WO2018095057A1 - Method, system and device for dsp-based dante digital audio processing - Google Patents

Method, system and device for dsp-based dante digital audio processing Download PDF

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WO2018095057A1
WO2018095057A1 PCT/CN2017/093282 CN2017093282W WO2018095057A1 WO 2018095057 A1 WO2018095057 A1 WO 2018095057A1 CN 2017093282 W CN2017093282 W CN 2017093282W WO 2018095057 A1 WO2018095057 A1 WO 2018095057A1
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audio signal
processing
input
dsp
audio
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PCT/CN2017/093282
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French (fr)
Chinese (zh)
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何欢潮
何伟峰
何图
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广州飞达音响股份有限公司
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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R5/00—Stereophonic arrangements
    • H04R5/04—Circuit arrangements, e.g. for selective connection of amplifier inputs/outputs to loudspeakers, for loudspeaker detection, or for adaptation of settings to personal preferences or hearing impairments

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  • the invention relates to the field of audio signal processing, and in particular to a Dante digital audio processing method, system and device based on DSP.
  • Ethernet-based digital audio transmission technology has become a technical focus of the professional audio industry, and is widely used in many projects because of its independent characteristics independent of the control system. On the one hand, it solves the problem of wiring difficulties in multiple lines, and also solves a series of problems that cannot be faced in the era of analog transmission, such as long-distance transmission, data backup, and automatic redundancy.
  • the more mature Ethernet audio transmission technologies mainly include CobraNet and EtherSound technologies, but both technologies have their own advantages.
  • Audinate has launched Dante, a digital audio transmission technology that incorporates many new technologies.
  • Dante is a digital audio transmission based on 3-layer IP network technology that provides a low latency, high precision and low cost solution for point-to-point audio connections.
  • Dante technology delivers high-precision clock signals as well as professional audio signals over Ethernet (100M or 1000M) and enables complex routing.
  • CobraNet and EtherSound such as uncompressed digital audio signal, which guarantees good sound quality; solves the complicated wiring problems in traditional audio transmission and reduces the cost;
  • Adapt to the existing network no special configuration is required; the audio signals in the network are marked in the form of "tags".
  • a combination of signals for audio processing is no analog audio signals, AES3 digital audio signals and Dante network audio in the prior art.
  • one of the objects of the present invention is to provide a DSP-based Dante digital audio processing method, which implements an optional input of an analog audio signal, an AES3 digital audio signal, and a Dante network audio signal. Independent audio level characteristic adjustment for four channels, and high-quality and low delay through multi-stage equalization adjustment.
  • a Dante digital audio processing method based on DSP comprising the following steps:
  • Step 1 Control and adjust an audio level characteristic of the input audio signal
  • Step 2 adjusting frequency response characteristics of each frequency band of the audio signal whose audio level characteristics are adjusted, so as to implement input equalization processing on the audio signal;
  • Step 3 performing frequency division processing on the audio signal after the input equalization processing
  • Step 4 Adjust frequency response characteristics of each frequency band of the frequency-divided audio signal to implement output equalization processing on the audio signal
  • the input equalization process is a 6-segment second-order IIR filter
  • the frequency division process includes a high-pass or low-pass selection, and gain and polarity adjustment control
  • the output equalization process is an 8-segment second-order IIR.
  • the filter and the 8-band second-order filter both adjust the frequency, bandwidth, and gain.
  • the audio signal input in step 1 is one or more of an analog audio signal, an AES3 digital audio signal, and a Dante digital audio signal.
  • the audio signal input in step 1 is divided into four channels through the channel routing system, and each channel performs independent audio level characteristic adjustment.
  • the input mode and the output mode are selectively selected, wherein the input mode includes stereo and signal addition, and the output modes include parallel and bridge.
  • step 4 the following steps are also included:
  • Step 5 performing delay processing on the audio signal after the output equalization processing, where the delay processing is adjusted by using time or length as an adjustment unit;
  • Step 6 Perform a limiting process on the delayed processed audio signal, and the limiting processing is one of off, -3 dB, -6 dB, and -12 dB.
  • Another object of the present invention is to provide a Dante digital audio processing system based on DSP, which realizes an optional input of an analog audio signal, an AES3 digital audio signal and a Dante network audio signal, and performs independent audio power through four channels.
  • the flat characteristic is adjusted, and the multi-stage equalization adjustment achieves high sound quality and low delay.
  • a DSP-based Dante digital audio processing system comprising:
  • a level control unit for controlling and adjusting an audio level characteristic of the input audio signal
  • the input equalizer unit is configured to adjust a frequency response characteristic of each frequency band of the audio signal whose audio level characteristic is adjusted, so as to implement input equalization processing on the audio signal;
  • a frequency dividing unit configured to perform frequency division processing on the audio signal after the input equalization processing
  • An output equalizer unit is configured to adjust a frequency response characteristic of each frequency band of the frequency-divided audio signal to implement output equalization processing on the audio signal;
  • the input equalization process is a 6-segment second-order IIR filter
  • the output equalization process is an 8-segment second-order IIR filter
  • the 8-segment second-order filter can adjust the frequency, the bandwidth, and the gain.
  • a delay unit configured to perform delay processing on the audio signal after the output equalization processing, where the delay processing is adjusted according to a time or length adjustment unit;
  • the limiting unit is configured to perform limiting processing on the delayed processed audio signal, and the limiting processing is one of off, -3dB, -6dB and -12dB.
  • Still another object of the present invention is to provide a DSP-based Dante digital audio processing device that implements an optional input of an analog audio signal, an AES3 digital audio signal, and a Dante network audio signal, by performing independent audio power on four channels.
  • the flat characteristic is adjusted, and the multi-stage equalization adjustment achieves high sound quality and low delay.
  • a DSP-based Dante digital audio processing device comprising:
  • An audio input module configured to input an audio signal to the DSP processor; wherein the audio input module includes:
  • a balanced analog signal input module connected to the DSP processor through an A/D converter for inputting an analog audio signal to the DSP processor;
  • An AES3 digital signal input module directly connected to the DSP processor for inputting an AES3 digital audio signal to the DSP processor;
  • a Dante network digital signal input module connected to the DSP processor through a digital audio processing module for inputting a Dante network digital audio signal to the DSP processor;
  • the power amplifier module is connected to the power amplifier module through a D/A converter to send the audio signal processed by the DSP processor to the power amplifier module for playing.
  • the measuring module the voltage detecting module, the current detecting module and the impedance detecting module, the temperature detecting module, the voltage detecting module, the current detecting module and the impedance detecting module are all connected to the DSP processor.
  • the touch control display module is further connected to the DSP processor for transmitting a touch control command to the DSP processor, and receiving the collection information of the temperature detecting module, the voltage detecting module, the current detecting module and the impedance detecting module. .
  • AES3 and Dante digital transmission card a variety of audio input methods, to meet the digital audio network transmission and amplification output; provide networked serial port, third-party remote control functions, also has automatic sleep and signal wake-up.
  • FIG. 1 is a schematic flow chart of a Dante digital audio processing method based on DSP according to an embodiment of the present invention
  • FIG. 2 is a schematic structural diagram of a DSP-based Dante digital audio processing device according to an embodiment of the present invention.
  • a DSP-based Dante digital audio processing method which includes the following step:
  • Step 1 signal input.
  • analog audio signal input AES3 digital audio signal input
  • Dante digital network audio signal input among which Dante digital network audio signal has low delay, no interference from traditional analog signal, and can distance up to 512 channel digital Simultaneous transmission of signals, point-to-point, or point-to-multipoint free-switching transmission.
  • the analog audio signal is sent to the DSP processor through the A/D converter, the AES3 digital audio signal directly through the cable and the Dante digital network audio signal through the corresponding Dante digital audio processing module, and the DSP processor performs the following steps 2-7. .
  • Step 2 Input mode (INPUT Mode) or output mode (OUTPUT Mode)
  • the channel routing algorithm is divided into four channels (CH1-CH4), and then the independent four channels are level-controlled.
  • the level control is the audio level characteristic of the input signal.
  • the input mode selection processing there are two modes: stereo, signal addition
  • the signal output mode selection processing there are two modes: Parallel, bridge mode.
  • Step 3 Adjust the frequency response characteristics of each frequency band of the audio signal whose audio level characteristics are adjusted, to perform input equalization (EQ) processing on the audio signal.
  • the input equalization consists of a 6-segment second-order IIR filter (referred to as 6-segment EQ) plus a high-low-pass filtering function.
  • the input 6-segment EQ, 2 of which are multi-type filter selection the multi-filter type includes (Linkwitz-Riley (or Butterworth or Bessel) / 6dB / oct, 12dB / oct, 18dB / oct, 24dB / oct /, 30dB / oct, 36dB / oct, 48dB / oct, etc.
  • Step 4 Perform frequency division (X-Over) processing on the audio signal after input equalization processing, and the type of frequency division has high-pass, low-pass selection, and control of gain and polarity adjustment.
  • Step 5 adjusting frequency response characteristics of each frequency band of the frequency-divided audio signal, To achieve output equalization (output EQ) processing of the audio signal.
  • the output equalization process is an 8-segment second-order IIR filter and the 8-band second-order filter can adjust the center frequency (frequency point) and the Q value (the Q value is equal to the center frequency divided by the bandwidth).
  • Step 6 Perform delay processing on the audio signal after the output equalization processing, and the delay may be up to 0.9 seconds, and the delay processing may be adjusted by adjusting the time or length.
  • Step 7 Perform a limiting process on the delayed audio signal, and the limiting processing is one of off, -3 dB, -6 dB, and -12 dB.
  • Step 8 The audio signal processed by the DSP processor is output (played) through the power amplifier unit.
  • the present invention also provides a DSP-based Dante digital audio processing system, which includes:
  • a level control unit for controlling an audio level characteristic of the adjusted input audio signal
  • the input equalizer unit is configured to adjust a frequency response characteristic of each frequency band of the audio signal whose audio level characteristic is adjusted, so as to implement input equalization processing on the audio signal;
  • a frequency dividing unit configured to perform frequency division processing on the audio signal after the input equalization processing
  • An output equalizer unit is configured to adjust a frequency response characteristic of each frequency band of the frequency-divided audio signal to implement output equalization processing on the audio signal;
  • a delay unit configured to perform delay processing on the audio signal after the output equalization processing, where the delay processing is adjusted according to a time or length adjustment unit;
  • the limiting unit is configured to perform limiting processing on the delayed processed audio signal, and the limiting processing is one of off, -3dB, -6dB and -12dB.
  • the present invention also provides a Dante digital audio processing device based on DSP. Set, which includes:
  • An audio input module configured to input an audio signal to the DSP processor; wherein the audio input module comprises: a balanced analog signal input module connected to the DSP processor through an A/D converter for inputting a simulation to the DSP processor Audio signal; AES3 digital signal input module, which is directly connected to the DSP processor through a cable for inputting AES3 digital audio signal to the DSP processor; Dante network digital signal input module receiving Dante network digital audio signal transmitted from outside RJ45 And 100M network control signal is received and connected to the digital network audio signal and control signal transmitted by the Dante digital audio processing module, the digital audio signal is transmitted to the DSP processing, the control signal and the MCU connection are connected to A DSP processor for inputting a Dante network digital audio signal to a DSP processor.
  • the DSP processor and the DSP processor process the input AES3 digital audio signal, the analog-to-digital converted analog audio signal, and the Dante network digital audio signal transmitted by the Dante network.
  • the processing includes digital audio algorithm processing including channel routing, input EQ, X-Over, output EQ, delay, phase adjustment, clipping, etc., that is, performing steps 2-7 above.
  • the power amplifier module is connected to the power amplifier module through a D/A converter to send the audio signal processed by the DSP processor to the power amplifier module for output.
  • the power amplifier module core detects and controls the power amplifier, the operating temperature of the power amplifier heat sink, and the power supply voltage and current.
  • the audio curve automatically determines the working condition of the connected speaker system to ensure that the user is in good normal use.
  • a touch control display module is connected to the DSP processor, configured to send a touch control command to the DSP processor, and receive the collection information of the temperature detecting module, the voltage detecting module, the current detecting module, and the impedance detecting module.
  • the DSP processor receives the data of the external control signal input from the touch control display module and transmits the data to the DSP processor or the DSP processor for real-time data transmission to the MCU for processing, and then displays the module 7 for the temperature, voltage, current, The size of the signal is analyzed and processed, and the MCU also analyzes and processes the impedance data of the amplifier load.
  • the DSP processor and the MCU can be implemented by the same DSP processor, or can be executed separately.

Abstract

Disclosed are a method, a system and a device for DSP-based Dante digital audio processing. The method comprises the following steps: step 1, controlling and adjusting the audio electric level property of inputted audio signals; step 2, adjusting the frequency response property of the respective frequency bands of the audio signals, the audio electric level property of which has been adjusted; step 3, performing frequency division of the audio signals on which input equalization has been performed; and step 4, adjusting the frequency response property of the respective frequency bands of the audio signals, the frequency division of which has been performed, to realize output equalization of the audio signals. The present invention realizes the selective inputting of analog audio signals, AES3 digital audio signals and Dante network audio signals, and achieves high tone quality and low latency by means of the independent adjustment of the audio electric level characteristic of four channels and the multi-band equalization adjustment.

Description

基于DSP的Dante数字音频处理方法、系统及装置Dante digital audio processing method, system and device based on DSP 技术领域Technical field
本发明涉及音频信号处理领域，尤其涉及一种基于DSP的Dante数字音频处理方法、系统及装置。The invention relates to the field of audio signal processing, and in particular to a Dante digital audio processing method, system and device based on DSP.
背景技术Background technique
基于以太网的数字音频传输技术已是专业音频行业的一个技术焦点，并以其不依赖于控制系统而独立存在的特性，广泛的应用到很多项目中。一方面它解决了多线路的布线困难问题，同时也解决了远距离传输、数据备份、自动冗余等一系列在模拟传输时代无法面对的问题。目前比较成熟的以太网音频传输技术主要有CobraNet和EtherSound技术，但这两种技术都各有千秋。在此基础上，为了更加迎合市场的需求，Audinate推出了Dante这种融合了很多新技术的数字音频传输技术。Ethernet-based digital audio transmission technology has become a technical focus of the professional audio industry, and is widely used in many projects because of its independent characteristics independent of the control system. On the one hand, it solves the problem of wiring difficulties in multiple lines, and also solves a series of problems that cannot be faced in the era of analog transmission, such as long-distance transmission, data backup, and automatic redundancy. At present, the more mature Ethernet audio transmission technologies mainly include CobraNet and EtherSound technologies, but both technologies have their own advantages. On this basis, in order to better meet the needs of the market, Audinate has launched Dante, a digital audio transmission technology that incorporates many new technologies.
Dante是一种基于3层的IP网络技术的数字音频传输方式，其为点对点的音频连接提供了一种低延时、高精度和低成本的解决方案。Dante技术可以在以太网(100M或者1000M)上传送高精度时钟信号以及专业音频信号并可以进行复杂的路由。与以往传统的音频传输技术相比，它继承了CobraNet与EtherSound所有的优点，如无压缩的数字音频信号，保证了良好的音质效果；解决了传统音频传输中繁杂的布线问题，降低了成本；适应现有网络，无需做特殊配置；网络中的音频信号，都以“标签”的形式进行标注等。现有技术中并未有将模拟音频信号、AES3数字音频信号和Dante网络音频 信号结合的进行音频处理的相关报道。Dante is a digital audio transmission based on 3-layer IP network technology that provides a low latency, high precision and low cost solution for point-to-point audio connections. Dante technology delivers high-precision clock signals as well as professional audio signals over Ethernet (100M or 1000M) and enables complex routing. Compared with the traditional audio transmission technology, it inherits all the advantages of CobraNet and EtherSound, such as uncompressed digital audio signal, which guarantees good sound quality; solves the complicated wiring problems in traditional audio transmission and reduces the cost; Adapt to the existing network, no special configuration is required; the audio signals in the network are marked in the form of "tags". There are no analog audio signals, AES3 digital audio signals and Dante network audio in the prior art. A combination of signals for audio processing.
发明内容Summary of the invention
为了克服现有技术的不足，本发明的目的之一在于提供一种基于DSP的Dante数字音频处理方法，其实现对模拟音频信号、AES3数字音频信号和Dante网络音频信号的可选性输入，通过对四通道进行独立的音频电平特性调节，并经多段均衡调节达到高音质和低延时。In order to overcome the deficiencies of the prior art, one of the objects of the present invention is to provide a DSP-based Dante digital audio processing method, which implements an optional input of an analog audio signal, an AES3 digital audio signal, and a Dante network audio signal. Independent audio level characteristic adjustment for four channels, and high-quality and low delay through multi-stage equalization adjustment.
为实现上述目的，本发明采用的技术方案是：In order to achieve the above object, the technical solution adopted by the present invention is:
一种基于DSP的Dante数字音频处理方法，包括以下步骤：A Dante digital audio processing method based on DSP, comprising the following steps:
步骤1、控制并调节输入的音频信号的音频电平特性；Step 1. Control and adjust an audio level characteristic of the input audio signal;
步骤2、对音频电平特性调节后的音频信号的各频段的频率响应特性进行调节，以实现对音频信号进行输入均衡处理；Step 2: adjusting frequency response characteristics of each frequency band of the audio signal whose audio level characteristics are adjusted, so as to implement input equalization processing on the audio signal;
步骤3、对输入均衡处理后的音频信号进行分频处理；Step 3: performing frequency division processing on the audio signal after the input equalization processing;
步骤4、对分频处理后的音频信号的各频段的频率响应特性进行调节，以实现对音频信号进行输出均衡处理；Step 4: Adjust frequency response characteristics of each frequency band of the frequency-divided audio signal to implement output equalization processing on the audio signal;
其中，所述输入均衡处理为6段二阶IIR滤波器，所述分频处理包括有高通或低通的选择、以及增益和极性调节的控制，所述输出均衡处理为8段二阶IIR滤波器且所述8段二阶滤波器均可对频率、带宽和增益进行调节。The input equalization process is a 6-segment second-order IIR filter, and the frequency division process includes a high-pass or low-pass selection, and gain and polarity adjustment control, and the output equalization process is an 8-segment second-order IIR. The filter and the 8-band second-order filter both adjust the frequency, bandwidth, and gain.
步骤1中输入的音频信号为模拟音频信号、AES3数字音频信号和Dante数字音频信号的一种或多种。The audio signal input in step 1 is one or more of an analog audio signal, an AES3 digital audio signal, and a Dante digital audio signal.
步骤1中输入的音频信号通过通道路由系统被分成四通道，每通道均进行独立的音频电平特性调节。 The audio signal input in step 1 is divided into four channels through the channel routing system, and each channel performs independent audio level characteristic adjustment.
步骤1中所述音频电平特性后，对输入模式和输出模式均进行择一选择，其中输入模式包括立体声和信号相加，输出模式包括并联和桥接。After the audio level characteristic in step 1, the input mode and the output mode are selectively selected, wherein the input mode includes stereo and signal addition, and the output modes include parallel and bridge.
在步骤4后还包括以下步骤：After step 4, the following steps are also included:
步骤5、对输出均衡处理后的音频信号进行延时处理，所述延时处理以时间或长度为调整单位进行调整；Step 5: performing delay processing on the audio signal after the output equalization processing, where the delay processing is adjusted by using time or length as an adjustment unit;
步骤6、对延时处理后的音频信号进行限幅处理，所述限幅处理为off、-3dB、-6dB和-12dB中的一种。Step 6. Perform a limiting process on the delayed processed audio signal, and the limiting processing is one of off, -3 dB, -6 dB, and -12 dB.
本发明的另一目的在于提供一种基于DSP的Dante数字音频处理系统，其实现对模拟音频信号、AES3数字音频信号和Dante网络音频信号的可选性输入，通过对四通道进行独立的音频电平特性调节，并经多段均衡调节达到高音质和低延时。Another object of the present invention is to provide a Dante digital audio processing system based on DSP, which realizes an optional input of an analog audio signal, an AES3 digital audio signal and a Dante network audio signal, and performs independent audio power through four channels. The flat characteristic is adjusted, and the multi-stage equalization adjustment achieves high sound quality and low delay.
为实现上述目的，本发明采用的技术方案是：In order to achieve the above object, the technical solution adopted by the present invention is:
一种基于DSP的Dante数字音频处理系统，其包括：A DSP-based Dante digital audio processing system, comprising:
电平控制单元，用于控制并调节输入的音频信号的音频电平特性；a level control unit for controlling and adjusting an audio level characteristic of the input audio signal;
输入均衡器单元，用于对音频电平特性调节后的音频信号的各频段的频率响应特性进行调节，以实现对音频信号进行输入均衡处理；The input equalizer unit is configured to adjust a frequency response characteristic of each frequency band of the audio signal whose audio level characteristic is adjusted, so as to implement input equalization processing on the audio signal;
分频单元，用于对输入均衡处理后的音频信号进行分频处理；a frequency dividing unit, configured to perform frequency division processing on the audio signal after the input equalization processing;
输出均衡器单元，用于对分频处理后的音频信号的各频段的频率响应特性进行调节，以实现对音频信号进行输出均衡处理；An output equalizer unit is configured to adjust a frequency response characteristic of each frequency band of the frequency-divided audio signal to implement output equalization processing on the audio signal;
其中，所述输入均衡处理为6段二阶IIR滤波器，所述输出均衡处理为8段二阶IIR滤波器且所述8段二阶滤波器均可对频率、带宽和增益进行调节。The input equalization process is a 6-segment second-order IIR filter, the output equalization process is an 8-segment second-order IIR filter, and the 8-segment second-order filter can adjust the frequency, the bandwidth, and the gain.
还包括： Also includes:
延时单元，用于对输出均衡处理后的音频信号进行延时处理，所述延时处理以时间或长度为调整单位进行调整；a delay unit, configured to perform delay processing on the audio signal after the output equalization processing, where the delay processing is adjusted according to a time or length adjustment unit;
限幅单元，用于对延时处理后的音频信号进行限幅处理，所述限幅处理为off、-3dB、-6dB和-12dB中的一种。The limiting unit is configured to perform limiting processing on the delayed processed audio signal, and the limiting processing is one of off, -3dB, -6dB and -12dB.
本发明的还有一目的在于提供一种基于DSP的Dante数字音频处理装置，其实现对模拟音频信号、AES3数字音频信号和Dante网络音频信号的可选性输入，通过对四通道进行独立的音频电平特性调节，并经多段均衡调节达到高音质和低延时。Still another object of the present invention is to provide a DSP-based Dante digital audio processing device that implements an optional input of an analog audio signal, an AES3 digital audio signal, and a Dante network audio signal, by performing independent audio power on four channels. The flat characteristic is adjusted, and the multi-stage equalization adjustment achieves high sound quality and low delay.
为实现上述目的，本发明采用的技术方案是：In order to achieve the above object, the technical solution adopted by the present invention is:
一种基于DSP的Dante数字音频处理装置，其包括：A DSP-based Dante digital audio processing device, comprising:
DSP处理器，用于执行上述的基于DSP的Dante数字音频处理方法；a DSP processor for performing the DSP-based Dante digital audio processing method described above;
音频输入模块，用于向DSP处理器输入音频信号；其中，所述音频输入模块包括：An audio input module, configured to input an audio signal to the DSP processor; wherein the audio input module includes:
平衡模拟信号输入模块，其通过A/D转换器连接至DSP处理器，用于向DSP处理器输入模拟音频信号；a balanced analog signal input module connected to the DSP processor through an A/D converter for inputting an analog audio signal to the DSP processor;
AES3数字信号输入模块，其直接连接至DSP处理器，用于向DSP处理器输入AES3数字音频信号；An AES3 digital signal input module directly connected to the DSP processor for inputting an AES3 digital audio signal to the DSP processor;
Dante网络数字信号输入模块，其通过数字音频处理模块连接至DSP处理器，用于向DSP处理器输入Dante网络数字音频信号；a Dante network digital signal input module connected to the DSP processor through a digital audio processing module for inputting a Dante network digital audio signal to the DSP processor;
功放模块，所述DSP处理器通过D/A转换器与该功放模块连接，以将DSP处理器处理后的音频信号发送至该功放模块进行播放。The power amplifier module is connected to the power amplifier module through a D/A converter to send the audio signal processed by the DSP processor to the power amplifier module for playing.
还包括对功放模块进行温度、电压、电流以及阻抗进行采集的温度检 测模块、电压检测模块、电流检测模块以及阻抗检测模块，所述温度检测模块、电压检测模块、电流检测模块以及阻抗检测模块均连接至DSP处理器。It also includes temperature detection for temperature, voltage, current, and impedance acquisition of the power amplifier module. The measuring module, the voltage detecting module, the current detecting module and the impedance detecting module, the temperature detecting module, the voltage detecting module, the current detecting module and the impedance detecting module are all connected to the DSP processor.
还包括触摸控制显示模块，所述触摸控制显示模块与DSP处理器相连，用于发送触摸控制指令给DSP处理器，并接收温度检测模块、电压检测模块、电流检测模块以及阻抗检测模块的采集信息。The touch control display module is further connected to the DSP processor for transmitting a touch control command to the DSP processor, and receiving the collection information of the temperature detecting module, the voltage detecting module, the current detecting module and the impedance detecting module. .
相比现有技术，本发明的有益效果至少如下：Compared with the prior art, the beneficial effects of the present invention are at least as follows:
1、通过高速的DSP音频处理算法将众多功能集成在一起，其中包括一个创新型的通道路由系统，内核工程调试实时分析系统，音箱负载的实时管理系统，以及内置两通道和四通道革命性的全功能数字音频DSP信号处理系统。1. Integrate many functions through high-speed DSP audio processing algorithms, including an innovative channel routing system, real-time analysis system for kernel engineering debugging, real-time management system for speaker load, and revolutionary built-in two-channel and four-channel. Full-featured digital audio DSP signal processing system.
2、可选的模拟、AES3和Dante数字传输卡多种音频输入方式，可满足数字音频的网络传输及放大输出；提供网络化串口、第三方远程控制功能，还具备自动休眠和信号唤醒等。2, optional analog, AES3 and Dante digital transmission card a variety of audio input methods, to meet the digital audio network transmission and amplification output; provide networked serial port, third-party remote control functions, also has automatic sleep and signal wake-up.
3、全触摸屏显示控制功能菜单，全编辑功能彩屏触摸控制，所有功能可以在显示屏内操作完成。3, full touch screen display control function menu, full editing function color screen touch control, all functions can be completed in the display.
附图说明DRAWINGS
图1是本发明实施例基于DSP的Dante数字音频处理方法流程示意图；1 is a schematic flow chart of a Dante digital audio processing method based on DSP according to an embodiment of the present invention;
图2是本发明实施例基于DSP的Dante数字音频处理装置的结构示意图。2 is a schematic structural diagram of a DSP-based Dante digital audio processing device according to an embodiment of the present invention.
具体实施方式detailed description
下面，结合附图以及具体实施方式，对本发明做进一步描述：The present invention will be further described below in conjunction with the drawings and specific embodiments.
请参照图1所示，一种基于DSP的Dante数字音频处理方法，其包括以下 步骤：Please refer to FIG. 1 , a DSP-based Dante digital audio processing method, which includes the following step:
步骤1、信号输入。信号输入有三种：模拟音频信号输入、AES3数字音频信号输入、Dante数字网络音频信号输入，其中，Dante数字网络音频信号具有低延时、无传统模拟信号的干扰、可远距离多达512通道数字信号同时传输、可以点对点，或点对多点的自由切换传输的功能。模拟音频信号通过A/D转换器、AES3数字音频信号直接通过电缆以及Dante数字网络音频信号通过相应的Dante数字音频处理模块送入DSP处理器进行处理，DSP处理器执行以下步骤2-7的操作。Step 1, signal input. There are three kinds of signal input: analog audio signal input, AES3 digital audio signal input, Dante digital network audio signal input, among which Dante digital network audio signal has low delay, no interference from traditional analog signal, and can distance up to 512 channel digital Simultaneous transmission of signals, point-to-point, or point-to-multipoint free-switching transmission. The analog audio signal is sent to the DSP processor through the A/D converter, the AES3 digital audio signal directly through the cable and the Dante digital network audio signal through the corresponding Dante digital audio processing module, and the DSP processor performs the following steps 2-7. .
步骤2、输入模式(INPUT Mode)或者输出模式(OUTPUT Mode)接收到步骤1的输入信号后，通过通道路由算法分成四通道(CH1-CH4)，然后独立的四通道均进行电平控制，所谓的电平控制，就是对输入信号的音频电平特性，电平控制处理后，对输入模式选择处理(有两种模式：立体声、信号相加)，信号输出模式选择处理(有两种模式：并联、桥接模式)。Step 2. Input mode (INPUT Mode) or output mode (OUTPUT Mode) After receiving the input signal of step 1, the channel routing algorithm is divided into four channels (CH1-CH4), and then the independent four channels are level-controlled. The level control is the audio level characteristic of the input signal. After the level control processing, the input mode selection processing (there are two modes: stereo, signal addition), and the signal output mode selection processing (there are two modes: Parallel, bridge mode).
步骤3、对音频电平特性调节后的音频信号的各频段的频率响应特性进行调节，以实现对音频信号进行输入均衡(EQ)处理。输入均衡包括6段二阶IIR滤波器(称之为6段EQ)加上一段高低通滤波的功能处理，其中，输入的6段EQ其中2段为多类型滤波器选择，多滤波器类型包括(Linkwitz-Riley(或Butterworth或Bessel)/6dB/oct、12dB/oct、18dB/oct、24dB/oct/、30dB/oct、36dB/oct、48dB/oct等。Step 3: Adjust the frequency response characteristics of each frequency band of the audio signal whose audio level characteristics are adjusted, to perform input equalization (EQ) processing on the audio signal. The input equalization consists of a 6-segment second-order IIR filter (referred to as 6-segment EQ) plus a high-low-pass filtering function. Among them, the input 6-segment EQ, 2 of which are multi-type filter selection, the multi-filter type includes (Linkwitz-Riley (or Butterworth or Bessel) / 6dB / oct, 12dB / oct, 18dB / oct, 24dB / oct /, 30dB / oct, 36dB / oct, 48dB / oct, etc.
步骤4、对输入均衡处理后的音频信号进行分频(X-Over)处理,分频的类型有高通、低通选择，还有增益和极性调节的控制。Step 4: Perform frequency division (X-Over) processing on the audio signal after input equalization processing, and the type of frequency division has high-pass, low-pass selection, and control of gain and polarity adjustment.
步骤5、对分频处理后的音频信号的各频段的频率响应特性进行调节， 以实现对音频信号进行输出均衡(输出EQ)处理。输出均衡处理为8段二阶IIR滤波器且该8段二阶滤波器均可对中心频率(频点)、Q值(Q值等于中心频率除以带宽)进行调节。Step 5: adjusting frequency response characteristics of each frequency band of the frequency-divided audio signal, To achieve output equalization (output EQ) processing of the audio signal. The output equalization process is an 8-segment second-order IIR filter and the 8-band second-order filter can adjust the center frequency (frequency point) and the Q value (the Q value is equal to the center frequency divided by the bandwidth).
步骤6、对输出均衡处理后的音频信号进行延时处理，延时最长可达0.9秒，延时处理可以时间或长度为调整单位进行调整。Step 6. Perform delay processing on the audio signal after the output equalization processing, and the delay may be up to 0.9 seconds, and the delay processing may be adjusted by adjusting the time or length.
步骤7、对延时处理后的音频信号进行限幅处理，所述限幅处理为off、-3dB、-6dB和-12dB中的一种。Step 7. Perform a limiting process on the delayed audio signal, and the limiting processing is one of off, -3 dB, -6 dB, and -12 dB.
步骤8、对上述DSP处理器处理后音频信号通过功放单元进行输出(播放)。Step 8. The audio signal processed by the DSP processor is output (played) through the power amplifier unit.
对应于该基于DSP的Dante数字音频处理方法，本发明还提供一种基于DSP的Dante数字音频处理系统，其包括：Corresponding to the DSP-based Dante digital audio processing method, the present invention also provides a DSP-based Dante digital audio processing system, which includes:
电平控制单元，用于控制调节输入的音频信号的音频电平特性；a level control unit for controlling an audio level characteristic of the adjusted input audio signal;
输入均衡器单元，用于对音频电平特性调节后的音频信号的各频段的频率响应特性进行调节，以实现对音频信号进行输入均衡处理；The input equalizer unit is configured to adjust a frequency response characteristic of each frequency band of the audio signal whose audio level characteristic is adjusted, so as to implement input equalization processing on the audio signal;
分频单元，用于对输入均衡处理后的音频信号进行分频处理；a frequency dividing unit, configured to perform frequency division processing on the audio signal after the input equalization processing;
输出均衡器单元，用于对分频处理后的音频信号的各频段的频率响应特性进行调节，以实现对音频信号进行输出均衡处理；An output equalizer unit is configured to adjust a frequency response characteristic of each frequency band of the frequency-divided audio signal to implement output equalization processing on the audio signal;
延时单元，用于对输出均衡处理后的音频信号进行延时处理，所述延时处理以时间或长度为调整单位进行调整；a delay unit, configured to perform delay processing on the audio signal after the output equalization processing, where the delay processing is adjusted according to a time or length adjustment unit;
限幅单元，用于对延时处理后的音频信号进行限幅处理，所述限幅处理为off、-3dB、-6dB和-12dB中的一种。The limiting unit is configured to perform limiting processing on the delayed processed audio signal, and the limiting processing is one of off, -3dB, -6dB and -12dB.
请参照图3所示，本发明还提供了一种基于DSP的Dante数字音频处理装 置，其包括：Referring to FIG. 3, the present invention also provides a Dante digital audio processing device based on DSP. Set, which includes:
音频输入模块，用于向DSP处理器输入音频信号；其中，所述音频输入模块包括：平衡模拟信号输入模块，其通过A/D转换器连接至DSP处理器，用于向DSP处理器输入模拟音频信号；AES3数字信号输入模块，其通过电缆直接连接至DSP处理器，用于向DSP处理器输入AES3数字音频信号；Dante网络数字信号输入模块，其接收从RJ45外部传输的Dante网络数字音频信号及100M的网络控制信号的接收并通过Dante数字音频处理模块(Dante数字音频处理模块对传输的数字网络音频信号和控制信号进行处理，数字音频信号传输给DSP处理，控制信号和MCU连接)连接至DSP处理器，用于向DSP处理器输入Dante网络数字音频信号。An audio input module, configured to input an audio signal to the DSP processor; wherein the audio input module comprises: a balanced analog signal input module connected to the DSP processor through an A/D converter for inputting a simulation to the DSP processor Audio signal; AES3 digital signal input module, which is directly connected to the DSP processor through a cable for inputting AES3 digital audio signal to the DSP processor; Dante network digital signal input module receiving Dante network digital audio signal transmitted from outside RJ45 And 100M network control signal is received and connected to the digital network audio signal and control signal transmitted by the Dante digital audio processing module, the digital audio signal is transmitted to the DSP processing, the control signal and the MCU connection are connected to A DSP processor for inputting a Dante network digital audio signal to a DSP processor.
DSP处理器，DSP处理器对输入的AES3数字音频信号、经过模数转换后的模拟音频信号及Dante网络传输的Dante网络数字音频信号进行处理。处理包括通道路由、输入EQ、X-Over，输出EQ，延时，相位调整，限幅等的数字音频算法处理，即执行上述步骤2-7。The DSP processor and the DSP processor process the input AES3 digital audio signal, the analog-to-digital converted analog audio signal, and the Dante network digital audio signal transmitted by the Dante network. The processing includes digital audio algorithm processing including channel routing, input EQ, X-Over, output EQ, delay, phase adjustment, clipping, etc., that is, performing steps 2-7 above.
功放模块，所述DSP处理器通过D/A转换器与该功放模块连接，以将DSP处理器处理后的音频信号发送至该功放模块进行输出。功放模块内核了功率放大器、功放散热器工作温度的管控、电源电压和电流的检测和管控。The power amplifier module is connected to the power amplifier module through a D/A converter to send the audio signal processed by the DSP processor to the power amplifier module for output. The power amplifier module core detects and controls the power amplifier, the operating temperature of the power amplifier heat sink, and the power supply voltage and current.
对功放模块进行温度、电压(电源电压)、电流以及阻抗进行采集的温度检测模块、电压检测模块、电流检测模块以及阻抗检测模块，所述温度检测模块、电压检测模块、电流检测模块以及阻抗检测模块均连接至DSP处理器。实时检测功放模块的工作温度、电压、电流、自动调整输出的功 率，使功放模块工作在最佳的工作状态；实时负载阻抗分析，自动判断功放所接的音箱负载情况，确定音箱是否有开路或都出现音箱线短路，并且可以在开机后，自动扫描整个音箱的音频曲线，自动判断分析所接音箱系统的工作情况，确保用户使用正常良好。A temperature detecting module, a voltage detecting module, a current detecting module, and an impedance detecting module for collecting temperature, voltage (supply voltage), current, and impedance of the power amplifier module, the temperature detecting module, the voltage detecting module, the current detecting module, and the impedance detecting The modules are all connected to the DSP processor. Real-time detection of the operating temperature, voltage, current of the power amplifier module, and automatic adjustment of the output power Rate, make the power amplifier module work in the best working condition; real-time load impedance analysis, automatically judge the speaker load condition of the power amplifier, determine whether the speaker has an open circuit or a short circuit of the speaker line, and can automatically scan the entire speaker after booting The audio curve automatically determines the working condition of the connected speaker system to ensure that the user is in good normal use.
以及触摸控制显示模块，所述触摸控制显示模块与DSP处理器相连，用于发送触摸控制指令给DSP处理器，并接收温度检测模块、电压检测模块、电流检测模块以及阻抗检测模块的采集信息。DSP处理器接收从触摸控制显示模块输入的外部控制信号的数据传送到DSP处理器或者DSP处理器的实时数据传递到MCU处理后到模块7进行显示，MCU还对功放模块温度、电压、电流、信号的大小进行分析和处理，MCU还对功放负载的阻抗数据进行分析和处理。其中，DSP处理器和MCU可以通过同一DSP处理器实现，也可以分开执行。And a touch control display module, the touch control display module is connected to the DSP processor, configured to send a touch control command to the DSP processor, and receive the collection information of the temperature detecting module, the voltage detecting module, the current detecting module, and the impedance detecting module. The DSP processor receives the data of the external control signal input from the touch control display module and transmits the data to the DSP processor or the DSP processor for real-time data transmission to the MCU for processing, and then displays the module 7 for the temperature, voltage, current, The size of the signal is analyzed and processed, and the MCU also analyzes and processes the impedance data of the amplifier load. Among them, the DSP processor and the MCU can be implemented by the same DSP processor, or can be executed separately.
对本领域的技术人员来说，可根据以上描述的技术方案以及构思，做出其它各种相应的改变以及形变，而所有的这些改变以及形变都应该属于本发明权利要求的保护范围之内。 Various other changes and modifications may be made by those skilled in the art in light of the above-described technical solutions and concepts, and all such changes and modifications are intended to fall within the scope of the appended claims.

Claims (10)

  1. 一种基于DSP的Dante数字音频处理方法，其特征在于，包括以下步骤：A Dante digital audio processing method based on DSP, characterized in that it comprises the following steps:
    步骤1、控制并调节输入的音频信号的音频电平特性；Step 1. Control and adjust an audio level characteristic of the input audio signal;
    步骤2、对音频电平特性调节后的音频信号的各频段的频率响应特性进行调节，以实现对音频信号进行输入均衡处理；Step 2: adjusting frequency response characteristics of each frequency band of the audio signal whose audio level characteristics are adjusted, so as to implement input equalization processing on the audio signal;
    步骤3、对输入均衡处理后的音频信号进行分频处理；Step 3: performing frequency division processing on the audio signal after the input equalization processing;
    步骤4、对分频处理后的音频信号的各频段的频率响应特性进行调节，以实现对音频信号进行输出均衡处理；Step 4: Adjust frequency response characteristics of each frequency band of the frequency-divided audio signal to implement output equalization processing on the audio signal;
    其中，所述输入均衡处理为6段二阶IIR滤波器，所述分频处理包括有高通或低通的选择、以及增益和极性调节的控制，所述输出均衡处理为8段二阶IIR滤波器且所述8段二阶滤波器均可对频率、带宽和增益进行调节。The input equalization process is a 6-segment second-order IIR filter, and the frequency division process includes a high-pass or low-pass selection, and gain and polarity adjustment control, and the output equalization process is an 8-segment second-order IIR. The filter and the 8-band second-order filter both adjust the frequency, bandwidth, and gain.
  2. 根据权利要求1所述的基于DSP的Dante数字音频处理方法，其特征在于，步骤1中输入的音频信号为模拟音频信号、AES3数字音频信号和Dante数字音频信号的一种或多种。The DSP-based Dante digital audio processing method according to claim 1, wherein the audio signal input in step 1 is one or more of an analog audio signal, an AES3 digital audio signal, and a Dante digital audio signal.
  3. 根据权利要求1所述的基于DSP的Dante数字音频处理方法，其特征在于，步骤1中输入的音频信号通过通道路由系统被分成四通道，每通道均进行独立的音频电平特性调节。The DSP-based Dante digital audio processing method according to claim 1, wherein the audio signal input in step 1 is divided into four channels by a channel routing system, and each channel performs independent audio level characteristic adjustment.
  4. 根据权利要求1所述的基于DSP的Dante数字音频处理方法，其特征在于，步骤1中所述音频电平特性后，对输入模式和输出模式均进行择一选择，其中输入模式包括立体声和信号相加，输出模式包括并联和桥接。 The DSP-based Dante digital audio processing method according to claim 1, wherein after the audio level characteristic in step 1, the input mode and the output mode are selectively selected, wherein the input mode includes stereo and signal. Add, the output modes include parallel and bridge.
  5. 根据权利要求1所述的基于DSP的Dante数字音频处理方法，其特征在于，在步骤4后还包括以下步骤：The DSP-based Dante digital audio processing method according to claim 1, further comprising the following steps after step 4:
    步骤5、对输出均衡处理后的音频信号进行延时处理，所述延时处理以时间或长度为调整单位进行调整；Step 5: performing delay processing on the audio signal after the output equalization processing, where the delay processing is adjusted by using time or length as an adjustment unit;
    步骤6、对延时处理后的音频信号进行限幅处理，所述限幅处理为off、-3dB、-6dB和-12dB中的一种。Step 6. Perform a limiting process on the delayed processed audio signal, and the limiting processing is one of off, -3 dB, -6 dB, and -12 dB.
  6. 一种基于DSP的Dante数字音频处理系统，其特征在于，其包括：电平控制单元，用于控制并调节输入的音频信号的音频电平特性；A Dante digital audio processing system based on DSP, characterized in that it comprises: a level control unit for controlling and adjusting an audio level characteristic of an input audio signal;
    输入均衡器单元，用于对音频电平特性调节后的音频信号的各频段的频率响应特性进行调节，以实现对音频信号进行输入均衡处理；The input equalizer unit is configured to adjust a frequency response characteristic of each frequency band of the audio signal whose audio level characteristic is adjusted, so as to implement input equalization processing on the audio signal;
    分频单元，用于对输入均衡处理后的音频信号进行分频处理；a frequency dividing unit, configured to perform frequency division processing on the audio signal after the input equalization processing;
    输出均衡器单元，用于对分频处理后的音频信号的各频段的频率响应特性进行调节，以实现对音频信号进行输出均衡处理；An output equalizer unit is configured to adjust a frequency response characteristic of each frequency band of the frequency-divided audio signal to implement output equalization processing on the audio signal;
    其中，所述输入均衡处理为6段二阶IIR滤波器，所述输出均衡处理为8段二阶IIR滤波器且所述8段二阶滤波器均可对频率、带宽和增益进行调节。The input equalization process is a 6-segment second-order IIR filter, the output equalization process is an 8-segment second-order IIR filter, and the 8-segment second-order filter can adjust the frequency, the bandwidth, and the gain.
  7. 根据权利要求6所述的基于DSP的Dante数字音频处理系统，其特征在于，还包括：The DSP-based Dante digital audio processing system according to claim 6, further comprising:
    延时单元，用于对输出均衡处理后的音频信号进行延时处理，所述延时处理以时间或长度为调整单位进行调整；a delay unit, configured to perform delay processing on the audio signal after the output equalization processing, where the delay processing is adjusted according to a time or length adjustment unit;
    限幅单元，用于对延时处理后的音频信号进行限幅处理，所述限幅处理为off、-3dB、-6dB和-12dB中的一种。The limiting unit is configured to perform limiting processing on the delayed processed audio signal, and the limiting processing is one of off, -3dB, -6dB and -12dB.
  8. 一种基于DSP的Dante数字音频处理装置，其特征在于，其包括： A Dante digital audio processing device based on DSP, characterized in that it comprises:
    DSP处理器，用于执行权利要求1-5任一项所述的基于DSP的Dante数字音频处理方法；a DSP processor for performing the DSP-based Dante digital audio processing method according to any one of claims 1-5;
    音频输入模块，用于向DSP处理器输入音频信号；其中，所述音频输入模块包括：An audio input module, configured to input an audio signal to the DSP processor; wherein the audio input module includes:
    平衡模拟信号输入模块，其通过A/D转换器连接至DSP处理器，用于向DSP处理器输入模拟音频信号；a balanced analog signal input module connected to the DSP processor through an A/D converter for inputting an analog audio signal to the DSP processor;
    AES3数字信号输入模块，其直接连接至DSP处理器，用于向DSP处理器输入AES3数字音频信号；An AES3 digital signal input module directly connected to the DSP processor for inputting an AES3 digital audio signal to the DSP processor;
    Dante网络数字信号输入模块，其通过数字音频处理模块连接至DSP处理器，用于向DSP处理器输入Dante网络数字音频信号；a Dante network digital signal input module connected to the DSP processor through a digital audio processing module for inputting a Dante network digital audio signal to the DSP processor;
    功放模块，所述DSP处理器通过D/A转换器与该功放模块连接，以将DSP处理器处理后的音频信号发送至该功放模块进行播放。The power amplifier module is connected to the power amplifier module through a D/A converter to send the audio signal processed by the DSP processor to the power amplifier module for playing.
  9. 根据权利要求8所述的基于DSP的Dante数字音频处理装置，其特征在于，还包括对功放模块进行温度、电压、电流以及阻抗进行采集的温度检测模块、电压检测模块、电流检测模块以及阻抗检测模块，所述温度检测模块、电压检测模块、电流检测模块以及阻抗检测模块均连接至DSP处理器。The DSP-based Dante digital audio processing device according to claim 8, further comprising a temperature detecting module, a voltage detecting module, a current detecting module, and an impedance detecting for collecting the temperature, the voltage, the current, and the impedance of the power amplifier module. The module, the temperature detecting module, the voltage detecting module, the current detecting module, and the impedance detecting module are all connected to the DSP processor.
  10. 根据权利要求9所述的基于DSP的Dante数字音频处理装置，其特征在于，还包括触摸控制显示模块，所述触摸控制显示模块与DSP处理器相连，用于发送触摸控制指令给DSP处理器，并接收温度检测模块、电压检测模块、电流检测模块以及阻抗检测模块的采集信息。 The DSP-based Dante digital audio processing device of claim 9, further comprising a touch control display module, the touch control display module being coupled to the DSP processor for transmitting a touch control command to the DSP processor, And receiving the collection information of the temperature detection module, the voltage detection module, the current detection module, and the impedance detection module.
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