268 results about "Digital audio signals" patented technology

The invention relates to the technical field of audio signal processing, and discloses a high-sampling-rate audio analysis optimization method and system, a storage medium and equipment, and the method comprises the steps: upgrading an FFTW (Fast Fourier Transform) library, and building an optimized memory management mechanism; carrying out adaptive framing processing on the input audio data, and optimizing the data access efficiency by adopting a multi-stage cache system; the calculation precision of the FFT fast Fourier transform is dynamically adjusted, and the calculation of the FFT fast Fourier transform is optimized by pre-calculating and caching an FFTW plan; performing clock synchronization optimization and jitter elimination on the COAX digital audio signal; establishing a multi-dimensional result cache matrix and an intelligent multiplexing decision tree, and reducing repeated operation through intelligent cache prediction and similarity calculation; system performance indexes are monitored in real time, adaptive parameter adjustment is realized based on an optimization objective function, audio quality and system stability are monitored in combination with subjective and objective evaluation mechanisms, repeated operation is remarkably reduced, and calculation resource allocation is optimized on the premise of ensuring sound quality.

The invention relates to the technical field of artificial intelligence and digital audio signal processing, and particularly discloses a Karaoke all-in-one machine intelligent tuning method based on AI sound effect optimization, and the method comprises the steps: obtaining an original mixed audio signal sung by a user, and carrying out the preprocessing; processing the preprocessed audio by using a multi-source audio separation model based on a deep neural network, and extracting a human voice track and accompaniment components; extracting time-frequency domain features from the human voice track, and performing quantitative evaluation on the voice tension degree of the user in combination with the trained voice state recognition model; dynamically adjusting an equalizer and a dynamic range compression parameter according to an evaluation result to realize adaptive sound effect optimization; synthesizing the optimized voice and accompaniment and outputting the synthesized voice and accompaniment to monitoring equipment; and finally, establishing a personalized user behavior feature vector by collecting user singing performance and historical tuning data, and performing online fine tuning on a system strategy based on a deep Q network reinforcement learning mechanism to continuously optimize a tuning effect.

The invention relates to the technical field of audio processing, and discloses a wireless audio data transmission and reading method and audio playing equipment, and the method comprises the steps: carrying out the signal processing of an input original audio signal, obtaining digital audio data, and building a waveform equation; performing parameter optimization to obtain an optimal waveform equation parameter; creating a signal processor and processing to obtain an optimized audio signal; performing data division to obtain a plurality of audio data packets, and solving the transmission optimization model to obtain a transmission scheduling scheme; performing coding processing to obtain a coded audio stream, and performing multi-layer mapping to obtain a multi-layer OFDM (Orthogonal Frequency Division Multiplexing) symbol; the method comprises the following steps: receiving the audio data, performing signal reconstruction processing to obtain a receiving end signal, decoding the receiving end signal to obtain decoded audio data, and recombining the decoded audio data to obtain a reconstructed digital audio signal, thereby improving the wireless audio data transmission and reading performance.

The invention relates to the technical field of digital audio signal processing, and discloses an audio synchronization multipoint PTS correction method and system, a storage medium and a device, and the method comprises the steps: setting PTS calibration points at a plurality of key nodes on an audio processing link of a master device and a slave device, and carrying out the systematic initialization of an initial PTS value of each calibration point; a clock synchronization mechanism between the master device and the slave device is established, and dynamic calibration of a slave device clock is achieved by periodically interacting time synchronization signals and calculating clock offset; performing multi-stage dynamic adjustment and compensation on the PTS value of the corresponding calibration point according to the network transmission delay fluctuation of the audio frame and the received actual coding duration and time deviation; the filling state and the PTS deviation value of the playing buffer area are monitored in real time, audio output synchronization between the master device and the slave device is maintained through a playing speed fine tuning strategy, optimization is carried out aiming at a lossy coding scene, and the limitation of a traditional single-point synchronization method is effectively solved.

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.

The invention discloses an interphone implementation system based on voice automatic recognition, and relates to the technical field of voice recognition, and the system comprises the steps: collecting a voice signal of a user through an interphone microphone, and converting the voice signal into an analog electric signal; converting the analog electric signal into a digital audio signal by using an AAC audio coding algorithm, and carrying out coding processing; voice data packets lost in the transmission process are repaired in a data redundancy mode by using a forward error correction technology; framing the audio signal to be processed by using an overlapped sliding window method; detecting a voice area endpoint in the voice frame by using a double-threshold judgment method; training a speech recognition hidden Markov model through the speech area data and the corresponding statements in the information base; and obtaining statement information corresponding to the voice signal of the user by using a Viterbi algorithm. By arranging the audio repairing module, the endpoint detection module, the model training module and the voice recognition module, the voice quality is improved, the call distance is increased, and thus the working efficiency of a user is improved.

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 technical field of audio processing, in particular to an airborne high-reliability intelligent voice call squelch reduction method, which comprises the following steps: performing ADC conversion on an acquired analog audio signal to obtain a digital audio signal; carrying out framing and FFT conversion on the digital audio signal, carrying out phase optimization, calculating a dynamic spectrum entropy weight, combining an auditory masking compensation factor, carrying out spectrum synthesis, and carrying out IFFT conversion and overlap elimination to obtain an audio signal after sound mixing processing; noise reduction processing is carried out on the audio signal after sound mixing processing through multi-mode feature extraction, and an audio signal after noise reduction processing is obtained; performing squelch judgment of Bayesian decision optimization on the audio signal subjected to noise reduction processing through energy feature extraction to obtain an audio signal subjected to squelch processing; performing automatic gain processing on the audio signal after the squelch processing to obtain an audio signal after automatic gain; and performing DAC conversion on the audio signal after automatic gain, and outputting a noise-reduced analog audio signal.

The invention discloses a bone conduction earphone and a bone conduction earphone system thereof, and relates to the technical field of bone conduction earphones, and the bone conduction earphone and the bone conduction earphone system thereof comprise a power supply module, a sound processing module and a connection module used for being connected with external equipment, the sound processing module comprises an audio input unit for receiving analog or digital audio signals from a microphone or other audio sources, the signal processing unit is used for processing the received audio signals; the vibration transmission unit is used for converting the processed audio signals into vibration signals and transmitting the vibration signals to auditory nerves of a user through skull bones; the signal processing unit comprises a noise reduction algorithm for reducing noise and a sound leakage prevention algorithm for reducing sound leakage of the bone conduction earphone; the noise reduction function and the sound leakage prevention function are achieved through the noise reduction algorithm and the sound leakage prevention algorithm respectively, and the noise reduction performance and the sound leakage prevention performance of the bone conduction earphone are improved.

There is provided a method for speech enhancement of a digital audio signal, wherein the input digital audio signal is in a digital waveform representation, the method comprising the steps of: a) based on a first trained artificial intelligence, Al, model creating a denoised digital audio signal in the waveform representation by removing energy from the input digital audio signal; b) based on a second trained Al model mapping the denoised digital audio signal in the waveform representation into a lower dimensional signal representation; c) based on a third trained Al model determining an enhanced digital audio signal in the lower dimensional signal representation such that the perceptual quality of the voice is elevated; and d) based on the second trained Al model mapping the enhanced digital audio signal in the lower dimensional signal representation back to the waveform representation; wherein each of the first, second, and third trained Al model is based on a respective Al model comprising a plurality of model parameters which are determined based on a training process.

Digital audio signal processing techniques used to provide an acoustic transparency function in a pair of headphones. A number of transparency filters can be computed at once, using optimization techniques or using a closed form solution, which are based on multiple re-seatings of the headphones and that are as a result robust for a population of wearers. In another embodiment, a transparency hearing filter of a headphone is computed by an adaptive system that takes into consideration the changing acoustic to electrical path between an earpiece speaker and an interior microphone of that headphone while worn by a user. Other embodiments are also described and claimed.

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.

The invention discloses a sound console digital audio signal processing method and system, a product and a medium, and relates to the field of voice or sound processing, and the method comprises the steps: obtaining a plurality of paths of audio input signals, carrying out the time-frequency domain conversion, analyzing the frequency distribution rule, extracting the fundamental frequency and overtone structure features, and building a tone feature fingerprint database. Then obtaining a preset standard audio signal, extracting a multi-dimensional feature vector, and matching the multi-dimensional feature vector with the fingerprint database to obtain a feature parameter group; on the basis, a coordination index system including the overall sound coordination degree, the sound level definition and the spatial sound image balance degree is constructed. A coordination index of an input signal is calculated in real time and compared with a standard signal, and an adjustment value is obtained. And finally, in combination with fingerprint database matching features, calculating a target audio processing parameter combination containing balanced, dynamic and spatial parameters. By implementing the method, the sound quality of multiple paths of audio signals 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 provides an audio noise reduction method and device, electronic equipment and a storage medium, and belongs to the technical field of audio signal processing, and the method comprises the steps: extracting audio features in an original audio signal, and dividing the original audio signal into a plurality of audio clips based on the audio features; adding a position label and a noise label to each audio clip; performing real-time sampling on all the audio clips to obtain a plurality of digital audio signal clips, and performing pre-emphasis processing on each digital audio signal clip based on a noise tag to obtain a first audio; processing the first audio based on a bimodal feature extraction algorithm and a target noise reduction model to obtain a second audio, and obtaining a denoised target audio based on the second audio and the position tag; the dual modes comprise a time domain mode and a frequency domain mode, and the target noise reduction model is a trained noise reduction model. The definition and naturalness of the audio signal can be remarkably improved, and the user experience is improved.

Embodiments of the present disclosure provide a voice wake-up method, apparatus, electronic device, and readable storage medium. The method includes: acquiring a deep feature encoded by a voice activity detection (VAD) encoding layer through a VAD decoding layer, the deep feature being extracted from a digital audio signal used for VAD encoding and voice wake-up encoding; determining whether each audio segment in the digital audio signal is a voice according to a probability that each audio frame in the digital audio signal is a voice frame, each audio segment including a plurality of audio frames; and when determining that the audio segment is the voice, indicating, by the VAD decoding layer, to input a deep feature of the audio segment into a voice wake-up decoding layer to perform voice wake-up. The embodiments of the present disclosure improve the accuracy of voice wake-up.

The invention discloses an AI voice interaction CD playing control method and equipment, and the method comprises the steps: collecting a voice signal of a user through calling a microphone of an AI voice recognition module, and carrying out the digital processing of the collected voice signal into a digital audio signal; preprocessing the digital audio signal; voice signal feature parameter extraction is carried out on the preprocessed digital audio signals, voice signal feature parameters comprise Mel-frequency cepstral coefficient feature vectors, and acoustic features of the voice signals are represented through the voice signal feature parameters; and inputting the extracted voice signal feature parameters into a pre-trained voice recognition model constructed based on a deep learning algorithm, and classifying and recognizing the input voice signal feature parameters through the voice recognition model to obtain a text control instruction for the CD playing equipment. According to the method, the voice instruction of the user can be accurately understood, playing control and music recommendation are automatically carried out according to personalized requirements and preferences of the user, and the pursuit of personalized audio experience of the user is met.

The invention relates to the field of Morse telegraph communication, and discloses an AI-driven Morse telegraph automatic reading and sending method and system, which realize automatic sending and automatic reading in the Morse telegraph communication process in the field of short-wave communication. The method is realized through a full-automatic message reading and sending process: automatically sending a message, converting a message text into a Morse code sequence, and controlling a radio station to send the Morse code sequence; automatically copying and reporting, sampling a Morse audio signal output by a short-wave radio station, and decoding, reasoning and predicting a message text based on an AI model; in communication control, a half-duplex communication protocol is adopted to coordinate a full-automatic transceiving process. The invention also designs an AI-driven Morse telegraph automatic reading and transmitting system, which comprises a central processing module for executing half-duplex communication protocol logic; the telegraph sending module is used for controlling the PTT interface of the radio station to generate a pulse for sending a telegraph; the telegraph receiving module is used for converting the Morse analog audio signal into a digital audio signal; and the AI decoding module is used for reasoning and predicting a message text through a training model.

The invention discloses an audio power amplifier circuit, an audio driving method and electronic equipment, and belongs to the technical field of audio driving. Wherein the audio power amplifier circuit comprises an audio driving circuit and a loudspeaker, the audio driving circuit is connected with the loudspeaker, the audio driving circuit comprises a power amplification module, and the power amplification module is connected with the loudspeaker; the voltage regulation module is connected with the power amplification module; and the envelope tracking module is connected with the voltage regulating module, and the envelope tracking module is used for receiving the digital audio signal and acquiring a signal parameter of the digital audio signal, and controlling the voltage regulating module to increase or decrease a voltage value of a power supply signal for supplying power to the power amplification module based on the signal parameter.

The invention belongs to the field of interphones, and provides a mobile phone auxiliary type intelligent interphone and an implementation method thereof, and the mobile phone auxiliary type intelligent interphone comprises a mobile phone end assembly which comprises a central processing unit, an audio processing chip, a first connection module, a loudspeaker and a microphone; the interphone end assembly comprises a main control chip, a wireless communication module, a second connection module and a power management module; a radio frequency transceiver chip is arranged in the wireless communication module, data transmission is carried out between the second connection module and the first connection module, and the audio processing chip adopts a CNN noise reduction model to process digital audio signals; according to the method, the digital audio signals are processed through the CNN noise reduction model, various complex background noises can be accurately identified and removed, clear voice signals can be restored, the voice definition and intelligibility are greatly improved, it is ensured that both communication parties can accurately transmit information in a noisy environment, and the user experience is improved. The problem of poor voice quality caused by noise interference in a traditional matching mode is effectively solved.

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.

Device and computer-implemented method for generating an anomaly, wherein a first representation (206) of a digital input (106), in particular a digital time series, a digital audio signal, or a digital image, preferably a video image, a radar image, a LiDAR image, an ultrasound image, an image from a motion sensor, or an infrared image, in a state space (208) is mapped to the digital input (106) by a first model (202) configured to map the first representation to the digital input (106), and the digital input (106) is mapped to a prediction (212) for content contained in the digital input (106) by a second model (204) configured to map the digital input (106) to the prediction (212), wherein a second representation in the state space (208) is determined depending on a measure that characterizes the quality of the prediction (212).which worsens the prediction (212), and wherein the anomaly is generated depending on the second representation.

Embodiments of the present disclosure provide a voice wake-up method, apparatus, electronic device, and readable storage medium. The method includes: acquiring a deep feature encoded by a voice activity detection (VAD) encoding layer through a VAD decoding layer, the deep feature being extracted from a digital audio signal used for VAD encoding and voice wake-up encoding; determining whether each audio segment in the digital audio signal is a voice according to a probability that each audio frame in the digital audio signal is a voice frame, each audio segment including a plurality of audio frames; and when determining that the audio segment is the voice, indicating, by the VAD decoding layer, to input a deep feature of the audio segment into a voice wake-up decoding layer to perform voice wake-up. The embodiments of the present disclosure improve the accuracy of voice wake-up.

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.

An audio processing apparatus having an echo canceling mechanism is provided. An audio transmission circuit receives an input digital audio signal from an external device. A DAC circuit performs conversion according to the input digital audio signal to generate an output analog audio signal to an external display device for power amplification and playback. An ADC circuit performs analog-to-digital conversion on an amplified signal generated by a power amplification circuit and a received audio signal generated by an audio receiving device to generate an amplified digital signal and a received digital audio signal. A processor implements an echo canceling algorithm to perform echo cancellation according to the amplified digital signal and the received digital audio signal to generate an output digital audio signal to be transmitted to the external device through the audio transmission circuit.

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.

There is described a computer-implemented method for obtaining M filters of a digital audio signal defined in the time domain, where M is an integer greater than or equal to 1, each filter being associated with a respective sound source, referred to as a secondary source, emitting a sound pressure field, the invention relates to a method for operating a sound pressure field that minimizes a reflected sound pressure field of a set of sound sources, referred to as primary sources, located in a room, comprising:-for each primary sound source and secondary sound source, obtaining (S601) an impulse response of reflected sound pressure at N respective different positions in the room, where all primary and secondary sound sources are placed in an operating position, the N positions are the same for all sound sources; determining (S602) a set of M filters by solving a regularization optimization problem based on said set of impulse responses and formulated in the time domain, the optimization problem being defined as (a) minimizing the norm of the sum of impulse responses at N positions; and (b) introducing a non-zero action delay for each filter; each secondary source is combined with a primary source referred to as an associated primary source.