Abstract: A method and device are provided for filtering digital audio signals using at least one ARMA filter, particularly during a filter change. The method includes the following steps: a step of receiving a first request to change filtering to or from filtering by a first ARMA filter; and, in response to the first request, a step of gradually switching, at each of a plurality of cascaded first filtering blocks, between digital-signal filtering by a first basic filtering cell and digital-signal filtering by another associated basic filtering cell, the first basic filtering cells of the plurality of first filtering blocks factorizing the first filter.

Abstract: A method and device are provided for detecting acoustic shocks in an audio stream. The method includes: breaking down the audio stream into audio frames; analyzing the audio frames in order to assign each audio frame a category value from among a plurality of predefined values; and determining the probability of an acoustic shock occurring in a current frame, based on a sequence of a given length of category values assigned to a set of frames, using a two-state Markov model, defined by a predetermined transition matrix and transmission matrix.

Abstract: A method and device are provided for filtering digital audio signals using at least one ARMA filter, particularly during a filter change. The method includes the following steps: a step of receiving a first request to change filtering to or from filtering by a first ARMA filter; and, in response to the first request, a step of gradually switching, at each of a plurality of cascaded first filtering blocks, between digital-signal filtering by a first basic filtering cell and digital-signal filtering by another associated basic filtering cell, the first basic filtering cells of the plurality of first filtering blocks factorizing the first filter.

Abstract: A method and device are provided for detecting acoustic shocks in an audio stream. The method includes: breaking down the audio stream into audio frames; analyzing the audio frames in order to assign each audio frame a category value from among a plurality of predefined values; and determining the probability of an acoustic shock occurring in a current frame, based on a sequence of a given length of category values assigned to a set of frames, using a two-state Markov model, defined by a predetermined transition matrix and transmission matrix.

Abstract: The invention relates to the processing of a signal that is compression encoded (COD) according to a predetermined encoding type applying a quantification operation (Q) and then decoded (DEC) so that the quantification noise is present in the decoded signal (S*). The signal processing of the invention comprises applying a quantification noise reduction (TBQ) to the decoded signal (S), preferably in the following manner: first obtaining information (INF) on the type of compression encoding, selecting a model for the reduction of the quantification noise adapted to said information by estimating the quantification noise (BQ) that the encoding may have generated; and applying to the decoded signal (S*) a processing for reducing the quantification noise (FIL) according to the selected model.

Abstract: The invention relates to the processing of a signal that is compression encoded (COD) according to a predetermined encoding type applying a quantification operation (Q) and then decoded (DEC) so that the quantification noise is present in the decoded signal (S*). The signal processing of the invention comprises applying a quantification noise reduction (TBQ) to the decoded signal (S), preferably in the following manner: first obtaining information (INF) on the type of compression encoding, selecting a model for the reduction of the quantification noise adapted to said information by estimating the quantification noise (BQ) that the encoding may have generated; and applying to the decoded signal (S*) a processing for reducing the quantification noise (FIL) according to the selected model.

Abstract: The invention relates to a method of processing a noisy sound signal and to a device for implementing said method. The inventive method includes the following steps which are performed for each frame of a noisy sound signal, consisting in: applying a frequency-domain transform; estimating a power spectral density (PSD) of the noise for the frame; calculating a first noise reduction filter from the estimated PSD of the noise and from an estimation of the PSD of a wanted signal corresponding to the frame; filtering the frame with the aid of the calculated noise reduction filter, in order to obtain a first noise-corrected estimation for the frame; and obtaining a frame of a second signal from the first noise-corrected estimation of the aforementioned frame, comprising harmonics which are essentially at the same positions as the wanted signal corresponding to the frame of the noisy sound signal.

Abstract: The device calculates a first frequency-dependent useful signal level estimator for the frame. The transfer function of a first noise-reducing filter is determined on the basis of the first useful signal level estimator and of a frequency-dependent noise level estimator. A second frequency-dependent useful signal level estimator for the frame is then calculated by combining the spectrum of the input signal and the transfer function of the first noise-reducing filter. The transfer function of a second noise-reducing filter is determined on the basis of the second useful signal level estimator and of the noise level estimator. The latter transfer function is used in a frame filtering operation to produce a signal with reduced noise.

Abstract: The present invention relates to a receiving system for a multi-sensor antenna and comprising: at least one set of channel filters (310j, 710k, 1 . . . N) filtering the signals Xi(t,f) received by the different antenna sensors, where these signals may include a desired signal (Si(t,f)); a summer (350, 720) summing the filtered signals [Vi(t,f)], where i=1 . . . N, by means of the channel filters and emitting an antenna output signal Y(t,f), at least one calculating module (320, 720) receiving either the sensor signals and aligning them in phase or the signals filtered by the channel filters, said module estimating the transfer function (W(t,f), W k(t,f)) of an optimum filter in a manner to minimize the square difference between the antenna output signals filtered by said filter and the desired signal characterized in that if furthermore comprises at least one statistical analysis module (330, 730) operating on the frequency values of the transfer function.

Abstract: The invention relates to a method of processing a noisy sound signal and to a device for implementing said method. The inventive method comprises the following steps which are performed for each frame (x(k,n)) of a noisy sound signal, consisting in: applying a frequency-domain transform; estimating a power spectral density (PSD) of the noise for the frame (?bb(k,f)): calculating a first noise reduction filter (H1(k,f)) from the estimated PSD of the noise and from an estimation of the PSD of a wanted signal corresponding to the frame (?ss (k,f)); filtering the frame (x(k,n); X(k,f)) with the aid of the calculated noise reduction filter, in order to obtain a first noise-corrected estimation for the frame (S1(k,f)); and obtaining a frame of a second signal (Sharmo(k,f)) from the first noise-corrected estimation of the aforementioned frame, comprising harmonics which are essentially at the same positions as the wanted signal corresponding to the frame of the noisy sound signal.

Abstract: The ratio of useful-signal power to the power of interference signals at an antenna (ANT) having plural sensors, and at least one sensor is obtained by filtering the antenna output signal with a filter having a transfer function (W(t), W(t, f)) equal to the ratio of two linear functions of the power ({circumflex over (p)}y(t), {circumflex over (p)}y(t, f)) at the output of the antenna to the power {circumflex over (p)}x(t), {circumflex over (p)}x(t, f)) at the input of the antenna (ANT).

Abstract: Interference is reduced in a reception system having a multi-pickup antenna and at least one path for deriving an antenna signal from signals received by the different pickups. The transfer function of a first filter, called a first postfilter, is estimated from a reference signal making it possible to regenerate the interference. The antenna signal is filtered by the first postfilter.

Abstract: The invention concerns a method which consists, when analysing an input signal in the frequency domain, in determining a noise level estimator and a useful signal level estimator in an input signal frame, thereby enabling to calculate the transfer function of a first noise-reducing filter, carrying out a second pass to fine-tune the useful signal level estimator, by combining the signal spectrum and the first filter transfer function, then to calculate the transfer function of a second noise-reducing filter on the basis of the fine-tuned useful signal level estimator and the noise level estimator. Said second noise-reducing filter is then used to reduce the noise level in the frame.

Abstract: The present invention relates to a receiving system for a multi-sensor antenna and comprising: