A method of real-time communication between a plurality of users each with respective communication devices having associated displays, the method comprising the steps of providing each of the plurality of users with collaboration initiation software at their communication devices and allowing at least first and second users to connect to at least one communication network by logging in at their respective communication devices. Further, the method comprises maintaining service records for at least the first and second logged in users, the service records including user identification information and an associated location where each user is logged in no matter where they are located. A display of a user identifier for at least the second user on the display of at least a first user's communication device is caused, thereby allowing the user to select the displayed second user's identifier and retrieve necessary addressing information of the second user so that a connection between the first and second users can be established, thereby enabling real-time communication displayed on the display of the first and second users.

Method for real-time communication between plural users

A hearing compensation system for the hearing impaired comprises a plurality of bandpass filters having an input connected to an input transducer and each bandpass filter having an output connected to the input of one of a plurality of multiplicative AGC circuits whose outputs are summed together and connected to the input of an output transducer. The multiplicative AGC circuits attenuate acoustic signals having a constant background level without the loss of speech intelligibility. The identification of the background noise portion of the acoustic signal is made by the constancy of the envelope of the input signal in each of the several frequency bands. The background noise that will be suppressed includes multi-talker speech babble, fan noise, feedback whistle, fluorescent light hum, and white noise.

Hearing aid device incorporating signal processing techniques

A CMOS integrated signal processing system for a sampling receiver includes a timing recovery circuit, wherein an on-chip numerically controlled oscillator is operative at periods T that are initially equal to the nominal baud rate of the signals controls a sinc interpolator receiving samples at the sampling rate. A loop filter is coupled to the sinc interpolator and to the numerically controlled oscillator. The arrangement is capable of handling various symbol rates. The system includes a circuit for carrier recovery, having a second on-chip numerically controlled oscillator, a digital derotation circuit responsive to the second numerically controlled oscillator, accepting an in phase component and a quadrature component of the sampled signals. An adaptive phase error estimation circuit is coupled in a feedback loop.

Signal processing system

A networked multimedia system (10) comprises a plurality of networks (40) and at least one storage server (100). A signal path interconnects the workstations (12) and the storage server (100). Each workstation (40) includes video and audio reproduction capabilities, as well as video and audio capture capabilities. Any given storage server (100) comprises a set of storage cells (120) that operate under the direction of a storage cell manager (160). A storage cell (120) may include one or more encoding (132) and transcoding converters configured to convert or transform audio and video signals originating at a workstation into a form suitable for storage. A storage cell (120) may further include one or more decoding converters (134) configured to convert stored signals into a form suitable for audio and video signal reproduction at a workstation. Each storage cell (120) additionally includes at least one storage device (150) and storage device controller (152) capable of storing, for later retrieval, signals generated by one or more converters. The storage cell controller responds to signals received from the workstations (40), and oversees the operation of the storage cells to facilitate the storage of converted audio and video signals in at least one file that can be simultaneously accessed by one or more application programs executing on one or more workstions

Scalable networked multimedia system and applications

A sampled amplitude read channel is disclosed for reading binary data from a computer disk storage system, wherein the read channel sub-samples an analog read signal at a rate lower than the baud rate and detects the binary data from the sub-sampled values using a sequence detector. In one embodiment, the sub-sampled values are interpolated to generate synchronous sample values which are processed by a conventional sequence detector. In another embodiment, the sequence detector is modified to detect the binary data directly from the sub-sampled values. In yet another embodiment, the sequence detector comprises a remodulator and an error pattern detector for detecting and correcting bit errors in the detected binary data. In addition, for the various embodiments a channel code increases the distance property of the sequence detector in order to compensate for the degradation in performance caused by sub-sampling.

Sub-sampled discrete time read channel for computer storage systems

The characteristics of a room in which a speakerphone (20) is located are measured by determining a time between a test signal and its first attack, and a number of sample periods between the first attack and a time when the average power in the echo falls below a threshold. The first-attack time determines a pre-filter delay and the number of sample periods determines a tap length for an adaptive echo-canceling filter (62). In a teleconferencing environment, an annoying initialization sequence is avoided by initializing filter coefficients for each microphone (140), and saving the initial filter coefficients generated thereby in a corresponding nonvolatile memory (104). In response to an off-hook signal, the coefficients are retrieved from the nonvolatile memory (104). During operation, the coefficients are constantly updated. If another microphone (141) is enabled, the stored coefficients corresponding to that microphone (141) are dynamically substituted for the present coefficients.

Apparatus and method for noise reduction for a full-duplex speakerphone or the like

In an asynchronous communication system such as a V.32 modern (80), an input signal is sampled at a near-end clock rate. Each sample is then interpolated in an interpolation filter (92) to provide corresponding interpolated values. The interpolation filter (92) uses a selected one of a predetermined number of sets of windowed sinc function coefficients, each set having a successively greater phase offset. A time drift between near-end and far-end clocks is measured by tracking the coefficient shift in a passband phase-splitting, fractionally-spaced equalizer (95). When the time drift exceeds a threshold, a subsequent set of windowed sinc function coefficients is selected. When the time drift exceeds the threshold after the last set of coefficients is used, the first set is again selected and an interpolated value is either dropped or repeated in forming the far-end data samples.

Method and apparatus for asynchronous timing recovery using interpolation filter

A method of dual-tone multifrequency (DTMF) detection which decimates and adaptively filters an input signal is provided to efficiently detect a presence of a DTMF signal. The input signal is provided to a half-band filter (14) to be decimated in frequency in accordance with Nyquist's theory. A decimated input signal is subsequently processed to form a low frequency component signal and a high frequency component signal. The low frequency component signal is again decimated by a decimator (24). The decimated low frequency component signal and the high frequency component signal are each filtered by an adaptive fir filter (22, 26) to provide a first and a second frequency parameter and a first and a second gain factor, respectively. The first and second frequency parameters and the first and second gain factors are then tested by a tone identifier (28) to determine if the input signal includes a valid DTMF signal.

DTMF detection having sample rate decimation and adaptive tone detection

A digital receiver, such as "C-QUAM" receiver (10), has phase error correction. In another form, a software program may be executed by a conventional digital signal processor to also implement phase error correction. A digital input signal is demodulated to form an in-phase and a quadrature component. The in-phase and quadrature components are processed by a digital envelope detector (24) to form a composite signal containing left and right audio channel information. The in-phase component and composite signal are both processed by a reciprocal cosine estimator (28) and a quadrature channel circuit (38) to provide a difference signal also containing left and right audio channel information. The difference signal is input to phase error correction circuitry (16, 22, 26) to estimate a phase error of the digital input signal. The estimated phase error is then used to correct an actual phase error of the digital input signal during demodulation.

Circuit and method for phase error correction in a digital receiver

A digital receiver (10) has a symmetrical base band processor (28, 30, 32, 34, and 36) which concurrently provides a left and a right channel of audio information. Because of the symmetrical design of the base band processor, each of the left and right channels of information has perfectly balanced phase and amplitude with respect to each other. The base band processor includes an adaptive gain compensator (28), multipliers (30, 32), and high pass filters (34, 36). Compensator (28) iteratively derives a gain factor which is multiplied by both an in-phase component and a quadrature component of a quadrature modulated input signal to respectively provide a composite of the left and right channels and a difference between the left and right channels. Subsequently, adders (44, 48) arithmetically manipulate the composite and difference of the channels to separate the left and the right audio information.

Dion M. Funderburk

Papers

Apparatus and method for noise reduction for a full-duplex speakerphone or the like

Method and apparatus for asynchronous timing recovery using interpolation filter

DTMF detection having sample rate decimation and adaptive tone detection

Circuit and method for phase error correction in a digital receiver

A symmetrically balanced phase and amplitude base band processor for a quadrature receiver