Abstract:
The audio signals associated with different co-located groups of talkers in a teleconference are detected (e.g., by comparing the voiceprint for the current talker group with stored voiceprints corresponding to all of the co-located teleconference participants) and processed using different and appropriate automatic gain control (AGC) levels, where each group has a corresponding stored AGC level. Depending on the embodiment, each group may have one or more participants.
Abstract:
The present invention provides a multi-party communication system in which the speaking party can be identified aurally and the speech contents can be accurately transmitted to the party at the receiving terminal. A multi-party communication server and a plurality of terminal devices with a communication function make up the multi-party communication system. Each terminal device with the communication function includes a speech right management unit, a speaking party name output unit and a buffer unit. The speaking party name output unit outputs the voice data of the speaking party identification information such as the name of the speaking party. The buffer unit accumulates the speech voice of the user as voice data. The speech right management unit controls the buffer unit to produce an output after the speaking party output unit. The speech right management unit issues a request to cancel the right to speak after completion of the output of the speech voice data.
Abstract:
A method for the voice-operated identification of the user of a telecommunications line in a telecommunications network is provided in the course of a dialog with a voice-operated dialog system. Utterances spoken by a caller from a group of callers limited to one telecommunications line are used during a human-to-human and/or human-to-machine dialog to apply a reference pattern for the caller. For each reference pattern, a user identifier is stored which is activated once the caller is identified, and, together with the CLI and/or ANI identifier of the telecommunications line, are made available to a server having a voice-controlled dialog system. On the basis of the CLI, including the user identifier, data previously stored for this user are ascertained by the system and made available for the dialog interface with the customer.
Abstract:
A voice authentication system using a removable voice ID card comprises: at server side, a voiceprint database for storing the voiceprints of all authorized users; a voiceprint updating means for updating the voiceprints in said voiceprint database; and a voiceprint digest generator for generating a voiceprint digest according to a request from a client; at client side, a voice ID card for storing the voiceprint of an authorized user; a validation means for validating the voiceprint in the voice ID card on the basis of the voiceprint digest from the server; an audio device for performing voice interaction with a user; and a voice authentication means for determining whether the voiceprint from said voice ID card is of the same speaker as the voice from said audio device. The present invention can significantly avoid the abuse of a voice ID card when it is lost or stolen by using the voiceprint digest stored at server side to verify the voiceprint in the voice ID card.
Abstract:
A home gateway system has a transceiver (70) capable of establishing a wireless local loop connection (72). A voice processing system (74) is coupled to the transceiver (70). The voice processing system (74) is capable of storing a message from an incoming call. A caller identification processing system (76) is coupled to the transceiver (70). The caller identification processing system (76) determines a telephone number of the incoming call and routes the incoming call to the voice processing system (74), if the telephone number belongs to a screened group of telephone numbers.
Abstract:
A system and method for recording a telephone conference and replaying a portion of the recording during the conference. Users participate by connecting through different types of networks using a device having a communication line connection. The recording can be in audio format, text format, or both. Thus, users can recall and replay textual information in addition to the recorded audio. Other information-such as time and user data-may also be recorded along with the audio and text. Users in the conference are identified to enable the association with them each user's contribution to the conference. The user or the user's device can assist by providing identification information. User identification may also be accomplished by associating each user's contribution with the particular line the user is calling from. Caller ID information may also be used to identify the user. Voice analysis may also performed to accomplish user identification.
Abstract:
Upon detecting an utterance period by a state decision part 14, a sound source position detecting part 15 detects the positions of sound sources 91 to 9K are detected by a sound source position detecting part 15, then covariance matrix of acquired signals are calculated by a covariance matrix calculating part 18 in correspondence to the respective sound sources, and stored in a covariance matrix storage part 18 in correspondence to the respective sound sources. The acquired sound level for each sound source is estimated by an acquired sound level estimating part 19 from the stored covariance matrix, and filter coefficients are determined by a filter coefficient calculating part 21 from the estimated acquired sound levels and the covariance matrices, and the filter coefficients are set in filters 121 to 12M. Acquired signals from the respective microphones are filtered by the filters, then the filtered outputs are added together by an adder 13, and the added output is provided as a send signal; by this, it is possible to generate send signals of desired levels irrespective of the positions of sound sources.
Abstract:
There is provided a system for facilitating a conference call. The system includes a module to generate a real-time voiceprint from a voice input of a participant in the conference call, and a module to provide information indicative of the participant based on the real-time voiceprint.
Abstract:
A communications terminal that displays the individual information of the caller at the start of a communication caused by an incoming call, comprising: a FLASH-ROM 10 forming a database that stores individual information for registered individuals and voice spectrum information for the individuals, in a mutually associated manner; a voice spectrum analyzing section 6 that extracts the voice spectrum information of a caller from the voice of the caller at the start of a communication caused by an incoming call; an MPU 7 that identifies a caller from among individuals in a database by comparing the voice spectrum information of the caller with voice spectrum information in the database; and an LCD 17 forming a display section that displays the individual information of the identified caller.
Abstract:
A method for alerting a participant in a conference call that the participant is speaking with insufficient volume is disclosed. The method includes determining that someone in a conference call between multiple endpoints is speaking with insufficient volume. The method further includes determining an active participant in the conference call, the active participants based on who is speaking or has spoken within a predetermined time interval and selectively communicating a speak-louder message to the active participant.