Patent Description:
Capture of audio signals from multiple sources and mixing of audio signals when these sources are moving in the spatial field requires significant effort. For example the capture and mixing of an audio signal source such as a speaker or artist within an audio environment such as a theatre or lecture hall to be presented to a listener and produce an effective audio atmosphere requires significant investment in equipment and training.

A commonly implemented system is where one or more 'external' microphones, for example a Lavalier microphone worn by the user or an audio channel associated with an instrument, is mixed with a suitable spatial (or environmental or audio field) audio signal such that the produced sound comes from an intended direction. This system is known in some areas a Spatial Audio Mixing (SAM).

The SAM system enables the creation of immersive sound scenes comprising "background spatial audio" or ambiance and sound objects for Virtual Reality (VR) applications. Often, the scene can be designed such that the overall spatial audio of the scene, such as a concert venue, is captured with a microphone array (such as one contained in the OZO virtual camera) and the most important sources captured using the 'external' microphones.

The SAM system typically employs a radio-based indoor positioning system (such as high accuracy indoor positioning -HAIP) which can provide estimates of a sound source direction of arrival (DOA) as azimuth and elevation values. However, for full positioning a distance estimate is required as well. In other words a distance estimate is required in order to obtain full spherical coordinates (azimuth, elevation, distance). The determination of a distance estimate would allow, for example, free listening point rendering of the sound sources, and rendering changes in the close-up audio signal as the distance changes. If distance information is not available, then free listening point audio rendering cannot be performed as when the listening position changes by a certain vector from its original position, the full original vector from the listening point to the sound source is needed to be able to determine the new vector from the new listening point to the sound source.

The lack of free listening point ability significantly limits the immersive audio experience since the user is only able to rotate their head not move their head to a different position within the sound scene and for example to listen to some sources more closely or explore different locations of the sound scene.

Moreover, without distance information then for fixed listening point audio rendering a distance (which is typically defined at <NUM> meters) is defined which is used to adjust the sound source volume or otherwise require a sound engineer to do this manually. Having full source positions sound rendering can be performed in such a way that the sound changes according to the distance of the sound source from the listener (so that the sound gets louder and the proportion of direct sound increases as the sound source approaches the listener).

<CIT> discloses apparatus comprising a processor configured to: receive a spatial audio signal associated with a microphone array configured to provide spatial audio capture and at least one additional audio signal associated with an additional microphone, the at least one additional microphone signal having been delayed by a variable delay determined such that the audio signals are time aligned; receive a relative position between a first position associated with the microphone array and a second position associated with the additional microphone; generate at least two output audio channel signals by processing and mixing the spatial audio signal and the at least one additional audio signal based on the relative position between the first position and the second position such that the at least two output audio channel signals present an augmented audio scene.

<CIT> discloses an apparatus for providing direction information based on a reproduced audio signal with an embedded watermark is provided. The apparatus comprises a signal processor, which is adapted to process at least two received watermarked audio signals recorded by at least two audio receivers at different spatial positions. The signal processor is adapted to process the received watermarked audio signals to obtain a receiver-specific information for each received watermarked audio signal. The receiver-specific information depends on the embedded watermarks embedded in the received watermarked audio signals. Moreover, the apparatus comprises a direction information provider for providing direction information based on the receiver-specific information for each received watermarked audio signal.

<CIT> discloses a sound monitoring system which provides autonomous and silent surveillance to monitor sound sources stationary or moving in 3D space and a blind separation of target acoustic signals. The underlying principle of this technology is a hybrid approach that uses: <NUM>) passive sonic detection and ranging method that consists of iterative triangulation and redundant checking to locate the Cartesian coordinates of arbitrary sound sources in 3D space, <NUM>) advanced signal processing to sanitize the measured data and enhance signal to noise ratio, and <NUM>) short-time source localization and separation to extract the target acoustic signals from the directly measured mixed ones.

"<NPL> discloses a three ring microphone array which estimates the horizontal/vertical direction and distance of sound sources and separates multiple sound sources for mobile robot audition. Arrangement of microphones is simulated and an optimized pattern which has three rings is implemented with <NUM> microphones. Sound localization and separation are achieved by delay and sum beam forming (DSBF) and frequency band selection (FBS). From on-line experiments results of sound horizontal and vertical localization, we confirmed that one or two sounds sources could be localized with an error of about <NUM> degrees and <NUM> to <NUM> in the case of the distance of about <NUM>. The off-line experiments of sound separation were evaluated by power spectrums in each frequency of separated sounds, and we confirmed that an appropriate frequency band could be selected by DSBF and FBS. The system can separate <NUM> different pressure speech sources without drowning out.

There is provided according to a first aspect an apparatus as indicated in claim <NUM>.

According to a second aspect there is provided a method as indicated in claim <NUM>.

The following describes in further detail suitable apparatus and possible mechanisms for the provision of effective sound source distance determination from the capture of audio signals from multiple devices. As discussed this is a desired parameter used in the mixing of the audio signals. In the following examples, audio signals and audio capture signals are described.

The concept as discussed in further detail hereafter is apparatus and methods for obtaining distance estimates for sound sources tracked with 'inside-out' radio-based positioning system. An 'inside-out' configuration generally features a radio-based positioning system (such as a high accuracy indoor positioning - HAIP) is only able to provide estimates of the direction of arrival such as azimuth, elevation. The 'inside-out' configuration has at least two significant benefits. Firstly it is has a very attractive form factor and can be deployed, for example, on the same camera stand as a virtual reality camera. Secondly it does not require any calibration but is ready to be used out-of-the-box. However, its limitation is that it does not provide us with distance estimates, which if required need to be obtained otherwise.

The embodiments as discussed hereafter thus utilize a microphone array, such as Nokia's OZO capture apparatus, to obtain distance estimates of sound sources which are also captured using external microphones. The system as discussed herein is able to identify which frequency bands of the audio signals generated from the array microphone capture apparatus likely contain frequency bands of each sound source, and then the fluctuation (variance) of direction of arrival (DOA) estimates of those frequency bands can be used to obtain a distance estimate for the sources. The distance information may then be combined with the azimuth and elevation (or DOA) information obtained from the radio-based positioning system, to generate full position information for sound sources tracked with an inside-out positioning configuration + a microphone array.

Furthermore in some embodiments the apparatus may be used to improve the reliability of radio-positioning based DOA estimates with audio-based DOA estimates.

The embodiments as discussed hereafter may be implemented as part of a Spatial Audio Mixing (SAM) system, where the sources and/or the listening (or viewing) position is able to be located based on a user or other input and as such the accurate full position information associated with each sound source is required to be able to spatially position the sound source within the sound scene relative to the listener. The concept as described herein thus may be considered to be enhancement to conventional Spatial Audio Capture (SPAC) technology. Spatial audio capture technology can process audio signals captured via a microphone array into a spatial audio format. In other words generating an audio signal format with a spatial perception capacity. The concept may thus be embodied in a form where audio signals may be captured such that, when rendered to a user, the user can experience the sound field as if they were present at the location of the capture device. Spatial audio capture can be implemented for microphone arrays found in mobile devices. In addition, audio processing derived from the spatial audio capture may be used employed within a presence-capturing device such as the Nokia OZO (OZO) devices.

A conventional approach to the capturing and mixing of audio sources with respect to an audio background or environment audio field signal would be for a professional producer to utilize a sound source microphone (which is also known as an external microphone, a close or Lavalier microphone worn by the user, or a microphone attached to an instrument or some other microphone) to capture audio signals close to the sound source, and further utilize a 'background' microphone or microphone array to capture a environmental audio signal. These signals or audio tracks may then be manually mixed to produce an output audio signal such that the produced sound features the sound source coming from an intended (though not necessarily the original) direction. Although capture and render systems may be separate, it is understood that they may be implemented with the same apparatus or may be distributed over a series of physically separate but communication capable apparatus. For example, a presence-capturing device such as the OZO device could be equipped with an additional interface for receiving location data and close microphone audio signals, and could be configured to perform the capture part. The output of a capture part of the system may be the microphone audio signals (e.g. as a <NUM> channel downmix), the close microphone audio signals (which may furthermore be time-delay compensated to match the time of the microphone array audio signals), and the position information of the close microphones (such as a time-varying azimuth, elevation, distance with regard to the microphone array).

The renderer as described herein may be an audio playback device (for example a set of headphones), user input (for example motion tracker), and software capable of mixing and audio rendering. In some embodiments the user input and audio rendering parts may be implemented within a computing device with display capacity such as a mobile phone, tablet computer, virtual reality headset, augmented reality headset etc..

Furthermore it is understood that at least some elements of the following mixing and rendering may be implemented within a distributed computing system such as known as the 'cloud'.

With respect to <FIG> an example system comprising apparatus suitable for implementing some embodiments is shown.

<FIG> shows an example sound source microphone (and tag) <NUM> which is configured to transmit HAIP signals which are received by the positioner <NUM> in order to determine the direction of arrival (DOA) and actual position of the sound source microphone <NUM> relative to the microphone array <NUM>. The sound source microphone may furthermore generate a sound source audio signal which is passed to the source separator <NUM>. Although in the following examples there is shown one example sound source microphone it is understood that there may be more than one sound source microphone. As described herein the sound source microphone <NUM> can be configured to capture audio signals associated with humans, instruments, or other sound sources of interest.

For example the sound source microphone <NUM> may be a Lavalier microphone. The sound source microphone may be any microphone external or separate to a microphone array which may capture the spatial audio signal. Thus the concept is applicable to any external/additional microphones be they Lavalier microphones, hand held microphones, mounted mics, or whatever. The sound source microphone can be worn/carried by persons or mounted as close-up microphones for instruments or a microphone in some relevant location which the designer wishes to capture accurately. A Lavalier microphone typically comprises a small microphone worn around the ear or otherwise close to the mouth. For other sound sources, such as musical instruments, the audio signal may be provided either by a Lavalier microphone or by an internal microphone system of the instrument (e.g., pick-up microphones in the case of an electric guitar) or an internal audio output (e.g., a electric keyboard output). In some embodiments the sound source microphone may be configured to output the captured audio signals to the source separator <NUM> wirelessly. The sound source microphone may in such embodiments be connected to a transmitter unit (not shown), which wirelessly transmits the audio signal to a receiver unit (not shown) within the source separator <NUM>.

In some embodiments the sound source microphone and thus the performers and/or the instruments that are being played positions may be tracked by using position tags located on or associated with the sound source microphone. Thus for example the sound source microphone comprises or is associated with a microphone position tag.

In some embodiments the system comprises a positioner <NUM>. The positioner <NUM> may be a configured to comprise a receiver which is configured to receive the radio signal transmitted by the microphone position tag such that the positioner <NUM> may determine information identifying the position or location of the sound source microphone. In some embodiments the positioner <NUM> is configured to determine the azimuth or the elevation of the sound source microphone relative to the positioner receiver. In some other embodiments the positioner <NUM> is configured to determine both the azimuth and the elevation of the sound source microphone relative to the positioner receiver.

Although the following examples show the use of the HAIP (high accuracy indoor positioning) radio frequency signal to determine the location of the close microphones it is understood that any suitable position estimation system may be used (for example satellite-based position estimation systems, inertial position estimation, beacon based position estimation etc.). The positioner <NUM> may be configured to output the 'radio-based' position estimate of the sound source microphone to the source separator <NUM>.

In some embodiments the system comprises a microphone array <NUM>. The microphone array <NUM> may comprise a plurality of microphones configured to capture a plurality of audio signals which represent the sound scene. The microphone array <NUM> may be configured to output the audio signals to the source separator <NUM> and furthermore to the audio based distance estimator <NUM>. In some embodiments the microphone array may be configured to output the captured audio signals to the source separator <NUM>/audio based distance estimator <NUM> wirelessly. The microphone array <NUM> may in such embodiments be connected to a transmitter unit (not shown), which wirelessly transmits the audio signal to a receiver unit (not shown) within the source separator <NUM>/audio based distance estimator <NUM>. In some embodiments the microphone array <NUM> is implemented as part of a presence capture apparatus or device such as a Nokia OZO.

The system in some embodiments comprises a source separator <NUM>. The source separator is configured in some embodiments to receive the audio signals from the sound source microphone <NUM>, the microphone array <NUM> and receive the radio based position estimates from the positioner <NUM>. The source separator <NUM> is then configured to identify parts of the microphone array audio signals which represent the audio signals generated by the sound source microphone. These identified parts may then be passed to the audio based distance estimator <NUM>. In some embodiments as described in detail hereafter the source separator <NUM> is configured to indicate frequency bands within the microphone array audio signals which are similar to the audio signals from the sound source microphone.

The system in some embodiments comprises an audio based distance estimator <NUM>. The audio based distance estimator in some embodiments is configured to receive the audio signals from the microphone array <NUM> and furthermore indications of which parts of the microphone array <NUM> audio signals are associated with the audio signals from the sound source microphone <NUM> in order to enable the audio based distance estimator <NUM> to generate a distance estimate of the sound source. The distance estimate may be passed in some embodiments to a position tracker <NUM>.

In some embodiments the system comprises a position tracker <NUM>. The position tracker <NUM> may be configured to receive the audio based distance estimate from the audio based distance estimator <NUM> and the positioner estimator estimates of the azimuth and/or elevation and combine these to generate a complete position estimate of the sound source. In some embodiments the position tracker <NUM> is further configured to update the radio based direction of arrival estimates using audio based direction of arrival estimates generated by the audio based distance estimator <NUM>.

With respect to <FIG> a schematic view of example source separator and audio based distance estimator apparatus is shown in further detail.

In some embodiments the source separator comprises a time to frequency domain transformer <NUM>. The time to frequency domain transformer <NUM> may be configured to receive the microphone array <NUM> audio signals and the sound source microphone <NUM> audio signal and apply a suitable time to frequency domain transform such as a Short Time Fourier Transform (STFT) in order to convert the input time domain signals into a suitable frequency domain representation.

The external microphone capture signal is subjected to the same STFT analysis as a channel in the array capture. The frequency domain representation is divided into B subbands. Thus for example the sound source microphone audio signal Y may be represented in the frequency domain as <MAT>.

The microphone array audio signals may be represented in the frequency domain representation by <MAT>.

The widths of the subbands can follow any suitable distribution. For example the Equivalent rectangular bandwidth (ERB) scale. The value of n represents a discrete frequency and k the microphone channel index.

The frequency domain representations may be output to a sub-band divider.

In some embodiments the source separator comprises a sub-band divider <NUM>. The sub-band divider is configured to receive the frequency domain representations of the microphone array audio signals and the sound source microphone audio signal and generate sub-band versions.

These sub-bands may be passed to a sub-band selector <NUM>.

In some embodiments the sound separator comprises a sub-band selector <NUM>. The sub-band selector is configured to receive the sub-band representations of the audio signals and select a sub-band for analysis.

The selection may for example be a loop control operation whereby a sub-band 'b' is selected and passed to the sub-band combiner <NUM> and a sub-band direction of arrival (DOA) estimator <NUM>.

In some embodiments the audio-based distance estimator comprises a sub-band direction of arrival (DOA) estimator <NUM>. The sub-band DOA estimator <NUM> in some embodiments is configured to receive the sub-band audio signal components from the source separator and determine an audio signal based direction of arrival estimation. This for example may be azimuth estimation and/or an elevation estimation. The estimator <NUM> may use any suitable estimation technique. For example in some embodiments the estimator <NUM> is configured to estimate the direction with two channels (in the example implementation channels <NUM> and <NUM>). The task is to find delay τb that maximizes the correlation between the two channels for subband b. In other words attempting to determine a DOA based on estimated impulse responses from one audio signal (from a first microphone) to another audio signal (from another microphone). This can be accomplished by creating time-shifted versions of the signal in channel <NUM>, and correlating these with the signal on channel <NUM>.

A time shift of τ time domain samples of <MAT> can be obtained as <MAT>.

Now the optimal delay τb is obtained from <MAT> where Re indicates the real part of the result and * denotes combined transpose and complex conjugate operations. <MAT> and <MAT> are considered vectors with length of nb+<NUM> - nb samples. The range of searching for the delay Dmax is selected based on the estimated maximum time delay of sound arrival to two microphones. Knowing the physical distribution of the channel microphones then permits the direction of arrival estimator to determine a first direction of arrival based on the delay.

By performing this analysis between further channels additional directions may be determined which improve the direction of arrival estimation and/or enable the direction of arrival estimation to resolve whether the direction of arrival is forwards of or to the rear of the microphone array, and further determine both an azimuth and elevation direction of arrival.

In some embodiments this delay information may be passed to the sub-band combiner <NUM>.

The sound separator in some embodiments further comprises a sub-band combiner <NUM>. The sub-band combiner <NUM> is configured to generate a combined microphone array audio signal for the sub band. Thus for example the sub-band combiner <NUM> may be configured to generate for the sub-band b: <MAT>.

In some embodiments the sub-band combiner <NUM> is configured to generate the combined (sum) signal using the following logic <MAT>.

In other word in some embodiments the sub-band combiner <NUM> is configured to receive the delay value from the sub-band DOA estimator <NUM> before generating the combined sub-band signal.

Having generated the sum or combined audio signal for the sub-band b, the sum may be passed to the sound source microphone determiner <NUM>.

In some embodiments the source separator <NUM> comprises a sound source determiner <NUM>. The sound source determiner <NUM> is configured to receive the sum or combined audio signal for the sub-band b and the sound source microphone(s) audio signal sub-band b components and determine whether the combined audio signals corresponds to the sound source microphone audio signal. In some embodiments where there are more than one sound source the sound source determiner may be configured to identify which sound source microphone the combined audio signal corresponds to.

In some embodiments the sound source microphone determiner is configured to identify such frequency bands in the microphone array capture which contain mostly the sound source microphone signal for calculating for the sub-band b <MAT>.

Where c is above a predetermined threshold, then the sound source microphone determiner is configured to conclude that <MAT> is dominated by Yb(n) and assign yi = b. The value of the threshold may in some embodiments be obtained by analyzing typical values of the correlation and selecting such a value which typically indicates a high probability of match (say <NUM>% of values above the threshold are true matches). It is noted that a band b does not need to correspond to any sound source microphone captured source, for example the audio signal may be dominated by noise. In this case it is not used for updating information for any sound source.

As a result of the determination, we obtain a subset {yi}, yi∈[<NUM>,. , B], of frequency bands b = <NUM>,. , B - <NUM> such that sound Y is likely the dominant sound captured by the microphone array at each selected band yi.

In some embodiments the audio based distance estimator <NUM> comprises a direction of arrival combiner <NUM>. The direction of arrival (DOA) combiner <NUM> may be configured to receive the DOA estimates for the sub-bands and the indications of which sub-bands are associated with which sound source microphone. The DOA combiner may then calculate the circular variance syi of the DOA estimates over the identified bands associated with a defined sound source microphone. This in some embodiments may be performed over the bands {yi}: <MAT> where card denotes the cardinality of a set. The variance may be output to a distance determiner <NUM>.

In some embodiments the audio based distance determiner comprises a distance determiner <NUM>. The distance determiner is configured to receive the variance values and from these determine the distance associated with the sound source. In some embodiments as the variance correlates with the distance of the sound source captured by Y, with increasing variance corresponding to increased distance and vice versa a model may be constructed which provides an estimate of the distance given a certain value of variance of the DOA. For example the method described in an earlier invention by Laitinen & Vilkamo to obtain an estimate of the distance for the source given the variance of the DOA.

In some embodiments the HAIP based DOA estimate can be used for selecting the set of bands which are used for correlation analysis (for finding whether the external mic captured sources is dominating these bands). In such embodiments the system may pick those bands b where the dominant DOA based on audio analysis matches the HAIP provided DOA, and the correlation analysis may be performed only on these bands.

The distance estimate may be passed to the position tracker.

With respect to <FIG> a flow diagram of the operation of the apparatus shown in <FIG> is shown.

The input microphone audio signals are first processed (if they are time domain audio signals) by the application of a STFT (or other suitable time to frequency domain transform).

The application of the STFT to the microphone audio signals is shown in <FIG> by step <NUM>.

The audio signals may then be divided into sub-bands. The dividing of the audio signals into sub-bands is shown in <FIG> by step <NUM>.

The selection of a sub-band to analyse is shown in <FIG> by step <NUM>. The step <NUM> is the first operation in a loop shown from step <NUM> to <NUM> and which may be repeated for all of the sub-bands.

The determination of the sub-band direction of arrival estimate is shown in <FIG> by step <NUM>.

The calculation of the sum or combined signal for the sub-band is shown in <FIG> by step <NUM>.

The determination of whether the sub-band combined audio signal corresponds to one of the sound source microphone audio signal (and which one if there is more than one) is shown in <FIG> by step <NUM>.

As discussed previously this loop may now pass back to step <NUM> to analyse a further sub-band.

The combination or generation of the set of DOA estimates associated with a specific sound source microphone is shown in <FIG> by step <NUM>.

The determination of the variance of the DOA estimates within the set is shown in <FIG> by step <NUM>.

The determination or updating of the distance based on the variance is shown in <FIG> by step <NUM>.

With respect to <FIG> a schematic view of the position tracker is shown in further detail.

In some embodiments the position tracker comprises a full-spherical co-ordinate determiner <NUM>. The full-spherical co-ordinate determiner <NUM> is configured to receive the sound source (or sound source microphones) direction of arrival information from the positioner and furthermore the distance determined by the audio based distance estimator and generate a full spherical co-ordinate identifier associated with the sound source. Thus in some embodiments the full-spherical co-ordinate determiner <NUM> is configured to output for each of the sound sources an orientation, elevation and distance estimate.

The operation of the full-spherical co-ordinate determiner <NUM> is shown with respect to <FIG>.

The operation of receiving the audio distance estimate and the positioner (HAIP) direction of arrival estimates is shown in <FIG> by step <NUM>.

The operation of generating or compiling the full spherical co-ordinates associated with the sound source based on the received the audio distance estimate and the positioner (HAIP) direction of arrival estimates is shown in <FIG> by step <NUM>.

In some embodiments the position tracker further comprises a direction of arrival (DOA) updater <NUM>. The DOA updater is configured in some embodiments to receive the positioner estimated direction of arrival values and furthermore the audio based direction of arrival values and generate an updated or combined direction of arrival value. The system may also pool the azimuth & elevation estimates together. In other words the position tracker may use the audio based DOA estimates for bands {yi} to improve the indoor-positioning based DOA estimates. For example, a weighted average of audio based DOA estimates and the audio based DOA estimates may be calculated and used.

<FIG> for example shows the operation of the DOA updater shown in <FIG> according to some embodiments.

First the positioning DOA estimates and the audio based DOA estimates are received as shown in <FIG> by step <NUM>.

Then the DOA estimate is updated using a combination of the positioning DOA estimates and the audio based DOA estimates as shown in <FIG> by step <NUM>.

With respect to <FIG> an example electronic device which may be used as the mixer and/or ambience signal generator is shown. The device may be any suitable electronics device or apparatus. For example in some embodiments the device <NUM> is a mobile device, user equipment, tablet computer, computer, audio playback apparatus, etc..

The device <NUM> may comprise a microphone <NUM>. The microphone <NUM> may comprise a plurality (for example a number N) of microphones. However it is understood that there may be any suitable configuration of microphones and any suitable number of microphones. In some embodiments the microphone <NUM> is separate from the apparatus and the audio signal transmitted to the apparatus by a wired or wireless coupling. The microphone <NUM> may in some embodiments be the microphone array as shown in the previous figures.

The microphone may be a transducer configured to convert acoustic waves into suitable electrical audio signals. In some embodiments the microphone can be solid state microphones. In other words the microphone may be capable of capturing audio signals and outputting a suitable digital format signal. In some other embodiments the microphone <NUM> can comprise any suitable microphone or audio capture means, for example a condenser microphone, capacitor microphone, electrostatic microphone, Electret condenser microphone, dynamic microphone, ribbon microphone, carbon microphone, piezoelectric microphone, or microelectrical-mechanical system (MEMS) microphone. The microphone can in some embodiments output the audio captured signal to an analogue-to-digital converter (ADC) <NUM>.

The device <NUM> may further comprise an analogue-to-digital converter <NUM>. The analogue-to-digital converter <NUM> may be configured to receive the audio signals from each of the microphone <NUM> and convert them into a format suitable for processing. In some embodiments where the microphone is an integrated microphone the analogue-to-digital converter is not required. The analogue-to-digital converter <NUM> can be any suitable analogue-to-digital conversion or processing means. The analogue-to-digital converter <NUM> may be configured to output the digital representations of the audio signal to a processor <NUM> or to a memory <NUM>.

In some embodiments the device <NUM> comprises a memory <NUM>. In some embodiments the at least one processor <NUM> is coupled to the memory <NUM>. The memory <NUM> can be any suitable storage means. In some embodiments the memory <NUM> comprises a program code section for storing program codes implementable upon the processor <NUM>. Furthermore in some embodiments the memory <NUM> can further comprise a stored data section for storing data, for example data that has been processed or to be processed in accordance with the embodiments as described herein. The implemented program code stored within the program code section and the data stored within the stored data section can be retrieved by the processor <NUM> whenever needed via the memory-processor coupling.

In some embodiments the device <NUM> comprises a user interface <NUM>. The user interface <NUM> can be coupled in some embodiments to the processor <NUM>. In some embodiments the processor <NUM> can control the operation of the user interface <NUM> and receive inputs from the user interface <NUM>. In some embodiments the user interface <NUM> can enable a user to input commands to the device <NUM>, for example via a keypad. In some embodiments the user interface <NUM> can enable the user to obtain information from the device <NUM>. For example the user interface <NUM> may comprise a display configured to display information from the device <NUM> to the user. The user interface <NUM> can in some embodiments comprise a touch screen or touch interface capable of both enabling information to be entered to the device <NUM> and further displaying information to the user of the device <NUM>. In some embodiments the user interface <NUM> may be the user interface for communicating with the position determiner as described herein.

In some implements the device <NUM> comprises a transceiver <NUM>. The transceiver <NUM> in such embodiments can be coupled to the processor <NUM> and configured to enable a communication with other apparatus or electronic devices, for example via a wireless communications network. The transceiver <NUM> or any suitable transceiver or transmitter and/or receiver means can in some embodiments be configured to communicate with other electronic devices or apparatus via a wire or wired coupling.

For example as shown in <FIG> the transceiver <NUM> may be configured to communicate with the renderer as described herein.

The transceiver <NUM> can communicate with further apparatus by any suitable known communications protocol. For example in some embodiments the transceiver <NUM> or transceiver means can use a suitable universal mobile telecommunications system (UMTS) protocol, a wireless local area network (WLAN) protocol such as for example IEEE <NUM>. X, a suitable short-range radio frequency communication protocol such as Bluetooth, or infrared data communication pathway (IRDA).

In some embodiments the device <NUM> may be employed as at least part of the renderer. As such the transceiver <NUM> may be configured to receive the audio signals and positional information from the microphone/close microphones/position determiner as described herein, and generate a suitable audio signal rendering by using the processor <NUM> executing suitable code. The device <NUM> may comprise a digital-to-analogue converter <NUM>. The digital-to-analogue converter <NUM> may be coupled to the processor <NUM> and/or memory <NUM> and be configured to convert digital representations of audio signals (such as from the processor <NUM> following an audio rendering of the audio signals as described herein) to a suitable analogue format suitable for presentation via an audio subsystem output. The digital-to-analogue converter (DAC) <NUM> or signal processing means can in some embodiments be any suitable DAC technology.

Furthermore the device <NUM> can comprise in some embodiments an audio subsystem output <NUM>. An example as shown in <FIG> shows the audio subsystem output <NUM> as an output socket configured to enabling a coupling with headphones <NUM>. However the audio subsystem output <NUM> may be any suitable audio output or a connection to an audio output. For example the audio subsystem output <NUM> may be a connection to a multichannel speaker system.

In some embodiments the digital to analogue converter <NUM> and audio subsystem <NUM> may be implemented within a physically separate output device. For example the DAC <NUM> and audio subsystem <NUM> may be implemented as cordless earphones communicating with the device <NUM> via the transceiver <NUM>.

Although the device <NUM> is shown having both audio capture, audio processing and audio rendering components, it would be understood that in some embodiments the device <NUM> can comprise just some of the elements.

Claim 1:
An apparatus comprising means for:
receiving at least two audio signals from a microphone array located within a sound scene;
receiving at least one further audio signal associated with at least one sound source from the sound scene;
determining (<NUM>) at least one portion of the at least two audio signals from the microphone array corresponding to the at least one further audio signal, the means for determining (<NUM>) at least one portion of the at least two audio signals from the microphone array corresponding to the at least one further audio signal being for identifying
at least one sub-band of the at least two audio signals, the at least one sub-band is associated with the at least one sound source; and
determining (<NUM>) a distance estimate from the microphone array to the at least one sound source based on the at least one portion of the at least two audio signals from the microphone array corresponding to the at least one further audio signal, wherein the means for determining (<NUM>) the distance estimate from the microphone array to the at least one sound source based on the at least one portion of the at least two audio signals from the microphone array corresponding to the at least one further audio signal further comprises means for:
determining (<NUM>) at least one audio sub-band positioning system direction of arrival estimate based on the identified sub-band of the at least two audio signals;
determining (<NUM>) a variance based on the at least one audio sub-band positioning system direction of arrival estimate; and
determining (<NUM>) the distance estimate from the microphone array to the at least one sound source based on the determined variance.