Miguel Arjona Ramírez
University of São Paulo
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Publication
Featured researches published by Miguel Arjona Ramírez.
Applied Physics Letters | 2005
Alexandre Zirpoli Simões; Miguel Arjona Ramírez; N. A. Perruci; C. S. Riccardi; Elson Longo; José Arana Varela
Ferroelectric Bi3.25La0.75Ti3O12 (BLT) thin films were deposited on Pt∕Ti∕SiO2∕Si substrates by the polymeric precursor method. The films present c-axis preferred orientation after annealing at 700°C for 2h in conventional furnace. All the capacitors showed good polarization fatigue characteristics at least up to 1×1010 bipolar pulse cycles and excellent retention properties up to 1×104s. We found that the polarization loss is insignificant with five write∕read voltages at a waiting time of 10 000 S. Independently of the applied electric field the retained switchable polarization approached a nearly steady-state value after a retention time of 10s.
IEEE Signal Processing Letters | 2006
Miguel Arjona Ramírez; Mário Minami
Linear prediction (LP) analysis, split in two stages, is proposed for a combined time-frequency analysis. The first-stage LP is used to obtain the residual signal and extract each one of its cycles, whose harmonic spectrum is then modeled by the second-stage estimate from discrete all-pole algorithms. Thus, harmonic cycle spectra are modeled with less than 1 dB in log spectral distortion (SD). Further, a method is proposed to approximate the log SD target. A linear approximation to the log power spectral ratio in the log SD gradient is shown to provide better model fit to harmonic cycle spectra.
IEEE Transactions on Speech and Audio Processing | 2000
Miguel Arjona Ramírez; Max Gerken
A joint position and amplitude search algorithm is proposed for algebraic multipulse codebooks to be used in code-excited linear predictive (CELP) coders. The joint search complexity is below one quarter that of the focused search and ranks below those of the G.729A and IS-641-A coders. Listening tests indicate an equivalence in perceived quality.
international conference on acoustics speech and signal processing | 1999
Miguel Arjona Ramírez; Max Gerken
A joint amplitude and position search procedure is proposed for searching algebraic multipulse codebooks. It is implemented within the reference G.723.1 codec as an example. This joint search method is shown to reduce down to one third the number of comparisons per subframe relative to the focused search over an extensive speech database. An efficient implementation of the joint search is derived which incorporates backward filtering of the residual target vector and precomputation of autocorrelation elements, bringing about a reduction in complexity of one-third in comparison to the focused search. The joint search performs about one thirtieth as many comparisons as the full position search.
ieee international telecommunications symposium | 1998
Miguel Arjona Ramírez; M. Gerken
A joint amplitude and position search is proposed for searching algebraic multipulse codebooks for CELP speech coding. It is implemented within the reference G.723.1 codec as an example. This joint search method is shown to reduce down to one-third the number of comparisons per subframe relative to the focused search over an extensive speech database. An efficient implementation of the joint search is derived which incorporates backward filtering of the residual target vector and precomputation of auto-correlation elements, bringing about a reduction in complexity of one third in comparison to the focused search. The joint search performs about one thirtieth as many comparisons as the full position search.
IEEE Signal Processing Letters | 2013
Miguel Arjona Ramírez
Vector quantization (VQ) of speech spectral vectors has been improved by techniques such as split VQ (SVQ), vector transforms and direction switching. This letter proposes Intra-Predictive Switched SVQ (IPSSVQ) with direction switching by a Gaussian Mixture Model (GMM), using at the frame level the prediction-based lower-triangular transform (PLT), which has lower complexity than the Karhunen-Loève transform (KLT). It is shown that equivalent results to GMM KLT SSVQ may be obtained in the quantization of line spectral frequency (LSF) vectors from wideband speech signals, such as transparent coding throughout the range from 46 bit/frame to 41 bit/frame, with about three-fourths as much operational complexity.
Archive | 2012
Marcelo Ornaghi Orlandi; Miguel Arjona Ramírez; Cesar R. Foschini; A. A. Felix; José A. Varela
There is a constant need in the modern electronic industry for capacitors with high capacity per volume in order to use in many applications such as memories devices, energy storage, microwave filters, among others. The synthesis, characterization and study of materials with a very high or giant dielectric constant are in particular important and have been studied by many investigators. Then the search for high dielectric materials has been driven, which must requires some special properties such as to keep dielectric constant almost independent of temperature and frequency and low dielectric loss. In recent years many material systems have been searched, including perovskite and others, aiming to achieve a reliable material for practical applications. In this chapter we review many classes of materials including titanates, nickeletes, cuprates, multiferroics and composites such as cermets, ceramic/polymer and ceramic/ceramic. To the end of this work is presented the most important models in the literature to explain mechanism for giant dielectric constant. The main task is to understand the mechanism that controls materials properties in order to synthesize an optimized material that maximizes those properties by using sol-gel or other chemical-like synthesis for many applications in the electronic industries.
international conference on digital signal processing | 2011
Miguel Arjona Ramírez
A method is proposed for split vector quantization (SVQ) that, while keeping up with the lower search complexity inherent to SVQ, reduces the attendant split loss. It operates partially in the training and encoding phases by normalizing the bandwidth covered by each split. The action is completed in the decoding phase by renormalizing the bandwidth spanned by each split of the codebook to that allowed by a nonoverlapping constraint with neighboring split bands. Therefore, the method is referred to as renormalized SPV (RSVQ). The performance of RSVQ is investigated in comparison to standard SVQ for coding line spectral frequency (LSF) vectors that parameterize the spectral envelope of wideband speech and it is found to save four bits, reaching transparent coding at 42 bit/frame.
2011 IEEE Workshop on Computational Intelligence in Biometrics and Identity Management (CIBIM) | 2011
Rodolfo Vertamatti; Miguel Arjona Ramírez
This study examines human fluctuating asymmetry as a tool to improve multibiometrics. A new technique called Bilateral Processing (BP) was implemented to analyze discordances in left and right trait sides. BP tested visible and infrared spectrum trait images using Cross-Correlation, Wavelets and Neural Networks. Selected traits were tooth, ear, iris, fingerprint, nose and cheek. Acoustic BP was also implemented for skin vibration asymmetry evaluation during voiced sounds and compared to MFCC plus Vector Quantization speaker recognition system. Image and acoustic BP investigated bilateral sides of 9 adult male brothers during one year. Results achieved 100% identification in all biometrics treated with BP, compared to a maximum 44% of correct identification without BP. Apart from ameliorating performance, the study concludes that body structure analyses incorporating biognostic mechanisms can complement traditional recognition approaches.
international conference on acoustics, speech, and signal processing | 2003
Miguel Arjona Ramírez
Waveform interpolation (WI) models for speech coding contain many parameters whose sampling rates may not be simply related so that most implementations tend to fix their rates right from the waveform extraction stage, thereby compromising quality by departing from perfect reconstruction. A waveform extractor is proposed which samples waveform cycles of the original prediction residual signal at their natural variable rate so that it can perfectly reconstruct the signal. The speech coder, which may operate at a uniform sampling rate, is coupled to the waveform extractor by means of an evolving waveform interpolator that may handle several interpolation methods and sampling rates for a variety of fixed and variable rate coders, including conventional WI coders.