Abstract: In one embodiment according to the invention, a digital cordless telephone includes a base unit adapted to couple to a telephone network, to transmit an outgoing message to an incoming caller, to store a digital message from the incoming caller, and to transmit and receive digital samples over an RF link. The digital cordless telephone of this embodiment also includes a handset adapted to transmit and receive digital samples over the RF link. According to this embodiment, the base unit and handset are further adapted to enable a user to monitor the stored digital message from the handset.

Abstract: A controller and a method for controlling the amount of loss in signal paths of a communication system are disclosed. The method includes sampling the impulse response of an echo cancelling circuit, detecting an abnormal signal condition by comparing the impulse response to a reference, and inserting loss into a first signal path when an abnormal signal condition is detected. One controller configuration for adjusting the amount of loss in the signal paths of a communication system has an adjustable attenuator in the signal path having a control signal input, a memory element which stores a reference, and a comparator having one input connected to the memory of the echo canceler circuit which samples the impulse response, another input connected to the memory which stores the reference, and an output connected to the control signal input of the attenuator.

Abstract: An adjunct Type II caller ID/call waiting (CIDCW) receiver unit is provided which has a greatly improved ability to detect tones and other call related information on a telephone line from a central office while the customer premises equipment is in an off-hook condition. The inventive adjunct CIDCW receiver unit is placed in series between the telephone line from the central office and the customer premises equipment, rather than in parallel with the customer premises equipment as in conventional adjunct CIDCW receiver units. A second telephone line interface (TLI) is included in the adjunct CIDCW receiver unit for connection to the customer premises unit to simulate the impedance of the telephone line. The adjunct CIDCW receiver unit has the ability to disconnect, mute or suppress the microphone signal from the customer premises equipment from being included in the signal received by the call related information receiver portion of the adjunct CIDCW receiver unit.

Abstract: The present invention provides a method and system to improving voice transmission quality within an environment containing multiple off-hook cordless telephones all residing on the same subscriber line. This is achieved by coordinating the sending, receiving, and acknowledging (i.e., handshaking) of control and telephone signals between the multiple cordless telephones and the base. The communication between cordless telephones takes place through a single household telephone line and the RF link existing between each cordless handset and its accompanying base station. When a single handset of a multiple handset cordless telephone system is registered as off-hook, the handshaking between that handset and base results in the transmission of voice signals between the off-hook handset and the base which are digitized using Adaptive Differential Pulse Code Modulation (ADPCM).

Abstract: A variable voice compression device and method in a voice messaging system is provided. The device includes a call related information/coding table or database which contains associations between call related information regarding desired calling parties, and selected coding techniques and effective data rates resulting in varying voice compression ratios. Upon receipt of an incoming call, call related information sent from the central office is compared to entries in the call related information/coding table or database. If a match is found, the subsequent voice message is encoded with a coding technique and effective data rate determined from the matching entry in the call related information/coding table or database. Otherwise, the voice message is encoded with a default coding technique and effective data rate.

Abstract: A silence compression system that improves data compression in a digital speech storage device, such as a digital telephone answering machine, without undue clipping of voice signals. Instead of employing only real-time compression, the inventive silence system analyzes and compresses or re-compresses digital speech samples stored previously, when the voice messaging system is off-line or otherwise in a low priority state. A method of silence compression comprises receiving real-time speech samples, storing the same in memory, and analyzing the stored speech samples at a later time to determine thresholds for periods of silence. The periods of silence are then compressed, and the silence compressed voice message is restored in memory. In this fashion, the processor is not required to make a silence period determination on-the-fly simultaneous with encoding and compression of the real-time voice message, and thus is not subjected to heavy processor loads typically encountered in real time.

Abstract: Type 2 caller ID/call waiting (CIDCW) customer premises equipment (CPE) is provided which has a greatly improved method of detecting tones indicating the availability of call information regarding an incoming call while the customer premises equipment is in an off-hook condition. A hybrid echo canceler (HEC) algorithm is added to the customer premises equipment (CPE) to suppress the signal from the microphone of the customer premises equipment from being included in the signal analyzed for the presence of tones. Thus, the microphone of the customer premises equipment need not be muted upon detection of the alerting CAS tone sequence in a Caller ID service. The HEC algorithm runs substantially continuously in a preferred embodiment, and may be combined with a conventionally operating HEC which cancels reflections due to the hybrid or telephone line interface.

Abstract: An echo canceller includes receive and send paths each having an input and output port, a first non-adaptive filter with first filter coefficients for generating a first replica signal in response to a receive input signal, a second adaptive filter with second filter coefficients for generating a second replica signal in response to the receive input signal, a first subtracter for generating a send output signal representing a difference between a send input signal and the first replica signal, a second subtracter for generating an error signal representing a difference between the send output signal and the second replica signal, and a controller for replacing the first filter coefficients by the sum of the first and second filter coefficients and for resetting the second filter coefficients in response to a calculation that includes a first quantity associated with the send output signal and a second quantity associated with the error signal.

Abstract: An echo canceller system includes first and second echo cancellers. Each echo canceller includes a foreground filter and an adaptive background filter, with the foreground filter providing the actual echo cancellation and the background filter updating the foreground filter. The echo canceller system also includes send and receive paths, a shared coefficient memory, and a controller for switching the shared coefficient memory between background filters in response to signals along the send and receive paths. The switching includes resetting the shared coefficient memory to prevent any transfer of filter coefficients between the background filters. The background filters operate one at a time, depending on which background filter has access to the shared coefficient memory, while the foreground filters operate simultaneously.

Abstract: In methods and apparatus for estimating the frequency response of telecommunications channels, signals carried on the channels are divided into pluralities of signal segments of limited duration, unvoiced signal segments are identified, and spectral components of the unvoiced signal segments are measured. The measured spectral components may be integrated over time to derive time-averaged measurements of the spectral components. The time-averaged measurements may be compared to corresponding components of expected unvoiced signal spectra to improve the accuracy of the frequency response estimates. Filters may be inserted in the channels to adjust the frequency responses, the filters having filter characteristics selected in response to the measured spectral components. The methods and apparatus can be used to compensate for poor bass frequency response in voice channels.

Abstract: A speech bandwidth extension method and apparatus analyzes narrowband speech sampled at 8 kHz using LPC analysis to determine its spectral shape and inverse filtering to extract its excitation signal. The excitation signal is interpolated to a sampling rate of 16 kHz and analyzed for pitch control and power level. A white noise generated wideband signal is then filtered to provide a synthesized wideband excitation signal. The narrowband shape is determined and compared to templates in respective vector quantizer codebooks, to select respective highband shape and gain. The synthesized wideband excitation signal is then filtered to provide a highband signal which is, in turn, added to the narrowband signal, interpolated to the 16 kHz sample rate, to produce an artificial wideband signal. The apparatus may be implemented on a digital signal processor chip.