Classifications

• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04S—STEREOPHONIC SYSTEMS 
  • H04S5/00—Pseudo-stereo systems, e.g. in which additional channel signals are derived from monophonic signals by means of phase shifting, time delay or reverberation 
• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
  • H04R5/00—Stereophonic arrangements
  • H04R5/04—Circuit arrangements, e.g. for selective connection of amplifier inputs/outputs to loudspeakers, for loudspeaker detection, or for adaptation of settings to personal preferences or hearing impairments
• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04S—STEREOPHONIC SYSTEMS 
  • H04S3/00—Systems employing more than two channels, e.g. quadraphonic
  • H04S3/008—Systems employing more than two channels, e.g. quadraphonic in which the audio signals are in digital form, i.e. employing more than two discrete digital channels
• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04S—STEREOPHONIC SYSTEMS 
  • 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
• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
  • H04R2205/00—Details of stereophonic arrangements covered by H04R5/00 but not provided for in any of its subgroups
  • H04R2205/021—Aspects relating to docking-station type assemblies to obtain an acoustical effect, e.g. the type of connection to external loudspeakers or housings, frequency improvement
• • 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
  • H04S2400/01—Multi-channel, i.e. more than two input channels, sound reproduction with two speakers wherein the multi-channel information is substantially preserved
• • 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
  • H04S2400/05—Generation or adaptation of centre channel in multi-channel audio systems
• • 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
  • H04S2400/07—Generation or adaptation of the Low Frequency Effect [LFE] channel, e.g. distribution or signal processing
• • 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
  • H04S2400/11—Positioning of individual sound objects, e.g. moving airplane, within a sound field
• • 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 
  • 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 audio signal processing for improving perception during sound reproduction.
• the international patent application WO2012088336 describes a method of processing an audio sound source to create four dimensions of spatialized sound.
• a virtual sound source can be moved along a path in a three-dimensional space over a specified period of time to obtain the location of the four-dimensional sound.
• the various embodiments described herein provide methods and systems for the existing mono, 2-channel and / or multi-channel conversion of audio signals into spatialized audio signals having two or more audio channels.
• the various embodiments also describe methods, systems, and apparatus for producing low frequency effects and center channel signals from incoming audio signals having one or more channels.
• Patent application WO9914983 discloses a device making it possible to create and use a pair of opposite loudspeakers of a headset, the feeling of a sound source being remote from the zone situated between said loudspeakers. .
• the device comprises:
• a first mixing matrix connected to the audio inputs and to a series of return inputs, which produces a predetermined combination of said audio inputs constituting intermediate output signals
• a filter system which filters said intermediate output signals and produces filtered intermediate output signals and the series of return inputs, and which comprises separate filters for filtering the forward response and the fast response and an approximation of the reverberated response, and to filter the return response so as to produce the return entries;
• a second mixing matrix which combines the filtered intermediate output signals to produce stereo right channel and left channel stereo outputs.
• European Patent EP2119306 discloses apparatus for processing an audio sound source to create four dimensions of spatialized sound.
• a virtual sound source can be moved along a path in a three-dimensional space over a specified period of time to obtain the location of the four-dimensional sound.
• a binaural filter for a desired spatial point is applied to the audio waveform to produce a spatialized waveform that, when the spatialized waveform is played from a pair of speakers, the sound appears to come from the chosen point space instead of speakers.
• a binaural filter for a point in space is simulated by interpolation of the nearest neighbor binaural filters selected from a plurality of predefined binaural filters.
• the audio waveform can be processed numerically by overlapping the data blocks with a short Fourier transform time.
• Localized sound can be processed later for Doppler shift and chamber simulation.
• the present invention relates to a method for processing an original audio signal of Nx channels, N being greater than 1 and x being greater than or equal to 0, comprising a multichannel processing step of said input audio signal by a multichannel convolution with a predefined imprint, said imprint being elaborated by the capture of a reference sound by a set of speakers arranged in a reference space characterized in that it comprises an additional step of selecting at least one of a plurality of fingerprints previously developed in different sound contexts.
• the patent application WO2012172264 describes a method of processing an original audio signal of Nx channels, N being greater than 1 and x being greater than or equal to 0, comprising a multichannel processing step of said input audio signal by a multichannel convolution with a predefined imprint, said imprint being elaborated by the capture of a reference sound by a set of speakers arranged in a reference space characterized in that it comprises an additional step of selecting at least one of a plurality of markings fingerprints previously developed in different sound contexts.
• Patent Application WO9725834 proposes another method and device for processing multichannel audio signals, each channel corresponding to a loudspeaker arranged at a particular point in a room so as to give, via headphones, the impression that multiples of 'Ghost' speakers are spread throughout the room.
• Heading Transfer Functions are selected by taking into consideration the height and azimuth of each speaker in relation to the listener.
• Each channel is HRTF filtered so that when these channels are combined in the left and right channels and output by headphones, the listener feels that the sound actually comes from ghost speakers distributed in the virtual room.
• HRTF coefficient sets entered in base data from a large number of individuals and the use of an optimal HRTF game for the listener provides listening impressions similar to those of an isolated listener listening to multiple high speakers distributed in the volume of a room.
• Applying an HRTF function to the output of the right and left channels allows, in the case of listening to the headphones, to give the impression of listening without headphones.
• the object of the present invention is to improve the perceived quality and in particular the extent of the spatialization, including with means of reproduction of average quality, such as docking stations of tablets or mobile phones ("docks"). ).
• the invention relates, in its most general sense, to a method of sound reproduction of a digital audio signal, characterized in that an oversampling step consisting in producing from a sampled signal at a first time is carried out.
• frequency F a signal sampled at a frequency NxF, where N corresponds to an integer greater than 1, and then to applying a convolution process on a first digital file sampled at a frequency NxF corresponding to the acquisition of the sound environment of a reference sound space, a second digital file sampled at an NxF frequency corresponding to the acquisition of the soundprint of a reference playback equipment, and third file digital sample sampled at an NxF frequency corresponding to the acquisition of an equalizer's audio fingerprint and a fourth file corresponding to said oversampled audio file, the resulting digital packets being then digitally converted to a digital sampling frequency F / M corresponding to the working frequency of the listening equipment.
• the processing is based on a mathematical convolution operation, and uses several prerecorded audio samples of the impulse response of the modeled space as well as an equalizer and playback equipment.
• the method comprises an additional step of recalculating the file corresponding to said sound footprint of the reference sound space, in order to modify the balance between the spatial channels of said sound footprint.
• FIG. 1 represents a schematic view of signal processing according to the invention.
• the treatment method according to the invention consists in producing different acoustic impressions of a sound source, with a view to convolving these different sound tracks.
• convolutions are a known technique of user capture, then the reproduction of the acoustic behavior of a place or an apparatus.
• convolution reverbs allow to propose to use the acoustics of many real places, famous concert halls or others: these acoustics, previously sampled, likely to be reused at will within the program.
• the principle is then to sample the acoustics of the sets in which the scenes of the film have been shot, in order to be able to easily apply this acoustics to the elements recorded a posteriori so that they integrate perfectly with the sounds from the direct catches. .
• the Impulse Response sensor to obtain the impulse response of a material or room constituting the sound footprint is based on the "deconvolution". It uses the excitation of the system by a known signal (here called f (t)). This signal is such that if we apply a transform (deconvolution function), the result is the Dirac function.
• the deconvolution function is chosen such that for the excitation signal f (t) and any function h (t):
• the types of signals used to capture impulse responses look like a Gaussian noise or a "white noise”.
• the excitation sequences are generated by a deterministic algorithm and are periodic (periods of the order of a few seconds or tens of seconds for our application) and constitute a pseudo-random signal. These sequences are created by linear feedback shift registers (LFSRs). This register structure, whose order is determined by the number of registers, is such that over its period it will produce the set of possible binary values for its order (if order 4 structure, there are 2 n possible values) .
• MLS Maximum Length Sequence
• the initial popularity of MLS stems from the ease of the deconvolution process.
• the signal MLS is such that for its deconvolution, one can use a transform called Hadamard transform, which simplifies the calculations and has the advantage of being computationally computationally using few resources.
• the first uses the passage in the frequency domain to make the calculations before returning to time.
• the second consists in non-periodically convolving the recorded signal with the excitation signal returned temporally:
• the method supports slight desynchronization: you can broadcast the sweep from one device and record with another without these two machines are synchronized by a clock.
• a sound impression of a listening medium for example a helmet
• Each of these impulse responses is captured from a reference signal at a high sampling rate, which is higher than the nominal sampling frequency of the playback equipment.
• the hallmark (3) is acquired from a white noise producing a file of 6 Mbytes per speaker, for a long time greater than 500 milliseconds, preferably between one and two seconds.
• the file corresponding to the impulse response is then compressed without loss (ZIP compression for example) and encrypted.
• the footprint of the headphones (1) is acquired in the same way with a white or pink signal of a duration of about 200 milliseconds, preferably between 100 and 500 milliseconds.
• the imprint of the equalizer (2) is acquired in the same way with a white or pink signal with a duration of about 200 milliseconds, advantageously between 100 and 500 milliseconds for each of the equalizer settings.
• a step (6) is performed to dynamically recalculate the left and right footprints according to the particularities of the playback equipment and, if appropriate, the sensory peculiarities of the listener. It has for example a setting means for changing the virtual spatial position. A modification of this setting controls the calculation of a new pair of fingerprints from the fingerprints initially provided, by morphing ("morphing"):
• This function can be controlled by the gyro sensor to create a dynamic displacement of the sound scene according to the user's movements It allows to center the voice in real time with respect to the head.

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• Engineering & Computer Science (AREA)
• Physics & Mathematics (AREA)
• Acoustics & Sound (AREA)
• Signal Processing (AREA)
• Multimedia (AREA)
• Stereophonic System (AREA)

Applications Claiming Priority (2)

Publications (2)

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• 2013
  • 2013-04-17 FR FR1353473A patent/FR3004883B1/fr not_active Expired - Fee Related
• 2014
  • 2014-04-09 CA CA2909580A patent/CA2909580A1/fr not_active Abandoned
  • 2014-04-09 WO PCT/FR2014/050846 patent/WO2014170580A1/fr active Application Filing
  • 2014-04-09 JP JP2016508209A patent/JP6438004B2/ja active Active
  • 2014-04-09 US US14/785,061 patent/US9609454B2/en not_active Expired - Fee Related
  • 2014-04-09 EP EP14721466.2A patent/EP2987339B1/de not_active Not-in-force
  • 2014-04-09 CN CN201480029770.7A patent/CN105308989B/zh not_active Expired - Fee Related

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