Efficient self-synchronised blind audio watermarking system based on time domain and FFT amplitude modification

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.

Estudi de casos pel control de síntesi d’instruments musicals virtuals mitjançant la veu cantada

Report for the scientific sojourn carried out at the Music Technology Area (Sound Processing and Control Lab), Faculty of Music, McGill University, Montreal, Canada, from October to December 2005.The aim of this research is to study the singing voice for controlling virtual musical instrument synthesis. It includes analysis and synthesis algorithms based on spectral audio processing. After digitalising the acoustic voice signal in the computer, a number of expressive descriptors of the singer are extracted. This process is achieved synchronously, thus all the nuance of the singer performance have been tracked. In a second stage, the extracted parameters are mapped to a sound synthesizer, the so-called digital musical instruments. In order achieve it, several tests with music students of the Faculty of Music, McGill University have been developed. These experiments have contributed to evaluate the system and to derive new control strategies to integrate: clarinet synthesis, bass guitar, visual representation of voice signals.

Disseny d'un sistema de filtrat d'àudio

Tant el medi transmissor com els equips d'enregistrament o reproducció de so introdueixen components de soroll d'alta freqüència als senyals. En aquest treball de final de carrera (TFC), s'ha dissenyat i implementat un sistema de filtrat d'àudio encaminat a filtrar aquestes components d'alta freqüència. Donat que l'oïda humana no pot percebre sons de més de 20 kHz, s'ha considerat aquest límit com a freqüència màxima a mantenir en la senyal.S'ha començat estudiant el senyal problema a través del seu espectre de freqüències simulat mitjançant la transformada discreta de Fourier (DFT, en anglès). Una vegada identificades les components d'alta freqüència a atenuar, s'han estudiat les diferents opcions de filtre passabaix.Inicialment, s'ha valorat la possibilitat del disseny de filtres analògics de Butterworth o Chebyshev, o de filtres digitals de tipus IIR (Infinite Impulse Response) basats en els primers. Tanmateix, malgrat assolir les especificacions en magnitud, mitjançant aquest filtres no s'obté una fase lineal en la banda de pas. Per això, s'ha realitzat un disseny de filtre digital tipus FIR (Finite Infinite Response) que compleix estrictament amb les especificacions i presenta una fase lineal en la banda de pas. S'ha simulat el comportament d'aquest filtre amb el senyal problema per tal d'assegurar el seu correcte funcionament.A continuació, s'ha implementat aquest últim disseny en llenguatge C i compilat per un microcontrolador de l'empresa Microchip. S'han realitzat proves de simulació mitjançant Stimulus del programa MPLAB. En definitiva, s'ha dissenyat un filtre passabaix de tipus FIR per acondicionar una senyal d'àudio que posteriorment s'ha implementat en un microcontrolador de Microchip.

Audio as medium for content distribution (providing access to knowledge)

Reaching and educating the masses to the benefit of all of mankind is the ultimate goal and through the use of this technology facility/tool many can be reached in their own language, in their own community, in their own time and at their own pace. Making this content available to those who will benefit from the information, is vital. These people who want to consume the content are not necessarily that interested in the qualification, they need the information. Making the content available in an auditory format may also help those who may not be as literate as others. The uses of audio/ recorded lessons have a number of uses and should not just be seen as a medium for content distribution to distant communities. Recording lectures makes it possible for a lecturer to present lectures to a vast number of students, while just presenting the lecture once.

ACCEDA y BUSTREAMING : Integración de audio y video en el proceso de autoarchivo del Repositorio de la ULPGC en acceso abierto

Presentació de les dues noves plataformes desenvolupades per la Biblioteca de la Universitat de Las Palmas de Gran Canaria: ACCEDA (documentació científica de la ULPGC en accés obert) i BUSTREAMING (documentació multimèdia). Aquestes són dues noves iniciatives de la Biblioteca Universitària emmarcades dins del moviment Open Access, que se sumen a la ja existent Memòria Digital de Canàries, que es va iniciar l'any 2003 amb molt d'èxit. Tot i que el programari utilitzat per al desenvolupament d'ACCEDA és DSpace, la Biblioteca Universitària ha implementat serveis als usuaris i administradors, com per exemple, la possibilitat de dipositar documents sense haver d'utilitzar l'autoarxivament i la incorporació de l'eina BUStreaming desenvolupada per la pròpia Biblioteca.

Non-peaky signals in wideband fading channels: achievable bit rates and optimal bandwidth

In the context of fading channels it is well established that, with a constrained transmit power, the bit rates achievable by signals that are not peaky vanish as the bandwidth grows without bound. Stepping back from the limit, we characterize the highest bit rate achievable by such non-peaky signals and the approximate bandwidth where that apex occurs. As it turns out, the gap between the highest rate achievable without peakedness and the infinite-bandwidth capacity (with unconstrained peakedness) is small for virtually all settings of interest to wireless communications. Thus, although strictly achieving capacity in wideband fading channels does require signal peakedness, bit rates not far from capacity can be achieved with conventional signaling formats that do not exhibit the serious practical drawbacks associated with peakedness. In addition, we show that the asymptotic decay of bit rate in the absence of peakedness usually takes hold at bandwidths so large that wideband fading models are called into question. Rather, ultrawideband models ought to be used.

Cross recurrence quantification for cover song identification

There is growing evidence that nonlinear time series analysis techniques can be used to successfully characterize, classify, or process signals derived from realworld dynamics even though these are not necessarily deterministic and stationary. In the present study we proceed in this direction by addressing an important problem our modern society is facing, the automatic classification of digital information. In particular, we address the automatic identification of cover songs, i.e. alternative renditions of a previously recorded musical piece. For this purpose we here propose a recurrence quantification analysis measure that allows tracking potentially curved and disrupted traces in cross recurrence plots. We apply this measure to cross recurrence plots constructed from the state space representation of musical descriptor time series extracted from the raw audio signal. We show that our method identifies cover songs with a higher accuracy as compared to previously published techniques. Beyond the particular application proposed here, we discuss how our approach can be useful for the characterization of a variety of signals from different scientific disciplines. We study coupled Rössler dynamics with stochastically modulated mean frequencies as one concrete example to illustrate this point.

Mutual information of IID complex Gaussian signals on block Rayleigh-faded channels

We present a method to compute, quickly and efficiently, the mutual information achieved by an IID (independent identically distributed) complex Gaussian signal on a block Rayleigh-faded channel without side information at the receiver. The method accommodates both scalar and MIMO (multiple-input multiple-output) settings. Operationally, this mutual information represents the highest spectral efficiency that can be attained using Gaussiancodebooks. Examples are provided that illustrate the loss in spectral efficiency caused by fast fading and how that loss is amplified when multiple transmit antennas are used. These examples are further enriched by comparisons with the channel capacity under perfect channel-state information at the receiver, and with the spectral efficiency attained by pilot-based transmission.

Mutual Information of IID complex Gaussian signals on block Rayleigh-faded channels

We present a method to compute, quickly and efficiently, the mutual information achieved by an IID (independent identically distributed) complex Gaussian signal on a block Rayleigh-faded channel without side information at the receiver. The method accommodates both scalar and MIMO (multiple-input multiple-output) settings. Operationally, this mutual information represents the highest spectral efficiency that can be attained using Gaussiancodebooks. Examples are provided that illustrate the loss in spectral efficiency caused by fast fading and how that loss is amplified when multiple transmit antennas are used. These examples are further enriched by comparisons with the channel capacity under perfect channel-state information at the receiver, and with the spectral efficiency attained by pilot-based transmission.

A Musically aware system for browsing and interacting with audio music collections

In the context of the CompMusic project we are developing methods to automatically describe/annotate audio music recordings pertaining to various music cultures. As away to demonstrate the usefulness of the methods we are also developing a system to browse and interact with specific audio collections. The system is an online web application that interfaces with all the data gathered (audio, scores plus contextual information) and all the descriptions that are automatically generated with the developed methods. In this paper we present the basic architecture of the proposed system, the types of data sources that it includes,and we mention some of the culture specific issues that we are working on for its development. The system is in a preliminary stage but it shows the potential that MIR technologies can have in browsing and interacting with musiccollections of various cultures.

An approach for linking score and audio recordings in Makam music of Turkey

The main information sources to study a particular piece of music are symbolic scores and audio recordings. These are complementary representations of the piece and it isvery useful to have a proper linking between the two of the musically meaningful events. For the case of makam music of Turkey, linking the available scores with the correspondingaudio recordings requires taking the specificities of this music into account, such as the particular tunings, the extensive usage of non-notated expressive elements, and the way in which the performer repeats fragmentsof the score. Moreover, for most of the pieces of the classical repertoire, there is no score written by the original composer. In this paper, we propose a methodology to pair sections of a score to the corresponding fragments of audio recording performances. The pitch information obtained from both sources is used as the common representationto be paired. From an audio recording, fundamental frequency estimation and tuning analysis is done to compute a pitch contour. From the corresponding score, symbolic note names and durations are converted to a syntheticpitch contour. Then, a linking operation is performed between these pitch contours in order to find the best correspondences.The method is tested on a dataset of 11 compositions spanning 44 audio recordings, which are mostly monophonic. An F3-score of 82% and 89% are obtained with automatic and semi-automatic karar detection respectively,showing that the methodology may give us a needed tool for further computational tasks such as form analysis, audio-score alignment and makam recognition.

Activation of Srk1 by the Mitogen-activated Protein Kinase Sty1/Spc1 Precedes Its Dissociation from the Kinase and Signals Its Degradation

Control of cell cycle progression by stress-activated protein kinases (SAPKs) is essential for cell adaptation to extracellular stimuli. The Schizosaccharomyces pombe SAPK Sty1/Spc1 orchestrates general changes in gene expression in response to diverse forms of cytotoxic stress. Here we show that Sty1/Spc1 is bound to its target, the Srk1 kinase, when the signaling pathway is inactive. In response to stress, Sty1/Spc1 phosphorylates Srk1 at threonine 463 of the regulatory domain, inducing both activation of Srk1 kinase, which negatively regulates cell cycle progression by inhibiting Cdc25, and dissociation of Srk1 from the SAPK, which leads to Srk1 degradation by the proteasome.

Continuous phase shift of sinusoidal signals using injection locked oscillators

This work presents an alternative to generate continuous phase shift of sinusoidal signals based on the use of super harmonic injection locked oscillators (ILO). The proposed circuit is a second harmonic ILO with varactor diodes as tuning elements. In the locking state, by changing the varactor bias, a phase shift instead of a frequency shift is observed at the oscillator output. By combining two of these circuits, relative phases up to 90 could be achieved. Two prototypes of the circuit have been implemented and tested, a hybrid version working in the range of 200-300 MHz and a multichip module (MCM) version covering the 900¿1000 MHz band.

Nonlinear relaxation time and the detection of weak signals

Laser systems can be used to detect very weak optical signals. The physical mechanism is the dynamical process of the relaxation of a laser from an unstable state to a steady stable state. We present an analysis of this process based on the study of the nonlinear relaxation time. Our analytical results are compared with numerical integration of the stochastic differential equations that model this process.