CN116996801A - Intelligent conference debugging speaking system with wired and wireless access AI - Google Patents
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Abstract
The invention relates to the technical field of electronics and computers, in particular to an intelligent debugging conference speaking system with wired and wireless access AI, which comprises an acquisition module, an analysis module, a processing module and an output module.
Description
Technical Field
The invention relates to the technical field of electronics and computers, in particular to an intelligent conference debugging speaking system with a wired and wireless access AI.
Background
In modern society, speaking devices in conference rooms are important tools to ensure efficient communication and exchange of conferences.
The meeting uses broadcasting equipment such as microphones, and microphone types on the market are various, and according to its price difference, performance, tone quality and stability of microphone are also different, and along with the increase of life, the performance of microphone also can have certain decay, and when actually meeting, in order to ensure the quality of meeting, need to use a large amount of equipment to carry out real-time control and adjustment to the conversation section of thick bamboo, in order to avoid furthest meeting in-process that the microphone breaks down and leads to meeting progress to receive the interference.
Due to the adoption of a large number of devices, the cost of a conference is increased, and the complicated adjustment process also causes a large hysteresis in the adjustment mode depending on manual intervention, so that the intervention adjustment of the dialogue barrel is difficult in time.
Disclosure of Invention
The invention provides an intelligent debugging conference speaking system with a wired and wireless access AI, which has the beneficial effects of avoiding manual debugging and achieving an automatic intervention adjustment effect by an AI algorithm, and solves the problems that the cost of the conference is increased, and the adjustment mode depending on manual intervention has larger hysteresis caused by a complicated adjustment process in the background art, so that the intervention adjustment of a dialogue barrel is difficult to be performed in time.
The invention provides the following technical scheme: the intelligent debugging conference speaking system with the wired and wireless access AI comprises an acquisition module, an analysis module, a processing module and an output module:
the acquisition module is used for acquiring necessary data for analysis and comprises: conference room data, microphone data, and external sound source data;
the analysis module is used for analyzing and calculating each data acquired by the acquisition module so as to acquire an intervention coefficientThe specific calculation mode is as follows:
in the middle ofThe size of the space representing the conference room, obtained from the conference room data set,representing the sound level of the microphoneThe integrated capability value representing the volume and tone of the microphone, is obtained from the microphone data,represents an external valueThe echo in the representative office is acquired by external sound source data;
in the middle ofAndall of the weight values are the weight values,is a correction coefficient, and，and (2) andandthe specific value of (2) is selected and set by the customer;
the analysis module applies the intervention coefficientsInputting the intervention coefficients into a processing module, and reprocessing the intervention coefficients by the processing module;
processing mouldAfter receiving the intervention coefficients calculated and acquired by the analysis module, the block calculates the intervention coefficients and the intervention threshold value recorded in the processing moduleThreshold of interventionComparing, thereby enabling the processing module to start and stop AI intelligent regulation;
the output module is used for outputting the data required by the conference to the playing device.
As an alternative scheme of the intelligent debugging conference speaking system with the wired and wireless access AI, the invention comprises the following steps: the acquisition module comprises a space detection unit, wherein the space detection unit comprises a laser detector and performs data acquisition on the length, width and height of an office through the following steps:
s1, standing at one side of a room, selecting a proper starting point, and aiming at a target point to be measured by using a display screen or visual assistance of a laser range finder;
s2, pressing a measuring button on the laser range finder, sending out a laser beam by the device, generating reflection on a target point, measuring the time of the laser beam emitted from the range finder to the target point and returning again by the laser range finder, and calculating the distance;
s3, moving the laser range finder to the next measuring point, typically a wall or corner connected to the last point, repeating S1 to S2, and measuring each part of the room;
acquiring the length, width and height of the office room through the steps S1-S3, and then summarizing the distance data to calculate the space size of the conference room。
As an alternative scheme of the intelligent debugging conference speaking system with the wired and wireless access AI, the invention comprises the following steps: the acquisition module also comprises a sound detection unit, wherein the sound detection unit comprises a recorder and sound analysis software;
the recorder is used for recording the sound of the microphone and analyzing the recorded audio file through sound analysis software in the computer so as to obtain the volume value of the microphoneAnd microphone tone quality valueThe different microphone models are respectively marked as、、、…、And、、、…、。
as an alternative scheme of the intelligent debugging conference speaking system with the wired and wireless access AI, the invention comprises the following steps: the collecting module further comprises a receipt unit, wherein the receipt unit is used for detecting external sound sources and comprises a decibel detector, the decibel detector can detect sound in the conference room and carry out auxiliary analysis in sound analysis software of a detection data transmission value, so that external sound source data in the conference room are obtained: including external echo values in offices。
As an alternative scheme of the intelligent debugging conference speaking system with the wired and wireless access AI, the invention comprises the following steps: the calculating unit is used for calculating and obtaining the microphone pitch valueAnd an external sound source value。
As an alternative scheme of the intelligent debugging conference speaking system with the wired and wireless access AI, the invention comprises the following steps: the microphone pitch valueThe method is obtained through calculation according to the following formula:
in (a)The code number representing the model isIs the volume of the microphone，The code number representing the model isIs the sound quality of the microphone；
In the middle ofAll of the weight values are the weight values,is a correction coefficient, and，and (2) andandthe specific value of (3) is selected by the customer.
As an alternative scheme of the intelligent debugging conference speaking system with the wired and wireless access AI, the invention comprises the following steps: the external valueThe method is obtained through calculation according to the following formula:
in the middle ofRepresenting the external echo value in the office,representing the size of the space of the conference room,in order to correct the coefficient of the coefficient,the specific value of (3) is selected by the customer.
As an alternative scheme of the intelligent debugging conference speaking system with the wired and wireless access AI, the invention comprises the following steps: the processing module comprises a comparison unit and an opening and closing unit, wherein the comparison unit calculates the acquired intervention coefficient by the analysis moduleIs recorded with the comparing unitIs an intervention threshold value of (a)Threshold of interventionAnd (3) carrying out numerical comparison so as to judge whether to start the intervention of the AI intelligent system, wherein the specific mode is as follows:
when the intervention coefficient isIntervention thresholdWhen the microphone is represented to perform well, the intervention of an AI intelligent system is not needed;
when the intervention threshold valueIntervention coefficientIntervention thresholdRepresenting the fluctuation of the microphone performance, and waking up an AI intelligent system to detect the microphone;
when the intervention coefficient isIntervention thresholdWhen the microphone is abnormal, the AI intelligent system is required to conduct intervention correction on the microphone.
The invention has the following beneficial effects:
1. the conference speaking system with the functions of integrating functions of wired and wireless access AI intelligent debugging, such as a wired analog microphone, a U-section wireless microphone, a wired digital microphone, a wireless digital microphone, input and output of a third-party sound source, no manual debugging, automatic intervention adjusting effect and the like is achieved by the aid of the AI algorithm, most microphone forms needed by conference room conference speaking are covered, high-quality sound transmission and speaking experience are guaranteed to be provided under different conference environments, and the intelligent conference speaking system can be intelligently adjusted according to specific conditions.
2. The intelligent conference speech debugging system with the wired and wireless access AI utilizes the decibel detector, the sound analysis software and the calculation formula to monitor and optimize the sound environment in the conference room, and the system can be customized according to different requirements by self-selecting and setting parameters of a client so as to ensure that the sound quality and the effect of the conference reach the optimal state.
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FIG. 1 is a schematic flow chart of the system structure of the present invention.
Fig. 2 is a schematic diagram of information transfer according to the present invention.
Detailed Description
The following description of the embodiments of the present invention will be made clearly and completely with reference to the accompanying drawings, in which it is apparent that the embodiments described are only some embodiments of the present invention, but not all embodiments. All other embodiments, which can be made by those skilled in the art based on the embodiments of the invention without making any inventive effort, are intended to be within the scope of the invention.
Example 1
In modern society, speaking devices in conference rooms are important tools to ensure efficient communication and exchange of conferences. The conventional conference speaking system generally needs a large amount of equipment and complicated manual debugging, in order to solve the problems, an intelligent debugging conference speaking system with wired and wireless access AI is provided, which has the input and output functions of various microphone modes, is provided with an AI algorithm, can realize automatic intervention adjustment effect, changes the conference room speaking equipment from a plurality of equipment to a host, please refer to fig. 1-2, and comprises an acquisition module, an analysis module, a processing module and an output module:
the acquisition module is used for acquiring necessary data for analysis and comprises: conference room data, microphone data, and external sound source data;
the analysis module is used for analyzing and calculating each data acquired by the acquisition module so as to acquire an intervention coefficientThe specific calculation mode is as follows:
in the middle ofThe size of the space representing the conference room, obtained from the conference room data set,representing the sound level of the microphoneThe integrated capability value representing the volume and tone of the microphone, is obtained from the microphone data,represents an external valueThe echo in the representative office is acquired by external sound source data;
in the middle ofAndall of the weight values are the weight values,is a correction coefficient, and，and (2) andandthe specific value of (2) is selected and set by the customer;
the analysis module applies the intervention coefficientsInputting the intervention coefficients into a processing module, and reprocessing the intervention coefficients by the processing module;
after receiving the intervention coefficients calculated and acquired by the analysis module, the processing module calculates the intervention coefficients and intervention threshold values recorded in the processing moduleThreshold of interventionComparing, thereby enabling the processing module to start and stop AI intelligent regulation;
the output module is used for outputting the data required by the conference to the playing device.
In this embodiment: the acquisition module is responsible for collecting the necessary data for subsequent analysis and processing. These data include: environmental data of conference rooms, audio data of microphones, and data of external sound sources.
The task of the analysis module is to analyze and calculate various data acquired by the acquisition module to obtain an intervention coefficient, which determines how the system adjusts the sound, the calculation of the intervention coefficient is based on a weighted combination of the conference room size, the microphone sound level value and the external sound source value, wherein the weight and the correction coefficient can be set by the user, and the analysis process ensures that the system can be intelligently adjusted according to specific conditions so as to provide the optimal sound quality.
The processing module receives the intervention coefficient from the analysis module and compares the intervention coefficient with an intervention threshold value and an intervention threshold value preset in the processing module, the comparison process determines whether the system starts AI intelligent regulation, and if the intervention coefficient is in the process, the system starts automatic regulation so as to ensure conference speaking quality.
The output module is responsible for outputting the intelligently adjusted sound data to the playing device so as to ensure that the participants can clearly hear the sound of the speaker, and the task of the module is to transmit the sound after the system optimization to the audience so as to provide the optimal conference experience.
The system integrates various technologies and intelligent algorithms, integrates the functions of wired analog microphone, U-section wireless microphone, wired digital microphone, wireless digital microphone, input and output of third-party sound source, and has the functions of automatic intervention adjustment effect and the like, and the conference speaking host machine covers most microphone forms required by conference room conference in speaking, so that high-quality sound transmission and speaking experience can be ensured under different conference environments, intelligent adjustment can be performed according to specific conditions, and manual adjustment can be reduced to a certain extent.
Example 2
Referring to fig. 1-2, the acquisition module includes a space detection unit, which includes a laser detector, and performs data acquisition on the length, width and height of an office by:
s1, standing at one side of a room, selecting a proper starting point, and aiming at a target point to be measured by using a display screen or visual assistance of a laser range finder;
s2, pressing a measuring button on the laser range finder, sending out a laser beam by the device, generating reflection on a target point, measuring the time of the laser beam emitted from the range finder to the target point and returning again by the laser range finder, and calculating the distance;
s3, moving the laser range finder to the next measuring point, typically a wall or corner connected to the last point, repeating the steps 1 to 2, and measuring each part of the room;
acquiring the length, width and height of the office room through the steps S1-S3, and then summarizing the distance data to calculate the space size of the conference room。
In this embodiment: the core component of the space detection unit is a laser detector, which adopts advanced laser ranging technology, and performs accurate data acquisition on the length, width and height of an office through the following steps.
First, the user stands on one side of the room, selects an appropriate starting point, and then aims at the target point to be measured using the display screen or visual aid of the laser rangefinder, which ensures the accuracy and precision of the measurement to the greatest extent.
The user presses a measurement button on the laser rangefinder, at which time the device will emit a laser beam that will reflect at the target point, the laser rangefinder will measure the time the laser beam is emitted from the rangefinder to the target point and back again, and the distance is calculated from the speed of light.
Next, the user moves the laser rangefinder to the next measurement point, typically the wall or corner of the room connected to the last point. Then, the user repeats steps 1 to 2 to measure various parts of the room.
Through the above steps, the spatial detection unit collects accurate data of the length, width and height of the office room, which can be used not only for calculating the spatial size of the conference room, but also for further sound analysis and optimization.
The space detection unit is used as one of key innovations of the intelligent debugging conference speaking system, not only improves the effect of sound optimization, but also provides possibility of more personalized settings, brings remarkable improvement to sound transmission and speaking experience in a conference room, further enables the intelligent debugging conference speaking system to be excellent in performance under different scenes, meets the requirements of different users, and improves conference efficiency and quality.
Example 3
Referring to fig. 1-2, the acquisition module further includes a sound detection unit, which includes a recorder and sound analysis software;
the recorder is used for recording the sound of the microphone and recordingThe audio file after being made is analyzed by sound analysis software in the computer, thereby obtaining the volume value of the microphoneAnd microphone tone quality valueThe different microphone models are respectively marked as、、、…、And、、、…、。
in this embodiment: the recorder is a key device in the sound detection unit, which has the task of recording the sound of the microphone, which captures the sound of the speaker in the conference, and stores the sound in the form of an audio file for further analysis and processing.
Sound analysis software is another important component of the sound detection unit. Its main task is to analyze recorded sound data to obtain two key parameters: microphone sound quality value and microphone sound quality value.
Microphone volume value: this parameter reflects the volume level of the microphone, i.e. the intensity of the sound captured by the microphone, which is important to ensure that the sound of the speaker is clearly transmitted to other participants in the conference room, and if the microphone volume value is too low, this may lead to unclear or low sound transmission, thereby affecting the conference effect.
Microphone tone quality value: this parameter measures the quality of the sound captured by the microphone, including the clarity, purity and other audio properties of the sound, the level of the microphone timbre directly affects the naturalness and audibility of the sound, a high quality microphone timbre means a clear, natural sound, while a low quality microphone timbre may lead to noisy, distorted or low fidelity sounds.
Different microphone models may have different microphone sound volume values and microphone sound quality values, so that the sound detection unit can record and distinguish between the microphones of different models. This feature enables the system to be intelligently tuned to provide the best sound quality depending on the microphone model used.
The introduction of the sound detection unit enhances the audio quality and adaptability of the intelligent debug conference system. Through the collaborative work of recorder and sound analysis software, the system can monitor and analyze the sound of microphone in real time to ensure that the sound transmission in the meeting is clear, natural, thereby improved effect and user experience of meeting, moreover, this system still possesses the ability of intelligent recognition different microphone models, with the characteristic according to different models carries out the sound adjustment, ensure that every time meeting can all go on with the best state, no matter under wired or wireless access condition, the characteristic of sound detection unit makes intelligent debugging conference system become an efficient, self-adaptation audio processing solution.
Example 4
Referring to fig. 1-2, the collecting module further includes a receipt unit, where the receipt unit is configured to detect an external sound source, and includes a db detector, where the db detector detects sound in the conference room and sends detection data to the sound analysis software to perform auxiliary analysis, so as to obtain external sound source data in the conference room: including external echo values in offices。
The analysis module comprises a calculation unit for calculating and acquiring microphone sound level valueAnd an external sound source value。
Microphone pitch valueThe method is obtained through calculation according to the following formula:
in the middle ofThe code number representing the model isIs the volume of the microphone，The code number representing the model isIs the sound quality of the microphone；
In the middle ofAll of the weight values are the weight values,is a correction coefficient, and，and (2) andandthe specific value of (3) is selected by the customer.
External valueThe method is obtained through calculation according to the following formula:
in the middle ofRepresenting the external echo value in the office,representing the size of the space of the conference room,in order to correct the coefficient of the coefficient,the specific value of (3) is selected by the customer.
In this embodiment: the system comprises a collecting module and an analyzing module, wherein the collecting module comprises a receipt unit and a calculating unit, the receipt unit is used for detecting an external sound source and comprises a decibel detector, the decibel detector is used for detecting sound in a conference room and sending detection data into sound analysis software to carry out auxiliary analysis, so that external sound source data in the conference room are obtained, the external sound source data comprise an external echo value in an office, the analyzing module comprises a calculating unit and is used for calculating and obtaining a microphone sound level value and an external sound source value, the microphone sound level value is obtained through a specific calculating formula, and the external sound source value is also obtained through another specific calculating formula.
The response piece unit is used for monitoring sound in the conference room through the decibel detector, which can help to identify and measure influence of external sound sources, particularly echo values, which is very important for improving conference quality, because external echoes can interfere with hearing experiences of participants, and the decibel detector sends detected data to sound analysis software so that the sound analysis software can assist in analyzing the sound data, which helps a system to better understand sound environment in the conference room and provides powerful support for subsequent debugging and optimization.
The task of the calculating unit in the analyzing module is to calculate a microphone sound level value and an external sound source value, wherein the microphone sound level value is an important parameter, and can be used for measuring the tone quality and the volume of a microphone. This is important to ensure that the speaker's voice is clearly audible.
The external sound source value can help to identify and quantify the external echo value in the office, the echo is the result of sound reflection, the sound quality of the conference can be disturbed to a certain extent, the system can better know the sound environment in the office through the external sound source value obtained by calculation, and appropriate measures are taken to reduce the interference of the echo to the conference.
The intelligent conference speech debugging system with the wired and wireless access AI provides a highly intelligent conference experience, monitors and optimizes the sound environment in the conference room by utilizing a decibel detector, sound analysis software and a calculation formula, and can be customized according to different requirements by self-selecting and setting parameters by a client so as to ensure that the sound quality and effect of the conference reach an optimal state.
Example 5
Referring to fig. 1-2, the processing module includes a comparing unit and an on-off unit, wherein the comparing unit calculates the obtained intervention coefficients by the analyzing moduleIntervention threshold value recorded by comparison unit itselfThreshold of interventionAnd (3) carrying out numerical comparison so as to judge whether to start the intervention of the AI intelligent system, wherein the specific mode is as follows:
when the intervention coefficient isIntervention thresholdWhen the microphone is represented to perform well, the intervention of an AI intelligent system is not needed;
when the intervention threshold valueIntervention coefficientIntervention thresholdRepresenting the fluctuation of the microphone performance, and waking up an AI intelligent system to detect the microphone;
when the intervention coefficient isIntervention thresholdWhen the microphone is abnormal, the AI intelligent system is required to conduct intervention correction on the microphone.
In this embodiment: the processing module comprises a comparing unit and an opening and closing unit, the comparing unit is responsible for analyzing and judging the performance of the microphone, and the opening and closing unit is responsible for deciding whether to start the intervention of the AI intelligent system according to the judgment of the comparing unit.
The comparison unit is provided with two important parameters, namely an intervention threshold value and an intervention threshold value, the two parameters are preset in the comparison unit and are used for being compared with an intervention coefficient, the intervention threshold value represents a critical point, when the intervention coefficient is smaller than or equal to the intervention threshold value, the system considers that the microphone performs well without intervention of an AI intelligent system, when the intervention coefficient exceeds the intervention threshold value, the system considers that the microphone performs abnormality, intervention correction of the AI intelligent system is needed, and the introduction setting of the two parameters represents the sensitivity and tolerance of the system to the microphone, so that the intervention adjustment is performed on the system by timely starting AI according to the actual condition of the microphone in the use process, and the flow degree of a conference and the conference quality are improved to a certain extent.
It is noted that relational terms such as first and second, and the like are used solely to distinguish one entity or action from another entity or action without necessarily requiring or implying any actual such relationship or order between such entities or actions. Moreover, the terms "comprises," "comprising," or any other variation thereof, are intended to cover a non-exclusive inclusion, such that a process, method, article, or apparatus that comprises a list of elements does not include only those elements but may include other elements not expressly listed or inherent to such process, method, article, or apparatus.
The foregoing is merely a preferred embodiment of the present invention, and it should be noted that it will be apparent to those skilled in the art that several modifications and variations can be made without departing from the technical principle of the present invention, and these modifications and variations should also be regarded as the scope of the invention.
Claims (8)
1. The utility model provides a possess wired wireless access AI intelligent debugging meeting system of speaking, includes collection module, analysis module, processing module, output module, its characterized in that:
the acquisition module is used for acquiring necessary data for analysis and comprises: conference room data, microphone data, and external sound source data;
the analysis module is used for analyzing and calculating each data acquired by the acquisition module so as to acquire an intervention coefficientThe specific calculation mode is as follows:
；
in the middle ofRepresenting the size of the room, acquired from the room data set, < >>Representing the sound level of the microphoneA comprehensive capacity value representing the volume and tone of the microphone, obtained from microphone data, ++>Represents an external valueThe echo in the representative office is acquired by external sound source data;
in the middle ofAnd +.>All are weight values, < >>Is a correction coefficient, and->，And->And +.>The specific value of (2) is selected and set by the customer;
the analysis module applies the intervention coefficientsInputting the intervention coefficients into a processing module, and reprocessing the intervention coefficients by the processing module;
after receiving the intervention coefficients calculated and acquired by the analysis module, the processing module calculates the intervention coefficients and intervention threshold values recorded in the processing moduleAnd intervention threshold->Comparing, thereby enabling the processing module to start and stop AI intelligent regulation;
the output module is used for outputting the data required by the conference to the playing device.
2. The intelligent conference call system with wired and wireless access AI of claim 1, wherein: the acquisition module comprises a space detection unit, wherein the space detection unit comprises a laser detector and performs data acquisition on the length, width and height of an office through the following steps:
s1, standing at one side of a room, selecting a proper starting point, and aiming at a target point to be measured by using a display screen or visual assistance of a laser range finder;
s2, pressing a measuring button on the laser range finder, sending out a laser beam by the device, generating reflection on a target point, measuring the time of the laser beam emitted from the range finder to the target point and returning again by the laser range finder, and calculating the distance;
s3, moving the laser range finder to the next measuring point, typically a wall or corner connected to the last point, repeating steps S1 to S2, and measuring each part of the room;
the office is obtained through the steps S1-S3After the length, width and height of the public room, the distance data are summarized to calculate the space size of the conference room。
3. The intelligent conference call system with wired and wireless access AI of claim 1, wherein: the acquisition module also comprises a sound detection unit, wherein the sound detection unit comprises a recorder and sound analysis software;
the recorder is used for recording the sound of the microphone and analyzing the recorded audio file through sound analysis software in the computer so as to obtain the volume value of the microphoneAnd microphone tone quality value->The different microphone models are respectively marked as、/>、/>、…、/>And +.>、/>、/>、…、/>。
4. The intelligent conference call system with wired and wireless access AI of claim 1, wherein: the collecting module further comprises a receipt unit, wherein the receipt unit is used for detecting external sound sources and comprises a decibel detector, the decibel detector can detect sound in the conference room and carry out auxiliary analysis in sound analysis software of a detection data transmission value, so that external sound source data in the conference room are obtained: including external echo values in offices。
5. The intelligent conference call system with wired and wireless access AI of claim 1, wherein: the analysis module comprises a calculation unit for calculating and acquiring microphone sound level valuesAnd an external sound source value +.>。
6. The intelligent conference call system with wired and wireless access AI of claim 1, wherein: the microphone pitch valueThe method is obtained through calculation according to the following formula:
；
in the middle ofThe code number of the expression model is->Is>，/>The code number of the expression model is->Is the sound quality of the microphone>；
In the middle ofAll are weight values, < >>Is a correction coefficient, and->，/>And->Andthe specific value of (3) is selected by the customer.
7. The intelligent conference call system with wired and wireless access AI of claim 1, wherein: the external valueThe method is obtained through calculation according to the following formula:
；
in the middle ofRepresents the external echo value in the office, +.>Representing the size of the room of the conference, +.>For correction factor +.>The specific value of (3) is selected by the customer.
8. The intelligent conference call system with wired and wireless access AI of claim 1, wherein: the processing module comprises a comparison unit and an opening and closing unit, wherein the comparison unit calculates the acquired intervention coefficient by the analysis moduleIntervention threshold value recorded in the comparison unit itself>And intervention threshold->And (3) carrying out numerical comparison so as to judge whether to start the intervention of the AI intelligent system, wherein the specific mode is as follows:
when the intervention coefficient isIntervention threshold->When the microphone is represented to perform well, the intervention of an AI intelligent system is not needed;
when the intervention threshold valueIntervention coefficient->Intervention threshold->Representing the fluctuation of the microphone performance, and waking up an AI intelligent system to detect the microphone;
when the intervention coefficient isIntervention threshold->When the microphone is abnormal, the AI intelligent system is required to conduct intervention correction on the microphone.
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