Patent Application
20070160216
The present invention relates to the synthesis of audio signals, in particular in applications for editing music, video games or even ring tones for cell phones.
More particularly, the invention relates to both acoustic synthesis techniques and three-dimensional (3D) sound techniques.
To offer innovative services, based on acoustic synthesis (to create ring tones, or even in the context of games on cell phones), efforts are currently focused on enhancing the acoustic synthesis methods. However, since the terminals are limited in terms of memory and computation power, it is preferable to develop methods that are both effective and economical in terms of complexity.
Acoustic Synthesis Techniques
Numerous acoustic synthesis techniques have been developed in recent years. It should be pointed out that, in reality, there is no universal technique capable of generating just any sound. In practice, the production models that have been created until now all have their restrictions. A classification established by Julius Smith in:
“Viewpoints on the History of Digital Synthesis”, Smith J. O; Keynote paper, Proc. Int. Comp. Music Conf. 1991, Montreal,
is outlined below.
The techniques are categorized in four groups:
Some techniques, depending on their use, may fall into a number of categories.
The choice of the synthesis technique suited to a terminal or to a rendition system can be made on the basis of three families of criteria, in particular criteria of the type of those proposed by the signal acoustics and processing laboratory of the University of Helsinki, as part of an assessment of the different synthesis methods:
“Evaluation of Modern Sound Synthesis Methods”, Tolonen T., Valimaki, V., Karjalainen M; Report 48, Espoo 1998.
A first family of criteria concerns the use of the following parameters:
The quality and diversity of the sounds that are produced determine the second family of criteria, according to the following parameters:
Finally, the third family of criteria deals with implementation solutions, with parameters such as:
It has recently emerged that the techniques relying on a spectral modeling (with reproduction of the spectral image perceived by a listener) or a physical modeling (with simulation of the physical origin of the sound) are the most satisfactory and offer wide potential for future systems.
However, currently, the methods based on wavetable synthesis are the most widely used. The principle of this technique is as follows. Firstly, all the natural audio signals can be broken down into four phases: attack, decay, sustain and release, generally grouped under the term “ADSR envelope” (Attack, Decay, Sustain, Release envelope), which will be described later.
The wavetable synthesis principle consists in taking one or more signal periods (corresponding to a recording or to a synthetic signal), then applying processes to it (with looping, change of fundamental frequency, etc.) and finally applying the abovementioned ADSR envelope to it. This very simple synthesis method makes it possible to obtain satisfactory results. A technique similar to wavetable synthesis is the one known as “sampling” which is, however, distinguished by the fact that it uses recordings of natural signals instead of synthetic signals.
Another example of simple synthesis is synthesis by frequency modulation, more widely known as “FM synthesis”. In this case, a frequency modulation is performed for which the frequency of the modulating signal and the carrier signal (fm and fc) is located in the audible range (20 to 20 000 Hz). It shall also be indicated that the respective amplitudes of the harmonics relative to the fundamental mode can be chosen to define a sound timbre.
There are different transmission formats for the information intended for the sound synthesizers. Firstly, a score can be transmitted in MIDI formats or according to the MPEG4-Structured Audio standard for it then to be synthesized by the chosen acoustic synthesis technique. In some systems, it is also possible to transmit information on the instruments to be used by the synthesizer, for example using the DLS format which allows the information necessary for wavetable sound synthesis to be transmitted. Similarly, algorithmic languages of the “CSound” or “MPEG-4 SAOL” type make it possible to represent the sounds by acoustic synthesis technique.
The present invention relates to the combination of acoustic synthesis with the spatialization of sounds obtained from this synthesis. A few known acoustic spatialization techniques are summarized below.
Acoustic Spatialization Techniques
These are methods of processing the audio signal applied to the simulation of acoustic and psycho-acoustic phenomena. These techniques are aimed at generating signals to be transmitted to loudspeakers or to headphones, in order to give the listener the auditory illusion of acoustic sources placed in a predetermined position around him. They are advantageously applied in the creation of virtual acoustic sources and images.
Among the acoustic spatialization techniques, there are mainly two categories.
The methods based on a physical approach generally consist in reproducing the acoustic field like the original acoustic field within an area of finite dimensions. These methods do not take into account a priori the perceptive properties of the auditory system, in particular in terms of auditory location. With such systems, the listener is thus immersed in a field that is at all points identical to that which he would have perceived in the presence of the actual sources and he can therefore locate the acoustic sources as in a real listening situation.
The methods based on a psycho-acoustic approach seek rather to exploit 3D sound perception mechanisms in order to simplify the sound reproduction process. For example, instead of reproducing the acoustic field over an entire area, it is possible to make do with reproducing it only at the two ears of the listener. Similarly, it is possible to impose a faithful reproduction of the acoustic field over a fraction of the spectrum only, in order to relax the constraint on the rest of the spectrum. The objective is to take account of the perception mechanisms of the auditory system in order to identify the minimum quantity of information to be reproduced to obtain a psycho-acoustic field identical to the original field, that is such that the ear, because of the limitations of its performance, is incapable of distinguishing them from each other.
In the first category, different techniques have been identified:
In the second category, there are, for example:
Each technique is characterized by a specific method of encoding and decoding spatialization information in an appropriate audio signal format.
The different acoustic spatialization techniques are also distinguished by the extent of the spatialization that they provide. Typically, a 3D spatialization such as ambisonic encoding, holophony, binaural or transaural synthesis (transaural synthesis being a transposition of the binaural technique on two remote loudspeakers), includes all the directions of the space. Moreover, a two-dimensional (2D) spatialization, such as stereophony, or a 2D restriction of holophony or of the ambisonic technique, is limited to the horizontal plane.
Finally, the different techniques are distinguished by their possible delivery systems, for example:
There is a wide range of current devices offering acoustic synthesis capabilities. These devices range from the musical instrument (such as a keyboard, a rhythm box, or other), mobile terminals, of PDA (Personal Digital Assistant) type, for example, or even computers with music editing software installed, or even effects pedal assemblies equipped with a MIDI interface. The sound reproduction systems (headset, stereo loudspeakers or multiple loudspeaker systems) and the quality of the acoustic synthesis systems vary widely, in particular according to the more or less limited computation capabilities and according to the environments in which such systems are used.
Systems capable of spatializing previously synthesized sounds, in particular by a cascading of an acoustic synthesis engine and a spatialization engine, are currently known. The spatialization is then applied to the output signal from the synthesizer (on a mono channel or two stereo channels) after mixing of the different sources. Implementations of this solution for then spatializing the sounds obtained from a synthesizer are thus known.
More generally known are implementations consisting in 3D rendition engines, which can be applied to any type of digital audio signals, whether synthetic or not. For example, the different musical instruments of a MIDI score (conventional acoustic synthesis format) can then be positioned in the acoustic space. However, to obtain such a spatialization, the MIDI signals must first be converted into digital audio signals and then a spatialization processing must be applied to the latter.
This implementation is particularly costly in terms of processing time and processing complexity.
One aim of the present invention is an acoustic synthesis method offering the possibility of directly spatializing the synthetic sounds.
More particularly, an aim of the present invention is to associate with the acoustic synthesis spatialization tools of satisfactory quality. However, this association compounds the complexity due to the acoustic synthesis with that of the spatialization, which makes it difficult to implement a spatialized acoustic synthesis on very restricted terminals (that is, terminals with relatively limited computation power and memory).
Another aim of the present invention is to achieve an optimization of the complexity involved in spatializing synthetic sounds according to the capabilities of the terminal.
To this end, the present invention firstly proposes an acoustic synthesis and spatialization method, in which a synthetic sound to be generated is characterized by the nature of a virtual acoustic source and by its position relative to a chosen origin.
The method according to the invention comprises a joint step consisting in determining parameters including at least one gain, for defining at the same time:
It will thus be understood that the present invention allows a sound spatialization technique to be integrated in an acoustic synthesis technique, so as to obtain a global processing using common parameters for the implementation of the two techniques.
In an embodiment, the spatialization of the virtual source is carried out in an ambisonic context. The method then includes a step for computing gains associated with ambisonic components in a spherical harmonics base.
In a variant, the synthetic sound is intended to be reproduced in a holophonic, or binaural, or transaural context, on a plurality of reproduction channels. It will be understood in particular that this “plurality of reproduction channels” can equally relate to two reproduction channels, in a binaural or transaural context, or even more than two reproduction channels, for example in a holophonic context. During said joint step, a delay between reproduction channels is also determined, to define at the same time:
In this embodiment, the nature of the virtual source is parameterized at least by a temporal loudness variation, over a chosen duration and including a sound triggering instant. In practice, this temporal variation can advantageously be represented by an ADSR envelope as described above.
Preferably, this variation comprises at least:
Of course, more complex envelope variations can be envisaged.
The spatialization of the virtual source is preferably performed by a binaural synthesis based on a linear breakdown of transfer functions, these transfer functions being expressed by a linear combination of terms dependent on the frequency of the sound and weighted by terms dependent on the direction of the sound. This measure proves advantageous in particular when the position of the virtual source can change over time and/or when a number of virtual sources are to be spatialized.
Preferably, the direction is defined by at least one bias angle (for a spatialization in a single plane) and, preferably, by a bias angle and an elevation angle (for a three-dimensional spatialization).
In the context of a binaural synthesis based on a linear breakdown of the transfer functions, the position of the virtual source is advantageously parameterized at least by:
Preferably, the nature of the virtual source is parameterized at least by one acoustic timbre, by associating the chosen relative loudnesses with harmonics of a frequency corresponding to a pitch of the sound. In practice, this modeling is advantageously carried out by an FM synthesis, described above.
In an advantageous embodiment, an acoustic synthesis engine is provided, specifically for generating spatialized sounds, relative to a predetermined origin.
Preferably, the synthesis engine is implemented in a music editing context, and a man-machine interface is also provided for, to place the virtual source in a chosen position relative to the predetermined origin.
To synthesize and spatialize a plurality of virtual sources, each source is assigned to a respective position, preferably by using a linear breakdown of the transfer functions in a binaural context, as indicated above.
The present invention aims also at a module for generating synthetic sounds, comprising in particular a processor, and including in particular a working memory specifically for storing instructions for implementing the above method, so as to process simultaneously a synthesis and a spatialization of the sound, according to one of the advantages of the present invention.
To this end, the present invention aims also at a computer program product, stored in a memory of a central processing unit or a terminal, in particular a mobile terminal, or on a removable medium specifically for cooperating with a drive of said central processing unit, and comprising instructions for implementing the above method.
Other characteristics and advantages of the invention will become apparent from examining the detailed description below, and the appended drawings in which:
and
It will be remembered that the present invention proposes to integrate a sound spatialization technique with an acoustic synthesis technique so as to obtain a global, optimized, spatialized acoustic synthesis processing. In the context of very restricted terminals, the pooling of certain of the acoustic synthesis operations, on the one hand, and acoustic spatialization operations on the other hand, proves of particular interest.
As a general rule, the function of an acoustic synthesis engine (typically a “synthesizer”) is to generate one or more synthetic signals, based on a sound synthesis model, a model that is driven based on a set of parameters, called “synthesis parameters” below. The synthetic signals generated by the synthesis engine can correspond to separate acoustic sources (which are, for example, the different instruments of a score) or can be associated with one and the same source, for example in the case of different notes from one and the same instrument. Hereinafter, the term “sound generator” denotes a module for producing a musical note. Thus, it will be understood that a synthesizer is made up of a set of sound generators.
Also as a general rule, an acoustic spatialization tool is a tool that accepts a given number of audio signals as input, these signals being representative of acoustic sources and, in principle, without any spatialization processing. It should be indicated in fact that, if these signals have already been subjected to a spatialized processing, this prior processing is not taken into account here. The role of the spatialization tool is to process the input signals, according to a scheme that is specific to the chosen spatialization technique, to generate a given number of output signals which define the spatialized signals representative of the acoustic scene in the chosen spatialization format. The nature and complexity of the spatialization processing depend obviously on the chosen technique, according to whether it is a rendition in stereophonic, binaural, holophonic or ambisonic format that is being considered.
More particularly, for many spatialization techniques, it appears that the processing can be reduced to an encoding phase and a decoding phase, as will be seen later.
The encoding corresponds to the sound pick-up of the acoustic field generated by the different sources at a given instant. This “virtual” sound pick-up system can be more or less complex depending on the acoustic spatialization technique adopted. Thus, a sound pick-up by a more or less large number of microphones with different positions and directivities is simulated. In all cases, the encoding amounts, for calculating the contribution of an acoustic source, at least to the application of gains and, more often than not, delays (typically in holophony or in binaural or transaural synthesis), to different copies of the signal sent by the source. There is one gain (and, where appropriate, one delay) per source for each virtual microphone. This gain (and this delay) depend on the position of the source relative to the microphone. If there is provided a virtual sound pick-up system equipped with K microphones, there are K signals output from the encoding system.
Referring to
The encoded signal Ej is given by the expression:
In this expression, it is assumed that L sources (i=1, 2, . . . , L) have to be processed, whereas the encoding format provides for K signals (j=1, 2, . . . , K). The gains and the delays depend on the position of the source i relative to the microphone j at the instant t. The encoding is therefore a representation of the acoustic field generated by the acoustic sources at that instant j. It is simply recalled here that, in an ambisonic context (consisting of a breakdown of the field in a spherical harmonics base), the delay does not actually contribute to the spatialization processing.
In the case where the acoustic sources are in a room, the image sources must be added. These are images of the acoustic sources reflected by the walls of the room. The image sources, by being reflected in turn on the walls, generate higher order image sources. In the above expression, L therefore no longer represents the number of sources, but the number of sources to which are added the number of image sources. The number of image sources is infinite, which is why, in practice, only the audible image sources and those for which the direction is perceived are kept. The image sources that are audible but for which the direction can no longer be perceived are grouped and their contribution is synthesized using an artificial reverberator.
The aim of the decoding step is to reproduce the encoded signals Ej on a given device, comprising a predetermined number T of acoustic transducers (headset, loudspeaker). This step consists in applying a matrix T×K of filters to the encoded signals. This matrix depends only on the rendition device, and not on the acoustic sources. Depending on the encoding and decoding technique chosen, this matrix can be very simple (for example identity) or very complex.
Then, it should be indicated that the invention advantageously provides a single step ETA for the joint synthesis and spatialization of the sound. As a general rule, a sound can be defined at least by:
Thus, in the example of a synthesizer with sensitive keypad, if the user plays a forte note, the loudness associated with the command Ci will be greater than the loudness associated with a piano note. More specifically, it should be indicated that the loudness parameter can, as a general rule, take into account the spatialization gain gi in a spatialization processing context, as will be seen below, according to one of the major advantages of the present invention.
Furthermore, a sound is, of course, also defined by its triggering instant. Typically, if the chosen spatialization technique is not an ambisonic processing, but rather binaural or transaural synthesis, holophony or other, the spatialization delay τi (which will be described in detail below) can be used to also control the triggering instant of the sound.
By referring again to
As indicated above, the latter two parameters gi and τi can be used jointly for the spatialization, but also for the very synthesis of the sound, when a loudness (or a pan in stereophony) and a triggering instant of the sound are defined.
More generally, it should be indicated that, in a preferred embodiment, the two modules M1 and M2 are grouped in one and the same module to allow for the definition in one and the same step of all the parameters of the signal si to be synthesized and spatialized: its frequency, its duration, its spatialization gain, its spatialization delay, in particular.
These parameters are then applied to an encoding module M3 of the acoustic synthesis and spatialization device D1. Typically, for example in binaural or transaural synthesis, this module M3 carries out a linear combination on the signals si which involves in particular the spatialization gains, as will be seen below. This encoding module M3 can also apply a compression mode encoding to the signals si to prepare for a transmission of the encoded data to a reproduction device D2.
It should be indicated, however, that this encoding module M3 is, in a preferred embodiment, directly incorporated in the modules M1 and M2 above, so as to create directly, within a single module D1 which would consist simply of an acoustic synthesis and spatialization engine, the signals Ej as if they were delivered by microphones j, as explained above.
Thus, the acoustic synthesis and spatialization engine D1 produces, at the output, K acoustic signals Ej representing the encoding of the virtual acoustic field that the different synthetic sources would have created if they had been real. At this stage, there is a description of an acoustic scene in a given encoding format.
Of course, provision can also be made to add (or “mix”) to this acoustic scene other scenes originating from an actual sound pick-up or from the output of other sound processing modules, provided that they are in the same spatialization format. The mixing of these different scenes then passes into a particular and unique decoding system M′3, provided at the input of a reproduction device D2. In the example represented in
There follows a description of a preferred embodiment of the invention, in this case applied to a mobile terminal and in the context of an acoustic spatialization by binaural synthesis.
On telecommunication terminals, mobile in particular, there is naturally provided an acoustic rendition with a stereophonic headset. The preferred acoustic source positioning technique is then binaural synthesis. It consists, for each acoustic source, in filtering the monophonic signal via acoustic transfer functions called HRTFs (for Head Related Transfer Functions), which model the transformations generated by the chest, the head and the auricle of the person hearing the signal originating from a acoustic source. For each position of the space, a pair of these functions (one function for the right ear, one function for the left ear) can be measured. The HRTFs are therefore functions of the position [θ,φ] (where θ represents the bias and p the elevation) and of the acoustic frequency f. There is then obtained, for a given subject, a database of 2M acoustic transfer functions representing each position of the space for each ear (M being the number of directions measured). Conventionally, this technique is implemented in so-called “bicanal” form.
Another binaural synthesis, based on a linear breakdown of the HRTFs, corresponds to an implementation which proves more effective in particular when a number of acoustic sources are spatialized, or in the case where the acoustic sources change position in time. In this case, the term “dynamic binaural synthesis” is used.
These two embodiments of binaural synthesis are described below.
“Bicanal” Binaural Synthesis
Referring to
It should be indicated that, in this implementation, the positions of the acoustic sources are not felt to change in time. However, if there is a desire to have the positions of the acoustic sources vary in the space over time, it is preferable to modify the filters used to model the left and right HRTFs. However, since these filters take the form of either finite impulse response (FIR) filters or infinite impulse response (IIR) filters, discontinuity problems on the left and right output signals appear, leading to audible “clicks”. The technical solution employed to overcome this problem is to rotate two sets of binaural filters in parallel. The first set simulates the first position [θ1,φ1] at an instant t1, the second the second position [θ2,φ2] at an instant t2. The signal giving the illusion of a movement between the first and second positions is then obtained by a sequenced fading of the left and right signals resulting from the first and second filtering processes. Thus, the complexity of the acoustic source positioning system is then multiplied by two compared to the static case. Furthermore, the number of filters to be implemented is proportional to the number of sources to be spatialized.
If N acoustic sources are considered, the number of filters needed is then 2.N for a static binaural synthesis and 4.N for a dynamic binaural synthesis.
A description of an advantageous variant is given below.
Binaural Synthesis Based on a Linear Breakdown of the HRTFs
It should first be indicated that such an implementation has a complexity that no longer depends on the total number of sources to be positioned in space. In practice, these techniques allow the HRTFs to be broken down on the basis of orthogonal functions, common to all the positions of the space, and therefore no longer depend only on the frequency f. Thus, the number of filters needed is reduced. More particularly, the number of filters is fixed and no longer depends on the number of sources to be positioned, so that adding an additional acoustic source requires only the application of a delay, followed by a multiplication operation by a number of gains dependent only on the position [θ,φ] and an addition operation, as will be seen with reference to
The aim of the linear breakdown of the HRTFs is to separate the space and frequency dependencies of the transfer functions. First, the phase excess of the HRTFs is extracted, then modeled in the form of a pure delay τ. The linear breakdown is then applied to the minimal phase component of the HRTFs. Each HRTF is expressed as a sum of P spatial functions Cj(θ,φ) and reconstruction filters Lj(f):
The implementation scheme for binaural synthesis based on a linear breakdown of the HRTFs is illustrated in
For each channel, the signals from the N sources are then added (step 43) then filtered (step 44) by the filter Lj(f) corresponding to the jth basic vector.
The same scheme is applied separately for the right and left channels.
The benefit of this implementation is that, unlike the “bicanal” binaural synthesis, the addition of an extra source does not require two additional filters (of FIR or IIR type). In other words, the P basic filters are shared by all the sources present. Furthermore, in the case of dynamic binaural synthesis, it is possible to vary the coefficients Cj(θi,φi) without provoking audible clicks at the output of the device. In this case, only 2.P filters are necessary, whereas 4.N filters were necessary for the dynamic bicanal implementation described above.
In other words, the delays τ and the gains C and D, which constitute the spatialization parameters and are specific to each acoustic source according to its position, can therefore be dissociated from the directional filters L(f) in the implementation of the binaural synthesis based on a linear breakdown of the HRTFs. Consequently, the directional filters are common to the N sources, independently of their position, their number or any movement by them. The application of the spatialization parameters then represents the spatial encoding proper, of the signals relative to the sources themselves, whereas the directional filters perform the actual spatial decoding processing, with a view to reproduction, which no longer depends on the position of the sources, but on the acoustic frequency.
Referring to
There now follows a description, with reference to
Of course, more complex envelope variations can be envisaged, including, for example, more than four phases.
It should, however, be indicated that most of the synthesized sounds can be modeled by an envelope variation as described above. Preferably, the parameters of the ADSR envelope are defined before performing the filterings provided for the spatialization processing, because of the time variables involved.
It will thus be understood that the maximum of the acoustic amplitude (in arbitrary units in
Reference is now made to
It will thus be understood that such a measure makes it possible to avoid, in a particularly advantageous way, generating, from a score in MIDI format, the sounds in a standard audio reproduction format (for example, in “wave” format) and encoding them again for a spatialization of the sound, as in the known implementations.
The present invention allows for the direct and concurrent implementation of the sound spatialization and synthesis steps. It will be understood in particular that any acoustic synthesis processing, requiring the definition of a loudness (and, where appropriate, a triggering instant of the sound), can be performed jointly with a spatialization processing, offering a gain (and, where appropriate, a delay) for each reproduction channel.
As a general rule, a sound synthesizer operates on the basis of reading a score which combines the information on the instruments to be synthesized, the instants at which the sounds must be played, the pitch of these sounds, their strength, etc. When reading this score, each sound is assigned a sound generator, as indicated above with reference to
Consideration is given first to the case where one and the same source plays a number of notes simultaneously. These notes, which originate from the same source, are spatialized in the same position and therefore with the same parameters. It is then preferred to combine the spatialization processing for the sound generators associated with the same source. In these conditions, the signals associated with the notes obtained from one and the same source are preferably summed first so as to apply the spatialization processing globally to the resultant signal, which, on the one hand, advantageously reduces the implementation cost and, on the other hand, advantageously ensures the consistency of the acoustic scene.
Furthermore, the gains and the delays can be applied by exploiting the structure of the synthesizer. On the one hand, the spatialization delays (left channel and right channel) are implemented in the form of delay lines. On the other hand, in the context of the synthesizer, the delays are managed by the triggering instants of the sound generators in accordance with the score. In the context of a spatialized acoustic synthesis, the two preceding approaches (delay line and control of the triggering instant) are combined so as to optimize the processing. There is therefore a saving of a delay line for each source, by adjusting the triggering instants of the sound generators. To this end, the difference between the delays of the left channel and the right channel for the spatialization is extracted. Provision is then made to add the smaller of the two delays to the triggering instant of the generator. It then remains to apply the time difference between the left and right channels to just one of the two channels, in the form of a delay line, it being understood that this delay difference can have positive or negative values.
With respect to the gains, the balance (or “pan”) parameter, which is typically associated with the stereophonic system, is no longer needed. It is therefore possible to eliminate the gains associated with balance. Furthermore, the volume parameter of the sound generator can be applied to the level of the various gains corresponding to the spatial encoding, as described above.
It should also be indicated that the present invention allows the acoustic spatialization to be applied, source by source, because of the fact that the spatialization tool is incorporated in the core of the acoustic synthesis engine. Such is not the case if, on the contrary, the chosen method is to simply cascade the synthesis engine with the spatialization tool. In this case, in practice, it should be remembered that the spatialization can be applied only globally to all the acoustic scene.
According to another advantage of the present invention, the acoustic synthesis and spatialization tools can be judiciously combined to produce an optimized implementation of a spatialized acoustic synthesis engine, with, in particular, an optimization of the combination of the synthesis and spatialization operations, taking into account in particular at least one spatialization gain and/or delay, or even a spatialization filter.
In the case where the synthesis parameters already apply one or more of these parameters (gain, delay, filter), the spatialization filters are advantageously taken into account by simply modifying the synthesis parameters, without modifying the synthesis model itself.
Moreover, by simply adding a gain and a delay to the acoustic synthesis engine, where necessary complemented with a filter, a spatialized acoustic synthesis, based on different possible spatialization techniques, can be obtained. These spatialization techniques (binaural/transaural synthesis, holophony, ambisony, etc.) can be of variable complexity and efficiency but overall offer a far richer and more comprehensive spatialization than stereophony, with, in particular, a natural and particularly immersive rendition of the acoustic scene. In practice, the inventive acoustic spatialization retains all the potential of a three dimensional acoustic rendition, in particular in terms of immersion, with genuine 3D spatialization.
Of course, it is also possible to provide for an integration of the spatialization and room effect processing, in the simplified form of at least one gain and/or one delay (where appropriate complemented with filters), and an artificial reverberator for the delayed reverberation.
| Filing Document | Filing Date | Country | Kind | 371c Date |
|---|---|---|---|---|
| PCT/FR03/03730 | 12/15/2003 | WO | 6/14/2006 |
