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Sonar systems are generally used underwater for range finding and detection. Active sonar emits an acoustic signal, or pulse of sound, into the water. The sound bounces off the target object and returns an echo to the sonar transducer. Unlike active sonar, passive sonar does not emit its own signal, which is an advantage for military vessels. But passive sonar cannot measure the range of an object unless it is used in conjunction with other passive listening devices. Multiple passive sonar devices must be used for triangulation of a sound source. No matter whether active sonar or passive sonar, the information included in the reflected signal can not be used without technical signal processing. To extract the useful information from the mixed signal, some steps are taken to transfer the raw acoustic data.
For active sonar, six steps are needed during the signal processing system.
To generate a signal pulse typical analog implementations are oscillators and voltage controlled oscillators (VCO) which are followed by modulators. Amplitude modulation is used to weight the pulse envelopes and to translate the signal spectrum up to some suitable carrier frequency for transmission.
First, in sonar system, the acoustic pressure field can be represented as . The field function includes four variables: time and spatial coordinate . Thus, according to the Fourier transform, in frequency domain [1]
In the formula is temporal frequency and is spatial frequency. We often define as elemental signal, for the reason that any 4-D can be generated by taking a linear combination of elemental signals. Obviously, the direction of gives the direction of propagation of waves, and the speed of the waves is
The wavelength is
In modern world, digital computers do contribute a lot to higher speed and efficiency in data analysis. Thus, it is necessary to convert an analog signal into a digital signal by sample the signal in time domain. The operation can be realized by three devices: a digital conversion device, a dynamic range controller and a digital conversion device.
For simplicity, the sampling is done at equal time intervals. In order to prevent the distortion (that is aliasing in frequency domain) after reconstructing the signal from sampled signal, one must sample at a faster rate.The sampling rate, which can well preserves the information content of an analog signal , is submitted to the Nyquist–Shannon sampling theorem. Assuming the sampling period is T, thus after temporal sampling, the signal is
n is the integer.
It is really an important part for good system performance in sonar system to have appropriate sensor array and beamformer. To infer information about the acoustic field it is necessary to sample the field in space and time. Temporal sampling has already been discussed in a previous section. The sensor array samples the spatial domain, while the beamformer integrate the sensor’s output in a special way to enhance detection and estimation performance of the system. The input to the beamformer is a set of time series, while the output of the beamformer is another set of time series or a set of Fourier coefficient.
For a desired direction , set .
Beamforming is one kind of filtering that can be applied to isolate signal components that are propagating in a particular direction.. In the picture is the most simple beamformer-the weighted delay-and-sum beamformer, which can be accomplished by an array of receivers or sensors. Every triangle is a sensor to sample in spatial domain. After spatial sampling, the sample signal will be weighted and the result is summing all the weighted signals. Assuming an array of M sensors distributed in space, such that the th sensor is located at the position of and the signal received by it is denoted .Thus after beamforming, the signal is
Bandshifting is employed in active and passive sonar to reduce the complexity of the hardware and software required for subsequent processing. For example,in active sonars the received signal is contained in a very narrow band of frequencies, typically about 2 kHz, centered at some high frequency, typically about 50 kHz. To avoid having to sample the received process at the Nyquist rate of 100 kHz, it is more efficient to demodulate the process to baseband and then employ sampling of the complex envelope at only 2 kHz.
Filters and smoothers are used extensively in modem sonar systems. After sampling, the signal is converted from analog signal into a discrete time signal, thus digital filters are only into consideration. What’s more, although some filters are time varying or adaptive, most of the filters are linear shift invariant. Digital filters used in sonar signal processors perform two major functions, the filtering of waveforms to modify the frequency content and the smoothing of waveforms to reduce the effects of noise. The two generic types of digital filters are FIR and infinite impulse response (IIR) filters. Input-output relationship of an FIR filter is
(1-D)
(2-D)
Input-output relationship of an IIR filter is
(1-D)
(2-D)
Both FIR filters and IIR filters have their advantages and disadvantages. First, the computational requirements of a sonar processor are more severe when implementing FIR filters. Second, for an IIR filter, linear phase is always difficult to obtain, so FIR filter is stable as opposed to an IIR filter. What’s more, FIR filters are more easily designed using the windowing technique.
In a word, the goal of the sonar is to extract the informations and data from acoustic space-time field, and put them into designed and prescribed process so that we can apply the different cases into one fixed pattern. Thus, to realize the goal, the final stage of sonar system consists of the following functions:
Linear filters process time-varying input signals to produce output signals, subject to the constraint of linearity. In most cases these linear filters are also time invariant in which case they can be analyzed exactly using LTI system theory revealing their transfer functions in the frequency domain and their impulse responses in the time domain. Real-time implementations of such linear signal processing filters in the time domain are inevitably causal, an additional constraint on their transfer functions. An analog electronic circuit consisting only of linear components will necessarily fall in this category, as will comparable mechanical systems or digital signal processing systems containing only linear elements. Since linear time-invariant filters can be completely characterized by their response to sinusoids of different frequencies, they are sometimes known as frequency filters.
In signal processing, a digital filter is a system that performs mathematical operations on a sampled, discrete-time signal to reduce or enhance certain aspects of that signal. This is in contrast to the other major type of electronic filter, the analog filter, which is typically an electronic circuit operating on continuous-time analog signals.
In signal processing, a finite impulse response (FIR) filter is a filter whose impulse response is of finite duration, because it settles to zero in finite time. This is in contrast to infinite impulse response (IIR) filters, which may have internal feedback and may continue to respond indefinitely.
Infinite impulse response (IIR) is a property applying to many linear time-invariant systems that are distinguished by having an impulse response that does not become exactly zero past a certain point but continues indefinitely. This is in contrast to a finite impulse response (FIR) system, in which the impulse response does become exactly zero at times for some finite , thus being of finite duration. Common examples of linear time-invariant systems are most electronic and digital filters. Systems with this property are known as IIR systems or IIR filters.
A sensor array is a group of sensors, usually deployed in a certain geometry pattern, used for collecting and processing electromagnetic or acoustic signals. The advantage of using a sensor array over using a single sensor lies in the fact that an array adds new dimensions to the observation, helping to estimate more parameters and improve the estimation performance. For example an array of radio antenna elements used for beamforming can increase antenna gain in the direction of the signal while decreasing the gain in other directions, i.e., increasing signal-to-noise ratio (SNR) by amplifying the signal coherently. Another example of sensor array application is to estimate the direction of arrival of impinging electromagnetic waves. The related processing method is called array signal processing. A third examples includes chemical sensor arrays, which utilize multiple chemical sensors for fingerprint detection in complex mixtures or sensing environments. Application examples of array signal processing include radar/sonar, wireless communications, seismology, machine condition monitoring, astronomical observations fault diagnosis, etc.
In signal processing, the Wiener filter is a filter used to produce an estimate of a desired or target random process by linear time-invariant (LTI) filtering of an observed noisy process, assuming known stationary signal and noise spectra, and additive noise. The Wiener filter minimizes the mean square error between the estimated random process and the desired process.
In digital signal processing, downsampling, compression, and decimation are terms associated with the process of resampling in a multi-rate digital signal processing system. Both downsampling and decimation can be synonymous with compression, or they can describe an entire process of bandwidth reduction (filtering) and sample-rate reduction. When the process is performed on a sequence of samples of a signal or a continuous function, it produces an approximation of the sequence that would have been obtained by sampling the signal at a lower rate.
Array processing is a wide area of research in the field of signal processing that extends from the simplest form of 1 dimensional line arrays to 2 and 3 dimensional array geometries. Array structure can be defined as a set of sensors that are spatially separated, e.g. radio antenna and seismic arrays. The sensors used for a specific problem may vary widely, for example microphones, accelerometers and telescopes. However, many similarities exist, the most fundamental of which may be an assumption of wave propagation. Wave propagation means there is a systemic relationship between the signal received on spatially separated sensors. By creating a physical model of the wave propagation, or in machine learning applications a training data set, the relationships between the signals received on spatially separated sensors can be leveraged for many applications.
Beamforming or spatial filtering is a signal processing technique used in sensor arrays for directional signal transmission or reception. This is achieved by combining elements in an antenna array in such a way that signals at particular angles experience constructive interference while others experience destructive interference. Beamforming can be used at both the transmitting and receiving ends in order to achieve spatial selectivity. The improvement compared with omnidirectional reception/transmission is known as the directivity of the array.
In signal processing, a filter bank is an array of bandpass filters that separates the input signal into multiple components, each one carrying a sub-band of the original signal. One application of a filter bank is a graphic equalizer, which can attenuate the components differently and recombine them into a modified version of the original signal. The process of decomposition performed by the filter bank is called analysis ; the output of analysis is referred to as a subband signal with as many subbands as there are filters in the filter bank. The reconstruction process is called synthesis, meaning reconstitution of a complete signal resulting from the filtering process.
The Goertzel algorithm is a technique in digital signal processing (DSP) for efficient evaluation of the individual terms of the discrete Fourier transform (DFT). It is useful in certain practical applications, such as recognition of dual-tone multi-frequency signaling (DTMF) tones produced by the push buttons of the keypad of a traditional analog telephone. The algorithm was first described by Gerald Goertzel in 1958.
In digital signal processing, a cascaded integrator–comb (CIC) is a computationally efficient class of low-pass finite impulse response (FIR) filter that chains N number of integrator and comb filter pairs to form a decimator or interpolator. In a decimating CIC, the input signal is first fed through N integrator stages, followed by a down-sampler, and then N comb stages. An interpolating CIC has the reverse order of this architecture, but with the down-sampler replaced with a zero-stuffer (up-sampler).
Geophysical survey is the systematic collection of geophysical data for spatial studies. Detection and analysis of the geophysical signals forms the core of Geophysical signal processing. The magnetic and gravitational fields emanating from the Earth's interior hold essential information concerning seismic activities and the internal structure. Hence, detection and analysis of the electric and Magnetic fields is very crucial. As the Electromagnetic and gravitational waves are multi-dimensional signals, all the 1-D transformation techniques can be extended for the analysis of these signals as well. Hence this article also discusses multi-dimensional signal processing techniques.
A digital delay line is a discrete element in a digital filter, which allows a signal to be delayed by a number of samples. Delay lines are commonly used to delay audio signals feeding loudspeakers to compensate for the speed of sound in air, and to align video signals with accompanying audio, called audio-to-video synchronization. Delay lines may compensate for electronic processing latency so that multiple signals leave a device simultaneously despite having different pathways.
Two dimensional filters have seen substantial development effort due to their importance and high applicability across several domains. In the 2-D case the situation is quite different from the 1-D case, because the multi-dimensional polynomials cannot in general be factored. This means that an arbitrary transfer function cannot generally be manipulated into a form required by a particular implementation. The input-output relationship of a 2-D IIR filter obeys a constant-coefficient linear partial difference equation from which the value of an output sample can be computed using the input samples and previously computed output samples. Because the values of the output samples are fed back, the 2-D filter, like its 1-D counterpart, can be unstable.
3D sound localization refers to an acoustic technology that is used to locate the source of a sound in a three-dimensional space. The source location is usually determined by the direction of the incoming sound waves and the distance between the source and sensors. It involves the structure arrangement design of the sensors and signal processing techniques.
In signal processing, multidimensional signal processing covers all signal processing done using multidimensional signals and systems. While multidimensional signal processing is a subset of signal processing, it is unique in the sense that it deals specifically with data that can only be adequately detailed using more than one dimension. In m-D digital signal processing, useful data is sampled in more than one dimension. Examples of this are image processing and multi-sensor radar detection. Both of these examples use multiple sensors to sample signals and form images based on the manipulation of these multiple signals. Processing in multi-dimension (m-D) requires more complex algorithms, compared to the 1-D case, to handle calculations such as the fast Fourier transform due to more degrees of freedom. In some cases, m-D signals and systems can be simplified into single dimension signal processing methods, if the considered systems are separable.
Beamforming is a signal processing technique used to spatially select propagating waves. In order to implement beamforming on digital hardware the received signals need to be discretized. This introduces quantization error, perturbing the array pattern. For this reason, the sample rate must be generally much greater than the Nyquist rate.
SAMV is a parameter-free superresolution algorithm for the linear inverse problem in spectral estimation, direction-of-arrival (DOA) estimation and tomographic reconstruction with applications in signal processing, medical imaging and remote sensing. The name was coined in 2013 to emphasize its basis on the asymptotically minimum variance (AMV) criterion. It is a powerful tool for the recovery of both the amplitude and frequency characteristics of multiple highly correlated sources in challenging environments. Applications include synthetic-aperture radar, computed tomography scan, and magnetic resonance imaging (MRI).
The perturbed γ-γ angular correlation, PAC for short or PAC-Spectroscopy, is a method of nuclear solid-state physics with which magnetic and electric fields in crystal structures can be measured. In doing so, electrical field gradients and the Larmor frequency in magnetic fields as well as dynamic effects are determined. With this very sensitive method, which requires only about 10–1000 billion atoms of a radioactive isotope per measurement, material properties in the local structure, phase transitions, magnetism and diffusion can be investigated. The PAC method is related to nuclear magnetic resonance and the Mössbauer effect, but shows no signal attenuation at very high temperatures. Today only the time-differential perturbed angular correlation (TDPAC) is used.