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Featured researches published by Hauke Krüger.


international conference on acoustics, speech, and signal processing | 2009

Candidate proposal for ITU-T super-wideband speech and audio coding

Bernd Geiser; Hauke Krüger; Heinrich W. Löllmann; Peter Vary; Deming Zhang; Hualin Wan; Hai Ting Li; Li Bin Zhang

This paper describes the speech and audio codec that has been submitted to ITU-T by Huawei and ETRI as a candidate for the upcoming super-wideband and stereo extensions of Rec. G.729.1 and G.718. The core codec in the current implementation is G.729.1 and the encoded frequency range is increased from 7 kHz to 14 kHz. Therefore, the maximum bit rate is raised from 32 kbit/s to 64 kbit/s by adding five bitstream layers. A comprehensive overview of the codec is presented with a focus on the mono coding components. The results of the listening tests that have been conducted during the ITU-T qualification phase are summarized. The proposed codec passes all quality requirements for mono input signals.


international symposium on information theory and its applications | 2008

On logarithmic spherical vector quantization

Hauke Krüger; Raimund Schreiber; Bernd Geiser; Peter Vary

Logarithmic spherical vector quantization (LSVQ) is a specific type of gain-shape vector quantization (VQ), where input vectors are decomposed into a gain and a shape component which are quantized independently. In this contribution, novel theoretical results on LSVQ are presented: It will be shown that, for high bit rates, with logarithmic (A-Law) scalar quantization (SQ) of the gain and spherical vector quantization (SVQ) of the shape component a signal-to-noise ratio (SNR) is achieved which is approximately independent of the input source distribution. In addition, a detailed theoretical analysis leads to a lower bound for the quantization distortion related to SVQ. By introducing approximations for the assumption of high bit rates, this bound is the basis for the computation of the optimal allocation of bit rate to the gain and the shape quantizer, respectively, and yields an estimate for the achievable SNR for LSVQ.


international conference on acoustics, speech, and signal processing | 2011

Active noise control in headsets: A new approach for broadband feedback ANC

Thomas Schumacher; Hauke Krüger; Marco Jeub; Peter Vary; Christophe Beaugeant

In this paper a novel approach for broadband feedback active noise control (ANC) is presented which is based on the combination of classical non-adaptive feedback and adaptive feedback ANC techniques. The non-adaptive part is suitable to attenuate low frequency ambient noise whereas the adaptive part attenuates periodic components of the ambient noise. The proposed technique yields a higher overall noise attenuation performance compared to a purely classical non-adaptive feedback or purely adaptive feedback ANC system. In addition to that, the combination of both techniques is also beneficial for practical realizations since the adaptive feedback ANC stabilizes the overall system. With regard to low cost headset devices, the impact of practical hardware constraints such as low-cost analog-to-digital and digital-to-analog converters (ADC, DAC) is discussed. As a conclusion, a mixed analog-digital realization of the new approach is proposed.


international conference on acoustics, speech, and signal processing | 2011

Gosset lattice spherical vector quantizationwith lowcomplexity

Hauke Krüger; Bernd Geiser; Peter Vary; Hai Ting Li; Deming Zhang

This paper introduces a novel and highly efficient realization of a spherical vector quantizer (SVQ), the “Gosset Low Complexity Vector Quantizer” (GLCVQ). The GLCVQ codebook is composed of vectors that are located on spherical shells of the Gosset lattice E8. A high encoding efficiency is achieved by representing the spherical vector codebook as aggregated permutation codes. Compared to previous algorithms, the computational complexity and memory consumption is further reduced by exploiting the properties of so called classleader root vectors and by a novel approach for the codevector-to- index-mapping. The GLCVQ concept can be generalized to vector dimensions that are multiples of eight. In particular, GLCVQ for 16-dimensional vectors is used in Amd. 6 to ITU-T Rec. G.729.1.


distributed simulation and real-time applications | 2008

RTPROC: A System for Rapid Real-Time Prototyping in Audio Signal Processing

Hauke Krüger; Peter Vary

In this contribution a new system for the rapid development of real-time prototypes for digital audio signal processing algorithms on Windows PCs and a digital signal processor (DSP) platform is presented. The goal of this system is to enable even unexperienced developers to transform the conceptual idea of a new algorithm into a standalone real-time demonstrator as quickly and conveniently as possible. In order to achieve this goal, a software architecture is defined where hardware and algorithm related programming issues are separated to allow the algorithm developer to completely focus on the implementation of the algorithm only. Compared to the earlier version of H.Kruger et al (2003), the new system supports all real-time prototype development phases from first Matlab simulations to the final highly efficient implementation fixed point arithmetic and covers all relevant aspects such as e.g. control interface generation, function verification, complexity measuring and real-time data tracking. Development of prototypes based on general purpose PCs(RTProcPC) and the ADSP-21369 EZKIT embedded DSP target (RTProcDSP) are currently supported. The system has been successfully used to implement various real-time prototypes such as noise reduction, acoustic echo compensation, and digital hearing aid simulation.


international conference on acoustics, speech, and signal processing | 2014

On noise propagation in closed-loop linear predictive coding

Hauke Krüger; Bernd Geiser; Peter Vary

A new noise production and propagation model for open- and closed loop linear predictive coding (LPC) is proposed in this paper. The model allows to accurately predict the overall SNR even at lower bit rates where the conventional high rate theory fails. Moreover, a source of LPC encoder instabilities is pointed out which is due to the interaction between the quantizer and the (filtered) feedback of the quantization error. The new model is verified by measurements.


multimedia signal processing | 2006

An Efficient Codebook for the SCELP Low Delay Audio Codec

Hauke Krüger; Peter Vary

The SCELP (spherical code excited linear prediction) audio codec has recently been proposed as a new candidate for low delay audio coding based on linear prediction (LP). The new codec applies closed-loop vector quantization employing a spherical code in a gain shape manner. The spherical code is based on the apple-peeling code construction rule and in general does not require a codebook table for the encoding and decoding process. In this contribution, however, we propose to employ auxiliary information gathered in advance to reduce the computational encoding and decoding complexity at runtime significantly. This auxiliary information can be considered as the SCELP codebook. Due to the consideration of the characteristics of the apple-peeling code construction principle, this codebook can be stored very efficiently in read-only-memory. With the proposed principle, low computational as well as low memory complexity can be achieved simultaneously in the SCELP codec


international conference on acoustics, speech, and signal processing | 2004

A versatile DSP-system for student-projects on embedded real-time audio signal processing

Hauke Krüger; Thomas Lotter; Peter Vary

In this contribution, a new software environment for an embedded system is presented that is part of a laboratory for teaching students in digital audio signal processing. This environment enables even students with only a limited background in programming embedded technology to achieve remarkable implementations of realtime projects, including runtime user interaction. Hardware and algorithm related programming issues are separated for audio processing and interactive messaging. While the hardware related part is provided along with the software environment, the students can completely focus on algorithmic aspects. A well defined interface builds the bridge between the two programming components. Examples are presented to demonstrate the flexibility and efficiency of the given software platform. The underlying projects were defined, implemented and finally demonstrated by students in an interactive real-time demonstration.


european signal processing conference | 2011

Robust dual-channel noise power spectral density estimation

Marco Jeub; Christoph Matthias Nelke; Hauke Krüger; Christophe Beaugeant; Peter Vary


Archive | 2010

Do We Need Dereverberation for Hand-Held Telephony?

Marco Jeub; Magnus Schäfer; Hauke Krüger; Christoph Matthias Nelke; Christophe Beaugeant; Peter Vary

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Marco Jeub

RWTH Aachen University

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