Method and system for obtaining at least one acoustic parameter of an environment

Abstract

At least one acoustic parameter of an environment is obtained by emitting a sound signal emitted, receiving a sound signal received, estimating a sampling frequency shift and of a temporal shift between the emission and reception devices on the basis of impulse responses obtained on the basis of portions of the sound signal received and of corresponding portions of the sound signal emitted, obtaining a corrected sound signal received by applying to the sound signal received a sampling frequency shift correction and a temporal shift correction on the basis of the estimating, determining the impulse response of the environment on the basis of portions of the corrected sound signal received and of corresponding portions of the sound signal emitted, and obtaining acoustic parameters of the environment on the basis of the impulse response.

Claims

1. A method for obtaining at least one acoustic parameter of an environment, the method comprising: emitting a sound signal emitted by an emission device; receiving a sound signal received by at least one reception device; and for each of said at least one reception device: determining an impulse response of the environment at a position of said reception device; and obtaining an acoustic parameter of the environment on the basis of the impulse response; wherein said method further comprises: estimating of a sampling frequency shift between the emission device and said reception device on the basis of at least one impulse response obtained on the basis of at least one portion of the sound signal received and of a corresponding portion of the sound signal emitted; estimating a temporal shift between the emission device and said reception device on the basis of at least one impulse response obtained on the basis of at least one portion of the sound signal received and of a corresponding portion of the sound signal emitted; obtaining a corrected sound signal received by applying, to the sound signal received by the reception device, a sampling frequency shift correction and a temporal shift correction on the basis of said estimations; determining said impulse response of the environment being obtained on the basis of at least one impulse response obtained on the basis of at least one portion of the corrected sound signal received and of a corresponding portion of the sound signal emitted, wherein the estimating the temporal shift between the emission device and said reception device is obtained on the basis of a comparison, for at least one portion of the sound signal emitted, between: the position of the maximum amplitude of the impulse response obtained on the basis of the portion of the sound signal emitted and of a corresponding portion of the sound signal received, and the position of the maximum amplitude of the ideal impulse response obtained on the basis of the portion of the sound signal emitted and of a theoretical sound signal received identical to the portion of the sound signal emitted.

2. The method as claimed in claim 1, further comprising a preliminary step of obtaining at least one characteristic of the environment, the at least one characteristic of the environment being taken into account to parametrize said sound signal emitted.

3. The method as claimed in claim 1, further comprising cutting the sound signal received into said portions of signal received, the cutting being performed on the basis of the corresponding portions of the sound signal emitted.

4. The method as claimed in claim 1, wherein the estimating the sampling frequency shift between the emission device and said reception device comprises the obtaining of a first estimation of the sampling frequency shift by a maximization, by varying the sampling frequency of the sound signal received of the maximum amplitude of the impulse response obtained on the basis of a portion of the sound signal emitted and of a corresponding portion of the sound signal received.

5. The method as claimed in claim 4, wherein the estimating of the sampling frequency shift between the emission device and said reception device comprises a refinement of said first estimation on the basis of a comparison, for several successive portions of the sound signal emitted, between: the position of the maximum amplitude of the impulse response obtained on the basis of the portion of the sound signal emitted and of a corresponding portion of the sound signal received re-sampled at the sampling frequency corresponding to said maximization, and the position of the maximum amplitude of the ideal impulse response obtained on the basis of the portion of the sound signal emitted and of a theoretical sound signal received identical to the portion of the sound signal emitted.

6. The method as claimed in claim 1, further comprising deleting invalid sequences of the corrected sound signal received, the invalid sequences corresponding to losses of packets of the sound signal emitted.

7. The method as claimed in claim 6, wherein the sound signal emitted comprises temporal tags and the invalid sequences of the corrected sound signal received are identified by comparing: a duration between two consecutive tags for the impulse response obtained on the basis of at least one portion of the sound signal emitted comprising the two tags and of a corresponding portion of the corrected sound signal received, and the duration between the two tags for the sound signal emitted.

8. The method as claimed in claim 6, wherein said impulse response of the environment is calculated on the basis of at least one portion of the corrected sound signal received in which the invalid sequences have been deleted and of a corresponding portion of the sound signal emitted in which the sequences corresponding to the invalid sequences have been deleted.

9. The method as claimed in claim 1, wherein the sound signal emitted comprises at least one portion comprising the repetition of at least one sequence, said impulse response of the environment being obtained by performing: a calculation of the impulse response on the basis of each of these sequences and of corresponding sequences of the sound signal received; an average of the impulse responses within each of said portions; and then an average of the averages for all said portions.

10. The method as claimed in claim 1, wherein the portions of the sound signal emitted comprise at least one sequence chosen from among: an MLS sequence (Maximum Length Sequence); an IRS sequence (Inverse Repeated Sequence); a TSP sequence (Time-Streched Pulses); and a Logarithmic SineSweep sequence.

11. A non-transitory computer-readable medium storing a computer program that, when executed by a computer, causes the computer to execute the method for obtaining at least one acoustic parameter as claimed in claim 1.

12. A terminal, comprising: a communication module able to dispatch a sound signal to an emission device able to play back the sound signal in the form of a sound emitted in an environment; a microphone able to receive a sound signal received; a processing module configured to: determine an impulse response of the environment at the position of said terminal; obtain an acoustic parameter of the environment on the basis of the impulse response; estimate a sampling frequency shift between the emission device and said terminal on the basis of at least one impulse response obtained on the basis of at least one portion of the sound signal received and of a corresponding portion of the sound signal emitted; estimate a temporal shift between the emission device and said terminal on the basis of at least one impulse response obtained on the basis of at least one portion of the sound signal received and of a corresponding portion of the sound signal emitted; obtain a corrected sound signal received by applying, to the sound signal received, a sampling frequency shift correction and a temporal shift correction on the basis of said estimations; and determine said impulse response of the environment on the basis of at least one impulse response obtained on the basis of at least one portion of the corrected sound signal received and of a corresponding portion of the sound signal emitted wherein the processing module is configured to obtain the temporal shift between the emission device and said reception device on the basis of a comparison, for at least one portion of the sound signal emitted, between: the position of the maximum amplitude of the impulse response obtained on the basis of the portion of the sound signal emitted and of a corresponding portion of the sound signal received, and the position of the maximum amplitude of the ideal impulse response obtained on the basis of the portion of the sound signal emitted and of a theoretical sound signal received identical to the portion of the sound signal emitted.

13. A system for obtaining at least one acoustic parameter of an environment, the system comprising: an emission device for emitting a sound signal emitted; at least one reception device for receiving a sound signal received; a module for determining an impulse response of the environment; and a module for obtaining an acoustic parameter of the environment on the basis of the impulse response; a module for estimating a sampling frequency shift between the emission device and said reception device on the basis of at least one impulse response obtained on the basis of at least one portion of the sound signal received and of a corresponding portion of the sound signal emitted; a module for estimating a temporal shift between the emission device and said reception device on the basis of at least one impulse response obtained on the basis of at least one portion of the sound signal received and of a corresponding portion of the sound signal emitted; and a module for obtaining a corrected sound signal received obtained by applying, to the sound signal received by the reception device, a sampling frequency shift correction and a temporal shift correction on the basis of said estimations, wherein the module for determining the impulse response of the environment being configured to determine said impulse response of the environment on the basis of at least one impulse response obtained on the basis of at least one portion of the corrected sound signal received and of a corresponding portion of the sound signal emitted, and wherein the module for estimating the temporal shift between the emission device and said reception device obtains the temporal shift on the basis of a comparison, for at least one portion of the sound signal emitted, between: the position of the maximum amplitude of the impulse response obtained on the basis of the portion of the sound signal emitted and of a corresponding portion of the sound signal received, and the position of the maximum amplitude of the ideal impulse response obtained on the basis of the portion of the sound signal emitted and of a theoretical sound signal received identical to the portion of the sound signal emitted.

Description

BRIEF DESCRIPTION OF THE DRAWINGS

(1) Other characteristics and advantages of the present invention will emerge from the description given hereinbelow, with reference to the drawings and to annex 1 which illustrate an exemplary embodiment thereof devoid of any limiting character and in which:

(2) FIG. 1 represents a terminal in accordance with the invention in a particular embodiment;

(3) FIG. 2 represents, in a schematic manner, a first exemplary implementation of the invention in a particular measurement environment;

(4) FIG. 3 represents a memory of a terminal in accordance with a particular embodiment of the invention;

(5) FIG. 4 represents an exemplary sound signal that may be used in a particular embodiment of the invention;

(6) FIG. 5 represents, in the form of a flowchart, the main steps of a method for obtaining acoustic parameters of an environment in a particular embodiment of the invention;

(7) FIG. 6 represents, in a schematic manner, a second exemplary implementation of the invention in a particular measurement environment; and

(8) annex 1 provides an exemplary sound signal emitted that may be used in a particular embodiment of the invention.

DETAILED DESCRIPTION OF A FIRST EMBODIMENT

(9) FIG. 1 represents in a schematic manner a terminal 10 in accordance with a particular embodiment of the invention. In this example, this terminal consists of a smartphone, offering the user, beyond telephone functions, and more generally communication functions, access to diverse applications as soon as the latter have been installed in the terminal.

(10) From a hardware point of view, this terminal comprises in particular a processor 11, a read-only memory of ROM type 12 in which are recorded system functions, especially drivers and the operating system of the terminal, a microphone 13, a loudspeaker 14, a screen 15, a sound card 16, one or more communication modules (3G, 4G, Bluetooth, WiFi, etc.) 17 and a rewritable nonvolatile memory 18 comprising applications APP and user data that are not represented in this figure, these elements being linked together by a bus system.

(11) In a known manner, the screen 15 constitutes a man-machine touch interface on which are represented icons I1, I2, IT, corresponding to the system applications and to the various applications APP installed by the user of the terminal.

(12) Among these icons, an icon IT allows the terminal to access remotely, via a telecommunications network, a portal of downloadable applications compatible with the operating system of the terminal and to install, optionally via payment and/or authentication, new applications APP in the rewritable nonvolatile memory 18.

(13) In the embodiment described here, a computer program PG in accordance with the invention can be downloaded from this applications portal, and an associated icon presented on the touch interface 15.

(14) With reference to FIG. 2, we will assume in this example that a user possessing two terminals RX1, RX2 that are identical or similar to the terminal 10 of FIG. 1 in respect of the functional parts in common with the latter (reception devices within the meaning of the invention), desires to assess in a single measurement, the acoustics of an environment 100 in two locations, this environment consisting in this example of a classroom comprising a podium and several rows of desks facing this podium.

(15) If this has not already been done, the user downloads and installs the computer program PG in the nonvolatile memory 18 of each of the terminals RX1, RX2, positions these terminals at the positions P1, P2 and places an enclosure TX (emission device within the meaning of the invention) at the level of the podium.

(16) This enclosure TX comprises communication means, not represented, compatible with those of the terminals RX1, RX2 so as to be able to receive from one of these terminals a sound signal S1.sub.k to be played back in the form of a sound signal emitted in the environment 100. In this example, the enclosure TX is an autonomous enclosure comprising in a conventional manner a loudspeaker, a battery, a power management circuit, a wireless connectivity module, a digital-analog converter, an audio amplifier.

(17) FIG. 3 represents a part of the nonvolatile memory 18 of each of the terminals RX1, RX2 after installation of the computer program PG. This memory comprises: a part CODE corresponding to the instructions, compiled or otherwise, of the computer program PG and interpretable by the processor 11, these instructions making it possible to implement steps E5 to E70 of the method according to the invention and represented in FIG. 5; a library LC of contexts CT.sub.k selectable by the user via the man-machine interface 15 of the terminal, each measurement context being defined by a pair {type of environment/use}; and a library LS of predefined or parametrizable sound signals S1.sub.k associated with these contexts.

(18) In the embodiment described here, updates of the computer program PG can be downloaded from the portal and installed in the memory 18, for example when new contexts CT.sub.k or new sound signals S1.sub.k are available.

(19) We will assume in this example that the computer program PG offers two modes of operation referenced Master/Slave, respectively implemented by the terminals RX1, RX2 in the example described here, and in which: the Master mode allows the terminal RX1 to choose a context CT.sub.1 in the library LC via the interface 15, to determine a sound signal hereinafter referenced S1.sub.1 associated with this context in the library of sounds LS, to dispatch the reference of this sound signal to the terminals RX2 in Slave mode by communication means 17, to dispatch the sound signal to the enclosure TX by communication means 17 so that the enclosure plays it back in the form of a sound signal emitted S1.sub.1 in the environment, to control the capture of a sound signal received S2.sub.1.sup.1 by the microphone 13 after a predetermined timeout, to record this sound signal received S2.sub.1.sup.1 in the nonvolatile memory 18 by the sound card 16, to determine acoustic parameters CA.sup.P1 of the environment at the position P1 of the terminal RX1 on the basis of the sound signal emitted S1.sub.1 by the enclosure TX and of the sound signal received S2.sub.1.sup.1, and to present an information message MSG to the user of the terminal generated on the basis of these parameters; the mode Slave allows the terminal RX2, to receive, from a Master terminal RX1, the reference of a sound signal S1.sub.1 by the communication means 17, to control the capture of a sound signal received S2.sub.1.sup.2 by the microphone 13 after a predetermined timeout, to record this sound signal received S2.sub.1.sup.2 in the nonvolatile memory 18 by the sound card 16, to determine acoustic parameters CA.sup.P2 of the environment at the position P2 of the terminal RX2 on the basis of the sound signal emitted S1.sub.1 emitted by the enclosure TX and of the sound signal received S2.sub.1.sup.2 and to present an information message MSG to the user of the terminal, generated on the basis of these parameters.

(20) In a general manner, the emitted sound signals S1.sub.k are preferentially sound signals having a non-negligible spectral density for all the frequencies lying between 10 Hz and 22050 Hz and exhibiting good autocorrelation properties. In particular, the emitted sound signals S1.sub.k are noise, such as white noise or pink noise.

(21) By way of example, it is possible to consider: that a sound signal exhibits a non-negligible spectral density for all the frequencies lying between 10 Hz and 22050 Hz when each frequency in this range exhibits a spectral density of greater than 10% of the maximum spectral density; and that a sound signal exhibits good autocorrelation properties when it exhibits a very pronounced autocorrelation peak, in particular the value of the autocorrelation at times greater than 0.1 s is less than 2%, preferably less than 1%, more preferably less than 0.5% of the value of the autocorrelation at 0 s.

(22) In the embodiment described here, and as described with reference to FIG. 4, the sound signal emitted S1.sub.1 comprises, after a filling header P0 (or padding bits), a first portion P1.sub.1.sup.1, and then third portions P1.sub.1.sup.3n bracketed by second portions or tags P1.sub.1.sup.2.

(23) Each of these portions P1.sub.1.sup.1, P1.sub.1.sup.2, P1.sub.1.sup.3n must be chosen to be of sufficiently long duration with respect to the presumed reverberation time of a sound signal in the environment 100.

(24) The first portion P1.sub.1.sup.1 is in this example used to allow: a first estimation DHi* of the sampling frequency shift DHi; and an estimation of the temporal shift DTi,
between the emission device TX and each of the reception devices RXi.

(25) This first portion P1.sub.1.sup.1 is longer than the estimated temporal shift between the emission device TX and each of the reception devices RXi, typically of the order of 2 seconds.

(26) The second portions or tags P1.sub.1.sup.2 of the sound signal emitted S1.sub.1 make it possible, in this embodiment: to refine the estimation DHi* of the sampling frequency shift DHi; and to detect possible packet losses which could falsify the impulse responses which will be obtained on the basis of the third portions P1.sub.1.sup.3n.

(27) The second portions P1.sub.1.sup.2 are a set of MLS sequences of shorter durations.

(28) In the embodiment described here, each third portion P1.sub.1.sup.3n consists of the repetition of at least one sequence P1.sub.1.sup.3nm, so as to resolve or decrease the effects of non-linearity problems. The duration of each of the third portions P1.sub.1.sup.3n must be considerably greater than the reverberation time in the environment 100 in order for the measurements to be accurate.

(29) With reference to FIG. 5, we shall now describe the method implemented by each of the terminals RX1, RX2 in this particular embodiment.

(30) In the mode of embodiment described here, the computer program PG comprises two processes, a process PM corresponding to the Master mode executed by the terminal RX1 and a process PE corresponding to the Slave mode executed by the terminal RX2.

(31) In the course of a step E5, the user selects on the man-machine interface 15 of the terminal RX1 a context CT.sub.1 Classroom from the library of contexts LC of this terminal.

(32) In the course of a step E10, the processor 11 of the terminal RX1 determines a sound signal S1.sub.1 associated with this context in the library LS of sound signals.

(33) In the course of a step E15, the terminal RX1 dispatches, by wireless communication means 17, the reference of the sound signal S1.sub.1 to the terminal RX2. The terminal RX2 receives this reference in the course of a step F15.

(34) In the course of a step E20, the Master terminal RX1 dispatches the sound signal S1.sub.1 to the enclosure TX so that the latter plays it back in the form of a sound signal emitted S1.sub.1 in the environment 100.

(35) In the embodiment described here, the method implemented by the Master RX1 and Slave RX2 terminals (hereinafter RXj) is thereafter identical.

(36) In the course of a step E25, the terminals RXj wait a predetermined duration before beginning to record the sound signal received S2.sub.1.sup.j. This duration allows in particular the processing of step F15 by the terminal RX2 and the configuration of the microphone 13 and of the sound card 16 at reception for each of the terminals. The duration of the header P0 of the sound signal emitted S1.sub.1 is chosen so that the terminals RXj actually begin to record the sound signal received S2.sub.1.sup.j before the enclosure TX plays back the first portion of the sound signal emitted P1.sub.1.sup.1.

(37) In the embodiment described here, the sampling frequency f.sub.TX of the emission device TX is 44100 Hz.

(38) In practice, and as is known to a person skilled in the art, the analog sound signal to be emitted is processed by the processor or a component of the emission device TX, so as to obtain a table of values whose interval corresponds to this frequency f.sub.TX and which defines, for each instant t.sub.i, an integer value corresponding to the discretized amplitude A.sub.i having to be dispatched at the instant t.sub.i to the loudspeaker of the device TX to make it vibrate.

(39) Each of the terminals RXj stores the sound signal received S2.sub.1.sup.j that it receives in the course of a step E30 in its rewritable nonvolatile memory 18.

(40) Calling f.sub.RXj the sampling frequency of the reception device RXj, the processor 11 or the sound card 16 of this device constructs a table of values whose interval corresponds to this frequency f.sub.RXj and which defines, for each instant t, an integer value corresponding to the discretized amplitude A(t) of the sound signal received S2.sub.1.sup.j by the microphone of the device RX.sub.j at the instant t.

(41) In the embodiment described here, in the course of a step E35, the sound signal received S2.sub.1.sup.j is cut up into portions P2.sub.1.sup.j1, P2.sub.1.sup.j2, P2.sub.1.sup.j3n corresponding to the portions P1.sub.1.sup.1, P1.sub.1.sup.2, P1.sub.1.sup.3n of the sound signal emitted S1.sub.1.

(42) In the course of a step E40, the processor 11 of the terminal RXj performs a first estimation DHj* of the sampling frequency shift DHj between the emission device TX and the reception device RXj.

(43) This step consists in particular in maximizing, by varying the sampling frequency of the sound signal received (for example between 44000 Hz and 44100 Hz), the maximum amplitude of the impulse response obtained on the basis of the first portion P1.sub.1.sup.1 of the sound signal emitted S1.sub.1 and of the corresponding first portion P2.sub.1.sup.j1 of the sound signal received S2.sub.1.sup.j.

(44) The sampling frequency of the sound signal received S2.sub.1.sup.j making it possible to obtain this maximum amplitude, which corresponds to a first estimation DHj* of the sampling frequency shift DHj, is stored in the course of this step E40.

(45) This first estimation DHj* of the sampling frequency shift produces, by re-sampling of the sound signal received, a new table of values T2.sub.j*, for example by linear interpolation of the values of the table of values T2.sub.j.

(46) In the course of a step E45, the processor 11 of the terminal RXj estimates the temporal shift DTj between the emission device TX and the reception device RXj.

(47) This step of estimating the temporal shift DTj consists in particular, in this embodiment, in comparing: on the one hand, the position of the maximum amplitude of the impulse response obtained on the basis of the first portion P1.sub.1.sup.1 of the sound signal emitted S1.sub.1 and of the corresponding first portion P2.sub.1.sup.j1* of the sound signal received S2.sub.1.sup.j re-sampled at the sampling frequency stored in step E40, and on the other hand, the position of the maximum amplitude of the ideal impulse response obtained on the basis of the first portion P1.sub.1.sup.1 of the sound signal emitted S1.sub.1 and of a theoretical sound signal received identical to this first portion P1.sub.1.sup.1.

(48) The temporal disparity between these two positions, which corresponds to an estimation of the temporal shift DTj, is stored in the course of this step E45.

(49) In the course of a step E50, the processor 11 of the terminal RXj refines the first estimation DHj* of the sampling frequency shift DHj between the emission device TX and the reception device RXj.

(50) This step E50 is advantageously carried out after having corrected the temporal shift DTj stored in step E45, that is to say after having temporally shifted the whole of the signal received S2.sub.1.sup.j so as to bring the maximum amplitude of the impulse response obtained on the basis of the first portion P1.sub.1.sup.1 of the sound signal emitted S1.sub.1 and of the corresponding first portion P2.sub.1.sup.j1* of the sound signal received S2.sub.1.sup.j into correspondence with the position of the maximum amplitude of the ideal impulse response obtained on the basis of the first portion P1.sub.1.sup.1 of the sound signal emitted S1.sub.1 and of a theoretical sound signal received identical to this first portion P1.sub.1.sup.1.

(51) Step E50 then consists in particular, in this embodiment, in comparing for the various successive second portions or tags P1.sub.1.sup.2 of the sound signal emitted S1.sub.1: on the one hand, the position of the maximum amplitude of the impulse response obtained on the basis of the second portion P1.sub.1.sup.2 of the sound signal emitted S1.sub.1 and of the corresponding second portion P2.sub.1.sup.j2* of the sound signal received S2.sub.1.sup.1 re-sampled at the sampling frequency stored in step E40, and on the other hand, the position of the maximum amplitude of the ideal impulse response obtained on the basis of the second portion P1.sub.1.sup.2 of the sound signal emitted S1.sub.1 and of a theoretical sound signal received identical to this second portion P1.sub.1.sup.2.

(52) If the temporal disparity between these two positions evolves over the various successive second portions or tags P1.sub.1.sup.2 of the sound signal emitted S1.sub.1, the estimation of the sampling frequency shift DHj can be refined by determining the sampling frequency for which a homogeneous temporal disparity is obtained between said two positions for all the successive second portions or tags P1.sub.1.sup.2 of the sound signal emitted S1.sub.1.

(53) The sampling frequency of the sound signal received S2.sub.1.sup.j making it possible to obtain this homogeneous temporal disparity for all the successive second portions or tags P1.sub.1.sup.2, which corresponds to a refined estimation of the sampling frequency shift DHj, is stored in the course of this step E50.

(54) In the course of this step E50, once the aforementioned temporal disparity is homogeneous for all the successive second portions or tags P1.sub.1.sup.2, it is also possible to refine the estimation of the temporal shift DTj between the emission device TX and the reception device RXj, by temporally shifting the whole of the signal received S2.sub.1.sup.j so as to bring the maximum amplitude of the impulse response obtained on the basis of each second portion P1.sub.1.sup.2 of the sound signal emitted S.sub.1 and of the corresponding second portion P2.sub.1.sup.j2* of the sound signal received S2.sub.1.sup.j into correspondence with the position of the maximum amplitude of the ideal impulse response obtained on the basis of each second portion P1.sub.1.sup.2 of the sound signal emitted S1.sub.1 and of a theoretical sound signal received identical to this second portion P1.sub.1.sup.2.

(55) In the course of a step E52, a sampling frequency shift correction and a temporal shift correction are applied to the sound signal received S2.sub.1.sup.j on the basis of the previously obtained estimations of the sampling frequency shift DHj and of the temporal shift DTj, so as to produce a corrected sound signal received S3.sub.1.sup.j* within the meaning of the invention.

(56) In the embodiment described here, the method according to the invention also comprises a step E55 of deleting invalid sequences of the corrected sound signal received S3.sub.1.sup.j*, these invalid sequences corresponding to losses of packets of the sound signal emitted S1.sub.1.

(57) In the embodiment described here, this detection is performed by verifying that the duration between two consecutive tags for the impulse response obtained on the basis of at least one portion of the sound signal emitted S1.sub.1 comprising these two tags and of a corresponding portion of the corrected sound signal received S3.sub.1.sup.j*, is substantially equal to the duration between the corresponding two tags P1.sub.1.sup.2 for the sound signal emitted S1.sub.1.

(58) When a duration between two consecutive tags for the impulse response deviates from the duration between the corresponding two tags P1.sub.1.sup.2 for the sound signal emitted S1.sub.1 beyond a predetermined threshold, it is considered that packets have been lost between these tags; the corresponding third portions of sound signal emitted P1.sub.1.sup.3n and P2.sub.1.sup.j3n lying between these tags are eliminated both in the sound signal emitted S1.sub.1 and in the corrected sound signal received S3.sub.1.sup.j.

(59) The corrected sound signal received S3.sub.1.sup.j* cleaned of these lost packets is denoted S3.sub.1.sup.j.

(60) It is recalled that, in this example, a third portion P1.sub.1.sup.3n of the sound signal emitted S1.sub.1, lying between two tags P1.sub.1.sup.2, consists of the repetition of at least one MLS sequence P1.sub.1.sup.3nm.

(61) In the embodiment described here, in the course of a step E60, the impulse response is calculated on the basis of each of these sequences P1.sub.1.sup.3nm, P2.sub.1.sup.j3nm, the average of these impulse responses within the third portion P1.sub.1.sup.3n is calculated for all the values of m, and then finally the average of these averages is calculated for all the values of n, stated otherwise for the set of third portions P1.sub.1.sup.3n, P2.sub.1.sup.j3n of the sound signals S1.sub.1, S3.sub.1.sup.j.

(62) The impulse response RI.sup.Pj of the environment 100 at the position of the reception device RXj (RI.sup.P1, RI.sup.P2 in this example) is thus obtained.

(63) This impulse response makes it possible to calculate, in the course of a step E65, one or more parameters CA.sup.Pj of the environment, for example acoustic parameters such as recalled previously defined by ISO standard 3382.

(64) In the particular embodiment described here, the method according to the invention comprises an expert system configured to produce, on the basis of this or these acoustic parameters, an information message relating to acoustic comfort, this message being explicit for a user who is not a specialist in acoustics. This message can for example inform the user about the comfort of the environment under certain conditions (sensitivity to noise from traffic, to nearby noise, etc.).

(65) This message is played back by the man-machine interface 15 of the terminal in the course of a step E70.

(66) In an advantageous manner, the method and the system according to the invention can also comprise a module configured to provide a user, on the basis of the data harvested about the environment and of the environment's acoustic parameters determined according to the invention, with a simulation of modified acoustic comfort after changing the insulation of the environment, in particular after renovation.

Second Embodiment

(67) In the first embodiment described hereinabove, the emission device is an enclosure TX able to excite the air in the environment 100. More precisely, the sound signal emitted S1.sub.1 is generated in the air by a membrane of the enclosure and propagates initially in the air, and then possibly in certain solids of the environment (partitions, floors, etc.), as far as the reception devices RX1, RX2.

(68) As a variant, in the second embodiment represented in FIG. 6, the emission device is a surface transducer TD able to excite a solid which it is adjoining in the environment 100. In particular, in the example represented in FIG. 6, the transducer TD is adjoining a floor of a first room of the environment 100. The signal emitted generated by the transducer TD in the solid with which it is associated (here the floor) is a sound signal within the meaning of the invention. This sound signal emitted S1.sub.1 propagates initially in the solid (floor), then in the air, and then possibly in other solids of the environment, as far as one or more reception devices.

(69) In the example of FIG. 6, the sound signal propagates in particular as far as a reception device RX1 positioned at a position P1 in a second room of the environment 100, situated under the floor excited by the transducer TD. Such an arrangement can make it possible to evaluate the structure-borne noise transmission through a partition (the floor in the example of FIG. 6), and to thus determine the properties of this partition in respect of impact noise insulation.

(70) In the example of FIG. 6, the transducer TD receives the sound signal to be emitted from a terminal T positioned in the first room, where the transducer TD is installed on the floor of this first room. The transducer TD comprises communication means, not represented, compatible with those of the terminal T, wireless communication means in particular, so as to be able to receive from this terminal a sound signal to be played back in the form of a sound signal emitted S1.sub.1 in the environment 100. In this example, the transducer TD is an autonomous device comprising in a conventional manner a vibrator, a battery, a power management circuit, a wireless connectivity module, a digital-analog converter, an audio amplifier. Moreover, the terminal T positioned in the first room and the reception device RX1 positioned in the second room are each identical or similar to the terminal 10 of FIG. 1 in respect of the functional parts in common with the latter.

(71) In the course of a step analogous to step E5 of FIG. 5, the user selects on the man-machine interface 15 of the terminal T positioned in the first room a context Rooms separated by a floor from the library of contexts LC of this terminal and, in the course of a step analogous to step E10 of FIG. 5, the processor 11 of the terminal T determines a sound signal associated with this context in the library LS of sound signals. In the course of a step analogous to step E15, the terminal T dispatches thereafter, by wireless communication means 17, the reference of the sound signal S1.sub.1 to the terminal RX1. The terminal RX1 receives this reference in the course of a step analogous to step F15. Thereafter, in the course of a step analogous to step E20 of FIG. 5, the terminal T dispatches the sound signal to the transducer TD so that the latter plays it back in the form of a sound signal emitted S1.sub.1 in the floor of the first room. The following steps of the method implemented by the terminal RX1 for receiving the sound signal received S2.sub.1.sup.1 positioned in the second room situated under the floor is identical to that detailed previously with reference to FIG. 5, comprising steps E25 to E70.

Third Embodiment

(72) In the embodiments described hereinabove, the user selects a measurement context from a predefined list of the man-machine interface 15.

(73) As a variant, in a third embodiment, the man-machine interface can present a more advanced menu, for example in an expert mode, allowing the user to define parameters of the environment, for example relating to the geometry (surface, volume) of the environment, to its structure (types of materials), or to the distance between the emission device TX, TD and the reception device or devices RX.

(74) In this third embodiment, the sound signal emitted S1.sub.k is optimized as a function of these parameters of the environment.

Fourth Embodiment

(75) In a fourth embodiment, the method comprises a step of filtering the sound signal emitted S1.sub.k and a corresponding step of filtering the corrected sound signal received S3.sub.k.sup.j.

(76) These steps can respectively be implemented after steps E10 and E55 of the method of FIG. 5. When this filtering step is performed, the Master terminal informs the Slave terminal thereof in the message dispatched in step E15.

(77) This filtering can in particular make it possible, by eliminating certain frequencies, to optimize the duration of the measurement and to render the sound signal emitted S1.sub.k more pleasant for the user by eliminating frequencies that are annoying to a human.

Fifth Embodiment

(78) In a fifth embodiment, the method differs from that of the first embodiment in that step E50 of refining the first estimation DHj* of the sampling frequency shift DHj between the emission device TX and the reception device RXj is implemented directly after the first estimation DHj* of the sampling frequency shift, without passing via the step of estimating the temporal shift DTj. Stated otherwise, step E50 of refining the first estimation DHj* of the sampling frequency shift is implemented without prior correction of the temporal shift DTj.

(79) This fifth embodiment requires, however, that the successive second portions or tags P1.sub.1.sup.2 of the sound signal emitted S1.sub.1 be longer than the temporal shift DTj, so that an overlap exists between each second portion P1.sub.1.sup.2 of the sound signal emitted S1.sub.1 and the corresponding second portion P2.sub.1.sup.j2* of the sound signal received S2.sub.1.sup.j re-sampled at the sampling frequency stored in step E40.

(80) The invention is not limited to the examples described and represented.

(81) In particular, in the first embodiment, two reception devices RX1 and RX2 are provided so as to determine acoustic parameters of the environment at two distinct positions. As a variant, a method and a system according to the invention can involve the reception of a sound signal received by a single reception device, in particular a single terminal which executes the steps of the process PM corresponding to the Master mode such as described previously, with the exception of step E15 of transmitting the reference of the sound signal S1.sub.1 to another reception device since in this case there is only a single reception device.

(82) Moreover, in the first embodiment described with reference to FIG. 2, the emission device TX is an autonomous enclosure. As a variant, the emission device TX can be a terminal which is identical or similar to the terminal 10 of FIG. 1 in respect of the functional parts in common with the latter, which emits the signal emitted S1.sub.1 by way of its loudspeaker 14.

(83) Moreover, in the case where several reception devices RXj are provided so as to determine acoustic parameters of the environment at distinct positions Pj, the method and the system according to the invention can be configured to present to the user on a central terminal (for example the Master terminal in the first embodiment) a single message which can comprise the set of local parameters CA.sup.Pj of the environment, optionally one or more characteristic global parameters of the environment, and optionally an information message relating to acoustic comfort such as mentioned previously.

(84) Finally, the method and the system according to the invention can be implemented to obtain at least one acoustic parameter in various types of environment, for example in a building or in a vehicle.

(85) Annex 1

(86) This annex provides the characteristics of a signal emitted S1.sub.1 that is usable in a particular embodiment of the invention.

(87) In this example, each portion P1.sub.1.sup.k of the sound signal emitted S1.sub.1 is composed of one or more sequences of maximum length of a linear feedback shift register of maximum length, also called a Maximum Length Sequence (MLS).

(88) For a description of the properties of these sequences, as well as for the way of generating them, the person skilled in the art will be able to refer to the document Shift register sequences, Solomon Wolf Golomb, Holden-Day, 1967.

(89) These sequences are characterized by a so-called characteristic polynomial, knowledge of which makes it possible to generate a unique sequence. Hereinafter, each polynomial will be characterized by a list of its non-zero degrees.

(90) For example, the polynomial x.sup.5+x.sup.4+x.sup.2+1 will be represented by [5, 4, 2, 0].

(91) For a sequence of degree n (maximum degree of the characteristic polynomial), its length will be 2.sup.n1. A sequence will then consist of a string of size 2.sup.n1 of 0s and 1s.

(92) In order to generate a sound signal, it is preferable (but not necessary) to multiply all the elements of this sequence by 2 and to deduct 1 from them, so as to obtain a string of 1 and 1s.

(93) An exemplary sequence typically used to determine the acoustic characteristics of an average room such as an office, consists of the following successive signal portions: Portion P0: empty Portion P1.sub.1.sup.1: string of 2 MLS sequences generated with [15, 14, 0]. In this particular case, each bit of the sequence is repeated 4 times: 001011->0000 0000 1111 0000 1111 1111 Portion P1.sub.1.sup.2: string of 5 MLS sequences generated with [13, 12, 11, 8, 0]. Portion P1.sub.1.sup.31: string of 5 MLS sequences such that: Portion P1.sub.1.sup.311: 1 MLS sequence generated with [16, 15, 13, 4, 0]. Portion P1.sub.1.sup.312: 1 MLS sequence generated with [16, 15, 12, 10, 0]. Portion P1.sub.1.sup.313: 1 MLS sequence generated with [16, 15, 12, 1, 0]. Portion P1.sub.1.sup.314: 1 MLS sequence generated with [16, 15, 10, 4, 0]. Portion P1.sub.1.sup.315: 1 MLS sequence generated with [16, 15, 9, 6, 0]. Portion P1.sub.1.sup.2: string of 5 MLS sequences generated with [13, 12, 11, 8, 0]. Portion P1.sub.1.sup.32: string of 5 MLS sequences such that: Portion P1.sub.1.sup.321: 1 MLS sequence generated with [16, 15, 9, 4, 0]. Portion P1.sub.1.sup.322: 1 MLS sequence generated with [16, 15, 7, 2, 0]. Portion P1.sub.1.sup.323: 1 MLS sequence generated with [16, 15, 4, 2, 0]. Portion P1.sub.1.sup.324: 1 MLS sequence generated with [16, 11, 13, 11, 0]. Portion P1.sub.1.sup.325: 1 MLS sequence generated with [16, 14, 13, 5, 0]. Portion P1.sub.1.sup.2: string of 5 MLS sequences generated with [13, 12, 11, 8, 0]. Portion P1.sub.1.sup.33: string of 5 MLS sequences such that: Portion P1.sub.1.sup.331: 1 MLS sequence generated with [16, 14, 12, 7, 0]. Portion P1.sub.1.sup.332: 1 MLS sequence generated with [16, 14, 11, 7, 0]. Portion P1.sub.1.sup.333: 1 MLS sequence generated with [16, 14, 9, 7, 0]. Portion P1.sub.1.sup.334: 1 MLS sequence generated with [16, 14, 9, 4, 0]. Portion P1.sub.1.sup.335: 1 MLS sequence generated with [16, 14, 8, 3, 0]. Portion P1.sub.1.sup.2: string of 5 MLS sequences generated with [13, 12, 11, 8, 0]. Portion P1.sub.1.sup.34: string of 5 MLS sequences such that: Portion P1.sub.1.sup.341: 1 MLS sequence generated with [16, 13, 12, 11, 0]. Portion P1.sub.1.sup.342: 1 MLS sequence generated with [16, 13, 12, 7, 0]. Portion P1.sub.1.sup.343: 1 MLS sequence generated with [16, 13, 11, 6, 0]. Portion P1.sub.1.sup.344: 1 MLS sequence generated with [16, 13, 9, 6, 0]. Portion P1.sub.1.sup.345: 1 MLS sequence generated with [16, 13, 6, 4, 0]. Portion P1.sub.1.sup.2: string of 5 MLS sequences generated with [13, 12, 11, 8, 0].