Programming is fast becoming a required skill set for students in every country. We present CS Bridge, a model for cross-border co-teaching of CS1, along with a corresponding open-source course-in-a-box curriculum made for easy localization. In the CS Bridge model, instructors and student-teachers from different countries come together to teach a short, stand-alone CS1 course to hundreds of local high school students. The corresponding open-source curriculum has been specifically designed to be easily adapted to a wide variety of local teaching practices, languages, and cultures.

In this paper, we suggest a novel group delay based method for the onset detection of pitched instruments. It is proposed to approach the problem of onset detection by examining three dimensions separately: phase (i.e., group delay), magnitude and pitch. The evaluation of the suggested onset detectors for phase, pitch and magnitude is performed using a new publicly available and fully onset annotated database of monophonic recordings which is balanced in terms of included instruments and onset samples per instrument, while it contains different performance styles. Results show that the accuracy of onset detection depends on the type of instruments as well as on the style of performance. Combining the information contained in the three dimensions by means of a fusion at decision level leads to an improvement of onset detection by about 8% in terms of F-measure, compared to the best single dimension.

Homomorphic analysis is a well-known method for the separation of non-linearly combined signals. More particularly, the use of complex cepstrum for source-tract deconvolution has been discussed in various articles. However there exists no study which proposes a glottal flow estimation methodology based on cepstrum and reports effective results. In this paper, we show that complex cepstrum can be effectively used for glottal flow estimation by separating the causal and anticausal components of a windowed speech signal as done by the Zeros of the Z-Transform (ZZT) decomposition. Based on exactly the same principles presented for ZZT decomposition, windowing should be applied such that the windowed speech signals exhibit mixed-phase characteristics which conform the speech production model that the anticausal component is mainly due to the glottal flow open phase. The advantage of the complex cepstrum-based approach compared to the ZZT decomposition is its much higher speed.

Synthesizing desired emotions using concatenative algorithms relies on collection of large databases. This paper focuses on the development and assessment of a simple algorithm to interpolate the intended vocal effort in existing databases in order to create new databases with intermediate levels of vocal effort. Three diphone databases in German with soft, modal, and loud voice qualities are processed with a spectral interpolation algorithm. A listening test is performed to evaluate the intended vocal effort in the original databases as well as the interpolated ones. The results show that the interpolation algorithm can create the intended intermediate levels of vocal effort given the original databases independent of the language background of the subjects. 1.

We propose a new spectral representation called the zeros of z-transform (ZZT), which is an all-zero representation of the z-transform of the signal. We show that separate patterns exist in ZZT representations of speech signals for the glottal flow and the vocal tract contributions. A decomposition method for source-tract separation is presented based on ZZT. The ZZT-decomposition consists in grouping the zeros into two sets, according to their location in the z-plane. This type of decomposition leads to separating glottal flow contribution (without a return phase) from vocal tract contribution in the z domain.

This study presents a method for estimation of glottal formant frequency (Fg) from speech signals. Our method is based on zeros of z-transform decomposition of speech spectra into two spectra: glottal flow dominated spectrum and vocal tract dominated spectrum. Peak picking is performed on the amplitude spectrum of the glottal flow dominated part. The algorithm is tested on synthetic speech. It is shown to be effective especially when glottal formant and first formant of vocal tract are not too close. In addition, tests on a real speech example are also presented where open quotient estimates from EGG signals are used as reference and correlated with the glottal formant frequency estimates. 1.

This study presents an improved version of our previously introduced formant tracking algorithm. The algorithm is based on processing the negative derivative of the argument of the chirp-z transform (termed as the differential phase spectrum) of a given speech signal. No modeling is included in the procedure but only peak picking on differential phase spectrum. We discuss the effects of roots of z-transform to differential phase spectrum and the need to ensure that all zeros are at some distance from the circle where chirp-z transform is computed. For that, we include an additional zero-decomposition step in our previously presented algorithm to improve its robustness. The final version of the algorithm is tested for analysis of synthetic speech and real speech signals and compared to two other formant tracking systems. 1.

This study proposes a new spectral decomposition method for source-tract separation. It is based on a new spectral representation called the Zeros of Z-Transform (ZZT), which is an all-zero representation of the z-transform of the signal. We show that separate patterns exist in ZZT representations of speech signals for the glottal flow and the vocal tract contributions. The ZZT-decomposition is simply composed of grouping the zeros into two sets, according to their location in the z-plane. This type of decomposition leads to separating glottal flow contribution (without a return phase) from vocal tract contributions in z domain.

Speech corpora design is one of the key issues in building high quality text to speech synthesis systems. Often read speech is used since it seems to be the easiest way to obtain a recorded speech corpus with highest control of the content. The main topic of this study is designing text for recording read speech corpora for concatenative text to speech systems. We will discuss application of the greedy algorithm for text selection by proposing a new way of implementing it and comparing with the standard implementation. Additionally, a text corpus design for Turkish TTS is presented. 1.