Low-frequency automatic calibration audio system

Abstract

An audio system is provided that includes at least two low-frequency transducers for projecting sound into a room and a portable device with at least two microphones for receiving sound from multiple directions at a first listening position. The microcontroller is programmed to provide a calibration command in response to a user input and to provide a measurement signal indicative of sound received by the microphone array. The processor is programmed to provide a test signal in response to receiving the calibration command, and each low-frequency transducer is adapted to generate a test sound in response to the test signal. The processor is further programmed to process the measurement signal to predict an acoustic response at a second listening position adjacent to the first listening position and to adjust acoustic settings associated with each low-frequency transducer to optimize sound at the first listening position and the second listening position.

Description

Technical Field

[0001] The present disclosure is directed to a system and method for automatically calibrating an audio system.

Background Art

[0002] An audio system typically includes loudspeakers that convert electrical signals into acoustic signals. A loudspeaker may include one or more transducers that generate a range of acoustic signals, such as high-frequency signals, mid-frequency signals, and low-frequency signals. One type of loudspeaker is a subwoofer that may include a low-frequency transducer that generates low-frequency signals.

[0003] An audio system may generate acoustic signals in various listening environments, such as a home listening room, a home theater, a movie theater, a concert hall, a vehicle interior, a recording studio, etc. The listening environment includes a plurality of listening positions for a person to hear the acoustic signals generated by the loudspeakers, for example, sofas in different sections within a home listening room.

[0004] The listening environment may affect the acoustic signals including low-frequency, mid-frequency, and / or high-frequency signals at the listening position. Depending on where the listener is in the room, the volume may vary in different tones. This can be particularly true for low frequencies in a small room in a house because the volume (measured in amplitude) of a particular tone or frequency can be artificially increased or decreased. Low frequencies can be important for enjoying music, movies, and most other forms of audio entertainment. In the example of a home theater, the room boundaries, including walls, curtains, furniture, furnishings, etc., may affect the acoustic signals as they travel from the loudspeakers to the listening positions.

[0005] The acoustic signal received at the listening position can be measured. One measure of the acoustic signal is a transfer function that can measure aspects of the acoustic signal including amplitude and / or phase at a single frequency, a discrete number of frequencies, or a frequency range. The transfer function can measure different ranges of frequencies. The amplitude of the transfer function is related to volume. Generally, the amplitude at a single frequency or frequency range is measured in decibels (dB). The amplitude deviation can be expressed as a positive or negative dB value relative to a specified target value. When the amplitude deviation is considered at multiple frequencies, the target curve can be flat or of any shape. The relative amplitude response is a measure of the amplitude deviation at one or more frequencies from the target value at one or more frequencies. The closer the amplitude value measured at the listening position is to the target value, the better the amplitude response. The deviation from the target value reflects the changes that occur when the acoustic signal interacts with the room boundaries. Peaks represent an increase in the amplitude deviation from the target value, and dips represent a decrease in the amplitude deviation from the target value.

[0006] These deviations in the amplitude response can depend on the frequency of the acoustic signal reproduced by the subwoofer, the position of the subwoofer, and the position of the listener. To the listener, low frequencies such as those in a soundtrack or movie may not sound as they are recorded on the recording medium but rather as if distorted by the room boundaries. Thus, the room can change the acoustic signal reproduced by the subwoofer, potentially degrading the frequency response performance, including the low-frequency performance of the audio system. Many techniques attempt to reduce or eliminate the amplitude deviation at a single listening position. Additional techniques attempt to reduce or eliminate the amplitude deviation at multiple listening positions. For example, U.S. Patent No. 7,526,093 to Devantier et al., assigned to Harman International Industries Inc., discloses a system for configuring an audio system using an acoustic field measurement approach that includes taking acoustic measurements from each subwoofer position and each listening position. Removing the amplitude deviation at multiple different listening positions is more difficult and typically relies on using multiple sound sources at different locations in the room.

Prior Art Documents

Patent Documents

[0007]

Patent Document 1

Summary of the Invention

[0008] In one embodiment, an audio system includes at least two low-frequency transducers for projecting sound into an indoor and a portable device. The portable device includes a microphone array having at least two microphones for receiving sound from a plurality of directions at a first listening position. A microcontroller is programmed to provide a calibration command in response to a user input and to provide a measurement signal indicative of the sound received by the microphone array. A processor is programmed to provide a test signal to each low-frequency transducer in response to receiving the calibration command, and each low-frequency transducer is adapted to generate a test sound in response to the test signal. The processor is further programmed to process the measurement signal to predict an acoustic response at a second listening position adjacent to the first listening position and to adjust acoustic settings associated with each low-frequency transducer to optimize the sound at the first and second listening positions.

[0009] In another embodiment, an audio system includes at least two low-frequency transducers, and each of the at least two low-frequency transducers is adapted to project sound into an indoor in response to receiving an audio signal. A controller is configured to provide a test audio signal to each low-frequency transducer in response to receiving a calibration command, to process a measurement signal indicative of the sound measured by at least two microphones at a first listening position in the indoor, to predict an acoustic response at a second listening position adjacent to the first listening position, and to adjust acoustic settings associated with each of the at least two low-frequency transducers to optimize the sound at the first and second listening positions.

[0010] In yet another embodiment, the audio system comprises at least two low-frequency transducers, a portable device, and a controller. Each of the at least two low-frequency transducers is adapted to project sound into the room in response to receiving an audio signal. The portable device includes at least two microphones for measuring sound from a plurality of directions at a first listening position, and a microcontroller programmed to provide a calibration command in response to a user input and to provide a measurement signal indicative of the sound measured by the at least two microphones. The controller is configured to, in response to receiving the calibration command, provide a first audio signal indicative of a predetermined sound sweep to each of the at least two low-frequency transducers, process the measurement signal to predict an acoustic response at a second listening position adjacent to the first listening position, and adjust the acoustic settings associated with each of the at least two low-frequency transducers in accordance with the first listening position to optimize the sound at the first and second listening positions. The controller is further configured to receive a music signal and provide a second audio signal indicative of the music signal to each of the at least two low-frequency transducers. number to each of the at least two low-frequency transducers and adjusted acoustic settings is further configured. BRIEF DESCRIPTION OF THE DRAWINGS

[0011]

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Embodiments for Carrying Out the Invention

[0012] If necessary, detailed embodiments of the present invention are disclosed herein, but it should be understood that the disclosed embodiments are merely illustrative examples of the present disclosure that can be embodied in various forms and alternative forms. The figures are not necessarily to scale, and some features may be exaggerated or minimized to show details of specific components. Therefore, the specific structural details and functional details disclosed herein should not be construed as limitations, but only as representative principles.

[0013] Referring to FIG. 1, an audio system according to one or more embodiments is shown and generally referred to by reference numeral 100. The audio system 100 is shown within a home listening environment such as a room 102. The audio system 100 includes a loudspeaker such as a sound bar 104 that includes one or more high-frequency transducers, mid-frequency transducers, and low-frequency transducers (e.g., subwoofers). The audio system 100 further includes a controller 106 and a portable measurement device 108. The audio system 100 may further include additional loudspeakers such as an external subwoofer 110 mounted at another location in the room 102. A user 112 is shown holding the portable measurement device 108 at a first listening position 114, such as a seat in the center of a sofa. There are two additional listeners adjacent to the user 112, one listener sitting at a second listening position 116 to the left of the user 112 and another listener sitting at a third listening position 118 to the right of the user 112. The audio system 100 automatically calibrates the sound projected to a plurality of locations within the room 102, such as the first, second, and third listening positions 114, 116, 118, by the sound bar 104 and the external subwoofer 110 in response to a command from the user 112 to activate the "one click" or the portable measurement device 108 and perform an acoustic measurement at the first listening position 114.

[0014] Referring to FIG. 2, the sound bar 104 includes a controller 106 that includes a processor 120, such as a digital signal processor (DSP), and a memory (not shown). The sound bar 104 includes, according to one or more embodiments, a high frequency (HF) transducer 122, a mid-frequency transducer 123, and a low frequency transducer, i.e., a subwoofer 124. In one or more embodiments, the subwoofer 124 provides sound between about 0 and 120 Hz, the mid-frequency transducer 123 provides sound between about 120 Hz and 2 kHz, and the high frequency (HF) transducer 122 provides sound between about 2 kHz and 20 kHz. The sound bar 104 further includes a transceiver 126, such as a low output radio frequency (RF) transceiver, connected to the controller 106 for wireless communication with other devices. The processor 120 receives an audio signal from a sound source 127, such as a television, media player, etc., and separates the audio signal into channels for each of the sound bar transducers 122, 123, and 124, and any additional transducers (e.g., the LF transducer 144 of the external subwoofer 110).

[0015] The portable measurement device 108 includes a microphone array 128 supported within a small housing 130 (e.g., a handheld remote control). According to one embodiment, the microphone array 128 is a primary array that includes two microphones, namely a left microphone 132 and a right microphone 134. The left and right microphones 132, 134 are relatively close to each other, for example, packaged about 10 cm apart, and are arranged in opposite directions, for example, left and right, to provide a directional sensor. Each microphone 132, 134 may be an omnidirectional microphone such as the Knowles MM20-33366-B116 microphone. In another embodiment, the microphone array 128 is a secondary array that includes three omnidirectional microphones, namely a left microphone 132, a right microphone 134, and a central microphone 136 located at the center between the left microphone 132 and the right microphone 134. In other embodiments of the audio system 100, the microphone array 128 includes a combination of different microphones, such as one or more acoustic cardioid microphones and one or more omnidirectional microphones, to form an array with left-facing lobes, optionally, with forward and rearward lobes as well. 次 The above arrays are formed.

[0016] The portable measurement device 108 includes a microcontroller 138 and a transceiver 140 (e.g., a low-power radio frequency (RF) transceiver). The transceiver 140 is connected to the microcontroller 138 for wireless communication with other devices such as the sound bar 104. The portable measurement device 108 further includes an externally accessible button 142 that communicates with the microcontroller 138 to initiate an auto-calibration sequence of the audio system 100. In one or more embodiments, some or all of the functions of the portable measurement device 108 may be provided by a smartphone or a tablet. For example, a smartphone may include a processor, a transceiver, and a touch screen (button) like the microcontroller 138, the transceiver 140, and the button 142.

[0017] The external subwoofer 110 includes one or more low-frequency transducers 144 and a subwoofer controller 146. The external subwoofer 110 further includes a transceiver 148 (e.g., a low-output radio frequency (RF) transceiver). The transceiver 148 is connected to the subwoofer controller 146 for wireless communication with other devices such as the sound bar 104 and the portable measurement device 108. In other embodiments, the external subwoofer 110 communicates with the sound bar 104 by wired communication.

[0018] The controller 106 includes a measurement module 150 for controlling a calibration sequence. According to one or more embodiments, the controller 106 further includes an optimization module 152 for adjusting the parameters of each audio channel or transducer, such parameters including individual channel delay, gain, polarity, filter, and the like.

[0019] Controller 106, microcontroller 138, and subwoofer controller 146 are each shown as a single controller, but may each include a plurality of controllers or may be embodied as software code within one or more other controllers. Controllers 106, 138, 146 collectively include any number of microprocessors, ASICs, ICs, memories (e.g., FLASH (registered trademark), ROM, RAM, EPROM, and / or EEPROM), and software code that cooperate with each other to perform a series of operations. Such hardware and / or software may be grouped into modules to perform specific functions. Any one or more of the controllers or devices described herein may include computer-executable instructions that can be compiled or interpreted from a created computer program using various programming languages and / or techniques. Generally, a processor (such as a microprocessor) receives instructions from, for example, a memory, a computer-readable medium, etc., and executes the instructions. The processing unit includes a non-transitory computer-readable storage medium capable of executing instructions of a software program. The computer-readable storage medium may be, but is not limited to, an electronic storage device, a magnetic storage device, an optical storage device, an electromagnetic storage device, a semiconductor storage device, or any suitable combination thereof. According to one or more embodiments, controllers 106, 138, 146 further include predetermined data stored in the memory, i.e., a "look-up table".

[0020] Referring to FIG. 3, the placement of the subwoofer and the listener in a small room, as well as the size and shape of the room, affect the resulting low-frequency response. FIG. 3 shows how standing waves look in room 102 when there is a soundbar 104 at one end. The subwoofer 124 of the soundbar 104 generates low-frequency sound, and three of the lowest-frequency standing waves are shown as the first mode 320, the second mode 322, and the third mode 324, with each mode corresponding to different frequencies, for example, 30 Hz, 60 Hz, and 90 Hz for a set of axial modes respectively. FIG. 3 represents the one-dimensional three-axis mode of room 102 at a certain moment. The maximum sound pressure exists at the boundaries of the room (i.e., both ends of room 102 in FIG. 2). The points where the sound pressure drops to the minimum value are generally called "nulls". Without mode attenuation, the sound pressure at the nulls drops to zero. However, in most actual rooms, the dip in the response at the nulls is in the range of about -20 dB.

[0021] Standing waves have peaks and dips at various positions throughout the room, and large amplitude deviations may occur depending on the position of the listener. Thus, since user 112 is at the null positions for both the first mode 320 and the third mode 324, the sound generated by the subwoofer 124 at these frequencies will sound much smaller than it actually is. Conversely, since user 112 is located at the peak of the second mode 322, the sound generated by the subwoofer 124 at this frequency will sound much larger than it actually is. Listeners at the second listening position 116 and the third listening position 118 are not placed at null positions with respect to any of the modes, so they can hear all three modes and obtain a more comfortable and accurate listening experience.

[0022] Referring to FIGS. 4A through 4B, one approach to addressing the standing wave problem in the single subwoofer scenario of FIG. 3 is to equalize the frequency response. FIG. 4A is a graph 400 having three curves 404, 406, 408 representing the frequency response of an acoustic measurement generated by a single subwoofer in a room, e.g., subwoofer 124 in room 102 of FIG. 3, according to one embodiment. The first curve 404 represents the frequency response of the sound measured at the first listening position 114. The second curve 406 represents the frequency response of the sound measured at the second listening position 116. The third curve 408 represents the spatial average of the first curve 404 and the second curve 406. As shown in FIG. 4A, since the first curve 404 and the second curve 406 rise and fall simultaneously at different frequencies, there is little or no difference between the listening positions, or between the seats, and the frequency response can be equalized to a desired target value by applying an equalizer filter to the parameters of the signal provided to each transducer.

[0023] FIG. 4B is a graph 410 including a first curve 414 representing the equalized frequency response of the sound measured at the first listening position, a second curve 416 representing the equalized frequency response of the sound measured at the second listening position, and a third curve 418 showing the spatial average of the first curve 414 and the second curve 416. The first curve 414, the second curve 416, and the third curve 418 are all substantially parallel to each other, which indicates that (as shown in FIG. 4A) when there is no variation between the listening positions, the frequency responses at both listening positions can be improved by equalizing the acoustic signal supplied to the subwoofer 124.

[0024] Referring to FIGS. 5A and 5B, the simple equalization approach of FIGS. 4A and 4B is not effective when there are differences between seats. FIG. 5A shows, according to another embodiment, a graph 500 having a first curve 504, a second curve 506, and a third curve 508 representing the frequency response of an acoustic measurement generated by a single subwoofer in a room, for example, subwoofer 124 in room 102 of FIG. 3. The first curve 504 represents the frequency response of the sound measured at the first listening position 114. The second curve 506 represents the frequency response of the sound measured at the second listening position 116. The third curve 508 represents the spatial average of the first curve 504 and the second curve 506. The spatial average curve 508 is approximately equal to the spatial average curve 408 of FIG. 4A. As shown in FIG. 5A, since the first curve 504 and the second curve 506 do not rise and fall simultaneously across the entire frequency range, there are differences between the listening positions.

[0025] FIG. 5B is a graph 510 including a first curve 514 representing the equalized frequency response of the sound measured at the first listening position 114, a second curve 516 representing the equalized frequency response of the sound measured at the second listening position 116, and a third curve 518 showing the spatial average of the first curve 514 and the second curve 516. The spatial average curves 408, 508 are approximately equal to each other, but the equalization curves 514 and 516 diverge from each other, indicating that such an equalization approach is not effective when there are differences between the listening positions (as shown in FIG. 5A). The fact that there are differences in the frequency response between the listening positions means that correcting the sound at one position with simple equalization may have an adverse effect on the sound at another position.

[0026] Referring to FIG. 6, another approach to addressing differences in sound quality between listening positions is to use multiple subwoofers at different positions within room 102 because the subwoofers at different positions can partially cancel out specific standing waves. FIG. 6 shows room 102 where both the subwoofer 124 of soundbar 104 and the external subwoofer 110 generate low-frequency modes from different positions, thereby canceling out two of the three modes, namely the first mode 620 and the third mode 624, but the second mode 622 at the first listening location 114 is not canceled. However, this approach requires an additional loudspeaker, such as the external subwoofer 110, and there are still nulls adjacent to the second and third listening positions 116, 118 within room 102.

[0027] FIG. 7 is a diagram showing an example of a multi-subwoofer, multi-receiver scenario in a room. Reference symbol I is an audio signal input to the audio system 100. The loudspeaker / room transfer functions from the subwoofer 124 (speaker 1) of the soundbar 104 and the external subwoofer 110 (speaker 2) in the room 102 to two receiving positions (e.g., the first listening position 114 and the second listening position 116) are represented by H 11 、H 12 、H 21 、and H 22 respectively, and R1 and R2 represent the transfer functions obtained at the receiving (listening) positions. There is a transmission path from each sound source to each receiver, and in this example, four transfer functions are generated. Assuming that the signals transmitted to each loudspeaker can be electrically modified as represented by M1 and M2, the modified signals can be added. Here, M is a complex modification factor that may or may not depend on frequency. To explain the complexity of the mathematical solution, the following equations solve the linear time-invariant system in the frequency domain.

[0028] R1(f)=IH 11 (f)M1(f) + IH 21 (f)M2(f) R2(f)=IH 12 (f)M1(f) + IH22 (f)M2(f) (1) Here, it is understood that all transfer functions and correction factors are complex numbers. This is recognized as a set of linear equations and can be represented more compactly in matrix form as follows.

[0029] [Number] Or simply, HM = R, (3) Here, it is assumed that the input I is 1.

[0030] A general goal of optimization is to make R equal to 1, that is, the signals of all receivers become identical to each other. R may be regarded as an objective function where both R1 and R2 are equal to 1. Solving equation (3) for M (the correction factor of the audio system), M = H -1 becomes the reciprocal of H. Since H is frequency-dependent, the solution for M is calculated at each frequency. However, the value of H may be such that the calculation of the reciprocal is difficult or the implementation is unrealistic (such as the gain of some loudspeakers at some frequencies being unrealistically high).

[0031] Since it is not always possible to determine an exact mathematical solution, in conventional approaches, attempts have been made to determine the best computable solution, such as the solution with the minimum error. The error function defines how close a particular configuration is to the desired solution, and the lowest error represents the best solution. However, this mathematical methodology requires a large amount of computational energy and can only solve the solution for two parameters. Acoustic problems involving more parameters are becoming increasingly difficult to solve. Some audio systems attempt to solve the problem by analyzing acoustic measurements taken at various locations within the listening room, but such an approach may be difficult for end-users in a home listening environment.

[0032] Referring to FIG. 8 and returning to FIG. 2 for reference, a method for automatically calibrating the audio system 100 is shown in accordance with one or more embodiments and is generally designated by the reference numeral 800. The method 800 is implemented using software code included within the controller 106 in accordance with one or more embodiments. The method is described using a flowchart shown in a number of sequential steps, and one or more steps may be omitted and / or performed in a different manner in one or more other embodiments. In other embodiments, the software code is distributed among multiple controllers, such as the controller 106 and the microcontroller 138.

[0033] In step 802, the user 112 initializes the calibration sequence by pressing a button 142 of the portable measurement device 108 while seated at the first listening position 114. In other embodiments, the calibration procedure may be initialized in response to an audio command or by sending a signal using a smartphone or tablet. The microcontroller 138 of the portable measurement device 108 generates an initialization command (CAL) and transmits the initialization command to the sound bar 104 via the transceiver 140.

[0034] In step 804, the controller 106 receives the initialization command via the transceiver 126, and the processor 120 activates the measurement module 150 to provide a sound sweep signal to the subwoofer 124 to emit as sound. In one embodiment, the sound sweep corresponds to a sound whose amplitude varies from -60 to 60 dB and whose frequency varies from 0 to 150 Hz. In step 806, the microphone array 128 of the portable measurement device 108 measures the sound sweep at the first listening position 114 and transmits sweep data (MIC) to the sound bar 104.

[0035] In step 808, the controller 106 processes the sweep data to predict responses at other listening positions, such as the second listening position 116 and the third listening position 118. The processor 120 may provide the predicted responses to the optimization module 152, which uses an optimization algorithm such as the sound field management algorithm described in U.S. Patent No. 7,526,093 to Devantier et al., the entirety of which is incorporated herein by reference for further processing of the data. In one or more embodiments, the controller 106 may execute multiple sweeps to repeat steps 804 through 808, or sample the background noise and adjust the excitation to give more energy to the noisier frequencies, or use other techniques or algorithms to increase the signal-to-noise ratio. Next, in step 810, the controller 106 adjusts the acoustic settings, such as the per-channel parameters including time delay, gain, polarity, and filter coefficients, based on the predicted responses.

[0036] FIG. 9 shows one embodiment of an audio system 100 including a primary microphone array that executes the auto-calibration method 800. Referring to FIG. 9 and back to FIG. 1 for reference, the microphone array 128 is a primary array including a left microphone 132 and a right microphone 134 according to one or more embodiments. The sound provided by the audio system 100 reflects off the surfaces within the room 102 and is similar to the sound provided by a plurality of virtual sound sources disposed at corresponding locations outside the room. The acoustic response at the first listening position 114 within the room 102 is similar to what would occur if there were no room and no cloud for such virtual sound sources. When the user 112 moves from the first listening position 114 to the second listening position 116, the user 112 approaches the virtual image on the immediate left by approximately one meter, i.e., nearly one meter between the centers of adjacent cushions on the sofa, and moves away from the virtual image on the immediate right by one meter. For virtual sound sources directly in front of or behind the user, the difference in distance is minimal or non-existent. For virtual sound sources in other directions, there is an intermediate difference in the distance to the virtual sound source.

[0037] FIG. 9 shows a method in which the left arrival sound and the right arrival sound are measured at step 806 using the directional microphones 132, 134, processed at step 808 by shifting the impulse response based on the estimated distance between the listening positions, and then recombined. At step 806, the portable measurement device 108 measures a sound sweep using the primary microphone array 128. The microphone array 128 is configured as a directional microphone in which the left and right microphones 132, 134 are arranged at a narrow interval in opposite directions along the axis A-A, for example, about 10 cm apart. FIG. 9 includes a left pole plot 902 representing the sound measured by the left microphone 132 and a right pole plot 904 representing the sound measured by the right microphone 134. In the illustrated embodiment, the left and right microphones 132, 134 are cardioid microphones that attenuate sound arriving from off-axis directions.

[0038] At step 808, the controller 106 of the sound bar 104 processes the sound sweep data. The processor 120 includes accurate signal delay components and gain components for each of the microphones 132, 134. The processor 120 decomposes the sound received by each of the microphones 132, 134 of the microphone array 128 into a left arrival component and a right arrival component, as shown by the left reflectogram 908 and the right reflectogram 910. The sound received directly from the sound bar 104 is received by the front lobe and the rear lobe (not shown) of the microphone array 128, and there is no time shift.

[0039] The measurement module 150 can predict the sounds existing at different listening positions, such as the second listening position 116 and the third listening position 118, by adjusting the acoustic settings at step 810 by shifting the time delays associated with the sound measured by the left microphone 132 (Δt L ) and the sound measured by the right microphone 134 (Δt R ) according to equations 4 and 5 shown below.

[0040] ΔtL = + / - d / c (4) Δt R = - / + d / c (5) Here, (d) represents the distance between the listening positions, for example, 1 meter, (c) represents the speed of sound, (-) is used to predict the sound at a position in the same direction as the microphone (for example, a position to the left of the left microphone 132), and (+) is used to predict the sound at a position in the opposite direction to the microphone (for example, a position to the right of the left microphone 132). For example, the audio system 100 subtracts d / c from each impulse measured by the left microphone 132 as referenced by reference numeral 916, and adds d / c to each impulse measured by the right microphone 134 as referenced by reference numeral 918 to predict the sound at the second listening position 116 facing the left of the first listening position 114. Next, the audio system 100 recombines the shifted signals represented by the simplified reflectogram as referenced by reference numeral 920 as a whole.

[0041] Figures 10 to 16 show a part of an automatic calibration method 800 executed by one embodiment of an audio system 100 including a secondary microphone array. The microphone array 128 is a secondary array including a left microphone 132, a right microphone 134, and a center microphone 136 according to one or more embodiments. Figures 10 to 12 show the basic theory behind the method 800 for automatically calibrating the audio system described with reference to Figure 8 by decomposing a complex sound field and then extrapolating the sound to predict the response at a new position.

[0042] Referring to FIG. 10, at any point in space, for example, at the first listening position 114, sound arrives from all directions as indicated by the converging arrows. Referring to FIG. 11, the audio system 100 uses the secondary microphone array 128 to simplify the complex sound field of FIG. 10 into its orthogonal components, namely, the left sound component 1102, the right sound component 1104, the front sound component 1106, and the rear sound component 1108. Next, referring to FIG. 12, the audio system 100 then extrapolates the sound and predicts the response at the new position by adding a delay to the components and summing the components.

[0043] FIGS. 13 through 15 show how the audio system 100 uses the directivity of the array to separate the directivity components in the left, right, and front / rear directions. FIG. 13 shows the secondary microphone array 128 including the left microphone 132, the right microphone 134, and the center microphone 136.

[0044] FIG. 14 shows an overlaid polar plot of the sound measured by each microphone. The polar plot includes a left polar plot 1402 representing the sound measured by the left microphone 132, a right polar plot 1404 representing the sound measured by the right microphone 134, and a center polar plot 1406 representing the sound measured by the center microphone 136. According to the illustrated embodiment, the left and right microphones 132, 134 are cardioid microphones that attenuate sound arriving from off-axis directions. However, the center microphone 136 is an omnidirectional microphone that measures omnidirectional sound. The center polar plot 1406 is generated by subtracting the audio data measured by the left microphone 132 and the right microphone 134 from the audio data generated by the center microphone 136. The audio system 100 performs this subtraction such that the sum of the combined directivity data from the microphones 132, 134, 136 is zero.

[0045] FIG. 15 shows a three-dimensional (3D) view of the pole plot. The 3D view includes a left cardioidal component 1512 representing the left pole plot 1402, a right cardioidal component 1514 representing the right pole plot 1404, and a central component 1516 representing the central pole plot 1406.

[0046] Referring to FIG. 16, at step 808, the audio system 100 processes the sweep data by simplifying the complex sound field of FIGS. 13 to 15 into its orthogonal components, namely the left sound component 1602, the right sound component 1604, the front sound component 1606, and the rear sound component 1608. Next, the audio system 100 then extrapolates the sound components 1602, 1604, 1606, 1608, adds a delay to the components, and sums the components to predict the response at the new position.

[0047] FIGS. 17 to 18 show a comparison of the performance of the audio system 100 with a primary microphone array and the performance of the audio system 100 with a secondary microphone array when executing the automatic calibration method 800. FIG. 17 is a graph 1700 including four curves 1702, 1704, 1706, and 1708 showing the amplitude response of the audio system 100, and FIG. 17A is an enlarged view of the graph 1700 between -20 and 20 dB and 50 and 150 Hz.

[0048] The first curve 1702 represents the actual sound present at the first listening position 114. The second curve 1704 represents the sound predicted by the audio system 100 at the second listening position 116 based on the sensor data obtained from the primary microphone array including the left microphone 132 and the right microphone 134 as described above with reference to FIG. 9. The third curve 1706 represents the sound predicted by the audio system 100 at the second listening position 116 based on the sensor data obtained from the secondary microphone array including the left microphone 132, the right microphone 134, and the central microphone 136 as described above with reference to FIGS. 10 to 16. The fourth curve 1708 represents the actual sound present at the second listening position.

[0049] The comparison between the second curve 1704 (primary array) and the third curve 1706 (secondary array) and the fourth curve 1708 shows that the performance of the secondary array is improved compared to that of the primary array. For example, at 85 Hz, the secondary curve 1706 is about 2 dB different from the actual acoustic curve 1708, while the primary curve 1704 is about 12 dB different from the actual acoustic curve. Similarly, at 110 Hz, the secondary curve 1706 is about 4 dB different from the actual acoustic curve 1708, while the primary curve 1704 is about 14 dB different from the actual acoustic curve. At both positions, the secondary array provides an improvement of about 10 dB compared to the primary array.

[0050] As indicated by reference numeral 1710 in FIG. 13, the amplitude response decreases at low frequencies, for example, below 25 Hz. This decrease depends on the microphone spacing, because the ability of the microphones to distinguish sounds with large wavelengths depends on there being sufficient spacing between the microphones themselves. The audio system 100 includes a correction of 6 dB per octave for the primary system and a correction of 12 dB for the secondary system to compensate for the decrease.

[0051] FIG. 18 is a graph 1800 including two curves 1802 and 1804 showing the phase response of the audio system 100. The first curve 1802 represents the difference between the actual sound at the second listening position 116 and the sound predicted by the audio system 100 at the second listening position 116 using the primary microphone array. The second curve 1804 represents the difference between the actual sound at the second listening position 116 and the sound predicted by the audio system 100 at the second listening position 116 using the secondary microphone array. The first curve 1802 varies greatly over the frequency range from 0 to 150 Hz. For example, the first curve is equal to about 200 degrees at 85 Hz and equal to about -200 degrees at 110 Hz. On the other hand, the second curve 1804 is approximately equal to zero over the entire frequency range, indicating that the phase response of the secondary system is much better than that of the primary system.

[0052] The automatic calibration method 800 can be extended to enable similar voice prediction in directions other than left / right by using a three - dimensional microphone array (i.e., four microphones) having a 3D arrangement of microphones. The 3D arrangement can predict responses at any position near the listening position, including up and down, to accommodate a room 102 having seats at different vertical positions, such as seats in a stadium. Method 800 is described as a time - domain approach, but similar calculations may be performed in the frequency domain.

[0053] Method 800 makes no assumptions about the acoustic environment based on extensive predetermined data and does not rely on complex room modeling or machine - learning methods. Rather, method 800 utilizes the indoor sound field measured by the microphone array 128. Thus, the audio system 100 does not require large - scale installation, such as many initial measurements, and thereby the user 112 can calibrate the system.

[0054] Although exemplary embodiments have been described above, these embodiments are not intended to describe all possible forms of the present disclosure. Rather, the terms used herein are terms of description rather than limitation, and it will be understood that various changes may be made without departing from the spirit and scope of the present disclosure. Additionally, the various features implementing the embodiments may be combined to form further embodiments.

Claims

1. At least two low-frequency transducers for projecting sound indoors, A portable device, Comprising at least two microphones, a microphone array for receiving the sound generated by each of the at least two low-frequency transducers from a plurality of directions at a first listening position, The portable device comprising a microcontroller programmed to provide a calibration command in response to a user input and to provide a measurement signal indicative of the sound received by the microphone array, A processor, Providing a test signal in response to the reception of the calibration command, wherein each of the at least two low-frequency transducers is adapted to generate a test sound in response to the test signal, Processing the measurement signal to predict an acoustic response at a second listening position adjacent to the first listening position, The processor programmed to adjust acoustic settings associated with each of the at least two low-frequency transducers to optimize the sound at the first listening position and the second listening position, an audio system comprising.

2. The audio system according to claim 1, wherein each of the at least two low-frequency transducers is adapted to generate a test sound of less than 120 Hz in response to the test signal.

3. The at least two microphones, A first microphone disposed on-axis, disposed in a first direction to receive incident sound and attenuating incident sound from off-axis, The audio system according to claim 1, further comprising a second microphone disposed on the axis and disposed in a second direction opposite to the first direction to receive incident sound and attenuating incident sound from off-axis.

4. The audio system according to claim 3, wherein the processor is further programmed to process the measurement signal to predict the acoustic response at the second listening position adjacent to the first listening position by shifting a time delay associated with the sound received by each of the first microphone and the second microphone based on a distance between the first listening position and the second listening position.

5. The audio system according to claim 3, wherein the microphone array further includes a third microphone disposed on the axis between the first microphone and the second microphone for receiving sounds from multiple directions.

6. The microcontroller of the portable device determines the directivity of the combined sound based on the difference between the sound received by the first and second microphones and the sound received by the third microphone, and is further programmed to provide the measurement signal based on the directivity of the combined sound, the audio system according to claim 5.

7. The processor separates the measurement signal into orthogonal components, and is further programmed to extrapolate the orthogonal components to the second listening position, the audio system according to claim 1.

8. The test signal indicates a predetermined sound sweep, the audio system according to claim 1.

9. The processor is further programmed to provide an audio signal indicating a music signal and the adjusted acoustic settings to each of the at least two low-frequency transducers, the audio system according to claim 1.

10. The portable device further includes an externally accessible button, and the microcontroller of the portable device is further programmed to provide the calibration command in response to a user pressing the externally accessible button, the audio system according to claim 1.

11. At least two low-frequency transducers, each of the at least two low-frequency transducers being adapted to project sound into a room in response to reception of an audio signal, and a controller, providing a test signal to each of the at least two low-frequency transducers in response to reception of a calibration command, processing a measurement signal indicating the sound received by at least two microphones at a first listening position in the room to predict an acoustic response at a second listening position adjacent to the first listening position, An audio system comprising the controller, configured to adjust acoustic settings associated with each of the at least two low-frequency transducers to optimize sound at the first listening position and the second listening position.

12. The controller is further configured to separate the measurement signal into orthogonal components and extrapolate the orthogonal components to the second listening position. The audio system according to claim 11.

13. The audio system according to claim 11, wherein the test signal represents a predetermined sound sweep.

14. The controller is further configured to provide an audio signal representing a music signal and the adjusted acoustic settings to each of the at least two low-frequency transducers. The audio system according to claim 11.

15. Further comprising a portable device comprising a microcontroller configured to be coupled to the at least two microphones and provide the measurement signal indicative of the sound received by the at least two microphones, wherein the at least two microphones are a first microphone disposed on-axis and in a first direction to receive incident sound and attenuate incident sound from off-axis; and a second microphone disposed on the axis and in a second direction opposite to the first direction to receive incident sound and attenuate incident sound from off-axis. The audio system according to claim 11.

16. The controller is further configured to process the measurement signal to predict the audio response at the second listening position adjacent to the first listening position by shifting a time delay associated with the sound received by each of the first microphone and the second microphone based on a distance between the first listening position and the second listening position. The audio system according to claim 15.

17. The audio system according to claim 15, further comprising a third microphone disposed on the axis between the first microphone and the second microphone for receiving sound from multiple directions.

18. The microcontroller of the portable device is Based on the difference between the sound received by the first and second microphones and the sound received by the third microphone, determine the directivity of the combined sound, The audio system according to claim 17, further configured to provide the measurement signal based on the directivity of the combined sound. **Claim 19** At least two low-frequency transducers, each of the at least two low-frequency transducers being adapted to project sound into a room in response to receiving an audio signal, the at least two low-frequency transducers, A portable device, At least three microphones adapted to receive sound at a first listening position, A microcontroller configured to provide a calibration command in response to a user input and provide a measurement signal indicative of the sound received by the at least three microphones, the portable device comprising the microcontroller, A controller, In response to receiving the calibration command, provide a first audio signal indicative of a predetermined sound sweep to each of the at least two low-frequency transducers, Process the measurement signal to predict an acoustic response at a second listening position adjacent to the first listening position, Adjust the acoustic settings associated with each of the at least two low-frequency transducers to optimize the sound at the first listening position and the second listening position, Receive a music signal, The controller, configured to provide the second audio signal indicative of the music signal and the adjusted acoustic settings to each of the at least two low-frequency transducers, an audio system comprising the controller. **Claim 20** The at least three microphones, A first microphone disposed on-axis and disposed in a first direction to receive incident sound and attenuate incident sound from off-axis, A second microphone disposed on-axis and disposed in a second direction opposite to the first direction to receive incident sound and attenuate incident sound from off-axis, A third microphone disposed on-axis between the first microphone and the second microphone to receive sound from multiple directions, comprising, The microcontroller of the portable device, Based on the difference between the sound received by the first and second microphones and the sound received by the third microphone, determine the directivity of the combined sound, The audio system according to claim 19, further configured to provide the measurement signal based on the directivity of the combined sound.

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