EP4184505B1 - Komplexitätsoptimierte klangraumisierung mit raumeffekt - Google Patents

Komplexitätsoptimierte klangraumisierung mit raumeffekt Download PDF

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EP4184505B1
EP4184505B1 EP22211949.7A EP22211949A EP4184505B1 EP 4184505 B1 EP4184505 B1 EP 4184505B1 EP 22211949 A EP22211949 A EP 22211949A EP 4184505 B1 EP4184505 B1 EP 4184505B1
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frequency
signal
transfer function
room
signals
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EP4184505A1 (de
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Grégory PALLONE
Marc Emerit
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Orange SA
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Orange SA
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    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04SSTEREOPHONIC SYSTEMS 
    • H04S7/00Indicating arrangements; Control arrangements, e.g. balance control
    • H04S7/30Control circuits for electronic adaptation of the sound field
    • H04S7/307Frequency adjustment, e.g. tone control
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04SSTEREOPHONIC SYSTEMS 
    • H04S7/00Indicating arrangements; Control arrangements, e.g. balance control
    • H04S7/30Control circuits for electronic adaptation of the sound field
    • H04S7/305Electronic adaptation of stereophonic audio signals to reverberation of the listening space
    • H04S7/306For headphones
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10KSOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K15/00Acoustics not otherwise provided for
    • G10K15/08Arrangements for producing a reverberation or echo sound
    • G10K15/12Arrangements for producing a reverberation or echo sound using electronic time-delay networks
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/008Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04SSTEREOPHONIC SYSTEMS 
    • H04S2400/00Details of stereophonic systems covered by H04S but not provided for in its groups
    • H04S2400/01Multi-channel, i.e. more than two input channels, sound reproduction with two speakers wherein the multi-channel information is substantially preserved
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04SSTEREOPHONIC SYSTEMS 
    • H04S2420/00Techniques used stereophonic systems covered by H04S but not provided for in its groups
    • H04S2420/01Enhancing the perception of the sound image or of the spatial distribution using head related transfer functions [HRTF's] or equivalents thereof, e.g. interaural time difference [ITD] or interaural level difference [ILD]
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04SSTEREOPHONIC SYSTEMS 
    • H04S2420/00Techniques used stereophonic systems covered by H04S but not provided for in its groups
    • H04S2420/07Synergistic effects of band splitting and sub-band processing

Definitions

  • the present invention relates to sound spatialization with room effect.
  • the invention finds an advantageous but non-limiting application to a processing of sound signals respectively coming from L channels associated with virtual speakers (for example in a multichannel representation, or even in an ambiophonic representation, of the sound to be reproduced), for spatialized restitution on real speakers (for example two earpieces of a headset in binaural reproduction, or two separate speakers in transaural reproduction).
  • L channels associated with virtual speakers for example in a multichannel representation, or even in an ambiophonic representation, of the sound to be reproduced
  • real speakers for example two earpieces of a headset in binaural reproduction, or two separate speakers in transaural reproduction.
  • the signal from one of these channels can be processed to have a first contribution on the left atrium and a second contribution on the right atrium, in binaural reproduction, in particular by applying a transfer function with effect of room for each of these contributions.
  • the application of these room effect transfer functions then helps to offer the listener a feeling of immersion allowing him to practically “locate in space” the virtual loudspeaker associated with this channel.
  • Such an achievement advantageously makes it possible to apply a common processing to all the signals, which corresponds, in a physical reality, to a “mixing” of the acoustic waves as the reverberations progress, therefore beyond a given duration. (characterizing the beginning of the presence of the diffuse field). Such an achievement then makes it possible to reduce the complexity of spatialization processing with room effect on several initial channels.
  • the document US2011/170721 A1 discloses a method of processing at least one input signal through a set of binaural filters such that the outputs can be played back on headphones to provide a sensation of listening to sound in a listening room via one or multiple virtual speakers.
  • the channel signals are received in encoded form, by a compression decoder.
  • This decoder sends the channel signals, once decoded, to a module spatialization for sound reproduction with room effect, on two speakers. It is then appropriate for this spatialization step (which follows the decoding of the received signals) to be of reduced processing complexity so as not to delay the overall set of decoding and spatialization steps upon reception of the signals before restitution.
  • the present invention improves the situation.
  • the invention proposes to reduce the complexity of the application of the room effect transfer function, in particular by reducing this complexity in the spectral domain.
  • the convolution by a transfer function becomes a multiplication of spectral components of the signal on the one hand, and of a filter representing the transfer function on the other hand ( figure 1 commented in detail later).
  • the invention then starts from the advantageous observation according to which, after direct propagation, a sound wave tends to attenuate in high frequencies due to progressive reflections on absorbing surfaces (typically walls, face of the listener, etc.). the wave especially in high frequencies. Additionally, air itself absorbs the higher frequency spectral components of sound as it propagate. This phenomenon is all the more increased for example for the diffuse sound field, for which it is not necessary to have a frequency representation for very high frequencies (for example above a frequency in a range of 5 to 15 kHz ).
  • this disclosure presents a sound spatialization method, comprising the application of at least one room effect transfer function to at least one sound signal, said application amounting to multiplying, in the spectral domain, spectral components of the sound signal by the spectral components of a filter corresponding to the aforementioned transfer function.
  • Each spectral component of the filter includes a temporal evolution in a time-frequency representation (as detailed below with reference to the Figure 3 ).
  • these spectral components of the filter are ignored, for the aforementioned multiplications of components, beyond a threshold frequency and after at least one given instant in said time-frequency representation.
  • the spectral components of the filter are taken into account up to a cutoff frequency which can be chosen for example between 5 and 15 kHz (depending on the room effect to be applied and/or the signal to be spatialized, as described below). Beyond the cutoff frequency, the multiplication is not even performed, which mathematically is the same as multiplying the signal by zero.
  • This given moment typically represents the moment when a sound wave begins to undergo reverberations (by successive reflections, or, later still, from the presence of a diffuse sound field).
  • the transfer function takes into account reverberations in the room effect (taking into account for example a diffuse sound field)
  • the aforementioned given instant can be chosen as a function of such reverberations.
  • the aforementioned given instant may be subsequent, in the room effect, to direct sound propagation with first reflections, and then correspond to the beginning of the presence of a diffuse sound field.
  • the aforementioned threshold frequency decreases as a function of time in said time-frequency representation. For example, if the signal is sampled over several successive blocks, it can be planned, for example, to keep the spectral components present in the signal, in the multiplication of the components, for a first block, then to ignore them at the same time. -beyond a first threshold frequency for a second block which follows the first block, then ignoring them beyond a second threshold frequency for a third block which follows the second block, etc., the second threshold frequency being more lower than the first.
  • the aforementioned given block may include, for example, samples located temporally at times which correspond to times when a sound wave has undergone one or more reflections, with even the beginning of the presence of a diffuse sound field.
  • the block which follows this given block (immediately or a few blocks later) can for example include samples located temporally after or from the beginning of the presence of a diffuse sound field.
  • the signal coming from the reverberations does not normally include spectral components of higher frequency than the initial signal.
  • the aforementioned threshold frequency cannot be greater than this highest frequency.
  • the aforementioned threshold frequency is chosen as being the minimum among a predetermined threshold frequency (for example between 5 and 15 kHz) and said highest frequency.
  • the highest frequency spectral component information can be provided by the decoder.
  • the spatialization is carried out with a module capable of supporting different signal formats, particularly in terms of sampling frequency of such signals, the highest frequency, mentioned above, cannot be greater than half of the sampling frequency, and, thus, the threshold frequency for the implementation of the invention can be further chosen as a function of this sampling frequency.
  • first and second transfer functions with room effect are applied respectively to these first and second channels, as explained above in the introduction (for example by adapting signals on surround channels to switch to binaural or transaural reproduction).
  • first and second transfer functions with room effect
  • one of the first and second transfer functions applies an ipsi-lateral acoustic path effect
  • the other of the first and second transfer functions applies a contra-lateral acoustic path effect
  • This “screening” frequency is explained by the fact that for a contralateral path between a virtual loudspeaker and a considered ear of the listener, the listener's head masks the acoustic path and absorbs the lower tones. highest frequencies of the acoustic wave (therefore eliminating the spectral components associated with the highest frequencies of the acoustic wave).
  • the aforementioned threshold frequency for the transfer function applying a contra-lateral path effect, can be chosen as a minimum among a predetermined threshold frequency (for example chosen between 5 and 15kHz) and this screening frequency. This achievement is advantageous to be applied already for the first block of samples. On the other hand, it does not exclude the possibility of increasing the threshold frequency again for the following block to simulate a first reflection on a wall located opposite the ear considered, this first reflection being received at this ear by a path ipsi-lateral.
  • the cutoff frequency can be chosen common to all the signals, in a possible embodiment, after a given instant which corresponds for example to the presence of the diffuse field.
  • At least one given instant is provided for the limitation of the taking into account of the frequency components up to a cut-off frequency, this given instant being located temporally at the start of a block different from a first block in a succession of blocks. This given instant therefore occurs after direct propagation, and at the time of sound reflections or the presence of a diffuse field.
  • FIG. 5 also illustrating, in an exemplary embodiment, a possible algorithm of a computer program that would be executed by a processor of a spatialization module operating the method within the meaning of the invention.
  • the present disclosure also relates, outside the scope of the claims, in general, to a computer program comprising instructions for implementing the above method, when they are executed by a processor.
  • the present invention relates to a sound spatialization module, comprising calculation means for applying at least one room effect transfer function to at least one input sound signal, said application amounting to multiplying, in the spectral domain, components spectral components of the sound signal by the spectral components of a filter corresponding to said transfer function, each spectral component of the filter comprising a temporal evolution in a time-frequency representation.
  • these calculation means are configured to ignore said spectral components of the filter for said multiplications of components, beyond a threshold frequency and after at least one given instant in said time-frequency representation.
  • This module can be integrated into a compression decoding device, or more generally into a restitution system.
  • the latter comprises in the example shown an input interface IN to receive the decoded signals, as well as calculation means such as a PROC processor and a working memory MEM cooperating with the IN/OUT interfaces to spatialize the signals I ( l ) and deliver through the output interface OUT only two signals O d and O g intended to power the respective earpieces of a CAS headset.
  • calculation means such as a PROC processor and a working memory MEM cooperating with the IN/OUT interfaces to spatialize the signals I ( l ) and deliver through the output interface OUT only two signals O d and O g intended to power the respective earpieces of a CAS headset.
  • a plurality of virtual speakers surround, in the example shown, the head TE of a listener.
  • Each of the HPV virtual speakers is initially powered by a signal I ( l ) with l ⁇ [1; L ] for example previously decoded as indicated above with reference to the Figure 6 .
  • the arrangement of the virtual speakers can concern a multichannel or also surround representation of the signals I ( l ) to be processed to restore them together in a spatialized manner with a room effect on a headset with CAS earpieces ( Figure 6 ).
  • a designer of such a filter can thus limit the components of his filter for the right ear up to the cutoff frequency F c d (0) (corresponding to a head screening frequency) even if the signal to be processed I( l ) can have higher spectral components and up to the frequency F c g (0) at least.
  • a designer of filters representing these transfer functions can plan to limit the components of the filters for the right ear up to the cutoff frequency F c d (1) and for the left ear up to the cutoff frequency F c g (1).
  • a designer of filters representing these transfer functions can plan to limit the components of the filters for the right ear up to the cutoff frequency F c d (2) and for the left ear up to the cutoff frequency F c g (2).
  • L input signals 1(1), I(2), ..., I(L) are transformed into the frequency domain, respectively at steps TF11, TF12, ..., TF1L.
  • input signals may already be available in frequency form (for example from the decoder).
  • step BA11 a complete spatialization impulse response (typically of the BRIR type for “Binaural Room Impulse Response”) in temporal form corresponding to signal 1(1) of channel 1 is stored in memory.
  • this impulse response is transformed into frequency form to obtain a corresponding filter in the spectral domain.
  • the filter is stored in its spectral form to avoid repeating the calculation of the transform. We then multiply this filter to the input signal in frequency form of channel 1 (which is equivalent to a convolution in the time domain). We therefore have the spatialized signal for signal 1(1) of channel 1.
  • the same operations are carried out for the other L-1 channels.
  • a similar treatment is carried out for the other earpiece.
  • the L spatialized channels are not accessible independently before summing: the single output signal is constructed by progressively summing each spatialized channel with the previous output signal.
  • the L input signals can typically correspond to the L channels of multichannel audio content intended to feed (“virtual”) speakers.
  • the L input signals can for example correspond to the L surround signals of audio content in surround representation.
  • step S21 L input signals 1(1), I(2), ..., I(L) are transformed into the frequency domain in step S21.
  • a spatialization impulse response A k ( l ) (typically of the BRIR type) corresponding to the signal l( l ) of the channel / is transformed into the spectral domain to obtain a frequency filter.
  • This multiplication is parameterized (as shown below with reference to the figure 4 ) so that certain frequency components are ignored, within the meaning of the invention. Typically, the highest frequency components will be ignored to limit the complexity of the calculations.
  • a cutoff frequency f cA(I) from which the frequency components are ignored is defined (for example the maximum frequency represented in the channel signal I( l ), or half of its sampling frequency).
  • a cutoff frequency is specific to an input signal, to an ear (therefore an output signal) and to a time block.
  • the summation is carried out in a particular way, because it takes into account a delay on the channels to characterize the reverberations (reflections and diffuse field), as detailed below.
  • the L spatialized channels are not accessible independently before summing: the single output signal is constructed by progressively summing each spatialized channel with the previous output signal.
  • the delay m is zero. In the case of a frequency representation, this delay corresponds generally at the size of a processed signal frame for the first block, and is interpreted as taking the previous input block in its frequency form.
  • an incomplete spatialization impulse response B k m ( l ) (typically BRIR type) corresponding to the signal l( I ) of channel 1 is transformed into the spectral domain to obtain a frequency filter.
  • the same operations are carried out for the L channels and the filter multiplication operations are repeated on the progressively delayed spectral signals by summing the contributions in step S25 repeated for each delay m until obtaining a single signal representing the L channels on the set M of the temporal blocks m considered.
  • the single output signal is constructed by progressively summing each spatialized channel with the previous output signal as we will now see with reference to the figure 4 .
  • step S26 we return to the time domain in step S26 to obtain an output signal intended to power one of the earpieces of the headset.
  • step S40 the output signal S is initialized to 0.
  • This output signal is expressed in the frequency domain. It has a limited size of a length greater than the cutoff frequency fc( l ). For example, this signal is set to [0; fs( l )/2], fs( l ) being the sampling frequency of this signal I( l ).
  • a first counting variable l is also initialized to 1. This first counting variable identifies one of the channel signals 1(1), I(2), ..., I( l ), ..., I(L) on the time block [0; N-1] for the right ear.
  • a second counting variable j is initialized to 0. This second counting variable identifies a frequency component of a signal I( l ) on the time block [0; N-1] for the right ear.
  • step S42 the coefficient c BRIR (j; l ) is stored. This coefficient corresponds to the frequency component j of the BRIR( l ) filter on the time block [0; N-1] for the right ear.
  • the coefficient c I (j; l ) is stored in memory. This coefficient corresponds to the frequency component j of the signal I( l ) on the time block [0; N-1] for the right ear.
  • the coefficients c BRIR (j; l ) and c I (j; l ) correspond to the same frequency component (identified by the variable j) and can thus be subsequently multiplied term by term (step S44).
  • a value MULT(j) is calculated corresponding to the multiplication of the coefficients c BRIR (j; l ) and c i (j; l ). These coefficients are indeed multiplied term by term because they correspond to the same frequency component j (for the same channel, on the same block and for the same ear).
  • step S45 this value MULT(j) is incremented to the signal S at the position of frequency j.
  • step S46 we increment the variable j and we resume at step S42. If the variable j is greater (for example or equal) to the cutoff frequency fc( l ), we go to test T48. Thus, we filled the signal S on the interval [0; fc( l )].
  • this signal can be defined over an interval larger than [0; fc( l )] (for example [0 ; fs( l )/2]).
  • this signal had been initialized to 0 over its entire definition interval. From then on, it is zero on the rest of the interval which has not been filled (for example [fc( l ); fs( l )/2]). The complexity is therefore improved here because steps of filling the signal S have not been carried out, which reduces the number of calculations necessary.
  • test T48 it is verified that the counting variable l corresponding to the signal I( l ) of channel l is less (for example strictly) than the number L of channels. If the variable l is less than or equal to L, the variable l is incremented in step S49 and the process is resumed in step S41. If the variable l is greater than L, the signal S corresponding to the spatialized signal for the time block [0; N-1] for the right ear is available in step S50.
  • This signal S corresponding to the time block [0; N-1] is then summed with other signals generated in a similar manner for other time blocks [N; 2N-1], [2N; 3N-1], etc., (and for which an appropriate delay has been applied in accordance with step DBD above of the figure 2 For example).
  • this low-pass filter is not applied to the audio signal, but to the BRIR filter (which is itself convolved with the audio signal) which is already composed of multiple reflections; the artifacts produced will therefore, at worst, be perceived as additional reflections of the original BRIR filter, and in practice rarely perceptible. It is, however, possible to attenuate these artifacts by slightly modifying the filter frequencies preceding the cutoff frequency (for example by gentle attenuation by applying a half Hanning window (fade out type)).
  • the weightings W k ( l ) and the gains G(I( l )) can be set to 1.
  • the gains G(I( l )) because this figure should be read as an integration of the gains with the weights 1/W k ( l ).
  • these two parameters are determined, fixed and multiplied together once and for all.

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  • Engineering & Computer Science (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
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  • Computational Linguistics (AREA)
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Claims (1)

  1. Klangverräumlichungsmodul, aufweisend Rechenmittel zur Anwendung wenigstens einer Übertragungsfunktion mit Raumeffekt auf wenigstens Eingangsklangsignal, wobei das Anwenden dem Multiplizieren, im Spektralbereich, der Spektralkomponenten des Klangsignals mit den Spektralkomponenten eines Filters entspricht, der der Übertragungsfunktion entspricht, wobei jede Spektralkomponente des Filters eine zeitliche Entwicklung in einer Zeit-Frequenz-Darstellung aufweist, dadurch gekennzeichnet, dass die Rechenmittel dazu ausgebildet sind, die Spektralkomponenten des Filters für die Multiplikationen von Komponenten über einer Schwellenfrequenz und nach wenigstens einem bestimmten Zeitpunkt in der Zeit-Frequenz-Darstellung zu ignorieren und dadurch, dass das Verräumlichungsmodul, das eine Mehrzahl von Eingangssignalen empfängt, wenigstens zwei Ausgangssignale ausgibt, wobei die Rechenmittel dazu ausgebildet sind, eine Übertragungsfunktion mit Raumeffekt auf jedes Eingangssignal anzuwenden, wobei das Modul dadurch gekennzeichnet ist, dass jedes der Ausgangssignale durch Anwendung der folgenden Formel geliefert wird: O k = l = 1 L I l 0 ; ; f k l A k l + + m = 1 M z iDDm l = 1 L G I l 1 Wkl Il 0 ; ; f k m B mean k m
    Figure imgb0008
    - wobei Ok ein Ausgangssignal und k der Index in Bezug auf ein Ausgangssignal ist,
    - wobei l ∈ [1; L] der Index in Bezug auf ein Eingangssignal von den Eingangssignalen ist, L die Anzahl von Eingangssignalen ist und I( l ) ein Eingangssignal von den Eingangssignalen ist,
    - wobei Ak(l) eine Übertragungsfunktion mit Raumeffekt ist, die spezifisch für ein Eingangssignal ist,
    - wobei B mean k m
    Figure imgb0009
    eine übergreifende Übertragungsfunktion mit Raumeffekt ist, die für die Eingangssignale gemeinsam ist,
    - wobei Wk(l) ein gewähltes Gewichtungsgewicht und G(I( l )), eine vorbestimmte Verstärkung zum Energieausgleich ist,
    - wobei z-iDDm eine Anwendung einer Verzögerung ist, die als Anzahl von Abtastwertblöcken gezählt wird, die einer Zeitabweichung zwischen einer Schallaussendung in einem Raum, die dem Raumeffekt entspricht, und einem Beginn des Vorhandenseins eines Diffusfelds in diesem Raum entspricht, wobei der Index m einer Anzahl von Abtastwertblöcken mit einer Dauer entspricht, die dieser Verzögerung entspricht, wobei M die Gesamtzahl von Blöcken ist, die eine Übertragungsfunktion in einer Zeit-Frequenz-Darstellung dauert,
    - wobei das Zeichen " . " die Multiplikation bezeichnet,
    - wobei das Zeichen "*[0;...;fk (l)] " den Faltungsoperator auf einer begrenzten Anzahl von Frequenzen bezeichnet und von einer tiefsten Frequenz zu einer Maximalfrequenz fk(l) reicht, die wenigstens vom Eingangssignal mit dem Index l abhängig ist, und
    - wobei das Zeichen "*[0;...;fk (m)]" den Faltungsoperator auf einer begrenzten Anzahl von Frequenzen bezeichnet und von einer tiefsten Frequenz zu einer Frequenz fk (m) reicht, die vom Abtastwerteblock mit dem Index m abhängig ist.
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FR1360185A FR3012247A1 (fr) 2013-10-18 2013-10-18 Spatialisation sonore avec effet de salle, optimisee en complexite
PCT/FR2014/052617 WO2015055946A1 (fr) 2013-10-18 2014-10-14 Spatialisation sonore avec effet de salle, optimisee en complexite
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EP2489206A1 (de) * 2009-10-12 2012-08-22 France Telecom Verarbeitung von in einer subbanddomäne codierten schalldaten
EP2503800B1 (de) * 2011-03-24 2018-09-19 Harman Becker Automotive Systems GmbH Räumlich konstanter Raumklang
EP2840811A1 (de) * 2013-07-22 2015-02-25 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Verfahren zur Verarbeitung eines Audiosignals, Signalverarbeitungseinheit, binauraler Renderer, Audiocodierer und Audiodecodierer

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KR20160073394A (ko) 2016-06-24
US20160269850A1 (en) 2016-09-15
ES2982054T3 (es) 2024-10-14
EP3058564B1 (de) 2023-07-26
EP4184505A1 (de) 2023-05-24
EP3058564A1 (de) 2016-08-24
JP6518661B2 (ja) 2019-05-22
US9641953B2 (en) 2017-05-02
CN105706162B (zh) 2019-06-11
FR3012247A1 (fr) 2015-04-24
ES2959534T3 (es) 2024-02-26
CN105706162A (zh) 2016-06-22
KR102156650B1 (ko) 2020-09-16
JP2016537866A (ja) 2016-12-01
WO2015055946A1 (fr) 2015-04-23

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