Multichannel acoustic echo cancellers aim at reducing the undesired acoustic coupling among microphones and loudspeakers in multichannel audio/video teleconferencing systems. In this scenario, double-talk refers to the situation in which there is an active speaker both in the remote room and in the local room. The detection of double-talk is one of the main issues to be faced in such systems to ensure a correct echo cancellation. Indeed, filters coefficients update has to be suppressed during double-talk in order to prevent the filter from diverging because of the presence of the local speaker. In this paper, a novel approach for double-talk detection is discussed based on the estimation of the fundamental frequency of the residual error signals through adaptive second-order notch filters. More specifically, simultaneous speaker activity in the remote and local rooms is revealed by exploiting the same frequency tracking introduced for multichannel decorrelation. Several experiments have been carried out to prove the effectiveness of the proposed approach also making comparison with other solutions of the state of the art.

The paper proposes a novel approach for measuring the room impulse response that is robust toward the nonlinearities affecting the power amplifier or the loudspeaker. The approach is implemented by modeling the acoustic path as a Legendre nonlinear filter and by measuring the first-order kernel using perfect periodic sequences and the cross-correlation method. Perfect sequences for Legendre filters are periodic sequences that guarantee the orthogonality of the Legendre basis functions over a period. They ensure the robustness of the first kernel measurement toward nonlinear distortions. The paper also explains how perfect periodic sequences for Legendre filters that are suitable for room impulse response identification can be developed. Experimental results involving both simulated and real environments illustrate the effectiveness and the robustness of the proposed approach.

Acoustic echo cancellers are used in teleconferencing systems in order to reduce undesired echoes due to coupling between microphones and loudspeakers. Stereophonic systems provide more realistic experience than single-channel systems, since listeners have spatial information that helps to identify the speaker position. Assuming this scenario, a suitable choice for the system parameters becomes essential to improve the audio reproduction quality. Error-driven optimization strategies are usually used to obtain an optimal system configuration but there is no relationship with the quality desired by the user. In this paper, an interactive evolutionary algorithm is adopted for a stereophonic acoustic echo cancellation system in order to meet subjective specifications in the optimization stage. In this way, the optimal system configuration is derived according to a user-driven approach in order to satisfy the quality requirements demanded by users availing such stereophonic systems. Experimental results prove the effectiveness of the proposed interactive architecture according to both objective and subjective measures.

In the field of immersive audio, crosstalk cancellation is adopted preserving the spatial characteristics of the reproduced signals. It is based on the use of HRTFs that can be approximated through a free-field model. However, this method assumes that the physical head is removed and replaced with two monopole receivers at the position of the ears and thus the source characteristics is not considered. In this context, a new approximation for the HRTFs is introduced taking into account the source directivity, i.e., the loudspeaker is modelled as a rigid sphere with a membrane vibrating on a spherical cap. A deep investigation on the effect of introducing the source directivity in the free-field model is reported in this paper. In particular, several experimental results are shown, considering both objective and subjective measurements, to validate the effectiveness of the proposed solution in terms of both channel separation and audio quality preservation.

Voice activity detection algorithms are intended to discriminate vocal segments from silence or other background noise periods. This is an important aspect in many speech communication systems and applications, e.g. teleconferencing systems. Therefore, a new method for detecting voiced segments in a multichannel scenario is proposed in this work by exploiting the fundamental frequency tracking through adaptive notch filters previously introduced for interchannel coherence reduction in multichannel acoustic echo cancellation systems. Novel expedients are presented in this paper for obtaining a more accurate tracking and thus for guaranteeing a correct voiced/unvoiced decision. The great advantage of the presented solution is related to the possibility of deriving a self-contained solution for multichannel acoustic echo cancellation where all the involved algorithms are based on the same fundamental frequency estimation. Experimental results prove the effectiveness of the proposed approach in correctly identifying vocal segments and silence and background noise periods.

Immersive speech communication systems have been gaining increasing attention due to their ability to reproduce enhanced acoustic images, and thus achieving good performance in terms of sound quality and accuracy. In this context , a fundamental role is played by intelligent acoustic interfaces (IAIs), which aim at acquiring and/or reproducing desired acoustic information with enhanced perception. The recent widespread availability of multimedia devices, equipped with different kind of sensors, has broadened the range of data processing methods, thus giving a chance for developing advanced IAIs. In this paper, we propose an immersive communication system composed of two IAIs: the first one exploits microphones and cameras, together with a signal processing system, to reduce unwanted noise and enhance the speech quality of the desired information in the transmitting room; the second one is an advanced reproduction system based on a loudspeaker array and on an effective wave field synthesis technique capable of reproducing the spatial perception of the desired speech source in the receiving room. The whole system has been assessed in simulated and real immersive communication scenarios: objective and subjective evaluations have been shown the effectiveness of the proposed system.

Multichannel sound reproduction systems aim at providing an optimal acoustical sensation to the listener by enhancing the listening experience and improving the spatial sound impression of the virtual scene. In this context, many audio applications requiring adaptive processing schemes have been developed in order to improve the performance of the audio reproduction. However, online multichannel systems identification requires the introduction of suitable solutions for overcoming the well-known non-uniqueness problem related to the correlation existing among the loudspeaker signals. This issue has been deeply investigated considering stereophonic systems but its extension to multichannel systems may not be so straightforward. In this paper, a novel solution for speech and audio signals is presented based on the “missing-fundamental” phenomenon. Experimental results proved the effectiveness of the approach also making comparisons with the existing state of the art.

Multichannel acoustic echo cancellers are used in teleconferencing systems in order to reduce undesired echoes and to introduce better performance in terms of sound localization in the presence of more than one participant. Already stereophonic systems obtain realistic performance, since listeners have spatial information that helps to identify the speaker position. Unfortunately, the correlation between the two channels prevents a correct identification of the echo paths and signal decorrelation is needed. In this letter, a decorrelation approach is proposed, combining the missing-fundamental theory with frequency shifting. Therefore, the proposed approach is based on the computation of an adaptive parameter related to the fundamental frequency that is also used for controlling the desired frequency shift value. Experiments confirm that high performance in terms of magnitude squared coherence reduction and convergence speed increase can be obtained, also comparing it with other approaches of the state of the art and outperforming previous results.

In teleconferencing systems, undesired echoes due to coupling between the loudspeaker and the microphone can be reduced using acoustic echo cancellers (AECs). In order to introduce better performance in terms of sound localization, at least stereophonic AECs (SAECs) have to be employed. In this case, the correlation between the two channels prevents a correct identification of the echo paths and a signal decorrelation is needed. In this paper, a novel mixed approach to signal decorrelation is proposed based on the estimation of the fundamental frequency. The principle of the proposed structure is to use an approach based on the “missing-fundamental” phenomenon at low frequencies (below 500 Hz), a time-varying all-pass filter controlled by the fundamental frequency at medium and high frequencies, and a time-varying multi-notch filter for improving performance in the upper part of the spectrum. Experiments confirm the effectiveness of the proposed approach in terms of correct identification of the echo paths and stereo perception preservation.

Acoustic echo cancelers (AECs) are used in teleconferencing systems to reduce undesired echoes originating from coupling between the loudspeaker and the microphone. Better performance in terms of sound localization can be achieved only by using multichannel AECs. By the use of a two-channel (stereo) system, it is already possible to obtain more realistic performance than the monochannel case because listeners have more spatial information that helps to identify the speaker position. In this paper, a novel approach to stereo AEC is proposed based on a combination of psychoacoustic effects. The principle of the proposed structure is to use an approach based on the “missing-fundamental” phenomenon at low frequencies (below 500 Hz) and a modified phase modulation approach at higher frequencies. Several results are presented in terms of magnitude-squared coherence, Itakura-Saito measure, convergence speed of adaptive filters, interaural cross-correlation, and subjective evaluation, in order to confirm the validity of the proposed approach.

Decorrelation is a well known issue in the context of Stereophonic Acoustic Echo Cancellation: it is related to the problem of uniquely identifying each pair of room acoustic paths, due to high inter-channel coherence. In this paper, a novel approach to decorrelate a stereo signal based on the missing fundamental phenomenon is proposed. An adaptive algorithm is employed to track the behavior of one of the two channels, ensuring a continuous decorrelation without affecting the stereo quality. Several results are presented comparing our approach with Masked Noise injection method in terms of Magnitude Square Coherence, Itakura Saito measure and Convergence Speed of adaptive filters in order to confirm the validity of the proposed approach.