Francisco Javier Casajús-Quirós

• Universidad Politécnica de Madrid

One may identify two independent sources of nonlinearities in digital-to-analog converters: (i) deviations at input sources and switches and (ii) nonidealities in the posterior summation circuit. The first are described by specific figures of merit, mainly integral nonlinearity and differential nonlinearity; the second, in terms of standard amplifi...

We describe how two different techniques: ultrasonic and ultrawide-band radio frequency inspections, can be used and combined to detect embedded structures in concrete. Joint analysis shall overcome limitations of the individual technologies, while providing further information on specimens. In this case, ultrasonic inspection achieves good spatial...

MLS-based identification of nonlinear systems is largely affected by deviations in the excitation signal amenable to the combined effect of DC-offset and an arbitrary gain. These induce orthogonality loss in the MLS filter bank output, thus invalidating the underlying identification construction. In this paper we present a correction algorithm to d...

A system for anchor-less self-positioning and simultaneous 2D estimation of rectangular room dimensions is proposed. The system is based on narrowband transceiver capable of simultaneous transmission and reception. The transmission is generated at omnidirectional antenna, while eight sectorized antennas with 45deg apertures in horizontal are used f...

A system for estimation of unknown rectangular room dimensions based on two radio transceivers, both capable of full duplex operations, is presented. The approach is based on CIR measurements taken at the same place where the signal is transmitted (generated), commonly known as self-to-self CIR. Another novelty is the receiver antenna design which...

A system for simultaneous 2D estimation of rectangular room and transceiver localization is proposed. The system is based on two radio transceivers, both capable of full duplex operations (simultaneous transmission and reception). This property enables measurements of channel impulse response (CIR) at the same place the signal is transmitted (gener...

A software tool, Localization of Sound Events (LSE), is presented which mimics the behavior of the auditory system for sound localization. This objective localization is accomplished by measuring a binaural signal in the horizontal plane and a monaural signal in the median plane. The LSE tool can simulate acoustic configurations such as a virtual a...

An instrumental method based on parametrically described spectral cues is proposed to estimate the direction of auditory events in the median plane. In contrast to a previous method where the shapes of two spectra are compared, the current method is based on known psychoacoustic features of the auditory system. The algorithm is described and the re...

Security in sensor networks is one of the most relevant research topics in resource constrained wireless devices and networks. Several attacks can be suffered in ad hoc and wireless sensor networks (WSN), which are highly susceptible to attacks, due to the limited resources of the nodes. In this paper, we propose innovative and lightweight localiza...

In this study, the performance of two asynchronous multi-access direct sequence code division multiple access (DS-CDMA) and multicarrier code division multiple access (MC-CDMA) systems with binary and polyphase long sequences is compared. An indoor multipath powerline channel model generated from random impedance networks is used. The powerline bac...

Security is a major concern in wireless sensor networks deployed in hostile environments. If the physical locations of the nodes are known, such information can be used as an aid in the detection of potential attacks and identification of intruders. The purpose of this paper is twofold: first, we survey some defensive techniques for sensor networks...

This paper presents an overview on the performance of hybrid data fusion and tracking algorithms evaluated in the WHERE consortium. The focus is on small scale indoor scenarios with ultra wideband (UWB) complementing cellular communication systems, mid scale indoor scenarios with Wi-Fi aiding cellular communication systems, and large scale outdoor...

This paper presents an overview on the performance of hybrid data fusion and tracking algorithms evaluated in the WHERE consortium. The focus is on small scale indoor scenarios with ultra wideband (UWB) complementing cellular communication systems, mid scale indoor scenarios with Wi-Fi aiding cellular communication systems, and large scale outdoor...

This paper describes the Software Defined Radio (SDR) platform that is being developed and implemented in the scope of the EU seventh Framework WHERE Project. The platform is based on one full-length PCI carrier board populated with one DSP module, one dual high-speed ADC/DAC module and one module corresponding to IF/RF front-end. It is equipped wi...

This deliverable presents a first performance evaluation of hybrid data fusion and tracking algorithms. The focus is on small scale indoor scenarios with ultra wideband (UWB) complementing cellular communication systems, mid scale indoor scenario with WIFI aiding cellular communication systems, and large scale outdoor scenario with global navigatio...

Pattern-matching methods for polyphonic transcription of piano sounds require a set of patterns that can be obtained by modeling the piano-sound spectra. The modeling should take into account not only the string stiffness but also the effect of the soundboard impedance on the string vibration. Studies on that effect corresponding to a wide range of...

A set of subjective experiments is presented, which were designed to determine the intelligibility of virtual acoustic opening environments when these environments are built around a source separation coding approach. The idea of virtual acoustic opening-based applications is to connect two distant rooms acoustically as if there were a real physica...

An adaptive receiver solution for an asynchronous DS-CDMA system in an indoor powerline network is proposed. A combination of different receiver structures with binary and complex-polyphase sequences is evaluated and compared in terms of bit error rate. The simulation results based on a complete powerline channel model are reported.

Future 4th Generation (4G) wireless multiuser communication systems will have to provide advanced multimedia services to an increasing number of users, making good use of the scarce spectrum resources. Thus, 4G system design should pursue both higher-transmission bit rates and higher spectral efficiencies. To achieve this goal, multiple antenna sys...

The reproduced sound event quality is very important in a WFS configuration that is used for an acoustic opening. One way of checking the subjective quality perceived by a listener is the ITU R. BS.1387 "Method for objective measurements of perceived audio quality", but this method does not provide information about the listener ability to localize...

A previous method presented by the authors carried out multi-pitch piano sound identification by using a pattern-matching process. In that method, the identification required, besides the matching-metric calculation, both a spectral predetection process and a validation step. Predetection allowed to select a subset out of the eighty-eight patterns,...

Polyphonic transcription of piano is a challenging task due to the specific characteristics of its sound spectrum. Pattern matching method, which compares the spectrum with a set of spectral patterns, has proven to give good results, although some limitations still exist mainly when analyzing notes of the lower octaves. Present research is oriented...

In order to achieve higher bit rates and gain diversity, recent trends in communication industry include the use of multiple antennas in transmitter and at the receiver. At the same time the terminals are getting smaller incrementing the level of electronic interferences in their interior. In this contribution, the signal distortion known as RF fro...

One of the recent trends in mobile communication industry is the use of multiple antennas in transmitter and at the receiver in order to gain diversity and achieve high bit rates. At the same time, as the mobile terminals are getting smaller, the number of electronic elements squeezed inside of them increases. Inside this hostile environment the co...

There is no doubt that powerline communication (PLC) networks are growing around home environments, to which we demand more reliability and data rate transmission speed. In order to design a powerline system it is necessary to understand and characterize the powerline network behavior through mathematical models. This would allow the transmitter /...

In this paper we address the implementation in FPGAs of a multiple-input multiple-output (MIMO) decoder embedded in a prototype of a 4G mobile receiver. This MIMO decoder is part of a multi-carrier code-division multiple-access (MC-CDMA) radio system, equipped with multiple antennas at both ends of the link, that is able to handle up to 32 users an...

In this paper we address the implementation on FPGAs of a 4G equaliser for a multiple-input multiple-output (MIMO) receiver. It is embedded in a multi-carrier code-division multiple-access (MC-CDMA) radio system, which is able to handle up to 32 users and provide transmission bit-rates up to 125 Mbps. We provide details on decisions taken for the d...

Multichannel acoustic cancellation problem requires work- ing with extremely large impulse responses. Multirate adap- tive schemes such as the partitioned block frequency-domain adaptive filter (PBFDAF) are good alternatives and are widely used in commercial echo cancellation systems nowa- days. However, being a Least Mean Square (LMS) derived algo...

Multichannel acoustic cancellation problem requires working with extremely large impulse responses. Multirate adaptive schemes such as the partitioned block frequency-domain adaptive filter (PBFDAF) are good alternatives and are widely used in commercial echo cancellation systems nowadays. However, being a Least Mean Square (LMS) derived algorithm,...

Partials of piano sounds are inharmonic. This inharmonicity is due either to string stiffness and to soundboard impedance. The last has not been widely documented. Two problems arise, to know the value of the impedance and to evaluate the frequency deviation the partial suffers. In this work, that deviation has been calculated either by using Morse...

The adaptive solution of a multichannel system is dependent of the input signal's correlation. An ill-conditioned system can converge to infinite solutions. However, in many applications, since the multichannel acoustic echo cancellation, the nearness between the sensors of the input signals to the system provokes this correlation. This disadvantag...

Sound reproduction is evolving towards multi-channel systems with a growing number of channels. Consequently, high quality multi-channel codecs are required. Last generation perceptual audio codecs, represented by MPEG advanced audio coder (AAC), can efficiently code typical surround multi-channel material but may benefit from a previous block of i...

The success of polyphonic detection varies depending on the instrument. Different approaches have been proposed. Piano polyphonic detection is particularly difficult. A special approach to detecting piano chords is described. Using a spectral pattern-matching method, chords containing up to four notes have been identified successfully. Chords throu...

Continuous phase modulation schemes, such as Gaussian minimum-shift keying (GMSK), are frequently used with limiter-discriminator (LD) detectors. This paper studies how the side information derived from the signal envelope can enhance the performance of a Viterbi algorithm (VA)-based receiver operating on the LD output of a GMSK scheme. By consider...

Automatic identification of chords is being investigated and several methods has been proved. A method using spectral pattern recognition has been presented previously by the authors. In that method, the chord is detected iteratively, note by note, by means of a set of patterns generated by an acoustical model of the piano. In each iteration step,...

Conventional decision feedback equalisation (DFE) is a common solution to the frequency selective multipath propagation problem that appears in high data rate radio communications because of its simplicity and good performance. However, when a limiter-discriminator (LD) is used for detection of GMSK signals, the global system has a nonlinear charac...

This paper describes the first version of a Java archive (a term basically equivalent to ‘library’ in other programming languages) that has been developed and made available as public domain software for the benefit of the DAFX community, and the COSTG6 web pages in particular. The library is available both as source code and ready to run bytecodes...

The separation of musical instruments acoustically mixed in one source is a very active field which has been approached from many different viewpoints. This article compares the blind source separation perspective and oscillatory correlation theory taking the auditory scene analysis as the point of departure (ASA). The former technique deals with t...

New formats for digital audio with enhanced resolution and sampling frequency make the limits of traditional approaches to audio effects noticeable. The processes applied during audio production and postproduction for media can be advanced to a higher level of quality thanks to a multiband approach to Digital Audio FX. Also completely new effects c...

The new version of the Digital Enhanced Cordless Telecommunication (DECT) standard contemplates the possibility of transmitting 64 kbps over the time division multiple access structure of the previous DECT standard. It is important to realise that transmission takes place by means of the single slots used today by most existing DECT systems for voi...

The conventional decision feedback equalizer (DFE) is a common solution to the frequency selective multipath propagation problem that appears in high data rate radio communications because of its simplicity and good performance. However, when limiter-discriminator (LD) is used for detection of GMSK signals, the global system has a nonlinear charact...

Maximum likelihood sequence detection (MLSD) using the Viterbi algorithm (VA) is a widely employed detection technique for both coherent and non-coherent detection of continuous phase modulation (CPM) signals due to the inherently nonlinear behaviour of the digital channel. In particular, its application to limiter-discriminator (LD) demodulation s...

A simplification of the maximum likelihood sequence detector (MLSD) algorithm with zero decision delay is presented for GMSK receivers based on limiter discriminator in a frequency selective fading environment. Good performance is obtained when the detector employs side envelope information

Due to the nonlinear characteristics of the signal at the output of the discriminator device, conventional equalizers are not very effective even under low delay spread propagation. We show that, due to the relation of the envelope information with the frequency noise and its dependence on the transmitted data when multipath transmission occurs, th...

The separation of musical instruments acoustically mixed in one source is a very active field which has been approached from many different viewpoints. This article compares the blind source separation perspective and oscillatory correlation theory taking the auditory scene analysis as the point of departure (ASA). The former technique deals with t...

In a digital FM transmitter-receiver system for data communications, demodulation can be carried out by a simple and cheap non-coherent demodulator scheme consisting of a hard-limiter block plus an FM demodulator (discriminator, FD), which is simple and insensitive to modulation index changes. Its application field includes partial response continu...

In this contribution, variable rate coding techniques are proposed to improve the speech quality of the DECT system in poor channel conditions. The basic idea is to estimate in advance the quality of the channel, and subsequently select the best source-channel rate combination. The channel quality is predicted from the characteristics of the signal...

An equalization-predistortion scheme for DECT applications in low time dispersive channels is described. It is intended for transmitter that use VCOs and receivers based on limiter-discriminators, therefore the digital channel is non-linear. This concentrates all the complexity of equalization at one end of the transmission (the base station). It h...

A new model for the spectral samples obtained in the multiband excitation speech coder (MBE) is introduced. Objective and subjective tests show that it compares favorably with the classical linear prediction (LP) model, specially for high pitched speakers. Strategies for efficiently quantizing the model parameters, suitable for low bit rate impleme...

Variable rate coding techniques are proposed to improve the speech quality of the DECT system in poor channel conditions. The basic idea is to estimate in advance the quality of the channel, and subsequently select the best source-channel rate combination. The channel quality is predicted from the characteristics of the signal received in previous...

Automatic score transcription goal is to achieve an score-like (notes pitches through time) representation from musical signals. Reliable pitch extraction methods for monophonic signals exist, but polyphonic signals are much more difficult, often ambiguous, to analyze. We propose a computationally efficient technique for automatic recognition of no...

A new phase coding algorithm working in the pitch-cycle waveform domain is introduced. It provides accurate phase coding at low bit cost, thus being suitable for low bit rate sinusoidal coders. Its performance is analysed inside a multiband excitation (MBE) coder with improved onset representation. In this context, the introduction of original phas...

Band-pass equalization for the Digital Enhanced Cordless Telecommunications (DECT) is described. Assuming the GFSK DECT signal is sampled at the Nyquist rate, 3.456 Msps, practical methods for the estimation of the modulation parameters are developed that use only the first 32 bits of the DECT slot. Total complexity can be assumed by modern signal...

We present a receiver capable of capturing two RF channels at the same time with a single RF front end and single IF stage. The idea is to use the lowest possible IF plus a digital image rejection unit that can separate two adjacent RF channels with a negligible cochannel's image interference. We analyze two procedures of compensating, in the IF ra...

An algorithm capable of reliably estimating the fundamental frequency of sound signals is described. Pitch estimation is carried out by means of a new loose harmonic matching algorithm which searches the spectrum of a sound for evenly-spaced peak patterns. This unaided algorithm is by itself capable of correctly estimating the fundamental frequency...

The multiband excitation coder (MBE) has been shown to be a suitable speech coding scheme for bit rates ranging from 2400 bps to 8000 bps. However the speech it produces is sometimes rough with a reverberating quality and its performance seems to be speaker dependent. This somewhat reduced quality (specially at low bit rates) is the main drawback o...

The authors describe the main topics related to a real-time implementation of the Proposed Federal Standard 1016 (PFS-1016) 4800 bps CELP (code excited linear prediction) voice coder together with some alternatives to extend real-time CELP schemes over a wide range of bit rates (8000, 6500, and 2400 bps are considered). Several procedures are evalu...

A speech coder is presented which combines vector excitation coding (VXC) with frequency-domain representations so as to obtain a high-quality efficient scheme at 4.8 kb/s based on the use of different coding strategies and bit allocations for different frames of speech and different frequency bands. Attention is focused on the representation of vo...

The main results from work on analysis techniques and quantization procedures for real-time implementation of the Proposed Federal Standard 1016 4800-b/s voice coder developed by the US DoD and AT&T Bell Laboratories are presented. Reduced search procedures are described for the adaptive and stochastic codebooks, two basic functions in code-excited...

The authors discuss various possible synthesis procedures for frequency-domain vector-excited coding (VXC). The algorithm, called vector adaptive transform coding (VATC), is a two-stage vector quantizer for the short-time Fourier transform (STFT) of speech. One stage represented by an adaptive codebook based on long-term prediction and the other ba...

A frequency-domain search procedure is presented that requires a computational load of approximately 1.5 MFLOPS, reduces the storage cost, and maintains the speech quality. A vector-excitation-coding (VXC) method in the transform domain called vector adaptive transform coding (VATC) is proposed. VATC provides a general framework for VXC in the tran...

Both multipulse and stochastic coders render telephonic, or close to telephonic quality, for speech signals. Stochastic coders are more efficient from the point of view of bit rate and multipulse coders are the most efficient ones if we are looking for low computational cost. In this paper we study some possibilities of coding speech using the mult...

In this paper, we propose a simplification of the model introduced by B.S. Atal and M.R. Schroeder for a code-excited linear predictor. Our scheme is based on a two-step search of the optimum code-word. In the first step, a sub-set in the code-book is selected by using a simple distance measure and taking into account the selections already made fo...

The bit rates of existing speech coders can be scaled down through the use of time-compression of speech signals. Of course this implies a small increase in coder complexity and some loss of quality. This work focuses on the use of dynamic time-compression in order to reduce the loss of quality. Long, stationary segments do exist in speech, therefo...

This paper presents a new adaptive algorithm for the long term stable operation of a digitally implemented fractionally-spaced adaptive equalizer. We show how previously reported solutions may become useless working with a too small word size. Then, we propose an alternative based on performing an equalizer tap weight normalization after each new a...

This paper presents a new model to generate simulated daily global solar radiation (DGSR) sequences. A DGSR value is the product of two factors: a seasonal low-frequency component, principally due to the sun's periodic movement, and a random component due to rapid fluctuations of the atmospheric environment. Methods are provided to automatically se...

Chorus and flanger are two classical effects that can be found in any current digital audio FX unit. Available chorus/flanger just translate into digital implementations the behaviour and parameters typical of cost-sensitive analogue realizations. But they can be greatly improved on a digital implementation with simple additions to the software. Af...

We analyze the requirements for digital signal processors used in audio applications. Methods for assessing the real computing power expected from them are presented. Proper requirements for a real-time development platform are also described. Code development tools are also included. Finally a review of common processors is done according to those...

A new architecture for musical distortion is proposed. Based on WaveShaping as the distortion generation element, a multiband front-end is used in order to extract simple (ideally monotonal) non-full band signals. Distortion pattern can be adjusted per band, with the benefit that intermodulation distortion is kept low, balancing the end result towa...

A piano sound database, called PianoUPM, is presented. It is intended to help the researching community in developing and testing transcription methods. A practical database needs to contain notes and chords played through the full piano range and it needs to be recorded from acoustic pianos rather than synthesized ones. The presented piano sound d...

This paper proposes a new scheme to develop efficient codification systems for future teleconference systems. Source separation approach along with modern Wave Field Synthesis (WFS) techniques to recreate a mechanical acoustic opening may provide us with a very useful tool to drastically reduce the number of channels to transmit. A multichannel, su...