• 专利附录
  • 专利说明
  • 权利要求
  • 类似技术
  • 同族专利
  • 引用文献
  • 法律状态
  • 优先权
    • 专利摘要:
    Apparatus (29) for receiving and playing audio signals, comprising: - a high frequency electromagnetic wave receiver configured to receive broadcast signals incorporating a plurality of audio tracks; a wireless communication interface, distinct from the receiver and configured to receive at least one setting parameter to be applied to at least one audio track of the received signal; a digital signal processor configured to apply the adjustment parameter to the audio track and produce a mixed signal.
    公开号:FR3032586A1
    申请号:FR1550923
    申请日:2015-02-05
    公开日:2016-08-12
    发明作者:Stephane Dufosse；Laurent Said；Frederic Paty；Stephanie Plasse
    申请人:Augmented Acoustics；
    IPC主号:
    专利说明:

    [0001] FIELD OF THE INVENTION The present invention relates generally to sound reinforcement systems for improving the sound experience of an auditor attending a show. The term "show" refers to any event that takes place in front of an audience (also referred to as an audience) and which includes one or more sound sources (such as voices, sounds, musical instruments). The show may be, for example, a concert, a play, a number (eg one-man show), a ballet, or a conference, a meeting, a meeting, a film, a television show or radio recorded live and in public. The show may be sounded, that is, the sounds of the show are broadcast (possibly amplified) to the attention of the audience by means of a (digital or analog) sound reproduction device. The show may be live (or "live") and take place on a stage, which designates a place where sound sources (eg musicians) are gathered or distributed; the audience may be gathered or distributed in a room, which in the case of a live performance may be adjacent to 20 (or distant from) the stage. The configuration of the room is arbitrary; it is not necessarily covered; the room can be located below the stage (or on the contrary overhanging). The room can be organized in tiers or on one level. The room can occupy the front of the stage (this is typically the case of a theater) or surround it (typically in a stadium). For some outdoor shows, the term "room" may in fact refer to an undefined space, such as a lawn or esplanade. The quality of the sound perceived by a listener can be affected by several objective factors (beyond the auditor's control): the acoustics of the place (presence of parasitic reflections, echoes, resonance modes), the quality of the system sound reinforcement, the constraints on sound engineers responsible for mixing, and the presence of noise (eg generated by a noisy audience). 3032586 2 Other subjective factors (related to the listener himself) may also affect the quality of the sound, including the listener's position in the room and his sensitivity to certain frequencies of the audio spectrum more than to other. Some measures can be taken to improve the quality of the sound perceived by the listeners. We can change the architecture of the place to improve the acoustics. It can also replace all or part of the equipment of the sound system. However, it is difficult to make a theater versatile. In particular, certain frequencies can be absorbed by the walls, or on the contrary totally reflected. A room can resonate at certain frequencies and generate parasitic vibrations to others. In addition, an acoustic suitable for chamber music may, for example, not be suitable for pop music, and vice versa. There is therefore a need to better adapt the sound reproduction to the expectations of the 15 listeners. Note also that, the auditory perception is subjective, a sound can be considered satisfactory by some listeners and otherwise unsatisfactory by others. US2014 / 0328485 discloses an audio system configured to provide enhanced sound effects to a listener equipped with a mobile terminal and headphones. For this purpose, this system comprises a console coupled on the one hand to a plurality of microphones and, on the other hand, to a wireless access point. The console is programmed to transfer to the mobile terminal, via the wireless access point, audio signals received on different channels from multiple microphones. A central server, at the request of each user, is responsible for performing the custom mix required by this user. This solution may seem to solve the above-mentioned difficulties, but it is however not free from defects. Firstly, the centralized client / server architecture that is described (a custom mix of the different audio channels is provided in point-to-point mode on request of each user) causes an overload of the central server, supposed to interact with each terminal (which forms control interface) and provide it with a mixed signal on demand. Admittedly, an alternative solution is briefly mentioned, which consists of processing the console signal directly at the terminal, but, as it is rightly indicated, this solution is likely to quickly exhaust the terminal battery. The document US2014 / 0328485 proposes, moreover, to avoid the superposition, in the listener's ears, of the console audio signal and the direct signal received from the sound system of the room, to introduce a delay in the return of the signal. audio from the console to synchronize it to the direct signal. Specifically, a given loudspeaker sends a pulsation (outside the audible spectrum) into the room to allow, via a microphone equipping the terminal, to calculate the distance from it to the loudspeaker and to deduce a delay to apply to the console signal. This procedure is, in practice, difficult to apply because the speaker emitting the pulsation is not necessarily the closest to the listener, so that the delay introduced in the restoration of the console signal may not allow synchronization to the direct signal. In addition, most mobile devices (especially smartphones) are equipped with low-pass filters that will eliminate the frequencies to which the pulses are sent. As a result, for the majority of users, the system described in this document will simply not work.
    [0002] A first objective is to improve for the listener the quality of the sound reproduction of a show, by avoiding, as much as possible, the factors likely to affect the quality of this restitution. A second objective is to allow an individualized personalization of the sound restored from the show.
    [0003] For this purpose, it is proposed, in the first place, an audio signal reception and reading apparatus which comprises: a high frequency electromagnetic wave receiver configured to receive broadcast signals incorporating a plurality of audio tracks; a wireless communication interface, distinct from the receiver and configured to receive at least one setting parameter to be applied to at least one audio track of the received signal; A digital signal processor configured to apply the setting parameter to the audio track and produce a mixed signal. According to various additional features, taken individually or in combination: the apparatus comprises an audio output for the sound reproduction of the signal mixed by the processor; the apparatus further comprises an integrated microphone configured to receive a sound signal in the vicinity of the apparatus; the processor is configured to calculate a time offset between the signal mixed by the processor and the sound signal picked up by the microphone; the processor is configured to apply to the mixed signal a delay equal to the calculated time offset. Secondly, it is proposed a headset incorporating an apparatus as presented above. This helmet may further include an acoustic noise reduction system. Thirdly, there is proposed a sound system for a show, this system comprising: a mixer that receives audio tracks; A transmitter connected to the mixer and configured to broadcast community audio tracks multichannel mode; at least one apparatus for receiving and reading the broadcast signal as presented above; a headphone connected to the device for the reproduction of the mixed signal.
    [0004] According to various additional features, taken individually or in combination: the show is live and includes sound sources, and the system includes microphones pointing to the sound sources to capture the sounds as an electrical signal forming an audio track. for each microphone. the system comprises a mobile terminal configured to transmit to the receiving apparatus, via its second communication interface, at least one setting parameter to be applied to at least one audio track. Other characteristics and advantages of the invention will appear more clearly and concretely on reading the following description of 5 preferred embodiments, with reference to the appended drawings in which: FIG. 1 is a diagrammatic view in perspective illustrating a sound system of a show, this sound system being equipped with a reception and reading apparatus; FIG. 2 is a schematic view detailing the functional architecture of the system. FIG. 1 shows a sound system 1 of a show emitting sounds (voices and / or noises and / or music) for the benefit of a public (or audience) composed of one or more listeners 2. This show can be live or "live" and live. The show can be a concert, a play, a number (eg one-man show), a ballet, a conference, a meeting, a meeting, a film, a live television or radio broadcast, and in public.
    [0005] When it is alive, the show takes place within a place 3 and includes, within this place 3, sound sources 4 (the show can, in addition to the 4 sound sources, include non-sonic visual sources ). In what follows, this place 3, which is not necessarily clearly delimited, is designated by the term "scene". In some cases, e.g. when the show is a concert, scene 3 designates a stage on which are gathered the artists who perform the show. In the illustrated example, the show is given by musicians (of whom two singers have been represented) who form 4 sound sources. Scene 3 is schematized by a dotted rectangle that frames the musicians 30 4. When the show is recorded (eg a movie) or broadcast away from the location in which it takes place, eg. a sports match taking place in a stadium and broadcast outside the stadium for the benefit of a supernumerary audience), the term "scene" may designate a screen on which the visual part of the show is projected. The listeners 2 may be gathered, or scattered, in front of or around the stage 3, or in front of screens retransmitting the show from a distance. The term "hall" usually means a place both covered and delimited. Here, we generalize the concept of the room, referenced under the number 5, to designate any place where the listeners are 2. In the simplest case (for example, in the case, illustrated, of a given concert on a stage 2), the room 5 is effectively covered and delimited, and extends 10 in front of the stage 3. But the room 5 is of any configuration; it may be an undefined space, not necessarily covered, such as a lawn or esplanade, eg. in the case of concerts given outdoors and / or in a stadium. The show is sounded, that is to say that the sounds from the sources 4 15 sound (the musicians in the illustrated example) are captured using the sound system 1, which includes for this purpose microphones 6 pointing to the sound sources 4 to capture the sounds in form, for each microphone 6, an electrical signal forming an audio track. Preferably, each sound source 4 is dedicated to a microphone 6. In FIG. 1, the microphones 6 are shown fixed, but at least some of them may be movable. In particular, a microphone 6 can be portable, especially if its source 4 is mobile, e.g. the singer or the guitarist of a musical group. In this case, the microphone 6 can be held by hand by the protagonist, or hooked to him or his instrument (or even integrated with it, eg in the case of an electric guitar). In a conventional manner, the system 1 may comprise a primary audio separator 7 which retrieves the audio track coming from each microphone 6 via an input audio line 8 - wired, as in the example illustrated, or at radio frequencies (RF) in the case of a wireless microphone.
    [0006] The splitter 7 comprises several inputs so that the signals generated by the different microphones 6 can be retrieved in parallel. The sound capture is therefore multitrack (ie on different tracks), in real time.
    [0007] The sound system 1 may comprise a first mixing table 9 connected to the primary separator 7 by a link 10 (wired or RF) by which the first table 9 receives different channels from the audio lines 8.
    [0008] This first mixing table 9 is connected, at the output, to a sound reproduction device 11, hereinafter more simply called sono, which comprises loudspeakers turned to the listeners 2. The sound system 11 may be organized in a in any way, preferably according to the habits of sound engineers; it can thus comprise loudspeakers mounted in a line array, arranged in a matrix, or grouped in front of the stage 3. For simplicity, the sound system 11 is represented in FIG. 1 by a single loudspeaker. The different channels from the primary separator 7 are mixed thanks to the first table 9 to form a first sound composition intended to be broadcast in free field to the public via the sound system 11. For this purpose, a sound engineer is responsible for mixing, on the first table 9, to apply to the room 5 settings applied to the various tracks, including the sound volume, balance, equalization.
    [0009] As can be seen in FIG. 1, the sound system 1 may comprise a second mixing table 12 connected to the primary separator 7 by a link 13 (wired or RF) by which the second table 12 also receives the channels coming from the lines 8 audio. The different channels from the primary separator 7 are mixed thanks to the second table 9 to form a second sound composition for stage 3 (via return speakers, not shown). For this purpose, a sound engineer (the same as before or another) is responsible for mixing on the second table 12, to apply to the scene 3 settings applied to the various tracks, 30 of which the volume sound, balance, equalization.
    [0010] According to an embodiment illustrated in FIG. 1, the sound system 1 further comprises a secondary separator 14 connected in parallel (or in cascade) with the primary separator 7 and which, like itself, retrieves the signal from each microphone 6 via the same lines 8 audio 5 input. At the output of the secondary splitter 14, the audio tracks may be identical to those from the primary splitter 7, or different, depending on the settings (identical or different) applied to the separators 7, 14. The sound system 1 also comprises a third table 15 10 connected to the secondary separator 14 by a link 16 (wired or RF) by which the third table 15 receives different channels from the audio lines 8. Alternatively, the system 1 is devoid of secondary separator 14, and the third table can be directly connected to one of the tables 9, 12, preferably the first table 9. In addition, in the case of a non living (eg a movie), the system is devoid of microphones, the audio lines being prerecorded. The sound system 1 comprises at least one broadcasting transmitter 17 to which the third table 15 is connected and which is configured to collect the various channels coming from the latter and broadcast them over the air in multichannel mode by means of an antenna. 18, e.g. according to the IEEE 802.11 standard. Care will be taken to cover the entire room 5 by adjusting the power of the transmitter 17 and, if necessary, by providing several transmitters 17 connected to the third table 15, distributed so as to maximize the coverage of the broadcast. Alternatively, the transmitter 17 and the third table 15 form a single device. The broadcasting carried out by the transmitter 17 is community-based, that is to say that the broadcast is common for a set of recipients, which are not necessarily identified by the transmitter 17 and which can capture the broadcast when 'they are configured for this purpose and located within range of the transmitter 17.
    [0011] The broadcasting mode is multichannel, that is to say that the signal broadcast by the transmitter 17 comprises each audio track taken separately from the others. The audio tracks can be multiplexed within the broadcast signal. The transmitter 17, illustrated in FIG. 2, comprises: an analog-to-digital converter (ADC), multichannel, configured to collect, by differential analog inputs, the signals analog signals from the third mixing table and converting them into digital signals; A digital signal processor 20 or DSP (Digital Signal Processor), programmed to process in real time the digital signals delivered to it by the ADC 19 by applying them, in particular, filters and conversion to signals; radio frequency; the signals are transmitted by the ADC 19 to the DSP 20 via a data bus 12S (integrated lnterchip Sound), which has the advantage of minimizing fluctuation phenomena (or jitter, of the English jitter) in the signal; the DSP 20 controls the ADC 19 via a synchronous serial data bus 22 or SPI (English acronym for the Serial Peripheral Interface), which makes it possible to perform a master-slave type control in which the ADC 19 is slaved to the DSP 20 which generates the clock signal for sending the data by the bus 12S 21; a modulator 23, which receives the processed audio stream of the DSP 20 via a 12S standard data bus 24 for in-phase and quadrature modulating (I / Q modulation); a radio transmitter 25 which receives the modulated signal of the modulator 23 via an I / Q bus 26 looping the radio transmitter 25 to the modulator 23 and which is connected to the antenna 18 for broadcasting the channels (over the air).
    [0012] According to a preferred embodiment illustrated in FIG. 2, the modulator 23 controls the radio transmitter 25 via a synchronous serial data bus 27 or SPI (acronym for the Serial Peripheral Interface), by means of which the radio transmitter 25 is slave to the modulator 23.
    [0013] In addition, according to a preferred embodiment, the radio transmitter 25 is connected to the antenna 18 via an amplifier 28 which applies to the signal from the radio transmitter 25 a coefficient of proportionality to increase the power (and therefore the range).
    [0014] The sound system 1 further comprises an apparatus 29 for receiving and reading the audio signals received from the transmitter 17. As illustrated in FIGS. 1 and 2, the apparatus 29 comprises a housing 30 and, in this housing 30 an antenna 31 and a receiver 32 of high frequency electromagnetic waves (HF), coupled to the antenna 31 and configured to receive the signals broadcast by the transmitter 17 (and incorporating, as we have seen, several audio tracks); a demodulator 33 which receives the modulated signal from the receiver 32 via an I / Q bus 34 looping the receiver 32 and the demodulator 33 and 15 which is programmed to demodulate this signal; according to a preferred embodiment illustrated in FIG. 2, the demodulator 33 controls the receiver 32 via a synchronous serial data bus or SPI (acronym for the Serial Peripheral Interface), whereby the receiver 32 is controlled by demodulator 33; A DSP 36, programmed to process in real time the digital signals delivered to it by the demodulator 33, in particular by applying them filters and settings which will be detailed below; the signals pass through a bus 37 12S; a microphone 38 integrated in the housing 30 and connected to the DSP 36; An audio output 39 (for example of the 3.5 mm jack type) integrated in the housing 30 and connected to the DSP 36 to enable the connection of a sound reproducing device such as an audio headset 40, for the sound reproduction. signals processed by the DSP 36; a wireless communication interface 41, separate from the receiver 32, e.g. ultra-high frequency (UHF) IEEE 802.15 standard (called "Bluetooth®"), connected to an antenna 42. The signals received by the interface 41 transit to the DSP 36 by a data bus 43 standard 12S; at the same time, the DSP 36 controls the interface 41 by an asynchronous circuit 44 of the UART (Universal Asynchronous Receiver Transmitter) type; Preferably a port 45 for connecting the apparatus 29 to peripherals (eg a programming computer), or to the mains for charging a built-in battery to the apparatus 29. Various other communication protocols The IEEE 802.11 standard (commonly known as WiFi) or its HiperLAN equivalent (Hlgh PERFORMANCE radio Local Area Network), the GPRS standard (General Packet Radio Service), the protocol IRDA (InfraRed Data Association) the WUSB (Wireless Universal Serial Bus).
    [0015] As can be seen in FIG. 2, the sound system 1 also comprises a mobile terminal 46 of the Smartphone, tablet or laptop type. The terminal 46 is configured to control the apparatus 29 and comprises: a processor 47; A primary wireless communication interface 48, connected to the processor 47 and to an antenna 49 and configured to communicate with a remote server 50 via a local network 51 (of the LAN: Local area network type), metropolitan (MAN type: Metropolitan area network) or even extended (WAN: Wide Area Network) such as the Internet, 20 via standard protocols such as 3G UMTS (universal mobile telecommunication system) or 4G LTE (Long term evolution ); a wireless communication interface 52, e.g. ultra-high frequency (UHF) IEEE 802.15 standard (called "Bluetooth®"), connected to an antenna 52 and the processor 47; a graphical interface 54, e.g. a touch screen. The other wireless communication protocols mentioned above may also be suitable for the implementation of the interface 52. The processor 47 integrates an application (which can be downloaded remotely via the processor 47 connecting for this purpose to the server 50 ) comprising instructions for providing the listener 2, via the graphical interface 54, with different settings for each track (such as a gain level and a level of panning) and for communicating via the antennas 53. , 42 and the interfaces 52, 41, the settings thus made to the processor 36 of the device 29 for it to apply to the different tracks before returning to the headphone 40 via the socket 39.
    [0016] The listener 2 therefore has a device 29, which can be lent to him when he enters the room 5 and until the end of the show, and a terminal 46, which may be his own smartphone. The wireless communication between the apparatus 29 and the terminal 46 is of the point-to-point type, so as to guarantee the uniqueness of the adjustments made by the listener 2 and to prevent these settings from being unwantedly extended to a greater or lesser extent. Other devices 29. For this purpose, the apparatus 29 and the terminal 46 are, prior to the establishment of a media session between them, matched by a conventional procedure in which the apparatus 29 and the terminal 46 exchange messages. of signaling containing their respective identifiers and URLs (uniform 15 resource locator). It is conceivable to match a terminal 46 with several devices 29, which are then controlled by the same terminal 46 so that the same setting controlled from it is deployed on all devices 29. Note that the media session between the terminal 46 and the apparatus 29 does not necessarily include the exchange of audio tracks. Indeed, it is sufficient for the application programmed in the terminal 46 to make adjustments to the broadcast audio tracks, the auditory experience of the auditor 2 sufficient to know if these settings are suitable. The application programmed in the terminal 46 may also, for this purpose, 25 be dedicated to the show in progress, and automatically integrate good audio tracks. Alternatively, the application is reusable for several shows, and includes a certain number of potentiometers, some of which can be activated or deactivated via a software update available for download by the listeners from the server 50, from the beginning and 30 during the duration of the show. As illustrated in FIG. 2, the setting proposed to the listener 2 for each track, via the graphical interface 54 of the terminal 46, may be in the form of a potentiometer emulation 55. The settings can be 3032586 13 also be grouped for all tracks, for example as a mixer graphic emulation. The listener 2 can thus, by performing for each audio track settings of his choice (including the sound level of each track), improve his sound experience by proceeding to his own mixing from the raw audio data received from the transmitter According to one embodiment, the various audio tracks are, at the level of the third mixing table, all preset at a medium level, allowing the listener 2 to have, from the beginning of the broadcast, an audio signal with a default mix (eg performed by a sound engineer) allowing a comfortable listening to which some auditors 2 not very demanding can estimate not having to retouch. A listener 2 equipped with a device 29 coupled to a mobile terminal 46 as described above, can apply to the various audio tracks (which are restored mixed in his headset 40) the adjustment parameters that suit him, which makes him allows you to customize your listening by adapting it to your needs, your tastes as well as your faculties (auditory acuity, inaudible frequencies, psycho-acoustic problems, asymmetrical sensitivity between the two ears for example).
    [0017] In other words, the multitrack broadcast allows the listener 2 to act at his convenience on one or more audio tracks conveyed on dedicated tracks to finally get a custom mix. The listener 2, which locally has all the multitrack audio contents (ie the tracks are isolated from each other at the device 29), can, via the application he has installed on his mobile 46 terminal, modify at will the sound of the contents. However, there may be a time lag between the audio signal perceived by the listener 2 in his headphones 40, and the audio signal perceived by the listener 2 coming from the sound system 11. The two signals are both real-time signals. , from the same 4 sonic sources, via the same microphones 6, but the signal of the headphones 40 traveled to the listener 2 at the speed of light (300 000 km / s), while the signal of the sono 11, which may appear synchronized to the signal of the headset 40 when the latter 30 is close to the sound 11, accuses when the headset 40 is distant from the sound 11 a delay due to the path separating the listener 2 of the sono 11 , traveled at the speed of sound (about 340 m / s). Thus, from the point of view of a listener 2 located at a distance of about 20 m from the sound system, the perceived time difference between the two signals is about 60 ms. This offset is important enough to be fully detected by the human ear and be an embarrassment for the listener 2 who will feel an unpleasant echo effect. According to one embodiment, in order to minimize (or eliminate) this echo effect, and this independently of the position of the listener 2 in the room 5, the apparatus 29 (and more precisely its DSP 36) is programmed to capture, via its integrated microphone 38, the sound environment in its vicinity (and, therefore, in the vicinity of the listener 2). The DSP 36 is programmed to: - estimate the phase difference between the acoustic signal picked up by the microphone 38 (ambient signal) and the signal intended for the headphones 40 (mixed signal), for example by calculating the delay between maxima; perform a time registration of the mixed signal on the ambient signal, so as to synchronize them.
    [0018] According to one embodiment, the estimate of the delay between the maxima is carried out at a predefined frequency of the sound spectrum, e.g. at a low frequency (corresponding to a dull sound such as a kick) or on the contrary at a high frequency (corresponding to a high-pitched sound such as a hi-hat cymbal tempo).
    [0019] The time synchronization between two acoustic signals is obtained by delaying the mixed signal of the estimated difference between the maxima of the two signals. This difference is preferably re-estimated regularly (eg at regular intervals of 1 second or less) to account for any factor that could change its value (such as the displacement of the listener 2 in the room 5). or a change in the physical configuration of the sound stage such as adding and / or removing an obstacle).
    [0020] The microphone 38 being integrated in the apparatus 29 and therefore permanently worn by the listener 2, it follows that the time registration automatically refreshes itself accordingly. Timing does not eliminate the physical sensations due to the sound pressure on the listener's body 2, especially in the low frequencies (so it is known that bass drum beats are reflected in volume loud sound, in the belly and ribcage of the listeners). But the physical sensations are part of the live experience and, to the extent that the sensations are synchronized with the mixed signal, the listener 2 can have the illusion that they are generated by the mixed quality audio signal that he hears in his headset 40. However, if the time adjustment makes it possible to attenuate (or even suppress) the echo effect, it does not correct, on the other hand, the quality defects of the ambient signal, in particular if the Room acoustics and / or the sound system 11 and / or the mixing of the signal delivered by it is / are degraded. Therefore, according to an advantageous embodiment, the 40 audio headset is equipped with a noise reduction system (in English Noise Canceling) to reduce the intensity of the ambient signal perceived by the listener 2 and thus make to predominate the mixed signal to improve the sound experience of the listener 2. There are models of fully equipped helmets, cf. eg. the helmet brand BOSE® marketed under the name QuiefComfort®, or the helmet brand PARROT® marketed under the name Zik®. According to one (non-limiting) embodiment, the signal delivered by the third mixer table is eight tracks, which the listener 2 can separately adjust from its terminal 46 and which are combined by the device 29 to from these settings to build a mixed audio signal according to the wishes of the listener 2. It is advantageous, to further enhance the sound experience of the listener 30 2, and in particular to enhance the live atmosphere in the mixed signal , to turn a microphone 6 to the room to capture the ambient noises, and especially the noise of participation (typically the resumption of choral lyrics or applause). The room noises are then allocated to a dedicated audio track, which the listener 2 can adjust in his mix. Similarly, it is advantageous to add in the mixed signal to the listener a reverb sound effect which gives the sound a characteristic acoustic coloration of the place.
    [0021] In addition, according to an interesting embodiment, the apparatus 29 is miniaturized and integrated with the audio headset 40. It follows from the above that the sound system 1 allows the listener 2: to improve his sound experience by enabling him to become, in a way, his own sound engineer; to overcome the acoustic constraints related to room 5; to enjoy an optimal sound quality (the sound from the third mixing table 15 and mixed at will by the listener 2); while enjoying a live experience with the benefits that are inherent to it (presence of artists, interactivity, festive atmosphere); to improve his relationship with the artist by making it more intimate; to immerse yourself more completely in the show thanks to a spatialized sound. By transferring the different sound tracks to the apparatus 29 and, therefore, delegating the audio processing to it as a personal box, the disadvantages (especially the computational burden) of a centralized architecture are overcome. However, the device 29 does not support the entire computational load since it is up to the terminal 46 (separate from the device 29) 25 to display the settings for the benefit of the listener 2. This avoids to equip the device 29 with a graphic interface consuming computational (and energy) resources, for the benefit of a great autonomy of operation. The apparatus 29 is preferably electrically self-contained, e.g. 30 provided with a battery (not shown), capable of being loaded 3032586 17 episodically by branch on the sector, in particular via a serial port type USB (Universal Serial Bus). Finally, thanks to its architecture, the sound system 1 allows the listener 2 to benefit from a constant experience and quality 5 whatever its position with respect to the stage 3 or compared to the sono 11. D ' an acoustic point of view, there is therefore for the listener 2 neither bad room nor bad placement.
    权利要求:
    Claims (10)
    [0001]
    REVENDICATIONS1. Apparatus (29) for receiving and playing audio signals which comprises: a high frequency electromagnetic wave receiver (32) configured to receive broadcast signals incorporating a plurality of audio tracks; a wireless communication interface (41), separate from the receiver (32) and configured to receive at least one setting parameter to be applied to at least one audio track of the received signal; a digital signal processor (36) configured to apply the adjustment parameter to the audio track and produce a mixed signal.
    [0002]
    2. Apparatus (29) according to claim 1, characterized in that it comprises an output (39) audio for the sound reproduction of the signal mixed by the processor (36).
    [0003]
    The apparatus (29) of claim 1 or claim 2 which further comprises an integrated microphone (38) configured to pick up a sound signal in the vicinity of the apparatus (29).
    [0004]
    Apparatus (29) according to one of claims 1 to 3, wherein the processor (36) is configured to calculate a time offset between the signal mixed by the processor (36) and the sound signal picked up by the microphone ( 38).
    [0005]
    The apparatus (29) of claim 4, wherein the processor (36) is configured to apply to the mixed signal a delay equal to the calculated time offset. 25
    [0006]
    6. Headphone (40) audio incorporating an apparatus (29) according to any one of the preceding claims.
    [0007]
    7. Headphone (40) audio according to claim 6, which further comprises an acoustic noise reduction system.
    [0008]
    A sound system (1) for a show, said system (1) comprising: a mixing table (15) which receives audio tracks; a transmitter (17) connected to the mixing table (15) and configured to communally broadcast the audio tracks in multichannel mode; At least one device (29) for receiving and reading the broadcast signal, according to one of claims 1 to 5; a headset (40) connected to the apparatus (29) for the reproduction of the mixed signal.
    [0009]
    9. System (1) according to claim 8, characterized in that, the spectacle being alive and comprising sources (4) sound, the system comprises microphones (6) pointing to the sources (4) sound to capture the as an electrical signal forming an audio track for each microphone (6).
    [0010]
    10. System according to claim 8 or claim 9, characterized in that it comprises a mobile terminal (46) configured to transmit to the reception apparatus (29), via its second communication interface (41), to the least one setting parameter to apply to at least one audio track.
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    KR20170115079A|2017-10-16|
    US20180035232A1|2018-02-01|
    EP3254389A1|2017-12-13|
    US9942681B2|2018-04-10|
    FR3032586B1|2018-03-16|
    JP2018512107A|2018-05-10|
    引用文献:
    公开号 | 申请日 | 公开日 | 申请人 | 专利标题
    CN2904481Y|2005-11-10|2007-05-23|晶通信息有限公司|Blue tooth stereophone with man-machine interface device operation and control functions|
    US20140328485A1|2013-05-06|2014-11-06|Nvidia Corporation|Systems and methods for stereoisation and enhancement of live event audio|
    US9319019B2|2013-02-11|2016-04-19|Symphonic Audio Technologies Corp.|Method for augmenting a listening experience|
    US20150124995A1|2013-11-04|2015-05-07|David Walter Defnet|Public address system with wireless audio transmission|WO2018051161A1|2016-09-16|2018-03-22|Augmented Acoustics|Method for producing and playing video and multichannel audio content|
    US10652398B2|2017-08-28|2020-05-12|Theater Ears, LLC|Systems and methods to disrupt phase cancellation effects when using headset devices|
    FR3070568B1|2017-08-28|2020-06-12|Theater Ears, LLC|SYSTEMS AND METHODS FOR REDUCING PHASE CANCELLATION EFFECTS WHEN USING HEADPHONES|
    FR3073694B1|2017-11-16|2019-11-29|Augmented Acoustics|METHOD FOR LIVE SOUNDING, IN THE HELMET, TAKING INTO ACCOUNT AUDITIVE PERCEPTION CHARACTERISTICS OF THE AUDITOR|
    KR102184918B1|2017-12-06|2020-12-02|주식회사 아세스|System and method for communication for aquatic activities|
    US10481859B2|2017-12-07|2019-11-19|Powerchord Group Limited|Audio synchronization and delay estimation|
    CN108848434A|2018-05-24|2018-11-20|深圳普罗声声学科技有限公司|Sound processing method and device, the hearing aid of apparatus for processing audio|
    CN113411719B|2021-06-17|2022-03-04|杭州海康威视数字技术股份有限公司|Microphone cascade system, microphone and terminal|
    法律状态:
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    2016-08-12| PLSC| Publication of the preliminary search report|Effective date: 20160812 |
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    优先权:
    申请号 | 申请日 | 专利标题
    FR1550923A|FR3032586B1|2015-02-05|2015-02-05|APPARATUS FOR RECEIVING AND READING AUDIO SIGNALS AND LIVE SOUND SYSTEM|
    FR1550923|2015-02-05|FR1550923A| FR3032586B1|2015-02-05|2015-02-05|APPARATUS FOR RECEIVING AND READING AUDIO SIGNALS AND LIVE SOUND SYSTEM|
    EP16707869.0A| EP3254389A1|2015-02-05|2016-02-05|Appliance for receiving and reading audio signals and live sound system|
    PCT/FR2016/050243| WO2016124865A1|2015-02-05|2016-02-05|Appliance for receiving and reading audio signals and live sound system|
    JP2017559923A| JP2018512107A|2015-02-05|2016-02-05|Apparatus and live acoustic system for receiving and reading sound signals|
    KR1020177024836A| KR20170115079A|2015-02-05|2016-02-05|Apparatus and live sound system for receiving and reading audio signals|
    US15/548,842| US9942681B2|2015-02-05|2016-02-05|Appliance for receiving and reading audio signals and live sound system|
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