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    • 专利摘要:
    AUDIO PRE-COMPENSATION REGULATOR MODEL USING A VARIABLE SET OF SUPPORT SPEAKERS. The invention is related to a basic idea of determining an audio pre-compensation regulator for an associated sound generating system, comprising a total of N (Major equal) 2 speakers, each having a speaker input. The audio pre-compensation regulator has a number of L (greater than equal) 1 inputs for L input signals, and N outputs for N output controller signals, one for each speaker. It is important to estimate, for each of at least a subset of the N speaker inputs, a pulse response at each measurement position. It is also important to specify for each of the L input signal (s), a speaker selected from the N speakers as the primary speaker, and a selected subset S, including at least one of the N speakers. , as support speakers. A key point is to specify for each primary speaker a target pulse response at each measurement position, with the target pulse response having an acoustic propagation delay, where the acoustic propagation delay is determined (...).
    公开号:BR112014018342B1
    申请号:R112014018342-2
    申请日:2012-03-22
    公开日:2021-03-16
    发明作者:Brãnnmark Lars-Johan；Ahlén Anders；Bahne Adrian
    申请人:Dirac Research Ab；
    IPC主号:
    专利说明:

    Technical Field of the Invention
    [001] The present invention, in general, is related to digital audio pre-compensation and, more particularly, to the model of a digital audio pre-compensation controller that generates several signals for a sound generating system, with the objective of modifying the dynamic response of the compensated system, as measured at various measurement positions, in a region of spatial interest, in a listening environment. Background of the Invention
    [002] A system to generate or reproduce sound, including amplifiers, cables, speakers and acoustic environments, will always affect the spectral, transient and spatial properties of the reproduced sound, usually in an undesired manner. Specifically, the acoustic reverberation of the room where the equipment is placed has a considerable and, usually, a detrimental effect on the perceived audio quality of the system. The reverb effect is usually described in a different way, depending on which region of frequency is considered. At low frequencies, reverberation is usually described in terms of resonances, vertical waves or so-called ambient modes, which affect the sound reproduced by introducing strong peaks and zero depth values at different frequencies at the low end of the spectrum. Under higher frequencies, reverberation is usually thought of as reflections reaching the listener's ears, some time after the sound comes directly from the speaker itself.
    [003] The reproduction of sound with a very high quality can generally be achieved by using leveled sets of cables, amplifiers and speakers of high quality, and by modifying the acoustic properties of the environment using, for example, acoustic diffusers, resonators Helmholtz and acoustically absorbing materials. However, these passive means of improving sound quality are cumbersome, expensive and sometimes impractical.
    [004] Other means to improve the quality of sound reproduction systems include active solutions based on digital filtration, usually referred to as pre-compensation, equalization or dereverberation.
    [005] A pre-compensation filter R, as shown in figure 1, is then placed between the original audio signal source and the audio equipment. The dynamic properties of the sound generating system can be measured and modeled by recording the system's response to known test signals in one or several positions in the environment. The filter R is then calculated and implemented to compensate for the measured properties of the system, symbolized by H in figure 1. In particular, it is desirable that the phase and amplitude response of the compensated system are close to a previously specified ideal response, symbolized by D in figure 1, in all measurement positions. In other words, it is required that the reproduction of compensated sound y (t) corresponds to the ideal yref (t), with some degree of precision. The pre-distortion generated by the pre-compensating device R is idealized to counterbalance the distortion due to the H system, so that the reproduction of the resulting sound has the characteristic of the sound of D. In order to obtain a pre-compensating device that is robust and of practical utility, it is important to imagine that the H model may not be a perfect description of the real system and the recordings of the system responses may contain disturbances, due, for example, to background noise. These measurement and modeling errors can, for example, be represented by adding a noise signal (e (t), in figure 1) to the system, producing the measured output of the system ym (t). As will be described below, modeling errors and uncertainties about the system can also be included in model H, which is then partially parameterized by random variables with specified probability distributions.
    [006] Therefore, up to the physical limits of the system, at least in theory, it is possible to obtain an improved quality of sound reproduction, without the high cost of using extremely advanced audio equipment. The purpose of the model may, for example, be to cancel the acoustic resonances and the diffraction effects caused by imperfectly constructed speaker boxes. Another application may be to minimize the effect of ambient modes (ie, low frequency resonance peaks and zero value peaks) at different locations in the listening environment. Yet another objective may be to obtain a balance of pleasant tonality and a detailed perceived stereo image.
    [007] Until then, the methods established for digital audio pre-compensation systems that exist commercially on the market and in the scientific literature are mainly single channel methods, see, for example, the bibliographic reference [17]. Single channel pre-compensation refers to the principle that the input signal to a speaker is processed by a single filter. When single signal pre-compensation is applied to a sound system containing more than one speaker channel, for example, a 5.1 cinema room system, having five broadband channels and a subwoofer (type of audio player). sound used to increase the sound of a speaker), which means that the filters for different speaker channels are determined individually and independently of each other. The proportion in which each compensated speaker actually obtains its ideal target response specified in all measurement positions depends mainly on the following two factors: 1. If the impulse response of the speaker and the environment is not entirely different. minimum phase characteristic, then, the compensation filter must be of the type called mixed phase, in order to correct non-minimum phase distortion components. Since practically all of the speaker-environment impulse responses do not contain minimum phase components [23], a minimum phase filter will be insufficient to compensate the system to obtain the target response fully. As the model of mixed phase filters for audio use is considerably less correct than the model of minimum phase filters, the products that exist most for digital audio pre-compensation make use of filters that are restricted to the minimum phase type . 2. If a speaker impulse response varies between different measurement positions, as is usually the case in an environment, then a single filter will not be able to fully correct the speaker response in all positions measurement, due to conflicting requirements in the different positions. In a medium sense, the response of the compensated system may be closer to the target, but due to the spatial variation of the system, there will always be errors remaining at each measurement position. In addition, if a mixed phase compensating device is used, then errors may occur in the form of so-called “pre-touches”, unless the compensating device is designed with great care [5]. Pre-touch errors are known to be noticeably much more objectionable than post-touch errors. In the bibliographic references [5, 6] it is shown how to design a mixed phase compensating device that alleviates the problem of pre-touch errors, correcting only the minimum phase distortion that is common to all measurement positions.
    [008] Thus, the single channel compensation method has a potential limitation, as it can only correct the impulse and frequency responses in the context of averaging, when multiple measurement positions are considered. In an acoustic environment, where the original response of a speaker varies markedly between measurement positions, this variability will also remain in the responses of the compensated speaker, although the performance of the compensated system, on average, is closer to the target performance. In addition, when designing a compensating device with respect to only one measurement position, this is not a realistic option, as it is well known that single point designs produce filters that are extremely weak and degrade the performance of the system in all other positions in the environment [13, 14].
    [009] Therefore, it can be concluded that single channel audio pre-compensation methods are more effective in correcting the degradations that are systematic with respect to the spatial region of interest, that is, the distortion of the components that are common or, at least, practically common, in all measurement positions. Typically, these systematic degradations are caused by the loudspeaker itself, or by reflective surfaces very close to the loudspeaker, or by the acoustics of the environment at low frequencies, where the wavelength is large, compared to the region of interest. If a sound reproduction system, including its acoustic environment, is such that its spatially variable distortion is dominant over its spatially common distortion, then, unfortunately, the improvement in sound quality offered by single channel methods will be less .
    [010] Considering the above, the question arises whether a higher performance audio pre-compensation strategy can be achieved, for example, through the use of speaker structures and filters in a more flexible way than suggested by the established single channel methods. In the research literature related to acoustics, some different strategies that are presented in addition to the traditional single channel filtration have been identified [2, 7, 9, 10, 11, 12, 18, 21, 22, 24, 25, 29, 33, 34]. In summary, the known methods can be grouped into the following categories: 1. The methods of the first category are based on the physical view related to the acoustics of the environment and, particularly, on the acoustic coupling between the speakers and the low resonance modes. frequency of the environment. It is well known that a carefully selected physical placement of the speakers and the use of several subwoofers are means that help to reduce the effect of the ambient modes [34]. 2. Another principle is the source-drain method [7, 8, 33], where the ambient modes are reduced by positioning a certain number of subwoofers symmetrically in the environment, after which time delay and gain adjustments are applied to the different subwoofer channels. According to this method, the subwoofers on all the front walls of the ambient room act as sound sources, while the subwoofers adjusted for phase delay and gain on the rear walls act as drains, that is, sound absorbers, which cancel reflections. frequency of the rear wall. However, the method is restricted to only work in the lower part of the spectrum (below 150 Hz) and the type of adjustment made to the subwoofer signals is quite primitive. 3. A third important method is modal equalization [16, 21], in which the modal resonances and their decay times are equalized by means of digital pre-filters. This method involves an explicit identification of the central frequencies and decay times of single ambient modes, being limited to operating at markedly low frequencies (typically only below 200 Hz), where the resonances of the ambient room are supposed to be differentiated and well separated on the frequency axis. Reference [16] discusses two possible approaches, Type I, which is that of a single channel equalizer, and Type II, which uses two or more channels to cancel the ambient modes. It is acknowledged in the reference [16] that the filter model for the Type II modal equalization is not correct when more than two channels are used, and an explicit solution for the case of the multiple channel model is not presented. All together, the approaches are unsatisfactory, since they are based on assumptions that, in general, are not met in a typical room environment, for example, that all modes that are subjected to equalization are well separated and that they can be estimated with great precision. 4. A fourth category of methods is based on the multi-channel filter model, with several objectives in mind. One objective is active noise control, in which the sound of one or several speakers is used to cancel unwanted acoustic disturbances, see, for example, the reference [11]. A second objective is to obtain an exact reproduction of the specific sound pressures in a small number of spatial positions, typically, the positions of the ears of a human listener. This approach is usually referred to as cross-talk cancellation, virtual acoustic image formation or trans-aural stereo [2, 22, 24, 25]. One drawback of this approach is that its performance is extremely sensitive to small movements of the listener and, in particular, is not robust in normal reverb environments. A third common objective concerns “holographic” audio training techniques, such as Wave Field Synthesis (WFS) and High Order Ambisonics (HOA) [10, 28, 30], which aim to reproduce arbitrary sound fields in large regions, in two or three dimensions, using massive loudspeaker arrangements of 50 or more speakers. A number of multi-channel filter models have been proposed in order to improve the performance of WFS and HOA techniques and correlated techniques, see, for example, references [9, 12, 18, 29]. A fourth objective concerns the minimization of the destructive phase interaction in the crossover frequency region, between the subwoofer and satellite speakers, in sound systems that employ the so-called bass management [3]. These mentioned multi-channel filter models are not suitable for the general speaker pre-compensation problem. First, they are significantly different in their objectives compared to single channel pre-compensation methods. Second, the proposed computation methods produce filters with unsatisfactory properties. For example, most methods design filters in the frequency domain without considering the behavior of the broadband filter, such as chance, the maximum delay allowed through the system and the level and duration of pre-touch errors.
    [011] None of the multi-channel filter projection methods described in the prior art are of use for the purpose of robust broadband speaker / room compensation, of an existing speaker installed for stereo audio playback or multiple channels. Summary of the Invention
    [012] It is a general objective of the invention, to provide an extended pre-compensation strategy, to improve the reproduction of stereo or multi-channel audio material relative to two or more speakers.
    [013] It constitutes a specific objective of the invention, provides a method for determining an audio pre-compensation controller for an associated sound generating system.
    [014] Another specific objective of the invention is to provide a system for determining an audio pre-compensation controller for an associated sound generating system.
    [015] Yet another specific objective of the invention is to provide a computer program product for determining an audio pre-compensation controller for an associated sound generating system.
    [016] It is also a specific objective of the invention to provide an improved audio pre-compensation controller, as well as an audio system comprising such an audio pre-compensation controller, and a digital audio signal generated by the audio controller. audio pre-compensation.
    [017] These and other objectives are achieved by the invention as defined by the appended claims.
    [018] A basic idea of the invention is to determine an audio pre-compensation controller for an associated sound generating system, comprising a total of N> 2 speakers, each having a speaker input. The audio pre-compensation controller has a number of L> 1 inputs for L input signals, and N outputs for N controller output signals, one for each speaker of the sound generator system, and the pre-controller - audio compensation generally has a number of adjustable filter parameters. It is important to estimate for each of at least a subset of said N speaker inputs, an impulse response in each of a plurality M> 2 of measurement positions, distributed in a region of interest in a listening environment, based on in sound measurements at said M measurement positions. It is also important to specify for each of the one or more L input signals a speaker, selected from the N speakers as a primary speaker, and a selected subset S, including at least one of the N speakers as a or more support speakers, where the primary speaker is not part of this subset. A fundamental point is to specify for each primary speaker a target impulse response in each of the M measurement positions, where the target impulse response has an acoustic propagation delay, where the acoustic propagation delay is determined based on the distance from the loudspeaker. primary speaker for the respective measurement position. The idea then is to determine for each of the one or more L input signals, based on the selected primary speaker and on the selected support speaker (s), filter parameters of the audio pre-compensation controller, so that a criterion function is optimized under the stability constraint of the dynamics principles of the audio pre-compensation controller. The criterion function includes a weighted sum of powers of differences between the estimated compensated impulse responses and the target impulse responses, with respect to the M measurement positions.
    [019] The different aspects of the invention include a method, a system and a computer program for determining an audio pre-compensation controller, a so-determined audio pre-compensation controller, an audio system incorporating said audio controller audio pre-compensation, as well as a digital audio signal generated by said audio pre-compensation controller.
    [020] The present invention offers the following advantages: - Improved design scheme for an audio pre-compensation controller; - Improved reproduction of stereo or multi-channel audio material, with respect to two or more speakers; - Better performance in rooms or listening environments, where the responses to the impulses of the speakers are variable with the spatial position; - Greater flexibility, in which performance improvements are not restricted by low frequencies; - Control related to types of problems, such as, casualty and pre-touch artifacts.
    [021] Other advantages and characteristics offered by the present invention will be observed when reading the following description of its modalities. Brief Description of Drawings
    [022] The invention, together with additional objectives and other advantages, can be better understood by referring to the following description, taken in correlation with the accompanying drawings, in which: - figure 1 describes a single channel compensating device R , which has a w (t) signal as an input signal. The compensating device produces a control signal u (t), which functions as an input for model H, stable, linear and dynamic, with single input and multiple outputs (SIMO) of the acoustic system. Model H has an input and M outputs, where the M outputs represent M measurement positions. The acoustic signals of the M measurement positions are represented by a column vector y (t). The desired properties of the dynamic system are specified by a stable model (D), type SIMO, which has an input and M outputs. When the signal o (t) is used as an input for D, the resulting output is a desired signal vector yref (t) with M elements. The M-dimensional signal vector ym (t) represents a measurement of y (t) and the signal vector e (t), which also has the dimension M, represents a possible measurement disturbance; figure 2 describes a multi-channel compensating device (R), which has a signal w (t) as an input signal. The compensating device produces a multi-channel u (t) control signal, with N elements, which acts as an input for the model (H), stable, linear and dynamic, with multiple inputs and multiple outputs (MIMO) of the acoustic system . Model (H) has N inputs and M outputs, where the N inputs represent the inputs for the N loudspeakers and the M outputs represent the M measurement positions. The acoustic signals at the M measurement positions are represented by a column vector y (t). The desired properties of the dynamic system are specified by a stable model (D), type SIMO, which has an input and M outputs. When the signal w (t) is used as an input for (D), the resulting output is a desired vector of signal yref (t) with M elements. The M-dimensional signal vector ym (t) represents a measurement of y (t) and the signal vector e (t), which also has the dimension M, represents a possible measurement disturbance; figure 3 is a schematic diagram illustrating an example of an audio system, including a sound generating system and an audio pre-compensation controller; figure 4 is a schematic block diagram of an example of a computer-based system, suitable for implementing the invention; figure 5 is a schematic flow diagram illustrating a method for determining an audio pre-compensation controller, according to an exemplary embodiment; - figure 6 illustrates the frequency responses of a speaker in an environment, measured in 64 positions (gray lines) and its quadratic mean (RMS) (black line); - figure 7 illustrates the frequency responses of the same speaker as figure 6, after a single channel pre-compensation filter has been applied to its input. The figure shows the frequency responses measured in 64 positions (gray lines) and their quadratic mean (RMS) (black line); - figure 8 shows the result of a multi-channel pre-compensation, where the speaker in figure 6 was used as the primary speaker, and 15 additional speakers were used as support speakers. The figure shows the frequency responses measured in 64 positions (gray lines) and their quadratic mean (RMS) (black line); - Figure 9 shows a cascade plot or cumulative spectral decay of the same speaker in Figure 6, when no pre-compensation was applied. The cascade shown in the figure is the cumulative spectral mean decay of the speaker impulse response at 64 positions; - figure 10 shows a cascade plot or spectral cumulative decay of the same speaker as figure 7, when a single channel pre-compensation filter was applied. The cascade shown in the figure is the cumulative spectral mean decay of the compensated impulse response of the speaker at 64 positions; - figure 11 shows a cascade plot or cumulative spectral decay of the same speaker as in figure 8, in which a multi-channel pre-compensation strategy was applied to compensate for the primary speaker, using 15 additional speakers. Support. The cascade shown in the figure is the cumulative spectral mean decay of the compensated impulse response of the speaker at 64 positions. Detailed Description of the Invention
    [023] In the presentation of the drawings, the same numerical references are used for similar or corresponding elements.
    [024] The proposed technology is based on the recognition that mathematical models of dynamic systems and model-based optimization of digital pre-compensation filters provide powerful tools for the filter model that improve the performance of various types of audio equipment, by modifying the input signals to the equipment. In addition, it is observed that suitable models can be obtained through measurements in a plurality of measurement positions, distributed in a region of interest in a listening environment.
    [025] As mentioned, a basic idea of the invention is to determine an audio pre-compensation controller for an associated sound generating system. As illustrated in the example in figure 3, the sound generating system comprises a total of N> 2 speakers, each having a speaker input. The audio pre-compensation controller has an L> 1 number of inputs, for the L input signal (s), and N outputs for the N controller output signals, one for each speaker in the system. sound generator. It should be understood that the controller output signals are directed to the speakers, that is, in the input path of the speakers. The controller output signals can be transferred to the speaker inputs via optional circuits (indicated by dashed lines), such as digital to analog converters, amplifiers and additional filters. Optional circuits may also include a wireless connection.
    [026] In general, the audio pre-compensation controller has a number of adjustable filter parameters, to be determined in the filter design scheme. The audio pre-compensation controller, when designed, must therefore generate N controller output signals for the sound generating system, in order to modify the dynamic response of the compensated system, when measured in a plurality M> 2 measurement positions, distributed in a region of interest in a listening environment.
    [027] Figure 5 is a schematic flow diagram illustrating a method for determining an audio pre-compensation controller, according to an exemplary modality. Step (S1) involves estimating for each of at least a subset of the N speaker inputs, an impulse response in each of a plurality of M> 2 measurement positions, distributed in a region of interest in a listening environment , based on the sound measurements in the M measurement positions. Step (S2) involves specifying for each of the one or more L input signals a speaker, selected from the N speakers as a primary speaker, and a selected subset S, including at least one of the N speakers. speakers as one or more supporting speakers, where the primary speaker is not part of the subset. Step (S3) involves specifying for each primary speaker a target impulse response in each of the M measurement positions, where the target impulse response has an acoustic propagation delay, where the acoustic propagation delay is determined based on the distance from the primary speaker to the respective measurement position. Step (S4) involves determining for each of the one or more L input signals, based on the selected primary speaker and on the selected support speaker (s), filter parameters of the audio pre-compensation controller, so that a criterion function is optimized under the stability constraint of the dynamics principles of the audio pre-compensation controller. The criterion function includes a weighted sum of powers of differences between the estimated compensated impulse responses and the target impulse responses, with respect to the M measurement positions.
    [028] Expressed differently, the audio pre-compensation controller is configured to control the acoustic response of P primary speakers, where P <L and P <N, through the combined use of P primary speakers and , for each primary speaker, an additional number of support speakers 1 <S <N-1 of the N speakers.
    [029] If there are two or more input signals, that is, L> 2, the method can also include the optional step (S5) of merging all the filter parameters, determined for the L input signals, within a set merged filter parameters for the audio pre-compensation controller. The audio pre-compensation controller, together with the fused set of filter parameters, is configured to operate on the L input signals, to generate the N controller output signals to the speakers, to obtain the impulse responses target.
    [030] For example, it may be desirable that the audio pre-compensation controller has the ability to produce zero or zero output for some of the N speakers, for some adjustment of its adjustable filter parameters.
    [031] Preferably, the target impulse responses are different from zero, and include adjustable parameters that can be modified within established limits. Thus, for example, the adjustable parameters of the target impulse responses, as well as the adjustable parameters of the audio pre-compensation controller can be adjusted together, in order to optimize the criterion function.
    [032] In a specific exemplary modality, the step of determining filter parameters of the audio pre-compensation controller is based on a Gaussian Linear Quadratic Optimization (LQG) of the parameters of a stable, linear and multivariable causal direct feed controller, based on in a given dynamic target system and in a dynamic model of the sound generating system. As mentioned, the controller output signals can be transferred to the speaker inputs via optional circuits. Thus, for example, each of the N controller output signals from the audio pre-compensation controller can be fed to a respective speaker through a pass-through filter, including a phase compensation component and a delay, producing N filtered controller output signals.
    [033] Optionally, the criterion function includes penalty terms, in which the penalty terms are such that the audio pre-compensation controller, obtained by optimizing the criterion function, produces levels of signals of restricted magnitude in a selected subset of the pre-compensation controller outputs, producing restricted signal levels at selected speaker inputs for the N speakers, for specified frequency ranges.
    [034] The penalty terms can be chosen differently a number of times, and the step of determining the audio pre-compensation controller filter parameters is repeated for each choice of penalty terms, resulting in a number of occurrences of the audio pre-compensation controller, each producing signal levels that individually restrict magnitudes to the supporting speakers, for specified frequency ranges.
    [035] In an optional additional mode, the criterion function contains a representation of possible errors in the estimated impulse responses. This representation of errors is designed as a set of models that describe the alleged range of errors. In this specific modality, the criterion function also contains an aggregation operation, which can be a sum, a weighted sum or a statistical expectation in relation to the said set of models.
    [036] In a particular example, the step of determining the filter parameters of the audio pre-compensation controller is also based on adjusting the filter parameters of the audio pre-compensation controller, in order to obtain a target response of frequency magnitude of the sound generating system, including the audio pre-compensation controller, in at least a subset of the M measurement positions.
    [037] For example, the step of adjusting the filter parameters of the audio pre-compensation controller is based on the evaluation of the frequency magnitude responses in at least a subset of the said M measurement positions and, after that, in determining a minimum phase model of the sound generating system, including the audio pre-compensation controller.
    [038] Preferably, in the estimation step for each of at least a subset of the N speaker inputs, an impulse response in each of the plurality M of measurement positions is based on a model that describes the dynamic response of the generating system of sound in the M measurement positions.
    [039] As understood by a person skilled in the art, the audio pre-compensation controller can be created by implementing the filter parameters in an audio filter structure. The audio filter structure is then, typically, incorporated together with the sound generator system to enable the generation of the target impulse response in the M measurement positions, in the listening environment.
    [040] The proposed technology can be used in several audio applications. Thus, for example, the sound generating system can be a car audio system, or a mobile studio audio system, and said listening environment can be part of a car or a mobile studio.
    [041] Other examples of a sound generating system include a cinema theater audio system, concert hall audio system, home audio system or a professional audio system, and the corresponding listening environment is part of a room cinema, a concert hall, a home, a studio, an auditorium or any other premises.
    [042] The proposed technology will now be described in greater detail with reference to several non-limiting exemplary modalities. Sound Field Control by Dynamic Linear Pre-compensation
    [043] Linear filters, systems or dynamic models that can have multiple inputs and / or multiple outputs are represented by function transfer matrices, as shown below, and are indicated by bold letters, such as, for example, H (q-1) or simply H. A special case of a function transfer matrix is a matrix that includes only FIR filters as elements. These matrices will be referred to as polynomial matrices and are indicated by capital letters in italics and bold, such as, for example, B (q-1) or simply B. In the present case, q-1 is a posterior displacement operator, which, when it operates a signal s (t) results in s (t-1), that is, q-1s (t) = s (t-1). Similarly, qs (t) = s (t + 1). When evaluating a polynomial or rotational matrix in the frequency domain, the complex variable (z) or (ejw) is replaced by (q). A causal matrix of FIR filters (polynomial matrix) B (q-1) operates only on input signals, which are present or have been passed with respect to the present time index (t). There will then be matrix elements that are polynomial only in the posterior displacement operator (q-1). Similarly, a polynomial matrix B (q, q-1) will operate with respect to future and past signals, while B (q) will operate only with respect to future signals. A superscript indicator (.) T, such as BT (q-1), or BT, means a matrix transpose and when used for a vector, a rotational or polynomial matrix means that a vector of the transposed row becomes a column vector, and the j-th row of a rotational or polynomial matrix becomes the j-th column of the same matrix. Similarly, a subscript indicator (.) * Means a conjugated complex transpose. This means that the vector, the rotational or polynomial matrix will be transposed, as explained above, and its elements will be complex or conjugated. Thus, for example, a complex transposed rotational matrix conjugated F (q-1) is indicated by F * (q). An identity matrix is a constant matrix with 1 digits diagonally. It is indicated by I or IN, if the dimension is N x N. Another constant matrix, for example, 0N, indicates a zero matrix of dimension N x N. In addition, diag ([F1 ... FN] T) indicates a diagonal matrix, with F1 ... FN diagonally, while trP indicates the trace of the matrix P, which is the sum of the diagonal elements of P.
    [044] The sound generation or reproduction system to be modified will be represented according to figure 2, by a time-invariant linear model and H, which describes the relationship in discrete time between a set of N input signals u (t) , set of M output signals modeled y (t):
    time (a unit sampling time is assumed, where, for example, (t + 1) means a sample time ahead of time t) and the sign y (t) is an M-dimensional column vector, representing the series of sound pressure time modeled in the M measurement positions. The operator H represents a model of the acoustic dynamic response, in the form of a function transfer matrix. This is a matrix of dimension M x N, whose elements are stable, linear and dynamic operators or transformers, for example, represented by FIR filters or IIR filters. These filters determine the response y (t) for an input vector u (t), N-dimensional, dependent on time. If the model H of M x N contains IIR filters as elements, then that model can be written in the so-called Matrix Fraction Description form on the right (MFD on the right), H (q-1) = B (q-1) A-1 (q-1) (2) where B (q-1) and A (q-1) are polynomial matrices of dimensions M x N and N x N, respectively [15]. The MFD form on the right, which will be used extensively in the description that follows includes the FIR filter matrix as a special case, by adjusting the denominator matrix for the identity matrix, that is, A = I.
    [045] The function transfer matrix H represents the effect of all or part of the sound generation system or sound reproduction system, including any pre-existing digital compensators, digital / analog converters, analog amplifiers, speakers , cables and the acoustic response of the environment. In other words, the function transfer matrix H represents the dynamic response of relevant parts of a sound generating system. The input signal u (t) for the system, which is an N-dimensional column vector, can represent input signals for N individual amplifier-speaker chains of the sound generating system. The signal ym (t) (where the subscript indicates “measurement”) is an M-dimensional column vector, representing the true (measured) sound time series in the M measurement positions and e (t) representing the background noise. , non-modeled reflections of the environment, effects of an incorrect model structure, non-linear distortion and other non-modeled contributions. Each M-dimensional column of H then represents the M transfer functions between one of the N speaker inputs and the M measurement positions.
    [046] Model H may also include uncertainties of an additive or multiplicative model, represented here by a rotational matrix ΔH. If, for example, the uncertainties of the ΔH model are parameterized by polynomial matrices with random coefficients, then a suitable model would be: H (q-1) = H0 (q-1) + ΔH (q-1) (3) where H0 (q-1) is the nominal model and ΔH (q-1) which is partially parameterized by random variables constitutes the model of uncertainties. Implementing the matrix fractions for H (q-1) and ΔH (q-1), the decomposition (3) of H (q-1) expands into:
    where Bo = BOAÍ, BÍ = BrA0, B = B0 + ΔBB1, and A = A0Ai. The matrices B0, ΔB and B are of dimension M x N, while B1, A0, A1 and A are of dimensions N x N. The matrices B0 and A0 refer to the nominal model H0 and the elements of ΔB are polynomial elements with variables stochastic as coefficients. For the sake of simplicity, we will assume that these coefficients have zero mean and unit variance. The B1A1-1 filter is used to model the spectral distribution of the stochastic uncertainty model. Also, it can be used to accommodate the variances of the different random coefficients of the unit. In the continuation, the denominators A0, A1 and A will, for reasons of simplicity, be assumed as if they are diagonal. If the system is represented as in (3), then H (q-1) can be viewed as a set of models, describing a range of possible errors in the measured response of the system. For a general introduction to the probabilistic modeling framework above, the reader should consult reference [27] and the respective references indicated there. The modeling of ΔH uncertainties can be performed in several ways, and the above formulation is just one example of how this can be achieved and used in a systematic way.
    [047] A general objective of sound field control is to modify the dynamics of the sound generating system, represented by (1), in relation to a reference dynamics. For this purpose, a reference matrix (or, in this case, a column vector) D of the dynamic systems is introduced:
    where a (t) is a signal that represents a live or recorded sound source, or even an artificially generated digital audio signal, including the test signals used to design the filter. The signal a (t) can, for example, represent a digitally recorded sound, or an analog source that has been sampled and digitized. In equation (5), the matrix D is a supposedly known function transferable column vector, of dimension M x 1. The linear discrete-time dynamic system must be specified by the designer. It represents the reference dynamics (desired target dynamics) of the vector y (t) in (1). In the compensated system, the signal a (t) will represent a signal outside the L fully input source signals. Their desired effect on the M measurement positions is represented by the elements D1, ..., DM of D in (5). System D can include a set of adjustable parameters. Alternatively, the system can be indirectly affected by this set, through its specification.
    [048] The audio pre-compensation controller is supposed to be implemented as a discrete, dynamic and multivariable time pre-compensation filter, generally referred to by R, which generates an input signal vector u (t) for the audio reproduction system (1), based on the dynamic linear processing of the signal a (t):

    [049] The present audio pre-compensation controller includes a set of adjustable parameters. These parameters must allow sufficient flexibility, in order to modify the dynamic input-output properties of the controller, for example, allowing some elements of R, or the integrity of R, to be equal to zero, for appropriate parameter adjustments. The optimization of R, however, should be restricted to parameter settings that make R a stable and dynamic input-output system.
    [050] The objective pursued by the present inventors is to build a stable transfer matrix of function R, of dimensions N x 1, designed to generate an input signal vector u (t) for the audio reproduction system (1) , so that its compensated model output y (t) also approaches the reference vector yref (t), according to a specified criterion. This objective can be achieved if: y (t) = H u (t) = HR a (t) = yref (t) = D a (t) (7)
    [051] The corresponding approximation error based on the model in the M measurement positions is represented by: ε (t) = yref (t) - y (t) = (D-HR) to (t) (8)
    [052] The true measured error vector will then be, according to figures 2 and 1, yref (t) - ym (t) = ε (t) - e (t). The approach (7) will never be exact in practice with a limited number of N speakers, a large number of M measurement positions and complicated dynamic broadband acoustic models in H. The quality of approach that can be obtained depends on the nature of the established problem. For a given fixed acoustic environment, the quality of the approach can, in general, be improved if the number of speaker channels N is increased. Quality can also be improved by increasing the number of measurement points M, within the intended listening region, as this provides a denser and more accurate sampling of the sound field as a function of space. The enlargement of the hearing region or the addition of regions to a fixed N number could, in general, result in greater approximation errors.
    [053] A scheme for calculating an appropriate approximation to the present problem will be highlighted below.
    [054] An important aspect to consider when designing a pre-compensation device is the relationship between the delay of the initial propagation of the system to be compensated and the delay of the initial propagation of the desired target dynamics. The delay in the initial propagation of a dynamic system is the time it takes for a signal to propagate, from the input to the output of the system. In other words, the delay of the initial propagation is provided by the time of the first non-zero coefficient of the impulse response of the system. A system H having an initial propagation delay of d samples can therefore be written as π = q ~ dH where at least one of the elements of JC has an impulse response that begins with a non-zero coefficient.
    [055] Next, the system of figure 2 will be considered, for example, and it is assumed that H has an initial propagation delay (d1), and D an initial propagation delay (d0). If (d1)> (d0), then, a causal compensator R, which uses only present and past values from a (t) cannot expect a satisfactory execution, because in time (t), the reference signal yref (t) will depend on the signal values a (t - d0 - k) for k> 0, while the output y (t) of the compensated system depends only on a (t - d1 - k), for k> 0, this that is, the reference signal depends on more recent data than that produced at the output of the system. The compensating device aims to direct y (t) in the direction of the reference signal yref (t), but, due to the time delay difference between H and D, the action of the control signal u (t) at the output of H will always get samples at least once, after what is needed. In order for the compensating device R to have a satisfactory performance in this case, it should be of the non-causal type, that is, it should be able to predict at least future values of the di - dence of the signal a (t). If the relationship between the initial delays is the opposite, that is, if (di) <(do), then the compensating device will have a much more satisfactory performance, due to the fact that, through the knowledge of D and a (t) o compensator will be able to predict future values of the reference signal. Therefore, the compensator can start to act on the H dynamics through the do - di samples in advance, in such a way that the output y (t) is more effectively directed in the direction of the reference signal yref (t). [o56] Therefore, in general, it is possible to improve the performance of a pre-compensator, by ensuring that the initial delay of target dynamics D is sufficiently large, compared to the initial delay of system H. Thus, for example, this can be obtained by adding a massive global delay q-do to the target, so that D = q-dl ^ D where D is the original intended target dynamics, and do is greater than or equal to the initial propagation delay of H. [o57] However, for the purposes of audio reproduction, allowing a large, massive q-do delay on the target is problematic. On the one hand, it is generally true that a large voluminous delay in the target dynamics helps to reduce the average reproduction error, for example: E {ll yref (t) -y (t) Hi}. On the other hand, as described above, a large voluminous delay in the target allows the compensator to act in the system in a predictable way, that is, the output y (t) may depend on the data of a (t), which are found “in the future ”, Compared to the data that make up the signal yref (t). Since the reproduction error yref (t) - y (t) is not necessarily zero, this prognostic behavior can cause errors that are perceived as pre-touches or pre-echoes in the compensated system. Technically, this means that the compensated system's impulse response contains sound energy that arrives before the intended voluminous delay (d0). Especially for impulsive and transient sounds, these pre-touch errors are perceived by humans as markedly unusual and irritating and, therefore, should be avoided, if possible. In the example above, the length of the time interval in which the pre-touch errors can occur is determined by the difference between the initial propagation delays of H and D. Therefore, it is of interest to use a large delay that is large enough to allow the compensator to function properly, but not so large that it causes the compensator to produce audible pre-touch errors. In other words, to minimize the effects of pre-touches you should use di ^ in the example above, with di as close to d0 as possible.
    [056] Therefore, in general, it is possible to improve the performance of a pre-compensator, by ensuring that the initial delay of target dynamics D is sufficiently large, compared to the initial delay of system H. Thus, for example, this can be obtained by adding a massive global delay q-d0 to the target, so that
    权利要求:
    Claims (20)
    [0001]
    1. Method for determining an audio pre-compensation controller for an associated sound generating system, comprising a total of N> 2 speakers, each having a speaker input, the audio pre-compensation controller having a number of L> 1 inputs for L signal or input signals and N outputs for N controller output signals, one for each speaker of the sound generating system, the audio pre-compensation controller having a number of adjustable filter parameters, the method characterized by the fact that it comprises the steps of: estimating (S1), for each of at least a subset of the N speaker inputs, an impulse response in each of a plurality M> 2 of measurement positions, distributed in a region of interest in a listening environment, based on sound measurements in the M measurement positions; specify (S2), for each of the one or more L input signals, a speaker selected from the N speakers as a primary speaker, and a selected subset S including at least one of the N speakers as one or more supporting speakers, where the primary speaker is not part of the subset; specify (S3), for each primary speaker a target impulse response in each of the M measurement positions with the target impulse response having an acoustic propagation delay, where the acoustic propagation delay is determined based on the distance from from the primary speaker to the respective measurement position; determine (S4), for each of the one or more L input signals, based on the selected primary speaker and the selected support speaker (s), filter parameters of the audio pre-compensation controller, of so that a criterion function is optimized under the dynamics stability constraint of the audio pre-compensation controller, with the criterion function including a weighted sum of powers of differences between compensated estimated impulse responses defined by the estimated compensated impulse responses by the controller audio pre-compensation and target impulse responses on the M measurement positions.
    [0002]
    2. Method, according to claim 1, characterized by the fact that L> 2 and the method comprises the step of merging (S5) all the filter parameters, determined for the L input signals, into a fused set of parameters filter for the audio pre-compensation controller, where the audio pre-compensation controller with the fused set of filter parameters is configured to operate on the L input signals to generate the N controller output signals for the speakers to get the target impulse responses.
    [0003]
    3. Method, according to claims 1 or 2, characterized by the fact that the audio pre-compensation controller is configured to control the acoustic response of P primary speakers, where P <L and P <N, by means of the combined use of the P primary speakers and, for each primary speaker, an additional number of supporting speakers 1 <S <N - 1 of the N speakers.
    [0004]
    4. Method according to any one of claims 1 to 3, characterized by the fact that the audio pre-compensation controller has the ability to produce zero output for some of the N speakers for some adjustments of their adjustable parameters of filter.
    [0005]
    5. Method according to any one of claims 1 to 4, characterized by the fact that the step (S4) of determining filter parameters of the audio pre-compensation controller is based on a Linear Quadratic Gaussian (LQG) optimization of the parameters of a stable, linear controller and multivariable causal direct power, based on a specific target dynamic system and on a dynamic model of the sound generating system.
    [0006]
    6. Method according to any one of claims 1 to 5, characterized in that each of the N controller output signals of the audio pre-compensation controller is fed to a respective speaker through a pass filter -everything, including a phase compensation component and a delay component, producing N filtered controller output signals.
    [0007]
    7. Method according to any one of claims 1 to 6, characterized by the fact that the criterion function includes penalty terms, in which the penalty terms are such that the audio pre-compensation controller, obtained through optimization of the criterion function, produces signal levels of restricted magnitude in a selected subset of the pre-compensation controller outputs, producing levels of restricted signals at selected speaker inputs for the N speakers for certain frequency ranges.
    [0008]
    8. Method according to claim 7, characterized by the fact that the penalty terms are chosen differently in a number of times, and the step of determining the filter parameters of the audio pre-compensation controller is repeated for each choice of penalty terms, resulting in a number of occurrences of the audio pre-compensation controller, each producing signal levels with magnitudes individually restricted to the supporting S speakers for certain frequency ranges.
    [0009]
    9. Method according to any one of claims 1 to 8, characterized by the fact that the criterion function includes, first, a set of models describing a range of possible errors in the estimated impulse responses and, second, a aggregation operation, where the aggregation operation is a sum, a weighted sum or a statistical expectation about the set of models.
    [0010]
    10. Method according to any one of claims 1 to 9, characterized in that the step (S4) of determining the filter parameters of the audio pre-compensation controller is also based on adjusting the filter parameters of the controller audio pre-compensation to obtain a target frequency magnitude response from the sound generating system, including the audio pre-compensation controller, in at least a subset of the M measurement positions.
    [0011]
    11. Method according to any one of claims 1 to 10, characterized by the fact that the target impulse responses are different from zero and include adjustable parameters that can be modified within established limits.
    [0012]
    12. Method, according to any of the preceding claims, characterized by the fact that the estimation step (S1) for each of at least a subset of the N speaker inputs, an impulse response in each of a plurality M of measurement positions is based on a model that describes the dynamic response of the sound generating system in the M measurement positions.
    [0013]
    13. Method, according to claim 1, characterized by the fact that the audio pre-compensation controller is created by implementing the filter parameters in an audio filter structure.
    [0014]
    14. Method, according to claim 13, characterized by the fact that the audio filter structure is incorporated in the sound generator system to enable generation of the target impulse response in the M measurement positions in the listening environment.
    [0015]
    15. System (100) for determining an audio pre-compensation controller (200) for an associated sound generating system comprising a total of N> 2 speakers, each having a speaker input, the audio controller audio pre-compensation (200) having a number of L> 1 inputs for L input signal (s) and N outputs for N controller output signals, one for each speaker of the sound generator system, the audio controller audio pre-compensation having a number of adjustable filter parameters, in which the system (100) is characterized by the fact that it comprises: - means for estimating, for each of at least a subset of the N speaker inputs, an impulse response in each of a plurality of M> 2 measurement positions, distributed in a region of interest in a listening environment, based on the sound measurements in the M measurement positions; - means for specifying, for each of the one or more L input signals, a speaker selected from the N speakers as a primary speaker, and a selected subset S including at least one of the N speakers as one or more supporting speakers, where the primary speaker is not part of the subset; - means for specifying, for each primary speaker, a target impulse response in each of the M measurement positions with the target impulse response showing an acoustic propagation delay, in which the acoustic propagation delay is determined based on the distance a from the primary speaker to the respective measurement position; - means for determining, for each of the one or more L input signals, based on the selected primary speaker and the one or more selected support speakers, filter parameters of the audio pre-compensation controller so that a criterion function is optimized under the dynamic stability constraint of the audio pre-compensation controller, with the criterion function including a weighted sum of powers of differences between the estimated compensated impulse responses defined by the estimated compensated impulse responses by the controller audio pre-compensation and target impulse responses on the M measurement positions.
    [0016]
    16. System according to claim 15, characterized by the fact that L> 2, and the system (100) comprises means for merging all filter parameters, determined for the L controller input signals, into a fused set of filter parameters for the audio pre-compensation controller, where the audio pre-compensation controller with the fused set of filter parameters is configured to operate on the L input signals, to generate the N output signals controller to the loudspeakers to obtain the target impulse responses.
    [0017]
    17. System according to claims 15 or 16, characterized by the fact that the means for determining filter parameters of the audio pre-compensation controller are configured to operate based on a Gaussian Linear Quadratic Optimization (LQG) of the parameters of a stable, linear and direct causal multivariable controller, based on a specific dynamic target system and on a dynamic model of the sound generating system.
    [0018]
    18. Computer-readable memory characterized by the fact that the method as defined in any one of claims 1 to 14 is recorded in it.
    [0019]
    19. Audio pre-compensation controller (200), characterized by the fact that it is determined by the use of the method as defined in any of claims 1 to 14.
    [0020]
    20. Audio system comprising a sound generator system and an audio pre-compensation controller (200) on the input path to the sound generator system, characterized by the fact that the audio pre-compensation controller is determined by the use of the method of agreement as defined in any one of claims 1 to 14.
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    法律状态:
    2018-12-18| B06F| Objections, documents and/or translations needed after an examination request according art. 34 industrial property law|
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    2021-01-05| B09A| Decision: intention to grant|
    2021-03-16| B16A| Patent or certificate of addition of invention granted|Free format text: PRAZO DE VALIDADE: 20 (VINTE) ANOS CONTADOS A PARTIR DE 22/03/2012, OBSERVADAS AS CONDICOES LEGAIS. |
    优先权:
    申请号 | 申请日 | 专利标题
    PCT/SE2012/050320|WO2013141768A1|2012-03-22|2012-03-22|Audio precompensation controller design using a variable set of support loudspeakers|
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