Abstract: A speech processing system (10) incorporates an analogue to digital converter (16) to digitize input speech signals for Fourier transformation to produce short-term spectral cross-sections. These cross-sections are compared with one hundred and fifty reference patterns in a store (34), the patterns having respective stored sets of formant frequencies assigned thereto by a human expert. Six stored patterns most closely matching each input cross-section are selected for further processing by dynamic programming, which indicates the pattern which is a best match to the input cross-section by using frequency-scale warping to achieve alignment. The stores formant frequencies of the best matching pattern are modified by the frequency warping, and the results are used as formant frequency estimates for the input cross-section. The frequencies are further refined on the basis of the shape of the input cross-section near to the chosen formants.

Abstract: In speech recognition it is advantageous to take account of noise levels both in recognition and training. In both processes signals reaching a microphone are digitized and passed through a filter bank to be separated into frequency channels. In training, a noise estimator and a masker are used with a recognizer to prepare and store probability density functions (p.d.f.s) for each channel partially defining Markov models of words to be recognized. The p.d.f.s are derived only from input signals above noise levels but derivation is such that the whole of each p.d.f. is represented. In recognition, "distance" measurements on which recognition is based are derived for each channel. If the signal in a channel is above noise then the distance is determined, by the recognizer, from the negative logarithm of the p.d.f. but if a channel signal is below noise then the distance is determined from the negative logarithm of the cumulative distance of the p.d.f. to the noise level.

Abstract: Input signals representative of speech are unreliable as inputs for speech recognition if processed conventionally by, among other processes, filtering into separate frequency bands. Further processing according to the invention takes the output from a filter bank and after operations of rectification and integration provides a process of median filtering and smoothing which significantly reduces the sampling rate of the filtered signals while retaining the important acoustic features of the input speech.