摘要:
The spatial sound conference system enables participants in a teleconference to distinguish between speakers even during periods of interruption and overtalk, identify speakers based on spatial location cues, understand low volume speech, and block out background noise using spatial sound information. Spatial sound information may be captured using microphones positioned at the ear locations of a dummy head at a conference table, or spatial sound information may be added to a participant's monaural audio signal using head-related transfer functions. Head-related transfer functions simulate the frequency response of audio signals across the head from one ear to the other ear to create a spatial location for a sound. Spatial sound is transmitted across a communication channel, such as ISDN, and reproduced using spatially disposed loudspeakers positioned at the ears of a participant. By inserting a spatial sound component in a teleconference, a speaker other than the loudest speaker may be heard during periods of interruption and overtalk. Additionally, speakers may be more readily identified when they have a spatial sound position, and the perception of background noise is reduced.
摘要:
The spatial sound conference system enables participants in a teleconference to distinguish between speakers even during periods of interruption and overtalk, identify speakers based on spatial location cues, understand low volume speech, and block out background noise using spatial sound information. Spatial sound information may be captured using microphones positioned at the ear locations of a dummy head at a conference table, or spatial sound information may be added to a participant's monaural audio signal using head-related transfer functions. Head-related transfer functions simulate the frequency response of audio signals across the head from one ear to the other ear to create a spatial location for a sound. Spatial sound is transmitted across a communication channel, such as ISDN, and reproduced using spatially disposed loudspeakers positioned at the ears of a participant. By inserting a spatial sound component in a teleconference, a speaker other than the loudest speaker may be heard during periods of interruption and overtalk. Additionally, speakers may be more readily identified when they have a spatial sound position, and the perception of background noise is reduced.
摘要:
A server in a network gathers textual information, such as news items, E-mail and the like. From that information, the server develops or identifies messages for use by individual subscribers. The same server that accumulates the text messages or another server in the network converts the textual information in each message to a sequence of speech synthesizer instructions. The converted messages, containing the sequences of speech synthesizer instructions, are transmitted to each identified subscriber's terminal device. A synthesizer in the terminal generates an audio waveform signal, representing the speech information, in response to the instructions. In the preferred embodiment, the terminals utilize concatenative type speech synthesizers, each of which has an associated vocabulary of stored fundamental sound samples. The instructions identify the sound samples, in order. The instructions also provide parameters for controlling characteristics of the signal generated during waveform synthesis for each sound sample in each sequence. For example, the instructions may specify the pitch, duration, amplitude, attack envelope and decay envelope for each sample. The division of the text to speech synthesis processing between the server and the terminals places the cost of the front end processing in the server, which is a shared resource. As a result, the hardware and software of the terminal may be relatively simple and inexpensive. Also, it is possible to upgrade the quality of the synthesis by upgrading the server software, without modifying the terminals.
摘要:
A method and system is disclosed for accessing a remote personalized secretarial platform that permits a wide variety of functions with high flexibility, while being easily usable by an individual telephone subscriber. The platform can be accessed whenever the subscriber telephone is off-hook through a voice recognition monitor that monitors the subscriber line and is responsive to a preselected utterance to generate an access signal. Placed at the telephone switch facility, a monitor module is speech responsive individually to a plurality of lines that are off hook to generate signals that effect switch functions including bridging to the platform and modifying a subscriber switch feature profile.
摘要:
A voice response unit (VRU) includes a library of content equivalent messages and prompts which may be substituted for one another to vary the presentation of messages provided to a user and thereby more closely simulate a human operator. Groups of content equivalent messages and prompts include multiple audio files, each with a slightly different wording or phraseology, but conveying substantially the same information. After a particular message content is selected, the corresponding group of messages is identified and a random number is generated and used to select one of the audio files of the group for playback. The VRU may be included as part of an automated dialer or auto attendant. In such a system, a calling party is greeted by the VRU and is prompted by a randomly selected prompt to speak the name of the called party. The system accesses a telephone directory, attempts to identify a name corresponding to the name spoken, and dials the number. The caller may interrupt or request alternative processing during a predetermined time period after the system has selected and read back a closest matching name or its corresponding telephone number. If processing is halted by the caller indicating that the name or telephone number selected by the system is incorrect, the system will attempt to identify a second closest guess, or if none is available, to ask the caller to reinput the name of the called party. Alternative processing includes hearing the telephone number without having it dialed, and diverting a call to voice mail.
摘要:
A telephone communications system Advanced Intelligent Network (AIN) platform provides a voice activated call dialing functionality through speaker independent phoneme speech recognition having a minimum volume of storage without requiring user template training. Speaker independent phoneme recognition identifies phoneme strings of caller spoken utterances which are then compared to phoneme string representations that previously have been stored in respective caller processing records (CPRs) for those subscribers listed in the ISCP database, or stored in an equivalent peripheral database with which the ISCP can communicate. Each stored phoneme string representation is associated in the CPR with a destination telephone number that may then be extracted to route a call.
摘要:
A mechanized directory assistance system for use in a telecommunications network includes multiple speech recognition devices comprising a word recognition device, a phoneme recognition device, and an alphabet recognition device. Also provided is a voice processing unit and a computer operating under stored program control. A database is utilized which may comprise the same database as used for operator directory assistance. The system operates as follows: A directory assistance caller is prompted to speak the city or location desired. The response is digitized and simultaneously inputted to the word and phoneme recognition devices which each output a translation signal plus a probability level signal. These are compared and the highest probability level translation is selected. The caller is prompted to speak the name of the sought party. The response is processed in the same manner as the location word. In the event that the probability level fails to meet a predetermined standard the caller is prompted to spell all or part of the location and/or name. The resulting signal is inputted to the alphabet device. When translations are obtained having a satisfactory probability level the database is accessed. If plural listings are located these are articulated and the caller is prompted to respond affirmatively or negatively as to each. When a single directory number has been located a signal is transmitted to the caller to articulate this number. The system also includes provision for DTMF keyboard input in aid of the spelling procedure.
摘要:
A communication service loads subscriber profile information corresponding to an intended call recipient in a call processing register of a switch serving a link to a destination station and transmits a distinct predetermined call alert signal over the link to the destination station based on the loaded subscriber service profile information. The distinct call alert signal identifies the intended recipient subscriber.
摘要:
Personal dial tone service is used to identify the user of a subscriber line to a telephone terminal and, based on that identification, the system and method dynamically configures that line with the personal profile of that user. Such a line is used in a roaming situation to provide voice mail service to the roamer through an emulation of the roamer's home voice mail interface. The emulation is accomplished by storage at the home locale of the roamer of object oriented script associated with both executable and non-executable data duplicating or emulating executable and non-executable data in the roamer's home voice mail system. The script directs the running of the executables using the non-executables at the roaming central office to provide to the roamer at that remote office voice mail service using virtually the same interface as the interface to which the roamer is accustomed at his home locale. The script is stored in an Intelligent Peripheral wherein the executables are run pursuant to the script. Voice mail messages may be stored either in the remote or home locals.
摘要:
A personal area network (PAN) device enables the communication of data using galvanic properties of the skin. A person can wear a processor coupled to a PAN device. When the person touches a sensor capable of communicating with the PAN, the processor sends and receive data through the PAN and the sensor. In accord with the invention, the processor stores personal information related to the wearer's telephone service, such as the person's identification and billing information. The processor also may store information relating to the person's telephone subscriber profile, defining that person's individualized telephone services. When the wearer touches a sensor on a pay telephone, the processor supplies the data through the PAN and the sensor to a processor in the telephone. The telephone communicates the data through the telephone network, to enable the network to provide personalized services. For example, the network uses the billing information to bill any calls that the person makes to the person's normal telephone account, in a manner analogous to a credit card type billing procedure. A feature of the invention is that virtually positive identification of a person is implemented preferably using biometric characteristics of the actual caller.