917 results about "Audio signal" patented technology

A headset antenna and a connector for the same are provided. The headset antenna includes an audio signal line, an antenna and a high impedance element in specified application frequency ranges. The audio signal line is adapted for transmitting an audio signal and the antenna is adapted for receiving an RF signal. The high impedance element is disposed on a transmission path of the audio signal and generates a high impedance at a specified frequency band of the RF signal, so that the audio signal line is equivalent to an open circuit and the antenna obtains a better receiving capability.

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.

A system for implementing interactive language learning includes a microphone, an audio output unit, a data storage unit, a display unit, and a controller. The controller generates an audio signal related to a question stored in the data storage unit, and controls the audio output unit to output the audio signal. In response to receipt of a speech signal, the controller outputs a text answer based on the speech signal. The controller determines whether the text answer corresponds with one of a plurality of predetermined answers associated with the question, generates one of an affirmative response and a guidance response based on the determination, transforms the one of the affirmative response and the guidance response into a form of speech, and controls the audio output unit to output the one of the affirmative response and the guidance response as a new audio signal.

The application discloses a baby crying sound recognition method for white noise equipment, relates to the field of audio processing and intelligent acoustic recognition, and comprises the following steps: collecting pure reference signals, first signals and second signals for synchronization and preprocessing, and obtaining mixed audio signals based on the first signals; calculating residual signals through a self-adaptive echo cancellation algorithm; calculating acoustic masking parameters based on the residual signals, the pure reference signals and environmental noise estimation; constructing a three-level recognition processing path comprising light feature detection, registered voiceprint comparison and multi-modal information fusion; selecting a recognition processing path based on the numerical range of the acoustic masking parameters, confirming the crying event of the target baby from the residual signals; and triggering corresponding graded alarms based on the crying event confirmation result. The three-level recognition path is scheduled through the acoustic masking parameters, multi-modal information is fused, and accurate and low-power baby crying monitoring under strong interference is realized.

This disclosure provides an apparatus for outputting audio signals. The apparatus may include a bone conduction unit configured to generate a bone conduction acoustic waveform. The apparatus may also include an air-tympanic conduction unit configured to generate an air-tympanic conduction acoustic waveform; the bone conduction acoustic waveform and the air-tympanic conduction acoustic waveform may represent the same audio signal. The apparatus may include a phase difference between the bone conduction acoustic waveform and the air-tympanic conduction acoustic waveform that may be less than a threshold. The apparatus may include a housing configured to accommodate at least a portion of the bone conduction unit and the air-tympanic conduction unit.

A control circuit for an audio amplifier includes a reference voltage generator and a voltage limiter. The reference voltage generator generates a first reference voltage and a second reference voltage based on a supply voltage of the audio amplifier. The voltage limiter receives an audio input signal, the first reference voltage, and the second reference voltage, and generates an adjusted audio signal based on the audio input signal, the first reference voltage, and the second reference voltage. When a voltage level of the audio input signal is higher than the first reference voltage, the adjusted audio signal is limited to the first reference voltage, and when the voltage level of the audio input signal is lower than the second reference voltage, the audio input signal is limited to the second reference voltage. The input voltage limiter is used to generate a truncated input signal proportional to the desired output signal when clipping occurs. This prevents the control loop from being affected by inaccurate feedback information and enables faster transient response and less distortion.

This invention provides an earphone control method and earphones that can improve the perceived auditory effect. [Solution] The method includes: collecting a first audio signal with a feedforward microphone and a second audio signal with a feedback microphone 202; determining the current scene type of the external scene in which the earphone is located based on the first audio signal and acquiring a target effect audio signal corresponding to the current scene type 204; determining current frequency spectrum relationship information between the first audio signal and the second audio signal, and target frequency spectrum relationship information between the first audio signal and the target effect audio signal 206; and repeatedly adjusting the filter parameters of the earphone based on the current frequency spectrum relationship information and the target frequency spectrum relationship information 208, wherein the error between the newly collected second audio signal and the target effect audio signal is smaller than a predetermined error.

Systems and methods for subband audio coding are disclosed. One example provides a method for encoding an input audio signal in a bitstream from a first device to a second device. The method includes obtaining, with the first device, a representation of the input audio signal in subbands, dividing a current frame into a current subset of the subbands excluding other subbands, and determining prediction parameters for the current subset of the subbands. The method further includes determining, with the first device, an active / passive flag for at least one of the subbands, and encoding the bitstream with the current subset of the subbands, a number of the subsets, the prediction parameters for the current subset of the subbands, and the active / passive flag for the other subsets of the subbands. The method can include decoding the bitstream with the second device.

The method comprises the following steps: a first terminal acquires a three-dimensional audio signal and performs signal layering on the three-dimensional audio signal to obtain a base layer audio signal and an enhancement layer audio signal; the base layer audio signal is encoded to obtain a base audio code stream, and the enhancement layer audio signal is encoded to obtain an enhancement audio code stream; a compatible code stream format pre-constructed is used to perform stream combining processing on the base audio code stream and the enhancement audio code stream to obtain a target audio code stream; and the target audio code stream is transmitted to a second terminal, so that the second terminal decodes the base audio code stream and the enhancement audio code stream and plays or only decodes the base audio code stream and plays. By using the method, only one audio code stream can be formed, effective encoding of the three-dimensional audio signal is realized, the storage cost and the bandwidth cost of an audio encoding end are reduced, and the encoding stability of an encoding end device is improved.

This invention relates to the fields of audio signal processing and artificial intelligence, specifically to a lyrics alignment method and electronic device based on automatic speech recognition. The lyrics alignment method based on automatic speech recognition in this application uses a pre-trained automatic speech recognition model to recognize audio samples and generate recognized text. The edit distance sequence between the recognized text and the lyrics text is calculated, and spoken segments are determined by analyzing the distance change pattern through a sliding window. These spoken segments are then replaced with silent segments, effectively eliminating spoken segments and avoiding interference from non-lyric content in the alignment process, thus laying the foundation for subsequent alignment.

This invention provides an audio transmission method, apparatus, and server method, relating to the technical field of audio adaptation. The method includes: establishing a communication connection between a sound card to be adapted and an embedded system to be adapted through an audio management unit, so as to transmit control signals generated by the embedded system to the sound card through the communication connection; and feeding back the audio signal generated by the sound card in response to the control signals to the embedded system through an audio transmission structure. This invention establishes a connection between the system and the sound card through an adaptation platform, enabling audio transmission between the system and the sound card.

A method and a decoder for generating concealment audio frame of an audio signal. The method includes performing a frequency domain analysis of a sequence of previously decoded audio signal to obtain a frequency spectrum and identifying peaks in the spectrum. The method further includes estimating a relative energy between the noise spectrum and the complete spectrum, determining an attenuation of the noise spectrum based on the relative energy, and applying the attenuation to the noise spectrum. The method includes applying an inverse transform to time domain on an error concealment spectrum, that includes the peaks and the attenuated noise spectrum.

This utility model relates to an ultra-low power, high-intensity directional acoustic warning device, including a front panel, a housing, a warning light, a sound-generating module, and a circuit board assembly. The circuit board assembly includes a power management module, a central management module, an audio algorithm processing unit, and a boost drive module. The power management module supplies power to the strobe light and the piezoelectric sound-generating unit. The central management module connects to the audio algorithm processing unit, the boost drive module, and the strobe light. The audio algorithm processing unit is connected to the boost drive module, which is also connected to the piezoelectric sound-generating unit. The central management module controls the audio algorithm processing unit to process the input audio signal and then drives the piezoelectric sound-generating unit to emit sound through the boost drive module. This utility model overcomes the high power consumption bottleneck of traditional electromagnetic loudspeakers and integrates the collaborative work of the strobe light and the piezoelectric sound source, enhancing target recognition through multi-dimensional warnings.

1. Name of the product in this design: Audio Amplifier. 2. Purpose of this design: to amplify audio signals. 3. The key design feature of this product is its shape. 4. The image or photograph that best illustrates the design's key points: 3D rendering 1. 5. The top view is already shown in perspective view 2, so it is omitted; the bottom view is already shown in perspective view 1, so it is omitted.

The present application relates to epidemic prevention communication technology field, especially to a duplex cluster communication interaction method and system and epidemic prevention helmet, which comprises: using the air guide pickup component arranged in the helmet body to obtain the voice signal of the wearer, and outputting the audio signal through the air guide sound production component arranged in the helmet body; performing echo cancellation and noise reduction processing on the obtained audio signal; using an adaptive filtering algorithm to suppress the acoustic feedback between the air guide pickup component and the air guide sound production component; based on a distributed channel access mechanism, allocating communication resources for multiple communication nodes to access in duplex; relying on a mesh self-organizing network topology without a central node to perform multi-hop forwarding and route maintenance of data packets, solving the problems of unclear communication, howling and signal attenuation in the strong noise, wind noise and shielding scene in the epidemic prevention helmet, effectively guaranteeing the clarity, stability and safety of cluster communication in harsh working environment.

The application discloses an end-to-end audio-visual source separation method based on dynamic gating fusion. The method uses a dynamically adjusted fusion mechanism to effectively fuse audio and visual features through a dynamic gating fusion module, and further separates the mixed audio signal through an audio-visual Transformer decoder. The method can accurately separate the target audio in a complex multi-source environment, significantly improving the quality of audio-visual source separation, especially in the presence of background noise and multiple sources. Experimental results show that the performance of the method on multiple benchmark datasets has been significantly improved, verifying its effectiveness and advantages in the audio-visual source separation task.

1. Name of the product in this design: Audio System (K901). 2. Purpose of this design: This design is used for the output of audio signals. 3. The key design feature of this product is its shape. 4. The image or photograph that best illustrates the design's key points: a 3D model.

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.

This application provides a method for controlling a vehicle, a device for controlling a vehicle, and a vehicle. The method relates to the field of vehicle control technology. The vehicle is equipped with a speaker array; the method includes: acquiring first position information of each speaker in the speaker array and second position information of a target object around the vehicle; determining audio output parameters of each speaker based on the first position information of each speaker and the second position information of the target object; and controlling each speaker to output a target audio signal based on the audio output parameters, the target audio signal being used to warn the target object. This method can improve the warning effect of the vehicle on the target object.

The application provides a recording processing method and related devices. The method can include: an electronic device can perform sound source positioning based on the sound collected by a microphone, obtain the position of a target sound source and the number of sound sources in the recording environment, and then perform sound source separation on the sound collected by the microphone according to the position of the target sound source and the number of sound sources in the recording environment to obtain the sound corresponding to the target sound source, i.e., a target audio signal. The electronic device can also determine the signal-to-noise ratio and display the current sound pickup quality to the user. This method can monitor and display the sound pickup quality to the user in real time, so that the user can adjust in time when the sound pickup quality is poor, thereby obtaining high-quality audio and improving the user experience.

Systems and methods of selecting digital platforms for execution of voice-based commands are provided. The system receives an application that performs an action associated with a service via digital platforms. The system debugs the application to validate parameters of the action on at least two platforms of the digital platforms. The system receives data packets comprising an input audio signal detected by a sensor of a client device, and parses the input audio signal to identify the action and the service. The system selects a first platform from the digital platforms to perform the action. The system initiates, responsive to selection of the first platform, an interactive data exchange to populate parameters of an action data structure corresponding to the action. The system executes the action via the selected platform using the action data structure.

A noise injury relief system is provided in the form of a portable handheld device configured for private therapeutic audio playback. The device includes an enclosure sized for pocket carry and having a front face with user controls and a cartridge insertion opening configured to receive a vaping cartridge. The front face includes an eleven-position slide switch for independently selecting internally stored recordings corresponding to eleven bells and a twelve-position slide switch positioned adjacent thereto for independently selecting twelve musical notes. An additional switch permits selection among different musical instruments for playback of the musical notes, and a volume control knob enables user adjustment of audio output. Internal audio signals are processed through a signal generator and transmitted wirelessly to a user audio device for private playback. The system is powered by an internal power supply and is configured for repeated use.

Methods and vehicle with sound visualization are provided. A vehicle includes a graphic projection display; exterior microphones mounted to the vehicle; and a processing device programmed to receive audio signals from the exterior microphones; retrieve map data in a region around the vehicle; identify from the audio signals a sound of interest; identify a location of a source of the sound of interest; determine a stationary status or a moving status of the source; when the source has the moving status, determining a direction and speed of movement of the source; ascertain from the map data and from the location, stationary status, moving status, direction, and / or speed, a preferred driving maneuver; determine a graphic exemplifying the preferred driving maneuver; and display the graphic upon the graphic projection display.

A mobile electronic code device includes a personal electronic device carried by a recipient user. The personal electronic device has a transceiver which receives a communication signal from an extrinsic communication network. An electronic memory is integrated into the personal electronic device which stores a language translation program. The language translation program translates a message in the communication signal into a coded audio signal to audibly communicate the message to the recipient user in a code that only the recipient user can understand thereby enhancing privacy for the recipient user. A speaker is integrated into the personal electronic device to audibly emit the coded audio signal to the recipient user through the plurality of sound holes.

The embodiment of the application provides an audio low-frequency signal gain control circuit and a related device, which comprise an input processing sub-circuit, a non-dynamic low-frequency signal processing sub-circuit, a dynamic low-frequency signal processing sub-circuit, a signal synthesis sub-circuit, an adaptive gain control sub-circuit and a bass speaker. The input processing sub-circuit is used for pre-processing an input stereo audio signal to obtain a first audio low-frequency signal; the non-dynamic low-frequency signal processing sub-circuit is used for processing the first audio low-frequency signal to obtain a non-dynamic audio low-frequency signal; the dynamic low-frequency signal processing sub-circuit is used for separating a dynamic audio low-frequency signal; the signal synthesis sub-circuit is used for merging the non-dynamic audio low-frequency signal and the dynamic audio low-frequency signal to obtain an audio low-frequency signal; the adaptive gain control sub-circuit is used for gain control on the audio low-frequency signal; and the bass speaker is used for outputting the audio low-frequency signal. In this way, the adaptive gain on the audio low-frequency signal is realized through an analog circuit.

This application discloses a vehicle including an audio system. The audio system includes at least a first speaker and a first piezoelectric sound-generating component. The first speaker is mounted at a first predetermined location in the vehicle, and the first piezoelectric sound-generating component is mounted at a second predetermined location in the vehicle. The first speaker outputs an audio signal in a first frequency band of the target audio signal; the first piezoelectric sound-generating component outputs an audio signal in a second frequency band of the target audio signal. The first and second frequency bands are different frequency bands. Thus, by combining the speaker and the piezoelectric sound-generating component to output audio signals in different frequency bands, the advantages of both the speaker and the piezoelectric sound-generating component can be better utilized, improving the overall sound quality.

1. The name of the design product: neck-hung sports Bluetooth earphone (R2). 2. The use of the design product: for playing audio signals. 3. The design points of the design product: in shape. 4. The picture or photo that best indicates the design points: perspective view 1.

The application discloses a kind of mill sound analyzers, receiver, for receiving the audio signal frequency band in the process of ball mill equipment operation;Filter module, amplification module, intelligent analysis module, output module;The processing analysis module includes: energy spectrum calculation module, carries out spectrum analysis to audio signal frequency band, extracts the energy of each frequency band as feature, and constructs feature vector in combination with historical state;Model inference module, using the constructed feature vector is handled to predict the current ball mill capacity.The application receives the audio signal of ball mill equipment in real time, and carries out processing and analysis to it, quickly responds the change of ball mill equipment operating state, realizes accurate prediction to ball mill capacity, optimizes the operating parameter of ball mill equipment by accurately predicting ball mill capacity, improves production efficiency and product quality, the application of intelligent analysis module and model inference module makes the management of ball mill equipment more intelligent and automated, reduces the complexity and cost of manual intervention.

This application provides a method, apparatus, device, and storage medium for selecting a master microphone. The method is applied to a recording host in a recording space, where multiple microphones are deployed at different locations within the recording space, and the recording host is connected to the multiple microphones. The method includes: acquiring user positioning data, which reflects the location of a target user in the recording space; and combining audio signal parameters acquired by the multiple microphones with the user positioning data to select a master microphone from the multiple microphones. This technical solution enables the selection of a master microphone and enhances the reliability of microphone selection.

The audio device includes a receiver 101 configured to receive an audio signal from which an audio beamformer 103 generates a beamformed output audio signal. The beamformer 103 includes a signal processor 201 that applies signal processing to the audio signal. Signal processing includes spatial beamforming and spatial decorrelation, which is often adapted spatial decorrelation. A decorrelation adapter 203 adapts decorrelation parameters depending on the audio signal, and a beamform adapter 205 adapts beamform parameters depending on the beamformed output audio signal. An equalizer 105 applies frequency equalization to the beamformed output audio signal to generate an output signal, and an equalization adapter 107 adapts frequency equalization depending on the beamform parameters, typically also depending on the decorrelation parameters. This approach provides improved fidelity to the desired sound source and attenuates interfering sound sources.