Patent classifications
H04R29/002
Operating method of smart audio system
There is provided a smart audio system including multiple audio devices and a central server. The central server confirms a model of every audio device and a position thereof in an operation area in a scan mode. The central server confirms a user position or a user state to accordingly control output power of a speaker of each of the multiple audio devices in an operation mode.
SYSTEMS AND METHODS TO ADJUST LOUDNESS OF CONNECTED AND MEDIA SOURCE DEVICES BASED ON CONTEXT
Systems and methods are disclosed for controlling one or more devices based on measured sound levels. A device management system may access managed devices on a network, determine if any managed devices are generating sound, identify the sound-generating device, access a loudness policy associated with the sound-generating device, select a sound measuring device near the sound-generating device, receive from the sound measuring device a measured sound level of the sound-generating device, determine if the measured sound level exceeds a threshold from the loudness policy, and issue a command to the sound-generating device to reduce the volume. The sound measuring device may be selected because it is close to or in the same room as the sound-generating device. A management system may set and store a sound level limit as a loudness policy for a device, room, building, community, or more.
Apparatus and method for speaker tuning and automatic digital signal processing configuration
A loudspeaker system including a loudspeaker array, an audio controller, and one or more insertable columns are provided. The loudspeaker array is configured to playback an audio output in a listening environment. The audio controller is configured to provide the audio output to the loudspeaker array. The one or more insertable columns is positioned between the audio controller and the loudspeaker to adjust a height of the loudspeaker array relative to the audio controller.
Condition-invariant feature extraction network
To generate substantially condition-invariant and speaker-discriminative features, embodiments are associated with a feature extractor capable of extracting features from speech frames based on first parameters, a speaker classifier capable of identifying a speaker based on the features and on second parameters, and a condition classifier capable of identifying a noise condition based on the features and on third parameters. The first parameters of the feature extractor and the second parameters of the speaker classifier are trained to minimize a speaker classification loss, the first parameters of the feature extractor are further trained to maximize a condition classification loss, and the third parameters of the condition classifier are trained to minimize the condition classification loss.
Inter-channel level difference based acoustic tap detection
A system configured to detect a tap event on a surface of a device only using microphone audio data. For example, instead of using a physical sensor to detect the tap event, the device may detect a tap event in proximity to a microphone based on a power level difference between two or more microphones. When a power ratio exceeds a threshold, the device may detect a tap event and perform an action. For example, the device may output an alarm and use a detected tap event as an input to delay or end the alarm. In some examples, the device may detect a tap event using a plurality of microphones. Additionally or alternatively, the device may distinguish between multiple tap events based on a location of the tap event, enabling the device to perform two separate actions depending on the location.
Integrated loudspeaker and control device
An example method of operation may include receiving audio data at one or more microphones disposed in a corresponding plurality of network devices, identifying amplitude values of the audio data at each of the plurality of network devices, and each of the amplitude values identified at each of the plurality of network devices are different from each other of the amplitude values, determining at each of the plurality of network devices a location of the audio data based on a direction and amplitude of the received audio data, modifying the audio data for output via a plurality of loudspeakers disposed in each of the plurality of network device, and outputting, via the plurality of loudspeakers, the modified audio data, and each loudspeaker outputs different versions of the modified audio data.
Speaker device for a motor vehicle seat
A speaker device for a vehicle seat comprises at least one loudspeaker, a wireless communication module operatively connected to the at least one loudspeaker, and an attachment support for fixing the speaker device to the vehicle seat.
ACOUSTIC PROCESSING DEVICE, METHOD, AND PROGRAM
The present technology relates to an acoustic processing device, method, and program capable of performing audio replaying with higher sound quality. An acoustic processing device includes: a first rendering processing unit that performs rendering processing on the basis of an audio signal and generates a first output audio signal for outputting sound from a plurality of first speakers; and a second rendering processing unit that performs rendering processing on the basis of an audio signal and generates a second output audio signal for outputting sound from a plurality of second speakers having a different replaying band from that of the first speakers. The present technology can be applied to an audio replaying system.
Audio Settings Based On Environment
Techniques described herein may involve audio settings based on an environment. An example implementation may involve a playback device playing back first audio content and during playback of at least a portion of the first audio content, detecting, via the at least one microphone, an audio signal. At least a portion of the detected audio signal may include a reflection of the first audio content played back by the playback device. The playback device may determine an equalization setting based on at least the determined one or more reflection characteristics and apply the determined equalization setting during playback of second audio content.
AUDIO PLAYING METHOD AND DEVICE, STORAGE MEDIUM, AND MOBILE TERMINAL
The present application provides an audio playing method, including: when an audio signal to be played is detected, storing the audio signal in a preset buffer while transmitting the audio signal to the auxiliary speaker to enable the auxiliary speaker to play the audio signal; initiating a timer to start timing in a process of storing the audio signal; and when a timing duration for which the timing lasts reaches a preset delay duration, obtaining the audio signal from the preset buffer and transmitting the audio signal to the main speaker to enable the main speaker to play the audio signal.