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In this paper, we consider the problem of recovering desired sound source signals from on-board microphone recordings on a noisy drone. Enhancement of source signal degraded by drone noise is considered to be a difficult task due to the strong noise generated from its motors and propellers causing an extremely low signal-to-drone noise ratio (SDNR). We propose a solution (i) by using a multichannel Wiener filter (MWF) to remove drone noise from microphone recordings, and (ii) further reduction of residual noise using a Gaussian mixture model (GMM) based dual-stage parametric Wiener filter (WF). The method exploits known statistics of motor current-specific drone noise. The theory developed is applicable to irregular microphone arrays embedded on a drone enabling realistic integration to most drones. We demonstrate the validity of the proposed framework through (i) experimental recordings from two different drone acoustics datasets and (ii) outdoor measurements from a hovering drone for a bioacoustic application. The results confirm improved performance in terms of SDNR, speech quality (PESQ), and intelligibility (STOI) at very low SDNR levels (up to -30 dB) and show a strong potential for signal enhancement applications using noisy drones.

Sound field reproduction algorithms require loudspeaker directivity, which is usually measured at discrete frequencies. A time domain model of loudspeaker directivity benefits broadband applications. This Letter proposes the concept of a directional wave front in the time domain, which could be linked to loudspeaker impulse responses measured on a spherical surface. The observed signal in the time domain at the boundary of a spherical region due to a propagating directional wave front is illustrated using a geometric model. Based on the geometric model, the spherical harmonic decomposition of the observed signal in the time domain is also derived.

In this paper, we consider the problem of recovering speech from audio recordings of on-board microphones in a noisy drone platform. Enhancement of speech degraded by drone noise is considered to be a difficult task due to the strong noise generated from its motors and propellers causing an extremely low signal-to-drone noise ratio (SdNR). We propose a solution by (i) developing a multichannel Wiener filter (MWF) to remove drone noise from microphone recordings, and (ii) further reduction of residual noise using a Gaussian mixture model (GMM) based dual-stage parametric Wiener filter (WF). The method exploits the known statistics of motor current-specific drone noise. The theory developed is applicable to irregular microphone array embedded in a drone enabling realistic integration to most drones. We evaluate the proposed framework using two different drone acoustics datasets under extreme SdNR levels including −30 dB. The experimental results confirm promising performance in terms of SdNR improvement, speech quality (PESQ), and intelligibility (STOI) and show a strong potential for speech enhancement applications using noisy drones.

Spatial audio has been studied for several decades, but has seen much renewed interest recently due to advances in both software and hardware for capture and playback, and the emergence of applications such as virtual reality and augmented reality. This renewed interest has led to the investment of increasing efforts in developing signal processing algorithms for spatial audio, both for capture and for playback. In particular, due to the popularity of headphones and earphones, many spatial audio signal processing methods have dealt with binaural reproduction based on headphone listening. Among these new developments, processing spatial audio signals recorded in real environments using microphone arrays plays an important role. Following this emerging activity, this paper aims to provide a scientific review of recent developments and an outlook for future challenges. This review also proposes a generalized framework for describing spatial audio signal processing for the binaural reproduction of recorded sound. This framework helps to understand the collective progress of the research community, and to identify gaps for future research. It is composed of five main blocks, namely: the acoustic scene, recording, processing, reproduction, and perception and evaluation. First, each block is briefly presented, and then, a comprehensive review of the processing block is provided. This includes topics from simple binaural recording to Ambisonics and perceptually motivated approaches, which focus on careful array configuration and design. Beamforming and parametric-based processing afford more flexible designs and shift the focus to processing and modeling of the sound field. Then, emerging machine- and deep-learning approaches, which take a further step towards flexibility in design, are described. Finally, specific methods for signal transformations such as rotation, translation and enhancement, enabling additional flexibility in reproduction and improvement in the quality of the binaural signal, are presented. The review concludes by highlighting directions for future research.

Successful recording of large spatial soundfields is a prevailing challenge in acoustic signal processing due to the enormous numbers of microphones required. This paper presents the design and analysis of an array of higher order microphones that uses 2D wavefield translation to provide a mode matching solution to the height invariant recording problem. It is shown that the use of $M$th order microphones significantly reduces the number of microphone units by a factor of $1/(2M + 1)$ at the expense of increased complexity at each microphone unit. Robustness of the proposed array is also analyzed based on the condition number of the translation matrix while discussing array configurations that result in low condition numbers. The white-noise gain (WNG) of the array is then derived to verify that improved WNG can be achieved when the translation matrix is well conditioned. Furthermore, the array’s performance is studied for interior soundfield recording as well as exterior soundfield recording using appropriate simulation examples.

Reproduction of a given sound field interior to a circular loudspeaker array without producing an undesirable exterior sound field is an unsolved problem over a broadband of frequencies. At low frequencies, by implementing the Kirchhoff-Helmholtz integral using a circular discrete array of line-source loudspeakers, a sound field can be recreated within the array and produce no exterior sound field, provided that the loudspeakers have azimuthal polar responses with variable first-order responses which are a combination of a two-dimensional (2D) monopole and a radially oriented 2D dipole. This paper examines the performance of circular discrete arrays of line-source loudspeakers which also include a tangential dipole, providing general variable-directivity responses in azimuth. It is shown that at low frequencies, the tangential dipoles are not required, but that near and above the Nyquist frequency, the tangential dipoles can both improve the interior accuracy and reduce the exterior sound field. The additional dipoles extend the useful range of the array by around an octave.

Spatial sound field recording and reproduction in reverberant rooms require the measurement of room transfer functions (RTF) and corresponding compensation such as room equalization to avoid unintended effects. Typically, the RTF rapidly varies over the room and hence requires a large number of point to point measurements to characterize the room. This paper provides (i) an efficient parameterization of the three dimensional acoustic transfer function over a region of space and (ii) a method to merge distributed multiple measurements by a higher order microphone (such as a spherical array) to analyze and synthesize the room transfer function over a region of space. The proposed method is shown to be a practical way of measuring the RTF over large areas with a significantly reduced number of measurements and improved robustness.

Spatial sound reproduction systems aim to produce a desired sound field over a volume of space. At high frequencies, the number of loudspeakers required is prohibitive. This paper shows that the use of loudspeakers with up to Nth order directivity allows reproduction over N times the bandwidth and produces a significantly attenuated exterior sound field. If the constraint on exterior cancellation of the field is removed, reproduction is possible over approximately 2N times the bandwidth. The use of higher order loudspeakers thus allows a significant reduction in the number of loudspeaker units, at the expense of increased complexity in each unit. For completeness, results are included for the generation of an exterior field with or without cancellation of the interior field.

Recording and reproduction of spatial sound fields over a large area is an unresolved problem in acoustic signal processing. This is due to the the inherent restriction in recording higher order harmonic components using practically realizable microphone arrays. As the frequency increases and as the region of interest becomes large the number of microphones needed in effective recording increases beyond practicality. In this paper, we show how to use higher order microphones, distributed in a large area, to record and accurately reconstruct 2D spatial sound fields. We use sound field coefficient translation between origins to combine distributed field recording to a single sound field over the entire region. We use simulation examples in (i) interior and (ii) exterior fields to corroborate our design.

Sound reproduction systems aim to produce a desired sound field over a region of space. At high frequencies, the number of loudspeakers required is prohibitive. This paper shows that the use of Nth order loudspeakers, in which each loudspeaker produces polar responses up to cos (NO) and sin (NO), produces accurate reproduction over N times the area of a first order array and can largely eliminate any exterior field. This allows a significant reduction in the number of loudspeaker units, at the expense of increased complexity in each loudspeaker unit.

Publication in the conference proceedings of EUSIPCO, Glasgow, Scotland, 2009

We investigate reduced-rank shift-invariant technique and its application for synchronization and channel identification in UWB systems. Shift-invariant techniques, such as ESPRIT and the matrix pencil method, have high resolution ability, but the associated high complexity makes them less attractive in real-time implementations. Aiming at reducing the complexity, we developed novel reduced-rank identification of principal components (RIPC) algorithms. These RIPC algorithms can automatically track the principal components and reduce the computational complexity significantly by transforming the generalized eigen-problem in an original high-dimensional space to a lower-dimensional space depending on the number of desired principal signals. We then investigate the application of the proposed RIPC algorithms for joint synchronization and channel estimation in UWB systems, where general correlator-based algorithms confront many limitations. Technical details, including sampling and the capture of synchronization delay, are provided. Experimental results show that the performance of the RIPC algorithms is only slightly inferior to the general full-rank algorithms.

In Cyclix Prefix OFDM (CP) systems, synchronization to the maximal multipath signal generally causes no problem, however, inter carrier interference (ICI) will be introduced in Zero-padding OFDM (ZP-OFDM) systems, e.g, multiband OFDM UWB (MB-OFDM) systems. In this paper, based on the overlap-and-add (ZP-OFDM-OLA) method typically applied in MB-OFDM, we analyze the cause of the ICI and the performance degradation due to the ICI. The expressions of the channel estimation error, signal-to-noise ratio and ICI are derived. Simulation results are also provided, which show good match with the theoretical results.

We present Cramér-Rao lower bounds (CRLBs) for the synchronization of UWB signals which should be tight lower bounds for the theoretical performance limits of UWB synchronizers. The CRLBs are investigated for both single-pulse systems and time-hopping systems in AWGN and multipath channels. Insights are given into the relationship between CRLBs for different Gaussian monocycles. An approximation method of the CRLBs is discussed when nuisance parameters exist. CRLBs in multipath channels are studied and formulated for three scenarios depending on the way multipath interference is treated. We find that a larger number of multipaths implies higher CRLBs and inferior performance of the synchronizers, and multipath interference on CRLBs cannot be eliminated completely except in very special cases. As every estimate of time delay could not be perfect, the least influence of the synchronization error on the performance of receivers is quantified.

Publication in the conference proceedings of EUSIPCO, Lisbon, Portugal, 2014