945 resultados para localizzazione audio array microfoni MATLAB simulazione digilent


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Melodic motifs form essential building blocks in Indian Classical music. The motifs, or key phrases, providestrong cues to the identity of the underlying raga in both Hindustani and Carnatic styles of Indian music. Automatic identification and clustering of similar motifs is relevant in this context. The inherent variations in various instances of a characteristic phrase in a bandish (composition)performance make it challenging to identify similar phrases in a performance. A nyas svara (long note)marks the ending of these phrases. The proposed method does segmentation of phrases through identification ofnyas and computes similarity with the reference characteristic phrase.

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Rapid amplification of cDNA ends (RACE) is a widely used approach for transcript identification. Random clone selection from the RACE mixture, however, is an ineffective sampling strategy if the dynamic range of transcript abundances is large. To improve sampling efficiency of human transcripts, we hybridized the products of the RACE reaction onto tiling arrays and used the detected exons to delineate a series of reverse-transcriptase (RT)-PCRs, through which the original RACE transcript population was segregated into simpler transcript populations. We independently cloned the products and sequenced randomly selected clones. This approach, RACEarray, is superior to direct cloning and sequencing of RACE products because it specifically targets new transcripts and often results in overall normalization of transcript abundance. We show theoretically and experimentally that this strategy leads indeed to efficient sampling of new transcripts, and we investigated multiplexing the strategy by pooling RACE reactions from multiple interrogated loci before hybridization.

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We report on two patients with de novo subtelomeric terminal deletion of chromosome 6p. Patient 1 is an 8-month-old female born with normal growth parameters, typical facial features of 6pter deletion, bilateral corectopia, and protruding tongue. She has severe developmental delay, profound bilateral neurosensory deafness, poor visual contact, and hypsarrhythmia since the age of 6 months. Patient 2 is a 5-year-old male born with normal growth parameters and unilateral hip dysplasia; he has a characteristic facial phenotype, bilateral embryotoxon, and moderate mental retardation. Further characterization of the deletion, using high-resolution array comparative genomic hybridization (array-CGH; Agilent Human Genome kit 244 K), revealed that Patient 1 has a 8.1 Mb 6pter-6p24.3 deletion associated with a contiguous 5.8 Mb 6p24.3-6p24.1 duplication and Patient 2 a 5.7 Mb 6pter-6p25.1 deletion partially overlapping with that of Patient 1. Complementary FISH and array analysis showed that the inv del dup(6) in Patient 1 originated de novo. Our results demonstrate that simple rearrangements are often more complex than defined by standard techniques. We also discuss genotype-phenotype correlations including previously reported cases of deletion 6p.

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We have constructed a forward modelling code in Matlab, capable of handling several commonly used electrical and electromagnetic methods in a 1D environment. We review the implemented electromagnetic field equations for grounded wires, frequency and transient soundings and present new solutions in the case of a non-magnetic first layer. The CR1Dmod code evaluates the Hankel transforms occurring in the field equations using either the Fast Hankel Transform based on digital filter theory, or a numerical integration scheme applied between the zeros of the Bessel function. A graphical user interface allows easy construction of 1D models and control of the parameters. Modelling results are in agreement with other authors, but the time of computation is less efficient than other available codes. Nevertheless, the CR1Dmod routine handles complex resistivities and offers solutions based on the full EM-equations as well as the quasi-static approximation. Thus, modelling of effects based on changes in the magnetic permeability and the permittivity is also possible.

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This paper proposes a novel high capacity robust audio watermarking algorithm by using the high frequency band of the wavelet decomposition at which the human auditory system (HAS) is not very sensitive to alteration. The main idea is to divide the high frequency band into frames and, for embedding, to change the wavelet samples depending on the average of relevant frame¿s samples. The experimental results show that the method has a very high capacity (about 11,000 bps), without significant perceptual distortion (ODG in [¿1 ,0] and SNR about 30dB), and provides robustness against common audio signal processing such as additive noise, filtering, echo and MPEG compression (MP3).

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Many audio watermarking schemes divide the audio signal into several blocks such that part of the watermark is embedded into each of them. One of the key issues in these block-oriented watermarking schemes is to preserve the synchronisation, i.e. to recover the exact position of each block in the mark recovery process. In this paper, a novel time domain synchronisation technique is presented together with a new blind watermarking scheme which works in the Discrete Fourier Transform (DFT or FFT) domain. The combined scheme provides excellent imperceptibility results whilst achieving robustness against typical attacks. Furthermore, the execution of the scheme is fast enough to be used in real-time applications. The excellent transparency of the embedding algorithm makes it particularly useful for professional applications, such as the embedding of monitoring information in broadcast signals. The scheme is also compared with some recent results of the literature.

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Peer-reviewed

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This paper describes an audio watermarking scheme based on lossy compression. The main idea is taken from an image watermarking approach where the JPEG compression algorithm is used to determine where and how the mark should be placed. Similarly, in the audio scheme suggested in this paper, an MPEG 1 Layer 3 algorithm is chosen for compression to determine the position of the mark bits and, thus, the psychoacoustic masking of the MPEG 1 Layer 3compression is implicitly used. This methodology provides with a high robustness degree against compression attacks. The suggested scheme is also shown to succeed against most of the StirMark benchmark attacks for audio.

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Process variations are a major bottleneck for digital CMOS integrated circuits manufacturability and yield. That iswhy regular techniques with different degrees of regularity are emerging as possible solutions. Our proposal is a new regular layout design technique called Via-Configurable Transistors Array (VCTA) that pushes to the limit circuit layout regularity for devices and interconnects in order to maximize regularity benefits. VCTA is predicted to perform worse than the Standard Cell approach designs for a certain technology node but it will allow the use of a future technology on an earlier time. Ourobjective is to optimize VCTA for it to be comparable to the Standard Cell design in an older technology. Simulations for the first unoptimized version of our VCTA of delay and energy consumption for a Full Adder circuit in the 90 nm technology node are presented and also the extrapolation for Carry-RippleAdders from 4 bits to 64 bits.

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The well-known structure of an array combiner along with a maximum likelihood sequence estimator (MLSE) receiveris the basis for the derivation of a space-time processor presentinggood properties in terms of co-channel and intersymbol interferencerejection. The use of spatial diversity at the receiver front-endtogether with a scalar MLSE implies a joint design of the spatialcombiner and the impulse response for the sequence detector. Thisis faced using the MMSE criterion under the constraint that thedesired user signal power is not cancelled, yielding an impulse responsefor the sequence detector that is matched to the channel andcombiner response. The procedure maximizes the signal-to-noiseratio at the input of the detector and exhibits excellent performancein realistic multipath channels.

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The subject of the thesis was the digital audio broadcasting technology developed in the Eureka project 147. The research was based on the literature on the subject. At first, some reasons for the digitisation of broadcasting technology were given. Next, the channel multiplexing and channel coding methods employed by digital radio were discussed. The design of these methods is based on certain phenomena related to the propagation of radio-frequency signals, and these phenomena were also described. After that, audio and data transfer mechanisms as well as the structure of digital radio network were explained. Furthermore, digital audio and data services were considered. Finally, the digital radio was examined from marketing and administrative aspects. From a merely technical point of view, the digital radio technology offers several improvements in comparison with analogue technology. However, the digital radio has not become as widespread as it was perhaps originally expected during its development.

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NURBS are widely used parametric approximation curves or surfaces. NURBS can be applied to the many applications. Examples of these applications are some computeraided design applications and some medical applications. Use of NURBS is very intuitive. The objective of this work was to implement the NURBS toolbox in the Matlab environment. Matlab is a program for many kinds of computational purposes. Matlab is also a programming language. NURBS toolbox implemented in this work offers a user an opportunity to use functions of this toolbox as parts of the user’s own programs. Current version of NURBS toolbox includes functions for NURBS curve and surface evaluation. The toolbox is designed such, that it allows extensions and enhancements in the future.