Classifications

• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04S—STEREOPHONIC SYSTEMS 
  • H04S7/00—Indicating arrangements; Control arrangements, e.g. balance control
  • H04S7/30—Control circuits for electronic adaptation of the sound field
  • H04S7/302—Electronic adaptation of stereophonic sound system to listener position or orientation
• • 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
  • 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
  • H04S—STEREOPHONIC SYSTEMS 
  • H04S2400/00—Details of stereophonic systems covered by H04S but not provided for in its groups
  • H04S2400/15—Aspects of sound capture and related signal processing for recording or reproduction
• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04S—STEREOPHONIC SYSTEMS 
  • H04S2420/00—Techniques used stereophonic systems covered by H04S but not provided for in its groups
  • H04S2420/11—Application of ambisonics in stereophonic audio systems
• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04S—STEREOPHONIC SYSTEMS 
  • H04S7/00—Indicating arrangements; Control arrangements, e.g. balance control
  • H04S7/30—Control circuits for electronic adaptation of the sound field
  • H04S7/302—Electronic adaptation of stereophonic sound system to listener position or orientation
  • H04S7/303—Tracking of listener position or orientation
  • H04S7/304—For headphones

Definitions

• the present invention relates to the field of sound signal processing.
• the most compact solution is the use of a network of microphones, such as the eigenmike of mh acoustics, the Soundfield of TSL Products, and the TetraMic Core Sound. Equipped with four to thirty-two microphones, these products are expensive and therefore reserved for professional use. Recent research has reduced the number of microphones (Palacino, JD, & Nicol, R. (2013).) "Spatial sound pick-up with a low number of microphones.” ICA 2013. Montreal, Canada. Microphones of reduced size and cost can be used such as those available to mobile phones.
• the shape of the microphone arrays remains standardized, however, from the dodecahedron for the EigenMike to the tetrahedron for the Soundfield and the TetraMic.
• This geometric shape makes it possible to use simple formulas to convert microphone signals into a format ambisonic, and were developed by Gerzon in 1975 (Gerzon, M. (1975). "The design of precisely coincide microphone arrays for stereo and surround sound.” 50th Audio Engineering Society Conference.).
• the ambisonic format is a set of audio channels that contains all the information necessary for the spatial reconstruction of the sound field.
• a novelty provided by this patent is the possibility of using any form of microphone array.
• the present invention intends to overcome the drawbacks of the prior art by proposing a method of processing the sound signal making it possible to capture the sound signal in all directions and then to restore the sound signal.
• the present invention relates, in its most general sense, to a sound signal processing method, characterized in that it comprises the following steps:
• the matrix calculation involves a matrix H calculated by the least squares method from the measured directivities of the N microphones and the ideal directivities of the ambison components.
• said microphones are arranged in a circle on a plane, spaced at an angle equal to 360 ° / N or at each corner of a mobile phone.
• said method implements four microphones spaced at an angle of 90 ° to the hoirzontal.
• said method implements a filter band pass filter from 100 Hz to 6 kHz.
• the order R of the ambisonic format is equal to one.
• a capture of said information relating to the orientation of the head of a user listening to the sound signal is performed by a sensor within a mobile phone or by a sensor located in a headset or a virtual reality helmet.
• the data in ambisonic format is transformed into data in binaural format.
• the present invention also relates to a system for processing the sound signal, comprising means for:
• FIGS 1 and 3 show the different steps of the process according to the present invention
• FIG. 2 illustrates the treatments applied in the context of the second step of the process according to the present invention
• ⁇ Figures 4a, 4b and 4c represent the ideal W, Y and X components of an ambisonic format of order 1 (on a horizontal plane);
• FIGS. 5a, 5b and 5c illustrate the approximated W, Y and X components of an ambisonic format of order 1;
• Figure 6 shows the placement of eight virtual speakers, each placed at 45 ° around a user.
• the present invention relates to a sound signal processing method, comprising the following steps:
• N being a natural integer greater than or equal to three
• FIGS 1 and 3 illustrate the different steps of the method according to the present invention.
• said microphones are arranged in a circle on a plane, spaced at an angle equal to 360 ° / N or at each corner of a mobile phone.
• the method according to the present invention implements four microphones spaced at an angle of 90 ° to the horizontal.
• the order R of the ambisonic format is one.
• the first step of the method according to the present invention consists in the recording of the sound signal.
• N microphones are used for this recording, N being a natural number greater than or equal to three, said microphones being arranged in a circle on a plane, spaced at an angle equal to 360 ° / N or at each corner of a mobile phone.
• N is equal to four and the microphones are spaced 90 °.
• These microphones are arranged in a circle on a plane.
• the radius of said circle is two centimeters, and the microphones are omnidirectional.
• the sound signal is picked up by said microphones, and digitized. This is a synchronous capture. At the end of this first step, four sampled digital signals are obtained.
• the second step of the method according to the present invention consists in the encoding of said four sampled digital signals, in an ambisonic format of order R, R being a natural integer greater than or equal to one.
• the ambisonic format is a standard format of multi-dimensional audio coding.
• the order R is equal to one.
• This order 1 makes it possible to represent the sound with the following notions: Forward - Back and Left - Right.
• Figures 4a, 4b and 4c represent the ideal W, Y and X components of an ambisonic format of order 1 (on a horizontal plane).
• Figures 5a, 5b and 5c illustrate the approximated W, Y and X components of an ambisonic format of order 1.
• Figure 2 illustrates the treatments applied in the context of the second step of the process according to the present invention. It can be observed in FIG. 2 that the input data are in the time domain, pass into the frequency domain following a Fast Fourier Transform (FFT) operation, and then the data in FIG. output are in the time domain following an inverse fast Fourier transform (IFFT) operation.
• FFT Fast Fourier Transform
• IFFT inverse fast Fourier transform
• Hanning windows with overlap are used by implementing an "overlap-add" type function.
• the input frequency data are modified using matrix multiplication. This matrix comprises weighting coefficients for each microphone signal and each frequency.
• filtering by means of a bandpass filter is performed on the data before the output.
• the method according to the present invention implements a filter bandpass filter from 100 Hz to 6 kHz. This eliminates the lower part and the acute part.
• impulse responses of the N microphones are measured, in this case four microphones, with a source positioned every 5 ° or every 10 ° around the microphone array.
• N is the number of microphones (four in the present embodiment)
• D is the number of measured source angular positions (108 in the present embodiment)
• V is the number of ambison channels (three in the present example) embodiment
• C DX N denotes the directivities of the microphones
• H Nx v denotes the matrix which transforms the directivities of the microphones into the desired directivities
• PDXV denotes the directivities prescribed by the ambisonic format (W, X and Y in the present example realization).
• H Nx v PDXV / CDXN for each index of frequency k if CDXN is invertible.
• the matrix H is defined once for future uses of the considered microphone array. Then, with each use, a matrix multiplication is performed in the frequency domain.
• Said matrix H has as many lines as microphones, therefore four in the present embodiment, and as many columns as required by the order of the ambisonic format used, therefore three columns in the present embodiment in which the order 1 is implemented on the horizontal plane.
• Out In x H, where H denotes the previously computed matrix, In denotes the input (audio channels coming from the microphones array, passed in the frequency domain) and Out denotes the output (Out being reconverted into the time domain for get the ambisonic format).
• the method according to the present invention implements, during this second step, an algorithm called least squares algorithm for each frequency, with for example 512 frequency points.
• data is obtained in the ambisonic format (in the present embodiment the signals W, X and Y).
• the third step of the method according to the present invention consists in the restitution of the sound signal, thanks to a transformation of the data in ambisonic format into two binaural channels.
• the information relating to the orientation of the head of the user listening to the sound signal is retrieved and used. This can be performed by a sensor in a mobile phone, a headset or a virtual reality headset.
• This orientation information consists of a vector comprising three angle values, in the English terminology “pitch”, “yaw” and “millet”.
• the angle value "yaw" is used on a plane.
• the ambisonic format is transformed into eight audio channels corresponding to a virtual placement of eight loudspeakers, each placed at 45 ° around the user.
• Figure 6 shows the placement of eight virtual speakers, each placed at 45 ° around a user.
• HRTF left ear and right ear filters
• HRTF left ear and right ear filters
• a pair of HRTF (left ear and right ear) filters are paired with each virtual speaker, and then all ("left ear” channels and all "right ear” channels together) are added to form two output channels .
• IIR Infinity Impulse Response
• FIG. 3 shows the different steps of the method according to the present invention.
• the present invention also relates to a system for processing the sound signal, comprising means for:
• N being a natural integer greater than or equal to three
• This sound signal processing system comprises at least one calculation unit and one memory unit.
• the invention is described in the foregoing by way of example. It is understood that the skilled person is able to realize different variants of the invention without departing from the scope of the patent.

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• Health & Medical Sciences (AREA)
• Otolaryngology (AREA)
• Physics & Mathematics (AREA)
• Engineering & Computer Science (AREA)
• Acoustics & Sound (AREA)
• Signal Processing (AREA)
• General Health & Medical Sciences (AREA)
• Stereophonic System (AREA)
• Circuit For Audible Band Transducer (AREA)

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Publications (2)

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• 2016
  • 2016-04-26 FR FR1653684A patent/FR3050601B1/fr not_active Expired - Fee Related
• 2017
  • 2017-04-20 US US16/096,339 patent/US10659902B2/en active Active
  • 2017-04-20 EP EP17725294.7A patent/EP3449643B1/de active Active
  • 2017-04-20 CN CN201780034334.2A patent/CN109661824A/zh active Pending
  • 2017-04-20 WO PCT/FR2017/050935 patent/WO2017187053A1/fr not_active Ceased

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