High fidelity speech encoding for telecommunications systems

• US 5,228,076 A
• Filed: 02/08/1991
• Issued: 07/13/1993
• Est. Priority Date: 06/12/1989
• Status: Expired due to Fees

• Current Assignee: Emil Hopner, Stanley M. Head
• Original Assignee: Emil Hopner, Stanley M. Head
• Inventors: Head, Stanley M., Hopner, Emil
• Primary Examiner(s): Dwyer, James L.
• Assistant Examiner(s): CHAN, WING F
• Application Number:
  US07/652,952
• Time in Patent Office: 886 Days
• Field of Search: 379/387, 379/201, 379/93, 379/94, 379/96-98
• US Class Current: 379/93.17
• CPC Class Codes: H04Q 11/04 for time-division multiplexing H04Q11/0001 takes precedence
• CPC Class Codes: H04Q 11/0407 using a stored programme control

Abstract

A user-orientated telephone instrument including high-fidelity speech encoding, encryption capability and many function presently provided by centralized call processing is described. Utilizing an ADPCM encoding process including forward error correction, a high quality relatively noise immune signal is provided. To the user, the instrument appears as a unique electronic telephone which offers many special services to facilitate communications with other people and with computers. Advanced features such as optional hands-off voice on/off-hook control, personal identification by voice print and remote chart presentation are provided.

Claims

1. A telephone for producing and receiving audible speech sound having no foldover distortion owing to the natural frequency spectrum of vowel sounds and having no lowpass filter, said telephone comprising:

a microprocessor;

first transducer means for producing, transmitting and receiving signals representing audible speech sound; and

signal processing means coupled to the microprocessor and to the transducer means, for converting said audible speech sound to digital signals having a predetermined data rate, said data rate including a preselected sampling rate and a preselected digitizing rate;

said preselected sampling rate selected at a rate for no foldover distortion of said vowel sounds and said preselected sampling rate is less than the Nyquist sampling rate of consonant sounds found in said audible speech sound;

said signal processing means including volume compression means for compressing said audible speech sound;

said preselected digitizing rate selected for use with said volume compression means for reducing quantization noise;

said preselected sampling rate and said preselected digitizing rate cooperating to provide high fidelity speech quality;

said signal processing means further including error correction means for producing digital signals representing error correction code at a preselected rate and for combining said last-mentioned digital signals with said digital signals representing audible speech sound;

said signal processing means further including conversion means for converting digital signals representing said audible speech sound to speech signals representing said audible speech sound.

2. A telephone as in claim 1 wherein said preselected digitizing rate comprise 3 bits of digital data.

3. A telephone as in claim 1 further including:

input means coupled to the microprocessor for entering information; and

second transducer means for producing, transmitting and receiving analog signals representing video data for display on said input means.

4. A telephone as in claim 2 wherein said preselected sampling rate is approximately 10,000 samples per second.

5. A telephone as in claim 4 wherein said preselected rate for said error correction code is approximately 2,000 bits per second.