38 results about "Latency (audio)" patented technology

This invention discloses a low-power edge-side simultaneous interpretation system and method based on audio-text synchronization and visual feature fusion, relating to the fields of multimodal human-computer interaction and machine translation technology. The system achieves high-precision audio-text timing alignment through an audio-text synchronization matching module, generates a lightweight visual feature stream using a visual feature processing module, and performs spatiotemporal fusion by a multimodal fusion inference module. Combined with edge-side heterogeneous computing power scheduling and dynamic power consumption control, it significantly reduces the power consumption of edge devices while ensuring low-latency translation of ≤20ms per frame. This invention fills the technological gap in edge-side low-power multimodal simultaneous interpretation and can be widely applied in edge scenarios such as mobile office, international communication, and smart wearables.

This invention provides an amplifier with improved bias tracking performance based on a DSP chip. It includes a file acquisition module that directly acquires the audio file to be played via an interface. The audio file is pre-input into the DSP for standard waveform analysis, which speeds up processing and reduces latency during actual playback. During actual playback, the actual played energy is compared with the pre-calculated energy to determine if it matches the pre-designed output. If it matches, the pre-designed output is used directly; otherwise, it is further input into a model for processing, ensuring operational stability and safety.

This disclosure presents embodiments of a model acceleration method, a video generation method, an apparatus, a device, a medium, and a product. One specific implementation of the method includes: determining a sample set and an original video generation model, wherein the samples in the sample set include reference images and audio; performing conditional distillation on the original video generation model based on the sample set to obtain a first video generation model, wherein the number of forward propagation steps of the first video generation model is less than the number of forward propagation steps of the original video generation model; and performing distribution matching distillation on the first video generation model based on the sample set to obtain a high-speed video generation model, wherein the number of inference steps of the high-speed video generation model is less than the number of inference steps of the original video generation model. This implementation relates to audio-driven video generation, reducing the computational complexity and inference latency of the model, and facilitating deployment and use in scenarios with limited computing resources.

This application discloses a simultaneous interpretation method, apparatus, device, and storage medium, relating to the field of audio processing technology. The aforementioned simultaneous interpretation method is applied to a simultaneous interpretation device, which includes a locally deployed AI service module. The method includes: acquiring first audio data in a first language; converting the first audio data into second audio data in a second language offline, based on the locally deployed AI service module; and transmitting the second audio data via wired transmission to a first communication device in a call state, wherein the second audio data is provided to a second communication device engaged in a call with the first communication device. This method enables low-latency transmission during simultaneous interpretation.

Provided is wireless communication method, apparatus and system, and a device. The method comprises receiving, by a wireless end device, first audio data transmitted by a wireless headset, in a first time slot through a first wireless communication channel therebetween; receiving task operation data input by each input device, in a corresponding time slot through a wireless communication channel therebetween, where the corresponding time slot is different from the first time slot; and executing at least one corresponding operation based on at least one of the first audio data and the task operation data. According to the present disclosure, various tasks including the game task are implemented in a wireless communication environment, and the first audio data and task operation data are received in different time slots during the execution of the tasks, such that the overall communication latency can be rather small.

The invention relates to the technical field of speech recognition, and provides an end-to-end speech recognition method and system based on a keyword attention enhancement mechanism, and the method comprises the steps: extracting entity keywords through a keyword searcher; the voice cache manager receives continuous air traffic control audio clips, converts the continuous air traffic control audio clips into an audio Mel spectrogram and then converts the audio Mel spectrogram into an audio embedded sequence; the keyword encoder unit maps the lexical element embedding representation sequence into a keyword embedding sequence; the audio transliteration decoder unit performs keyword attention enhancement calculation and converts splicing vectors of the audio embedding sequence, the transliteration start mark and the lexical element embedding representation sequence into a prediction text sequence; and inputting the new lexical element embedded representation sequence into a text translator through autoregression until the audio transwriting decoder unit outputs a transwriting end mark, and outputting the predicted text sequence as a speech recognition text. According to the invention, real-time identification of the streaming input voice is realized, and the method has the advantages of high identification precision, low response delay, flexible deployment and the like.

An exemplary process for wirelessly streaming audio data to a head-mounted display (HMD) with low latency and reduced auditory distortion is described. This includes determining one or more statistics indicating the latency associated with audio data wirelessly received by the HMD and stored in a buffer within the HMD, and determining, at least in part, to adjust the size of the buffer based on these statistics. To reduce the buffer size, segments of audio data can be replaced with synthesized audio data segments to obtain modified audio data in the buffer. To increase the buffer size, synthesized audio data segments can be added to the audio data to obtain modified audio data in the buffer. Audio content can then be output via one or more speakers of the HMD, at least in part, based on the modified audio data.

This application relates to a low-latency Bluetooth audio transmission method, apparatus, device, and medium. The Bluetooth audio transmission method includes: a data access step, initiating an audio data transmission request to a microphone; after receiving the audio data sent by the microphone via Bluetooth, the host device parses the audio data through the Bluetooth protocol stack and writes the parsed audio data as raw data into the operating system's data interface; an audio processing step, reading the raw data from the data interface through an audio processing library, processing the raw data, and outputting the processed audio data for playback; wherein the audio processing library is located in the operating system's hardware abstraction layer. This invention achieves low-latency and stable Bluetooth audio playback by processing audio data at the hardware abstraction layer and combining it with a playback clock adaptive adjustment mechanism.

This invention relates to the field of audio circuit hardware technology, and in particular to a low-latency ARC circuit configuration method and system based on I2S audio switching. External video data is received through an ARC circuit compatibility interface and input to a video conversion chip for data separation, achieving synchronous separation of I2S audio signals and MIPI video signals. The separated I2S audio signals are then simulated and sent to the host SOC and external I / O terminals respectively by a first-stage audio switching chip. Through link delay analysis and instantaneous delay judgment, the audio path is dynamically configured to achieve switching control between the internal processing path and the direct path. A second-stage audio switching chip realizes multi-channel audio multiplexing and direct transmission simulation, achieving a low-latency, highly compatible ARC audio output scheme while ensuring audio and video timing alignment. This invention can significantly reduce the direct transmission latency of ARC audio while ensuring audio and video synchronization and system compatibility, and improves the flexibility and reliability of audio path configuration.

This invention discloses a multimodal data fusion coding method for constructing a unified underlying language for national big data, belonging to the fields of national big data architecture, multimodal governance, and domestically produced unified coding. Based on the mathematical fundamental feature theory and the STE domestic coding system, coupled with the GZ-BigData-RISC-V domestic dedicated chip and a multimodal AI fusion model, it achieves normalized access to multimodal data, unified extraction of fundamental features, unified underlying language coding, lossless fusion, and secure storage. Quantitative thresholds are set for cross-modal fusion efficiency ≥85%, data consistency ≥99.5%, coding latency ≤1ms, and lossless fusion rate 100%, completing the unified coding and semantic association of text, image, audio, video, time-series, and geospatial data. This method breaks down data heterogeneity barriers, constructs the only nationally interoperable unified underlying language for big data, and is 100% domestically produced and controllable throughout the entire process, supporting the construction of a national integrated big data center and promoting the market-oriented circulation of data elements.

Features described herein generally relate to providing dynamic spatial audio. Particularly, audio data is received, first audio frames are generated from the received audio data, the audio frames are transmitted to an audio playback device using a wireless link in a dynamic spatial audio mode, at least one condition associated with the audio playback device is detected, second audio frames are generated from the received audio data; and the second audio frames is transmitted to the audio playback device using the wireless link in a basic audio mode.

The application relates to a low-delay playing method and device, electronic equipment and a storage medium, the method being applied to an audio playing device with a playing storage area, the playing storage area comprising a first storage area and a second storage area, the method comprising the following steps: reading the playing storage area from a first preset position of the first storage area and playing when a playing trigger time Tt arrives; triggering decoding at a first time T1 to obtain first data, and storing the first data into the playing storage area according to a first storage rule; triggering decoding at a second time T2 to obtain second data, and storing the second data into the playing storage area according to a second storage rule; wherein T1 is equal to or later than a time Tr1 at which the audio playing device receives the first data, T2 is equal to or later than a time Tr2 at which the audio playing device receives the second data, and T2 is later than Tt; the first data starts playing before the second storage area is read for the first time since Tt, so that the delay of audio playing is reduced, and the continuity of playing is ensured.

This invention discloses an audio processing method, system, and apparatus. Upon receiving an audio stream from a client, a two-layer speech activity detection is performed in parallel. When the duration of silence detected exceeds a relatively short first time threshold, speech recognition is performed asynchronously on the received audio stream to determine the corresponding pre-recognition text. When the duration of silence detected exceeds a relatively long second time threshold, the audio stream input is determined to have ended. The recognition status of the pre-recognition text is then queried. If recognition is successful, the pre-recognition text is used as the final speech recognition text. This method introduces a shorter first time threshold to trigger pre-recognition speech recognition, providing a pre-trigger time closer to the end of the user's speech before the termination judgment (i.e., the second time threshold). Furthermore, it uses an asynchronous approach to start the pre-recognition speech recognition task to complete some key computational tasks in advance, reducing processing latency caused by the waiting time for the termination judgment.

In this disclosure, a method (50) is proposed for switching from a first Bluetooth audio source (20) to a second Bluetooth audio source (21) in a user-friendly manner, for example, reducing latency. This method is performed by an audio rendering system (10), which includes a primary wireless speaker (11) and a secondary wireless speaker (12). The audio rendering system receives a first audio stream from the first audio source using a first audio topology, the first audio topology including a first set of wireless links. Upon receiving a request to switch to a second audio stream using the second audio topology (51), (52) a second set of wireless links is established, and reception of the second audio stream (53) begins, while (54) the audio rendering system and the first audio source are maintained connected via the first set of wireless links. The wireless speaker can be a wireless earbud, bookshelf speaker, floorstanding speaker, outdoor speaker, subwoofer, or headphone speaker.

The present application relates to the technical field of audio data processing, and relates to an audio data processing method for a personal sound amplification product, and a personal sound amplification product. The method comprises: an adaptive feedback canceller eliminates acoustic feedback in an audio signal; an adaptive notch filter suppresses a howling signal in the audio signal; a transform module converts an audio signal acquired by a microphone into a time-frequency signal; a multi-subband dynamic range compressor performs dynamic range adjustment on the time-frequency signal to obtain gains of frequency points; a filter conversion module calculates filter coefficients on the basis of the gains of the frequency points; a time domain filter performs, on the basis of the filter coefficients, time domain filtering on an audio signal outputted by the adaptive notch filter; an equalizer adjusts a frequency response of the filtered audio signal; an amplitude limiter limits an amplitude of the obtained audio signal to be within a preset range; a digital-to-analog converter converts the audio signal into an electrical signal; and a speaker converts the electrical signal into sound for playback. The present application can reduce PSAP latency.

This invention discloses an audio spectrum fluidized interactive presentation method and system based on shader computing power, belonging to the field of image processing technology. It includes: real-time acquisition of audio streams and user interaction data to extract features; generation of the total dynamic field of audio interaction through nonlinear coupling in a GPU parallel computing shader; inputting this field into a preset neural network model for forward inference in a fragment shader to complete the fluid neurophysical evolution; separating the audio semantic layer in the rendering pipeline and performing competitive visual attribute mapping to generate images; and asynchronous scheduling of each processing pipeline through an asynchronous computing engine. This invention employs full-pipeline GPU computing and neural operator evolution, effectively resolving the contradiction between fluid simulation and real-time rendering, eliminating frequent data transmission, and achieving low-latency interactive feedback while ensuring physical realism, significantly improving the immersiveness and expressiveness of audio visualization.

This invention relates to the field of artificial intelligence technology, solving the response latency problem in the content generation process of existing artificial intelligence technologies. It provides a method for optimizing generation efficiency based on intent recognition and memory. The method includes: inputting raw audio data into a pre-trained intent recognition and analysis model to obtain text information, urgency level, and content complexity level corresponding to the raw audio data; determining a target generation mode based on the urgency level and content complexity level; acquiring memory feature information matching the text information through a preset memory enhancement network; generating target generative content based on the text information, memory feature information, and the target generation mode; obtaining user feedback on the target generative content; and adjusting the weights of the memory enhancement network based on the feedback information to optimize generation efficiency.

Described are in-ear monitoring (IEM) systems configured for audio performance environments requiring low audio latency and high scalability. IEM systems can include an audio channel allocation device that determines audio channel allocation for transmitting audio payload to IEM devices. Audio payload may be allocated to a radio frame based on, e.g., bit rate, modulation and coding scheme, latency / fidelity requirements, etc. IEM devices can include audio driver(s) configured to generate an audio output, a circuit configured to control audio output generation by the driver(s), in-ear portion(s), and a bodypack receiver. IEM devices can receive the audio allocation information, configure its circuit accordingly, receive audio payload carried in a carrier wave based on the allocation information, and generate the audio output based on the audio payload.

This application discloses a digital human voice interaction method and system, relating to the field of speech synthesis and recognition technology. The method includes: acquiring raw speech, network latency, and computing power status, and aligning them to obtain the terminal status; extracting energy and silence intervals to generate pause boundary sequences, speech segment tables, and target rhythm sequences; determining the recognition mode to obtain a standardized text stream; generating a response text segment table and a raw audio segment table; obtaining a continuous audio stream through time scaling, timbre baseline extraction, and envelope adjustment; and generating lip-sync rhythm sequences and facial expression preservation sequences accordingly. This application improves the rendering consistency and continuity of cross-terminal voice interaction for two-dimensional digital humans.

This utility model relates to the field of audio and video recorder technology, specifically to an audio and video recorder with image stabilization. It includes a recorder body and a built-in attitude sensor circuit that senses the device's motion in space in real time. Combined with image processing algorithms, it dynamically compensates for image shake caused by external vibrations, effectively eliminating or reducing image jitter and significantly improving video stability and clarity. By integrating an LCM display module onto the recorder body, it enables real-time display and touch operation of the camera feed, enhancing the convenience and intuitiveness of human-computer interaction. The camera module and LCM display module, connected via a MIPI interface, ensure high-speed transmission and low-latency display of high-definition video data. The attitude sensor circuit and battery level gauge communicate with the main control circuit board via an I2C interface, enabling real-time monitoring of the device's motion and accurate detection of battery power.

This application discloses a digital human voice interaction method and system, relating to the field of speech synthesis and recognition technology. The method includes: acquiring raw speech, network latency, and computing power status, and aligning them to obtain the terminal status; extracting energy and silence intervals to generate pause boundary sequences, speech segment tables, and target rhythm sequences; determining the recognition mode to obtain a standardized text stream; generating a response text segment table and a raw audio segment table; obtaining a continuous audio stream through time scaling, timbre baseline extraction, and envelope adjustment; and generating lip-sync rhythm sequences and facial expression preservation sequences accordingly. This application improves the rendering consistency and continuity of cross-terminal voice interaction for two-dimensional digital humans.

This application belongs to the field of artificial intelligence, specifically relating to a visual diagnostic method for equipment anomalies, an electronic device, and a storage medium. The method includes: acquiring video containing the equipment's operating status, text-based consultation questions, and prior metadata of the equipment to generate a multimodal data packet; extracting visual embedding vectors of keyframes from the video frame stream and extracting audio event tags from the audio frame stream; performing coarse detection based on the visual embedding vectors of the keyframes to determine the time interval where a preset device exists; performing fine localization based on image data and text interaction commands of the keyframes within the time interval to output fault location information; and constructing a multimodal temporal semantic graph based on the fault location information, audio event tags, and text interaction commands for multimodal reasoning to generate structured fault diagnosis data. This application's method achieves low-latency, high-accuracy diagnosis of on-site equipment video by deeply fusing multimodal information through a large visual model and a large language model.

The application belongs to the technical field of network communication, and particularly relates to a high-definition low-latency audio and video long-distance transmission device. The high-definition low-latency audio and video long-distance transmission device comprises a high-definition audio and video interface box and a high-definition audio and video adapter which are used in cooperation, wherein the high-definition audio and video interface box and the high-definition audio and video adapter each comprise a cover plate assembly, a casing assembly and an internal module; the cover plate assembly is connected with the casing assembly through a connecting piece and forms a sealed cavity; and the internal module is placed in the sealed cavity and is overlapped with the casing assembly. The device adopts a whole machine sealing mode, the cover plate adopts a double-layer sealing design, and the jack assembly adopts an integrated processing forming mode, so that a complete sealed shielding chamber is formed in the device. The internal module is located in the casing assembly and is physically isolated from the external environment. Meanwhile, the internal module is subjected to a reinforcing design, so that the device can be normally used in a vibration environment such as a vehicle-mounted environment or a ship-mounted environment.

This invention relates to the field of communication technology, specifically to a low-latency, high-reliability converged communication audio and video dynamic adaptation encoding and decoding transmission system. It includes: a link data acquisition and storage unit; a dynamic adaptation control unit employing a lightweight causal timing prediction engine with an improved temporal convolution + sparse attention architecture; an encoding and decoding unit; a transmission adaptation unit; and a reliability verification and feedback unit. This invention uses a lightweight causal timing prediction module to perform feedforward prediction on continuous link state sequences, and uses the prediction results as feedforward compensation signals input to the adaptation parameter decision module, replacing the traditional passive backward adjustment logic. This allows for early adaptation to time-varying link fluctuations. Simultaneously, combined with a confidence decay and mode switching mechanism, it automatically reverts to a reactive adjustment mode when the prediction confidence falls below a set threshold, mitigating audio and video stuttering and excessive latency issues caused by sudden changes in link state.

This invention discloses an audio steganography method for secure communication in vehicle-to-everything (V2X) networks based on reinforcement learning. The method involves windowing and framing the in-vehicle audio signal and performing a short-time Fourier transform to construct time-domain and frequency-domain state vectors. A time-domain agent decides whether to embed information in the current frame. If embedding is selected, a frequency-domain agent is activated to determine the frequency band, embedding strength, and encoding method, and encoded ciphertext noise is superimposed on the amplitude spectrum. By combining multi-dimensional reward functions considering capacity, robustness, concealment, latency, and detection risk, the agent's policy convergence is driven. This invention enables secure, concealed transmission with high capacity, low latency, and high robustness while ensuring audio perception quality and real-time communication. It is suitable for concealed audio carrier communication in vehicle-to-vehicle and vehicle-to-infrastructure (V2I) scenarios.

This application discloses an audio processing chip and a vehicle, belonging to the field of audio chip technology. Through the technical solution provided in this application, after the modem receives external audio data, it directly transmits it to the audio digital data processor for echo cancellation processing. The processed audio data is directly transmitted to the audio amplifier through the I2S port, and after digital-to-analog conversion, drives the speaker to produce sound. The entire audio data transmission path is completed within a single chip, eliminating the protocol conversion required for cross-chip communication. The hardware-level data transmission mechanism of the I2S interface replaces the Ethernet transmission process, significantly reducing audio data relay time and lowering audio data transmission latency.

A computing system includes first and second client computing devices accessing a communications network to establish a communications session. The first client computing device operates an audio analysis agent to determine network latency within the communications session based on communications with an audio analysis agent in the second client computing device. In response to the network latency exceeding a latency threshold, audio input from a user of the first client computing device is analyzed to determine a speaking status of the user. The audio analysis agent generates an indicator command message for the second client computing device based on the determined speaking status of the user. The second client computing device displays an indicator based on the indicator command message indicating when a user of the second client computing device can speak to avoid speech confliction with the user of said first client computing device.

The application belongs to the field of artificial intelligence, and particularly relates to a device abnormality visual diagnosis method, an electronic device and a storage medium. The method comprises the following steps: acquiring a video containing a device running state field, a text consultation question and device prior metadata, and generating a multi-modal data package; extracting a key frame visual embedding vector from a video frame stream and extracting an audio event label from an audio frame stream; performing coarse detection based on the key frame visual embedding vector, and determining a time interval in which a preset device exists; performing fine positioning based on image data of the key frame in the time interval and a text interaction instruction, and outputting fault positioning information; constructing a multi-modal time sequence semantic graph based on the fault positioning information, the audio event label and the text interaction instruction to perform multi-modal reasoning, and generating fault diagnosis structured data. The method of the application realizes low-latency and high-accuracy diagnosis of a device field video through deep fusion of multi-modal information by a visual large model and a large language model.

This invention belongs to the field of audio signal processing and privacy protection technology, specifically relating to an adaptive ultrasonic anti-recording interference method based on local AI voice feature analysis. The method involves: acquiring environmental audio signals through an audio acquisition module; performing voice feature analysis on the pre-processed audio signals; establishing a dual-microcontroller collaborative working mode; controlling the ultrasonic emission module to generate and drive the ultrasonic transducer array to emit ultrasonic interference signals; the second microcontroller dynamically adjusting interference parameters based on a closed-loop feedback compensation mechanism; and a power management module switching system power consumption modes according to the voice detection status to achieve multi-mode power management. This invention relies on a local lightweight AI model to complete voice feature extraction, achieving real-time voice detection and interference parameter generation without relying on a network. This eliminates network latency and avoids the privacy leakage risks of uploading audio data to the cloud, making it suitable for offline use scenarios of portable devices.

This application provides a service-aware Bluetooth HCI interface dynamic flow control optimization method and device, relating to the field of wireless communication technology. The method includes: a host in a Bluetooth communication device communicates with a controller via an HCI interface; based on the total buffer capacity of the controller and combined with the available audio context declared in the published audio capability service, it dynamically calculates and configures independent credit pools with different weights for isochronous streams (ISO) and asynchronous data streams (ACLs); based on link quality and data queue status, the controller performs predictive credit early return for ISO services before sending an acknowledgment response at the physical layer, so that the host can send the next frame of audio data in advance; and it uses byte-block granularity for buffer idle state reporting and flow control scheduling. This application solves the problems of delayed credit return and coarse flow control granularity in traditional Bluetooth HCI flow control mechanisms, ensuring low-latency, low-jitter transmission of real-time services while improving the utilization rate of the controller buffer.