A personal communications system is described which includes components of software and hardware operating in conjunction with a personal computer. The user interface control software operates on a personal computer, preferably within the Microsoft Windows® environment. The software control system communicates with hardware components linked to the software through the personal computer serial communications port. The hardware components include telephone communication equipment, digital signal processors, and hardware to enable voice, fax and data communication with a remote site connected through a standard telephone line. The functions of the hardware components are controlled by control software operating within the hardware component and from the software components operating within the personal computer. The major functions of the system are a telephone function, a voice mail function, a fax manager function, a multi-media mail function, a show and tell function, a terminal function and an address book function. The telephone function allows the present system to operate, from the users perspective, as a conventional telephone using either hands-free, headset or handset operation. The telephone function is more sophisticated than a standard telephone in that the present system converts the voice into a digital signal which can be processed with echo cancellation, compressed, stored as digital data for later retrieval and transmitted as digital voice data concurrent with the transfer of digital information data.

Computer-based multifunction personal communications system.

The present invention includes software and hardware components to enable digital data communication over standard telephone lines. The present invention converts analog voice signals to digital data, compresses that data and places the compressed speech data into packets for transfer over the telephone lines to a remote site. A voice control digital signal processor (DSP) operates to use one of a plurality of speech compression algorithms which produce a scaleable amount of compression. The rate of compression is inversely proportional to the quality of the speech the compression algorithm is able to reproduce. The higher the compression, the lower the reproduction quality. The selection of the rate of compression is dependant on such factors as the speed or data bandwidth on the communications connection between the two sites, the data demand between the sites and amount of silence detected in the speech signal. The voice compression rate is dynamically changed as the aforementioned factors change. A negotiation handshake protocol is described which enables the two sites to negotiate the compression rate based on such factors.

Dynamic selection of compression rate for a voice compression algorithm in a voice over data modem

A loudspeaking telephone station (speakerphone) includes a loudspeaker and one or more microphones within the same housing. The microphones are directional, each having a polar response characteristic that includes a major lobe, one or more side lobes, and nulls between pairs of lobes. The loudspeaker is positioned in the null of the polar response characteristic that resides between the major lobe and an adjacent side lobe. The microphone apparatus is positioned so that its major lobe is aimed in a direction that is generally perpendicular to the direction that the loudspeaker is aimed. Means are provided for increasing the distance between input sound ports of a first-order-gradient (FOG) microphone and thereby improving its sensitivity. A pair of such improved FOG microphones are used in assembling a second-order-gradient microphone. Full duplex operation is achieved when a pair of echo cancelers are added to further reduce the coupling between the transmit and receive directions of the speakerphone.

Loudspeaking telephone station including directional microphone

A personal communications system enables the operator to simultaneously transmit voice and data communication to a remote site. The personal communications system is equipped with two telephone line interfaces to allow connection between two remote sites. The connection between the first remote site and the local site may operate in a voice over data communications mode to simultaneously send compressed voice and data. A digital transmission protocol which is consistent with current packet standards is used to create an independent channel through use of a modified supervisory packet for negotiating communication parameters, including the speech compression algorithm, the speech compression ratio, the communication multiplex scheme, and other operations needed for control of remote hardware interfaces.

Voice over data modem with selectable voice compression

A system and a method for echo suppression which use two or more microphones (22, 36 and/or 68) for beamforming of an echo signal (L') corresponding to an incoming speech signal (L) that is output by a loudspeaker (20). The outputs of the microphones (22, 36 and/or 68) are appropriately filtered and then linearly combined (Figs. 3-7 and 10-11) to cancel the echo signal (L'), even if the incoming speech signal (L) has been non-linearly distorted (e.g., in a digital-to-analog converter (16), an amplifier (18) and/or the loudspeaker (20). Also, the microphones (22, 36 and/or 68) may be selectively positioned or pointed (e.g., as shown in Figs. 8-9) to allow linear cancellation of ambient noise (N).

Echo canceler for non-linear circuits

A full-duplex digital speakerphone 10 includes a transmit signal path having an output coupled to a telephone trunk and a receive signal path having an input coupled to the telephone trunk and an output coupled to a loudspeaker. The speakerphone further includes a room echo cancellation adaptive filter 56 and a trunk echo cancellation adaptive filter 66. Serially coupled within the transmit signal path is a selective suppression block 50 for suppressing a component of a Mu-Law or an A-Law quantization error signal. A second selective suppression block 52 is serially coupled within the receive signal path. Suppression of non-linearities due to Mu-Law or A-Law signal conversion is also accommodated by providing a non-linear signal processing block 40 at an input to an adaptive filter and an optional non-linear signal processing block at an output of the adaptive filter. Each of the blocks emulates and compensates for signal converter non-linearity. The speakerphone facilitates adaptive filter coefficient initialization by beginning a call in a half-duplex mode and switching to full-duplex when filter coefficients are adapted. The speakerphone also has a variable adaptation step size which is a function of a short-term estimate of signal power within the associated transmit or receive signal paths.

Ygal Arbel

Papers

Full-duplex digital speakerphone