CN111586530A - Sound box processor - Google Patents
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- CN111586530A CN111586530A CN202010405961.6A CN202010405961A CN111586530A CN 111586530 A CN111586530 A CN 111586530A CN 202010405961 A CN202010405961 A CN 202010405961A CN 111586530 A CN111586530 A CN 111586530A
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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
- H04R3/00—Circuits for transducers, loudspeakers or microphones
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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
- H04R3/00—Circuits for transducers, loudspeakers or microphones
- H04R3/04—Circuits for transducers, loudspeakers or microphones for correcting frequency response
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Abstract
The invention relates to a sound box processor, comprising: the device comprises a control module, a communication control interface, a display module, an audio input interface, an audio processing module and an audio output interface; the audio output interface, the audio processing module and the audio output interface are electrically connected in sequence, the audio processing module and the communication control interface are electrically connected with the control module, and the communication control interface is electrically connected with the display module. The system has the advantages that the professional sound box processor and the acoustic measurement system are integrated, and the system becomes smaller, lighter and better to use.
Description
Technical Field
The invention belongs to the technical field of electronics and circuits, and particularly relates to a sound box processor, which is a digital audio application and measurement technology of the sound box processor.
Background
For a long time, the engineering installation of a professional sound system is a very complicated process, and especially in the tuning link of listening environment, a professional environment test tool and a professional tuning technical engineer are needed, firstly, a sound box can be completed through the measurement and compensation of the frequency response of a sound-deadening room and then the heavy and repeated tuning and listening on site, and secondly, the professional sound system and the measurement system thereof have large volume and heavy weight, and are very inconvenient to use in some occasions.
Disclosure of Invention
In view of the above, the present invention provides a speaker processor to solve the above problems.
In order to achieve the purpose, the invention adopts the following technical scheme: a loudspeaker processor comprising: the device comprises a control module, a communication control interface, a display module, an audio input interface, an audio processing module and an audio output interface;
the audio input interface, the audio processing module and the audio output interface are electrically connected in sequence, the audio processing module and the communication control interface are electrically connected with the control module, and the communication control interface is electrically connected with the display module;
the sound box processor further comprises a power supply and a fan, the power supply provides electric energy for the control module, the communication control interface, the display module, the audio input interface, the audio processing module, the audio output interface and the fan, and the fan is electrically connected with the display module.
The display module is an LCD display module, the audio processing module is a DSP audio processing module, the control module is an MCU control module, and the MCU control module comprises: the DSP audio processing module is controlled in real time, internal parameters of the DSP audio processing module are configured, the configured parameters are written into the external FLASH memory, and the parameters of the DSP audio processing module can also be stored on a computer, so that data transfer and safe backup are facilitated.
Moreover, the self-contained listening environment field test system has the function, FIR WIZARD in LPP Series computer software provides FIR coefficient leading-in and downloading functions for a 1024 tap FIR filter, so that automatic compensation and correction of frequency response and acoustic phase of a loudspeaker, a sound box and a field environment are realized, a frequency response curve of the whole listening environment tends to be more flat, and the tone quality is better restored. In addition, the IR can be introduced into a third-party frequency response curve to provide a good reference curve for a user to manually adjust an acoustic compensation curve;
the communication control interface comprises a serial port (UART), a USB, an RS485 and an ETHERNET;
the LCD display module comprises an MCU, an external FLASH (SPI) memory, an LCD, a 5-direction navigation knob LED display, keys and a serial port (UART);
the audio input interface receives analog audio signals from the differential amplification input circuit, the analog audio signals are sampled and quantized by the A/D audio encoder into I2S digital signals, the I2S digital signals are transmitted to the DSP audio processing module for audio algorithm processing, the processed signals are transmitted to the audio output interface through an I2S bus, the D/A audio codec is converted into differential audio signals by the D/A audio codec and then output, the differential amplification output circuit converts the analog audio signals and outputs, and the signals are transmitted to the DANTE OUT digital audio module through the SRC, so that multi-channel digital signal output is realized;
the DSP audio processing module comprises a special DSP audio processing module M716 supporting multi-channel audio processing, an SRAM memory, a clock circuit and a DSP audio algorithm running in the DSP;
the SRC in the DSP audio processing module receives digital audio signals from the DANTE and AES/EBU modules in the audio input interface, converts the digital audio signals into I2S digital signals with corresponding sampling rates, and transmits the digital signals to the DSP audio processing module for audio algorithm processing, and the SRC provides self-adaptive capacity to audio signals with different sampling rates in the whole I2S audio signal processing process.
The communication control interface is a USB communication circuit, an RS485 communication circuit or an ETHERNET communication circuit, the USB communication circuit adopts an STM32 chip built-in USB interface, the RS485 communication circuit adopts an MAX485 level conversion chip, and the external FLASH memory adopts a 25Q64JVS1Q chip of WINBOND company.
The clock circuit comprises a DSP clock connected with the DSP audio processing module, an MCU clock connected with the MCU control circuit, and an I2S clock supplied to the AD/DA audio codec.
In the clock circuit, the DSP clock adopts an external 12.288MHz active crystal oscillator, the MCU clock adopts an external 25MHz passive crystal oscillator, and the I2S clock is output by the DSP audio processing module.
The audio input interface is designed by adopting an NE5532+4580 operational amplifier, and the audio output interface is designed by adopting an NE5532+ OP1652 operational amplifier.
The A/D audio analog-to-digital converter adopts a PCM4220 chip, and the D/A audio digital-to-analog converter adopts a PCM1794A chip.
The DSP audio processing module is a SAM5xxx (customized plate M716 of the company) series 24-bit/56-bit, 800M MAC/sec multichannel special DSP audio processing module of the France DREAM company, and the SRAM adopts a LY6251216ML chip of the LYONTEK company. The multiple channels include, but are not limited to, 1 in multiple out, 2 in 4 out, 2 in 6 out, 3 in 6 out, 4 in 8 out, 8 in 8 out, and the like.
The MCU control module adopts an enhanced serial chip STM32F107VCT6 with a 32-bit ARM Cortex M3 kernel.
The LCD display module adopts an FSTN LCD with LM9091BCT320x96 resolution and a 5-direction navigation knob, and adopts MCU STM32F070RBT6+25Q16JVS1Q chip external storage of ST company to display system menus and user input control.
An ETHERNET communication circuit in the communication control interface adopts a DP83848C chip;
the DANTE module in the audio input interface adopts an ULT-01-004 chip, a network M88E6320 chip, a storage FT24C256 chip and a clock Si5351 chip of DANTE company, and the SRC adopts an AK4127 chip;
an AK4113VF chip of AKM company is adopted by an AES/EBU module in the audio input interface, and an AK4127 chip is adopted by the SRC;
the DSP audio algorithm supports up to 8 groups of 1024 tap FIR filters, FIR coefficient introduction and downloading are supported, up to 8 groups of X-OVER which can switch FIR/IIR are supported by an output channel, and PEQ of the input and output channels all support the function of introducing the third-party measurement parameters of Ext.
The invention has the beneficial effects that:
1. the sound box processor (LPP Series) software of the invention has rich, simple and powerful functions, and the software in the machine is provided with FIR WIZARD to solve the acoustic measurement problem in the prior art! The frequency response and phase test of the listening environment can be automatically completed only by clicking the next step and a few options, FIR filter coefficients are generated and stored in a 1024 tap FIR filter structure of the machine, compensation and correction are automatically carried out on the sound box and the environment, the frequency response curve of the whole listening environment tends to be flatter, and the tone quality is better restored. Provide the possibility of extending professional sound systems to more non-professional applications! The audio processing system and the acoustic measurement system are integrated together, become smaller, lighter and better, so that more non-professionals can use the system, high-fidelity sound experience can be easily obtained, and the goal pursued by LPP Series is achieved.
2. The sound box processor has a listening environment frequency response test function and an automatic phase FIR compensation correction function. As shown in fig. 4, firstly, the LPP Series computer software starts the FIR WIZARD test function guide in the software to generate the frequency sweep test signal, and the frequency sweep test signal is output from the XLR audio output interface module to the power amplifier via the XLR audio input interface module of the LPP Series machine to push various models of speakers to emit test audio, and the test audio is transmitted, reflected, diffused, etc. through the listening environment, picked up by the microphone, passed through the ASIO low-delay sound card, reentered into the LPP Series computer software, and through the cooperative operation of the DSP and the LPP Series computer software, the finally calculated FIR coefficient is written into the FIR filter of the DSP module, so as to complete the compensation and correction of the frequency response and phase of the listening environment, and achieve the best audio reproduction effect.
Drawings
FIG. 1 is an overall schematic diagram of the sound box processor of the present invention.
Fig. 2 is a block diagram of the electrical connections of the internal modules of the sound box processor of the present invention.
FIG. 3 is a signal flow diagram of the DSP 8 channel audio algorithm within the sound box processor of the present invention.
FIG. 4 is a functional block diagram of a test system for a sound box processor listening environment according to the present invention.
Detailed Description
In order to make the objects, technical solutions and advantages of the present invention clearer, the technical solutions of the present invention will be further clearly and completely described below with reference to the embodiments of the present invention. It should be noted 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.
The sound box processor in the embodiment has a frequency response test function of listening environment and an automatic phase FIR compensation correction function. As shown in fig. 1, the sound box processor of the present embodiment includes: the device comprises a control module, a communication control interface, a display module, an audio input interface, an audio processing module and an audio output interface;
the audio input interface, the audio processing module and the audio output interface are electrically connected in sequence, the audio processing module and the communication control interface are electrically connected with the control module, and the communication control interface is electrically connected with the display module.
As shown in fig. 4, the LPP Series computer software starts the FIR WIZARD test function guide in the software to generate the frequency sweep test signal, and the frequency sweep test signal is outputted from the XLR audio output interface module to the power amplifier via the XLR audio input interface module of the LPP Series machine to push various models of speakers to emit test audio, which is picked up by the microphone after transmission, reflection, diffusion, etc. of the listening environment, and then enters the LPP Series computer software again through the ASIO low-delay sound card, and finally calculated FIR coefficients are written into the FIR filter of the DSP module through the cooperative operation of the DSP and the LPP Series computer software, so that the frequency response and phase compensation and correction of the listening environment are completed, and the optimal audio reproduction effect is achieved.
The present invention will be further described with reference to the following detailed description, wherein the drawings are provided for illustrative purposes only and are not intended to be limiting; to better illustrate the embodiments of the present invention, some parts of the drawings may be omitted, enlarged or reduced, and do not represent the size of an actual product; it will be understood by those skilled in the art that certain well-known structures in the drawings and descriptions thereof may be omitted.
As shown in fig. 2, the sound box processor in this embodiment further includes a power supply and a fan, the power supply provides electric energy for the control module, the communication control interface, the display module, the audio input interface, the audio processing module, the audio output interface and the fan, and the fan is electrically connected to the display module.
The audio input interface adopts a NE5532 operational amplifier of TI company and a JRC4580 two-way operational amplifier of JRC company; the differential amplification output circuit adopts a NE5532+ OP1652 two-way operational amplifier design of TI company, has low noise, low distortion and low crosstalk, has stable gain and can provide excellent dynamic performance in a wide load range. The differential signal has more advantages in the wiring aspect than the common single-ended signal, and firstly, the differential signal has stronger anti-interference capability, more accurate signal time sequence and good signal wire coupling, and can effectively inhibit EMI (electro-magnetic interference), because the polarities of the two signal wires are opposite, electromagnetic fields radiated outwards can be mutually offset.
Analog audio signals are converted into digital signals through sampling, quantizing and encoding processes and then transmitted to a DSP processor for processing, 2-channel Delta-SIGMA24-Bit A/D conversion chips are used for completing the conversion process, a PCM4220 chip with high performance integration of Texas Instruments (TI) company can be adopted as an A/D conversion circuit, the sampling frequency can reach 216kHz at most, a differential analog structure is adopted, the dynamic range of 123dB is wide, the signal-to-noise ratio is high, the sound fidelity is high, and the digital audio signal processing method is widely applied to the fields of broadcast television, entertainment, performance, professional audio recording and broadcasting and the like.
The digital audio signal needs to be converted into an analog signal again after being processed by the DSP processor, a 2-channel Delta-SIGMA24-Bit D/A conversion chip is used for completing the conversion process, and a PCM17 1794A chip which is high in performance and integrated by Texas Instruments (TI) company can be adopted as a D/A conversion circuit, so that the digital audio signal has excellent dynamic performance and enhanced clock jitter tolerance. The balanced current output is provided, the sampling frequency can reach 200kHz at most, a differential analog structure is adopted, the 129dB dynamic range is wide, the signal-to-noise ratio is high, the sound reduction degree is high, and the balanced current output is widely applied to the fields of broadcast television public address systems, entertainment systems, performance systems, professional recording and broadcasting and the like.
This audio amplifier treater MCU adopts 32-bit ARM Cortex M3 kernel's high performance, low cost, low-power consumption microprocessor, the maximum frequency is 72MHz, instruction execution speed is up to 1.25DMIPS/MHz, can adopt ST company STM32 enhancement mode series chip STM32F107VCT6, it has 64 to 256K FLASH to integrate on the piece, 64K SRAM is up to 4 general 16 bit timers more, there are 2I 2C, 3 SPI and 5 serial ports, interface resource is abundant, mainly be responsible for DSP system initialization in the system, data operation, storage, communication management. STM32F070RBT6 mainly completes user operation interface control, level display and the like.
The RS485 communication interface adopts an MAX485 chip, and the ETHERNET circuit adopts a chip DP83848C chip.
The DSP clock of the sound box processor adopts an external 12.288MHz active crystal oscillator, the timing sequence accuracy is high, the MCU clock adopts an external 25MHz passive crystal oscillator, the I2S clock of the AD D/A conversion/DA conversion chip is output by the DSP, the sampling clock is 96K, and the sampling digit is 24-Bit.
In order to increase the number of files in the system, while using the integrated false sh space on chip, an external extended FLASH memory is required, which is mainly used to store files for configuring DSP parameters, and in this document, 25Q64JVS1Q and AT24C256A are selected as memory expansion chips, connected with an MCU, and read and write data through SPI/I2C bus, thereby realizing storage of up to 120 sound boxes and setting of environmental parameters.
The core processing part of the processor is a DSP processor, the DSP processor adopts SAM5xxx (custom plate M716 of the company) series 24-bit/56-bit of the France DREAM company, 800M MAC/sec multichannel special DSP audio processor, the core frequency is 200MHz, 32kx24/72kx16 data/effect RAM and 32kx24 DSP parameter RAM are integrated on a chip, the DSP processor is completed by a complete 56-bit double-precision mode, the digital signal processing performance is excellent, and the MARANI audio algorithm is well operated. It is externally controlled and parameter configured through a PARALLEL (PARALLEL) port.
Analog audio signals are input into an A/D conversion chip through a differential amplification input circuit, amplified and input into the A/D conversion chip, sampled and quantized into I2S digital signals by the A/D conversion chip, transmitted to a DSP processor for audio algorithm processing, transmitted through an I2S bus, returned to the D/A conversion chip, converted into differential audio signals by the D/A conversion chip for output, converted into analog audio signals by a differential amplification output circuit for output, and input, conversion, processing and output processes of the signals are completed. In the whole process, the system MCU is connected with computer software or APP through USB, RS485 or ETHERNET, and the configuration parameters of the whole machine can be generated in real time and archived to an externally expanded FLASH memory or stored to a computer.
As shown in fig. 3, the DSP audio processing module can support multi-channel audio processing, and after the digital signal enters the DSP, the DSP can configure: a noise generator, loudness, an input route and input linkage; at most 8 input channels are equipped with various DSP audio algorithms, including noise gate, gain, mute, delay, phase, 13-segment 17-filtering-type selectable parametric equalization, RMS compressor and software features of the invention (at most 8 branch 1024-tap FIR filter carriers which can be imported or automatically guided) for compensating and correcting defects of listening environment and sound frequency response, phase and the like; at most 8 output channels are equipped with various DSP audio algorithms including gain, mute, delay, phase, 7-segment 17-filter-type selectable parametric equalization, RMS compressor, PEAK limiter, output routing, 4-path DANTE output tone source selection, output linkage, and with the software features of the invention (8-path selectable FIR/IIR electronic frequency division), the frequency division requirements for different purposes, such as low phase shift division, may employ FIR filters, up to 512 taps can be provided for selection, and the software feature (ext. ir file import function) of the invention is used for importing the tested frequency response curve of the listening environment test or the frequency response curve of the sound box of a third party as the reference curve for compensating parameter equalization, so that the user can set the parameters in a silent state, thereby simplifying the environmental noise problem caused by the frequency sweep test signal during the installation of the audio system.
The sound box processor has the advantages of high integration level, good stability, strong expansibility, programmable control, remote application upgrade support and the like, and is widely applied to the fields of professional recording and broadcasting systems, professional sound equipment, performance systems, entertainment systems, digital power amplifiers and the like at present.
The above-mentioned embodiments only express several embodiments of the present invention, and the description thereof is more specific and detailed, but not construed as limiting the scope of the present invention. It should be noted that, for a person skilled in the art, several variations and modifications can be made without departing from the inventive concept, which falls within the scope of the present invention. Therefore, the protection scope of the present patent shall be subject to the appended claims.
Claims (10)
1. A loudspeaker processor, comprising: the device comprises a control module, a communication control interface, a display module, an audio input interface, an audio processing module and an audio output interface;
the audio input interface, the audio processing module and the audio output interface are electrically connected in sequence, the audio processing module and the communication control interface are electrically connected with the control module, and the communication control interface is electrically connected with the display module.
2. A sound box processor according to claim 1, wherein the display module is an LCD display module, the audio processing module is a DSP audio processing module, and the control module is an MCU control module, the MCU control module comprising: the system comprises a clock circuit, an external EEP/FLASH storage, an MCU and a serial port communication circuit, wherein the MCU control module realizes double-MCU linkage through the serial port communication circuit and an LCD display module, and is connected with a terminal through a USB, an RS485 and an ETHERNET communication control interface;
the sound box processor also comprises a power supply and a fan, the control module, the communication control interface, the display module, the audio input interface, the audio processing module, the audio output interface and the fan are electrically connected with the power supply, and the fan is electrically connected with the display module;
the communication control interface comprises a serial port, a USB, RS485 and an ETHERNET;
the display module comprises an external FLASHSPI memory, an LCD, 5-direction navigation knob LED display and keys;
the audio input interface receives an analog audio signal from a differential amplification input circuit, the analog audio signal is sampled and quantized into an I2S digital signal through an A/D audio encoder and then transmitted to the DSP audio processing module for audio algorithm processing, the processed I2S digital signal is transmitted to the audio output interface through an I2S bus, a D/A audio codec is converted into a differential audio signal by a D/A audio codec and then output, the differential amplification output circuit converts the analog audio signal output, and the digital audio signal is transmitted to a DANTE OUT digital audio module in the audio output interface through SRC in the DSP audio processing module to realize multi-channel digital signal output;
the DSP audio processing module comprises an M716, an SRAM memory, a clock circuit and a DSP audio algorithm running in the DSP audio processing module, and the M716 is a special DSP audio processing module supporting multi-channel audio processing;
the SRC receives digital audio signals from the DANTE digital audio module and the AES/EBU digital module, converts the digital audio signals into I2S digital signals with corresponding sampling rates, and transmits the I2S digital signals to the DSP audio processing module for audio algorithm processing;
the communication control interface is a USB communication circuit, an RS485 communication circuit or an ETHERNET communication circuit, the USB communication circuit adopts an STM32 chip with a built-in USB interface, the RS485 communication circuit adopts an MAX485 level conversion chip, and the external FLASH memory adopts a 25Q64JVS1Q chip.
3. A sound box processor according to claim 2, wherein the clock circuit comprises a DSP clock connected to the DSP audio processing module, an MCU clock connected to the MCU control circuit, and an I2S clock.
4. The speaker processor of claim 3, wherein the DSP clock employs an external 12.288MHz active crystal oscillator, the MCU clock employs an external 25MHz passive crystal oscillator, and the I2S clock is output by the DSP audio processing module.
5. The sound box processor of claim 1, wherein the audio input interface is designed using an NE5532+4580 operational amplifier, and the audio output interface is designed using an NE5532+ OP1652 operational amplifier.
6. The speaker processor as claimed in claim 1, wherein the a/D audio adc of the audio input interface is a PCM4220 chip, and the D/a audio dac of the audio output interface is a PCM1794A chip.
7. The sound box processor of claim 2, wherein the DSP audio processing module is a SAM5xxx series 24-bit/56-bit, 800M MAC/sec multichannel dedicated DSP audio processing module, and the SRAM memory employs a LY6251216ML chip; the multiple channels include a 1 in multiple out channel, a2 in 4 out channel, a2 in 6 out channel, a 3 in 6 out channel, a 4 in 8 out channel, and an 8 in 8 out channel.
8. The sound box processor of claim 2, wherein the MCU control module employs an enhanced series of chips STM32F107VCT6 with a 32-bit armport M3 kernel.
9. The sound box processor of claim 2, wherein the LCD display module uses an FSTN LCD of LM9091BCT320x96 resolution to display system menus and user input controls, and the 5-way navigation knob uses an MCUSTM32F070RBT6+25Q16JVS1Q chip for external storage.
10. The sound box processor of claim 2, wherein the ETHERNET communication circuit in the communication control interface employs a DP83848C chip;
the DANTE digital audio module comprises an ULT-01-004 chip, a network M88E6320 chip, a storage FT24C256 chip and a clock Si5351 chip, wherein the SRC adopts an AK4127 chip;
the AES/EBU digital module adopts AK4113VF chip, the SRC adopts AK4127 chip;
the DSP audio algorithm supports 8 groups of 1024 tap FIR filters and the leading-in and downloading of FIR coefficients, the output channel of the audio output interface supports 8 groups of X-OVER capable of switching FIR/IIR, and the PEQ of the input channel in the audio input interface and the PEQ of the output channel in the audio output interface both support the leading-in of the third-party measurement parameters of Ext.
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Application publication date: 20200825 |