Showing papers on "Speech coding published in 1968"

Predictive coding of speech signals

Abstract: Predictive coding of signals, i.e., the reduction or redundancy in a signal by subtracting from it that part which can be predicted from its past, is a well-known technique for reducing the channel capacity required to transmit a signal with specified fidelity. It has been widely applied to signals, such as television signals which have regularly repeating intervals of information, but has not been satisfactorily applied to signals, such as speech, which exhibit characteristics that vary from speaker to speaker and from time to time for one speaker. According to this invention, an adaptive predictor is employed which is readjusted periodically to match the time-varying characteristics of a speech signal.

Intelligibility of speech signals spectrally compressed by a sampling-synthesizing technique

Abstract: The purpose of this study was 1) to measure the intelligibility of speech signals processed with a frequency converter designed to effect a spectral compression by a sampling-synthesizing technique, and 2) to investigate the effect of a short training period on the perception of the speech signals so processed. The input signals were composed of 71 CVC words recorded by an American female speaker. The highest center frequency for the input of the 22 channels was 9500 Hz. The four compression ratios of the output to the fixed input frequency spectrum were 1.0, 0.7 0.5, and 0.4 for all 22 channels used. Eighty listeners were randomly selected and assigned to one of the four conditions of spectral compression. The intelligibility score of each condition of spectral compression was obtained from a multiple choice test of six possible choices for each stimulus. The average intelligibility score for all English phonemes was obtained for each condition of compression. The intelligibility of each vowel and consonant was also calculated for all conditions. Only 15 minutes of programmed training in listening to this processed speech was found to be significantly effective for improving perception in all four conditions.

Problems of diver communication

Abstract: The major problems of underwater communication are microphone and earphone construction speech distortion caused by high pressure and/or mixed gas breathing environments, and face mask design. Microphone and earphone elements have recently been developed that are relatively unaffected by increased ambient pressure. These elements are constructed of piezoelectric ceramics encased in a unique rubber housing to prevent water leakage. Preliminary tests indicate that the frequency response of these elements is changed less than 3 dB by pressures exceeding 500 lbf/in2gauge. Extensive research in speech distortion under extreme pressures with various gas mixtures includes voice spectrum analysis, study of adaptation changes in speech, and intelligibility measurements of helium speech before and after processing or "unscrambling." These basic data provide information for the design of microphones, masks, and electronic and mechanical means of normalizing distorted speech. Face mask design must include consideration of breathing requirements, as well as speaking requirements. The design of a speaking cavity has, of necessity, been based on empirical data. These designs often conflict with the primary requisite of proper breathing design. Electronic compensation is often employed to correct or improve deficiencies in mask characteristics.

Method of multiplex speech synthesis

Abstract: The invention relates to a method of channel vocoder speech synthesis. For representing the speech signals for several independent speech lines, digital speech data stored in data processors are used. In addition, the method provides for the storage of the time-sampled digital description of the transient behaviors of n spectrum channel band-pass filters. Only one such description is needed for synthesizing the speech signals for m speech lines. For a given speech line the transient responses of the band-pass filters are modulated by the frequency function for the given line. The modulated transient values are added for corresponding time samples and stored in a delay line for the given speech line. The stored value of a speech line is released at points in time defined by the excitation function, thus releasing a digital description of the transient response of the set of band-pass filters as if they were excited by a unit pulse and modulated by the frequency function of the given speech line. The digital description is demodulated to an analogue form by conventional means.