Structure of a sampling rate converter based on elementary signal processing blocks

2008 ◽  
Vol 18 (4) ◽  
pp. 480-487
Author(s):  
Eduardo Martos-Naya ◽  
Jesús López-Fernández ◽  
Luis Díez del Río ◽  
José Tomás Entrambasaguas Muñoz
2020 ◽  
Vol 10 (19) ◽  
pp. 6956
Author(s):  
Yisak Kim ◽  
Juyoung Park ◽  
Hyungsuk Kim

Acquisition times and storage requirements have become increasingly important in signal-processing applications, as the sizes of datasets have increased. Hence, compressed sensing (CS) has emerged as an alternative processing technique, as original signals can be reconstructed using fewer data samples collected at frequencies below the Nyquist sampling rate. However, further analysis of CS data in both time and frequency domains requires the reconstruction of the original form of the time-domain data, as traditional signal-processing techniques are designed for uncompressed data. In this paper, we propose a signal-processing framework that extracts spectral properties for frequency-domain analysis directly from under-sampled ultrasound CS data, using an appropriate basis matrix, and efficiently converts this into the envelope of a time-domain signal, avoiding full reconstruction. The technique generates more accurate results than the traditional framework in both time- and frequency-domain analyses, and is simpler and faster in execution than full reconstruction, without any loss of information. Hence, the proposed framework offers a new standard for signal processing using ultrasound CS data, especially for small and portable systems handling large datasets.


1994 ◽  
Vol 116 (4) ◽  
pp. 396-402 ◽  
Author(s):  
Dj. Boussaa ◽  
K. Dang Van ◽  
P. Labbe´ ◽  
H. T. Tang

Three dynamic tests on pressurized elbows involving fatigue-ratcheting as the major failure mode are investigated. The fatigue analysis is carried out with two approaches: the first is global and consists of Markl’s equation; the second is local and is based on local fatigue criteria combined with a suggestion by Coffin to take ratcheting into account. The implementation of these approaches required some elementary signal processing to separate the cyclic part of the strain from ratcheting. Results on narrow-band processes were used to assess cumulative damage. Finite element computations extended the data into uninstrumented locations. The great sensitivity of the implemented criteria to their arguments is discussed and exemplified. Despite the complexities of the issue, computed results match well with the experimental data.


2010 ◽  
Vol 44-47 ◽  
pp. 2569-2572 ◽  
Author(s):  
Xiao Wei Du ◽  
Li Rong Li

A higher performance signal processing module is demanded because of the high sensitivity of fiber-optic sensor probe. Excellent signal processing module is expected have the function of restore the sensor signal and very little to the introduction of noise. In this paper, improved demodulation algorithm is proposed based on the general demodulation algorithm through changing certain parameters of the algorithm aimed to improve the quality of the signal after demodulation. The output signals of scene sensor probe is sampling, sampling data through filtering. By using the improved fiber optic sensor signal demodulation algorithm, the system can fully improve the quality of the signal after demodulation under the premise of maintains the minimum sampling rate.


Author(s):  
Anatoly V. Bychkov ◽  
Irina Yu. Bychkova ◽  
Nadezhda N. Suslova ◽  
Kurbangali K. Alimov

The apparatus of artificial neural networks (ANN) is proposed to be used for signal processing in active ultrasonic (US) vibration control of electrical equipment. A feature of the applied neural network algorithm is that the required information about vibration parameters is embedded in the ultrasound signal’s phase change at its constant amplitude. Under these conditions, traditional spectral analysis of signals requires a high sampling rate and a significant recording duration. When using the direct propagation’s ANN with three hidden layers, it was shown that it is sufficient to use a sampling frequency of 5-6 points for the period of an ultrasonic wave and a recording duration of 4-5 periods to estimate the nonstationary frequency and amplitude of the vibration signal. Estimates of the error in determining the amplitude, frequency and phase of vibrations are obtained. The root-mean-square errors of the neural network algorithm do not exceed units of percent. The possibilities of using a trained neural network for signal processing in a «sliding window» are demonstrated. The accuracy characteristics of the proposed neural network algorithm of signal processing and the possibility of its optimization for electrical equipment’s vibration control are discussed.


Author(s):  
Gordana Jovanovic Dolecek

Digital signal processing (DSP) is an area of science and engineering that has been rapidly developed over the past years. This rapid development is a result of significant advances in digital computers technology and integrated circuits fabrication (Mitra, 2005; Smith, 2002). Classical digital signal processing structures belong to the class of single-rate systems since the sampling rates at all points of the system are the same. The process of converting a signal from a given rate to a different rate is called sampling rate conversion. Systems that employ multiple sampling rates in the processing of digital signals are called multirate digital signal processing systems. Sample rate conversion is one of the main operations in a multirate system (Harris, 2004; Stearns, 2002).


Author(s):  
Ljiljana D. Milic

A multirate filter can be defined as a digital filter in which the input data rate is changed in one or more intermediate points. With the efficient multirate approach, computations are evaluated at the lowest possible sampling rate, thus improving the computational efficiency, increasing the computation speed, and lowering the power consumption. Multirate filters are of essential importance for communications, image processing, digital audio, and multimedia. The role of multirate filtering in modern signal processing systems is threefold: Firstly, they are used whenever there is a need to preserve the signal properties when connecting two systems operating at different sampling rates. Secondly, multirate techniques are used for constructing filters with stringent spectral constraints that are very difficult, even impossible, to be solved otherwise. Thirdly, multirate filters are used in constructing multirate filter banks.


2009 ◽  
Author(s):  
Jonas Bentell ◽  
Dirk Uwaerts ◽  
Jonathan Cloots ◽  
Thomas Bocquet ◽  
Joel Neys ◽  
...  

2013 ◽  
Vol 373-375 ◽  
pp. 579-582
Author(s):  
Jin Lun Chen

The auditory filter-bank is the key component of auditory model, and its implementation involves a lot of computations. The time spent by an auditory filter-bank to finish its work has a significant effect on the real-time implementation of auditory model-based audio signal processing systems. In this paper, a multi-rate implementation of auditory filter bank is presented. Through using low sampling rate for the filters with low centre frequency, and using high sampling rate for the filters with high centre frequency, we can greatly reduce the computation requirement.


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