61 results about "Programming" patented technology

An audio system is provided that includes a loudspeaker, at least one microphone, and at least one controller. The loudspeaker transmits an audio signal into a listening environment defined by at least one wall in a room. The at least one microphone is positioned on the loudspeaker and is configured to capture a reverberated audio signal including a plurality of reverberations and a plurality of peaks. The reverberated audio signal is indicative of the audio signal being reflected from the at least one wall. The at least one controller is programmed to apply a confidence score to the plurality of peaks to obtain a maximum score which is indicative of a maximum peak of the audio signal that is reflected from the at least one wall and determine a distance between the loudspeaker and the wall based at least on the maximum score.

A system for real-time monitoring of user-generated audio content for audio anomaly and a related method are disclosed. In some embodiments, the system is programmed to receive, in real time, audio data generated by a first mobile device, such as a smartphone. The system is programed to determine, in real time, whether an audio anomaly has occurred from the audio data. The system is programmed to cause, in real time, a presentation of an alert to a second mobile device, which could be the same smartphone, in response to detecting an occurrence of audio anomaly.

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 present invention discloses a graphical programming system and a graphical programming method for scientific education. The method comprises: an embedded device downloads a program corresponding to a graphical code in a code area of ​​the graphical programming platform from a graphical programming platform, wherein the program records the duration of an audio file to be played; the embedded device sets the duration of a timer based on the duration of the audio file, and at the same time, instructs the graphical programming platform to play the audio to be played through a serial port, wherein the duration of the timer is equal to the duration in the audio file; when the timer reaches the set duration, the embedded device executes the related operation indicated by the second programming instruction block after the first programming instruction block or the embedded device instructs the graphical programming platform to execute the related operation indicated by the second programming instruction block after the first programming instruction block. The present invention solves the technical problem of poor interaction between the graphical programming platform and the embedded device.

Techniques for facilitating voice based dictation of programming code within a context of an IDE are disclosed. Programming code is fed to a text-to-speech (TTS) model. The TTS model generates an audio file associated with the code. The audio file is then fed to a speech-to-text (STT) model. The STT model generates a transcription file associated with the audio file. Each respective line of code included in the programming code is mapped to a corresponding line of code included in the transcription file, resulting in generation of a list of phrase pairings. These phrase pairings represent relationships between actual code and how that actual code sounds if read out loud. An LLM then ingests the list of phrase pairings. The LLM identifies correlations between programming vocabulary that has specific meaning within the context of the IDE and how that programming vocabulary sounds if read out loud.

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.

The present invention claims protection for an audio noise reduction accelerator system and method based on the expansion of RISC v custom instruction set, which belongs to the field of integrated circuit technology, and mainly includes: E203_CORE, NICE_CORE, NICE_Interface, E203_SOC, audio codec WM8731 module, and audio noise reduction FxLMS algorithm. Among them, E203_CORE is connected to NICE_CORE through NICE_Interface, E203_CORE, NICE_CORE and related peripheral ports together form E203_SOC, E203_SOC is connected to the audio codec WM8731 module, and the audio noise reduction FxLMS algorithm is downloaded to the RISC v processor core through software programming for operation. The innovation point is that compared with the processor of ARM instruction set architecture, the processor using RISC v custom instruction set can accelerate the specific operation part of the audio noise reduction FxLMS algorithm; the present invention can further optimize the problems of area, power consumption, granularity, etc., while improving the flexibility and feasibility of the algorithm.

The invention relates to a stability test method and system for Bluetooth audio playing of an Android vehicle-mounted system, and the method comprises the steps: S1, generating a single-frequency sound file through audio production software, and importing the single-frequency sound file into a smart phone; s2, programming to realize an audio feature recognition algorithm; s3, compiling a test script, defining information such as test times, and testing; s4, the vehicle-mounted system and the mobile phone set the volume used for calibrating the threshold value, the specific main control device module collects the sound played by the vehicle-mounted loudspeaker, the picture of the vehicle-mounted central control host and the picture of the intelligent mobile phone in real time, and whether the actually played audio frequency is consistent with the audio information displayed by the vehicle-mounted central control host collected by the picture or not is judged; and S5, recording a test result, executing the next test script, and repeating until the set test times. According to the invention, an audio feature recognition algorithm is adopted to calibrate a sound detection threshold value, and sound data generated by a large number of audio playing operations of the vehicle-mounted Bluetooth music system are automatically detected through the test script, so that the test stability is ensured.

A lighting device for a musical instrument includes an LED configured to attach to the musical instrument, a microphone, a communication interface configured to communicate with a mobile device, and a controller. The controller is selectable between a plurality of operating modes including at least one tone-responsive mode and at least one performance mode. In the tone-responsive mode, the controller receives an ambient audio signal from the microphone, detects a tone of the musical instrument based on the ambient audio signal, and dynamically controls a lighting characteristic output by the LED based on the detected tone. In the performance mode, the controller wirelessly receives a set cuing signal from the communication interface for changing between a predefined group of sets, and controls the LED during each of the predefined group of sets according to respective preprogrammed lighting patterns for each of the predefined group of sets.

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.

Techniques are disclosed for facilitating voice-based dictation of programming code within the context of an IDE. The programming code is fed to a text-to-speech (TTS) model. The TTS model generates an audio file associated with the code. The audio file is then fed to a voice-to-text (STT) model. The STT model generates a transcription file associated with the audio file. Each respective code line included in the programming code is mapped to a corresponding code line included in the transcription file, thereby causing generation of a phrase pair list. The phrase pairs represent a relationship between the actual code and how the actual code listens to when read out by loud sound. The LLM then ingests a list of phrase pairs. The LLM identifies a correlation between a programming vocabulary having a particular meaning within the context of the IDE and how the programming vocabulary listens to when read out by loud sound.

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.

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

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.

An interactive doll assembly for entertaining and educating a child includes a doll and a housing. A panel is attached to the doll and defines a pocket, into which the housing is selectively positionable. A battery, a microprocessor, and a receiver are attached to the housing and are positioned in an interior space defined by the housing. A speaker is attached to the housing. The microprocessor is operationally engaged to the battery, the receiver, and the speaker. Oversight programming code, which is selectively positionable on an electronic device of a user, enables the electronic device to store a plurality of audio files and user selection of a respective audio file for transmission to the microprocessor, via the receiver. The microprocessor is enabled to actuate the speaker to broadcast the audio.