CN114584659A - Audio processing unit of conference system - Google Patents

Abstract

The invention discloses an audio processing unit of a conference system, comprising: the system comprises a first chip, a second chip, an MCU and a host; the system comprises a first chip, a second chip, an MCU and a host; the first chip is used for sending and receiving audio data; the second chip is used for collecting and monitoring audio data; the MCU is used for transmitting the audio data of the first chip to the second chip for monitoring, and is also used for transmitting the audio data collected by the second chip to the first chip and then transmitting the audio data to the host. The invention overcomes the problems of high cost and easy data packet drop existing in the audio processing by adopting the Dante protocol through the action of the first chip and the second chip, and has the advantage of low-delay audio transmission.

Description

Audio processing unit of conference system

Technical Field

The present invention relates to the field of communications technologies, and in particular, to an audio processing unit of a conference system.

Background

With the continuous development of network technology, the conference system gradually converts pure analog audio into network digital audio. The network digital audio transmission mainly comprises two modes of a wired network and a wireless network, wherein the wireless conference system network mainly comprises 2.4G, U segments, infrared and WiFi; the wired conference unit mainly adopts a Dante module to perform network transmission and audio decoding of audio. However, the wireless 2.4G and U sections have poor anti-interference capability and confidentiality, and the transmission distance is very limited; the infrared wireless conference system can only transmit in a sight line range, and has strong limitation; the Dante module is not only costly, but also is prone to packet loss in transmission once the data size is large.

Disclosure of Invention

The invention aims to provide an audio processing unit of a conference system, which aims to solve the problems of high cost and high data volume and easy transmission and packet drop when a digital wired conference unit in the prior art adopts a Dante module to decode audio.

To achieve the above object, the present invention provides an audio processing unit of a conference system, comprising:

the system comprises a first chip, a second chip, an MCU and a host;

the first chip is used for sending and receiving audio data; the second chip is used for collecting and monitoring audio data;

the MCU is used for transmitting the audio data of the first chip to the second chip for monitoring, and is also used for transmitting the audio data collected by the second chip to the first chip and then transmitting the audio data to the host.

Further, preferably, the second chip is further configured to:

and converting the analog signal of the audio data into a digital signal after the analog signal is amplified by the operational amplifier circuit, and transmitting the digital signal to the MCU.

Further, preferably, the MCU is further configured to:

and sending the digital signal of the audio data sent by the host to the second chip, converting the digital signal into a corresponding analog signal, and outputting the analog signal through the operational amplifier circuit.

Further, preferably, the first chip is a LAN 8720.

Further, preferably, the second chip is WM 8978.

Further, preferably, the second chip is further configured to collect and monitor audio data in a 16-bit 48KHz sampling frequency and 15mS audio delay manner.

Further, preferably, the first chip and the host use a UDP protocol for audio data interaction.

Further, preferably, the MCU is further configured to process the message queue in a blocking manner, including:

converting the audio data collected by the second chip into sound decibels;

and determining whether to transmit audio data according to the sound decibel.

Further, preferably, the MCU is further configured to:

when the sound decibel is greater than or equal to a first preset value, recording the current moment;

and when the sound decibel is smaller than a first preset value and lasts for a preset time, zeroing the current moment and confirming that audio data transmission is not carried out.

Compared with the prior art, the invention has the beneficial effects that:

1) the 16-bit 48KHz sampling frequency and the audio delay within 15mS are adopted, so that the characteristic of no delay in audio transmission is ensured;

2) the uncompressed network audio transmission adopts UDP protocol to carry out data interaction with the host in the Ethernet, and has a mechanism for controlling data retransmission, thereby preventing the problem of data lack of reception; supporting unicast and multicast communication with a host;

3) the conference system has rich conference application functions, supports speaking and voting functions, and meets the requirements of daily conference application through a conference sign-in function;

4) the voting function is supported, the data is supported to be updated and displayed in real time in the background of the PC end, and the result can be displayed in a text mode, a histogram mode and a pie mode; the voting result is supported to be projected, amplified and displayed;

5) by adopting a Poe power supply mode, a plurality of switches can be expanded to realize a distributed effect;

6) a multiple unit simultaneous speak function may be performed.

Drawings

In order to more clearly illustrate the technical solution of the present invention, the drawings needed to be used in the embodiments will be briefly described below, and it is obvious that the drawings in the following description are only some embodiments of the present invention, and it is obvious for those skilled in the art that other drawings can be obtained according to the drawings without creative efforts.

Fig. 1 is a schematic structural diagram of an audio processing unit of a conference system according to an embodiment of the present invention;

fig. 2 is a schematic structural diagram of an audio processing unit of a conference system according to yet another embodiment of the present invention.

Detailed Description

The technical solutions in the embodiments of the present invention will be clearly and completely described below with reference to the drawings in the embodiments of the present invention, and it is obvious that the described embodiments are only a part of the embodiments of the present invention, and not all of the embodiments. All other embodiments, which can be derived by a person skilled in the art from the embodiments given herein without making any creative effort, shall fall within the protection scope of the present invention.

It should be understood that the step numbers used herein are for convenience of description only and are not intended as limitations on the order in which the steps are performed.

It is to be understood that the terminology used in the description of the invention herein is for the purpose of describing particular embodiments only and is not intended to be limiting of the invention. As used in the specification of the present invention and the appended claims, the singular forms "a," "an," and "the" are intended to include the plural forms as well, unless the context clearly indicates otherwise.

The terms "comprises" and "comprising" indicate the presence of the described features, integers, steps, operations, elements, and/or components, but do not preclude the presence or addition of one or more other features, integers, steps, operations, elements, components, and/or groups thereof.

The term "and/or" refers to and includes any and all possible combinations of one or more of the associated listed items.

Referring to fig. 1, an embodiment of the invention provides an audio processing unit of a conference system. As shown in fig. 1, the audio processing unit of the conference system includes:

the system comprises a first chip 02, a second chip 04, an MCU03 and a host 01;

the first chip 02 is used for transmitting and receiving audio data; the second chip 04 is used for collecting and monitoring audio data;

the MCU03 is configured to transmit the audio data of the first chip 02 to the second chip 04 for monitoring, and transmit the audio data collected by the second chip 04 to the first chip 02 and then to the host 01.

The first chip 02 employs a LAN8720, and the second chip 04 employs WM 8978.

It should be noted that, the existing network for conference audio transmission includes segment 2.4G, U, infrared, WiFi, etc., but each of them has its own disadvantages, which specifically include:

1) the biggest defects of wireless 2.4G and U sections are poor interference resistance and confidentiality, the problems of audio distortion, noise, secret leakage and the like are easily caused, and signals are weakened along with the increase of transmission distance, so the transmission distance is short.

2) The infrared wireless conference system has a short transmission distance because the optical characteristics of infrared rays cannot penetrate through opaque obstacles and can only transmit in a sight range.

3) The non-digital wired conference unit audio transmits the audio in an analog mode, a hardware circuit cannot be modularized, the transmission distance is far, the non-digital wired conference unit audio is easily interfered, a control instruction uses 485 communication, although the 485 communication has certain anti-interference performance, the 485 communication belongs to half-duplex communication, and phenomena such as packet drop and the like are easily caused when the data volume is large.

4) At present, the Dante modules adopted by most of the digital wired conference units on the market carry out network transmission and audio decoding of audio, the MCU03 controls the Dante modules through serial ports, the data volume is large, the problem of packet drop can be caused, the chip cost of the Dante scheme is higher, and the production cost of products is increased.

5) In addition to the Dante protocol, the AVB protocol is also used, but the AVB protocol is based on the transmission of Layer2 and can only be used by a special switch, which is not beneficial to popularization.

Therefore, in order to overcome the defects in the various manners described above, an embodiment of the present invention provides an audio processing unit of a conference system, including: the system comprises a first chip 02, a second chip 04, an MCU03 and a host 01; the first chip 02 employs a LAN8720, and the second chip 04 employs WM 8978.

In one embodiment, the second chip 04 is further configured to:

and converting the analog signal of the audio data into a digital signal after being amplified by the operational amplifier circuit, and transmitting the digital signal to the MCU 03.

The MCU03 is further configured to send the digital signal of the audio data sent by the host 01 to the second chip 04, convert the digital signal into a corresponding analog signal, and output the analog signal through the operational amplifier circuit.

Further, the second chip 04 is also used for collecting and monitoring audio data by adopting a 16-bit 48KHz sampling frequency and an audio delay within 15 mS.

Specifically, the implementation of the audio acquisition process in this embodiment includes: an analog signal input by a microphone is amplified by an operational amplifier circuit and then is connected to a LINEIN end of WM S, the analog signal is converted into a digital signal by WM S and then is transmitted to an MCU03 through I2S of WM S, and the MCU03 is transmitted to a host 01 through a network; the audio monitoring process mainly comprises the following steps: the MCU03 receives audio digital signal data sent by the host 01 through the network, transmits the audio digital signal data to the WM 03 through the I2S interface to convert the audio digital signal data into an analog signal, outputs the analog signal through the operational amplifier circuit, and finally realizes monitoring.

In one embodiment, when an audio processing unit of the conference system is used for acquiring audio, software starts an ADC configuration of a WM8978 sound card to sample audio data. Preferably, WM8978 receives the original audio in 16-bit 48KHz sampling frequency with audio delay of 15 mS. The software starts I2S to receive interrupt, starts DMA (direct memory access) data double buffer, after the I2S receives interrupt and receives data, the received data is sent out through a message queue, the speaking host 01 allocates an IP far-end IP to the speaking unit each time, and the unit sends audio data to the far-end IP.

It should be noted that DMA (direct memory access) is used to provide high-speed data transfer between the peripheral and the memory and between the memory and the memory, and data can be moved quickly by DMA without any CPU operation, thus saving CPU resources for other devices.

Further, when the receiving interruption comes, the data acquisition is automatically transferred to another buffer area, and the currently acquired audio data is generated to a message queue.

In one embodiment, the MCU03 is also configured to handle message queues in a blocking manner, including:

converting the original audio into sound decibels;

confirming whether audio data transmission is carried out according to the sound decibel, and specifically comprising the following steps:

when the sound decibel is greater than or equal to a first preset value, recording the current moment;

and when the sound decibel is smaller than a first preset value and lasts for a preset time, zeroing the current moment and confirming that audio data transmission is not carried out.

In practical application, for the message queue processing thread, the MCU03 mainly receives the message from the message queue in a blocking manner, converts the PCM audio data into sound decibels after receiving the message, and determines whether to transmit the audio data according to the sound decibels. And if the current time T is recorded and the sound decibel is continuously less than 50 decibels 2S after exceeding 50 decibels, setting the T to 0 and not transmitting data any more. Therefore, under the environment without speaking of the user, the noise in the environment is automatically filtered without transmitting data.

In an alternative embodiment, the first chip 02 and the host 01 use UDP protocol for audio data interaction.

In this embodiment, the first chip 02 and the host 01 mainly use a UDP protocol for data interaction. Specifically, when an audio processing unit of the conference system monitors audio, software starts the DAC configuration of the WM 03 sound card, the unit receives audio data sent by the host 01 through a UDP protocol, and the MCU03 transmits the received data to the WM 03 through the I2S interface, controls the monitoring volume through the digital tuning chip, amplifies the data through the analog circuit, and transmits the amplified data to the monitoring speaker for monitoring.

Therefore, with the audio processing unit of the conference system provided by the embodiment, at least the following functions can be implemented:

(1) the ADC adopts multi-bit feedback and a higher sampling rate to reduce pulse jitter and high-frequency noise, high-quality audio data of 48KHz is sampled at the maximum, and an audio decibel screening algorithm is used to filter out signals with lower decibels, so that bottom noise interference is further prevented, and more fidelity tone quality is ensured;

(2) the conference system has rich conference application functions, supports speaking and voting functions, and meets the requirements of daily conference application through a conference sign-in function;

(3) the uncompressed network audio transmission adopts UDP protocol to carry out data interaction with the host in the Ethernet, and has a mechanism for controlling data retransmission, thereby preventing the problem of data lack of reception; unicast and multicast communication with the host is supported;

(4) the voting function is supported, the data is supported to be updated and displayed in real time in the background of the PC end, and the result can be displayed in a text mode, a histogram mode and a pie mode; the voting result is supported to be projected, amplified and displayed;

(5) by adopting a Poe power supply mode, a plurality of switches can be expanded to realize a distributed effect;

(6) at most 6 units can speak simultaneously; the units can be a chairman representative unit, a chairman speaking unit, a representative unit and a speaking unit. The chairman unit has a priority control right and a platform mode, and in addition, the chairman representative unit also has a right of starting voting, and voting, election and rating modes are supported by voting;

(7) and a 2.4-inch oled high-definition screen is adopted to display the state information, the ID number, the unit type and the like of the unit in real time.

In the several embodiments provided in the present application, it should be understood that the disclosed system, apparatus and method may be implemented in other ways. For example, the above-described apparatus embodiments are merely illustrative, and for example, the division of the units is only one logical functional division, and there may be other divisions in actual practice, for example, multiple units or page components may be combined or integrated into another system, or some features may be omitted, or not executed. In addition, the shown or discussed mutual coupling or direct coupling or communication connection may be an indirect coupling or communication connection of devices or units through some interfaces, and may be in an electrical, mechanical or other form.

The units described as separate parts may or may not be physically separate, and parts displayed as units may or may not be physical units, may be located in one position, or may be distributed on multiple network units. Some or all of the units can be selected according to actual needs to achieve the purpose of the solution of the embodiment.

In addition, functional units in the embodiments of the present application may be integrated into one processing unit, or each unit may exist alone physically, or two or more units are integrated into one unit. The integrated unit may be implemented in the form of hardware, or in the form of hardware plus a software functional unit.

The integrated unit implemented in the form of a software functional unit may be stored in a computer readable storage medium. The software functional unit is stored in a storage medium and includes several instructions to enable a computer device (which may be a personal computer, a server, or a network device) or a processor (processor) to execute some steps of the methods according to the embodiments of the present application. And the aforementioned storage medium includes: various media capable of storing program codes, such as a usb disk, a removable hard disk, a Read-Only Memory (ROM), a Random Access Memory (RAM), a magnetic disk, or an optical disk.

Finally, it should be noted that the above embodiments are only used for illustrating the technical solutions of the present application, and not for limiting the same; although the present application has been described in detail with reference to the foregoing embodiments, it will be understood by those skilled in the art that various changes in form and details may be made therein without departing from the spirit and scope of the invention; and such modifications or substitutions do not depart from the spirit and scope of the corresponding technical solutions in the embodiments of the present application.

Claims (9)

1. An audio processing unit of a conferencing system, comprising:

the system comprises a first chip, a second chip, an MCU and a host;

the first chip is used for sending and receiving audio data; the second chip is used for collecting and monitoring audio data;

the MCU is used for transmitting the audio data of the first chip to the second chip for monitoring, and is also used for transmitting the audio data collected by the second chip to the first chip and then transmitting the audio data to the host.

2. The audio processing unit of the conference system as claimed in claim 1, wherein said second chip is further configured to:

and converting the analog signal of the audio data into a digital signal after the analog signal is amplified by the operational amplifier circuit, and transmitting the digital signal to the MCU.

3. The audio processing unit of the conference system of claim 1, wherein the MCU is further configured to:

and sending the digital signal of the audio data sent by the host to the second chip, converting the digital signal into a corresponding analog signal, and outputting the analog signal through the operational amplifier circuit.

4. The audio processing unit of the conference system as in claim 1, wherein said first chip is a LAN 8720.

5. Audio processing unit for a conferencing system according to any of the claims 1-4, wherein the second chip is WM 8978.

6. The audio processing unit of the conference system as claimed in claim 1, wherein said second chip is further configured to collect and listen to audio data in a 16-bit 48KHz sampling frequency with 15mS audio delay.

7. The audio processing unit of the conference system as claimed in claim 1, wherein the first chip and the host adopt UDP protocol for audio data interaction.

8. The audio processing unit of the conference system according to claim 1, wherein said MCU is further configured to process a message queue in a blocking manner, comprising:

converting the audio data collected by the second chip into sound decibels;

and determining whether to transmit audio data according to the sound decibel.

9. The audio processing unit of the conference system as claimed in claim 8, wherein said MCU is further configured to:

when the sound decibel is greater than or equal to a first preset value, recording the current moment;

and when the sound decibel is smaller than a first preset value and lasts for a preset time, zeroing the current moment and confirming that audio data transmission is not carried out.

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