Classifications

• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
  • G10L19/008—Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing

Definitions

• the present invention relates to the field of coding / decoding multi-channel digital audio signals. More particularly, the present invention relates to the parametric encoding / decoding of multi-channel audio signals.
• This type of coding / decoding is based on the extraction of spatialization parameters so that at decoding, the spatial perception of the listener can be reconstituted.
• Such a coding technique is known as "Binaural Cue
• Coding in English (BCC) which aims on the one hand to extract and then code the auditory spatialization indices and on the other hand to code a monophonic or stereophonic signal from a mastering of the original multichannel signal.
• This parametric approach is a low rate coding.
• the main advantage of this coding approach is to allow a better compression ratio than conventional multi-channel digital audio compression methods while ensuring the backward compatibility of the compressed format obtained with existing coding formats and broadcasting systems.
• FIG. 1 describes such a coding / decoding system in which the coder 100 constructs a sum signal ("downmix" in English) S s by matrixing in
• the 110 channels of the original multichannel signal S and provides via a parameter extraction module 120, a reduced set of parameters P which characterize the spatial content of the original multichannel signal.
• the multichannel signal is reconstructed (S ') by a synthesis module 160 which takes into account both the sum signal and the transmitted parameters P.
• the sum signal has a reduced number of channels. These channels can be encoded by a conventional audio encoder before transmission or storage.
• the sum signal has two channels and is compatible with conventional stereo broadcasting. Before transmission or storage, this sum signal can thus be encoded by any conventional stereo encoder. The signal thus coded is then compatible with the devices comprising the corresponding decoder which reconstruct the sum signal while ignoring the spatial data.
• This coding scheme is based on a tree structure that allows the processing of only a limited number of channels simultaneously.
• this technique is satisfactory for coding and decoding signals of reduced complexity used in the audiovisual field, for example for 5.1 signals.
• it does not make it possible to obtain a satisfactory quality for more complex multichannel signals, for example for signals originating from direct multichannel sound taps or even surround signals.
• the present invention improves the situation.
• the method proposes a method of encoding a multichannel audio signal representing a sound scene comprising a plurality of sound sources.
• the method is such that it comprises a step of decomposing the multichannel signal into frequency bands and the following steps per frequency band:
• the information being representative of the spatial distribution of the sound source in the sound scene
• the source-related directivity information gives not only the direction of the source but also the shape, or spatial distribution, of the source, ie the interaction that this source can have with the others. sources of the sound stage.
• Knowing this directivity information associated with the sum signal will allow the decoder to obtain a signal of better quality which takes into account interchannel redundancies in a global manner and the probable phase oppositions between channels.
• the sum signal from the coding according to the invention can be decoded by a standard decoder as known in the state of the art, thus providing interoperability with existing decoders.
• the method further comprises a step of encoding secondary sources among the unselected sources of the sound scene and inserting coding information of the secondary sources in the bit stream.
• the coding of the secondary sources will thus make it possible to provide additional precision on the decoded signal, in particular for complex signals of the type, for example, ambiophonic ones.
• the coding information of the secondary sources may be, for example, coded spectral envelopes or coded time envelopes which may constitute parametric representations of the secondary sources.
• the coding of secondary sources comprises the following steps:
• pseudo-sources representing at least part of the secondary sources, by decorrelation with at least one main source and / or at least one coded secondary source;
• part of the secondary sources or diffuse sources can then be represented by pseudo-sources. In this case, it is then possible to code this representation without increasing the coding rate.
• the coding of the directivity information is performed by a parametric representation method.
• This method is of low complexity and adapts particularly to the case of synthetic sound stage representing an ideal coding situation.
• These parametric representations may comprise, for example, arrival direction information, for the reconstruction of a simulating directivity a plane wave or directivity pattern selection indicia in a dictionary of directivity shapes.
• the coding of the directivity information is performed by a principal component analysis method delivering basic directivity vectors associated with gains allowing the reconstruction of the initial directivities.
• the coding of the directivity information is performed by a combination of a principal component analysis method and a parametric representation method.
• the present invention also relates to a method for decoding a multichannel audio signal representing a sound scene comprising a plurality of sound sources, from a bit stream and a sum signal.
• the method is such that it comprises the following steps:
• the decoding method thus makes it possible to reconstruct the multichannel signal of high quality for a faithful reproduction of the spatialized sound taking into account interchannel redundancies globally and the probable phase oppositions between channels.
• the method further comprises the following steps:
• the method further comprises the following step:
• decoding the secondary sources by using a source actually transmitted and a predefined decorrelator to reconstruct pseudosources representative of at least a part of the secondary sources.
• the method further comprises the following steps:
• decoding secondary sources by using the source and the decorrelator index to reconstruct pseudo-sources representative of at least a part of the secondary sources. This makes it possible to find pseudo-sources representing part of the original secondary sources without degrading the sound reproduction of the decoded sound scene.
• the present invention also relates to an encoder of a multichannel audio signal representing a sound scene having a plurality of sound sources.
• the encoder is such that it comprises:
• a directivity information obtaining module capable of obtaining this information by sound source of the sound scene and by frequency band, the information being representative of the spatial distribution of the sound source in the sound scene;
• a coding module for the directivity information and a module for forming a bit stream comprising the coded directional information, the bit stream being able to be transmitted parallel to the sum signal.
• This decoder is such that it comprises:
• a means of storage readable by a computer or a processor, whether or not integrated into the encoder, possibly removable, stores a computer program implementing an encoding method and / or a decoding method according to the invention.
• FIG. 1 illustrates a coding / decoding system of the state of the art of standard MPEG Surround system type
• FIG. 2 illustrates an encoder and a coding method according to one embodiment of the invention
• FIG. 3a illustrates a first embodiment of the coding of the directivities according to the invention
• FIG. 3b illustrates a second embodiment of the coding of the directivities according to the invention
• FIG. 4 represents examples of directivities used by the invention
• FIG. 5 illustrates a decoder and a decoding method according to one embodiment of the invention
• FIG. 6 represents an alternative embodiment of an encoder and a coding method according to the invention.
• FIG. 7 represents an alternative embodiment of a decoder and a decoding method according to the invention.
• FIGS. 8a and 8b respectively represent an exemplary device comprising an encoder and an exemplary device comprising a decoder according to the invention.
• FIG. 2 illustrates in the form of a block diagram, an encoder according to one embodiment of the invention as well as the steps of a coding method according to one embodiment of the invention.
• the encoder thus illustrated comprises a time-frequency transform module 210 which receives as input an original multichannel signal representing a sound scene comprising a plurality of sound sources.
• This module therefore performs a step T of calculating the time-frequency transform of the original multichannel signal.
• This transform is carried out for example by a short-term Fourier transform.
• each of the n x channels of the original signal is window on the current time frame, then the Fourier transform F of the window signal is calculated using a fast calculation algorithm on ⁇ FFT points.
• ⁇ FFT X n x containing the coefficients of the original multichannel signal in the frequency space.
• the subsequent processing by the encoder is done by frequency band. This is done by cutting the matrix of coefficients X into a set of sub-matrices X j each containing the frequency coefficients in the j th band.
• the signal is thus obtained for a given frequency band Sg.
• a module for obtaining directional information 220 makes it possible to determine, by an OBT step, on the one hand, the directivities associated with each of the sources of the sound scene and, on the other hand, to determine the sources of the sound scene for the given frequency band.
• the directivities are vectors of the same dimension as the number n s of channels of the multichannel signal S m .
• Each source is associated with a vector of directivity.
• the directivity vector associated with a source corresponds to the weighting function to be applied to this source before playing it on a loudspeaker, so as to best reproduce a direction of arrival and a source width . It is easily understood that for a very large number of regularly spaced loudspeakers, the directivity vector will faithfully represent the radiation of a sound source.
• the vector of directivity will be obtained by the application of an inverse spherical Fourier transform on the components of the ambiophonic orders.
• the ambiophonic signals correspond to a decomposition into spherical harmonics, hence the direct correspondence with the directivity of the sources.
• the set of directivity vectors therefore constitutes a large amount of data that would be too expensive to transmit directly for low coding rate applications.
• two methods of representing the directivities can for example be used.
• the Cod.Di coding module 230 for directivity information can thus implement one of the two methods described below or a combination of the two methods.
• a first method is a parametric modeling method that makes it possible to exploit knowledge a priori on the signal format used. It consists of transmitting only a very small number of parameters and reconstructing the directivities according to known coding schemes.
• the associated directivity is known as a function of the direction of arrival of the sound source.
• a search for peaks in the directivity diagram (by analogy with sinusoidal analysis, as explained for example in the document "Computer modeling of musical sound (analysis, transformation, synthesis)" by Sylvain Marchand, PhD thesis, University Bordeaux 1 , allows to detect the direction of arrival relatively faithfully.Other methods like the "matching pursuit", as presented in S.
• a parametric representation can also use a simple form dictionary to represent the directivities.
• FIG. 4 gives some simple forms of directivity (in azimuth) that can be used.
• directivity in azimuth
• FIG. 4 gives some simple forms of directivity (in azimuth) that can be used.
• the directivities one associates with an element of the dictionary, the corresponding azimuth and a gain allowing to play on the amplitude of this vector of directivity of the dictionary.
• One can thus, from a dictionary of form of directivity, to deduce the best form or combination of forms that will best reconstruct the initial directivity.
• the directivity coding module 230 comprises a parametric modeling module which outputs P directionality parameters. These parameters are then quantized by the quantization module 240.
• This first method makes it possible to obtain a very good level of compression when the scene corresponds to an ideal coding. This will be particularly the case on synthetic soundtracks. However, for complex scenes or microphonic sound, it is necessary to use more generic coding models, involving the transmission of a greater amount of information.
• the representation of the directivity information is in the form of a linear combination of a limited number of basic directivities.
• This method is based on the fact that the set of directivities at a given moment generally has a reduced dimension. Indeed, only a small number of sources is active at a given moment and the directivity for each source varies little with the frequency. It is thus possible to represent all the directivities in a group of frequency bands from a very small number of well-chosen basic directivities.
• the parameters transmitted are then the basic directivity vectors for the group of bands considered, and for each directivity to be coded, the coefficients to be applied to the basic directivities to reconstruct the directivity considered.
• PCA Principal component analysis
• LT. Jolliffe in "Principal Component Analysis", Springer, 2002.
• the application of the principal component analysis to the coding of the directivities is carried out as follows: firstly, a matrix of the initial directivities Di is formed, whose number of rows corresponds to the total number of sources of the sound stage, and the number of columns is the number of channels of the original multichannel. Then, one carries out properly the principal component analysis which corresponds to the diagonalization of the covariance matrix, and which gives the matrix of the eigenvectors. Finally, we select the eigenvectors carrying the most important information and corresponding to the eigenvalues of higher value.
• the number of eigenvectors to keep may be fixed or variable in time depending on the available flow.
• This new base thus gives the matrix D B T.
• the representation of the directivities is therefore made from basic directivity.
• the matrix of directivities Di is written as the linear combination of these basic directivities.
• Di G D D B
• D B is the matrix of the basic directivities for all the bands
• G D the matrix of the associated gains.
• the number of rows of this matrix represents the total number of sources of the sound stage and the number of columns represents the number of basic directivity vectors.
• basic directivities are sent by group of considered bands, in order to more accurately represent the directivities. It is possible, for example, to provide two basic directivity groups: one for low frequencies and one for high frequencies. The limit between these two groups can for example be chosen between 5 and 7 kHz.
• the coding module 230 comprises a main component analysis module delivering basic directivity vectors D B and associated coefficients or gain vectors G D -
• FIG. 3a illustrates in a detailed manner, the directivity coding block 230, in a first variant embodiment.
• This coding mode uses the two diagrams of representation of the directivities.
• a module 310 performs parametric modeling as previously explained to provide directional parameters (P).
• a module 320 performs principal component analysis to provide both basic directivity vectors (D B ) and associated coefficients (G D ).
• a selection module 330 selects frequency band per frequency band, the best mode of coding for the directivity by choosing the best compromise reconstruction of the directivities / flow.
• the choice of the representation chosen (parametric representation or by linear combination of basic directivities) is done in order to optimize the efficiency of the compression.
• a selection criterion is, for example, the minimization of the mean squared error.
• a perceptual weighting may possibly be used for the choice of the directivity coding mode. This weighting is intended for example to promote the reconstruction of the directivities in the frontal area, for which the ear is more sensitive.
• the error function to be minimized in the case of the ACP encoding model can be in the following form:
• the directivity parameters from the selection module are then quantized by a step Q by the quantization module 240 of FIG. 2.
• the two coding modes are cascades.
• Figure 3b illustrates in detail this block of coding.
• a parametric modeling module 340 performs a modeling for a certain number of directivities and outputs at the same time directivity parameters (P) for the modeled directivities and unmodelled directivities or residual directivities DiR .
• D residual directivities
• main component analysis module 350 which outputs basic directional vectors (D B ) and associated coefficients (G D ).
• D B basic directional vectors
• G D associated coefficients
• the directivity parameters, the basic directivity vectors as well as the coefficients are provided at the input of the quantization module 240 of FIG. 2.
• Quantization Q is performed by reducing the accuracy as a function of perception data and then applying entropy coding. Also, the possibility of exploiting the redundancy between frequency bands or between successive frames can reduce the flow. Intra-frame or inter-frame predictions on the parameters can therefore be used. In general, the standard methods of quantification can be used. On the other hand, the vectors to be quantified being orthonormed, this property can be exploited during the scalar quantization of the components of the vector. Indeed, for a vector of dimension N, only N-I components will have to be quantified, the last component being able to be recalculated.
• a construction module of a bit stream 250 inserts this coded direction information in a bit stream Fb according to the step Con.Fb.
• the encoder as described here further comprises a selection module 260 able to select in the step Select main sources (S p ⁇ n c ) among the sources of the sound scene to be encoded (S tot ).
• a particular embodiment uses a principal component analysis method, ACP, in each frequency band in block 220 to extract all the sources of the sound scene (S tot ).
• ACP principal component analysis method
• the sources of greater importance are then selected by the module 260 to constitute the main sources (S p ⁇ nc ), which are then stamped in step M by the module 270 to construct a sum signal (S Sf ,) (or "downmix" in English).
• the number of main sources (S pr j nc ) is chosen according to the number of channels of the sum signal. This number is chosen less than or equal to the number of channels. Preferably, a number of main sources is chosen equal to the number of channels of the sum signal.
• the matrix M is then a predefined square matrix.
• This sum signal per frequency band undergoes an inverse time-frequency transformation T 1 by the inverse transform module 290 in order to provide a time sum signal (S s ).
• This sum signal is then encoded by a speech coder or an audio coder of the state of the art (for example: G.729.1 or MPEG-4 AAC).
• Secondary sources (S sec ) can be coded by a coding module
• bit stream construction module 250 280 and added to the bit stream in the bit stream construction module 250.
• the coding module 280 which may in one embodiment be a short-term Fourier transform coding module. These sources can then be separately encoded using the aforementioned audio or speech coders.
• the coefficients of the transform of these secondary sources can be coded directly only in the bands considered to be important.
• Secondary sources can be encoded by parametric representations; these representations may be in the form of spectral envelope or temporal envelope.
• This method of encoding a multichannel signal as described is particularly interesting in that the analysis is made on windows that can be of short length.
• this coding model causes a low algorithmic delay allowing its use in applications where control of the delay is important.
• the encoder as described implements an additional preprocessing step P by a preprocessing module 215.
• This module performs a basic changeover step in order to express the sound scene using the flat wave decomposition of the acoustic field.
• the original surround signal is seen as the angular Fourier transform of a sound field.
• the first plane wave decomposition operation therefore corresponds to taking the omnidirectional component of the ambiophonic signal as representing the zero angular frequency (this component is therefore a real component).
• the following surround components order 1, 2, 3, etc. are combined to obtain the complex coefficients of the angular Fourier transform.
• the first component represents the real part
• the second component represents the imaginary part.
• O For a two-dimensional representation, for an order O, we obtain O + 1 complex components.
• a Short Term Fourier Transform (on the time dimension) is then applied to obtain the Fourier transforms (in the frequency domain) of each angular harmonic. This step then integrates the transformation step T of the module 210. the complete angular transform by recreating the harmonics of negative frequencies by Hermitian symmetry.
• an inverse Fourier transform is carried out on the dimension of the angular frequencies to pass in the domain of the directivities.
• This pre-processing step allows the coder to work in a signal space whose physical and perceptual interpretation is simplified, which makes it possible to more effectively exploit knowledge of spatial auditory perception and thus improve coding performance.
• the encoding of the surround signals remains possible without this pre-processing step. For non-surround signals, this step is not necessary. For these signals, the knowledge of the recording system or reproduction associated with the signal makes it possible to directly interpret the signals as a plane wave decomposition of the acoustic field.
• Figure 5 now describes a decoder and a decoding method in one embodiment of the invention.
• This decoder receives as input the bit stream F b as constructed by the encoder described above as well as the sum signal S 5 .
• the first decoding step consists of realizing the time-frequency transform T of the sum signal S s by the transform module 510 to obtain a sum signal per frequency band, S sf,.
• This transform is carried out using, for example, the short-term Fourier transform. It should be noted that other transforms or filterbanks may also be used, including non-uniform filterbanks according to a perception scale (e.g. Bark). It may be noted that in order to avoid discontinuities during the reconstruction of the signal from this transform, a recovery addition method is used. For the time frame considered, the step of calculating the transform of
• Fourier in the short term is to window each of the n f channels of the sum signal S s using a window w of length greater than the time frame, and then to calculate the Fourier transform of the window signal with the help of a fast calculation algorithm on npFT points.
• a complex matrix F of size npF T xn f containing the coefficients of the sum signal in the frequency space is thus obtained.
• the entire processing is done in frequency bands.
• the matrix coefficients F is cut into a plurality of submatrices F j each containing the frequency coefficients in the j th band.
• Different choices for the frequency division of the bands are possible.
• symmetrical bands with respect to the zero frequency in the Fourier transform are chosen in the short term.
• the description of the decoding steps performed by the decoder will be made for a given frequency band. The steps are of course carried out for each of the frequency bands to be processed.
• the module 520 performs a dematrix N of the frequency coefficients of the signal transform sum of the frequency band considered so as to find the main sources of the sound scene. More specifically, the S p ⁇ ⁇ c matrix of frequency coefficients for the current frequency band of the n p ⁇ nc main sources is obtained according to the relation:
• N is of dimension n f xn p ⁇ nc and B is a matrix of dimension n bm xn f where n bm is the number of components (or bins) frequency retained in the frequency band considered.
• N I.
• the number of rows of the matrix N corresponds to the number of channels of the sum signal, and the number of columns corresponds to the number of main sources transmitted.
• the dimensions are inverted, I being an identity matrix of dimensions n p ⁇ nc xn p ⁇ nc .
• the lines of B are the frequency components in the current frequency band, the columns correspond to the channels of the sum signal.
• the lines of S p ⁇ n c are the frequency components in the current frequency band, and each column corresponds to a main source.
• the number of main sources n p ⁇ nc is preferably less than or equal to the number n f of channels of the sum signal to ensure that the operation is invertible, and may possibly be different for each frequency band.
• the number of sources to be reconstructed in the current frequency band to obtain a satisfactory reconstruction of the scene is greater than the number of channels of the sum signal.
• additional or secondary sources are coded and then decoded from the bitstream for the current band by the module 550 for decoding the bitstream.
• This decoding module decodes the information contained in the bit stream and in particular, the directional information and, where appropriate, the secondary sources.
• the decoding of the secondary sources is carried out by the inverse operations that those which were carried out with the coding. Whatever the coding method that has been chosen for the secondary sources, if reconstruction data or coding information of the secondary sources has been transmitted in the bit stream for the current band, the corresponding data is decoded to reconstruct the dry matrix S frequency coefficients in the current band of the n sec secondary sources.
• the shape of the dry matrix S is similar to the matrix S pnnc , that is, the lines are the frequency components in the current frequency band, and each column corresponds to a secondary source.
• Ssupp according to S relation (S pntlL S mpp j therefore S is a matrix of dimension n b, n tot xn - Also, the form is identical to S p ⁇ matrices n c and S supp: lines are the frequency components in the current frequency band, each column is a source, with n tot sources in total.
• the directivity information is extracted from the bit stream at the Decod step. Fb by the module 550.
• the possible outputs of this decoding module of the bitstream depend on the coding methods of the directivities used in the coding. They can be in the form of basic directivity vectors D B and associated coefficients G D and / or modeling parameters P.
• This data is then transmitted to a directional information reconstruction module 560 which performs the decoding of the directional information by reverse operations from those performed in the coding.
• the number of directivities to be reconstructed is equal to the number n tot of sources in the frequency band considered, each source being associated with a directional vector.
• the matrix of directivities Di is written as the linear combination of these basic directivities.
• Di G D D B
• D B is the matrix of the basic directivities for all the bands
• G D the matrix of the associated gains.
• This gain matrix has a number of lines equal to the total number of sources n tot , and a number of columns equal to the number of basic directivity vectors.
• basic directivities are decoded by group of frequency bands considered, in order to more accurately represent the directivities.
• group of frequency bands considered in order to more accurately represent the directivities.
• a vector of gains associated with the basic directivities is then decoded for each band.
• Y SD T , where Y is the reconstructed signal in the band.
• the rows of the matrix Y are the frequency components in the current frequency band, and each column corresponds to a channel of the multichannel signal to be reconstructed.
• the corresponding time signals are then obtained by inverse Fourier transform T ', using a fast algorithm implemented by the inverse transform module 540. This gives the multichannel signal S m on the current time frame.
• the different time frames are then combined by conventional overlap-add (or overlap-add) method to reconstruct the complete multichannel signal.
• temporal or frequency smoothing of the parameters can be used both for analysis and synthesis to ensure smooth transitions in the sound scene.
• a sign of sudden change of the sound stage may be reserved in the bit stream to avoid smoothing the decoder in the case of detection of a rapid change in the composition of the sound stage.
• conventional methods of adapting the resolution of the time-frequency analysis can be used (change in the size of the analysis and synthesis windows over time).
• a base change module can perform a pre-processing P "1 to obtain a plane wave decomposition of the signals, a base change module 570 performs the reverse operation from the signals. in plane waves to find the original multichannel signal.
• the coding of the embodiment described with reference to FIG. 2 makes it possible to obtain efficient compression when the complexity of the scene remains limited.
• the complexity of the scene is greater, ie when the scene contains a high number of active sources in a frequency band, or important diffuse components, a large number of sources and associated directivity becomes necessary for to obtain a good quality of restitution of the scene. The effectiveness of compression is then reduced.
• the encoder as represented in FIG. 6 comprises the modules 215, 210, 220, 230, 240 as described with reference to FIG. 2.
• This encoder also includes the modules 260, 270 and 290 as described with reference to FIG.
• This encoder comprises a coding module of the secondary sources 620, which differs from the module 280 of Figure 2 in the case where the number of secondary sources is important.
• this coding module 620 a method of parametric coding secondary sources is implemented by this coding module 620.
• the field is perceptibly comparable to a diffuse field, and the representation of the field by one or more statistical characteristics of the field is sufficient to reconstruct a perceptually equivalent field.
• the spatially diffuse components of the sound scene can be perceptively reconstructed from the simple knowledge of the corresponding directivity, and by controlling the coherence of the created field. This can be done by using pseudo-sources constructed by decorrelation, from a limited number of transmitted sources and by using the diffuse component directivity estimated on the original multichannel signal. The objective is then to reconstruct a sound field statistically and perceptually equivalent to the original, even if it consists of signals whose waveforms are different.
• a number of secondary sources are not transmitted and are replaced by pseudo-sources obtained by decorrelation of the transmitted sources, or by any other artificial source decorrelated sources transmitted. This avoids the transmission of data corresponding to these sources and significantly improves coding efficiency.
• a source to be transmitted to the decoder and a predefined decorrelator known from both the encoder and the decoder, to be applied to the transmitted source to select pseudo-sources for the decoder, are chosen.
• a parametric representation of the secondary sources is obtained by the coding module of the secondary sources 620 and is also transmitted to the construction module of the bitstream.
• This parametric representation of secondary sources or diffuse sources is effected for example by a spectral envelope.
• a time envelope can also be used.
• the pseudo-sources are calculated by a decorrelation module 630 which calculates the decorrelated sources from at least one main source or with at least one coded secondary source to be transmitted.
• decorrelators and several initial sources can be used, and one can select the initial source associated with a type of decorrelator giving the best reconstruction result.
• These decorrelation data such as the index of the correlator used and the choice data of the initial source as the index of the source, are then transmitted to the building module of the bit stream to be inserted. The number of sources to transmit is reduced while maintaining a good perceptive quality of the reconstructed signal.
• FIG. 7 represents a decoder and a decoding method adapted to the coding according to the variant embodiment described in FIG. 6.
• This decoder comprises the modules 510, 520, 530, 540, 570, 560 as described with reference to FIG. 5. This decoder differs from that described in FIG. information decoded by the decoding module of the bit stream 720 and the decorrelation calculation block 710.
• the module 720 obtains, in addition to directional information from sources of the sound scene and, if appropriate, decoded secondary sources, parametric representation data of certain secondary sources or diffuse sources and possibly information on the decorrelator and the transmitted sources. to use to rebuild the pseudo-sources.
• the latter information is then used by the decorrelation module 710 which makes it possible to reconstruct the secondary pseudo-sources which will be combined with the main sources and the other potential secondary sources in the spatialization module as described with reference to FIG.
• the encoders and decoders as described with reference to FIGS. 2, 6 and 5, 7 can be integrated into a multimedia equipment of the living room decoder type, a computer or even communication equipment such as a mobile telephone or personal electronic organizer.
• FIG. 8a represents an example of such multimedia equipment or coding device comprising an encoder according to the invention.
• This device comprises a PROC processor cooperating with a memory block BM having a storage and / or working memory MEM.
• the memory block may advantageously comprise a computer program comprising code instructions for implementing the steps of the coding method within the meaning of the invention, when these instructions are executed by the processor PROC, and in particular the steps of
• the information being representative of the spatial distribution of the sound source in the sound scene
• bit stream comprising the coded directional information
• the description of FIG. 2 repeats the steps of an algorithm of such a computer program.
• the computer program can also be stored on a memory medium readable by a reader of the device or downloadable in the memory space of the equipment.
• the device comprises an input module adapted to receive a multichannel signal representing a sound scene, either by a communication network, or by reading a content stored on a storage medium.
• This multimedia equipment may also include means for capturing such a multichannel signal.
• the device comprises an output module able to transmit a bit stream
• FIG. 8b illustrates an example of multimedia equipment or decoding device comprising a decoder according to the invention.
• This device comprises a PROC processor cooperating with a memory block BM having a storage and / or working memory MEM.
• the memory block may advantageously comprise a computer program comprising code instructions for implementing the steps of the decoding method in the sense of the invention, when these instructions are executed by the processor PROC, and in particular the steps of: - extraction in the bitstream and decoding information of directivities representative of the spatial distribution of the sources in the sound scene;
• the computer program can also be stored on a memory medium readable by a reader of the device or downloadable in the memory space of the equipment.
• the device comprises an input module adapted to receive a bit stream Fb and a sum signal S s coming for example from a communication network. These input signals can come from a reading on a storage medium.
• the device comprises an output module capable of transmitting a multichannel signal decoded by the decoding method implemented by the equipment.
• This multimedia equipment may also include speaker-type reproduction means or communication means capable of transmitting this multi-channel signal.
• Such multimedia equipment may include both the encoder and the decoder according to the invention.
• the input signal then being the original multichannel signal and the output signal, the decoded multichannel signal.

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• 2009
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