Piñero Sipán, María Gemma
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- PublicationRepresentación temporal y frecuencial de una señal OFDM(Universitat Politècnica de València, 2008-05-12T06:46:43Z) Piñero Sipán, María Gemma; Escuela Técnica Superior de Ingeniería de Telecomunicación; Departamento de Comunicaciones; Instituto Universitario de Telecomunicación y Aplicaciones MultimediaLas figuras muestran 3 símbolos temporales OFDM de longitud Sdata, y el espectro de una secuencia de Nsym símbolos OFDM, respectivamente. Los símbolos originales son símbolos QPSK aleatorios. Sdata debe ser potencia de 2 para una realización eficiente del modulador mediante ifft, pero si no se introduce así, la propia simulación escoge la potencia de 2 más cercana en un rango de 16 a 512. El número de símbolos Nsym está en el rango (2,12).
- PublicationDistributed Affine Projection Algorithm Over Acoustically Coupled Sensor Networks(Institute of Electrical and Electronics Engineers, 2017) Ferrer Contreras, Miguel; González Salvador, Alberto; Diego Antón, María de; Piñero Sipán, María Gemma; Escuela Técnica Superior de Ingeniería de Telecomunicación; Departamento de Comunicaciones; Instituto Universitario de Telecomunicación y Aplicaciones Multimedia; Generalitat Valenciana; Ministerio de Economía, Industria y Competitividad[EN] In this paper, we present a distributed affine projection (AP) algorithm for an acoustic sensor network where the nodes are acoustically coupled. Every acoustic node is composed of a microphone, a processor, and an actuator to control the sound field. This type of networks can use distributed adaptive algorithms to deal with the active noise control (ANC) problem in a cooperative manner, providing more flexible and scalable ANC systems. In this regard, we introduce here a distributed version of the multichannel filtered-x AP algorithm over an acoustic sensor network that it is called distributed filtered-x AP (DFxAP) algorithm. The analysis of the mean and the mean-square deviation performance of the algorithm at each node is given for a network with a ring topology and without constraints in the communication layer. The theoretical results are validated through several simulations. Moreover, simulations show that the proposed DFxAP outperforms the previously reported distributed multiple error filtered-x least mean square algorithm.
- PublicationFuncion de Array del Filtro Optimo(Universitat Politècnica de València, 2008-05-12T06:46:43Z) Piñero Sipán, María Gemma; Escuela Técnica Superior de Ingeniería de Telecomunicación; Departamento de Comunicaciones; Instituto Universitario de Telecomunicación y Aplicaciones MultimediaLa primera figura muestra la Función de array o Beampattern de un beamformer basado en filtro óptimo. Utiliza para ello un array lineal de M antenas separadas media longitud de onda. La segunda figura muestra los símbolos QPSK a la salida del beamformer cuando la señal deseada llega por la dirección de llegada ¿theta_s¿ y la señal interferente llega por la dirección ¿theta_i¿. Se puede comprobar que (1) La Función de Array del filtro óptimo depende de "theta_s" y "theta_i", y (2) Cuando las señales tienen direcciones de llegada (DOAs) cercanas, la señal QPSK a la salida del beamformer se recibe en buenas condiciones.
- PublicationFast exact variable order affine projection algorithm(Elsevier, 2012-09) Ferrer Contreras, Miguel; González Salvador, Alberto; Diego Antón, María de; Piñero Sipán, María Gemma; Escuela Técnica Superior de Ingeniería de Telecomunicación; Departamento de Comunicaciones; Instituto Universitario de Telecomunicación y Aplicaciones Multimedia; Generalitat ValencianaVariable order affine projection algorithms have been recently presented to be used when not only the convergence speed of the algorithm has to be adjusted but also its computational cost and its final residual error. These kind of affine projection (AP) algorithms improve the standard AP algorithm performance at steady state by reducing the residual mean square error. Furthermore these algorithms optimize computational cost by dynamically adjusting their projection order to convergence speed requirements. The main cost of the standard AP algorithm is due to the matrix inversion that appears in the coefficient update equation. Most efforts to decrease the computational cost of these algorithms have focused on the optimization of this matrix inversion. This paper deals with optimization of the computational cost of variable order AP algorithms by recursive calculation of the inverse signal matrix. Thus, a fast exact variable order AP algorithm is proposed. Exact iterative expressions to calculate the inverse matrix when the algorithm projection order either increases or decreases are incorporated into a variable order AP algorithm leading to a reduced complexity implementation. The simulation results show the proposed algorithm performs similarly to the variable order AP algorithms and it has a lower computational complexity. © 2012 Elsevier B.V. All rights reserved.
- PublicationSteady-state mean square performance of the multichannel filtered-X affine projection algorithm(Institute of Electrical and Electronics Engineers (IEEE), 2012-06) Ferrer Contreras, Miguel; Diego Antón, María de; González Salvador, Alberto; Piñero Sipán, María Gemma; Escuela Técnica Superior de Ingeniería de Telecomunicación; Departamento de Comunicaciones; Instituto Universitario de Telecomunicación y Aplicaciones Multimedia; Generalitat Valenciana; Ministerio de Ciencia e Innovación; Universitat Politècnica de ValènciaThis paper provides a unified analysis of the steady-state behavior of different multichannel filtered-x affine projection algorithms when used in active noise control (ANC) systems. The analysis deals with two different filtering schemes: the modified filtered-x affine projection (MFXAP) algorithm, with the modified filtered-x structure embedded, that has been studied in previous works, and the filtered-x affine projection (CFXAP) algorithm, with the conventional filtered-x structure embedded, that becomes the main contribution of this work. This study is based on energy conservation principles and does not require a specific signal distribution. The derived theoretical models allow to accurately predict the steady-state performance of the considered algorithms for moderate AP orders and low step sizes mu. Simulation results obtained in practical ANC systems validate both the analysis and the achieved expressions, showing a relative good match between theory and practice. © 2012 IEEE.
- PublicationFiltros adaptativos: Algoritmo LMS(2023-06-14T09:58:01Z) Piñero Sipán, María Gemma; Escuela Técnica Superior de Ingeniería de Telecomunicación; Departamento de Comunicaciones; Instituto Universitario de Telecomunicación y Aplicaciones MultimediaAlgoritmo LMS utilizado en los filtros adaptativos. Obtiene los coeficientes de un filtro FIR (Finite Impulse Response) que actúa de forma similar a un filtro óptimo, pero basándose en las muestras de la señal.
- PublicationHigh performance lattice reduction on heterogeneous computing platform(Springer Verlag (Germany), 2014-11) Jozsa, Csaba M; Domene Oltra, Fernando; Vidal Maciá, Antonio Manuel; Piñero Sipán, María Gemma; González Salvador, Alberto; Escuela Técnica Superior de Ingeniería de Telecomunicación; Departamento de Comunicaciones; Instituto Universitario de Telecomunicación y Aplicaciones Multimedia; Generalitat Valenciana; Agencia Estatal de Investigación; Pázmány Péter Catholic UniversityThe lattice reduction (LR) technique has become very important in many engineering fields. However, its high complexity makes difficult its use in real-time applications, especially in applications that deal with large matrices. As a solution, the modified block LLL (MB-LLL) algorithm was introduced, where several levels of parallelism were exploited: (a) fine-grained parallelism was achieved through the cost-reduced all-swap LLL (CR-AS-LLL) algorithm introduced together with the MB-LLL by Jzsa et al. (Proceedings of the tenth international symposium on wireless communication systems, 2013) and (b) coarse-grained parallelism was achieved by applying the block-reduction concept presented by Wetzel (Algorithmic number theory. Springer, New York, pp 323-337, 1998). In this paper, we present the cost-reduced MB-LLL (CR-MB-LLL) algorithm, which allows to significantly reduce the computational complexity of the MB-LLL by allowing the relaxation of the first LLL condition while executing the LR of submatrices, resulting in the delay of the Gram-Schmidt coefficients update and by using less costly procedures during the boundary checks. The effects of complexity reduction and implementation details are analyzed and discussed for several architectures. A mapping of the CR-MB-LLL on a heterogeneous platform is proposed and it is compared with implementations running on a dynamic parallelism enabled GPU and a multi-core CPU. The mapping on the architecture proposed allows a dynamic scheduling of kernels where the overhead introduced is hidden by the use of several CUDA streams. Results show that the execution time of the CR-MB-LLL algorithm on the heterogeneous platform outperforms the multi-core CPU and it is more efficient than the CR-AS-LLL algorithm in case of large matrices.
- PublicationFIWARE based low-cost wireless acoustic sensor network for monitoring and classification of urban soundscape(Elsevier, 2021-09-04) Arce Vila, Pau; Salvo, David; Piñero Sipán, María Gemma; González Salvador, Alberto; Escuela Técnica Superior de Ingeniería de Telecomunicación; Departamento de Comunicaciones; Instituto Universitario de Telecomunicación y Aplicaciones Multimedia; GENERALITAT VALENCIANA; AGENCIA ESTATAL DE INVESTIGACION; European Regional Development Fund; COMISION DE LAS COMUNIDADES EUROPEA[EN] This work presents a wireless acoustic sensor network (WASN) that monitors urban environments by recognizing a given set of sound events or classes. The nodes of the WASN are Raspberry Pi devices that not only record the ambient sound, but also detect and recognize different sound events. All the signal processing tasks, from the recording to the classification carried out by a convolutional neural network (CNN), are run on Raspberry Pi devices. Due to the low cost of the proposed acoustic nodes, the system exhibits a very high potential scalability. Regarding the underlying WASN, it has been designed according to the open standard FIWARE, thus the whole system can be deployed without the need of proprietary software. Regarding the performance of the sound classifier, the proposed WASN achieves similar accuracy compared to other WASNs that make use of cloud computing. However, the proposed WASN significantly minimizes the network traffic since it does not exchange audio signals, but only contextual information in form of labels. On the other hand, most of the time the class reported by the WASN nodes is the "background'' soundscape, which usually contains no event of interest. This is the case when monitoring the soundscape of big avenues, where four events have been identified: "traffic'', "siren'', "horn'' and "noisy vehicles'', being the "traffic'' class associated to the background soundscape. In this paper, the use of a simple pre-detection stage prior to the CNN classification is proposed, with the aim of saving computation and power consumption at the nodes. The pre-detection stage is able to differentiate the other three relevant sounds from the "traffic'' and activates the classifier only when some of these three events is likely occurring. The proposed pre-detection stage has been validated through data recorded in the city of Valencia (Spain), achieving a reduction of the Raspberry Pi CPU's usage by a factor of six.
- PublicationParallel Implementation Strategies for MIMO ID-BICM Systems(2013) Simarro Haro, Mª de los Angeles; Ramiro Sánchez, Carla; Martínez Zaldívar, Francisco José; Vidal Maciá, Antonio Manuel; González Téllez, Alberto; Piñero Sipán, María Gemma; García Mollá, Víctor Manuel; Escuela Técnica Superior de Ingeniería de Telecomunicación; Departamento de Sistemas Informáticos y Computación; Departamento de Comunicaciones; Instituto Universitario de Telecomunicación y Aplicaciones Multimedia; Escuela Técnica Superior de Ingeniería Informática; Generalitat Valenciana; Ministerio de Ciencia e Innovación; Ministerio de Economía y Competitividad; Universitat Politècnica de València[EN] One of the current techniques proposed for multiple transmit and receive antennas wireless communication systems is the use of error control coding and iterative detection and decoding at the receiver. These sophisticated techniques produce a significant increase of the computational cost and require large computational power. The use of modern computer facilities as multicore and multi-GPU (Graphics Processing Unit) processors can decrease the computational time required, representing a promising solution for the receiver implementation in these systems. In this paper we explain how iterative receivers can improve the performance of suboptimal detectors. We also introduce a novel parallel receiver scheme based on a hybrid computing model where CPUs and GPUs work together to accelerate the detection and decoding steps; this design comes to exploit the features of the GPU NVIDIA Kepler architecture respect to the previous one in order to optimize the communication system performance.
- PublicationImproving speech intelligibility in hearing aids. Part I: Signal processing algorithms(Instituto de Telecomunicaciones y Aplicaciones Multimedia (ITEAM), 2014) Ayllón, D.; Gil Pita, Roberto; Rosa Zurera, Manuel; Padilla, L.; Piñero Sipán, María Gemma; Diego Antón, María de; Ferrer Contreras, Miguel; González Téllez, Alberto; Escuela Técnica Superior de Ingeniería de Telecomunicación; Departamento de Comunicaciones; Instituto Universitario de Telecomunicación y Aplicaciones Multimedia; Ministerio de Economía y Competitividad[EN] The improvement of speech intelligibility in hearing aids is a traditional problem that still remains open and unsolved. Modern devices may include signal processing algorithms to improve intelligibility: automatic gain control, automatic environmental classification or speech enhancement. However, the design of such algorithms is strongly restricted by some engineering constraints caused by the reduced dimensions of hearing aid devices. In this paper, we discuss the application of state-of-theart signal processing algorithms to improve speech intelligibility in digital hearing aids, with particular emphasis on speech enhancement algorithms. Different alternatives for both monaural and binaural speech enhancement have been considered, arguing whether they are suitable to be implemented in a commercial hearing aid or not.