346 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 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.

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 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.

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 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 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 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 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.

A system for platform-independent visualization of audio content, in particular audio tracks utilizing a central computer system in communication with user devices via a computer network. The central system utilizes various algorithms to identify spoken content from audio tracks and identifies “great moments” and / or selects visual assets associated with the identified content. Audio tracks, for example Podcasts, may be segmented into topical audio segments based upon themes or topics, with segments from disparate podcasts combined into a single listening experience, based upon certain criteria, e.g., topics, themes, keywords, and the like.

The embodiment of the invention discloses an audio transcriptor, an audio processing method and device and a readable storage medium, and the audio transcriptor is used for an intelligent terminal. The system comprises a DSP digital audio module, an SPI recording audio processing module, a TF storage module and at least one external microphone. The DSP digital audio module is provided with a first DSP interface, a second DSP interface, a third DSP interface and a fourth DSP interface; the SPI recording audio processing module is provided with a first SPI interface, a second SPI interface and at least two microphone input interfaces; the first DSP interface is used for being connected with an intelligent terminal, the second DSP interface is used for being connected with an earphone and / or a microphone, the third DSP interface is connected with the first SPI interface, and the second SPI interface is connected with the TF storage module; the at least two microphone input interfaces are used for connecting microphones; wherein the signal transmission frequency bands of the at least two microphone input interfaces and the third DSP interface are different.

A projector, such as a toy projector, in which stored content or information is delivered both as digital images and digital audio content from an interchangeable character which is interchangeably attached to the projector. The stored content or information can be specific to the content and / or theme being displayed and is delivered through an electrical connection to the projector.

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.

Multimodal digital audio generation techniques are described that leverage multimodal inputs such as a digital image and text to generate digital audio using machine learning. In one or more examples, a digital image and text are received. Image semantic information is extracted from the digital image using machine learning. Digital audio is generated using generative machine learning based on the text and the image semantic information. The digital audio is then rendered and output by a digital audio output device.

A system and a method are disclosed for keyword spotting in a digital audio stream. The method includes processing the digital audio stream to extract a feature matrix; applying a set of one-dimensional temporal convolutions to the feature matrix to obtain a first convolved feature matrix; transposing time and frequency dimensions of the feature matrix to obtain a transposed matrix; applying a set of one-dimensional frequency convolutions to the transposed matrix to obtain a second convolved feature matrix; identifying a presence of a keyword based on further processing of a combination of the first and second convolved feature matrices; and performing a function in response to the presence of the keyword.

The invention belongs to the technical field of medical instrument control, and particularly discloses a single-line digital audio output control system of high-frequency surgical equipment, which comprises the steps of constructing a multi-dimensional parameter fused state perception and semantic audio feedback mechanism, accurately identifying the working mode, the power level and the output stability of the equipment through auditory sense, and controlling the output of the high-frequency surgical equipment according to the state perception and semantic audio feedback mechanism. The operation burden and the misjudgment risk in a complex operation environment are effectively reduced; transmission of audio, data and a power supply is integrated by adopting a single-line digital audio bus, so that the internal structure and the electromagnetic compatibility of equipment are optimized fundamentally; by establishing a set of self-adaptive safety closed loop from communication quality monitoring to power and audio cooperative control, the health condition of a communication link can be actively monitored, and when it is detected that the communication quality is reduced, protective measures such as power reduction and safety alarm are automatically triggered, and the potential control failure risk is eliminated in the bud.

The invention provides a digital audio noise reduction method based on time-frequency mask separation, relates to the technical field of digital signal processing, and not only can adapt to the change of environmental noise in real time and optimize the noise reduction effect, but also can accurately distinguish different sound sources in a complex noise environment through a dynamic mask adjustment technology so as to improve the noise reduction efficiency. Therefore, the voice recognition and processing capability of the vehicle-mounted audio system is improved. By combining multi-modal data acquisition and a comprehensive noise modeling technology, a more detailed noise prediction and processing basis is provided for the system, so that the definition and accuracy of audio processing are improved while noise interference is reduced.

Examples relate to generating an editable representation of a DJ performance. A computing system accesses at least one data stream representing the DJ performance and analyzes it to identify audio elements within the performance. The system generates a digital representation of the audio elements editable in a digital audio workstation, wherein the digital representation comprises a structured data format including information related to tracks, effects, and timing. The digital representation may include track objects for tracks played, clip objects representing segments of tracks, processor objects representing effects applied to tracks, and tempo objects representing tempo information. The system stores the digital representation, which can be loaded into a digital audio workstation application for editing using tools specific for DJ editing workflows.

The invention relates to the technical field of artificial intelligence and digital audio processing, and discloses a music element dynamic recombination and automatic sound mixing data processing and making system. Comprising a music information deep representation and semantic understanding module, a music element dynamic recombination module, an intelligent adaptive sound mixing module, a multi-agent collaborative optimization and evaluation learning module, a user interaction and collaborative creation interface module and a knowledge base and learning result storage module. According to the method, deep understanding of music content is realized through multi-modal music feature extraction, structure analysis and emotion semantic dynamic modeling, sound mixing parameters are adaptively adjusted in real time according to context factors by utilizing a dynamic weight matrix mechanism, and a creation closed loop is formed by combining multi-agent collaborative optimization; the problems that in the prior art, music element recombination adaptability is poor, the sound mixing effect is homogenized, and cross-module collaboration is insufficient are solved, and the music creation efficiency and the emotion expression accuracy can be improved.

A system including: a plurality of ID tags with a unique identifier, the ID tag being configured to be scannable by a scanner to identify the unique identifier, at least some ID tags being physically attached to a plurality of physical books and at least some ID tags not being physically attached to physical books, a network connected service, the service including a database which includes data representing a plurality of content links between unique identifiers of ID tags and files including digital content, the system being arranged to allow a plurality of digital devices to interact with the system, each of the digital devices including a speaker and / or a display to play and / or display digital audio content and / or digital visual content respectively, a communication unit to allow the digital device to communicate with the network connected service and a scanner arranged to scan an ID tag.

The dashed lines in the drawings delineate parts that do not form part of the design sought to be registered. The design is characterized by a combination of shape and pattern. Figure 1.1 is a front, right, top perspective view of a digital audio player showing the new design. Figure 1.2 is a rear, right, top perspective view thereof. Figure 1.3 is a rear, left, bottom perspective view thereof. Figure 1.4 is a front view thereof. Figure 1.5 is a rear view thereof. Figure 1.6 is a left side view thereof. Figure 1.7 is a right side view thereof. Figure 1.8 is a top view thereof. And Figure 1.9 is a bottom view thereof. 1.1) Perspective view; 1.2) Perspective view; 1.3) Perspective view; 1.4) Front view; 1.5) Rear view; 1.6) Left side view; 1.7) Right side view; 1.8) Top view; 1.9) Bottom view

A projector, such as a toy projector, in which stored content or information is delivered both as digital images and digital audio content from an interchangeable character which is interchangeably attached to the projector. The stored content or information can be specific to the content and / or theme being displayed and is delivered through an electrical connection to the projector.