40 results about "Programming" patented technology

The invention relates to the technical field of AI audio chips, and discloses an audio processing model quantification method and system, and the method comprises the steps: applying an incremental voltage pulse to a multi-layer memristor cross array, measuring a conductivity value, and generating a memristor quantification configuration table; programming the neuron units in the multi-layer memristor cross array to obtain quantized neuron units; calculating a synaptic weight matrix; the audio signal to be processed is input into the quantization neuron unit to execute pulse calculation, the audio features are obtained, and the audio recognition result is generated on the end-side processor, so that the problems of nonlinearity and element difference of the memristor are effectively solved, the robustness of the system is improved, and the optimal balance of precision and power consumption can be realized in different scenes.

A method performed by a programmed processor of an electronic device. The device obtains an audio signal, obtains, using one or more microphones, a microphone signal that includes audio of an environment in which the electronic device is located. The device determines a context of the device, and selects a volume compensation model from several models based on the determined context. The device processes the audio signal according to the selected volume compensation model and the microphone signal, and uses the processed audio signal to drive one or more speakers of the device.

The invention relates to the field of sensors, in particular to a pressure-tactile sensor based on a common electrode structure and application thereof, and the pressure-tactile sensor comprises a first electrode layer, an air dielectric layer, a conductive layer, a mixed pressure sensitive layer and a second electrode layer from top to bottom, wherein the air dielectric layer is formed at the middle position by arranging insulating cushion blocks at the edge between the first electrode layer and the conductive layer, and an output electrode layer is arranged between the insulating cushion block at one side and the conductive layer. The sensor can be used in wearable health equipment to realize health monitoring, can be integrated to a signal acquisition and processing system, and is combined with a graphical programming platform to realize accurate control of electronic piano key tones and volumes. Good application prospects and important technical development potential are shown in the fields of intelligent interaction interfaces, intelligent musical instruments, human-computer interaction and the like.

In at least one embodiment, an emulator system including at least one first audio controller is provided. The at least one first audio controller includes a primary field programmable gate array (FPGA), a universal audio network interface sub-board and a secondary FPGA. The primary FPGA is programmed to one of transmit and receive at least one audio signal in a vehicle. The universal audio network interface sub-board is operably coupled to the primary FPGA and is programmed to one or more of transmit or receive the at least one audio signal in accordance with one or more audio network interfaces. The secondary FPGA includes a first communication block that is programmed to enable the primary FPGA to one of transmit or receive the at least one audio signal in accordance with the one or more audio network interfaces to or from the universal audio network interface.

An audio power amplifier (1) defining a plurality of characteristics relating to the corresponding capabilities and functions of the audio amplifier (1) itself comprises a control unit (18) comprising instructions, where the instructions comprise setting parameters. The control unit (18) is programmed to: set a characteristic of a plurality of characteristics according to a set parameter value; digital information representing updated values of the plurality of setting parameters is received, and settings of the plurality of characteristics are updated according to the updated values.

Methods, apparatus, systems, and articles of manufacture are disclosed. An example system includes: interface circuitry; programmable circuitry; and instructions to program the programmable circuitry to: obtain audio of a media presentation; obtain ambient noise in an area associated with the media presentation; determine an intensity of a difference between the audio and the ambient noise; and determine an engagement level of an audience of the media presentation based on a duration the intensity satisfies a threshold value.

A hearing aid system is disclosed that automates selection or creation of hearing profiles using a dynamically customizable vivo adaptare audiogram. The hearing aid and a smart device communicate via a programming interface, where the smart device runs processor-executable instructions for a listening intelligence and frequency enhancement function. An artificial intelligence module evaluates ambient sound, compares it to stored hearing profiles specifying frequency adjustments, and either selects one or merges multiple profiles to form a new profile. The chosen profile is activated in real time, with the option to store it for later use. Available modes include “Test and Store,” capturing ambient sound to refine a profile, and real-time adaptation, which continuously applies incremental changes without repeated user input.

The present invention discloses a 3D printing system and method for a music box cylinder. The system part includes: a collection module, a programming module, and a printing module. The collection module is used to collect cylinder information and corresponding note information. The programming module is used to construct a printing model based on the cylinder information and note information. The printing module is used to realize 3D printing of different music box cylinders based on the printing model. The present invention can design the appearance and groove layout of the cylinder according to different needs and preferences, realizing the production of personalized music box cylinders. At the same time, different cylinders can be replaced according to needs to change the music repertoire of the music box, increasing the user's music selection. Compared with the traditional mechanical processing method for making cylinders, the 3D printing system of the present invention has a lower cost and a simpler and faster production process.

In at least one embodiment, an apparatus for generating audio in a listening room of a vehicle is provided. The apparatus includes a memory device and at least one audio controller. The at least one audio controller is programmed to receive an incoming audio signal comprising primary audio data from an audio source for playback, and to receive a first control signal indicative of a desired location for playback of the primary audio data. The at least one audio controller is further programmed to adjust reverberation of at least the incoming audio signal to play the primary audio data in a desired location of the vehicle and to receive a second control signal indicative of a desired natural sound to play in the listening room. The at least one audio controller is further programmed to play the desired natural sound along with primary audio data in the desired venue.

In at least one embodiment, an emulator system including at least one first audio controller is provided. The at least one first audio controller includes a primary field programmable gate array (FPGA), a universal audio network interface sub-board and a secondary FPGA. The primary FPGA is programmed to one of transmit and receive at least one audio signal in a vehicle. The universal audio network interface sub-board is operably coupled to the primary FPGA and is programmed to one or more of transmit or receive the at least one audio signal in accordance with one or more audio network interfaces. The secondary FPGA includes a first communication block that is programmed to enable the primary FPGA to one of transmit or receive the at least one audio signal in accordance with the one or more audio network interfaces to or from the universal audio network interface.

An application programming interface, API, and its implementing functions provide software code that can be incorporated into a simulated reality application and that determine a total calibration gain that is to be applied during run time of the application, to the audio signal of a particular digital audio asset that has been selected by the author of the simulated reality application. The total calibration gain includes at least a loudness correction gain that is determined based on a user-specified sound level parameter indicated through the API by the author. Other aspects are also described and claimed.

Computer-readable medium containing instructions that configure a computer to: present an application programming interface (API), wherein the API includes a user-specified sound level parameter; receive, via the API, one or more values ​​for the user-specified sound level parameter, which are assigned to a digital audio asset;and incorporating code into a simulated reality application that is currently being assembled, wherein the code determines a loudness correction gain for the digital audio asset by determining a difference between i) the one or more values ​​for the user-specified sound level parameter and ii) a parameter of the playback hardware device that represents a known or predefined sound level that can be produced by a playback hardware device on which the simulated reality application is running, wherein the loudness correction gain compensates for the difference, and the loudness correction gain is then to be applied to the digital audio asset during the runtime of the simulated reality application.

A system is programmed and operated to perform a method of indicating health, status, or condition of a monitored target. In certain implementations, the method involves: obtaining at least one metric associated with the monitored target; determining a level of audio degradation to be applied to a source digital media file that conveys default audio content, wherein the level of audio degradation is determined as a function of the obtained at least one metric; modifying the source digital media file in accordance with the determined level of audio degradation, resulting in modified audio data; and controlling playback of the modified audio data to serve as an audible indicator of the health, status, or condition of the monitored target.

In at least one embodiment, an audio system is provided. The audio system includes one or more microphones and at least one controller. The one or more microphones are positioned within a listening space to provide a captured audio input signal. The at least one controller is programmed to receive the captured audio input signal and to receive a first signal indicative of an audio level in the listening space. The at least one controller determines a gain amount based on the audio level and to adjusts the captured audio input signal.

The application discloses an audio downloading and playing method and device of an open-source Hongmeng system, a terminal and a medium. The audio downloading and playing method of the open-source Hongmeng system is executed by a commercial display multimedia terminal, and an audio downloading and playing framework based on an open-source Hongmeng system kernel is pre-implanted in the commercial display multimedia terminal, and the method comprises the following steps: based on the audio downloading and playing framework, obtaining audio file associated data to be sent sent by a digital signage server; wherein the audio downloading and playing framework comprises a JS layer, an application programming interface, a message management module, a transmission control protocol connection establishment module, a transmission control protocol socket module and an open-source Hongmeng system kernel which are sequentially connected; determining audio data to be displayed according to the audio downloading and playing framework and the audio file associated data to be sent, and performing audio display based on the audio data to be displayed. The technical scheme of the embodiment of the application can enable the commercial display multimedia terminal to perform audio playing based on the open-source Hongmeng system.

In at least one embodiment, an audio system for providing a fault recovery mechanism is provided. The system includes at least one digital signal processor (DSP) and at least one host processor. The at least one DSP is programmed to process an audio input signal to transmit an audio output signal to an amplifier for playing back the audio output signal in a listening environment and to transmit metering information indicative of audio attributes employed by the at least one DSP while processing the audio input signal. The at least one host processor is programmed to receive the metering information; and to control an amplifier to deactivate playing back the audio output signal in the listening environment at least based on the at least one host processor is not receiving the metering information within a predetermined time frame.

An application programming interface, API, and its implementing functions provide software code that can be incorporated into a simulated reality application and that determine a total calibration gain that is to be applied during run time of the application, to the audio signal of a particular digital audio asset that has been selected by the author of the simulated reality application. The total calibration gain includes at least a loudness correction gain that is determined based on a user-specified sound level parameter indicated through the API by the author. Other aspects are also described and claimed.

A method performed by a programmed processor, the method including receiving an input noise signal, generating, using a machine learning model that has an input based on the input noise signal, a mono audio signal that includes a sound and a spatial parameter for the mono audio signal, and generating spatial audio data by spatially encoding the mono audio signal according to the spatial parameter.

In at least one embodiment, a simulator system including at least one first audio controller is provided. The at least one first audio controller includes a main field-programmable gate array (FPGA), a universal audio network interface (UNIC) daughterboard, and an auxiliary FPGA. The main FPGA is programmed to transmit and receive at least one audio signal in a vehicle. The UNIC daughterboard is operatively coupled to the main FPGA and programmed to transmit or receive at least one audio signal via one or more audio network interfaces. The auxiliary FPGA includes a first communication block configured to enable the main FPGA to transmit or receive at least one audio signal to or from the UNIC daughterboard via one or more audio network interfaces.

Methods, apparatus, systems, and articles of manufacture are disclosed. An example system includes: interface circuitry; programmable circuitry; and instructions to program the programmable circuitry to: obtain audio of a media presentation; obtain ambient noise in an area associated with the media presentation; determine an intensity of a difference between the audio and the ambient noise; and determine an engagement level of an audience of the media presentation based on a duration the intensity satisfies a threshold value.

A computing device includes a display; and at least one processor programmed to transmit a multimedia signal including one or more of audio and video data for output via one or more of a loudspeaker and the display; receive a first user input via on a first interface of the display, the first user input being indicative of a user selection to create personalized content including a start position and an end position corresponding to a portion of the multimedia signal; present at least one attribute of the personalized content on a second interface for selection by a user; and associate the at least one attribute to the personalized content of the multimedia signal to provide a desired attribute in response to a second user input being received at the second interface, the second user input being indicative of selecting at the at least one attribute.

A hearing aid system is disclosed that automates selection or creation of hearing profiles using a dynamically customizable vivo adaptare audiogram. The hearing aid and a smart device communicate via a programming interface, where the smart device runs processor-executable instructions for a listening intelligence and frequency enhancement function. An artificial intelligence module evaluates ambient sound, compares it to stored hearing profiles specifying frequency adjustments, and either selects one or merges multiple profiles to form a new profile. The chosen profile is activated in real time, with the option to store it for later use. Available modes include “Test and Store,” capturing ambient sound to refine a profile, and real-time adaptation, which continuously applies incremental changes without repeated user input.

A method and system that includes at least one processor, the at least one computer in communication with at least one data storage unit, the at least one computer programmed and / or configured to: generate a plurality of dance sequences; associate or cause the association of each of the plurality of dance sequences with at least one dance corpus classification; generate, for each of the plurality of users, a user profile; receive, from each of the plurality of users, at least one preference data; associate or causing the association of each of the user's profile with at least one preference data; and generate a recommendation of a dance sequence for each of the plurality of users.

The present disclosure relates to an audio processing unit, a method performed by the audio processing unit, and a storage medium. A device and method for generating an encoded audio bitstream including substream structure metadata (SSM) and / or program information metadata (PIM) and audio data in a bitstream. Other aspects are a device and method for decoding such a bitstream, and an audio processing unit (e.g., an encoder, decoder, or post-processor) configured (e.g., programmed) to perform the method or including a buffer memory storing at least one frame of an audio bitstream generated according to any embodiment of the method.

A kind of hearing instead of vision's guide blind method relates to electronic information field.The image of about 3 meters in front is collected by camera, and is sent into image to music module simultaneously to carry out image to music conversion, first, the image RGB space is converted to HSV space, and the value of each pixel point H, S, V is obtained.The value of H, S, V is mapped to the parameter of midi format file of music, there are six mapping modes, and the best mapping mode is that H value is given to the pitch parameter of midi file, S value is given to the volume parameter of midi file, and V value is given to the delta-time parameter of midi file.The midi parameter construction is carried out by the obtained parameter value of midi file, and then the midi music file is obtained.The H, S, V value is matched with volume, pitch and delta-time parameter after mapping, the converted music is obtained, the human ear listens to this sound, judges the existence condition of the obstacle in front, and the purpose of guide blind is achieved.The hearing instead of vision algorithm is realized by programming, and cheap and effective assistance is provided for blind people.

In at least one embodiment, an audio system is provided. The audio system includes one or more microphones and at least one controller. The one or more microphones are positioned within a listening space to provide a captured audio input signal. The at least one controller is programmed to receive the captured audio input signal and receive a first signal indicative of an audio level in the listening space. The at least one controller determines an amount of gain based on the audio level and adjusts the captured audio input signal.

Methods, systems, articles of manufacture, apparatus, and methods for generating multimodal flow and audio outputs are disclosed. An example apparatus includes interface circuitry, machine-readable instructions, and at least one processor circuit programmed by the machine-readable instructions to train an unsupervised image model to generate flow tensors based on a reference frame and a driver frame, wherein the flow tensors represent at least one of rotation information or translation information. The example apparatus also includes at least one processor circuit programmed by the machine-readable instructions to train a denoising diffusion probability model (DDPM) based on (a) the flow tensors, (b) audio distributions, and (c) prompt signals, wherein the trained DDPM temporally aligns the flow tensors with the audio distributions.

There is disclosed a network-based public address (PA) receiver. The disclosed network-based PA receiver includes: a system-on-chip (SoC) including a processing system (PS) and programmable logic (PL); wherein the PS includes a processor that executes a software program that outputs audio data encoded in a particular audio data representation format based on first PA audio data received via a network; wherein the PL is programmed to provide first audio interface logic and second audio interface logic; wherein the first audio interface logic organizes the encoded audio data into first formatted audio data according to a first audio interface format; and wherein the second audio interface logic organizes an audio data payload of a particular transport protocol, extracted from second PA audio data received via the network, into second formatted audio data according to a second audio interface format.

The invention discloses a digital musical instrument audio-visual interaction system, which belongs to the technical field of digital musical instruments and is characterized in that a player can continue to use a familiar playing mode and lower the learning threshold by providing a playing interaction terminal for a user, and then playing data of the playing interaction terminal is transmitted to data processing equipment through routing equipment; the method comprises the following steps of: acquiring audio-visual interaction data by adopting a preset data mapping rule on the basis of the playing data of the playing interaction terminal, so that a user can easily adjust control parameters without programming, the use threshold is reduced, and finally, the audio-visual interaction data is used as the basis, so that the user experience is improved. An audio-visual interaction function is provided for a user, visual feedback and sound synthesis are highly integrated, and more immersive experience is provided for a player.