Host signal processor modem and telephone

Details Host signal processor modem and telephone Host signal processor modem and telephone

1999-06-08

2001-06-26

Pham, Chi (Department: 2631)

Pulse or digital communications

Miscellaneous

Reexamination Certificate

active

06252920

ABSTRACT:

REFERENCE TO MICROFICHE APPENDIX
The present specification comprises a microfiche appendix. The total number of microfiche sheets in the microfiche appendix is one. The total number of frames in the microfiche appendix is 48.
BACKGROUND OF THE INVENTION
1. Field of the Invention
This invention relates to systems including multipurpose modems which implement modem and telephone functions using resources such as processors and audio hardware which are native to host computers.
2. Description of Related Art
Multipurpose modems often incorporate digital data and fax functions with speakerphone and answering machine capabilities. Such modems typically have components which serve multiple purposes, such as a speaker which provides an audible sound for monitoring a handshake sequence between connecting modems and provides sound during telephone communications, but many functions require special hardware within the modem which increases the cost of the modem.
Host signal processor (HSP) modems have been developed which eliminate signal processors and other hardware in the modems in favor of software executed by the processor in a host computer. Software for such HSP modems performs many of the signal and data conversions required of a modem. Use of the host processor reduces modem hardware costs by reducing hardware in the modem. An efficient and inexpensive way to expand an HSP modem to provide standard telephone communications is desired.
SUMMARY OF THE INVENTION
In accordance with the invention, a communication system uses host signal processor (HSP) modem hardware, native audio hardware in a host computer, and procedures executed by the host computer to operate as a modem and a telephone. The native audio hardware provides audio output and/or input for telephone functions. The HSP modem hardware provides the required interface with telephone lines. Operating system protocols allow software communications between the HSP modem hardware and the audio hardware without any direct hardware connections between the HSP modem hardware and the audio hardware even though the audio hardware and HSP modem hardware operate asynchronously with independent sample clocks. A clock recovery procedure executed by the host computer matches data transfer rates and compensates for differences between the independent sample clocks.
The native audio hardware not only provides resources for expanding the HSP modem to include speakerphone functions but can also make a modem handshake sequence audible for user monitoring. Accordingly, no speakers or speaker driver circuits for monitoring handshake are required in the HSP modem hardware.
In one embodiment of the invention, a communication system includes an analog-to-digital converter or a codec which converts an analog signal from an input line such as a telephone line to digital samples accessible to a host computer having native audio hardware. Software executed by the host computer transfers the digital samples from the converter to the native audio hardware to provide audible sounds from the signal received on the input line. Digital samples from the host computer (i.e. from a program executed by the host computer or from the audio hardware) are converted to an analog output signal transmitted on an output line.
In one embodiment, a first index indicates where in a buffer new samples from the HSP modem hardware are transferred, and a second index indicates which samples in the buffer are transferred to the audio hardware. The first index leads the second index to provide a margin between where samples are written and where samples are read. Comparing the first index to the second index indicates whether a difference between the rates at which samples are written and read has increased or decreased the margin. Rates are equalized by duplicating or deleting samples to increase or decrease the margin. This prevents overflow or underflow of the buffer.


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