Abstract: A method includes defining a transition band for a signal having a spectrum within a first frequency band, where the transition band is defined as a portion of the first frequency band, and is located near an adjacent frequency band that is adjacent to the first frequency band. The method analyzes the transition band to obtain a transition band spectral envelope and a transition band excitation spectrum; estimates an adjacent frequency band spectral envelope; generates an adjacent frequency band excitation spectrum by periodic repetition of at least a part of the transition band excitation spectrum with a repetition period determined by a pitch frequency of the signal; and combines the adjacent frequency band spectral envelope and the adjacent frequency band excitation spectrum to obtain an adjacent frequency band signal spectrum. A signal processing logic for performing the method is also disclosed.

Abstract: A method includes defining a transition band for a signal having a spectrum within a first frequency band, where the transition band is defined as a portion of the first frequency band, and is located near an adjacent frequency band that is adjacent to the first frequency band. The method analyzes the transition band to obtain a transition band spectral envelope and a transition band excitation spectrum; estimates an adjacent frequency band spectral envelope; generates an adjacent frequency band excitation spectrum by periodic repetition of at least a part of the transition band excitation spectrum with a repetition period determined by a pitch frequency of the signal; and combines the adjacent frequency band spectral envelope and the adjacent frequency band excitation spectrum to obtain an adjacent frequency band signal spectrum. A signal processing logic for performing the method is also disclosed.

Abstract: A method and apparatus are provided for reproducing a speech sequence of a user through a communication device of the user. The method includes the steps of detecting a speech sequence from the user through the communication device, recognizing a phoneme sequence within the detected speech sequence and forming a confidence level of each phoneme within the recognized phoneme sequence. The method further includes the steps of audibly reproducing the recognized phoneme sequence for the user through the communication device and gradually highlighting or degrading a voice quality of at least some phonemes of the recognized phoneme sequence based upon the formed confidence level of the at least some phonemes.

Abstract: An electronic device (400) for speech dialog includes functions that receive (405, 205) a speech phrase that includes an instantiated variable (315), generate pitch and voicing characteristics (330) of the instantiated variable, and performs voice recognition (410, 220) of the instantiated variable to determine a most likely set of recognition acoustic states (335). A trained map (358) is established (115) that maps recognition feature vectors derived from training speech (105) to synthesis feature vectors derived from the same training speech (110). Recognition feature vectors that represent the most likely set of recognition acoustic states for the recognized instantiated variable are converted to a most likely set of synthesis acoustic states (420) in accordance with the map.

Abstract: An echo canceling circuit comprising a double talk detector, an upper band signal filter configured to pass only near-end upper band signals to the double talk detector and remove lower band signals, an adaptive filter circuit, a control circuit operatively coupled to the double talk detector and to the adaptive filter circuit, and a threshold estimator configured to iteratively calculate an upper adaptive decision threshold value and a lower adaptive decision threshold value. The double talk detector declares near-end speech to be present if an estimated power level of the upper band signals exceeds the upper adaptive decision threshold value, and declares the near-end speech to be absent if the estimated power level of the upper band signals falls below the lower adaptive decision threshold value for a predetermined number of iterative cycles.

Abstract: A speech communication system provides a speech encoder that generates a set of coded parameters representative of the desired speech signal characteristics. The speech communication system also provides a speech decoder that receives the set of coded parameters to generate reconstructed speech. The speech decoder includes an equalizer that computes a matching set of parameters from the reconstructed speech generated by the speech decoder, undoes the set of characteristics corresponding to the computed set of parameters, and imposes the set of characteristics corresponding to the coded set of parameters, thereby producing equalized reconstructed speech.

Abstract: A method and apparatus for speech reconstruction within a distributed speech recognition system is provided herein. Missing MFCCs are reconstructed and utilized to generate speech. Particularly, partial recovery of the missing MFCCs is achieved by exploiting the dependence of the missing MFCCs on the transmitted pitch period P as well as on the transmitted MFCCs. Harmonic magnitudes are then obtained from the transmitted and reconstructed MFCCs, and the speech is reconstructed utilizing these harmonic magnitudes.

Abstract: In a distributed voice recognition system, a back-end pattern matching unit 27 can be informed of voice activity detection information as developed through use of a back-end voice activity detector 25. Although no specific voice activity detection information is developed or forwarded by the front-end of the system, precursor information as developed at the back-end can be used by the voice activity detector to nevertheless ascertain with relative accuracy the presence or absence of voice in a given set of corresponding voice recognition features as developed by the front-end of the system.

Abstract: A method and apparatus for prediction in a speech-coding system is provided herein. The method of a 1st order long-term predictor (LTP) filter, using a sub-sample resolution delay, is extended to a multi-tap LTP filter, or, viewed from another vantage point, the conventional integer-sample resolution multi-tap LTP filter is extended to use sub-sample resolution delay. This novel formulation of a multi-tap LTP filter offers a number of advantages over the prior-art LTP filter configurations. Particularly, defining the lag with sub-sample resolution makes it possible to explicitly model the delay values that have a fractional component, within the limits of resolution of the over-sampling factor used by the interpolation filter. The coefficients of such a multi-tap LTP filter are thus largely freed from modeling the effect of delays that have a fractional component.

Abstract: A Fast Fourier Transform (FFT) based voice synthesis method 110, program product and vocoder. Sounds, e.g., speech and audio, are synthesized from multiple sine waves. Each sine wave component is represented by a small number of FFT coefficients 116. Amplitude 120 and phase 124 information of the components may be incorporated into these coefficients. The FFT coefficients corresponding to each of the components are summed 126 and, then, an inverse FFT is applied 128 to the sum to generate a time domain signal. An appropriate section is extracted 130 from the inverse transformed time domain signal as an approximation to the desired output. FFT based synthesis 110 may be combined with simple sine wave summation 100, using FFT based synthesis 110 for complex sounds, e.g., male voices and unvoiced speech, and sine wave summation 100 for simpler sounds, e.g., female voices.

Abstract: A method and apparatus for noise suppression within a distributed speech recognition system is provided herein. Mel-frequency cepstral coefficients (MFCCs) values are converted to filter bank outputs (F′0 through F′22). The filter bank outputs are then used by a noise suppressor (303) for channel energy estimation, noise energy estimation, etc. Noise-suppression takes place on F′0 through F′22 and the noise-suppressed filter bank outputs F″0 through F″22 are converted back to MFCC values.

Abstract: A method and apparatus for speech reconstruction within a distributed speech recognition system is provided herein. Missing MFCCs are reconstructed and utilized to generate speech. Particularly, partial recovery of the missing MFCCs is achieved by exploiting the dependence of the missing MFCCs on the transmitted pitch period P as well as on the transmitted MFCCs. Harmonic magnitudes are then obtained from the transmitted and reconstructed MFCCs, and the speech is reconstructed utilizing these harmonic magnitudes.

Abstract: In a distributed voice recognition system, a back-end pattern matching unit 27 can be informed of voice activity detection information as developed through use of a back-end voice activity detector 25. Although no specific voice activity detection information is developed or forwarded by the front-end of the system, precursor information as developed at the back-end can be used by the voice activity detector to nevertheless ascertain with relative accuracy the presence or absence of voice in a given set of corresponding voice recognition features as developed by the front-end of the system.