438 results about "Voice communication" patented technology

The invention discloses an AI-based illumination control system self-adaptive dimming method, which comprises the following steps of: acquiring a two-dimensional coordinate, a motion track curve and voice communication data of a personnel position, and identifying social interaction scene categories of different time slices; determining a social interaction strength score according to the voice communication data, and predicting the most adaptive spectrum configuration data in the current social interaction scene in combination with the social influence factor matrix and the social interaction scene type; according to a historical dimming log and an energy consumption metering curve, identifying a short-term high-frequency dimming segment, and adjusting a real-time response strategy and a disturbance classification processing mechanism of the spectrum configuration data; and adjusting the illumination power distribution values of the high-frequency use area and the low-frequency use area according to the occupation raster data generated by the millimeter-wave radar and the infrared array in combination with the lamp area mapping relation and the power budget constraint. Through multi-source data fusion and a self-adaptive regulation and control mechanism, unification of real-time performance, pertinence and energy-saving performance of spectrum regulation is realized, and the dynamic response capability and the user perception experience of the intelligent illumination system are improved.

The invention discloses a voice QoS optimizing and playing method, device and equipment for end-to-end information source encryption and a medium, and relates to the field of instant messaging, and the method comprises the steps: storing a received encrypted voice data packet into a jitter buffer area in a ciphertext form; determining a current target buffer water level of the jitter buffer based on the current network jitter degree and a delay factor introduced by the current decryption processing; generating a play control strategy and a decryption scheduling strategy according to the current actual buffer water level and the current target buffer water level of the jitter buffer; the decryption scheduling strategy comprises a decryption operation frequency, a triggering opportunity and an execution priority, and the decryption scheduling strategy is adjusted based on a real-time playing control strategy and a jitter buffer area state; and scheduling the corresponding encrypted voice data packet from the jitter buffer area by using a decryption scheduling strategy for decryption, and decoding and playing the decrypted voice data packet according to a playing control strategy. According to the invention, the overall fluency of encrypted real-time voice communication is improved.

The invention relates to the technical field of wireless communication, in particular to a public network interphone with a quantum chip. According to the interphone, a master control module and a quantum security chip work cooperatively, a quantum random number generator is used for dynamically deriving a session key, and offline security distribution of a group master key is realized by encrypting a two-dimensional code; in a communication process, a system adaptively switches a voice coding mode according to a network condition, end-to-end authentication encryption is carried out on voice data by using a hardware encryption engine, and low-delay transmission is guaranteed through a network priority mark; the device establishes a complete key life cycle management mechanism, supports regular update and forward security of session keys and instant update and backward security of a group master key when members change, and ensures that all key operations are completed in a quantum chip. According to the invention, various eavesdropping and tampering attacks are effectively resisted, and safe, real-time and clear high-confidentiality voice communication in a public network environment is realized.

The invention discloses a real-time voice communication system and method based on a movement prediction and standby path mechanism, and aims to solve the problems of voice packet loss, delay and call interruption caused by frequent link interruption, large route reconstruction delay and high control overhead when an existing outdoor dynamic small-scale team performs real-time voice communication in a high-speed movement scene. The system comprises a voice sending end, a voice receiving end, a path management module and a mobile prediction module. The path management module constructs and maintains a main path tree for data transmission and a pre-generated standby path tree; the movement prediction module indirectly predicts the movement trend of the node through the change trend of the link quality, and constructs a movement prediction quality parameter through the change of the link packet loss rate, the delay and the jitter so as to dynamically optimize the structure of the path tree; when the main path is subjected to link breakage or the communication quality, such as packet loss rate, delay and jitter deterioration, is below threshold values, the system can be seamlessly switched to the standby path tree, so that the continuity and real-time performance of voice communication are guaranteed; according to the invention, the lightweight design is adopted, the control packet overhead is reduced, the method is suitable for an outdoor real-time voice communication scene with a small number of nodes, high moving speed and limited bandwidth and power consumption, and the communication robustness and the user experience are remarkably improved.

Systems and methods detect and prevent vishing attacks through an integrated framework combining SIP header customization, STIR / SHAKEN frameworks, AI / ML analysis, and real-time speech analysis using the Viterbi algorithm. The system begins with call initiation, embedding authentication information in the SIP header. The SIP data is transmitted and verified using STIR / SHAKEN frameworks, ensuring the authenticity of the caller's identity. Verified data is cross-referenced with third-party databases and analyzed by an AI / ML engine to detect anomalies. If potential fraud is detected, the call is blocked, and the customer is notified. Calls that pass initial checks are further analyzed using the Viterbi algorithm, which converts speech to text and identifies suspicious patterns. An anomaly pattern detector processes the converted text to detect vishing indicators, terminating the call if a match is found. This multi-layered approach ensures robust protection against vishing, enhancing the security and reliability of voice communications while safeguarding users from fraud.

An apparatus for audio communication includes a memory configured to receive audio data from a user of a voice communication, and one or more processors in communication with the memory. The one or more processors are configured to receive the audio data for the voice communication, which includes voice data of the user and background audio data. The processors are further configured to classify the background audio data into a plurality of types of background audio data, determine to not suppress a subset of the plurality of types of background audio data, process the audio data to not suppress the subset of the plurality of types of background audio data to generate output audio data, and transmit the output audio data.

The invention provides an interactive voice response processing method and device, equipment and a storage medium, and relates to the technical field of voice communication, and the method comprises the steps: obtaining IVR process configuration information related to an incoming call request under the condition that the incoming call request of a client is received through a telephone soft switch platform, iVR branch process configuration information in the IVR process configuration information comprises information of a plurality of nodes and execution conditions among the nodes; executing a starting node in the IVR process configuration information, and searching a to-be-executed node according to an execution condition between nodes in the IVR branch process configuration information related to the starting node; the to-be-executed node is executed, the next execution node of the to-be-executed node is searched according to the execution condition between the nodes in the IVR branch process configuration information related to the to-be-executed node and / or the key information of the client responding to the to-be-executed node, and the scheme of the invention can improve the IVR processing efficiency and improve the user experience.

A user interface system enables communication through inaudible speech by detecting physiological signals associated with inner voice communication. The system includes a wearable device positioned in, on, or around a user's ear, housing one or more sensor modalities such as electromyography (EMG) sensors for detecting muscle movements, functional near-infrared spectroscopy (fNIR) sensors for brain activity, and / or detection-and-ranging systems using sonar or radar for micro-deformations. Pre-processing modules condition the sensor data, which is then analyzed by one or more machine learning models to reconstruct the content of the user's inaudible communication and produce output representations. The system enables high-bandwidth natural language interaction without audible vocalization, maintaining privacy while supporting applications including AI assistant interaction, real-time translation, secure authentication, and inter-party communication.

The invention discloses a voice interaction method and device and electronic equipment. The method comprises the following steps: identifying service data sent by a terminal to obtain an identification result; under the condition that the identification result indicates that the service data is the voice service, semi-persistent scheduling (SPS) configuration information is sent to the terminal through radio resource control (RRC) signaling, and SPS configuration corresponding to the SPS configuration information is activated through control signaling; and after the SPS configuration is successfully activated, voice data sent by the terminal according to the SPS configuration information is received through the satellite, and data transmission optimization is carried out according to the propagation characteristics of the satellite in the voice data receiving process. According to the method and the device, the technical problems of low voice communication quality and stability caused by the fact that the standard native voice service in the IoT NTN is not supported and complex propagation characteristics in a high-orbit satellite communication environment are solved.

The invention relates to the technical field of earphones, in particular to a microphone structure, an earphone and a test method. The microphone structure comprises a shell and a microphone assembly, the shell comprises a pickup hole, an exhaust hole and a containing cavity, the microphone assembly is located in the containing cavity, the microphone assembly and the outer wall of the shell are arranged in a spaced mode, the microphone assembly divides the containing cavity into a first cavity and a second cavity which are independent from each other, the pickup hole is communicated with the first cavity, and the exhaust hole is communicated with the second cavity. The earphone comprises the microphone structure. The testing method is used for testing the acoustic performance of the earphone. According to the microphone structure, the earphone and the test method, the noise reduction effect can be improved, and the definition of voice communication is ensured.

The invention discloses a radio analog telephone communication encryption method, device and equipment and a medium, and the method comprises the steps: generating a frequency spectrum segment number and a pseudo-random sorting sequence through a preset encryption algorithm; and carrying out controllable segmentation on the total spectrum of the analog voice according to a continuous sequence from low frequency to high frequency, carrying out sequence recombination on each sub-spectrum according to a pseudo-random sequence, finally combining and outputting encrypted voice, and recovering the original voice by utilizing the same pseudo-random sequence reverse recombination at a receiving end. Therefore, the time-frequency continuity and statistical characteristics of the voice spectrum in the traditional radio analog communication are fundamentally changed. According to the method, frequency spectrums are recombined as a whole according to a pseudo-random rule, so that an eavesdropper can only obtain disordered voice signals after the frequency spectrums are rearranged even if the eavesdropper intercepts the signals, and the difficulty of passive cracking means such as frequency spectrum analysis, filtering reduction and voice recognition is remarkably increased; the problem that in the prior art, radio analog telephone communication is insufficient in encryption and easy to crack is solved.

To provide a system for preventing an accident by detecting a recognition discrepancy or an operation mistake between a controller and a pilot in real time.SOLUTION: A system comprising: means for receiving voice communication data between an air traffic controller and a pilot of an aircraft; means for converting the received voice communication data into text data; means for extracting instruction content from the converted text data; means for generating a predicted action based on the extracted instruction content; means for collecting current position data of the aircraft from a GPS; means for monitoring an actual action of the aircraft based on the collected GPS data; means for comparing the monitored action of the aircraft with the predicted action to detect an abnormality; and means for generating and notifying an alert when the abnormality is detected.SELECTED DRAWING: Figure 1

The utility model discloses a guest room control system with service reminding and sleep aiding sound effects, and relates to the field of intelligent services. The guest room control system with the service reminding and sleep-aiding sound effects comprises a reminding service module which is arranged on the wall of a guest room and is used for carrying out voice reminding and sending different reminding music at the set time of a user after the customer sets the reminding type and the reminding time; the music playing module is arranged on the wall of the guest room and is used for providing various types of sleep-aiding music for the guest room; the voice conversation module is arranged beside a bedside table of a guest room and is used for voice communication between a worker and a customer; and the control module is arranged at the reception desk of the hotel, and the control module is in communication connection with the reminding service module, the music playing module and the voice conversation module. According to the system, a manager can provide various intelligent services, such as reminding, music and voice communication services, in a guest room for a customer, the individual requirements of the customer are met, and the check-in experience feeling of the customer is improved.

The invention discloses a voice communication method and device, terminal equipment and a computer readable storage medium, and the method comprises the steps: obtaining the information of a target cell corresponding to a voice communication request when the voice communication request is received; determining support state information of the target cell for the first call mode according to the information of the target cell; and when the support state information is first state information, establishing a voice call in a second call mode, the first state information representing that the target cell does not support establishment of the voice call in the first call mode. By adopting the method provided by the invention, whether the target cell supports the establishment of the voice call in the first call mode or not can be predicted to obtain the support state information, and if it is predicted that the target cell does not support the establishment of the voice call in the first call mode, the voice call is directly established in the second call mode. Repeated attempts to establish the voice call in the first call mode are not needed, the voice call establishment time is saved, and the user experience is improved.

The invention relates to the technical field of cooperative communication, in particular to a digital interphone and satellite telephone cooperative communication control method and system, and the method comprises the steps: obtaining the working time sequence of a geophysical exploration instrument and the attribute information of each digital interphone; dividing the digital interphones into a first interfered group and a second interfered group according to the attribute information; establishing an expected link available state sequence according to the working time sequence; incorporating the voice frame sequence into a first or second co-processing queue; performing time correspondence on the voice frames in the first co-processing queue and an expected link available state sequence to obtain a target voice frame sequence, and forwarding the target voice frame sequence to a second interfered group through a satellite link or an intercom link; and according to the expected link available state sequence corresponding to the first interfered group, performing link selection sending control on the voice frames in the second cooperative processing queue. According to the invention, the problem of unstable talkback communication under the condition of strong transient electromagnetic interference is overcome, and the reliability and continuity of on-site voice communication are improved.

A virtual reality (VR) system including a head mounted display (HMD), a sensor or input device, and options, selection, and display modules. The HMD displays a VR environment including sports content of a live game, a prerecorded game, or a sports channel. The options module generates an options signal to show viewing options in the VR environment. The viewing options define viewing aspects for viewing the sports content within the VR environment. The sensor or input device generates signals based on detection of movement of the HMD, movement of an eyeball of a user, user voice communication, or user contact with the input device. The selection module, based on the signals: selects the sports content and a viewing option; and generates a selection signal. The display module displays the sports content in the VR environment based on the selection signal.

This invention belongs to the field of communication technology and discloses a voice communication system based on deep ultraviolet light. The system includes: a voice communication terminal, a transmitting information processing module, an electro-optical conversion module, a transmitting optical system, a receiving optical system, a photoelectric conversion module, and a receiving information processing module. Compared with traditional radio frequency communication, ultraviolet light has excellent anti-interference and anti-interception capabilities; compared with point-to-point communication in the deep infrared band, ultraviolet light has non-line-of-sight characteristics, enabling communication even in jungle warfare. This invention uniquely employs FMCW (Frequency Modulated Continuous Wave) modulation. FMCW encodes bit information as a continuous increase or decrease in frequency. The receiving end obtains the difference frequency signal through mixing, which can be demodulated using matched filtering or Fourier transform, significantly improving processing gain and noise immunity. Even under weak scattered light conditions, signals can be reliably extracted from strong backgrounds, greatly improving communication link margin.

This product is a portable information and communication terminal equipped with authentication and communication functions. The touchpad allows for fingerprint authentication and input operations on an external display, while the microphone / speaker allows for voice communication. This product can be carried by attaching it to various attachments, as illustrated in the "Reference Diagram Showing Usage" and "Reference Image Diagram 1 to 3 Showing Usage."

1. The name of the design product: earphone (mermaid). 2. The use of the design product: for receiving and playing audio signals, voice communication, etc. 3. The design points of the design product: in shape. 4. The picture or photo that best indicates the design points: perspective view 1.

This allows for the assurance of good call quality across multiple frequency bands without the need for tuning for each individual frequency band. [Solution] This is an acoustic signal processing device that transmits a signal in a predetermined frequency band, the receiving frequency band, acquired via a communication device that performs voice communication, to a sound generating unit via a receiving-side signal path to produce sound from the sound generating unit, and acquires voice input via a sound detection sensor via a transmitting-side signal path and outputs it to the communication device. It comprises a band detection unit that detects whether the receiving frequency band is a narrowband first frequency band or a second frequency band wider than the first frequency band, based on a reference signal transmitted through the receiving-side signal path, and a band-specific countermeasure unit that takes countermeasures according to the frequency band detected by the band detection unit.

This invention provides an end-to-end voice encryption method, apparatus, and Bluetooth headset suitable for multi-hop lossy channels. The method includes front-end noise reduction, framing, and segmentation of the acquired voice at the transmitting end. Based on the session mode, a group public key or a point-to-point private key is automatically selected as the master key from a pre-configured key set. Then, multiple time-domain segments within each frame are scrambled, subjected to segment-by-segment Fast Fourier Transform, and frequency-domain encryption based on subkeys, according to the master key. The encrypted voice is then generated through inverse transformation and overlapping weighted smoothing concatenation, and transmitted through a multi-hop voice communication channel containing multi-level lossy encoding and decoding. At the receiving end, the reverse process is performed: framing and segmentation, transformation, inverse frequency-domain encryption, and inverse time-domain scrambling. Combined with U-shaped neural network noise reduction at both the front-end and back-end, the voice is restored to intelligible speech. This method enables flexible key management and high-quality secure voice transmission in scenarios where group calls and point-to-point private calls coexist and undergo multiple lossy compression operations.

1. Name of the product in this design: Fire Helmet. 2. Purpose of this design: To protect the head of firefighters, capture and display video of the rescue scene, and conduct voice communication. 3. The key design feature of this product is its shape. 4. The image or photograph that best illustrates the design's key points: a 3D model.

The application relates to the technical field of underwater acoustic communication, and discloses a method and system for dynamically allocating resources of an underwater voice communication system. The method comprises the following steps: setting a multi-user priority strategy according to a formation modular grouping; adopting a hybrid access mechanism of fusion of time division multiple access and carrier sense multiple access / collision avoidance, dividing communication channels into inter-group time slots and intra-group time slots, and competing for access to idle sub-carrier clusters in the time slots through a CSMA / CA mechanism; predicting a future state of an underwater acoustic channel; based on the predicted future state, adaptively switching a modulation and coding scheme according to a preset signal-to-noise ratio threshold; distributing sub-carrier bands to users with the largest channel gain based on a channel gain matrix; and combining the multi-user priority strategy, the hybrid access mechanism and the modulation and coding scheme to jointly optimize and distribute sub-carriers, bit loading and transmission power. Through dynamic resource allocation and adaptive transmission optimization, the application effectively improves the resource utilization rate and transmission quality of underwater voice communication.

The application discloses a voice networking service method based on SIP protocol and belongs to the technical field of multimedia voice communication. The application firstly constructs a DNS module, an NMS module and a VRS module, wherein the communication protocol between the NMS module and an external voice dispatching platform adopts the SIP protocol, the communication protocol between the DNS, the NMS and the VRS module adopts the TCP protocol, the protocol adopted by the communication between the NMS modules is the TCP protocol, the voice dispatching platform is used for accessing a user terminal, is connected with the user terminal through the SIP protocol and is connected with the NMS module through the SIP protocol; under the premise that a calling user calls a called user in a networking, the platform addressing is completed by the DNS module, and the voice dispatching platform is instructed by the NMS module to establish the call connection between the calling user and the called user. The application can provide the networking service between different voice dispatching platforms, can dynamically configure a routing path and has functions such as port convergence. The application provides an extended function interface, greatly improves the flexibility and expansibility of the networking.

The invention relates to the technical field of fire-fighting masks, in particular to a fire-fighting protective mask with a bidirectional voice communication function, which comprises a fire-fighting mask body and a filter, a protective cylinder is in threaded connection with the front side of the filter, an adjusting cover is rotatably arranged at the front end of the outer side of the protective cylinder, and the fire-fighting protective mask further comprises a transition gear; the adjusting frame is rotationally arranged between the inner wall of the adjusting cover and the outer wall of the protective cylinder. According to the invention, the gear and gear ring transmission mechanism is driven by rotating the adjusting cover, so that the opening and closing angles of the plurality of sealing baffles on the front surface of the protective cylinder can be synchronously and accurately controlled, and the sectional area of the air inlet can be continuously adjusted; by means of the design, a wearer can buffer hot airflow impact in a small opening state in the initial wearing process and provide sufficient airflow contact time for the canister so as to improve the filtering efficiency, the mask can be switched to a large opening state when high-intensity operation is needed so as to remarkably reduce inspiration resistance, the mask wearing comfort is improved, and meanwhile the breathing safety of the wearer is guaranteed.

The utility model provides a voice gateway device comprising an upper housing, and the surface of the upper housing is provided with heat radiation fins. The bottom shell is fixedly connected with the upper shell, a boss is arranged on the inner side of the bottom shell, and the upper shell and the bottom shell are fixedly connected to form a containing space; the main board is positioned in the accommodating space and is fixed on the boss; the battery is electrically connected with the main board, and the battery is fixed on the main board; the light guide part is fixed on the inner side of the bottom shell and is electrically connected with the mainboard; the antenna part is fixed on the outer side of the bottom shell and is electrically connected with the mainboard; a clamping groove is formed in the first side of the bottom shell; the second side of the bottom shell is provided with a plurality of interface holes. The voice gateway equipment is good in heat dissipation and simple to install, does not depend on an external power supply for power supply, and can ensure the continuity of voice communication under the condition of no external power supply for power supply.

A kind of based on magnetic attraction wearing language conversion machine translation device and method, belong to machine translation technical field, to solve the operation of the existing translation device generally is suitable for face-to-face communication when translating, translation work is more cumbersome when online, the present application passes through translation control terminal and includes input module, input module electrically connected conversion selection module, conversion selection module electrically connected language conversion module, language conversion module electrically connected output module, input module electrically connected communication equipment, when carrying out translation work, utilize input module can be through voice input or text input mode input text, then utilize conversion selection module selects the kind needed to be translated, again utilize language conversion module conversion text, finally utilize output module through text output and voice output mode presents translation result, and utilize communication equipment bluetooth connection input module, can realize the purpose of real-time translation when user voice communication.

The present application belongs to the technical field of speech processing, and particularly relates to a double-talking detection and acoustic echo cancellation cooperative processing method based on double CUDA streams. The method comprises the following steps: creating a first CUDA stream and a second CUDA stream on the same graphics processing unit; establishing a shared memory control package for cross-stream control between the first CUDA stream and the second CUDA stream; organizing a microphone near-end signal and a far-end reference signal into continuous frames with a preset frame length and frame shift; in the second CUDA stream, recording a synchronization event after writing sub-band control parameters and frame-level mode markers into the shared memory control package; in the first CUDA stream, performing partition blocking frequency domain sub-band adaptive filtering to update filter coefficients and generate echo suppression output. The present application realizes a highly cooperative, low-delay, high-robust real-time processing system for double-talking detection and echo cancellation, and has operation efficiency, control accuracy and engineering realizability, and is suitable for voice communication, remote conference and intelligent voice interaction systems.

1. Name of the product in this design: Landline Telephone. 2. Purpose of this design: This design is used for voice communication. 3. The key design feature of this product is its shape. 4. The image or photograph that best illustrates the design's key points: a 3D model.

The invention relates to the technical field of vehicle-mounted voice communication, and provides a multi-priority CAN (Controller Area Network) bus voice interaction method and a multi-priority CAN bus voice interaction system, which ensure that the alarm voice with the highest priority can be immediately broadcasted when a vehicle is in any state through dynamic priority arbitration, eliminate the safety risk caused by voice channel blockage and improve the safety of the vehicle. Bus conflicts are reduced by adopting an intelligent voice scheduling strategy, the bandwidth utilization efficiency of the CAN bus is improved, the fluency of a voice system is ensured, intelligent arbitration is realized, and CAN bus voice interaction of timely output of high-priority voice is ensured; meanwhile, the system can understand scenes (for example, navigation is more important during high-speed driving) so as to provide more intelligent voice interaction experience better meeting user requirements, the design is based on a standard CAN protocol, hardware does not need to be changed, only software upgrading is needed, and popularization and application are easy.