Low-frequency automatic calibration sound 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 technology]

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

[0003] An acoustic system may generate acoustic signals in a variety of listening environments, such as a home listening room, a home theater, a movie theater, a concert hall, in a car, a recording studio, etc. The listening environments include multiple listening positions for a person to hear the acoustic signals generated by the loudspeakers, for example, a sofa in different sections within a home listening room.

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

[0005] The acoustic signal received at the listening position may be measured. One measure of the acoustic signal is a transfer function, which may 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 may measure various ranges of frequencies. The amplitude of the transfer function is related to the volume. Generally, the amplitude of a single frequency or a frequency range is measured in decibels (dB). The amplitude deviation may be expressed as a positive or negative decibel value relative to a specified target value. If the amplitude deviation is considered at multiple frequencies, the target curve may be flat or of any shape. The relative amplitude response is a measure of the amplitude deviation at one or more frequencies from a 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. A peak represents an increase in the amplitude deviation from the target value, and a dip represents a decrease in the amplitude deviation from the target value.

[0006] These deviations in amplitude response may depend on the frequency of the sound signal reproduced by the subwoofer, the position of the subwoofer, and the position of the listener. The listener may hear low frequencies of a soundtrack, movie, etc., distorted by room boundaries, rather than as they were heard on the recording medium. Thus, the room may alter the sound signal reproduced by the subwoofer, adversely affecting the frequency response performance, including the low frequency performance, of the sound system. Many techniques attempt to reduce or eliminate amplitude deviations at a single listening position. Additional techniques attempt to reduce or eliminate amplitude deviations 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 a sound field measurement approach that includes taking acoustic measurements from each subwoofer position and each listening position. Eliminating amplitude deviations at multiple different listening positions is more difficult and typically relies on using multiple sound sources in different locations in the room. [Prior art documents] [Patent documents]

[0007] [Patent Document 1] U.S. Patent No. 7,526,093 Summary of the Invention

[0008] In one embodiment, the audio system comprises at least two low-frequency transducers for projecting sound into a room and into a portable device. The portable device includes a microphone array comprising 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 the sound received by the microphone array. The 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 an acoustic setting associated with each low-frequency transducer to optimize the sound at the first listening position and the second listening position.

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

[0010] In yet another embodiment, an audio system includes at least two low-frequency transducers, a handheld device, and a controller. Each of the at least two low-frequency transducers is adapted to project sound into a room in response to receiving an audio signal. The handheld device includes at least two microphones for measuring sound from multiple 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. In response to receiving the calibration command, the controller is configured to 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 acoustic settings associated with each of the at least two low-frequency transducers for the first listening position to optimize sound at the first listening position and the second listening position. The controller is further configured to receive the music signal and provide the music signal and a second audio signal indicative of the adjusted sound setting to each of the at least two low-frequency transducers. [Brief description of the drawings]

[0011] [Figure 1] FIG. 1 is a top view of an audio system including a portable measurement device according to one or more embodiments. [Diagram 2] FIG. 2 is a system diagram of the audio system in FIG. 1. [Diagram 3] FIG. 2 is a diagram illustrating three axial modes produced by one loudspeaker of the audio system of FIG. 1, showing the positions of three listeners relative to the loudspeaker. [Figure 4A] 1 is a graph showing the amplitude response of sound produced by one loudspeaker of an audio system and measured at two listening positions in a room, where there is no difference in the amplitude response between the two listening positions. [Figure 4B]1 is a graph showing the amplitude response of equalized sound produced by one loudspeaker of an audio system and measured at two listening positions in a room, where there is no difference in the amplitude response between the two listening positions. [Figure 5A] 1 is a graph showing the amplitude response of sound produced by one loudspeaker of an audio system and measured at two listening positions in a room, with differences in the amplitude response between the two listening positions. [Figure 5B] 1 is a graph showing the amplitude response of equalized sound produced by one loudspeaker of an audio system and measured at two listening positions in a room, with differences in the amplitude response between the two listening positions. [Figure 6] FIG. 2 is a diagram illustrating three axial modes produced by the two loudspeakers of the audio system of FIG. 1, showing the positions of three listeners relative to the loudspeakers. [Figure 7] FIG. 1 illustrates a multi-subwoofer, multi-receiver scenario in a room. [Figure 8] 2 is a flow chart illustrating a method for automatically calibrating the audio system of FIG. 1. [Figure 9] 9 shows the audio system of FIG. 1 including a primary microphone array, which performs part of the method of FIG. 8. [Figure 10] FIG. 1 is a diagram showing sounds arriving at a listening position from all directions. [Figure 11] FIG. 11 shows the complex sound field of FIG. 10 simplified into its orthogonal components. [Figure 12] FIG. 12 illustrates an extrapolation of the sound components of FIG. 11 to predict the response at a new listening position. [Figure 13] FIG. 2 shows a secondary microphone array. [Figure 14] 14 is a graph of a polar plot of sound measured by the secondary microphone array of FIG. 13. [Figure 15] FIG. 15 shows a three-dimensional model of the polar plot of FIG. 14. [Figure 16] FIG. 15 shows the complex sound field of FIG. 14 simplified into its orthogonal components. [Figure 17] 2 is a graph showing the amplitude response of sound produced by the audio system of FIG. 1; [Figure 17A] FIG. 18 is an enlarged view of a portion of the graph in FIG. 17. [Figure 18] 2 is a graph illustrating the phase response of a predicted sound produced by the audio system of FIG. 1. DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS

[0012] Where necessary, detailed embodiments of the present invention are disclosed herein, but it should be understood that the disclosed embodiments are merely exemplary of the present disclosure, which may be embodied in various and alternative forms. The figures are not necessarily to scale, and some features may be exaggerated or minimized to show details of particular components. Thus, specific structural and functional details disclosed herein are not to be construed as limiting, but merely as representative principles.

[0013] 1, an audio system according to one or more embodiments is shown and generally referenced by the numeral 100. The audio system 100 is shown in a home listening environment, such as a room 102. The audio system 100 includes a loudspeaker, such as a soundbar 104, including one or more high-frequency, mid-frequency, and low-frequency transducers (e.g., a subwoofer). The audio system 100 further includes a controller 106 and a portable measurement device 108. The audio system 100 may further include an additional loudspeaker, such as an external subwoofer 110, mounted elsewhere in the room 102. A user 112 is shown holding the portable measurement device 108 at a first listening position 114, e.g., the center seat of a sofa. Next to the user 112 are two additional listeners, one sitting at a second listening position 116 to the left of the user 112 and another sitting at a third listening position 118 to the right of the user 112. In response to a command from the user 112 to activate the “one-click” or portable measurement device 108 and take an acoustic measurement at the first listening position 114, the audio system 100 automatically calibrates the sound projected by the soundbar 104 and external subwoofer 110 to multiple positions in the room 102, e.g., first, second and third listening positions 114, 116, 118.

[0014] 2, the soundbar 104 includes a controller 106 including a processor 120, such as a digital signal processor (DSP), and a memory (not shown). The soundbar 104, according to one or more embodiments, includes a high frequency (HF) transducer 122, a mid frequency transducer 123, and a low frequency transducer or 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 soundbar 104 further includes a transceiver 126, e.g., a low power radio frequency (RF) transceiver, connected to the controller 106 for wireless communication with other devices. The processor 120 receives an audio signal from an audio source 127, such as a television, media player, etc., and separates the audio signal into channels for each soundbar transducer 122, 123, and 124, and any additional transducers (e.g., 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 including two microphones, a left microphone 132 and a right microphone 134. The left and right microphones 132, 134 are packaged relatively close to each other, e.g., about 10 cm apart, and are positioned in opposite directions, e.g., left and right, to provide a directional sensor. Each microphone 132, 134 may be an omnidirectional microphone, such as a Knowles MM20-33366-B116 microphone. In another embodiment, the microphone array 128 is a secondary array including three omnidirectional microphones, a left microphone 132, a right microphone 134, and a central microphone 136 located centrally between the left microphone 132 and the right microphone 134. Other embodiments of the audio system 100 include a microphone array 128 and a dual-band microphone array including a combination of different microphones, such as one or more acoustic cardioid microphones and one or more omnidirectional microphones, with left and right facing lobes, and optionally forward and rear facing lobes. 次 The above array is 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 soundbar 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 functionality of the portable measurement device 108 may be provided by a smartphone or tablet. For example, a smartphone may include a processor, a transceiver, and a touch screen (button), such as 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-power radio frequency (RF) transceiver). The transceiver 148 is connected to the subwoofer controller 146 for wireless communication with the soundbar 104 and other devices, such as the portable measurement device 108. In other embodiments, the external subwoofer 110 communicates with the soundbar 104 via wired communication.

[0018] The controller 106 includes a measurement module 150 for controlling the calibration sequence. According to one or more embodiments, the controller 106 further includes an optimization module 152 for adjusting parameters of each audio channel or transducer, such parameters including individual channel delays, gains, polarities, filters, etc.

[0019] Although the controller 106, the microcontroller 138, and the subwoofer controller 146 are each shown as a single controller, each may include multiple controllers or may be embodied as software code within one or more of the other controllers. The controllers 106, 138, 146 generally include any number of microprocessors, ASICs, ICs, memory (e.g., FLASH, 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 include computer-executable instructions that may be compiled or interpreted from computer programs written using various programming languages ​​and / or technologies. In general, a processor (such as a microprocessor) receives instructions from, for example, a memory, a computer-readable medium, etc., and executes the instructions. A 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, the controller 106, 138, 146 further includes predetermined data stored in 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 a room 102 with a soundbar 104 at one end. The subwoofer 124 of the soundbar 104 produces low-frequency sound, and three of the lowest frequency standing waves are shown as a first mode 320, a second mode 322, and a third mode 324, each mode corresponding to a different frequency, e.g., 30 Hz, 60 Hz, and 90 Hz, respectively, for a set of axial modes. FIG. 3 represents the one-dimensional three-axial modes of the room 102 at a given moment. The maximum sound pressure is at the boundaries of the room (i.e., at both ends of the room 102 in FIG. 2). The points where the sound pressure drops to a minimum are commonly referred to as "nulls." If there is no modal attenuation, the sound pressure at the nulls drops to zero. However, in most real-world rooms the dip in the response at the null is in the range of about -20dB.

[0021] The standing waves have peaks and dips at various locations throughout the room, and may experience large amplitude deviations depending on the listener's location. Thus, since the user 112 is at a null for both the first mode 320 and the third mode 324, the sound generated by the subwoofer 124 at these frequencies will sound much quieter than it should. Conversely, since the user 112 is at the peak of the second mode 322, the sound generated by the subwoofer 124 at this frequency will sound much louder than it should. The listeners at the second listening position 116 and the third listening position 118 are not located at a null for any of the modes, and therefore hear all three modes, resulting in a more comfortable and accurate listening experience.

[0022] 4A-4B, one approach to address the standing wave problem in the single subwoofer scenario of FIG. 3 is to equalize the frequency response. FIG. 4A is a graph 400 with three curves 404, 406, 408 representing the frequency response of acoustic measurements produced 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, the first curve 404 and the second curve 406 rise and fall simultaneously at different frequencies, so that there is little or no variation between listening positions or between 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] 4B is a graph 410 including a first curve 414 representing an equalized frequency response of sound measured at a first listening position, a second curve 416 representing an equalized frequency response of sound measured at a 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 approximately parallel to each other, indicating that in the absence of variation between the listening positions (as shown in FIG. 4A), the frequency response of both listening positions may be improved by equalizing the acoustic signal provided to the subwoofer 124.

[0024] Referring to Figures 5A and 5B, the simple equalization approach of Figures 4A and 4B is not effective when there are differences from seat to seat. Figure 5A is 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, the subwoofer 124 in the room 102 of Figure 3, according to another embodiment. 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 Figure 4A. As shown in FIG. 5A, the first curve 504 and the second curve 506 do not rise and fall simultaneously across the frequency range, so differences exist between listening positions.

[0025] 5B is a graph 510 including a first curve 514 representing an equalized frequency response of the sound measured at the first listening position 114, a second curve 516 representing an 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. While the spatial average curves 408, 508 are approximately equal to each other, 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 differences in frequency response between the listening positions mean that correcting the sound at one position with simple equalization may adversely affect 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 in the room 102, as subwoofers at different positions can partially cancel certain standing waves. Fig. 6 shows a room 102 in which both the subwoofer 124 of the soundbar 104 and the external subwoofer 110 generate low frequency modes from different positions, which cancels two of the three modes, the first mode 620 and the third mode 624, but does not cancel the second mode 622 at the first listening position 114. However, this approach requires an additional loudspeaker, e.g., the external subwoofer 110, and there are still nulls in the room 102 adjacent to the second and third listening positions 116, 118.

[0027] 7 shows an example of a multi-subwoofer, multi-receiver scenario in a room. Reference I is an audio signal input to the audio system 100. The loudspeaker / room transfer function from the subwoofer 124 (Speaker 1) of the soundbar 104 and the external subwoofer 110 (Speaker 2) to two receiving positions (e.g., a first listening position 114 and a second listening position 116) in the room 102 is given by H 11 , H 12 , H 21 , and H 22 It is represented by R 1 and R 2 represents the transfer function obtained at the receiving (listening) position. Each sound source has a transmission path to each receiver, resulting in four transfer functions in this example. The signal sent to each loudspeaker is 1 and M 2 The modified signals can be added assuming that the inputs are electrically modifiable as represented by: where M is a complex modification factor that may or may not be frequency dependent. To illustrate the complexity of the mathematical solution, the following equation solves a linear time-invariant system in the frequency domain:

[0028] R 1 (f)=IH 11 (f)M 1(f) + IH 21 (f)M 2 (f) R 2 (f)=IH 12 (f)M 1 (f) + IH 22 (f)M 2 (f) (1) Here, all the transfer functions and correction factors are understood to be complex. This can be viewed as a set of simultaneous linear equations, which can be expressed more compactly in matrix form as follows:

[0029]

number

[0030] A common optimization goal is to have R equal to 1, i.e., the signals at all receivers are identical to each other. R is a function of R 1 and R 2 may be considered as an objective function where both are equal to 1. Solving equation (3) for M, the correction factor of the audio system, gives M = H -1 , the inverse of H. Since H is frequency dependent, a solution for M is computed at each frequency. However, there may be values ​​of H that are difficult to inverse or impractical to implement (e.g., the gain of some loudspeakers at some frequencies is impractically high).

[0031] Since it is not always feasible to determine an exact mathematical solution, traditional approaches have attempted to determine the best computable solution, such as the solution with the smallest error. An error function defines how close a particular configuration is to the desired solution, with the lowest error representing the best solution. However, this mathematical methodology requires a large amount of computational energy, yet can only solve two-parameter solutions. Acoustic problems that examine more parameters are becoming increasingly difficult to solve. Some audio systems attempt to solve the problem by analyzing acoustic measurements taken at different locations in the listening room, but such an approach can be difficult for end users in a home listening environment.

[0032] 8 and referring back to FIG. 2, a method for automatically calibrating audio system 100 is shown in accordance with one or more embodiments and generally designated 800. Method 800 is implemented using software code contained within controller 106 in accordance with one or more embodiments. The method is described using a flow chart shown with several sequential steps, 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 across multiple controllers, for example controller 106 and microcontroller 138.

[0033] At step 802, the user 112 initializes the calibration sequence by pressing the button 142 on the portable measurement device 108 while seated at the first listening position 114. In other embodiments, the calibration procedure may be initiated in response to a voice 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 soundbar 104 via the transceiver 140.

[0034] At step 804, the controller 106 receives an 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 radiate as sound. In one embodiment, the sound sweep corresponds to a sound varying in amplitude from -60 to 60 dB and in frequency from 0 to 150 Hz. At step 806, the microphone array 128 of the portable measurement device 108 measures the sound sweep at the first listening position 114 and transmits the sweep data (MIC) to the soundbar 104.

[0035] In step 808, the controller 106 processes the sweep data to predict responses at other listening positions, e.g., 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., incorporated herein by reference in its entirety, to further process the data. In one or more embodiments, the controller 106 may perform multiple sweeps and repeat steps 804 through 808, or use other techniques or algorithms to increase the signal-to-noise ratio, such as sampling background noise and adjusting the excitation to give more energy to noisy frequencies. Then, in step 810, the controller 106 adjusts the acoustic settings, e.g., parameters for each individual channel, including time delay, gain, polarity, and filter coefficients, based on the predicted responses.

[0036] FIG. 9 illustrates one embodiment of an audio system 100 including a primary microphone array performing an automatic calibration method 800. Referring to FIG. 9 and referring back to FIG. 1, 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 surfaces in the room 102 and resembles the sound provided by a number of virtual sound sources located at corresponding positions outside the room. The acoustic response at the first listening position 114 in the room 102 is similar to that which would occur if such virtual sound sources were absent of the room and clouds. As the user 112 moves from the first listening position 114 to the second listening position 116, the user 112 moves approximately one meter closer to the virtual image immediately to the left, i.e., one meter the distance between the centers of adjacent cushions on a sofa, and one meter away from the virtual image immediately to the right. For virtual sound sources directly in front of or behind the user, there is minimal or no difference in distance. For virtual sound sources in other directions, there is an intermediate difference in the distance to the virtual sound source.

[0037] FIG. 9 illustrates how the left and right arriving sounds are measured in step 806 using directional microphones 132, 134, processed in step 808 by shifting the impulse responses based on the estimated distance between the listening positions, and then recombined. In step 806, the portable measurement device 108 measures the sound sweep using the primary microphone array 128. The microphone array 128 is configured as a directional microphone with left and right microphones 132, 134 closely spaced, for example about 10 cm apart, in opposite directions along the axis AA. FIG. 9 includes a left polar plot 902 representing the sound measured by the left microphone 132 and a right polar 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, which attenuate sounds coming from off-axis directions.

[0038] At step 808, the controller 106 of the soundbar 104 processes the sound sweep data. The processor 120 includes the correct signal delay and gain components for each microphone 132, 134. The processor 120 decomposes the sound received at each microphone 132, 134 of the microphone array 128 into left and right arriving components, as shown by the left reflectogram 908 and the right reflectogram 910. The sound received directly from the soundbar 104 is received by the front and rear lobes (not shown) of the microphone array 128 and is not time shifted.

[0039] The measurement module 150 calculates the sound (Δt L ) and the sound measured by the right microphone 134 (Δt R ), the acoustic settings can be adjusted in step 810 to predict the sounds present at different listening positions, e.g., the second listening position 116 and the third listening position 118.

[0040] Δt L =+ / -d / c (4) Δt R =- / +d / c (5) Here, (d) represents the distance between the listening positions, e.g., 1 meter, (c) represents the speed of sound, (-) is used to predict the sound at a position in the same direction as the microphone (e.g., 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 (e.g., a position to the right of the left microphone 132). For example, the audio system 100 predicts the sound at the second listening position 116, which is facing to the left of the first listening position 114, by subtracting d / c from each impulse measured by the left microphone 132, as referenced at 916, and adding d / c to each impulse measured by the right microphone 134, as referenced at 918. The audio system 100 then recombines the shifted signals, represented by a simplified reflectogram, as referenced generally at 920.

[0041] Figures 10 to 16 show a portion of an automatic calibration method 800 performed by one embodiment of an audio system 100 that includes a secondary microphone array. The microphone array 128 is a secondary array that includes 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 an audio system as described with reference to Figure 8 by decomposing a complex sound field and then extrapolating the sound to predict the response at new positions.

[0042] With reference to Figure 10, at any point in space, for example the first listening position 114, sounds come from all directions as indicated by the converging arrows. With reference to Figure 11, the audio system 100 utilizes a secondary microphone array 128 to simplify the complex sound field of Figure 10 into its orthogonal components: left sound component 1102, right sound component 1104, front sound component 1106, and rear sound component 1108. Now, with reference to Figure 12, the audio system 100 then extrapolates the sound by adding delays to the components and summing them to predict the response at the new position.

[0043] 13 to 15 show how the audio system 100 uses the array's directivity to separate left, right, and front / rear directional components. Figure 13 shows a secondary microphone array 128 that includes a left microphone 132, a right microphone 134, and a center microphone 136.

[0044] 14 shows overlaid polar plots of the sound measured by each microphone. The polar plots include 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 central 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 sounds coming from off-axis directions. However, the center microphone 136 is an omnidirectional microphone that measures sounds in all directions. The central polar plot 1406 is generated by subtracting the sound data measured by the left microphone 132 and the right microphone 134 from the sound data generated by the center microphone 136. The audio system 100 performs this subtraction such that the sum of the combined directional data from the microphones 132, 134, 136 is zero.

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

[0046] 16, the audio system 100 processes the sweep data in step 808 by simplifying the complex sound field of Figures 13 to 15 into its orthogonal components: left sound component 1602, right sound component 1604, front sound component 1606, and rear sound component 1608. The audio system 100 then predicts the response at the new position by extrapolating the sound components 1602, 1604, 1606, 1608, adding delays to the components, and summing the components.

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

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

[0049] A comparison of the second curve 1704 (primary array) and the third curve 1706 (secondary array) with the fourth curve 1708 shows the improvement in performance of the secondary array over that of the primary array. For example, at 85 Hz, the secondary curve 1706 differs from the actual acoustic curve 1708 by about 2 dB, whereas the primary curve 1704 differs from the actual acoustic curve by about 12 dB. Similarly, at 110 Hz, the secondary curve 1706 differs from the actual acoustic curve 1708 by about 4 dB, whereas the primary curve 1704 differs from the actual acoustic curve by about 14 dB. At both locations, the secondary array provides an improvement of about 10 dB over the primary array.

[0050] The amplitude response falls off at low frequencies, e.g., below 25 Hz, as shown at 1710 in Figure 13. This falloff depends on the microphone spacing, since the ability of the microphones to distinguish between sounds with large wavelengths depends on the microphones themselves being spaced sufficiently apart. To compensate for the falloff, audio system 100 includes a 6 dB per octave correction for first-order systems and a 12 dB correction for second-order systems.

[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 at the second listening position 116 by the audio system 100 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 at the second listening position 116 by the audio system 100 using the secondary microphone array. The first curve 1802 varies significantly over the frequency range from 0 to 150 Hz. For example, the first curve is equal to about 200 degrees at 85 Hz and is equal to about -200 degrees at 110 Hz. On the other hand, the second curve 1804 is almost equal to zero over the entire frequency range, which indicates that the phase response of the secondary system is much better than that of the primary system.

[0052] The auto-calibration method 800 can be extended to allow similar sound predictions in directions other than left / right by using a tertiary microphone array (i.e., four microphones) with a 3D placement of microphones. The 3D placement may predict the response at any position near the listening position, including above and below, to accommodate rooms 102 with seats in different vertical positions, such as stadium seating. Although the method 800 is described as a time domain approach, similar calculations may be performed in the frequency domain.

[0053] Method 800 does not make any assumptions about the acoustic environment based on extensive pre-defined data, nor does it rely on complex room modeling or machine learning methods or the like. Rather, method 800 utilizes the sound field in the room as measured by microphone array 128. Thus, audio system 100 does not require extensive installation, e.g., many initial measurements, allowing user 112 to calibrate the system.

[0054] Although exemplary embodiments have been described above, it is not intended that these embodiments 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. In addition, various features of the implementations of 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, and 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, 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, the portable device, 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 audio 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 attenuate 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 attenuate 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 voice 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 comprises 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 a directivity of a combined sound based on a 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 audio system according to claim 1, wherein the processor is further programmed to provide an audio signal indicating a music signal and the adjusted acoustic setting to each of the at least two low-frequency transducers.

10. The portable device further comprises a button accessible from the outside, and the microcontroller of the portable device is further programmed to provide the calibration command in response to a user pressing the button accessible from the outside. 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 receiving an audio signal, the at least two low-frequency transducers, A controller, In response to receiving a calibration command, provide a test signal to each of the at least two low-frequency transducers, Process a measurement signal indicative of the sound received by at least two microphones at a first listening position in the room to predict an audio response at a second listening position adjacent to the first listening position, An audio system comprising the controller, the controller being 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, Separate the measurement signal into orthogonal components, The audio system according to claim 11, further configured to extrapolate the orthogonal components to the second listening position.

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

14. The controller is further configured to provide an audio signal indicative of 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 a first microphone disposed on the axis, disposed in a first direction to receive incident sound and attenuate incident sound from off-axis; and a second microphone disposed on the axis, disposed 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 determines the directivity of the combined sound based on a difference between the sound received by the first and second microphones and the sound received by the third microphone, and is further configured to provide the measurement signal based on the directivity of the combined sound, the audio system according to claim 17.

19. At least two low-frequency transducers, each of the at least two low-frequency transducers being adapted to project sound indoors in response to reception of 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 to 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 reception of 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 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, Receive a music signal, The controller configured to provide a 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.

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 the 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 the axis between the first microphone and the second microphone to receive sound from multiple directions, the portable device comprising the third microphone; 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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