A method and apparatus for processing a signal to extract selected information therefrom is provided. According to the method, an input signal is converted into a sequence of data samples, and these data samples are applied sequentially through a convolver having first and second sections, the output of the first section forming the input of the second section. A data sample in the second convolver section is then compared to each data sample in the first convolver section to produce an output signal of a plurality of data points, each of the data points representative of a midpoint position within the convolver between a pair of compared data samples. This comparison step is repeated for each data sample in the second convolver section and the output signals form a histogram from which the selected information is extracted. This method may be advantageously utilized in a speech recognition process for extracting various features from an input speech signal.
Inspection of solder paste on a printed circuit board using a histogram of an image before printing (pre-application image) to normalize an image after printing of the printed circuit board (post-application image) is described. Existing lighting and optics used for alignment of the screen-printing stencil to the printed circuit board are used for the solder paste inspection.
In a method of and an apparatus for the segmentation of speech, an acoustic speech signal is converted into N signals S.sub.i (f), each signal pertaining to a time interval i of N successive time intervals (1.ltoreq.i.ltoreq.N). For successive time intervals i a function c.sub.ij (FIG. 7) is then derived which is a measure of the agreement between the signals in the time intervals i and j. The middle m.sub.i of the pertaining function C.sub.ij is determined for each time interval i and those values i are determined which at least approximately correspond to zero-axis crossings with the same sign in a function d.sub.i (see FIG. 3b), d.sub.i indicating the difference between m.sub.i and i. The signal segmented in this way can be used for the derivation of diphones so that a library of diphones can be built up (see FIG. 1), or it can be used for the recognition of the speech utterance (see FIG. 2).
A method and apparatus for generating a signal transformation useful in signal processing. According to the preferred embodiment, a signal, e.g., a speech waveform, is first converted into a sequence of digital data samples, and a reference position along a first sub-part of the sequence is then selected. A "weighted" histogram corresponding to the reference position is then generated according to a correlation function. Thereafter, a new reference position is selected, for example, at a sub-part of the sequence located a pitch period of the signal from the original reference position, and an additional histogram is generated for this sub-part. The plurality of histograms comprise the transformation of the signal, which retains a substantial part of the informational content of the original signal. Therefore, the transformation is then used as the signal itself in signal processing applications such as speech compression and synthesis.
A compact, self-contained portable computing apparatus is provided which is completely supported by a user for hands-free retrieval and display of information for the user. The computing apparatus includes a voice-recognition module, in communication with a processor, for receiving audio commands from the user, for converting the received audio commands into electrical signals, for recognizing the converted electrical signals and for sending the recognized electrical signals to the processor for processing, the voice-recognition module being supported by the user. The computing apparatus further includes a display in communication with the processor for receiving information from the processor and for displaying the received information for the user, the display being supported by the user whereby the user may operate the computing apparatus to display information in a hands-free manner utilizing only audio commands.
A method and apparatus for converting the reproducing speed of an acoustic signal where, of the acoustic signals held in a data recording section 1, the input acoustic signal s1 (sampled for max. pitch cycle.times.2) is read from a process-start position P. A low-pass filter 7 controls the high-band component of the acoustic signal s1. A decimation section 8 performs appropriate down-sampling on a signal output from the low pass filter 7. The signal, thus down-sampled, is read into a signal buffer section 9. A down-sampled, input acoustic signal s2 is transferred from the signal buffer section 9 to a pitch-calculating section 3, which calculates a pitch cycle s3.