain.

[0028] In a preferred embodiment, the step of processing the audio lines one by one and transmitting them to the upper layer network through the audio window to obtain the target audio line includes:

[0029] Formulate a corresponding adjustment plan, and process the audio lines one by one based on the adjustment plan to obtain the processed audio lines;

[0030] The processed audio line is transmitted to the corresponding cloud server based on the audio window, and the processed audio line is used as an additional audio line and stored in the primary service chain and the secondary service chain in the corresponding upper network;

[0031] The additional audio line is copied and separated from the upper network, and a communication relationship is established between the follow-up point in the additional audio line and the follow-up point in the corresponding audio line in the cloud server to obtain the target audio line.

[0032] In a preferred embodiment, the step of combining target audio lines to obtain target audio includes:

[0033] Temporarily combining the target audio lines based on the play time to obtain temporary audio;

[0034] Bind all follow points in the temporary audio to get the target audio.

[0035] The present invention also provides an audio processing system suitable for controlling sound fields in different functional spaces, including:

[0036] An acquisition module is used to acquire the audio to be processed, extract information from the audio to be processed to obtain multiple audio lines, and establish a corresponding relationship between the multiple audio lines as a lower layer network;

[0037] A setting module, connected to the acquisition module, for setting the same number of audio windows for the multiple audio lines, respectively, associating adjacent audio windows in a single audio line with each other, and communicating and connecting the audio windows with the corresponding cloud server based on an upper layer network, wherein the cloud server serves as the upper layer network;

[0038] A corresponding module, connected to the setting module, for corresponding the audio windows on the multiple audio lines based on the corresponding relationship between the multiple audio lines, and associating the cloud servers corresponding to the audio windows on the corresponding multiple audio lines;

[0039] The processing module is connected to the corresponding module and is used to process the audio lines one by one and transmit them to the upper network through the audio window to obtain the target audio line;

[0040] The combining module is connected to the processing module and is used for combining the target audio lines to obtain the target audio.

[0041] In the above technical solution, the technical effects and advantages provided by the present invention are:

[0042] The additional audio line of the present invention can be copied and used without the upper network, which is convenient for audio transmission. At the same time, the target audio line can also obtain the original audio line in the cloud server through the communication relationship between the follow point and the follow point in the corresponding audio line in the cloud server. At the same time, according to the audio window in the original audio line, the most original audio of the audio line that has not been modified can be obtained, which is convenient for subsequent reprocessing of the audio, has better audio processing and transmission functions, can understand the information of other audio lines based on the information of one audio line, and can ensure the safe storage of audio and the accuracy of the original audio. BRIEF DESCRIPTION OF THE DRAWINGS

[0043] In order to more clearly illustrate the embodiments of the present application or the technical solutions in the prior art, the drawings required for use in the embodiments will be briefly introduced below. Obviously, the drawings described below are only some embodiments recorded in the present invention. For ordinary technicians in this field, other drawings can also be obtained based on these drawings.

[0044] Figure 1 The figure is a flow chart of the method of the present invention.

[0045] Figure 2 It is a system block diagram of the present invention. DETAILED DESCRIPTION

[0046] In order to make the purpose, technical solution and advantages of the embodiments of the present invention clearer, the technical solution in the embodiments of the present invention will be clearly and completely described below in conjunction with the drawings in the embodiments of the present invention. Obviously, the described embodiments are part of the embodiments of the present invention, not all of the embodiments. Based on the embodiments of the present invention, all other embodiments obtained by ordinary technicians in this field without creative work are within the scope of protection of the present invention.

[0047] Example 1, please refer to Figure 1 As shown, the audio processing method applicable to sound field control of different functional spaces described in this embodiment includes the following steps:

[0048] S1. Obtain the audio to be processed, extract information from the audio to be processed to obtain multiple audio lines, and establish a corresponding relationship between the multiple audio lines as a lower layer network;

[0049] S2. respectively setting the same number of audio windows for the corresponding multiple audio lines, associating adjacent audio windows in a single audio line with each other, and communicating and connecting the audio windows with the corresponding cloud server based on the upper layer network, wherein the cloud server serves as the upper layer network;

[0050] S3, based on the correspondence between the multiple audio lines, the audio windows on the multiple audio lines are matched with each other, and the cloud servers corresponding to the audio windows on the corresponding multiple audio lines are associated with each other;

[0051] S4, processing the audio lines one by one and transmitting them to the upper network through the audio window to obtain the target audio line;

[0052] S5, combining the target audio lines to obtain the target audio;

[0053] As described in the above steps S2-S24, with the development of society, audio is an important way of information communication in society. In the use of audio, audio processing is necessary to improve the quality of audio and the demand for information acquisition. In the process of audio processing, it is difficult to associate the processed audio with the original audio. Audio processing mostly involves multiple departments. Therefore, in the process of processing, audio transmission is required for collaborative audio processing. In the audio processing process, the original audio is the basis for processing audio and is very important basic data. After the processed audio is transmitted to others, the original audio also needs to be transmitted to exchange audio. The process is relatively complicated and it is difficult to ensure the information integrity of the original audio during transmission. In the present application, the additional audio line can be copied and used without the upper network, which is convenient for audio transmission. At the same time, the target audio line can also obtain the original audio line in the cloud server through the communication relationship between the follow point and the follow point in the corresponding audio line in the cloud server. At the same time, according to the audio window in the original audio line, the audio line can be obtained without modification. The most original audio is convenient for subsequent reprocessing of the audio, has better audio processing and transmission effects, can understand the information of other audio lines based on the information of one audio line, and can ensure the safe storage of audio and the accuracy of the original audio.

[0054] In audio quality processing, it is necessary to refer to multiple acoustic indicators to judge the audio quality and adjust the parameters to obtain audio with optimized quality, including:

[0055] STI is the Speech Transmission Index, an important indicator for measuring the quality of speech transmission. It combines multiple factors such as reverberation time, signal-to-noise ratio, echo, system distortion, and psychoacoustic effects (such as masking effects). It is expressed as a value between 0 and 1. The larger the value, the better the clarity. However, STI itself is not a parameter that can be adjusted directly, but needs to be indirectly affected by adjusting the relevant parameters of the audio system or audio processing process. The following STI directly obtains the STI value by measuring the modulation index m by the direct method;

[0056] For example, the STI acquisition method includes placing a microphone at the position to be tested for STI, collecting the audio signal after passing through the transmission medium, and collecting the frequency response, reverberation time, signal-to-noise ratio, echo, etc.;

[0057] The collected audio signal is band-pass filtered to remove unnecessary frequency components; then, the filtered signal is envelope-detected to extract its modulation characteristics;

[0058] By analyzing the modulation characteristics, the modulation index m is calculated, where modulation index m = [maximum audio amplitude - minimum audio amplitude] / [maximum audio amplitude + minimum audio amplitude];

[0059] According to the conversion relationship between the modulation index m and STI, the modulation index and the collected frequency response, reverberation time, signal-to-noise ratio, echo, etc. are combined to obtain the STI value. (The modulation index m reflects the depth or amplitude of the modulated signal. In the frequency modulation (FM) signal, the larger the m value, the larger the frequency deviation and the corresponding increase in the signal bandwidth. A larger m value may mean richer voice details and higher clarity, but it may also bring the risk of signal distortion and increased noise. The STI value also needs to refer to the frequency response, reverberation time, signal-to-noise ratio, echo, etc. of the transmission system);

[0060] The calculated STI value is recorded and evaluated according to the IEC 60268-16 standard or other relevant standards. The closer the STI value is to 1, the better the clarity of the speech signal; otherwise, it means that the clarity is poor;

[0061] After obtaining the STI, the STI that does not meet the audio parameter quality threshold is optimized. The optimization methods include:

[0062] Optimize the sound field environment: 1. Control the reverberation time: Too long or too short a reverberation time will affect the STI value. According to the specific environment in which the audio is used (such as conference rooms, theaters, concert halls, etc.), the reverberation time can be controlled by adjusting the sound-absorbing materials and layout in the room to keep it within a suitable range.

[0063] 2. Reduce echo: Echoes will interfere with the original sound and reduce the STI value. Echoes can be reduced by installing acoustic materials such as sound-absorbing panels and diffusers, or by adjusting the position of the sound source and the audience;

[0064] Improve signal quality: 1. Improve signal-to-noise ratio: The signal-to-noise ratio refers to the ratio of signal to noise. Improving the signal-to-noise ratio can reduce the interference of noise on speech, thereby improving the STI value. This can be achieved by using high-quality audio equipment, optimizing signal transmission paths, and adopting noise suppression technology.

[0065] Avoid signal distortion: Signal distortion will cause the loss of useful information in the voice signal and reduce the STI value. Therefore, in the audio processing process, avoid excessive compression, clipping and other operations that cause distortion; use audio processing software or equipment: 1. Equalizer: By adjusting the equalizer to change the spectrum distribution of the audio signal, the voice signal will be more prominent in the key frequency band, thereby increasing the STI value. However, be careful not to over-adjust to avoid introducing new distortion.

[0066] Compressor: Proper use of compressor can balance the dynamic range of audio signals, making the sound smoother and easier to understand, but it is also necessary to avoid distortion caused by over-compression; In addition to the optimization of STI in audio, the following parameters are also optimized:

[0067] EDT and RT10, the full name in Chinese is "Early Decay Time". It is defined as the reverberation time determined by the slope of the decay curve from 0dB to -10dB in the early part of the decay process of the indoor sound field after the sound source stops making sound;

[0068] RT60: RT60 refers to the "reverberation time" of a room. Specifically, it refers to the time it takes for the sound field to decay by 60dB, measured in seconds. The larger the RT60, the longer the reverberation time of the room, that is, the longer it takes for the sound to disappear.

[0069] T20, T30: They represent the RT60 calculated based on the time taken for the sound field to decay by 20 and 30 dB respectively;

[0070] Clarity and intelligibility parameters: The main feature of the speech intelligibility index is to calculate different weights for speech sounds in different frequency bands, thereby defining the distribution of speech sounds at different frequencies;

[0071] C80: C80 refers to the logarithm of the ratio of the sound energy within 80ms to the sound energy after 80ms. This index represents the clarity of music.

[0072] D50: D50 refers to the logarithm of the ratio of sound energy within 50ms to total sound energy. D50 represents the clarity of speech and is positively correlated with STI. The maximum value of d50 is 1. The closer it is to 1, the higher the clarity of speech in the hall.

[0073] The ALCONS value is the consonant loss, which is an indicator of clarity;

[0074] AI and SII: AI clarity index. When fitting hearing aids, we often use the clarity index and speech clarity index to guide and evaluate the adjustment of hearing aid parameters and their effects, in order to improve the patient's audibility of speech.

[0075] RASTI stands for Room Acoustic Speech Transmission Index, which was developed for special cases, such as verifying a speaker's speech in a small room without echoes;

[0076] STIPA: The full meaning of STIPA is the Language Transmission Index of Public Broadcasting System;

[0077] TS: Center of gravity time ts, represents the relationship between clarity and reverberation, and is a secondary reference indicator;

[0078] Acoustic indicator parameters corresponding to the audio in the system are obtained according to the above acoustic indicators, multiple audio parameter quality thresholds are set corresponding to the multiple acoustic indicator parameters, and the acoustic indicator parameters corresponding to the audio parameter quality thresholds that do not meet the adjustment are adjusted to obtain the adjusted and optimized audio;

[0079] In one embodiment, the step S1 of establishing the correspondence between the plurality of audio lines as the lower layer network includes:

[0080] S11, setting an audio frame, wherein the audio frame includes a main audio frame and multiple sub-audio frames, and setting multiple follow columns based on the audio frame, wherein the follow columns are respectively configured in the sub-audio frames, wherein the follow column is a collection of multiple follow points, and there is a one-to-one correspondence between the follow column and the sub-audio frame;

[0081] S12, determining the audio to be processed, putting the audio to be processed into an audio frame, and extracting the type of sound in the audio to obtain human voice audio and environmental audio;

[0082] S13, the follow points mark the human voice audio and the ambient audio respectively according to the timbre, the follow points in the follow column mark the audio with the same timbre, the follow points in the same follow column are associated, and the human voice audio and the ambient audio are split based on the follow column to obtain multiple audio lines;

[0083] S14, based on the follow-up point, the plurality of audio lines are matched according to the audio playing time to obtain a corresponding relationship between the plurality of audio lines as a lower layer network;

[0084] As described in the above steps S11-S14, the audio frame here is a space for processing audio, and multiple follow columns are set in the audio frame, wherein the follow column is a collection of multiple follow points. When the audio is subsequently split, a follow column corresponds to an audio line of the timbre, the audio to be processed is determined, and the audio to be processed is put into the audio frame, and the type of sound in the audio is extracted to obtain human voice audio and environmental audio. Here, the human voice and environmental sound in the audio are first separated, and then different timbres are split according to the type of the human voice audio and environmental audio, and the audio of different timbres is followed by the follow column. The follow points in the follow column follow the audio of the same timbre. The follow points mark the human voice audio and the ambient audio respectively according to the timbre. The follow points in the follow column mark the audio of the same timbre. The follow points in the same follow column are associated. The human voice audio and the ambient audio are split based on the follow column to obtain multiple audio lines. Then, the multiple audio lines are matched according to the audio playback time based on the follow points to obtain the corresponding relationship between the multiple audio lines as the lower layer network. Here, it is used for subsequent data transmission and storage between the cloud server in the upper layer network, which has a good information management function.

[0085] In one embodiment, the follow points in the follow column mark the audio with the same timbre, and the step S13 of associating the follow points in the same follow column includes:

[0086] S131, playing human voice audio and environmental audio, and identifying sounds in the human voice audio and environmental audio based on timbre to obtain multiple timbre;

[0087] S132, one-to-one correspondence is established between the follow column and the sounds of multiple timbres in the audio, the human voice audio and the ambient audio are played continuously, and the follow point in the follow column is marked starting from the sound start point of the corresponding timbre based on the marking condition, and the position marked on the sound of the corresponding timbre is used as the marking position, and the marking position is bound to the follow point;

[0088] S133, using the play time recorded at the marked position of the follow point, the sound content between adjacent follow points is identified to obtain a single segment of information and stored together in the adjacent follow points;

[0089] S134, encoding the single-segment information and assigning it to the corresponding follow-up point to obtain a follow-up point label, and sorting and associating the follow-up points according to the follow-up point labels;

[0090] As described in the above steps S131-S134, multiple follow columns are respectively set in the sub-audio frame, and then the sounds in the human voice audio and the environmental audio are respectively identified to obtain multiple timbres. The follow column enters the main audio frame and corresponds to the sounds of the multiple timbres in the audio one by one, and the human voice audio and the environmental audio are continued to be played. Based on the marking condition, the follow point in the follow column is marked starting from the sound starting point of the corresponding timbre, and the position of the mark on the sound of the corresponding timbre is used as the marking position, and the marking position is bound to the follow point. The marking condition here is to be separated by a certain playback time interval. When the sound of the timbre exceeds T time, the mark appears. The interrupt mark is not marked according to the playback time interval, and when it appears again, it is marked with the timbre. The starting point of the sound is marked according to the playing time interval, and the follow-up point is marked at the end point of the sound. The playing time at the marked position is recorded by the follow-up point, and the sound content between adjacent follow-up points is identified to obtain a single-segment information and stored in adjacent follow-up points. The follow-up point can record the content of the sound of the same timbre and arrange the emission time to avoid confusion. Then, the follow-up point label is obtained by encoding based on the single-segment information and assigning it to the corresponding follow-up point. The follow-up points are sorted and associated by the follow-up point label, so that the follow-up points in the follow-up column can mark and sort the sounds of the same timbre, and can split sounds of multiple timbre, with a good division effect;

[0091] In one embodiment, the step S2 of setting the same number of audio windows for the corresponding multiple audio lines, associating adjacent audio windows in a single audio line with each other, and communicating and linking the audio windows with the corresponding cloud server based on the upper layer network includes:

[0092] S21, virtually extending multiple audio lines based on the playback time of the audio to be processed, setting the same number of division windows on the virtually extended multiple audio lines, and the division windows on the virtually extended single audio line are in a connected state;

[0093] S22, obtaining the position of the divided window and the follow-up point label inside the divided window, and establishing the position information between the divided window and the follow-up point inside the divided window based on the follow-up point label, wherein the position of the divided window is the playback time segment of the audio line defined by the divided window, and the position information is the time position relationship between the playback time of the follow-up point inside the divided window and the playback time segment;

[0094] S23, using the divided window, the location information corresponding to the divided window, and the location of the divided window as an audio window;

[0095] S24, associating adjacent audio windows in a single audio line with each other, and communicating and connecting the audio windows with corresponding cloud servers based on an upper layer network;

[0096] As described in the above steps S21-S24, since the starting points of the sounds of multiple audio lines are inconsistent and the interruptions in the middle are also inconsistent, the multiple audio lines are virtually extended to the length of the playback time of the audio to be processed according to the playback time of the audio to be processed, and then the same number of division windows are set on the multiple virtually extended audio lines, and the division windows on the virtually extended single audio line are in a contact state, and the division windows are set closely together, and then the position of the division window and the follow point label inside the division window are obtained, and the position information between the division window and the follow point inside the division window is established based on the follow point label, wherein the position of the division window is the playback time segment of the audio line delimited by the division window, and the position information is the position information inside the division window. The time position relationship between the playback time of the following points and the playback time segment is determined, where the playback time segments corresponding to the playback time at both ends of the audio line divided by the dividing window are used as the position of the dividing window, and there may be one or more following points or no following point in the dividing window. When there is no following point, there is no point information in the dividing window and only the position of the dividing window exists. The dividing window, the position information corresponding to the dividing window, and the position of the dividing window are used as audio windows, and adjacent audio windows in a single audio line are associated with each other. Based on the upper network, the audio window is communicated and linked with the corresponding cloud server, and the corresponding cloud server is used as an upper network for connecting with the corresponding audio window. The step of associating adjacent audio windows in a single audio line includes:

[0097] The identification codes are set on both sides of the audio window corresponding to the setting order of the audio window;

[0098] Identify and associate adjacent audio windows based on the identification code;

[0099] It can better set the audio window to a stable level, ensuring the accuracy of the audio and facilitating subsequent audio adjustments;

[0100] In one embodiment, the step S3 of corresponding the audio windows on the multiple audio lines based on the corresponding relationship between the multiple audio lines and associating the cloud servers corresponding to the audio windows on the corresponding multiple audio lines includes:

[0101] S31, connecting the cloud servers corresponding to the audio windows in the single audio line through a protocol to obtain a first-level service chain;

[0102] S32, corresponding audio windows at the same divided window positions on multiple audio lines as chained audio windows, and connecting and associating cloud servers corresponding to the chained audio windows through a protocol to obtain a secondary service chain;

[0103] As described in the above steps S31 and S32, the cloud servers corresponding to the audio windows in a single audio line are connected through a protocol to obtain a primary service chain, which is a blockchain for storing information. After that, the audio windows corresponding to the points of the same divided windows on multiple audio lines are matched with each other, and the corresponding cloud servers are also associated and connected through a protocol. As a secondary service chain, it is also another blockchain. There is an intersection between the two blockchains, and the information between the primary service chain and the secondary service chain is isolated from each other. For example, the first audio window in the primary service chain and an audio window in the secondary service chain are an intersection. Therefore, except for the audio window itself information (the audio window as the intersection) in the secondary service chain, the information of the audio window itself is not included in the secondary service chain. The information of other audio windows in the secondary service chain other than the original information of the audio in the audio window is hidden from the primary service chain. The information of other audio lines can be learned based on the information of one audio line, and the safe storage of audio and the accuracy of the original audio can be guaranteed. For example, the audio window carries the playback time recorded by the follow point and the sound content between adjacent follow points, the location of the divided window, the follow point label inside the divided window, and the location information between the divided window and the follow point inside the divided window. If this information changes, the original audio may be modified. In addition, the original audio is stored by the blockchain and is stored jointly by the primary service chain and the secondary service chain to ensure the security of information.

[0104] In one embodiment, the step S4 of processing the audio lines one by one and transmitting them to the upper layer network through the audio window to obtain the target audio line includes:

[0105] S41, formulating a corresponding adjustment scheme, and processing the audio lines one by one based on the adjustment scheme to obtain processed audio lines;

[0106] S42, transmitting the processed audio line to the corresponding cloud server based on the audio window, and storing the processed audio line as an additional audio line in the first-level service chain and the second-level service chain in the corresponding upper network;

[0107] S43, copying the additional audio line and separating it from the upper network, establishing a communication relationship between the follow-up point in the additional audio line and the follow-up point in the corresponding audio line in the cloud server to obtain the target audio line;

[0108] As described in the above steps S41-S43, a corresponding adjustment plan is formulated for the corresponding audio line, wherein the adjustment plan includes sound volume, denoising and other clarification processing or sound elimination and cutting operations, to obtain the processed audio line, and the processing is used as the internal audio processing as the inner loop stage, and then further processing is performed according to the audio requirements in the environment. The processing here may include operations such as speech clarity and speech sound enhancement, and finally the audio is adjusted according to the audio parameters required by the environment as the outer loop stage, and the outer loop and the inner loop are integrated for audio processing to obtain the final audio, where cutting or adding audio are automatically matched with other audio lines for the same playback time, and then the processed audio line is transmitted to the corresponding cloud server based on the audio window, and the primary service chain and the secondary service chain corresponding to the processed audio line are used as additional audio, and the follow point in the additional audio is matched with the corresponding one in the cloud server. A communication relationship is established between the follow points in the audio line and they are stored together in the upper network to obtain the target audio line. Here, the audio lines are adjusted one by one, and then an additional audio is generated as the additional audio line and stored in the primary service chain and the secondary service chain accordingly. Here, the storage in the primary service chain and the secondary service chain is also an additional storage, which does not affect the original data of the original audio line. A communication relationship is established between the follow point in the additional audio line and the follow point in the corresponding audio line in the cloud server to obtain the target audio line. The additional audio line here can be copied and used away from the upper network, which is convenient for audio transmission. At the same time, the target audio line can also obtain the original audio line in the cloud server through the communication relationship between the follow point and the follow point in the corresponding audio line in the cloud server. At the same time, according to the audio window in the original audio line, the audio line can be obtained as the most original audio that has not been modified, which is convenient for subsequent reprocessing of the audio, and has a good audio processing and transmission effect;

[0109] In one embodiment, the step S5 of combining target audio lines to obtain target audio includes:

[0110] S51, temporarily combining the target audio lines based on the playing time to obtain temporary audio;

[0111] S52, binding all follow points in the temporary audio to obtain the target audio;

[0112] As described in the above steps S51 and S52, the target audio lines are temporarily combined according to the audio playback time to obtain temporary audio. The temporary audio here is a combination of multiple audio line sounds, which is unstable. Therefore, it is necessary to bind all the follow points in the temporary audio to obtain the target audio, so that the audio is more stable. During the transmission process, the stability of the audio can be better guaranteed to avoid confusion and audio line deviation. All the follow points in the temporary audio are bound to determine the mutual information between the follow points. The information here determines the time interval between the follow points and the corresponding target audio lines.

[0113] Example 2, please refer to Figure 2 As shown, the audio processing system applicable to sound field control of different functional spaces described in this embodiment includes:

[0114] An acquisition module is used to acquire the audio to be processed, extract information from the audio to be processed to obtain multiple audio lines, and establish a corresponding relationship between the multiple audio lines as a lower layer network;

[0115] A setting module, connected to the acquisition module, for setting the same number of audio windows for the multiple audio lines, respectively, associating adjacent audio windows in a single audio line with each other, and communicating and connecting the audio windows with the corresponding cloud server based on an upper layer network, wherein the cloud server serves as the upper layer network;

[0116] A corresponding module, connected to the setting module, for corresponding the audio windows on the multiple audio lines based on the corresponding relationship between the multiple audio lines, and associating the cloud servers corresponding to the audio windows on the corresponding multiple audio lines;

[0117] The processing module is connected to the corresponding module and is used to process the audio lines one by one and transmit them to the upper network through the audio window to obtain the target audio line;

[0118] The combining module is connected to the processing module and is used for combining the target audio lines to obtain the target audio.

[0119] The above is only a specific implementation of the present application, but the protection scope of the present application is not limited thereto. Any person skilled in the art who is familiar with the present technical field can easily think of changes or substitutions within the technical scope disclosed in the present application, which should be included in the protection scope of the present application. Therefore, the protection scope of the present application should be based on the protection scope of the claims.

Claims

1. An audio processing method suitable for controlling sound fields in different functional spaces, characterized in that: The following steps are involved: Acquire the audio to be processed, extract information from the audio to be processed to obtain multiple audio lines, and establish a corresponding relationship between the multiple audio lines as a lower layer network; The same number of audio windows are respectively set corresponding to the multiple audio lines, and the adjacent audio windows in a single audio line are associated with each other, and the audio windows are communicatively connected with the corresponding cloud server based on the upper layer network, wherein the cloud server serves as the upper layer network; Based on the correspondence between the multiple audio lines, the audio windows on the multiple audio lines are corresponded to each other, and the cloud servers corresponding to the audio windows on the corresponding multiple audio lines are associated with each other; Process the audio lines one by one and transmit them to the upper network through the audio window to obtain the target audio line; Combining the target audio lines to obtain the target audio; The step of establishing the correspondence between the plurality of audio lines as the lower layer network comprises: Setting an audio frame, wherein the audio frame includes a main audio frame and a plurality of sub-audio frames, and setting a plurality of follow columns based on the audio frame, wherein the follow columns are respectively configured in the sub-audio frames, wherein the follow columns are a collection of a plurality of follow points, and there is a one-to-one correspondence between the follow columns and the sub-audio frames; Determine the audio to be processed, put the audio to be processed into the audio frame, and extract the type of sound in the audio to obtain human voice audio and environmental audio; The follow point marks the human voice audio and the ambient audio respectively according to the timbre, the follow points in the follow column mark the audio with the same timbre, the follow points in the same follow column are associated, and the human voice audio and the ambient audio are split based on the follow column to obtain multiple audio lines; Based on the follow-up point, the multiple audio lines are matched according to the audio playing time to obtain the corresponding relationship between the multiple audio lines as the lower layer network.

2. The audio processing method for controlling sound fields in different functional spaces according to claim 1, characterized in that: The following points in the following column mark the audio with the same timbre, and the step of associating the following points in the same following column comprises: Play human voice audio and environmental audio, and identify the sounds in the human voice audio and environmental audio based on the timbre to obtain multiple timbre; Make a one-to-one correspondence between the follow column and the sounds of multiple timbres in the audio, continue to play the human voice audio and the ambient audio, and mark the follow point in the follow column starting from the sound start point of the corresponding timbre based on the marking condition, and use the marked position on the sound of the corresponding timbre as the marked position, and bind the marked position to the follow point; By recording the playing time of the follow point at the marked position, the sound content between adjacent follow points is identified to obtain a single segment of information and stored together in the adjacent follow points; The single-segment information is encoded and assigned to the corresponding follow-up point to obtain a follow-up point label, and the follow-up points are sorted and associated according to the follow-up point label.

3. The audio processing method for controlling sound fields in different functional spaces according to claim 1, characterized in that: The steps of respectively setting the same number of audio windows for the corresponding multiple audio lines, associating adjacent audio windows in a single audio line with each other, and communicating and connecting the audio windows with the corresponding cloud server based on an upper layer network include: Based on the playback time of the audio to be processed, multiple audio lines are virtually extended to correspond to each other, the same number of division windows are set on the multiple virtually extended audio lines, and the division windows on the single virtually extended audio line are in a connected state; Acquire the position of the divided window and the follow point tag inside the divided window, and establish the position information between the divided window and the follow point inside the divided window based on the follow point tag, wherein the position of the divided window is the playback time segment of the audio line defined by the divided window, and the position information is the time position relationship between the playback time of the follow point inside the divided window and the playback time segment; Using the position information corresponding to the divided window and the position of the divided window as the audio window; Adjacent audio windows in a single audio line are associated with each other, and the audio windows are communicatively connected with the corresponding cloud server based on the upper layer network.

4. The audio processing method for controlling sound fields in different functional spaces according to claim 1, characterized in that: The step of corresponding the audio windows on the multiple audio lines based on the corresponding relationship between the multiple audio lines, and associating the cloud servers corresponding to the audio windows on the corresponding multiple audio lines, includes: The cloud servers corresponding to the audio windows in a single audio line are connected through a protocol to obtain a first-level service chain; The audio windows at the same divided window locations on multiple audio lines are matched as chained audio windows, and the cloud servers corresponding to the chained audio windows are connected and associated through a protocol to obtain a secondary service chain.

5. The audio processing method for controlling sound fields in different functional spaces according to claim 1, characterized in that: The step of processing the audio lines one by one and transmitting them to the upper layer network through the audio window to obtain the target audio line includes: Formulate a corresponding adjustment plan, and process the audio lines one by one based on the adjustment plan to obtain the processed audio lines; The processed audio line is transmitted to the corresponding cloud server based on the audio window, and the processed audio line is used as an additional audio line and stored in the primary service chain and the secondary service chain in the corresponding upper network; The additional audio line is copied and separated from the upper network, and a communication relationship is established between the follow-up point in the additional audio line and the follow-up point in the corresponding audio line in the cloud server to obtain the target audio line.

6. The audio processing method for controlling sound fields in different functional spaces according to claim 1, characterized in that: The step of combining target audio lines to obtain target audio includes: Temporarily combining the target audio lines based on the play time to obtain temporary audio; Bind all follow points in the temporary audio to get the target audio.

7. An audio processing system suitable for controlling sound fields in different functional spaces, used to implement the audio processing method suitable for controlling sound fields in different functional spaces as claimed in any one of claims 1 to 6, characterized in that: include: An acquisition module is used to acquire the audio to be processed, extract information from the audio to be processed to obtain multiple audio lines, and establish a corresponding relationship between the multiple audio lines as a lower layer network; The acquisition module comprises: Setting an audio frame, wherein the audio frame includes a main audio frame and a plurality of sub-audio frames, and setting a plurality of follow columns based on the audio frame, wherein the follow columns are respectively configured in the sub-audio frames, wherein the follow columns are a collection of a plurality of follow points, and there is a one-to-one correspondence between the follow columns and the sub-audio frames; Determine the audio to be processed, put the audio to be processed into the audio frame, and extract the type of sound in the audio to obtain human voice audio and environmental audio; The follow point marks the human voice audio and the ambient audio respectively according to the timbre, the follow points in the follow column mark the audio with the same timbre, the follow points in the same follow column are associated, and the human voice audio and the ambient audio are split based on the follow column to obtain multiple audio lines; Based on the follow point, the multiple audio lines are matched according to the audio playing time to obtain the corresponding relationship between the multiple audio lines as the lower layer network; A setting module, connected to the acquisition module, for setting the same number of audio windows for the multiple audio lines, respectively, associating adjacent audio windows in a single audio line with each other, and communicating and connecting the audio windows with the corresponding cloud server based on an upper layer network, wherein the cloud server serves as the upper layer network; A corresponding module, connected to the setting module, for corresponding the audio windows on the multiple audio lines based on the corresponding relationship between the multiple audio lines, and associating the cloud servers corresponding to the audio windows on the corresponding multiple audio lines; The processing module is connected to the corresponding module and is used to process the audio lines one by one and transmit them to the upper network through the audio window to obtain the target audio line; The combining module is connected to the processing module and is used for combining the target audio lines to obtain the target audio.

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