Abstract: Methods and devices for a low complex inter-channel difference estimation are provided. A method for the estimation of inter-channel differences (ICDs), comprises applying a transformation from a time domain to a frequency domain to a plurality of audio channel signals, calculating a plurality of ICD values for the ICDs between at least one of the plurality of audio channel signals and a reference audio channel signal over a predetermined frequency range, each ICD value being calculated over a portion of the predetermined frequency range, calculating, for each of the plurality of ICD values, a weighted ICD value by multiplying each of the plurality of ICD values with a corresponding frequency-dependent weighting factor, and calculating an ICD range value for the predetermined frequency range by adding the plurality of weighted ICD values.

Abstract: The invention relates to an audio signal processing apparatus for processing an audio signal, the audio signal processing apparatus comprising: a converter configured to convert a stereo audio signal into a binaural audio signal; and a determiner configured to determine upon the basis of an indicator signal whether the audio signal is a stereo audio signal or a binaural audio signal, the indicator signal indicating whether the audio signal is a stereo audio signal or a binaural audio signal, the determiner being further configured to provide the audio signal to the converter if the audio signal is a stereo audio signal.

Abstract: The invention relates to a method for rendering a stereo audio signal over a first loudspeaker and a second loudspeaker with respect to a desired direction, the stereo audio signal comprising a first audio signal component (L) and a second audio signal component (R), the method comprising: providing a first rendering signal based on a combination of Land a first difference signal obtained based on a difference between L and R to the first loudspeaker, and providing a second rendering signal based on a combination of R and a second difference signal obtained based on the difference between L and R to the second loudspeaker, such that both difference signals are different with respect to sign and one difference signal is delayed by a delay compared to the other difference signal to define a dipole signal, wherein the delay is adapted according to the desired direction.

Abstract: The invention relates to a portable electronic device, comprising: a housing comprising at least one hole; and at least two directional microphones mounted in the housing and placed coincidentally for stereo sound pickup, each one of the microphones defining a main sound axis and each one of the two directional microphones defining a direct sound direction and a opposite sound direction which describe opposite directions of the main sound axis, wherein the at least one hole is a common hole shared between the at least two directional microphones such that the main sound axis of the at least two directional microphones are pointing through the common hole in different directions.

Abstract: Methods and devices for a low complex inter-channel difference estimation are provided. A method for the estimation of inter-channel differences (ICDs), comprises applying a transformation from a time domain to a frequency domain to a plurality of audio channel signals, calculating a plurality of ICD values for the ICDs between at least one of the plurality of audio channel signals and a reference audio channel signal over a predetermined frequency range, each ICD value being calculated over a portion of the predetermined frequency range, calculating, for each of the plurality of ICD values, a weighted ICD value by multiplying each of the plurality of ICD values with a corresponding frequency-dependent weighting factor, and calculating an ICD range value for the predetermined frequency range by adding the plurality of weighted ICD values.

Abstract: An audio rendering system is provided that comprises a plurality of loudspeakers arranged to approximate a desired spatial sound field within a predetermined reproduction region, wherein the loudspeakers are configured to approximate the sound field based on a weighted series of orthonormal basis functions for the reproduction region.

Abstract: A method for reconstructing at least one target signal comprises determining a first set of feature vectors from the input signal, the first set of feature vectors forming a non-negative input matrix; determining a second set of feature vectors, the second set of feature vectors forming a non-negative noise matrix; decomposing the input matrix into a sum of a first matrix and a second matrix, the first matrix representing a product of a non-negative bases matrix and a non-negative weight matrix, and the second matrix representing a combination of the noise matrix and a noise weight vector; and reconstructing the at least one target signal based on the non-negative bases matrix and the non-negative weight matrix.

Abstract: The invention relates to an audio signal synthesizer, the audio signal synthesizer comprises a transformer for transforming the down-mix audio signal into frequency domain to obtain a transformed audio signal; a signal generator for generating a first auxiliary signal, for generating a second auxiliary signal, and for generating a third auxiliary signal upon the basis of the transformed audio signal; a de-correlator for generating a first de-correlated signal, and for generating a second de-correlated signal from the third auxiliary signal, the first de-correlated signal and the second de-correlated signal being at least partly de-correlated; and a combiner for combining the first auxiliary signal with the first de-correlated signal to obtain a first audio signal, and for combining the second auxiliary signal with the second de-correlated signal to obtain the second audio signal, the first audio signal and the second audio signal forming the multi-channel audio signal.

Abstract: The invention relates to a portable electronic device, comprising: at least two directional microphones for stereo sound pickup, each one of the two directional microphones defining a direct sound direction and an opposite sound direction towards which the directional microphones are directed; and a housing comprising for each of the directional microphones a first hole and a second hole, the first hole being located at a different side of the portable electronic device than the second hole.

Abstract: The invention relates to a portable electronic device, comprising: a housing comprising at least one hole; and at least two directional microphones mounted in the housing and placed coincidentally for stereo sound pickup, each one of the microphones defining a main sound axis and each one of the two directional microphones defining a direct sound direction and a opposite sound direction which describe opposite directions of the main sound axis, wherein the at least one hole is a common hole shared between the at least two directional microphones such that the main sound axis of the at least two directional microphones are pointing through the common hole in different directions.

Abstract: A method for coding a multi-channel audio signal representing a sound scene comprising a plurality of sound sources is provided. This method comprises decomposing the multi-channel signal into frequency bands and, per frequency band, obtaining directivity information per sound source of the sound scene, the information being representative of the spatial distribution of the sound source in the sound scene, of selecting a set of sound sources of the sound scene constituting principal sources, of matrixing the selected principal sources to obtain a sum signal with a reduced number of channels and, of coding the directivity information and of forming a binary stream comprising the coded directivity information, the binary stream being transmittable in parallel with the sum signal. A decoding method is also provided that is able to decode the sum signal and the directivity information to obtain a multi-channel signal, to an adapted coder and an adapted decoder.

Abstract: A method of binary allocation in an enhancement coding/decoding for improving a hierarchical coding/decoding of digital audio signals, including a core coding/decoding in a first frequency band and a band extension coding/decoding in a second frequency band. For a predetermined number of bits to be allocated for the enhancement coding/decoding, a first number of bits is allocated to a coding/decoding for correcting the core coding/decoding in the first frequency band and according to a first mode of coding/decoding and a second number of bits is allocated to an enhancement coding/decoding for improving the extension coding/decoding in the second frequency band and according to a second mode of coding/decoding. Also provided are an allocation module implementing the method and a coder and decoder including this module.

Abstract: In accordance with an embodiment, a method of generating an encoded audio signal, the method includes estimating a time-frequency energy of an input audio signal from a time-frequency filter bank, computing a global variance of the time-frequency energy, determining a post-processing method according to the global variance, and transmitting an encoded representation of the input audio signal along with an indication of the determined post-processing method.

Abstract: A method for encoding and decoding a digital audio signal is provided, said method comprising the steps of: encoding a first sequence of samples of the digital signal according to a transform encoding; encoding a second sequence of samples of the digital signal according to a predictive encoding; wherein the second sequence starts before the end of the first sequence, a subsequence common to the first and second sequences being thus encoded both by predictive encoding and by transform encoding.

Abstract: 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 (hL,L, hL,R) 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.

Abstract: A method for processing sound data is provided for the reconstruction of multi-channel audio data on the basis at least of data on a reduced number of channels and of spatialization data. A test is carried out to determine whether the spatialization data received are valid. If the test is positive, a spatialization value is predicted according to a per respective model of a plurality of models. A prediction model is chosen on the basis of the spatialization values thus predicted and on the basis of the spatialization data received, to permit, in case of subsequent reception of defective spatialization data, a prediction according to this chosen model of a spatialization value and to use this predicted spatialization value for the reconstruction of the multi-channel audio data.

Abstract: An apparatus and a method for encoding an input signal applied to the apparatus comprising: a transient detector adapted to detect whether the applied input signal comprises a transient; at least two transient signal encoders adapted to encode the applied input signal if a transient is detected by the transient detector; and a selection unit adapted to select a transient signal encoder among the transient signal encoders according to at least one predetermined selection criterion.

Abstract: A method is provided for coding a multi-channel audio signal representing a sound scene comprising a plurality of sound sources. The method comprises decomposing the multi-channel signal into frequency bands and the following performed per frequency band: obtaining data representative of the direction of the sound sources of the sound scene, selecting a set of sound sources constituting principal sources, adapting the data representative of the direction of the selected principal sources, as a function of restitution characteristics of the multi-channel signal, determining a matrix for mixing the principal sources as a function of the adapted data, matrixing the principal sources by the matrix determined so as to obtain a sum signal with a reduced number of channels and coding the data representative of the direction of the sound sources and forming a binary stream comprising the coded data, the binary stream being transmittable in parallel with the sum signal.

Abstract: A method of hierarchical coding of a digital audio frequency input signal into several frequency sub-bands, including a core coding of the input signal according to a first throughput and at least one enhancement coding of higher throughput, of a residual signal. The core coding uses a binary allocation according to an energy criterion. The method includes for the enhancement coding: calculating a frequency-based masking threshold for at least part of the frequency bands processed by the enhancement coding; determining a perceptual importance per frequency sub-band as a function of the masking threshold and as a function of the number of bits allocated for the core coding; binary allocation of bits in the frequency sub-bands processed by the enhancement coding, as a function of the perceptual importance determined; and coding the residual signal according to the bit allocation. Also provided are a decoding method, a coder and a decoder.

Abstract: An embodiment of the present invention provides a method for generating a downmixed signal, including: performing a time-frequency transform on a received left sound channel signal and a received right sound channel signal to obtain a frequency domain signal, and dividing the frequency domain signal into several frequency bands; calculating a sound channel energy ratio and a sound channel phase difference of each frequency band; calculating a phase difference between the downmixed signal and a first sound channel signal in each frequency band according to the sound channel energy ratio and the sound channel phase difference; and calculating a frequency domain downmixed signal according to the left sound channel signal, the right sound channel signal, and the phase difference between the downmixed signal and the first sound channel signal in each frequency band. This method effectively improves quality of stereo encoding and decoding.