Classifications

• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
  • H04R1/00—Details of transducers, loudspeakers or microphones
  • H04R1/20—Arrangements for obtaining desired frequency or directional characteristics
  • H04R1/32—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only
  • H04R1/40—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
  • H04R1/406—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers microphones
• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
  • H04R1/00—Details of transducers, loudspeakers or microphones
  • H04R1/20—Arrangements for obtaining desired frequency or directional characteristics
  • H04R1/32—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only
  • H04R1/40—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
  • H04R1/403—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers loud-speakers
• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
  • H04R3/00—Circuits for transducers, loudspeakers or microphones
  • H04R3/005—Circuits for transducers, loudspeakers or microphones for combining the signals of two or more microphones
• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
  • H04R2201/00—Details of transducers, loudspeakers or microphones covered by H04R1/00 but not provided for in any of its subgroups
  • H04R2201/40—Details of arrangements for obtaining desired directional characteristic by combining a number of identical transducers covered by H04R1/40 but not provided for in any of its subgroups
  • H04R2201/401—2D or 3D arrays of transducers
• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
  • H04R2201/00—Details of transducers, loudspeakers or microphones covered by H04R1/00 but not provided for in any of its subgroups
  • H04R2201/40—Details of arrangements for obtaining desired directional characteristic by combining a number of identical transducers covered by H04R1/40 but not provided for in any of its subgroups
  • H04R2201/403—Linear arrays of transducers
• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
  • H04R2201/00—Details of transducers, loudspeakers or microphones covered by H04R1/00 but not provided for in any of its subgroups
  • H04R2201/40—Details of arrangements for obtaining desired directional characteristic by combining a number of identical transducers covered by H04R1/40 but not provided for in any of its subgroups
  • H04R2201/405—Non-uniform arrays of transducers or a plurality of uniform arrays with different transducer spacing

Definitions

• the present invention relates to the field of sound recording and sound reproduction. It relates in particular to a system which can be used as a sound sensor only, or else, simultaneously for capturing and restoring sound.
• the targeted applications are:
• the system makes it possible to replace one or more microphones, guaranteeing a high quality of the sound picked up, and allowing arbitrary directivity diagrams
• microphones microphones
• sound reproduction microphones
• the invention aims to remedy the above problems by an effective solution which does not require a pointing solution.
• the invention relates, according to its most general meaning, to a system for processing a sound signal comprising an amplifier, an analog-digital conversion circuit or a digital-analog conversion circuit, a filtering circuit, elements for end processing and a circuit for summing the filtered signals, so as to deliver at least one output signal, said end processing elements being sound sensors or loudspeakers, characterized in that the filtering circuit is optimized as a function of the specific characteristics of the end processing elements and as a function of their geometric location and in that it comprises at least one directional lobe and an output circuit delivering an output signal by directional lobe.
• the processing system is a system for acquiring a sound signal and in that the end processing elements are sound sensors.
• the sound sensors are distributed along a plurality of concentric axes.
• the sound sensors are distributed along a plurality of coplanar axes.
• the sound sensors are distributed along three axes forming an angle of 120 ° between them.
• the sound sensors are distributed along five axes oriented respectively with an angle of 0 °, ⁇ 30 °, ⁇ 110 ° relative to a reference axis.
• the output circuit delivers several output signals each corresponding to a directional lobe, these different output signals being recomposed to form a single signal corresponding to the sum of the directional signals weighted by the energy of the corresponding lobe.
• the output circuit delivers several output signals each corresponding to a directional lobe, these different output signals being recomposed to form a single signal corresponding to:
• WHERE designates the amplitude of the output signal as a function of the frequency
• L n ⁇ (f) designates a weighting factor as a function of the identifier of the first lobe, and of the angle of the source
• L n + 1 / ⁇ (f) designates a weighting factor as a function of the identifier of the second lobe, and of the angle of the source
• S n + 1 (f) denotes the amplitude of the signal from the second lobe.
• the output circuit delivers several output signals each corresponding to a directional lobe, these different output signals each being interpreted in a differentiated manner without recomposition.
• the processing system is a system for restoring a sound signal and in that the processing elements at the ends are loudspeakers.
• the loudspeakers are distributed along a plurality of concentric axes.
• the loudspeakers are distributed along a plurality of coplanar axes.
• the loudspeakers are distributed along three axes forming an angle of 120 ° between them.
• the loudspeakers are distributed along five axes oriented respectively with an angle of 0 °, ⁇ 30 °, ⁇ 110 ° relative to a reference axis.
• the output circuit diffuses several output signals each corresponding to a directive lobe, each output signal being diffused in its directive lobe.
• the processing system is a sound acquisition and restitution system, in that the end processing elements are loudspeakers or sound sensors and in that it comprises filters for acquisition and restitution filters.
• the acquisition filters are optimized so as not to receive the signal emitted by the loudspeakers and in that the reproduction filters are optimized so as not to send a signal to the sound sensors.
• the geometry of the loudspeakers is positioned on an axis perpendicular to the plane of the sound sensors.
• the sound sensors are placed on hyperbolas located in vertical planes.
• FIG. 1 shows a schematic view of the logarithmic placement of the sensors on a straight line
• - Figure 2 shows a schematic view of the placement of the sensors on a set of lines from a common origin
• - Figure 3 shows a schematic view of the placement of the speakers relative to the microphones
• FIG. 4 shows a schematic view of the placement of the sensors on a hyperbola
• FIG. 5 shows a schematic view of a structure of the acquisition device with a single output channel
• - Figure 6 shows a schematic view of a structure of the acquisition device with several output channels
• - Figure 7 shows a schematic view of a structure of the channel selection device by energy
• - Figure 8 shows a schematic view of a structure of the channel selection device by position
• - Figure 9 shows a schematic view of a structure of the rendering device
• FIG. 10 shows a schematic view of a structure of the complete device with acquisition on a channel and restitution
• - Figure 11 shows a schematic view of a structure of the complete device with acquisition on several channels and restitution
• FIG. 12 represents a schematic view of the hypotheses on the natural noise of the sensors
• FIG. 13 represents a schematic view of the restitution device with several exit channels, each being diffused in a directive lobe.
• the invention relates to a system for sound recording, or for sound recording and sound reproduction. It implements a plurality of sensors, in particular microphones, forming a one, two or three-dimensional network.
• This system is digital and consists of: - a part intended for taking sounds, a part intended for the reproduction of sounds, a part ensuring the coupling between the two parts above.
• the invention lies in particular in the placement of these sensors in space, to form a network with well-defined geometric characteristics.
• FIG. 6 The part ensuring the taking of sounds is shown in FIG. 6 schematically. It consists of a set of M sensors (1, 11, 21, 31). These sensors are made up of microphonic cells, for example with electrets providing an analog electrical signal from the captured sound. Each sensor is connected to an optional pre-amplification stage (2, 12, 22, 32), also ensuring high-pass filtering if necessary in order to attenuate the components whose frequencies are between 0 Hz and the cut-off frequency of this high-pass filter, this cut-off frequency generally being between 20 Hz and 200 Hz.
• These analog-digital converters are connected to a digital delay line type circuit (4, 14, 24, 34) making it possible to delay each signal by a fixed number of sampling periods, the number possibly being different for each signal.
• Each digital delay line is connected to a circuit (5, 15, 25, 35) enabling these signals to be distributed to one or more signal processing microprocessors (DSP for "Digital Signal Processor"),
• DSP signal processing microprocessor
• Each DSP is connected to a circuit allowing the digital signal from the DSP to be transmitted to, for example, a recorder or a mixing console (this circuit could for example be a transmitter in AES-3 format) or even to a digital telecommunications network (of ISDN type, or of type TCP / IP).
• the processing carried out by the DSP (s) consists in: filtering by a digital filter, each signal from a sensor, - adding all of the - ⁇ filtered signals,
• the part ensuring the reproduction of sounds is represented by figure 9. It consists of a signal processing processor (DSP) receiving the digital signal to be reproduced, and manufacturer using P filters digital (100, 110, 120, 130), P filtered versions of this signal to be reproduced.
• DSP signal processing processor
• a digital delay line type circuit (101, 111, 121, 131) makes it possible to delay each signal by a fixed number of sampling periods, the number possibly being different for each signal.
• Each digital delay line is connected to a digital-analog converter (102, 112, 122, 132) responsible for transforming the series of samples at the rate of F e per second, into an analog electrical signal,
• the converters are connected to an optional amplification stage (103, 113, 123, 133).
• Each of the amplifiers is connected to a speaker providing sound from the analog electrical signal.
• the part ensuring the reproduction of the sounds is presented in FIG. 13 in a variant which allows several signals to be diffused simultaneously, each of them being diffused in a directive lobe which is specific to it.
• 3 different signals are shown by way of example, but the device can be produced with any number of input signals. This number will generally be lower than the number P of the loudspeakers (otherwise, the directional lobes would partially overlap).
• the amplification stage (103, 113, 123, 133) is no longer optional, and moreover each amplifier is a summator which performs the sum of the signals present on its inputs and amplifies this sum.
• the signal 1 to be restored is received by a signal processing processor (DSP) which processes it as in the case of FIG. 9.
• DSP signal processing processor
• the DSP manufactures using the P digital filters (100, 110, 120, 130) the P filtered versions of signal 1 to be restored. These filtered versions are delayed by a digital delay line type circuit (101, 111, 121, 131). They are then transformed into an analog electrical signal by the converters (102, 112, 122, 132) and these electrical signals are supplied to one of the inputs of the amplifiers (103, 113, 123, 133) effecting the sum of their inputs.
• P digital filters make the P versions which are delayed in the digital delay lines (201, 211, 221, 231), then converted into analog electrical signals from converters (202, 212, 222, 232) and presented at the input of amplifiers (103, 113, 123, 133).
• P digital filters manufacture the P versions which are delayed in the digital delay lines (301, 311, 321, 331), then converted into an analog electrical signal by the converters (302, 312, 322, 332) and presented at the input of the amplifiers (103, 113, 123, 133).
• the positions of the sensors must respect two contradictory constraints, and the placement of the sensors will result from a compromise between these constraints.
• the minimum spacing between sensors will be 0.024 m for a direction of sight orthogonal to the axis carrying the sensors, and 0.012 m for a direction of sight in the axis of the sensors. These spacings are indicative and may be modified depending on constraints related to the physical size of the sensors used. It is desirable that the sensors are as far apart as possible, in order to guarantee the best possible spatial resolution. Typically, the sensors would have to cover a space the size of which is several times the longest wavelength in the signal.
• the minimum size of the sensor network will be 3.36 m. Such a size will often be incompatible with the real constraints of the application. This leads to a logarithmic distribution of the sensors.
• FIG. 1 represents a first example of placement of sensors along an axis to form a one-dimensional network.
• the spacing of two consecutive microphones (1 to M) obeys a law of logarithmic progression.
• the sensors will be placed in such a way that the ratio between two successive spacings is constant:
• An effective alternative embodiment of the device includes an optimized placement of the sensors. We suggest that the microphonic sensors be located on a set of rays from a point of origin.
• Figure 2 shows the placement of the sensors in this configuration.
• Two examples are illustrated (12 sensors distributed over 3 rays, and 16 sensors distributed over 5 rays, with a sensor in the center).
• the first example can make it possible to form 6 tracks pointing in the directions 0 °, 60 °, 120 °, 180 °, 240 ° and 300 °.
• the second example can make it possible to form 5 channels pointing to the directions 0 °, 30 °, 110 °, 250 ° and 330 °, which are the preferred directions for the reproduction of surround signals, or 5.1.
• Another possible embodiment of the device consists in placing the sensors on a circle or on several concentric circles. We can then choose to distribute them uniformly on each circle. So if we have Q circles, and if the circle q has M q sensors, the angular difference between two successive sensors on this circle will be constant and equal to:
• the speakers are located on a radius from an origin point.
• a common origin will advantageously be chosen for the spokes carrying the microphonic sensors, and for the ray carrying the loudspeakers.
• Figure 3 shows the placement of the sensors in this configuration.
• Microphonic sensors can also be found on hyperbolas.
• the sensors are then located on a hyperboloid whose loudspeakers 1 and 2 constitute the foci.
• Figure 4 illustrates the placement of the sensors in this configuration.
• the two speakers h x and h 2 are located on a vertical line.
• Other sensors can be located on one or more other identical hyperbolas, located in other vertical planes passing through the two loudspeakers.
• V m ( m (* - ⁇ m ), me [1,].
• the received signals X m (f) are first delayed to give the signals Y m (f) which are then filtered to give the signals Z m (f); these are then summed to give the output S (f):
• Filters are optimized using a set of directions: - the direction of the target source,
• a source If a source is in one of the directions of the neighborhood, it will be received with a gain close to 1 (in general slightly less than 1). If a source is in one of the directions to be rejected, it will be treated so as to have a very low gain.
• the optimization of the filters is done under the constraint that the directivity diagram has a value of 1 in the direction of the target source.
• the object of these constraints is to allow the directivity diagram to keep a constant shape for all the frequencies above the reference frequency. Without this constraint, the minimization of the criterion J would lead to values of the directivity diagram which would be greater than 1 for the directions in the region.
• the sensors are all affected by noise, called own noise, which comes from the operation of the sensor itself and not from the sounds picked up.
• the processing undergone by the signals picked up in our device risks increasing the level of the own noise of the sensors.
• the signal received is the sum of two signals, that corresponding to the source Y u (f) and that corresponding to the noise B (£)
• Y (f) Y h (f) + Y u (f).
• the signal received from the useful source is expressed as the signal U (f) propagating in the channel whose frequency response is C ⁇ (f), then amplified by the gain of the sensors
• the own noise of the sensors does not propagate in the acoustic channel, being born inside the sensor, but it is also amplified by the gain of the sensors, delayed on each sensor, which represents the matrix V (f), before being processed by the filters H (f).
• the output channels are indexed by ne [1, N].
• y n, m (t), me [1, M], ne [1, N] is filtered by a digital filter (finite impulse response filter of length L) and the result is written:
• This processing structure is shown in FIG. 6.
• N parallel channels When the acquisition is done on N parallel channels, a single channel is extracted from these N channels which will be the signal leaving the device (for example in a video conference or audio conference application, this will be the signal which after compression will be sent to the distant correspondent).
• the objective is to reconstruct a signal s (t) by linear combination between the signals s n (t), ne [1, N].
• the selected combination will be noted:
• N s (t) ⁇ w n (f) s n (t)
• W t + i W t + ⁇ Grac ⁇ W RAW,
• Grad w W t ⁇ R t W t 2R t W t , again:
• W t + 1 W t + 2 ⁇ R t W t .
• This processing structure is shown in FIG. 7.
• the source position can be identified by conventional methods, for example that presented by Y. Grenier, P.Loubaton, "Localization of broadband sources by temporal methods", 12th Symposium on Signal Processing and its Applications, GRETSI, Juan-les-Pins, pp 457-460, 1989.
• Each exit route is associated with a direction
• FIG. 8 A simplified version is obtained by supposing that the functions L n ⁇ (f) and L n + l ⁇ (f) are independent of " the f equence and reduced to two gains L n ⁇ and L n + l ⁇ . The minimization of the criterion is then done on all frequencies and no longer frequency by frequency. The two gains are then solution of:
• This processing structure is shown in FIG. 9.
• the signal to be restored R (/) is converted into P offset versions U p (f), which are then filtered to give the signals V (/) which will be restored.
• the received signal R ⁇ (f) is the sum of the contributions of these P restored signals filtered by the corresponding acoustic channel C p ⁇ (f) (channel between the speaker p and the measuring point at position ⁇ ):
• R e (f) ⁇ C pt ⁇ (f) H p (f J2 ⁇ fî> R (f).
• G (f) represents the gain of the loudspeaker at frequency f.
• the optimization of the filters is done from a set of directions: the preferred return direction, - the return directions which are located within a neighborhood, constituted by the directions that define the main lobe, of the directivity diagram, the directions of the sources to reject, (it is important that the cardinal of this set is greater than the number of speakers).
• the optimization of the filters is done under the constraint that the directivity diagram has a value of 1 in the direction of the target source.
• terminals ⁇ ⁇ will generally be made by making them equal to the squares of the amplitude of the directivity diagram at the reference frequency f r :
• the object of these constraints is to allow the directivity diagram to keep a constant shape for all the frequencies above the reference frequency.
• the directivity diagrams apply to each of the signals to be reproduced.
• the device comprising both the acquisition and the reproduction of sounds, an acoustic coupling occurs between the reproduction part and the acquisition part.
• the signal output from the speakers will be picked up by the microphones.
• Another more efficient echo cancellation algorithm was presented in the article E. Moulines, O. Ait Amrane, Y. Grenier, "The Generalized Multi-delay Adaptive Filter: Structure and Convergence Analysis.”, IEEE Trans. on Signal Processing, Vol. 43, no. 1, pp 14-18, January 1995.
• FIG. 10 shows a complete view of the device, including the acquisition of the sound with an output channel (as in FIG. 5), an acoustic echo cancellation block, the reproduction of the sound as in FIG. 9, and a mixing block as in figure 7 or figure 8.
• FIG. 11 shows a complete view of the device, including the acquisition of the sound with several output channels (as in FIG. 6), an acoustic echo cancellation block, the reproduction of the sound as in FIG. 9, and a mixing block as in figure 7 or figure 8.
• Joint frequency optimization criteria When the device performs both acquisition and restitution, the two criteria are modified to take account of this situation.
• the optimization of the acquisition will have an additional goal: to cancel the signals received from the loudspeaker.
• the optimization of the reproduction will have the additional aim of not restoring the signal to the sensors for acquisition.
• C mp (f) denote the transfer function (Fourier transform of the impulse response) of the acoustic channel between the sensor m and the speaker p.

Landscapes

• Health & Medical Sciences (AREA)
• Otolaryngology (AREA)
• Physics & Mathematics (AREA)
• Engineering & Computer Science (AREA)
• Acoustics & Sound (AREA)
• Signal Processing (AREA)
• General Health & Medical Sciences (AREA)
• Circuit For Audible Band Transducer (AREA)
• Electrophonic Musical Instruments (AREA)
• Push-Button Switches (AREA)

Applications Claiming Priority (3)

Publications (2)

Family

ID=8868780

Family Applications (1)

Country Status (4)

Families Citing this family (5)

Family Cites Families (5)

• 2001
  • 2001-10-26 FR FR0113896A patent/FR2831763B1/fr not_active Expired - Fee Related
• 2002
  • 2002-10-25 WO PCT/FR2002/003685 patent/WO2003037034A1/fr not_active Application Discontinuation
  • 2002-10-25 EP EP02791898A patent/EP1438871B1/de not_active Expired - Lifetime
  • 2002-10-25 AT AT02791898T patent/ATE540537T1/de active

Non-Patent Citations (1)

Also Published As

Similar Documents

Legal Events