317 results about "Speech technology" patented technology

A digital audio file search method and apparatus for digital audio files is provided that allows a user to navigate the audio files by generating speech sounds related to the information of the audio files to facilitate searching and playback. The digital audio file search method and apparatus searches for audio files in a portable digital audio player in combination with an automobile audio system through speech sounds by utilizing text-to-speech processing and by prompting response from a user in response to the generated speech sounds. The text-to-speech technology is utilized to generate the speech sound based on tag-data of the audio files. When hearing the speech sounds, the user gives instruction for searching the files without being distracted from driving the automobile.

A method is provided that allows data access service provider subscribers to manage their telephone service through a data connection. The subscriber is enabled to obtain call data information and is provided real time control. During a data call, a visual incoming call indicator informs the subscriber, through a popup window, connected to the data access service provider that there is a call attempt. A visual message waiting indicator allows a subscriber, connected to the data access service provider to be notified of a pending message on the voice message system. A visual call disposition allows the subscriber, through the data connection, to dispose of calls. The call disposition options include forwarding a call to voice mail, playing an announcement to the calling party, forwarding the call to another line, sending a text message which could be converted to speech using text to speech technology, answering the call using voice over data call or terminating the data connection in order to accept the call.

A digital audio file search method and apparatus for digital audio files is provided that allows a user to navigate the audio files by generating speech sounds related to the information of the audio files to facilitate searching and playback. The digital audio file search method and apparatus searches for audio files in a portable digital audio player in combination with an automobile audio system through speech sounds by utilizing text-to-speech processing and by prompting response from a user in response to the generated speech sounds. The text-to-speech technology is utilized to generate the speech sound based on tag-data of the audio files. When hearing the speech sounds, the user gives instruction for searching the files without being distracted from driving the automobile.

The present invention takes advantage of Voice over Internet Protocol (VoIP) technology by introducing VoIP-based call center telephony equipment that is software-based and runs on inexpensive off-the-shelf personal computer (PC) systems. With the VoIP-based call center system of the present invention, the traditional Public Switched Telephone Network (PSTN) is coupled to a Voice over Internet Protocol (VoIP) gateway in order to convert all incoming traditional telephone communication into VoIP based telephony telecommunication. This is performed using the well-known SIP telephony protocol set forth in RFC 3261. Once converted to the VoIP format, the incoming VoIP-based calls are directed to the VoIP based Call Center Server system. The Call Center Server system provides all the sophisticated call center features that were formerly only available in large call centers created with specialized expensive telephone equipment

Methods and systems for consolidating the workflow of various devices into a wireless, voice-enabled workflow. A method includes establishing a connection between a wireless, voice-enabled device and a data system using an interface and accessing the data system using voice commands via the connection between the wireless communication device and the data system. Voice commands may be used to facilitate data acquisition, data retrieval, order entry, dictation, audio playback, voice over IP conferencing, paging, and / or data analysis, for example. A plurality of connections may be established between the wireless, voice-enabled device and a plurality of data systems. WiFi wireless technology or other standard for voice and data transfer between devices without use of cables, for example, may be used to facilitate hands-free hygienic, centralized operation of a plurality of data systems using the wireless, voice-enabled device and the interface.

A voice-enabled system for online content access and delivery provides a voice and telephony interface, as well a text and graphic interface, for browsing and accessing requested content or shopping over the Internet using a browser or a telephone. The system allows customers to access an online data application, search for desired content items, select content items, and finally pay for selected items using a credit card, over a phone line or the Internet. A telephony-Internet interface converts spoken queries into electronic commands for transmission to an online data application. Markup language-type pages transmitted to callers from the online data application are parsed to extract selected information. The selected information is then reported to the callers via audio messaging. A voice-enabled technology for mobile multi-modal interaction is also provided.

A method for switching between speech recognition technologies. The method includes reception of an initial recognition request accompanied by control information. Recognition characteristics are determined using the control information and then a switch is configured based upon the particular characteristic. Alternatively, the switch may be configured based upon system load levels and resource constraints.

Systems and methods for providing uninterrupted transfer of voice telephony provided by a first service provider to a derived voice technology over a digital subscriber line provided by a second service provider are disclosed are disclosed. The system generally comprises a first telephone line configured to connect to a first and a second voice switch of the first and second service providers, respectively, having a same assigned telephone number and a derived voice customer premise equipment configured to connect to the first and second telephone lines and to selectively connect a telephone to the second voice switch. The method generally comprises establishing connectivity between a telephone and a first and a second voice switch of the first and second service providers via a first and a second line, respectively, having a same assigned telephone number and selectively connecting the telephone to the second voice switch via a client premise equipment.

A system method of advertising using a multi-media application system is disclosed. The multi-media application relates to the delivery of multi-media messages using animated entities that audibly deliver messages created by a sender using text-to-speech technologies. The method provides targeted advertising based on information learned about both the sender of a multi-media message and the recipient of the multi-media message. The information may relate to an analysis of a text message created by the sender, emoticons chosen by the sender and inserted into the text of the message, the choice by the sender of an animated entity, or other parameters such as background music chosen for which template is chosen by the sender. Advertising messages may be delivered before the recipient receives the multi-media message, during the reception by the recipient of the multi-media message or following the reception of the multi-media message. A decision regarding whether to include an advertising message may be based on a text analysis or an analysis of the emoticons or other tags inserted into the text by the sender. Further, animated entities such as professionally designed face models, templates, additional emoticons, animation or sound effects may also be purchased by the sender for a limited number of multi-media messages, for limited amount of time or longer for use in creating multi-media messages. The system comprises a server to handle the reception and processing of sender multi-media messages and client software for both creating multi-media messages and receiving multi-media messages.

This invention relates to the management, in an interactive voice response system, of a plurality of speech technology modules. In particular it relates to an apparatus and a method for dynamically determining which of a plurality of speech technology modules to use during voice interaction between the system and a user. In prior art IVR systems each speech technology module is configured for a specific application or task. Most speech technology modules have different lexicons for the range of functions but it is the full lexicon which can determine an engine's suitability for a language. For instance, one type of speech recognition engine is preferred for certain languages whereas IBM ViaVoice is a good general all rounder. Choosing one speech recognition module according to application or function alone is not entirely satisfactory and there is a need for improvement. The present solution is to select, for each interaction, one of the speech technology modules from the plurality of the modules to be used by the application according to the environment property of the interaction.

The process of text generation / creation is automated. The text to be processed is used as seed for the text generation process. The text to be processed can be in any language and can be passed to text generation process through any internal / external application or process, through speech technology or through manual entry. At the first step, word(s) are extracted from the text. Each word is considered as seed and this seed is grown up into different word(s) / sentence(s) lists according to the selected criteria. The generated lists are then processed and combined / jointed through a simple mechanism to generate text. This generated text then can be saved, analyzed, filtered or searched on the internet, intranet, extranet, in database(s) or in user defined data repositories again according to the criteria selected by the user or by some external application or process.

The invention discloses a pronunciation quality evaluation method of language study machine in the computer subsidiary language study and phonetic technique domain, which is characterized by the following: extracting exercise phonetic feature; exercising standard pronunciation model; forming standard pronunciation network; detecting phonetic end; extracting evaluation phonetic feature; searching optimum path; calculating the mark of pronunciation quality. The method displays objective and stable evaluation, which constitutes imbedded English study system and mutual human-machine education and oral English self-detection.

A hand held device that captures information with the capability to read only the captured information, display only the captured information, or simultaneously read and display the captured information. The device includes text-to-voice technology, a flat-panel display, a computer processor, a headphone for private receipt of transmitted information, microphone to receive dictated information, and storage. The device enables blind and / or visual impaired persons to read information anytime and anywhere.

The invention relates to a system suitable for a blind person to surf the Internet, comprising an automatic server news downloading system and a client application system. A server system can realize real-time news downloading, store the news into a server and realizes real-time updating; and a client system can correspondingly respond to voice input of the blind person by virtue of voice recognition and synthesis and outputs a voice. For special case of the blind person, the system realizes three core functions of the Internet, namely information acquisition, knowledge learning and interaction. By applying the system provided by the invention, the blind person can effectively listen to news by virtue of the Internet, listen to an electronic books, inquire encyclopedia nouns and report and listen to a post, wherein the news listening system can support the blind person to sequentially listen to news in programs, and the blind person can pay close attention to interested news by keyword search and related news search.

An examining and approving management system for administrative information comprises an automatic number-sending subsystem, a document-handling management subsystem, a process management subsystem, a certificate printing subsystem, a short message management subsystem, an one-card service subsystem, a reply subsystem, an electronic archive integration subsystem and a voice center subsystem. The examining and approving management system for the administrative information achieves coordinated service from certificate application to certificate printing for an applicant, builds complete personal information system for the applicant through intelligent card technology, omits repeated recording of the personal information, achieves multi-channel distribution and processing of the information through a digital certificate, a wireless application protocol (wap) network and independent voice technology, and overcomes shortcomings of dispersed former examining and approving departments, long consumed time, undefined examining and approving responsibilities and irregular charge, and has remarkable effects.

The present invention belongs to voice field of technology, pronunciation quality evaluation method of computer Computer Aided Language Learning system includes: calculation of the matching fraction, calculation of the sensing fraction based on Mel frequency scale, calculation of the segment length fraction and calculation of the keynote fraction, and processing fusion after mapping the above fractions; the pronunciation quality evaluation method of the invention has better robustness, high pertinency with the expert evaluation, used for interactive language learn and automatic spoken language test.

A medical dictation workflow system can be customized from the selection of available user application programs. A voice interface OCX can interface speech technologies with the selected user application programs of the medical dictation workflow system. The medical dictation workflow system may be directed to generating reports through filling out defined fields. The fields can be generated through a tracking system subscribing to a core reporting system and requesting certain information be captured or through a user. The voice interface OCX can provide macros so a user can customize the fields, navigate among the fields, or fill in the fields with data through a voice recognition engine or a wave player control. The data entered into the fields can be automatically entered into corresponding database elements of a database.

Systems and methods for offering base band voice telephony while using derived voice over data technology such as VoATM or VoIP are disclosed. The system may generally comprise a derived voice over data termination device located outside of the client premise and a connection between the client premise and the derived voice over data termination device, where the connection between the client premise and the derived voice over data termination device is a base-band analog voice loop. The derived voice over data termination device is configured to convert between base band signals and derived voice over data signals utilizing derived voice over data technology. The method generally comprises providing a derived voice over data termination device in a wire center, providing a base band connection between the client telephone and the derived voice over data termination device, transmitting base-band analog signals between the client telephone and the derived voice over data termination device, and transmitting derived voice over data signals between the derived voice over data termination device and a voice gateway connected to a public switched telephone network.

The invention discloses a voice over IP (VoIP) hiding communication method based on stream encryption, which belongs to the field of secure communication, is applied to the communication employing the voice over IP as carrier, and aims to improve the security performance of hiding communication while ensuring the real-time communication of VoIP. The method comprises the following steps: (1) the step of on-line negotiation; (2) the step of stream encryption; (3) the step of information hiding; and (4) the step of hidden information extraction. Before hiding the information in the VoIP stream, the hidden information is segmented and subjected to the bitwise XOR operation with a super-random number, thereby effectively preventing decryption and ensuring the security. The VoIP hiding communication method ensures the security while maintaining the real-time property of the VoIP system, and is applied to the transmission of large blocks of hidden data.

The invention discloses an administrative information mobile approval management system, which comprises an automatic number-issuing sub-system, an event-handling management sub-system, a flow management sub-system, a certificate printing sub-system, a short message management sub-system, a one-card service sub-system, a mobile approval sub-system, an electronic archive integrated sub-system and a voice center sub-system. Therefore, one-package service from event-handling application to certificate printing is provided for an applicant. By adoption of an intelligent card technology, a perfect personal information system is established for the applicant, and repeated recording of personal information is eliminated. By adoption of a digital certificate, a wireless application protocol (WAP) network and an automatic voice technology, information can be distributed in a multi-channel mode and processed. The administrative information mobile approval management system has the advantages that: the defects that approval departments are scattered, approval time is long, approval responsibility is not clear and fees are arbitrarily charged are overcome; the space limitation and time limitation of the conventional approval mode are eliminated; and an obvious effect is achieved.

Some embodiments relate to a cellular network which better utilizes packet-switched (PS) voice technologies, such as VoLTE, for roaming user equipment (UE) devices. When a roaming UE associated with a home cellular carrier that does not support PS to CS handover (SRVCC) desires to make a VoLTE call, the cellular network may determine probability of such a handover during the the call. The cellular network may selectively accept or reject the packet-switched wireless voice call based on the handover probability. If the probability of handover is high, the cellular network may reject the packet-switched wireless voice and trigger the UE to fall back to a circuit-switched network and re-originate the wireless voice call on the circuit-switched network. In the case of a mobile terminated call, the cellular network may provide signaling to the UE to perform a fallback to a circuit-switched network in order to receive the mobile terminated call.

There is provided a method of quantifying a voice quality in a telecommunication system including a first gateway in communication with a second gateway over a packet network. The method comprises deriving speech parameters from a first speech signal of a first talker received by the first gateway over a first communication line and a second speech signal of a second talker received by the first gateway from the second gateway over the packet network, determining a conversational impairment index using the speech parameters, deriving technology parameters based on voice technology and components in the telecommunication system, determining a technology impairment index using the technology parameters, and mapping the conversational impairment index and the technology impairment index into a conversational quality index to quantify the voice quality in the telecommunication system.

The invention provides pronunciation evaluation equipment, method and system, data processing equipment and method, voice processing equipment and method, and a mobile terminal, aiming at overcoming the defect that a provided pronunciation score is not accurate since the pronunciation importance of each word in a sentence is not differentially treated when the pronunciation situation of a user is evaluated in an existing voice technology. The pronunciation evaluation equipment comprises a user voice receiving unit, a score calculation unit, a word weight determination unit and a pronunciation evaluation unit, wherein the user voice receiving unit is used for receiving a user voice recorded by a user in allusion to a preset text; the score calculation unit is used for calculating the pronunciation score of a voice block, corresponding to each word of the preset text, in the user voice; the word weight determination unit is used for determining the weight of each word of the preset text on the basis of reference voice characteristics; the pronunciation evaluation unit is used for carrying out weighted calculation on the pronunciation score of the corresponding voice block of each word in the user voice in the sentence according to the determined weight so as to obtain the total score of the corresponding voice part of the sentence in the user voice. All the technologies provided by the invention can be applied to the technical field of voice.

The invention relates to the technical field of speech technology, and discloses a method and a system for processing users' speech signals. The method includes the steps of, by a server, receiving users' speech signals which are mixture of external speech received by a speech terminal through a microphone and double-tone multi-frequency key tone of the speech terminal; subjecting the received users' speech signals to spectral analysis by the server; judging whether the preset key is pushed or not during talking according to double-tone multi-frequency target frequency component corresponding to the preset key in the frequency spectrum; if the key is pushed, then determining that current user speech input is over. By the method and the system for processing users' speech signals, whether speech is over or not can be determined accurately effectively.

The invention relates to a deaf children speech rehabilitation method and a system based on three-dimensional head portrait and belongs to medical device field; the key technique of the invention includes that the three-dimensional modeling is combined with visual speech technology to create a parameter-driven three-dimensional lip movement model and a three-dimensional Chinese speech-aid visual voice database for deaf children rehabilitation, on the basis of creating three-dimensional conversation heat portraits, the speech recognition technology is combined with the image recognition technology to correct the pronunciation of the deaf children and help deaf children to rehabilitate Chinese speech ability.

The invention discloses a voice service control method, a voice service control device, a storage medium and an air conditioner. The method comprises the following steps of obtaining evaluation information fed back in the process of using voice service of equipment to be controlled by a user, wherein the evaluation information includes at least one of voice evaluation information, character evaluation information and press key evaluation information; and optimizing the voice service according to the evaluation information, so that the goal of using the optimized voice service by the user in anext time is achieved. By using the scheme, the problem of use inconvenience by the users due to user interaction inconformity of voice products and voice technology protocol differences can be solved; and the effects of improving the use convenience and accelerating the speed are achieved.

The invention discloses a voice interaction method and device, an electronic device and a storage medium, and belongs to the technical field of voice. The voice interaction method comprises the stepsthat a voice instruction of a user is analyzed, at least one piece of instruction information of the voice instruction, user portrait information of the user and emotion information of the user is obtained; based on at least one piece of instruction information of the voice instruction, user portrait information of the user and emotion information of the user, target sound attribute information isobtained; and target voice response with the target sound attribute information is provided. According to the voice interaction method and device, the electronic device and the storage medium, the voice instruction of the user is analyzed, at least one piece of instruction information of the voice instruction, user portrait information of the user and emotion information of the user is obtained;and what voice response of the sound is used for feeding back the voice instruction is determined based on at least one piece of information, the sound attribute of the voice response is not fixed, the relevance of the voice instruction with the voice response is high, interest, intelligence and diversity are additionally arranged in the voice interaction process, and the response effect is good.

A method for converting packet-based voice data of a first format directly to packet-based voice data of a second format, and vice versa. Data from networks using non-compatible packet-based voice technologies, for example, VoATM and VoIP, are interworked for direct conversion. Connection is set between an edge gateway of a first voice packet network, having data in a first format, and an interworking unit (IWU). Another connection is set between this IWU and an edge gateway of a second voice packet network, having data in the second format. The IWU is controlled by a single call agent that co-ordinates the conversion, at the IWU, between the two packet formats. Because it has this capability, this call agent is also called the “conversion server”. This call agent may be identical to the call agent used to control one or both edge gateways that use different packet based technologies.

A custom-content audible representation of selected data content is automatically created for a user. The content is based on content preferences of the user (e.g., one or more web browsing histories). The content is aggregated, converted using text-to-speech technology, and adapted to fit in a desired length selected for the personalized audible representation. The length of the audible representation may be custom for the user, and may be determined based on the amount of time the user is typically traveling.