78 results about "Live voice" patented technology

In an automated leads-and-bids matching system, bid profiles are defined to describe desires of lead buyers. Received leads are matched to active ones of the bid profiles whose specifications the leads substantially match. Lead segmentation data is captured on-the-fly for example by inducing potential consumers to navigate their way through tree-organized web sites that categorize the consumers according to their geographic location, income / revenue range, class of products desired and / or other attributes. Live voice or other telecommunication connections to the pre-classified consumers are coupled to corresponding, pre-classified telecommunication nodes of a call processing system. The call processing system deduces the segmentation data of the consumers from the identities of the pre-classified nodes through which their connections pass. The deduced segmentation data is passed to an automated matching system or auctioning subsystem that finds the highest bids for each given lead. A quality rating database rates the quality of leads provided by different sellers. A price discounting engine discounts the amount paid to sellers who are hot-rated as sources of inferior leads.

The present invention is directed to a call admission controller that is operable to: (a) determine at least one of (i) a bandwidth utilization level for a first path including a first link; (ii) an available bandwidth level for the first path; and (iii) one or more Quality of Service or QoS metrics for the first path; (b) compare the at least one of (i) a bandwidth utilization level; (ii) an available bandwidth level; and (iii) one or more Quality of Service or QoS metrics to one or more selected thresholds to determine whether a new live voice communication may be set up with a first selected codec; and (iii) when a new live voice communication may not be set up with the first selected codec, perform at least one of the following operations: (i) select a second different codec from among a plurality of possible codecs for the new live voice communication, wherein the second codec has a lower bit rate than the first codec; (ii) change an existing live voice communication from the first codec to the second codec; and (iii) redirect the new live voice communication from the first path to a second different path, wherein the second path does not include the first link.

Current voice message systems typically operate by recording a message that people leave for a user. However, these current systems do not process the message in a manner where significant information, such as phone numbers, meeting times, and locations, is identified, extracted, and output. The present invention solves the current system's limitations and extracts pertinent data from a voice message. In addition, the present invention is versatile because it is applicable to a recorded or live voice message, and any type of phone can actuate the extraction mechanism. The invention builds on voice recognition technology by identifying data, for instance, common phrases and strings of numbers from a voice message. Next, the invention provides an interface for actuating extraction technology. In one embodiment, the extraction technology removes information based on user commands according to a series of keystrokes. In another embodiment, the invention extracts information by a user's voice commands. Consequently, when a telephone user encounters a situation where he / she is unable to write down a phone number or other information, the user will be able to quickly access the invention to find the number or other piece of relevant information and save it for retrieval later. This invention saves the user time and frustration from constantly replaying the same message. It also saves money by reducing fees from repetitive and redundant telephone calls. With simple keystrokes or voice commands, a user can immediately access extracted information and make his / her desired telephone call.

A system to help patients manage their healthcare using a comprehensive package of healthcare reminders administered by remote computing and wireless communications devices. Healthy lifestyle, preventive care, chronic condition maintenance care, medication compliance and custom reminders are set according to a schedule and delivered via email, SMS text message, IVR phone message, live voice phone or other communications methods. Participants respond to reminders with their actual progress toward achieving the goal or with data concerning their condition which is recorded and reported. Tools allow users to schedule healthcare provider appointments and receive related reminders and follow-ups, refill prescriptions and generate a variety of reports to track their own progress and condition and which can be shared with a physician. Participating physicians have direct access to patient records to monitor progress and compliance in real time and to communicate with and manage their patients. Integration of personal health records and electronic medical records systems when available permit detailed analysis to identify potential issues and recommend certain reminders to improve the overall quality and integration of care.

The present invention is directed to a telecommunication device 174 or 180 including (a) an input operable to receive from an endpoint, as part of a live voice communication, a text message and an audio message, the audio message being a voiced version of the text message; (b) a character display; (c) at least one activator; and (d) a graphical display agent 184 operable to (i) display the text message on an associated character display 300 and (ii) audibly play the audio message.

A digital telephone or an analog telephone system with VOIP conversion is linked into the surveillance system and transmissions therefrom are treated as an event requiring an emergency response from the system. The system supports integration of emergency telephone systems into a multi-media surveillance system to permit comprehensive and multiple responses to an event triggered by receipt of the emergency phone call transmission. The telephone transmission is captured and transmitted to a surveillance system processor. Upon receipt, the system can react with any of numerous responses. First, the telephone call is transmitted to a manned station and may be answered in the normal fashion with a live voice response. The receipt of the call also activates the entire system in the same manner as a triggering signal from any other sensor or appliance. In addition, various other responses may be activated, either manually or in a pre-programmed hierarchy.

An improved earbud design providing for full modularity; improved and variable hearing protection, sound quality, comfort, fit, aesthetics, and signal connectivity; and the ability to maintain environmental sound directionality comprised of a multitude of new features with variable vents and membranes which dilute the harmful pneumatic effects of sound while improving its acoustic quality. A location-based transmission system provides event attendees to mix live sound with streamed sound through Ambrose Earbuds for reduced hearing risk and no quality detriments due to timing gaps, occlusion or ear tip spectral broadening, and enables noise pollution-free musical performances. A displacement-based digital compression algorithm caps maximum output air displacement as well as sound pressure level. Thus, an earbud is provided that through adjustments and modularity can act as a personal listening device, a hearing protection device and as a personal aesthetic statement with customized fit and comfort.

Authenticating a purported user attempting to access a secure resource includes enrolling a user's voice sample by requiring the user to orally speak preselected enrollment utterances, generating prompts and respective predetermined correct responses where each question has only one correct response, presenting a prompt to the user in real time, and analyzing the user's real time live response to determine if the live response matches the predetermined correct response and if voice characteristics of the user's live voice sample match characteristics of the enrolled voice sample.

A system to automate networked routing of audio in a live sound system includes a computing device having a processor and memory and in which the live sound system is configurable for a venue. The memory includes instructions executable by the processor to route audio signals over a network based on user selection of source audio signals from at least one transmitting audio device and on logical grouping of loudspeakers by the user from within a graphical user interface. The processor, for each receiving audio device channel of the group of loudspeakers (such as bandpass inputs), identifies power amplifiers and / or digital signal processors (DSPs) (which may be integrated within a powered loudspeaker) through which to route the selected source audio signals; and then routes the selected source audio signals from the selected transmitting audio device over the network to the receiving audio device channels through channels of the identified power amplifiers and / or DSPs.

An improved system and method to provide important information either, as voice, text, picture, holographic image or video from the calling party to a called party before the called party answers a telephone. The information may be a live or recorded voice, text, image or video message. The calling party may send a voice message, for example, and send it along with a call set-up message. The voice message may be recorded beforehand, recorded as part of the dialing process, or sent live before the call is answered. Instead of the called party's telephone ringing, the recorded voice message is played or the real-time data / bearer stream is heard. The called party hears the voice message instead of the ring tone normally heard when there is an incoming telephone call. If the telephone is not answered, the calling party is sent to voicemail or mms-storage system, where the calling party is given the option to record a voice message which may or may not include the original recorded or live voice message sent along with the call set-up message. Alternatively if the called party has a follow me service, the other contact numbers are attempted as directed in the user preferences before sending the calling party to voicemail. If the calling and called party have telephones, such as mobile phones, IP phones, IP softphones, PDA with telephony capabilities, etc., which allow for multimedia messages such as text, images and video messages, the calling party's recorded voice message may include text, images or video.

A computerized system for calling a plurality of cell phones in compliance with the TCPA, comprises a call center database; call center workstations operated by transfer agents; client workstations operated by client associates; and a computer processor that retrieves a cell phone number of a consumer to call, in response to an initiate call instruction from a respective transfer agent, dials the cell phone number, connects the call to the respective transfer agent if a live voice is detected; receives a transfer call request from the respective transfer agent, transfers the call with the one consumer to a client associate, using the cell number as the origination number of the call to identify the one consumer to the client associate, disconnects the connection between the one consumer and the respective transfer agent, and identifies a next consumer to call upon request by the transfer agent.

A method for listening to and comparing high-frequency sounds such as those produced by birds and insects by people suffering from high-frequency hearing loss, common among the aging population. Live sound is received by a microphone to produce an electrical signal. The signal is then digitized and transformed such that high frequency sounds are shifted to lower frequencies making them easier to hear. Recordings of known sources are also transformed for comparison.

A system allows an agent to execute an interaction protocol, to create an interaction between the agent and a contact. The agent or a computer dialing system initiates a call to a contact and the agent selectively interleaves responses to the contact with a live voice or with a recorded script. The interleaving of responses to the contact from the human agent and the recorded script approximates seamless transitions. The system includes a script player for outputting scripted voice waveforms over a phone line to a contact and a signal processor configured to provide a normalized signal selected from the output device and the script player.

The present invention, in one embodiment, is directed to a spam filter 132 that identifies, based on a likely interest of a user, electronic text messages addressed to the user as being spam or non-spam and determines a likely interest of the user using at least one of the following: (i) at least part of a search query provided to a search engine; (ii) content of at least one web page selected by the user during a browsing session; (iii) content of a cookie; (iv) a live voice electronic address; and (v) an entry in an electronic calendar of the user.

The invention discloses a voice interaction method, and the method comprises the steps: determining a first manual service agent after a call line between a call object and a call center system is insignal connection; searching a TTS real live voice library of the first manual service agent from a pre-stored TTS live voice library; calling a voice recognition module to perform voice recognition of a voice signal after receiving the voice signal of the call object, so as to obtain the call content of the call object; calling a preset voice response rule algorithm, and determining a response script file corresponding to the call content; calling the TTS live voice library of the first manual service agent, and replying the content of the response script file to the call object. The method employs the TTS live voice library to obtain a synthesized voice with a live voice tone, and effectively improves the language affinity and user experience. The invention also discloses the call centersystem, a voice interaction device and equipment thereof, and a computer readable storage medium, which also have the above-mentioned beneficial effects.

The invention discloses remote attendance equipment and a corresponding remote attendance system. In the remote attendance equipment, a moving part receives a control command from a control part; a video part collects on-site images and sends the images to a cloud server; an audio part collects on-site sounds and sends the sounds to the cloud server; and the audio and video parts receive audio and video signals from the cloud server for restoring sounds and images. The audio part adopts a multi-microphone array for collecting the on-site sounds. Sounds collected by each microphone are subjected to primary filtering, audio frequencies are subjected to wave beam processing for forming single signals, and then the single signals are subjected to secondary filtering, and thus the position of a main sound source is obtained. According to the invention, a user is allowed to start the remote attendance equipment at any place capable of accessing to the Internet, and to appear at an expected place for on-site communication. Thus, the user is prevented from frequently attending various on-site conferences, and wasted travel time is reduced.

The invention discloses a live broadcast method and device for chess games and a storage medium, and relates to the field of online live broadcast. The method comprises the following steps: obtainingchessboard synchronization data and voice live broadcast data of a chess game; Displaying a user interface of the application program according to the chessboard synchronous data, displaying a digitalchessboard on the user interface, and arranging a chess piece distribution diagram corresponding to the chessboard synchronous data on the digital chessboard; playing Live broadcast voice of the chess game according to the voice live broadcast data; And when an operation signal corresponding to the digital chessboard is received, pausing to synchronize the digital chessboard according to the chessboard synchronization data, and changing chess piece distribution on the digital chessboard according to the operation signal. Therefore, when a user listens to the live voice of the chess game, theuser can perform forward, backward and picture-changing operations on the digital chessboard by himself / herself, and meanwhile, the live listening requirement of the user and the chess research requirement are met.

An embodiment of the invention discloses an unmanned aerial vehicle (UAV) field sound acquisition method, an audio video realization method and related devices. The audio video realization method comprises the steps of: acquiring a sound signal of a UAV working environment; acquiring noise caused by the UAV in operation; filtering out the noise from the sound signal to obtain field sounds of the UAV working environment; acquiring video images of the UAV working environment; and synchronously synthesizing the field sounds and the video images of the UAV working environment to obtain an audio video. The UAV field sound acquisition method, the audio video realization method and the related devices can enable a user to shoot the audio video through operating a UAV, and the user experience is enhanced.

The invention discloses a literal live broadcasting method, comprising the steps of: performing voice recognition on live voice audio to generate ordered characters; determining the home of the generated ordered characters according to a preset voiceprint feature; and generating ordered characters including the home as live characters. The invention further discloses a literal live broadcasting platform.

Computer-generated personalized voice messages are created by concatenating data files of audio that were pre-recorded in the voice of an individual whose live voice is to be simulated during the delivery of the voice message. A call to a person or list of persons is placed by a computer. Common identifiers of each person to be called are read from a data base of data files, and matched with a separate data base containing recorded voice phrases, each of which is a digitization of the individual speaking content corresponding to the identifier—such as the person's first name. The recorded voice phrase audio is concatenated with at least one other audio file, which is a digitization of a message to be delivered to the called person.

To-be-commented video data is received by a terminal device, and a video sequence number of the to-be-commented video data is determined, which steps further include comprising, by the video data, a first correlation 2-tuple of a global base sequence number and the video sequence number, in which a sequence number of a live voice in the video data is the global base sequence number, and the video sequence number of the video data is determined according to the global base sequence number; and determining a video sequence number of the to-be-commented video data according to the first correlation 2-tuple. The to-be-commented video data is played, a commentary voice is acquired, and a correspondence between a voice sequence number of the commentary voice and the video sequence number is determined. The correspondence between the voice sequence number and the video sequence number is sent to a video playback end.

The invention discloses an ultrahigh pressure water jet velocity optimal control method extracted based on voice characteristics. A plurality of background noise signals are collected before a workpiece is cut, time domain characteristics and frequency domain characteristics are obtained through analysis, orthogonal wavelet decomposition of five scales is carried out on respective discrete series of selected N group water jet cutting field voice signals, approximate reflection voice signals are obtained, and amplitude envelope is determined; corresponding water jet cutting starting time and water jet cutting stopping time are obtained to obtain cutting time and calculate signal energy; the cutting time and the signal energy serve as input quantity, and optimum nozzle sidesway velocity is obtained though manual experience to serve as output quantity to build a least square support vector machine model; when the workpiece is cut, cutting field voice signals are collected on line, the cutting time and the signal energy are extracted to be input in the least square support vector machine model, the optimum nozzle sidesway velocity is obtained, and work efficiency of water jet cutting and quality of cut products are improved.

The invention relates to an association classroom realization method which comprises the steps of arranging an image positioning detector in a teaching classroom to collect real-time pictures of a teacher and students on site and unidirectionally output to a tracking host, allowing the tracking host to unidirectionally output a control signal to a direction control console according to an internalalgorithm, allowing the direction control console to output the control signal to a recording work station according to an internal control mode, allowing the recording work station to control a camera to collect and transmit video pictures of the teacher and the students to the recording work station in real time based on the control signal of the tracking host, allowing an intelligent tone tuning system to transmit collected site sound to the recording work station in real time, allowing the recording work station to transmit received audio and video signals to an interaction teaching server, and allowing the interaction teaching server to transmit the received audio and video signals or audio and video information taken from a comprehensive video application management platform to an interaction classroom. The association classroom system realization method can achieve remote teaching and obtains a good interaction effect.

The invention discloses a state detection method, device and equipment based on audio characteristics. According to the method, a state detection model is established through deep learning, and the state detection model is used for detecting whether operation field sound is the sound emitted by equipment in a normal operation mode, so that it is determined whether the operation of the working equipment is normal or not. According to the device and the equipment provided by the invention, the operation state of the equipment can be quickly judged, and a quantitative basis is provided for overhauling and safe operation of the equipment.

The invention discloses a visual cellar wine jar sealing system and a visual cellar wine jar sealing method based on an Internet of Things multifunctional label. The visual cellar wine jar sealing system comprises mobile networking equipment, the Internet of Things multifunctional label, a server and an Internet of Things singing recording pen, wherein an ultra-high-frequency passive RFID (radio frequency identification) chip is arranged in the Internet of Things multifunctional label, a one-dimension bar code, an in-site recording pen touch press key and a sign writing region are arranged on the Internet of Things multifunctional label, and an Internet of Things round code is arranged inside the sign writing region. The visual cellar wine jar sealing system and the visual cellar wine jar sealing method based on the Internet of Things multifunctional label solve the problems that the function of the existing RFID electronic label is single, and the user hand writing sign chirography data collection, the in-site sound data collection and the video collection are difficult to realize.

The invention discloses an AI control interaction system for a singing music fountain. The AI control interaction system comprises a wireless voice input unit, a song requesting unit, a power amplifier unit, a full-color large screen, a music input module, a multi-frequency audio signal collection controller, an AI main controller, a man-machine input interface, a digital control channel, an analogue control channel, a light audio control channel, a remote analogue-digital integrated processing function module, a remote digital-analogue control distribution board, a live voice-operated light drive array, a pump, a converter, a motor and digital quantity equipment, wherein the multi-frequency audio signal collection controller collects multiple segments of spectrum signals and transmits thespectrum signals to the AI main controller; and the AI main controller carries out spectrum analysis, communication control, analogue-digital output and light music control, processes and analyzes the spectrum signals and then separates out a digital control signal, an analogue control signal and a light color change control signal. According to the AI control interaction system, interaction of aman and the fountain can be achieved, and specific water form and light can be automatically turned on and manual editing is not needed.

The invention relates to a public broadcasting sound pressure self-adaptation control method. Through a magnitude squared coherence (MSC) function, the coherence of site sound signals and broadcast source frequency spectrums is computed, an obtained MSC coefficient is used as an MSC signal-to-noise ratio representation value to indirectly represent the site signal-to-noise ratio, and accordingly public broadcasting sound pressure is adjusted in a self-adaptation mode. According to the public broadcasting sound pressure self-adaptation control method, the MSC function is used for estimating the coherence of the site sound signals and the broadcast frequency spectrums, so that the site signal to noise ratio (SNR) information is obtained, under the condition that sound pollution is not increased, after sound pressure level adjusting, site mixed signals are obviously clearer than original site mixed signals, and broadcast content can be clearly heard.