Abstract: An interactive voice actuated control system for a testing machine such as a tensile testing machine is described. Voice commands are passed through a user-command predictor and integrated with a graphical user interface control panel to allow hands-free operation. The user-command predictor learns operator command patterns on-line and predicts the most likely next action. It assists less experienced operators by recommending the next command, and it adds robustness to the voice command interpreter by verbally asking the operator to repeat unlikely commanded actions. The voice actuated control system applies to industrial machines whose normal operation is characterized by a nonrandom series of commands.

Abstract: Method and systems to voice-enable a user interface using a voice extension module are provided. A voice extension module includes a preprocessor, a speech recognition engine, and an input handler. The voice extension module receives user interface information, such as, a hypertext markup language (HTML) document, and voice-enables the document so that a user may interact with any user interface elements using voice commands.

Abstract: A signal modification technique facilitates compact voice coding by employing a continuous, rather than piece-wise continuous, time warp contour to modify an original residual signal to match an idealized contour, avoiding edge effects caused by prior art techniques. Warping is executed using a continuous warp contour lacking spatial discontinuities which does not invert or overly distend the positions of adjacent end points in adjacent frames. The linear shift implemented by the warp contour is derived via quadratic approximation or other method, to reduce the complexity of coding to allow for practical and economical implementation. In particular, the algorithm for determining the warp contour uses only a subset of possible contours contained within a sub-range of the range of possible contours. The relative correlation strengths from these contours are modeled as points on a polynomial trace and the optimum warp contour is calculated by maximizing the modeling function.

Abstract: A method for detecting a transient in a discrete-time audio signal is performed completely in the time domain and includes the step of segmenting the discrete-time audio signal so as to generate consecutive segments of the same length with unfiltered discrete-time audio signals xs(T−1). The discrete-time audio signal in a current segment is subsequently filtered. Then either the energy of the filtered discrete-time audio signal in the current segment can be compared with the energy of the filtered discrete-time audio signal in a preceding segment or a current relationship between the energy of the filtered discrete-time audio signal in the current segment and the energy of the unfiltered discrete-time audio signal in the current segment can be formed and this current relationship compared with a preceding corresponding relationship. On the basis of the one and/or the other of these comparisons it is detected whether a transient is present in the discrete-time audio signal.

Abstract: A client/server text-to-speech synthesis system and method divides the method optimally between client and server. The server stores large databases for pronunciation analysis, prosody generation, and acoustic unit selection corresponding to a normalized text, while the client performs computationally intensive decompression and concatenation of selected acoustic units to generate speech. The units are transmitted from the client to the server in a highly compressed format, with a compression method selected based on the predetermined set of potential acoustic units. This compression method allows for very high-quality and natural-sounding speech to be output at the client machine.

Abstract: A method and an apparatus for processing a sound signal in which a useful signal and an interference signal are specified, the sound signal being transformed into the frequency domain and a change in the profile of the frequency being represented by an envelope for at least one frequency over a time. By segmenting the envelope, a maximum is obtained for each segment, the smallest maximum, weighted by a factor, being subtracted from the sound signal. It is also possible to take account of the minimum for the purpose of reducing the interference signal.

Abstract: The systems and methods described herein allow dictation and associated routing and formatting information to be forwarded to a transcription system. The transcription system converts the information into a document. The additional information associated with the dictation is then applied to the document to ensure proper formatting, routing, or the like. The completed document is returned to the original dictator for review and proofing. Upon approval, the document is distributed via the transcription system in accordance with distribution information associated with the document.

Abstract: A decoder for packetized speech with differential quantization of line spectral frequencies and fixed-codebook gain conceals erased frames with interpolation of future and past frames by reconstruct future frame predicted parameters from presumed interpolations of erased frame parameters.

Abstract: A method of composing messages for speech output and the improvement of the quality of reproduction of speech outputs. A series of original sentences for messages is segmented and stored as audio files with search criteria. The length, position, and transition values for the respective segments can be recorded and stored. A sentence to be reproduced is transmitted in a format corresponding to the format of the search criteria. It is determined whether the sentence to be reproduced can be fully reproduced by one segment or a succession of stored segments. The segments found in each case are examined using their entries as to how far the individual segments match as regards speech rhythm. The audio files of the segments in which the examination resulted in the pre-requisites for optimal maintaining of the natural speech rhythm are combined and output for reproduction.

Abstract: In a method of synthesizing voiced speech from pitch prototype waveforms by time-synchronous waveform interpolation (TSWI), one or more pitch prototypes is extracted from a speech signal or a residue signal. The extraction process is performed in such a way that the prototype has minimum energy at the boundary. Each prototype is circularly shifted so as to be time-synchronous with the original signal. A linear phase shift is applied to each extracted prototype relative to the previously extracted prototype so as to maximize the cross-correlation between successive extracted prototypes. A two-dimensional prototype-evolving surface is constructed by unsampling the prototypes to every sample point. The two-dimensional prototype-evolving surface is re-sampled to generate a one-dimensional, synthesized signal frame with sample points defined by piecewise continuous cubic phase contour functions computed from the pitch lags and the phase shifts added to the extracted prototypes.

Abstract: A method for processing a misrecognition error in an embedded speech recognition system during a speech recognition session can include the step of speech-to-text converting audio input in the embedded speech recognition system based on an active language model. The speech-to-text conversion can produce speech recognized text that can be presented through a user interface. A user-initiated misrecognition error notification can be detected. The audio input and a reference to the active language model can be provided to a speech recognition system training process associated with the embedded speech recognition system.

Abstract: A low power audio processor is disclosed which includes: a bit stream processing unit for performing bit processing for an applied audio stream and for decoding the bit processed audio stream to have a format conducive to digital signal processing; a digital signal processing unit for receiving the decoded data from the bit stream processing unit to perform digital signal processing; a post processing unit for post processing audio data from the digital signal processing unit to output final audio data; and a host interface unit for interfacing with an external device to provide an audio parallel stream from the external device to the bit stream processing unit.

Abstract: An interactive multimedia book provides hands-on multimedia instruction to the user in response to voiced commands. The book is implemented on a computer system and includes both text and audio/video clips. The interactive multimedia book is accessed by voiced commands and natural language queries as the primary user input. The displayed text is written in a markup language and contains hyperlinks which link the current topic with other related topics. The user may command the book to read the text and, as the text is read by the voice synthesizer, a word which is also a hyperlink will change its attributes upon being spoken. The user will be able to observe or hear this and simply utter the word which is the hyperlink to navigate to the linked topic.

Abstract: A speech compression system with a special fixed codebook structure and a new search routine is proposed for speech coding. The system is capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech. The codebook structure uses a plurality of subcodebooks. Each subcodebook is designed to fit a specific group of speech signals. A better way is used to calculate a criterion value, minimizing an error signal in a minimization loop as part of the coding system. An external signal sets a maximum bitstream rate for delivering encoded speech into a communications system. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. Each codec is selectively activated to encode and decode the speech signals at different bit rates to enhance overall quality of the synthesized speech at a limited average bit rate.

Abstract: The invention relates to a method of identifying a language in which a text is composed in the form of a string of characters, and also to a method of controlling a speech reproduction unit and to a communication device. To be able to carry out language identification with little expenditure, it is provided according to the invention that a frequency distribution (h1(x), h2(x,y), h3(x,y,z)) of letters in the text is ascertained, the ascertained frequency distribution (h1(x), h2(x,y), h3(x,y,z)) is compared with corresponding frequency distributions (l1(x), l2(x,y), l3(x,y,z)) of available languages, in order to ascertain similarity factors (s1, S2, s3) which indicate the similarity of the language of the text with the available languages, and the language for which the ascertained similarity factor (S1, S2, S3) is the greatest is established as the language of the text.

Abstract: Methods and an apparatus for enhancement of source coding systems utilizing high frequency reconstruction (HFR) are introduced. The problem of insufficient noise contents is addressed in a reconstructed highband, by using Adaptive Noise-floor Addition. New methods are also introduced for enhanced performance by means of limiting unwanted noise, interpolation and smoothing of envelope adjustment amplification factors. The methods and apparatus used are applicable to both speech coding and natural audio coding systems.

Abstract: An apparatus and a method for encoding an input signal on the time base through orthogonal transform involves removing the correlation of signal waveform on the basis of the parameters obtained by means of linear predictive coding (LPC) analysis and pitch analysis of the input signal on the time base prior to the orthogonal transform. The time base input signal from input terminal is sent to a normalization circuit section and a LPC analysis circuit. The normalization circuit section removes the correlation of the signal waveform and takes out the residue by an LPC inverse filter and pitch inverse filter and sends the residue to an orthogonal transform circuit section. The LPC parameters from the LPC analysis circuit and the pitch parameters from the pitch analysis circuit are sent to a bit allocation calculation circuit.

Abstract: A method of converting text to speech in a communication device includes providing a code table containing coded speech parameters. Next steps include inputting a text message into a communication device, and dividing the text message into phonics. A next step includes mapping each of the phonics against the code table to find the coded speech parameters corresponding to each of the phonics. A next step includes processing the coded speech parameters corresponding to each of the phonics to provide an audio signal. In this way, text can be mapped directly to a vocoder table without intermediate translation steps.

Abstract: A speech-coding device includes a fixed codebook, an adaptive codebook, a short-term enhancement circuit, and a summing circuit. The short-term enhancement circuit connects an output of the fixed codebook to a summing circuit. The summing circuit adds an adaptive codebook contribution to a fixed codebook contribution. The short-term enhancement circuit can also be connected to a synthesis filter to emphasize the spectral formants in an encoder and a decoder.

Abstract: A system and method for providing transformed web pages to users with special needs is presented. In one aspect of the system and method, a Translator/Mediator Server is located between the user and the web site. The Translator/Mediator Server translates and transforms the web pages that the user requests from the web site. The translation and transformation of the web pages is directed towards the particular needs of the user.