Duplex voice talkback method and system

Abstract

The invention relates to the technical field of communication, in particular to a duplex voice talkback method and system, and the method comprises the steps: S1, collecting the audio data of a user, starting a voice talkback session, and configuring the audio configuration setting of a talkback; s2, judging whether the audio configuration setting adopts a state mode of an intercom or not; s3, determining a trigger signal of the intercom when starting the voice intercom session, and establishing full-duplex communication according to the radio frequency of the preset channel frequency interval; s4, processing the radio frequency of the intercom and the corresponding equipment according to the specific condition of the intercom during calling, and setting a communication channel according to the priority of communication; s5, verifying the full-duplex signals under the communication channels, verifying the communication condition in each communication channel, and completing the communication comparison of the full-duplex signals; and the communication efficiency, reliability and accuracy of the intercom are improved.

Description

Technical Field

[0001] The invention relates to the technical field of communications, and in particular to a duplex voice intercom method and system. Background Art

[0002] Although traditional full-duplex communication can send and receive signals at the same time, it is prone to self-interference when working in the same frequency band. Existing solutions usually rely on complex filter design and echo cancellation algorithms, but these methods often increase the complexity and cost of the system. At the same time, in a multi-user environment, how to effectively manage and allocate RF frequency resources is also a key issue. Different users may have different priority requirements. For example, emergency calls should be handled first, while daily business calls can be processed later. Existing systems usually lack a flexible priority scheduling mechanism, resulting in high-priority calls being blocked by low-priority calls, affecting communication efficiency and reliability.

[0003] For example, Chinese patent publication number CN109246626A discloses a voice intercom method, system and terminal. In the method, system and terminal, the server uses the FreeSWITCH module that can realize the full-duplex voice conference function as the platform of voice soft switch, and realizes the voice intercom in half-duplex mode by pre-configuring the dialing plan, SIP client access conference number, and canceling or restoring the SIP client mute.

[0004] The prior art describes how to implement intercom communication, but in a scenario where there are multiple conversations, it cannot be adjusted according to the situation and scenario of the conversations, resulting in conversation congestion and reducing the efficiency and quality of the conversations. Summary of the invention

[0005] In order to solve the above technical problems, the technical solution adopted by the present invention is: a duplex voice intercom method, comprising: S1, collecting audio data of the user, starting a voice intercom session, and configuring the audio configuration settings of the intercom.

[0006] S2, determining whether the audio configuration setting adopts the state mode of the intercom.

[0007] S3, determining a trigger signal for the intercom to start a voice intercom session, and establishing full-duplex communication according to a radio frequency in a preset channel frequency interval.

[0008] S4, according to the specific situation of the intercom when calling, the radio frequency frequency of the intercom and the corresponding device is processed, and the communication channel is set according to the priority of the communication.

[0009] S5, verifying the full-duplex signal under the communication channel, and verifying the communication status in each communication channel, and completing the communication comparison of the full-duplex signal.

[0010] A duplex voice intercom system includes: an audio data acquisition module for collecting user audio data and determining the strength and source distance of the audio data. If the audio data is normal and the distance is less than a first preset distance, the audio data is sent; otherwise, a trigger signal is adjusted and the audio data is re-modulated.

[0011] The state judgment configuration module is used to judge whether the device adopts a specific state mode and configure the transmission mode of the audio data according to the state mode.

[0012] The priority management module is used to determine the priority of the current dialing according to the dialing sequence and compare it with the call status and radio frequency; when the call status changes, the priority is dynamically adjusted.

[0013] The communication quality optimization module is used to calculate the signal strength, signal-to-noise ratio and spectrum occupancy rate under each communication channel, analyze the correlation between signal strength, signal-to-noise ratio and spectrum occupancy rate, output the first effect factor and the second effect factor, and complete the communication comparison.

[0014] The beneficial effects of the present invention are as follows: 1. The present invention can improve the quality of currently transmitted audio data by accurately collecting and processing audio data, obtaining the distance and trigger signal of the audio data, and processing according to the waveform generated by the audio data, and filter the noise existing in the audio data, so that the transmitted audio data maintains the corresponding intensity and frequency, ensuring that only high-quality audio data is sent, reducing noise and interference in communication; and enhancing the user's communication experience in complex environments.

[0015] 2. The present invention dynamically adjusts the radio frequency and priority management to ensure that the transmitted voice information can guarantee certain periodic characteristics, improves the quality of voice information, and dynamically adjusts the dialing priority to ensure that when the call status changes, it can respond quickly and reallocate communication resources, thereby improving the continuity and stability of communication; at the same time, by comparing the radio frequency of the intercom with that of the corresponding device, it ensures that the two parties of the call can be accurately connected, avoids communication interruption or misconnection, and ensures that key information can be transmitted in a timely and effective manner.

[0016] 3. The present invention can comprehensively evaluate the communication quality according to indicators such as signal strength, signal-to-noise ratio and spectrum occupancy, and output the first effect factor and the second effect factor, thereby providing an objective and accurate basis for communication comparison and helping to promptly discover and solve communication problems. BRIEF DESCRIPTION OF THE DRAWINGS

[0017] The present invention is further described below in conjunction with the accompanying drawings and embodiments.

[0018] Figure 1 The invention is a flowchart of a duplex voice intercom method.

[0019] Figure 2 The present invention is a flowchart of step S1 of a duplex voice intercom method.

[0020] Figure 3 The present invention is a flowchart of step S12 of a duplex voice intercom method.

[0021] Figure 4 The present invention is a flowchart of step S11 of a duplex voice intercom method.

[0022] Figure 5 The present invention is a flowchart of step S2 of a duplex voice intercom method.

[0023] Figure 6 The present invention is a flowchart of step S3 of a duplex voice intercom method.

[0024] Figure 7 The present invention is a flowchart of step S4 of a duplex voice intercom method.

[0025] Figure 8 The present invention is a flowchart of step S43 of the duplex voice intercom method.

[0026] Fig. 9 The present invention is a flowchart of step S5 of a duplex voice intercom method.

[0027] Fig.10 It is a system framework diagram of a duplex voice intercom system. DETAILED DESCRIPTION

[0028] The embodiments of the present invention are described in detail below. The embodiments described below are exemplary and are only used to explain the present invention, and should not be construed as limiting the present invention. If no specific techniques or conditions are specified in the embodiments, the techniques or conditions described in the literature in the art or the product specifications are used.

[0029] See also Figure 1 , a duplex voice intercom method, comprising: S1, collecting audio data of a user, starting a voice intercom session, and configuring audio configuration settings of an intercom.

[0030] S2, determining whether the audio configuration setting adopts the state mode of the intercom.

[0031] S3, determining a trigger signal for the intercom to start a voice intercom session, and establishing full-duplex communication according to a radio frequency in a preset channel frequency interval.

[0032] S4, according to the specific situation of the intercom when calling, the radio frequency frequency of the intercom and the corresponding device is processed, and the communication channel is set according to the priority of the communication.

[0033] S5, verifying the full-duplex signal under the communication channel, and verifying the communication status in each communication channel, and completing the communication comparison of the full-duplex signal.

[0034] In one embodiment of the present invention, when the intercom starts a voice intercom session, it collects the audio data of the current user, sends the audio data to the cloud server, automatically checks the audio data of the user, and configures the audio configuration settings of the intercom after the check is completed.

[0035] So when you start a voice intercom session, Figure 2 As shown, the implementation method of step S1 also includes: S11, obtaining audio data of the intercom within a first preset distance, and judging the strength of the audio data, and considering the judged audio data as a processing result of the first preset distance; the first preset distance represents the distance at which the current intercom normally makes a call, and this distance generally refers to the distance between the sound emitted by the current user and the intercom when the current user uses the intercom. This distance is generally set to within one meter, and one meter is used as the first preset distance, which represents the sound source distance of the user's normal audio. When this distance is exceeded, the audio received by the intercom may be the generated echo or the noise existing in the current scene; judging the strength of the audio data is to judge whether the current sound received by the user is normal. This will be compared with the current pronunciation according to the interval of the user's normal pronunciation when using the intercom to prevent the situation where the sound is small and affects the data transmission. When the strength of the audio data is greater than the minimum value of the interval of the user's normal pronunciation when using the intercom, the audio data of the current intercom within the first preset distance is regarded as normal data.

[0036] S12, when the processing result of the first preset distance is normal, determine whether there is audio data whose acquisition distance is greater than the first preset distance, and send the audio data after the determination is completed.

[0037] At this time, it is mainly determined whether the audio data at the first preset distance is received normally. Figure 3 As shown, step S12 may also include the following implementation method: S121, if there is audio data acquisition distance greater than the first preset distance, a trigger signal is set for each frame of audio data, and the waveform conformity of the trigger signal is calculated according to the waveform of the trigger signal; this indicates that the currently acquired audio data may contain noise, and the set trigger signal is used to extract audio features present in the audio data, such as amplitude, phase and period, and to determine whether the waveform of the trigger signal is the same as the waveform of the preset noise in the database, and the cosine similarity between the waveform of the trigger signal and the waveform of the preset noise is set as the waveform conformity of the trigger signal.

[0038] S122, modulate the user's audio data into target audio information according to the waveform conformity of the trigger signal, and send it according to the preset radio frequency. In this step, the purpose of the processing result of the first preset distance and the trigger signal is to distinguish the currently received audio data and output these data as target audio information after processing them according to the different sound source distances, and finally complete the sending of the current audio data.

[0039] In step S122, the waveform conformity of the trigger signal is adjusted to the target audio information by filtering the trigger signal in the audio data whose waveform conformity is greater than the preset waveform conformity, and adjusting the filtered audio data to the target audio information.

[0040] S123: If there is no audio data acquisition distance greater than the first preset distance, modulate the user's audio data into target audio information and send it according to a preset radio frequency frequency.

[0041] Regarding the judgment method of the audio distance of the current user and the first preset distance, the Doppler effect and positioning based on intensity difference can be used to judge the approximate distance of the current audio data; the principle of positioning based on intensity difference is: under ideal conditions, the intensity of sound waves decays according to the inverse square law with the propagation distance. By comparing the difference in sound intensity from different directions, the distance of the sound source can be roughly estimated; since this method is easily affected by factors such as environmental noise and reflection, the accuracy is low, and it is mainly used for approximate estimation to obtain an approximate sound source distance. This distance will be compared with the first preset distance to determine what processing method should be used for the current audio data. The principle of the Doppler effect is: if the sound source moves relative to the receiver, the received sound frequency will change, that is, the Doppler effect; by analyzing this change, the speed of the sound source and the relative distance change trend can be inferred. This method is suitable for dynamic scenes and requires accurate measurement of frequency changes.

[0042] The application of Doppler effect in intercom can be described using the following case.

[0043] At a large construction site, the intercom devices worn by workers are not only used for daily communication, but also integrate Doppler effect analysis to monitor the speed of surrounding moving machinery. When heavy machinery (such as excavators or bulldozers) approaches, the system can determine the approaching speed of these machines by analyzing the changes in the received sound frequency and issue a warning signal.

[0044] Corresponding descriptive data: Sound source speed: Assume that an excavator moves towards the worker at a speed of 5 m / s.

[0045] Speed ​​of sound: The speed of sound in air is about 340 m / s.

[0046] Frequency change: If the original sound frequency of the excavator is 1000Hz, then according to the Doppler effect formula f′=f×(v+v 0 ) / (vv s ), where f′ is the frequency heard by the observer, f is the frequency emitted by the sound source, v is the speed of sound, and v 0 is the velocity of the observer (0 in this case), v s is the speed of the sound source. Substituting this into the numerical calculation, we can get a new frequency of approximately 1014.7 Hz.

[0047] The algorithm built into the intercom recognizes the increase in frequency and calculates the approach speed of the excavator. Based on this approach speed, the relevant noise in the intercom can be found and filtered out. At this time, the server connected to the intercom will work together with nearby devices to improve the overall audio quality of the final communication.

[0048] The application of intensity difference-based positioning on intercoms can also be described using the following case.

[0049] In an outdoor scenario, multiple intercom devices are distributed in different locations. These devices can capture environmental sounds through built-in microphone arrays and use sound intensity differences to locate and enhance the voice of a specific speaker. When a speaker speaks, the system uses the received sound intensity differences to determine the speaker's approximate location and adjust the sound output to ensure that all participants can hear the speech clearly.

[0050] Reception strength: Assuming that everyone's handheld walkie-talkies are distributed throughout the area, each walkie-talkie is equivalent to a distributed microphone node.

[0051] Device A (front left): 60dB.

[0052] Device B (front right): 55dB.

[0053] Device C (left rear): 50dB.

[0054] Device D (right rear): 45dB.

[0055] By comparing the sound intensity received by each device, it is determined that the main direction of the speaker is the front left. Subsequently, the sound from this direction is enhanced, while the background noise from other directions is suppressed to ensure the best listening experience; this describes the scenario when the intercom is used in multiple close positions, and when the intercom emits different sound intensities, the intercom itself performs corresponding processing of enhancement and filtering.

[0056] These two methods demonstrate how to apply physical principles to practical scenarios to improve the functionality and user experience of walkie-talkies. Although the positioning accuracy based on intensity difference is limited, it can still provide valuable auxiliary information in certain specific environments; and the Doppler effect is particularly suitable for monitoring speed and distance changes in dynamic scenes. The combination of the two or in conjunction with other technologies can further improve the overall performance of walkie-talkies.

[0057] It should be noted that the above cases are only examples of intercom implementation in different environments, and the actual functions implemented are more than what is described above.

[0058] At this time, the Doppler effect and positioning based on intensity difference are used together to determine the corresponding distance of the audio data, such as Figure 4 As shown, the method for screening audio data within the first preset distance in step S11 also includes: S111, using the intercom to continuously monitor the user's audio data and record the frequency change of the audio data.

[0059] S112, for each intercom, measuring the intensity of the received audio data, and comparing the relative intensity change of the audio data; the relative intensity change refers to whether the audio data generates an excessive change in sound intensity for a single channel or sound after being received.

[0060] S113, compare the relative change in the intensity of the audio data with the preset intensity difference, and estimate the distance estimation value of the source of the audio data. Here, the distance estimation value obtains the sound intensity at the reference distance under normal reference conditions, and divides the sound intensity under the reference condition by the current sound intensity, then takes the square root and multiplies it by the reference distance to obtain a distance estimation value. The distance estimation value is mainly used to judge the size of the sound intensity to verify whether the current intercom is receiving audio data normally. At the same time, the reference distance is set using the average distance between the handheld intercom and the mouth when the user uses it normally.

[0061] S114, using the Doppler effect and combining the frequency change of the audio data, the distance estimation value is adjusted. When the adjusted distance estimation value is less than the first preset distance, the processing result of the first preset distance is determined to be normal.

[0062] To adjust the distance estimate, the frequency offset value of the frequency change is first calculated. This will calculate the difference between the currently received audio frequency and the preset audio frequency. The frequency offset value is then multiplied by the current user's movement speed and the time interval corresponding to the movement speed, and then divided by the preset audio frequency to obtain an adjustment value. The distance estimate is then subtracted from the adjustment value to obtain the adjusted distance estimate. This value is finally calculated to present a distance value, which is compared with the first preset distance to determine the main source of sound currently received by the walkie-talkie.

[0063] After completing the distance judgment of these audio data, the noise that occurs when the user is talking can be identified, and at the same time, it can be judged whether the current communication situation is good, so as to improve the communication quality of subsequent full-duplex communication.

[0064] In one embodiment of the present invention, Figure 5 As shown, step S2 is mainly used to determine whether the settings of the intercom are normal, and to communicate with the intercom that is set normally. Step S2 may include the following implementation methods: S21, if the state mode of the intercom is adopted, the user's audio data is sent to the corresponding device according to the audio configuration settings.

[0065] S22, if the state mode of the intercom is not adopted, the state mode of the intercom is verified, audio parameters are parsed from the user's audio data, and the audio parameters are sent to the corresponding device. The corresponding device here represents the device with which the current intercom is communicating, that is, the called intercom. The called intercom will check the audio configuration settings between the two intercoms in communication, determine whether the calling device is currently allowed to communicate, and establish communication after completing the verification.

[0066] State mode refers to a mechanism by which a device automatically adjusts its audio configuration parameters such as volume, frequency response, etc. according to its current operating mode or condition. This is usually based on the device's usage scenario, such as: mute, normal call, emergency call, etc., environmental noise level, battery power, etc. Audio configuration settings refer to the data set by the state mode by default, based on which the use of the intercom is controlled.

[0067] The status mode may include the following: working mode, audio quality setting, noise suppression and echo cancellation, power management strategy, ringtone and prompt tone setting, and network status adaptability. When using the intercom, these contents will be configured according to the parameters set in advance by the intercom. When starting a voice intercom session, if these status modes change to a certain extent, the actual audio data received and sent by a group of intercoms in communication will be different in a certain form after verification. At this time, the voice data needs to be adjusted according to the status mode of the intercom to timely reflect the corresponding communication situation, thereby completing the duplex voice intercom of the intercom.

[0068] Working mode: such as full-duplex and half-duplex mode selection.

[0069] Audio quality settings: including sampling rate, bit depth, etc., which affect sound clarity and file size.

[0070] Noise suppression and echo cancellation: Adjust the strength of these features based on the ambient noise level.

[0071] Power management policy: Adjust performance based on remaining battery power to extend usage time.

[0072] Volume Level: Adjust output volume based on ambient noise level or user preference.

[0073] Ringtone and reminder tone settings: Different states may correspond to different reminder sound effects.

[0074] Adaptability to network conditions: Adjust the transmission rate or encoding method according to the quality of the network connection.

[0075] In one embodiment of the present invention, step S3 verifies the specific situation of the current audio data according to the trigger signal that is verified at the beginning, and establishes full-duplex communication according to the preset radio frequency. At this time, the establishment of full-duplex communication will superimpose the corresponding audio data in the trigger signal to obtain an enhanced audio data, thereby establishing a clear conversation between the two parties.

[0076] like Figure 6 As shown, step S3 also includes the following implementation method: S31, according to the RF frequency of the preset channel interval, obtain the frequency pair of the current voice intercom session; this frequency pair represents a group of RF frequencies for communicating according to the RF frequency in the preset channel interval, and the audio data to be sent are interacted according to this RF frequency, but when interacting, it is necessary to verify the audio data represented by the frequency pair to determine the transmission status of the voices of both parties.

[0077] S32, based on the frequency pair of the current voice intercom session, convert the user's audio data into voice information, and divide the voice information into first voice information and second voice information according to the trigger signal of the voice intercom session. The trigger signal during voice intercom mainly indicates the content of corresponding noise in the audio data. At this time, the voice information will be divided into two types of information according to the existing noise; the first voice information represents the voice information converted from the audio data with more audio features, and the second voice information is the voice information converted from the audio data with fewer audio features; that is, the first voice information has more noise when divided, and the second voice information has less noise.

[0078] S33, comparing the first voice information with the second voice information, determining the periodic characteristics of the first voice information and the second voice information, and outputting the determined first voice information and the second voice information.

[0079] When comparing the first voice information with the second voice information, the signal-to-noise ratios in the first voice information and the second voice information are mainly compared, and it is determined whether the current first voice information and the second voice information have periodic features when the signal-to-noise ratio is calculated. If so, the first voice information and the second voice information are superimposed, and the superimposed first voice information and the second voice information are used as the first voice information and the second voice information after the determination is completed.

[0080] The superposition method can use weighted superposition to superimpose the parts of the first voice information and the second voice information with periodic features to strengthen the parts with periodic features. The periodic features mainly use the autocorrelation function to calculate the periodic components in the first voice information and the second voice information, or use Fourier change to convert the data corresponding to the first voice information and the second voice information into a spectrum diagram, and identify the peaks in the spectrum diagram to find the obvious periodic components therein; if the autocorrelation function has a significant peak at a non-zero delay, it is considered that the corresponding voice information has periodicity, and if there is a stable frequency peak in the spectrum diagram, it is considered that the corresponding voice information has periodicity. After superimposing these periodic first voice information and second voice information, the quality of voice communication is improved, and a better user experience can be provided in a complex environment; to reduce the situation where voice quality is reduced in complex situations. For the signal-to-noise ratio, it is calculated using the ratio of signal power to noise power, so this content will not be described in detail at this time.

[0081] In one embodiment of the present invention, step S4 mainly describes the situation of the intercom during the call, and adjusts the priority of the intercom according to the radio frequency used during the call, and completes the setting of the communication channel.

[0082] At this time, the two parties of the call can realize point-to-group and group-to-group calls, and dial according to the preset buttons during the call to reduce the call delay caused by dialing. The specific situation during the call is more inclined to determine the call classification and the priority of the corresponding group communication to complete the management of the audio data in the communication channel, and then set the corresponding progress according to the dialing and communication situation, and finally complete the fast intercom and communication priority processing.

[0083] like Figure 7 As shown, the implementation of step S4 also includes: S41, dialing according to a preset dialing sequence to obtain the priority of the current dialing.

[0084] S42, comparing the call status of the current dialing and the radio frequency frequency of the intercom and the corresponding device according to the priority of the current dialing.

[0085] S43, when the call status changes, according to the change form of the call status, the radio frequency of the intercom and the corresponding device is used as the first judgment basis to adjust the priority of the current dialing.

[0086] At this time, the way to adjust the dialing priority for the radio frequency frequency of the intercom and the corresponding device is to compare whether the radio frequency frequency has changed, and at the same time compare the priority level of the current dialing in the initial state, and then adjust according to this priority to complete the hierarchical management of intercom calls in the group.

[0087] like Figure 8 As shown, at this time, step S43 also includes: S431, identifying the radio frequency frequency of the intercom and the corresponding device. If the radio frequency frequency of the intercom and the corresponding device is inconsistent with the radio frequency frequency of the preset channel frequency interval, it is determined that the call state has changed, and the priority of the current dial is reset; at this time, when the radio frequency frequency of the intercom and the corresponding device is inconsistent with the radio frequency frequency of the preset channel frequency interval, it means that the call is over, or the call is abnormally interrupted. Only when the two radio frequency frequencies are consistent, the current dialing call is established. Resetting the priority of the current dialing is actually adjusting the priority of the current dialing, because the current dialing scenario is subject to more interference and poor communication quality, and the inconsistent frequency will cause conflicts in priority management. At this time, the readjustment can make the priority and the corresponding frequency match to complete a call for a certain priority.

[0088] First, the priority of the current dial will be initially defined, and divided into high priority: such as emergency rescue, command, etc. Medium priority: such as daily task arrangement, progress report, etc. Low priority: such as non-critical notification, background music, etc. Different RF frequencies will be set here according to the priority. This RF frequency will be pre-stored in the preset channel frequency range, such as setting a high frequency band (such as 450-460MHz) for high priority calls; medium frequency band (such as 460-470MHz) for medium priority calls; low frequency band (such as 470-480MHz): for low priority calls.

[0089] At this time, the priority of the current dialing is reset, and it will be adjusted according to the different forms of RF frequency expression, such as receiving two call requests at the same time.

[0090] Call A: A security alert from an on-site security officer, initially with a high priority, but currently using the mid-band frequency (460-470MHz).

[0091] Call B: A request for supplies from the logistics team, initially medium priority, but currently using the high frequency band (450-460MHz).

[0092] At this time, two different frequencies are detected, and after comparing the priorities, the priorities are readjusted; for call A, although the initial priority is high, it is temporarily regarded as medium priority because it uses the medium frequency band; for call B, although the initial priority is medium, it is upgraded to high priority because it uses the high frequency band. At this time, the priority is reset.

[0093] S432, if the radio frequency of the intercom and the corresponding device is consistent with the radio frequency of the preset channel frequency range, the current dial is connected according to the obtained priority of the current dial, and the communication channel is set. At this time, the frequency is consistent, indicating that the priority and frequency are correct. After the corresponding call is connected, the corresponding device will maintain this frequency to complete the setting of the communication channel between the current intercom and the corresponding device.

[0094] In one embodiment of the present invention, step S5 mainly checks the intercom communicating under the communication channel to complete the final communication quality check. At this time, the full-duplex signal existing in the communication channel represents the intercom that is communicating. After comparing the actual status of this intercom, it can be obtained whether the conversation of the current intercom is stable, thereby completing the intercom call in a complex environment and reducing the impact of surrounding noise and low conversation communication quality.

[0095] like Fig. 9 As shown, the implementation method of step S5 also includes: S51, calculating the signal strength, signal-to-noise ratio and spectrum occupancy rate under each communication channel; at this time, the signal strength, signal-to-noise ratio and spectrum occupancy rate are calculated to determine whether the currently established communication channel can communicate normally and whether there will be corresponding data loss problems.

[0096] S52, identify the negative correlation factor corresponding to the signal strength, signal-to-noise ratio and spectrum occupancy as the first effect factor; the first effect factor is the data that there is a negative correlation between the signal strength, signal-to-noise ratio and spectrum occupancy, and these data will decrease due to the increase of any one of the signal strength, signal-to-noise ratio and spectrum occupancy. At this time, the data of the negatively correlated signal strength, signal-to-noise ratio and spectrum occupancy when the communication channel is tested is obtained, and the absolute difference average value of this part of the data is set as the first effect factor. The absolute difference is calculated by subtracting this part of the negatively correlated data from the standard value set in the historical data, and then taking the average value to quantify the error that the negatively correlated data can produce. This part of the error will highlight the part where there is a problem in the communication. After identifying this part, the corresponding data can be sent to the server to facilitate subsequent staff to perform corresponding optimization.

[0097] S53, calculate the Pearson correlation coefficients of signal strength, signal-to-noise ratio and spectrum occupancy in turn, and take the sum of the Pearson correlation coefficients of signal strength, signal-to-noise ratio and spectrum occupancy as the second effect factor; the second effect factor is essentially the positive correlation between signal strength, signal-to-noise ratio and spectrum occupancy, and is quantified using the Pearson correlation coefficient. The second effect factor obtained later will highlight this part of the correlation to indicate the quality of the communication status of the communication channel, so as to complete the verification of the communication status in the communication channel.

[0098] S54, outputting the values ​​of the first effect factor and the second effect factor to complete the communication comparison of the full-duplex signal. In this step, after obtaining the values ​​of the first effect factor and the second effect factor, the specific situation of the communication channel can be known according to the two values, and then the two values ​​are output to the external server to complete the communication comparison of the full-duplex signal.

[0099] like Fig.10 As shown, the present invention also provides a duplex voice intercom system, including: an audio data acquisition module, a state judgment configuration module, a priority management module and a communication quality optimization module; the output end of the audio data acquisition module is connected to the state judgment configuration module, the output end of the state judgment configuration module is connected to the priority management module, and the output end of the priority management module is connected to the communication quality optimization module.

[0100] The audio data acquisition module is used to collect the user's audio data and determine the intensity and source distance of the audio data. If the audio data is normal and the distance is less than a first preset distance, it is sent; otherwise, the trigger signal is adjusted and the audio data is re-modulated.

[0101] The state judgment configuration module is used to judge whether the device adopts a specific state mode and configure the transmission mode of the audio data according to the state mode.

[0102] The priority management module is used to determine the priority of the current dialing according to the dialing sequence and compare it with the call status and radio frequency; when the call status changes, the priority is dynamically adjusted.

[0103] The communication quality optimization module is used to calculate the signal strength, signal-to-noise ratio and spectrum occupancy rate under each communication channel, analyze the correlation between signal strength, signal-to-noise ratio and spectrum occupancy rate, output the first effect factor and the second effect factor, and complete the communication comparison.

[0104] Although the embodiments of the present invention have been shown and described above, it is to be understood that the above embodiments are exemplary and are not to be construed as limitations on the present invention. A person skilled in the art may make changes, modifications, substitutions and variations to the above embodiments within the scope of the present invention and they are still covered by the protection scope of the present invention.

Claims

1. A duplex voice intercom method, characterized in that: include: S1, collects the user's audio data, starts a voice intercom session, and configures the audio configuration settings of the intercom; S2, determining whether the audio configuration setting adopts the state mode of the intercom; S3, determining a trigger signal for the intercom to start a voice intercom session, and establishing full-duplex communication according to a radio frequency in a preset channel frequency interval; S4, according to the specific situation of the intercom during the call, the radio frequency frequency of the intercom and the corresponding device is processed, and the communication channel is set according to the priority of the communication; S5, verifying the full-duplex signal under the communication channel, and verifying the communication status in each communication channel, and completing the communication comparison of the full-duplex signal.

2. A duplex voice intercom method according to claim 1, characterized in that: The implementation of step S1 also includes: S11, obtaining audio data of the intercom within a first preset distance, determining the strength of the audio data, and treating the determined audio data as a processing result of the first preset distance; S12, when the processing result of the first preset distance is normal, determine whether there is audio data whose acquisition distance is greater than the first preset distance, and send the audio data after the determination is completed.

3. A duplex voice intercom method according to claim 2, characterized in that: Step S12 may also include the following implementations: S121, if the acquisition distance of the audio data is greater than the first preset distance, setting a trigger signal for each frame of audio data, and calculating the waveform conformity of the trigger signal according to the waveform of the trigger signal; S122, modulating the user's audio data into target audio information according to the waveform conformity of the trigger signal, and sending it according to a preset radio frequency; S123: If there is no audio data acquisition distance greater than the first preset distance, modulate the user's audio data into target audio information and send it according to a preset radio frequency frequency.

4. A duplex voice intercom method according to claim 2, characterized in that: Step S11 also includes: S111, using the intercom to continuously monitor the audio data of the user and record the frequency change of the audio data; S112, for each intercom, measuring the strength of the received audio data, and comparing the relative changes in the strength of the audio data; S113, comparing the relative change in the intensity of the audio data with a preset intensity difference, and estimating a distance estimation value of the source of the audio data; S114, using the Doppler effect and combining the frequency change of the audio data, the distance estimation value is adjusted. When the adjusted distance estimation value is less than the first preset distance, the processing result of the first preset distance is determined to be normal.

5. A duplex voice intercom method according to claim 1, characterized in that: Step S2 includes the following implementations: S21, if the intercom status mode is adopted, the user's audio data is sent to the corresponding device according to the audio configuration settings; S22: If the state mode of the intercom is not adopted, verify the state mode of the intercom, parse audio parameters from the user's audio data, and send the audio parameters to the corresponding device.

6. A duplex voice intercom method according to claim 1, characterized in that: Step S3 includes the following implementations: S31, obtaining a frequency pair of the current voice intercom session according to the radio frequency of the preset channel interval; S32, based on the frequency pair of the current voice intercom session, converting the user's audio data into voice information, and dividing the voice information into first voice information and second voice information according to a trigger signal of the voice intercom session; S33, comparing the first voice information with the second voice information, determining the periodicity characteristics of the first voice information and the second voice information, and outputting the determined first voice information and the second voice information.

7. A duplex voice intercom method according to claim 1, characterized in that: The implementation of step S4 also includes: S41, dialing according to a preset dialing sequence to obtain the priority of the current dialing; S42, comparing the call status of the current dialing and the radio frequency frequency of the intercom and the corresponding device according to the priority of the current dialing; S43, when the call status changes, according to the change form of the call status, the radio frequency of the intercom and the corresponding device is used as the first judgment basis to adjust the priority of the current dialing.

8. A duplex voice intercom method according to claim 7, characterized in that: Step S43 also includes: S431, identifying the radio frequency frequencies of the intercom and the corresponding device, if the radio frequency frequencies of the intercom and the corresponding device are inconsistent with the radio frequency frequencies in the preset channel frequency interval, determining that the call status has changed, and resetting the priority of the current dialing; S432: If the radio frequency of the intercom and the corresponding device is consistent with the radio frequency of the preset channel frequency interval, the current dial is connected according to the obtained priority of the current dial, and the communication channel is set.

9. A duplex voice intercom method according to claim 1, characterized in that: The implementation of step S5 also includes: S51, calculating the signal strength, signal-to-noise ratio and spectrum occupancy rate of each communication channel; S52, identifying a negative correlation factor corresponding to signal strength, signal-to-noise ratio, and spectrum occupancy as a first effect factor; S53, calculating the Pearson correlation coefficients of the signal strength, the signal-to-noise ratio, and the spectrum occupancy rate in sequence, and taking the sum of the Pearson correlation coefficients of the signal strength, the signal-to-noise ratio, and the spectrum occupancy rate as the second effect factor; S54, outputting the values ​​of the first effect factor and the second effect factor to complete the communication comparison of the full-duplex signal.

10. A duplex voice intercom system, characterized in that: include: The audio data acquisition module is used to collect the user's audio data and determine the strength and source distance of the audio data. If the audio data is normal and the distance is less than a first preset distance, it is sent; otherwise, the trigger signal is adjusted and the audio data is re-modulated; A state judgment configuration module is used to judge whether the device adopts a specific state mode and configure the transmission mode of the audio data according to the state mode; The priority management module is used to determine the priority of the current dialing according to the dialing sequence and compare it with the call status and radio frequency; when the call status changes, the priority is dynamically adjusted; The communication quality optimization module is used to calculate the signal strength, signal-to-noise ratio and spectrum occupancy rate under each communication channel, analyze the correlation between signal strength, signal-to-noise ratio and spectrum occupancy rate, output the first effect factor and the second effect factor, and complete the communication comparison.

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