Lucent Technologies Inc.

Agere Systems Inc.

Agere Systems Guardian Corp.

Vasu Iyengar

James Charles Popa

Richard M Ubowski

Paul Joseph Davis

James Albert Johanson, Iii

Joseph M Cannon

Peter Kroon

Syed S Ali

Ronald E Crochiere

Glenn A Ehrich

A digital telephone answering device having speakerphone capability allows a recorded message to be played back and heard by the far-end party as well as over the local speakerphone by the near-end party during speakerphone operation as if it were a normal receive signal. Moreover, normal speakerphone conversation is possible during this conversational playback mode allowing the far-end party and/or near-end party to break-in over the played back pre-recorded message as desired and be heard by the other party. Both message and conversation signals are preferably (but not necessarily) at similar levels. Also, the message playback at the near end is preferably (but not necessarily) subject to the same speakerphone digital volume control as that received at the far end.

Message playback concurrent with speakerphone operation

Transmitting message playback concurrent with speakerphone operation

A configurable adaptive echo canceler which can perform acoustical echo cancellation (AEC) or hybrid echo cancellation (HEC) under software control. Particular parameters based on the particular application can be associated with either a suitable AEC or a suitable HEC, and the parameters are made selectable through the use of a control register. Exemplary parameters are accuracy of filter coefficients, a span select to manipulate the expected length of the echo path, a step size to adjust the adaptation of the echo cancellation, enablement of a tap update, selection of adaptation mode, center clipper selection, and sub-band or full-band echo cancellation. With the use of audio line switching, the configurable echo canceler can allow various modes of operation of the telephone, some of which may require an AEC in the installed unit and/or some of which may require an HEC. For instance, in a cordless telephone application, in an intercom mode, a configurable echo canceler in a cordless handset receiver and a configurable echo canceler in a base unit may each be configured as AECs. In a base speakerphone mode, the configurable echo canceler in the base unit may be configured to provide both HEC and AEC, while the configurable echo canceler in the cordless handset receiver may be configured as an AEC. In a handset speakerphone mode, the configurable echo canceler in the base unit may be configured as an HEC, while the configurable echo canceler in the cordless handset receiver may be configured as an AEC.

Configurable echo canceller

A digital cordless telephone combines speakerphone control information together with voice data information in each of a plurality of data packets transmitted between the remote handset and the base unit while the remote handset is operating in a speakerphone mode. To allow half-duplex speakerphone functionality in the digital remote handset, the speakerphone control information includes break-in information relating to a break in by the user of the remote handset into an ongoing conversation. To allow full-duplex, echo cancellation mode operation of the speakerphone functionality at the digital remote handset, the speakerphone control information includes coordinating information relating to an amount of echo cancellation performed by the base unit and by the remote handset. In the disclosed embodiment, the speakerphone control information further includes switching and signal level information.

Digital cordless telephone with speakerphone in a remote handset

A telephone answering device including two separate coders, a first coder for encoding/decoding fixed voice prompts spoken by a single speaker, and a second coder for encoding/decoding incoming and outgoing voice messages spoken by multiple speakers. The first coder uses a first set of codebooks trained based on a first set of utterances spoken by a single speaker, while the second coder uses a second set of codebooks trained based on a second set of utterances spoken by multiple speakers. Because the first set of utterances is significantly smaller in size than the second set of utterances, and the range of pitch period is significantly smaller in size for the first set of utterances spoken by a single speaker in comparison to that of the second set of utterances spoken by multiple speakers, the size of the first set of codebooks is significantly reduced relative to the size of the second set of codebooks. As a result, the fixed voice prompt messages may be compressed at a lower bit rate with a relatively high quality of encoding, thereby optimizing the codebook and reducing the amount of necessary memory capacity for storing the encoded fixed voice prompts. The memory required for the encoded first voice prompts is so small that they can be stored in a low cost DSP ROM.

Efficient compression of VROM messages for telephone answering devices

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. In an alternative embodiment according to the invention, a method of accessing a digital message stored in one of a base unit and a handset of a cordless telephone from the other of the base unit and the handset includes the steps of transmitting a control signal from the accessing device to the storing device, retrieving the digital message, in encoded form, from a memory in the storing device, and transmitting at least a portion thereof to the accessing device. The method further includes the step of decoding the encoded digital message in the accessing device.

Handset access of message in digital cordless telephone

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. Another controller configuration has a program running on a digital signal processor within the communication system, with the program analyzing the impulse response of the echo canceler so as to generate a control signal, and a call programmable gain attenuator in the signal path which is responsive to the control signal from the program to selectively insert or remove loss into the signal path.

Communication system based on echo canceler tap profile

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.

Call related information receiver unit

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). When one or more handsets become active and are registered as off-hook, the transmission of voice signals between the off-hook handsets and base is accomplished through the use of speech coding at a lower rate than that of ADPCM coding. The successful switching between ADPCM and speech coding within a multiple handset environment allows for speech quality to be maximized at all times.

Apparatus for enhanced voice quality in multiple cordless handset environment and method

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.

Speech coding selection based on call related information

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. This enables more efficient compression of speech samples, lighter duty processors, and improved voice quality upon reproduction by eliminating undesired clipping of the voice signal encountered in prior systems after periods of silence. The silence compressed speech samples are stored in a storage device for subsequent playback.

Silence compression for recorded voice messages

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.

Performance enhancement of tone detection and FSK detection using hybrid

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. The echo canceller system is well-suited for use in loudspeaking telephone sets, with the first echo canceller canceling a line echo through a hybrid transformer, and the second echo canceller canceling an acoustic echo between a loudspeaker and a microphone. The coefficient memory may be switched to the first background filter in response to a near-end signal without a far-end signal (i.e., transmit state), and switched to the second background filter in response to a far-end signal without a near-end signal (i.e., receive state).

Echo canceller system with shared coefficient memory

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.

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Allentown, PA, United States of America

Vasu Iyengar