143 and/or containers from container set 144. It is understood that these VCEs may be stored as images and may be transferred among and between the various physical machine hosts, either as images or after instantiation of the VCE. Cloud orchestration module 141 manages the transfer and storage of images, deploys new instantiations of VCEs and manages active instantiations of VCE deployments. Gateway 140 is the collection of computer software, hardware, and firmware that allows public cloud 105 to communicate through WAN 102.

Some further explanation of virtualized computing environments (VCEs) will now be provided. VCEs can be stored as “images.” A new active instance of the VCE can be instantiated from the image. Two familiar types of VCEs are virtual machines and containers. A container is a VCE that uses operating-system-level virtualization. This refers to an operating system feature in which the kernel allows the existence of multiple isolated user-space instances, called containers. These isolated user-space instances typically behave as real computers from the point of view of programs running in them. A computer program running on an ordinary operating system can utilize all resources of that computer, such as connected devices, files and folders, network shares, CPU power, and quantifiable hardware capabilities. However, programs running inside a container can only use the contents of the container and devices assigned to the container, a feature which is known as containerization.

PRIVATE CLOUD 106 is similar to public cloud 105, except that the computing resources are only available for use by a single enterprise. While private cloud 106 is depicted as being in communication with WAN 102, in other embodiments a private cloud may be disconnected from the internet entirely and only accessible through a local/private network. A hybrid cloud is a composition of multiple clouds of different types (for example, private, community or public cloud types), often respectively implemented by different vendors. Each of the multiple clouds remains a separate and discrete entity, but the larger hybrid cloud architecture is bound together by standardized or proprietary technology that enables orchestration, management, and/or data/application portability between the multiple constituent clouds. In this embodiment, public cloud 105 and private cloud 106 are both part of a larger hybrid cloud.

Preferably, the present invention provides a method, and system, wherein the set of audio parameters comprise at least one from a list, the list comprising: sample rate; bit resolution; and filter setting.

Preferably, the present invention provides a method, and system, wherein, in response to determining a second distance from the object to the avatar, determining a further closest oscillator; updating the embedded data structure based on the second distance; and inserting the updated embedded data structure at a rate determined by the further closest oscillator.

Preferably, the present invention provides a method, and system, wherein the virtual environment is one of a list, the list comprising: virtual reality, VR, augmented reality, AR; and mixed reality, MR.

Preferably, the present invention provides a method, and system, wherein the audio parameters are dynamic, dependent on the distance.

Preferably, the present invention provides a method, and system, wherein the filter setting comprises a cutoff frequency, CF, set to no more than a Nyquist frequency of the audio data stream, wherein CF is a non-linear function of the distance.

Preferably, the present invention provides a method, and system, further comprising sending a request to receive an audio data stream from each object within a threshold range from the avatar.

Preferably, the present invention provides a method, and system, wherein the avatar is at a first location, an object of the set of objects is at a second location, and in response to determining that the first location is different from a second location, unsubscribing from the object.

Preferably, the present invention provides a method, and system, wherein each of the first location and the second location define volumes of the virtual environment.

Advantageously, the present invention provides a more accurate representation of audio in a metaverse that doesn't discard audio elements in the environment.

Advantageously, distribution of sounds are optimized for all users present. Advantageously, only the minimum required audio data is transmitted to the client, which minimizes bandwidth.

Advantageously, the present invention reduces the bandwidth of audio emitting objects through continuous adjustment of the sample rate Rs, bit resolution Rb and a filter setting Fs. The present invention transfers a resolution specification which is updated continuously and embedded into the stream at a certain rate. Advantageously, preferred embodiments of the present invention provide optimization of bandwidth for accurate and complete audio perception using a resolution specification embedded into a dynamic audio data stream to continuously adjust sampling frequency, bit resolution Rb and filter setting Fs of audio emitting objects in metaverse rather than just truncating/thresholding objects that are far away.

Advantageously, sound is reproduced in different ways depending on how far off the object is.

Although preferred embodiments are described in the context of a virtual reality (VR) environment, the skilled person will understand that the present invention also applies to other models of digital representations, such as AR and MR.

For the benefit of illustration, the following terms will be used.

A user refers to a human interacting with the VR environment. A client refers to the user's hardware/software system. An avatar is a virtual, digital representation of the user in the VR environment on a display of the user's client.

VR uses a number of sound techniques, such as Spatial Audio. Spatial audio aims to replicate how sound behaves in the real world. This involves simulating the direction, distance, and movement of sound sources relative to the user's position. Spatial audio techniques include: Binaural Audio; Head-Related Transfer Functions (HRTFs); Real-Time Rendering; Environmental Effects (for example, the sounds of busy city streets, or running water); Interactive Sound Design (for example, footsteps of moving user); and Integration with Visuals (for example, matching footsteps with associated animations).

Sample frequency (also known as sampling frequency) determines the rate at which a continuous audio signal is converted into a discrete digital representation and represents the number of samples captured per second.

To digitize analog audio signals, a continuous waveform is sampled at regular intervals determined by the sample frequency using an analogue to digital converter (ADC). The analog signal is measured at these discrete points in time, and each sample represents the amplitude of the signal at that instant.

Using the Nyquist-Shannon sampling theorem, to accurately represent a continuous signal in digital form, the sample frequency must be at least twice the highest frequency component present in the signal. This is known as the Nyquist frequency. Therefore, the sample frequency determines the maximum frequency that can be accurately represented in the digital signal.

When a digital audio signal is played back through a digital-to-analog converter (DAC), the samples are reconstructed into a continuous waveform. The sample frequency determines the rate at which these samples are converted back into analog voltages, influencing the fidelity and accuracy of the reconstructed signal.

The sample frequency also affects the frequency response of digital audio systems. Due to the Nyquist-Shannon theorem, frequencies above half the sample rate Rs (Nyquist frequency) cannot be accurately represented and may cause aliasing effects. Therefore, the sample frequency sets the upper limit of the frequency range that can be faithfully reproduced in the digital audio signal.

Higher sample frequencies generally result in higher audio quality and resolution, as they provide more samples per second to capture the nuances of the original analog signal. However, higher sample rate also requires more storage space and computational resources.

Bit resolution Rb in digital audio refers to the number of bits used to represent each sample of an audio signal. Bit resolution Rb determines the dynamic range, resolution, and precision of the digital audio signal, by specifying the number of quantization levels available to represent the amplitude of the signal. Bit resolution Rb is typically expressed as the number of bits per sample and is often referred to as the audio's “bit depth.” A higher bit depth allows for a greater dynamic range and more precise representation of the original analog signal. For example, a 16-bit audio signal has 16 bits per sample, allowing for 65,536 quantization levels.

Bit resolution Rb directly impacts the dynamic range of the digital audio signal. The dynamic range is the ratio between the loudest and quietest parts of the audio signal. A higher bit resolution Rb allows for a greater dynamic range because it provides more discrete amplitude levels to represent the signal's varying loudness levels.

Bit resolution Rb also affects the precision with which the amplitude of the audio signal can be represented. With a higher bit resolution Rb, the quantization levels (the discrete steps between minimum and maximum amplitude values) are smaller, resulting in finer detail and more accurate representation of the original analog signal.

Quantization noise refers to the error introduced when the continuous analog signal is quantized into discrete digital values. Higher bit resolutions result in smaller quantization errors and lower quantization noise, improving the signal-to-noise ratio and overall audio quality.

Higher bit resolution generally leads to higher perceived audio quality, as they allow for more accurate representation of the audio signal's dynamic range and nuances. However, the audible difference between different bit resolutions may vary depending on factors such as the quality of the audio equipment and the listener's sensitivity.

Common sample frequencies/bit resolutions used in digital audio include 44.1 kHz/16-bit (CD quality), 48 kHz/24-bit (standard for digital video and audio production), 96 kHz/32-bit (high-resolution audio). The choice of sample frequency/bit rate depends on factors such as the intended application, desired audio quality, and compatibility with other audio equipment and standards.

The bit rate of a digital audio signal refers to the number of bits processed or transmitted per unit of time, typically expressed in bits per second (bps) or kilobits per second (kbps). In digital audio, the bit rate is influenced by factors such as the bit depth, sample rate Rs, and the number of audio channels (mono vs. stereo, etc.). In uncompressed audio formats, bit rate is calculated by multiplying the sample rate Rs (in Hz) by the bit depth (in bits per sample) and the number of channels (e.g., mono=1, stereo=2).

In compressed audio formats, bit rate is typically specified as the average number of bits used to represent one second of audio data after compression.

The ‘cutoff frequency’ of a filter is the frequency at which the magnitude response of a filter is reduced by a certain level compared to its passband or peak magnitude. This level is typically defined as the half-power point, where the magnitude is decreased by −3 decibels (dB), or approximately 70.7% of the peak value.

The cutoff frequency is a critical parameter because it determines the transition between the passband (where the filter allows signals to pass with little attenuation) and the stopband (where the filter attenuates signals). In a low-pass filter, the cutoff frequency is the point beyond which higher frequencies are attenuated, while lower frequencies are allowed to pass. In a high-pass filter, the cutoff frequency is the point beyond which lower frequencies are attenuated, while higher frequencies are allowed to pass. In a band-pass filter, there are two cutoff frequencies: the lower cutoff frequency, and the upper cutoff frequency. These frequencies define the range of frequencies that are allowed to pass through the filter.

The specific definition of the cutoff frequency may vary depending on the type of filter and the design parameters chosen. Additionally, the term “cutoff frequency” can also be used in other contexts, such as in the context of signal analysis, where it refers to the highest frequency of interest in a signal or a system.

In analog systems, the cut-off frequency typically refers to the frequency at which a filter attenuates the signal by a certain amount, often −3 dB. However, in digital audio, filters are implemented using digital signal processing techniques, and the cut-off frequency is determined by the design of the digital filter. There are various types of digital filters used in audio processing, such as finite impulse response (FIR) filters and infinite impulse response (IIR) filters. Each type of filter has its own characteristics and methods for specifying the cut-off frequency.

The bandwidth of a digital audio signal is the range of frequencies that can be accurately represented or transmitted by the digital audio system. It is typically defined as the difference between the highest and lowest frequencies present in the audio signal.

Bandwidth is influenced by several factors in digital audio systems, including the sample rate Rs and the bit depth. As a consequence of the Nyquist-Shannon theorem, the bandwidth is limited by the sample rate Rs, extending from 0 Hz (DC) up to the Nyquist frequency.

For example, using the example values discussed above, in high-resolution audio with a sample rate Rs of 96 kHz, the Nyquist frequency is 48 kHz, so the bandwidth is 0 Hz to 48 kHz.

Additionally, the bit depth of a digital audio system affects its ability to accurately represent dynamic range, which indirectly influences the perceived bandwidth. Higher bit depths allow for more precise representation of amplitude levels, which can contribute to better resolution in the frequency domain and potentially extend the effective bandwidth of the system.

Bandwidth in digital audio refers to the frequency range that can be accurately represented or transmitted by a digital audio system and is influenced by factors such as the sample rate Rs and bit depth.

One common method of generating sound in VR is modal synthesis. Modal synthesis is a technique used in digital signal processing and computer music synthesis to simulate the sound of vibrating objects or systems, such as musical instruments or mechanical structures, by modelling their modes of vibration. In this method, the vibration of the object is represented by a set of vibrational modes, each with its own frequency, damping factor, and shape.

In modal synthesis, the sound of a complex vibrating object is decomposed into its individual vibrational modes and then the sound is synthesized by independently controlling each mode. This allows for a detailed and flexible synthesis of realistic and physically accurate sounds, making it particularly useful for simulating resonant systems.

Modal synthesis is commonly used in various applications, including virtual acoustic instruments, sound design for film and video games, and physical modelling synthesis in electronic music production. It offers a powerful and versatile approach to generating a wide range of sounds with realistic and expressive qualities.

Embodiments of the present invention optimize the distribution of audio data streams in a metaverse by using nodes in the environment, the nodes comprising low frequency oscillators (LFOs) that specify the characteristics of the environment in terms of how often the rate of individual objects can be updated, each object emitting an audio data stream with an embedded specification that adheres to the rate, the specification carrying a set of audio properties. An avatar, representing a user, switches to a new node (LFO) in the environment as the avatar moves in 3D space, affecting the stream and the specification of the audio emitting objects.

Embodiments of the present invention measures the distance between the object and the avatar. Audio is emitted from an object using a dynamic sample rate Rs, and a dynamic precision (bit resolution Rb). A lowpass filter is applied to the object, with the cutoff frequency (CF) set to no more than the Nyquist frequency, where CF is a non-linear function of the distance to the object.

Along with the emitted audio data stream, a “resolution specification” is transferred at a rate which is controlled by the appropriate LFO. The resolution specification is a data structure. The rate of the low frequency oscillator can be updated to cope with dynamic environments and events where sudden changes need to be captured correctly. For example, an event which includes many moving objects requires an update rate that is faster than a conference call.

The further away an object is from a user, the less bandwidth emitted audio needs to occupy. That is because a high sample rate Rs for objects that are far away is not required, they should only be noticeable in the distance, thus frequency content of such sounds can be filtered. Neither does the accuracy need to be high, so the number of bits (resolution) are reduced too. This mirrors real life; the farther away an object is, the more emitted sound is filtered by the surroundings.

In practice, an avatar in a VR environment will receive audio signals from multiple sources, which will need to be mixed according to the spatial and dynamic arrangement of object audio elements. Such mixing will involve the spatial audio techniques referred to above.

In a typical metaverse environment, users connect to a central server or a network of servers (herein referred to as the server) that host the virtual world using a multi-client multi-object (MCMO) model. This model supports multiple clients interacting with the same virtual environment, where each client 204 can interact with multiple objects simultaneously. The server acts as a central authority for managing and synchronizing the actions and movements of all users within the virtual space. Each user's device (such as a computer, VR headset, or mobile device) continuously sends positional and movement data to the server. This data includes information about the user's position, orientation, velocity, and actions within the virtual environment. The server receives positional data from all connected users and processes this information in real-time. The server broadcasts updates to all other users in the virtual space, ensuring that the world view of all users remains consistent and synchronized. A networking infrastructure allows communication and interaction between multiple clients and virtual objects. The infrastructure typically includes client-server communication protocols or peer-to-peer protocols.

The server is responsible for detecting and resolving collisions between users or objects within the environment. This can also include the server carrying out simulations according to the laws of physics of interactions between objects and users. The server calculates the effects of forces, collisions, and other physical interactions based on the positional data received from users and updates the virtual environment.

The server is also responsible for authenticating users and enforcing security measures to prevent unauthorized access within the metaverse environment.

For illustration purposes, the invention will be described in the context of users interacting with client software/hardware devices (hereinto termed ‘clients’). The clients interact with servers acting as the central authority.

In a preferred embodiment of the present invention, audio is delivered using a push model following a client-initiated subscription request. The server or the object responds to the subscription by streaming or transmitting the audio data to the client, which then plays the audio through the user's audio output device (e.g., headphones, speakers). This approach allows for more efficient use of network bandwidth and resources since audio data is only transmitted as required by the client. The server or object pushes audio data to users without the need for explicit requests from the client. Objects emitting audio continuously stream or transmit audio data to users or clients within their range or field of view. As users move within the virtual environment, they automatically receive audio data from nearby objects that emit sound. This method eliminates the need for the client to identify objects and then send a request.

In a typical VR MCMO model, both server and clients play their part in controlling sound. The client usually handles audio management, including volume control based on the user's perspective and position within the virtual environment. Clients use various techniques, such as spatial audio processing algorithms, to simulate sounds realistically, including Doppler effects as objects move relative to the user. Client-side sound management allows for a more appropriate processing based on the user's hardware capabilities. Server management of audio is also essential so that audio is consistent across multiple users.

An audio data stream 706 is a never-ending stream of data, compared to a file which has a defined end. The resolution specification 702 is defined by the object 212 when the client 204 uses the object's API to subscribe on an audio data stream 706. The audio data stream 706 receives the client's ID and can look up where in the VR environment the client 204 is located relative to the audio object 212.

Preferred embodiments of the present invention will be described with reference to a virtual environment comprising a set of avatars interacting with objects within the VR environment. Multiple users 202 on respective clients 204 can interact with multiple objects 212 simultaneously. For simplicity it is assumed that one user 202 is represented by one avatar 210 on one client 204. However, the skilled person would understand that a user could control multiple avatars 210 on a single client 204. The invention will be described in the context of one client 204 interacting with one server. In practice, a VR environment comprises multiple clients 204 and multiple servers 218.

FIG. 2 depicts levels of a VR environment 200, according to a preferred embodiment of the present invention. FIG. 3 depicts an exemplary VR environment 300, according to a preferred embodiment of the present invention.

FIG. 4, which should be read in conjunction with FIGS. 2, 3, 5 to 8, depicts a high-level exemplary schematic flow diagram of a method 400 depicting operation method steps for managing audio distribution in a virtual environment, according to a preferred embodiment of the present invention.

FIG. 5 also depicts a high-level exemplary schematic flow diagram 500 depicting method steps of FIG. 4 illustrating client and server steps, according to a preferred embodiment of the present invention.

FIG. 6 depicts an exemplary schematic diagram 600 of software elements, according to a preferred embodiment of the present invention.

FIG. 7 depicts an exemplary schematic diagram depicting further software constructs, according to a preferred embodiment of the present invention. FIG. 7 comprises software components 704, 712, 714 of software functionality 201. FIG. 7 also comprises elements of an audio data stream 706, and time illustration of an audio data stream 706-1. For clarity purposes audio data stream 706-1 does not depict changing images 208 and metadata 708.

FIG. 2 depicts levels of a VR environment (also known as a VR, AR, MR ‘world’), In a VR environment there are a number of logical/physical levels. At the highest level (not depicted), there is the VR environment itself. At a first level L1, a user interacts with the VR environment using a client 204 and displays an image 208 of the VR environment on a display 206 of the client 204. For simplicity, FIG. 4 depicts one avatar 210 and a clock object 212. The user 202 controls the avatar 210. The avatar 210 and the clock object 212 are in a first location 220 of the virtual environment. Such a location 220, 222 are also known as ‘rooms’ of the VR environment. Also depicted is a second location 222. Although the VR environment is depicted in two dimensions, VR environments may also be defined in three dimensions.

At a second level L2, a client agent 214 running on the client 204 interacts with a server agent 216 typically running on a server 218. The client agent 214 manages interactions that the user 202 has with the VR environment, as well as interacting with the server agent 216 that acts as the central authority discussed above. At a third level L3, the client 204 interact with the server 218.

Preferred embodiments will be described with reference to a ‘sound’ coming from the object 212 and being received by the avatar 210. The skilled person will understand that underlying this are interactions between a client agent 214, and a server agent 216, and between a client 204 and the server 218. A ‘sound’ apparently coming from an object is, in reality, a first audio data stream being transmitted from the server 218 to the client 204, processed on the client 204 and played though the sound device (for example, headphones, speakers, etc.) of the client 204, as an analogue signal. The sound device plays the sound by translating a second audio file by transmission of waves in some medium, such as air or water. The user 202 senses the energy waves and interprets them as the sound. The first audio file and the same audio file may be the same, or different if the client 204 manipulates the first audio file.

FIG. 3 depicts a VR environment. The VR environment comprises two avatars, avatar_1 210-1, and avatar_2 210-2. The VR environment also comprises four objects 212 that can act as sound sources. Object_1 212-1 issues human speech, containing high quality audio. Object_2 212-2 contains high frequency noise. Object_3 212-3 contains the full frequency range. Object_4 212-4 is moving object containing mid and low frequency noise.

Referring to FIG. 4, the method starts at step 402. At step 404 a user 202 logs in using the client 204 and authenticates with the VR environment. Step 406 represents a set up step for each avatar 210. At step 406, a receive component 602 of the client 204 receives a stream of audio data stream 706 from a stream agent component 704 of the server 218 with an image 208, objects 212, and other data. The image 208 also depicts the user's avatar 210-1, and other users' respective avatars 210-2. The other users' avatars 210-2 may also act as objects 212 with respect to the user's avatar 210-1. Included in the stream of image 208 is metadata 708 for each object specifying whether the object provides audio data 710. As part of step 406, the receive component receives 408 an image 208. At step 410 a render component 604 of the client 204 renders the image 208 on the display 206 of the client 204.

At step 412, a distance component 606 of the client 204 computes how far away their avatar 210 is to all audio emitting objects 212 that are within a threshold range in the VR environment.

At step 414 a subscribe component 608 of the client 204 subscribes to each object 212-1, 212-2, 212-3, 212-4 to receive an audio data stream 706 comprising audio data 710 according to a ‘resolution specification 702’. A subscription request may also comprise a set of audio parameters of the client, so that the resolution specification 702 can be appropriate. For example, if a client can only reproduce low frequencies, there is no point in sending high frequencies. A create agent 714 of the server agent 216 creates the resolution specification based on a distance between avatar and object, the distance, the resolution specification comprising a set of appropriate audio parameters.

The resolution specification 702 comprises information about the audio data 710, including, but not limited to, sample rate Rs, accuracy (bit resolution Rb) and filter setting Fs. The skilled person would understand that other audio parameters could be used. An embodiment of the invention uses a subscription based model, with events being posted to the client 204, acting as a subscriber. In this context the system works in a push mode. Each subscribed audio emitting object 212 gets an identification (ID) of the client 204 when the client 204 subscribes with the object 212, enabling the audio emitting object 212 to look up in the VR environment where the avatar 210 is located. The designer of an audio emitting object 212 provides an interface, which the client 204 uses to subscribe to audio data 710 according to a relationship that the client 204 has with the object 212. For example, if the distance is far away, there may be no need to reproduce high frequencies, etc. As part of the subscription the client 204 requests an optimum (lowest possible) sample rate Rs, bit resolution Rb and filter frequency.

For example, if there are only objects 212 that are not very close, the sample rate Rs, bit resolution Rb, and filter setting Fs can be reduced. Similarly, if there are only human objects 212 and low frequency sounds, then the sample rate Rs, bit resolution Rb, and filter setting Fs can be reduced. If they are far away, it can be reduced even more. In contrast if the avatar 210-1 is located close to an object 212 representing somebody who is, for example, playing an instrument, full fidelity might be needed.

The resolution specification 702 is embedded into the stream 706 and is updated at a certain interval. Thus, a stream 706 of audio data 710-1 is followed by a resolution specification 702-1, followed by another stream of data 710-2, followed by another resolution specification 702-2, etc. As can be seen in FIG. 7, the rate in which the embedded resolution specifications occur is controlled by the LFO 618.

When an object 212-3 moves away from the avatar 210-1 there is less need to transfer a full bandwidth audio data 710, because as the object 212-3 moves away it is naturally less audible to the avatar 210-1.

Avatar_1 210-1 is close to object_1 212-1, so Avatar_1 210-1 will subscribe to object_1 212-1 to request an audio data stream from that object_1 212-1 which has a high quality. However, object_1 212-1 doesn't need to reproduce more than, say, up to 16 kHz: object_1 212-1 is a speaker and human formants, and any noise emitted by a speaker's consonants are far below that frequency.

Avatar_1 210-1 is far away from object_4 212-4 and thus object_4 212-4 can emit audio with a lower sampling frequency and reduced bit resolution Rb, and it will use a lower filter setting for that object, to eliminate folding distortion and at the same time simulate the larger distance.

Avatar_2 210-2 is close to object_4 212-4, but since object_4 212-4 contains only mid and low frequency noise, object_4 212-4 will transmit with a low sampling frequency, low bit resolution Rb and a low filter setting. Both Avatar_1 210-1 and Avatar_2 210-2 users are far away from object_3 212-3, so although the source contains the full frequency range, it doesn't need to reproduce the full frequency range with full bit resolution Rb and the sampling frequency can be reduced and the filter setting Fs reduced.

Object_4 212-4 is a moving object and if Object_4 212-4 moves away from avatar_2 210-2, then resolution specification 702 of Object_4 212-4 will be updated accordingly as it transmits its stream to avatar_2 210-2. Object_4 212-4 is a moving object 212 so the audio data stream 706-1 needs to be updated at a reasonable rate. As object_4 212-4 has a low and mid frequency range, if it moves far away, only the low frequency range needs to be captured.

Objects 212 in the VR environment need not be visible to a user 202.

In parallel with the steps 404-422 a server method manages the VR environment. At step 424, the server agent 216 identifies LFOs 618 in the VR environment, and identifies, amongst other factors, the closest LFO 618 to each object 212. The server agent 216 will use the frequency rate from the closest LFO 618 to determine at which rate the server agent 216 will be insert the embedded data structure (that is the resolution specification) into the audio data 710. At step 426, a client tracker component 712 associated with each object 212 tracks each avatar 210-1, 210-2 in the VR environment. The client tracker component 712 calculates the distance to each avatar 210 and updates the next available slot in the audio data stream 706-1 with a resolution specification 702 that reflects the current situation. The resolution specification 702-2 is updated by the server agent 216, embedded in the audio data stream 706-1 and sent to the client 204 in the audio data stream 706.

The client tracker component 712 can also identify if the client 204 leaves a local environment, resulting in the client 204 being unsubscribed from the object 212 and halting streaming of audio data streams 706 to the client 204. Conversely, the client tracker component 712 can also identify if an object 212 leaves a local environment, resulting in the client 204 being unsubscribed from the object 212 and halting streaming of audio data streams 706 to the client 204. In both cases, a first location 220 of the client 204 is different from a second location 222 of the object 212. Locations 220, 222 can be measured with a number of granularities. For example, a location 220, 222 can define a point in the VR environment. Alternatively, a location 220, 222 can define a VR environment ‘room’, and so be defined as a two dimensional area, or a three dimensional volume of the VR environment.

One way of tracking in a VR environment is known as “lighthouse” tracking, developed by Valve Corporation for its HTC Vive VR system. In an immersive VR environment, two base stations emit infrared light in a specific sequence and timing, which is detected by sensors on the VR headset and controllers. To determine where the client 204 is located relative to the audio object 212, the audio object 212 can access a central repository or in multiple repositories, for example one repository per local “room”, according to the architecture of the VR environment.

Continuously, the audio object 212 then obtains the ID of the client 204 and gets various properties of the client 204, for example where the client 204 is located. The object 212 streams audio data 710 to the client and the resolution specification 702 is included at an interval which is determined by the nearest LFO.

At step 416, the object 212 streams raw audio data 710 in a stream 706 to each subscribed client 204-1, 210-2, along with the resolution specification 702. The audio emitting object determines the quality at which a specific subscription will be produced.

At step 418 a process component 610 of the client 204 processes the audio data stream 706 to produce a processed audio file. The client 204 interprets what is received from the subscribed object 212. Each client 204 comprises a low pass filter 616. As part of the process step 418, the received audio data 710 is passed through the low pass filter 616 to avoid folding distortion of frequencies above the Nyquist frequency. Folding distortion occurs when frequencies above the Nyquist frequency are incorrectly represented in a digital system.

At step 420, a mix component