A wireless microphone system may be configured to compensate for a radio frequency (RF) loss incurred in a wired communication path. The system may comprise a receiver configured to receive an downlink RF signal from a wireless microphone, and a transmission line accessory configured to receive an uplink test signal. The RF test signal may be generated from an RF source and may be measured by the transmission line accessory to determine configuration information for an adjustable RF gain circuit.
Systems and methods configured to generate talker coordinates for directing a camera towards an active talker in the presence of acoustic reflections are disclosed. One method comprises receiving sound location information for a detected audio source from a microphone; determining, based on the sound location information, a first set of coordinates representing an estimated talker location; determining, based on the sound location information and a height of the environment, a second set of coordinates representing a corrected talker location; calculating a weighted height coordinate based on a first height coordinate of the first set of coordinates, a second height coordinate of the second set of coordinates, and stored height coordinates from previously detected audio sources; and transmitting, to a camera, a third set of coordinates comprising the weighted height coordinate and representing a final talker location, to cause the camera to point its image capturing component towards the received location.
Acoustic echo cancellation systems and methods are provided that can automatically adjust a threshold of a non-linear processor based on the state of a conferencing session, such as a far end single talk condition or a doubletalk condition. The state of the conferencing session may be detected based on various combinations of metrics that are measured from a microphone signal and a remote audio signal. The systems and methods can improve the removal of residual echo and therefore enhance the overall performance of the acoustic echo cancellation system.
G10K 11/175 - Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound
Systems and methods are disclosed for providing a more consistent background noise for mixed audio transmitted from a local environment to a remote location. The systems and methods may generate and add aggregate comfort noise to a mixed audio signal, and the aggregate comfort noise may be based on noise spectral estimates from each audio device in an environment. Metrics related to speech and noise levels may be shared between the audio devices in an environment to modify the noise reduction processing of audio signals on each audio device.
Acoustic echo cancellation systems and methods are provided that can automatically adjust a threshold of a non-linear processor based on the state of a conferencing session, such as a far end single talk condition or a doubletalk condition. The state of the conferencing session may be detected based on various combinations of metrics that are measured from a microphone signal and a remote audio signal. The systems and methods can improve the removal of residual echo and therefore enhance the overall performance of the acoustic echo cancellation system.
H04M 9/08 - Two-way loud-speaking telephone systems with means for conditioning the signal, e.g. for suppressing echoes for one or both directions of traffic
Systems and methods are provided for receiving, from at least one microphone (102), boundary information defining one or more boundaries (118) for an audio pick-up region (108); receiving, from the at least one microphone, sound location information indicating a detected sound position of an audio source (106) located within the audio pick-up region; identifying, based on the sound location information and the boundary information, a first boundary of the one or more boundaries as being located near the detected sound position; calculating a first distance between the detected sound position and the first boundary; determining, based on the first distance, a depth of field parameter for at least one camera (104); and providing the depth of field parameter and the sound location information to the at least one camera.
Aspects of the disclosure relate to use of aggregated signal transmission. An aggregated signal may comprise multiple packets, reception of which may be acknowledged via a block acknowledgment. A packet may be repeated in successive aggregated signal transmissions, with a maximum quantity of repetitions being based on a packet class associated with a packet. Repetition of a packet may be based on receiving a negative acknowledgment indicating failed packet reception and/or decoding. A frequency hopping protocol may be used for the successive aggregated signal transmissions.
H04L 1/1867 - Arrangements specially adapted for the transmitter end
H04L 1/08 - Arrangements for detecting or preventing errors in the information received by repeating transmission, e.g. Verdan system
H04L 1/1607 - Arrangements for detecting or preventing errors in the information received by using return channel in which the return channel carries supervisory signals, e.g. repetition request signals - Details of the supervisory signal
A virtual USB interface can enable a software module, e.g., conferencing software, to interface with a software audio endpoint as if the software audio endpoint were a physical USB endpoint. Signals conforming to the USB standard can be transceived and adapted to and from non-USB signals using the virtual USB interface. Both the signals conforming to the USB standard and the non-USB signals may include a media channel and/or a control channel.
Techniques are disclosed herein for providing audio enhancement and optimization of an immersive audio experience. Examples may include generating an audio feature set for a transduced audio stream captured in an environment, inputting the audio feature set to a neural network model configured to generate an audio isolation mask associated with the transduced audio stream, and generating isolated audio for the transduced audio stream based at least in part on the audio isolation mask.
H04H 60/04 - Studio equipment; Interconnection of studios
H04R 1/40 - Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
G10L 21/0216 - Noise filtering characterised by the method used for estimating noise
10.
CONFERENCING SYSTEMS AND METHODS FOR ADJUSTING CAMERA TRACKING BASED ON MICROPHONE COVERAGE
Conferencing systems and methods configured to more accurately and optimally position cameras towards a talker are disclosed. The positioning of cameras may be based on the coverage area of microphone arrays in an environment and through the use of error regions surrounding camera presets.
H04L 12/18 - Arrangements for providing special services to substations for broadcast or conference
H04N 23/62 - Control of parameters via user interfaces
H04N 23/69 - Control of means for changing angle of the field of view, e.g. optical zoom objectives or electronic zooming
H04N 23/90 - Arrangement of cameras or camera modules, e.g. multiple cameras in TV studios or sports stadiums
H04R 1/40 - Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
Techniques are disclosed herein for providing audio enhancement and optimization of an immersive audio experience. Examples may include generating an audio feature set for a transduced audio stream captured in an environment, inputting the audio feature set to a neural network model configured to generate an audio isolation mask associated with the transduced audio stream, and generating isolated audio for the transduced audio stream based at least in part on the audio isolation mask.
G10L 21/0216 - Noise filtering characterised by the method used for estimating noise
G10L 25/30 - Speech or voice analysis techniques not restricted to a single one of groups characterised by the analysis technique using neural networks
G10L 25/57 - Speech or voice analysis techniques not restricted to a single one of groups specially adapted for particular use for comparison or discrimination for processing of video signals
Systems and methods are provided for receiving, from at least one microphone, boundary information defining one or more boundaries for an audio pick-up region; receiving, from the at least one microphone, sound location information indicating a detected sound position of an audio source located within the audio pick-up region; identifying, based on the sound location information and the boundary information, a first boundary of the one or more boundaries as being located near the detected sound position; calculating a first distance between the detected sound position and the first boundary; determining, based on the first distance, a depth of field parameter for at least one camera; and providing the depth of field parameter and the sound location information to the at least one camera.
A software-based conferencing platform is provided. The platform comprises a plurality of audio sources providing input audio signals, the audio sources including a virtual audio device driver configured to receive far-end input audio signals from a conferencing software module, and a network audio library configured to receive near-end input audio signals from one or more near-end audio devices. The platform further comprises a digital signal processing component configured to receive the input audio signals from the audio sources and generate audio output signals based the received signals, the digital signal processing component comprising an acoustic echo cancellation module configured to apply acoustic echo cancellation techniques to one or more of the near-end input audio signals.
H04L 7/00 - Arrangements for synchronising receiver with transmitter
H04L 43/08 - Monitoring or testing based on specific metrics, e.g. QoS, energy consumption or environmental parameters
H04R 1/40 - Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
Various embodiments of the present disclosure provide methods, apparatus, systems, devices, and/or the like for improved group extensibility through distributed local group configuration data storage. An example method includes receiving, from one or more devices located within a network proximity, shared group configuration data, and, responsive to determining that a device conflict exists associated with the shared group configuration data, selecting as current shared group configuration data received shared group configuration data associated with a most recent update timestamp; and transmitting the current shared group configuration data to each device located within the network proximity.
H04L 41/22 - Arrangements for maintenance, administration or management of data switching networks, e.g. of packet switching networks comprising specially adapted graphical user interfaces [GUI]
A microphone may comprise a housing and a rotatable positioning member. The rotatable positioning member may be configured to aid in the specific placement of the microphone capsule relative to the surface of an audio output device. The rotatable positioning member may offset a force caused by a tension in a cable connected to the housing based on the rotatable positioning member being rotated from a first position to a second position.
A videoconferencing device (1300) may include an elongated housing (1302) having two ends, a microphone array (104), one or more loudspeakers (110), and a plurality of cameras (1330, 1332), where at least one of the cameras is disposed at one end of the housing at least another of the cameras is disposed at an opposite end of the housing. The videoconferencing device may also include a microphone array disposed within an acoustical cavity (106), where the cameras and loudspeakers are disposed outside of the acoustical cavity. The videoconferencing devices may provide viewing angle diversity of the cameras to better capture images and/or video of participants in a videoconference, and also provide more optimal audio capture of the participants.
A videoconferencing device may include an elongated housing having two ends, a microphone array, one or more loudspeakers, and a plurality of cameras, where at least one of the cameras is disposed at one end of the housing at least another of the cameras is disposed at an opposite end of the housing. The videoconferencing device may also include a microphone array disposed within an acoustical cavity, where the cameras and loudspeakers are disposed outside of the acoustical cavity. The videoconferencing devices may provide viewing angle diversity of the cameras to better capture images and/or video of participants in a videoconference, and also provide more optimal audio capture of the participants.
An example immersive audio signal processing system and a computer-implemented method for generating a target arena environment audio stream are provided. The example immersive audio signal processing system includes a plurality of multi-lobe digital sound wave capture devices positioned within the arena environment. The plurality of multi-lobe digital sound wave capture devices is configured to direct first beamformed lobes to a playing region of the arena environment, second beamformed lobes to a spectator region of the arena environment, and third beamformed lobes to a noise source region of the arena environment. A digital signal processor is configured to isolate noise audio components originating from at least the spectator region or the noise source region from the audio signal stream and generate a target arena environment audio stream.
G06T 7/70 - Determining position or orientation of objects or cameras
G10L 25/57 - Speech or voice analysis techniques not restricted to a single one of groups specially adapted for particular use for comparison or discrimination for processing of video signals
Array microphone systems and methods for enabling predistortion of the aperture of an array microphone at various steering vectors are provided. The predistortion of the aperture can regularize the apparent aperture of the array microphone to be closer to that of the base aperture of the array microphone, while minimizing the number of microphone elements used. Improved directivity and more optimal pickup and capture of sound in the environment may result through the use of these systems and methods, which can help to avoid undesired noise and/or to more efficiently cover audio sources.
H04R 1/40 - Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
H04R 3/04 - Circuits for transducers for correcting frequency response
20.
ARRAY MICROPHONE APERTURE PREDISTORTION FOR IMPROVED DIRECTIVITY
Array microphone systems and methods for enabling predistortion of the aperture of an array microphone at various steering vectors are provided. The predistortion of the aperture can regularize the apparent aperture of the array microphone to be closer to that of the base aperture of the array microphone, while minimizing the number of microphone elements used. Improved directivity and more optimal pickup and capture of sound in the environment may result through the use of these systems and methods, which can help to avoid undesired noise and/or to more efficiently cover audio sources.
H04R 1/40 - Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
H04R 3/04 - Circuits for transducers for correcting frequency response
Embodiments include an array microphone, comprising: a plurality of microphone boards arranged in a linear pattern along a first axis and comprising a plurality of microphone elements configured to cover a plurality of frequency bands, each microphone board comprising: a first linear array comprising a first microphone element of the plurality of microphone elements and one or more second microphone elements of the plurality of microphone elements, the first microphone element located on the first axis and the one or more second microphone elements located on a second axis orthogonal to the first axis, and a second linear array comprising the first microphone element and one or more third microphone elements of the plurality of microphone elements, the one or more third microphone elements located on a third axis orthogonal to the first axis and to the second axis.
H04R 1/40 - Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
22.
BEAMFORMING FOR A MICROPHONE ARRAY BASED ON A STEERED RESPONSE POWER TRANSFORMATION OF AUDIO DATA
Techniques are disclosed herein for providing beamforming for at least one microphone array based at least in part on a steered response power (SRP) transformation of audio data. Examples may include receiving audio data from multiple audio capture devices comprising at least one microphone array located within an audio environment. Examples may also include generating an SRP transformation of the audio data. The SRP transformation may comprise a set of SRP weights for a spatial coordinate grid representing the audio environment. Examples may also include performing, based at least in part on a signal-to-noise ratio (SNR) estimate associated with the SRP transformation, one or more of beamforming steering or beamforming selection with respect to the at least one linear array microphone.
H04R 1/40 - Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
G10L 21/0216 - Noise filtering characterised by the method used for estimating noise
Embodiments include an array microphone, comprising: a plurality of microphone boards arranged in a linear pattern along a first axis and comprising a plurality of microphone elements configured to cover a plurality of frequency bands, each microphone board comprising: a first linear array comprising a first microphone element of the plurality of microphone elements and one or more second microphone elements of the plurality of microphone elements, the first microphone element located on the first axis and the one or more second microphone elements located on a second axis orthogonal to the first axis, and a second linear array comprising the first microphone element and one or more third microphone elements of the plurality of microphone elements, the one or more third microphone elements located on a third axis orthogonal to the first axis and to the second axis.
H04R 1/26 - Spatial arrangement of separate transducers responsive to two or more frequency ranges
H04R 1/40 - Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
24.
DETECTION AND ATTENUATION OF VOCAL ARTIFACTS IN AUDIO
Automatic detection and attenuation of vocal artifacts in an audio are described herein. First and second amplitude measurements may be generated for respective first and second frequency bands of an audio signal. A vocal artifact indication may be generated based on a ratio of the first and second amplitude measurements. The vocal artifact indication may be used to attenuate at least one of the first frequency band or the second frequency band. An attenuated signal may be provided in which one or more vocal artifacts have been attenuated.
G10L 21/0264 - Noise filtering characterised by the type of parameter measurement, e.g. correlation techniques, zero crossing techniques or predictive techniques
25.
DIRECTIONALLY DEPENDENT ACOUSTIC STRUCTURE FOR AUDIO PROCESSING RELATED TO AT LEAST ONE MICROPHONE SENSOR
Techniques for providing a directionally dependent acoustic structure for audio processing related to at least one microphone sensor are discussed herein. Examples may include transforming an augmented audio signal defined by a directionally dependent acoustic structure positioned proximate to at least one microphone sensor into at least one audio data object set, inputting the at least one audio data object set to a model configured to generate at least one spatialization data structure indicative of spatialization information for at least one audio source located within the audio environment, and generating audio processing data based at least in part on the at least one spatialization data structure.
Embodiments provide for employing an audio transformation model to resynthesize speech signals associated with a speaking entity. Examples can receive audio signals comprising speech signals that are captured by an audio capture device. Examples can divide the audio signals into audio segments and input the audio segments into an audio transformation model to generate a voice vector representation and a speech vector representation. The voice vector representation comprises characteristics related to a speaking voice associated with the speaking entity and the speech vector representation comprises one or more words spoken by the speaking entity. The one or more words comprised in the speech vector representation are associated with respective contextual attributes associated with the one or more words. The audio transformation model can utilize the voice vector representation and the speech vector representation to regenerate the speech signals associated with the speaking entity.
G10L 25/30 - Speech or voice analysis techniques not restricted to a single one of groups characterised by the analysis technique using neural networks
Methods and apparatuses for capturing and encoding ambisonic audio are described herein. An example ambisonic microphone may comprise a first microphone capsule oriented substantially toward a first vertex of a notional tetrahedron, a second microphone capsule oriented substantially toward a second vertex of the notional tetrahedron, a third microphone capsule oriented substantially toward a third vertex of the notional tetrahedron, and a fourth microphone capsule oriented substantially toward a fourth vertex of the notional tetrahedron.
Methods and apparatuses for capturing and encoding ambisonic audio are described herein. An example ambisonic microphone may comprise a first microphone capsule oriented substantially toward a first vertex of a notional tetrahedron, a second microphone capsule oriented substantially toward a second vertex of the notional tetrahedron, a third microphone capsule oriented substantially toward a third vertex of the notional tetrahedron, and a fourth microphone capsule oriented substantially toward a fourth vertex of the notional tetrahedron.
H04R 1/40 - Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
A soft decision audio decoding system for preserving audio continuity in a digital wireless audio receiver is provided that deduces the likelihood of errors in a received digital signal, based on generated hard bits and soft bits. The soft bits may be utilized by a soft audio decoder to determine whether the digital signal should be decoded or muted. The soft bits may be generated based on the detected point and a detected noise power, or by using a soft-output Viterbi algorithm. The value of the soft bits may indicate confidence in the strength of the hard bit generation. The soft decision audio decoding system may infer errors and decode perceptually acceptable audio without requiring error detection, as in conventional systems, as well as have low latency and improved granularity.
G10L 19/00 - Speech or audio signal analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
G10L 19/005 - Correction of errors induced by the transmission channel, if related to the coding algorithm
H03M 13/41 - Sequence estimation, i.e using statistical methods for the reconstruction of the original codes using the Viterbi algorithm or Viterbi processors
30.
PERSONAL STAGE MONITORING SYSTEM WITH PERSONAL MIXING
Aspects of the disclosure relate to personal stage monitoring (PSM) systems and methods. A PSM system may include a transmitter and a receiver, where the transmitter may transmit audio data to the receiver to provide performers with monitoring feedback. The PSM transmitter may include audio processing and/or personal mixing devices that provide a customized mix of audio signals that may be transmitted to the receiver. The integration of the audio processing and/or mixing devices with the PSM transmitter advantageously reduces the complexity of the overall audio system and improves audio quality and performance by reducing noise and latency of the audio system.
H03G 1/02 - Remote control of amplification, tone, or bandwidth
G10H 1/00 - ELECTROPHONIC MUSICAL INSTRUMENTS; INSTRUMENTS IN WHICH THE TONES ARE GENERATED BY ELECTROMECHANICAL MEANS OR ELECTRONIC GENERATORS, OR IN WHICH THE TONES ARE SYNTHESISED FROM A DATA STORE - Details of electrophonic musical instruments
H03G 9/00 - Combinations of two or more types of control, e.g. gain control and tone control
Aspects of the disclosure relate to an audio system architecture designed for sound reinforcement. The audio system may comprise one or more input device(s), one or more output device(s), one or more wireless hubs, and one or more user computing devices. Connections between the input device(s), output device(s), and the one or more wireless hubs may be wireless and automated and/or managed via the one or more user computing devices. The user computing devices or the wireless hub may adjust a configuration of each of the devices in the audio system. One or more of the devices in the audio system may be capable of a network connection which may enable recording, live-streaming to remote audiences, system management and operation, cloud-based storage and/or processing, and more.
Embodiments include a microphone array comprising a first plurality of directional microphone elements arranged in a first cluster formed by directing a front face of the microphone elements towards a center of the first cluster, and a second plurality of directional microphone elements arranged in a second cluster formed by directing a front face of the elements away from a center of the second cluster, wherein the first cluster is disposed vertically above the second cluster. Also provided is a microphone comprising a first microphone array comprising a plurality of directional microphone elements arranged in close proximity to each other and configured to capture near-field sounds within a first range of frequencies, and a second microphone array disposed concentrically around the first microphone array, the second array comprising a plurality of omnidirectional microphone elements configured to capture near-field sounds within a second range of frequencies higher than the first range.
G10L 15/22 - Procedures used during a speech recognition process, e.g. man-machine dialog
G10L 15/28 - Constructional details of speech recognition systems
G10L 25/51 - Speech or voice analysis techniques not restricted to a single one of groups specially adapted for particular use for comparison or discrimination
H04R 1/40 - Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
Aspects of the disclosure relate to an audio system architecture designed for sound reinforcement. The audio system may comprise one or more input device(s), one or more output device(s), one or more wireless hubs, and one or more user computing devices. Connections between the input device(s), output device(s), and the one or more wireless hubs may be wireless and automated and/or managed via the one or more user computing devices. The user computing devices or the wireless hub may adjust a configuration of each of the devices in the audio system. One or more of the devices in the audio system may be capable of a network connection which may enable recording, live-streaming to remote audiences, system management and operation, cloud-based storage and/or processing, and more.
Embodiments include a processing device communicatively coupled to a plurality of audio devices comprising at least one microphone and at least one speaker, and to a digital signal processing (DSP) component having a plurality of audio input channels for receiving audio signals captured by the at least one microphone, the processing device being configured to identify one or more of the audio devices based on a unique identifier associated with each of the one or more audio devices; obtain device information from each identified audio device; and adjust one or more settings of the DSP component based on the device information. A computer-implemented method of automatically configuring an audio conferencing system, comprising a digital signal processing (DSP) component and a plurality of audio devices including at least one speaker and at least one microphone, is also provided.
Systems and methods are provided herein for deploying, using at least one microphone, a first microphone lobe towards a first location, the first microphone lobe configured to capture one or more first audio signals from a first audio source located within a first audio pick-up region; deploying, using the at least one microphone, a second microphone lobe towards a second location, the second microphone lobe configured to capture one or more second audio signals from a second audio source located outside the first audio pick-up region; and removing, using at least one processor, off-axis noise from the one or more first audio signals by applying, to the one or more first audio signals, a mask determined based on the one or more second audio signals.
The present disclosure describes a digital-to-analog converter (DAC) comprising a programmable integrated circuit, a resampler, and a clock. The programmable integrated circuit may be configured to perform delta-sigma modulation to convert a digital signal to a coarse representation of an analog signal. The coarse representation of the analog signal may be sent to a first-stage of the resampler. After processing the signal, the first-stage of the resampler may pass a continuous signal to the second-stage of the resampler. The second-stage of the resampler produce a clean sampling instant for the signal that represents the point in time where the signal transitions from discrete time to continuous time. The analog signal may then be outputted to one or more receivers.
The present disclosure describes a digital-to-analog converter (DAC) comprising a programmable integrated circuit, a resampler, and a clock. The programmable integrated circuit may be configured to perform delta-sigma modulation to convert a digital signal to a coarse representation of an analog signal. The coarse representation of the analog signal may be sent to a first-stage of the resampler. After processing the signal, the first-stage of the resampler may pass a continuous signal to the second-stage of the resampler. The second-stage of the resampler produce a clean sampling instant for the signal that represents the point in time where the signal transitions from discrete time to continuous time. The analog signal may then be outputted to one or more receivers.
Systems and methods are provided herein for deploying, using at least one microphone, a first microphone lobe towards a first location, the first microphone lobe configured to capture one or more first audio signals from a first audio source located within a first audio pick-up region; deploying, using the at least one microphone, a second microphone lobe towards a second location, the second microphone lobe configured to capture one or more second audio signals from a second audio source located outside the first audio pick-up region; and removing, using at least one processor, off-axis noise from the one or more first audio signals by applying, to the one or more first audio signals, a mask determined based on the one or more second audio signals.
G10K 11/175 - Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound
Techniques for predicted audio immersion related to audio capture devices within an audio environment are discussed herein. Examples may include receiving audio stream inputs associated with audio capture devices positioned within one or more audio capture areas of an audio environment, where the audio environment comprises the one or more audio capture areas and one or more non-audio capture areas. Additionally, the audio stream inputs are transformed into respective audio feature sets. The respective audio feature sets are then input to a hybrid neural network model configured to generate respective augmented signal path data vectors for the respective audio stream inputs. Respective neural network paths of the hybrid neural network model may process one or more of the respective audio feature sets based on respective digital signal processing augmentation networks integrated within the hybrid neural network model.
Partially adaptive audio beamforming systems and methods are provided that enable improved acoustic echo cancellation of sound played on a loudspeaker that is in close proximity to a microphone array in an audio device. A stored beamformer parameter, such as an inverse covariance matrix, can be utilized by a frequency domain beamformer to generate a beamformed signal. The overall performance and resource usage by the audio device can be optimized.
Partially adaptive audio beamforming systems and methods are provided that enable improved acoustic echo cancellation of sound played on a loudspeaker that is in close proximity to a microphone array in an audio device. A stored beamformer parameter, such as an inverse covariance matrix, can be utilized by a frequency domain beamformer to generate a beamformed signal. The overall performance and resource usage by the audio device can be optimized.
Array microphone systems and methods that can automatically focus and/or place beamformed lobes in response to detected sound activity are provided. The automatic focus and/or placement of the beamformed lobes can be inhibited based on a remote far end audio signal. The quality of the coverage of audio sources in an environment may be improved by ensuring that beamformed lobes are optimally picking up the audio sources even if they have moved and changed locations.
Systems and methods are described herein for receiving, at a plurality of audio channels, respective audio signals captured by one or more microphones; based on a speech quality determination for each signal, identifying, in real time, a first subset of the audio channels as capturing speech audio, and a second subset of the audio channels as capturing noise audio, wherein the first subset comprises one or more audio channels and the second subset comprises one or more other audio channels; generating, using a first mixer, a mixed audio output that includes the signals received at the one or more audio channels; generating, using a second mixer, a noise mix that includes the signals received at the one or more other audio channels; and removing off-axis noise from the mixed audio output by applying, to that output, a mask determined based on the noise mix.
Systems and methods are described herein for receiving, at a plurality of audio channels, respective audio signals captured by one or more microphones; based on a speech quality determination for each signal, identifying, in real time, a first subset of the audio channels as capturing speech audio, and a second subset of the audio channels as capturing noise audio, wherein the first subset comprises one or more audio channels and the second subset comprises one or more other audio channels; generating, using a first mixer, a mixed audio output that includes the signals received at the one or more audio channels; generating, using a second mixer, a noise mix that includes the signals received at the one or more other audio channels; and removing off-axis noise from the mixed audio output by applying, to that output, a mask determined based on the noise mix.
G10L 25/60 - Speech or voice analysis techniques not restricted to a single one of groups specially adapted for particular use for comparison or discrimination for measuring the quality of voice signals
Systems, apparatuses, and methods are described for increasing a performance of a small wireless device comprising one or more conductive elements electrically coupled with an antenna. A user interface element of the wireless device may be configured to operate as a radiative component of an antenna structure. A power source element may be electrically isolated from a ground, resulting in decreased coupling with the antenna. Moreover, an antenna bandwidth may be increased, via modification of one or more antenna characteristics and/or the addition of one or more novel antenna components, in order to increase the detune-resiliency of the antenna. In addition, the radio frequency performance and reliability of the wireless device may be improved by maintaining fixed positioning between various components of the wireless device. Thus, the antenna and the one or more conductive elements may be configured to increase the device performance.
Array microphone systems and methods that can automatically focus and/or place beamformed lobes in response to detected sound activity are provided. The automatic focus and/or placement of the beamformed lobes can be inhibited based on a remote far end audio signal. The quality of the coverage of audio sources in an environment may be improved by ensuring that beamformed lobes are optimally picking up the audio sources even if they have moved and changed locations.
G10L 21/0216 - Noise filtering characterised by the method used for estimating noise
H04R 1/40 - Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
48.
ARRAY MICROPHONE SYSTEM AND METHOD OF ASSEMBLING THE SAME
Embodiments include a microphone assembly comprising an array microphone and a housing configured to support the array microphone and sized and shaped to be mountable in a drop ceiling in place of at least one of a plurality of ceiling tiles included in the drop ceiling. A front face of the housing includes a sound-permeable screen having a size and shape that is substantially similar to the at least one of the plurality of ceiling tiles. Embodiments also include an array microphone system comprising a plurality of microphones arranged, on a substrate, in a number of concentric, nested rings of varying sizes around a central point of the substrate. Each ring comprises a subset of the plurality of microphones positioned at predetermined intervals along a circumference of the ring.
H04R 1/40 - Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
A printed circuit board (PCB) assembly supports selecting alternate printed antenna geometries of an antenna by selectively placing common PCB components (for example, zero-ohm resistors) on the PCB. The desired zero-ohm resistors may be placed using automated part placement equipment commonly used to place surface mount components. The configurable antenna comprises at least one antenna section having a plurality of antenna components. Zero-ohm resistors are selectively placed in series along the antenna section to couple the desired conductor components when manufacturing the PCB assembly. With some embodiments, a configurable antenna includes a low frequency antenna section that may be selectively coupled with a high frequency antenna section antenna through one or more zero-ohm resistors, where each antenna section has a plurality of antenna components. With this approach, a common printed circuit board may be used to support a plurality of antenna variations spanning different frequency bands.
H01Q 1/27 - Adaptation for use in or on movable bodies
H01Q 1/38 - Structural form of radiating elements, e.g. cone, spiral, umbrella formed by a conductive layer on an insulating support
H01Q 5/321 - Individual or coupled radiating elements, each element being fed in an unspecified way using frequency dependent circuits or components, e.g. trap circuits or capacitors within a radiating element or between connected radiating elements
H01Q 9/42 - Resonant antennas with feed to end of elongated active element, e.g. unipole with folded element, the folded parts being spaced apart a small fraction of the operating wavelength
50.
Band Selectable Geometry for Printed Circuit Board Antennas
A printed circuit board (PCB) assembly supports selecting alternate printed antenna geometries of an antenna by selectively placing common PCB components (for example, zero-ohm resistors) on the PCB. The desired zero-ohm resistors may be placed using automated part placement equipment commonly used to place surface mount components. The configurable antenna comprises at least one antenna section having a plurality of antenna components. Zero-ohm resistors are selectively placed in series along the antenna section to couple the desired conductor components when manufacturing the PCB assembly. With some embodiments, a configurable antenna includes a low frequency antenna section that may be selectively coupled with a high frequency antenna section antenna through one or more zero-ohm resistors, where each antenna section has a plurality of antenna components. With this approach, a common printed circuit board may be used to support a plurality of antenna variations spanning different frequency bands.
H01Q 5/335 - Individual or coupled radiating elements, each element being fed in an unspecified way using frequency dependent circuits or components, e.g. trap circuits or capacitors at the feed, e.g. for impedance matching
H01Q 9/26 - Resonant antennas with feed intermediate between the extremities of the antenna, e.g. centre-fed dipole with folded element or elements, the folded parts being spaced apart a small fraction of operating wavelength
H01Q 15/00 - Devices for reflection, refraction, diffraction or polarisation of waves radiated from an antenna, e.g. quasi-optical devices
51.
FULL-BAND AUDIO SIGNAL RECONSTRUCTION ENABLED BY OUTPUT FROM A MACHINE LEARNING MODEL
Techniques are disclosed herein for providing full-band audio signal reconstruction enabled by output from a machine learning model trained based on an audio feature set extracted from a portion of the audio signal. Examples may include generating a model input audio feature set for a first frequency portion of an audio signal defined based on a hybrid audio processing frequency threshold. Examples may also include inputting the model input audio feature set to a machine learning model configured to generate a frequency characteristics output related to the first frequency portion of the audio signal. Examples may also include applying the frequency characteristics output to at least a second frequency portion of the audio signal to generate a reconstructed full-band audio signal.
G10L 19/06 - Determination or coding of the spectral characteristics, e.g. of the short-term prediction coefficients
G10L 19/02 - Speech or audio signal analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
G10L 25/21 - Speech or voice analysis techniques not restricted to a single one of groups characterised by the type of extracted parameters the extracted parameters being power information
52.
FULL-BAND AUDIO SIGNAL RECONSTRUCTION ENABLED BY OUTPUT FROM A MACHINE LEARNING MODEL
Techniques are disclosed herein for providing full-band audio signal reconstruction enabled by output from a machine learning model trained based on an audio feature set extracted from a portion of the audio signal. Examples may include generating a model input audio feature set for a first frequency portion of an audio signal defined based on a hybrid audio processing frequency threshold. Examples may also include inputting the model input audio feature set to a machine learning model configured to generate a frequency characteristics output related to the first frequency portion of the audio signal. Examples may also include applying the frequency characteristics output to at least a second frequency portion of the audio signal to generate a reconstructed full-band audio signal.
A wireless audio system using low overhead in-band control and audio transmission is provided. The wireless audio system includes a first wireless audio device configured to operate separate physical layer channels for audio data and control data, and transmit the audio data and control data using a single wideband carrier. The wireless audio system also includes one or more second wireless audio devices configured to receive the audio data and control data, and execute an instruction based on the control data.
H04W 4/80 - Services using short range communication, e.g. near-field communication [NFC], radio-frequency identification [RFID] or low energy communication
H04L 1/00 - Arrangements for detecting or preventing errors in the information received
H04L 5/00 - Arrangements affording multiple use of the transmission path
Acoustic echo cancellation systems and methods are provided that improve the quality of the audio transmitted from by an audio device a near end to a far end when a doubletalk condition is present, including allowing certain subbands of an echo-cancelled signal to be less attenuated by overriding certain gains of subbands of the echo-cancelled audio signal in a non-linear processor, and compressing and applying makeup gain to a remote audio signal.
H04M 9/08 - Two-way loud-speaking telephone systems with means for conditioning the signal, e.g. for suppressing echoes for one or both directions of traffic
A wireless audio system including a transmitter using multiple antenna diversity techniques for different signal types is provided. Multipath performance may be optimized, along with improved spectral efficiency of the system.
H04B 7/06 - Diversity systems; Multi-antenna systems, i.e. transmission or reception using multiple antennas using two or more spaced independent antennas at the transmitting station
H04L 1/06 - Arrangements for detecting or preventing errors in the information received by diversity reception using space diversity
H04L 7/00 - Arrangements for synchronising receiver with transmitter
Acoustic echo cancellation systems and methods are provided that improve the quality of the audio transmitted from by an audio device a near end to a far end when a doubletalk condition is present, including allowing certain subbands of an echo-cancelled signal to be less attenuated by overriding certain gains of subbands of the echo-cancelled audio signal in a non-linear processor, and compressing and applying makeup gain to a remote audio signal.
H04M 9/08 - Two-way loud-speaking telephone systems with means for conditioning the signal, e.g. for suppressing echoes for one or both directions of traffic
57.
TRANSMIT REDUNDANCY AND AUTONOMOUS RECEIVER CHANNEL SELECTION IN A WIRELESS IN-EAR MONITOR AUDIO SYSTEM
Techniques for providing transmit redundancy for a wireless audio system, such as an in-ear monitor audio system, are discussed herein. Some embodiments may include tuning to a first wireless communication channel associated with a first carrier frequency to receive a radio frequency signal. Based on a determination that a communication channel condition for the first wireless communication channel satisfies a communication channel condition threshold, various embodiments include tuning to a second wireless communication channel associated with a second carrier frequency to receive the radio frequency signal, where the second carrier frequency is different from the first carrier frequency.
Techniques for providing transmit redundancy for a wireless audio system, such as an in-ear monitor audio system, are discussed herein. Some embodiments may include tuning to a first wireless communication channel associated with a first carrier frequency to receive a radio frequency signal. Based on a determination that a communication channel condition for the first wireless communication channel satisfies a communication channel condition threshold, various embodiments include tuning to a second wireless communication channel associated with a second carrier frequency to receive the radio frequency signal, where the second carrier frequency is different from the first carrier frequency.
H04B 1/10 - Means associated with receiver for limiting or suppressing noise or interference
H04H 60/32 - Arrangements for monitoring conditions of receiving stations, e.g. malfunction or breakdown of receiving stations
H04H 60/43 - Arrangements for identifying or recognising characteristics with a direct linkage to broadcast information or to broadcast space-time, e.g. for identifying broadcast stations or for identifying users for identifying broadcast time or space for identifying broadcast space, i.e. broadcast channels, broadcast stations or broadcast areas for identifying broadcast channels
Audio-visual systems and methods are configured to determine a first talker location based on a first group of sound locations corresponding to audio detected by the microphone in association with one or more talkers; receive a new sound location for new audio detected by the microphone in association with at least one talker; determine a proximity of the new sound location to the first group of sound locations; based on the new sound location being in close proximity to one or more of the sound locations in the first group, determine a second talker location based on the new sound location and the first group of sound locations; determine a second proximity of the second talker location to the first talker location; provide the second talker location to the camera if the second proximity meets or exceeds a threshold; and otherwise, provide the first talker location the camera.
H04N 23/695 - Control of camera direction for changing a field of view, e.g. pan, tilt or based on tracking of objects
H04N 23/62 - Control of parameters via user interfaces
H04N 23/68 - Control of cameras or camera modules for stable pick-up of the scene, e.g. compensating for camera body vibrations
G10L 25/51 - Speech or voice analysis techniques not restricted to a single one of groups specially adapted for particular use for comparison or discrimination
61.
SYSTEM AND METHOD FOR CAMERA MOTION STABILIZATION USING AUDIO LOCALIZATION
Audio-visual systems and methods are configured to determine a first talker location (Pa) based on a first group of sound locations corresponding to audio detected by the microphone (104) in association with one or more talkers (102); receive a new sound location for new audio detected by the microphone in association with at least one talker; determine a proximity of the new sound location to the first group of sound locations; based on the new sound location being in close proximity to one or more of the sound locations in the first group, determine a second talker location (Pb) based on the new sound location and the first group of sound locations; determine a second proximity of the second talker location to the first talker location; provide the second talker location to the camera (106) if the second proximity meets or exceeds a threshold; and otherwise, provide the first talker location the camera.
Transducer steering and configuration systems and methods using a local positioning system are provided. The position and/or orientation of transducers, devices, and/or objects within a physical environment may be utilized to enable steering of lobes and nulls of the transducers, to create self-assembling arrays of the transducers, and to enable monitoring and configuration of the transducers, devices, and objects through an augmented reality interface. The transducers and devices may be more optimally configured which can result in better capture of sound, better reproduction of sound, improved system performance, and increased user satisfaction.
An audio device may be connected to a communication network. The audio device may send or receive audio data via a network, based on a network clock that may be synchronized with other audio device connected to the network. The audio device may buffer, convert between digital audio signals and analog audio signals, encrypt, decrypt, packetize, depacketize, compress, and/or decompress audio data using a local asynchronous media clock using a relatively lower precision clocking technology such as a crystal-based oscillator.
An audio device may be connected to a communication network. The audio device may send or receive audio data via a network, based on a network clock that may be synchronized with other audio device connected to the network. The audio device may buffer, convert between digital audio signals and analog audio signals, encrypt, decrypt, packetize, depacketize, compress, and/or decompress audio data using a local asynchronous media clock using a relatively lower precision clocking technology such as a crystal-based oscillator.
A wireless microphone system utilizes an antenna structure that supports dual frequency bands. With some embodiments, the wireless microphone system comprises a first apparatus (for example, a wireless transmitter) and an attached second apparatus (for example, an attached microphone), where the chassis of the first apparatus and the housing of the second apparatus support the antenna structure through an electrical connector (for example, through an outer shell of a Cannon (XLR) connector) between the two apparatuses. The chassis and the housing may support first and second halves of a dipole antenna, respectively. The first apparatus supports first and second electrical circuits that are operational within a first and second frequency band, respectively, and that connect to first and second input ports of a combining circuit (for example, a diplexer filter). The output port of the combining circuit connects to the housing through the electrical connector.
Conferencing systems and methods configured to generate talker coordinates for directing a camera towards talker locations in an environment are disclosed, as well as talker tracking using multiple microphones and multiple cameras. One method includes determining, using a first microphone array and based on audio associated with a talker, a first talker location in a first coordinate system relative to the first microphone array; determining, using a second microphone array and based on the audio associated with the talker, a second talker location in a second coordinate system relative to the second microphone array; determining, based on the first talker location and the second talker location, an estimated talker location in a third coordinate system relative to a camera; and transmitting, to the camera, the estimated talker location in the third coordinate system to cause the camera to point the camera towards the estimated talker location.
A microphone may digitize multiple analog audio channels into multiple digital audio channels, digitally process the digital audio channels, and output the digital audio channels to another device while maintaining channel separation. The microphone may also include a touch-sensitive user interface that may have multiple live meter modes and a user-selectable color theme.
Conferencing systems and methods configured to generate talker coordinates for directing a camera towards talker locations in an environment are disclosed, as well as talker tracking using multiple microphones and multiple cameras. One method includes determining, using a first microphone array (104) and based on audio associated with a talker (102), a first talker location (P1) in a first coordinate system relative to the first microphone array; determining, using a second microphone array (104) and based on the audio associated with the talker (102), a second talker location (P2) in a second coordinate system relative to the second microphone array; determining, based on the first talker location and the second talker location, an estimated talker location (P3) in a third coordinate system relative to a camera (106); and transmitting, to the camera, the estimated talker location in the third coordinate system to cause the camera to point the camera towards the estimated talker location.
An example immersive audio signal processing system and a computer-implemented method for generating a target arena environment audio stream are provided. The example immersive audio signal processing system includes a plurality of multi-lobe digital sound wave capture devices positioned within the arena environment. The plurality of multi-lobe digital sound wave capture devices is configured to direct first beamformed lobes to a playing region of the arena environment, second beamformed lobes to a spectator region of the arena environment, and third beamformed lobes to a noise source region of the arena environment. A digital signal processor is configured to isolate noise audio components originating from at least the spectator region or the noise source region from the audio signal stream and generate a target arena environment audio stream.
H04R 1/40 - Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
H04N 21/8547 - Content authoring involving timestamps for synchronizing content
70.
MULTI-LOBE DIGITAL MICROPHONE ENABLED AUDIO CAPTURE AND SPATIALIZATION FOR GENERATING AN IMMERSIVE ARENA BASED AUDIO EXPERIENCE
An example immersive audio signal processing system and a computer-implemented method for generating a target arena environment audio stream are provided. The example immersive audio signal processing system includes a plurality of multi-lobe digital sound wave capture devices positioned within the arena environment. The plurality of multi-lobe digital sound wave capture devices is configured to direct first beamformed lobes to a playing region of the arena environment, second beamformed lobes to a spectator region of the arena environment, and third beamformed lobes to a noise source region of the arena environment. A digital signal processor is configured to isolate noise audio components originating from at least the spectator region or the noise source region from the audio signal stream and generate a target arena environment audio stream.
Embodiments include a wireless microphone comprising an elongated main body configured for handheld operation of the microphone; a display bezel area included in the main body; a first antenna positioned at a bottom end of the main body; and a second antenna integrated into the display bezel area. Embodiments also include a wireless handheld microphone comprising a main body having a conductive housing and a tubular shape configured for handheld operation of the microphone; an opening included on a side surface of the conductive housing; a non-conductive cover coupled to the conductive housing and configured to cover the opening; and an antenna positioned adjacent to the non-conductive cover.
H01Q 1/22 - Supports; Mounting means by structural association with other equipment or articles
H04R 1/00 - LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS - Details of transducers
H04R 1/04 - Structural association of microphone with electric circuitry therefor
H01Q 5/50 - Feeding or matching arrangements for broad-band or multi-band operation
H01Q 1/36 - Structural form of radiating elements, e.g. cone, spiral, umbrella
H01Q 1/50 - Structural association of antennas with earthing switches, lead-in devices or lightning protectors
Methods and apparatuses for automatic analysis and optimization of an audio signal are described herein. An example method may comprise receiving, by an audio signal optimizer, a first indication to perform an audio signal optimization, receiving an audio signal from an input device, recording a sample of the audio signal, analyzing the sample of the audio signal for at least one audio parameter, and performing, based on an analysis of the sample of the audio signal, the audio signal optimization of the audio signal, wherein the audio signal optimization comprises a configuration of a gain level of the audio signal.
The present disclosure describes a wireless microphone system that allows one or more microphones to wirelessly communicate with one or more wireless receivers. The wireless microphone system may allow for a plurality of microphones to be used interchangeably with the one or more receivers. The communications between the one or more wireless microphones and the one or more wireless receivers may utilize more than one wireless communication protocol, such as Bluetooth Low Energy (BLE) as well as a proprietary protocol such as a non-Bluetooth 2.4 GHz wireless communication protocol, and the system may switch between the wireless communication protocols. The one or more wireless microphones may further synchronize communications with the one or more receivers.
H04R 1/00 - LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS - Details of transducers
Methods and apparatuses for automatic analysis and optimization of an audio signal are described herein. An example method may comprise receiving, by an audio signal optimizer, a first indication to perform an audio signal optimization, receiving an audio signal from an input device, recording a sample of the audio signal, analyzing the sample of the audio signal for at least one audio parameter, and performing, based on an analysis of the sample of the audio signal, the audio signal optimization of the audio signal, wherein the audio signal optimization comprises a configuration of a gain level of the audio signal.
Various embodiments of the present disclosure provide methods, apparatus, systems, devices, and/or the like for inferring characteristics of a physical enclosure using a plurality of audio signals. The plurality of audio signals may be processed using a feature extraction framework to generate structured audio event data sets, which may be processed using an audio event framework to determine the characteristics of the physical enclosure.
Techniques for isolating audio signals related to audio sources within an audio environment are discussed herein. Examples may include receiving a plurality of audio data objects. Each audio data object includes digitized audio signals captured by a capture device positioned within an audio environment. Examples may also include inputting the audio data objects to a source localizer model that is configured to generate, based on the audio data objects, one or more audio source position estimate objects. Examples may also include inputting the audio data objects and each audio source position estimate object to a source generator model of one or more source generator models. The source generator model is configured to generate, based on the audio source position estimate object, a source isolated audio output component. The source isolated audio output component may include isolated audio signals associated with an audio source within the audio environment.
Techniques for isolating audio signals related to audio sources within an audio environment are discussed herein. Examples may include receiving a plurality of audio data objects. Each audio data object includes digitized audio signals captured by a capture device positioned within an audio environment. Examples may also include inputting the audio data objects to a source localizer model that is configured to generate, based on the audio data objects, one or more audio source position estimate objects. Examples may also include inputting the audio data objects and each audio source position estimate object to a source generator model of one or more source generator models. The source generator model is configured to generate, based on the audio source position estimate object, a source isolated audio output component. The source isolated audio output component may include isolated audio signals associated with an audio source within the audio environment.
A microphone may comprise a microphone element for detecting sound, and a digital signal processor configured to process a first audio signal that is based on the sound in accordance with a selected one of a plurality of digital signal processing (DSP) modes. Each of the DSP modes may be for processing the first audio signal in a different way. For example, the DSP modes may account for distance of the person speaking (e.g., near versus far) and/or desired tone (e.g., darker, neutral, or bright tone). At least some of the modes may have, for example, an automatic level control setting to provide a more consistent volume as the user changes their distance from the microphone or changes their speaking level, and that may be associated with particular default (and/or adjustable) values of the parameters attack, hold, decay, maximum gain, and/or target gain, each depending upon which DSP is being applied.
Various embodiments of the present disclosure provide methods, apparatus, systems, devices, and/or the like for reducing defects of audio signal samples by using at least one of audio source feature separation machine learning models, audio generation machine learning models, and/or audio source feature classification machine learning models.
A wireless power transfer system is provided, comprising a first device having a power supply and configured to wirelessly transmit electric power from the power supply, and a second device having one or more electrical components and configured to wirelessly receive the electric power and provide it to the one or more electrical components for consumption. The second device further comprises an alignment module configured to activate an indicator if a measured level of the received power is greater than a threshold level. Also provided is a wireless system comprising a first device having a power supply and a wireless transmitter configured to transmit a data signal and an electric power signal, and a second device having a wireless receiver configured to receive the electric power signal and data signal, and one or more electrical components configured to consume the received power and process the received data.
H02J 50/90 - Circuit arrangements or systems for wireless supply or distribution of electric power involving detection or optimisation of position, e.g. alignment
H04W 4/80 - Services using short range communication, e.g. near-field communication [NFC], radio-frequency identification [RFID] or low energy communication
G08B 5/00 - Visible signalling systems, e.g. personal calling systems, remote indication of seats occupied
Various embodiments of the present disclosure provide methods, apparatus, systems, devices, and/or the like for reducing defects of audio signal samples by using at least one of audio source feature separation machine learning models, audio generation machine learning models, and/or audio source feature classification machine learning models.
Described are systems, methods, apparatuses, and computer program products for wireless in-ear-monitoring (IEM) of audio. A system includes transmitter(s) configured to map orthogonal sub-carriers of a digital signal to narrowband receivers to form receiver-allocated audio channels, modulate the digital signal, and transmit the signal as an ultra-high frequency (UHF) analog carrier wave comprising the orthogonal sub-carriers to the nearby receiver. A narrowband receiver is configured to demodulate and sample the sub-carriers allocated to the receiver. Sub-carriers can be positioned orthogonal to one another in adjacent sub-bands of the frequency domain and beacon symbols and pilot signals can be iteratively provided in the same portion of the frequency domain for each channel. The receiver can use non-data-aided and data-aided approaches for synchronization of the time domain and frequency domain waveforms of the received signal to the transmitted signal prior to sampling the allocated sub-carriers.
Embodiments include an audio system comprising an audio device, a speaker, and a processor. The audio system is configured to receive data from one or more sensors corresponding to persons in a room and/or characteristics of a room, and responsively take action to modify one or more characteristics of the audio system, share the information with other systems or devices, and track data over time to determine patterns and trends in the data.
H04L 12/18 - Arrangements for providing special services to substations for broadcast or conference
H04R 1/40 - Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
Described are in-ear monitoring (IEM) systems configured for audio performance environments requiring low audio latency and high scalability. IEM systems can include an audio channel allocation device that determines audio channel allocation for transmitting audio payload to IEM devices. Audio payload may be allocated to a radio frame based on, e.g., bit rate, modulation and coding scheme, latency/fidelity requirements, etc. IEM devices can include audio driver(s) configured to generate an audio output, a circuit configured to control audio output generation by the driver(s), in-ear portion(s), and a bodypack receiver. IEM devices can receive the audio allocation information, configure its circuit accordingly, receive audio payload carried in a carrier wave based on the allocation information, and generate the audio output based on the audio payload.
G10H 1/00 - ELECTROPHONIC MUSICAL INSTRUMENTS; INSTRUMENTS IN WHICH THE TONES ARE GENERATED BY ELECTROMECHANICAL MEANS OR ELECTRONIC GENERATORS, OR IN WHICH THE TONES ARE SYNTHESISED FROM A DATA STORE - Details of electrophonic musical instruments
88.
IN-EAR MONITORING (IEM) SYSTEM CONFIGURED FOR AUDIO PERFORMANCE ENVIRONMENTS REQUIRING LOW AUDIO LATENCY AND HIGH SCALABILITY
Described are in-ear monitoring (IEM) systems configured for audio performance environments requiring low audio latency and high scalability. IEM systems can include an audio channel allocation device that determines audio channel allocation for transmitting audio payload to IEM devices. Audio payload may be allocated to a radio frame based on, e.g., bit rate, modulation and coding scheme, latency/fidelity requirements, etc. IEM devices can include audio driver(s) configured to generate an audio output, a circuit configured to control audio output generation by the driver(s), in-ear portion(s), and a bodypack receiver. IEM devices can receive the audio allocation information, configure its circuit accordingly, receive audio payload carried in a carrier wave based on the allocation information, and generate the audio output based on the audio payload.
A microphone module comprises a housing, an audio bus, and a first plurality of microphones in communication with the audio bus. The microphone module further comprises a module processor in communication with the first plurality of microphones and the audio bus. The module processor is configured to detect the presence of an array processor in communication with the audio bus, detect the presence of a second microphone module in communication with the audio bus, and configure the audio bus to pass audio signals from both the first plurality of microphones and the second microphone module to the array processor.
H04R 1/40 - Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
H04M 3/56 - Arrangements for connecting several subscribers to a common circuit, i.e. affording conference facilities
An audio system that synchronizes multiple audio devices with reduced timing overhead. For example, the audio system may utilize a multi-dimensional buffer to efficiently receive, synchronize, process, and send audio data.
The present disclosure describes systems, apparatuses, methods, and computer-readable media for multi-channel software-defined radio (SDR) audio transceivers, multi-user audio systems using the same, and methods of using the same. SDR audio transceivers can comprise an antenna and/or radio transmitter. SDR audio transceivers can be configured for multi-channel audio transmission. The multi-channel audio transmission can include wideband channels or channels allocated across a wideband of spectrum. The multi-channel audio transmission can include narrowband channels or channels allocated to a relatively narrower band of spectrum.
H04L 41/40 - Arrangements for maintenance, administration or management of data switching networks, e.g. of packet switching networks using virtualisation of network functions or resources, e.g. SDN or NFV entities
A backplate assembly for a condenser microphone. The backplate may be coated with a parylene configured to help reduce the flatness deviation of the backplate across the diameter of the backplate. A plurality of openings may extend from the top portion of the backplate to the bottom portion of the backplate.
The present disclosure describes systems, apparatuses, methods, and computer-readable media for multi-channel software-defined radio (SDR) audio transceivers, multi-user audio systems using the same, and methods of using the same. SDR audio transceivers can comprise an antenna and/or radio transmitter. SDR audio transceivers can be configured for multi-channel audio transmission. The multi-channel audio transmission can include wideband channels or channels allocated across a wideband of spectrum. The multi-channel audio transmission can include narrowband channels or channels allocated to a relatively narrower band of spectrum.
H04B 1/00 - TRANSMISSION - Details of transmission systems not characterised by the medium used for transmission
H04B 1/20 - Circuits for coupling gramophone pick-up, recorder output, or microphone to receiver
H04B 7/08 - Diversity systems; Multi-antenna systems, i.e. transmission or reception using multiple antennas using two or more spaced independent antennas at the receiving station
A microphone may digitize multiple analog audio channels into multiple digital audio channels, digitally process the digital audio channels, and output the digital audio channels to another device while maintaining channel separation. The microphone may also include a touch-sensitive user interface that may have multiple live meter modes and a user-selectable color theme.
The present disclosure describes a wireless microphone system that allows one or more microphones to wirelessly communicate with a receiver. Additionally, the wireless microphone system may allow for a plurality of microphones to be used interchangeably with the receiver. To ensure communication between the receiver and the one or more microphones, the receiver may occasionally transmit a synchronization signal to the one or more microphones. In response to receiving the synchronization signal, at least one of the one or more microphones may determine that a clock of the at least one microphone is drifting from the master audio clock of the receiver. The at least one microphone may then adjust the microphone's audio clock to re-synchronize the audio clock of the microphone with the master audio clock of the receiver.
The present disclosure describes a wireless microphone system that allows one or more microphones to wirelessly communicate with a receiver. Additionally, the wireless microphone system may allow for a plurality of microphones to be used interchangeably with the receiver. To ensure communication between the receiver and the one or more microphones, the receiver may occasionally transmit a synchronization signal to the one or more microphones. In response to receiving the synchronization signal, at least one of the one or more microphones may determine that a clock of the at least one microphone is drifting from the master audio clock of the receiver. The at least one microphone may then adjust the microphone's audio clock to re-synchronize the audio clock of the microphone with the master audio clock of the receiver.
H04R 1/00 - LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS - Details of transducers
Array microphone systems and methods that can automatically focus and/or place beamformed lobes in response to detected sound activity are provided. The automatic focus and/or placement of the beamformed lobes can be inhibited based on a remote far end audio signal. The quality of the coverage of audio sources in an environment may be improved by ensuring that beamformed lobes are optimally picking up the audio sources even if they have moved and changed locations.
A wireless earphone incorporates a wire antenna having a form factor driven innovative antenna shape that minimizes antenna detuning caused by user interactions with the earphones. The wire shape, diameter, and distance of the wire antenna from the printed circuit board (PCB) are selected for an acceptable tradeoff between antenna bandwidth and radiated efficiency. By inserting an end through a through-hole of the PCB, the wire antenna is electrically connected to a multi-layer PCB without traditional approaches such as springs, pogo pins, and the like. An antenna holder further secures the antenna within a thin profile housing for precise placement and manufacturing consistency. A PCB-specific RF VIA geometry is also utilized for partial impedance matching of a transmission line to the wire antenna. In addition, a more constant impedance is maintained along the transmission line connecting a radio device with the wire antenna.
Embodiments include a method of reducing echo in an audio system comprising a microphone, an acoustic echo canceller (AEC), and at least one processor, the method comprising receiving, by the at least one processor, an audio signal detected by the microphone; deploying, by the at least one processor, a microphone lobe towards a first location associated with the detected audio signal; obtaining, by the at least one processor, one or more AEC parameters for the first location, the one or more AEC parameters being stored in a memory in communication with the at least one processor; initializing, by the at least one processor, the AEC using the one or more AEC parameters; and generating, by the at least one processor, an echo-cancelled output signal using the initialized AEC and based on the detected audio signal and a reference signal provided to the AEC.
G10K 11/178 - Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound by electro-acoustically regenerating the original acoustic waves in anti-phase
100.
SYSTEM AND METHOD FOR OPTIMIZATION OF ACOUSTIC ECHO CANCELLATION CONVERGENCE
Embodiments include a method of reducing echo in an audio system comprising a microphone, an acoustic echo canceller (AEC), and at least one processor, the method comprising receiving, by the at least one processor, an audio signal detected by the microphone; deploying, by the at least one processor, a microphone lobe towards a first location associated with the detected audio signal; obtaining, by the at least one processor, one or more AEC parameters for the first location, the one or more AEC parameters being stored in a memory in communication with the at least one processor; initializing, by the at least one processor, the AEC using the one or more AEC parameters; and generating, by the at least one processor, an echo-cancelled output signal using the initialized AEC and based on the detected audio signal and a reference signal provided to the AEC.
H04M 9/08 - Two-way loud-speaking telephone systems with means for conditioning the signal, e.g. for suppressing echoes for one or both directions of traffic
