Heinrich W. Löllmann

Low delay noise reduction and dereverberation for hearing aids

A new system for single-channel speech enhancement is proposed which achieves a joint suppression of late reverberant speech and background noise with a low signal delay and low computational complexity. It is based on a generalized spectral subtraction rule which depends on the variances of the late reverberant speech and background noise. The calculation of the spectral variances of the late reverberant speech requires an estimate of the reverberation time (RT) which is accomplished by a maximum likelihood (ML) approach. The enhancement with this blind RT estimation achieves almost the same speech quality as by using the actual RT. In comparison to commonly used post-filters in hearing aids which only perform a noise reduction, a significantly better objective and subjective speech quality is achieved. The proposed system performs time-domain filtering with coefficients adapted in the non-uniform (Bark-scaled) frequency-domain. This allows to achieve a high speech quality with low signal delay which is important for speech enhancement in hearing aids or related applications such as hands-free communication systems.

Uniform and warped low delay filter-banks for speech enhancement

A versatile filter-bank concept for adaptive subband filtering is proposed, which achieves a significantly lower algorithmic signal delay than commonly used analysis-synthesis filter-banks. It is derived as an efficient implementation of the filter-bank summation method and performs time-domain filtering with coefficients adapted in the uniform or non-uniform frequency-domain. The frequency warped version of the proposed filter-bank has a lower computational complexity than the usual warped analysis-synthesis filter-bank for most parameter configurations. The application to speech enhancement shows that the same quality of the enhanced speech can be achieved but with lower signal delay. For systems with tight signal delay requirements, modifications of the new filter-bank design are discussed to further decrease its signal delay by approximating the original time-domain filter by an FIR or IIR filter of lower degree. This approach can achieve a very low signal delay and reduced computational complexity with almost no loss for the perceived speech quality.

A blind speech enhancement algorithm for the suppression of late reverberation and noise

This paper proposes a new speech enhancement algorithm for the suppression of background noise and late reverberation without a priori knowledge. A generalized spectral weighting rule based on estimations for the spectral variances of the late reverberant speech and background noise is used for speech enhancement. This allows to suppress speech distortions due to late room reflections without knowledge about the complete room impulse response. In contrast to existing methods, all needed quantities are estimated entire blindly from the reverberant and noisy speech signal. The new algorithm has also a low signal delay which is important, e.g., for speech enhancement in mobile communication devices or hearing aids.

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

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.

Post-Filter Design for Superdirective Beamformers with Closely Spaced Microphones

In this paper, the post-filter design for superdirective beamformers with small microphone arrays is investigated, which can be used for speech enhancement systems in mobile communication devices or digital hearing aids. It is shown that coherent noise sources can be well suppressed by a multi-channel controlled post-filter. However, a sufficient suppression of diffuse noise sources can not be achieved by this. Such noise can be further reduced by a single-channel controlled post-filter. This combined post-filter design leads to a significantly better speech quality compared to the related approach of Le Bouquin et al.

Least-Squares Design of DFT Filter-Banks Based on Allpass Transformation of Higher Order

The allpass transformation of higher order is a very general concept to construct a frequency warped analysis-synthesis filter bank (AS FB) with nonuniform time-frequency resolution. In contrast to the more common allpass transformation of first order, the delay elements of the analysis filter bank are substituted by allpass filters of higher order to achieve a more flexible control over its frequency selectivity. Known analytical closed-form designs for the synthesis filter bank can ensure perfect reconstruction (PR), but the synthesis subband filters are not necessarily stable and exhibit no distinctive bandpass characteristic. These problems are addressed by a new least-squares error (LSE) filter bank design. The coefficients of the finite-impulse-response (FIR) synthesis filters are determined simply by a linear set of equations where the signal delay is an adjustable design parameter. This approach can achieve a perfect signal reconstruction with synthesis filters which are inherently stable and feature a bandpass characteristic. The proposed filter bank is of interest for various subband processing systems requiring nonuniform frequency bands.

Least-squares design of subsampled allpass transformed DFT filter-banks with LTI property

A new design approach for an allpass transformed analysis-synthesis filter-bank (AS FB) with subsampling is proposed, which can be used for adaptive subband processing with non-uniform time-frequency resolution. Signal reconstruction is performed by an FIR synthesis filter-bank whose coefficients are determined by two different linear least-squares (LS) error designs. The presented unconstrained LS optimization leads to perfect reconstruction (PR), but an insufficient bandpass characteristic for the synthesis subband filters. This problem is solved by a second, equality constrained least-squares (CLS) error design, which achieves near-perfect reconstruction (NPR). In the process, the linear distortions are minimized with the constraint for a linear, time-invariant (LTI) overall transfer function. The aliasing-free reconstruction error is significantly lower than for known designs of allpass transformed DFT filter-banks with NPR. Moreover, the new filter-bank design allows for a low and adjustable signal delay.

Design of IIR QMF banks with near-perfect reconstruction and low complexity

A novel design for a two-channel IIR quadrature-mirror filter (QMF) bank with near-perfect reconstruction (NPR) is presented. The analysis filter-bank is given by an efficient polyphase network (PPN) implementation based on allpass filters. The arising phase distortions are almost compensated by stable allpass filters, designed via analytical closed-form expressions. In a first design, the remaining aliasing, amplitude and phase distortions become arbitrarily small in dependence of the tolerable system delay and algorithmic complexity, respectively. In a second design, aliasing and amplitude distortions are completely canceled and phase distortions are minimized at the expense of an additional signal delay. The proposed QMF banks have a lower algorithmic complexity than comparable designs.

Estimation of the frequency dependent reverberation time by means of warped filter-banks

An improved approach for the estimation of the frequency dependent reverberation time (RT) by means of allpass transformed filter-banks is presented. It is shown that by means of these warped filter-banks, a much more accurate RT estimation at lower frequencies can be obtained than by octave filter-banks, which are commonly used for the estimation of the frequency dependent RT. Furthermore, allpass transformed filter-banks can achieve a much better approximation of the non-uniform frequency resolution of the human auditory system than octave filter-banks. A uniform or non-uniform (auditory) frequency resolution can thereby be simply adjusted by a single allpass coefficient. The RT estimation can be done with an allpass transformed DFT or DCT filter-bank. The warped DCT filter-bank is of special interest as it provides real-valued subband signals. This facilitates the use of a maximum-likelihood (ML) estimator for either a non-blind estimation of the frequency dependent RT from a room impulse response or a blind estimation from a reverberant speech signal.

IIR QMF-bank design for speech and audio subband coding

A new speech and audio codec has been submitted recently to ITU-T by a consortium of Huawei and ETRI as candidate proposal for the super-wideband and stereo extensions of ITU-T Rec. G.729.1 and G.718. This hierarchical codec with bit rates from 8-64 kbit/s relies on a subband splitting by means of a quadrature-mirror filter-bank (QMF-bank). For this, an allpass-based QMF-bank is used whose design and implementation is presented in this contribution. This IIR filter-bank allows to achieve a significantly lower signal delay in comparison to the traditional FIR QMF-bank solution without a compromise for the speech and audio quality.