1078results 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.

Examples described herein relate to transitioning a playback session between portable playback devices such as “smart” headphones, earbuds, and handheld speakers with playback devices of a zone-based media playback system. Exemplary techniques facilitate continuity of playback when transitioning between locations (e.g., from at home to on-the-go or vice versa) or between listening paradigms (e.g., personal or out-loud playback of audio content). An example implementation includes detecting a swap trigger, determining the source playback device(s) and target playback device(s), and performing a playback session swap between the source playback device(s) and target playback device(s).

Turntable control systems are disclosed. A playback device can include an analog audio source component such as a turntable. The playback device is configured to cause playback of first audio based on analog audio signals generated by the analog audio source component. Based on an indication that analog audio signals are no longer being generated, the playback device requests, via a network interface, audio content from one or more remote computing devices. The request can include identifying related content based on the first audio. The playback device receives, via the network interface, second audio content from, and causes playback of second audio based on the second audio content.

An audio session management method for an audio environment having multiple audio devices may involve receiving, from a first device implementing a first application and by a device implementing an audio session manager, a first route initiation request to initiate a first route for a first audio session. The first route initiation request may indicate a first audio source and a first audio environment destination. The first audio environment destination may correspond with at least a first person in the audio environment, but in some instances will not indicate an audio device. The method may involve establishing a first route corresponding to the first route initiation request. Establishing the first route may involve determining a first location of at least the first person in the audio environment, determining at least one audio device for a first stage of the first audio session and initiating or scheduling the first audio session.

Turntable control systems are disclosed. A playback device can include an analog audio source component such as a turntable. The playback device is configured to cause playback of first audio based on analog audio signals generated by the analog audio source component. Based on an indication that analog audio signals are no longer being generated, the playback device requests, via a network interface, audio content from one or more remote computing devices. The request can include identifying related content based on the first audio. The playback device receives, via the network interface, second audio content from, and causes playback of second audio based on the second audio content.

The invention provides a control method and device of a vehicle-mounted external loudspeaker and a vehicle-mounted external loudspeaker system. The control method comprises the following steps that in response to starting of a vehicle-mounted loudspeaker box mode, the relative position relation between a user and a vehicle is obtained at a preset time interval, and the vehicle is provided with a plurality of vehicle-mounted external loudspeakers making sounds outwards; determining a target loudspeaker and a corresponding sound channel from the plurality of vehicle-mounted external loudspeakers at least based on the relative position relationship; and controlling each target loudspeaker to play the audio data of the corresponding sound channel. The invention provides a mode of taking the vehicle as a large mobile sound box capable of providing a surround sound field, and the provided surround sound field can be adjusted according to the relative position relationship between the user and the vehicle, so that the hearing effect of the user when using the vehicle-mounted sound box is improved, the application scene of the vehicle is expanded, and the user experience is improved.

Example techniques relate to primary-ambient playback of surround audio by audio playback devices. Example playback devices described herein may include multiple speakers, such as a forward-firing audio transducer and side-firing audio transducers. Using example techniques, such playback devices may perform primary-ambient decomposition to separate surround channels into primary′ and ambient channels. The playback devices play back the ambient channel(s) via the side-firing transducers and the primary channel via the forward-firing transducer.

To provide a compact voice-activated electronic device with speakers, electronic components and a dual-purpose waveguide / heatsink structure that directs sound from the speakers to an outside of the device and to dissipate heat from the electronic components to the outside of the device.SOLUTION: A compact voice-activated electronic device 404 comprises: a speaker assembly 436; a bottom housing 408 on which the speaker assembly is mounted; an upper housing 406 serving as a speaker grill; a printed circuit board 432 on which electronic components are mounted; and an acoustic waveguide / heatsink 434 configured to direct sound from the speaker assembly to an outside of the voice-activated electronic device and to dissipate heat generated in the electronic components to the outside of the device.SELECTED DRAWING: Figure 4E

A playback device comprising a first spectrum radio for communication via a first wireless spectrum and a second spectrum radio for communication via a second wireless spectrum operates in a first mode in which the first spectrum radio is enabled and the second spectrum radio is disabled. While operating in the first mode, the playback device receives a command to join a synchrony group with a display device. Based on the command, the playback device begins to operate in a second mode in which the second spectrum radio is enabled. The playback device receives audio content for playback, wherein the audio content corresponds to video content being displayed at a display device. Using the second spectrum radio, the playback device coordinates with the display device for playing back the audio content in synchrony with the video content.

Provided herein are system, method, and computer program product embodiments for providing an interactive visual representation of loudspeaker sound coverage in a venue. An embodiment enables temporal, spectral, and spatial audio metering from digital audio to venue acoustics. This technology can visualize audio capabilities of one or more loudspeaker arrays using light to provide visualization grouping of large audio channel outputs. This can allow spatial information to be visually represented by mapping directional sound to light beams.

A system, method, and computer-readable medium for performing a smart audio operation. The smart audio operation includes detecting a plurality of audio devices available for use by an information handling system; monitoring information handling system audio related information; analyzing the information handling system audio related information using a trained smart audio artificial intelligence model; and, automatically connecting the information handling system to a particular audio device of the plurality of audio devices based upon the information handling system audio related information.

Images of an actual rendering environment are acquired through image sensors operating in conjunction with a media consumption system. The acquired images of the actual rendering environment are used to predict audio characteristics of objects present in the actual rendering environment. Spatial audio rendered, to a user in the actual rendering environment, by audio speakers operating in conjunction with the media consumption system is adjusted or modified based at least in part on the audio characteristics of the objects present in the actual rendering environment.

A system and method to adjust power levels between interconnected loudspeaker assemblies is described herein, the system comprising: a primary loudspeaker assembly, and at least one secondary loudspeaker assembly interconnected with the primary loudspeaker assembly and adapted to receive an audio signal from the primary loudspeaker assembly; an interconnection verification circuit located in the primary loudspeaker assembly; and a controller located in the primary loudspeaker assembly adapted to receive the status signal from the interconnection verification circuit, wherein the controller comprises—at least one processor communicatively coupled to the interconnection verification circuit; a memory operatively connected to the at least one processor, wherein the memory stores computer-executable instructions that, when executed by the at least one processor, cause the at least one processor to execute a method that comprises: generating an interconnection test signal by the interconnection verification circuit, wherein the interconnection test signal determines if the primary loudspeaker assembly is properly connected to the at least one secondary loudspeaker assembly; generating a connected status signal if the interconnection test signal determines that the primary loudspeaker assembly is properly connected to the at least one secondary loudspeaker assembly; receiving the connected status signal by the controller located in the primary loudspeaker assembly; and generating a first set of command signals by the controller to split power between the primary loudspeaker assembly and the at least one secondary loudspeaker assembly.

An electronic apparatus, may include: a communication interface; at least one processor; and a memory configured to store instructions, when executed by the at least one processor, cause the electronic apparatus to: based on a user input for a sound play being received, identify a target device capable of performing the sound play; identify a power state of the target device; based on the power state of the target device being identified as an off state, transmit a first wakeup signal for changing a power state of a speaker in the target device to an on state, to the target device through the communication interface; and based on a first response signal corresponding to the first wakeup signal being received, transmit a first control command for the sound play to the target device through the communication interface.

The invention provides a howling / feedback suppression method and device in a sound reinforcement scene, a medium and a product. The method comprises the following steps: acquiring a microphone signal of a current frame; frequency domain self-adaptive filtering processing is carried out on the microphone to be processed, and a predicted target signal is output to the loudspeaker, and the frequency domain self-adaptive filtering processing comprises the following steps: multiplying the estimated acoustic transfer function of the previous frame by the loudspeaker signal of the previous frame in the frequency domain to obtain a feedback estimation signal, and then subtracting the feedback estimation signal from the microphone signal to obtain a predicted target signal; a residual signal is obtained; the residual signal is used as a prediction target signal to be output to a loudspeaker after being subjected to inverse transformation, meanwhile, iteration is carried out on an estimated acoustic transfer function of a previous frame, an estimated acoustic transfer function of a current frame is obtained, and the step length of iteration is controlled through a recurrent neural network model. According to the method, the accuracy of acoustic feedback estimation in various actual scenes can be improved, the distortion degree in a sound reinforcement scene is improved, and the maximum gain is improved.

An audio conference system for automatically forming a single combined physical microphone array aperture across associated and / or disassociated ad-hoc microphone elements in a shared 3D space is provided. The audio conference system includes a plurality of microphone / speaker units, each including at least one microphone and / or at least one speaker and a system processor communicating with the microphone / speaker units. The system processor instructs the microphone / speaker units to transmit unique calibration signals sequentially or simultaneously and to calculate time difference of arrival (TDOA) between the microphone / speaker units. A physical array structure of the microphone / speaker units is obtained based on TDOA between the microphone / speaker units, and a consolidated target coverage zone common to the microphone / speaker units is generated based on the physical array structure.

Methods, apparatus, systems, and articles of manufacture are disclosed to generate spatial audio based on computer vision. An example apparatus includes at least one memory, instructions in the apparatus, and processor circuitry to execute the instructions to determine a position of an audio source based on an image generated via a camera, and apply an audio spatialization filter to an audio signal generated by a microphone based on the position of the audio source.

A method is provided that is performed at a system including multiple speech collectors to collect speech from a talker to produce corresponding ones of multiple audio signals that each convey speech energy: for each audio signal: separating high-frequency speech energy from low-frequency speech energy; and determining a first energy level of the high-frequency speech energy; and determining a preferred audio signal among the multiple audio signals for subsequent processing at least based on the first energy level of each audio signal.

An integrated audio microphone system simultaneously provides in-room amplification and remote communication during a collaboration session with one or more local participants in a shared space and one or more remote participants. The integrated audio microphone system includes at least one microphone array and speaker system located in the shared space, at least one microphone system for amplification located in the shared space, an audio processor connected to the microphone system and the microphone array and speaker system, and a computer connected to the audio processor. The audio processor is configured to generate a plurality of virtual microphone bubbles and to amplify received audio signals from the microphone system for in-room audio amplification. The amplified audio signals are output to speakers of the microphone array and speaker system. Audio signals from the microphone system and the arrays of microphones are output to the computer to send the audio signals to one or more remote participants.

Methods and apparatus for identifying a music service based on a user command. A content type is identified from a received user command and a music service is selected that supports the content type. A selected music service can then transmit audio content associated with the content type for playback.

A conferencing system includes a plurality of microphones and an audio processing system that performs blind source separation operations on audio signals to identify different audio sources. The system processes the separated audio sources to identify or classify the sources and generates an output stream including the source separated content.

Embodiments provide for a system and method that address challenges associated with audio routing of an incoming call between two devices operatively coupled via a short-range wireless connection. Ultrasonic watermark signals are generated and sent between the two devices. Detection and non-detection of the ultrasonic watermark signals controls the audio routing. The ultrasonic watermark signals can be modified based on a predetermined operational characteristic known only to the two devices for increased robustness to the system.

Embodiments described herein provide for detecting presence of an object in proximity to a playback device and responsively performing one or more operations. In an example implementation, a playback device detects, via a proximity detector of the playback device, presence of an object in proximity to the playback device. In response to detecting the presence of the object, the playback device performs one or more operations.

Multichannel audio playback devices and associated systems and methods are disclosed herein. In some examples, a first playback device is configured to receive a source stream of audio content comprising left, right and center input channels. In a first mode, the first playback device is configured to play back audio via a plurality of transducers based on the left, right, and center input channels. In a second mode, in which the first playback device is bonded to second and third playback devices, the first playback device is configured to (i) play back audio via the plurality of transducers based on at least the center input channel, (ii) cause audio to be played via the second playback device based on at least the right input channel, and (iii) cause audio to be played via the third playback device based on at least the left input channel.

This disclosure provides methods, devices, and systems for audio signal mixing. The present implementations more specifically relate to mixing audio signals from a microphone array by performing fixed beamforming to generate beams, reducing noise on the beams, and mixing the beams to generate a final audio signal for playback. In some aspects, an audio mixing system includes a fixed beamformer to generate beams from audio signals from a microphone array and noise reduction units (NRUs) to reduce a noise component of each audio beam. The system also includes logic to calculate a signal characteristic of each reduced noise audio beam to determine, based on the signal characteristics, the reduced noise audio beams that include a speech component. The logic also generates a gain for each audio beam based on the selection, with the gains used in beam mixing. In some aspects, the NRU includes a neural network noise reduction unit.

A multi-room audio synchronous playback method, a system, a device and a storage medium are provided. The multi-room audio synchronous playback method includes: creating an ad-hoc network using a plurality of audio devices supporting a target wireless communication protocol via point-to-point connection, the ad-hoc network covering a plurality of rooms, the target wireless communication protocol operating at a frequency band of 5 GHz to 7 GHz; dynamically analyzing link quality to select an optimum transmission path, and dynamically optimizing a network topology of the ad-hoc network via multi-hop relay; transmitting target audio data between the audio devices of the ad-hoc network based on the optimum transmission path, and performing real-time interaction and dynamic compensation based on a synchronization signal to obtain a synchronization parameter; and controlling each audio device of the ad-hoc network based on the synchronization parameter to synchronously play the target audio data.

A method for adaptive additive sound includes receiving ambient sound data corresponding to ambient sound in a first zone acquired by a microphone in the first zone, analyzing the ambient sound data from the first zone, generating audio signal data for the second zone based at least in part on the ambient sound data from the first zone, and transmitting the audio signal data for the second zone to a speaker in the second zone. The first zone is separate from the second zone within a space.