Christian Schörkhuber

Kunstuniversität Graz | KUG · Institute of Electronic Music and Acoustics

Ambisonics has been widely adopted for conveying immersive audio experiences via headphone-based sound scene reproduction. It is well known, however, that the inherent reduction of spatial resolution of order-limited Ambisonic scenes leads to blurred source images, reduced spaciousness and direction-dependent timbral artifacts when rendered for bin...

Headphone-based reproduction of Ambisonic sound scenes is an active research area. Potential applications are to be found in gaming and virtual reality, simulation and auralization, or music production. Binaural rendering can be achieved via Ambisonic decoding for an array of virtual loudspeakers and subsequent convolution with the respective HRTF,...

(Open Access: https://doi.org/10.1121/1.5040489) Binaural rendering of Ambisonic signals is of great interest in the fields of virtual reality, immersive media, and virtual acoustics. Typically, the spatial order of head-related impulse responses (HRIRs) is considerably higher than the order of the Ambisonic signals. The resulting order reduction o...

A method for acoustic scene playback comprises: providing recording data comprising microphone signals of one or more microphone setups positioned within an acoustic scene and microphone metadata of the one or more microphone setups, wherein each of the one or more microphone setups comprises one or more microphones and has a recording spot which i...

Typically, the frequency-dependent transformation matrix to convert the signals of a spherical or circular microphone array to Ambisonic signals is obtained by a norm-constrained least-squares approach, where either modelled or measured microphone responses are used. These methods do not enforce orthonormality of the obtained directivity patterns,...

A sound signal processing apparatus including a plurality of microphones, where each microphone is configured to receive the sound signal from the target source, a processor configured to estimate a first power measure on the basis of the sound signal from the target source received by a first microphone of the microphones and a second power measur...

The aim of immersive teleconferencing is to convey a realistic sound field impression to a remote participant. To this end, the spatial distribution of talkers as well as room information needs to be captured by the near-end system and accurately reproduced on the far-end. We consider a setup where high speech quality is obtained by means of severa...

Everyday situations are rich in numerous acoustic events emerging from different origins. Such acoustic scenes may comprise discussions of our fellow human beings, chirping birds, cars, cyclists, and many more. So far, no recording or scene analysis technique for this rich and dynamically changing acoustic environment exists, though it would be nee...

In this paper, we propose a time-frequency representation where the frequency bins are distributed uniformly in log-frequency and their Q-factors obey a linear function of the bin center frequencies. The latter allows for time-frequency representations where the bandwidths can be e.g. constant on the log-frequency scale (constant Q) or constant on...

Distributed microphone arrays exploit the spatial diversity of an acoustic scene and obtain higher signal-to-noise ratios than compact microphone arrays that sample the sound field only locally. However, as distances between distributed microphones grow, wired connections become infeasible and Wireless Acoustic Sensor Networks (WASN) need to be emp...

Pitch shifting of polyphonic music is usually performed by manipulating the time-frequency representation of the input signal. Most approaches proposed in the past are based on the Fourier transform although its linear frequency bin spacing is known to be inadequate to some degree for analyzing and processing music signals. Recently invertible cons...

Pitch-scale modifications of polyphonic music are usually performed by manipulating the time-frequency representation of the input signal. Most approaches proposed in the past are thereby based on the Fourier transform although its linear frequency bin spacing is known to be inadequate to some degree for analysing and processing music signals. Rece...

This paper proposes a computationally efficient method for computing the constant-Q transform (CQT) of a time-domain signal. CQT refers to a time-frequency represen-tation where the frequency bins are geometrically spaced and the Q-factors (ratios of the center frequencies to band-widths) of all bins are equal. An inverse transform is pro-posed whi...