Bioavailable sustained release oral opioid analgesic dosage forms, comprising a plurality of multiparticulates produced via melt extrusion techniques are disclosed.

Melt-extruded orally administrable opioid formulations

A unit dose sustained-release oral dosage form containing a plurality of melt-extruded particles, each consisting essentially of a therapeutically active agent, one or more retardants, and an optional water-insoluble binder is disclosed. The particles have a length of from about 0.1 to about 12 mm and can be of varying diameters and each unit dose provides a release of therapeutically active agents over at least about 8 hours. Methods of preparing the unit doses as well as extrusion processes and methods of treatment are also disclosed.

Extruded multi-particulates

Melt-extrusion multiparticulates

In a communications network, voice and facsimile calls are established between first and second and second terminals via first and second media gateways in a communications network. Each gateway is capable of autonomous transitioning between a voice over IP (VoIP) call mode and a facsimile over IP (FoIP) call mode. A bearer path for the call is established via a media gateway controller. During call setup, the media gateways exchange confirmation of each gateway's capability to perform autonomous transitioning between VoIP and FoIP call modes. The call is established as a VoIP call, and, on a determination by the media gateways that a facsimile call is required, each media gateway autonomously transitions between its VoIP and FoIP call modes without the intervention of the media gateway controller.

Voice and fax over IP call establishment in a communications network

An improved method and apparatus for filtering an electronic environment with relatively high accuracy and efficiency and relatively short flow-through delay ("latency") is disclosed. Embodiments of the invention may be applied to digital filters implemented in software, hardware or a combination of both for applications such as audio filtering or electronic modelling of acoustic system characteristics. The method disclosed is broadly applicable in the field of signal processing and may be used to advantage, for example, in adaptive filtering; audio reverberation processing; adaptive echo cancellation; spatial processing; virtual reality audio; correlation, radar; radar pulse compression; deconvolution; seismic analysis; telecommunications; pattern recognition; robotics; 3D acoustic modelling; audio post production (including auralization and auto reverberant matching); audio equalization; compression; sonar; ultrasonics; secure communication systems; digital audio broadcast, acoustic analysis, surveillance; noise cancellation; and echo cancellation.

Method and apparatus for filtering an electronic environment with improved accuracy and efficiency and short flow-through delay

An adaptive nonrecursive digital filter includes a first filter unit having N controllable filter coefficients. A control unit controls the filter coefficients as a function of an input signal applied to the first filter unit at a given sampling rate and as a function of a difference between a reference signal and an output signal output by the first filter unit, in accordance with the least mean error square in each case. The first filter unit and the control unit sample the input signal and the reference signal continuously, with a cyclically varying delay, at a correspondingly increased sampling rate. A second filter unit is operated at the given sampling rate and has N controllable filter coefficients. The second filter unit receives the input signal, outputs a further output signal and has filter coefficients also being controlled by the control unit.

Adaptive nonrecursive digital filter and method for forming filter coefficients therefor

The invention permits the improvement of the quality of an audio transmission in which audio data including samples of an audio signal are asynchronously transmitted in data packets via a packet-oriented communication network, such as a LAN. According to the invention, the audio data to be transmitted and/or already transmitted are digitally converted such that their sampling rate is altered and/or the duration of an audio signal represented by the audio data is modified while largely retaining its pitch. The conversion is performed based on a transmission situation currently detected, in such a manner that a quality of service of the audio transmission is optimized.

Method for improving the quality of an audio transmission via a packet-oriented communication network and communication system for implementing the method

The invention relates to a method and a device for reproducing a binaural output signal generated from a monaural input signal and comprising a first output signal and a second output signal via at least a first and a second speaker of a binaural headset particularly for VoIP applications.

Method and device for reproducing a binaural output signal generated from a monaural input signal

A switching center is provided which has an internal broadband bus having broadband interfaces coupling both peripheral devices and internal system components of the switching center. In addition, a data packet communications protocol is provided which can be used for the direct and specifically directed transfer of communication data, which are to be relayed over connections, via the broadband bus between a respective peripheral device and a respective internal system component and also between the peripheral devices themselves.

Switching center and central switching controller with internal broadband bus

The method involves receiving symbols (Z) as a signal (S) which contains a mixture of two audio frequencies, selected from two different groups (HG, LG) of frequencies. The signal is sampled with a predetermined sampling rate, and is supplied to a first decimation filter (DO), in which the signal is low-pass filtered (DFO), and its sampling rate is halved. The filtered signal is supplied to a second decimation filter (D1), which divides it under further halving of the sampling rate, through band pass filtering (DF1), into a high-pass filtered signal (S1) and a low-pass filtered signal (S2). The filtered signals are respectively supplied to a first and second adaptive digital notch filter (ANWDF1, ANWDF2), whose filter coefficients (kl, k2) are a measure for the frequencies of the signal, and to a first and second level detector (PD1, PD2), after further halving of the sampling rate, to determine the amplitude (al, a2) of the respective signal. The transmitted symbol is determined in a discriminator unit (Rw), from the determined amplitudes of the high-pass and the low-pass filtered signal, and the adaptive filter coefficients. The first and second decimation filters are preferably recursive bridged digital filters of seventh and third degree, respectively.

Gonzalo Lucioni

Papers

Adaptive nonrecursive digital filter and method for forming filter coefficients therefor

Method for improving the quality of an audio transmission via a packet-oriented communication network and communication system for implementing the method

Method and device for reproducing a binaural output signal generated from a monaural input signal

Switching center and central switching controller with internal broadband bus

Method and apparatus for evaluating Multifrequency tone signals using an adaptive wave digital notch filter