Abstract: A voice recognition system applies speaker-dependent modification functions to acoustic feature vectors prior to voice recognition pattern matching against a speaker-independent acoustic model. An adaptation engine matches a set of acoustic feature vectors X with an adaptation model to select a speaker-dependent feature vector modification function f( ), which is then applied to X to form a modified set of acoustic feature vectors f(X). Voice recognition is then performed by correlating the modified acoustic feature vectors f(X) with a speaker-independent acoustic model.

Abstract: A method and system for voice recognition combines different types of voice recognition (VR) engines in order to recognize user-defined digits and control words, predefined digits and control words, and nametags. Speaker-independent VR engines are combined with speaker-dependent VR engines. A Hidden Markov Model (HMM) VR engine is combined with Dynamic Time Warping (DTW) engines.

Abstract: A distributed voice recognition system includes a digital signal processor (DSP), a nonvolatile storage medium, and a microprocessor. The DSP is configured to extract parameters from digitized input speech samples and provide the extracted parameters to the microprocessor. The nonvolatile storage medium contains a database of speech templates. The microprocessor is configured to read the contents of the nonvolatile storage medium, compare the parameters with the contents, and select a speech template based upon the comparison. The nonvolatile storage medium may be a flash memory. The DSP may be a vocoder. If the DSP is a vocoder, the parameters may be diagnostic data generated by the vocoder. The distributed voice recognition system may reside on an application specific integrated circuit (ASIC).

Abstract: A novel and improved method and an accompanying apparatus provide for a distributed voice recognition (VR) capability in a remote device (201). Remote device (201) decides and controls what portions of the VR processing may take place at remote device (201) and what other portions may take place at a base station (202) in wireless communication with remote device (201).

Abstract: A voice recognition rejection scheme for capturing an utterance includes the steps accepting the utterance, applying an N-best algorithm to the utterance, or rejecting the utterance. The utterance is accepted if a first predefined relationship exists between one or more closest comparison results for the utterance with respect to a stored word and one or more differences between the one or more closest comparison results and one or more other comparison results between the utterance and one or more other stored words. An N-best algorithm is applied to the utterance if a second predefined relationship exists between the one or more closest comparison results and the one or more differences between the one or more closest comparison results and the one or more other comparison results.

Abstract: The present invention is a novel and improved method and apparatus for performing multiple applications on a microprocessor or DSP. One embodiment of the invention includes a car kit for interfacing with a cellular telephone. The care kit includes a digital signal processor for performing digital signal processing and a cradle unit for holding the cellular telephone and for providing a link to said digital signal processor. Additionally, an on-chip memory is provided, located on an integrated circuit with the digital signal processor, for storing a limited function program that control said DSP. Off-chip memory stores multiple limited function programs and for loading any one of the limited function programs to the on-chip memory.

Abstract: A method and apparatus for communicating both voice and control data between a communication device (such as a cellular phone) and an external accessory (such as a hands-free kit) over a data bus. The method includes formatting a sequence of bits into a repeating sequence of first time slots and second time slots, transmitting the voice data in the first time slot, and transmitting the control data in the second time slot. Notably, a first bit of each of the second time slots comprises a clock bit that alternates between a high value and a low value (e.g. a ‘1’ or a ‘0’) as between consecutive second time slots.

Abstract: An apparatus for accurate endpointing of speech in the presence of noise includes a processor and a software module. The processor executes the instructions of the software module to compare an utterance with a first signal-to-noise-ratio (SNR) threshold value to determine a first starting point and a first ending point of the utterance. The processor then compares with a second SNR threshold value a part of the utterance that predates the first starting point to determine a second starting point of the utterance. The processor also then compares with the second SNR threshold value a part of the utterance that postdates the first ending point to determine a second ending point of the utterance. The first and second SNR threshold values are recalculated periodically to reflect changing SNR conditions. The first SNR threshold value advantageously exceeds the second SNR threshold value.

Abstract: A system and method for forming a segmented speech signal from an input speech signal having a plurality of frames. The input speech signal is converted from a time domain signal to a frequency domain signal having a plurality of speech frames, wherein each speech frame in the frequency domain signal is represented by at least one spectral value associated with the speech frame. A spectral difference value is then determined for each pair of adjacent frames in the frequency domain signal, wherein the spectral difference value for each pair of adjacent frames is representative of a difference between the at least one spectral value associated with each frame in the pair of adjacent frames. An initial cluster boundary is set between each pair of adjacent frames in the frequency domain signal, and a variance value is assigned to each cluster in the frequency domain signal, wherein the variance value for each cluster is equal to one of the determined spectral difference values.

Abstract: A method and apparatus for eighth-rate random number generation for speech coders includes a random number generator configured to generate values of a first random variable. A lookup table is used to store values of a second random variable. The lookup table is addressed with the values of the first random variable. The second random variable is an inverse transform of a cumulative distribution function of the first random variable. A codec encodes input silence frames with the values of the first and second random variables, and regenerates the silence frames with the values of the first and second random variables. The speech coder may be an enhanced variable rate coder, and the silence frames may be encoded at eighth rate. The random variables are advantageously Gaussian random variables with values that are uniformly distributed between zero and one.

Abstract: A method and apparatus for eighth-rate random number generation for speech coders includes a random number generator configured to generate values of a first random variable. A lookup table is used to store values of a second random variable. The lookup table is addressed with the values of the first random variable. The second random variable is an inverse transform of a cumulative distribution function of the first random variable. An codec encodes input silence frames with the values of the first and second random variables, and regenerates the silence frames with the values of the first and second random variables. The speech coder may be an enhanced variable rate coder, and the silence frames may be encoded at eighth rate. The random variables are advantageously Gaussian random variables with values that are uniformly distributed between zero and one.

Abstract: A voice recognition system having a feature extraction apparatus is located in a remote station. The feature extraction apparatus extracts features from an input speech frame and then provides the extracted features to a central processing station. In the central processing station, the features are provided to a word decoder which determines the syntax of the input speech frame.

Abstract: A method and apparatus for implementing a vocoder in a application specific integrated circuit (ASIC) is provided. The apparatus contains a DSP core that performs computations in accordance with a reduced instruction set (RISC) architecture. The circuit further includes a specifically designed slave processor to the DSP core referred to as the minimization processor. The apparatus further comprises a specifically designed block normalization circuitry.

Abstract: A method and apparatus for implementing a vocoder in a application specific integrated circuit (ASIC) is provided. The apparatus contains a DSP core that performs computations in accordance with a reduced instruction set (RISC) architecture. The circuit further includes a specifically designed slave processor to the DSP core referred to as the minimization processor. The apparatus further comprises a specifically designed block normalization circuitry.