1457results about "Signal processing" patented technology

A controller configured to control a sound producing module includes a volume controlling unit configured to determine a demodulation amplitude and a modulation amplitude corresponding to a target volume. The sound producing module comprises a driving circuit and an air-pulse generating device. The driving circuit generates a demodulation driving signal according to the demodulation amplitude and generates a modulation driving signal according to the modulation amplitude, so as to drive the air-pulse generating device. The air-pulse generating device produces sound via generating a plurality of air pulses at an ultrasonic pulse rate.

A mixing device is provided. The mixing device includes: an input unit, inputting sounds; multiple output units, outputting the sounds; a sound setting operator, setting a sound setting value that is a setting value relating to the sounds; a selection operator, selecting the output unit of a target for setting the sound setting value; a setting value acquisition means, acquiring the sound setting value set by the sound setting operator with respect to the output unit selected by the selection operator; and a mixing means, for each of the output units, mixing the sounds input at the input unit based on the sound setting value corresponding to the output unit and being the sound setting value acquired by the setting value acquisition means.

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.

A calibration method comprises adjusting an operating frequency of an air-pulse generating device, such that a sound pressure level (SPL) of the air-pulse generating device is within a specific range. The sound producing module comprises the air-pulse generating device configured to produce sound via generating a plurality of air pulses at a pulse rate corresponding to the operating frequency.

Disclosed in the present invention are an automatic delay calibration method for a smart speaker system, a device, and a storage medium. The method comprises: initializing a smart speaker system and performing environmental testing to preliminarily configure system parameters; exciting a target sound channel and playing a target test signal; locating a first test signal on the basis of a voice activity detection (VAD) algorithm, determining the start and end of the first test signal, and identifying a time difference of the first test signal to obtain a delay of main channels; locating a time of arrival of a second test signal on the basis of a fast peak search algorithm, and identifying a delay of a subwoofer; and performing delay calibration on the subwoofer and the main channels, and dynamically adjusting delay calibration parameters of the channels on the basis of a delay calibration amount, thereby achieving accurate time alignment between the subwoofer and the main channels. The present invention solves the problem of low-frequency signal delays that are difficult to process in conventional technologies, and particularly makes a breakthrough progress in the coordination of a subwoofer and other sound channels.

In aspects of adjusting audio output in a vehicle, a vehicle audio system implements an audio playback manager that detects a location of a first person in the vehicle, the first person located in proximity of a first speaker device configured for audio output. The audio playback manager also detects an additional location of a second person in the vehicle, the second person located in proximity of a second speaker device configured for the audio output. The audio playback manager determines a preference for the first person related to the audio output and an additional preference for the second person related to the audio output. Based on the preference of the first person or the additional preference of the second person, the audio playback manager adjusts a volume of at least one of the first speaker device or the second speaker device.

A zonal audio system for a vehicle may include a microcontroller unit (MCU), a non-volatile storage communicatively coupled with the MCU that stores audio files for different types of audio functions, and a digital signal processing (DSP) module communicatively coupled with the MCU. The DSP module may process multiple types of audio functions simultaneously through a layered architecture that consolidates and prioritizes different audio streams including emergency calls, acoustic vehicle alerts, chimes, and infotainment.

A computer-implemented sound calibration method, a system and a storage medium are provided. A test audio is played by a subwoofer and a main speaker, transmitted audio is collected by an audio collection device at a primary listening position, a sound generation delay and a volume parameter difference are analyzed, and the subwoofer is calibrated based on of a frequency response curve of a scenario, so as to solve the problem in the related art where the user experience is adversely affected due to low sound calibration accuracy between the subwoofer and the main speaker.

The invention relates to the technical field of audio driving, in particular to a driving circuit and method and electronic equipment. The method comprises the following steps: arranging two boosting units at the output end of the power amplifier, respectively receiving two paths of signals with opposite phases output by the power amplifier so as to realize that boosted signals output by the two boosting units are also signals with opposite phases, arranging one boosting unit to be connected with the positive electrode of the piezoelectric ceramic module, and arranging the other boosting unit to be connected with the negative electrode of the piezoelectric ceramic module; and the other boosting unit is connected with the negative electrode of the piezoelectric ceramic module, so that the driving voltage of the piezoelectric ceramic module is increased. The boost unit is simple in structure, occupies a small area of a circuit board, and can be effectively applied to miniaturized electronic equipment.

The invention provides a sound effect control method and electronic equipment, and relates to the technical field of terminals. The audios played by the electronic equipment have similar sound effects in different use states, so that the sound effect stability of the electronic equipment is ensured. The method is applied to the folding screen electronic equipment, and comprises the following steps: the electronic equipment plays a first audio with a first sound effect through a first loudspeaker and a second loudspeaker in a first use state, the first loudspeaker performs playing by adopting a first sound effect parameter, and the second loudspeaker performs playing by adopting a second sound effect parameter. And in response to user operation, switching to a second use state, playing the first audio with a second sound effect through the first loudspeaker and the second loudspeaker, the first loudspeaker playing with the third sound effect parameter, and the second loudspeaker playing with the fourth sound effect parameter. The first sound effect parameter is different from the third sound effect parameter, the deviation between the second sound effect and the first sound effect is smaller than a first threshold value, and the first use state and the second use state are related to the folding state of the electronic equipment.

The present disclosure relates to the technical field of audio control. Provided are a computer-implemented method for audio control, and an electronic device and a storage medium. The computer-implemented method for audio control comprises: on the basis of a specified measurement signal, performing acoustic effect testing on a target space, in order to obtain environmental response information, wherein the environmental response information comprises impulse response information and post-separation impulse response information; analyzing the environmental response information, in order to obtain acoustic characteristic information of the target space; on the basis of the acoustic characteristic information, performing compensation processing on audio to be processed, in order to obtain regulated audio; regulating gains of multiple channels corresponding to the regulated audio, in order to obtain re-regulated audio; and performing multi-channel collaborative optimization processing on the re-regulated audio, in order to obtain audio having a target effect. The method achieves precise compensation for spatial acoustic defects, resolves inter-channel crosstalk, improves the acoustic image localization and spatial perception, significantly improves the perceptual audio quality, and optimizes an output.

A first electrical signal is received from a first cardioid microphone capsule of a microphone assembly. The first cardioid microphone capsule is oriented to face in a first direction. A second electrical signal is received from a second cardioid microphone capsule of the microphone assembly. The second microphone capsule is oriented to face in a second direction that is opposite the first direction. A polar pattern is formed for the microphone assembly by converting the first electrical signal into a first audio signal and converting the second electrical signal into a second audio signal. A first weight is applied to the first audio signal and a second weight is applied to the second audio signal. The polar pattern is generated for the microphone assembly by summing the first audio signal and the second audio signal.

The invention discloses a sound power adjusting system based on vehicle-mounted battery state self-adaption, and relates to the technical field of automobile electronic control, and the system comprises a vehicle state sensing and data collection module which is used for analyzing a bus message to obtain a power parameter and a load state, and collecting a battery physical parameter; the decision and strategy control module is connected to the sensing module and is used for calculating a sound equipment allowable output power threshold value under the current working condition and generating a control instruction according to the power priority mapping table and a battery state power calculation model; the execution and audio processing module is used for adjusting the carrier frequency and gain of a power amplifier according to the instruction and executing audio frequency band processing; and the cross-domain isolation power supply protection module is used for providing isolation power supply and switching to the emergency energy storage element when the voltage drops. Through a load preloading mechanism and dynamic internal resistance evaluation, adaptive matching of the sound power and the vehicle battery state is realized, and the system is effectively prevented from resetting due to voltage drop.

A system can operate a speech-controlled device to perform user engagement detection (UED) processing to detect when speech represented in audio data is directed to the device. For example, the device may extract audio features from the audio data and process these audio features using a classifier to estimate an orientation of the user's head, which may be used as a proxy for user engagement. Thus, if the head orientation is within an engagement zone (which varies based on distance to the user), the device may determine that the user is engaged with the device and perform language processing on input speech. In contrast, if the head orientation is outside of the engagement zone, the device may determine that the user is not engaged and ignore the input speech. To enable additional functionality, the classifier may optionally output a coarse estimate of the head orientation along with the UED determination.

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.

Devices, systems, and methods are disclosed for customizing audio experiences associated with operation of a ridable device. An electronic device having a processor detects contextual information such as speed, acceleration, ambient noise level, environment type, or location of the ridable device. Sensor data from accelerometers, gyroscopes, microphones, cameras, or positioning systems may be processed to identify these conditions. Based on the detected context, the electronic device determines audio configuration parameters, including selection of audio content, adjustment of volume, blending of tracks, or modification of tempo or pitch. The determined parameters are applied to output audio through one or more speakers of the electronic device, the ridable device, or associated accessories. In some implementations, playback is synchronized with other riders using a common time reference, geo-fence triggers are employed to provide location-specific content, or warning sounds are generated when nearby objects are detected, enhancing both rider enjoyment and environmental awareness.

Systems can include an acoustic sensor array configured to receive acoustic signals, an illuminator configured to emit electromagnetic radiation, an electromagnetic imaging tool configured to receive electromagnetic radiation, a distance measuring tool, and a processor. The processor can illuminate the target scene via the illuminator, receive electromagnetic image data from the electromagnetic imaging tool representative of the illuminated scene, receive acoustic data from the acoustic sensor array, and receive distance information from the distance measuring tool. The processor can be further configured to generate acoustic image data of the scene based on the received acoustic data and received distance information and generate a display image comprising combined acoustic image data and electromagnetic image data. The processor can determine depths of various acoustic signals within a scene and generate a representation of the scene the shows the determined depths, including floorplan and volumetric representations.

Example embodiments relate to an apparatus, method and computer program for output of stereo or spatial audio. The method may for example comprise providing first data representing stereo or spatial audio, providing second data representing other audio and determining an orientation and / or configuration of an apparatus. The method may also comprise outputting the first data to selected first and second loudspeakers of the apparatus, comprising at least three loudspeakers based, at least in part, on the orientation and / or configuration of the apparatus; and outputting the second data to at least a third loudspeaker of the at least three loudspeakers other than the first and second loudspeakers.

Some systems include an acoustic sensor array configured to receive acoustic signals, an electromagnetic imaging tool configured to receive electromagnetic radiation, a user interface, a display, and a processor. The processor can receive electromagnetic data from the electromagnetic imaging tool and acoustic data from the acoustic sensor array. The processor can generate acoustic image data of the scene based on the received acoustic data, generate a display image comprising combined acoustic image data and electromagnetic image data, and present the display image on the display. The processor can receive an annotation input from the user interface and update the display image based on the received annotation input. The processor can be configured to determine one or more acoustic parameters associated with the received acoustic signal and determine a criticality associated with the acoustic signal. A user can annotated the display image with determined criticality information or other determined information.

Signal processing circuitry configured to receive an input signal and to output a processed output signal, wherein the signal processing circuitry is configured to: receive an indication of a temporal location of a transient in the input signal; and provide, in the processed output signal, a masking signal bridging the temporal location of the transient to mask the transient.

The embodiment of the invention provides a method executed by electronic equipment, the electronic equipment and a storage medium, and relates to the field of artificial intelligence. The method comprises the following steps: acquiring an input first audio signal; based on direction of arrival (DOA) information of a first audio signal, target direction information and spatial information of a physical microphone (RM) of the electronic equipment, obtaining the number and positions of virtual microphones (VM); a second audio signal is obtained based on the first audio signal and the number and location of VMs. Optionally, the method performed by the electronic device may be performed using an artificial intelligence model.

A device to perform location-based audio signal compensation includes a memory configured to store instructions and one or more processors. The one or more processors are configured to execute the instructions to receive an audio input signal corresponding to sound received from a second device, determine position data indicative of a position of the device, and generate, based on the audio input signal, a compensation filter to be applied to an audio playback signal prior to playout from the second device, to at least partially compensate for distortion associated with sound propagation from the second device to the position of the device.

A method performed by an audio system that includes a headset for playing back audio, the method includes receiving an audio signal having audio content, determining a characteristic of the audio content, determining a parameter of the headset, producing a filtered audio signal by applying an equalization (EQ) filter to the audio signal based on the characteristic of the audio content and the parameter of the headset, and driving one or more speakers of the headset with the filtered audio signal.

This application provides a sound-field control method and device, an electronic device and a computer-readable storage medium. The method includes: obtaining a target sound field parameter, calculating a control parameter of a sound-generating device according to the target sound field parameter, and controlling operation of the sound-generating device according to the control parameter.

Features described herein generally relate to detecting and correcting audible distortion due to speaker port occlusion. Particularly, an audio signal is output, an impedance curve for the audio signal is calculated, a change in a resonance frequency for the speaker based on an impedance curve model for the speaker and the impedance curve for the speaker is detected, and a gain of the amplifier is set based on the detected change in the resonance frequency for the speaker.

In one embodiment, a computer-implemented method includes receiving an input audio signal for playback by at least one speaker and at least one tactile transducer, receiving a control signal associated with a user-specified setting for the at least one tactile transducer, and generating a processed audio signal to drive the at least one tactile transducer, wherein the processed audio signal is adjusted based on the control signal independent of a speaker output signal driving the at least one speaker.

A method and electronic consumer device for creating an individual spatial listening zone for a user provided. The method includes receiving an input audio signal and processing it to generate first and second ultrasonic electric audio signals. The ultrasonic signals are provided to first and second ultrasonic transducers which convert them into directional ultrasonic acoustic signals. These signals are directed to a spatial listening zone where the user is located. Through non-linear interaction of the ultrasonic signals in air within the spatial listening zone, an audible sound signal is produced for the user that corresponds to the desired audio based on the input signal. The audible sound is substantially confined to the spatial listening zone and inaudible outside it. Various techniques for dynamically adjusting the spatial listening zone based on user position are disclosed. The system allows for creation of personalized audio experiences while minimizing disturbance to others nearby.

A communication system includes a plurality of full duplex transceiver assemblies each having an ON condition and an OFF condition, and configured to be worn by a different user, each of the plurality of full duplex transceiver assemblies having a housing and printed circuit board coupled to the housing, the printed circuit board including a transceiver having a microprocessor. Each microprocessor is configured to emit a different stream of controlling data when the plurality of full duplex transceiver assemblies are in the ON condition, thereby allowing each of the plurality of full duplex transceiver assemblies to communicate among a plurality of different logical channels. Embedded with each different stream of controlling data is a unique identification number for grouping each of the plurality of full duplex transceiver assemblies together.

The invention provides a method, a device and a system for audio signal mixing. The present implementation more particularly relates to mixing audio signals from a microphone array to generate beams by performing fixed beamforming, 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 a noise reduction unit (NRU) to reduce noise components of each audio beam. The system also includes logic to compute a signal characteristic of each de-noised audio beam to determine a de-noised audio beam including a speech component based on the signal characteristic. The logic also generates a gain for each audio beam based on the selection, wherein the gains are used in beam mixing. In some aspects, the NRU includes a neural network noise reduction unit.