Patent Description:
The present disclosure is generally related to audio systems with limited array signals.

<CIT> discloses a system for improving low frequency output of small loudspeakers without causing distortion or damaging the loudspeaker.

The present invention relates to an audio system according to claim <NUM>. Advantageous embodiments are recited in dependent claims of the appended claim set.

Other features, objects, and advantages will be apparent from the description and the drawings, and from the claims.

In an audio system with an array of speakers, each speaker will typically have an associated limiter designed to prevent the signal input to the speaker from exceeding a predetermined threshold. The predetermined threshold is normally set to prevent the input signal from overdriving the speaker. Because each speaker input (e.g., each output channel from the audio system) will have a respective gain, once a limiter limits the input to one speaker of the speaker array-the remaining speakers of the speaker array not being similarly limited-the relative gains of the speakers will be altered, undesirably "breaking" the speaker array. There exists a need in the art for limiting the signal input to the speaker array while preserving the relative gains of the speaker array when a particular input signal exceeds a limiter threshold.

<FIG> shows a block diagram of an example audio system <NUM> that includes a filter <NUM>, a limiter <NUM>, and a speaker array <NUM>. In the example shown, an input signal is received at the filter <NUM>. The input signal may be from at least one sensor, such as a microphone, or other audio source, such as the output of a noise-cancellation system, and may be comprised of multiple input signals from a variety of sources, e.g., multiple microphones, multiple audio sources, etc..

In this example shown, the filter <NUM> performs two functions: expanding the input signal into multiple output channels and applying equalization filtering to the respective output channels in order to tune the output signals for a particular desired speaker array output. The equalization filtering will apply some gain to each output channel, adjusting the magnitude and phase of the input signal as a function of frequency. The magnitude and phase response of the equalization filtering will typically vary from output channel to output channel in order to achieve the desired audio mix within the predefined volume-although it is conceivable that the same equalization filtering will be applied two or more output channels. For example, one output channel may have reduced or increased gain in one frequency band and another output signal may have reduced or increased gain in another frequency band. Filter <NUM> will thus receive an input signal and output multiple filtered output signals from multiple channels (the output signals being the input signals having some filtering applied to them). Further, it should be understood that filter may receive more than one input signal, and may expand each received input signal into multiple channels (as shown, for example, in <FIG>).

Filter <NUM> may be implemented as a single filter or a combination of filters. In an example, filter <NUM> may be an finite impulse response (FIR) filters, infinite impulse response (IIR) filters, or some combination of the two.

As mentioned, each output signal from filter <NUM> may be directed to a respective individual speaker of a speaker array <NUM> (the speaker array, in <FIG>, is represented as a single speaker for the sake of simplicity). The speakers of the speaker array <NUM> may be arranged within a predefined volume, such as, for example, the cabin of a vehicle. Each speaker receives the respective output signal and transduces it to an audio signal that is transmitted through the predefined space. Although speakers are referred to in this disclosure, it should generally be understood that each speaker may be implemented as any type of acoustic transducer.

The output signals may be output to limiter <NUM>, disposed between the speaker array <NUM> and the filter <NUM>, which limits each respective output signal to a particular threshold. More specifically, each limiter <NUM> will apply some value of gain reduction to attenuate the input signal, such that the signal will not exceed the threshold. The threshold may be defined to avoid overdriving each respective speaker of speaker array <NUM>.

As shown in <FIG>, filter may be conceptualized as a combination of two filters: an array filter <NUM> and equalization filter <NUM>. In an example, the array filter <NUM> receives the input signal and expands it two or more output channels. In various embodiments, the array filter may also add some gain and/or filtering to each channel. The equalization filter <NUM> adjusts the magnitude and phase of each channel as a function of frequency in order to tune the output to the speaker array <NUM>, as described above.

One implementation of audio system <NUM> is shown in <FIG>. In this example, array filter <NUM> is depicted expanding the input signal into two channels. Although two channels are depicted in the figures, the array filter <NUM> may expand the input signal to any number of channels. The equalization filters <NUM>, here shown as each being associated with a respective array filter output channel, adjust the magnitude and phase of the associated output signal as a function of frequency and delay. Each equalization filter <NUM> outputs a signal to an associated limiter <NUM> and speaker of the speaker array <NUM>.

Each respective limiter <NUM> will limit the magnitude of the output of the associated equalization filter <NUM> to within a threshold value.

The audio system of <FIG> and any of the audio systems described in this disclosure may be implemented with a nontransitory storage medium and processor. In an embodiment, the non-transitory storage medium may store program code that, when executed by processor, implements the various filters, limiters, compressors, etc. Alternatively the audio systems may be implemented hardware or firmware, or any combination of hardware, firmware, or software, as will be understood by a person of ordinary skill in the in conjunction with a review of this disclosure. For example, any audio system may be implemented by an FPGA, an ASIC, or other suitable hardware.

As described above, the output signals of filter <NUM> may, by design, have different magnitudes and phases due to the equalization filtering. As a result, one or more limiters <NUM> may reach the threshold before the others, so that a subset of the output signals are attenuated while the other output signal remains at the gain set by the equalization filter <NUM>, altering the gain difference between the output signals, and, consequently, "breaking" the array output. To avoid limiting a subset of the output signals, the input signal may, instead, be limited. By limiting the input signal, each filter output signal will be reduced by the same gain, and thus the array is maintained.

This may be accomplished, for example, through the example audio-system <NUM> shown in <FIG>. In this example, filter <NUM> is duplicated in a parallel side-chain. The duplicate filter <NUM>, which is identical in processing to filter <NUM>, receives and processes the input signal while the input to filter <NUM> is delayed by a delay <NUM>. The output signals of the duplicate filter are received at a level detector <NUM> which determines whether any of the output signals exceed a predetermined threshold. The threshold of the level detector <NUM> may be selected to be comparable to the magnitude at which output limiters would be set, e.g., in <FIG>. The level detector <NUM> is in communication with input limiter <NUM>, which applies the gain necessary to limit the magnitude of the input signal (having now passed the delay block) by at least the maximum amount that the output signals exceeded the threshold of the level detector. (The level detector, may, for example, set the threshold of the limiter such that appropriate gain reduction is applied to the input signal. ) To the extent that two or more filter <NUM> output signals exceed the threshold at once, the limiter <NUM> gain is set according to the magnitude of the greater output signal. Stated differently, the gain reduction of the limiter <NUM> is determined by the maximum magnitude of all received filter <NUM> output signals.

Reducing the input signal by the magnitude that the output signal of filter <NUM> exceeds the threshold effectively lowers the gain of each output signal of filter <NUM> such that no single output signal of filter <NUM> exceeds the threshold of an output limiter (as shown, for example, in <FIG>). This obviates the need for output limiters, since the input signal has been limited at the input.

An expanded example of <FIG> is shown in <FIG>, depicting constituent filters of filter <NUM> and duplicate filter <NUM>: specifically, duplicate array filter <NUM> and duplicate equalization filters <NUM> reproduce the response of filter <NUM>, and output two (or more) output signals to level detector <NUM>.

The audio system <NUM> described in connection with <FIG> and <FIG>, however, is computationally expensive, as it requires duplicating filter and, thus, performing the filtering twice. To avoid the computational complexity, various lower-cost audio systems may be implemented.

An example of such a lower-cost audio system is shown in <FIG>. This example is similar to the example of <FIG>, however each output limiter <NUM> is coupled together, such that, when one limiter <NUM> exceeds the threshold and applies a gain reduction, the remaining limiters <NUM> apply the same gain reduction, maintaining the gain difference between the output channels. If multiple limiter <NUM> thresholds are exceeded at once, then all limiters <NUM> implement the maximum gain reduction implemented by any of the limiters. Each limiter <NUM> thus applies the greater of a gain based on the magnitude the received output signal of filter <NUM> exceeds a predetermined threshold or the gain applied by another limiter. In this way, the maximum gain applied by any limiter <NUM> is applied by all limiters <NUM>.

This solution, however, suffers if any of the output signals of filter <NUM> are offset in time with respect to the others. Take, for example, a two-channel filter <NUM>, one output channel of filter that has high gain and no delay and another output channel that has lower gain but is delayed by one sample. When the output channel with higher gain exceeds its threshold, by, for example, <NUM> dB, then a <NUM> dB reduction is applied to both channels. But because of the delay, the reduction for the delayed channel needs to be also delayed by one sample. However, because the delay is likely frequency-dependent, the array may still not be maintained.

To account for this, the limiters <NUM> may be configured to have a long release time (i.e., the period the limiter is increasing gain back to unity). The long release time ensures that all delays in any given channel are fully resolved before the limiters stop applying gain. The release time may be set, for example, to ensure that <NUM> after the peak signal, the gain has decayed toward unity by no more than 1dB, where <NUM> is substantially more time than the delay in the filters. A long release time is not necessarily ideal, however, because the output channels will likely be limited, attenuating the array <NUM> output audio signal, for a longer period than necessary.

An embodiment according to the invention, audio system <NUM>, is shown in <FIG>. This example is similar to the look-ahead design of the ideal solution depicted in FIG. 3A-3B, however, approximation circuitry <NUM> has been substituted in place of the duplicate filter <NUM>. The approximation circuitry <NUM> approximates the filtering performed by the filter <NUM>, but, in certain examples, demands fewer processing resources than the duplicate filter <NUM> and is thus more computationally efficient.

In the simplest example, the approximation circuitry <NUM> may apply a gain (e.g., with an amplifier, although other methods of applying a gain are contemplated) substantially equal (e.g., within <NUM> dB) to the maximum gain applied by filter <NUM>. To be effective, the maximum gain is the largest gain applied between the output signal of filter <NUM> and the input signal received after the limiter <NUM>. This solution will not approximate any frequency or phase characteristics of the filter <NUM>, but will still largely approximate the maximum output magnitude of filter <NUM>.

In more complex variants, a relatively cheap infinite-impulse-response (IIR) filter could be employed for approximation circuitry <NUM> to approximate the characteristics of filter <NUM> (here, "relatively cheap" means requiring less computational resources than filter). This IIR filter could, for example, be a minimum-phase filter that would capture the only magnitude response of the filter <NUM>, but not the phase response, which generally presents the greatest computational complexity. Alternatively, an IIR filter that approximates the magnitude and phase response of the filter <NUM> may be used. To avoid adding too much complexity, the IIR filter could be limited in the number of stages implemented.

Approximation circuit <NUM> may also comprise an FIR filter. In an example, the FIR filter may be a minimum-phase filter that, like the minimum-phase IIR implementation, captures only the magnitude response of filter <NUM>. Alternately, the FIR may be implemented in a manner substantially the same as filter <NUM> (and thus the same as duplicate filter <NUM>). The FIR filter implementation may thus require fewer or the same amount of computational resources as filter <NUM>.

An example, according to the invention, expanded version of <FIG> is provided in <FIG>. As shown, equalization filters <NUM> are approximated with computationally-cheaper approximating IIR filters <NUM>. Similarly, approximating array filter <NUM> may be implemented with cheaper IIR filters, thus reducing the computation demands made by audio system <NUM>.

To simplify the IIR filter (minimum-phase or otherwise) only the filtering of the output signal of filter <NUM> having the greatest applied gain by filter <NUM> may be approximated. Stated differently, if the gain of the filtering applied to each output signal of filter <NUM> is larger for one output signal with respect to the others, only the filtering applied to that output signal may be approximated, since that output signal will likely exceed the threshold of an output limiter <NUM> before the others.

Because an approximation circuit <NUM> is used, the output signals may still overdrive the speaker array <NUM> in instances where the approximation fails to accurately capture the magnitude of the output signal. As a result, an output limiter <NUM> may still be used to limit such instances and prevent overdriving the speaker array <NUM>. Of course, this means that, in such instances, the array would not be maintained. The quality of the approximation will dictate the frequency that the output limiter <NUM> is met.

Variations of audio system <NUM> are contemplated by this disclosure. For example, rather than duplicating array filter <NUM>, array filter <NUM> may be disposed in front of the delay <NUM>, thus providing duplicate input signals to limiter <NUM> and to the approximation circuitry <NUM>. If the limiter <NUM> applies a gain reduction, it will be applied to both input signals equally, which are output to filter <NUM>.

A third example is shown in <FIG>. In this example, the threshold of the input limiter <NUM> is set according to feedback <NUM> received from the filter <NUM> at the level detector <NUM>. The input limiter <NUM> applies the gain necessary to limit the magnitude of the input signal by at least the maximum amount that the output signals of filter <NUM> exceeded the threshold of the level detector <NUM>. This example avoids processing-heavy side chains by setting the gain of input limiter <NUM> according to the actual output of filter <NUM>.

This example, however, fails to catch rapid transients, as these will be output to the speakers <NUM> before the threshold of the limiter <NUM> can be set/changed. (The transients that will not be limited at the input by this example will be determined, at least in part, by the propagation delay through the filter <NUM> and feedback loop <NUM>.

To mitigate this, gain <NUM> may be added to the feedback loop <NUM> (e.g., with an amplifier disposed between the filter output and the level detector), artificially increasing the output of filter <NUM>, resulting in an earlier gain reduction by the input limiter <NUM>. Adding gain to the feedback loop <NUM>, however, will result in applying limiting more than often necessary. Alternatively, and to the same result, the threshold of the limiter <NUM> may be lowered by a predetermined amount.

An example expanded version of is depicted in <FIG>. As shown, the output signals from each output channel are provided to the level detector <NUM>, which determines whether the magnitude of either output signal exceeds the threshold value. The limiter <NUM> then applies gain based on the amount the magnitude of either output signal exceeds the threshold. If both output signals exceed the threshold, the gain of limiter <NUM> is set according to the amount the greater of the two output signals exceeds the threshold.

Any of the above-described examples may be combined together into a single solution, as shown in, for example, in <FIG>. In this example, the coupled output limiters described in connection with <FIG>, the low-cost approximation look-ahead filter described in connection <FIG>, and the feedback loop described in connection with <FIG> have been combined into a single system. Because both of the approximation circuit <NUM> and the filter <NUM> both output a signal to the level detector <NUM>, the greatest magnitude signal from either approximation circuit <NUM> or filter <NUM> may be used to set the gain of limiter <NUM>. It should be recognized that any two of the above-described audio systems may be combined together, rather than all three.

In addition, of the above-described solutions may be configured for multiband limiting. Multiband limiting employs multiple limiters, each receiving a particular frequency band of signal and is most useful when one frequency or band of frequencies receives more gain than others. For example, if a filter has high gain at high frequencies and low gain at low frequencies, a wideband input signal will reach the threshold of the limiter due to the high gain of the high frequencies, even though the low frequency filter output was not above the limiter threshold. To mitigate this, the filter outputs are divided across different frequency bands and only one band is input to a particular limiter. The frequency bands may be selected according the particular filter characteristics (e.g., grouped into high gain frequency bands and low or lower gain frequency bands). The granularity offered by the number of frequency bands and limiters employed may be balanced against the additional computational complexity added by duplicated limiters/filters.

Multi-band limiting may be implemented in a number of ways. For example, as shown in <FIG> the filtering and limiting any of the above-described audio systems (shown collectively in <FIG> as filtering and limiting block <NUM>) may be duplicated N-times (e.g., three times), each receiving a different frequency band (there may be some overlap between the bands), the outputs of each duplicated audio system being summed together and output to speaker array <NUM>. For example, as shown in <FIG>, a low-pass filter <NUM> may be employed at the input to one chain, to limit that audio system to one band, while a high-pass filter <NUM> may be employed at the input to another chain. It should be understood that the examples of low-pass <NUM> and high-pass filter <NUM> as used here (and in <FIG>) are merely provided as examples-the shape and cut-off frequency (or frequencies) of the input filters may be selected to achieve any desired multi-band limiting.

However, the example of <FIG> requires duplicating the filter <NUM> multiple times, which, as described above, is computationally expensive. Accordingly, the limiters <NUM>, <NUM> of filtering and limiting <NUM> rather than the limiters <NUM>, <NUM>, <NUM> and filter <NUM>, may be duplicated and the inputs/feedback signals to the duplicated limiters <NUM>, <NUM>, <NUM> may be constrained to certain frequency bands.

For example, <FIG> shows an example, audio system <NUM>, of the filtering and limiting of <FIG> modified for multi-band limiting. As shown, filter <NUM> is not duplicated. Instead, each output channel is respectively input to a high-pass <NUM> and low-pass filter <NUM> (again, these filter types are merely provided as examples) each output from a high-pass <NUM> and low-pass filter <NUM> is fed to a respective limiter <NUM>. Each limiter <NUM> associated with a particular filter output channel is summed together. Each limiter <NUM> associated with a particular frequency band is coupled together such that the same gain is applied to each. For example, if one limiter <NUM> receiving high-frequency outputs exceeds its threshold and applies some gain, the other limiter <NUM> receiving high-frequency output (of another channel) may apply the same gain in order to maintain the array, but the limiters <NUM> receiving the low-frequency outputs will not apply the same gain (unless, of course, one of their respective thresholds is met as well).

<FIG> shows an an embodiment according to the invention, audio system <NUM>, of <FIG> modified for multi-band limiting as well. Again, the filter <NUM> is not duplicated, rather, the input signal is passed through a high pass filter <NUM> and low pass filter <NUM>, which are situated in parallel. The output of the high pass filter <NUM> and low pass filter <NUM> are respectively input to duplicated side-chains, which compute whether the approximated signal for the particular band of frequencies exceeds the predetermined threshold and sets the gain of a respective duplicated limiter <NUM> accordingly. The limiters <NUM> each respectively receive the output of the input filter, and thus limit only the respective band for which their gain has been adjusted. The outputs of the limiters <NUM> are summed together and input to filter <NUM>.

<FIG> shows an example, audio system <NUM>, of <FIG> modified for multi-band limiting. As shown, the input signal is received through one of input high-pass filter <NUM> or low pass filter <NUM> at the duplicated limiters <NUM>. The gain applied by the limiters <NUM> is set by the feedback signal from filter <NUM>, the feedback signal also being filtered (respectively by feedback high-pass filter <NUM> and low-pass filter <NUM>) according to the input signal frequency band limiter <NUM> receives. This ensures that the limiter <NUM> will not respond to the magnitude of frequencies outside of the particular frequency band of the input signal the limiter receives at the input. The outputs of the duplicated limiters <NUM> are summed together and input to filter <NUM>.

The above topologies depicted and described for multi-band limiting are merely provided as examples; each audio system could be modified for multi-band limiting differently in other examples. Further, each example may be further expanded for limiting any number of frequency bands: the two bands are merely provided as examples. Also, in each of the above multi-band limiting examples, there may be some overlap between the various frequency bands, but the bands are generally not coextensive.

As shown in <FIG>, and as mentioned above, filter may receive multiple inputs and expand each input into two or more channels, each channel being input to a respective speaker of the speaker array. This, however, may result in a configuration in which a limiter <NUM>, associated with one speaker, is more sensitive to one input signal than the other. For example, if, in the example of <FIG>, one filter <NUM> applies a maximum gain factor of two, while the remaining filters <NUM> apply a maximum gain factor of one (these gain factors may, for example, be frequency and delay dependent), the limiter <NUM> receiving the signal from the filter <NUM> with the gain factor of two is more likely to reach the limiter threshold as a result of one input signal than the other.

To compensate for this, coupled compressors <NUM> may be added to each respective input signal. Each compressor <NUM> will apply some compression (e.g., attenuate the magnitude of the signal) to the respective input signal. The compression curve may be tailored so that an input signal does not reach the limiter <NUM> threshold as frequently. The compressors <NUM> may each be configured to apply the same compression, so that the relative gains of the output channels of filter <NUM> are not affected by the compression.

Although two compressors <NUM> are shown in <FIG>, each compressing a respective input signal, it should be understood that any number of compressors <NUM> may be associated with any number of inputs. In an example, a single input signal may be compressed by a single compressor <NUM>. Furthermore, in an alternate example (when multiple compressors <NUM> are used with multiple respective inputs), each compressor <NUM> may respectively apply different compression curves, rather than the same compression curve to each respective input signal.

The functionality described herein, or portions thereof, and its various modifications (hereinafter "the functions") can be implemented, at least in part, via a computer program product, e.g., a computer program tangibly embodied in an information carrier, such as one or more non-transitory machine-readable media or storage device, for execution by, or to control the operation of, one or more data processing apparatus, e.g., a programmable processor, a computer, multiple computers, and/or programmable logic components.

Actions associated with implementing all or part of the functions can be performed by one or more programmable processors executing one or more computer programs to perform the functions of the calibration process. All or part of the functions can be implemented as, special purpose logic circuitry, e.g., an FPGA and/or an ASIC (application-specific integrated circuit).

Components of a computer include a processor for executing instructions and one or more memory devices for storing instructions and data.

Claim 1:
An audio system (<NUM>), comprising:
at least one approximating circuit (<NUM>), the at least one approximating circuit configured to apply an approximated gain to an input signal, the at least one approximating circuit outputting an approximated signal;
a limiter (<NUM>) configured to receive the delayed input signal and to reduce a magnitude of the delayed input signal by a gain determined by an amount the magnitude of the approximated signal exceeds a predetermined threshold, the limiter outputting a limited input signal;
a filter (<NUM>) configured to receive the limited input signal and to respectively apply a filter gain, the filter outputting a plurality of filtered signals, wherein the approximated gain is substantially similar to the filter gain and the approximating circuit is configured to approximate the filtering performed by the filter, in a way that demands fewer processing resources and thus is more computationally efficient; and
a speaker array (<NUM>) comprising a plurality of speakers, each speaker configured to receive a respective filtered signal of the plurality of filtered signals and to transduce the respective filtered signal to an audio signal.