SINUSOIDAL NOISE CANCELATION DENGAN MENGGUNAKAN DIGITAL SIGNAL PROCESSING STARTER KIT TMS320C6713

Original speech signal is needed both in telecommunications and in some instruments in a variety of fields. Not infrequently, the original audio signal is damaged due to noise. This noise can cause the original signal changes in the actual form. In the final project will be designed FIR filter to remove noise by using TMS320C6713 DSK. Sound signal to be input to the mixed noise removal filter system noise. The mixed voice signal will be searched by subtracting the signal difference to noise signal output FIR filter to get the signal e (n), and then do an adaptation resulting filter coefficients. Results of the adaptive filter coefficients would be put back to calculate the noise signal output next FIR filter. Original voice signal used is the word "sinus" uttered by teenage boys, teenage girls, boys and girls. Girl's voice had the highest frequency with an average 522.50 Hz, the frequency of the sound of the boys 462.63 Hz, the sound frequency of 222.58 Hz girls and voice frequencies teenage boys at 201.49 Hz. Noise signal used is 100 Hz sinusoidal noise. From the test results obtained for the system output signal SNR sound input teenage boy was 19.94 dB. SNR output signal to the input of 21.39 dB girls. SNR signal input output system for boys was 34.70 dB. SNR output signal to the input system daughters of 35.52 dB

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Modified GSC Method to Reduce the Distortion of the Enhanced Speech Signal Using Cross-Correlation and Sidelobe Neutralization

Applied Sciences ◽

10.3390/app11146288 ◽

2021 ◽

Vol 11 (14) ◽

pp. 6288

Author(s):

Hang Su ◽

Chang-Myung Lee

Keyword(s):

Speech Signal ◽

Output Signal ◽

Cross Correlation ◽

Acoustic Noise ◽

Audio Signal ◽

Noise Signal ◽

Lms Algorithm ◽

Least Mean Square ◽

Experiment Data ◽

Noise Component

The generalized sidelobe canceller (GSC) method is a common algorithm to enhance audio signals using a microphone array. Distortion of the enhanced audio signal consists of two parts: the residual acoustic noise and the distortion of the desired audio signal, which means that the desired audio signal is damaged. This paper proposes a modified GSC method to reduce both kinds of distortion when the desired audio signal is a non-stationary speech signal. First, the cross-correlation coefficient between the canceling signal and the error signal of the least mean square (LMS) algorithm was added to the adaptive process of the GSC method to reduce the distortion of the enhanced signal while the energy of the desired signal frame was increased suddenly. The sidelobe pattern of beamforming was then presented to estimate the noise signal in the beamforming output signal of the GSC method. The noise component of the beamforming output signal was decreased by subtracting the estimated noise signal to improve the denoising performance of the GSC method. Finally, the GSC-SN-MCC method was proposed by merging the above two methods. The experiment was performed in an anechoic chamber to validate the proposed method in various SNR conditions. Furthermore, the simulated calculation with inaccurate noise directions was conducted based on the experiment data to inspect the robustness of the proposed method to the error of the estimated noise direction. The experiment data and calculation results indicated that the proposed method could reduce the distortion effectively under various SNR conditions and would not cause more distortion if the estimated noise direction is far from the actual noise direction.

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The Design and Implementation of Infrared Communication System

Applied Mechanics and Materials ◽

10.4028/www.scientific.net/amm.496-500.2091 ◽

2014 ◽

Vol 496-500 ◽

pp. 2091-2094

Author(s):

Jia Min Zhou ◽

Ren Er Yang ◽

Xu Yan Ni

Keyword(s):

Communication System ◽

Audio Signal ◽

Data Communication ◽

Digital Signal ◽

Single Chip ◽

Wireless Data ◽

Infrared Communication ◽

Voice Signal ◽

Temperature Signal ◽

Wireless Data Communication

As a kind of wireless communication, Infrared wireless data communication has been developing rapidly and has been widely used in the close wireless data communication. In this paper, there is the design and achievement of infrared communication system, which has realized the short distance transmission of voice and digital signal (temperature signal). The modem part of the system is implemented by the phase locked circuit composed of LM567. At the same time, with the single chip processor as the core, the temperature information which has been encoded is added to the audio signal and then to be transmitted to the receiver to be decoded to restore the temperature information. After the test of implementation, this system can transmit the speech signal and digital signal directly and the voice signal received has no obvious distortion. This system is simple in making and has a good application prospect.

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Eliminating unwanted signals in sound by using digital signal processing system

Indonesian Journal of Electrical Engineering and Computer Science ◽

10.11591/ijeecs.v18.i2.pp829-834 ◽

2020 ◽

Vol 18 (2) ◽

pp. 829

Author(s):

Amer T Saeed ◽

Zaid Raad Saber ◽

Ahmed M. Sana ◽

Musa A. Hameed

Keyword(s):

Signal Processing ◽

Digital Signal Processing ◽

Digital Filters ◽

Processing System ◽

Digital Signal ◽

Noise Signal ◽

Time Range ◽

Signal Processing System ◽

Digital Signal Processing System ◽

Sound Files

<a name="_Hlk536186602"></a>Unwanted signals or noise signals in sound files are considered one of the major challenges and issues for a thousand users. It is impossible to reduce or remove these noise signals without identifying their types and ranges. Therefore, to address one of the big problems in the digital or analogue communication, which is noise signals or unwanted signals, an adaptive selection method and noise signal removal algorithm are proposed in this research. The proposed algorithm is done through specifying the types of undesirable signals, frequency, and time range, then utilizing digital signal processing system which includes design several types of digital filters based on the types and numbers of unwanted signals. Four digital filters are used in this research to remove noise signals from the sound file by implementing the proposed algorithm using Matlab Code. Results show that our proposed algorithm was done successfully and the whole noise signals were removed without any negative consequence in the output sound signal. Unwanted signals or noise signals in sound files are considered one of the major challenges and issues for a thousand users. It is impossible to reduce or remove these noise signals without identifying their types and ranges. Therefore, to address one of the big problems in the digital or analogue communication, which is noise signals or unwanted signals, an adaptive selection method and noise signal removal algorithm are proposed in this research. The proposed algorithm is done through specifying the types of undesirable signals, frequency, and time range, then utilizing digital signal processing system which includes design several types of digital filters based on the types and numbers of unwanted signals. Four digital filters are used in this research to remove noise signals from the sound file by implementing the proposed algorithm using Matlab Code. Results show that our proposed algorithm was done successfully and the whole noise signals were removed without any negative consequence in the output sound signal.

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Voice re-education and its evaluation through digital signal processing: A case study

Loquens ◽

10.3989/loquens.2017.040 ◽

2017 ◽

Vol 4 (1) ◽

pp. 040

Author(s):

Zulema Santana-López ◽

Óscar Domínguez-Jaén ◽

Jesús B. Alonso ◽

María Del Carmen Mato-Carrodeguas

Keyword(s):

Voice Quality ◽

Quality Measurement ◽

Digital Signal ◽

Communication Process ◽

Time Frequency ◽

Voice Signal ◽

Functional Dysphonia ◽

Before And After ◽

Voice Pathologies ◽

The Voice

Voice pathologies, caused either by functional dysphonia or organic lesions, or even by just an inappropriate emission of the voice, may lead to vocal abuse, affecting significantly the communication process. The present study is based on the case of a single patient diagnosed with myasthenia gravis (Erb-Goldflam syndrome). In this case, this affection has caused, among other disruptions, a dysarthria. For its treatment, a technique for the education and re-education of the voice has been used, based on a resonator element: the cellophane screen. This article shows the results obtained in the patient after applying a vocal re-education technique called the Cimardi Method: the Cellophane Screen, which is a pioneering technique in this field. Changes in the patient’s voice signal have been studied before and after the application of the Cimardi Method in different domains of study: time-frequency, spectrum, and cepstrum. Moreover, parameters for voice quality measurement, such as shimmer, jitter and harmonic-to-noise ratio (HNR), have been used to quantify the results obtained with the Cimardi Method. Once the results were analyzed, it has been observed that the Cimardi Method helps to produce a more natural and free vocal emission, which is very useful as a rehabilitation therapy for those people presenting certain vocal disorders.

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Secure Communication of Fractional Complex Chaotic Systems Based on Fractional Difference Function Synchronization

Complexity ◽

10.1155/2019/7242791 ◽

2019 ◽

Vol 2019 ◽

pp. 1-10 ◽

Cited By ~ 5

Author(s):

Jiaxun Liu ◽

Zuoxun Wang ◽

Minglei Shu ◽

Fangfang Zhang ◽

Sen Leng ◽

...

Keyword(s):

Secure Communication ◽

Chaotic Systems ◽

Digital Signal ◽

Fractional Difference ◽

Voice Signal ◽

Difference Function ◽

Image Cryptosystem ◽

Definition Of ◽

Complex Chaotic Systems ◽

Function Synchronization

Fractional complex chaotic systems have attracted great interest recently. However, most of scholars adopted integer real chaotic system and fractional real and integer complex chaotic systems to improve the security of communication. In this paper, the advantages of fractional complex chaotic synchronization (FCCS) in secure communication are firstly demonstrated. To begin with, we propose the definition of fractional difference function synchronization (FDFS) according to difference function synchronization (DFS) of integer complex chaotic systems. FDFS makes communication secure based on FCCS possible. Then we design corresponding controller and present a general communication scheme based on FDFS. Finally, we respectively accomplish simulations which transmit analog signal, digital signal, voice signal, and image signal. Especially for image signal, we give a novel image cryptosystem based on FDFS. The results demonstrate the superiority and good performances of FDFS in secure communication.

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A Low-Power CMOS Wireless Acoustic Sensing Platform for Remote Surveillance Applications

Sensors ◽

10.3390/s20010178 ◽

2019 ◽

Vol 20 (1) ◽

pp. 178

Author(s):

Yong Wang ◽

Ranran Zhou ◽

Zhenyue Liu ◽

Bingbo Yan

Keyword(s):

Low Power ◽

Maximum Output Power ◽

Audio Signal ◽

Digital Signal ◽

Maximum Output ◽

Acoustic Sensing ◽

Sensing Platform ◽

Remote Surveillance ◽

Standby Mode ◽

Surveillance Applications

A low-power wireless acoustic sensing platform for remote surveillance applications based on a 180 nm CMOS technology is proposed in this paper. The audio signal, which is acquired by a microphone, is first amplified and filtered. Then, the analog signal is converted to a digital signal by a 10-bit analog-to-digital converter (ADC). A digital automatic gain control module is integrated to obtain an optimal input of the ADC. The digital signal is modulated and transmitted at the 433 MHz ISM band after being repacked and encoded. To save power for portable applications, the chip switches to standby mode when no audio is detected. The wireless sensing platform occupies a chip area of 1.76 mm 2 . The supply voltage is 2.5 V for the power amplifier and 1.8 V for other circuits. The measured maximum output power is 5.7 dBm and the transmission distance is over 500 m for real application scenarios. The chip consumes 25.1 mW power in normal work mode and 0.058 mW in standby mode. Compared to existing wireless acoustic sensors, the proposed wireless acoustic sensing platform can achieve features such as compactness, power efficiency, and reliability.

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Audio Signal and Troubleshooting System Based on Wireless Sensor

International Journal of Online Engineering (iJOE) ◽

10.3991/ijoe.v14i06.8702 ◽

2018 ◽

Vol 14 (06) ◽

pp. 113

Author(s):

Min Shen ◽

Zhiling Tang

Keyword(s):

Wireless Sensor Networks ◽

Sensor Networks ◽

Detection System ◽

Transmission Rate ◽

Audio Signal ◽

Wireless Sensor ◽

Stable Operation ◽

Voice Signal ◽

Sound Detection ◽

Monitoring Software

To explore the application of audio signal in the troubleshooting system, a sound detection system was designed. The system consists of three parts: voice acquisition node, aggregation node and host computer monitoring software. Time protocol synchronization (TPSN) algorithm was used to realize the synchronization between nodes. The algorithm and the trilateration method were applied to the system. The applications of the sound detection system in the field of fault sound source localization were realized.Wireless sensor networks were used in sound detection systems, which had many advantages. On the one hand, complicated wiring was avoided. It had the advantages of easy to set up and easy to move.On the other hand, some self-organizing and adaptive features in wireless sensor networks and some methods of synchronization and localization could be introduced. Thewhole system was more flexible and its application was more extensive.Using monitoring software, users can remotely access to the scene of the voice signal.The results showed that the system had high transmission rate, stable operation and small positioning error. Therefore, it has good application prospects.

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Approximate Compressor-Based Multiplier Design Methodology for Error-Resilient Digital Signal Processing

Journal of Circuits System and Computers ◽

10.1142/s0218126620502333 ◽

2020 ◽

Vol 29 (14) ◽

pp. 2050233

Author(s):

Zhixi Yang ◽

Xianbin Li ◽

Jun Yang

Keyword(s):

Image Processing ◽

Signal Processing ◽

Digital Signal Processing ◽

Resource Utilization ◽

Finite Impulse Response ◽

Digital Signal ◽

Fir Filter ◽

Electrical Performance ◽

Approximate Computing ◽

Dsp Applications

As many digital signal processing (DSP) applications such as digital filtering are inherently error-tolerant, approximate computing has attracted significant attention. A multiplier is the fundamental component for DSP applications and takes up the most part of the resource utilization, namely power and area. A multiplier consists of partial product arrays (PPAs) and compressors are often used to reduce partial products (PPs) to generate the final product. Approximate computing has been studied as an innovative paradigm for reducing resource utilization for the DSP systems. In this paper, a 4:2 approximate compressor-based multiplier is studied. Approximate 4:2 compressors are designed with a practical design criterion, and an approximate multiplier that uses both truncation and the proposed compressors for PP reduction is subsequently designed. Different levels of truncation and approximate compression combination are studied for accuracy and electrical performance. A practical selection algorithm is then leveraged to identify the optimal combinations for multiplier designs with better performance in terms of both accuracy and electrical performance measurements. Two real case studies are performed, i.e., image processing and a finite impulse response (FIR) filter. The design proposed in this paper has achieved up to 16.96% and 20.81% savings on power and area with an average signal-to-noise ratio (SNR) larger than 25[Formula: see text]dB for image processing; similarly, with a decrease of 0.3[Formula: see text]dB in the output SNR, 12.22% and 30.05% savings on power and area have been achieved for an FIR filter compared to conventional multiplier designs.

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Design of the Control System for Rehabilitation Horse Based on MCU STC89C52

Advanced Engineering Forum ◽

10.4028/www.scientific.net/aef.2-3.234 ◽

2011 ◽

Vol 2-3 ◽

pp. 234-238

Author(s):

Hai Tao Qi ◽

Guang Lei Feng ◽

Hong Wang

Keyword(s):

Control System ◽

Real Time ◽

Digital Signal ◽

Human Machine Interface ◽

Analog Signal ◽

The Real ◽

User Input ◽

Speed Controller ◽

Machine Interface ◽

Signal Output

It introduces a design of the control system for rehabilitation horse based on MCU STC89C52. The system’s control core is an 8-bit MCU STC89C52. First, the user input commands through the keyboard, then send commands to the DA conversion chip PCF8591 which can achieve the digital signal to analog signal output after dealing with MCU. Finally, PCF8591 send analog signal to the speed controller of DC motor in order to control the DC motor’s speed. Meanwhile, it builds a human-machine interface (HMI) to display the real-time speed of the horse through LCD.

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Research and Application of a FPGA-Based Dynamic Distributed Algorithm

Applied Mechanics and Materials ◽

10.4028/www.scientific.net/amm.58-60.1696 ◽

2011 ◽

Vol 58-60 ◽

pp. 1696-1700

Author(s):

Wei Zheng Ren ◽

Ying Gao ◽

Yan Song Cui

Keyword(s):

Signal Processing ◽

Digital Signal Processing ◽

Processing Speed ◽

Distributed Algorithm ◽

Digital Signal ◽

Fir Filter ◽

Effective Solution ◽

Data Bus

A dynamic distributed algorithm (DDA) with a look-up dynamic table instead of ROM was put forward based on the theory of signed distributed algorithm, in order to improve processing speed and flexibility of product sum on FPGA. Since the DDA occupies few hardware resources, performs fast operation and realizes programmable coefficient, the limitation of digital signal processing speed on fixed data bus width and sequential operation was avoided by using the algorithm. At the same time, an effective solution to realizing coefficient programmable FIR filter was presented.

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