Patent Application
20230308822
The present invention generally relates to utilizing positional 3D spatial sound power information for the purpose of deterministic positional based automatic gain control to adjust one or more dynamically configured microphone arrays in at least near real-time for multi-user conference situations for optimum audio signal and ambient sound level performance.
Obtaining high quality audio at both ends of a conference call is difficult to manage due to, but not limited to, variable room dimensions, dynamic seating plans, roaming participants, unknown number of microphones and locations, unknown speaker system locations, known steady state and unknown dynamic noise, variable desired sound source levels, and unknown room characteristics. Because of the complex needs and requirements, solving the problems has proven difficult and insufficient within the current art.
To provide an audio conference system that addresses dynamic room usage scenarios and the audio performance variables discussed above, microphone systems need to be thoughtfully designed, installed, configured, and calibrated to perform satisfactorily in the environment. The process starts by placing an audio conference system in the room utilizing one or more microphones. The placement of microphone(s) is critical for obtaining adequate room coverage which must then be balanced with proximity of the microphone(s) to the participants to maximize desired vocal audio pickup while reducing pickup of speakers and undesired sound sources. In a small space where participants are collocated around a table, simple audio conference systems can be placed on the table to provide adequate performance and participant audio room coverage. Larger spaces require multiple microphones of various form factors which may be mounted in any combination of, but not limited to, the ceiling, tables, walls, etc., making for increasingly complex and difficult installations. To optimize audio performance of the audio conference system, various compromises are typically required based on, but not limited to, limited available microphone mounting locations, inability to run connecting cables, room use changes requiring a different microphone layout, seated vs agile and walking participants, location of undesired noise sources and other equipment in the room, etc. all affecting where and what type of microphones can be placed in the room.
Once mounting locations have been determined and the system has been installed, the audio system will typically require a manual calibration process run by an audio technician to complete setup up. Examples of items checked during the calibration include: the coverage zone for each microphone type, gain structure and levels of the microphone inputs, feedback calibration and adjustment of speaker levels and echo canceler calibration. It should be noted in the current art, the microphone systems do not have knowledge of location information relative to other microphones and speakers in the system, so the setup procedure is managing for basic signal levels and audio parameters to account for the unknown placement of equipment. As a result, if any part of the microphone or speaker system is removed, replaced, or new microphone and speakers are added, the system would need to undergo a new calibration and configuration procedure. Even though the audio conference system has been calibrated to work as a system, the microphone elements operate independently of each other requiring complex switching and management logic to ensure the correct microphone system element is active for the appropriate speaking participant in the room. The impact of this is overlapping microphone coverage zones, coverage zone boundaries that cannot be configured for, or controlled precisely resulting in microphone element conflict with desired sound sources, unwanted undesired sound source pick up, to little coverage zone for the room, coverage zone extension beyond the preferred coverage area and inconsistent gain management as sound sources move across the various microphone coverage regions. This can result in the microphone system having to deal with a wide dynamic range of audio signals and ambient background noise levels based on the sound source position in the room relative to any one of the physical microphones installed. This issue is further complicated and magnified when two or more microphone array systems are utilized to provide sufficient room coverage.
In the currently known art, there have been various approaches to solving the complex issue of managing wide dynamic range audio signals with acceptable ambient sound level performance from multi-location based sound and signal sources. Typically, this is accomplished using heuristic-based automatic gain control techniques to enhance audio conferencing system performance in a multi-user room. Automatic gain control is used to bring the desired signal, which in this case may be but is not limited to a speaking participant in the room, to within an acceptable dynamic range to be transmitted to remote participants through third party telephone, network and/or teleconference software such as Microsoft Skype, for example. If automatic gain control was not implemented the conversations would be hard to hear with the sound volume levels swinging from very low level to very loud levels. The communication system may not be able to manage the signal properly, with too little signal strength to be heard clearly or too much signal strength, which would overdrive the system resulting in clipping of the signal and adding significant distortion. Either scenario would not be acceptable in an audio conference situation. If the signal is within a sufficient range to propagate through the system, the resulting dynamic range swings would require the remote participants to continually adjust their volume control to compensate for the widely variable level differences that would be present for each individual speaking participant. An unwanted byproduct of typical automatic gain control circuits is the ambient sound levels also tracking in proportion to volume changes by the remote participant.
Automatic gain control is typically applied as a post-processing function within a variable gain amplifier or after the analog digital converter in a digital signal processor isolated from the microphone processing logic. The automatic gain control does not know a key parameter such as the position of the sound source, or the configuration of the specific microphone system at that location in the room that was used to pick up the sound source audio, which means the automatic gain control will need to operate on heuristic principals, assumptions, and configuration limits. This is problematic because the automatic gain control solutions have to work on heuristic principals because the actual location of the sound and ambient sound sources are not known in relation to the microphone system's various microphone elements, which means the performance of the automatic gain control is not deterministic. This results in serious shortcomings by not being able to adapt to and provide consistent performance and acceptable end user experiences.
Automatic gain control systems which need to deal with large dynamic range signals end up having to adjust the gain of the system, which can show up as sharp unexpected changes in background ambient sound levels. The automatic gain control will appear to hunt for the right gain setting so there can be a warbling and inconsistent sound levels making it difficult to understand the person speaking. The automatic gain control is trying to normalize to preset parameters that may or may not be suitable to the actual situation, as designers cannot anticipate all scenarios and contingencies that an automatic gain control function must handle. Third party conference and phone software such as but not limited to Microsoft Skype, for example, have specifications that need to be met to guarantee compatibility, certifications, and consistent performance. Automatic gain controls in the current art do not know the distance and the actual sound levels of the sound source they are trying to manage, resulting in inconsistent sound volume when switching sources and fluctuating ambient sound level performance. This makes for solutions that are not deterministic and do not provide a high level of audio performance and user experience.
Thus, the current art is not able to provide consistent performance in regard to a natural user experience regarding desired source signal level control and consistent ambient sound level performance.
An approach in the prior art is to utilize various methods to determine source location targeting parameters to determine Automatic Gain Control (AGC) settings. However, the systems in the prior art address a gain adjustment method that does not adequately manage the ambient noise levels to a consistent level, regardless of targeted AGC parameters, which is problematic for maintaining a natural audio listening experience with consistent ambient noise levels for conference participants.
The optimum solution would be a conference system that is able to automatically determine and adapt an optimized combined coverage zone for shape, size, position, and boundary dimensions in real-time utilizing all available microphone elements in shared space as a single physical array and create a gain map based on the position of each virtual microphone in the coverage zone in relation to each microphone array element. However, fully automating the dynamic gain structure coverage zone process and creating a single dimensioned, positioned, and shaped coverage zone grid from multiple individual microphones that is able to fully encompass a 3D space including limiting the coverage area to derived boundaries and solving such problems has proven difficult and insufficient within the current art.
An automatic calibration process is preferably required which will detect microphones attached or removed from the system, locate the microphones in 3D space to sufficient position and orientation accuracy to form a single cohesive microphone array element out of all the in-room microphones. With all microphones operating as a single physical microphone element, effectively a microphone array, the system will be able to derive a single cohesive position based, dimensioned and shaped coverage map that contains gain specific parameters for each virtual microphone position relative to the individual microphone elements that is specific and adapts to the room in which the microphone system is installed. This approach improves, for example, the management of audio signal gain, tracking participants, maintaining ambient noise levels, minimizing unwanted sound sources, reduction of ingress from other spaces and sounds source bleed through from coverage grids that extend beyond wall boundaries and wide-open spaces while accommodating a wide range of microphone placement options; one of which is being able to add or remove microphone elements in the system and have the audio conference system integrate the changed microphone element structure into the microphone array in real-time and preferably adapting the coverage pattern accordingly.
Systems in the current art do not automatically derive, establish and adjust their specific coverage zone parameters and virtual microphone gain parameter details based on specific microphone element positions and orientations, and instead rely on a manual calibration and setup process to setup the audio conference system requiring complex DSP switching and management processors to integrate independent microphones into a coordinated microphone room coverage selection process based on the position and sound levels of the participants in the room. Adapting to the addition or removal of a microphone element is a complex process. The audio conference system will typically need to be taken offline, recalibrated, and configured to account for coverage patterns as microphones are added or removed from the audio conference system. Adapting and optimizing the coverage area to a specific size, shape and bounded dimensions is not easily accomplished with microphone devices used in the current art which results in a scenario where either not enough of the desired space is covered or too much of the desired space is covered extending into an undesired space and undesired sound source pickup.
Therefore, the current art is not able to provide a dynamically formed virtual microphone coverage grid with individual virtual microphone gain parameters in real-time accounting for individual microphone position placement in the space during audio conference system setup that takes into account multiple microphone-to-speaker combinations, multiple microphone and microphone array formats, microphone room position, addition and removal of microphones, in-room reverberation, and return echo signals.
An object of the present embodiments is, in real-time, upon auto-calibration of the combined microphone array system after automatically determining and positioning the microphone coverage grid for the optimal dispersion of virtual microphones for grid placement, size and geometric shape relative to a reference point in the combined microphone array and to the position of the other microphone elements in the combined microphone array to create a virtual microphone position based gain map. More specifically, it is an object of the invention to preferably derive virtual microphone specific gain parameters for one or more physical microphone elements that are specific to each microphone element relative to each virtual microphone position in the coverage map in the shared 3D space.
The present invention provides a real-time adaptable solution to undertake creation of a dynamically determined coverage zone grid containing positional based gain parameters for each virtual microphone based on the installed microphone's position, orientation, and configuration settings in the 3D space.
These advantages and others are achieved, for example, by a system for positional based automatic gain control to adjust a dynamically configured combined microphone array in a shared 3D space for optimum audio signal and ambient sound level performance. The system includes a combined microphone array and a system processor communicating with the combined microphone array. The combined microphone array includes one or more of individual microphones and/or microphone arrays each including a plurality of microphones. The microphones in each microphone array are arranged along a microphone axis. The system processor is configured to perform operations including: obtaining predetermined locations of the microphones of the combined microphone array throughout the shared 3D space, obtaining predetermined coverage zone dimensions based on the locations of the microphones of the combined microphone array, populating the coverage zone dimensions with virtual microphones, identifying locations of sound sources in the shared 3D space based on the virtual microphones, computing one or more positional based gain control (PBGC) parameter values for one or more virtual microphones based on the locations of the virtual microphones, and combining microphone signals into desired channel audio signals by applying the PBGC parameters to adjust microphones to control positional based microphone gains based on the location information of the sound source.
The PBGC parameter values may be stored in one or more computer-readable media. The PBGC parameter values may include gains for the microphones and the virtual microphones. The adjusting microphones may include adjusting a gain value for each microphone. The PBGC parameters may be pre-computed based on locations of the virtual microphones. The PBGC parameters may be computed in real-time when a new sound source location is determined and corresponding virtual microphone receives focus. The operations may further include creating processed audio signals from raw microphone signals, and applying gain values to processed audio signals by using the PBGC parameters. the positional based microphone gains may be obtained on a per microphone basis. The microphones in the combined microphone array may be configured to form a 2D plane in the shared 3D space. The microphones in the combined microphone array may be configured to form a hyperplane in the shared 3D space.
The preferred embodiments comprise both algorithms and hardware accelerators to implement the structures and functions described herein.
The present invention is directed to apparatus and methods that enable groups of people (and other sound sources, for example, recordings, broadcast music, Internet sound, etc.), known as “participants”, to join together over a network, such as the Internet, in a remotely-distributed real-time fashion employing personal computers, network workstations, and/or other similarly connected appliances, often without face-to-face contact, to engage in effective audio conference meetings that utilize large multi-user rooms (spaces) with distributed participants.
Advantageously, embodiments of the present apparatus and methods provide an ability for remote participants to experience in room sound sources (participants) of a conference call at a consistent volume level, regardless of their location with respect to the microphone array, while always maintaining consistent ambient sound levels, regardless of the number of microphone elements distributed throughout the space.
A notable challenge to picking up sound clearly in a room, cabin, or confined space is the dynamic nature of the sound sources, resulting in a wide range of sound pressure levels, while maintaining realistic and consistent ambient sound levels for the remote participant(s). Creating a dynamically shaped and positioned virtual microphone bubble map that contains gain parameters for each virtual microphone location from ad-hoc located microphones in a 3D space requires reliably placing and sizing the 3D virtual microphone bubble map with sufficient accuracy to position the virtual microphone bubble map in proper context to the room boundaries, physical microphones installed locations and the participants usage requirements all without requiring a complex manual setup procedure, the merging of individual microphone coverage zones, directional microphone systems or complex digital signal processing (DSP) logic. Instead, preferably using a microphone array system that is aware of its constituent microphones locations relative to each other in the 3D space as well as each microphone device having configuration parameters that facilitate coverage zone boundary determinations on a per microphone basis allowing for a microphone array system that is able to automatically and dynamically derive and establish room specific installed coverage zone areas and constraints to optimize the coverage zone area and gain structure parameters for each virtual microphone in each individual room automatically without the need to manually calibrate and configure the microphone system.
A “microphone” in this specification may include, but is not limited to, one or more of, any combination of transducer device(s) such as, microphone element, condenser mics, dynamic mics, ribbon mics, USB mics, stereo mics, mono mics, shotgun mics, boundary mic, small diaphragm mics, large diaphragm mics, multi-pattern mics, strip microphones, digital microphones, fixed microphone arrays, dynamic microphone arrays, beam forming microphone arrays, and/or any transducer device capable of receiving acoustic signals and converting to electrical signals, and or digital signals.
A “microphone point source” is defined for the purpose of this specification as the center of the aperture of each physical microphone. The microphones are considered to be omni-directional as defined by their polar plot and can essentially be considered an isotropic point source. This is required for determining the geometric arrangement of the physical microphones relative to each other. The microphones will be considered to be a microphone point source in 3D space.
A “Boundary Device” in this specification may be defined as any microphone and/or microphone arrangement that has been defined as a boundary device. A microphone can be configured and thus defined as a boundary device through automatic queries to the microphone and/or through a manual configuration process. A boundary device may be mounted on a room boundary such as a wall or ceiling, a tabletop, and/or a free-standing microphone offset from or suspended from a mounting location that will be used to define the outer coverage area limit of the installed microphone system in its environment. The microphone system will use microphones configured as boundary devices to derive coverage zone dimensions in the 3D space. By default, if a boundary device is mounted to a wall or ceiling it will define the coverage area to be constrained to that mounting surface which can then be used to derive room dimensions. As more boundary devices are installed on each room boundary in a space the accuracy of determining the room dimensions increases with each device and can be determined to a high degree of accuracy if all room boundaries are used for mounting. By the same token a boundary device can be free standing in a space such as a microphone on a stand or suspended from a ceiling or offset from a wall or other structure. The coverage zone dimension will be constrained to that boundary device which is not defining a specific room dimension but is a free air dimension that is movable based on the boundary devices' current placement in the space. These can be used to define a boundary constraint of 1, 2 or 3 planes based on the location of the boundary device. Boundary constraints are defined as part of the boundary device configuration parameters to be defined in detail within the specification. Note that a boundary device is not restricted to create a boundary at its microphone location. For example, a boundary device that consists of a single microphone hanging from a ceiling mount at a known distance could create a boundary at the ceiling by off-setting the boundary from the microphone by that known distance.
A “microphone arrangement” may be defined in this specification as a geometric arrangement of all the microphones contained in the microphone system. Microphone arrangements are required to determine the virtual microphone distribution pattern. The microphones can be mounted at any point in the 3D space, which may be a room boundary, such as a wall, ceiling or floor. Alternatively, the microphones may be offset from the room boundaries by mounting on stands, tables or structures that provide offset from the room boundaries. The microphone arrangements are used to describe all the possible geometric layouts of the physical microphones to either form a microphone axis (m-axis), microphone plane (m-plane) or microphone hyperplane (m-hyperplane) geometric arrangement in the 3D space.
A “microphone axis” (m-axis) may be defined in this specification as an arrangement of microphones that forms and is constrained to a single 1D line.
A “microphone plane” (m-plane) may be defined in this specification as an arrangement containing all the physical microphones that forms and is constrained to a 2D geometric plane. A microphone plane cannot be formed from a single microphone axis.
A “microphone hyperplane” (m-hyperplane) may be defined in this specification as an arrangement containing all the physical microphones that forms a 3-dimensional hyperplane structure between the microphones. A microphone hyperplane cannot be formed from a single microphone axis or microphone plane.
Two or more microphone aperture arrangements can be combined to form an overall microphone aperture arrangement. For example, two microphone axes arranged perpendicular to each other will form a microphone plane and two microphone planes arranged perpendicular to each other with form a microphone hyperplane.
A “virtual microphone” in this specification represents a point in space that has been focused on by the combined microphone array by time-aligning and combining a set of physical microphone signals according to the time delays based on the speed of sound and the time to propagate from the sound source each to physical microphone.
A “Coverage Zone Dimension” in the specification may include physical boundaries such as wall, ceiling and floors that contain a space with regards to the establishment of installing and configuring a microphone system coverage patterns and dimensions. The coverage zone dimension can be known ahead of time or derived with a number of sufficiently placed microphone arrays also known as boundary devices placed on or offset from physical room boundaries.
A “combined array” in this specification can be defined as the combining of two more individual microphone elements, groups of microphone elements and other combined microphone elements into a single combined microphone array system that is aware of the relative distance between each microphone element to a reference microphone element, determined in configuration, and is aware of the relative orientation of the microphone elements such as a m-axis, m-plane and m-hyperplane sub arrangements of the combined array. A combined array will integrate all microphone elements into a single array and will be able to form coverage pattern configurations as a combined array.
A “conference enabled system” in this specification may include, but is not limited to, one or more of, any combination of device(s) such as, UC (unified communications) compliant devices and software, computers, dedicated software, audio devices, cell phones, a laptop, tablets, smart watches, a cloud-access device, and/or any device capable of sending and receiving audio signals to/from a local area network or a wide area network (e.g. the Internet), containing integrated or attached microphones, amplifiers, speakers and network adapters. PSTN, Phone networks etc.
A “communication connection” in this specification may include, but is not limited to, one or more of or any combination of network interface(s) and devices(s) such as, Wi-Fi modems and cards, internet routers, internet switches, LAN cards, local area network devices, wide area network devices, PSTN, Phone networks, etc.
A “device” in this specification may include, but is not limited to, one or more of, or any combination of processing device(s) such as, a cell phone, a Personal Digital Assistant, a smart watch or other body-borne device (e.g., glasses, pendants, rings, etc.), a personal computer, a laptop, a pad, a cloud-access device, a white board, and/or any device capable of sending/receiving messages to/from a local area network or a wide area network (e.g., the Internet), such as devices embedded in cars, trucks, aircraft, household appliances (refrigerators, stoves, thermostats, lights, electrical control circuits, the Internet of Things, etc.).
A “participant” in this specification may include, but is not limited to, one or more of, any combination of persons such as students, employees, users, attendees, or any other general groups of people that can be interchanged throughout the specification and construed to mean the same thing. Who gathering into a room or space for the purpose of listening to and or being a part of a classroom, conference, presentation, panel discussion or any event that requires a public address system and a UCC connection for remote participants to join and be a part of the session taking place. Throughout this specification a participant is a desired sound source, and the two words can be construed to mean the same thing.
A “desired sound source” in this specification may include, but is not limited to, one or more of a combination of audio source signals of interest such as: sound sources that have frequency and time domain attributes, specific spectral signatures, and/or any audio sounds that have amplitude, power, phase, frequency and time, and/or voice characteristics that can be measured and/or identified such that a microphone can be focused on the desired sound source and said signals processed to optimize audio quality before delivery to an audio conferencing system. Examples include one or more speaking persons, one or more audio speakers providing input from a remote location, combined video/audio sources, multiple persons, or a combination of these. A desired sound source can radiate sound in an omni-polar pattern and/or in any one or combination of directions from the center of origin of the sound source.
An “undesired sound source” in this specification may include, but is not limited to, one or more of a combination of persistent or semi-persistent audio sources such as: sound sources that may be measured to be constant over a configurable specified period of time, have a predetermined amplitude response, have configurable frequency and time domain attributes, specific spectral signatures, and/or any audio sounds that have amplitude, power, phase, frequency and time characteristics that can be measured and/or identified such that a microphone might be erroneously focused on the undesired sound source. These undesired sources encompass, but are not limited to, Heating, Ventilation, Air Conditioning (HVAC) fans and vents; projector and display fans and electronic components; white noise generators; any other types of persistent or semi-persistent electronic or mechanical sound sources; external sound source such as traffic, trains, trucks, etc.; and any combination of these. An undesired sound source can radiate sound in an omni-polar pattern and/or in any one or combination of directions from the center of origin of the sound source.
A “system processor” is preferably a computing platform composed of standard or proprietary hardware and associated software or firmware processing audio and control signals. An example of a standard hardware/software system processor would be a Windows-based computer. An example of a proprietary hardware/software/firmware system processor would be a Digital Signal Processor (DSP).
A “communication connection interface” is preferably a standard networking hardware and software processing stack for providing connectivity between physically separated audio-conferencing systems. A primary example would be a physical Ethernet connection providing TCPIP network protocol connections.
A “Unified Communication Client” (UCC) is preferably a program that performs the functions of but not limited to messaging, voice and video calling, team collaboration, video conferencing and file sharing between teams and or individuals using devices deployed at each remote end to support the session. Sessions can be in the same building and/or they can be located anywhere in the world that a connection can be establish through a communications framework such but not limited to Wi-Fi, LAN, Intranet, telephony, wireless or other standard forms of communication protocols. The term “Unified Communications” may refer to systems that allow companies to access the tools they need for communication through a single application or service (e.g., a single user interface). Increasingly, Unified Communications have been offered as a service, which is a category of “as a service” or “cloud” delivery mechanisms for enterprise communications (“UCaaS”). Examples of prominent UCaaS providers include Dialpad, Cisco, Mitel, RingCentral, Twilio, Voxbone, 8×8, and Zoom Video Communications.
An “engine” is preferably a program that performs a core function for other programs. An engine can be a central or focal program in an operating system, subsystem, or application program that coordinates the overall operation of other programs. It is also used to describe a special-purpose program containing an algorithm that can sometimes be changed. The best-known usage is the term search engine which uses an algorithm to search an index of topics given a search argument. An engine is preferably designed so that its approach to searching an index, for example, can be changed to reflect new rules for finding and prioritizing matches in the index. In artificial intelligence, for another example, the program that uses rules of logic to derive output from a knowledge base is called an inference engine.
As used herein, a “server” may comprise one or more processors, one or more Random Access Memories (RAM), one or more Read Only Memories (ROM), one or more user interfaces, such as display(s), keyboard(s), mouse/mice, etc. A server is preferably apparatus that provides functionality for other computer programs or devices, called “clients.” This architecture is called the client-server model, and a single overall computation is typically distributed across multiple processes or devices. Servers can provide various functionalities, often called “services”, such as sharing data or resources among multiple clients, or performing computation for a client. A single server can serve multiple clients, and a single client can use multiple servers. A client process may run on the same device or may connect over a network to a server on a different device. Typical servers are database servers, file servers, mail servers, print servers, web servers, game servers, application servers, and chat servers. The servers discussed in this specification may include one or more of the above, sharing functionality as appropriate. Client-server systems are most frequently implemented by (and often identified with) the request-response model: a client sends a request to the server, which performs some action and sends a response back to the client, typically with a result or acknowledgement. Designating a computer as “server-class hardware” implies that it is specialized for running servers on it. This often implies that it is more powerful and reliable than standard personal computers, but alternatively, large computing clusters may be composed of many relatively simple, replaceable server components.
The servers and devices in this specification typically use the one or more processors to run one or more stored “computer programs” and/or non-transitory “computer-readable media” to cause the device and/or server(s) to perform the functions recited herein. The media may include Compact Discs, DVDs, ROM, RAM, solid-state memory, or any other storage device capable of storing any of the one or more computer programs, data and parameters.
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For clarity purposes, a single remote user 101 is illustrated. However, it should be noted that there may be a plurality of remote users 101 connected to the conference system 110 which can be located anywhere a communication connection 123 is available. The number of remote users is not germane to the preferred embodiment of the invention and is included for the purpose of illustrating the context of how the audio conference system 110 is intended to be used once it has been installed and calibrated. The room 112 is configured with examples of, but not limited to, ceiling, wall, and desk mounted microphones 106 and examples of, but not limited to, ceiling and wall mounted speakers 105 which are connected to the audio conference system 110 via audio interface connections 122. In-room participants 107 may be located around a table 108 or moving about the room 112 to interact with various devices such as the touch screen monitor 111. A touch screen/flat screen monitor 111 is located on the long wall. A microphone 106 enabled webcam 109 is located on the wall beside the touch screen 111 aiming towards the in-room participants 107. The microphone 106 enabled web cam 109 is connected to the audio conference system 110 through common industry standard audio/video interfaces 122. The complete audio conference system 110 as shown is sufficiently complex that a manual setup for the microphone system is most likely required for the purpose of establishing coverage zone areas between microphones, gain structure and microphone gating levels of the microphones 106, including feedback and echo calibration of the system 110 before it can be used by the participants 107 in the room 112. As the participants 107 move around the room 112, the audio conference system 110 will need to determine the microphone 106 with the best audio pickup performance in real-time and adjust or switch to that microphone 106. Problems can occur when microphone coverage zones overlap between the physically spaced microphones 106. This can create microphone 106 selection confusion especially in systems relying on gain detection and level gate thresholding to determine the most appropriate microphone 106 to activate for the talking participant at any one time during the conference call. Some systems in the current art will try to blend individual microphones 106 through post processing means, which is also a compromise trying to balance the signal levels appropriately across separate microphone elements 106 and can create a comb filtering effect and reduced signal to noise ratio if the microphones 106 are not properly aligned and summed in the time domain. Since the microphone 106 elements are potentially part of a different array and/or location, the audio signal level in general and the audio signal level with respect to the ambient noise will be affected. This is especially pronounced in situations with high dynamic range, participants that are close to and then move away from or are far from the microphone system and the microphone system switches between the participants 107. The microphone system AGC circuit will attempt to compensate for dynamic changes in the source signal which will have a direct effect on the perceived ambient noise levels 701 as the AGC circuits adjust the gain to compensate. Conference systems 110 that do not compensate the system gain based on the specific speaking participant location 702 can never really be optimized for all dynamic situations in the room 112.
For this type of system, the specific 3D location (x, y, z) of each microphone element in space is not known, nor is it determined through the manual calibration procedure. Signal levels and thresholds are measured and adjusted for based on a manual setup procedure using computer 103 running calibration software by a trained audio technician (not shown). If the microphones 106 or speakers 105 are relocated in the room, removed or more devices are added the audio conference, manual calibration will need to be redone by the audio technician.
The size, shape, construction materials and the usage scenario of the room 112 dictates situations in which equipment can or cannot be installed in the room 112. In many situations the installer is not able to install the microphone system 106 in optimal locations in the room 112 and compromises must be made. To further complicate the system 110 installation as the room 112 increases in size, an increase in the number of speakers 105 and microphones 106 is typically required to ensure adequate audio pickup and sound coverage throughout the room 112 and thus increases the complexity of the installation, setup, and calibration of the audio conference system 110.
The speaker system 105 and the microphone system 106 may be installed in any number of locations and anywhere in the room 112. The number of devices 105, 106 required is typically dictated by the size of the room and the specific layout and intended usages. Trying to optimize all devices 105, 106 and specifically the microphones 106 for all potential room scenarios can be problematic.
It should be noted that microphone 106 and speaker 105 systems can be integrated in the same device such as tabletop devices and/or wall mounted integrated enclosures or any combination thereof and is within the scope of this disclosure as illustrated in
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It is important for the combined microphone system to be able to determine its microphone arrangement during the building of the combined microphone array. The microphone arrangement determines how the virtual microphones 301 can be arranged, placed, and dimensioned in the 3D space 112. Once the virtual microphones 301 have been placed in a position 702 relative to the combined array, the gain structure parameters for each virtual microphone 301 can be determined. Since each virtual microphone 301 has a known position 702 to the combined array a positional based gain control strategy can be implemented. With that it is important to understand the various microphone arrangements and how virtual microphones 301 are distributed in each scenario. The preferred embodiment of the invention will be able to utilize the automatically determined microphone arrangement for each unique combined microphone array 124 to dynamically optimize the virtual microphone 301 coverage pattern for the particular microphone 106 arrangement of the combined microphone array 124 installation. As more microphone elements 106 and/or arrays 124 also known as boundary devices are incrementally added to the system the combined microphone system can further optimize the coverage dimensions of the virtual microphone 301 bubble map to the specific room 112 dimensions and/or boundary device locations 702 relative to each other thus creating an extremely flexible and scalable array architecture that can automatically determine and adjust its coverage area, and gain structure eliminating the need for manual configuration and the dependence on heuristic AGC algorithms, the usage of independent microphone arrays with overlapping coverage areas and complex handoff and cover zone mappings. The microphone arrangement of the combined array allows for a contiguous virtual microphone 301 map with known virtual microphone 301 locations 702 across all the installed devices 106, 124. It is important to understand the various microphone arrangements and the coverage zone specifics that the preferred embodiment of the invention uses.
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The geometric layout of the virtual microphones 301 will be equally represented in the reflected virtual microphone plane behind the wall. The virtual microphone 301 distribution geometries are symmetrical as represented by front of wall 307a and behind the wall 307b. The number of virtual microphones 301 can be configured to the y-axis dimensions, front of wall depth 307a and the horizontal-axis, width across the front of wall 307a. As stated previously, the same dimensions will be reflected behind the wall. For example, the y-axis coverage pattern configuration limit 308a will be equally reflected behind the wall in the y-axis in the opposite direction 308b. The z-axis cannot be configured due to the toroid 308 shape of the virtual microphone geometry. Put another way the number of virtual microphones 301 can be configured in the y-axis and x-axis but not in the z-axis for the m-axis 201 arrangement. As mentioned previously the m-axis 201 arrangement is well suited to a boundary mounting scenario where the reflected virtual microphones 302 can be ignored and the z-axis is not critical for the function of the array 124 in the room 112. The preferred embodiment of the invention can position the virtual microphone 301 map in relative position to the m-axis 201 orientation and can be configured to constrain the width (x-axis) and depth (y-axis) of the virtual microphone 301 map if the room boundary dimensions are known relative to the m-axis 201 position in the room 112.
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For simplicity the illustration of the m-hyperplane 203 is shown as cubic however it is not constrained to a cubic geometry for virtual microphone 301 coverage map form factor and instead is meant to represent that the virtual microphones 301 are not distributed on an axis or a plane and thus incurring the limitations of those geometries. The virtual microphones 301 can be distributed in any geometry and pattern supported by the hardware and mounting locations of the individual arrays 124 within the combined array and be considered within the scope of the invention.
With reference to
With reference to
Once the virtual microphone 301 map has been determined, the positional based gain control (PBGC) 920
With reference to
With reference to
For the purpose of this embodiment, the microphone array 124 is positioned against a wall; however, the position of the microphone array 124 can be against any wall, ceiling mounted, suspended, tabletop mounted and/or placed on a mounting stand such as tripod in the room 112. There are notionally three participants illustrated in the room, Participant 1 107, Participant 2 107 and Participant 3 107. Participant(s) and sound source(s) can and will be used interchangeably and in this context mean substantially the same thing. Each participant 107 illustrates, but is not limited to, an example of the variability of position within a room 112. The embodiments are designed to adjust for and accommodate such positions (stationary and/or moving). For example, each participant 107 may be moving, and thus have varying location coordinates in the X, Y, and Z directions. Also illustrated is an ambient sound 701, which may be present and propagated throughout the room 112, such that it is relatively constant for each participant 107, locations. For example, the room ambient noise 701 may be one or more of HVAC noise, TV noise, outside noise, etc.
Also illustrated is a Minimum Threshold Distance (MTD) 703 and a Configurable Threshold Distance (CTD) 704. The area inside the CTD 704 is the microphone array 124 configuration zone. In that zone, utilizing the specific distance P2 d(m) 705 (e.g., distance in metric) of participant 2 107, the array 124 will be configured for individual gain and microphone 106 selection to stabilize the array 124 volume output and ambient sound level 701 relative to the Participant 2 location 107. Within the CTD 704 there is preferably enough positional 702 resolution of the system to utilize distance path loss 705 to tune the array 124 for individual microphone 106 gain-weighted measurements. Within the zone of the CTD 704 and the MTD 703, the microphone array 124 is dynamically configured to utilize between one and twelve of the microphones 106, based on the position 702 of the sound source 107.
For participants 107 outside the CTD 704, preferably all microphones 106 are used. As the sound source 107 gets further from the CTD 704, its perceived volume will drop off. This is the preferred behavior as it may be undesirable to pick up people far away and have them sound as if they are in the room.
For participants 107 in the zone between the MTD 703 and the CTD 704, the system will preferably pick the n+1 microphones 124b which are closest to the location 702 of the sound source 107 to act as the microphone array (e.g., one of them will only be fractionally on) and the remainder are preferably turned off.
When a participant 1 107 is within the MTD 703 distance P1 d(m) 706, the system will preferably select a pair of microphones 106 in the array 124, so that the ambient sound level 701 can be maintained with one microphone 124b fully on and one fractionally on, e.g., 5%, 10%, 15%, 20%, 25%, 30%, 35%, 40%, 45%, 50%, or any value between 0% and 99%. When the participant 107 gets within the MTD 703 of the closest microphone 106, the array 124 will preferably no longer use that microphone 106. Instead, the system preferably uses one or more other microphones 106 further away, that are outside the closest-microphone MTD 703 to control the gain of the sound source 107. If the microphones 106 are spaced close enough, there will usually exist a microphone in the range where n=1. The maximum microphone 106 spacing allowed is preferably (sqrt(2)−1)*MTD 703.
Beyond the CTD 704 such as Participant 3 107 at distance P3 d(m) 707, all 12 microphones (or however many microphones are in the array, e.g., any number between 2 and 100; and the “array” may be a one-dimensional array, a two-dimensional matrix array, or a three-dimensional linear or matrix array having certain microphones 106 at different distances from a Z-axis baseline) of the microphone array 124 are preferably sequentially enabled as the positional information 702 (obtained from the system) becomes too granular and the best performance is realized with all 12 microphones 106 in operation. Both the MTD 703 and the CTD 704 are preferably system-configurable parameters that are set based on the microphone array 124 parameters and the room 112 parameters.
The prior art invention described in U.S. Pat. No. 10,387,108 considers a single (minimum) distance from the sound source 107 to a single (closest) microphone within the microphone array. This single distance is a reasonably good approximation for the remaining microphones contained in the same array 124. However, when two or more arrays 124 are used the single distance used in prior art equations is largely inadequate for modeling the contribution of microphones 106 in separate arrays 124 positioned at significant distances from each other. In contrast, current invention overcomes this limitation by accurately accounting for the plurality of distances from the sound source 107 to the plurality of microphone elements 106 by defining a generalized mathematical treatment.
With reference to
For the purpose of this embodiment, the microphone arrays 124a, 124b and 124c are positioned against walls, ceiling mounted, suspended, tabletop mounted and/or placed on a mounting stand such as tripod in the room 112 and the microphone arrays can be placed on any wall in the room 112, there can be plurality of microphone arrays on the same wall surface and ceiling of the room 112. There are notionally three participants 107 illustrated in the room 112, Participant 1 107, Participant 2 107 and Participant 3 107. Participant(s) and sound source(s) can and will be used interchangeably and in this context mean substantially the same thing. Each participant 107 illustrates, but is not limited to, an example of the variability of position 702 within a room 112. The embodiments are designed to adjust for and accommodate such positions 702 (stationary and/or moving). For example, each Participant 107 may be moving, and thus have varying location 702 coordinates in the X, Y, and Z directions. Also illustrated is an ambient sound 701, which may be present and propagated throughout the room, such that it is relatively constant for each participant 107 locations. For example, the room 112 ambient noise 701 may be one or more of HVAC noise, TV noise, outside noise, etc.
Also illustrated in
For participants 107 outside the CTD 704 additional microphone arrays 124b, 124c and 106 are activated and a sufficient number of microphones 106 within each of the arrays are utilized to stabilize the volume output and ambient sound at location 702. As the sound source 107 moves further away from arrays 124b, 124c and 106 and all available microphones 106 are already utilized, the audio level will start to drop off. This is the preferred behavior as it may be undesirable to pick up people far away such as outside of the bubble map coverage zone and have them sound as if they are in the room 112. At certain locations of the source 107 the invention may preferably determine that one or more of the arrays 124a, 124b, 124c and 106 based on specific distances to the source P21, P24, P23 and P22 respectively are contributing adversely to stabilizing the volume output and ambient sound 701 and consequently direct that array to turn all microphones 106 off. This aspect of the invention is referred to as the microphone engagement criteria. For the example of participant 2 source at location 702 to the array 124b may be the first choice of the array to be turned off due to the largest relative distance P24.
When only a subset of microphones 106 within each array 124a, 124b, 124, and 106 is utilized the system may pick the microphones 106 that are closest to the source location 702 or microphones 106 starting from the center of the array 124 or other similar methods of microphone allocation (
When two or more array CTD 704 zones overlap due to proximity of the arrays 124, the microphone 106 allocation proceeds as outlined above but resulting signal weights allows for the microphones 106 from both overlapping arrays to be active simultaneously. Such case is illustrated with the arrays 124b and 124c and the sound source 107 participant 3 in
When the sound source inside the CTD zone 704 moves closer to the microphone array 124 and the number of microphones 106 utilized according to the invention decreases to 1 the distance to the array is called the Minimum Threshold Distance (MTD 703) and denoted with symbol Dm in the equations. It should be noted that the MTD 703 in the invention is a significant improvement over prior art and results in improved microphone array 124 performance in this region as will be further described in the specification. This boundary is illustrated in
Each of the arrays 124a, 124b, 124c, and 106 may contain generally a different number of microphone elements 106; and the “array” may be a one-dimensional array (m-axis 201), a two-dimensional matrix array (m-plane 202), or a three-dimensional linear or matrix array (m-hyperplane 203) having certain microphones 106 at different distances from a Z-axis baseline. CTD 704 is preferably a system-configurable parameter denoted by distance Dc while MTD 703 zone is derived from the selected CTD 704 as outlined by S1023 in
Similarly, a closest array 124 with more than N1 microphone elements will maintain the desired output level beyond the CTD 704 range. The nominal microphone array size N1 is preferably a system-configurable parameter. For the example system in
Note that the microphone allocation scheme such as those presented in
With reference to
With reference to
With reference to
One embodiment may comprise the processor described and depicted in U.S. Pat. No. 10,063,987, the entire contents of which are incorporated herein by reference.
The sound pressure level (SPL) of the sound waves follows a very predictable loss pattern where the SPL is inversely proportional to the distances P21 d(m) 802, P22, P23 and P24 from the source Participant 2 107 to each of the microphone arrays 124a, 124b, 124c, and 106. Since the positional information 908 derived from the Target Processor 902 is known, the distance P21, P22, P23 and P24 can be calculated, and the PBGC within 920 calculates the gain required, on a per microphone 106 basis, based on the distances 802 to each microphone 106 of the microphone arrays 124a, 124b, 124c, and 106. In the preferred implementation the gain table covering every possible virtual microphone 301 location 702 is pre-calculated in PBGC within Array Configuration and Calibration process 901 step 920 and sent via 940 to the Gain Weight Processor 926 where the gain values are loaded and applied to the microphone signals. Alternatively, the PBGC invention can operate inside the Gain Weight Processor 926 while the Array Configuration and Calibration 901 provides only the physical array 124 and microphone 106 locations directly to 926 via the connection 916.
The Target Processor 902 utilizing the Microphone Array signals 918 preferably determines the substantially exact positional location 702 (X,Y,Z) coordinates of the sound source 107 with the highest processing gain. This is the sound source 107 that the microphone array 124 will focus on. The Target Processor 902 preferably runs independent of the Audio Processor 903. The Target Processor 902 preferably communicates the positional information 908 to the Audio Processor 903, which comprises the Delay Processor 932 and the Gain Weight Processor 926 which loads the PBGC gains from the gain table 940 for the virtual microphone 301 location 702 selected by 908. The Audio Processor 903 preferably runs at the required sample rates (e.g., 24 kHz) to support the desired frequency response specifications, meaning the sample rates are not limited by the invention implementation in the embodiments.
Once the Gain Weight parameters 928 Alpha (α=the multiplication factor to be applied to each of the fully-on microphone signals. f*α=the multiplication factor to be applied to the fractionally-on microphone signal (f is preferably a value between 0 and 1)); and the Pa parameters have been calculated, they are multiplied 929 with the individual Microphone 106 signals 936, resulting in weighted output parameters 931 that have been gain-compensated based on the actual distances 802 (P21, P22, P23 and P24) to each microphone 106 of the microphone arrays 124a, 124b, 124c, and 106. This process accomplishes the specific automatic gain control function, which adjusts the microphone 106 levels 931 that are preferably sent to the delay elements.
The delays in the microphone arrays 124a, 124b, 124c, and 106 are calculated using the positional information 908 from the Target Processor 901 in the Delay Processor 932. The Delay Processor 932 preferably calculates the individual direct path delays d(m) for each microphone 106 relative to the sound source 107 location 702. It then preferably adds the extra DELAY into each microphone path of D-d(m) so that the overall DELAY between the sound source 107 and the summer 933 through all the microphone paths is preferably a constant D. The value constant D would typically be the delay through the longest path between a microphone 106 and a position monitored by the Target Processor 902, measured in milliseconds. For example, if the longest distance between the 17 microphones 106 and the 8192 virtual microphone 301 points monitored by the Target Processor 902 is 10m, then then the value of D would be that distance converted into a delay, about 30 ms. The result is that signals from all microphones 106 are aligned in the time domain, allowing for maximum natural gain of all direct signal path signals to the microphone arrays 124a, 124b, 124c, and 106. All the output signals 934 are preferably summed at the Summer 933 and output for further system processing. The resulting delays are applied to all of the microphones whether they will be used by the Gain Weight Processor 926 or not.
Note that the Target Processor 902 can identify one or multiple sound source locations. The number of locations corresponds to the number of output channels provided by the audio processor. Each channel c would have its own set of weights and delays for its given location.
To provide gain control of the desired signal without affecting the ambient sound level is preferably accomplished through the following methods. This is accomplished by controlling the processing gain of the microphone arrays 124, 124b, 124c and 106. Processing gain is how much the arrays 124a, 124b, 124c, and 106 boosts the desired signal source relative to the undesired signal sources. As illustrated with the microphone arrays 124 in
In this embodiment, the maximum gain that can be achieved with all 17 microphones is 4.12, and the minimum gain (when reduced to a single microphone) is 1. This gives a 12.3 dB gain range. Inside the CTD 704 of individual arrays 124a, 124b, 124c, and 106 typically only the microphones 106 of the closest array are enabled and individually turned off as the sound source gets closer to the array. Depending on the specific locations of arrays 124a, 124b, 124c, and 106 in the room 112 often the desired level is maintained well outside the CTD 704 due to the activation of microphones in plurality of arrays 124a, 124b, 124c, and 106 and the processing gain they provide. When distances P21, P22, P23 and P24 from sound source 107 to microphone arrays 124a, 124b, 124c, and 106 increase further the sound level will drop off with the inverse distance law.
To optimize the implementation embodiments, it is not preferred to just switch microphones 106 in and out, since this may cause undesirable jumps in the sound volume. To make the adjustments continuous, it is preferable to assign some number of microphones 106 to be fully turned on and one microphone 106 to be partially turned on. The partially turned-on microphone 106 allows a smooth transition from one set of microphones 106 to another, and to implement any arbitrary gain within the limits.
For the calculation of microphone gain parameters, it is preferred to determine a specific gain, Gf, for the focused signal while keeping the background gain, Gbg, at unity. To do this, it is preferred to turn n microphones 106 in system microphone array combining 124a, 124b, 124c, and 106 on fully and have one microphone 106, of any one of the available microphones 106, on fractionally with a constant f that is somewhere between 0 and 1. Each microphone 106 signal is preferably weighted by the common constant α. Given the assumptions that the background signals are orthogonal, so they add by power when combined, and that the levels of the signals arriving at each microphone 106 are aligned in phase due to the action of the delay processor 932, the rms gain of n signal with a gain of a and one signal with a gain of f*α is:
G
bg=α√{square root over (n+f2)} (1)
Setting Gbg to unity to keep it constant gives:
α=1/√{square root over (n+f2)} (2)
Logic flow of the positional based gain control (PBGC) algorithm is captured in
d(i)=∥M(i)−Pt∥ (3)
Where operator ∥ . . . ∥ represents the Euclidean distance calculation on position vector 702.
The desired effective gain Geff is a system configurable setting based on the CTD distance 704 denoted here as Dc (see
The effective gain combines the effects of sound propagation over distances d(i) and the processing gain delivered by the Gain Weight Processor 926, and multipliers 929. The system stabilizes the output sound by maintaining the effective gain Geff at all locations close enough to microphone arrays 124a, 124b, 124c, and 106 where this is possible. When distances increase further and no additional microphones 106 are available to reinforce the sound, Geff cannot be maintained, and the output sound level will drop off. The value Gm is system configurable and preferably set to Gm=√{square root over (N1)} where N1 is the number of microphones within a single microphone array device, for example 6 in array 124. This definition for Gm provides an intuitive property that Geff can be maintained in the vicinity of an array with N1 microphones to at least the extent of the CTD 704 and possibly farther depending on the proximity of other microphone arrays 124a, 124b, 124c, and 106. Note that for arrays with fewer microphones 106 than N1 the range where Geff can be maintained is reduced and vice versa.
The position based gain processor 920 will meet the desired effective gain when sufficient number of microphones n and a fractional microphone f are enabled so that the following equation is met,
where distances d(j(1 . . . n)) are distances to n microphones and d(j(n+1)) is the distance to the fractional microphone f where all the microphones are allocated according to the active microphone allocation scheme described in
With reference to
Initially the number of microphones is n=1. After the first microphone j(1) is allocated in step S1003 according to the current allocation scheme (
Depending on system configurable parameter “MTD method” S1024 either a single fractional microphone j(1) is used (fractional method S1025) or the allocation j(1) is changed so that the distance d(j(1)) exceeds the threshold Dm (push-away method S1026). In the case of the fractional MTD method S1024 with processing step S1025 the output sound level is stabilized at the desired effective gain by setting the gain weight processor w(j(1)) to the following value,
When using the fractional MTD method S1024, the effective gain Geff is achieved and consequently the level of the source signal remains constant with the decreasing distance. However, the background signal level will decrease because the preferred unity gain according to the equations (1), (2) is not maintained. In contrast the use of the push-away MTD method S1024, processing step S1026 achieves both the desired effective gain and the background signal unity gain property provided that the size of the array, microphone spacing and the selection of parameters CTD/MTD 703, 704 allow for the re-allocation of the microphone 106 outside the MTD 703 zone per S1026. Examples of solutions for gain weight values w(j) for methods S1025, S1026 are illustrated in graphics 1001 and 1002 respectively. After push-away MTD method S1026 reassigns j(1) and the procedure control is returned to the main logic flow in
The core processing steps S1012, S1013, S1014, S1015, S1016 act to evaluate the addition of one microphone at a time to find a sufficient set of microphones 106 that satisfies the equation (5). Step S1010 checks that there are unused microphones 106 still available for the processing loop to continue. When source position Pt is far from all microphone arrays and distances d(i) are large the right side of the equation (5) cannot meet the target focus gain Gf (left side of equation) which results in all microphones being activated (n=N). In such conditions, the final gains w(i) are calculated according to S1011 and the gain calculation procedure ends.
If unused microphones 106 are available in S1010 then next microphone j(n+1) is selected in step S1012 according to the desired allocation scheme (
Where equation (8) shows that the (n+1) value can be more efficiently calculated recursively from the previous Sd(n) value based on all previously allocated microphones 1 . . . n. The values of Sd(n) and Sd(n+1) are used in the next step S1014 to compute the new microphone (n+1) engagement status according to the following equation,
When true, this microphone engagement status indicates that the inclusion of the new microphone (n+1) increases the source signal level relative to the background signal (SNR) and that equation (5) is closer to being satisfied. The engagement status is checked in step S1015 in the logic flow diagram. If the new microphone 106 does not engage, i.e. equation (9) is false, then no additional microphones 106 will be allocated, and the gain processor weights are computed in S1020 from the subset of n from the total of N microphones 106 present in the system.
When n microphones are fully activated in S1020, there is no fractional microphone and the gains w(i) for the remaining microphones are set to 0. Intuitively speaking, the microphone (n+1) fails to engage when the distance d(j(n+1)) is large relative to distances of the currently allocated microphones 1 . . . n thus providing poor quality signal to the PBGC processor 920 (
Conversely, when the engage status S1015 is true we proceed to checking if focus gain Gf(left hand side of equation (5)) has been met or exceeded by the additional gain provided by the microphone (n+1). The sufficient gain status is evaluated in S1016 according to equation,
When the sufficient gain status (11) is false the new microphone (n+1) is appended to the list of fully active microphones in step S1017 and the processing loop continues at step S1010 where preferably the next microphone is evaluated. However, if sufficient gain is reached, i.e., condition in equation (11) is true, then we proceed to S1018 to solve the quadratic equation (14) for the unknown fractional microphone value f. The equation (14) is constructed by first making substitutions defined in equations (12) and (13).
With the known values n (number of fully active microphones) and f (fraction assigned to the last microphone (n+1)) the weights 928 of the gain weight processor 926 are calculated in step S1019 as following,
Then values a and f are used to calculate the fully activated microphone gains as w(j(1 . . . n))=a, the fractional microphone gains w(j(n+1))=α*f while the remaining microphone gains are set to w(i)=0 as described in step S1019 of the logic flow diagram in
With reference to
With reference to
In
Using the same room 112 configuration and positioning of arrays as in
The calculations of microphone gain parameters as described above (
Additional examples of the spatial arrangements of the microphone activation regions are shown in
The example in
Examples of gain weight processor weights calculated according to the PBGC logic flow logic and equations in
In
In
In
In
In
With reference to
The embodiments described in this application have been presented with respect to use in one or more conference rooms preferably with multiple users. However, the present invention may also find applicability in other environments such as: 1. Commercial transit passenger and crew cabins such as, but not limited to, aircraft, busses, trains and boats. All of these commercial applications can be outfitted with microphones and can benefit from consistent desired source volume and control of the ambient sound conditions which can vary from moderate to considerable; 2. Private transportation such as cars, truck, and mini vans, where command and control applications and voice communication applications are becoming more prominent; 3. Industrial applications such as manufacturing floors, warehouses, hospitals, and retail outlets to allow for audio monitoring and to facilitate employee communications without having to use specific portable devices; and 4. Drive through windows and similar applications, where ambient sounds levels can be quite high and variable, can be controlled to consistent levels within the scope of the invention. Also, the processing described above may be carried out in one or more devices, one or more servers, cloud servers, etc.
While the present invention has been described with respect to what is presently considered to be the preferred embodiments, it is to be understood that the invention is not limited to the disclosed embodiments. To the contrary, the invention is intended to cover various modifications and equivalent arrangements included within the spirit and scope of the appended claims. The scope of the following claims is to be accorded the broadest interpretation so as to encompass all such modifications and equivalent structures and functions.
This application claims priority to U.S. Provisional Patent Application No. 63/324,452, filed Mar. 28, 2022, the entire contents of which are incorporated herein by reference.
| Number | Date | Country | |
|---|---|---|---|
| 63324452 | Mar 2022 | US |
