Orange

France Telecom

Alexandre Guerin

Julien Faure

Claude Marro

Mathieu Baque

David Virette

Marc Emerit

Rozenn Nicol

Grégory Pallone

Srdan Kitic

Gregory Pallone

Jean-Luc Garcia

Arnaud Lefort

Clément Gaultier

Henrique Tomaz-Amorim

Mathieu BaquÉ

A method and apparatus for processing sound data acquired by a plurality of microphones. The method includes: on the basis of the signals acquired by the plurality of microphones, determining a direction of arrival of a sound originating from at least one sound source of interest; applying spatial filtering to the sound data as a function of the direction of arrival of the sound; estimating ratios, in the time-frequency domain, in a magnitude representative of a signal amplitude, between the filtered sound data on the one hand and the acquired sound data on the other hand; and as a function of the estimated ratios, producing a weight mask to be applied in the time-frequency domain to the acquired sound data in order to construct an acoustic signal representing the sound originating from the source of interest but enhanced relative to the ambient noise.

Estimating an optimized mask for processing acquired sound data

The processing of a signal y(t) coming from a microphone of an equipment item including a loudspeaker intended to be supplied a signal x(t), limits an echo effect induced by the microphone capturing a sound emitted by the loudspeaker. This sound and any of its acoustic reflections follow an acoustic path w from the loudspeaker to the microphone. To limit the echo effect, the processing includes determining ŝ(t) a useful signal s(t) by subtracting from the signal y(t) an estimate of an echo signal x(t)*ŵ(t) given by applying a filter ŵ(t) to the signal x(t). The filter ŵ(t) is adaptive by variable step size to account for a change over time in the acoustic path w(t). The adaptive filter ŵ(t) is produced at each frame k of samples as a function of an update ΔWto the acoustic path w for this frame k and by applying a normalization Λ satisfying a criterion chosen for minimal variance.

Method and device for variable pitch echo cancellation

Processing acoustic signals to detect sound sources in a sound scene. The method includes: obtaining a plurality of signals representative of the sound scene, captured by a plurality of microphones of predefined positions; based on the signals captured by the microphones and on the positions of the microphones, applying a quantization of directional measurements of sound intensity and establishing a corresponding acoustic activity map in a sound source localization space, the space being of dimension N; constructing at least one vector basis of dimension less than N; projecting the acoustic activity map onto at least one axis of the vector basis; and searching for at least one local peak of acoustic activity in the map projection, an identified local peak corresponding to the presence of a sound source in the scene.

Localization of sound sources in a given acoustic environment

A method for processing sound data for separating N sound sources of a multichannel sound signal sensed in a real medium. The method includes: separating sources to the sensed multichannel signal and obtaining a separation matrix and a set of M sound components, with M&#x2265;N; calculating a set of bi-variate first descriptors representative of statistical relations between the components of the pairs of the set obtained of M components, calculating a set of uni-variate second descriptors representative of characteristics of encoding of the components of the set obtained of M components; and classifying the components of the set of M components, according to two classes of components, a first class of N direct components corresponding to the N direct sound sources and a second class of M&#x2212;N reverberated components, by calculating probability of membership in one of the two classes, dependent on the sets of first and second descriptors.

Processing of sound data for separating sound sources in a multichannel signal

A method implemented by a processing device for processing an ambisonic content including a plurality of ambisonic components of a plurality of orders defining a succession of ambisonic channels in each of which an ambisonic component is represented. The method includes: frequency filtering of the ambisonic components in a plurality of frequency bands, compiling an ambisonic decoding matrix, processing the ambisonic decoding matrix in order to extract, by matrix dimension reduction, a plurality of ambisonic decoding sub-matrices each associated with an ambisonic order and a frequency band selected for this ambisonic order, respective applications of the decoding sub-matrices to the ambisonic components in each selected frequency band, and a reconstruction, band by band, of the results of said respective applications, in order to deliver a plurality of decoded signals, each associated with a sound source.

Processing in sub-bands of an actual ambisonic content for improved decoding

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.

Method and device for filtering during a change in an ARMA filter

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.

Method and device for detecting acoustic shocks

The invention is aimed at improving the quality of the filtering by transfer functions of HRTF type of signals (L, R) compressed in a transformed domain, for binaural playing on two channels (L-BIN, R-BIN), using a combination of HRTF filters (h, h) including a decorrelated version (HRTF-C*, HRTF-E*) of a few of these filters. For this purpose, a decorrelation cue is given with spatialization parameters (SPAT) accompanying the compressed signals (L, R). The Decorrelation comprises applying a different phase shift to each subband of the input signal combined with addition of an overall delay. The invention makes it possible to improve the broadening in the binaural rendition of audio scenes initially in a multi-channel format.

Binaural spatialization of compression-encoded sound data utilizing phase shift and delay applied to each subband

Coefficients of an adaptive filter representative of an acoustic channel between an emitted acoustic signal and a microphone signal are determined and smoothed in time. An echo is then estimated by filtering the emitted acoustic signal with the smoothed coefficients. Properties of the estimated echo and of the microphone signal are estimated. The echo cancellation filter is controlled as a function of a comparison between the properties of the estimated echo and those of the microphone signal so as to take into account the potential presence of a signal other than an echo signal in the microphone signal.

Control of echo cancellation filters

The invention concerns a method for three-dimensional spatialization of audio channels from a filter BRIR incorporating a theater effect. For a specific number N of samples corresponding to the size of the pulse response of the BRIR filter, it consists in breaking down (A) the BRIR filter into at least a set of delay and amplitude values associated with the times of arrival of reflections; extracting (B) on the number of B samples at least one spectral module of the BRIR filter; and constituting (C) from each successive delay, its amplitude and its spectral module associated with an elementary BRIR filter (BRIR) directly applied to the audio channels in the time, frequency or transformed domain. The invention is applicable to binaural or multichannel spatialization.

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Rennes, France

Alexandre Guerin