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: A device for processing speech signal is disclosed comprising a speech signal input and speech signal digitizer, a control signal input and control signal digitizer, a digital signal processor, a transmitter to transmit processed digital signals in a cordless fashion to a base station, and a storage means, which is connected upstream of the transmitter and by which, depending on the quality of the communication link between the device and the base station, buffers the processed digital signals to be supplied and to be transmitted between the device and the base station.

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: A model is provided for formants found in human speech. Under one aspect of the invention, the model is used to synthesize speech. Under this aspect of the invention, the formant model is used to identify a most likely formant track for the synthesized speech. Based on this track, a series of resonators are used to introduce the formants into the speech signal.

Abstract: The excitation in a CELP-like speech coder is recursively calculated. For a given bitrate and a given complexity, the recursive approach described lowers the complexity with minimum impact on speech quality. The excitation signal is a sum of at least three vector terms, each vector term being a product of a codebook vector zk and an associated gain term gk. A first vector term g0z0 is determined that is representative of a target excitation vector x. Each remaining vector term is recursively determined as a vector term gkzk representative of the difference between the target excitation vector x and the sum of previously determined vector terms, ∑ i = 0 k - 1 ⁢ g i ⁢ z i .

Abstract: A system and method for improving the accuracy of a speech recognition program. The system is based on a speech recognition program that automatically converts a pre-recorded audio file into a written text. The system parses the written text into segments, each of which can be corrected by the system and saved in a retrievable manner in association with the computer. The standard speech files are saved towards improving accuracy in speech-to-text conversion by the speech recognition program. The system further includes facilities to repetitively establish an independent instance of the written text from the pre-recorded audio file using the speech recognition program. This independent instance can then be broken into segments and each erroneous segment in said independent instance replaced with the corrected segment associated with that segment. In this manner, repetitive instruction of a speech recognition program can be facilitated.

Abstract: A method and apparatus for the variable rate coding of a speech signal. An input speech signal is classified and an appropriate coding mode is selected based on this classification. For each classification, the coding mode that achieves the lowest bit rate with an acceptable quality of speech reproduction is selected. Low average bit rates are achieved by only employing high fidelity modes (i.e., high bit rate, broadly applicable to different types of speech) during portions of the speech where this fidelity is required for acceptable output. Lower bit rate modes are used during portions of speech where these modes produce acceptable output. Input speech signal is classified into active and inactive regions. Active regions are further classified into voiced, unvoiced, and transient regions. Various coding modes are applied to active speech, depending upon the required level of fidelity. Coding modes may be utilized according to the strengths and weaknesses of each particular mode.

Abstract: In a voice pitch normalization device equipped in a voice recognition device VRAp for recognizing an incoming command voice Sva uttered by any speaker, and used to normalize the incoming command voice to be in an optimal pitch for voice recognition, a target voice generator produces a target voice signal by changing the incoming command voice Svd on the basis of a predetermined degree. A probability calculator calculates a probability indicating a degree of coincidence among the target voice signal and a plurality of words in sample data. A voice pitch changer repeatedly changes the target voice signal in voice pitch until a maximum probability becomes a predetermined probability or greater.

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: 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 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.

Abstract: A speech device for detecting a speech signal in a received signal and for determining a speech time slot, the device including a switch-on threshold detector for detecting certain detection information in relation to a threshold, and an information processing means for receiving and processing the detection information and for terminating the production of speech detection information featuring a speech time slot if the certain detection information was received during a first switch-off period, while the information processing means are arranged for additionally terminating the delivery of speech detection information if the certain detection information was not received during a second switch-off period and/or if certain detection information was received during a third switch-off period.

Abstract: An adaptive speech recognition method with noise compensation is disclosed. In speech recognition, optimal equalization factors for feature vectors of a plurality of speech frames corresponding to each probability density function in a speech model are determined based on the plurality of speech frames of the input speech and the speech model. The parameters of the speech model are adapted by the optimal equalization factor and a bias compensation vector, which is corresponding to and retrieved by the optimal equalization factor. The optimal equalization factor is provided to adjust a distance of the mean vector in the speech model. The bias compensation vector is provided to adjust a direction change of the mean vector in the speech model.

Abstract: A time-separated speech coder that codes a transitional signal of voiced/unvoiced sound through harmonic speech coding, the coder including a transitional excitation signal analyzer/synthesizer for coding the transitional signal by extracting the harmonic model parameters of both transitional analyzers after detecting a transitional point and generating sinusoidal waveforms according to a variable transitional point separating both transitional analyzers. By the transitional point at which energy varies abruptly and the time-separated coding based on the transitional point, more improved speech quality than in the general harmonic speech coder can be obtained using the time-separated speech coder by increasing the representation capability of the transitional signal with large energy variation, after adapting it to the variable transitional point.

Abstract: The invention provides a method of coding an information signal. An information signal is represented by a sequence of pulses. A plurality of pulse parameters are determined based on the sequence of pulses including a non-zero pulse parameter corresponding to a number of non-zero pulse positions in the sequence of pulses. The non-zero pulse parameter is coded using a variable-length codeword.

Abstract: Techniques and systems for identifying coding rates of transmitted frames are described. Unused bits in rate adapted frames are used to carry frame type indicator patterns. Maximal rate frames (i.e., with a highest coding rate) need not include a frame type indicator.

Abstract: A method for adjusting a speech menu interface in a speech recognition system. The method can include a series of steps which can include identifying one or more menu items from a data structure in memory for presentation using the speech menu interface. The step of retrieving the one or more menu items from the data structure can be included. Also, the step of reading at least one attribute corresponding to the one or more menu items with menu item criteria can be included. Based on the comparing step, the method can include selecting a presentation style for presentation of the one or more menu items using the speech menu interface.

Abstract: The invention provides a method and apparatus for automatically generating a summary or key phrase for a song. The song, or a portion thereof, is digitized and converted into a sequence of feature vectors, such mel-frequency cepstral coefficients (MFCCs). The feature vectors are then processed in order decipher the song's structure. Those sections that correspond to different structural elements are then marked with corresponding labels. Once the song is labeled, various heuristics are applied to select a key phrase corresponding to the song's summary. For example, the system may identify the label that appears most frequently within the song, and then select the longest duration of that label as the summary.