Abstract: A system and method for inputting Chinese characters using Pinyin without requiring the entry of a delimiter key between Pinyin entries in a reduced keyboard is disclosed. The system searches for all possible single or multiple Pinyin spellings based on the entered Latin alphabets. Once the user has completed the inputting of the Pinyin spellings for desired Chinese phrase or characters, all possible matching phrases or characters are displayed on screen and off-screen due to screen size. The user then scrolls through a list of matching phrases or characters and selects the desired one by clicking.

Abstract: A system and method of updating automatic speech recognition parameters on a mobile device are disclosed. The method comprises storing user account-specific adaptation data associated with ASR on a computing device associated with a wireless network, generating new ASR adaptation parameters based on transmitted information from the mobile device when a communication channel between the computing device and the mobile device becomes available and transmitting the new ASR adaptation data to the mobile device when a communication channel between the computing device and the mobile device becomes available. The new ASR adaptation data on the mobile device more accurately recognizes user utterances.

Abstract: A system and method enrolls a speaker with an enrollment utterance and authenticates a user with a biometric analysis of an authentication utterance, without the need for a PIN (Personal Identification Number). During authentication, the system uses the same authentication utterance to identify who a speaker claims to be with speaker recognition, and verify whether is the speaker is actually the claimed person. Thus, it is not necessary for the speaker to identify biometric data using a PIN. The biometric analysis includes a neural tree network to determine unique aspects of the authentication utterances for comparison to the enrollment authentication. The biometric analysis leverages a statistical analysis using Hidden Markov Models to before authorizing the speaker.

Abstract: An approach for efficiently reducing background noise from speech signal in real-time applications is presented. A noisy input speech signal is processed through an inverse filter when the spectrum tilt of the input signal is not that of a pure background noise model the noisy input signal is also filtered in order to reduce the spectrum valley areas of the noisy input signal when the background noise is present.

Abstract: A frame erasure concealment device and method that is based on reestimating gain parameters for a code excited linear prediction (CELP) coder is disclosed. During operation, when a frame in a stream of received data is detected as being erased, the coding parameters, especially an adaptive codebook gain gp and a fixed codebook gain gc, of the erased and subsequent frames can be reestimated by a gain matching procedure. By using this technique with the IS-641 speech coder, it has been found that the present invention improves frame erasure concealment device and method improve the speech quality under various channel conditions, compared with a conventional extrapolation-based concealment algorithm.

Abstract: In a transcription device (1) for transcribing a spoken text (GT) into a recognized text (ET) and for editing incorrectly recognized parts of the recognized text (ET), marking means (12, 15, 17) are provided that are arranged for the partly automatic and partly manual marking of parts of the spoken text (GT) and/or of the recognized text (ET) that have a common characteristic. As a result, subsequent unified processing of marked parts of the text that have common characteristics becomes possible.

Abstract: An automated speech recognition filter is disclosed. The automated speech recognition filter device provides a speech signal to an automated speech platform that approximates an original speech signal as spoken into a transceiver by a user. In providing the speech signal, the automated speech recognition filter determines various models representative of a cumulative signal degradation of the original speech signal from various devices along a transmission signal path and a reception signal path between the transceiver and a device housing the filter. The automated speech platform can thereby provide an audio signal corresponding to a context of the original speech signal.

Abstract: A list of integer values is generated from n-grams of a user input. The list of integer values is sorted. Differences between adjacent integer values in the list are calculated. Each calculated difference is encoded using a Golomb code. A Golomb compressed language model is accessed to identify likely matches.

Abstract: In a method of entering text into a device a first character input is provided that is indicative of a first character of a text entry. Next, a vocalization of the text entry is captured. A probable word candidate is then identified for a first word of the vocalization based upon the first character input and an analysis of the vocalization. Finally, the probable word candidate is displayed for a user.

Abstract: A multi-mail system and method is disclosed in which a sender may convey and a recipient can realize emotional aspects associated with substantive content of a multi-mail message by receiving a message that is more than textual in nature. Voice recognition technology and programmatic relation of sound and graphics may be used to produce a talking image. In one embodiment, the image may include the user's own visual and/or audio likeness. In an alternate embodiment, the image may comprise any available visual and/or audio display selected by the user. The multi-mail message may be inputted by a user in a text format and transposed into a format including the selected image and/or voice. In an alternate embodiment, a spoken message may be converted into a format including the selected image and/or voice. The formatted messages are then stored and/or transmitted via an email system or some other electronic network.

Abstract: A method to eliminate discontinuities in an adaptively filtered signal includes filtering a beginning portion of a current signal frame using a past set of filter coefficients, thereby producing a first filtered frame portion. The method also includes filtering the beginning portion of the current signal frame using a current set of filter coefficients, thereby producing a second filtered frame portion. The method also includes modifying the second filtered frame portion with the first filtered frame portion so as to smooth a possible filtered signal discontinuity between the second filtered frame portion and a past filtered frame produced using the past filter coefficients.

Abstract: Pitch estimation and classification into voiced, unvoiced and transitional speech were performed by a spectro-temporal auto-correlation technique. A peak picking formula was then employed. A weighting function was then applied to the power spectrum. The harmonics weighted power spectrum underwent mel-scaled band-pass filtering, and the log-energy of the filter's output was discrete cosine transformed to produce cepstral coefficients. A within-filter cubic-root amplitude compression was applied to reduce amplitude variation without compromise of the gain invariance properties.

Abstract: Methods and apparatus are provided for communicating an audio stream. A perceptual mask is estimated for an audio stream, based on the perceptual threshold of the human auditory system. A hidden sub-channel is dynamically allocated substantially below the estimated perceptual mask based on the characteristics of the audio stream, in which additional payload is transmitted. The additional payload can be related to components of the audio stream that would not otherwise be transmitted in a narrowband signal, or to concurrent services that can be accessed while the audio stream is being transmitted. A suitable receiver can recover the additional payload, whereas the audio stream will be virtually unaffected from a human auditory standpoint when received by a traditional receiver. A coding scheme is also provided in which a portion of a codec is used to code an upper-band portion of an audio stream, while the narrowband portion is left uncoded.

Abstract: A method for transferring real time information on a record carrier, typically bitstream audio on an optical disc, which method comprises encoding consecutive segments of the real time information to compressed real time data in frames, and determining a buffer occupancy for at least one frame, which buffer occupancy is indicative of an amount of compressed real time data to be present in the playback buffer at the start of decoding said frame. A signal is transmitted carrying the compressed real time data and the buffer occupancy, which data are received, stored in a playback buffer and finally decoded. The retrieving and/or the decoding is controlled in dependence on said transferred buffer occupancy. A playback buffer can be used effectively without risk for underflow or overflow. Also a method for recording audio information on a record carrier, a recording device, a record carrier and a playback device are described.

Abstract: Systems and methods for dynamically analyzing temporality in an individual's speech in order to selectively categorize the speech fluency of the individual and/or to selectively provide speech training based on the results of the dynamic analysis. Temporal variables in one or more speech samples are dynamically quantified. The temporal variables in combination with a dynamic process, which is derived from analyses of temporality in the speech of native speakers and language learners, are used to provide a fluency score that identifies a proficiency of the individual. In some implementations, temporal variables are measured instantaneously.

Abstract: For determining an estimate of a need for information units for encoding a signal, a measure for the distribution of the energy in the frequency band is taken into account in addition to the admissible interference for a frequency band and an energy of the frequency band. With this, a better estimate of the need for information units is obtained, so that coding can be done more efficiently and more accurately.

Abstract: New techniques and systems may be implemented to improve error correction in speech recognition. These new techniques and systems may be implemented to correct errors in speech recognition systems may be used in a standard desktop environment, in a mobile environment, or in any other type of environment that can receive and/or present recognized speech.

Abstract: Text-to-speech streaming data is output to an end user using a distributed network based message processing system. The distributed network system includes a user access server that controls access of registered users to the data retrieval system. An internetwork data retrieval system server retrieves raw data from an internetwork. A text-to-speech server converts the raw data to an audible speech data. A memory storage output device stores a streaming media file containing the audible speech data and a streaming media server transmits the audible speech data to the registered users via the internetwork.

Abstract: A method of communicating product use instructions to patient includes: (a.) providing a group of product container with microprocessors, and, (b.) providing a central processor separate from the product containers. Each product container has the microprocessor attached to the container. The microprocessor includes: (a)(i) a wave file receiving chip; (a)(ii) a wave file storage means; (a)(iii) a wave file audio playback means; (a)(iv) an audio playback start means; and (a)(v) a power supply within the microprocessor. The central processor includes: (b)(i) user input means; (b)(ii) text-to-speech means; (b)(iii) wave file means to create a wave file from the text-to-speech means; and (b)(iv) wireless transmission means to transmit the wave file to the microprocessor wave file receiving chips of a plurality of OTC containers simultaneously.

Abstract: A computer-based system generates alternative phonetic transcriptions for a target word or phrase corresponding to specific phonological processes that replace individual phonemes or clusters of two or more phonemes with replacement phonemes. The system compares a user's speech with a list of possible transcriptions that includes the base (i.e., correct) transcription of the test target as well as the different alternative transcriptions, to identify the transcription that best matches the user's. In a speech therapy application, the system identifies the phonological process(es), if any, associated with the user's speech and generates statistics over multiple test targets that can be used to diagnose the user's specific phonological disorders. The system can also be implemented in other contexts such as foreign language instruction and automated attendant applications to cover a wide variety and range of accents and/or phonological disorders.