356results about "Public address systems" patented technology

A method for synchronizing audio reproduction in collocated end devices is presented. Each of the devices auto-correlates using noise to determine a threshold prior to the antenna receiving an audio signal. When the devices receive a common audio signal, they provide audio outputs. Each device cross-correlates its audio output with the audio outputs of the other devices. The timing of the audio output of each device is then adjusted such that the audio outputs of all of the devices align temporally with the lagging or leading device.

Digital signal processing (DSP) may be distributed across a plurality of endpoints and one or more aggregators in an audio system. An aggregator may perform DSP and / or may control DSP features performed by endpoints based on DSP feature capability of the endpoints and the aggregator. Links may be established between the aggregator and the endpoints to communicate audio signals and related information for performing the DSP features.

An audiovisual system uses a spatial position detection module to process audio signals and determine the location and orientation of a speaking participant. Based on this data, the system dynamically controls sensors to optimize audio and video capture. Behavioral and contextual information may also be used to train intelligence models for improved system performance. Further, sensor arrays may be utilized to identify gaze vectors of participants to select a camera sensor of the sensor array. Meeting content collected with the sensor array may be organized into meeting metrics in accordance with an analytics strategy before training an intelligence module with the meeting metrics. Meeting content may also be configured into digital tiles in accordance with a tile strategy. At least one of the digital tiles may be altered in response to a meeting condition detected by the sensor array.

Systems and methods are disclosed in which a playback device transmits a first sound signal including a predetermined waveform. In one example, the playback device receives a second sound signal including at least one reflection of the first sound signal. The second sound signal is processed to determine a location of a person relative to the playback device, and a characteristic of audio reproduction by the playback device is selected, based on the determined location of the person.

The invention provides a sound amplification device based on an M-H metamaterial and a sound transmission method using the sound amplification device, the sound amplification device at least comprises an upper shell, a lower shell and a waveguide located between the upper shell and the lower shell, and the waveguide is made of the Mie metamaterial; the waveguide comprises an outer wall, a plurality of sound wave inlets are formed in the outer wall, and the upper shell, the lower shell and the outer wall enclose a waveguide space; a through hole penetrating through the waveguide and the upper shell is formed in the waveguide space, and the through hole is a sound wave outlet in the sound amplification device; the waveguide comprises a plurality of fan parts which are arranged around the through hole in a circumferential radiation manner; a waveguide channel penetrating through the sound wave outlet and the sound wave inlet is formed between the adjacent fan parts; the control current of the sound source is determined by the size of the sound amplifying device and the scattering distribution condition of the sound amplifying device in different resonance modes, the sound at the sound wave outlet is larger than the sound at the sound source in energy level, and an included angle exists in the propagation direction. By determining the control current at the sound source which is most matched with the size of the sound amplifying device, the sound propagation control with low cost, high accuracy and wide application range is realized.

In a wireless stereo system including first and second battery-powered speakers, if low power is sensed in the second speaker, the user may be given the option of shifting the first speaker to mono mode, or maintain the first speaker in stereo mode, playing its assigned channel, or alter a left / right stereo balance between the speakers. Speaker setting may also change depending on whether a speaker is on or off a charging dock.

The utility model relates to the technical field of loudspeakers, and particularly discloses an invisible sound reinforcement plate which comprises a middle core layer which is a composite base material formed by interweaving carbon fibers and aramid fibers. The supporting layer is arranged at the upper end of the middle core layer, and the supporting layer is composed of a plurality of frameworks with the sections of regular hexagon structures and is of a honeycomb structure as a whole; the sound reinforcement plate has the advantages of high strength, good tensile effect, support of ultrahigh-frequency sound wave conduction, light weight and reduction of the overall weight of the sound reinforcement plate; the supporting layer is composed of a plurality of frameworks with the sections of regular hexagon structures, the whole supporting layer is of a honeycomb structure, rigidity is good, bending resistance is high, deformation is not prone to being caused in the assembling and using process, and therefore the using performance of the sound reinforcement plate is improved. The supporting ribs form a continuous force transmission path and disperse local stress, the trapezoid cross section reduces the stress concentration coefficient, and the buffering wrapping edge is made of silica gel materials, prevents external dust from invading the sound reinforcement plate interlayer and meanwhile plays a role in buffering and shock resistance.

Systems and techniques for localization of disturbances are illustrated. One embodiment includes a method for detecting a disturbance in an area of operation, the area of operation including a plurality of transmitter-receiver pairs. The method obtains, based on a first plurality of signal transmissions between the plurality of transmitter-receiver pairs, at least one baseline channel impulse response (CIR) measurement. The method obtains, based on a second plurality of signal transmissions between the plurality of transmitter-receiver pairs, at least one additional CIR measurement; The method obtains, based on the at least one additional CIR measurement and the at least one baseline CIR measurement, at least one residual CIR measurement. The method derives, based on a sufficient similarity between the at least one residual CIR measurement and at least one particular key of a plurality of keys, localization data for at least one disturbance detected within the area of operation.

Example techniques may involve multiple calibrations for one or more playback devices. An example implementation may involve detecting, via a microphone, calibration sounds as emitted by one or more playback devices during a calibration sequence, perhaps by recording first samples while the microphone is in motion through a given environment and recording second samples while the microphone is stationary at one or more particular locations. The implementation may also include determining a first calibration for the one or more playback devices based on at least the first samples of the calibrations sounds and determining a second calibration for the one or more playback devices based on at least the second samples of the calibrations sounds. The implementation may further include applying at least one of (a) the first calibration or (b) the second calibration to playback by the one or more playback devices.

The present invention provides a method for processing music audio data, comprising the steps of providing input audio data representing a first piece of music containing a mixture of predetermined musical timbres, decomposing the input audio data to generate at least a first audio track representing a first musical timbre selected from the predetermined musical timbres, and a second audio track representing a second musical timbre selected from the predetermined musical timbres, applying a predetermined first audio effect to the first audio track, applying no audio effect or a predetermined second audio effect, which is different from the first audio effect, to the second audio track, and obtaining recombined audio data by recombining the first audio track with the second audio track.

An apparatus comprises receivers (101, 102) receiving microphone signals and a video signal capturing a scene. An image generator (113) generates an image signal comprising images generated from the video signal and a zoom determiner (115) determines a zoom property, such as a zoom factor, for the images. An audio component generator (107) generates a stereo signal component for a given mono audio signal to comprise the mono audio signal positioned at a position in a stereo image of the stereo signal component. A combiner (109) generates the stereo audio signal to include at least one stereo signal component and an output (111) generates the audiovisual signal to comprise the stereo audio signal and the image signal. In the approach the position is determined in dependence on the zoom property, and typically also dependent on a direction of arrival for the given mono audio signal. The approach may allow closer consistency between visual and auditory spatial cues.

An over-ear wearable device includes a housing with a first portion configured to be worn behind an ear of a user and a second portion configured to extend over the ear to position an output near an ear canal of the user. A memory is disposed within the housing and stores map data for an airport. A navigation sensor is disposed within the housing and configured to generate location data. A multi-button user input interface is coupled to the first portion. A processor is disposed within the housing and configured to responsive to selection of a first button of the multi-button user input interface, iterate through a list of destinations identified in the map data, responsive to selection of a second button of the multi-button user input interface, designate a target destination, and generate verbal navigation instructions based on the location data, the map data, and the target destination.

A computing apparatus with the proposed Method HRTF- for soundstage extension, upon which some inherent issues of the conventional Method HRTF such as the need for pre-measuring or simulating, highly individual dependence, small sweet spot, etc., have been effectively resolved. With the approaches that decoupling the individual-dependent segment from the entire HRTF chain, moving the intersection-point forward outside the ear, and simplifying the frequency response with relative-quantity approximations, the goal- "listening with one's own ear" is achieved with robustness improved and sweet spot enlarged. The present invention can be deployed in diverse scenarios: the stereo speakers; a foundational algorithmic module for advanced applications such as home theater systems, etc.; and an audio file in a particular encoding format can be also generated upon the invention that the soundstage can be extended readily through replaying with conventional stereo speakers.

Generative media content (e.g., generative audio) can be played back across multiple playback devices concurrently. A generative content group coordinator device can receive input parameters, which can include sensor data, media content, or other such input. The coordinator device can generate first and second generative media content streams, each of which can be transmitted to first and second playback devices, respectively. The first and second playback devices can play back the first and second streams of generative media content concurrently.

Various embodiments set forth techniques for positioning speakers within a venue. The techniques include generating, via a machine learning model, at least one of a two-dimensional (2D) representation or a three-dimensional (3D) representation of a venue based on one or more images of the venue. The techniques further include determining one or more parameters associated with one or more speakers to be placed within the venue based on the at least one of the 2D representation or the 3D representation.

Audio samples are obtained from a plurality of microphones in a conference room that includes a plurality of participants of an online communication session. A cross correlation is calculated between audio samples for each microphone pair of the plurality of microphones. For each participant, a distance between the participant and each microphone is estimated and, for each microphone pair, an expected delay between when microphones in a microphone pair receive audio from a participant is calculated based on the distance. For each participant, a score is computed based on the cross correlation for each microphone pair and the expected delay for each microphone pair and the participant that is speaking is identified based on the score computed for each participant.

This disclosure describes the classification (hereinafter referred to as "UNCLR classification") and crosstalk detection (hereinafter referred to as "XTALK detection") of irrelevant stereo content in an input stereo audio signal. This disclosure also describes stereo mode selection, such as automatic LRTD / DFT stereo mode selection. Furthermore, this disclosure uses the classification to select one of a first stereo mode and a second stereo mode for encoding a stereo audio signal including the left and right channels; to detect crosstalk in the stereo audio signal including the left and right channels in response to features extracted from the stereo audio signal including the left and right channels; or to classify irrelevant stereo content in the stereo audio signal including the left and right channels in response to features extracted from the stereo audio signal including the left and right channels.

This application relates to a method and processor for generating a calibrated audio value for a loudspeaker driver. The method includes: at a given time: receiving an input audio value of an input audio signal; generating a position value of a diaphragm of the loudspeaker driver, the position value indicating a predicted displacement of the diaphragm at or after receiving the input audio value; generating a velocity value of the diaphragm, the velocity value indicating a predicted velocity of the diaphragm at or after receiving the input audio value; using the input audio value, the position value, and the velocity value to generate a calibrated audio value; and sending the calibrated audio value to the loudspeaker driver to mitigate vibration of the loudspeaker driver.

An audio control system includes: an obtainer that obtains sound data indicating a sound picked up by a microphone disposed in a second space located in an area surrounding a first space used by a person; a storage in which sound source data is stored; and a controller that causes a loudspeaker disposed in the first space or the second space to reproduce an audio obtained from the sound source data more loudly in the second space than in the first space. The controller determines a loudness the sound indicated in the sound data obtained, and adjusts a sound volume of the audio outputted from the loudspeaker to be lower with an increase in the loudness of the sound determined.

In bidirectional communication between a plurality of wearable terminals, necessary sounds are transmitted to other wearable terminals. A method according to one embodiment of the present disclosure includes a step of bidirectionally communicating sound between a first wearable terminal including a first microphone, a first storage unit, and a first speaker, and a second wearable terminal including a second microphone, a second storage unit, and a second speaker. The first wearable terminal, in which the sound output by the second speaker or the sound from the direction of the head of the wearer of the second wearable terminal is collected by the first microphone, sets the sound from the direction of the head of the wearer of the first wearable terminal as a sound larger than the other sounds. A step of recording the data in the first storage unit; and a step in which the recorded sound is transmitted to the second wearable terminal.

Synchronized output of audio on a group of devices can comprise sending audio data from an audio distribution master device to one or more slave devices in the group. Scores can be assigned to respective audio playback devices, the scores being indicative of a performance level of the respective audio playback devices acting as a master device. The device with the highest score is designated as a candidate master device and one or more remaining devices are designated as a candidate slave(s). A throughput test is conducted with the highest scoring device acting as the candidate master device. The results of the throughput test are used to determine a master device for a group of devices. Latency of the throughput test can be reduced by using a prescribed time period for completion of the throughput test, and / or by selecting a first group configuration to pass the throughput test.

For synchronized media streaming in a device chain including first, second and third Bluetooth media devices (400) having a controller (425b) and time synchronization algorithm (422a or 424b), the first device formats a first media packet including received media data, second device address, and synchronization information including packet start time including a delay time and cumulative clock drift counted thus far. The second device receives the first packet and formats a second packet including the media data, third device address, updated synchronization information including the second packet's play start time, local clock time, clock drift, and a cumulative clock drift counted thus far. The second device transmits the second packet to the third device. The synchronization algorithms begin synchronized playing of the media data including the first device playing after the delay time, second BT device playing after the first packet's play start time, and the third device playing after the second packet's play start time.

System for a real-time radio receiver network, wherein the system comprises: a remote digital antenna digital receiver (“RDADR”) with a processor; a first remote digital antenna (“RDA”) comprising a processor, at least one receiver and at least one antenna; a second RDA with a processor, at least one receiver and at least one antenna, an audio source that is coupled to the transmitter; and an audio output device coupled with the RDADR, where the RDADR, the first RDA and the second RDA are coupled to each other via a digital bus, wherein the first RDA and the second RDA attempt to receive one or more digital signals from a transmitter, and if the one or more digital signals are received without error by the first RDA or the second RDA, the RDA that has received the one or more error-free digital signals transmits the one or more error-free digital signals via the digital bus to the RDADR, where the audio source generates one or more digital signals, and where the audio source is at least a microphone or a musical instrument.

A method for using an individualized endpoint location of a user to modify an audio signal includes determining a position of a user listening to a sound, based on an output from a GPS component. The method includes determining a distance of the user from a source of the sound. The method includes determining a corrected speed of sound when accounting for at least one environmental factor of an area associated with the source of the sound. The method includes determining a delay between the position and the source of the sound responsive to the determined position, distance, and corrected speed of sound. The method includes modifying an audio signal generated at the source of the sound, wherein modifying includes applying the determined delay to the audio signal.

One or more embodiments of the present disclosure provide a method for an audio playback device, an audio playback device, an audio playback system, and a computer-readable storage medium, where the method for a first audio playback device includes: connecting with an audio acquisition device and at least one second audio playback device; sending connection information of the first audio playback device with the audio acquisition device to the at least one second audio playback device for the second audio playback device to receive, based at least on the connection information, an audio acquisition signal sent by the audio acquisition device; receiving the audio acquisition signal sent by the audio acquisition device; and playing first audio corresponding to the audio acquisition signal synchronously with the at least one second audio playback device. The method for an audio playback device according to the present disclosure can reduce delay between sound propagated through air and sound propagated after being processed by an audio acquisition device and an audio playback device, thereby improving auditory experience of a user.

This invention is related to signal processing of music. The invention specifically concerns removal of unwanted artifacts from a music signal. The invention provides a method for attenuating resonances in a sound signal, which method uses subsampling of at least one sub-band of the original signal to save computing resources. As a coarse overview description, the method comprises at least steps in which an incoming signal block is filtered (120) into at least two different frequency sub-band signals, at least one of the sub-band signals is subsampled (121), a resonance analysis step is performed (125) for each sub-band signal, said step resulting in a filter response, a sub-band signal from each sub-band is convolved (130) with the corresponding filter response to produce a filtered sub-band signal, for each subsampled sub-band, the filtered sub-band signal is upsampled (140), and finally the upsampled sub-band signals of subsampled sub-bands and filtered sub-band signals of any non-subsampled sub-bands are combined (150) to form a result signal block. This method saves computing resources by performing analysis based on subsampled sub-band blocks.

Provided in the present application are a deep learning-based echo cancellation method, a device and a readable storage medium. A specific implementation solution comprises: acquiring a far-end microphone signal corresponding to a far-end room, and acquiring a near-end microphone signal corresponding to a near-end room; using the far-end microphone signal as a reference signal, acquiring a first compressed complex spectrum corresponding to the reference signal, and acquiring a second compressed complex spectrum corresponding to the near-end microphone signal; inputting the first compressed complex spectrum and the second compressed complex spectrum into a trained neural network model for echo cancellation processing, to output a near-end speech compressed complex spectrum; and performing an inverse short-time Fourier transform on the near-end speech compressed complex spectrum to obtain a pure near-end speech signal. In the implementation of the present application, the microphone signal of each far-end room serves as neural network reference information to restore the near-end pure speech signal, effectively suppressing echo components and thus implementing echo suppression of virtual spatial sound systems.

Some methods may involve receiving one or more audio signals, making an estimation of the direct sound level for each loudspeaker of the set of loudspeakers in a listening area, making an estimation of the loudness of each loudspeaker in the listening area and rendering one or more audio signals for reproduction via the set of loudspeakers. The rendering may involve determining, for each audio signal, a relative activation of each loudspeaker based, at least in part, on the estimation of the direct sound level for each loudspeaker in the listening area. A combined level of activations across loudspeakers may be based, at least in part, on the estimation of the loudness of each loudspeaker in the listening area.

Various implementations include a portable speaker configured to adjust an audio output based on a detected input connection. In a particular case, the portable speaker includes an enclosure housing at least one electro-acoustic transducer for providing an audio output, a processor coupled to the transducer, an audio input module coupled to the processor for receiving an audio input signal, and a battery configured to power the at least one transducer, the processor, and the audio input module, an input channel for receiving a wired audio input connection, and at least one wireless input channel for receiving audio input from a source device via a wireless connection, where the processor is configured to adjust the audio signal received from the wired audio input connection if the source device is already connected via the wireless connection.