Abstract: A frame erasure concealment technique for a bitstream-based feature extractor in a speech recognition system particularly suited for use in a wireless communication system operates to “delete” each frame in which an erasure is declared. The deletions thus reduce the length of the observation sequence, but have been found to provide for sufficient speech recognition based on both single word and “string” tests of the deletion technique.

Abstract: The method and preprocessor enhances the intelligibility of narrowband speech without essentially lengthening the overall time duration of the signal. Both spectral enhancements and variable-rate time-scaling procedures are implemented to improve the salience of initial consonants, particularly the perceptually important formant transitions. Emphasis is transferred from the dominating vowel to the preceding consonant through adaptation of the phoneme timing structure. In a further embodiment, the technique is applied as a preprocessor to a speech coder.

Abstract: Speech at the beginning of a talkspurt in a discontinuous transmission (DTX) packet telephony system is speeded up to help make up for an access delay incurred during channel allocation. Incoming speech frames are buffered, a pitch period for a current portion of the signal is estimated, and then a pitch period=s worth of the signal is cut from that portion. This is continued until the original access delay, as estimated from the time lag between the commencement of voice input for the talkspurt, and notification that a channel is available, is eliminated. The remainder of the talkspurt is then transmitted without such compression.

Abstract: A system and method are disclosed for extending the bandwidth of a narrowband signal such as a speech signal. The method applies a parametric approach to bandwidth extension but does not require training. The parametric representation relates to a discrete acoustic tube model (DATM). The method comprises computing narrowband linear predictive coefficients (LPCs) from a received narrowband speech signal, computing narrowband partial correlation coefficients (parcors) using recursion, computing Mnb area coefficients from the partial correlation coefficient, and extracting Mwb area coefficients using interpolation. Wideband parcors are computed from the Mwb area coefficients and wideband LPCs are computed from the wideband parcors.

Abstract: A system and method that enhance and code a digitized speech signal by breaking the digitized speech signal into constituent parts. The method comprises applying at least two speech enhancement processes to produce at least two enhanced digitized speech signals and computing a coded speech signal by processing the at least two enhanced digitized speech signals.

Abstract: A feature extraction process for use in a wireless communication system provides automatic speech recognition based on both spectral envelope and voicing information. The shape of the spectral envelope is used to determine the LSPs of the incoming bitstream and the adaptive gain coefficients and fixed gain coefficients are used to generate the “voiced” and “unvoiced” feature parameter information.

Abstract: Method and system for delivering a message over a telecommunications network to a recipient comprising transmitting a message over the telecommunications network to the recipient when a predetermined energy/silence condition is detected, performing echo cancellation on a signal communicated from the telecommunications network, and monitoring the signal to detect the energy/silence condition.

Abstract: Recognizing a stream of speech received as speech vectors over a lossy communications link includes constructing for a speech recognizer a series of speech vectors from packets received over a lossy packetized transmission link, wherein some of the packets associated with each speech vector are lost or corrupted during transmission. Each constructed speech vector is multi-dimensional and includes associated features. Potentially corrupted features within the speech vector are indicated to the speech recognizer when present. Speech recognition is attempted at the speech recognizer on the speech vectors when corrupted features are present. This recognition may be based only on certain or valid features within each speech vector. Retransmission of a missing or corrupted packet is requested when corrupted values are indicated by the indicating step and when the attempted recognition step fails.

Abstract: A system and method for providing telephony and high speed data access over a broadband access network, comprising a network interface unit (NIU) coupled to a backup local exchange carrier (LEC) line, the broadband access network coupled to the NIU, an intermediate point-of-presence (IPOP) coupled to the broadband access network, and at least one external access network coupled to the IPOP. The system also provides for a fail-safe mode in which the NIU supports the LEC line for lifeline services.

Abstract: A system and method of providing a location-based translation service is disclosed. A wireless device receives from a wireless network an updated priority target language list according to the location of the wireless device. The wireless device and/or wireless network are capable of automatic speech recognition (ASR), text-to-speech (TTS) services and language translation services. A language and location database module associates probable target languages in a given area. The wireless network node receives the location of the wireless device through a network-based location means or through a GPS system and compares the location of the wireless device to the language and location database. The wireless network transmits to the wireless device an updated priority language list based on the data in the language and location database such that the wireless device pre-selects the most likely target language if the user launches the language translation application.

Abstract: An apparatus and method for data processing that improves estimation of spectral parameters of speech data and reduces algorithmic delay in a data coding operation. Estimation of spectral parameters is improved by adaptively adjusting a gain function used to enhance data based on whether the data contains information speech and noise or noise only. A determination is made concerning whether the speech signal to be processed represents articulated speech or a speech pause and a gain is formed for application to the speech signal. The lowest value the gain may assume (i.e., its lower limit) is determined based on whether the speech signal is known to represent articulated speech or not. The lower limit of the gain during periods of speech activity is constrained to be lower than the lower limit of the gain during speech pause. Also, the gain that is applied to a data frame of the speech signal is adaptively limited based on limited a priori signal-to-noise (SNR) values.

Abstract: A system and method are disclosed for extending the bandwidth of a narrowband signal such as a speech signal. The method applies a parametric approach to bandwidth extension but does not require training. The parametric representation relates to a discrete acoustic tube model ATH. The method comprises computing narrowband linear predictive coefficients (LPCs) from a received narrowband speech signal, computing narrowband partial correlation coefficients (parcors) using recursion, computing Mnb area coefficients from the partial correlation coefficient, and extracting Mwb area coefficients using interpolation. Wideband parcors are computed from the Mwb area coefficients and wideband LPCs are computed from the wideband parcors.

Abstract: An apparatus and method for data processing that improves estimation of spectral parameters of speech data and reduces algorithmic delay in a data coding operation. Estimation of spectral parameters is improved by adaptively adjusting a gain function used to enhance data based on whether the data contains information speech and noise or noise only. Delay is reduced by extracting coding parameters using incompletely processed data. This data is formed by multiplying a less current portion of an input data frame with a synthesis window and a more current portion of the data frame with an inverse analysis window, and performing an overlap-add process on the data frame and a similarly processed previous data frame.

Abstract: A frame erasure concealment technique for a bitstream-based feature extractor in a speech recognition system particularly suited for use in a wireless communication system operates to “delete” each frame in which an erasure is declared. The deletions thus reduce the length of the observation sequence, but have been found to provide for sufficient speech recognition based on both single word and “string” tests of the deletion technique.

Abstract: An apparatus and method for data processing that improves estimation of spectral parameters of speech data and reduces algorithmic delay in a data coding operation. Estimation of spectral parameters is improved by adaptively adjusting a gain function used to enhance data based on whether the data contains information speech and noise or noise only. Delay is reduced by extracting coding parameters using incompletely processed data.

Abstract: A speech enhancement system receives noisy speech and produces enhanced speech. The noisy speech is characterized by a spectral amplitude spanning a plurality of frequency bins. The speech enhancement system modifies the spectral amplitude of the noisy speech without affecting the phase of the noisy speech. The speech enhancement system includes a core estimator that applies to the noisy speech one of a first set of gains for each frequency bin. A noise adaptation module segments the noisy speech into noise-only and signal-containing frames, maintains a current estimate of the noise spectrum and an estimate of the probability of signal absence in each frequency bin. A signal-to-noise ratio estimator measures an a-posteriori signal-to-noise ratio and estimates an a-priori signal-to-noise ratio based on the noise estimate.

Abstract: A feature extraction process for use in a wireless communication system provides automatic speech recognition based on both spectral envelope and voicing information. The shape of the spectral envelope is used to determine the LSPs of the incoming bitstream and the adaptive gain coefficients and fixed gain coefficients are used to generate the “voiced” and “unvoiced” feature parameter information.

Abstract: Method and system for delivering a message over a telecommunications network to a recipient comprising transmitting a message over the telecommunications network to the recipient when a predetermined energy/silence condition is detected, performing echo cancellation on a signal communicated from the telecommunications network, and monitoring the signal to detect the energy/silence condition.

Abstract: Method and system for delivering a message over a telecommunications network to a recipient comprising transmitting a message over the telecommunications network to the recipient when a predetermined energy/silence condition is detected, performing echo cancellation on a signal communicated from the telecommunications network, and monitoring the signal to detect the energy/silence condition.

Abstract: A speech coder separates input digitized speech into component parts on an interval by interval basis. The component parts include gain components, spectrum components and excitation signal components. A set of speech enhancement systems within the speech coder processes the component parts such that each component part has its own individual speech enhancement process. For example, one speech enhancement process can be applied for analyzing the spectrum components and another speech enhancement process can be used for analyzing the excitation signal components.