Abstract: This article introduces a digital anti-noise module based on a dedicated DSP chip and using a unique software anti-noise algorithm to achieve a speech clarity of no less than 98 in a 120-dB noise environment. This module has been successfully used in China's airborne communication equipment.
Overview The current domestic third-generation anti-noise products use dynamic noise reduction (DNR) technology. DNR technology is to dynamically adjust the output voice switch through changing voice peaks, so as to achieve the purpose of noise reduction. Although it is currently a better anti-noise analog processing technology, it also has some limitations, including light note dropout and strong noise noise tailing; the noise reduction effect is focused on low frequencies; noise reduction is completely implemented by hardware circuits. Problems such as troublesome debugging and maintenance. Due to these problems, a large number of applications of analog DNR noise reduction products are restricted. With the rapid development of digital signal processing technology, products of digital anti-noise technology supported by digital signal processors and related algorithms continue to emerge. The digital anti-noise module proposed in this paper is to apply modern digital signal processing (DSP) technology and its high-speed real-time processing operation characteristics, and use corresponding software algorithms to process the voice and noise in a high-noise environment to complete the high-noise environment. Voice communication function.
The performance advantages of this module include:
a) The software uses an adaptive filtering algorithm. The digital anti-noise processor's noise suppression is generally above 50 decibels, and the output voice is stable without word leakage and noise tailing.
b) The digital anti-noise processor equalizes the amount of noise reduction in the entire voice band (300 ~ 3000Hz).
c) The digital anti-noise processor can meet different anti-noise requirements by changing the software algorithm, which is convenient for product upgrading.
d) Hardware cost is lower than analog DNR products.
e) Using software encryption technology, products are not easy to be infringed and counterfeited, which is conducive to protecting the interests of manufacturers.
Main index requirements and overall solution ideas As a part of the JK-DP10 digital anti-noise processor, this digital anti-noise processing module is mainly used for communication in the noisy environment such as on-board communication terminal equipment. Its frequency range is 300 ~ 3400Hz , The flatness is not more than 2dB. The noise reduction performance is: when the module input terminal adds 3mV, 2 seconds intermittent sine wave signal (frequency is 300Hz, 700Hz, 1000Hz, 1500Hz, 2000Hz, 2500Hz, 3000Hz) and joins 3mV, 120dB continuous white noise signal, the module The difference in output level is not less than 50dB.
First, choose a suitable DSP device. Requires low power consumption, high-speed data operation and throughput (above 40 MIPS), including A / D, D / A, and Flash (16KB). Then establish an effective noise model, design adaptive filtering structure and related software algorithms. Next, the electromagnetic compatibility (EMC) of the digital anti-noise processor is designed, and an anti-noise microphone device that can adapt to 120 dB of environmental noise is selected. The combination of DSP hardware and related software algorithms enables the digital anti-noise processor to achieve a speech clarity of not less than 98 in a 120dB high noise environment.
figure 1
Software and hardware design scheme Main working principle The processor mainly completes the high-definition communication function of voice under high noise environment. The voice signal and environmental noise are input to the pre-amplifier stage through the MIC. The function of the pre-amplifier stage is to amplify the voice and environmental noise to the amplitude that can be recognized by the A / D in the dedicated DSP chip, so that the A / D can convert the signal normally. The analog signal is converted into a 12-bit digital signal after A / D conversion and enters the arithmetic unit of the DSP. The DSP completes the measurement of the surrounding environmental noise and establishes a mathematical model within the first 3 seconds, and then processes the voice and noise according to the given algorithm , The processing result is sent to D / A through the data bus, and then sent to the post-amplifier after smoothing filtering. The role of the post-amplifier is to meet the input requirements of the related equipment.
DSP chip selection The JK-DP10 digital anti-noise processor designed in this paper has higher requirements for the digital signal processor chip. The chip must have a strong real-time processing performance, but also must have a high computing speed and data throughput capacity; also requires low power consumption, external A / D, D / A and Flash flash memory is best integrated within the DSP, to Reduce product volume. Therefore, one of the TMS320C5XX series DSP chips is selected as the processing chip, and high-speed A / D, D / A and 32KB Flash are used for program loading.
Software algorithm scheme Digital anti-noise processor is realized by adaptive filter. The adaptive filter has the ability to track the changes of the signal and noise, so that the characteristics of the filter also change with the change of the signal and noise to achieve the optimal filtering effect.
The characteristic change of the adaptive filter is realized by the adaptive algorithm by adjusting the filter weight coefficients. In general, the adaptive filter consists of two parts, one is the filter structure, and the other is the adaptive algorithm for adjusting the filter coefficients. The structure of the adaptive filter adopts the FIR structure. The processing of in-band white noise can not achieve the optimal noise reduction effect with the classic LMS algorithm, but also use the auto-correlation characteristics of the noise and the power spectral density characteristics, and make appropriate modifications based on the LMS algorithm to achieve the best Noise reduction effect.
The DSP realization structure of the digital anti-noise microphone group is shown in Figure 1.
The original input signal d (n) includes signal and noise, and x (n) is the reference noise input. This adaptive filter essentially completes the noise estimation in d (n), and subtracts the estimated value y (n) from the original channel to achieve the result of noise cancellation. Of course, the estimated value y (n) and the original input The signal is not a simple algebraic subtraction, but a set of corresponding software algorithms, such as power spectrum analysis of related power.
In Figure 1, the adaptive filter uses a horizontal structure, and the filter output y (n) is expressed as:
N- 1
y (n) = ∑ Wi (n- i)
i = 0
N is the order of the filter.
Software Design The complexity of an adaptive filter implementation is usually measured by the number of multiplications and orders it requires. Based on the adaptive filter system implemented by DSP, the data throughput and data processing speed of its DSP chip are also very important. This digital anti-noise processor uses a 120-order adaptive digital filter, and selects a DSP chip with a calculation speed of 40MIPS as the main processor. Because the DSP chip contains A / D and D / A and 16KB of flash memory, these On-chip resources make the implementation of adaptive filters more efficient.
According to the autocorrelation characteristics and power density of the noise, in addition to the LMS algorithm in the FIR filter of the traditional symmetric horizontal structure, the power spectral density of the noise and signal is estimated, that is, the 16 values ​​of the sample code Carry out square accumulation, find its average power value, and compare it with the same point power value. The difference after the comparison is divided by the set noise threshold value. Smaller, the signal output amplitude becomes larger, if the result is less than or equal to 1, the filter weight becomes larger, and the signal output amplitude becomes smaller.
Customized special anti-noise DSP chip After the debugging work is completed, it is handed over to the company specializing in the production of DSP chips to make a DSP special chip with anti-noise function. After actual measurement, the power consumption of the whole machine is not more than 70mA, and the pin of the DSP chip is reduced to 64 pins, which greatly reduces the area of ​​the printed circuit board. Because the software code is masked once in the chip, the trouble of writing code every time is eliminated, and the workload of debugging is reduced. Under normal circumstances, the module can be completed as long as 3 points of debugging, which greatly reduces the debugging cost and is beneficial to mass production.
Conclusion The digital anti-noise module uses a DSP chip and uses adaptive technology, which not only improves the anti-noise performance of communication products, but also reduces production costs. This module has been successfully applied to China's airborne communication equipment.
Filter in air purifiers are used to reduce the concentration of contaminants (including dust, pollen, pet dander, mold spores, dust mite feces, smoke particles and volatile organic compounds) found in the environment. These contaminants can act as allergens and pose a health risk. So, Air Purifier is useful and even necessary for people.
Air Filter,Voc Air Filter,Air Purifier Filters,Air Filtration System
Dongguan V1 Environmental Technology Co., Ltd. , https://www.v1airpurifier.com