Multi-mode wireless audio control method and device, computer equipment and medium

Abstract

The invention relates to a multi-mode wireless audio control method and device, computer equipment and a medium, and the method comprises the steps: synchronously collecting an original voice signal and monitoring audio data through a wireless microphone, and carrying out the processing of the original voice signal and the monitoring audio data based on a split-track coding mode, generating first audio data for local recording and second audio data for monitoring; carrying out frame-level comparison on the first audio data and the second audio data, executing real-time difference analysis, and carrying out differential compression on a superposed part to generate a dual-channel compressed audio data packet; analyzing the two-channel compressed audio data packet, and then sending the two-channel compressed audio data packet to a wireless monitoring earphone; and the wireless monitoring earphone collects audio receiving quality parameters in real time and generates receiving feedback information, further returns the receiving feedback information to the microphone receiver, and dynamically adjusts configuration information of the UHF communication link according to the receiving feedback information. The method has the effect of improving the monitoring stability and recording consistency of the wireless audio system.

Description

Technical Field

[0001] The present application relates to the technical field of wireless audio transmission and monitoring processing, and in particular to a multi-mode wireless audio control method, apparatus, computer equipment, and medium. Background Art

[0002] At present, wireless microphone systems are widely used in various scenarios that require high-quality voice capture, such as performance recording, live interviews, and video shooting. Among them, the real-time audio feedback capability of monitoring headphones is one of the key factors to ensure recording quality and operational adjustments.

[0003] Existing wireless audio systems usually use microphones to collect voice and then transmit it to a local recording module and monitoring headphones respectively. The recording module is mainly responsible for saving the complete audio, while the monitoring headphones receive real-time audio signals through an independent transmission path. In this structure, the monitoring link and the recording link are often completely separated in terms of processing flow and encoding method. Although this design achieves parallel transmission, the lack of a unified processing mechanism for the two audio channels in terms of timing, frame structure, and compression method leads to problems such as uncontrollable delays, content asynchrony, and inconsistent playback experience in the monitoring process. In addition, existing monitoring headphones generally do not have the ability to feedback link quality, which makes the system unable to automatically identify and adjust when facing interference or link quality deterioration, which is prone to monitoring interruptions and sound quality degradation.

[0004] The above-mentioned existing technical solutions have the following defects: the existing monitoring link and recording link processing are separated, and there is a lack of feedback loop and synchronization control mechanism, which leads to unstable monitoring experience and poor consistency of recording restoration. Therefore, there is room for improvement. Summary of the Invention

[0005] In order to improve the monitoring stability and recording consistency of a wireless audio system, the present application provides a multi-mode wireless audio control method, apparatus, computer equipment and medium.

[0006] The above-mentioned invention objective of this application is achieved through the following technical solutions: A multi-mode wireless audio control method, the method comprising: Establishing a UHF communication link between the wireless microphone and the microphone receiver, and a low-latency monitoring link between the microphone receiver and the wireless monitoring headset; Synchronously collecting an original voice signal and monitored audio data through the wireless microphone, and processing the original voice signal and the monitored audio data based on a track encoding method to generate first audio data for local recording and second audio data for monitoring; performing frame-level comparison on the first audio data and the second audio data, performing real-time difference analysis, and performing differential compression on the overlapping portions to generate a dual-channel compressed audio data packet; Sending the dual-channel compressed audio data packet to the microphone receiver via the UHF communication link for parsing, and then sending the parsed monitoring portion to the wireless monitoring headset via the low-latency monitoring link; The wireless monitoring headset collects audio reception quality parameters in real time and generates reception feedback information, and then returns the reception feedback information to the microphone receiver to dynamically adjust the configuration information of the UHF communication link according to the reception feedback information.

[0007] By adopting the above technical solution, by establishing a UHF communication link between the wireless microphone and the microphone receiver, and a low-latency monitoring link between the microphone receiver and the wireless monitoring headphones, high-speed intercommunication of audio data between the microphone and the headphones is achieved, avoiding the link fragmentation and data asynchrony problems in traditional monitoring systems, thereby improving monitoring synchronization and transmission stability. By synchronously collecting the original voice signal and the monitoring audio data and performing track-by-track encoding processing, the audio content used for local recording and monitoring can be effectively distinguished, avoiding mutual interference between the two audio processing flows, thereby improving local audio quality and enhancing monitoring controllability. By performing frame-level comparison of the recorded audio data and the monitoring audio data and performing real-time difference analysis and overlap compression, redundant content can be effectively removed and the data transmission volume can be compressed, thereby reducing communication load and saving bandwidth resources. By sending dual-channel compressed data to the receiver and transmitting the monitoring portion via the monitoring link, while the monitoring headphones collect reception quality parameters and generate feedback information, the UHF link status can be dynamically evaluated and parameter adjusted, thereby improving the audio link's anti-interference capability and monitoring continuity.

[0008] In one example, the present application may be further configured as follows: processing the original voice signal and the monitoring audio data based on the track encoding method to generate the first audio data for local recording and the second audio data for monitoring specifically includes: Obtaining the original sample streams of the original voice signal and the monitored audio data, and performing bandpass filtering and noise suppression processing on the original sample streams; Performing high-precision linear encoding on the processed original sample stream, dividing it into a plurality of main channel audio frames according to a preset frame structure, wherein the main channel audio frames constitute the first audio data for local recording; Time domain compression and bandwidth limitation processing are performed on the processed original sampling stream, and a rate adaptive encoding method is used to generate monitoring channel audio frames, and the monitoring channel audio frames constitute the second audio data used for monitoring.

[0009] By adopting the above technical solution, by obtaining the original sampling stream of the original voice signal and the monitoring audio data, and performing bandpass filtering and noise suppression processing, the signal-to-noise ratio of the audio signal can be effectively improved, and environmental interference and unnecessary frequency bands can be filtered out, thereby ensuring the basic quality of subsequent encoded data; by performing high-precision linear encoding on the processed sampling stream and dividing the main channel audio frame, it can be ensured that the audio data used for recording has high fidelity and structural clarity, thereby improving the efficiency of later restoration and editing; by performing time domain compression and bandwidth limitation processing on the processed sampling stream and adopting rate adaptive coding, it is possible to generate audio frames that meet the real-time transmission characteristics of the monitoring channel, thereby ensuring low transmission delay while maintaining an acceptable range of sound quality, thereby improving the monitoring experience.

[0010] In one example, the present application may be further configured as follows: performing frame-level comparison on the first audio data and the second audio data, performing real-time difference analysis, and performing differential compression on the overlapping portions to generate a dual-channel compressed audio data packet, specifically including: Extracting a synchronization identification field from each frame of the first audio data and the second audio data, and pairing the main channel audio frame with the monitoring channel audio frame one by one based on the synchronization identification field; Calculating the difference of frame content for each pair of paired frames to obtain a difference score for the corresponding frames, and generating a frame difference marking table based on the difference score; performing a frame content replacement operation on the monitoring channel audio frames whose difference is lower than a preset threshold according to the frame difference marking table, thereby generating a monitoring channel differential frame set; The monitoring channel differential frame set and the main channel audio frame set in the first audio data are arranged in time sequence to generate the dual-channel compressed audio data packet.

[0011] By adopting the above technical solution, by extracting the synchronization identification fields in the audio frames of the main channel and the monitoring channel and pairing them frame by frame, a structured correspondence can be established, providing a data basis for subsequent difference analysis, thereby ensuring that the compression analysis is based on content correspondence rather than rough time matching; by calculating the difference scores of the paired frames and generating a frame difference marking table, repeated and redundant content can be accurately identified, thereby achieving targeted differential compression and improving compression efficiency; by replacing low-difference frames and generating a monitoring channel differential frame set, and then organizing this frame set and the main channel frame set into a dual-channel compressed data packet, the monitoring data transmission volume can be greatly reduced while maintaining content synchronization, thereby improving the overall data processing efficiency and communication adaptability of the system.

[0012] In one example, the present application can be further configured as follows: the dual-channel compressed audio data packet includes a channel identification field, a frame index mapping table and a time synchronization information field, and the time synchronization information field is used to restore the playback synchronization rhythm of the monitoring channel and the main channel audio frame in the receiver.

[0013] By adopting the above technical solution, by introducing the channel identification field, frame index mapping table and time synchronization information field in the dual-channel compressed audio data packet, it is possible to achieve a clear structural division of the monitoring frame and the main channel frame, establish a reference relationship and control the playback timing, thereby supporting the receiver end to accurately restore the consistent rhythm of the monitoring content and the main channel content, avoid the audio playback misalignment problem, and enhance the ability to quickly reconstruct the differential frames in the transmission process, thereby improving the coherence and accuracy of the monitoring audio.

[0014] In one example, the present application may be further configured as follows: the real-time collection of audio reception quality parameters and the generation of reception feedback information, and then the return of the reception feedback information to the microphone receiver, so as to dynamically adjust the configuration information of the UHF communication link according to the reception feedback information, specifically including: Parsing the received monitoring channel audio frames in each receiving cycle, and synchronously recording the signal strength value, packet loss count value and audio decoding delay duration in the current cycle; Based on the parameter recording results of multiple consecutive receiving cycles, a sliding time window is constructed, and a fitting calculation is performed on the signal strength change trend and the packet loss density change trend to obtain a reception quality evaluation vector, and then the reception feedback information is generated and sent to the microphone receiver; Analyze the various indicator parameters in the received feedback information, and dynamically adjust the transmission power, level tolerance limit value and frequency hopping channel distribution of the UHF communication link according to the preset link optimization strategy table.

[0015] By adopting the above technical solution, by parsing the monitoring channel audio frames in each receiving cycle and synchronously recording the signal strength, packet loss count and decoding delay, the transmission quality performance of the monitoring link in the time domain can be comprehensively monitored, thereby providing a real parameter basis for subsequent dynamic tuning; by constructing a sliding window based on multiple consecutive cycles and fitting the signal and packet loss change trends to generate a quality assessment vector, short-term anomalies can be combined with long-term trends to make stability judgments, thereby avoiding excessive parameter adjustment caused by occasional fluctuations; by sending the evaluation results to the microphone receiver and the receiver dynamically adjusting the UHF link configuration accordingly, it is possible to achieve optimal control of transmission power, fault tolerance settings and frequency hopping scheduling based on real-time link conditions, thereby significantly improving communication robustness and monitoring stability.

[0016] In one example, the present application may be further configured as follows: dynamically adjusting the transmit power, level tolerance limit value, and frequency hopping channel distribution of the UHF communication link according to a preset link tuning strategy table, specifically including: Performing an indicator weight score on the received feedback information, constructing a link abnormality level identifier based on the signal strength weight, packet loss rate weight, and delay fluctuation weight, and determining the current UHF link tuning priority according to the abnormality level identifier; When the abnormality level is medium, the transmit power gain adjustment operation is performed first, and the transmit power is gradually increased based on the preset safe power level difference within the current frequency until the signal strength returns to the target threshold or reaches the transmit limit boundary; When the abnormality level is marked as high, the fault tolerance strategy adjustment process is started to reconfigure the redundancy ratio of the frame-level error correction coding and the data frame tolerance error threshold to enhance the robustness against high packet loss and sudden interference.

[0017] By adopting the above technical solution, by weighting the three indicators in the received feedback information and constructing a link anomaly level identification, it is possible to achieve quantitative judgment of the link status under different interference situations, thereby providing an executable basis for link parameter adjustment decisions; when the anomaly level is medium, the transmission power gain adjustment operation is performed, which can improve the signal coverage quality without changing the frequency point, thereby quickly repairing the signal weakening problem caused by insufficient power; when the anomaly level is high, the fault tolerance strategy adjustment process is started, and the number of error correction bits and frame tolerance threshold are reconfigured, which can improve the data frame survival rate in strong interference scenarios, thereby ensuring the sustainable transmission and minimum quality assurance of monitoring audio in complex environments.

[0018] The second object of the present invention is achieved through the following technical solutions: A multi-mode wireless audio control device, comprising: A link establishment module for establishing a UHF communication link between the wireless microphone and the microphone receiver, and a low-latency monitoring link between the microphone receiver and the wireless monitoring headset; an audio acquisition and encoding module, configured to synchronously acquire an original voice signal and monitored audio data through the wireless microphone, and process the original voice signal and the monitored audio data based on a track encoding method to generate first audio data for local recording and second audio data for monitoring; a difference analysis and compression module, configured to perform frame-level comparison on the first audio data and the second audio data, perform real-time difference analysis, and perform differential compression on the overlapping portions to generate a dual-channel compressed audio data packet; A data transmission and monitoring module, configured to send the dual-channel compressed audio data packet to the microphone receiver via the UHF communication link for parsing, and then send the parsed monitoring portion to the wireless monitoring headset via the low-latency monitoring link; A monitoring feedback adjustment module is used for the wireless monitoring headset to collect audio reception quality parameters in real time and generate reception feedback information, and then return the reception feedback information to the microphone receiver to dynamically adjust the configuration information of the UHF communication link according to the reception feedback information.

[0019] By adopting the above technical solution, by establishing a UHF communication link between the wireless microphone and the microphone receiver, and a low-latency monitoring link between the microphone receiver and the wireless monitoring headphones, high-speed intercommunication of audio data between the microphone and the headphones is achieved, avoiding the link fragmentation and data asynchrony problems in traditional monitoring systems, thereby improving monitoring synchronization and transmission stability. By synchronously collecting the original voice signal and the monitoring audio data and performing track-by-track encoding processing, the audio content used for local recording and monitoring can be effectively distinguished, avoiding mutual interference between the two audio processing flows, thereby improving local audio quality and enhancing monitoring controllability. By performing frame-level comparison of the recorded audio data and the monitoring audio data and performing real-time difference analysis and overlap compression, redundant content can be effectively removed and the data transmission volume can be compressed, thereby reducing communication load and saving bandwidth resources. By sending dual-channel compressed data to the receiver and transmitting the monitoring portion via the monitoring link, while the monitoring headphones collect reception quality parameters and generate feedback information, the UHF link status can be dynamically evaluated and parameter adjusted, thereby improving the audio link's anti-interference capability and monitoring continuity.

[0020] The third objective of this application is achieved through the following technical solutions: A computer device includes a memory, a processor, and a computer program stored in the memory and executable on the processor. When the processor executes the computer program, the steps of the multi-mode wireless audio control method are implemented.

[0021] The fourth objective of this application is achieved through the following technical solutions: A computer-readable storage medium stores a computer program, which implements the steps of the multi-mode wireless audio control method when executed by a processor.

[0022] In summary, this application has the following beneficial technical effects: 1. By establishing a UHF communication link between the wireless microphone and microphone receiver, and a low-latency monitoring link between the microphone receiver and wireless monitoring headphones, high-speed audio data exchange between the microphone and headphones is achieved, avoiding the link fragmentation and data asynchrony issues common in traditional monitoring systems, thereby improving monitoring synchronization and transmission stability. By synchronously collecting original voice signals and monitoring audio data and performing separate-track encoding, the audio content used for local recording and monitoring can be effectively distinguished, preventing interference between the two audio processing flows, thereby improving local audio quality and enhancing monitoring controllability. By performing frame-level comparison of recorded and monitored audio data and performing real-time difference analysis and overlap compression, redundant content can be effectively removed and data transmission volume can be compressed, reducing communication load and saving bandwidth resources. By sending dual-channel compressed data to the receiver and transmitting the monitoring portion via the monitoring link, while the monitoring headphones collect reception quality parameters and generate feedback information, the UHF link status can be dynamically assessed and parameter adjustments can be made, thereby improving the audio link's anti-interference capability and monitoring continuity. 2. By obtaining the original sampling streams of the original voice signal and the monitored audio data and performing bandpass filtering and noise suppression processing, the signal-to-noise ratio of the audio signal can be effectively improved, and environmental interference and unnecessary frequency bands can be filtered out, thereby ensuring the basic quality of the subsequent encoded data; by performing high-precision linear encoding on the processed sampling stream and dividing the main channel audio frames, the audio data used for recording can be ensured to have high fidelity and structural clarity, thereby improving the efficiency of later restoration and editing; by performing time domain compression and bandwidth limitation processing on the processed sampling stream and adopting rate adaptive coding, audio frames that meet the real-time transmission characteristics of the monitoring channel can be generated, thereby ensuring low transmission delay while maintaining acceptable sound quality and improving the monitoring experience; 3. By extracting the synchronization identification fields in the audio frames of the main channel and the monitoring channel and pairing them frame by frame, a structured correspondence can be established, providing a data basis for subsequent difference analysis, thereby ensuring that the compression analysis is based on content correspondence rather than rough time matching; by calculating the difference scores of the paired frames and generating a frame difference marker table, repeated and redundant content can be accurately identified, thereby achieving targeted differential compression and improving compression efficiency; by replacing low-difference frames and generating a monitoring channel differential frame set, and then organizing this frame set and the main channel frame set into a dual-channel compressed data packet, the monitoring data transmission volume can be greatly reduced while maintaining content synchronization, thereby improving the overall data processing efficiency and communication adaptability of the system. BRIEF DESCRIPTION OF THE DRAWINGS

[0023] Figure 1 This is a flow chart of a multi-mode wireless audio control method according to an embodiment of the present application; Figure 2This is a flowchart for implementing step S20 in the multi-mode wireless audio control method in one embodiment of the present application; Figure 3 This is a flowchart for implementing step S30 in the multi-mode wireless audio control method in one embodiment of the present application; Figure 4 This is a flowchart for implementing step S50 in the multi-mode wireless audio control method in one embodiment of the present application; Figure 5 This is a flowchart for implementing step S53 in the multi-mode wireless audio control method in one embodiment of the present application; Figure 6 This is a principle block diagram of a multi-mode wireless audio control device in one embodiment of the present application; Figure 7 It is a schematic diagram of a device in one embodiment of the present application. DETAILED DESCRIPTION

[0024] The present application is further described in detail below with reference to the accompanying drawings.

[0025] In one embodiment, if Figure 1 As shown, the present application discloses a multi-mode wireless audio control method, which specifically includes the following steps: S10: Establishing a UHF communication link between the wireless microphone and the microphone receiver, and a low-latency monitoring link between the microphone receiver and the wireless monitoring headset.

[0026] Specifically, when establishing a UHF communication link, the wireless microphone is controlled to read the preset working frequency band and device identification information from the internal configuration table during the startup phase, generate a synchronous handshake signal packet and transmit it on the target frequency, and the microphone receiver scans the available frequency list and returns a handshake response after detecting the corresponding identification information, completing the frequency pairing, address registration and initial synchronization frame mark setting of the two-way link; when establishing a low-latency monitoring link, the microphone receiver activates the monitoring channel broadcast mode and broadcasts the monitoring connection request. After receiving the matching broadcast, the monitoring headset completes the binding handshake and switches to the low-latency transmission protocol mode under the receiver's instructions to complete the monitoring link initialization, including rate negotiation, buffer depth setting and control channel mapping configuration.

[0027] S20: Synchronously collect the original voice signal and the monitoring audio data through the wireless microphone, and process the original voice signal and the monitoring audio data based on the track encoding method to generate first audio data for local recording and second audio data for monitoring.

[0028] Specifically, during the acquisition process, the main channel input interface and the monitoring channel input interface of the microphone are controlled to be enabled at the same time. The main channel input interface is connected to the microphone unit to collect voice signals, and the monitoring channel input interface is connected to the power amplifier post-stage to collect ear return signals. The two channels are sampled in real time by independent ADC channels and attached with a unified timestamp field. The two original sampling data frames are written into the buffer queue in sequence through the data cache management module, and the encoding process module is started to process the two data according to the track structure to generate dual-track structured audio data.

[0029] S30: performing frame-level comparison on the first audio data and the second audio data, executing real-time difference analysis, and performing differential compression on the overlapping parts to generate a dual-channel compressed audio data packet.

[0030] Specifically, when performing difference analysis on two channels of audio data, the frame header information and timestamp identifier of each frame in the first audio data and the second audio data are extracted, and a frame comparison relationship is established in chronological order. The frame-by-frame input comparison module performs content similarity calculation and outputs the difference score result. A judgment threshold for content overlap is set and the monitoring frames with a difference lower than the threshold are screened to perform inter-frame differential encoding processing. Other non-overlapping frames are encoded in the original form. Finally, the control data encapsulation module combines the monitoring audio frames after differential processing with the main channel audio frames into a mixed format data packet and generates a channel identification index table for marking channel attribution.

[0031] S40: Send the dual-channel compressed audio data packet to the microphone receiver via the UHF communication link for parsing, and then send the parsed monitoring part to the wireless monitoring headset via the low-latency monitoring link.

[0032] Specifically, before sending the data packet, the UHF communication interface is controlled to start the link status check and refresh the currently available channel information, the generated compressed audio data packets are loaded into the waiting queue in sequence, and the segment sequence identifier and check field are set, and the link transmission control unit packages and encrypts the packets frame by frame and transmits them to the microphone receiver. After decoding, the receiver separates the monitoring audio part from the compressed packet, writes it into the monitoring buffer, and then sends this part of the data to the monitoring headphone receiving buffer through a low-latency link before the next monitoring frame cycle arrives, and marks the synchronization identifier in the frame control field to facilitate subsequent headphone timing alignment and playback reconstruction.

[0033] S50: The wireless monitoring headset collects audio reception quality parameters in real time and generates reception feedback information, and then returns the reception feedback information to the microphone receiver to dynamically adjust the configuration information of the UHF communication link according to the reception feedback information.

[0034] Specifically, when receiving each frame of monitoring audio data, the monitoring headset records the current reception time, signal strength RSSI value, check frame error rate and frame parsing delay time, and uses a sliding window structure to count the reception quality parameters of a group of continuous frames in each feedback cycle. The reception quality evaluation result is constructed by combining the average level of historical frames and the change trend of the current cycle. The feedback information frame is then assembled through the control channel of the low-latency link, and the fields including the timestamp, the unique number of the monitoring headset, the current link status indication and three parameter indicator fields are written. The feedback information is sent before the next receiver sends a control frame confirmation. After being received by the receiver, it is written into the link optimization buffer pool and waits for the parameter adjustment processing process to be called.

[0035] In one embodiment, if Figure 2 As shown, in step S20, the original voice signal and the monitoring audio data are processed based on the track encoding method to generate the first audio data for local recording and the second audio data for monitoring, which specifically includes: S21: Acquire the original sampling stream of the original voice signal and the monitored audio data, and perform bandpass filtering and noise suppression processing on the original sampling stream.

[0036] Specifically, after audio sampling, each frame of original voice signal and monitoring signal data is first input into the bandpass filtering module respectively. The effective frequency range of 80Hz to 10kHz is set for the voice channel, and an IIR filter group is used to remove noise components outside the frequency band. At the same time, the monitoring signal is introduced into the dynamic noise suppression module to analyze the background sound energy change trend and adjust the attenuation parameter in real time, thereby suppressing environmental interference such as wind noise and current noise. The audio stream after filtering and noise reduction is pushed to the encoding processing module for subsequent encoding process.

[0037] S22: Perform high-precision linear encoding on the processed original sampling stream, and divide it into multiple main channel audio frames according to a preset frame structure. The main channel audio frames constitute the first audio data for local recording.

[0038] Specifically, the encoding controller is called to enable the 16-bit linear PCM encoding mode for the main channel voice sampling stream that has been filtered and denoised. Every 1024 sampling points are divided into a frame, and the frame header is accompanied by a timestamp, frame number and channel identification field. The main channel audio frame sequence is constructed and written into the first audio data buffer. Subsequently, it is arranged in sequence and uniformly output as a frame data set for local recording.

[0039] S23: Performing time domain compression and bandwidth limitation processing on the processed original sampling stream, and generating monitoring channel audio frames using a rate adaptive encoding method. The monitoring channel audio frames constitute the second audio data for monitoring.

[0040] Specifically, after the monitoring channel sampling stream is input into the compression engine, frame segmentation processing is performed and the dynamic coding rate calculation module is called. The appropriate coding gear is selected according to the current monitoring link bandwidth status and the remaining transmission frame length, and the compression operation is performed using ADPCM encoding or LPC encoding. At the same time, frame-level truncation and bit rate limiting strategies are added to the encoding process to ensure low latency requirements. Each compressed frame is marked with a channel number and frame sequence number and written into the monitoring channel output buffer to form the second audio data.

[0041] In one embodiment, if Figure 3 As shown, in step S30, the first audio data and the second audio data are compared at the frame level, a real-time difference analysis is performed, and the overlapping parts are differentially compressed to generate a dual-channel compressed audio data packet, which specifically includes: S31: Extracting the synchronization identification field of each frame in the first audio data and the second audio data, and pairing the main channel audio frame with the monitoring channel audio frame one by one based on the synchronization identification field.

[0042] Specifically, each frame of the first audio data and the second audio data is read, the synchronization timestamp and frame sequence number in the frame header are extracted, a frame index table is constructed, and a one-to-one mapping relationship between the main channel audio frame and the monitoring channel audio frame is established in chronological order. The successfully matched frame pairs are sequentially input into the comparison queue for subsequent difference calculation.

[0043] S32: Calculating the difference of frame content for each pair of paired frames to obtain difference scores for the corresponding frames, and generating a frame difference marking table based on the difference scores.

[0044] Specifically, the mean square error method is used to compare the amplitude differences of paired frames in the time domain, and the similarity coefficients between the spectral morphologies are calculated. The two indicators are combined to generate a difference score, which is compared with the set threshold. Frame pairs below the threshold are marked as "compressible", and frame pairs above the threshold are marked as "keep original frames". The scores and marking results of all frame pairs are written into the frame difference marking table for reference in subsequent compression processing.

[0045] S33: performing a frame content replacement operation on the monitoring channel audio frames whose difference is lower than a preset threshold according to the frame difference marking table, thereby generating a monitoring channel differential frame set.

[0046] Specifically, the listening channel frames marked as "compressible" in the frame difference mark table are traversed, the corresponding main channel frame numbers are extracted and their amplitude difference sequences are calculated, the difference vectors and the main channel frame numbers are encapsulated into a differential frame structure and the original listening frame is replaced, and the corresponding listening frames in the retained marks remain unchanged. Finally, a listening channel differential frame set consisting of a mixture of differential frames and original listening frames is constructed.

[0047] S34: Arrange the monitoring channel differential frame set and the main channel audio frame set in the first audio data in chronological order to generate a dual-channel compressed audio data packet.

[0048] Specifically, all frame data in the monitoring channel differential frame set and the main channel audio frame set are read, reordered according to the timestamp field and written alternately into the encoding buffer, and channel identification and frame classification information are appended before each frame. After all frames are sorted, a unified encapsulation operation is performed to generate a dual-channel compressed audio data packet containing a channel label area, a synchronization field area and a payload area.

[0049] In one embodiment, if Figure 4 As shown, in step S50, the audio reception quality parameters are collected in real time and reception feedback information is generated, and then the reception feedback information is returned to the microphone receiver to dynamically adjust the configuration information of the UHF communication link according to the reception feedback information, specifically including: S51: parsing the received monitoring channel audio frame in each receiving cycle, and synchronously recording the signal strength value, packet loss count value and audio decoding delay duration in the current cycle.

[0050] Specifically, each time the monitoring headset receives a new monitoring audio frame, it parses the frame header and reads its sequence number, timestamp, and verification status. It calculates the packet loss based on the deviation from the expected frame number, and measures the buffer processing time of the receiving end as a delay indicator. It uses the RSSI module to record the wireless signal strength when the current frame is received, and writes the three parameter records into the data table of the current feedback cycle for subsequent trend analysis.

[0051] S52: Based on the parameter recording results of multiple continuous receiving cycles, a sliding time window is constructed, and the signal strength change trend and the packet loss density change trend are fitted and calculated to obtain a reception quality evaluation vector, and then the reception feedback information is generated and sent to the microphone receiver.

[0052] Specifically, a sliding time window of fixed size is set to perform linear fitting on the signal strength and packet loss rate sequences recorded in the most recent receiving cycles, and its first-order derivative is calculated as the trend parameter. At the same time, a three-dimensional reception quality evaluation vector is constructed in combination with the average delay value. The evaluation result, the current headset identification information, and the current timestamp are encoded into a feedback information packet and sent to the microphone receiver through the control channel of the monitoring link for the tuning module to call.

[0053] S53: Analyze various indicator parameters in the received feedback information, and dynamically adjust the transmission power, level tolerance limit value and frequency hopping channel distribution of the UHF communication link according to the preset link optimization strategy table.

[0054] Specifically, after receiving the feedback information, the microphone receiver parses the signal strength change rate, packet loss density trend value and average delay index, assigns a scoring weight to each indicator according to the link tuning strategy table and calculates the total scoring result. Based on the scoring value, the current link adjustment priority and processing path are determined, and power adjustment, fault tolerance amplitude update or frequency hopping channel switching operations are selectively performed, and the new parameters are written into the link configuration register to update the transmission status of the current UHF communication link.

[0055] In one embodiment, if Figure 5 As shown, in step S53, the transmission power, level tolerance limit value and frequency hopping channel distribution of the UHF communication link are dynamically adjusted according to the preset link optimization strategy table, specifically including: S531: Perform indicator weight scoring on the received feedback information, construct a link abnormality level identifier based on the signal strength weight, packet loss rate weight, and delay fluctuation weight, and determine the current UHF link tuning priority based on the abnormality level identifier.

[0056] Specifically, the three parameter values ​​in the feedback information are read and a single score is calculated for each parameter according to the scoring rules set in the strategy table. The weighted sum is used as the total abnormality score value. According to the mapping relationship between the total score value and the level threshold, an identification result of "normal", "medium abnormality" or "high level abnormality" is generated and used as the judgment condition for the subsequent execution path.

[0057] S532: When the abnormality level is marked as medium, the gain adjustment operation of the transmission power is performed first, and the transmission power is gradually increased based on the preset safe power level difference within the current frequency point until the signal strength recovers to the target threshold or reaches the transmission limit boundary.

[0058] Specifically, the UHF power adjustment module is called to increase the transmission power according to the step curve without changing the frequency. The adjustment amplitude is provided by the level corresponding gain table in the strategy table. After each increase, an RSSI verification is performed to determine whether it has returned to the stable range. If two consecutive verifications meet the target signal strength, the power configuration is maintained and the adjustment process is exited.

[0059] S533: When the abnormality level is marked as high, the fault tolerance strategy adjustment process is started to reconfigure the redundancy ratio of the frame-level error correction coding and the data frame tolerance error threshold to enhance the robustness against high packet loss and sudden interference.

[0060] Specifically, according to the fault-tolerant configuration scheme corresponding to high-level anomalies in the strategy table, the number of frame error correction bits and the tolerance error limit of the encoding module are adjusted, the redundancy ratio is increased to the maximum tolerance range to increase the probability of data recovery, and the frame discard threshold is lowered to ensure that data can still alleviate playback jams and maintain monitoring continuity in weak signal conditions. The new fault-tolerant parameters take effect immediately from the next encoding cycle.

[0061] It should be understood that the size of the serial numbers of the steps in the above embodiments does not mean the order of execution. The execution order of each process should be determined by its function and internal logic, and should not constitute any limitation on the implementation process of the embodiments of this application.

[0062] In one embodiment, a multi-mode wireless audio control device is provided, which corresponds to the multi-mode wireless audio control method in the above embodiment. Figure 6 As shown, the multi-mode wireless audio control device includes a link establishment module, an audio acquisition and encoding module, a difference analysis and compression module, a data transmission and monitoring module, and a monitoring feedback adjustment module. The functional modules are described in detail as follows: A link establishment module for establishing a UHF communication link between the wireless microphone and the microphone receiver, and a low-latency monitoring link between the microphone receiver and the wireless monitoring headset; An audio acquisition and encoding module is used to synchronously acquire original voice signals and monitored audio data through a wireless microphone, and process the original voice signals and monitored audio data based on a track-by-track encoding method to generate first audio data for local recording and second audio data for monitoring; a difference analysis and compression module, configured to perform frame-level comparison on the first audio data and the second audio data, perform real-time difference analysis, and perform differential compression on the overlapping portions to generate a dual-channel compressed audio data packet; The data transmission and monitoring module is used to send the dual-channel compressed audio data packets to the microphone receiver through the UHF communication link for parsing, and then send the parsed monitoring part to the wireless monitoring headset through the low-latency monitoring link; The monitoring feedback adjustment module is used for the wireless monitoring headset to collect audio reception quality parameters in real time and generate reception feedback information, and then return the reception feedback information to the microphone receiver to dynamically adjust the configuration information of the UHF communication link according to the reception feedback information.

[0063] Optionally, the audio acquisition and encoding module specifically includes: The sampling preprocessing submodule is used to obtain the original sampling stream of the original voice signal and the monitored audio data, and perform bandpass filtering and noise suppression processing on the original sampling stream; A main channel encoding submodule, configured to perform high-precision linear encoding on the processed original sample stream, dividing it into a plurality of main channel audio frames according to a preset frame structure, wherein the main channel audio frames constitute first audio data for local recording; The monitoring channel encoding submodule is used to perform time domain compression and bandwidth limitation processing on the processed original sampling stream, and use a rate adaptive encoding method to generate monitoring channel audio frames, which constitute the second audio data for monitoring.

[0064] Optional, the difference analysis and compression module specifically includes: A frame pairing submodule, configured to extract the synchronization identification field of each frame in the first audio data and the second audio data, and pair the main channel audio frame with the monitoring channel audio frame one by one based on the synchronization identification field; A difference analysis submodule is used to calculate the difference of frame content for each pair of paired frames, obtain a difference score for the corresponding frames, and generate a frame difference marking table based on the difference score; The differential replacement submodule is used to perform a frame content replacement operation on the monitoring channel audio frames whose difference is lower than a preset threshold according to the frame difference mark table, thereby generating a monitoring channel differential frame set; The data encapsulation submodule is used to arrange the monitoring channel differential frame set and the main channel audio frame set in the first audio data in time sequence to generate a dual-channel compressed audio data packet.

[0065] Optionally, the monitoring feedback adjustment module specifically includes: The receiving parameter acquisition submodule is used to parse the received monitoring channel audio frames in each receiving cycle and synchronously record the signal strength value, packet loss count value and audio decoding delay time in the current cycle; The quality assessment submodule is used to construct a sliding time window based on the parameter recording results of multiple consecutive reception cycles, fit the signal strength change trend and the packet loss density change trend, obtain the reception quality assessment vector, and then generate reception feedback information and send it to the microphone receiver; The link parameter adjustment submodule is used to analyze the various indicator parameters in the received feedback information and dynamically adjust the transmission power, level tolerance limit value and frequency hopping channel distribution of the UHF communication link according to the preset link tuning strategy table.

[0066] Optionally, the link parameter adjustment submodule specifically includes: The anomaly assessment unit is used to score the received feedback information by indicator weight, construct a link anomaly level identifier based on the signal strength weight, packet loss rate weight and delay fluctuation weight, and determine the current UHF link tuning priority based on the anomaly level identifier; The power adjustment unit is used to prioritize the transmit power gain adjustment operation when the abnormality level is marked as medium, and gradually increase the transmit power based on the preset safe power level difference within the current frequency until the signal strength returns to the target threshold or reaches the transmission limit boundary; The fault-tolerance optimization unit is used to start the fault-tolerance strategy adjustment process when the abnormality level is marked as high, reconfigure the redundancy ratio of the frame-level error correction coding and the data frame tolerance error threshold to enhance the robustness against high packet loss and sudden interference.

[0067] For the specific limitations of the multi-mode wireless audio control device, please refer to the limitations of the multi-mode wireless audio control method above and will not be repeated here. The various modules in the above-mentioned multi-mode wireless audio control device can be implemented in whole or in part through software, hardware, or a combination thereof. The above-mentioned modules can be embedded in or independent of the processor in the computer device in hardware form, or can be stored in the memory of the computer device in software form, so that the processor can call and execute the corresponding operations of the above-mentioned modules.

[0068] In one embodiment, a computer device is provided. The computer device may be a server, and its internal structure diagram may be as follows: Figure 7 As shown. The computer device includes a processor, memory, network interface, and database connected via a system bus. The processor of the computer device is used to provide computing and control capabilities. The memory of the computer device includes a non-volatile storage medium and an internal memory. The non-volatile storage medium stores an operating system, a computer program, and a database. The internal memory provides an environment for the operation of the operating system and computer program in the non-volatile storage medium. The network interface of the computer device is used to communicate with an external terminal via a network connection. When the computer program is executed by the processor, it implements a multi-mode wireless audio control method.

[0069] In one embodiment, a computer device is provided, including a memory, a processor, and a computer program stored in the memory and executable on the processor. When the processor executes the computer program, the following steps are performed: Establishing a UHF communication link between the wireless microphone and the microphone receiver, and a low-latency monitoring link between the microphone receiver and the wireless monitoring headset; Synchronously collecting original voice signals and monitored audio data through a wireless microphone, and processing the original voice signals and monitored audio data based on a track encoding method to generate first audio data for local recording and second audio data for monitoring; Performing frame-level comparison on the first audio data and the second audio data, performing real-time difference analysis, and performing differential compression on the overlapping parts to generate a dual-channel compressed audio data packet; The dual-channel compressed audio data packet is sent to the microphone receiver via the UHF communication link for parsing, and the parsed monitoring part is then sent to the wireless monitoring headphones via the low-latency monitoring link; The wireless monitoring headset collects audio reception quality parameters in real time and generates reception feedback information, which is then returned to the microphone receiver to dynamically adjust the configuration information of the UHF communication link according to the reception feedback information.

[0070] In one embodiment, a computer-readable storage medium is provided, on which a computer program is stored. When the computer program is executed by a processor, the following steps are implemented: Establishing a UHF communication link between the wireless microphone and the microphone receiver, and a low-latency monitoring link between the microphone receiver and the wireless monitoring headset; Synchronously collecting original voice signals and monitored audio data through a wireless microphone, and processing the original voice signals and monitored audio data based on a track encoding method to generate first audio data for local recording and second audio data for monitoring; Performing frame-level comparison on the first audio data and the second audio data, performing real-time difference analysis, and performing differential compression on the overlapping parts to generate a dual-channel compressed audio data packet; The dual-channel compressed audio data packet is sent to the microphone receiver via the UHF communication link for parsing, and the parsed monitoring part is then sent to the wireless monitoring headphones via the low-latency monitoring link; The wireless monitoring headset collects audio reception quality parameters in real time and generates reception feedback information, which is then returned to the microphone receiver to dynamically adjust the configuration information of the UHF communication link according to the reception feedback information.

[0071] Those skilled in the art will appreciate that all or part of the processes in the above-described method embodiments can be implemented by instructing the relevant hardware through a computer program. The computer program can be stored in a non-volatile computer-readable storage medium. When executed, the computer program can include the processes of the above-described method embodiments. Any reference to memory, storage, database, or other media used in the various embodiments provided herein may include non-volatile and / or volatile memory. Non-volatile memory may include read-only memory (ROM), programmable ROM (PROM), electrically programmable ROM (EPROM), electrically erasable programmable ROM (EEPROM), or flash memory. Volatile memory may include random access memory (RAM) or external cache memory. By way of illustration and not limitation, RAM is available in various forms, such as static RAM (SRAM), dynamic RAM (DRAM), synchronous DRAM (SDRAM), double data rate SDRAM (DDRSDRAM), enhanced SDRAM (ESDRAM), synchronous link DRAM (SLDRAM), RAMbus direct RAM (RDRAM), direct RAMbus dynamic RAM (DRDRAM), and RAMbus dynamic RAM (RDRAM).

[0072] Those skilled in the art will clearly understand that for the sake of convenience and brevity of description, only the division of the above-mentioned functional units and modules is used as an example. In actual applications, the above-mentioned functions can be distributed and completed by different functional units and modules as needed, that is, the internal structure of the device can be divided into different functional units or modules to complete all or part of the functions described above.

[0073] The above-described embodiments are only used to illustrate the technical solutions of the present application, rather than to limit them. Although the present application has been described in detail with reference to the aforementioned embodiments, those skilled in the art should understand that they can still modify the technical solutions described in the aforementioned embodiments, or make equivalent replacements for some of the technical features therein. These modifications or replacements do not deviate the essence of the corresponding technical solutions from the spirit and scope of the technical solutions of the various embodiments of the present application, and should all be included in the scope of protection of the present application.

Claims

1. A multi-mode wireless audio control method, characterized in that: The method comprises: Establishing a UHF communication link between the wireless microphone and the microphone receiver, and a low-latency monitoring link between the microphone receiver and the wireless monitoring headset; Synchronously collecting an original voice signal and monitored audio data through the wireless microphone, and processing the original voice signal and the monitored audio data based on a track encoding method to generate first audio data for local recording and second audio data for monitoring; performing frame-level comparison on the first audio data and the second audio data, performing real-time difference analysis, and performing differential compression on the overlapping portions to generate a dual-channel compressed audio data packet; Sending the dual-channel compressed audio data packet to the microphone receiver via the UHF communication link for parsing, and then sending the parsed monitoring portion to the wireless monitoring headset via the low-latency monitoring link; The wireless monitoring headset collects audio reception quality parameters in real time and generates reception feedback information, and then returns the reception feedback information to the microphone receiver to dynamically adjust the configuration information of the UHF communication link according to the reception feedback information.

2. The multi-mode wireless audio control method according to claim 1, wherein: The processing of the original voice signal and the monitoring audio data based on the track encoding method to generate first audio data for local recording and second audio data for monitoring specifically includes: Obtaining the original sample streams of the original voice signal and the monitored audio data, and performing bandpass filtering and noise suppression processing on the original sample streams; Performing high-precision linear encoding on the processed original sample stream, dividing it into a plurality of main channel audio frames according to a preset frame structure, wherein the main channel audio frames constitute the first audio data for local recording; Time domain compression and bandwidth limitation processing are performed on the processed original sampling stream, and a rate adaptive encoding method is used to generate monitoring channel audio frames, and the monitoring channel audio frames constitute the second audio data used for monitoring.

3. The multi-mode wireless audio control method according to claim 2, wherein: The performing frame-level comparison on the first audio data and the second audio data, performing real-time difference analysis, and performing differential compression on the overlapping parts to generate a dual-channel compressed audio data packet specifically includes: Extracting a synchronization identification field from each frame of the first audio data and the second audio data, and pairing the main channel audio frame with the monitoring channel audio frame one by one based on the synchronization identification field; Calculating the difference of frame content for each pair of paired frames to obtain a difference score for the corresponding frames, and generating a frame difference marking table based on the difference score; performing a frame content replacement operation on the monitoring channel audio frames whose difference is lower than a preset threshold according to the frame difference marking table, thereby generating a monitoring channel differential frame set; The monitoring channel differential frame set and the main channel audio frame set in the first audio data are arranged in time sequence to generate the dual-channel compressed audio data packet.

4. The multi-mode wireless audio control method according to claim 3, wherein: The dual-channel compressed audio data packet includes a channel identification field, a frame index mapping table and a time synchronization information field. The time synchronization information field is used to restore the playback synchronization rhythm of the monitoring channel and the main channel audio frame in the receiver.

5. The multi-mode wireless audio control method according to claim 2, wherein: The real-time collection of audio reception quality parameters and generation of reception feedback information, and then returning the reception feedback information to the microphone receiver to dynamically adjust the configuration information of the UHF communication link according to the reception feedback information, specifically includes: Parsing the received monitoring channel audio frames in each receiving cycle, and synchronously recording the signal strength value, packet loss count value and audio decoding delay duration in the current cycle; Based on the parameter recording results of multiple consecutive receiving cycles, a sliding time window is constructed, and a fitting calculation is performed on the signal strength change trend and the packet loss density change trend to obtain a reception quality evaluation vector, and then the reception feedback information is generated and sent to the microphone receiver; Analyze the various indicator parameters in the received feedback information, and dynamically adjust the transmission power, level tolerance limit value and frequency hopping channel distribution of the UHF communication link according to the preset link optimization strategy table.

6. The multi-mode wireless audio control method according to claim 5, characterized in that: The method of dynamically adjusting the transmit power, level tolerance limit value, and frequency hopping channel distribution of the UHF communication link according to the preset link optimization strategy table specifically includes: Performing an indicator weight score on the received feedback information, constructing a link abnormality level identifier based on the signal strength weight, packet loss rate weight, and delay fluctuation weight, and determining the current UHF link tuning priority according to the abnormality level identifier; When the abnormality level is medium, the transmit power gain adjustment operation is performed first, and the transmit power is gradually increased based on the preset safe power level difference within the current frequency until the signal strength returns to the target threshold or reaches the transmit limit boundary; When the abnormality level is marked as high, the fault tolerance strategy adjustment process is started to reconfigure the redundancy ratio of the frame-level error correction coding and the data frame tolerance error threshold to enhance the robustness against high packet loss and sudden interference.

7. A multi-mode wireless audio control device, characterized in that: The device comprises: A link establishment module for establishing a UHF communication link between the wireless microphone and the microphone receiver, and a low-latency monitoring link between the microphone receiver and the wireless monitoring headset; an audio acquisition and encoding module, configured to synchronously acquire an original voice signal and monitored audio data through the wireless microphone, and process the original voice signal and the monitored audio data based on a track encoding method to generate first audio data for local recording and second audio data for monitoring; a difference analysis and compression module, configured to perform frame-level comparison on the first audio data and the second audio data, perform real-time difference analysis, and perform differential compression on the overlapping portions to generate a dual-channel compressed audio data packet; A data transmission and monitoring module, configured to send the dual-channel compressed audio data packet to the microphone receiver via the UHF communication link for parsing, and then send the parsed monitoring portion to the wireless monitoring headset via the low-latency monitoring link; A monitoring feedback adjustment module is used for the wireless monitoring headset to collect audio reception quality parameters in real time and generate reception feedback information, and then return the reception feedback information to the microphone receiver to dynamically adjust the configuration information of the UHF communication link according to the reception feedback information.

8. The multi-mode wireless audio control device according to claim 7, characterized in that: The audio acquisition and encoding module specifically includes: a sampling preprocessing submodule, configured to obtain the original sampling streams of the original speech signal and the monitoring audio data, and perform bandpass filtering and noise suppression processing on the original sampling streams; a main channel encoding submodule, configured to perform high-precision linear encoding on the processed original sample stream, and divide the processed original sample stream into a plurality of main channel audio frames according to a preset frame structure, wherein the main channel audio frames constitute the first audio data for local recording; The monitoring channel encoding submodule is used to perform time domain compression and bandwidth limitation processing on the processed original sampling stream, and generate monitoring channel audio frames using a rate adaptive encoding method. The monitoring channel audio frames constitute the second audio data used for monitoring.

9. A computer device comprising a memory, a processor, and a computer program stored in the memory and executable on the processor, wherein: When the processor executes the computer program, the steps of the multi-mode wireless audio control method according to any one of claims 1 to 6 are implemented.

10. A computer-readable storage medium storing a computer program, characterized in that: When the computer program is executed by a processor, the steps of the multi-mode wireless audio control method according to any one of claims 1 to 6 are implemented.

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