Virtual 3-D sound can be easily delivered to a listener by binaural audio signals that are reproduced via headphones, which guarantees that only the correct signals reach the corresponding ears. Reproducing the binaural audio signal by two or more loudspeakers introduces the problems of crosstalk on the one hand, and, of reverberation on the other hand. In crosstalk cancellation, the audio signals are fed through a network of prefilters prior to loudspeaker reproduction to ensure that only the designated signal reaches the corresponding ear of the listener. Since room impulse responses are very sensitive to spatial mismatch, and since listeners might slightly move while listening, robust designs are needed. In this paper, we present a method that jointly handles the three problems of crosstalk, reverberation reduction, and spatial robustness with respect to varying listening positions for one or more binaural source signals and multiple listeners. The proposed method is based on a multichannel room impulse response reshaping approach by optimizing a -norm based criterion. Replacing the well-known least-squares technique by a -norm based method employing a large value for allows us to explicitly control the amount of crosstalk and to shape the remaining reverberation effects according to a desired decay.

Directional Audio Coding (DirAC) represents an efficient scheme to analyze and reproduce spatial sound; the coded stream consists of a single-channel audio signal and few parameters. Estimation of these parameters ideally relies on figure-of-eight microphones and one omnidirectional microphone. The figure-of-eight directivity can be efficiently approximated by first-order differential microphone arrays. However, due to spatial sampling, there are considerable deviations from the required directivity at high frequencies. These deviations lead to incorrect spatial parameter estimates; especially, the instantaneous direction-of-arrival (DOA) becomes biased and - beyond a certain aliasing frequency - ambiguous. Parameters like the DOA drive subsequent processing units, e. g., for spatial filtering. In this paper we propose a novel strategy to compensate bias and ambiguity not directly w. r. t. the DOA but the output parameters of a spatial filtering processing unit. Simulation results confirm the benefits of the novel method, which does not involve additional computational load at run-time.

In this contribution various objective measures that can be used to evaluate speech dereverberation algorithms by means of listening-room compensation (LRC) are compared to subjective listening tests. It is shown that technical measures describing the impulse responses are suitable for evaluation of such algorithms. Most signal-based objective measures fail to judge the specific distortions that may be introduced by LRC algorithms like late reverberation since these artifacts are small in amplitude but perceptually relevant due to the loss of masking of the room impulse response. Only one signal-based measure, the so-called perceptual similarity measure (PSM), showed high correlation with subjective rating for the given test setup.

The purpose of room impulse response (RIR) shortening and reshaping is usually to improve the intelligibility of the received signal by prefiltering the source signal before it is played with a loudspeaker in a closed room. In an alternative, but mathematically equivalent setting, one may aim to postfilter a recorded microphone signal to remove audible echoes. While least-squares methods have mainly been used for the design of shortening/reshaping filters for RIRs until now, we propose to use the infinity- orp-norm as optimization criteria. In our method, design errors will be uniformly distributed over the entire temporal range of the shortened/reshaped global impulse response. In addition, the psychoacoustic property of masking effects is considered during the filter design, which makes it possible to significantly reduce the filter length, compared to standard approaches, without affecting the perceived performance.

Directional audio coding (DirAC) is an efficient technique to capture and reproduce spatial sound. The analysis step outputs a mono DirAC stream, comprising an omnidirectional microphone pressure signal and side information, i.e., direction of arrival and diffuseness of the sound field expressed in time-frequency domain. This contribution proposes a method to merge two or more mono DirAC streams for a joint playback at the reproduction side. This problem arises in applications such as immersive spatial audio teleconferencing. With respect to a trivial direct merging, the proposed method is more efficient as it does not require the synthesis step. From this follows the benefit that the loudspeaker setup at the reproduction side does not have to be known in advance. Simulations and informal listening tests confirm the absence of any artifacts and that the proposed method is practically indistinguishable from the ideal merging.

Directional audio coding (DirAC) is a recent method for spatial audio processing, based on a perceptually motivated representation of spatial sound. Due to its efficiency, DirAC has already been proposed for spatial audio teleconferencing scenarios. Modern hands-free communication systems usually include beamforming techniques to improve speech intelligibility by suppressing diffuse background noise and interfering sources. In this paper, we propose a novel spatial filtering method which can be integrated into the DirAC spatial codec. It uses a spectral weighting of the recorded audio signal, where the design of the corresponding spatial filter transfer function is based on the DirAC parameters, i. e., direction-of-arrival and diffuseness of the sound field. Simulation results show that compared to a standard beamformer the novel technique offers significantly higher interference attenuation, while introducing similar distortion of the desired signal.

The purpose of room impulse response (RIR) shortening is to improve the intelligibility of the received signal by prefiltering the source signal before it is played with a loudspeaker in a closed room. In this paper, we propose to use the infinity-norm as optimization criterion for the design of shortening filters of RIRs. Similar to the equiripple filter design method, design errors will be uniformly distributed over the unwanted temporal range of the shortened global impulse response. The D50 measure is exploited during the design of the shortening filter, which makes it possible to significantly reduce the length of the prefilter without affecting the perceived performance.

Crosstalk cancellation is a well-known technique to generate virtual 3D sound via loudspeakers. Usually, headphones are used to playback audio material, which has been filtered with HRTFs. Audio playback in a reverberant room using loudspeakers and crosstalk cancellation is our intended application. This raises the need for a robust design, since listeners might slightly move their heads during a listening session. The novel design is based on a known least squares crosstalk canceller design technique, but with added robustness. The robustness is achieved with the help of assumed stochastic perturbation systems, which lie in parallel to the actual propagation impulse responses, during the design process.

This paper addresses the usability of channel shortening equalizers known for data transmission systems for the equalization of acoustic systems. In multicarrier systems, equalization filters are used to shorten the channel's effective length to the size of a cyclic prefix or the guard interval. In most applications the equalizer succeeds the channel. In acoustic systems, an equalizer is placed in front of a playback loudspeaker to generate a desired impulse response for the concatenation of the equalizer, a loudspeaker, a room impulse response, and a reference microphone. In this paper, we show that shaping the desired impulse response to a shorter reverberation time is more appropriate for acoustical systems than trying to exactly truncate it to a maximum length. Investigations are carried out using a multi-loudspeaker-multi-microphone system

In this contribution, we introduce a novel framework for combining approaches for acoustic echo cancellation and beamforming. Classical schemes of combination incorporate the simple concatenation of both subsystems. However, if the echo cancellers come first, they cannot exploit the noise reduction capabilities of the beamformer. In the other setup, a time-variant beamformer can heavily disturb the succeeding echo canceller's convergence. The new approach establishes a possibility to smoothly switch between two possible target functions: the mean squared errors at the beamformer's inputs and its output, respectively. Thus, one combined system benefits from advantages of both classical schemes. The paper contains a theoretical analysis of possible solutions for preceding echo cancellers, which are adapted using the beamformer output. The appropriate modification of the normalized least mean squares algorithm is derived too.

Post-filters are a powerful extension to improve echo attenuation when combined with the well-known echo canceller. In order to guarantee high quality of the transmitted speech signal, the primary purpose of a post-filtering system is to estimate the power spectral density (PSD) of the residual echo at the output of the echo canceller as accurately as possible. We introduce a novel technique to estimate the residual echo by using a microphone array. The robustness against double-talk and other additive interferences is reached by means of minimum statistics and further enhanced by exploiting spatial information. Simulation results show that the new methods are able to estimate the residual echo even under adverse conditions.

In this paper, we compare different combinations of a multi-channel non-adaptive noise reduction unit (NRU) and an acoustic echo cancellation unit (AEC) for a standard single-channel voice transmission. The results show that the NRU and the AEC-unit can be interchanged without increasing the computational complexity of the combined system, The length of an AEC's adaptive filter can be greatly reduced, if a multi-channel AEC-unit in front of a multichannel NRU is used. Furthermore, the control information, like the step-size of the NLMS-algorithm, has to be computed only once.