Background
The teleconference is a communication service which is widely applied at present, can effectively solve the problem of voice communication among multiple people in different places, and is popular with enterprises and individual users. For example, business people in an enterprise often need to actively conference a telephone with other multiple people in different places to communicate voice on a business trip. Therefore, they need a way to be able to hold a conference call anytime anywhere.
At present, the used conference machine is very easy to generate howling phenomenon in the conversation process, and the conference machine receives many audio signals of other conference machines and is not clear, so that the using effect is poor.
Disclosure of Invention
The technical problem to be solved by the present invention is to provide a noise reduction conference machine, which can effectively avoid the occurrence of howling phenomenon, and can make the conversation sound clear and have good use effect.
In order to solve the technical problems, the technical scheme of the invention is as follows:
a noise-reducing conferencing machine, comprising:
the sound pickup module comprises a plurality of microphones which are arranged in an array mode and is used for collecting sound sources and environment sounds and converting the sound sources and the environment sounds into first audio signals;
the WiFi module is used for being in interactive connection with other conference machines through a wide area network, receiving second audio signals sent by the other conference machines and sending final audio signals;
the direction key is used for a user to manually specify the direction of the sound source;
the CPU processing module calculates the sound source direction according to the first audio signal; performing sound source directivity voice enhancement processing on the first audio signal according to the calculated sound source direction or a sound source direction manually designated by a user; carrying out ambient sound filtering processing on the first audio signal subjected to the sound source directional speech enhancement processing; filtering the second audio signal to remove environmental sounds;
the loudspeaker module plays the second audio signal after the environmental sound filtering processing;
pickup module, wiFi module, position button and loudspeaker module all with CPU processing module electric connection, final audio signal is the first audio signal after sound source directionality speech enhancement processing and environmental sound filtering processing.
Preferably, the sound source is the voice of the user speaking.
Preferably, the first audio signal includes a plurality of audio electrical signals obtained by acquiring and converting the same sound source by a plurality of microphones, respectively.
As a preferable scheme, the method for acquiring the sound source orientation by the CPU processing module according to the first audio signal includes the following steps:
1) randomly selecting one of the microphones as a reference microphone, and acquiring a sound source by the reference microphone to convert the sound source into a reference audio electrical signal;
2) acquiring the same sound source by using other microphones, and comparing other audio electric signals obtained by conversion with the reference audio electric signals obtained in the step 1) respectively to obtain amplitude differences and phase differences between the other audio electric signals and the reference audio electric signals;
3) calculating the time difference of the sound of the same sound source transmitted to the reference microphone and other microphones according to the amplitude difference and the phase difference obtained in the step 2);
4) and calculating the sound source orientation according to the time difference obtained in the step 3) and the distance between the reference microphone and the other microphones corresponding to the time difference.
As a preferable scheme, the method for performing sound source directional speech enhancement processing on a first audio signal by the CPU processing module includes the following steps:
1) the CPU processing module selects a microphone closest to a sound source as a target microphone according to the calculated sound source position or a sound source position manually specified by a user, and the target microphone collects and converts the sound source to obtain a target audio electrical signal;
2) the CPU processing module converts other audio electric signals obtained by collecting the same sound source by other microphones and converting the same sound source and the target audio electric signals obtained in the step 1) into other frequency spectrums and target frequency spectrums, and makes the other frequencies and the target frequency spectrums compare;
3) the CPU processing module amplifies the amplitude corresponding to the frequency of the target frequency spectrum with the phase earlier than that of other frequency spectrums, and simultaneously, the CPU processing module reduces the amplitude corresponding to the frequency of the target frequency spectrum with the phase lagging behind that of other frequency spectrums.
As a preferred scheme, the ambient sound includes the sound of the second audio signal that the loudspeaker module broadcast was handled through the ambient sound filtering, first audio signal still includes the third audio signal that the pickup module gathered and converted the ambient sound.
As a preferred scheme, the method for performing, by the CPU processing module, ambient sound filtering processing on the first audio subjected to the sound source directivity speech enhancement processing includes the following steps:
1) the CPU processing module compares the first audio signal and the second audio signal which are subjected to the sound source directivity voice enhancement processing;
2) and the CPU processing module filters a third audio signal similar to the second audio signal in the first audio signal after the sound source directional speech enhancement processing to obtain a final audio signal.
As a preferred scheme, the method for performing the ambient sound filtering processing on the second audio signal by the CPU processing module includes the following steps:
1) the CPU processing module converts the second audio signal into a first frequency spectrum;
2) the CPU processing module filters the signals of the first frequency spectrum obtained in the step 1) within the frequency range of 200-3000 Hz.
As a preferred scheme, the noise reduction conference machine further comprises an orientation indicator light, and the orientation indicator light is connected with the CPU processing module.
As a preferred scheme, the noise reduction conference machine further comprises an input module for inputting IP addresses of other conference machines, a sending module for sending the IP addresses of the other conference machines to the noise reduction conference machine, and a storage module for storing conference party information, wherein the input module, the sending module, and the storage module are integrated to form an application installed on the intelligent terminal.
The invention has the beneficial effects that: through CPU processing module, pickup module, wiFi module, position button and loudspeaker module make this meeting function of making an uproar of falling carry out sound source directionality speech enhancement to first audio signal according to the sound source position or the manual appointed sound source position of user that calculate and handle, carry out environment sound filtering processing and carry out environment sound filtering processing to second audio signal to the first audio signal after the sound source directionality speech enhancement is handled, thereby can effectively avoid the phenomenon of whistling to take place, and enable the conversation sound clear, excellent in use effect.
Detailed Description
The structural and operational principles of the present invention are explained in further detail below with reference to the accompanying drawings.
As shown in fig. 1, a noise reduction conference machine includes:
the sound pickup module 1 comprises a plurality of microphones which are arranged in an array mode and used for collecting sound sources and environment sounds and converting the sound sources and the environment sounds into first audio signals;
the WiFi module 2 is used for being in interactive connection with other conference machines through a wide area network, receiving second audio signals sent by the other conference machines and sending final audio signals;
an orientation key 3 for a user to manually specify the orientation of the sound source;
the CPU processing module 4 calculates the sound source direction according to the first audio signal; performing sound source directivity voice enhancement processing on the first audio signal according to the calculated sound source direction or a sound source direction manually designated by a user; carrying out ambient sound filtering processing on the first audio signal subjected to the sound source directional speech enhancement processing; filtering the second audio signal to remove environmental sounds;
the loudspeaker module 5 plays the second audio signal after the environmental sound filtering processing;
pickup module 1, wiFi module 2, position button 3 and loudspeaker module 5 all are connected with CPU processing module 4, final audio signal is the first audio signal after sound source directionality speech enhancement processing and environmental sound filtering processing.
As a preferable scheme, the noise reduction conference machine further comprises an orientation indicator lamp 6, wherein the orientation indicator lamp 6 is connected with the CPU processing module 4, so that the orientation indicator lamp 6 is turned on when a user presses the orientation key 3, and the user is informed of which orientation the user speaks most clearly, thereby facilitating the use of the user.
As a preferred scheme, the noise reduction conference machine further includes an input module 7 for inputting IP addresses of other conference machines, a sending module 8 for sending the IP addresses of the other conference machines to the noise reduction conference machine, and a storage module 9 for storing conference party information, the input module 7, the sending module 8, and the storage module 9 are integrated to form an application installed on the intelligent terminal 10, and the intelligent terminal 10 is connected with the noise reduction conference machine through a WiFi wide area network. The intelligent terminal 10 is a mobile phone, a tablet or a computer.
The method for accessing the noise reduction conference machine to the WiFi wide area network for the first time comprises the following steps:
the WiFi module 2 of the noise reduction conference machine is opened, then an account number and a password which are connected with a WiFi wide area network are input into the intelligent terminal 10, then the intelligent terminal 10 sends the account number and the password to the CPU processing module 4 of the noise reduction conference machine in a WiFi broadcasting mode, the CPU processing module 4 controls the WiFi module 2 to be connected with the WiFi wide area network after receiving the account number and the password, the account number and the password are stored in the CPU processing module 4, and the automatic connection can be realized when the same WiFi wide area network is connected next time.
The method for starting the teleconference between the noise reduction conference machine and other conference machines comprises the following steps:
1) the intelligent terminal 10 is connected with the noise reduction conference machine through a WiFi wide area network;
2) the IP addresses of other conference machines are input through the applied input module 7, then the IP addresses are sent to the noise reduction conference machine through the sending module 8, the noise reduction conference machine is connected with the other conference machines after receiving the IP addresses, and the telephone conference is started.
The sending module 8 of the application sends the IP address to the noise reduction conference machine, and the storage module 9 of the application stores the IP address and can be used for the user to input other information, so that the user can recognize and group the information conveniently, and the use by the user is convenient.
Preferably, the sound source is the voice of the user speaking.
Preferably, the first audio signal includes a plurality of audio electrical signals obtained by acquiring and converting the same sound source by a plurality of microphones, respectively.
As shown in fig. 2, the method for acquiring the sound source bearing by the CPU processing module 4 according to the first audio signal includes the following steps:
1) randomly selecting one of the microphones as a reference microphone, and acquiring a sound source by the reference microphone to convert the sound source into a reference audio electrical signal;
2) acquiring the same sound source by using other microphones, and comparing other audio electric signals obtained by conversion with the reference audio electric signals obtained in the step 1) respectively to obtain amplitude differences and phase differences between the other audio electric signals and the reference audio electric signals;
3) calculating the time difference of the sound of the same sound source transmitted to the reference microphone and other microphones according to the amplitude difference and the phase difference obtained in the step 2);
4) and calculating the sound source orientation according to the time difference obtained in the step 3) and the distance between the reference microphone and the other microphones corresponding to the time difference.
Since the microphones in the sound pickup module are arranged in an array manner, and the distances between the microphones are known, taking the reference microphone as the origin of the coordinates, the coordinates of other microphones are also known, according to the formula:
wherein, (x, y) represents the sound source coordinates, (x)
i,y
i) Coordinate values representing a reference microphone, (x)
j,y
j) Representing one of the pick-up modulesCoordinates of other microphones; c represents the propagation speed of sound waves in air; while
Represents the time difference between the propagation of the sound wave to the reference microphone and one other microphone; thus, the coordinates of the sound source can be obtained, and the position information of the sound source and each microphone and the distance information between the sound source and each microphone can be known.
It should be noted that, in the above description, in addition to the reference microphone as the origin of the coordinates, the geometric center positions of the reference microphone and other microphones may also be selected as the origin of the reference coordinates.
As shown in fig. 3, the method for performing sound source directional speech enhancement processing on a first audio signal by the CPU processing module 4 includes the following steps:
1) the CPU processing module 4 selects a microphone closest to a sound source as a target microphone according to the calculated sound source position or a sound source position manually designated by a user, and the target microphone collects and converts the sound source to obtain a target audio electrical signal;
2) the CPU processing module 4 converts other audio electric signals obtained by collecting and converting the same sound source by other microphones and the target audio electric signal obtained in the step 1) into other frequency spectrums and target frequency spectrums, and compares the other frequencies with the target frequency spectrums;
3) the CPU processing module 4 amplifies the amplitude corresponding to the frequency of the target spectrum that is earlier in phase than the other spectra, and the CPU processing module 4 reduces the amplitude corresponding to the frequency of the target spectrum that is later in phase than the other spectra.
As a preferred scheme, the environment sound includes that speaker module 5 plays the sound of the second audio signal after environment sound filtering processing and other sounds around the sound source, and other sounds around the sound source in the environment sound are disposed after the above-mentioned sound source directionality speech enhancement processing of first audio signal, and only leave the sound of the second audio signal after environment sound filtering processing played by speaker module 5 in the environment sound, first audio signal still includes that pickup module 1 gathers the third audio signal that the environment sound converted into.
Since it is possible that both the opposite party and the own party are speaking during the conversation, when the speaker module 5 plays the second audio frequency after the ambient sound filtering processing, the sound pickup module 1 collects the voice of the own party and also collects the sound emitted by the loudspeaker module 5, that is, the first audio signal at this time includes the third audio signal, if the first audio signal without being filtered by the environmental sound is directly sent out, which is equivalent to transmitting the voice of the other party back and forth, so howling is easily formed, resulting in unclear voice during the call, therefore, the CPU processing module 4 needs to filter the third audio signal portion of the first audio signal, so that the final audio signal does not include the sound emitted from the speaker module 5 (i.e. the environmental sound), only the sound of the user (i.e. the sound source) remains, therefore, the howling phenomenon can be avoided, the call can be clear, and a better use experience is provided for the user.
As shown in fig. 4, the method for filtering out the ambient sound from the first audio processed by the CPU processing module 4 through the sound source directional speech enhancement includes the following steps:
1) the CPU processing module 4 compares the first audio signal and the second audio signal after the sound source directivity voice enhancement processing;
2) the CPU processing module 4 filters a third audio signal similar to the second audio signal in the first audio signal after the sound source directivity voice enhancement processing, and obtains a final audio signal.
As shown in fig. 5, the method for performing the ambient sound filtering processing on the second audio signal by the CPU processing module 4 includes the following steps:
1) the CPU processing module 4 converts the second audio signal into a first frequency spectrum;
2) the CPU processing module 4 filters the signals of the first frequency spectrum obtained in the step 1) within the frequency range of 200-3000 Hz.
Because the voice frequency of the speaking of the person is mainly concentrated between 200 and 3000Hz, the normal hearing range of the ears of the person can reach 30-18000Hz, and the conference machine adopted by the opposite party is not necessary to have the noise reduction function, the voice transmitted by the opposite party may contain other environmental sounds except the voice of the speaking of the person, so that the voice transmitted by the opposite party is unclear, therefore, the CPU processing module 4 extracts the frequency band of 200 and 3000Hz, and filters the frequency bands except 200 and 3000Hz, so that the environmental sound can be effectively filtered, and the voice transmitted by the opposite party is ensured to be clear.
The above description is only a preferred embodiment of the present invention, and all the minor modifications, equivalent changes and modifications made to the above embodiment according to the technical solution of the present invention are within the scope of the technical solution of the present invention.