136 results about "Digital audio signals" patented technology

The invention relates to the technical field of digital playing equipment, and provides a lossless control method for the volume of a digital player and the digital player. The method comprises the following steps of: after receiving a target volume parameter, executing dynamic multiplying power oversampling and high-order noise shaping on an input digital audio signal, and injecting preset jitter to suppress in-band quantization distortion so as to generate an optimized digital audio stream; meanwhile, a high-resolution digital reference current control instruction corresponding to the target volume is independently generated, output of a reference current source is adjusted according to the high-resolution digital reference current control instruction, and a high-precision differential current signal is obtained by combining nonlinearity, temperature excursion and quantization errors of a closed-loop feedback compensation device through a multi-bit current segmentation network; performing bias stabilization processing on the signal, and converting the signal into a differential voltage signal for suppressing common-mode noise through a precision differential resistance network; and finally, low-distortion differential analog audio signals capable of directly driving a load are output through low-distortion gain, impedance conversion and power driving, so that low-distortion, high-signal-to-noise-ratio and wide-dynamic-range output in a full-volume range is realized.

A system for latency-aware orchestration and performance optimization in AI-driven telephone communication, consisting of: a speech capture unit configured to capture an analog audio signal from a telephone interface and convert the analog audio signal into a digital audio signal stream; a feature extraction unit that is operationally coupled with the speech acquisition unit and is configured to generate a feature representation of the digital audio signal stream through spectral decomposition, noise reduction, and temporal segmentation; an AI inference processor communicatively connected to the feature extraction unit, configured to run one or more AI models for automatic speech recognition, natural language understanding, and emotion recognition on the feature representation to generate intermediate results for inference; a latency orchestration controller coupled to the AI ​​inference processor, wherein the latency orchestration controller is configured to monitor latency across multiple processing stages, predict cumulative delay propagation using a hybrid latency estimation model, and orchestrate the execution scheduling of the AI ​​inference processor based on the predicted latency deviation; a performance optimization unit coupled with the latency orchestration controller and configured to dynamically adjust computational accuracy, inference batch size, and feature processing resolution based on latency thresholds and quality constraints set by the latency orchestration controller; and a transmission synchronization array configured to time-align the processed output generated by the AI ​​inference processor and transmit it to a remote communication node, with the transmission synchronization array maintaining deterministic time coordination between successive packets and the orchestrated inference results.

The invention relates to the field of audio signal processing, in particular to a multi-channel audio signal hybrid coding method and device and a storage medium, and the method comprises the steps: collecting digital audio signals through a plurality of audio input channels, carrying out the anti-aliasing filtering of the collected digital audio signals, so as to obtain audio signals in a 48kHz / 16bit format, and outputting the audio signals in the 48kHz / 16bit format. An audio signal in a 48kHz / 16bit format is processed through a loudness weighted average method, the audio signal after loudness weighted average is processed through a frequency domain mixing technology to obtain a mixed signal, and dynamic range compression is performed on the mixed signal based on calculation of an auditory masking curve to obtain a compressed signal. And carrying out hybrid coding on the compressed signal to obtain a compressed data stream, and packaging and outputting the compressed data stream. Through dynamic mixing and compression processing, the problem of signal distortion caused by a traditional linear superposition method is effectively solved, reasonable superposition of all paths of signals is ensured through dynamic mixing, and level overload caused by simple addition is avoided.

The invention provides a data missing compensation method, system and device for a digital audio conference system and a medium, and the method comprises the steps: obtaining a real-time audio data stream in the digital audio conference system, inputting the real-time audio data stream into a dynamic cache region through a dynamic cache technology, and obtaining the data of the dynamic cache region; missing type detection is carried out on sampling point time sequence intervals of the dynamic cache region data, missing type marks are obtained, and the missing type marks comprise single-point missing marks and / or whole-frame missing marks; determining a compensation strategy according to the missing type mark, and compensating the digital audio signal based on the compensation strategy to obtain compensation data, the compensation data comprising a single-point compensation value and / or frame compensation data; and mapping the compensation data to an original audio time sequence position in the real-time audio data stream for interpolation to obtain a continuous audio output stream. By adopting the method, the phase continuity of the audio waveform can be kept, and the call quality and auditory experience in a complex transmission scene can be improved.

A computer assisted method for classifying digital audio files based on features of a digital audio signal comprised in the file, comprising: storing the audio file in a digital memory; determining a portion (p) of drop of the audio file, for example the portion with the highest subjective loudness; classifying said audio file based on features of said drop.

The invention relates to a wireless broadcast connection system and method and a method for automatically configuring an emergency alarm area. The system comprises a wireless broadcast receiving device and an intelligent terminal, the wireless broadcast receiving device is configured to be used for receiving broadcast signals and wirelessly transmitting broadcast audio streams to the intelligent terminal, and the wireless broadcast receiving device comprises a broadcast receiving processing unit, a wireless communication module and a digital signal processing unit. The digital signal processing unit is connected with the broadcast receiving and processing unit and the wireless communication module, and is used for encoding the digital audio signal into an audio stream in a compressed format and carrying out transmission buffer management on the audio stream; wherein the digital signal processing unit is provided with a self-adaptive transmission control module, and the self-adaptive transmission control module has a bidirectional communication capability and is used for receiving a playing buffer area state feedback signal from the intelligent terminal and dynamically adjusting the transmission rate of sending the compressed audio stream to the intelligent terminal according to the feedback signal; therefore, synchronous control of buffer areas of a sending end and a receiving end is realized.

Methods and systems are described for mitigating acoustic feedback via an open ear device. In various examples, systems or methods may receive, via a first microphone and a second microphone positioned on an open ear device, a first audio signal and a second audio signal, respectively. The first audio signal and second audio signal may be converted to a first digital audio signal and a second digital audio signal. The first digital audio signal and second digital audio signal may be independently processed by an AFC to reduce acoustic feedback. A beamformer may adjust the first digital audio signal and second digital audio signal based on a target direction to create a beamformer digital audio signal. The beamformer digital audio signal may be processed via feedforward processing to create a target audio. The target audio may be phase shifted and transmitted to a loudspeaker positioned on the open ear device.

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 utility model relates to a vehicle-mounted sound equipment testing device, which belongs to the field of vehicle-mounted audio testing, and comprises an MCU (Microprogrammed Control Unit) module, which is used for coordinating the operation of each module in equipment and ensuring the ordered transmission of data streams, communicates with an A2B module through an I2C (Inter-Integrated Circuit) protocol, and performs audio data transmission and equipment state control with a PC (Personal Computer) through a USB (Universal Serial Bus) interface; the DSP module is used for receiving input signals from the optical fiber interface and the audio analog interface, processing the received audio signals and transmitting the processed signals to the A2B module and the MCU module; the FLASH memory is used for storing data required by equipment operation; and the CAN module is used for enabling the MCU module to communicate with a CAN / CAN-FD bus of the vehicle, receiving a vehicle control signal or sending debugging data to the vehicle. The system is suitable for the fields of vehicle-mounted audio debugging, audio testing and data communication, and can simultaneously acquire and synchronously process control signals and digital audio signals, thereby improving the debugging efficiency and reducing the development period.

A watermark-based audio processing method for use in the audio player including a receiver, an analog-to-digital converter, a processor, a digital-to-analog converter and a transmitter includes: the receiver receiving an audio input signal, the analog-to-digital converter converting the audio input signal into a digital input signal including a keyword, the processor detecting an energy of the digital input signal, if the energy exceeds a threshold, the processor determining whether a watermark is included in the digital input signal, if not, the processor embedding the watermark into a predefined digital audio signal to generate a digital output signal, then the digital-to-analog converter converting the digital output signal into an audio output signal, and finally, the transmitter outputting the audio output signal for playback.

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.

A system is provided.SOLUTION: A system, comprising: means for converting a user's speech into a digital signal; means for transmitting the digital signal to a server; means for converting the speech digital signal into character string text on the server; means for translating the converted text into a specified language in real time; means for converting the translated text into speech data that retains characteristics of the original speech; means for transmitting the speech data to a terminal; and means for playing the speech data on the terminal.SELECTED DRAWING: Figure 1

The utility model provides an audio interaction host, and relates to the technical field of audio equipment, the audio interaction host of one embodiment comprises a processor, and an analog audio signal input device, a digital audio signal input device and an audio output device which are in communication connection with the processor; the analog audio signal input equipment is used for acquiring analog audio signals, converting the acquired analog audio signals into first digital audio signals and outputting the first digital audio signals to the processor; the digital audio signal input device is used for obtaining a second digital audio signal and outputting the obtained second digital audio signal to the processor, and the processor is used for processing the accessed digital audio signal, obtaining a processed digital audio signal and outputting the obtained processed digital audio signal to the audio output device; and the audio output equipment is used for converting the received processed digital audio signal into an analog audio signal and outputting the analog audio signal. According to the scheme of the embodiment, the interaction performance of the audio interaction equipment is improved.

There is provided an audio signal signature module (500) comprising: an input (502) configured to receive a digital audio signal (121); and a block divider (130) configured to divide the received digital audio signal (121) into a sequence of audio blocks (131) having a block length. A signature unit (140) generates a signature (144) associated with the current audio block (131) by applying a private key (142) to the extracted audio feature (151). The combined information is provided as signed authentication information (141) and stored in a predetermined storage location in a memory (600). The predetermined storage location may be used as authentication access information (162) and a current audio block (131) embedded by an information embedder. An audio output (190) provides an authenticable audio signal (501) comprising a sequence of audio blocks with embedded authentication access information (162). A block divider (130) adjusts the block length of the audio block in accordance with the size of authentication access information to be embedded in the audio block.

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.

An apparatus, data structure and computer-implemented method for digital content processing, where the method comprises providing (202) a first data set, where the first data set comprises elements, providing (204) a second data set, where the second data set comprises elements, where the digital content of a respective one of the elements comprises a digital image, the method includes generating (206) a first set of descriptions, such as video images, radar images, LiDAR images, ultrasound images, motion images or thermal images, or wherein the content of respective ones of the elements includes digital audio signals, in particular using a data-to-text model, where the first set includes element-aspect descriptions of elements of a first data set, where the first set includes element-aspect descriptions of the elements of the first data set, and where the first set includes element-aspect descriptions of the elements of the second data set. Wherein a description of a respective element of the first dataset is determined from the content of the respective element of the first dataset, in particular generated (208) with a data-to-text model, a second set of descriptions wherein the second set comprises an element-aspect description of an element of the second dataset.

The invention discloses an electric lute audio processing system and processing method and an intelligent electric lute, and relates to the technical field of electric lutes, the electric lute audio processing system comprises a pickup unit, a main control unit, a DSP sound effect processing module used for sound effect processing and an output module used for outputting digital audio signals, an analog electric signal acquired by the pickup unit is amplified and then sent to the main control unit; the analog electric signal is converted into a digital signal through the analog-to-digital converter and then is sent to the DSP sound effect processing module; the DSP sound effect processing module is used for performing timbre processing and switching on the collected original sound signals; the processed digital audio signal is output through the output unit; multiple functions of Bluetooth accompaniment, multi-timbre simulation, a tuner, an equalizer, power amplification and charging management are integrated into one body, the integration degree is high, and the functions are rich. Various digital audio processing units are combined with a traditional lute structure, and the application scene and expressive force of the musical instrument are expanded.

The invention provides a hearing aid device, method and device with a self-adaptive hearing compensation function, the hearing aid device comprises a wireless communication module, a main control chip and an audio output module, and a pre-training hearing compensation model is built in the main control chip. A full-band continuous personalized frequency response gain curve with gain amplitude positively correlated with hearing loss is output through model reasoning, an audio output module performs full-band frequency-point-by-frequency balanced linear gain compensation on an original digital audio signal played by the device based on the curve, the audio dynamic range is not compressed in the whole process, the original spectrum structure is not changed, and the gain amplitude of the original digital audio signal is not changed. And the high-fidelity tone quality is kept. Different from a traditional hearing aid and a conventional audio playing device, only for audio processing inside the device, pickup and noise reduction are not needed, hearing features of a hearing impaired user can be accurately adapted, clear and balanced listening can be achieved, tone quality distortion can be avoided, and compensation precision, use comfort and scene universality are considered.

The present application relates to the technical field of hearing aids, in particular to a method and system for enhancing the audio clarity directivity of a hearing aid in two channels, comprising: sending a detection signal to identify a mobile phone power supply protocol, receiving and identifying a UAC protocol, outputting corresponding safe voltage and current according to the type of mobile phone power supply protocol, triggering three-level protection, identifying the current audio output mode, sampling rate and bit depth of the mobile phone by analyzing the UAC protocol, and obtaining a standard audio stream format; receiving a scene mode instruction, controlling the dual-microphone channel according to the scene mode instruction to collect analog signals, and converting the analog signals into digital audio signals with source identification respectively; generating enhanced digital audio according to the type of digital audio signal; reading the connection state parameters fed back by the mobile phone in real time, and encapsulating the enhanced digital audio into a Bluetooth signal according to the connection state parameters and transmitting the Bluetooth signal to a Bluetooth hearing aid. The present application realizes high-precision audio enhancement by combining dynamic adaptive filtering with multi-frequency band regulation.

The present application belongs to the technical field of audio processing, and particularly relates to an audio reproduction method and system for live sound field data collection and digital signal processing. The method comprises collecting sound field data of a real acoustic space and generating a dedicated sound field parameter model; receiving an original digital audio signal and a user instruction, obtaining a dedicated sound field parameter model corresponding to the user instruction as a target dedicated sound field parameter model; performing convolution processing on the original digital audio signal and the target dedicated sound field parameter model to obtain a processed audio signal with enhanced spatial sense; and outputting the processed audio signal after multi-channel digital amplification, thereby realizing audio reproduction of the acoustic response characteristics of the dedicated sound field. The present application generates a dedicated sound field parameter model based on impulse response data collected in a real acoustic space, and the reproduced reverberation characteristics, spatial sense and sound image positioning are highly consistent with the target space, and the sound field restoration accuracy is significantly improved.

This invention relates to a method for modeling audio response curves in a web environment, comprising the following steps: S1: Generating frequency and gain based on screen click events, and forming control points by user input of Q values; or obtaining three parameters—frequency, gain, and Q value—input by the user to form control points; S2: Converting the frequency, gain, and Q value of the control points into parameters of a Gaussian function and generating connection points, smoothing the connection points with a Bézier curve, and then drawing the audio response curve; S3: Mapping the Gaussian function parameters to a Web Audio API and obtaining a digital audio signal through a DSP algorithm, and then converting it into an analog audio signal. By simulating the peak value and gain characteristics of the audio response curve in local regions using a Gaussian function, a natural transition of the audio response curve in local regions is ensured; by using a Bézier curve to smoothly connect the Gaussian curves, a highly accurate and sufficiently smooth audio response curve is obtained.

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.

In various applications, the system provides a method for processing audio signals, including: receiving, by a processor, a digital audio signal from a recorded audio file; analyzing, by the processor, the digital audio signal to identify pitch distortion caused by changes in momentary sound level; determining, by the processor, an amount of compensation of the audio signal to correct the identified pitch distortion; dynamically adjusting, by the processor, the digital audio signal by the compensation amount to correct the identified pitch distortion; and outputting, by the processor, the digital audio signal to an audio transducer device of a listener to improve a listening experience for the listener of the recorded audio file.

The invention discloses a UPS fan fault early warning method and system, and belongs to the technical field of uninterruptible power supplies. The method comprises the following steps: S1, audio signal acquisition: carrying out initial configuration on an early warning system, and then acquiring an audio signal of a UPS fan during operation in real time through a microphone array arranged in a UPS case; s2, audio signal preprocessing: performing amplification, filtering and analog-to-digital conversion processing on the fan audio signal to obtain a digital audio signal; s3, fault intelligent identification: inputting the preprocessed digital audio signal into a pre-trained one-dimensional convolutional neural network model for real-time analysis, and identifying an early fault type of the fan to obtain a fault prediction result; and S4, early warning execution: according to a fault prediction result, generating and executing corresponding local and remote early warning actions. According to the method, the strong feature learning capability of the 1D-CNN model is used, early weak fault features can be accurately recognized from complex background sounds, and the diagnosis accuracy is far higher than that of a traditional threshold value judgment method.

The invention is suitable for the technical field of electronic microphone audio signal processing, and provides an electronic microphone audio signal processing system and method, and the system comprises a server which is connected with a signal collection module, a digital signal processing module, a digital-to-analog conversion module, and an output module. The signal acquisition module comprises a sound sensor, an analog front end and an analog-to-digital converter; the sound sensor adopts an MEMS microphone array and converts sound wave pressure change into an analog electric signal; a pre-amplifier, a biasing circuit and an anti-aliasing filter are arranged in the analog front end; the pre-amplifier amplifies an electric signal output by the microphone; the anti-aliasing filter is used for filtering high-frequency components higher than the Nyquist frequency before analog-digital conversion, and aliasing distortion is prevented. According to the invention, digital audio signals can be accurately processed in real time, echoes are effectively eliminated, noise is suppressed, the volume is stabilized, the voice definition and the tone quality are improved, and the audio signal processing quality of the electronic microphone is greatly improved.

A device, a datastructure, and a computer implemented method for digital content processing. The method includes providing a first dataset; providing a second dataset; wherein a digital content of a respective element of the elements of the first and second datasets include a digital image or a digital audio signal; generating, with a data-to-text model, a first set of descriptions, wherein the first set comprises an element-wise description of the elements of the first dataset, wherein the description of the respective element of the first dataset is determined depending on the content of the respective element of the first dataset; generating, with the data-to-text model, a second set of descriptions, wherein the second set comprises an element-wise description of the elements of the second dataset, wherein the description of the respective element of the second dataset is determined depending on the content of the respective element of the second dataset.

The invention relates to a virtual bass enhancement method and device. The method comprises the following steps: acquiring a digital audio signal; filtering the digital audio signal to obtain a low-frequency signal; performing amplitude regulation on the low-frequency signal to obtain a signal after amplitude regulation; performing noise extraction on the signal after amplitude regulation to obtain a noise reference signal; generating a harmonic enhancement signal based on the signal after amplitude regulation and the noise reference signal; and adjusting the harmonic intensity of the harmonic enhancement signal to obtain an enhanced audio signal.

The invention discloses a station group article intelligent broadcasting system and method based on multi-mode emotional speech synthesis, and relates to the technical field of speech synthesis, and the method comprises the steps: extracting paragraph-level semantic and emotional characteristics, carrying out the dimension mapping of the semantic and emotional characteristics, and generating an emotional compensation parameter set; based on the emotion compensation parameter set, hidden layer state features generated in the preorder text paragraph rhythm prediction stage are stored according to paragraph granularity; when rhythm prediction is executed on a current text paragraph, backtracking and introducing corresponding preorder paragraph hidden layer state features, participating in rhythm prediction of the current paragraph, and generating a cross-paragraph consistent rhythm prediction signal; and performing cross-modal alignment fusion on the rhythm prediction signal and the emotion compensation parameter set to generate an acoustic feature prediction result, converting the acoustic feature prediction result into a Mel spectrum feature, and generating a corresponding digital audio signal stream, thereby improving the voice naturalness and emotion consistency of long text automatic broadcast.

There is provided a microphone (100) comprising: a microphone head (110) adapted to capture an audio signal; an AD converter (120) configured to convert the captured audio signal (111) into a digital signal (121); a block generator (130) configured to divide the digital audio signal (121) into a plurality of audio blocks (131); a private key signature unit (140) configured to generate at least one signature (141) based on at least one audio block of the plurality of audio blocks (131) using a private key (142) associated with the microphone (100); and an audio output (190) configured to output the at least one audio block (131) and the generated at least one signature (151).

An electronic device outputting a sound according to embodiments of the disclosure may comprise memory storing at least one instruction, a processor executing the at least one instruction, at least one speaker, and an audio processing module processing the input audio signal and outputting the processed input audio signal to the at least one speaker. The audio processing module may include a digital amplifier amplifying a digital audio signal, a digital-to-analog converter (DAC) converting the digital audio signal into an analog audio signal, an analog amplifier amplifying the analog audio signal, a boost amplifier boosting the analog audio signal to a reference high voltage or more, and at least one noise gate removing noise included in the analog audio signal. The digital amplifier may apply a positive (+) digital gain to the digital audio signal. The analog amplifier may apply a negative (−) analog gain to the analog audio signal.