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Dive into the research topics where Andreas Niedermeier is active.

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Featured researches published by Andreas Niedermeier.


international conference on acoustics, speech, and signal processing | 2015

Spectral envelope reconstruction via IGF for audio transform coding

Christian Helmrich; Andreas Niedermeier; Sascha Disch; Florin Ghido

In low-bitrate audio coding, modern coders often rely on efficient parametric techniques to enhance the performance of the waveform preserving transform coder core. While the latter features well-known perceptually adapted quantization of spectral coefficients, parametric techniques reconstruct the signal parts that have been quantized to zero by the encoder to meet the low-bitrate constraint. Large numbers of zeroed spectral values and especially consecutive zeros constituting gaps often lead to audible artifacts at the decoder. To avoid such artifacts the new 3GPP Enhanced Voice Services (EVS) coding standard utilizes noise filling and intelligent gap filling (IGF) techniques, guided by spectral envelope information. In this paper the underlying considerations of the parametric energy adjustment and transmission in EVS and its relation to noise filling, IGF, and tonality preservation are presented. It is further shown that complex-valued IGF envelope calculation in the encoder improves the temporal energy stability of some signals while retaining real-valued decoder-side processing.


european signal processing conference | 2015

Low-complexity semi-parametric joint-stereo audio transform coding

Christian Helmrich; Andreas Niedermeier; Stefan Bayer; Bernd Edler

Traditional audio codecs based on real-valued transforms utilize separate and largely independent algorithmic schemes for parametric coding of noise-like or high-frequency spectral components as well as channel pairs. It is shown that in the frequency-domain part of coders such as Extended HE-AAC, these schemes can be unified into a single algorithmic block located at the core of the modified discrete cosine transform path, enabling greater flexibility like semi-parametric coding and large savings in codec delay and complexity. This paper focuses on the stereo coding aspect of this block and demonstrates that, by using specially chosen spectral configurations when deriving the parametric side-information in the encoder, perceptual artifacts can be reduced and the spatial processing in the decoder can remain real-valued. Listening tests confirm the benefit of our proposal at intermediate bit-rates.


international conference on acoustics, speech, and signal processing | 2016

Harmonic-percussive-residual sound separation using the structure tensor on spectrograms

Richard Füg; Andreas Niedermeier; Jonathan Driedger; Sascha Disch; Meinard Müller

Harmonic-percussive-residual (HPR) sound separation is a useful preprocessing tool for applications such as pitched instrument transcription or rhythm extraction. Recent methods rely on the observation that in a spectrogram representation, harmonic sounds lead to horizontal structures and percussive sounds lead to vertical structures. Furthermore, these methods associate structures that are neither horizontal nor vertical (i.e., non-harmonic, non-percussive sounds) with a residual category. However, this assumption does not hold for signals like frequency modulated tones that show fluctuating spectral structures, while nevertheless carrying tonal information. Therefore, a strict classification into horizontal and vertical is inappropriate for these signals and might lead to leakage of tonal information into the residual component. In this work, we propose a novel method that instead uses the structure tensor-a mathematical tool known from image processing-to calculate predominant orientation angles in the magnitude spectrogram. We show how this orientation information can be used to distinguish between harmonic, percussive, and residual signal components, even in the case of frequency modulated signals. Finally, we verify the effectiveness of our method by means of both objective evaluation measures as well as audio examples.


IEEE Transactions on Broadcasting | 2017

Development of the MPEG-H TV Audio System for ATSC 3.0

Robert Bleidt; Deep Sen; Andreas Niedermeier; Bernd Czelhan; Simone Füg; Sascha Disch; Jürgen Herre; Johannes Hilpert; Max Neuendorf; Harald Fuchs; Jochen Issing; Adrian Murtaza; Achim Kuntz; Michael Kratschmer; Fabian Küch; Richard Füg; Benjamin Schubert; Sascha Dick; Guillaume Fuchs; Florian Schuh; Elena Burdiel; Nils Günther Peters; Moo-Young Kim

A new TV audio system based on the MPEG-H 3D audio standard has been designed, tested, and implemented for ATSC 3.0 broadcasting. The system offers immersive sound to increase the realism and immersion of programming, and offers audio objects that enable interactivity or personalization by viewers. Immersive sound may be broadcast using loudspeaker channel-based signals or scene-based components in combination with static or dynamic audio objects. Interactivity can be enabled through broadcaster-authored preset mixes or through user control of object gains and positions. Improved loudness and dynamic range control allows tailoring the sound for best reproduction on a variety of consumer devices and listening environments. The system includes features to allow operation in HD-SDI broadcast plants, storage, and editing of complex audio programs on existing video editor software or digital audio workstations, frame-accurate switching of programs, and new technologies to adapt current mixing consoles for live broadcast production of immersive and interactive sound. Field tests at live broadcast events were conducted during system design and a live demonstration test bed was constructed to prove the viability of the system design. The system also includes receiver-side components to enable interactivity, binaural rendering for headphone, or tablet computer listening, a “3D soundbar” for immersive playback without overhead speakers, and transport over HDMI 1.4 connections in consumer equipment. The system has been selected as a proposed standard of ATSC 3.0 and is the sole audio system of the UHD ATSC 3.0 broadcasting service currently being deployed in South Korea.


Journal of The Audio Engineering Society | 2016

Intelligent Gap Filling in Perceptual Transform Coding of Audio

Sascha Disch; Andreas Niedermeier; Christian Helmrich; Christian Neukam; Konstantin Schmidt; Ralf Geiger; Jérémie Lecomte; Florin Ghido; Frederik Nagel; Bernd Edler


Archive | 2017

PROCESSOR, METHOD AND COMPUTER PROGRAM FOR PROCESSING AN AUDIO SIGNAL USING TRUNCATED ANALYSIS OR SYNTHESIS WINDOW OVERLAP PORTIONS

Guillaume Fuchs; Markus Multrus; Matthias Neusinger; Andreas Niedermeier; Markus Schnell


Archive | 2016

Audio encoder for encoding a multichannel signal and audio decoder for decoding an encoded audio signal

Sascha Disch; Guillaume Fuchs; Emmanuel Ravelli; Christian Neukam; Konstantin Schmidt; Conrad Benndorf; Andreas Niedermeier; Benjamin Schubert; Ralf Geiger


Archive | 2016

Context-based entropy coding of sample values of a spectral envelope

Florin Ghido; Andreas Niedermeier


Archive | 2016

Apparatus and method for decoding or encoding an audio signal using energy information values for a reconstruction band

Andreas Niedermeier; Christian Ertel; Ralf Geiger; Florin Ghido; Christian Helmrich


Archive | 2015

Decoder for generating a frequency enhanced audio signal, method of decoding, encoder for generating an encoded signal and method of encoding using compact selection side information

Frederik Nagel; Sascha Disch; Andreas Niedermeier

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Christian Helmrich

University of Erlangen-Nuremberg

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Florin Ghido

Tampere University of Technology

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Guillaume Fuchs

Université de Sherbrooke

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Achim Kuntz

University of Erlangen-Nuremberg

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Christian Ertel

University of Erlangen-Nuremberg

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