Jacob Donley

An efficient approach to dynamically weighted multizone wideband reproduction of speech soundfields

This paper proposes and evaluates an efficient approach for practical reproduction of multizone soundfields for speech sources. The reproduction method, based on a previously proposed approach, utilises weighting parameters to control the soundfield reproduced in each zone whilst minimising the number of loudspeakers required. Proposed here is an interpolation scheme for predicting the weighting parameter values of the multizone soundfield model that otherwise requires significant computational effort. It is shown that initial computation time can be reduced by a factor of 1024 with only -85dB of error in the reproduced soundfield relative to reproduction without interpolated weighting parameters. The perceptual impact on the quality of the speech reproduced using the method is also shown to be negligible. By using pre-saved soundfields determined using the proposed approach, practical reproduction of dynamically weighted multizone soundfields of wideband speech could be achieved in real-time.

Improving speech privacy in personal sound zones

This paper proposes two methods for providing speech privacy between spatial zones in anechoic and reverberant environments. The methods are based on masking the content leaked between regions. The masking is optimised to maximise the speech intelligibility contrast (SIC) between the zones. The first method uses a uniform masker signal that is combined with desired multizone loudspeaker signals and requires acoustic contrast between zones. The second method computes a space-time domain masker signal in parallel with the loudspeaker signals so that the combination of the two emphasises the spectral masking in the targeted quiet zone. Simulations show that it is possible to achieve a significant SIC in anechoic environments whilst maintaining speech quality in the bright zone.

Multizone reproduction of speech soundfields: A perceptually weighted approach

In this paper a method for the reproduction of multizone speech soundfields using perceptual weighting criteria is proposed. Psychoacoustic models are used to derive a space-time-frequency weighting function to control leakage of perceptually unimportant energy from the bright zone into the quiet zone. This is combined with a method for regulating the number of basis planewaves used in the reproduction to allow for an efficient implementation using a codebook of predetermined weights based on desired soundfield energy in the zones. The approach is capable of improving the mean squared error for reproduced speech in the bright zone by -10.5 decibels. Results also show that the approach leads to a significant reduction in the spatial error within the bright zone whilst requiring 65% less loudspeaker signal power for the case where the soundfield in this zone is in line with, and hence partially directed to, the quiet zone.

Analysing the Quality of Experience of multisensory media from measurements of physiological responses

This paper investigates the Quality of Experience (QoE) of multisensory media by analysing biosignals collected by electroencephalography (EEG) and eye gaze sensors and comparing with subjective ratings. Also investigated is the impact on QoE of various levels of synchronicity between the sensory effect and target video scene. Results confirm findings from previous research that show sensory effects added to videos increases the QoE rating. While there was no statistical difference observed for the QoE ratings for different levels of sensory effect synchronicity, an analysis of raw EEG data showed 25% more activity in the temporal lobe during asynchronous effects and 20-25% more activity in the occipital lobe during synchronous effects. The eye gaze data showed more deviation for a video with synchronous effects and the EEG showed correlating occipital lobe activity for this instance. These differences in physiological responses indicate sensory effect synchronicity may affect QoE despite subjective ratings appearing similar.

Reproducing personal sound zones using a hybrid synthesis of dynamic and parametric loudspeakers

This paper proposes a hybrid approach to personal sound zones utilising multizone soundfield reproduction techniques and parametric loudspeakers. Crossover filters are designed, to switch between reproduction methods, through analytical analysis of aliasing artifacts in multizone reproductions. By realising the designed crossover filters, wideband acoustic contrast between zones is significantly improved. The trade-off between acoustic contrast and the bandwidth of the reproduced soundfield is investigated. Results show that by incorporating the proposed hybrid model the whole wideband bandwidth is spatial-aliasing free with a mean acoustic contrast consistently above 54.2dB, an improvement of up to 24.2dB from a non-hybrid approach, with as few as 16 dynamic loudspeakers and one parametric loudspeaker.

Active speech control using wave-domain processing with a linear wall of dipole secondary sources

In this paper, we investigate the effects of compensating for wave-domain filtering delay in an active speech control system. An active control system utilising wave-domain processed basis functions is evaluated for a linear array of dipole secondary sources. The target control soundfield is matched in a least squares sense using orthogonal wavefields to a predicted future target soundfield. Filtering is implemented using a block-based short-time signal processing approach which induces an inherent delay. We present an autoregressive method for predictively compensating for the filter delay. An approach to block-length choice that maximises the soundfield control is proposed for a trade-off between soundfield reproduction accuracy and prediction accuracy. Results show that block-length choice has a significant effect on the active suppression of speech.

Towards real-time source counting by estimation of coherent-to-diffuse ratios from ad-hoc microphone array recordings

Coherent-to-diffuse ratio (CDR) estimates over short time frames are utilized for source counting using ad-hoc microphone arrays to record speech from multiple participants in scenarios such as a meeting. It is shown that the CDR estimates obtained at ad-hoc dual (two channel) microphone nodes, located at unknown locations within an unknown reverberant room, can detect time frames with more than one active source and are informative for source counting applications. Results show that interfering sources can be detected with accuracies ranging from 69% to 89% for delays ranging from 20 ms to 300 ms, with source counting accuracies ranged from 61% to 81% for two sources and the same range of delays.