Classifications

• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04S—STEREOPHONIC SYSTEMS 
  • H04S1/00—Two-channel systems
  • H04S1/007—Two-channel systems in which the audio signals are in digital form
• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04S—STEREOPHONIC SYSTEMS 
  • H04S3/00—Systems employing more than two channels, e.g. quadraphonic
  • H04S3/008—Systems employing more than two channels, e.g. quadraphonic in which the audio signals are in digital form, i.e. employing more than two discrete digital channels
• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04S—STEREOPHONIC SYSTEMS 
  • H04S3/00—Systems employing more than two channels, e.g. quadraphonic
  • H04S3/02—Systems employing more than two channels, e.g. quadraphonic of the matrix type, i.e. in which input signals are combined algebraically, e.g. after having been phase shifted with respect to each other

Definitions

• the invention relates to the spatialization, so-called 3D rendering, of compressed audio signals.
• Such an operation is for example performed during the decompression of a compressed 3D audio signal for example, represented on a number of channels, to a number of different channels, two for example, to allow the reproduction of the 3D audio effects on a headphones.
• the term "binaural” refers to the reproduction on a stereophonic headphones of a sound signal with nevertheless spatialization effects.
• the invention is however not limited to the aforementioned technique and applies, in particular, to techniques derived from the "binaural”, such as the so-called technical rendering techniques TRANSAURAL ® , that is to say on top of remote speakers.
• TRANSAURAL ® is a registered trademark of COOPER BAUCK CORPORATION.
• Such techniques can then use a "cross-talk cancellation", which consists in canceling the crossed acoustic paths, so that a sound, thus processed and then emitted by the loudspeakers , can be perceived only by one of the two ears of a listener.
• the invention also relates to the transmission and reproduction of multichannel audio signals and their conversion to a rendering device, transducer, imposed by the equipment of a user.
• a rendering device transducer
• This is for example the case for the reproduction of a 5.1 sound stage by an audio headset, or by a pair of loudspeakers.
• the invention also relates to the reproduction, in the context of a game or video recording for example, of one or more samples sound stored in files, with a view to their spatialization.
• the two-channel binaural synthesis consists, with reference to FIG. 1a, in filtering the signal of the different sound sources. If it is desired to position, at the restitution, at a position in space, via functions acoustic transfer signals HRTF-I and RHTF-r right in the frequency domain corresponding to the appropriate direction, defined in polar coordinates [O 1 , C 1 ).
• the transfer functions HRTF for "Head Related Transfer Functions" in English, are the acoustic transfer functions of the head of the listener between the positions of the space and the auditory canal.
• HRIR for "Head Related Impulse Response” is referred to as their temporal form. These functions may further include a room effect. We obtain, for each sound source If two left and right signals which are then added to the left and right signals from the spatialization of other sound sources, to finally give the L and R signals broadcast to the left and right ears of the listener.
• the number of necessary filters or transfer functions is then 2.N for a static binaural synthesis and 4.N for a dynamic binaural synthesis, N designating the number of sound sources or audio streams to be spatialized.
• the binaural filter implementation is generally in the form of two minimal phase filters and a pure delay, corresponding to the difference of the left and right delays applied to the ear furthest away from the source. This delay is usually implemented using a delay line.
• the minimum phase filter is a finite impulse response filter and can be executed in the time or frequency domain. Infinite impulse response filters can be searched to approximate the minimum phase ⁇ RTF filter module.
• FIG. 1b in the non-limiting context of a 5.1 spatialized sound scene, with a view to restoring it to the headphones of a human HB.
• the sound emanating from the loudspeaker Lf affects the left ear LE through an HRTF filter A but this same sound reaches the right ear RE modified by a HRTF filter B.
• the position of the speakers relative to the aforementioned HB individual may be symmetrical or not.
• Each ear therefore receives the contribution of the 5 loudspeakers in the form modeled below:
• Bl ALf + CC + BRf + DSI + ESr
• Bl is the binauralized signal for the left ear LE
• Br is the binauralized signal for the right ear RE.
• the filters A, B, C, D and E are modeled, most often, by linear digital filters and it is therefore necessary, in the configuration shown in FIG. 1b, to have 10 filtering functions to be applied, which can be reduced to 5. , considering the symmetries.
• the aforementioned filtering operations can be performed in the frequency domain, for example by virtue of a fast convolution performed in the Fourier domain.
• a Fast Fourier Transform (FFT) Fourier Transform is then used to perform binauralization effectively.
• the HRTF filters A, B, C, D and E can be simplified as a frequency equalizer and a delay.
• the HRTF filter A can be realized as a simple equalizer, since it is a direct path, while the HRTF filter B includes an additional delay.
• the HRTF filters can be decomposed into a minimum phase filter and a pure delay. The delay for the ear closest to the source can be taken as zero.
• the reconstruction operation by spatial decoding of a 3D audio sound scene, from a reduced number of transmitted channels, as represented in FIG. 1c, is also known from the state of the art.
• the configuration represented in FIG. 1c is that relating to the decoding of a coded sound channel having location parameters in the frequency domain, in order to reconstruct a spatialized sound scene 5.1.
• the aforementioned reconstruction is performed by a frequency subband sub-frequency decoder, as represented in FIG. 1c.
• the coded audio signal m undergoes 5 spatialization processing steps, which are controlled by parameters or complex coefficients of spatialization CLD and ICC calculated by the encoder and which, by means of decorrelation operations and gain correction, to realistically reconstruct the sound stage composed of six channels, the five channels represented in FIG. 1b, to which is added a low frequency effect channel Ife.
• each OTT module corresponding to a matrix of decoding coefficients must then be converted into the Fourier domain, at the cost of an approximation, because the operations are not performed in the same domain.
• the complexity is further increased because the synthetic operation "Synth" is followed by three FFT transformations.
• the HRTF filterings are complex to perform because they require the use of subband filters, the minimum length of which is fixed and which must take into account the phenomenon of spectral folding of the subbands.
• the object of the present invention is to overcome the numerous drawbacks of the above-mentioned prior art of sound spatialization of 3 D audio scenes, in particular transauralisation or binauralization of 3 D audio scenes.
• an objective of the present invention is the execution of a specific filtering of spatially coded audio signals or channels in the frequency subband domain of a spatial decoding, in order to limit the number of transformations two by two, while reducing the filtering operations to a minimum, but maintaining a good quality of source spatialization, including transauralisation or binauralization.
• the execution of the aforementioned specific filtering is based on the equalizer-delay form of the spatialization filters, transaural or binaural, for a direct application of filtering by equalization-delay in the domain of the sub-bands.
• Another objective of the present invention is to obtain a 3D rendering quality very close to that obtained from modeling filters such as original HRTF filters, by the sole addition of a transaural spatial processing of very low complexity, following a classical spatial decoding in the transformed domain.
• an objective of the present invention is a new source spatialization technique applicable not only to the transaural or binaural rendering of a monophonic sound, but also to several monophonic sounds and in particular to the multiple channels of 5.1, 6.1, 7.1, 8.1 or 5.1 stereo sounds. higher.
• the subject of the present invention is thus a method for sound spatialisation of an audio scene comprising a first set comprising a number greater than or equal to the unit of audio channels coded spatially over a number of sub-bands of determined frequencies, and decoded in a transformed domain, in a second set comprising a number greater than or equal to two of sound reproduction channels in the time domain, from acoustic propagation modeling filters of the audio signals of the first set of channels.
• this method is remarkable in that, for each modeling filter converted into at least one gain and a delay applicable in the transformed domain, it consists in performing at least, for each sub-band frequency of the transformed domain:
• a filtering by equalization-delay of the signal in sub-band by applying a gain respectively of a delay on the signal in sub-band, to generate from the spatially coded channels, an equalized and delayed component of a value determined in the frequency subband considered,
• the method which is the subject of the invention is also remarkable in that the filtering by equalization-delay of the signal in sub-band includes at least the application of a phase shift and, if appropriate, a pure delay by storage, for the at least one of the frequency sub-bands.
• the method which is the subject of the invention is also remarkable in that it includes filtering by equalization-delay in a hybrid transformed domain, comprising an additional step of frequency cutting into additional subbands, with or without decimation.
• the method which is the subject of the invention is finally remarkable in that to convert each modeling filter into a gain value or a delay value in the transformed domain, it consists at least in associating as a gain value with each subband a real value. defined as the average of the modeling filter module in this sub-band and to associate as delay value with each sub-band a delay value corresponding to the reception delay between the left ear and the right ear for different positions.
• the subject of the present invention is correspondingly to a sound spatialization device of an audio scene comprising a first set comprising a number, greater than or equal to one, of audio channels coded spatially over a number of sub-bands of determined frequencies, and decoded in a transformed domain into a second set comprising a number greater than or equal to two of time domain rendering sound channels, from sound propagation modeling filters audio signals of the first subset of channels.
• this device is remarkable in that, for each frequency sub-band of a spatial decoder in the transformed domain, this device comprises in addition to this spatial decoder:
• a filtering module by equalization-delay of the signal in sub-band by applying a gain respectively of a delay on the signal in sub-band, for generating from each of the spatially audio-coded channels an equalized component and delayed by a determined delay value in the sub-frequency band considered, a module for adding a subset of equalized and delayed components to create a number of filtered signals in the transformed domain corresponding to the number of the second set greater than or equal to two of the reproduction sound channels in the time domain,
• FIG. 2a represents an illustrative flow diagram of the implementation steps of the sound spatialization method which is the subject of the invention
• FIG. 2b represents by way of illustration, an alternative embodiment of the method that is the subject of the invention represented in FIG. 2a, obtained by creating additional subbands, in the absence of decimation;
• FIG. 2c represents by way of illustration, an alternative embodiment of the method that is the subject of the invention represented in FIG. 2a obtained by creating additional subbands, in the presence of decimation;
• FIG. 3a represents, by way of illustration, a stage, for a frequency sub-band of a spatial decoder, of a sound spatialization device which is the subject of the invention
• FIG. 3b represents, by way of illustration, a detail of implementation of a filter by equalization-delay allowing the implementation of the device of the invention shown in FIG. 3a;
• FIG. 4 represents by way of illustration, an example of implementation of the device according to the invention in which the calculation of the delay equalization filters is delocalized.
• the method according to the invention applies to an audio scene such as an audio scene 3 D represented by a first set comprising an N number of audio channels coded spatially greater than or equal to unity, N> 1, on a number of frequency subbands determined and decoded in a transformed domain.
• the transformed domain is a transformed frequency domain such as Fourier domain, PQMF domain or any hybrid domain derived from them by creating additional frequency subbands, whether or not subjected to a temporal decimation process. Consequently, the spatially coded audio channels constituting the first set N of channels are represented in a nonlimiting manner by the channels F1, Fr, Sr, SI, C, Ife previously described in the description and corresponding to a decoding mode of a 3 D audio scene in the corresponding transformed domain, as previously described in the description. This mode is none other than the 5.1 mode previously mentioned.
• these signals are decoded in the aforementioned transformed domain according to a determined number of sub-bands suitable for decoding, the set of sub-bands being noted.
• k denotes the rank of the subband considered.
• the method which is the subject of the invention makes it possible to transform all the spatially encoded audio channels mentioned above into a second set comprising a number, greater than or equal to two, of sound reproduction channels in the time domain, the sound reproduction channels being noted Bl and Br for the left binaural channels respectively right, without limitation in the context of Figure 2a. It is understood, in particular, that instead of two binaural channels, the method which is the subject of the invention applies to any number of channels greater than two, allowing, for example, the real-time sound reproduction of the 3D audio scene, as represented and described in the description in conjunction with FIG. 1b.
• this is implemented using acoustic propagation modeling filters of the audio signals of the first set of spatially coded audio channels, taking into account a conversion in the form of at least one gain and delay applicable in the transformed domain, as will be described later in the description.
• the modeling filters will be designated HRTF filters in the following description.
• the method according to the invention consists, for each frequency sub-band of the transformed domain of rank k, to perform a filtering in step A by equalization-delay of the signal in subband by application a gain g k respectively of a delay d k on the sub-band signal, to generate from the spatially-referenced coded channels, that is to say the channels Fl, C, Fr, Sr, SI and Ife, an equalized and delayed component of a determined delay value in the frequency subband SBk considered of rank k.
• CEDk x ⁇ FI, C, Fr, Sr, SI, lfe ⁇ (gkx, dkx).
• FEBk x denotes each equalized and delayed component obtained by applying the gain g kx and the delay d ⁇ on each of the spatially coded audio channels, that is to say the channels Fl, C, Fr, Sr , SI, Ife. Consequently, and in the aforementioned symbolic relation, x, for the corresponding rank k sub-band, can actually take the values F1, C, Fr, Sr, SI, Ife.
• Step A is then followed in the transformed domain of a step B of adding a subset of equalized and delayed components to create a number of filtered signals in the transformed domain corresponding to the number N 'of the second set, greater than or equal to 2, sound channels of restitution in the time domain.
• step B of FIG. 2a the addition operation is given by the symbolic relation:
• F (FI, C, Fr, Sr, SI, Ife) denotes the subset of the filtered signals in the transformed domain obtained by summation of a subset of equalized and delayed components CED kx .
• the subset of equalized and delayed components may consist of adding five of these components equalized and delayed for each ear to obtain the number N 'equal to 2 of filtered signals in the transformed domain, as will be described in more detail later in the description.
• the aforementioned addition step B is then followed by a step C of synthesizing each of the filtered signals in the transformed domain by a synthesis filter to obtain the second set of number N 'greater than or equal to two of sound signals of restitution in the time domain.
• step C of FIG. 2a the corresponding synthesis operation is represented by the symbolic relation:
• the method that is the subject of the invention can be applied to any 3D audio scene composed of N varying from 1 to infinity of audio channels or channels coded spatially to N 'varying from 2 to the infinity of sound channels of restitution.
• this step consists more specifically of adding a subset of components delayed differently by the different delays to generate the N 'components for each sub-band.
• the filtering by equalization-delay of the signal in sub-band includes at least the application of a phase shift supplemented if necessary by a pure delay by storage, for at least one sub-band. frequency bands.
• the transformed domain may, as previously mentioned in the description, correspond to a hybrid transformed domain as will be described in connection with FIG. 2b in the case where no frequency decimation is applied in the corresponding sub-band.
• the filtering by equalization delay shown in step A of Figure 2a is then performed in three substeps A1, A2, A3 shown in Figure 2b.
• the step A comprises an additional step of frequency-cutting in additional sub-bands without decimation, to increase the number of applied gain values and thus the frequency accuracy, followed by a subgrouping step. additional bands to which the aforementioned gain values have been applied.
• Frequency cutting and then grouping operations are represented in substeps Ai and A 2 of FIG. 2b.
• the step of the frequency cuts is represented in the substep Ai by the relation: HRTF ⁇ ⁇ g ⁇ , dkz ⁇ *: f.
• the gain and delay values for the sub-band of rank k considered are subdivided into Z corresponding gain values, a gain value gkz for each additional subband and the sub-step 1 2 it is understood that the grouping of the additional subbands is performed from the corresponding coded audio channels for the corresponding index x which has been applied the gain value gkz in the additional subband considered.
• [GCEDkz] ⁇ : fx is the grouping of the additional subbands to which the gain values for the additional subbands were applied.
• the sub-step A 2 is then followed by a sub-step A 3 consisting of applying the delay to the grouped additional subbands and in particular to the spatially coded audio channels of corresponding index x via the delay d kx of similar to Step A of Fig. 2a.
• the method which is the subject of the invention may also consist in performing a delay-equalization filtering in a hybrid transformed domain comprising an additional step of frequency cutting into additional sub-bands with decimation, as shown in FIG. 2c.
• step A'i of FIG. 2c is identical to step A 1 of FIG. 2b, to execute the creation of additional subbands with decimation.
• step A 1 of FIG. 2c the decimation operation in step A 1 of FIG. 2c is executed in the time domain.
• Step A 1 is then followed by a step A 2 corresponding to a grouping of the additional subbands to which the above-mentioned gain values have been applied in view of the decimation.
• Grouping step A ' 2 is itself preceded or followed by application of the delay dkx thus represented by the double reversing arrow of steps A 2 and A' 3 .
• this operation may advantageously consist in associating, as a gain value with each subband of rank k, a real value defined as the average of the corresponding HRTF filter module and to associate, as a delay value with each subband of rank k, a delay value corresponding to the delay of propagation between the left ear and the right ear of a listener for different positions.
• each subband SB k is associated with a delay value corresponding to the propagation delay between the left ear and the right ear of a listener for different positions.
• each band is associated with a real value.
• the HRTF filter module it is possible from the HRTF filter module, to calculate, for each subband, the average of the module of the aforementioned HRTF filter. Such an operation is similar to an octave band or Bark analysis of HRTF filters.
• the delay to be applied for the indirect channels that is to say the delay values which are more particularly applicable to the channels whose delay is not minimum, is determined.
• ITD Interaural Time Difference
• the most common method estimates the arrival time as the time when the HRIR time filter exceeds a given threshold.
• the arrival time may correspond to the time for which the response of the HRIR filter reaches 10% of its maximum.
• the application of a gain in the complex PQMF domain consists in multiplying the value of each sample of the subband signal, represented by a complex value, by the gain value formed by a real number.
• the application of a delay in the PQMF transformed domain consists, for each sample of the subband signal represented by a complex value, of introducing a rotation in the complex plane by multiplication of this sample by a complex exponential value depending on the rank of the sub-band considered, the sub-sampling rate in the sub-band considered and a delay parameter related to the interaural delay difference of a listener.
• the rotation in the complex plane is then followed by a pure time delay of the sample after rotation.
• This pure time delay is a function of the difference in the interaural delay of a listener and the sub-sampling rate in the subband considered.
• the aforementioned delays are applied to the resulting signals, ie the equalized signals and in particular to the subsets of these signals or channels which do not benefit from a direct path.
• M is the sub-sampling rate in the sub-band considered, M wants to be taken equal to 64, for example;
• - y (k, n) is the value of the output sample after application of the pure delay on the n-rank time sample of sub-band SBk of rank k, i.e. sample x (k, n) to which the delay B is applied.
• D * M + d corresponds to the interaural delay calculated previously, d can take negative values which makes it possible to simulate a phase advance instead of a delay.
• the processing implemented consists in performing a complex multiplication between an exponential complex and a subband sample formed by a complex value.
• the method which is the subject of the invention can also be implemented in a hybrid transformed domain.
• This hybrid transformed domain is a frequency domain in which the PQMF bands are advantageously redécoupées by a bank of filters decimated or not.
• the decimation means a decimation in time, so the introduction of a delay advantageously follows the procedure including a pure delay and a phase shifter.
• the delay may be applied only once during the synthesis. It is indeed useless to apply the same delay on each of the branches because the synthesis is a linear operation, without subsampling.
• the application of the gains remains the same, these being simply more numerous, as previously described in connection with FIG. 2b for example, and thus make it possible to follow the more precise cutting in frequency.
• a real gain is then applied per additional subband.
• the method according to the invention is repeated for at least two equalization-delay pairs and the signals obtained are summed to obtain the sound channels in the time domain.
• a more detailed description of a sound spatialization device of an audio scene comprising a first set comprising a number greater than or equal to the unit of audio channels spatially coded on a number of frequency subbands determined and decoded in a domain converted into a second set comprising a number greater than or equal to 2 of sound reproduction channels in the time domain, according to the subject of the present invention, will now be described in connection with Figures 3a and 3b.
• the device the invention is based on the principle of conversion in the form of at least one gain and a delay applicable in the transformed domain of modeling filters of the acoustic propagation of the audio signals of the first set of channels mentioned above.
• the device according to the invention allows the sound spatialization of an audio scene, such as a 3D audio scene, into a second set comprising a number, greater than or equal to two, of sound reproduction channels in the time domain.
• the device according to the invention shown in FIG. 3a relates to a stage of this device specific to each sub-band SB k of rank k decoding in the transformed domain.
• stage, for each subband of rank k shown in FIG. 3a is in fact replicated for each of the sub-bands to finally constitute the sound spatialization device according to the subject of the present invention.
• the stage represented in FIG. 3a will hereinafter be referred to as the sound spatialization device object of the invention.
• the device according to the invention as represented in FIG. 3a comprises, in addition to the spatial decoder shown, comprising the OTT modules 0 to OTT 4 substantially corresponding to a spatial decoder SD of the prior art such as 1c, but in which a sum of the front channel C and the low frequency channel Ife is additionally effected by a summator S, in a manner known per se from the state of the art. 1 filtering by equalization-delay of the signal in sub-band by applying a gain respectively a delay on the subband signal.
• the application of a gain is represented on each of the spatially coded audio channels, represented by amplifiers 1 0 to 1 8 , the latter generating an equalized component which may or may not be delayed by the intermediate delay elements noted 1g to I 12 to generate from each of the spatially coded audio channels an equalized and delayed component of a determined delay value in the frequency subband SB k .
• the gains of the amplifiers 1 0 to 1 8 have arbitrary values A, B, B, A > C, D, E 1 E, D respectively.
• the delay values applied by the delay modules 1 9 to 1i 2 have the values Df, Bf, Ds, Ds.
• the structure of the gains and delays introduced is symmetrical. A non-symmetrical structure can be implemented without departing from the scope of the subject of the invention.
• the device according to the invention also comprises a module 2 for adding a subset of equalized and delayed components to create a number of filtered signals in the transformed domain corresponding to the number N 'of the second set greater than or equal to two sound channels of restitution in the time domain.
• the device which is the subject of the invention comprises a module 3 for synthesizing each of the filtered signals in the transformed domain to obtain the second set comprising a number N 'greater than or equal to two of sound reproduction signals in the time domain.
• the synthesis module 3 thus comprises, in the embodiment of Figure 3a, a synthesizer 3 0 and 3i which allow each to grant a playback sound signal in the time domain Bi louse binaural left signal, respectively B n for binaural signal law.
• the equalized and delayed components in the embodiment of FIG. 3a are obtained in the following manner with:
• - B [k] denotes the gain of the amplifier I 11 I 2 represented in FIG. 3a
• - C [k] denotes the gain of the amplifier 1 4
• each amplifier, 1 0 to 1 8 delivers the following equalized components successively: - A [k] * FI [k] [n],
• the delays introduced by the delay elements 1g, 1i, 1n and I 12 are applied to the aforementioned equalized components to generate the equalized and delayed components.
• these delays are applied to the subset that does not have a direct path. These are, in the description of FIG. 3a, the signals which have undergone the multiplications by the gains B [k] and E [k] applied by the amplifiers or multipliers 11 12 and 16 and 17.
• the corresponding filtering element comprises a numerical multiplier, that is to say one of the multipliers or amplifiers 1 0 to 1 8 and represented by the gain value g kx in FIG. 3b, this multiplier allowing the multiplication of any complex sample of each coded audio channel of index x corresponding to the channels Fl, Fr, Clfe,
• the filtering element represented in FIG. 3b comprises at least one complex numerical multiplier making it possible to introduce a rotation in the complex plane of any sample of the signal in subband by a complex exponential value, the value exp (-j (k, SS k )) where ⁇ (k, SS k ) denotes a phase value which is a function of the sub-sampling rate of the sub-band considered and the rank of the sub-band considered k.
• ⁇ ⁇ k, SS k ) ⁇ * (k + 0.5) * d / M.
• the complex numerical multiplier is followed by a delay line denoted LAR introducing a pure delay of each sample after rotation, to introduce a pure time delay according to the difference of the interaural delay of a listener and the subsampling rate M in the sub-band SB k considered.
• the values of d and D are such that these values correspond to the application of a delay D * M + d in the non-sampled time domain and that the delay D * M + d corresponds to the interaural delay previously mentionned.
• the equalized and delayed components that is to say the addition of the components:
• the aforementioned signals can then feed a digital-to-analog converter, to allow the listening of sounds left Bi and right B r on an audio headset for example.
• the operation of synthesis performed by the synthesis modules 3 0 and 3i includes, where appropriate, the hybrid synthesis process as described above in the description.
• the method which is the subject of the invention may advantageously consist in dissociating the equalization and delay operations, which may relate to frequency sub-bands in a different number.
• the equalization can for example be performed in the hybrid domain and the delay in the PQMF domain.
• the method and the device that are the subject of the invention can also be applied to effect the trans-scaling, ie the rendering of a 3d sound field on a pair of tops or to convert in an uncomplicated manner a representation of N audio channels or sound sources from a spatial decoder or from several monophonic decoders to N 'available audio channels at the rendering level.
• the filtering operations can then be multiplied if necessary.
• the method and the device which are the subject of the invention can be applied to the case of an interactive 3D game in the sounds emitted by the different objects or sound sources, which can then be spatialized as a function of their relative position in relation to the listener. Sound samples are then compressed and stored in different files or memory areas. To be played and spatialised, they are partially decoded in order to remain in the coded domain and are filtered in the coded domain by suitable binaural filters advantageously using the writing method according to the object of the present invention.
• the invention finally covers a computer program comprising a sequence of instructions stored on a storage medium for execution by a computer or a dedicated sound spatialization device, which during this execution performs the addition filtering and synthesis as described in connection with Figures 2a to 2c and 3a, 3b previously in the description. It is understood in particular that the operations shown in the above figures can advantageously be implemented on complex digital samples via a central processing unit, a working memory and a program memory, not shown. in the drawing of Figure 3a. Finally, the calculation of the gains and delays constituting the equalization-delay filters can be performed externally to the device object of the invention shown in FIGS. 3a and 3b, as will be described hereinafter in FIG. connection with Figure 4.
• a first spatial coding and rate reduction coding unit I is considered, including a device according to the invention as represented in FIG. 3a, 3b, making it possible to carry out the aforementioned spatial coding at from an audio scene in 5.1 mode for example and coded audio transmission, on the one hand, and spatial parameters, on the other hand, to a decoding unit and spatial decoding II.
• the calculation of the delay equalization filters can then be performed by a separate unit III, which from the modeling filters, HRTF filters, calculates the gain and delay equalization values and transmits them to the coding unit I. spatial and spatial decoding unit II.
• Spatial coding can thus take into account the HRTFs that will be applied to correct its spatial parameters and improve 3D rendering.
• the rate reduction encoder can use these HRTFs to measure the perceptual effects of frequency quantization.
• the process implemented by the device and method of the invention thus makes it possible to execute a sound spatialization of an audio scene in which the first set comprises a determined number of spatially coded audio channels and the second set has a lower number of time domain rendering sound channels. It also allows decoding to perform an inverse transformation of a number of spatially coded audio channels to a set having a greater or equal number of time domain rendering sound channels.

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