17,565 research outputs found
Adaptive interpolation of discrete-time signals that can be modeled as autoregressive processes
This paper presents an adaptive algorithm for the restoration of lost sample values in discrete-time signals that can locally be described by means of autoregressive processes. The only restrictions are that the positions of the unknown samples should be known and that they should be embedded in a sufficiently large neighborhood of known samples. The estimates of the unknown samples are obtained by minimizing the sum of squares of the residual errors that involve estimates of the autoregressive parameters. A statistical analysis shows that, for a burst of lost samples, the expected quadratic interpolation error per sample converges to the signal variance when the burst length tends to infinity. The method is in fact the first step of an iterative algorithm, in which in each iteration step the current estimates of the missing samples are used to compute the new estimates. Furthermore, the feasibility of implementation in hardware for real-time use is established. The method has been tested on artificially generated auto-regressive processes as well as on digitized music and speech signals
The NLMS algorithm with time-variant optimum stepsize derived from a Bayesian network perspective
In this article, we derive a new stepsize adaptation for the normalized least
mean square algorithm (NLMS) by describing the task of linear acoustic echo
cancellation from a Bayesian network perspective. Similar to the well-known
Kalman filter equations, we model the acoustic wave propagation from the
loudspeaker to the microphone by a latent state vector and define a linear
observation equation (to model the relation between the state vector and the
observation) as well as a linear process equation (to model the temporal
progress of the state vector). Based on additional assumptions on the
statistics of the random variables in observation and process equation, we
apply the expectation-maximization (EM) algorithm to derive an NLMS-like filter
adaptation. By exploiting the conditional independence rules for Bayesian
networks, we reveal that the resulting EM-NLMS algorithm has a stepsize update
equivalent to the optimal-stepsize calculation proposed by Yamamoto and
Kitayama in 1982, which has been adopted in many textbooks. As main difference,
the instantaneous stepsize value is estimated in the M step of the EM algorithm
(instead of being approximated by artificially extending the acoustic echo
path). The EM-NLMS algorithm is experimentally verified for synthesized
scenarios with both, white noise and male speech as input signal.Comment: 4 pages, 1 page of reference
Bitwise Source Separation on Hashed Spectra: An Efficient Posterior Estimation Scheme Using Partial Rank Order Metrics
This paper proposes an efficient bitwise solution to the single-channel
source separation task. Most dictionary-based source separation algorithms rely
on iterative update rules during the run time, which becomes computationally
costly especially when we employ an overcomplete dictionary and sparse encoding
that tend to give better separation results. To avoid such cost we propose a
bitwise scheme on hashed spectra that leads to an efficient posterior
probability calculation. For each source, the algorithm uses a partial rank
order metric to extract robust features that form a binarized dictionary of
hashed spectra. Then, for a mixture spectrum, its hash code is compared with
each source's hashed dictionary in one pass. This simple voting-based
dictionary search allows a fast and iteration-free estimation of ratio masking
at each bin of a signal spectrogram. We verify that the proposed BitWise Source
Separation (BWSS) algorithm produces sensible source separation results for the
single-channel speech denoising task, with 6-8 dB mean SDR. To our knowledge,
this is the first dictionary based algorithm for this task that is completely
iteration-free in both training and testing
Improving subband spectral estimation using modified AR model
It has already been shown that spectral estimation can be improved when applied to subband outputs of an adapted filterbank rather than to the original fullband signal. In the present paper, this procedure is applied jointly to a novel predictive autoregressive (AR) model. The model exploits time-shifting and is therefore referred to as time-shift AR (TSAR)
model. Estimators are proposed for the unknown TS-AR parameters and the spectrum of the observed signal. The TS-AR model yields improved spectrum estimation by taking advantage of the correlation between subseries that after decimation. Simulation results on signals with continuous and line spectra that demonstrate the performance of the proposed method are provided
- …