Gernot Kubin

Nonlinear prediction of mobile radio channels: measurements and MARS model designs

The rapid time variation of mobile radio channels is often modeled as a random process with second order moments reflecting vehicle speed, bandwidth and the scattering environment. These statistics typically show that there is little room for prediction of channel properties such as received power or complex taps of the impulse response coefficients, at least when linear predictor structures are considered. We use mutual information estimation to measure statistical dependencies in sequences of wideband mobile radio channel data and find significant nonlinear dependencies, far exceeding the linear component. Based on these upper limits for the predictability of channel evolution over time intervals up to 30 ms ahead, we develop practical nonlinear predictor systems using multivariate adaptive regression splines (MARS). We demonstrate computationally efficient schemes that increase the prediction horizon beyond 10 ms, compared to less than 4 ms with linear predictors at comparable prediction gains.

On speech coding in a perceptual domain

For speech coders which fall within the class of waveform coders, the reconstructed signal approaches the original with increasing bit rate. In such coders, the distortion criterion generally operates on the speech signal or a signal obtained by adaptive linear filtering of the speech signal. To satisfy computational and delay constraints, the distortion criterion must be reduced to a very simple approximation of the auditory system. This drawback of conventional approaches motivates a new speech coding paradigm in which the coding is performed in a domain where the single-letter squared-error criterion forms an accurate representation of perception. The new paradigm requires a model of the auditory periphery which is accurate, can be be inverted with relatively low computational effort, and which represents the signal with relatively few parameters. We develop such a model of the auditory periphery and discuss its suitability for speech coding. The results indicate that the new paradigm in general and our auditory model in particular form a promising basis for the coding of both speech and audio at low bit rates.

A waveguide model for slapbass synthesis

Starting from the waveguide model for plucked strings, a new digital signal processing model for the slapping technique on electric bass guitars is derived. The model includes amplitude limitations for the string at the frets and/or the fingerboard. These highly nonlinear elements are realized by conditional reflections which depend on the local string displacement. A model of the string dynamics for the two slapbass techniques-knocking the string with the thumb knuckle and plucking very strong with the index or middle finger-has been implemented both as MATLAB and C simulations and synthesizes sounds close to the natural instrument.

Speaker verification security improvement by means of speech watermarking

This paper presents a security enhanced speaker verification system based on speech signal watermarking. Our proposed system can detect several situations where a playback speech, a synthetically generated speech, a manipulated speech signal or a hacker trying to imitate the speech is fooling the biometric system. In addition, we have generated a watermarked speech signals database from which we have obtained relevant conclusions about the influence of this technique on speaker verification rates. Mainly we have checked that biometrics and watermarking can coexist simultaneously minimizing the mutual effects. Experimental results show that the proposed speech watermarking system can suffer A-law coding with a message error rate lower than 2x10^-^4 for SWR higher than 20dB at a message rate of 48bits/s.

Speech Watermarking for Analog Flat-Fading Bandpass Channels

We present a blind speech watermarking algorithm that embeds the watermark data in the phase of non-voiced speech by replacing the excitation signal of an autoregressive speech signal representation. The watermark signal is embedded in a frequency subband, which facilitates robustness against bandpass filtering channels. We derive several sets of pulse shapes that prevent intersymbol interference and that allow the passband watermark signal to be created by simple filtering. A marker-based synchronization scheme robustly detects the location of the embedded watermark data without the occurrence of insertions or deletions. In light of a potential application to analog aeronautical voice radio communication, we present experimental results for embedding a watermark in narrowband speech at a bit-rate of 450 bit/s. The recursive least-squares (RLS) equalization-based watermark detector not only compensates for the vocal tract filtering, but also recovers the watermark data in the presence of nonlinear phase and bandpass filtering, amplitude modulation, and additive white Gaussian noise (AWGN), making the watermarking scheme highly robust.

Speaker identification security improvement by means of speech watermarking

This paper presents a security enhanced speaker identification system based on speech signal watermarking. Our proposed system can detect several situations where a playback speech, a synthetically generated speech, or a hacker trying to imitate the speech is fooling the biometric system. It is also suitable for forensic experts, who sometimes have to demonstrate in front of a court that a digital recording has neither been manipulated nor edited. In addition, we demonstrate that this watermark can coexist simultaneously with biometric speaker identification based on Gaussian mixture models (GMM), minimizing the mutual effects.

Poincaré pitch marks

A novel approach for pitch mark determination based on dynamical systems theory is presented. Pitch marks are used for speech analysis and modification, such as jitter measurement or time scale modification. The algorithm works in a pseudo-state space and calculates the Poincare section at a chosen point in the state space. Pitch marks are then found at the crossing of the trajectories with the Poincare plane of the initial point. The procedure is performed frame-wise to account for the changing dynamics of the speech production system. The system is intended for real-time use, so higher-level processing extending over more than one frame is not used. The processing delay is, therefore, limited to one frame. The algorithm is evaluated by calculating an average pitch value for 10ms frames and using a small database with pitch measurements from a laryngograph signal. The results are compared to a reference correlation-based pitch mark algorithm. The performance of the proposed algorithm is comparable to the reference algorithm, but in contrast correctly follows the pitch marks of diplophonic voices.

Spectral shaping of timing mismatches in time-interleaved analog-to-digital converters

We introduce spectral shaping of timing mismatches in time-interleaved analog-to-digital converters (TIADCs). We show how to reorder the channel ADC sequence in order to achieve spectral shaping and provide optimization criteria. We present a simple channel sorting algorithm, which significantly improves the signal-to-noise and distortion ratio (SINAD) and the spurious free dynamic range (SFDR) by applying a simple low-pass filter. The power consumption of the presented timing mismatch compensation method is very low compared to other methods and does not significantly scale with the number of channel ADCs. Furthermore, the compensation method does not depend on the absolute accuracy of the identified timing mismatches, which vastly relaxes the requirements on timing mismatch identification methods.

Relative information loss in the PCA

In this work we analyze principle component analysis (PCA) as a deterministic input-output system. We show that the relative information loss induced by reducing the dimensionality of the data after performing the PCA is the same as in dimensionality reduction without PCA. Furthermore, we analyze the case where the PCA uses the sample covariance matrix to compute the rotation. If the rotation matrix is not available at the output, we show that an infinite amount of information is lost. The relative information loss is shown to decrease with increasing sample size.

On Interfaces for Mobile Information Retrieval

We consider the task of retrieving online information in mobile environments. We propose question-answering as a more appropriate interface than page-browsing for small displays. We assess different modalities for communicating using a mobile device with question-answering systems, focusing on speech. We then survey existing research in spoken information retrieval, present some new findings, and assess the difficulty of the endeavor.