CN218550164U

CN218550164U - Easy-to-use video conference audio digital mixer

Info

Publication number: CN218550164U
Application number: CN202223110684.XU
Authority: CN
Inventors: 周波
Original assignee: Shenzhen Guanglian Yunzhi Technology Co ltd
Current assignee: Shenzhen Guanglian Yunzhi Technology Co ltd
Priority date: 2022-11-23
Filing date: 2022-11-23
Publication date: 2023-02-28
Anticipated expiration: 2032-11-23

Abstract

An easy-to-use audio digital mixer for video conference belongs to the technical field of audio signal processing. Including the DSP unit, its characterized in that: the DSP unit comprises an automatic mixer module, a noise gate module, an equalizer module and a matrix mixing module which are sequentially connected, wherein a microphone interface is connected with the input end of the automatic mixer module, an audio input interface is connected with the noise gate module, and the output end of the matrix mixing module is connected with an audio output interface; the MCU is connected with a plurality of adjusting knobs for adjusting the input or output volume of different ports at the input end of the MCU. In the easy-to-use audio digital mixer for the video conference, the volume of the input port or the output port is adjusted only by the knobs for adjusting the volume, the defect that a sound console and an audio matrix processor in the prior art are difficult to directly adjust is overcome, and the use difficulty of non-professional users can be effectively and greatly reduced.

Description

Easy-to-use video conference audio digital mixer

Technical Field

An easy-to-use audio digital mixer for video conference belongs to the technical field of audio signal processing.

Background

Video conferencing systems have become the mainstream means of remote communication in recent years. In video conferencing, audio interactive communication is very important besides video images, and the importance of audio interactive communication is even more important than video images. When a video conference is started, a microphone and a loudspeaker of a notebook computer can be used for intercommunication for individual users, but the microphone and the loudspeaker of the notebook computer are too small in sound pickup distance and loudspeaker power for various conference rooms, so that a conference room sound system is needed.

The sound system of a conference room generally includes main devices such as an audio signal input (e.g., a microphone), an audio signal conditioning (e.g., a sound console or an audio matrix processor), and an audio signal output (e.g., sound). When a video conference needs to be held in a conference room, a video conference hardware terminal or a computer generally needs to be in butt joint with a sound console or an audio matrix processor in a sound system of the conference room.

When a sound console (audio matrix processor) is adjusted, the sound console (audio matrix processor) comprises an input part and an output part, wherein the input part comprises a plurality of channels, each channel comprises a plurality of interfaces, a knob, a key and a push rod according to a path through which a signal flows, the MIC IN interface and the LINE IN interface respectively (the two interfaces of the sound console are made into a composite interface and provided with PAD attenuation buttons), a GAIN GAIN adjusting knob, a high pitch, a middle pitch and a low pitch GAIN adjusting knob AUX1-4, a front push rod key, a rear push rod key, a sound direction adjusting selection knob, a mute key, a push rod, a grouping 1-4 key, a sending main output key, a monitoring key and the like. The output part is divided into a main output channel and a push rod, an AUX auxiliary output channel and a main adjusting knob, a grouping output channel and a push rod, etc.

Therefore, there is no function of adjusting a certain volume in the sound console, when a certain volume is adjusted, adjustment of a plurality of function knobs (keys and push rods) is required, and various buttons, push rods, signal lamps with various colors and interfaces with different functions are dazzlingly disturbed, so that it is difficult for operators who are not audio professionals. However, the operation manual is too specialized, the operation difficulty is high, the operation is difficult, and other problems, and in addition, the sound console or the audio matrix needs to be connected with the audio part of the video conference terminal, so that the debugging is very complicated. Therefore, it is a technical solution to design a method for facilitating audio adjustment by non-professionals for video conferencing, which is a problem to be solved in the field.

Disclosure of Invention

The to-be-solved technical problem of the utility model is: the digital audio processing system overcomes the defects of the prior art, realizes the adjustment of the volume of the input port or the output port only through a plurality of knobs for adjusting the volume, overcomes the defects that in a sound console and an audio matrix processor in the prior art, complicated and numerous knobs and audio matrix panels cannot be directly adjusted, can effectively and greatly reduce the use difficulty of non-professional users, and can quickly build the digital audio processing system of the video conference audio system.

The utility model provides a technical scheme that its technical problem adopted is: this easy video conferencing audio digital mixer who uses, including the DSP unit, its characterized in that: the DSP unit comprises an automatic mixer module, a noise gate module, an equalizer module and a matrix mixing module which are sequentially connected, wherein a microphone interface is connected with the input end of the automatic mixer module, an audio input interface is connected with the noise gate module, and the output end of the matrix mixing module is connected with an audio output interface; the intelligent conference system is characterized in that the intelligent conference system is also provided with an MCU unit which is bidirectionally connected with the DSP unit, the input end of the MCU unit is connected with a plurality of adjusting knobs which are used for adjusting the input or output volumes of different ports, and the five adjusting knobs are respectively used for adjusting the volume of a local microphone, the volume of local audio, the volume of sound of a remote conference place, the volume of a microphone sent to the remote conference place and the volume of audio sent to the remote conference place.

Preferably, a USB interface unit which is connected with the DSP unit in a bidirectional mode is arranged, the USB interface unit is connected with the DSP unit in the bidirectional mode through a USB controller module, and the USB interface unit comprises a first USB interface and a second USB interface.

Preferably, the microphone interface and the audio input interface are respectively connected with the automatic mixer module and the noise gate module through an analog-to-digital conversion module.

Preferably, the matrix sound mixing module is connected with the audio output interface through a digital-to-analog conversion module.

Preferably, an HDMI input interface and an HDMI output interface are arranged, the HDMI input interface and the HDMI output interface are connected with the video signal processing module, and the video signal processing module is in bidirectional connection with the USB controller module.

Preferably, the output end of the MCU unit is connected with a plurality of indicator lamps.

Preferably, a PAD signal attenuation button connected with the DSP unit is further provided.

Compared with the prior art, the utility model discloses the beneficial effect who has is:

in the easy-to-use video conference audio digital mixer, the volume of the input port or the output port is adjusted only through the knobs for adjusting the volume, the defects that in a sound console and an audio matrix processor in the prior art, complicated and numerous knobs and audio matrix panels cannot be directly adjusted are overcome, the complicated operation of the sound console and the audio matrix can be greatly improved, the use difficulty of non-professional users can be effectively and greatly reduced, and a video conference audio system is rapidly built.

In this easy-to-use video conference audio digital mixer, only 5 adjustment knobs remain: the system comprises a local microphone volume adjusting knob, a local audio volume adjusting knob, a sound volume adjusting knob of a remote meeting place, a microphone volume adjusting knob sent to the remote meeting place and an audio volume adjusting knob sent to the remote meeting place.

In the easy-to-use video conference audio digital mixer, only necessary interfaces are reserved, the reserved interfaces can be directly connected with audio input and output interfaces of video conference terminals, most hardware video conference terminals can be directly connected, like the Brillop series, the Hua HDX series, the Hua TE series, the BOX series and the like, and audio local output interfaces are also reserved for connecting local audio equipment. The system can be directly connected with a computer, a mobile phone and a tablet computer through a USB (universal serial bus) line, and supports most software video conference terminals such as tench conferences, nails, zoom conferences and the like.

Drawings

Fig. 1 is a schematic diagram of a front panel of a video conference audio digital mixer that is easy to use.

Fig. 2 is a schematic diagram of a back panel of a video conference audio digital mixer for easy use.

Fig. 3 is a schematic block diagram of an easy-to-use video conference audio digital mixer control circuit.

Wherein: 1. the device comprises an indicator lamp 2, a shell 3, an adjusting knob 4, a power interface 5, a 6.35 output interface 6, a PAD signal attenuation button 7, a first USB interface 8, a second USB interface 9, an HDMI interface 10, a first audio interface 11, a second audio interface 12, a microphone interface 13 and a linear audio output interface.

Detailed Description

Fig. 1~3 is a preferred embodiment of the present invention, which is further illustrated in conjunction with 1~3.

Example 1:

an easy-to-use audio digital mixer for video conference (hereinafter referred to as digital audio processing system) comprises a housing 2 and a control circuit arranged in the housing 2, as shown in 1~2, a plurality of indicator lamps 1 are arranged on one side of a front panel of the digital audio processing system, and a plurality of adjusting knobs 3 are arranged on the side portions of the indicator lamps 1 side by side. In the digital audio processing system, five adjusting knobs 3 are provided for adjusting the local microphone volume, the local audio volume, the sound volume of the remote conference room, the microphone volume sent to the remote conference room, and the audio volume sent to the remote conference room, respectively. The indicator light 1 and the adjusting knob 3 are respectively connected into the control circuit.

The middle of the back of the shell 2 is provided with ase:Sub>A USB interface unit, the USB interface unit comprises ase:Sub>A second USB interface 8, the second USB interface 8 is ase:Sub>A USB-B type interface and is used for continuously clicking ase:Sub>A computer, two first USB interfaces 7 are arranged above the second USB interface 8 in parallel, and the first USB interfaces 7 are USB-A type interfaces and are used for connecting ase:Sub>A USB peripheral. Two sides of the upper first USB interface 7 are respectively provided with one HDMI interface 9, and the two HDMI interfaces 9 are respectively used for inputting and outputting video signals.

Two first audio interfaces 10 and two second audio interfaces 11 are arranged side by side from the top on the right side of the USB unit, the two first audio interfaces 10 are used for accessing linear audio signals, and the two second audio interfaces 11 are only used for inputting locally output linear audio signals. On the right side of the second audio interface 11 a plurality of microphone interfaces 12 are arranged side by side.

The linear audio output interface 13 is arranged on the left side of the USB unit in parallel up and down, and the two 6.35 output interfaces 5 are arranged on the left side of the linear audio output interface 13 in parallel up and down. A PAD signal attenuation button 6 is provided on the right side of the linear audio output interface 13 and the 6.35 output interface 5, respectively. 6.35 the right side of the output interface 5 is the power interface 4.

As shown in fig. 3, the control circuit includes a DSP unit and an MCU unit, and an input/output port of the DSP unit is bidirectionally connected to an input/output port of the MCU unit. The adjusting knob 3: the first knob, the second knob, the … … and the fifth knob are respectively connected with the analog-to-digital conversion module through the following steps: the first knob analog-to-digital conversion, the second knob analog-to-digital conversion, … … and the fifth knob analog-to-digital conversion are connected with different input ends of the MCU unit, the output end of the MCU unit is connected with the indicator lamp 1, and the USB debugging interface 4 is connected with the MCU unit in a bidirectional mode. The MCU unit may be implemented using a microprocessor commonly available in the art, such as an ST32 series microprocessor.

In the DSP unit, the microphone interface 12 is connected to an input terminal of the first analog-to-digital conversion module, an output terminal of the first analog-to-digital conversion module is connected to an input terminal of the automatic mixer module, and an output terminal of the automatic mixer module is connected to an input terminal of the noise gate module. The output end of the noise gate module is connected with the input end of the equalizer module, the output end of the equalizer module is connected with the input end of the matrix sound mixing module, the output end of the matrix sound mixing module is connected with the input end of the digital-to-analog conversion module, and the output end of the digital-to-analog conversion module is connected with the audio output interface (comprising a linear audio output interface 13 and a 6.35 output interface 5).

The audio input interface (including the first audio interface 10, the second audio interface 11 is connected to the input end of the second analog-to-digital conversion module, the output end of the second analog-to-digital conversion module is connected to the other input end of the noise gate module, the USB socket unit (the first USB interface 7 and the second USB interface 8) is bidirectionally connected to the DSP unit through the USB controller module.

The USB controller module comprises a USB audio codec. The USB video coder-decoder and the USBI/O port, the first USB interface 7 and the second USB interface 8 are communicated with the USB controller module through the USBI/O port. The HDMI interface 9 (including HDMI input interface and HDMI output interface) is connected to the video signal processing module, and the video signal processing module is connected to the USB controller module in both directions. The two PAD signal attenuation buttons 6 are simultaneously connected to the MCU unit and act on the devices accessed via the 6.35 output interface 5 and the linear audio output interface 13, respectively.

In the DSP unit, an automatic mixer module increases the input gain of a microphone by 3dB every time one more microphone is opened in a meeting place according to the sound amplification principle, and the howling problem between the microphone and a sound box is easily caused along with the increase of the opening number. The automatic microphone mixing module can automatically control the gains of a plurality of microphones (including but not limited to 4 microphones) in real time, even in the case that a plurality of speakers speak simultaneously, the automatic microphone mixing module can keep the integral output of the automatic mixing module equal to the gain when only one microphone is turned on, thereby obviously reducing howling caused by positive feedback and environment noise and comb filtering generated by adjacent microphones and not changing the sound quality.

In the noise gate module in the DSP unit, when a plurality of microphone signals are automatically mixed, the signal-to-noise ratio of the signals is not changed, and the signals sent by the microphones to the automatic mixing module include the own background noise and the environmental noise of the microphones, and these noises are present although the volume of the noises is small. The noise gate is a volume valve between the automatic sound mixing module and the ADC conversion module, and the main function of the noise gate is a valve set for preventing background noise from entering a sound amplification system of a local or remote meeting place, which is simply understood as follows: when the input signal of the microphone is large, the volume valve is opened, and when the input signal of the microphone is small, the volume valve is closed.

When the microphone is opened, the noise volume reaches a set threshold value, the noise gate cannot be opened, the noise gate can reject background noise and meeting place environment sound picked up by the microphone, when the speaker speaks, the volume exceeds the threshold value, the noise gate is opened, signals enter the next module through the noise gate, and the input signal-to-noise ratio of the microphone is greatly improved due to the existence of the noise gate.

The equalizer in the DSP unit is used for modifying sound and making up the defects caused by the acoustic environment, and the design of the invention is mainly aimed at conferences, and the reproduction of human voice is the main target to be pursued, so the equalizer does not modify the sound, and makes up the defects caused by the acoustic environment, and is simple to process the howling point and prevent the howling.

The matrix sound mixing module in the DSP unit has the main function of sending one or more paths of audio input signals to one or more paths of output interfaces at will. And respectively sending the audio input to the local audio output interface and the remote audio output interface through the matrix audio mixing module.

The specific working process and working principle are as follows:

in the in-service use, when needing to adjust the volume, operating personnel rotates corresponding adjust knob 3, adjust knob 3 rotates the back, and its resistance changes, and the signal of telecommunication (voltage value or current value) of inputing to the MCU unit changes, and after the MCU unit received this volume signal regulating value, according to the volume signal regulating value, send corresponding regulating instruction to the DSP unit again, the DSP unit carries out corresponding regulation to the volume of output interface according to the volume signal regulating value that the MCU unit sent, specifically:

when the volume of the microphone transmitted to the remote meeting place needs to be adjusted, the corresponding adjusting knob 3 is rotated. When the adjusting knob 3 is rotated, the resistance value of the potentiometer in the adjusting knob 3 also changes correspondingly, the resistance value can be collected by the MCU system and converted into a digital signal by the ADC module, the converted digital signal enters the MCU system module, the MCU system can send the digital signal to the DSP system module, the DSP system module receives the value of the digital signal sent by the MCU, the volume of a microphone (input through the microphone interface 12) in the DSP system module is adjusted and then output to the matrix sound mixer, the volume is sent to the DAC module again and converted into an analog signal, and finally the analog signal is output through the 6.35 output interface 5.

When the volume of the audio transmitted to the remote conference room needs to be adjusted, the corresponding adjusting knob 3 is rotated. When the adjusting knob 3 is rotated, the resistance value of the potentiometer in the adjusting knob 3 also changes correspondingly, the resistance value can be collected by the MCU system and converted into a digital signal by the ADC module, the converted digital signal enters the MCU system module, the MCU system can send the digital signal to the DSP system module, the DSP system module receives the value of the digital signal sent by the MCU, the volume of a linear signal (through the second audio interface 11) entering the DSP system module is adjusted and then output to the matrix sound mixer, the linear signal is sent to the DAC module and converted into an analog signal, and finally the analog signal is output through the interface 6.35 output interface 5.

When the local audio volume needs to be adjusted, the corresponding adjusting knob 3 is rotated. When the adjusting knob 3 is rotated, the resistance value of the potentiometer in the adjusting knob 3 also changes correspondingly, the resistance value can be collected by the MCU system and converted into a digital signal by the ADC module, the converted digital signal enters the MCU system module, the MCU system can send the digital signal to the DSP system module, the DSP system module receives the value of the digital signal sent by the MCU, the volume of a linear audio signal entering the DSP system module or a USB audio signal from a computer is adjusted and then output to the DAC module to be converted into an analog signal, and finally the analog signal is output through the 6.35 output interface 5.

When the local microphone volume needs to be adjusted, the corresponding adjusting knob 3 is rotated. When the adjusting knob 3 is rotated, the resistance value of the potentiometer in the adjusting knob 3 also changes correspondingly, the resistance value can be collected by the MCU system and converted into a digital signal by the ADC module, the converted digital signal enters the MCU system module, the MCU system can send the digital signal to the DSP system module, the DSP system module receives the value of the digital signal sent by the MCU, the volume of a microphone (input through the microphone interface 12) in the DSP system module is adjusted and then output to the matrix sound mixer, and then the volume is sent to the DAC module to be converted into an analog signal, and finally the analog signal is output through the linear audio output interface 13.

Example 2:

this example differs from example 1 in that: in this embodiment, the automatic mixer module, the noise gate module, the equalizer module, and the matrix mixing module may be implemented by conventional analog devices without using a DSP unit.

While the invention has been described with reference to a preferred embodiment, it will be understood by those skilled in the art that various changes in form and detail may be made without departing from the spirit and scope of the invention. However, any simple modification, equivalent change and modification made to the above embodiments according to the technical substance of the present invention still belong to the protection scope of the technical solution of the present invention.

Claims

1. An easy-to-use audio digital mixer for video conferencing, comprising a DSP unit, characterized in that: the DSP unit comprises an automatic mixer module, a noise gate module, an equalizer module and a matrix mixing module which are connected in sequence, wherein a microphone interface (12) is connected with the input end of the automatic mixer module, an audio input interface is connected with the noise gate module, and the output end of the matrix mixing module is connected with an audio output interface; the intelligent conference system is characterized in that the intelligent conference system is also provided with an MCU unit which is bidirectionally connected with the DSP unit, the input end of the MCU unit is connected with a plurality of adjusting knobs (3) which are used for adjusting the input or output volumes of different ports, and the five adjusting knobs (3) are respectively used for adjusting the volume of a local microphone, the volume of local audio, the volume of sound of a remote conference place, the volume of a microphone sent to the remote conference place and the volume of audio sent to the remote conference place.

2. An easy to use video conferencing audio-digital mixer according to claim 1, wherein: the USB interface unit is connected with the DSP unit in a bidirectional mode and comprises a first USB interface (7) and a second USB interface (8).

3. An easy to use video conferencing audio-digital mixer according to claim 1, wherein: the microphone interface (12) and the audio input interface are respectively connected with the automatic mixer module and the noise gate module through the analog-to-digital conversion module.

4. An easy-to-use video conferencing audio-digital mixer according to claim 1, wherein: and the matrix sound mixing module is connected with the audio output interface through the digital-to-analog conversion module.

5. Easy to use video conferencing audio-digital mixer according to claim 2, characterized in that: the USB controller module is provided with an HDMI input interface and an HDMI output interface, the HDMI input interface and the HDMI output interface are connected with the video signal processing module, and the video signal processing module is in bidirectional connection with the USB controller module.

6. An easy-to-use video conferencing audio-digital mixer according to claim 1, wherein: and the output end of the MCU unit is connected with a plurality of indicator lamps.

7. An easy-to-use video conferencing audio-digital mixer according to claim 1, wherein: and a PAD signal attenuation button connected with the DSP unit is also arranged.