Abstract: An apparatus and method for the robust recognition of speech during a call in a noisy environment is presented. Specific background noise models are created to model various background noises which may interfere in the error free recognition of speech. These background noise models are then used to determine which noise characteristics a particular call has. Once a determination has been made of the background noise in any given call, speech recognition is carried out using the appropriate background noise model.

Abstract: Automated capture of a string of intended alphabetic or alphanumeric characters is provided by using indications of telephone keys respectively selected in accordance with the intended characters and also using signals respectively representing utterances of each of the intended characters to generate selected characters. The telephone key indication is a dual tone multifrequency signal or an utterance representing a number corresponding to the telephone key. A selected character corresponding to one of the telephone key indications is generated based on the utterance signal corresponding to the one telephone key indication.

Abstract: A customer service system includes a natural language device, a remote device remotely coupled to the natural language device over a network and a database coupled to the natural language device. The database has a plurality of answers stored on it that are indexed to natural language keys. The natural language device implements a natural language understanding system. The natural language device receives a natural language question over the network from the remote device. The question is analyzed using the natural language understanding system. Based on the analysis, the database is then queried. An answer to the question is received based on the query, and the answer is provided to the remote device over the network.

Abstract: Simultaneous and temporal masking of digital speech data is applied to an MBE-based speech coding technique to achieve additional, substantial compression of coded speech over existing coding techniques, while enabling synthesis of coded speech with minimal perceptual degradation relative to the human auditory system. A real-time perceptual coder and decoder is disclosed in which speech may be sampled at 10 kHz, coded at an average rate of less than 2 bits/sample, and reproduced in a manner that is perceptually transparent to a human listener. The coder compresses speech segments that are inaudible due to simultaneous or temporal masking, while audible speech segments are not compressed.

Abstract: A Dynamic Time/Frequency Warping (DTFW) technique is disclosed for speaker verification, speech recognition and channel normalization, among other uses. The DTFW technique utilities best path dynamic programming methods using a 3-dimensional time frequency array representing the spectral differences between a test utterance (the utterance being analyzed) and a reference utterance (template). The array is created by summing the squares of the differences of each feature in each frame of the template with each feature in each frame of the utterance in question. Dynamic programming techniques are then used to find the minimal distance path matching the test utterance and the template so as to optimize the time and frequency warping paths.