TECHNOLOGICAL FIELD

Examples of the disclosure relate to a loudspeaker system and a calibration method for the loudspeaker system. Some relate to the loudspeaker system defining an omnidirectional speaker for acoustic assessments.

BACKGROUND

Some environments, such as buildings and auditoria, are required to satisfy certain acoustic performance requirements.

Compliance with these requirements may be assessed in an objective manner. For example, speech quality is assessed via a Speech Transmission Index (STI) score which is a measure of speech intelligibility in an environment.

During an assessment, the assessor places an omnidirectional speaker at a representative sound source location, to produce a reference sound. The assessor places microphones at representative listening locations, to record the resulting sound.

The assessor evaluates the recordings to determine whether the acoustic performance requirements have been satisfied.

It is important that the omnidirectional speaker renders a ‘perfect sphere’ of sound across the required frequency range, to enable a proper acoustic characterization of the environment. Designing such a speaker is challenging and has been the subject of research. Hardware solutions have focused on reducing the sizes of the electroacoustic transducers (audio transducers) of the omnidirectional speaker, to maximise coupling between adjacent audio transducers and therefore minimise comb filtering.

BRIEF SUMMARY According to an aspect of the invention, there is provided a loudspeaker system comprising a loudspeaker enclosure, a plurality of audio transducers secured to the loudspeaker enclosure and each facing in a different direction, the loudspeaker system further comprising an audio signal processor configured to apply decorrelation filters to decorrelate signals output by the plurality of audio transducers.

According to a further aspect of the invention, there is provided a calibration method for a loudspeaker system, the loudspeaker system comprising a loudspeaker enclosure, a plurality of audio transducers secured to the loudspeaker enclosure and each facing in a different direction, the loudspeaker system further comprising an audio signal processor, the calibration method comprising: determining impulse responses of the audio transducers to input signals to which a set of decorrelation filters have been applied, each impulse response associated with a different angular position ‘m’ defined by a microphone apparatus and the loudspeaker enclosure; determining whether a performance metric condition is satisfied, in dependence on differences between the impulse responses for the different angular positions; in dependence on the performance metric condition not being satisfied, modifying the set of decorrelation filters and then repeating the above steps of determining impulse responses and determining whether the performance metric condition is satisfied; and in dependence on the performance metric condition being satisfied, providing the set of decorrelation filters for implementation in the audio signal processor of the loudspeaker system.

According to a further aspect of the invention, there is provided an apparatus comprising: at least one processor such as an audio signal processor; and at least one memory including computer program code the at least one memory and the computer program code configured to, with the at least one processor, cause the apparatus at least to perform the calibration method. According to a further aspect of the invention, there is provided a computer program that, when run on a computer, performs the calibration method.

BRIEF DESCRIPTION

Some examples will now be described with reference to the accompanying drawings in which:

FIG. 1 illustrates a first example of a loudspeaker system and a microphone apparatus; FIG. 2 illustrates a second example of a loudspeaker system and a microphone apparatus;

FIG. 3 illustrates an example method;

FIG. 4 illustrates an example apparatus; and

FIG. 5 illustrates an example non-transitory computer-readable storage medium.

DETAILED DESCRIPTION

FIGS. 1 and 2 illustrate a loudspeaker system 100 to be calibrated by the methods described herein. The loudspeaker system 100 comprises an omnidirectional loudspeaker 101 , an audio signal processor 142, and an amplifier 144. FIGS. 1 and 2 also show calibration equipment in the form of a microphone apparatus 200 and a controller 400 comprising at least one memory for storing audio recorded by the microphone apparatus 200.

The loudspeaker system 100 is configured to be used in acoustic assessments. The illustrated omnidirectional speaker 101 comprises an icosahedral loudspeaker enclosure 102 to which a plurality of audio transducers 104 are secured. Each audio transducer 104 is a drive unit.

Each face of the loudspeaker enclosure 102 comprises a single audio transducer 104.

In another embodiment, the omnidirectional loudspeaker 101 is a different polyhedron with more or fewer faces and/or comprises more or fewer audio transducers 104. For example, the omnidirectional loudspeaker 101 can be an octahedron or dodecahedron. The loudspeaker enclosure 102 can comprise any appropriate mount (not shown) enabling its attachment to a stand.

If one of the 20 faces of the icosahedral loudspeaker enclosure 102 comprises the mount, then each of the remaining 19 faces may comprise an audio transducer 104. If the mount is instead at one of the vertices, then each of the 20 faces may comprise an audio transducer 104.

The large number of small audio transducers 104, in close proximity to each other and each facing a different direction than the other, increases their spatial density and therefore increases their mutual coupling and the frequency at which comb filtering becomes notable.

The amplifier 144 and the audio signal processor 142 define an audio driving system 140 and can either be integrated or separate from each other.

As shown in FIGS. 1-2, the amplifier 144 and the audio signal processor 142 may be outside the loudspeaker enclosure 102. Alternatively, one or both of them may be inside the loudspeaker enclosure 102.

As shown in FIGS. 1-2, the loudspeaker system 100 comprises a plurality of audio channels 106, each represented by a line connecting the audio driving system 140 to the loudspeaker enclosure 102. Each line can be in the form of an electrical wire or other electrical conduit. The amplifier 144 has a number of outputs corresponding to the number of audio channels.

Four audio channels are shown. Each audio channel is connected to a subset of four or five audio transducers 104. Each subset is an array of different audio transducers 104.

The use of fewer audio channels than audio transducers 104 reduces the number of amplifier circuits required in the amplifier 144, and reduces the time required to calibrate the loudspeaker system 100 because fewer calibrations are required. In another embodiment, the number of audio channels is the same as the number of audio transducers 104. Each audio channel is connected to one audio transducer 104.

Once calibrated, the end user of the calibrated loudspeaker system 100, conducting acoustic measurements, may provide an input sound signal for acoustic measurement to all of the audio channels simultaneously.

The use of four channels for 19 or 20 audio transducers 104 enables the special case where four arrays/subsets of audio transducers 104 can be interleaved such that each audio transducer 104 connected to a given audio channel is non-adjacent to any other audio transducer 104 connected to the same audio channel.

This is beneficial because no pair of adjacent audio transducers 104 will cause comb filtering due to spatial aliasing only.

If the loudspeaker enclosure is instead a dodecahedron (12-sided), 12 separate audio channels may be needed.

The calibration method 300 described in relation to FIG. 3 calibrates each audio channel and therefore calibrates each array/subset of audio transducers 104.

The calibration method 300 described below advantageously calibrates decorrelation filters known as Temporally Diffuse Impulses (TDIs) in the audio signal processor 142, to decorrelate the audio transducers 104 (audio channels). TDIs are described first.

The advantage of using TDIs compared to other decorrelation filters is that the reference sound is better preserved, without additional implementation complexity. Firstly, this is because all-pass behaviour is maintained. Secondly, this is because time smearing is minimised. Thirdly, there is more control over the balance between the length of the filter required for low frequency decorrelation, and the effects of time smearing at high frequencies. In some implementations, a different decorrelation filter than TDIs can be used. TDIs are synthesised by summing exponentially decaying, random phase cosine signals of increasing frequency up to the Nyquist frequency of the system. The use of frequency-dependent decays avoids the time smearing that occurs with relatively long finite impulse response (FIR) filters.

The lowest and highest frequencies requiring decorrelation can be user-defined.

The time constant for the lowest frequency may be in the order of 10 ^A -1 seconds. The time constant for the highest frequency may be in the order of 10 ^A -2 or 10 ^A -3 seconds. The time constants may represent the time it takes for the amplitude of the exponential decay curve to reach a factor of 0.368 (1/e).

In TDI generation, the time constants of the intermediate frequencies are generated through an interpolation process, and stored in a vector once the time constants of the lowest and highest frequencies are defined.

The time constants in the vector are then converted to a vector of decay constants (DC), by:

When generating a separate TDI for each audio channel, each TDI is generated using the same DC vector but a vector of random phase values is defined for each TDI. The phase response for each TDI is set using a random sequence of length N/2 with values between ±x, where x is a value selected from the range 0 to TT.

This random ‘seed’ enables decorrelation of signals output/rendered by the audio transducers 104. The random phase values (P) can be generated by a processor via a probability density function (PDF).

Once the vector of random phase values has been defined, the TDIs are then synthesised. This is achieved by summing exponentially decaying cosine signals of increasing frequency, each cosine signal being of a random phase drawn from the vector of random phase values. Each cosine signal is multiplied by an exponential decay window having a decay constant from the DC vector. where:

I = the synthesised impulse r = vector of length N with linearly spaced values from n= 0 to n=N-1 o = standard deviation

Pk = vector of random phase values

DC = vector of decay constants k = impulse currently being synthesised

The generated impulses are then equalised by a minimum-phase equalisation function or similar. According to a minimum-phase equalisation function, the frequency response of the impulse is normalised to the minimum-phase Fourier transform of the impulse. This ensures an all-pass magnitude response.

The minimum-phase Fourier transform T(f) of the unequalised impulse is obtained from the complex conjugate of the Hilbert transform of the log magnitude response: y(y) = _e H(iog(|x(/)l))

Y(f) = minimum-phase Fourier transform of unequalised impulse

X(f) = Fourier transform of unequalised impulse

H = Hilbert transform operator

The final TDI is: where:

T ¹ = inverse Fourier transform

IR = real part of complex number The calibration method 300 is now described, with reference to FIGS. 1-3. FIGS. 1 and 2 illustrate example arrangements of the loudspeaker system 100 and a microphone apparatus 200 for calibration.

FIG. 1 illustrates a first embodiment in which the loudspeaker enclosure 102 is at least partially surrounded by microphones 202 of a microphone apparatus 200. The location may be an anechoic chamber which is any chamber configured to suppress sound reflections.

The microphone apparatus 200 is connected to a controller 400 comprising at least one memory to store audio signals captured by the microphone apparatus 200, for use in calibration.

The audio transducers 104 are connected to the audio driving system 140 by the audio channels.

The microphones 202 are arranged in an array. Each microphone position is referred to as an angular position relative to an orientation of the loudspeaker enclosure 102. The angular position is denoted by the symbol ‘m’. While the audio transducers 104 render a reference sound, the microphones 202 are activated sequentially or simultaneously to record the reference sound from each angular position.

In some examples, each microphone 202 has a substantially omnidirectional pick-up pattern, and is therefore known as a ‘measurement microphone’.

The microphones 202 are of equal distance from the loudspeaker enclosure 102. The microphones 202 are oriented so that the loudspeaker enclosure 102 is ‘on-axis’ with respect to each microphone 202. The microphones 202 may therefore be arranged in an arc or circle.

Each microphone 202 is at a distance from the loudspeaker enclosure 102 of at least three diameters and/or no greater than six diameters. The diameter refers to the circleequivalent diameter or other characteristic dimension of the loudspeaker enclosure 102. An upper limit of distance reduces effects such as reflections from nearby surfaces. These reflections will not be important if the measurement room is an anechoic chamber.

The angular spacing of each adjacent microphone 202 relative to the loudspeaker enclosure 102 at the centre, may be a value no greater than 30 circular degrees. A high spatial density of angular positions enables calibration at higher frequencies. The number of angular positions m may be dictated by the upper frequency to calibrate for. The angular spacing may be less than A/2 at the maximum frequency.

The recommended minimum number of angular positions (microphones 202) is 20, but as few as 12 could be used as long as the angular spacing of adjacent microphones 202 is not too high. The number of angular positions can be different than the number of audio channels n.

The angular positions may lie on a common plane such as a horizontal plane. Additionally, or alternatively, calibration could include angular positions on a different plane. The calibration may therefore be in two or three dimensions.

FIG. 2 illustrates a second embodiment in which a rotator 106 is configured to rotate the loudspeaker enclosure 102. This enables the microphone apparatus 200 to have only one microphone 202. The rotator 106 can comprise a turntable or the like. The rotator 106 may be controlled automatically or manually, as part of the later-described method 300.

Therefore, in FIG. 2 the angular position is defined by an orientation of the loudspeaker enclosure 102, whereas in FIG. 1 the angular position is defined by the microphone position.

In a third embodiment (not shown), one microphone 202 is used as in FIG. 2, but a rotator 106 is not provided so the microphone 202 is moved manually or automatically.

FIG. 3 is a flowchart illustrating an example calibration method 300 for the loudspeaker system 100. The blocks in the method 300 are optional unless explicitly required to be essential by the independent claims. Some, but not necessarily all blocks of the method 300 may be executed automatically.

Block 302 comprises setting up the loudspeaker system 100 and the microphone apparatus 200, in the manner shown in the first embodiment (FIG. 1), second embodiment (FIG. 2), or third embodiment.

Decision block 304 comprises determining whether correct connectivity has been achieved.

In an example, determining whether correct connectivity has been achieved comprises rendering a reference sound through each audio channel, and recording the reference sound by the microphone apparatus 200.

The reference sound may be of a low frequency (<300Hz) such as 250Hz or 200Hz. The reference sound may be band-limited. When outputting low frequencies, the audio transducers are expected to be coherent.

The controller 400 may be configured to implement a relative amplitude threshold. If the recording of one set of audio channels has an amplitude greater than the recording of another set of audio channels exceeding the threshold value, there may be a problem with connection polarity. The threshold may be a value selected from the range 3dB to 12dB.

A speaker was tested using noise and measuring the 250Hz octave band. A connection polarity within the loudspeaker casing (not at the amplifier) showed (a) a non-spherical polar response, and (b) a reduction in level (250Hz octave band) of between 4 dB and 9 dB, depending on the measurement position.

If decision block 304 is satisfied, the method 300 proceeds to block 306. Block 306 comprises selecting a first one of the angular positions ‘m’, such as by selecting a first microphone 202 (FIG. 1) or setting the loudspeaker enclosure 102 to a first orientation (FIG. 2). Block 308 comprises activating a selected audio channel to activate the subset of one or more audio transducers 104 connected to that audio channel. Activating the selected audio channel can comprise causing rendering of audio via the selected audio channel, while the other audio channels are inactive.

The rendered audio can comprise standard test signal characteristics such as pink noise characteristics.

Block 310 comprises measuring and storing an impulse response of the audio transducers to the audio rendered by the loudspeaker system 100, as detected by one of the microphone(s) 202. The impulse response captures the transfer function for the subset of audio transducers 104.

Block 310 may be implemented by the controller 400. The controller 400 may first apply a microphone transfer function to compensate for microphone effects.

Decision block 312 comprises determining whether the measured impulse response satisfies a validity condition.

In an implementation, the validity condition requires the impulse response to include an above-threshold peak. Therefore, if the stored audio data is just noise it can be discarded.

Blocks 314-320 together define feedback loops for measuring and storing the impulse response for each audio channel n for each angular position m.

Block 314 comprises incrementing the value of the selected audio channel n by 1 , which means selecting the next audio channel. Decision block 316 determines whether n is greater than N, which means determining whether the last audio channel N has been measured.

If the currently selected audio channel n is not a value greater than N, then the currently selected audio channel exists so the method 300 loops back to blocks 308-312 to measure and store its impulse response. If n is greater than N, then the currently selected audio channel does not exist and the method 300 proceeds to block 318.

Block 318 comprises incrementing the value of the currently selected angular position m by 1 , which means selecting the next angular position. This can comprise selecting a different microphone 202 (FIG. 1) and/or rotating the loudspeaker enclosure 102 (FIG. 2). Decision block 320 determines whether m is greater than M, which means determining whether the last angular position M has been measured.

If the currently selected angular position m is not a value greater than M, then the currently selected angular position m exists and the method 300 loops back to blocks 306-316 to measure and store the impulse responses of each audio channel n for the currently selected angular position m.

If m is greater than M, then the currently selected angular position does not exist, meaning that all of the impulse responses have been measured and stored. The method 300 proceeds to block 322 accordingly, to commence a TDI generation part of the method 300 which relies on the stored impulse responses.

Blocks 322-330 iteratively generates TDIs, applies them to the measured impulse responses to predict performance, and outputs the TDIs when satisfactory performance is predicted. This is done automatically (without user intervention) by a controller 400 via predictive methods, rather than by repeating physical experiments. This finds the optimal TDIs within seconds, at most.

The controller 400 may be the same controller 400 or a different controller than that shown in FIGS. 1-2.

Block 322 comprises generating a set of ‘N’ TDIs, where N is the number of audio channels. The TDIs are generated in the above-described manner, which includes generating the random phase values. Therefore, each TDI is statistically random with respect to each other TDI, notwithstanding their shared user-defined time constraints and equalisation. Blocks 324 and 326 comprise applying the TDIs to input signals in the form of the previously-stored measured impulse responses, and determining (calculating) the resulting predicted/expected impulse responses with the TDIs applied.

Calculating the predicted impulse responses comprises obtaining the convolution of the measured impulse responses with the set of TDIs, wherein the measured impulse responses are the input signals. This corresponds to multiplying the measured impulse responses by the TDIs in the frequency domain.

The predicted impulse response for each angular position m is calculated by summating (adding) the predicted impulse responses of all the audio channels N corresponding to that angular position m. The predicted impulse response therefore predicts what impulse response would be measured by a microphone 202 at a particular angular position and distance, when all of the audio channels are simultaneously rendering the same audio through all of their respective audio transducers 104, based on the currently-generated TDIs.

At blocks 328-330, the method 300 then determines whether a predicted performance metric condition is satisfied, referred to herein as a performance metric condition. This is based on the predicted impulse responses. The performance metric condition is configured to allow TDIs which minimise the predicted correlations between the audio transducers 104.

Block 328 comprises calculating the predicted performance metric and decision block 330 comprises determining whether the predicted performance metric satisfies the performance metric condition.

The performance metric condition is based on the differences between the set of M predicted impulse responses each corresponding to a different one of the angular positions m. The differences may be statistical differences. The statistical differences may relate to a measure of statistical dispersion in the frequency domain, such as variance. This is because variance of the predicted impulse responses in the frequency domain across the angular positions is an indicator of comb filtering due to correlation between the audio transducers 104.

Calculating the predicted performance metric can comprise determining the value of the measure of statistical dispersion, such as variance, in the frequency domain, for the ensemble/set of M predicted impulse responses.

In some implementations, the predicted performance metric is calculated for each frequency or frequency bin/band of interest, resulting in a set of F values of the predicted performance metrics. A measure of central tendency, such as average value, may then be calculated and taken as the value. This means that inconsistent performance across different frequencies can be rejected.

Determining whether the performance metric condition is satisfied can comprise determining whether the value of the measure of statistical dispersion is less than a threshold.

If the performance metric condition is not satisfied (threshold exceeded), the method 300 may loop back to block 322 to generate a new set of TDIs. The new set of TDIs may be random with respect to the previous set of TDIs.

If the performance metric condition is satisfied, then the calibration is complete subject to an optional validation experiment from blocks 332-342.

The validation experiment is now described. As with previous operations, block 332 comprises determining impulse responses to audio rendered by the audio transducers 104, via the microphone apparatus 200 and controller 400.

The purpose of the validation experiment is to investigate whether significant comb filtering/correlation will occur in use, when all the audio transducers 104 would be used simultaneously. The purpose of optimising the TDIs is to replace a few wide comb filters that significantly degrade the omnidirectional nature of the source with many very-narrow comb filters that have a much lower impact. The reference audio signal is now output simultaneously through multiple/all audio channels N, rather than through just one audio channel n at a time. The reference audio signal may be the same, repeatable signal across all audio channels such as the Maximum Length Sequence (MLS) or a swept sine.

In this experiment, the TDIs that satisfied the performance metric condition are implemented as software and/or hardware in the audio signal processor 142, to minimise the correlation between the audio transducers 104.

Each time block 332 measures an impulse response, the method 300 proceeds to optional block 334 which checks the validity of the measurements in the same manner as block 312.

The next blocks 336-338 create a loop in which the angular position m is incremented until a set of M experimentally determined impulse responses has been created. These blocks work in the same manner as blocks 318-320, the only difference being that these blocks loop back to block 332.

Blocks 340-342 relate to calculating an experiment performance metric and determining whether an experiment performance metric condition is satisfied. The metric and the condition may be the same as defined earlier for blocks 328-330, the difference being that it is performed on experimentally derived impulse responses rather than predicted impulse responses.

If decision block 342 is not satisfied, then the TDIs are not suitable and the method 300 loops back to block 322 to generate new TDIs. Testing revealed that this is unlikely to happen due to the reliability of the prediction in blocks 324-330.

Once the (or each) performance metric condition is satisfied, the method 300 proceeds to the final block 344 which comprises providing the set of TDIs for implementation in the audio signal processor 142 of the loudspeaker system 100.

This can take the form of an automatic installation of the TDIs into the software of the audio signal processor 142. Alternatively, this can be a manual process in which the method 300 records the set of TDIs in the memory and/or renders the TDIs on a display. The user can then install the TDIs in to the software of the audio signal processor 142.

The above method 300 utilises predictive and experimental techniques to rapidly find the optimal TDIs. It would be appreciated that the method 300 could be performed wholly experimentally by omitting blocks 324-330, or wholly predictively by omitting blocks 332-342.

Fig 4 illustrates an example of a controller 400 suitable for use in an apparatus such as the audio system 100. Implementation of a controller 400 may be as controller circuitry. The controller 400 may be implemented in hardware alone, have certain aspects in software including firmware alone or can be a combination of hardware and software (including firmware).

As illustrated in Fig 4 the controller 400 may be implemented using instructions that enable hardware functionality, for example, by using executable instructions of a computer program 406 in a general-purpose or special-purpose processor 402 that may be stored on a computer readable storage medium (disk, memory etc) to be executed by such a processor 402.

The processor 402 is configured to read from and write to the memory 404. The processor 402 may also comprise an output interface via which data and/or commands are output by the processor 402 and an input interface via which data and/or commands are input to the processor 402.

The memory 404 stores a computer program 406 comprising computer program instructions (computer program code) that controls the operation of the apparatus 100 when loaded into the processor 402. The computer program instructions, of the computer program 406, provide the logic and routines that enables the apparatus to perform the methods illustrated in the accompanying Figs. The processor 402 by reading the memory 404 is able to load and execute the computer program 406. The apparatus 100 comprises: at least one processor 402; and at least one memory 404 including computer program code the at least one memory 404 and the computer program code configured to, with the at least one processor 402, cause the apparatus 100 at least to perform: determining 326/332 impulse responses of the audio transducers to input signals to which a set of Temporally Diffuse Impulses has been applied, each impulse response associated with a different angular position ‘m’ defined by a microphone apparatus 200 and the loudspeaker enclosure 102; determining 330/342 whether a performance metric condition is satisfied, in dependence on differences between the impulse responses for the different angular positions; in dependence on the performance metric condition not being satisfied, modifying 322 the set of Temporally Diffuse Impulses and then repeating the above steps of determining impulse responses and determining whether the performance metric condition is satisfied; and in dependence on the performance metric condition being satisfied, providing 344 the set of Temporally Diffuse Impulses for implementation in the audio signal processor 142 of the loudspeaker system 100.

As illustrated in Fig 5, the computer program 406 may arrive at the apparatus 100 via any suitable delivery mechanism 408. The delivery mechanism 408 may be, for example, a machine-readable medium, a computer-readable medium, a non-transitory computer-readable storage medium, a computer program product, a memory device, a record medium such as a Compact Disc Read-Only Memory (CD-ROM) or a Digital Versatile Disc (DVD) or a solid-state memory, an article of manufacture that comprises or tangibly embodies the computer program 406. The delivery mechanism may be a signal configured to reliably transfer the computer program 406. The apparatus 100 may propagate or transmit the computer program 406 as a computer data signal.

Computer program instructions for causing an apparatus to perform at least the following or for performing at least the following: causing determining 326/332 impulse responses of the audio transducers to input signals to which a set of Temporally Diffuse Impulses has been applied, each impulse response associated with a different angular position ‘m’ defined by a microphone apparatus 200 and the loudspeaker enclosure 102; causing determining 330/342 whether a performance metric condition is satisfied, in dependence on differences between the impulse responses for the different angular positions; in dependence on the performance metric condition not being satisfied, causing modifying 322 the set of Temporally Diffuse Impulses and then repeating the above steps of determining impulse responses and determining whether the performance metric condition is satisfied; and in dependence on the performance metric condition being satisfied, causing providing 344 the set of Temporally Diffuse Impulses for implementation in the audio signal processor 142 of the loudspeaker system 100.

The computer program instructions may be comprised in a computer program, a non- transitory computer readable medium, a computer program product, a machine- readable medium. In some but not necessarily all examples, the computer program instructions may be distributed over more than one computer program.

Although the memory 404 is illustrated as a single component/circuitry it may be implemented as one or more separate components/circuitry some or all of which may be integrated/removable and/or may provide permanent/semi-permanent/ dynamic/cached storage.

Although the processor 402 is illustrated as a single component/circuitry it may be implemented as one or more separate components/circuitry some or all of which may be integrated/removable. The processor 402 may be a single core or multi-core processor.

References to ‘computer-readable storage medium’, ‘computer program product’, ‘tangibly embodied computer program’ etc. or a ‘controller’, ‘computer’, ‘processor’ etc. should be understood to encompass not only computers having different architectures such as single /multi- processor architectures and sequential (Von Neumann)/parallel architectures but also specialized circuits such as field- programmable gate arrays (FPGA), application specific circuits (ASIC), signal processing devices and other processing circuitry. References to computer program, instructions, code etc. should be understood to encompass software for a programmable processor or firmware such as, for example, the programmable content of a hardware device whether instructions for a processor, or configuration settings for a fixed-function device, gate array or programmable logic device etc.

The blocks illustrated in the accompanying Figs may represent steps in a method 300 and/or sections of code in the computer program 406. The illustration of a particular order to the blocks does not necessarily imply that there is a required or preferred order for the blocks and the order and arrangement of the block may be varied. Furthermore, it may be possible for some blocks to be omitted.

Where a structural feature has been described, it may be replaced by means for performing one or more of the functions of the structural feature whether that function or those functions are explicitly or implicitly described.

The recording of data may comprise only temporary recording, or it may comprise permanent recording or it may comprise both temporary recording and permanent recording, Temporary recording implies the recording of data temporarily. This may, for example, occur during sensing or image capture, occur at a dynamic memory, occur at a buffer such as a circular buffer, a register, a cache or similar. Permanent recording implies that the data is in the form of an addressable data structure that is retrievable from an addressable memory space and can therefore be stored and retrieved until deleted or over-written, although long-term storage may or may not occur. The use of the term ‘capture’ in relation to an image relates to temporary recording of the data of the image. The use of the term ‘store’ in relation to an image relates to permanent recording of the data of the image.

The term ‘comprise’ is used in this document with an inclusive not an exclusive meaning. That is any reference to X comprising Y indicates that X may comprise only one Y or may comprise more than one Y. If it is intended to use ‘comprise’ with an exclusive meaning then it will be made clear in the context by referring to “comprising only one...” or by using “consisting”. In this description, the wording 'connect’, 'couple’ and 'communication’ and their derivatives mean operationally connected/coupled/in communication. It should be appreciated that any number or combination of intervening components can exist (including no intervening components), i.e., so as to provide direct or indirect connection/coupling/communication. Any such intervening components can include hardware and/or software components.

As used herein, the term "determine/determining" (and grammatical variants thereof) can include, not least: calculating, computing, processing, deriving, measuring, investigating, identifying, looking up (for example, looking up in a table, a database or another data structure), ascertaining and the like. Also, "determining" can include (for example, information), accessing (for example, accessing data in a memory), obtaining and the like. Also, " determine/determining" can include resolving, selecting, choosing, establishing, and the like.

In this description, reference has been made to various examples. The description of features or functions in relation to an example indicates that those features or functions are present in that example. The use of the term ‘example’ or ‘for example’ or ‘can’ or ‘may’ in the text denotes, whether explicitly stated or not, that such features or functions are present in at least the described example, whether described as an example or not, and that they can be, but are not necessarily, present in some of or all other examples. Thus ‘example’, ‘for example’, ‘can’ or ‘may’ refers to a particular instance in a class of examples. A property of the instance can be a property of only that instance or a property of the class or a property of a sub-class of the class that includes some but not all of the instances in the class. It is therefore implicitly disclosed that a feature described with reference to one example but not with reference to another example, can where possible be used in that other example as part of a working combination but does not necessarily have to be used in that other example.

Although examples have been described in the preceding paragraphs with reference to various examples, it should be appreciated that modifications to the examples given can be made without departing from the scope of the claims. Features described in the preceding description may be used in combinations other than the combinations explicitly described above.

Although functions have been described with reference to certain features, those functions may be performable by other features whether described or not.

Although features have been described with reference to certain examples, those features may also be present in other examples whether described or not.

The term ‘a’, ‘an’ or ‘the’ is used in this document with an inclusive not an exclusive meaning. That is any reference to X comprising a/an/the Y indicates that X may comprise only one Y or may comprise more than one Y unless the context clearly indicates the contrary. If it is intended to use ‘a’, ‘an’ or ‘the’ with an exclusive meaning then it will be made clear in the context. In some circumstances the use of ‘at least one’ or ‘one or more’ may be used to emphasis an inclusive meaning but the absence of these terms should not be taken to infer any exclusive meaning.

The presence of a feature (or combination of features) in a claim is a reference to that feature or (combination of features) itself and also to features that achieve substantially the same technical effect (equivalent features). The equivalent features include, for example, features that are variants and achieve substantially the same result in substantially the same way. The equivalent features include, for example, features that perform substantially the same function, in substantially the same way to achieve substantially the same result.

In this description, reference has been made to various examples using adjectives or adjectival phrases to describe characteristics of the examples. Such a description of a characteristic in relation to an example indicates that the characteristic is present in some examples exactly as described and is present in other examples substantially as described.

The above description describes some examples of the present disclosure however those of ordinary skill in the art will be aware of possible alternative structures and method features which offer equivalent functionality to the specific examples of such structures and features described herein above and which for the sake of brevity and clarity have been omitted from the above description. Nonetheless, the above description should be read as implicitly including reference to such alternative structures and method features which provide equivalent functionality unless such alternative structures or method features are explicitly excluded in the above description of the examples of the present disclosure.

Whilst endeavoring in the foregoing specification to draw attention to those features believed to be of importance it should be understood that the Applicant may seek protection via the claims in respect of any patentable feature or combination of features hereinbefore referred to and/or shown in the drawings whether or not emphasis has been placed thereon. l/we claim: