System, device and method for multi-dimensional adaptive microphone-loudspeaker array set for room correction and equalization

Abstract

In at least one embodiment, an audio system is provided. The audio system includes: a plurality of speakers; a plurality of microphones; and an audio controller. The plurality of speakers transmit audio signals in a listening environment. The plurality of microphones detect the audio signals in the listening environment. The at least one audio controller is configured to determine a first psychoacoustic perceived loudness (PPL) of the audio signal when the audio signal is played back through a first speaker among the plurality of speakers, and to determine a second PPL of the audio signal when the audio signal is sensed by a first microphone among the plurality of microphones. The at least one audio controller is further configured to map the first speaker among the plurality of speakers to the first microphone among the plurality of microphones based at least on the first PPL and the second PPL.

Description

Technical Field

[0001] Aspects disclosed herein may generally relate to systems, devices, and methods for multidimensional adaptive microphone-loudspeaker array sets for room correction and room equalization. In one aspect, the disclosed systems, devices, and / or methods may map a listening room to a microphone and loudspeaker array set according to criteria based on human sound and psychoacoustic perception. These and other aspects will be discussed in more detail below. Background Art

[0002] When the sound is reproduced by one or more loudspeakers, the perception of the desired auditory illusion is modified by the listening environment. The sound reproduction system may also introduce undesirable artificial phenomena. Room response equalization (RRE) aims to improve the sound reproduction in the room by designing an equalizer based on one or more measurements of the room response using advanced digital signal processing techniques. Various established techniques can be used to solve the RRE problem, including homomorphic filtering, linear predictive coding (LPC), least squares optimization, frequency domain deconvolution, and multiple input / multiple output inverse theorem (MINT) solutions. Various preprocessing methods are usually used to improve RRE techniques, such as non-uniform frequency resolution, complex smoothing, frequency bending, Kautz filters, and multi-rate filters. Current room correction and equalization techniques can be classified as single-position or multi-position monitoring with fixed or adaptive room equalizers. As more functions are supported, their complexity may increase exponentially. Therefore, such systems may become an obstacle to successful implementation on real-time processors. In addition, these solutions may not utilize psychoacoustics, which involves studying sound perception and audiology based on the way humans perceive various sounds.

[0003] Current speaker-microphone array set mapping techniques rely on proximity analysis to determine the influential speakers at a given listening position. In other words, speakers are mapped to the listening area based on their physical distance from the microphones in the listening area. These techniques are inefficient in small enclosed spaces (such as car cabins) where a large number of speakers may become equally close to more than one listening position, thereby increasing computational complexity and reducing the benefits of room equalization. Furthermore, proximity analysis may exclude influential speakers that are beyond a given distance from the listening position. Summary of the invention

[0004] In at least one embodiment, an audio system is provided. The audio system includes: a plurality of speakers; a plurality of microphones; and an audio controller. The plurality of speakers transmit audio signals in a listening environment. The plurality of microphones detect the audio signals in the listening environment. The at least one audio controller is configured to determine a first psychoacoustic perceived loudness (PPL) of the audio signal when the audio signal is played back through a first speaker among the plurality of speakers, and to determine a second PPL of the audio signal when the audio signal is sensed by a first microphone among the plurality of microphones. The at least one audio controller is further configured to map the first speaker among the plurality of speakers to the first microphone among the plurality of microphones based at least on the first PPL and the second PPL.

[0005] In at least one embodiment, an audio system is provided. The audio system includes: a plurality of speakers; a plurality of microphones; and at least one audio controller. The plurality of speakers are configured to transmit audio signals in a listening environment. Each of the microphones is positioned at a corresponding listening position in the listening environment. The plurality of microphones are configured to detect the audio signals in the listening environment. The at least one audio controller is configured to determine a first psychoacoustic perceived loudness (PPL) of each of the plurality of speakers and determine a second PPL of each of the plurality of microphones to equalize the audio signals in the listening environment using an adaptive process.

[0006] In at least one embodiment, a method for equalizing an audio signal in a listening environment using an adaptive process is provided. The method includes: transmitting an audio signal in a listening environment through a plurality of speakers; and detecting the audio signal in the listening environment through a plurality of microphones positioned in the listening environment. The method includes: determining a first psychoacoustic perceived loudness (PPL) of each of the plurality of speakers; and determining a second PPL of each of the plurality of microphones to equalize the audio signal in the listening environment using an adaptive process based on the first PPL and the second PPL. BRIEF DESCRIPTION OF THE DRAWINGS

[0007] Embodiments of the present disclosure are particularly pointed out in the appended claims. However, other features of the various embodiments will become more apparent and are best understood by referring to the following detailed description in conjunction with the accompanying drawings, in which:

[0008] Figure 1 A system for providing audio to a two-dimensional arbitrary microphone and speaker room array is shown;

[0009] Figure 2A system for providing audio to a two-dimensional microphone and speaker room array according to one embodiment is shown;

[0010] Figure 3 A method for performing calibration to map one or more speakers to one or more microphones according to one embodiment is shown;

[0011] Figure 4 According to one embodiment, a method for Figure 3 A system for assigning loudspeakers to microphones during a calibration method;

[0012] Figure 5 A system for performing an adaptive runtime process for room correction and equalization according to one embodiment is shown; and

[0013] Figure 6 A method for providing Figure 5 A method for performing an adaptive run-time process for a room correction and equalization system. DETAILED DESCRIPTION

[0014] As required, detailed embodiments of the present invention are disclosed herein; however, it should be understood that the disclosed embodiments are merely examples of the present invention that may be embodied in various and alternative forms. The drawings are not necessarily drawn to scale; some features may be exaggerated or minimized to show details of particular components. Therefore, the specific structural and functional details disclosed herein should not be interpreted as limiting, but merely as a representative basis for teaching those skilled in the art to apply the present invention in different ways.

[0015] It should be appreciated that the controllers disclosed herein may include various microprocessors, microcontrollers, digital signal processors (DSPs), integrated circuits, memory devices (e.g., flash memory, random access memory (RAM), read-only memory (ROM), electrically programmable read-only memory (EPROM), electrically erasable programmable read-only memory (EEPROM), or other suitable variations thereof) and software that cooperate with each other to perform one or more operations disclosed herein. In addition, such controllers disclosed utilize one or more microprocessors to execute a computer program embodied in a non-transitory computer-readable medium, the computer program being programmed to perform any number of the disclosed functions. In addition, one or more controllers as provided herein include a housing and various numbers of microprocessors, integrated circuits, and memory devices (e.g., flash memory, random access memory (RAM), read-only memory (ROM), electrically programmable read-only memory (EPROM), electrically erasable programmable read-only memory (EEPROM)) positioned within the housing. The one or more controllers disclosed also include hardware-based inputs and outputs for receiving data from and transmitting data to other hardware-based devices discussed herein, respectively.

[0016] To obtain a successful immersive and high-fidelity listening experience in an enclosed space, such as, for example, a vehicle cabin, room equalization or correction may be necessary. The process of room equalization (RE) involves, among other things, compensating for unwanted room sound artifacts, such as early reflections, reverberant reflections, surrounding material properties, and speaker imperfections. Again, RE can be performed in a fixed manner or in an adaptive manner. In fixed RE, calibration is initially performed, and filter coefficients are calculated and used after calibration with little or no updating. In adaptive RE, calibration is initially performed to determine some initial conditions, after which runtime adaptation is performed to update the filter coefficients to track changing room conditions in real time.

[0017] Most room enclosures are considered to be weakly stationary environments. For example, room conditions vary as a function of room geometry (i.e., furniture, fixtures, luggage, etc.), room capacity (i.e., number of people, pets, etc.), and room environment (i.e., temperature, humidity, etc.). Therefore, it may be necessary to adapt the RE filter coefficients to these changing conditions to improve performance.

[0018] Aspects disclosed herein map a listening room to a set of microphone and speaker arrays based on human perception of sound (e.g., psychoacoustics). After this initial calibration phase, a runtime process continuously updates the equalization filter coefficients to adapt to the changing conditions of the room. The disclosed techniques relate to the nature of human perception and are flexible enough to implement these operating modes depending on the application. In particular, the disclosed techniques can map any number of speakers to any number of listening positions. In addition, the equalization filter coefficients can be fixed when performing the calibration process, and the filter coefficients can be adaptive when performing the runtime process in addition to calibration.

[0019] Figure 1 An exemplary audio system 100 is shown. System 100 includes a speaker array 102a-102g (eg, "102") located in a listening environment 104. Listening environment 104 may correspond to, for example, a vehicle cabin, a living room, a concert hall, etc. Figure 1 When the speakers 102 are depicted surrounding the microphone array, such microphones correspond to simulated listening positions of users 106a-106p (e.g., "106") in the listening environment 104. An audio controller 108 is operably coupled to the speaker array 102 for providing audio input into the listening environment 104 through the speakers 102. It should be appreciated that the positions of the speakers 102 and the listening positions of the users 106 can be fixed or variable. The speakers 102 and the listening positions of the users 106 generally form a two-dimensional array. It may be desirable to map corresponding speakers 102 to one or more listening positions 106 to enable the users to experience the best sound perception of the audio.

[0020] Figure 2An audio system 200 according to one embodiment is shown. The audio system 200 includes a speaker array 202a-202f ("202") and a microphone array 204a-204c ("204") positioned in a listening environment 205. Each of the microphones 204a-204d is positioned at a corresponding listening position of a user 206a-206d, respectively, in the listening environment. At least one audio controller 208 (hereinafter referred to as "audio controller") is operably coupled to the speaker array 202 for providing audio input into the listening environment 205 through the speaker 202. The audio input includes a signal having an acoustic frequency in the audible and / or ultrasonic range. For example, the audio input may include a test signal or an audio recording such as a sine wave, a chirp wave, a Gaussian noise, a pink noise, etc. It should be appreciated that the position of the speaker 202 and the listening position of the user 206 may be fixed or variable. It is desirable to map the corresponding speaker 202 to one or more of the listening positions 206. The microphones 204 are shown and provided in the listening environment 205 to enable the audio controller 208 to perform a calibration to map each speaker 202 to one or more microphones 204 (ie, or one or more listening positions 206 ).

[0021] As noted above, it is desirable to map corresponding speakers 202 to one or more of the listening positions 206 so that the user can experience an optimal listening experience. Mapping a particular speaker 202 to one or more of the listening positions 206 to achieve optimal audio playback (i.e., audio perception) can be based on, for example, psychoacoustic perceived loudness (PPL) and the distance of the speaker 202 to the listening position 206. PPL is a measure of perceptually relevant information contained in any audio recording. PPL represents the theoretical limit of how much sound energy the human ear can perceive at different time intervals or frames. PPL is defined as follows:

[0022]

[0023] Wherein E(k) is the energy in the kth psychoacoustic critical band and is a complex value, T(k) is the masking threshold in the kth psychoacoustic critical band and is a real value, and CB is the number of psychoacoustic critical bands. The masking threshold of critical band k provides a power level below which any sound energy is inaudible to the listener. Sound energy in critical band k above the masking threshold is audible to the listener. The calculation of E(k) and T(k) follows the technology developed in the field of digital audio perceptual coding. For example, the audio signal is first windowed and transformed to the frequency domain. A mapping from the frequency domain to the psychoacoustic critical band domain is performed. The masking threshold is then obtained using perceptual rules. The frequency domain transform is performed by first multiplying a segment of audio input or frame (defined on a time interval) with a window function (e.g., Hamming, Hann, Blackman, etc.), followed by a time-to-frequency transform (such as FFT, DFT, DCT, wavelet, etc.). The frequency domain signal is then multiplied by a linear or nonlinear mapping matrix from the frequency domain to the psychoacoustic critical band domain. The psychoacoustic critical band domain includes perceptual scales such as the equivalent rectangular bandwidth (ERB) scale, the Bark scale, or the Mel scale. The masking threshold T(k) can be estimated by first calculating the power in each critical band, i.e., P(k) = Real(E(k)) 2 +Imaginary((E(k)) 2 , applying a spreading function (SF), and then calculating various psychoacoustic metrics such as spectral flatness measure (SFM), tonality coefficients, and masking shift.

[0024] The ideal PPL (PPL_I) (or first psychoacoustic perceived loudness) is typically calculated from the audio input (and at the speakers 202a-202c) at certain time intervals or over the entire audio sequence. The measured PPL (PPL_M) (or second psychoacoustic perceived loudness) is calculated at similar time intervals at the microphone input. The PPL loss (PPL_L) is the difference between PPL_I and PPL_M and measures the amount of acoustic energy deviation from the ideal value due to room conditions and speaker imperfections. PPL_L is calculated as a complex value and therefore contains information about level deviation (amplitude) and arrival time deviation (phase),

[0025]

[0026] It is reasonable to assume that the measured masked threshold TM(k) should be equal to the ideal masked threshold TI(k), so the PPL_L in the above equation is approximately:

[0027]

[0028] This approximation avoids computationally intensive masking threshold calculations while still providing accurate results for the magnitude and phase of acoustic deviations from the ideal.

[0029] Therefore, in order to account for the PPL of the speaker 202 to the listening position 206 (e.g., microphone 204), the audio controller 208 performs the following calibration process. For each microphone 204 or "m", the audio controller 208 measures the PPL_I from each speaker 202 while playing back the calibration audio signal in the listening environment. This produces a (m*s)PPL_M quantity at the microphone, which measures the impact of room conditions and speaker design on each microphone 204 according to the following equation:

[0030] Targeting |E I (k)|>T I (k)

[0031] Targeting |E M (k)|>T M (k)

[0032] |PPL L |=|PPL I |-|PPL M |

[0033] where || is a complex amplitude operator and only includes energy above the hearing threshold of its corresponding critical band.

[0034] For Figure 2For each microphone 204 positioned in the array as shown, the audio controller 208 uses PPL to determine the loudest speaker group within the speaker array 202. For example, |PPL_I| of an input audio waveform is calculated by the audio controller 208 over the entire track length of the audio or at some time intervals (e.g., exemplary every 10 milliseconds) as the audio is played sequentially through the speakers 202a, 202b, 202c, 202d, 202e, and 202f. At the same time, |PPL_M| is measured at microphone 204a at similar time intervals as the audio is being played through each speaker 202. For each speaker 202, |PPL_L| is calculated as the difference between |PPL_I| and |PPL_M|, which determines the perceived audible deviation at the listening position 206a. The magnitude of PPL_L determines the deviation from the ideal perceived audio loudness level at the listening position 206a. A programmable threshold level of perceived loudness amplitude loss is used to distinguish between an affected speaker 202 and an unaffected speaker at a listening position 204a. The audio controller 208 may assign any given microphone 204 to one or more speakers 202. For example, the audio controller 208 may assign speakers 202a and 202b to the microphone 204a based on psychoacoustic perceived loudness. The audio controller 208 may assign speakers 202b and 202c to the microphone 204b based on loudness (e.g., based on PPL and PPL loss).

[0035] Figure 3 A method 300 is shown for performing calibration to map one or more speakers 202 (e.g., speaker array 202) to one or more microphones 204 (e.g., microphone array 204) according to one embodiment. In operation 302, the audio controller 208 cycles through the number of microphones 204 positioned within the listening environment 205. In this operation, the audio controller 208 stores data corresponding to the total number of microphones 204 positioned in the listening environment 205.

[0036] In operation 304, the audio controller 208 loops over the number of speakers 202 positioned within the listening environment 205. In this operation, the audio controller 208 stores data corresponding to the total number of speakers 202 positioned in the listening environment 205.

[0037] In operation 306 , the audio controller 208 iteratively calculates |PPL_I|, |PPL_M|, and |PPL_L| losses for each microphone and speaker set.

[0038] In operation 308, the audio controller 208 compares |PPL_L| to a programmable threshold level of perceived loudness amplitude loss, which is used to distinguish between the affected speakers 202 and the unaffected speakers at the listening position 204a. As noted above, |PPL_L| is calculated as the difference between |PPL_I| and |PPL_M|, which determines the perceived audible deviation at the listening position 206a. If |PPL_L| is less than the programmable threshold level, the method 300 moves to operation 310. If not, the method 300 moves to operation 312.

[0039] At operation 310, the audio controller 208 determines whether all speakers 202 in the array have generated |PPL_I|, |PPL_M|, and |PPL_L| losses (e.g., operations 306 and 308 are performed for each speaker 202 in the array). If this condition is met, the method 300 moves to operation 302. If not, the method returns to operation 304.

[0040] In operation 312, the audio controller 208 assigns the corresponding speakers 202 to one or more microphones 204. In operation 314, the audio controller 208 stores the RE calibration fixed coefficients, which are determined from PPL_L (i.e., psychoacoustic perceived loudness loss). Once the speaker-microphone array set mapping is completed and the fixed calibration coefficients are calculated and stored, RE is then performed by applying these coefficients to the input of the speaker, such as Figure 4 shown.

[0041] Figure 4 A reference is shown according to one embodiment. Figure 3 The operation 312 of the method 300 of the present invention allocates the speaker 202 to the system 400 of the microphone 204. The system 400 includes an audio controller 208 and an array 460 having one or more speakers 202 and one or more microphones 204. The one or more microphones 204 can be positioned close to the corresponding listening positions of the users 206a-206b. The audio controller 208 includes a memory 209 for storing the RE fixed coefficients derived from the method 300 during calibration. The one or more microphones 204 can be positioned close to the corresponding listening positions of the users 206. The audio controller 208 includes a first plurality of filter banks 450a-450b, a matrix mixer 451, a plurality of multiplier circuits 452a-452c, a second plurality of filter banks 454a-454b and a function block 472.

[0042] With regard to assigning the loudspeaker 202 to the microphone 204, the audio controller 208 may assign the loudspeakers 202a, 202b to the microphone 204a at the listening position 206. The audio controller 208 may assign the loudspeakers 202b, 202c to the microphone 204b at the listening position 206b. The first plurality of filter banks 450a-450b may be implemented as analysis filter banks and are configured to transform the stereo audio input (e.g., input R and L) into a psychoacoustic critical band domain. The matrix mixer 451 generates three channels from a stereo 2-channel audio input. Figure 3 The calibration method generates 4 sets of fixed calibration coefficients (W(M1, S1), W(M1, S2), W(M2, S2) and W(M2, S3)) to perform RE in the listening environment 205. Function block 472 receives the calibration coefficients W(M1, S2) and W(M2, S3) and combines (or merges) them to generate a single output fed to the multiplier circuit 472. Function block 472 combines the calibration coefficients W(M1, S2) and W(M2, S3) to combine the responses from the microphones 204a, 204b, such as but not limited to maximum, minimum, average, smoothing. The second plurality of filter banks (or synthesis filter banks) 454a-454c are configured to filter the outputs (e.g., compensated signals) from the multiplier circuits 452a-452c, respectively. The compensated signals are transformed back into the time domain using synthesis filter banks 454a-454c before being sent to speakers 202a-202c.

[0043] Return to reference Figure 3 In operation 314, the audio controller 208 stores the assignment of one or more speakers 202 to each microphone 204 in its memory 209. Figure 2 As noted, each of the microphones 204a - 204d is positioned at one or more corresponding listening positions of the users 206a - 206d , respectively, in the listening environment 205 .

[0044] Figure 5 A system 500 for performing an adaptive runtime process for room correction and equalization to occur in real time is shown according to one embodiment. Figure 3The disclosed calibration process is static in mapping one or more speakers 202 to one or more microphones 204 (or listening positions 206) to enable the user to experience the best sound perception of the audio. It should be recognized that numerous room conditions change dynamically as a function of room geometry (i.e., furniture, fixtures, luggage, etc.), room capacity (i.e., number of people, number of pets, etc.), and room environment (i.e., temperature, humidity, etc.) to affect the listening experience of the user in the listening environment 205. The system 500 is generally configured to employ a continuous runtime algorithm to account for these changing room conditions. This may be advantageous for, but not limited to, listening environments within a vehicle.

[0045] System 500 generally includes Figure 4 The various features set forth in the system 400 of the present invention (e.g., the audio controller 208, the first plurality of filter banks 450a-450b, the matrix mixer 451, the plurality of multiplier circuits 452a-452c, the second plurality of filter banks 454a-454b, and the array 460. The system 500 also includes a plurality of first delay blocks 453a-453c, a plurality of first delay blocks 456a-456c, a first plurality of psychoacoustic building blocks 458a-458c, a first plurality of difference blocks 459a-459c, a third plurality of filter banks 461a-461b, a second plurality of psychoacoustic building blocks 462a-462b, a plurality of comparators 470a-470d, and a function block 482.

[0046] The adaptive process performed by the system 500 may begin with each of the microphones 204a-204b providing an audio input signal to the plurality of filter banks 450a-450b, respectively. In this case, the microphones 204a-204b generate outputs indicative of the audio being played in the listening environment 205 through playback from the speakers 202a-202c. For example, and as Figure 4 As indicated, loudspeakers 202a and 202b may be assigned to microphone 204a (or listening position 206a), and loudspeakers 202b and 202c may be assigned to microphone 204b (or listening position 206b). As indicated above, a plurality of filter banks (or analysis filter banks) 450a-450b transform stereo audio input into audio in a psychoacoustic critical band domain. Matrix mixer 451 generates three channels from a stereo 2-channel audio input. A plurality of first delay blocks 453a-453c delay the output from matrix mixer 451.

[0047] The compensation circuit 452a-452c is generally configured to compensate for the amplitude or phase (or both amplitude and phase) of the received audio input. The second plurality of filter banks 454a-454c are configured to filter the output from the compensation circuit 452a-452c respectively. The plurality of filter banks 454a-454c are configured to transfer the compensated signal from the compensation circuit 452a-452c to the time domain before the audio is transmitted to the loudspeaker 202a-202c. The loudspeaker 202a-202c replays the audio provided by the second filter block 254a-254c in the listening environment 205. The microphone 204a-204b senses the audio played back in the listening environment 205, and the sensed audio is output to the third plurality of filter banks (or analysis filter banks) 461a-461b for filtering. Psychoacoustic building blocks 462a-462b convert the filtered sensed audio and calculate the energy in each critical sub-band of the psychoacoustic band represented by EM(m, j), where m corresponds to the number of microphones and j corresponds to the critical bands from critical band number 1 to critical band number CB covering the audible acoustic frequency range (e.g., from 0 to 20kHz) in the psychoacoustic frequency scale. For example, psychoacoustic building blocks 462a-462b generate EM(1, j) and EM(2, j), respectively. Psychoacoustic building block 462a provides EM(1, j) to comparators 470a, 470b. Psychoacoustic building block 462b provides EM(2, j) to comparators 470c and 470d. The relevance of comparators 470a-470d will be discussed in more detail below.

[0048] While providing audio input to the first plurality of filter banks 450a-450b, audio input is also provided to delay blocks 456a-456b. Delay blocks 456a-456b delay audio input by, for example, 10 milliseconds to 20 milliseconds. Delayed audio input is provided to psychoacoustic building blocks 458a-458c. Delay blocks 456a-456c are applied to both microphone path and speaker path to provide frame synchronization between microphone path and speaker path. It should be appreciated that the tuning of the delay value utilized in delay blocks 456a-456c may be necessary to achieve frame synchronization between microphone path and speaker path (e.g., there will be a delay between speaker 202 playback audio and then microphone 204 captures the audio played back by speaker 202). The psychoacoustic building blocks 458a-458c convert the delayed audio input and calculate the energy in each critical subband of the psychoacoustic band represented by ES(s,j), where s corresponds to the number of speakers and j corresponds to the critical band numbers from critical band number 1 to critical band number CB in the psychoacoustic frequency scale covering the audible acoustic frequency range (e.g., from 0 to 20 kHz).

[0049] For speaker 202a, comparator 470a generates sub-band coefficients WS(s,j) or WS(1,j), which generally correspond to the difference between the psychoacoustic bands of speaker 202a and microphone 204a. For example, sub-band coefficients WS(1,j)=ES(1,j)-EM(1,j), which are transmitted to compensation circuit 452b to modify the audio input to speaker 202a. Similarly, for speaker 202b, comparator 470b generates sub-band coefficients WS 1 (s,j) or WS 1 (2, j), the sub-band coefficients generally correspond to the first difference between the psychoacoustic bands of the loudspeaker 202b and the microphone 204a. For example, the sub-band coefficients WS 1 (2,j)=ES(2,j)-EM(1,j) is transmitted to the function block 482. In addition, for the speaker 202b, the comparator 470c generates the sub-band coefficient WS 2 (s,j) or WS 2 (2, j), the sub-band coefficients generally correspond to the second difference between the psychoacoustic bands of the loudspeaker 202b and the microphone 204b. The sub-band coefficients WS 2 (2, j) = ES (2, j) - EM (2, j) is transmitted to the function block 482. The function block 482 combines the responses of the microphones 204a, 204b by taking the maximum, minimum, average, or smoothing of the responses of the two microphones 204a, 204b. The function block 482 transmits the output of the function corresponding to the psychoacoustic frequency band of the speaker 202b and the two microphones 204a, 204b to the compensation circuit 452c to modify the audio input to the speaker 202b.

[0050] For speaker 202c, comparator 470d generates sub-band coefficients WS(s,j) or WS(3,j), which generally correspond to the difference between the psychoacoustic bands of speaker 202c and microphone 204b. For example, sub-band coefficients WS(3,j)=ES(3,j)-EM(2,j) are transmitted to compensation circuit 452a to modify the audio input to speaker 202c. Compensation circuits 452a, 452b, 452c apply complex factors (e.g., by phase or amplitude).

[0051] The above-described adaptive process provides room equalization or correction, which can provide a successful immersive and high-fidelity listening experience in an enclosed space such as, for example, a vehicle cabin. The process of room equalization (RE) involves, among other things, compensating for unwanted room sound artifacts such as early reflections, reverberant reflections, surrounding material properties, and speaker imperfections.

[0052] Psychoacoustically perceived loudness (or PPL), which is a subjective perception of sound pressure, can be calculated using different techniques, such as equal loudness contours, absolute hearing threshold (ATH), A-weighting, K-weighting relative to full scale (LKFS), etc. PPL can also be calculated using the psychoacoustic definitions presented herein. An advantage of the embodiments disclosed herein is that both amplitude information and phase information of room impairments can be obtained through the complex nature of critical subband analysis. For example, the loudspeaker 202a has the following transmitted loudness and received loudness at the listening position 206a of the microphone 204a, respectively:

[0053] PPL TX =∑ j ΔS(1,j)=∑ j |ES(1,j)-TS(1,j)| (Equation 1)

[0054] PPL RX =∑ j ΔM(1,j)=∑ j |EM(1,j)-TS(1,j)| (Equation 2)

[0055] Equation 1 may be performed by difference blocks 459a-459c, such as in combination with Figure 5 The PPLs defined in Equations 1 and 2 (eg, for the adaptive process) are similar to those cited above in connection with static calibration. For example, PPL TX Similar to the ideal PPL (PPL_I) (or first psychoacoustic perceived loudness) and PPL RX Similar to (PPL_M) ((or the second psychoacoustic perceived loudness) (e.g., each of the ideal PPL (PPL_I) and (PPL_M) has been indicated above). For simplicity, the PPL referenced in conjunction with Equations 1 and 2 is being redefined. ES(s,j) generally corresponds to a critical sub-band within the psychoacoustic frequency range of the loudspeaker, and TS(s,j) generally corresponds to a psychoacoustic hearing threshold for each critical sub-band. If S(s,j) is greater than 0, the audio content in sub-band j is audible to the listener. If S(s,j) is less than 0, the audio content in sub-band j is inaudible to the listener.

[0056] PPL due to room acoustic artifacts 损失 Defined as:

[0057] PPL 损失 =PPL TX -PPL RX =ES(1,j)-EM(1,j) (Equation 3)

[0058] Equation 3 may be determined or implemented by various comparators 470a-470d, such as in combination with Figure 4 As indicated. PPL 损失 Similar to PPL_L (ie psychoacoustic perceived loudness loss) as noted above. 损失 is a complex quantity with information about amplitude and phase, as shown directly below.

[0059] Mag(PPL 损失 ) = |ES(1, j) - EM(1, j)| (Equation 4)

[0060]

[0061] If the PPL at the critical sub-band j 损失 has a positive amplitude, then the main room impairments are caused by absorption and / or dissipation. On the other hand, if the PPL at the critical sub-band j is 损失 With negative amplitude, the main room impairments are caused by reflections and reverberation.

[0062] For the PPL at critical sub-band j 损失 In the case of positive amplitude, room equalization may involve amplifying the critical sub-band of attenuation. In general, the compensator circuits 452a-452c may determine the PPL at the critical sub-band j. 损失 Whether it has a positive amplitude, if it has a positive amplitude, the audio input transmitted to the speakers 202a-202c respectively is amplified.

[0063] For the PPL at critical sub-band j 损失 In the case of negative amplitude, room equalization may involve attenuating the amplified critical sub-band. In general, the compensator circuits 452a-452c may determine the PPL at the critical sub-band j. 损失 Whether it has a negative amplitude, if it has a negative amplitude, the audio input transmitted to the speakers 202a-202c respectively is amplified.

[0064] For the PPL at critical sub-band j RX The phase and PPL at the critical sub-band j TX If the phase difference of the received critical sub-band exceeds a certain threshold (or predetermined threshold), a phase correction can be applied by rotating the received critical sub-band phase to match its transmitted counterpart. This complex multiplication is performed as indicated above. For example, when the PPL at critical sub-band j RX The phase and PPL at the critical sub-band j TX When the phase difference of exceeds a certain threshold (or predetermined threshold), the compensation circuits 454a-454c perform complex multiplication. If a particular speaker 202 is shared by more than one microphone 204, then the PPL RXThe microphone phase performs mathematical operations such as maximum, minimum, average, smoothing, etc. Examples of this are Figure 4 The first function block 472 shown in FIG. 4 is not shown in FIG. 4 because the speaker 202b is shared by the microphones 204a and 204b.

[0065] Figure 6 A method for providing Figure 5 The method 600 of performing an adaptive runtime process for the room correction and equalization system 500 of the present invention. In operation 602, the audio controller 208 determines the PPL of each speaker 202a, 202b, 202c in the array. For example, the audio controller 208 determines the PPL of each speaker 202a, 202b, 202c based on equation 1 as indicated above. The audio controller 208 also determines the PPL of each microphone 204a, 204b in the array based on equation 2 as indicated above.

[0066] In operation 604, the audio controller 208 determines the PPL loss due to acoustic artifacts that may be present in the listening environment 205. For example, the audio controller 208 determines the PPL loss due to acoustic artifacts based on equation 3 as noted above. In operation 606, the audio controller 208 determines whether the magnitude of the PPL loss of the speakers 202a-202c and microphones 204a-204b is positive or negative. For example, in operation 606, the audio controller 208 determines the magnitude of the PPL loss based on equation 4 and then determines whether the magnitude is positive or negative. If the magnitude is positive, the method 600 proceeds to operation 610. If the magnitude is negative, the method 600 proceeds to operation 612.

[0067] In operation 610, the audio controller 208 determines that the PPL loss at the critical sub-band j has a positive magnitude and that the primary listening room impairments are due to absorption and / or dissipation present in the listening environment 205. In this case, the audio controller 208 amplifies the audio input provided to the speakers 202a-202c in the listening environment 205.

[0068] In operation 612, the audio controller 208 determines that the PPL loss at the critical sub-band j has a negative magnitude and that the primary listening room impairments are due to reflections and reverberation in the listening environment 205. In this case, the audio controller 208 attenuates the audio input provided to the speakers 202a-202c in the listening environment 205.

[0069] In operation 614, the audio controller 208 determines the phase of the PPL loss of the microphones 204a-204b and the speakers 202a-202c based on Equation 5 as noted above. For example, the audio controller 208 determines whether the phase of the PPL loss of the speakers 202a-202c differs from the phase of the PPL loss of the microphones 204a-204b by a predetermined threshold. If this condition is true, the method 600 moves to operation 618. If not, the method 600 returns to operation 602. In operation 618, the audio controller 208 applies a phase correction to the critical sub-band phase (e.g., ES(s,j)) of the speakers 202a-202c or the critical sub-band phase of the microphones 204a-204b by rotating the received critical sub-band phase (e.g., the critical sub-band phase of the microphones 204a-204b) to match its transmitted counterpart (e.g., the critical sub-band phase of the speakers 202a-202c).

[0070] Although exemplary embodiments are described above, it is not intended that these embodiments describe all possible forms of the present invention. Rather, the words used in this specification are descriptive rather than restrictive, and it should be understood that various changes can be made without departing from the spirit and scope of the present invention. In addition, the features of the various embodiments can be combined to form other embodiments of the present invention.

Claims

1. An audio system, comprising: a plurality of speakers configured to transmit audio signals in a listening environment; a plurality of microphones, each of the plurality of microphones being positioned at a corresponding listening position in the listening environment, the plurality of microphones being configured to detect the audio signal in the listening environment; At least one audio controller, the at least one audio controller being configured to: determining a first psychoacoustic perceived loudness (PPL) of the audio signal while the audio signal is being played back through a first speaker of the plurality of speakers, the PPL corresponding to a limit on the amount of acoustic energy perceived by a human ear at different time intervals or frames; determining a second PPL of the audio signal when the audio signal is sensed by a first microphone of the plurality of microphones; and The first speaker of the plurality of speakers is mapped to the first microphone of the plurality of microphones based at least on the first PPL and the second PPL. 2 . The audio system of claim 1 , wherein the at least one audio controller is further configured to determine a first magnitude of the first PPL and a second magnitude of the second PPL before mapping the first speaker to the first microphone. 3 . The audio system of claim 2 , wherein the at least one audio controller is further configured to obtain a difference between the first amplitude of the first PPL and the second amplitude of the second PPL before mapping the first speaker to the first microphone. 4 . The audio system of claim 3 , wherein the difference between the first amplitude of the first PPL and the second amplitude of the second PPL corresponds to a PPL loss indicative of a perceived audible deviation at a listening position in the listening environment. 5 . The audio system of claim 4 , wherein the audio controller is further configured to compare the PPL loss to a programmable threshold to determine whether to map the first speaker to the first microphone. 6 . The audio system of claim 5 , wherein the audio controller is further configured to map the first speaker to the first microphone in response to the PPL loss being less than the programmable threshold.

7. The audio system of claim 1, wherein the audio controller is further configured to apply an adaptive process to equalize the audio signal in the listening environment based at least on the first PPL and the second PPL.

8. The audio system of claim 7, wherein the audio controller is further configured to determine the first PPL of the audio signal when the audio signal is being played back through each of the plurality of speakers, and to determine the second PPL of the audio signal for each of the plurality of microphones when the audio signal is sensed by each of the plurality of microphones. 9 . The audio system of claim 8 , wherein the audio controller is further configured to determine a PPL loss for each speaker of the plurality of speakers and each microphone of the plurality of microphones based on a difference between the first PPL and the second PPL.

10. The audio system of claim 9, wherein the at least one audio controller is further configured to amplify the audio input signals to the plurality of speakers if the magnitude of the PPL loss for each of the speakers and each of the microphones is positive to account for absorption and / or dissipation present in the listening environment.

11. The audio system of claim 9, wherein the at least one audio controller is further configured to attenuate audio input signals to the plurality of speakers if the magnitude of the PPL loss of each of the speakers and each of the microphones is negative to account for reflections and reverberation in the listening environment.

12. The audio system of claim 9, wherein the at least one audio controller is further configured to determine whether a phase of the PPL loss of each of the speakers differs from the PPL of each of the microphones by a predetermined amount.

13. The audio system of claim 12, wherein the at least one audio controller is further configured to apply a phase correction to a critical sub-band phase of the plurality of speakers or a critical sub-band phase of the plurality of microphones when the phase of the PPL loss of each of the speakers differs from the PPL of each of the microphones by the predetermined amount.

14. An audio system comprising: a plurality of speakers configured to transmit audio signals in a listening environment; a plurality of microphones, each of the plurality of microphones being positioned at a corresponding listening position in the listening environment, the plurality of microphones being configured to detect the audio signal in the listening environment; as well as At least one audio controller configured to determine a first psychoacoustic perceived loudness (PPL) for each of the plurality of speakers and to determine a second PPL for each of the plurality of microphones to employ an adaptive process to equalize the audio signal in the listening environment, the PPL corresponding to a limit of an amount of acoustic energy perceived by a human ear at different time intervals or frames. 15 . The audio system of claim 14 , wherein the audio controller is further configured to determine a PPL loss for each speaker of the plurality of speakers and each microphone of the plurality of microphones based on a difference between the first PPL and the second PPL.

16. The audio system of claim 15, wherein the at least one audio controller is further configured to amplify the audio input signals to the plurality of speakers if the magnitude of the PPL loss for each of the speakers and each of the microphones is positive to account for absorption and / or dissipation present in the listening environment.

17. The audio system of claim 15, wherein the at least one audio controller is further configured to attenuate audio input signals to the plurality of speakers if the magnitude of the PPL loss of each of the speakers and each of the microphones is negative to account for reflections and reverberation in the listening environment.

18. The audio system of claim 15, wherein the at least one audio controller is further configured to determine whether a phase of the PPL loss of each of the speakers differs from the PPL of each of the microphones by a predetermined amount.

19. The audio system of claim 18, wherein the at least one audio controller is further configured to apply a phase correction to a critical sub-band phase of the plurality of speakers or a critical sub-band phase of the plurality of microphones when the phase of the PPL loss of each of the speakers differs from the PPL of each of the microphones by the predetermined amount.

20. A method for equalizing an audio signal in a listening environment using an adaptive process, the method comprising: transmitting audio signals in the listening environment via a plurality of speakers; detecting the audio signal in the listening environment via a plurality of microphones positioned in the listening environment; determining a first psychoacoustic perceived loudness PPL for each of the plurality of speakers, the PPL corresponding to a limit of an amount of acoustic energy perceived by a human ear at different time intervals or frames; as well as A second PPL for each of the plurality of microphones is determined to employ an adaptive process to equalize the audio signal in the listening environment based on the first PPL and the second PPL.

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