WO2004014105A1 - Systeme de traitement audio - Google Patents

Systeme de traitement audio Download PDF

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Publication number
WO2004014105A1
WO2004014105A1 PCT/IB2003/003389 IB0303389W WO2004014105A1 WO 2004014105 A1 WO2004014105 A1 WO 2004014105A1 IB 0303389 W IB0303389 W IB 0303389W WO 2004014105 A1 WO2004014105 A1 WO 2004014105A1
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WO
WIPO (PCT)
Prior art keywords
microphone
audio processing
processing system
signals
location
Prior art date
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Ceased
Application number
PCT/IB2003/003389
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English (en)
Inventor
Josep A. Rodenas
Roy Irwan
Ronaldus M. Aarts
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Koninklijke Philips NV
Original Assignee
Koninklijke Philips Electronics NV
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by Koninklijke Philips Electronics NV filed Critical Koninklijke Philips Electronics NV
Priority to AU2003250413A priority Critical patent/AU2003250413A1/en
Priority to JP2004525694A priority patent/JP2005535217A/ja
Publication of WO2004014105A1 publication Critical patent/WO2004014105A1/fr
Anticipated expiration legal-status Critical
Ceased legal-status Critical Current

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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; ELECTRIC HEARING AIDS; PUBLIC ADDRESS SYSTEMS
    • H04R5/00—Stereophonic arrangements
    • H04R5/027—Spatial or constructional arrangements of microphones, e.g. in dummy heads
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10H—ELECTROPHONIC MUSICAL INSTRUMENTS; INSTRUMENTS IN WHICH THE TONES ARE GENERATED BY ELECTROMECHANICAL MEANS OR ELECTRONIC GENERATORS, OR IN WHICH THE TONES ARE SYNTHESISED FROM A DATA STORE
    • G10H1/00—Details of electrophonic musical instruments
    • G10H1/36—Accompaniment arrangements
    • G10H1/361—Recording/reproducing of accompaniment for use with an external source, e.g. karaoke systems
    • G10H1/366—Recording/reproducing of accompaniment for use with an external source, e.g. karaoke systems with means for modifying or correcting the external signal, e.g. pitch correction, reverberation, changing a singer's voice
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S1/00—Two-channel systems
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10H—ELECTROPHONIC MUSICAL INSTRUMENTS; INSTRUMENTS IN WHICH THE TONES ARE GENERATED BY ELECTROMECHANICAL MEANS OR ELECTRONIC GENERATORS, OR IN WHICH THE TONES ARE SYNTHESISED FROM A DATA STORE
    • G10H2210/00—Aspects or methods of musical processing having intrinsic musical character, i.e. involving musical theory or musical parameters or relying on musical knowledge, as applied in electrophonic musical tools or instruments
    • G10H2210/155—Musical effects
    • G10H2210/265—Acoustic effect simulation, i.e. volume, spatial, resonance or reverberation effects added to a musical sound, usually by appropriate filtering or delays
    • G10H2210/295—Spatial effects, musical uses of multiple audio channels, e.g. stereo
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S1/00—Two-channel systems
    • H04S1/002—Non-adaptive circuits, e.g. manually adjustable or static, for enhancing the sound image or the spatial distribution
    • H04S1/005—For headphones
    • 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

Definitions

  • the present invention relates in general to an audio processing system, especially suitable for presenting music and/or songs to an audience. More specifically, the present invention relates to an audio processing system suitable for karaoke.
  • the phrase “driver” will be used for a device capable of converting electric input power to output sound waves, hi some conventions, such device is also indicated as speaker or loudspeaker, but in the context of the present invention the phrase “loudspeaker” will be used for an assembly comprising a housing or cabinet and one or more drives mounted in said housing, while the phrase “speaker” will be used for a person, such as for instance a person speaking, or singing, or playing an instrument; however, in order to try to avoidmisunderstanding, usually the phrase "singer” will be used for such person.
  • Stereo audio processing systems for presenting sound or song to an audience are generally known. Basically, such system comprises two or more loudspeakers driven by a stereo amplifier device, which may receive a conventional audio signal from a conventional source, such as a recording on CD.
  • stereo systems have a problem relating to the fact that listeners are capable of perceiving a direction from which sounds originates.
  • a listener will perceive the sound as originating from a virtual source having a virtual source location which depends on the location of the listener.
  • a listener positioned in a median plane with respect to said loudspeakers will perceive the virtual source location at a central location between said two loudspeakers.
  • a listener positioned outside said median plane will perceive the virtual source location as substantially coinciding with the location of the closest loudspeaker.
  • the speaker or singer may be moving on a stage or the like, so that in general the physical location of the speaker or singer does not correspond to the virtual source location as perceived by a listener.
  • the experience of seeing a person in one location and perceiving his voice originating from another location is a strange experience, lacking reality.
  • an audio processing system comprises actual source location detection means for generating a signal indicative of the actual location of an audio source such as a speaker, and processing means responsive to this detecting means to adapt the balance of the driving means for the loudspeakers such that the virtual source location is shifted such as to substantially correspond to the speaker location as detected.
  • the actual source location detecting means comprise a signal transmitter associated with the microphone or carried by the speaker/singer.
  • the detection means may further comprise a system of receivers adapted to receive the signal emitted by the sender, and a processor adapted to combine the receiver signals such as to determine the sender location (for instance by triangulation method).
  • the audio processing system is adapted to generate a microphone-bearing signal, preferably having a frequency outside the audible range, which microphone-bearing signal is picked-up by the microphone and sent back to the audio processing system, which is capable of calculating the distance from the microphone to each loudspeaker and hence to calculate the location of the microphone in space with respect to the coordinate system as defined by the loudspeakers.
  • an audio processing system capable of producing audio with one set of loudspeakers in such a way that two or even more virtual source locations are generated, i.e. (at least) one virtual music source location and (at least) one virtual singer source location , wherein the virtual music source location is a fixed location independent from the actual singer location, and wherein the virtual singer source location moves with the position of the singer as he moves around the stage.
  • an audio system comprises at least two sound processing channels, a first frequency channel substantially comprising music signals and a second channel substantially comprising microphone signals. Audio signals within the first channel are treated such as to result in a fixed virtual source location. In contrast, audio signals within the second channel are treated such as to result in a variable source location corresponding to the variable singer location.
  • Another problem relates to the speaker or singer himself. Since he can hear his own voice being reproduced by the loudspeakers, he can be considered as being one of the listeners, with the important difference that he will generally be located closer to the loudspeakers than the actual audience. He also will perceive a virtual source location, but due to his being close to the loudspeakers, the virtual source location as perceived by the singer will typically coincide with the location of the closest loudspeaker.
  • the present invention also aims to provide a solution to this problem.
  • the balance of the loudspeakers is adapted such that the virtual singer source location as perceived by the singer is a fixed location, for instance corresponding to the center C between the two loudspeakers, or, alternatively such that the virtual singer source location as perceived by the singer is a variable location moving along with the moving singer.
  • the singer wears a set of earphones, and the balance of signal supplied to the set of earphones is set such that the virtual singer source location as perceived by the singer is a fixed location halfway the two earphones.
  • Fig. 1 is a schematic top view of an arrangement of loudspeakers and listeners
  • Fig. 2 is a schematic top view of an arrangement of loudspeakers and listeners, for illustrating an adaptable virtual source location according to the invention
  • Figs. 3A and 3B are schematical block diagrams illustrating microphone location detection means
  • Fig. 4 is a schematical block diagram illustrating an audio processing system
  • Fig. 5 is a schematical block diagram illustrating another embodiment of an audio processing system.
  • the present invention will first be explained for the case of one speaker/singer/musician; later, the case of a plurality of speakers/singers/musicians will be discussed.
  • Fig. 1 schematically shows a sound producing system 1 comprising an arrangement of two loudspeakers LS L and LS R , arranged next to each other, at a certain mutual distance.
  • each loudspeaker may comprise one or more individual drivers.
  • a central location halfway between said two loudspeakers LSL and LSR is indicated at C.
  • a median plane between said two loudspeakers LSL an LS R is indicated at M. More specifically, the two loudspeakers LS L and LS R are placed in a symmetrical arrangement with respect to this median plane M.
  • the subscripts L and R denote left and right, as seen by a listener.
  • the figure shows a first listener LI located in a position coinciding with said median plane M, and a second listener L2 located besides the first listener LI, at a certain distance from said median plane M.
  • the figure also shows radiation diagrams 1 L and R1 R of said two loudspeakers LS L and LS R , respectively.
  • a radiation diagram indicates the relative intensity of sound generated by the corresponding loudspeaker into a certain direction, assuming that all sound originates from one point S , S R , respectively, which point is taken as origin of a polar coordinate system.
  • line through said point S L parallel to said median plane M, is taken as X-axis in this coordinate system.
  • a position A in front of the left loudspeaker LS L is defined in polar coordinates r and ⁇ , wherein r is the distance
  • the intensity of sound, generated by a loudspeaker into a direction ⁇ , is indicated by the length of a line piece from said one originating point to the intersection with the corresponding radiation diagram.
  • the relative intensity of sound into the direction of the first listener LI is indicated by the length
  • the radiation diagrams R1 L and R1 R constitute a (part of a) circle.
  • a listener when the two loudspeakers produce the same (mono) sound, a listener will perceive the sound as originating from a virtual source having a virtual source location VSL which depends on the location of the listener.
  • VSL virtual source location
  • the two loudspeaker systems LS and LS R have uniform radiation diagrams R1 L and R1 R , and that they are driven in a symmetrical way, the first listener LI will perceive the virtual source location VSL at said central location C, whereas the second listener L2 will perceive the virtual source location as substantially coinciding with the location of the closest loudspeaker, i.e. the right-hand loudspeaker LS R in Fig. 1.
  • the first listener LI will perceive good stereo quality, involving a good separation between left-hand channel sound and right-hand channel sound, whereas the second listener L2 will perceive all sound as originating from the right-hand loudspeaker LS R , which seriously affects or even eliminates the stereo perception.
  • the listener L2 receives sound from the closest loudspeaker LS R earlier than the sound from the more remote loudspeaker LSL- Second, the listener L2 receives sound from the closest loudspeaker LS R with larger intensity than the sound from the more remote loudspeaker LS L .
  • the area in space where good stereo quality is perceived is indicated as the "sweet spot".
  • the sweet spot will more or less coincide with the median plane M.
  • the two loudspeakers are arranged symmetrically, while further said two non-uniform radiation characteristics are mutually mirror-symmetrical, so that the situation does not change for listeners located along the median plane M.
  • the second listener L2 receives sound from the closest loudspeaker LS R earlier than the sound from the more remote loudspeaker LS L -
  • the second listener L2 receives sound from the closest loudspeaker LSR with relatively less intensity as compared to the intensity of sound received from the more remote loudspeaker LS L -
  • This intensity difference can be more than 10 dB.
  • this intensity difference at the location of the second listener L2 can counter-act the effect of the earlier arriving sound wave from the closest loudspeaker, such that the listener L2 will also perceive a virtual source location at C.
  • the extent of the area (sweet spot) where all listeners will perceive the same virtual source location has increased.
  • Fig. 1 also shows schematically a movable audio source, indicated at W, which source is free to move on a stage, as indicated by arrows in an Y-direction.
  • this source W is a person, who may be speaking, singing, or playing his instrument. Sound generated by this person W is picked up by a microphone 11 carried by this person W.
  • the audio signal produced by the microphone is processed by the audio processing system, and sound is emitted from the loudspeakers LS L and LS R .
  • the microphone signal S M is treated as "normal" mono signal, and is fed, after suitable amplification, to both loudspeakers LSL and LS R in equal intensities.
  • a balance ratio parameter P L R will be defined as the ratio G I /G R , wherein G L and G R indicate the gain of the left-hand audio channel and the right-hand audio channel, respectively. More particularly, the balance ratio parameter P LR may depend on signal frequency f.
  • the balance ratio parameter P R may be specified for a certain frequency, which will be indicated as PL R (I) - GL(f)/G (f).
  • the balance ratio parameter P LR may also be specified as a substantially constant value for all frequencies within a frequency range from a first frequency fl to a second frequency f2, which will be indicated as
  • the balance ratio P R 1; this means in practice that, if a mono signal is received, this signal is applied to the left-hand loudspeaker and to the right-hand loudspeaker with equal amplification. As explained above, this will lead to a virtual source location NSL located at said center C, at least for listeners within the sweet spot.
  • the present invention is partly based on the understanding that the virtual source location NSL can be shifted towards either loudspeaker by changing the balance ratio parameter P L .
  • a sound production system 2 in accordance with the present invention will be described, which is capable of providing a movable virtual source location.
  • the invention will be explained for the case of only one source.
  • such sound production system 2 is particularly desirable in case the source of the audio is mobile and visible to the listeners.
  • Such sound production system is already useful in the case of a relatively narrow sweet spot, but preferably it implements the widened sweet spot technology as described above.
  • the sound production system 2 comprises an arrangement of two loudspeakers LS L and LSR, connected to loudspeaker outputs 17 and 18, respectively, of an audio processing system 10.
  • Fig. 2 further indicates a person W, equipped with a microphone 11, who is free to move on a stage.
  • the microphone 11 generates a microphone signal S M which is transferred, either through a wire-coupling or a wireless coupling such as known per se, to a microphone input 16 of said audio processing system 10.
  • Said audio processing system 10 receives and processes the microphone signal S M and drives the loudspeakers LS L and LSR accordingly. As explained above, listeners will perceive the microphone sound as originating from a virtual source location.
  • said audio processing system 10 provides for an adaptive virtual singer source location ANSSL. More particularly, said audio processing system 10 is capable of generating sound in such a way that the virtual source location as perceived by a listener corresponds to the actual location of person W, as indicated in Fig. 2. Thus, if this person W is moving, indeed, said audio processing system 10 controls its output signals to the loudspeakers such that the adaptive virtual singer source location ANSSL is actually moving along with said person W.
  • the audio processing system 10 utilizes the effect that the virtual source location shifts towards the loudspeaker with the highest sound intensity.
  • the audio processing system 10 according to the present invention would generate its left-hand output drive signal at a lower magnitude than its right-hand output drive signal.
  • the audio processing system 10 according to the present invention is adapted to change the balance ratio P LR in accordance with the actual location of person W.
  • the audio processing system 10 may be used in combination with various types of loudspeakers.
  • the loudspeakers are of a type having an asymmetric radiation diagram R2L, R2R, as explained above with reference to Fig. 1, such that the shifting of the virtual source location ANSSL will be perceived by all listeners in relatively large sweet spot, while the shifted virtual source location will be approximately the same for all listeners in the increased sweet spot.
  • the shifting of the virtual source location will be such that the shifted virtual source location substantially coincides with the actual source location of person W.
  • the audio processing system 10 needs to have information on the actual source location.
  • the audio processing system 10 is provided with actual source location detection means 40.
  • such actual source location detection means 40 comprise a transmitter 41 carried by the person W or, preferably, associated with the microphone 11, and at least one receiver 42 coupled to the audio processing system 10 for sending a receiver signal S42 to the audio processing system 10.
  • the transmitter 41 is adapted to emit a predefined signal S41, which is received by the receiver 42.
  • Said predefined signal S41 is such that the receiver 42, or alternatively the audio processing system 10, is capable of calculating the actual location of the transmitter 41.
  • the transmitter 41 may be associated with a GPS module (not shown), and the predefined signal S41 may actually communicate the GPS-coordinates to the audio processing system 10.
  • the system comprises an array of receivers 42 coupled to the audio processing system 10, and that the transmitter 41 is adapted to emit a pulsed signal, which may be a light signal or a radio signal but which preferably is a sound signal, more preferably an ultrasound signal.
  • a pulsed signal which may be a light signal or a radio signal but which preferably is a sound signal, more preferably an ultrasound signal.
  • the audio processing system 10 will receive a plurality of receiver signals S42, the relative order and time differences being representative for the actual location of the transmitter 41 with respect to the receivers 42.
  • the audio processing system 10 is designed to generate predetermined microphone bearing signals S43 through the loudspeakers LS L and LS R , typically pulsed signals.
  • these location bearing signals S43 are generated within a frequency range inaudible to the human ear but within the capabilities of the loudspeakers and the microphone.
  • the emitted signals S43 are picked up by the microphone 11, the respective times of arrival depending on the actual distances between the microphone 11 and the respective loudspeakers LS L and LS R .
  • the signals S43 as picked up by the microphone 11 are converted into electrical signals S44 and, as part of the normal microphone signal S M , are transmitted to the input 16 of the audio processing system 10 through the normal microphone channel 12 (which may be a wireless channel).
  • the signals S43 as emitted by the respective loudspeakers LS L and LS R are coded such as to make possible a distinction between the different loudspeakers.
  • the audio processing system will receive a plurality of microphone pickup signals S44 L and S44 R , arriving in a certain relative order and with a certain time difference. It is noted that the audio processing system 10 is also aware of the emission times, and is consequently capable of calculating traveling times. The audio processing system 10 is designed to filter out these microphone pickup signals S44 L and S44R and process them in order to calculate the actual location of the detecting microphone 11 with respect to the respective locations of the loudspeakers LS L and LS R , on the basis of the respective traveling times of the microphone pickup signals S44L and S44 R .
  • the coding of the microphone bearing signals S43 may be any suitable coding, suitable for adequate distinction by the audio processing system 10.
  • the microphone bearing signals S43 L and S43 R may be mutually identical but emitted at different times for different loudspeakers. In this respect, the repetition time between successive signals emitted from different loudspeakers may even be longer than the travelling and processing time.
  • the microphone bearing signals S43L and S43R may comprise pulse trains having mutually different carrier frequencies.
  • the microphone bearing signals S43 L and S43R may comprise pulse trains containing a pulse width coding or a pulse distance coding.
  • a suitable coding is a coding of which the auto correlation function resembles a pulse as much as possible, such as a Barker code. It is also possible to use pairs of codes, such as for instance Golay codes; for more detailed information on Golay codes, reference is made to the article "The Merit Factor Of Long Low Autocorrelation Binary Sequences" by MJ.E. Golay in IEEE Transactions on Information Theory, ( May 1982), vol.28, nr.3, p.543-549.
  • the audio processing system 10 is adapted to derive, from the signals S42 or S44 as received at its input 16, a signal or parameter value indicating the actual source location of person W.
  • a signal or parameter value indicating the actual source location of person W.
  • the implementation of said signal or value is not relevant: it may for instance be an analog value such as a voltage level, or alternatively it may be a digital value. In this respect, it is noted that it is not specifically necessary to actually calculate such location. It will be sufficient if the audio processing system 10 has access to a relationship between on the one hand the received microphone signals S42 or S44 and on the other hand an adequate balance ratio P LR .
  • This relationship may be stored in a memory 13 associated with the audio processing system 10, for instance in the form of a translation table, h order to allow for different arrangements of the loudspeakers, it is preferred that the relationship is adjustable, such that in a learning phase, for various microphone locations, adequate control settings can be determined and stored in said memory.
  • the microphone signal S M also contains sound signals, indicated as voice signals SN, to be processed by the audio processing system 10 for reproduction through the loudspeakers LS L and LS R , for which purpose the audio processing system 10 generates loudspeaker drive signals SD L and SD R , respectively, at its outputs 17 and 18.
  • these loudspeaker drive signals SD L and SD R respectively, comprise individual drive signals for the individual drivers of the loudspeakers.
  • the audio processing system 10 is adapted to control the balance ratio P LR of the loudspeaker drive signals SD L and SD R to the loudspeakers LSL and LS R such that a virtual spot location ANSSL will be perceived corresponding to the actual source location of person W as indicated by the received microphone signals S42 or S44.
  • the audio processing system 10 will decrease the balance ratio P R , either by a relative amplification of the loudspeaker drive signals SD R to the right-hand loudspeaker LSR, or by a relative attenuation of the loudspeaker drive signals SD L to the left-hand loudspeaker LS L , or both.
  • the present invention has been explained for the case of a listener LI, L2 in an audience, i.e. at a relatively large distance from the loudspeakers.
  • the invention provides in correlating the sound perception to the visual perception of viewing the person W.
  • This person W also is a listener in the sense that he will hear his own voice (or the music produced by his instrument) being reproduced by the loudspeaker systems.
  • the speaker/singer W suffers from the strange perception of hearing his own voice (music) coming from outside himself, from a location not coinciding with his own location.
  • the speaker/singer W will hear his own voice (music) as coming from a location (virtual source location) coinciding with the closest loudspeaker, unless he is located in the median plane M.
  • the technical measures proposed for providing a singer-related solution are different from the technical measures proposed for providing an audience-related solution, but it is noted that in both cases the technical measures are based on the same inventive concepts.
  • an audio processing system implementing the technical measures proposed for providing a singer- related solution may be physically distinct from an audio processing system implementing the technical measures proposed for providing an audience-related solution.
  • the audio processing system 10 comprises a selection switch 14, and the audio processing system 10 is responsive to a signal received from this switch 14 to operate either in an "audience" mode as described above, or in a "singer” mode as will be discussed below.
  • the audio processing system 10 operates oppositely with respect to the “audience” mode.
  • the audio processing system 10 shifts the balance ratio P LR towards the loudspeaker (LS R ) closest to the speaker/singer W.
  • the audio processing system shifts the balance ratio P LR towards the loudspeaker (LS L ) which is the most remote from the speaker/singer W.
  • This balance shift may be such that the speaker/singer W perceives a virtual source location independent of his actual location, such as for instance the center C between the two loudspeakers.
  • the balance shift is such that the speaker/singer W perceives his own voice (music) as coming from his own location, i.e. an adaptive virtual singer source location ANSSL coinciding with his actual location.
  • the audio processing system needs to be provided with location detection means in order to be able to calculate the required amount of balance shift.
  • location detection means and the way in which a relation between detection signals and balance shift is determined, the same applies as mentioned above with respect to the "audience" mode.
  • the audio processing system is capable of solving the problem for the speaker/singer W as well as for the audience listeners 11, 12.
  • the speaker/singer W is equipped with a headset 48 of earphones, and the audio processing system 10 has a further output 19 coupled to a transmitter 47 for transmitting, preferably wireless, an output signal to the headset 48.
  • the selection switch 14 maybe omitted in this case.
  • the audio processing system 10 With respect to its outputs 17 and 18 driving the loudspeakers LS L and LS R , the audio processing system 10 operates in the "audience" mode as described above.
  • the audio processing system 10 may operate in a mono mode, such that the speaker/singer W hears his own voice with equal timing and equal intensity from the left-hand earphone and from the right-hand earphone and will perceive his own voice as coming from his own location.
  • the virtual source location as perceived by the speaker/singer W moves along with the actual source location, i.e. the actual location of the speaker/singer W.
  • the transmitter 41 may be associated with the headset 48 instead of the microphone 11.
  • a moving source loudspeaker/singer
  • a stationary source such as for instance in case of a singer being accompanied by music.
  • the stationary source may be a live orchestra, or a recording played from, for instance, CD.
  • the recording may even comprise singing.
  • stationary audio will be called “background”; in contrast, the audio produced by the person W will be called “foreground”.
  • the present invention also provides solutions to these further complications.
  • Fig. 4 illustrates an embodiment of the audio processing system 10 comprising an input 51 for receiving background signals, for instance from a CD player. It is assumed that input 51 is a stereo input; therefore, two signal lines are shown. Although the audio processing system 10 is capable of receiving foreground signals from a stereo microphone, it is assumed here that microphone input 16 is a mono input; therefore, only one signal line is shown.
  • the recorded background may comprise audio of the same nature as the foreground.
  • the person W may be a singer, and the recorded background may comprise singing.
  • the person W may be a musician such as a violin player, and the recorded background may comprise violin music.
  • the sound signals corresponding to this background audio having the same nature as the foreground will be indicated as "same nature background audio signals".
  • these same nature background audio signals may be suppressed by a band reject filter 52.
  • a suitable frequency range is, for example, 300 - 4500 Hz for the case of voice audio such as singing.
  • the microphone signals received at microphone input 16 may contain background, for instance because the microphone picks up the sound of an accompanying music band; if desired, these background signals may be suppressed by an echo feedback suppressor 53 or, alternatively, a band pass filter.
  • a suitable frequency range is, for example, 300 - 4500 Hz.
  • the audio processing system 10 comprises music processing means 54 for processing the background signals in a suitable manner; this background processing means 54 may be a conventional processing means, and is not discussed in great detail.
  • the background processing means 54 has outputs 56L and 56R (stereo).
  • the audio processing system 10 comprises foreground processing means 55 for processing the microphone signals (voice; music) in a suitable manner; this foreground processing means 55 may be a conventional processing means, and is not discussed in great detail.
  • the foreground processing means 55 may have two different outputs 57L and 57R for handling a stereo signal; however, in this embodiment, the foreground processing means 55 has one output 57.
  • the audio processing system 10 further comprises two controllable amplifiers 59L, 59R, both having their input connected to the output 57 of the foreground processing means 55.
  • the audio processing system 10 comprises control means 60 generating control signals to the controllable amplifiers 59L, 59R such as to set the gains G 59 L, G 59R of said controllable amplifiers 59L, 59R in response to the microphone bearing signals S42 or S44, the output signal from mode selection switch 14, and the information in memory 13, as will be clear to a person skilled in the art, such as to produce weighted foreground signals S 57 L and S 57R .
  • a first adder 61 has inputs connected to the left-hand output 56L of the background processing means 54 and to the output of left-hand amplifier 59L, and has its output connected to first output 17 to provide the left-hand loudspeaker drive signal SD L .
  • a second adder 62 has inputs connected to the right-hand output 56R of the background processing means 54 and to the output of right-hand amplifier 59R, and has its output connected to second output 18 to provide the right-hand loudspeaker drive signal SD R .
  • the ratio G 59L G5 R of said two gains corresponds to the balance ratio pLR.
  • the embodiment shown in Fig. 4 is also capable of providing a headset drive signal SDH to a headset output 19.
  • a third adder 63 has inputs connected to the output 57 of the foreground processing means 55 and to the left-hand output 56L and the right-hand output 56R of the background processing means 54. All these signals are added without weighing.
  • the mono signal at output 19 will generate a virtual source location between the earphones of the headset 48.
  • the audio processing system 10 illustrated in Fig. 4 comprises at least two signal processing paths for processing the movable microphone signals differently from the stationary source signals.
  • Fig. 5 schematically illustrates a more complicated embodiment of audio processing system 10, capable of handling an arrangement of two or more movable microphones.
  • Fig. 5 is comparable to Fig. 4, but the filters 52, 53 and processing units 54, 55 are omitted for sake of convenience.
  • the audio processing system 10 has a plurality of microphone input ports 16i (in the example shown: three), and separate signal processing paths for each microphone signal, such signal processing path including controllable amplifiers 59Li and 59Ri.
  • the audio processing system 10 will comprise corresponding control units 60i, responsive to a corresponding microphone location detection means (not shown).
  • each microphone signal is individually processed as described earlier, in correspondence with the actual location of the corresponding microphone.
  • each microphone signal component is reproduced by the loudspeakers LS L and LS R such that a corresponding adaptive virtual singer source location results, actually corresponding to the actual location of the corresponding singer.
  • the audio processing system 10 has a plurality of headset outputs 19i, each for supplying a headset output signal SDHi to a headset to be used by one respective singer.
  • a processing unit 70 receives all weighted microphone signals S57Li and S57RL All headset output signals may be mono, and may be identical to each other. However, in a more sophisticated processing, the illusion of directivity regarding the other singers is created in each headset output signal. To that end, all headset output signals are preferably stereo.
  • the corresponding weighted microphone signals S57Li and S57Ri are added and supplied in mono. Also, possible background from background input 51 is supplied in mono.
  • the microphone components of the other singers may be supplied with such left/right weighing that virtual source locations are perceived corresponding to the actual locations of the other singers.

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

Abstract

L'invention concerne un système de production sonore (1) comprenant deux haut-parleurs (LSL; LSR) disposés l'un à côté de l'autre sur des côtés opposés d'un plan médian (M). Chaque haut-parleur présente des caractéristiques de rayonnement (R2L; R2R) conçues de manière à obtenir un point idéal relativement large. Ce système de production sonore comprend également un système de traitement audio (10) doté d'une entrée de microphone (16) et de sorties de haut-parleur (17, 18), et est conçu de manière à générer un signal d'actionnement de haut-parleur (SDL ; SDR) présentant un certain rapport d'équilibrage (ςLR). Des moyens de détection d'emplacement source (40) sont associés au système de traitement audio (10) afin de générer un signal d'emplacement représentant l'emplacement source actuel d'un microphone (11). Ce système de traitement audio (10) réagit au signal d'emplacement reçu depuis les moyens de détection d'emplacement source (40) afin de modifier le réglage dudit rapport d'équilibrage (ςLR).
PCT/IB2003/003389 2002-07-31 2003-07-31 Systeme de traitement audio Ceased WO2004014105A1 (fr)

Priority Applications (2)

Application Number Priority Date Filing Date Title
AU2003250413A AU2003250413A1 (en) 2002-07-31 2003-07-31 Audio processing system
JP2004525694A JP2005535217A (ja) 2002-07-31 2003-07-31 オーディオ処理システム

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
EP02078147.2 2002-07-31
EP02078147 2002-07-31

Publications (1)

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WO2004014105A1 true WO2004014105A1 (fr) 2004-02-12

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PCT/IB2003/003389 Ceased WO2004014105A1 (fr) 2002-07-31 2003-07-31 Systeme de traitement audio

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JP (1) JP2005535217A (fr)
KR (1) KR20050047085A (fr)
CN (1) CN1672463A (fr)
AU (1) AU2003250413A1 (fr)
WO (1) WO2004014105A1 (fr)

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WO2006037197A3 (fr) * 2004-10-01 2006-05-04 Agm Academia De Ginastica Move Dispositif rythmique servant a produire et a jouer des sons, a realiser un accompagnement et a evaluer la performance de l'utilisateur
EP1703772A1 (fr) 2005-03-15 2006-09-20 Yamaha Corporation Système de détection de position, système de haut-parleur et dispositif terminal d'utilisateur
US20100322435A1 (en) * 2005-12-02 2010-12-23 Yamaha Corporation Position Detecting System, Audio Device and Terminal Device Used in the Position Detecting System
CN1797538B (zh) * 2004-12-01 2011-04-06 创新科技有限公司 用于使用户能够修改音频文件的方法和装置
WO2024146888A1 (fr) * 2023-01-04 2024-07-11 Snap Inc. Système et procédé de reproduction audio

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JP5082327B2 (ja) * 2006-08-09 2012-11-28 ソニー株式会社 音声信号処理装置、音声信号処理方法および音声信号処理プログラム
CN102238356A (zh) * 2010-05-07 2011-11-09 Tcl集团股份有限公司 分体式平板电视机音质控制的方法及电视机
JP5664581B2 (ja) * 2012-03-19 2015-02-04 カシオ計算機株式会社 楽音発生装置、楽音発生方法及びプログラム
KR20160122835A (ko) * 2014-03-18 2016-10-24 로베르트 보쉬 게엠베하 적응형 음향 강도 분석장치
CN104698973B (zh) * 2015-01-23 2017-11-21 中国矿业大学 一种并联式矿用破碎站的自动对中系统及对中方法
EP3547718A4 (fr) 2016-11-25 2019-11-13 Sony Corporation Dispositif de reproduction, procédé de reproduction, dispositif de traitement d'informations, procédé de traitement d'informations, et programme
CN106412770A (zh) * 2016-12-16 2017-02-15 齐旭辉 一种基于无线定位的音响声像再现系统及工作方法
CN106535058A (zh) * 2017-02-07 2017-03-22 黄光瑜 舞台现场扩音重现立体声的方法
CN111903143B (zh) * 2018-03-30 2022-03-18 索尼公司 信号处理设备和方法以及计算机可读存储介质
CN114389732A (zh) * 2022-03-08 2022-04-22 深圳德威音响有限公司 一种数字调音台调音系统和方法
CN116347322B (zh) * 2023-03-17 2025-11-14 深圳市龙芯威半导体科技有限公司 基于uwb定位的立体声合成方法、系统、设备及存储介质

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WO2006037197A3 (fr) * 2004-10-01 2006-05-04 Agm Academia De Ginastica Move Dispositif rythmique servant a produire et a jouer des sons, a realiser un accompagnement et a evaluer la performance de l'utilisateur
CN1797538B (zh) * 2004-12-01 2011-04-06 创新科技有限公司 用于使用户能够修改音频文件的方法和装置
EP1703772A1 (fr) 2005-03-15 2006-09-20 Yamaha Corporation Système de détection de position, système de haut-parleur et dispositif terminal d'utilisateur
US7929720B2 (en) 2005-03-15 2011-04-19 Yamaha Corporation Position detecting system, speaker system, and user terminal apparatus
US20100322435A1 (en) * 2005-12-02 2010-12-23 Yamaha Corporation Position Detecting System, Audio Device and Terminal Device Used in the Position Detecting System
EP1962558A4 (fr) * 2005-12-02 2013-06-19 Yamaha Corp Système de détection de position, dispositif audio et dispositif terminal utilisés dans le système de détection de position
WO2024146888A1 (fr) * 2023-01-04 2024-07-11 Snap Inc. Système et procédé de reproduction audio

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KR20050047085A (ko) 2005-05-19
JP2005535217A (ja) 2005-11-17
AU2003250413A1 (en) 2004-02-23

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