Deep Sen

Iterative Phase Estimation for the Synthesis of Separated Sources From Single-Channel Mixtures

In this letter, we propose a novel method of refining the time-domain synthesis of individual source estimates from a single channel mixture. Employing a closed-loop architecture, the algorithm refines the synthesis of each source by iteratively estimating the phase of the sources, given the estimates of the source magnitude spectra and a single channel time-domain mixture. The performance of the algorithm is evaluated for harmonic musical mixtures, and considerable improvements to the synthesized estimates are obtained relative to phase binary masking, given accurate source magnitude spectra.

Performance Enhancement of Vibration Sensing Employing Multiple Phase-Shifted Fiber Bragg Grating

Multiple phase-shifted fiber Bragg gratings (MPS-FBGs) are proposed for a novel intensity-type vibration sensing scheme with broadband source based interrogation. In this scheme, the collective and simultaneous operation of the subchannels of MPS-FBG multiplies the total acquired signal power change at a particular channel, considerably enhancing sensitivity. Compared to a standard FBG scheme, our experiment scheme attained significant sensitivity enhancement of nearly 20 dB using 17 phase-shifted FBG. These experimental results agree with the theoretical prediction based on cross-correlation relations. The sensitivity enhancement while retaining the system simplicity would be an attractive option for an economical coarse wavelength division multiplexed (CWDM) vibration sensing system based on a single broadband source.

Use of an auditory model to improve speech coders

A method for incorporating an auditory masking model in a speech coder using traditional articulatory models is presented. The auditory model attempts to model the frequency selectivity and masking properties of the human cochlea. Coding gain is achieved by analyzing the perceptual content of each sample in the spectrum. The scheme is thus able to introduce selective distortion that is a direct function of human hearing perception and is thus optimally matched to the hearing process. It is shown that good coding gain can be obtained with excellent speech quality. The algorithm can be used on its own or as a front end for traditional vocoders. it can also be implemented with very little computational overhead and low coding delay.<<ETX>>

Spectral envelope sensitivity of musical instrument sounds.

It is well known that the spectral envelope is a perceptually salient attribute in musical instrument timbre perception. While a number of studies have explored discrimination thresholds for changes to the spectral envelope, the question of how sensitivity varies as a function of center frequency and bandwidth for musical instruments has yet to be addressed. In this paper a two-alternative forced-choice experiment was conducted to observe perceptual sensitivity to modifications made on trumpet, clarinet and viola sounds. The experiment involved attenuating 14 frequency bands for each instrument in order to determine discrimination thresholds as a function of center frequency and bandwidth. The results indicate that perceptual sensitivity is governed by the first few harmonics and sensitivity does not improve when extending the bandwidth any higher. However, sensitivity was found to decrease if changes were made only to the higher frequencies and continued to decrease as the distorted bandwidth was widened. The results are analyzed and discussed with respect to two other spectral envelope discrimination studies in the literature as well as what is predicted from a psychoacoustic model.

Predicting foreground SH, SL and BNH DAM scores for multidimensional objective measure of speech quality

Current objective measures of speech quality (Rix et al. (2000), Beerends et al. (1994) attempt to evaluate degraded speech by calculating a single distance measure between the original signal and the synthesized signal being evaluated. The distance measure is usually carried out after both the original and synthesized signal have been transformed to represent the effect of the auditory periphery. However, the fact that the subjective judgement of quality is based on a multidimensional perceptual space representation suggests that a measure that is based on predicting a multitude of independent perceptual characteristics, would yield better results and be applicable to a wider range of distortions and speech synthesis systems. This paper presents such a multidimensional approach to objective evaluation of speech quality and is directly motivated by the work of Voiers (2001) from which the subjective evaluation procedure known as diagnostic acceptability measure (DAM) was created. While the DAM is a subjective measure of the detectability of the distortions identified by Voiers, this work reports on the first baby steps taken for objective evaluation of a subset of those same parametric distortions determined to be the principal components of the quality space from a previous statistical analysis (Sen (2001)).

Sensitivity Enhancement in Composite Cavity Fiber Laser Hydrophone

We report an application of composite cavity fiber laser (CCFL) for hydrophone sensitivity enhancement. While most of the sensitivity enhancement methods rely on amplification of acoustic signal by the coating design, our proposed scheme exploits the inherently nonlinear phase condition of the CCFL. A particular point of interest for CCFL hydrophone application is the proposed partial cavity sensing scheme that, when only one cavity is made responsive, a substantial sensitivity enhancement can be achieved. Theoretical analysis shows that this scheme can significantly enhance sensitivity, achieving as high as 40 dB excess to the standard response of a bare fiber. In experiment, this scheme produced a mean responsivity of -41.1 dB re rad Pa-1, representing an improvement in sensitivity by 14 dB compared to the standard response. Evidently, this CCFL sensing scheme provides an additional means for sensitivity improvement in conjunction to the conventional packaging technique.

Objective evaluation of speech signal quality by the prediction of multiple foreground diagnostic acceptability measure attributes

A methodology is described to objectively diagnose the quality of speech signals by predicting the perceptual detectability of a selected set of distortions. The distortions are a statistically selected subset of the broad number of distortions used in diagnostic acceptability measure (DAM) testing. The justification for such a methodology is established from the analysis of a set of speech signals representing a broad set of distortions and their respective DAM scores. At the heart of the ability to isolate and diagnose the perceptibility of the individual distortions is a physiologically motivated cochlear model. The philosophy and methodology is thus distinct from traditional objective measures that are typically designed to predict mean opinion scores (MOS) using well versed functional psychoacoustic models. Even so, a weighted sum of these objectively predicted set of distortions is able to predict accurate and robust MOS scores, even when the reference speech signals have been subject to the Lombard effect.

Sinusoidal Frequency Estimation based on the Time Derivative of the STFT Phase Response

This paper presents the phase derivative FFT (PDFFT)-a computationally efficient algorithm for estimating the frequency of a sinusoid from the short time Fourier transform (STFT). Upon obtaining initial coarse estimates from the FFT of a given frame, the PDFFT makes further refinement to the frequency estimate using only the time derivative of the phase response. The algorithm is derived and is shown to require only 4 multiplies per peak. Single frequencies in the presence of noise are resolved well, outperforming the commonly used quadratically interpolated FFT (QIFFT) method even with zero-padding. The algorithm is then used to separate two sinusoids of close frequency proximity that appear as a single peak in the magnitude spectrum

Comparison of ARMA modelling methods for low bit rate speech coding

There are two main parts of parametric speech coding algorithms such as codebook-excited linear prediction (CELP): the determination of the vocal tract filter parameters and the selection of the excitation signals based on a perceptual error criterion. The vocal tract includes the oral and nasal cavities depending on the type of speech segments (e.g. nasals and unvoiced fricatives). The contribution from the nasal tract suggests the need for an ARMA (or pole-zero) model instead of the conventional AR (pole only) model. The paper compares the performance of several ARMA modelling techniques in estimating the vocal tract filter parameters. The best method in terms of spectral fit and computational complexity is then applied to a CELP-type speech coding algorithm, with results which are superior to conventional AR models.<<ETX>>

Output power and threshold gain of apodized DFB fiber laser

Effects of apodization on distributed feedback fiber laser (DFB FL) output power and threshold gain are theoretically investigated by employing the transfer matrix method. Three distinct types of profile are investigated: the gaussian, flat or nonapodize, and sigmoid profile. The gaussian and sigmoid profiles are the two extreme cases examined; the former has a strong profile around a centrally located phase shift, while the latter is with a weaker profile. Findings indicate that the tradeoff between output power and higher order mode threshold performance are resulting from the interplay between these profile shapes. The comprehensive results presented in this paper should assist the development of high performance DFB FLs.