Now showing items 1-20 of 137

  • A bit more on the ability of adaptation of speech signals 

    Ballesteros, Dora Maria; Moreno Aróstegui, Juan Manuel (2013-03)
    Article
    Open Access
    Some traditional digital signal processing techniques encompass enhancement, filtering, coding, compression, detection and recognition. Recently, it has been presented a new hypothesis of signal processing known as the ...
  • A comparative study of techniques for HMM-based noisy speech recognition in noisy car environment 

    Hernando Pericás, Francisco Javier; Nadeu Camprubí, Climent; Mariño Acebal, José Bernardo (Springer, 1993)
    Conference report
    Open Access
    The performance of existing speech recognition systems degrades rapidly in the presence of background noise when training and testing cannot be done under the same ambient conditions. The aim of this paper is to report the ...
  • A continuously adaptive vector predictive coder (AVPC) for speech encoding 

    Masgrau Gómez, Enrique José; Mariño Acebal, José Bernardo; Vallverdú Bayés, Sisco (Institute of Electrical and Electronics Engineers (IEEE), 1986)
    Conference report
    Open Access
    In this work we present a waveform speech coding system including vector quantization. This system can be seen as a vector version of the scalar ADPCM speech coder. In such system the speech samples are grouped in vectors ...
  • Adaptive prediction and bit-assignment in subband coding of speech 

    Mariño Acebal, José Bernardo; Martí Ros, Jaume (1985)
    Conference report
    Open Access
    The combination of time-domain harmonic scaling (TDHS) and sub-band coding (SBC) provides an encoding approach which allows 9.6 Kb/s speech encoding with good communication quality. Starting from this structure, this paper ...
  • Adaptive vector predictive speech coding with sample by sample update at 16 Kbps 

    Masgrau Gómez, Enrique José; Mariño Acebal, José Bernardo (1986)
    Conference report
    Open Access
    A vectorial generalization of the ADPCM system is introduced. Once the speech signal is grouped in vectors, they are coded using a vector predictor (VP) and a vector quantizer (VQ). Both subsystems are continously adaptive; ...
  • Almacenamiento en nodos de redes inalámbricas de sensores 

    Pérez Rodríguez, Iria (Universitat Politècnica de Catalunya, 2013-01-21)
    Master thesis (pre-Bologna period)
    Open Access
    [ANGLÈS] The aim of this project is to store data inside a Wireless Sensor Network node, in order to transmit it when the conditions are favourable, using a certain hardware and software stack. This necessity comes from ...
  • An HMM-Based Approach to the INTERSPEECH 2011 Speaker State Challenge 

    Nogueiras Rodríguez, Albino (2011)
    Conference lecture
    Restricted access - publisher's policy
    The current main trend in paralinguistic information recognition is the so-called static classification. In this kind of classification the low level descriptors are pooled togethr by means of statistical functionals ...
  • APVQ encoder applied to wideband speech coding 

    Salavedra Molí, Josep; Masgrau Gómez, Enrique José (Institute of Electrical and Electronics Engineers (IEEE), 1996)
    Conference report
    Open Access
    The paper describes a coding scheme for broadband speech (sampling frequency 16 KHz). The authors present a wideband speech encoder called APVQ (adaptive predictive vector quantization). It combines subband coding, vector ...
  • AR modeling of the speech autocorrelation to improve noisy speech recognition 

    Hernando Pericás, Francisco Javier; Nadeu Camprubí, Climent (1992)
    Conference report
    Open Access
    Speech recognition in noisy environments remains an unsolved problem even in the case of isolated word recognition with small vocabularies. Recently, several techniques have been proposed to alleviate this problem. Concretely, ...
  • A spectral estimator of vocal jitter 

    Mas Soro, Pol (Universitat Politècnica de Catalunya, 2011-09-09)
    Master thesis (pre-Bologna period)
    Open Access
    Covenantee:  Université libre de Bruxelles
    English: The purpose of this thesis is to study and implement a spectral method for short-time jitter estimation. Jitter consists in rapid perturbations of the vocal cycle lengths, which can be observed from one cycle to ...
  • A speech enhancement system using higher order ar estimation in real environments 

    Salavedra Molí, Josep; Masgrau Gómez, Enrique José; Moreno Bilbao, M. Asunción (1993)
    Conference report
    Open Access
    We study some speech enhancement algorithms based on the iterative Wiener filtering method due to Lim-Oppenheim [2], where the AR spectral estimation of the speech is carried out using a second-order analysis. But in our ...
  • Audio classification experiments in a neonatal intensive care unit 

    Sólvez Pérez, Sergi (Universitat Politècnica de Catalunya, 2014-06-25)
    Master thesis (pre-Bologna period)
    Open Access
    [ANGLÈS] Newborns delivered at a gestational age of 24-32 weeks commonly have health problems. The use of a Neonatal Intensive Care Unit (NICU) is, in most of the cases, crucial for their survival. Nowadays, it is known ...
  • Audiovisual head orientation estimation with particle filtering in multisensor scenarios 

    Canton Ferrer, Cristian; Segura Perales, Carlos; Casas Pla, Josep Ramon; Pardàs Feliu, Montse; Hernando Pericás, Francisco Javier (2008-01)
    Article
    Open Access
    This article presents a multimodal approach to head pose estimation of individuals in environments equipped with multiple cameras and microphones, such as SmartRooms or automatic video conferencing. Determining the individuals ...
  • Bandwidth extension of narrowband speech 

    Expósito Pérez, Miquel; Salavedra Molí, Josep (Universidad Politécnica de Valencia, 2014)
    Conference report
    Open Access
    Recently, 4G mobile phone systems have been designed to process wideband speech signals whose sampling frequency is 16 kHz. However, most part of mobile and classical phone network, and current 3G mobile phones, still ...
  • Bit-slice implementation of a linear predictive vocoder 

    Vázquez Grau, Gregorio; Gasull Llampallas, Antoni (1985)
    Conference report
    Open Access
    A digital 16-bit high-speed general-purpose signal-processor is shown. The main objective has been the implementation of a linear predictive vocoder for obtaining real-time speech compression. For real-time digital speech ...
  • Building synthetic voices in the META-NET framework 

    Garcia Casademont, Emília; Bonafonte Cávez, Antonio; Moreno Bilbao, M. Asunción (2012)
    Conference report
    Restricted access - publisher's policy
    METANET 4 U is a European project aiming at supporting language technology for European languages and multilingualism. It is a project in the META-NET Network of Excellence, a cluster of projects aiming at fostering the ...
  • CDHMM speaker recognition by means of frequency filtering of filter-bank energies 

    Hernando Pericás, Francisco Javier; Nadeu Camprubí, Climent (1997)
    Conference report
    Open Access
    Recently, the set of spectral parameters of every speech frame that result from filtering the frequency sequence of mel-scaled filter-bank energies with a simple first-order high-pass FIR filter have proved to be an efficient ...
  • Codificación APVQ de voz en banda ancha para velocidades entre 16 y 32 KBPS 

    Salavedra Molí, Josep; Masgrau Gómez, Enrique José (1996)
    Conference report
    Open Access
    This paper describes a coding scheme for broadband speech (sampling frequency 16KHz). We present a wideband speech encoder called APVQ (Adaptive Predictive Vector Quantization). It combines Subband Coding, Vector Quantization ...
  • Codificación APVQ de voz en banda ancha usando asignación dinámica de bits 

    Salavedra Molí, Josep (Universidad de Valladolid, 1995)
    Conference report
    Open Access
    This paper describes a coding scheme for broadband speech. It can be seen as a vectorial extension of a conventional ADPCM encoder. In this scheme, signal vector is formed with one sample of the normalized prediction error ...
  • Codificación APVQ-extendida de voz de banda ancha 

    Masgrau Gómez, Enrique José; Salavedra Molí, Josep (1994)
    Conference report
    Open Access
    This paper describes a coding scheme for broadband speech. It can be seen as a vectorial extension of an conventional ADPCM encoder. In this scheme, the vector signal is formed with one sample of the normalizaed prediction ...