466 results about "Digital audio" patented technology

A collaborative cloud-based digital music composition, performance and production studio system network employing machine-intelligent digital audio workstation (DAW) systems that are supported by cloud-based automated music composition, performance, production and publishing services. Such services enable improved workflows and enhanced productivity, while ensuring that the music IP rights of all parties involved in the AI-assisted music creation process are respected and responsibly managed in the best economic interests of individual artists, performers, producers, publishers and consumers alike. The digital music studio system network can be realized in various ways supporting the DAW systems in ways suited to different user preferences and application environments.

The invention relates to the technical field of digital audio processing, in particular to a digital audio processing method and system for sound equipment. The system comprises an audio signal acquisition module, a signal analysis module, an adaptive adjustment module, a howling suppression module and a sound effect optimization feedback module. The audio signal acquisition module acquires data such as frequency spectrum and amplitude of the sound equipment in real time. And the signal analysis module generates a signal analysis report based on a non-negative matrix factorization algorithm and a generative adversarial network algorithm. And the adaptive adjustment module automatically adjusts audio parameters through a fuzzy logic control algorithm. The howling suppression module suppresses howling noise by using an adaptive feedback cancellation algorithm. And the sound effect optimization feedback module adopts a Bezier curve to enhance the sound effect and realize frequency, phase and space positioning optimization. Through cooperative processing of audio acquisition, accurate analysis, adaptive adjustment, howling suppression and spatial sound effect optimization, the audio quality and stability are improved, and the audio spatial effect and user experience are enhanced.

The invention relates to the technical field of multi-channel audio processing, and discloses a multi-channel audio processing method and device for an ultra-low power consumption end side AI chip, and the method comprises the steps: carrying out the ADC conversion of a multi-channel analog audio signal from a microphone array, and obtaining a multi-channel digital audio data stream; performing frame segmentation and frequency domain transformation processing on the multi-channel digital audio data stream to obtain a multi-channel frequency domain feature matrix; executing variational Bayesian inference to obtain an enhanced voice signal matrix; implementing two-stage cascade processing to obtain a voice event processing result; and executing an adaptive federated learning algorithm among the distributed devices to obtain globally optimized model parameters, and deploying the globally optimized model parameters to an ultra-low power consumption end side AI chip to execute multi-channel audio processing. And a complex audio processing algorithm can be efficiently operated on an end-side AI chip with limited computing resources.

The invention belongs to the field of intelligent music generation, and relates to an intelligent music production system and method based on a model context protocol, and the system comprises a large language model, the model context protocol, and a digital audio workstation execution layer. The large language model is used for deconstructing a natural language instruction into structured operation logic and generating a semantic instruction based on the music knowledge base; the semantic instruction at least comprises a device parameter and a chord sequence; the model context protocol is used for compiling the semantic instruction into a JSON command and synchronizing working state data of the digital audio workstation in the digital audio workstation execution layer to a large language model so as to update a model context; the model context comprises track configuration, effect chain parameters and time sequence positioning information of the digital audio workstation; the digital audio workstation execution layer is used for converting the JSON command into music operation to obtain a music operation file; a man-machine cooperative music production mode is reconstructed, and the quality of music production is improved.

The invention belongs to the technical field of data processing, relates to a modular digital audio multi-channel IO control system and method for aviation communication, and aims to solve the problems of poor environmental adaptability, static synchronization parameters, calibration lag and insufficient intelligent fault-tolerant capability of a synchronization technology in the aviation field. The method comprises the steps that a scene sensing module fuses multi-domain data to generate a scene vector, synchronous parameters are generated through a look-ahead control module, clock skew of multiple IO channels is calibrated in combination with a synchronous execution module, and synchronous control is achieved; the health monitoring module quantifies the full-link health degree and dynamically adapts to the self-healing strategy through the self-healing learning module. According to the method, high-precision synchronization and high-robustness fault tolerance are balanced through perspective prediction and response type self-healing closed-loop cooperation, the system is endowed with intelligent self-adaptive evolution ability, and the synchronization performance and task reliability of the aviation audio system in a complex dynamic environment are improved.

An AI-based personality and behavior analytic and assessment system includes an input interface configured for receiving input data containing audio speech, an automatic speech recognition module configured to receive the input data, recognize the spoken words in the input data, and generate an output representing the spoken words attributed to the individual, the output including speaking moments and audio slices, a machine learning text-based feature generation pipeline configured to receive the speaking moments and generate a numerical text-based feature set, a machine learning audio-based feature generation pipeline configured to receive the audio slices and generate a numerical audio-based feature set, a machine learning inference processor configured to receive at least one of the numerical text-based feature set and the numerical audio-based feature set, develop inferences from the feature sets, and generate scores representing the probabilities for a number of personality and behavioral traits of the individual.

The invention relates to the technical field of serial deserializer design, and discloses an FPGA-based direct-current balanced digital audio and video signal serial deserializer, which comprises a serializer, a deserializer, a control configuration module, a power management module, a configuration storage module and a time sequence control module, the serializer is used for converting parallel digital audio and video data into high-speed serial data streams; the deserializer is used for detecting and repairing abnormal data in the high-speed serial data stream by using a machine learning technology, and converting the repaired high-speed serial data stream back to parallel digital audio and video data; the control configuration module is used for configuring working parameters of the serializer and the deserializer through an external interface; the time sequence control module is used for managing the time sequence relation in the system and ensuring synchronization of data flow between different modules. According to the method, the defects of a traditional deserializer in the aspects of anomaly detection and repair, signal quality evaluation and parameter optimization are overcome, and the stability and quality of high-resolution digital audio and video data transmission are improved.

The invention relates to the technical field of equipment anomaly recognition, and particularly discloses an equipment anomaly behavior recognition method and system based on audio and video streams, and the method comprises the steps: collecting equipment operation audio decibel values, preprocessing the audio decibel values to obtain digital audio streams, converting the digital audio streams to obtain a detection spectrogram, calculating a wave crest offset ratio and a wave crest number ratio, and obtaining section anomaly values; the method comprises the following steps: acquiring a video image of an equipment connection part, calculating a position and an angle offset ratio by using an algorithm to obtain a stable value of a part, judging and generating an abnormal analysis signal, further calculating an abnormal difference ratio, a stable difference ratio and an equipment abnormal ratio based on the abnormal analysis signal, drawing a change curve, and calculating an angle offset ratio, variance and the like to obtain linear values. The system comprises an audio acquisition and analysis module, a video analysis module and an abnormity judgment module. The device abnormal behavior can be effectively identified, and the device maintenance efficiency is improved.

A system for analyzing audio includes a memory configured to store known digital audio representation containing known fraudulent audio streams and a processor operably coupled to the memory. The processor receives a portion of an audio stream from an external device and produces a transcript of the portion of the audio stream. The processor then determines a timing score, an emotional score, a background score, and a content score by analyzing the portion of an audio stream and the corresponding transcript and comparing them to the known digital audio representations and transcripts. The processor then determines if the audio stream is malicious by combining the timing score, emotional score, background score, and content score to produce a combined score and comparing the combined score to a threshold. The processor notifies a user that the call may be fraudulent when the combined score is greater than the threshold.

The invention discloses an AI-driven convergence media digital audio and video content security auditing method, which comprises the following steps: acquiring audio and video contents published by a platform and user information of a publisher in real time, the audio and video contents comprising live streaming, video files, audio files and image-text contents; feature extraction is carried out on the video file, the text data, the video data and the audio data, and then fusion and vectorization processing are carried out to obtain a comprehensive feature vector; and inputting the comprehensive feature vector into an AI model for detection, determining whether illegal content exists or not, and processing according to a detection result. The AI technology is introduced, intelligent auditing of various modal contents such as videos, audios and images and texts is achieved, the auditing efficiency and accuracy are improved, meanwhile, the auditing strategy is dynamically adjusted according to different scenes and user behaviors, the diversified content auditing requirements of the convergence media platform are met, and the safety and compliance of digital audio and video contents are ensured.

The invention relates to the technical field of sound box synchronous playing, and discloses a Bluetooth earphone and sound box synchronous playing method and system supporting stereo synthesis, and the method comprises the steps: carrying out the digital sampling of an input audio signal, and obtaining high-frequency digital audio data and low-frequency digital audio data; performing scanning and stereo function identification on peripheral equipment based on a Bluetooth 5.0 protocol to obtain a target audio equipment list; distributing a main data buffer area and a standby data buffer area; distributing the high-frequency digital audio data to Bluetooth earphone equipment for transmission, distributing the low-frequency digital audio data to sound box equipment for transmission, and when detecting that clock drift among equipment exceeds a preset target value, adjusting a reading rate of a main data buffer area for clock synchronization to obtain an equipment synchronization state; and the signal gain and the equalization parameter of the equipment are dynamically adjusted, and the target synchronous playing audio is output, so that the accurate distribution of the high-frequency data and the low-frequency data is realized, and the overall playing effect of the system is improved.

The invention relates to an audio SOC chip, an audio subsystem interface module thereof, and electronic equipment, and the audio subsystem interface module comprises a plurality of downlink path units which are used for carrying out the downlink processing of at least one path of digital audio data obtained from an application processor chip and / or a digital signal processing chip through a peripheral bus interface module, the audio data is output through the audio data transmission interface module; the plurality of uplink path units are used for performing uplink processing on at least one path of digital audio data input through the audio data transmission interface module and outputting the processed data to the application processor chip and / or the digital signal processing chip through the peripheral bus interface module; and the plurality of loopback path units are used for converging the at least one path of digital audio data input through the audio data transmission interface module and the digital audio data obtained by performing downlink processing through any downlink path unit in the plurality of downlink path units, and outputting the digital audio data through the audio data transmission interface module, so that the power consumption of a chip is reduced.

A specialized audio / instrument cable with built-in digital signal processing capabilities that adds digital audio sampling capabilities that allows the user to trigger synthesized sounds or virtual musical instruments from within the cable itself to affect the sound generated from an instrument or microphone such that the cable is the only connection needed between the instrument or microphone and an output device. Using voice recognition, the specialized cable can select an audio effects chain algorithm and / or sampled sound algorithm extrapolated from a musical digital audio fingerprint (MDAF) created from a desired musician's instrument to alter the sound of the input instrument's audio.

The invention discloses a digital audio watermark processing method and device, equipment and a storage medium, and belongs to the technical field of artificial intelligence. The method comprises the following steps: receiving a watermark adding instruction, and carrying out Fourier transform on original host voice to obtain an audio signal; identifying a watermark information adding bit in the audio signal through the audio watermark model, and embedding a watermark message in the watermark information adding bit; the interfering audio signal and the audio signal carrying the watermark message are combined. Receiving the message extraction signal, and separating an interference audio signal in the combined audio signal; a watermark information adding bit in the audio signal is identified, and a watermark message is extracted from the watermark information adding bit. The invention also relates to the technical field of block chains, and the host voice is stored on a block chain network. According to the method, the watermark information adding position is automatically identified through the audio watermark model, so that the watermark information is added and extracted, the dependence on expert knowledge is remarkably reduced, and the watermark processing speed and efficiency are improved.

Provided herein are system, method, and computer program product embodiments for providing an interactive visual representation of loudspeaker sound coverage in a venue. An embodiment enables temporal, spectral, and spatial audio metering from digital audio to venue acoustics. This technology can visualize audio capabilities of one or more loudspeaker arrays using light to provide visualization grouping of large audio channel outputs. This can allow spatial information to be visually represented by mapping directional sound to light beams.

The systems and methods taught herein are generally directed to facilitating a social network of a group of performers that occurs in a distributed computing environment having remote processing devices linked through a communications network in a streaming audio / video setting, including a dynamic point referencing of an audiovisual performance for an accurate and precise selection and controlled cycling of portions of the performance. The dynamic point referencing can be used by a learning artist, for example, in analyzing or performing a portion of the work through an accurate and precise digital audio / video instructional method having the controlled cycling feature. Such systems and methods will be appreciated, for example, by musicians, dancers, and other enthusiasts of the performing arts.

The invention discloses a parallel constant-voltage digital audio power amplifier and a broadcasting system. Comprising a line input module, a line output module, a first analog-to-digital conversion circuit, a second analog-to-digital conversion circuit, a third analog-to-digital conversion circuit, a digital signal processor, a single-chip microcomputer, a digital-to-analog conversion circuit, an error comparator, a modulator, an integrating circuit, a pulse width modulation input half-bridge output stage, and a first carrier frequency selection and gain adjustment module. A second carrier frequency selection and gain adjustment module, a first LC filter, a high-frequency compensation network, a communication bus and a power switch. The technical problems that an existing multi-channel parallel digital constant-voltage power amplifier is only limited to power amplifier channels in the same power amplifier, the parallel connection mode only uses the first channel as a main power amplifier and only uses the other channels as slave power amplifiers, and certain limitation exists are solved.

The invention discloses a digital audio processing system based on a DSP, and relates to the field of digital audio processing, and the system comprises an acquisition module which is used for capturing vehicle body vibration sound waves, environment incoming noise and in-cabin original audio signals, and generating a multi-dimensional acoustic data stream through voiceprint feature extraction so as to output an original sound signal matrix; the analysis module is used for mapping the original sound signal matrix to a phase space, extracting singular attractor features of noise evolution through a chaos theory, and generating a noise dynamic model; according to the method, various sound waves can be accurately captured and converted into multi-dimensional data, deep analysis and targeted counteracting of noise are achieved on the basis, meanwhile, an acoustic adjustment scheme matched with the environment in the cabin can be generated, the acoustic environment can be dynamically optimized in combination with the physiological state of passengers, and potential noise is pre-judged in advance for suppression.

The invention discloses an audio and video transmission method and device based on multi-mode AI enhancement and dynamic end-to-end encryption, equipment and a storage medium, and relates to the technical field of digital audio and video processing and secure transmission, and the method comprises the steps: obtaining an original audio and video stream, and separating the original audio and video stream into audio data, video data and subtitle text data; performing multi-mode AI enhancement processing on the audio data, the video data and the subtitle text data to obtain enhanced audio and video data; generating an encryption decision instruction based on the real-time network state, and performing encryption processing on the enhanced audio and video data according to the encryption decision instruction to generate encrypted data with a tamper-proof watermark; and when the encrypted data passes integrity verification and copyright legality judgment, the encrypted data is transmitted through a secure transmission channel, and collaborative optimization of security and efficiency is realized through a core technology chain of multi-modal separation enhancement, dynamic encryption decision, hardware-level watermark binding and credible verification transmission.

The invention discloses a sleep assistance composite audio generation method and system based on multi-frequency brain waves and a medium, and belongs to the technical field of digital audio processing, and the technical scheme is characterized by comprising the following steps: S1, collecting a brain wave signal of a user in real time through a brain wave monitoring module; s2, dynamically selecting a target frequency band combination according to the sleep stage and setting an amplitude change rule; s4, pink noise with the frequency spectrum range of 20-200 Hz is injected into the composite audio, the energy proportion of the pink noise is 5%-15%, and the amplitude is dynamically adjusted according to the environmental sound pressure level SPLenv. The method has the effect that the limitation of traditional sleep assisting audio is solved through the technologies of multi-frequency brain wave synchronization, personalized customization, dynamic adjustment, diversified audio combination and the like.

The systems and methods taught herein are generally directed to facilitating a performing arts contest in a distributed computing environment having remote processing devices linked through a communications network in a streaming audio / video setting, including a dynamic point referencing of an audiovisual performance for an accurate and precise selection and controlled cycling of portions of the performance. The dynamic point referencing can be used by a learning artist, for example, in analyzing or performing a portion of the work through an accurate and precise digital audio / video instructional method having the controlled cycling feature. Such systems and methods will be appreciated, for example, by musicians, dancers, and other enthusiasts of the performing arts.

The disclosure describes systems and methods for improving latency for wireless speakers. The system can establish a buffer pipe between a wireless chip of a media player and a wireless chip of a wireless speaker using an inter-IC sound protocol. The system can receive samples of audio data through the media player. The system can communicate the audio data to the wireless speaker using the buffer pipe to bypass the transport layer stack of the media player and the wireless speaker. The system can cause a transmitter of the wireless speaker to provide the audio data to a digital audio converter for output to a speaker of the wireless speaker.

The invention discloses a low-delay lossless digital audio transmission method, a low-delay lossless digital audio transmission system and a storage medium, which relate to the technical field of digital audio transmission and are used for monitoring network environment data in real time and dynamically adjusting audio transmission parameters according to changes of network bandwidth Bw, network delay Dr, packet loss rate Pd and network jitter Wd. Comprising an audio transmission rate Rsv, a data packet size Spd and a redundancy coding ratio Rbf. The adaptive adjustment can ensure that the data transmission quantity is reduced and the packet loss probability is reduced when the network condition is poor, and meanwhile, the transmission efficiency is improved and the audio quality is optimized when the bandwidth condition is good. Compared with a traditional fixed transmission parameter method, lossless audio transmission can be achieved under different network conditions, and meanwhile unnecessary delay is reduced. According to the invention, intelligent compression of the audio data is realized, the data volume is reduced under the condition that the bandwidth is limited, the network burden is reduced, and the stability and integrity of audio transmission in a low-bandwidth environment are ensured.

The broken lines in the drawings illustrate the portions of the design that form no part of the claim; the characteristic features of the design lie in the combination of the shape and pattern; FIG. 1.1 is a front, right side, top perspective view of the digital audio player showing a new design; FIG. 1.2 is a rear, left side, top perspective view thereof; FIG. 1.3 is a front view thereof; FIG. 1.4 is a rear view thereof; FIG. 1.5 is a left side view thereof; FIG. 1.6 is a right side view thereof; FIG. 1.7 is a top view thereof; and FIG. 1.8 is a bottom view thereof.

Systems and methods for digital audio measurement such as perceived loudness. An example of the method includes dividing, by a processor, a digital audio file into a plurality of blocks in a time sequence at a first time length; determining, by the processor, a respective Loudness Units relative to Full Scale (LUFS) for each block; determining, by the processor, a difference between the LUFS of each block with a reference value; and generating, by the processor, an indicator from one or more of the differences of the plurality of blocks within a second time length.

The invention relates to an analog-digital hybrid SRAM (Static Random Access Memory) memory audio noise reduction method and system, and the method comprises the following steps: carrying out the differential sampling quantization processing of an input audio signal, and obtaining a multi-bit-width digital audio code stream; inputting the multi-bit-width digital audio code stream into a preset SRAM (Static Random Access Memory) memory chip to carry out mixed mode write-in operation to obtain an analog-digital mixed memory matrix; performing frequency domain feature extraction on the analog-digital hybrid storage matrix to obtain frequency domain extraction features; performing noise identification and suppression on the audio signal based on the frequency domain extraction feature to obtain a time-varying noise suppression coefficient; and performing audio signal recovery processing on the time-varying noise suppression coefficient to obtain a denoised output audio signal, thereby solving the technical problem that power consumption and hardware cost are greatly increased due to the fact that a traditional digital signal processing method generally depends on a high-precision analog-to-digital converter and a high-performance processor.

The present application provides a method, system, device and medium for generating audio and video of a digital human, the method comprising: first, obtaining text information and an initial image; determining the emotion category corresponding to the text information based on an emotion analysis model; identifying the face area in the initial image to obtain the gender category corresponding to the digital human in the initial image; generating audio data corresponding to the digital human based on the emotion category and the gender category; generating an initial video of the digital human based on the audio data and the initial image; inputting the initial video of the digital human into a lip alignment model to obtain an output video; aligning the mouth shape of the digital human in the output video with the audio data. Using the emotion analysis model and image recognition technology, the emotion category of the text information and the gender category of the digital human in the initial image can be accurately identified, and a digital human video with emotional expression can be generated.

The invention belongs to the field of artificial intelligence, particularly relates to an intelligent sound field adaptive system of digital professional sound equipment, and aims to solve the problems of inaccurate sound field regulation and control, response lag, dependence on manual tuning and the like in a complex acoustic environment. The system comprises a sound field sensing module, an acoustic modeling and analysis module, a self-adaptive sound field regulation and control engine, a multi-channel digital audio processing unit and a feedback optimization module, and dynamic sound field modeling, real-time audio processing and environment self-adaptive regulation and control are realized through distributed sensing, hybrid modeling, multi-target optimization and closed-loop feedback. The system supports rapid re-calibration, multi-scene memory and user preference learning, ensures voice clarity and music fidelity, balances full-field hearing consistency, and significantly reduces manual intervention requirements.

The utility model provides a digital audio playing platform, which comprises a master control module, a digital audio processing module, a digital audio playing module, a digital audio playing module and a digital audio playing module, the digital audio processing module is used for decoding, sampling, format conversion and data processing of digital audio files; the digital audio output module is used for outputting digital audio data in corresponding formats through a plurality of digital audio output channels; the left sound channel audio circuit module and the right sound channel audio circuit module are two independent working modules with consistent circuit structures, are respectively used for conversion, adjustment and gain control of audio signals in a left sound channel and a right sound channel, and output multiple paths of analog audio signals to the analog audio output module; and the analog audio output module is used for realizing balanced and single-ended double-channel earphone output. According to the utility model, by designing the circuit module and adopting high-performance devices, the effects of lossless high-fidelity tone quality, abundant audio output interfaces, multi-tone selection and more convenient operation mode are achieved.

The invention relates to the technical field of hearing-aid communication terminals, and discloses a hearing-aid operation method, system and terminal for communication data for old people. The hearing-aid operation method is applied to a terminal host integrated with a hearing-aid unit, an audio unit, a communication unit, a positioning unit, an AI processing unit and a wireless connection unit, and specifically comprises the following steps: S101, obtaining a user start request, synchronously collecting an environment simulation sound signal through a terminal microphone array, and converting the environment simulation sound signal into a high-resolution digital audio stream through an ADC module; according to the invention, a hardware end deeply integrates hearing aid, communication and an AI processing unit to integrate an environment sound enhancement function and a family interaction function into the same data processing stream, and a self-adaptive multi-filtering technology based on scene perception is adopted, so that noise reduction, frequency band enhancement and signal shaping parameters can be dynamically adjusted according to a real-time acoustic environment; and speech definition and auditory safety in a complex environment are obviously improved.