Signal Processing

This chapter has presented ultrasonic speech as a novel application of ultrasound in speech augmentation. Ultrasonic speech, operating by replacing the natural excitation in audible speech with an LF ultrasonic signal, has applications in speech augmentation for the speech rehabilitation and secure communications communities. This chapter has studied the requirements in modelling ultrasonic speech as a linear system of sound propagation and has proven that LPA, a major tool in the analysis of normal speech, is also extendible to ultrasonic speech. In pursuing this aim, we first introduced the attributes of ultrasonic propagation in a linear lossless gas medium. We observed that if the sound propagation is an adiabatic procedure and the gas obeys the ideal gas law and with small disturbances in the medium as a result of wave propagation, the gas medium can be considered a linear lossless medium for ultrasound propagation. We then discussed deviations of these conditions for ultrasound propagation in the air medium. Subsequently, LF ultrasound was introduced, and the impacts of the deviations of linear acoustic behaviour were numerically analyzed for propagation of low frequency ultrasound in the vocal tract. Then we considered the application of LF ultrasound in speech augmentation and discussed the aspects of system design which seek more attention. By a review of previous implementations, we investigated how they had addressed these aspects including the injection points and methods of down-conversion to audible domain. Afterwards we considered the physiology and anatomy of the human speech production mechanism and how we can substitute the natural excitation with an ultrasonic waveform in speech augmentation. We also stated that the ultrasonic excitation could be applied as a supplement to natural excitation to provide additional data for speech processing applications. The chapter then demonstrated a linear modelling scheme in addition to the fact that speech LPA tools can be extended to sound propagation at lower ultrasonic frequencies. Starting with basic wave equations, and making several simplifying assumptions such as rigid walls for closed glottis and VT, relatively small signal disturbance, and a spatially flat (uniform) excitation source, the VT has been shown to be LTI with the transfer function in the form of a pole-zero IIR filter. By means of this derivation, the conventional source-filter model was proven to be extendable for an ultrasonic speech production system, and thus the powerful tools of LPA can be used. In this chapter we have tried to bridge from audible speech processing methods to ultrasonics by mathematically and physically demonstrating that the extension of principles of audible speech processing to the analysis of ultrasonic speech is plausible. This significantly simplifies ultrasonic speech processing. The currently neglected area of LF ultrasonics research in speech analysis and processing can now be explored with relative ease. Further research effort is necessary, and welcomed in this area, as it moves toward further maturity and future real-life applications Preface......Page 5 Mariane R. Petraglia......Page 9 Sebastian Miron, Xijing Guo and David Brie......Page 27 Ramin Pichevar, Hossein Najaf-Zadeh, Louis Thibault and Hassan Lahdili......Page 45 Daniela Roşca and Jean-Pierre Antoine......Page 67 Faisal Darbari, Robert W. Stewart and Ian A. Glover......Page 85 Armando J. Pinho, António J. R. Neves, Daniel A. Martins, Carlos A. C. Bastos and Paulo J. S. G. Ferreira......Page 125 Ewa Skubalska-Rafajłowicz and Ewaryst Rafajłowicz......Page 139 Makoto Nakashizuka......Page 159 A. Suvichakorn, H. Ratiney, S. Cavassila, and J.-P Antoine......Page 175 Minoru Kuribayashi......Page 205 T. Trigano, U. Isserles, T. Montagu and Y. Ritov......Page 225 Simone Milani......Page 249 Hiroshi Ijima and Akira Ohsumi......Page 271 Alfonso Fernandez-Vazquez and Gordana Jovanovic Dolecek......Page 283 Hachem Kadri, Manuel Davy and Noureddine Ellouze......Page 315 Adriana Vasilache......Page 329 Rodrigo Torres, Eduardo Simas Filho, Danilo de Lima and José de Seixas......Page 345 Lorenzo Cappellari......Page 367 Nobutaka Ono and Shigeki Sagayama......Page 393 Hitoshi Kiya and Izumi Ito......Page 405 Robert M. Nickel, Tomohiro Sugimoto and Xiaoqiang Xiao......Page 423 António J. R. Neves and Armando J. Pinho......Page 437 Takao Hinamoto, Keijiro Kawai, Masayoshi Nakamoto andWu-Sheng Lu......Page 457 Rodolphe J. Gentili, Hyuk Oh, Trent J. Bradberry, Bradley D. Hatfield and José L. Contreras-Vidal......Page 469 Farzaneh Ahmadi and Ian Mcloughlin......Page 511 In this paper we introduced a novel algorithm for DOA estimation for polarized sources, based on a four-way PARAFAC representation of the data covariance. A quadrilinear alternated least squares procedure is used to estimate the steering vectors and the polarization vectors of the sources. Compared to MUSIC for polarized sources, the proposed algorithm ensures the mixture model identifiability; thus it avoids the exhaustive grid search over the parameters space, typical to eigestructure algorithms. An upper bound on the minimum number of sensors needed to ensure the identifiability of the mixture model is derived. Given the symmetric structure of the data covariance, our algorithm presents a smaller complexity per iteration compared to three-way PARAFAC applied directly on the raw data. In terms of estimation, the proposed algorithm presents slightly better performance than MUSIC and ESPRIT, thanks to its higher dimensionality and it clearly outperforms the three-way algorithm when the number of temporal samples becomes important. The variance of our algorithm decreases with an increase in the sample size while for the three-way method it tends asymptotically to a constant value