• I am currently Senior Scientist at the the Advanced Audio Research Group in Fraunhofer IIS, focusing on the new Moving Picture Experts Group standard for audio for virtual and augmented reality (VR/AR), MPEG-I. The standard has now reached International Standard status.
  • Before that, I’ve spent some years on subjective and objective audio quality assessment for state-of-the-art perceptual audio codecs. During this time I’ve worked on extending an established ITU-R objective quality assessment method (BS.1387-1, PEAQ) for parametric codecs and multi-channel audio.
  • Before that, as a part of the Audio and Multimedia Technologies Division, I researched and designed audio processing algorithms for embedded devices and digital signal processors, maintaining and expanding the Core Design Kit. This involved collaboration with international partners such as Google, Apple, Amazon and others, and NHK Japan.
  • I have been involved in the development, implementation, optimization, and testing of cross-platform audio coding software in line with MPEG and 3GPP standards, as well as the design and development of automated software tests and deployment planning.
  • I’ve filed patents

MPEG‑I Immersive Audio

I was part of the core development team of the standard.

Articles, interviews, press coverage and standard resources

MPEG-H Personalized Immersive Sound

I was part of the team that produced the first embedded MPEG-H audio encoder of Fraunhofer. In particular, I was in charge of the fixed-point architecture integration of the core audio encoder and audio quality testing.

Articles, interviews, and standard resources

Delay-reduced mode of MPEG-4 Enhanced Low Delay AAC (AAC-ELD)

The MPEG-4 AAC Enhanced Low Delay (AAC-ELD) coder is well established in high quality communication applications, such as Apple’s FaceTime, as well as in professional live broadcasting. Both applications require high interactivity, which typically demands an algorithmic codec delay between 15 ms and 35 ms. Recently, MPEG finalized a new delay-reduced mode for AAC-ELD featuring only a fraction of the regular algorithmic delay. This mode operates virtually at higher sampling rates while maintaining standard sampling rates for I/O. Supporting this feature, AAC-ELD can address even more delay critical applications, like wireless microphones or headsets for TV.
In this paper, main details of the delay-reduced mode of AAC-ELD are presented and application scenarios are outlined. Audio quality aspects are discussed and compared against other codecs with a delay below 10 ms.

Other Projects