22 results about "Digital audio signals" patented technology

Embodiments of the present application provide a soft mute control method and device for a radio device, a medium and a product. By extracting the spectral deviation feature and the time domain energy feature of the digital audio signal of the radio device, the spectral deviation feature can reflect the frequency spectrum similarity between the signal and the noise, and the time domain energy feature can reflect the signal strength. Combined with multi-state decision and energy decision, multi-dimensional decision of the signal state is performed to realize accurate identification of the signal state, and then the output gain is dynamically adjusted. Compared with the prior art, not only the signal receiving strength index is relied on for adjustment, but also the problem of large and small sound caused by signal strength fluctuation due to position change is avoided. The spectral deviation feature is used to effectively distinguish between effective signals and noise, improve the accuracy of soft mute decision, and the time domain energy feature is used as auxiliary decision to enhance the reliability of the system in a low energy scene, so that the volume is smooth and continuous, and the user's radio experience is improved as a whole.

The application discloses a Bluetooth earphone system, which comprises a charging bin and at least one Bluetooth earphone body accommodated in the charging bin, and a separable electric connection structure is arranged between the charging bin and the Bluetooth earphone body; the charging bin comprises a Bluetooth audio module, an analog audio interface module and a charging interface module; the Bluetooth audio module is integrated with an analog-digital conversion module and a wireless communication module; the charging interface module is integrated with a digital audio interface module connected with the Bluetooth audio module; the analog audio interface module and the digital audio interface module are connected with a non-Bluetooth audio device through signal lines; analog audio signals output by the non-Bluetooth audio device are converted into digital signals, and / or digital audio signals output by the non-Bluetooth audio device are processed by the digital audio interface module, and then the analog audio signals and the digital audio signals are wirelessly transmitted to the Bluetooth earphone body by the Bluetooth audio module. Thus, a user wearing the Bluetooth earphone body can realize wireless monitoring of the non-Bluetooth audio device, the operation is simple, and user experience is improved.

The present invention belongs to the technical field of electroacoustics in electronics, and relates to an audio system formed by a speaker-enclosure planar array, and a preparation method for the audio system. Different from the prior art, the present invention uses a plurality of small / -micro-aperture speakers to form a special planar array audio device, such that by using the plurality of small- / micro-aperture speakers, audio signals with frequencies of 20-120 Hz or even higher, which are typically cut off in small-aperture speaker systems, are caused, by means of synchronous superposition operation, to output powerful, clear and pure bass sound waves. Moreover, an algorithm for a linear digital filter is used to perform frequency division and surround sound processing on digital audio signals of the speakers without generating a phase shift. Therefore, a series of functional defects in conventional large-aperture subwoofers are overcome, thereby enabling the planar-array audio system to achieve a surround sound effect in a three-dimensional or two-dimensional space.

This utility model discloses an acoustic testing system for a directional sound-emitting device, including an audio analyzer, an audio processing unit, a power amplifier, a transformer unit, a DC power supply, and a pickup unit. The audio analyzer provides audio, which is then processed as a digital audio signal by the audio processing unit. The processed audio is amplified by the power amplifier, and after exiting the power amplifier, it is amplified again by the transformer. With the DC bias provided by the DC power supply, it is then connected to the directional sound-emitting device under test. The sound emitted by the directional sound-emitting device is received by the pickup unit and returned to the audio analyzer, which displays the frequency response and distortion curves. This utility model provides a novel acoustic testing platform that enables rapid testing of the acoustic performance of directional sound-emitting devices.

The utility model provides a kind of sound mixing device of bluetooth FM radio function, including signal input unit, sound mixing processing unit, front-stage amplification unit, filter circuit unit, rear-stage power amplification unit, audio playback unit;At least one signal input unit input digital audio signal is mixed sound after being handled by sound mixing processing unit, amplifies by front-stage amplification unit, then is filtered by filter circuit unit, amplifies by rear-stage power amplification unit after entering audio playback unit and plays;The input unit includes bluetooth receiving module, digital FM radio unit and at least one microphone.The utility model uses digitization, high intelligentization, integrated chip, product volume is small, electrical performance is good.

The application discloses a choir balance guidance method and system based on voiceprint recognition, relates to the technical field of digital audio signal processing, and comprises the following steps: extracting acoustic energy features and ontology voiceprint matching features of a current time window of multiple pickup channels, and calculating time sequence change rates of both relative to historical time windows; performing orthogonal state deviation judgment based on the time sequence change rates, and locking a target pickup channel interfered by a howling sound; determining a howling sound emission source channel and an absolute propagation time delay by cross-correlation ranging within a preset time delay constraint range; inversely calculating a physical propagation distance according to the time delay, and constructing a spatial spectrum dissipation filter in combination with a nonlinear acoustic dissipation model; performing phase shift and frequency response deformation constraint on a frequency domain signal of the howling sound source by using the time delay and the filter, and generating a physically deformed howling sound prediction signal; performing targeted cancellation on the frequency domain signal of the target channel and the prediction signal, reconstructing a pure acoustic energy parameter, and generating a balance guidance instruction; and the application can solve the balance guidance distortion problem caused by howling sound aliasing.

This invention discloses an AI-driven dynamic audio fence control method, specifically including the following steps: Step 1: Signal acquisition and preprocessing using a beamforming microphone array and a DSP processor; Step 2: AI feature extraction and speech noise classification of standardized multi-channel digital audio signals; Step 3: MVDR dynamic beamforming processing based on speech noise classification labels and noise feature masks; Step 4: 3D audio rendering of the beamformed single-channel target speech signal; Step 5: Effect monitoring and parameter iteration based on the target speech signal and feedback signals from the audio output module; Step 6: Pure target speech output playback based on the 3D rendered time-domain target speech signal. The advantages of this invention are: ensuring a continuous improvement in the target speech signal-to-noise ratio (SNR), ultimately transmitting pure target speech to the speaker, forming a personal sound bubble-like virtual audio fence around the target object.

1. Name of the product in this design: Decoding Headphone Amplifier (JM6PRO 2 Visual Portable Decoding Headphone Amplifier). 2. Purpose of this design: To convert digital audio signals from devices such as mobile phones and computers into analog audio signals to provide sufficient power to drive headphones, enabling them to produce a full, powerful, and well-controlled sound. 3. The key design feature of this product is its shape. 4. The image or photograph that best illustrates the design's key features: the front view.

A computer implemented method for generating digital audio signals for stimulating a user's brain, comprising the steps of: generating a first audio signal which changes frequency at a first rate for playback to a patient through a first speaker of a stereo headset; receiving an input which identifies a stressor frequency for the patient; generating a second audio signal which changes frequency at a second rate for playback to the patient through a second speaker of the stereo headset, while simultaneously playing an audio signal at the stressor frequency through the first speaker of the stereo headset, wherein an initial frequency for the second audio signal is selected to be within a range extending from the stressor frequency to a frequency 40 Hz higher than the stressor frequency; receiving an input which identifies a disruptor frequency for the user; and, storing the identified stressor frequency and the identified disruptor frequency for subsequent playback to the user.

Training of an audio foundation model (AFM) is performed using a dataset constructed using low-level audio property control and high-level composition planning. A plurality of digital audio compositions are generated, using a large language model (LLM) as a planner agent. The planner agent is prompted to generate composition plans defining logical combinations of foreground and background digital sounds, event occurrences within the compositions, and digital sound properties. The foreground and background digital sounds have consistent audio quality. An audio composition tool generates the plurality of digital audio compositions according to the composition plans. Descriptive text is generated for each of the digital audio compositions using a summarizer agent. The summarizer agent is implemented as an LLM, prompted to describe the digital audio compositions. The compositions and the corresponding descriptive text are combined to form audio-text pairs. An AFM is trained to interpret digital audio signals using the audio-text pairs.

A microphone hub and a method of operating one or more microphone hubs, The microphone hub including an input operable to receive one or more analog audio signals from one or more microphones, an audio network interface (ANIF), at least one analogue to digital converter (ADC) coupled to the input and to the ANIF, and a power system operable to provide power to the microphone hub. The ADC is operable to convert each of the one or more analog audio signals to one or more digital audio signals. The ADC is further operable to send the one or more digital audio signals to the ANIF.

An audio signal signature module (500) is provided, comprising: an input (502) for receiving a digital signal (121); a block divider (130) for dividing the received digital audio signal into a sequence of multiple audio blocks (131); an audio feature extractor (150) for extracting audio features from the current audio block; and a signature unit for generating a signature associated with the current audio block by applying a private key to the audio features. The signature unit (140) also provides signed authentication information including the extracted audio features and the signature, wherein the signed authentication information is output to a memory. The audio signal signature module (500) further includes an information embedder (160) for generating an audio block with embedded information by embedding authentication access information into the current audio block. The authentication access information includes information about how the signed authentication information can be retrieved from the memory and / or where the signed authentication information can be retrieved from the memory. In addition, an audio output (190) is provided for outputting an authenticateable audio signal comprising a sequence of audio blocks with embedded authentication access information.

An AI voice interaction CD player control method and device. The method comprises the following steps: calling a microphone for an AI voice recognition module to collect the user's voice signal, and digitizing the collected voice signal into a digital audio signal; preprocessing the digital audio signal; extracting voice signal feature parameters for the preprocessed digital audio signal; the voice signal feature parameters include Mel-frequency cepstral coefficient feature vectors and represent the acoustic features of the voice signal; inputting the extracted voice signal feature parameters into a pre-trained voice recognition model built based on a deep learning algorithm, and classifying and recognizing the input voice signal feature parameters through the voice recognition model to generate text control commands for a CD player. The invention can accurately understand the user's voice commands, and automatically perform playback control and music recommendation based on the user's personalized needs and preferences.

1. The name of the design product: portable decoding earphone (MD3). 2. The use of the design product: for digital audio signal to analog signal processing and analog signal amplification. 3. The design points of the design product: in shape. 4. The picture or photo that best shows the design points: perspective view 1.

The application relates to the field of audio compensation technology and provides an audio device, a method, equipment and a medium based on DAC module optimized delay difference. The audio device comprises a clock generation circuit which generates a reference clock signal; a clock distribution circuit which buffers the reference clock signal to obtain a first working clock to a DAC module and performs frequency division to obtain a second working clock to a digital audio processing circuit; the digital audio processing circuit processes a digital audio signal according to the second working clock, outputs an audio data stream and a synchronous reference clock signal; the DAC module detects a phase difference between the first working clock and the reference clock signal, generates a phase error signal and filters the phase error signal to obtain a phase adjustment value, adjusts the phase of the first working clock to generate a sampling clock according to the phase adjustment value, makes a sampling time point located in a data stable interval and compensates for a sampling deviation caused by the propagation delay of the first and second working clocks. The application eliminates clock delay mismatch and improves audio linearity and fidelity.

The audio acquisition device comprises a first audio acquisition module, a second audio acquisition module, a logic scheduling module and a processing module. The first audio acquisition module works in an I2S master mode, outputs an N-bit first digital audio signal, a reference bit clock signal and a first left-right channel control signal; the second audio acquisition module works in an I2S slave mode and outputs an N-bit second digital audio signal under the triggering of a second left-right channel control signal. The logic scheduling module generates a second left-right channel control signal with a K-bit clock period offset based on the reference bit clock signal, and forms a combined data stream by making two signals time-sharing occupy the same physical data channel through hardware selection logic according to the combined logic state of the first left-right channel control signal and the second left-right channel control signal. The processing module receives the combined data stream and parses multiple original audio data. The scheme realizes multi-channel acquisition without increasing the number of physical channels, reduces the interface resource occupation and hardware cost.

The application discloses a kind of photovoltaic support health monitoring system and method, and the vibration audio of photovoltaic support is collected by digital audio collector, and is converted into digital audio signal and is transmitted to audio terminal processing system, then the vibration voiceprint features of photovoltaic support are extracted;Meanwhile, the pressure between photovoltaic support connecting components and the pressure change are monitored by pressure voltage sensor, are converted into voltage signal and are transmitted to voltage terminal processing system, then voltage waveform features are extracted;Again, the vibration voiceprint features and voltage waveform features extracted are superimposed and integrated by double-wave signal processing subsystem, and double-mode monitoring data are obtained;Finally, the health status of photovoltaic support is analyzed based on the double-mode monitoring data.The present application can improve the inspection efficiency and monitoring accuracy of photovoltaic support, thereby improving the safety of photovoltaic power station.

The application discloses a stereo width enhancement method and device, a storage medium and a program product, and relates to the technical field of digital audio signal processing. The method comprises the following steps: extracting a central channel time domain signal based on an input left channel signal and a right channel signal; extracting a lateral channel signal based on the left channel signal, the right channel signal and the central channel time domain signal; dividing the lateral channel signal into continuous processing blocks according to time; for each processing block, detecting the peak value of each sampling point in the processing block, and determining the actual gain of the processing block based on the peak value; and based on the actual gain of the processing block, amplifying the lateral channel signal in the processing block, and merging the amplified lateral channel signal with the central channel time domain signal to obtain an enhanced stereo signal. Through block-level peak value detection and smooth gain variation, the upper limit of the gain is limited, the dynamic enhancement of the stereo width is realized without clipping distortion, and the dynamic response is natural without sound image jumping.

This invention discloses a method and system for intelligent mixing with human voice spectrum protection, belonging to the field of digital audio signal processing technology. The method extracts speech semantic weights using a lightweight NLP model, combines speech rate, pause duration, continuous listening duration, and environmental noise parameters, and calculates a real-time listening status score using a nonlinear mapping function. Based on the score, it adaptively adjusts the background binaural beat audio signal parameters, locking the 250Hz-4kHz core human voice frequency band. Gain boosting, notch filtering, and dynamic compensation are used to achieve human voice spectrum isolation protection, completing the intelligent mixing output. The system includes modules for speech parsing, status acquisition, nonlinear comprehensive evaluation, adaptive control, intelligent mixing engine, audio playback, and environmental noise acquisition. It is compatible with multiple terminals such as learning apps, online classes, smart headphones, and hearing aids. This invention effectively solves the problem of human voice spectrum masking, improves voice clarity and auditory experience, adjusts only the audio signal frequency, avoids patent infringement risks, and offers a complete and practical technical solution suitable for various intelligent audio mixing scenarios.

A data processing method performed by a processor that receives data signals from a plurality of devices that utilize different protocols from each other receives data signals, respectively, from each of the plurality of devices, and generates multitrack audio data, in which audio data of a plurality of channels is stored by storing a first data string of a digital audio signal, received from a first device of the plurality of devices that utilizes a first protocol, in a first channel of the plurality of channels, and storing a data string of a digital signal, related to the digital audio signal and received from a second device of the plurality of devices that utilizes a second protocol different from the first protocol, as a second data string of the digital audio signal in a second channel of the plurality of channels.