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 as 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.

One of the aspects of SAM system is the generation and use of volumetric virtual sound sources. The term volumetric virtual sound source refers to a virtual sound source with a spatial volume, whereas a point-like virtual source is perceived from a single point in space. Volumetric virtual sound sources are useful in various applications including virtual and augmented reality and computer gaming. They enable creative opportunities for sound engineers and facilitate more realistic representation of sounds with a natural size, such as large sound-emitting objects. Consider, for example, a fountain, sea, or a large machine. Such volumetric virtual sound sources are discussed in Pihlajamäki et al. Synthesis of Spatially Extended Virtual Sources with Time-Frequency Decomposition of Mono Signals, JAES <NUM>.

The creation of volumetric virtual sound sources can be implemented by creation of sounds with perceived spatial extents as the ability of humans to perceive sounds at different distances is not good. A sound with a perceived spatial extent may be surrounding the listener or it may have a specific width.

An effect in human hearing called summing localization enables humans to perceive simultaneously presented coherent audio signals as a virtual sound source between the original sources. If the coherence is lower, the signals may be perceived as separate audio objects or as a spatially extended auditory effect. Coherence can be measured with the interaural cross-correlation value between signals (IACC). When played identical signals, where the IACC value equals one, from both headphones, humans will perceive an auditory event in the center of the head. When played non-identical signals, where the IACC value equals zero, from both headphones, one auditory event will be perceived near each ear. When the IACC value is between one and zero, humans may perceive a spatially extended or spread auditory event inside the head, with the extent varying according to the IACC value.

To synthesize a sound source with a perceived spatial extent, one approach is to divide to signal into non-overlapping frequency bands, and then present the frequency bands at distinct spatial positions around the listener. The area from which the frequency bands are presented may be used to control the perceived spatial extent. Special care needs to be taken on how to distribute the frequency bands, such that no degradation in the timbre of the sound occurs, and that the sound is perceived as a single spatially extended source rather than several sound objects.

When spatially extending a sound source the audio is split into frequency bands. These bands are then rendered from a number of different directions defined by the desired spatial extent. The frequency bands are divided into the different directions using what is called a low-discrepancy sequence, e.g., a Halton sequence. The sequence provides random-looking uniformly-distributed frequency-component sets for the different directions. Thus, for each direction we have a filter which selects frequency components of the original signal based on the Halton sequence. Using these filters, we have signals for the different directions that, ideally, have similar frequency content (shape) as the original signal, but do not contain common frequency components with each other. This results in the sound being heard as having spatial extent.

However, the signal content to be spatially extended can sometimes be less than optimal for performing the effect. Percussive or impulsive sounds lose the "impact" in their onsets, speech and vocal content can become less intelligible, and timbre of "peaky" sounds can change in ways that is undesired. These additional effects are often undesired and should be avoided.

<NPL> discloses that a synthesis of volumetric virtual sources is a useful technique for auditory displays and virtual worlds. In the article, this technique is revisited in detail and the effect of different parameters is examined to ultimately achieve optimal quality and perception in all situations. The results of a series of informal and formal experiments are presented and suggest that the method is very viable in many cases. Furthermore, it is shown that different distribution widths can be produced with the method as well.

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

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

The following describes in further detail suitable apparatus and possible mechanisms for the provision of effective audio signal generation including the generation of volumetric virtual sound sources from the capture of audio signals. The volumetric virtual sound sources being suitable for processing and/or mixing for generation of immersive sound scenes.

A conventional approach to the capturing and mixing of sound sources with respect to an audio background or environment audio field signal would be for a professional producer to utilize 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.

The concepts as discussed in detail hereafter is a system that detects whether an input or captured audio signal comprises any 'problematic' attributes (e.g., impulsiveness and peakiness). The input signal can be a recorded or synthetic monophonic sound with any suitable content. If the input signal is determined to be problematic, the embodiments as described herein may be configured to modify parameters associated with a spatial extent effect and further be configured to mix the spatial extent processed audio signal with the original audio signal to avoid adverse effects to the timbre of the spatial extent processing being output. This approach preserves the timbre which is generally a desired action and one of the main factors used by audio professionals in determining whether they are willing to use an effect.

With respect to <FIG> is shown an example system for controlling a spatial extent processing of an input audio signal.

The system comprises an audio signal input. In the example shown in <FIG> the audio signal input is a mono audio signal. The mono audio signal may be one from a microphone such as an external microphone. The external microphone may be any microphone external or separate to a microphone array (for example a Lavalier microphone) which may capture a 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 external microphones 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 close microphone may be configured to output the captured audio signals to a mixer. The external microphone may be connected to a transmitter unit (not shown), which wirelessly transmits the audio signal to a receiver unit (not shown).

In some embodiments the external microphone, mic sources 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 microphone source. Thus for example the external microphone comprises or is associated with a microphone position tag. The microphone position tag may be configured to transmit a radio signal such that an associated receiver may determine information identifying the position or location of the close microphone. It is important to note that microphones worn by people can be freely moved in the acoustic space and the system supporting location sensing of wearable microphone has to support continuous sensing of user or microphone location. The close microphone position tag may be configured to output this signal to a position tracker. 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.).

Although in the following example the mono audio signal is determined from an external microphone, the audio signal may be a stored audio signal or a synthetic (for example a generated or significantly processed audio signal).

The system comprises an attribute analyser <NUM>. The attribute analyser is configured to receive the mono audio signal from the microphone. The attribute analyser <NUM> may be configured to comprise one or more 'problematic' attribute determiners configured to determine whether the input audio signal comprises a 'problematic' component and be configured to output an attribute parameter value which may be passed to an attribute mapper <NUM>.

Thus for example as shown in <FIG>, the attribute analyser <NUM> may comprise a peakiness analyser <NUM>. The peakiness analyser <NUM> may be configured to determine whether the input audio signals comprises a peakiness component and the degree of peakiness that the input audio signal contains. Peakiness is signal feature that measures how "spiky" the time domain signal is. A peaky signal usually has many frequency components in phase and can be also described as audibly "buzzy". This timbral quality may be lost quickly where the phase of the audio signal is 'touched' or the signal is otherwise modified. Thus special care should be taken when processing audio signals comprising significant peakiness components. Peakiness and the related phase-alignment can be analyzed using a simple hearing model as shown, for example, in <NPL>.

The hearing model converts the input audio signal into neural activation patterns. A peaky signal, using this model, has a high time alignment in activations and modifying or processing the signal can reduce this alignment. Thus, modification or processing an audio signal with determined peakiness components can produce more perceptual effects than an audio signal with less peakiness. Thus, high peakiness is considered as a problematic attribute.

Furthermore as shown in <FIG>, the attribute analyser <NUM> may comprise an impulsiveness analyser <NUM>. The impulsiveness analyser <NUM> may be configured to determine whether the input audio signals comprises an impulsiveness component and the degree of impulsiveness that the input audio signal contains.

The impulsiveness analyser <NUM> in some embodiments may analyse the impulsiveness of the input audio signal with a two-window detector. The analyser <NUM> may thus be configured to generate two running rectangular time domain energy windows with equal area but with different lengths. For example in some embodiments the analyser comprises a first 'long' window of <NUM> time samples and a second 'short' window of <NUM> time samples. The analyser may then be configured to calculate the energy (for example by determining a sum of squared sample values) for both windows. The short window energy value may be scaled to match the long window energy value. For example in some embodiments the time samples are multiplied by <NUM> before determining the energy value or the output energy value scaled similarly. The resulting energy values can be directly compared. Where the energy of the short window values is significantly larger (such as defined by threshold value) than the long window energy values, then the signal is very likely to have an impulse detected by the short window.

Also as shown in <FIG>, the attribute analyser <NUM> may comprise a speech or voice activity detector <NUM>. The speech detector <NUM> may be configured to determine whether the input audio signals comprise speech components. Speech within the input audio signal may be problematic as human hearing is tuned to listening for it. For example, processed audio signals comprising speech require special care otherwise they can sound very unnatural and disturbing. In some embodiments the detector <NUM> comprises a signal classifier and corresponding outputs from this classifier may be mapped to parameters to control the mixing process so that the output audio signal sounds spatially extended but still natural. For example in some embodiments the detector comprises a deep neural network trained to classify between speech and other signal types. This training may be for example, using any of the above features or by signal spectrum analysis directly. In some embodiments a voice activity detector or speech/music discriminator may be implemented to complement or replace the classifier described above. Thus in some embodiments a Voice Activity Detector (VAD) may be configured to first perform a noise reduction, calculate some features or quantities from a section of the input signal, and then apply a classification rule to classify the section as speech or non-speech. In some embodiments this classification rule is based on determining a value exceeds a threshold. In some embodiments there may be some feedback in this sequence, in which the VAD decision is used to improve the noise estimate in the noise reduction stage, or to adaptively vary the threshold(s). These feedback operations improve the VAD performance in non-stationary noise (i.e. when the noise varies a lot). Some VAD methods may formulate the decision rule on a frame by frame basis using instantaneous measures of the divergence distance between speech and noise. The different measures which are used in VAD methods may include spectral slope, correlation coefficient, log likelihood ratio, cepstral, weighted cepstral, and modified distance measures. In some embodiments the signal classifier may comprise a percussiveness detector. The percussiveness detector may be configured to perform an analysis of percussiveness, using, for example the pulse-metric characterization such as described within CONSTRUCTION AND EVALUATION OF A ROBUST MULTIFEATURE SPEECH/MUSIC DISCRIMINATOR, Speech & music discrimination, pulse-metric feature available from https://www. edu/~dpwe/papers/ScheiS97-mussp.

In some embodiments the system may comprise an attribute mapper <NUM>. The attribute mapper <NUM> may be configured to receive the output from the attribute analyser <NUM> (and the one or more 'problematic' attribute determiners) and be configured to determine at least one mix/control parameter based on the determined at least one attribute parameter value from the analyser <NUM>. In some embodiments the attribute mapper comprises at least one attribute mapper element, where each of the elements is associated with a specific attribute analyser element and is configured to output separate mix/control parameters for each element. The at least one mix/control parameter may be passed to a mixer controller/extent compensator <NUM>.

Thus for example in some embodiments the attribute mapper <NUM> comprise elements which map the output of the analyser elements/detector to a range from <NUM> to <NUM>.

Thus for example the attribute mapper <NUM> comprises a peakiness mapper <NUM> configured to receive the output of the peakiness analyser <NUM> and map the output to a range between <NUM> to <NUM>. Thus the peakiness mapper <NUM> may be configured to define the peakiness attribute parameter associated with an input audio signal to be between <NUM> and <NUM> with <NUM> being fully peaky, i.e., the above model has full time alignment in neural activations, and <NUM> being "unpeaky", i.e., the neural activations are completely misaligned in time.

Similarly the attribute mapper comprises an impulsiveness mapper <NUM>. The impulsiveness mapper <NUM> may be configured to count the number of detected impulses from the impulsiveness analyser <NUM> for a defined time frame of consecutive windows. This number can be used as the estimate of impulsiveness of the signal. A high impulsiveness may be considered as a problematic attribute for the signal. In some embodiments the analyser may be configured to define the impulsiveness parameter value to be a value between <NUM> and <NUM>. An impulsiveness parameter value of <NUM> signifies that there were no detected impulses in the signal within the defined time frame and the impulsiveness parameter value <NUM> signifies that there was maximum allowed number of impulses (for example the 'maximum' number may be <NUM>) in the input signal within the defined time frame. This 'maximum' number limit may in some embodiments be user defined or may be a predefined value that is deemed as a good limit for impulses in a certain time frame. An example time frame is <NUM>.

Also the attribute mapper may comprise a speech mapper <NUM>. The speech mapper may receive the output of the speech detector <NUM> and output a speech or voice attribute parameter with a value between <NUM> and <NUM>. In some embodiments as signal type is relatively constant measure the classified signal types has a constant value that they are mapped to. This ensures that an appropriate amount of later algorithm modification is performed for each signal type. For example, input audio signals with determined speech components could have a value of <NUM> and input audio signals with determined drum components may have a value of <NUM>.

The system comprises a mixer controller/extent compensator <NUM>. The mixer controller/extent compensator <NUM> may be configured to receive the outputs from the attribute mapper in the form of mix/control parameters and generate suitable mix controls for a mapper <NUM> and/or spatial extent compensation control for a spatial extent synthesizer <NUM>.

In some embodiments the mixer controller/extent compensator <NUM> may be configured to combine multiple mapped attribute values into one 'problematic' signal estimate. This combination may be a maximum value from all of the mapped attribute values as thus would be the maximum requirement for any spatial extent algorithm modification to avoid undesired changes in signal timbre. The mixer controller/extent compensator <NUM> is configured to generate at least one control signal based on the determined mapped attribute values and these control signals used to control the operation of the spatial extent synthesizer and/or the mixer. In other words the spatial extent synthesizer operation may be controlled based on the attribute values and/or the mixer is similarly controlled based on the attribute values.

The system comprises a spatial extent synthesizer <NUM>. The spatial extent synthesizer <NUM> is configured to receive the input audio signal and generate a spatially extended audio signal. In some embodiments the spatial extent synthesizer <NUM> is configured to receive a further input from the mixer controller/extent compensator <NUM>. The spatial extent synthesizer <NUM> is configured in some embodiments to output the spatially extended audio signal to a mixer <NUM>.

The system comprises a mixer <NUM> (or an adaptive combiner). The mixer <NUM> is configured to receive the input audio signal, the spatially extended audio signal and a mixer control input from the mixer controller/extent compensator <NUM>. The mixer is configured to combine or mix the input audio signals and the spatially extended audio signal based on the mixer control input to generate a suitable output audio signal.

The mixer <NUM> is configured to mix the original input signal and spatially extended signal together. As the spatial extent effect does not cause delay, the signals can be mixed directly together. The original signal may be mixed into the centre direction of the spatially extended signal and this direction will be the one from which the sound is perceived to come from. Based on the output of the mixer controller (which is based on the attribute analyser and attribute mapper) the balance between signals is varied. If the signal is not problematic in other words the 'combined' attribute value is low and closer to <NUM> then little to none of the original signal is added to the mix. On the other hand, if the signal is very problematic, in other words the 'combined' attribute value is high and closer to <NUM>, the mixer is configured to raise the amount of original signal in the mix so that it is equally loud (for example a <NUM>/<NUM> mix) compared to the spatially extended signal. This mix control attempts to reduce perceivable problems and makes the timbre of the sound more similar to the original signal. However, it may also reduce the perceived spatial extent of the combined signal and may require extent compensation in the extent synthesis module by modifying the parameters as described hereafter.

With respect to <FIG> an example spatial extent synthesiser <NUM> is shown in further detail. As described herein the spatial extent synthesiser <NUM> receives the input audio signal and spatially extends the audio signal to a defined (for example <NUM> degree) spatial extent using methods for spatial extent control. In other words it takes as input a mono sound source audio signal and spatial extent parameters (width, height and depth).

In some embodiments where the audio signal input is a time domain signal the spatial extent synthesiser <NUM> comprises a suitable time to frequency domain transformer. For example as shown in <FIG> the spatial extent synthesiser <NUM> comprises a Short-Time Fourier Transform (STFT) <NUM> configured to receive the audio signal and output a suitable frequency domain output. In some embodiments the input is a time-domain signal which is processed with hop-size of <NUM> samples. A processing frame of <NUM> samples is used, and it is formed from the current <NUM> samples and previous <NUM> samples. The processing frame is zero-padded to twice its length (<NUM> samples) and Hann windowed. The Fourier transform is calculated from the windowed frame producing the Short-Time Fourier Transform (STFT) output. The STFT output is symmetric, thus it is sufficient to process the positive half of <NUM> samples including the DC component, totalling <NUM> samples. Although the STFT is shown in <FIG> any suitable time to frequency domain transform may be used.

In some embodiments the spatial extent synthesiser <NUM> further comprises a filter bank <NUM>. The filter bank <NUM> is configured to receive the output of the STFT <NUM> and using a set of filters generated based on a Halton sequence (and with some default parameters) generate a number of frequency bands <NUM>. In statistics, Halton sequences are sequences used to generate points in space for numerical methods such as Monte Carlo simulations. Although these sequences are deterministic, they are of low discrepancy, that is, appear to be random for many purposes. In some embodiments the filter bank <NUM> comprises set of <NUM> different distribution filters, which are used to create <NUM> different frequency domain signals where the signals do not contain overlapping frequency components. These signals are denoted Band <NUM> F <NUM><NUM> to Band <NUM> F <NUM><NUM> in <FIG>. The filtering can be implemented in the frequency domain by multiplying the STFT output with stored filter coefficients for each band.

In some embodiments the spatial extent synthesiser <NUM> further comprises a spatial extent input <NUM>. The spatial extent input <NUM> may be configured to define the spatial extent of the audio signal.

Furthermore in some embodiments the spatial extent synthesiser <NUM> may further comprise an object position input/determiner <NUM>. The object position input/determiner <NUM> may be configured to determine the spatial position of sound sources. This information may be determined in some embodiments by the sound object processor.

In some embodiments the spatial extent synthesiser <NUM> may further comprise a band position determiner <NUM>. The band position determiner <NUM> may be configured to receive the outputs from the object position input/determiner <NUM> and the spatial extent input <NUM> and from these generate an output passed to the vector base amplitude panning processor <NUM>.

In some embodiments the spatial extent synthesiser <NUM> may further comprise a vector base amplitude panning (VBAP) processor <NUM>. The VBAP <NUM> may be configured to generate control signals to control the panning of the frequency domain signals to desired spatial positions. Given the spatial position of the sound source (azimuth, elevation) and the desired spatial extent for the source (width in degrees), the system calculates a spatial position for each frequency domain signal. For example, if the spatial position of the sound source is zero degrees azimuth (front), and spatial extent <NUM> degrees, the VBAP may position the frequency bands at positions azimuth <NUM>, <NUM>, <NUM>, <NUM>, <NUM>, -<NUM>, -<NUM>, -<NUM>, -<NUM> degrees. Thus, we use a linear allocation of bands around the source position, with the span defined by the spatial extent.

The VBAP processor <NUM> may therefore be used to calculate a suitable gain for the signal, given the desired loudspeaker positions. VBAP processor <NUM> may provide gains for a signal such that it can be spatially positioned to a suitable position. These gains may be passed to a series of multipliers <NUM>. In the following example the spatial extent synthesiser (or spatially extending controller) is implemented using a vector based amplitude panning operation. However it is understood that the spatial extent synthesis or spatially extending control may be implementation agnostic and any suitable implementation used to generate the spatially extending control. For example in some embodiments the spatially extending control may implement direct binaural panning (using Head related transfer function filters for directions), direct assignment to the output channel locations (for example direct assignment to the loudspeakers without using any panning), synthesized ambisonics, and wave-field synthesis.

In some embodiments the spatial extent synthesiser <NUM> may further comprise a series of multipliers <NUM>. In <FIG> is shown one multiplier for each frequency band. Thus the series of multipliers comprise multipliers <NUM><NUM> to <NUM><NUM>, however any suitable number of multipliers may be used. Each frequency domain band signal may be multiplied in the multiplier <NUM> with the determined VBAP gains.

The products of the VBAP gains and each frequency band signal may be passed to a series of output channel sum devices <NUM>.

In some embodiments the spatial extent synthesiser <NUM> may further comprise a series of sum devices <NUM>. The sum devices <NUM> may receive the outputs from the multipliers and combine them to generate an output channel band signal <NUM>. In the example shown in <FIG>, a <NUM> loudspeaker format output is implemented with outputs for front left (Band FL F <NUM><NUM>), front right (Band FR F <NUM><NUM>), rear left (Band RL F <NUM><NUM>), and rear right (Band RR F <NUM><NUM>) channels which are generated by sum devices <NUM><NUM>, <NUM><NUM>, <NUM><NUM> <NUM><NUM> respectively. In some other embodiments other loudspeaker formats or number of channels can be supported.

Furthermore in some embodiments other panning methods can be used such as panning laws, or the signals could be assigned to the closest loudspeakers directly.

In some embodiments the spatial extent synthesiser <NUM> may further comprise a series of inverse Short-Time Fourier Transforms (ISTFT) <NUM>. For example as shown in <FIG> there is an ISTFT <NUM><NUM> associated with the FL signal an ISTFT <NUM><NUM> associated with the FR signal, an ISTFT <NUM><NUM> associated with the RL signal output and an ISTFT <NUM><NUM> associated with the RR signal. In other words it provides N component audio signals to be played from different directions based on the spatial extent parameters. The signals are subjected to Inverse Short-Time Fourier Transform (ISTFT) and overlap-added to produce time-domain outputs.

As adding the original signal into the mix draws perception of that sound source towards that direction, the resulting spatial extent perception is less than intended. This mix 'narrowing' can be compensated by making the spatial extent actually wider than the originally specified or determined value.

Thus for example where there is more original signal in the mix then the mixer controller/extent compensator may be configured to compensate for the narrowing by controlling the spatial extent synthesiser <NUM> to produce a wider or compensated spatially extended audio signal.

This increase in extent may for example be performed in some embodiments by increasing the extent directly from the mixing value so that where there is <NUM>% original signal there is no increase in extent and where the mix comprises <NUM>% of the original audio signal the spatial extent synthesiser would always controlled to produce a full <NUM>° extent spatially extended audio signal. In some embodiments the increase is linear interpolation between these values.

In some embodiments the spatial extent synthesiser <NUM> is configured to apply a predefined lookup table based on the current extent that has been perceptually evaluated to be correct. This lookup works so that there is a defined modified extent value for each pair of original extent and current mixing value.

In some embodiments a predefined extent value is applied to the spatial extent synthesiser associated with a specific signal type.

Examples of this extending of the spatial extent processing is shown in <FIG>. For example where the analyser determines that the input audio signal has a maximum attribute value of <NUM> shown by arrow <NUM> in <FIG>, then a normal spatial extent synthesis can be performed as shown in <FIG> by the extended arc <NUM>. When the input has a maximum attribute of <NUM> shown by arrow <NUM> then the output of the spatial extent synthesis is configured to extend the spatial extension, as shown by the further extended arc <NUM> and the direction arrow <NUM> representing the original direction. When the input has a maximum attribute of <NUM> then the spatial extent synthesis may be configured to extend the spatial extension to produce a <NUM> degree spatially extended audio signal as represented by the full circle <NUM> and the direction arrow representing the original signal.

Example use cases of the system shown in <FIG> which are produced when a user, who may be an audio professional who can aurally detect changes in a sound signal when it is ran through an audio effect, is operating the system is described hereafter. These changes may not necessarily be perceived by a non-professional user.

In some embodiments the user may want to create a spatial extent for a trumpet track. This signal is detected as peaky. When it goes normally through the system, it would become less sharp and the user does not want that. They decide to apply the mixer where <NUM>% of the original signal is mixed in to the output to retain better the qualities of the trumpet.

A further example could be where a user wants to create surrounding spatial extent for a drumming track with congas. These sounds are detected to be clearly impulsive and would lose the impact if processed normally. Instead, the user uses the proposed system and add <NUM>% of the original signal into the output mix and makes the extent part completely surrounding. The resulting perceived extent is not completely surrounding but still very wide.

An additional use may be where the user wants to create spatially extended speech. Running speech directly through the spatial extent system creates an 'otherworldly' sound which is perceived as disturbing. A user however may use the system as it classifies this input audio signal as speech and instead adds <NUM>% of the original signal into the mix to retain the naturalness of the speech.

With respect to <FIG> an example electronic device which may be used as the mixer and/or system 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 for generating at least one audio signal associated with a sound scene, the apparatus comprising means configured to:
receive at least one audio signal;
analyse (<NUM>, <NUM>) the at least one audio signal to determine at least one attribute parameter, wherein the at least one attribute parameter is at least one of: a detection of voice activity within the at least one audio signal; a determination of peakiness within the at least one audio signal in the time domain; and a determination of impulsiveness within the at least one audio signal by comparing energy values of different time windows;
determine (<NUM>) at least one control signal based on the at least one attribute parameter;
generate (<NUM>) a spatially extended audio signal from the at least one audio signal based on the at least one control signal; and
combine (<NUM>), based on the at least one control signal, the at least one audio signal and the spatially extended audio signal to generate the at least one audio signal associated with the sound scene.