Abstract: The present disclosure relates to an intelligent network management system. The intelligent network management system includes: a plurality of terminals, including at least one management terminal and at least one application terminal; and at least one server, coupled to the management terminal and the application terminal, and including a data management circuit, a service management circuit, and an application management circuit, the management terminal being configured to send a data management signal, a service management signal and an application management signal to the server, the data management circuit performing data layer management on the application terminal in response to the data management signal, the service management circuit performing service layer management on the application terminal in response to the service management signal, and the application management circuit performing application layer management on the application terminal in response to the application management signal.

Abstract: A method for providing a context awareness service is provided. The method includes defining a control command for the context awareness service depending on a user input, triggering a playback mode and the context awareness service in response to a user selection, receiving external audio through a microphone in the playback mode, determining whether the received audio corresponds to the control command, and executing a particular action assigned to the control command when the received audio corresponds to the control command.

Abstract: A system for synchronizing a local audio processing clock rate of a digital signal processor (DSP) to an audio clock rate of a network to which the DSP is connected. The system includes an adjustable clock synthesizer that is configured to establish the local audio processing clock rate of the DSP. The DSP is configured to generate events that are associated with the local audio processing clock rate of the DSP. The DSP is further configured to monitor the generated events over time and based on the monitored events cause the adjustable clock synthesizer to adjust the local audio processing clock rate of the DSP to better match the network audio clock rate.

Abstract: Methods and systems for providing message playback using a shared electronic device is described herein. In response to receiving a request to output messages, a speech-processing system may determine a group account associated with a requesting device, and may determine messages stored by a message data store for the group account. Speaker identification processing may also be performed to determine a speaker of the request. A user account associated with the speaker, and messages stored for the user account, may be determined. A summary response indicating the user account's messages and the group account's message may then be generated such that the user account messages are identified prior to the group account's messages. The messages may then be analyzed to determine an appropriate voice user interface for the requester such that the playback of the messages using a shared electronic device is more natural and conversational.

Abstract: Techniques for processing a voice initiated request by a web server are presented. The techniques may include receiving, by a web server, request data representing a voice command to a user device, the request data including an identification of a requested webpage; determining, by the web server, that a response to the request data will continue a voice interaction; and providing, by the web server and to the user device, data for a voice enabled webpage associated with the requested webpage, where the data for the voice enabled webpage is configured to invoke a voice interface for the user device.

Abstract: Various embodiments include methods and devices for implementing automatic grammar augmentation for improving voice command recognition accuracy in systems with a small footprint acoustic model. Alternative expressions that may capture acoustic model decoding variations may be added to a grammar set. An acoustic model-specific statistical pronunciation dictionary may be derived by running the acoustic model through a large general speech dataset and constructing a command-specific candidate set containing potential grammar expressions. Greedy based and cross-entropy-method (CEM) based algorithms may be utilized to search the candidate set for augmentations with improved recognition accuracy.

Abstract: Query history expansion may be provided. Upon receiving a spoken query from a user, an adapted language model may be applied to convert the spoken query to text. The adapted language model may comprise a plurality of queries interpolated from the user's previous queries and queries associated with other users. The spoken query may be executed and the results of the spoken query may be provided to the user.

Abstract: Methods and systems for voice and video messaging are provided here. In some embodiments a method for voice and video messaging may include establishing a telecommunication session with a calling device associated with a first user in response to an indication that a call request from the calling device directed to a called identifier associated with a second user failed, providing to the calling device an option to leave a video message and an option to leave an audio message, receiving a selection from the calling device indicating the selection of the option to leave a video message, disconnecting the established telecommunication session, and receiving a video message from the calling device.

Abstract: Query history expansion may be provided. Upon receiving a spoken query from a user, an adapted language model may be applied to convert the spoken query to text. The adapted language model may comprise a plurality of queries interpolated from the user's previous queries and queries associated with other users. The spoken query may be executed and the results of the spoken query may be provided to the user.

Abstract: A microprocessor includes a memory interface to obtain data envelopes of a first length, and control logic to implement an instruction to load an initial data envelope of a stream of data values into a buffer, each data value having a second length shorter than the first length, the stream of data values being disposed across successive data envelopes at the memory interface. Another instruction merges current contents of the buffer and the memory interface such that each invocation loads one of the data values into a first register, and moves at least a remainder of the current contents of the memory interface into the buffer for use in a successive invocation. Another instruction loads a reversed representation of a set of data values obtained via the memory interface into a second register. Another instruction implements an FIR computation including a SIMD operation involving multiple data values of the stream and the reversed representation.

Abstract: A method and a device for data processing in a digital subscriber line environment, wherein a flat noise margin is determined based on an actual noise determined and based on a maximum noise expected. A device is initialized using the flat noise margin. Also, a communication system is described that includes such a device.

Abstract: Described is a system and method for processing a voice mail. The method comprises receiving a voice mail, converting the voice mail into a text message using a predefined speech-to-text conversion algorithm and transmitting the text message to a wireless computing device.

Abstract: A method and a processing device for managing an interactive speech recognition system is provided. Whether a voice input relates to expected input, at least partially, of any one of a group of menus different from a current menu is determined. If the voice input relates to the expected input, at least partially, of any one of the group of menus different from the current menu, skipping to the one of the group of menus is performed. The group of menus is different from the current menu include menus at multiple hierarchical levels.

Abstract: A computer-implemented system and method for processing audio in a voice response environment is provided. A database of host scripts each comprising signature files of audio phrases and actions to take when one of the audio phrases is recognized is maintained. The host scripts are loaded and a call to a voice mail server is initiated. Incoming audio buffers are received during the call from voice messages stored on the voice mail server. The incoming audio buffers are processed. A signature data structure is created for each audio buffer. The signature data structure is compared with signatures of expected phrases in the host scripts. The actions stored in the host scripts are executed when the signature data structure matches the signature of the expected phrase.

Abstract: A multimedia conferencing system includes a loud speaker system, one or more microphones for receiving a local audio signal and a remote audio signal, a state machine and an echo canceller that operates in conjunction with two reference signals to remove substantially all of a feedback signal component in the local audio signal that results from reinforcing and playing the local audio signal over the loud speaker system. The state machine operates to detect that only the local audio is active, and if so controls the operation of the echo canceller such that only the feedback component of the local audio signal is removed and the local audio signal is not suppressed.

Abstract: A method and a processing device for managing an interactive speech recognition system is provided. Whether a voice input relates to expected input, at least partially, of any one of a group of menus different from a current menu is determined. If the voice input relates to the expected input, at least partially, of any one of the group of menus different from the current menu, skipping to the one of the group of menus is performed. The group of menus different from the current menu include menus at multiple hierarchical levels.

Abstract: A device for voice identification including a receiver, a segmenter, a resolver, two advancers, a buffer, and a plurality of IIR resonator digital filters where each IIR filter comprises a set of memory locations or functional equivalent to hold filter specifications, a memory location or functional equivalent to hold the arithmetic reciprocal of the filter's gain, a five cell controller array, several multipliers, an adder, a subtractor, and a logical non-shift register. Each cell of the five cell controller array has five logical states, each acting as a five-position single-pole rotating switch that operates in unison with the four others. Additionally, the device also includes an artificial neural network and a display means.

Abstract: Systems and methods for secure recording in a customer center environment are provided. The system receives data related to a communication at a media distribution device. A key is provided by a key server to the media distribution device to encrypt the received data. In addition, an identifier may be associated with the received data. The key and the identifier are stored in a database associated with the key server. The encrypted data is recorded the a recorder, where it is accessible to authorized users within the customer center. Upon request, the key is provided to the authorized user to decrypt the encrypted data for play back. The customer center environment may include a user interface for viewing customer contacts. The contact may be organized into folders and annotations associated with the customer contacts may be received.

Abstract: A DFE circuit for use in a semiconductor memory device and an initializing method thereof. In the method of initializing a DFE circuit used in a semiconductor memory device having a discontinuous data transmission, the DFE circuit may be used for changing a sampling reference level in response to a level of previous data and sampling transmission data. The method includes terminating a data channel having a transmission of the transmission data at a predefined termination level, and controlling a sampling start time point of the transmission data as a time point preceding a transmission time point of the transmission data by a predefined time. Further, an initialization may be performed of the previous data on the basis of initialization data obtained through a pre-sampling of the data channel at a sampling start time point of the transmission data, thereby obtaining an initialization of the DFE circuit and compensating for a feedback delay.

Abstract: This disclosure relates to noise suppression in data transfers in transceivers of wireless devices.

Abstract: A method includes receiving data in a first data processing module, and enabling a second data processing module when at least one signal time slot of the received data comprises data that complies with a first data transmission standard. The method also includes exchanging signals between the first data processing module and software executing in a processor, and determining that a software configuration of the second data processing module has been completed. The method also includes processing the data in the second data processing module for the at least one signal time slot, and enabling a third data processing module upon a completion of processing at least one data block in the second data processing module, and determining that a software configuration of the third data processing module has been completed, the at least one data block comprising multiple signal time slots.

Abstract: A network appliance (100) is provided having a network controller subsystem (110) for coupling the appliance (100) to a data network for providing and receiving data packets to and from a packet data network. A digital signal processing subsystem (120) is coupled to the network controller subsystem (110). A signal conversion subsystem (130) is coupled to the digital signal processing subsystem (120) and a user interface subsystem (160) is coupled to both the signal conversion subsystem (130) and the digital signal processing subsystem (120). The digital signal processing subsystem (120) operates under the control of a computer program which is capable of detecting incoming calls, initiating call sessions, and preferably, implementing advanced telephony features.

Abstract: A method may include selecting a voice mail out-dial access number corresponding to a local access transport area (LATA) boundary, storing the voice mail out-dial access number on a voice mail system, and providing, by the voice mail system, voice mail out-dial service based on the LATA boundary associated with the voice mail out-dial access number. A method for providing toll-free access to the voice mail system that may be used by customers from both on-network and off-network end points.

Abstract: The voice mixing method includes a first step for selecting voice information from a plurality of voice information, a second step for adding up all the selected voice information, a third step for obtaining a voice signal totaling the voice signals other than one voice signal, of the selected voice signals, a fourth step for encoding the voice information obtained in the second step, a fifth step for encoding the voice signal obtained in the third step, and a sixth step for copying the encoded information obtained in the fourth step into the encoded information in the fifth step.

Abstract: Described is a system and method for processing a voice mail. The method comprises receiving a voice mail, converting the voice mail into a text message using a predefined speech-to-text conversion algorithm and transmitting the text message to a wireless computing device.

Abstract: The subject matter of this disclosure can be implemented in, among other things, a device that includes processors, means for receiving a voicemail greeting template having a default portion and a placeholder portion, wherein the default portion is included in voicemail greetings generated based on the voicemail greeting template, and a user status propagation module executable by the one or more processors. In these examples, the user status propagation module receives status information for a user or a device, wherein the status information includes a state of the user or a state of the device, and replaces the placeholder portion of the voicemail greeting template with the received status information to generate a personalized voicemail greeting of the user that includes the default portion and the received status information, wherein output of the personalized voicemail greeting is initiated in response to an incoming phone call.

Abstract: Apparatus and method to allow retrieval of voice messages deleted from the voice message memory of a voice messaging system. A voice messaging system such as a telephone answering device includes a deleted voice message memory for storing voice messages deleted from the voice message memory. The deleted voice messages stored in the deleted voice message memory are retrievable by the user for review subject to rules for permanent deletion of the deleted voice messages (e.g., after a period of time, when the deleted voice message memory approaches capacity, periodically, etc.).

Abstract: A computer-implemented voice mail method includes obtaining an electronic audio file of a recorded user message directed to a telephone user, automatically generating a transcript of the recorded user message, and identifying locations in the transcript in coordination with timestamps in the recorded user message so that successive portions of the transcript can be highlighted in coordination with playing of the recorded user message. The method also include identifying one or characteristics of the message using meta data relating to the recorded user message, and storing the recorded user message and information about the identified locations of the recorded user message.

Abstract: A method is provided for multi-pass echo residue detection. The method includes detecting audio data, and determining whether the audio data is recognized as speech. Additionally, the method categorizes the audio data recognized as speech as including an acceptable level of residual echo, and categorizes categorizing unrecognizable audio data as including an unacceptable level of residual echo. Furthermore, the method determines whether the unrecognizable audio data contains a user input, and also determines whether a duration of the user input is at least a predetermined duration, and when the user input is at least the predetermined duration, the method extracts the predetermined duration of the user input from a total duration of the user input.

Abstract: A telephone call is established between a local telephone (110) associated with a local party and a remote telephone (104) associated with a remote party. The local telephone (110) receives a voice signal from the remote telephone (104) responsive to establishing the telephone call. The voice signal represents directory assistance information, a name, a phone number or directions provided by the remote party. The local telephone (110) stores the voice signal in a memory unit (212) in the local telephone (110) responsive to receiving the voice signal. The local telephone (110) converts the voice signal to a text signal responsive to storing the voice signal. The local telephone (110) stores the text signal in the memory unit (214) in the local telephone (110) responsive to converting the voice signal to a text signal. The local telephone (110) edits the stored text signal in the memory unit responsive to storing the text signal.

Abstract: A system and method for creating and distributing messages including dynamic content is provided. In one form, an entity provides a plurality of content segments for use in building a dynamic message. The entity also provides a listing of end users to receive a message and information associated with the user, such as demographic or prior purchasing information. In an alternate form, the service connects to the user in a communication session and collects information from the user directly. The service then uses this information, whether provided by the entity or the end user, as the basis for selection of various content segments which will make up a dynamic message for presentation to the user.

Abstract: An intelligent image smoke/flame/complex intelligent image smoke/flame sensor, and an intelligent image smoke/flame detection system formed of such an intelligent image smoke/flame/complex intelligent image smoke/flame sensor. The intelligent image smoke/flame sensor is formed of an image sensor, for example, CCD (Charge-coupled Device), a DSP Processor, an input/output circuit, an infrared lamp and filter assembly, a casing, a window glass, a front cover, and a rear cover. The intelligent image smoke/flame detection system accurately detects fire characteristics including smoke and flame, producing an early alarm to lower catastrophe.

Abstract: A computer implemented method and computer program product for managing conversational input. The process authenticates a user on a device in a set of communication devices for manipulating conversation stubs. The process monitors an exchange of conversation for a triggering event on the device in a set of communication devices. Responsive to detecting the triggering event, the process generates a conversation stub that contains information relating to the topic of conversation. The conversation stub is stored in a storage device located remotely to the set of communication devices. Responsive to receiving a selection of the conversation stub, the process presents the conversation stub on a communication device selected from the set of communication devices.

Abstract: A method for providing a context specific speaker adaptation user interface is disclosed. The method allows a specific user of a voice activated system to train a voice recognition system to understand specific uttered commands by receiving a training sequence initiation input, sending at least one command to be trained, receiving a selection input, prompting a verbal input, recording the verbal input, and linking a voice input with an instruction corresponding with the command input.

Abstract: Apparatus and method for simultaneously providing multiple telephone-type services to any/all POTS-type devices on a single wire pair at a user premises. The present invention provides for the ability to add separately addressable POTS devices on a single service loop. This can be accomplished in at least two ways: first by the use of a multipoint protocol or second by Frequency Division Multiplexing.

Abstract: A system and a method are disclosed for facilitating sending of DTMF (dual-tone multi-frequency) signals to a destination after connecting to a system by placing a phone call. An entry associated with an application running on a mobile computing device is automatically scanned to detect a telephone number and any string of characters associated with the telephone number. The mobile computing device presents one or more candidate strings of characters that are associated with the telephone number. A user may select the strings of characters to send the DTMF signals after connecting to a destination telephone system. By automatically extracting and displaying the string of characters associated with the telephone number, the user's manual operation associated with accomplishing a task on a remote telephone system is minimized.

Abstract: A media system is provided. The media system may include a media player, an input, and a processor. The media player may have a scrub bar and a playhead. The scrub bar may be operable to represent a time line of a media recording that is being played back and the playhead may indicate a portion of the media recording that is being played back. The input may be operable to receive user behavior. The processor may be operable to dynamically adjust a speed of playback of the media recording based on the user behavior.

Abstract: A joint detection system and associated methods are provided. The joint detection system is configured to perform joint detection of received signals and includes a joint detector accelerator and a programmable digital signal processor (DSP). The joint detector accelerator is configured to perform front-end processing of first data inputted to the joint detector accelerator and output second data resulting from the front-end processing. The joint detector accelerator is further configured to perform back-end processing using at least third data inputted to the joint detector accelerator. The programmable DSP is coupled to the joint detector accelerator, and the programmable DSP is programmed to perform at least one intermediate processing operation using the second data outputted by the joint detector accelerator. The programmable DSP is further programmed to output the third data resulting from the intermediate processing operation to the joint detector accelerator.

Abstract: A method for providing menu tree assistance includes receiving a voice request from a user via a telematics unit and a wireless network and determining a voice menu tree based on the voice request. The method further includes receiving at least one response based on the voice menu tree, determining a failure or confirmation based on the at least one response and converting the response to data based on the determined confirmation. The method further includes recording the response based on the determined failure and providing the recorded response and data to an advisor at a call center. A system and a computer readable medium including computer program code are also disclosed.

Abstract: A method and apparatus for managing protocol information used in exchanging communications is provided. One or more protocol profiles are stored. Each of the protocol profiles is associated with a protocol. One or more transaction profiles are stored. Each of the transaction profiles identifies a combination of protocol profiles. One or more trading partner transaction profiles are stored. Each of the trading partner transaction profiles identifies a configuration of protocols for exchanging communications with a receiving party. After storing the profiles, a particular parameter value of a particular protocol profile is updated to identify an updated value without updating any of the transaction profiles. Updating the particular parameter value causes the updated value to be inherited by at least one of the transaction profiles. An application may obtain the updated value by accessing a profile that references the particular protocol profile.

Abstract: An audio recorder comprises: a microphone for receiving analog audio signals; a removable memory cartridge for digitally storing audio message data entries; and a microcontroller for converting the analog audio signals to digital signals and storing the digital signals in the removable memory cartridge.

Abstract: Voice conversations by way of communications devices are conducted by transmitting symbols representative of a user's voice from a transmitting communications device and recreating the user's voice at a receiving communications device. The communications devices each include a processing engine responsive to a user's voice input for generating speech sample data indicative of predetermined portions of the user's voice. A storage device is coupled to the processing engine and stores the speech sample data. The processing engine also includes a communication module that generates transmission data, indicative of the user's voice spoken during a communication session as a function of the speech sample data and causes transmission of the transmission data to a remotely located recipient of the communication session.

Abstract: Apparatus and method to allow retrieval of voice messages deleted from the voice message memory of a voice messaging system. A voice messaging system such as a telephone answering device includes a deleted voice message memory for storing voice messages deleted from the voice message memory. The deleted voice messages stored in the deleted voice message memory are retrievable by the user for review subject to rules for permanent deletion of the deleted voice messages (e.g., after a period of time, when the deleted voice message memory approaches capacity, periodically, etc.).

Abstract: An audio message recording and playback system including a recorder-imprinter device, a self-contained voice module for playback of an audio message and a number of input devices for initially recording the audio message to the recorder-imprinter device. An internet web site is established for taking orders from customers, where customer data such as billing and shipping information is obtained. The web site further includes software that permits a user to create a voice recording to be delivered along with an accompanying product such as a greeting card, gift, flowers, or an ornament. The voice recording is accessed by a proprietor who confirms the billing and delivery information, and then transfers the voice recording to a recorder-imprinter that creates a transferable voice module for incorporation into the intended delivery.

Abstract: A telephone and voice mail (voice processing) system is implemented using only a single processing means for controlling operations of both the telephone system and the voice mail system. The single processing means communicates with a hard disk, which stores programs for running the various operations of the system, voice prompts and all voice mail messages. The single processing means is coupled to signal processing circuitry, which emulates analog electronics that would be used for filters, tone decoder, generators, etc. The single processing means and signal processing circuitry are coupled to central office lines and station lines by a digital cross-point matrix, which can connect any voice path to another voice path. The system integrates call processing and voice processing into one system controlled by one set of software and a single processing means.

Abstract: An short message service switching private branch exchange system and a method can reduce costs. The method can include certifying whether a usable DSP exists or not when an office line and a speech path are connected to each other, and transmitting an short message service signal transmitted from the office line to the DSP through a PCM channel if the usable DSP exists. A main processor of the private branch exchange can certify an extension line terminal that will receive short message service, generate an short message service signal having a format corresponding to a kind of the extension line terminal by using the DSP, and transmit the formatted SMS signal to an extension line interface unit of a corresponding extension line terminal. The system switches an SMS signal between an office line and an extension line by sharing the DSP, for example, by time division.

Abstract: A method and apparatus for building a training set for an automated speech recognition-based system, which determines the statistically optimal number of frequently requested responses to automate in order to achieve a desired automation rate. The invention may be used to select the appropriate tokens and responses to train the system and to achieve a desired “phrase coverage” for all of the many different ways human beings may phrase a request that calls for one of a plurality of frequently-requested responses. The invention also determines the statistically optimal number of tokens (spoken requests) required to train a speech recognition-based system to achieve the desired phrase coverage and optimal allocation of tokens over the set of responses that are to be automated.

Abstract: The invention relates to a distribution device (2) in a data signal processing installation (1), and a data signal processing installation (1). The distribution device comprises a distribution block (3) which has a receiving device containing elements to which data signal lines (4, 5, 6) can be connected, the functional elements having a circuit to distribute signals transmitted by the data signal lines (4, 5, 6). The distribution device also comprises a data signal editing unit with active and/or passive electronic components, in which the data signals transmitted from the data signal lines (3, 4, 5) are edited in a pre-determined manner. The data signal editing unit is integrated into the components of the distribution block (3).

Abstract: A system and method are disclosed for the licensing and managing media resources in a converged services platform. A novel central pool of resource points representing a licensed level of media resource capability is created, the central pool for the storage of all default and any additional licensed resource points represented by the converged services platform, where portions of the central pool are dynamically allocated to one or more media resource cards.

Abstract: An apparatus for taking an electronic message from a user and actively notifying a receiver is disclosed. The apparatus includes a message input device for receiving the electronic message from the user, a message storing device for storing the electronic message therein, a message playing device for playing the electronic message, a vibration detector for detecting a certain level of vibration, a controller electrically connected to the message input device, the message storing device, the message playing device and the vibration detector, and a case accommodating the message input device, the message storing device, the message playing device, the vibration detector and the controller therein. The vibration detector sends out a trigger signal in response to the vibration, and the controller allows the electronic message to be read out from the message storing device and played by the message playing device in response to the trigger signal.