Current Issue : January - March Volume : 2012 Issue Number : 1 Articles : 5 Articles
Frequency-domain blind source separation (BSS) performs poorly in high reverberation because the independence assumption collapses at each frequency bins when the number of bins increases. To improve the separation result, this paper proposes a method which combines two techniques by using beamforming as a preprocessor of blind source separation. With the sound source locations supposed to be known, the mixed signals are dereverberated and enhanced by beamforming; then the beamformed signals are further separated by blind source separation. To implement the proposed method, a superdirective fixed beamformer is designed for beamforming, and an interfrequency dependence-based permutation alignment scheme is presented for frequency-domain blind source separation. With beamforming shortening mixing filters and reducing noise before blind source separation, the combined method works better in reverberation. The performance of the proposed method is investigated by separating up to 4 sources in different environments with reverberation time from 100?ms to 700?ms. Simulation results verify the outperformance of the proposed method over using beamforming or blind source separation alone. Analysis demonstrates that the proposed method is computationally efficient and appropriate for real-time processing....
We propose a methodology to design and evaluate environmental sounds for virtual environments. We propose to combine physically modeled sound events with recorded soundscapes. Physical models are used to provide feedback to users' actions, while soundscapes reproduce the characteristic soundmarks of an environment. In this particular case, physical models are used to simulate the act of walking in the botanical garden of the city of Prague, while soundscapes are used to reproduce the particular sound of the garden. The auditory feedback designed was combined with a photorealistic reproduction of the same garden. A between-subject experiment was conducted, where 126 subjects participated, involving six different experimental conditions, including both uni- and bimodal stimuli (auditory and visual). The auditory stimuli consisted of several combinations of auditory feedback, including static sound sources as well as self-induced interactive sounds simulated using physical models. Results show that subjects' motion in the environment is significantly enhanced when dynamic sound sources and sound of egomotion are rendered in the environment....
When a number of speakers are simultaneously active, for example in meetings or noisy public places, the sources of interest need to be separated from interfering speakers and from each other in order to be robustly recognized. Independent component analysis (ICA) has proven a valuable tool for this purpose. However, ICA outputs can still contain strong residual components of the interfering speakers whenever noise or reverberation is high. In such cases, nonlinear postprocessing can be applied to the ICA outputs, for the purpose of reducing remaining interferences. In order to improve robustness to the artefacts and loss of information caused by this process, recognition can be greatly enhanced by considering the processed speech feature vector as a random variable with time-varying uncertainty, rather than as deterministic. The aim of this paper is to show the potential to improve recognition of multiple overlapping speech signals through nonlinear postprocessing together with uncertainty-based decoding techniques....
In multiway loudspeaker systems, digital signal processing techniques have been used to correct the frequency response, the propagation time, and the lobbing errors. These solutions are mainly based on correcting the delays between the signals coming from loudspeaker system transducers, and they still show limited performances over the overlap frequency bands. In this paper, we propose an enhanced optimization of relevant directivity characteristics of a multiway loudspeaker system such as the frequency response, the radiation pattern, and the directivity index over an extended transducers' frequency overlap bands. The optimization process is based on applying complex weights to the crossover filter transfer functions by using an iterative approach....
To overcome harmonic structure distortions of complex tones in the low frequency range due to the frequency to electrode mapping function used in Nucleus cochlear implants, two modified frequency maps based on a semitone frequency scale (Smt-MF and Smt-LF) were implemented and evaluated. The semitone maps were compared against standard mapping in three psychoacoustic experiments with the three mappings; pitch ranking, melody contour identification (MCI) and instrument recognition. In the pitch ranking test, two tones were presented to normal hearing (NH) subjects. The MCI test presented different acoustic patterns to NH and CI recipients to identify the patterns. In the instrument recognition (IR) test, a musical piece was played by eight instruments which subjects had to identify. Pitch ranking results showed improvements with semitone mapping over Std mapping. This was reflected in the MCI results with both NH subjects and CI recipients. Smt-LF sounded unnaturally high-pitched due to frequency transposition. Clarinet recognition was significantly enhanced with Smt-MF but the average IR decreased. Pitch ranking and MCI showed improvements with semitone mapping over Std mapping. However, the frequency limits of Smt-LF and Smt-MF produced difficulties when partials were filtered out due to the frequency limits. Although Smt-LF provided better pitch ranking and MCI, the perceived sounds were much higher in pitch and some CI recipients disliked it. Smt-MF maps the tones closer to their natural characteristic frequencies and probably sounded more natural than Smt-LF....
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