34 results about "Audio Signal Processor" patented technology

Apparatus for retrieving an acoustic room information from a database, comprising: a test fingerprint processor (10) for deriving a test fingerprint from a test acoustic room information; and a database processor (20) for retrieving a reference acoustic room information having a reference fingerprint matching with the test fingerprint, wherein the test fingerprint processor (10) comprises; a filter bank configured for decomposing the test acoustic room information into a plurality of acoustic room information sub-bands; and a parameter calculator (12) configured for calculating a sub-band parameter for each acoustic room information sub-band of the plurality of acoustic room information sub-bands, the sub-band parameter indicating an energy change over time in the acoustic room information sub-band, wherein the test fingerprint is based on the sub-band parameters for the acoustic room information sub-bands.

An electronic device for characterizing and / or monitoring an operation with an inhaler device is disclosed. The electronic device comprises a memory, an interface and a processor comprising predictor circuitry configured to operate according to a prediction model. The processor is configured to obtain operation data (such as operation data), where the operation data is indicative of an audio signal representing an operation with the inhaler device. The processor is configured to determine, based on the operation data, using the predictor circuitry, a predicted operation parameter indicative of a prediction of an operation (such as inhalation flow and / or an exhalation flow) with the inhaler device. The processor is configured to determine, based on the predicted operation parameter, an operation representation.

Embodiments of the present application provide a voice interaction method, the method comprising: collecting user voice, and inputting the user voice into a first voice model of an audio signal processor to obtain a preliminary screening confidence; when the preliminary screening confidence is greater than or equal to a preliminary screening confidence threshold, inputting the user voice into a second voice model of an application processor to obtain a target confidence; calculating, by the application processor, a sound source angle corresponding to the user voice, and determining a target confidence threshold corresponding to the sound source angle; when the target confidence is greater than or equal to the target confidence threshold, determining that the user voice is a voice instruction, and performing voice interaction in response to the voice instruction. Embodiments of the present application preliminarily screen the user voice through the first voice model of the audio signal processor, thereby reducing system power consumption, and the second voice model of the application processor can accurately determine whether the user voice is a voice instruction.

The utility model discloses a USB high-fidelity decoder with a vibration unit, which comprises an integrated board. According to the utility model, the USB assembly, the system control assembly, the Bluetooth assembly, the first motor assembly, the first vibration driving assembly, the audio signal processor assembly, the loudspeaker assembly, the second motor assembly, the second vibration driving assembly, the earphone assembly and the high-fidelity digital-to-analog conversion assembly are integrated in the integrated board, so that the integrated board forms an integral module; the module can be attached to the surface of an object such as a mobile phone and can also be embedded into a mobile phone shell, the design that sound is experienced through the touch sense of the human body is achieved, a user can feel the sound in multiple dimensions on the basis of traditional auditory sense, the overall experience of entertainment equipment is enhanced, and meanwhile brand new understanding of music / sound is achieved.

An audio signal processor for generating a two-channel audio signal, comprising: an input interface (100) for providing single-channel acoustic data describing an acoustic environment; a two-channel synthesizer (200) for synthesizing two-channel acoustic data from the single-channel acoustic data using the listener position or rotation; and a sound generator (300) for generating a two-channel audio signal from the audio signal and the two-channel acoustic data, where the input interface (100) is configured to acquire (150) an original representation related to the single-channel acoustic data, and deriving (151) single-channel acoustic data using the original representation and additional data stored in or accessible by the audio signal processor.

An audio signal processor for generating a two-channel audio signal, comprising: an input interface (100) for providing single-channel acoustic data describing an acoustic environment; a two-channel combiner (200) for synthesizing two-channel acoustic data from single-channel acoustic data using listener position or rotation; and a sound generator (300) for generating a two-channel audio signal from an audio signal and two-channel acoustic data, wherein the two-channel combiner (200) is configured to separate the single-channel acoustic data into at least two parts consisting of a direct sound portion and at least one of an early reflection portion and a late reverberation portion (210), and to process at least two parts individually for each part to generate two-channel acoustic data (220, 230, 240), and the two-channel combiner (200) is configured to compute two-channel acoustic data for the early reflection portion using a specular reflection portion describing a separate early reflection and a diffusion portion describing the diffusion effect within the early reflection portion (230).

A device and method for audio applications. The device includes a memory and a processor configured to receive a first audio signal, receive metadata associated with the first audio signal, create a first profile associated with the first audio signal, the first profile including a plurality of adjustable audio parameters, and process the first audio signal using the plurality of adjustable audio parameters of the first profile. The processor is configured to send the first audio signal processed using the first profile to one or more playback devices; receive one or more first user inputs from one or more sensors, analyze the one or more first user inputs, and adjust one or more of the plurality of audio parameters of the first profile based on the analysis, the metadata, or a combination of the analysis and the metadata; process the first audio signal using the adjusted first profile, and send the first audio signal processed using the adjusted first profile to one or more sound channels.

A hub device includes a plurality of inputs configured to receive one or more audio signals from an audio device, a processor, and a non-transitory computer-readable storage medium storing instructions. The instructions, when executed by the processor, cause the processor to map the one or more audio signals received at the plurality of inputs to a plurality of audio channels, and generate a plurality of audio streams corresponding to the plurality of audio channels, wherein different streams of the plurality of audio streams correspond to different channels of the plurality of audio channels. The hub device further includes one or more audio outputs configured to transmit the plurality of audio streams to a plurality of wireless speakers.

An audio signal processor includes a harmonic signal generator to generate a harmonic signal that includes a signal generated by frequency scaling components of a target audio signal by a factor of k, a left audio signal generator to generate a left audio signal by convolving a head-related transfer function from a target localization position of the target audio signal to a left ear of a listener with the target audio signal, a right audio signal generator to generate a right audio signal by convolving a head-related transfer function from the target localization position to a right ear of the listener with the target audio signal, a left synthesizer to generate a left channel audio signal by synthesizing the left audio signal and the harmonic signal, and a right synthesizer to generate a right channel audio signal by synthesizing the right audio signal and the harmonic signal.

This invention discloses an integrated audio signal processor for high-speed data transmission via 40G / 10G Ethernet and SRIO, belonging to the field of integrated audio signal processing technology. It includes a signal classification module that dynamically divides the audio signal stream into fast and slow signal streams; a heterogeneous transmission module comprising a 40G Ethernet pass-through channel, a 10G Ethernet channel, and an SRIO board interconnection channel; an intelligent routing module connecting the signal classification module and the heterogeneous transmission module; a container resource scheduling module for dynamically partitioning GPU memory and allocating computing resources for multi-container parallel tasks; a heterogeneous computing offloading module for generating kernel-level scheduling instructions; and a closed-loop optimization module that collects network latency, jitter, and GPU utilization data in real time and generates an optimization strategy package containing channel switching instructions and resource reallocation instructions. This invention achieves low-overhead communication, dynamic path selection, container-level resource isolation, and GPU virtualization collaborative optimization.

The invention discloses a method for processing an audio signal and an audio signal distortion detection system. Comprising the steps that an audio amplifier receives an audio signal output by an audio signal processor in real time, generates a first pulse width signal according to the audio signal and sends the first pulse width signal to a distortion compensation device; the distortion compensation device generates a second pulse width signal according to a preset distortion degree threshold value of the audio signal; the distortion compensation device generates a level signal according to the first pulse width signal and the second pulse width signal, and sends the level signal to the audio signal processor; the audio signal processor adjusts the strength of the audio signal according to the high-low state of the received level signal, so that the distortion degree of the audio signal is smaller than or equal to the preset distortion degree, the flexible adjustment of the distortion degree of the audio signal is realized, and the distortion degree of the audio signal can better meet the requirement; and the control precision of the audio signal is improved.

The utility model discloses a sound system for a vehicle and the vehicle, and belongs to the technical field of sound. The sound system comprises a multimedia host which comprises an audio signal processor and a built-in power amplifier, and a first audio signal output end of the audio signal processor is connected with a signal input end of the built-in power amplifier; the signal input end of the first external power amplifier is connected with the second audio signal output end of the audio signal processor; the first loudspeaker is connected with the signal output end of the built-in power amplifier; and the second loudspeaker is connected with the signal output end of the first external power amplifier. According to the invention, the high-configuration requirement of the user can be quickly met.

An audio signal processor for generating a 2-channel audio signal, comprising: an input interface (100) for providing single-channel acoustic data describing an acoustic environment; a 2-channel synthesizer (200) for synthesizing 2-channel acoustic data from single-channel acoustic data using listener position or rotation; and a sound generator (300) for generating a 2-channel audio signal from an audio signal and 2-channel acoustic data, wherein the 2-channel synthesizer (200) separates the single-channel acoustic data into at least two parts (210) consisting of a direct sound portion and at least one of an early reflection portion and a late reverberation portion, and generates 2-channel acoustic data for each part Audio signal processor configured to process at least two parts individually to generate (220, 230, 240), wherein a two-channel combiner (200) is configured to use the amplitude spectrum of the early reflection portion or single-channel acoustic data without the direct tone portion or late reverberation portion and a first channel noise phase spectrum to obtain the first channel of the two-channel acoustic data, and to use the amplitude spectrum of the early reflection portion or single-channel acoustic data without the direct tone portion or late reverberation portion and a second channel noise phase spectrum to calculate the two-channel diffuse portion of the early reflection portion or single-channel acoustic data without the direct tone portion or late reverberation portion.

An audio signal processor for generating a 2-channel audio signal, comprising: an input interface (100) for providing single-channel acoustic data describing an acoustic environment; a 2-channel synthesizer (200) for synthesizing 2-channel acoustic data from single-channel acoustic data using listener position or rotation; and a sound generator (300) for generating a 2-channel audio signal from an audio signal and 2-channel acoustic data, wherein the 2-channel synthesizer (200) separates the single-channel acoustic data into at least two parts (210) consisting of a direct sound portion and at least one of an early reflection portion and a late reverberation portion, and processes at least two parts individually for each part to generate 2-channel acoustic data. Audio signal processor configured such as (220, 230, 240), wherein a two-channel combiner (200) segments early reflections into multiple segments (231), determines multiple image source positions representing the source positions of reflected sound (232), associates image source positions with segments using a matching operation, the matching operation comprising: calculating the sound arrival time of each image source to the listener position; and associating the image source position with a corresponding segment having a time delay in the corresponding segment that best matches the sound arrival time of the corresponding image source position (234); and uses the image source positions associated with the segments to calculate two-channel acoustic data of direct sound.

An audio signal processor for generating a 2-channel audio signal comprises an input interface (100) for providing single-channel acoustic data describing an acoustic environment, a 2-channel synthesizer (200) for synthesizing 2-channel acoustic data from the single-channel acoustic data using listener position or rotation, and a sound generator (300) for generating a 2-channel audio signal from an audio signal and 2-channel acoustic data, wherein the 2-channel synthesizer (200) separates the single-channel acoustic data into at least two parts consisting of a direct sound portion and at least one of an early reflection portion and a late reverberation portion (210), processes at least two parts individually to generate 2-channel acoustic data for each part (220, 230, 240), determines the directivity information of the sound source with respect to the listener position and the source position or orientation of the sound source (222), and is configured to use the directivity information in the calculation of the 2-channel acoustic data of the direct sound portion (220).

An apparatus includes at least three audio sensors, at least three analog processing modules, and a processor circuit. The at least three audio sensors may be configured to generate electronic signals in response to audio sounds in an environment surrounding a vehicle. The at least three analog processing modules may be configured to process the electronic signals generated by the at least three audio sensors to generate one or more processed audio signals. The processor circuit may be configured to process the one or more processed audio signals generated by the at least three analog processing modules to determine a source direction of the audio sounds relative to the vehicle and identify a type characteristic of the audio sounds. The at least three analog processing modules generally process amplitude and pitch of the electronic signals using separate analog processing paths.

Systems and methods use a processor to obtain a signal data signatures (SDS) of a forced cough vocalizations (FCV). The processor controls a display present an instruction to a user to produce an FCV. The processor controls a recording device to record audio during the FCV so as to receive an audio signal that captures the FCV. The processor pre-processes the audio signal and outputs FCV signal data representative of the FCV. The processor uses a cough signature model to ingest the FCV signal data and output an SDS representative of isolated FCV-related signal data. The processor validates one or more SDSs by classifying and testing the SDS against SDS criteria (baseline) that establish a minimum quality7 of the SDS. Based on the SDS failing to achieve the SDS criteria, the processor deletes the SDS and controls the display to present a new instruction to the user for a new FCV.

Embodiments of the present application provide a voice interaction method and related equipment, the method comprises: collecting user voice, and inputting the user voice into a first voice model of an audio signal processor to obtain a preliminary screening confidence; when the preliminary screening confidence is greater than or equal to a preliminary screening confidence threshold, inputting the user voice into a second voice model of an application processor to obtain a target confidence; calculating the sound source angle corresponding to the user voice through the application processor to determine the target confidence threshold corresponding to the sound source angle; when the target confidence is greater than or equal to the target confidence threshold, determining that the user voice is a voice instruction, and responding to the voice instruction to perform voice interaction. Embodiments of the present application preliminarily screen the user voice through the first voice model of the audio signal processor, thereby reducing the system power consumption, and the second voice model of the application processor can accurately determine whether the user voice is a voice instruction.

A noise cancelling microphone configured to cancel ambient noise is disclosed herein. The noise cancelling microphone includes a first micro-electro-mechanical systems (MEMs) microphone coupled to a first surface of a printed circuit board (PCB), a second MEMs microphone coupled to a second surface of the PCB opposite the first surface, and an audio signal processor electrically coupled to the first MEMs microphone and the second MEMs microphone. The first MEMs microphone is configured to receive a first set of one or more sound waves approaching the noise cancelling and output a first electrical signal. The second MEMs microphone is configured to receive a second set of one or more sound waves and output a second electrical signal. The audio signal processor is configured to receive the first electrical signal and the second electrical signal, process the first electrical signal and the second electrical signal, and output a noise canceled signal.

An audio signal processor for generating a two-channel audio signal, comprising: an input interface (100) for providing single-channel acoustic data describing an acoustic environment; a two-channel synthesizer (200) for synthesizing two-channel acoustic data from single-channel acoustic data using listener position or rotation; and a sound generator (300) for generating a two-channel audio signal from an audio signal and two-channel acoustic data, wherein the two-channel synthesizer (200) synthesizes the single-channel acoustic data into at least two parts consisting of a direct sound portion and at least one of an early reflection portion and a late reverberation portion. An audio signal processor configured to separate into minutes (210), process at least two parts individually to generate two-channel acoustic data for each part (220, 230, 240), determine the separation time in single-channel acoustic data between the direct sound part and the early reflection part or between the early reflection part and the late reverberation part (601), extend at least one of the two parts of the separation time by a certain number of samples so that an overlap in the separation time is achieved (602), and window at least one of the extended parts using a specific window function that takes sample extension into account (603).

An audio signal processor for generating a two-channel audio signal, comprising: an input interface (100) for providing single-channel acoustic data describing an acoustic environment; a two-channel synthesizer (200) for synthesizing two-channel acoustic data from the single-channel acoustic data using listener position or rotation; and a sound generator (300) for generating a two-channel audio signal from an audio signal and the two-channel acoustic data, wherein the input interface (100) is configured to acquire a raw representation related to the single-channel acoustic data (150) and derive the single-channel acoustic data using the raw representation and additional data stored in or accessible by the audio signal processor (151).

Disclosed are a system and method for verifying a power amplifier of a vehicular audio system. The system includes: a TDM converter for converting an audio signal into a TDM signal; an audio signal processor for receiving the TDM signal and generating and outputting a test signal for transmitting to the power amplifier, the audio signal processor having N audio channels, N>1; a channel configurator having M configuration units, M≥N, each of the audio channels corresponding to one of the configuration units, and each of the configuration units having a selected state and a non-selected state; and a controller for determining that each of the configuration units is in the selected state or non-selected state and controlling the audio signal processor to place the TDM signal in audio channel(s) corresponding to configuration unit(s) in the selected state.

Method and system for a cochlear implant that includes receiving, by a processor, an electric audio signal; processing, by the processor, the electric audio signal, thereby providing a processed electric audio signal; receiving, by a transducer driver, the processed electric audio signal; transmitting, by the transducer driver, the processed electric audio signal to electro-acoustic transducer elements that are at least arranged within a cochlea; and vibrating at least one electro-acoustic transducer element of the electro-acoustic transducer elements in response to the at least one electro-acoustic transducer element receiving the processed electric audio signal, thereby providing a vibrating stimulus based on the processed electric audio signal.

Audio signal processor for generating a two-channel audio signal, comprising: an input interface for providing single-channel acoustic data describing an acoustic environment; a two-channel synthesizer for synthesizing two-channel acoustic data from the single-channel acoustic data using a listener position or rotation; and a sound generator for generating the two-channel audio signal from an audio signal and the two-channel acoustic data, wherein the input interface is configured to acquire a raw representation related to the single-channel acoustic data, and to derive the single-channel acoustic data using the raw representation and additional data stored in the audio signal processor or accessible by the audio signal processor.

An audio signal processor for generating a two-channel audio signal, comprising: an input interface (100) for providing single-channel acoustic data describing an acoustic environment; a two-channel combiner (200) for synthesizing two-channel acoustic data from single-channel acoustic data using listener position or rotation; and a sound generator (300) for generating a two-channel audio signal from an audio signal and two-channel acoustic data, wherein the two-channel combiner (200) separates the single-channel acoustic data into at least two parts (210) consisting of a direct sound portion and at least one of an early reflection portion and a late reverberation portion, and generates at least two parts for each portion to generate two-channel acoustic data Audio signal processor comprising two physically separated devices (901, 902), wherein the first of the two physically separated devices (901) is configured to process at least one of the direct sound portion and the early reflection portion (220, 230), and the second of the two physically separated devices (903) is configured to process at least one of the early reflection portion and the late reverberation portion (230, 240), and the first device (901) and the second device (902) are connected via a transmission interface (918, 925) and have separate power supplies (917, 924).

Systems, methods, and servers are provided. A system may include a user interface to receive audio input data from a user input, an audio input device to receive multiple audio signals of multiple audio signal types, a processor circuit and a memory including machine-readable instructions that, when executed by the processor circuit, cause the processor circuit to receive, by an artificial intelligence (AI), the user input and the audio signals that include ones of the audio signal types. The processor is further caused to generate, by a music generator model that receives the user input and the audio signals, output audio content that corresponds with the music generator model, and store the output audio content for subsequent delivery to a user.

In an approach for road section recognition using multi-modal cognitive mechanism, a processor receives an audio signal from a road test. A processor processes the audio signal to generate an acoustic spectrum density distribution map to identify a respective at least one road section switching point in a first mode. A processor processes a spectrogram of the audio signal to identify the respective at least one road section switching point in a second mode. A processor uses a machine learning model to predict an expected sound at each frame of the audio signal, to calculate a similarity between the expected sound and an actual sound, and to identify the respective at least one road switching point when the similarity is lower than a pre-set similarity threshold in a third mode. A processor combines results of the three modes to obtain a final set of road section switching points.

A hearing aid comprises a) an ITE-part adapted for being located at or in an ear canal of the user, b) a forward path for processing sound from the environment of the user. The forward path comprises b1) at least one first input transducer providing at least one first electric input signal representing said sound as received at the respective at least one first microphone, said at least one first input transducer being located to allow picking up sound from the environment of the user, b2) an audio signal processor comprising a gain unit for applying a frequency and / or level dependent prescribed gain to compensate for a hearing impairment of the user to said at least one first electric input signal, or a signal or signals originating therefrom, and configured to provide a processed signal in dependence thereof, b3) an output transducer for providing stimuli perceivable as sound to the user in dependence of said processed signal. The hearing aid further comprises c) at least one second input transducer providing at least one second electric input signal representing said sound as received at the at least one second input transducer, the at least one second input transducer being located in said ITE-part to pick up sound at the eardrum of the user, d) a correlator configured to determine a correlation measure between the at least one second electric input signal, or a signal originating therefrom, and a signal of the forward path and e) a gain modifier configured to modify said gain of the gain unit in dependence of said correlation measure. A method of operating a hearing aid is further disclosed.