Digital speech processing system having reduced encoding bit requirements
First Claim
1. In a linear prediction speech processing system wherein a digital speech signal is divided in the time domain into sections and each section is analyzed to determine the parameters of a speech model filter, a volume parameter and a pitch parameter, a method for coding the determined parameters to reduce bit requirements and increase the frame rate of transmission of the parameter information for subsequent synthesis, comprising the steps of:
- combining the determined parameters of at least two successive speech sections into a block of information;
coding the determined parameters for the first speech section in said block in complete form to represent their magnitudes; and
coding at least some of the parameters in the remaining speech sections in said block in a form representation of their relative difference in magnitude from the corresponding parameters in said first speech section.
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Abstract
A digitized speech signal is divided into sections and each section is analyzed by the linear prediction method to determine the coefficients of a sound formation model, a sound volume parameter, information concerning voiced or unvoiced excitation and the period of the vocal band base frequency. In order to improve the quality of speech without increasing the data rate, redundance reducing coding of the speech parameters is effected. The coding of the speech parameters is performed in blocks of two or three adjacent speech sections. The parameters of the first speech section are coded in a complete form, and those of the other speech sections in a differential form or in part not at all. The average number of bits required per speech section is reduced to compensate for the increased section rate, so that the overall data rate is not increased.
48 Citations
15 Claims
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1. In a linear prediction speech processing system wherein a digital speech signal is divided in the time domain into sections and each section is analyzed to determine the parameters of a speech model filter, a volume parameter and a pitch parameter, a method for coding the determined parameters to reduce bit requirements and increase the frame rate of transmission of the parameter information for subsequent synthesis, comprising the steps of:
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combining the determined parameters of at least two successive speech sections into a block of information; coding the determined parameters for the first speech section in said block in complete form to represent their magnitudes; and coding at least some of the parameters in the remaining speech sections in said block in a form representation of their relative difference in magnitude from the corresponding parameters in said first speech section. - View Dependent Claims (2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13)
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14. Apparatus for analyzing a speech signal using the linear prediction process and coding the results of the analysis for transmission, comprising:
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means for digitizing a speech signal and dividing the digitized signal into blocks containing at least two speech sections; a parameter calculator for determining the coefficients of a model speech filter based upon the energy levels of the speech signal, and a sound volume parameter for each speech section; a pitch decision stage for determining whether the speech information in a speed section is voiced or unvoiced; a pitch computation stage for determining the pitch of a voiced speech signal; and coding means for encoding the filter coefficients, sound volume parameter, and determined pitch for the first section of a block in a complete form to represent their magnitudes and for encoding at least some of the filter coefficients, sound volume parameter and determined pitch for the remaining sections of a block in a form representative of their difference from the corresponding information for the first section. - View Dependent Claims (15)
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Specification