Abstract: A method for implementing a speech recognition system for use during conditions with background noise includes the steps of calculating, in real-time, sequential short-term delta energy parameters for speech energy from a spoken utterance, determining threshold values in the speech energy, and identifying a beginning point and an ending point for the spoken utterance based on the relationship between the threshold values and the short-term delta energy parameters.

Abstract: A method of creating a user menu (AM1) in a display (2, 22) in a user unit (1, 21, 41). The user menu (AM1) denotes an accessory unit (5, 25a) connected to the user unit (1, 21, 41). Information (I1, I2) is stored in a menu storage unit (3, 23) in the user unit (1, 21). The information (I1, I2) is used to create a standard menu (SM1). The accessory unit (5, 25a) is connected to the user unit (1, 21), and a menu selection code (MCODE) is sent from the accessory unit (5, 25a) to the menu storage unit (3, 23) in the user unit (1, 21). A standard menu (SM1), which corresponds to the menu selection code (MCODE), is pointed out in the menu storage unit (3, 23). The pointed out standard menu (SM1) is adapted to a user menu (AM1) corresponding to the menu selection code (MCODE), and the user menu (AM1) is activated in the display (2, 22).

Abstract: A system and method for accurately determining the position of a mobile unit operating within a predefined service area is disclosed. Three embodiments of the present invention are disclosed which teach the use of one, two and three narrow beam base transceiver stations in the determination of a mobile unit's position. Where one base station is utilized, an information map of signal attributes is used in the position determination. Where two and three base stations are used, signal strength measurements in combination with the time difference of arrival of a signal at the various base stations are used in the position determination.

Abstract: A dictation system is disclosed comprising a hand held dictation device (1) for storing a speech signal in memory means (15,20), the device comprising data compression means (30) for data compressing the speech signal into a data compressed speech signal and storing means for storing the data compressed speech signal in the memory means. The data compression means (30) are adapted to carry out a data compression step on the speech signal in one of at least two different data compression modes, the at least two different data compression modes resulting in different data compression ratios when applied to the same speech signal, the said at least two different data compression modes being selectable by a user.

Abstract: A voice recording and/or reproducing device includes a plurality of coders having different bit rates for coding voice to provide coded voice data, a voice recording mode change over switch for selecting one of the plurality of coders, and a system controller. The system controller stores coding selection data obtained by the change over of the voice recording mode and coded voice data obtained from the selected coder, to a storing medium, and reduces a deterioration of the voice due to the change over. The voice recording and/or reproducing device also includes a detector for detecting the coding selection data, and a plurality of decoders for decoding the coded voice data at the bit rate corresponding to the detected coding selection data.

Abstract: A receiving device comprises a demodulating section for demodulating transmission signal into a demodulated signal having a digital speech signal and a error detecting code. The demodulating section divides the digital speech signal into first through N-th subframe speech signals and detects a received field intensities of the first through the N-th subframe speech signals to produce first through N-th field detection signals representative of the received field intensities of the first through the N-th subframe speech signals, respectively, where N represents a positive integer which is greater than one.

Abstract: A radio frequency audio distortion measurement system for a motor vehicle radio. The system includes a generator that generates test RF signals and transmits the signals to the motor vehicle radio through a wireless communication link. The system includes a controller coupled to the generator that controls the transmission of the RF signals to the radio. The system also includes a distortion analyzer coupled to the RF receiver and the controller that senses and analyzes the audio distortion caused by motor vehicle system components, such as motor vehicle electrical and electronic system components, in response to commands output from the controller. The system of the present invention thereby provides an objective way of analyzing audio distortion caused by radiated conducted and/or coupled interference from electrical and electronic systems in a controlled test environment.

Abstract: A cellular phone case provided with a sound collecting means for collecting the sound from a hearing portion of a cellular phone. The sound collecting means comprises a cover having an sound collecting portion being protrudly formed to surround an ear and a first opening being formed at the bottom of the sound collecting portion, the interior of the sound collection portion being empty, a cushion member positioned in the interior of the sound collecting portion, and a base sheet having a second opening therein being formed substantially coincide with the first opening to encircle the hearing portion of the cellular phone, said base sheet being attached at the bottom of the cover and the edge thereof being attached to the cellular phone case.

Abstract: An improved system for identifying the loudest speech signal in a G.723.1 based audio teleconferencing link is disclosed. The system selects the loudest of several analog audio signals by directly analyzing the encoded G.723.1 bit streams representing those signals, rather than by decoding the encoded speech signal in the G.723.1 bit streams and then re-encoding the signal as a selected output bit stream. The system uses the excitation gain parameters encoded in G.723.1 frames to approximate frame gains for respective bit streams and then estimates a short term speech energy for each bit stream by averaging the approximate frame gains over time. The system then compares the estimated speech energy levels and outputs to each conference participant the signal with the highest estimated speech energy as the next portion of an output signal.

Abstract: The present invention teaches an economical disposable emergency cellular phone. A further object of the invention is a new technique for having a large number of cellular phones share the same small group of access numbers and serial numbers in order to reduce the monthly charges to zero for the end consumer.

Abstract: Mobile radio telephone service is provided to a plurality of members of a subscriber group. Members of the subscriber group, which may comprise, for example, members of a family or employees of a small business, share a single subscription to the mobile radio telephone network which subscription limits the number of traffic channels which can be simultaneously occupied by the members. The group subscription avoids each of the individual members having to have a separate mobile radio system subscription and at the same time allows the network operator to spread subscriber traffic over a larger period of time during the day.

Abstract: A scalable audio coding/decoding method and apparatus are provided.

Abstract: A method including the steps of receiving a location area change notification from a mobile, customizing a custom location area index table based on the location area change notification, and transferring the custom location area index table to the mobile if the mobile supports the custom location area index table.

Abstract: A speech coding device in which an excitation signal of speech signals is expressed as a sum of a plurality of pulse strings, and positions of the pulse strings are selected from predetermined pulse position candidates to determine the excitation signal so that distortion between an input speech signal and a reproduced speech signal obtained by exciting a synthetic filter using the excitation signal may be minimized, resulting in obtaining reproduced speech signals with high quality in a small operational amount. In a pulse searcher, a pulse generating section outputs a plurality of pulse strings, and a pulse searching section sequentially searches the pulse strings to determine the positions of the plurality of pulse strings constituting the excitation signal. One pulse searching section searches using a Viterbi algorithm. Another pulse searching section preliminarily searches in a tree shape of pulse position candidates. Another pulse searching section searches every pulse position candidate group.

Abstract: The invention provides a device, method (400,500,600), and system (100) to improve compression efficiency when coding audio for bitrate scalability. It includes at least one of an encoder and a decoder and is applicable when utilizing perceptual coding for an upper bitrate. The encoder includes a hybrid psychoacoustic modeling unit, coupled to receive lowband audio and diffband audio, for determining psychoacoustic data, and a quantizer control and zero-flagging unit, coupled to receive psychoacoustic data and diffband audio, for determining explicit quantizer stepsize parameters and at least one of: 1) implicit quantizer stepsize parameters and 2) implicit zero-flags.

Abstract: Voice-generating information, comprising discrete voice data for velocity or pitch of a voice is made by dispensing the discrete data so that the voice data is not dependent on a time lag between phonemes and at the same time is present at a relative level against a reference thereof. The said information includes data on plural types of voice tone, and is stored in a voice-generating information storing section. Voice tone data indicating sound parameters for each voice element, such as phoneme for each voice tone type, is stored in a voice tone storing section. Voice data, corresponding to the type of voice tone in the voice-generating information stored in the voice-generating storing section, is selected from a plurality of voice type data stored in the voice tone storing section under control by a control section. Meter patterns, which occur successively in the direction of a time axis, are developed according to the voice-generating information.

Abstract: Methods and apparatus for performing translation between different language are provided. The present invention includes a translation system that performs translation having increased accuracy by providing a three-dimensional topical dual-language database. The topical database includes a set of source-to-target language translations for each topic that the database is being used for. In one embodiment, a user first selects the topic of conversation, then words spoken into a telephone are translated and produced as synthesized voice signals from another telephone so that a near real-time conversation may be had between two people speaking different languages. An additional feature of the present invention is the addition of a computer terminal that displays the input and output phrases so that either user may edit the input phrases, or indicate that the translation was ambiguous and request a rephrasing of the material.

Abstract: A radio communicating apparatus for communicating via a radio signal comprises: a connecting unit to connect a terminal and a communication channel; and a controller for controlling the connecting unit so as to disconnect the communication channel when a control signal indicating that the communication is being executed is not received at a predetermined timing.

Abstract: In a preferred embodiment, a digital cordless telephone system includes a public switched telephone network (PSTN) connected to general telephones, a public switched data network for receiving and transmitting designated packet data, a plurality of digital cordless telephones, and a plurality of public base stations. Each base station is assigned to a designated service area and controls the digital cordless telephones located in its service area. A plurality of local control units are connected to respective public base stations through the PSTN, each local control unit storing information concerning the public base stations in its service area.

Abstract: An unsupervised, discriminative, sentence level, HMM adaptation based on speech-silence classification is presented. Silence and speech regions are determined either using a speech end-pointer or the segmentation obtained from the recognizer in a first pass. The discriminative training procedure using a GPD or any other discriminative training algorithm, employed in conjunction with the HMM-based recognizer, is then used to increase the discrimination between silence and speech.