Abstract: A Digital Subscriber Line Access Multiplexer (DSLAM) is provided which comprises: a network side interface adapted to receive a plurality of communication signals conveyed along a broadband network: a storage means adapted to receive at least some of the communication signals, associate one or more corresponding priorities therewith, and store their content in accordance with at least one of the associated priorities: multiplexing means operative to retrieve content stored in the storage means and multiplex the content received with run time data received at the network side interface into a multiplexed product while ensuring at least a minimum pre-defined quality of service for the multiplexed product: a subscribers' output interface adapted to receive the multiplexed product from the multiplexing means and forward it along a communication line extending towards a subscriber; and a subscribers' input interface adapted to receive information generated by subscribers.

Abstract: Dividing a multiplexed optical signal transmitted along a first transmission path into two portions and diverting one of the two portions to a second transmission path and forwarding the other portion of the multiplexed optical signal along that first transmission path, while the energy of the diverted portion comprises the entire spectrum of the multiplexed optical signal. Preferably, the diverted first portion is further divided into at least two parts, each comprising a wavelength range substantially different than the others, and at least one of the two parts is forwarded towards an optical receiver.

Abstract: The arrangement of the invention allows for modifying amplitude, frequency or phase characteristics, for example, of an audio or video signal in a compressed signal system without altering the encoder or decoder employed by each conferee in communications information conferencing. A control network inserted between the encoder and decoder permits demultiplexing the compressed bitstream signal into sensitive parameters of the signal to be controlled, as well as into those parameters whose characteristics are not of concern. The control is then effected on the parameters of concern, in a way so as to eliminate the signal degradation associated with general tandeming signal decoding and coding operations of all characteristics in establishing the scaling control desired.

Abstract: Simplified methods of searching a codebook table are provided. These methods perform a codebook search for a plurality of pulses, one pulse at a time, in order of increasing to decreasing pulse significance, wherein pulse significance is defined as the relative contribution a given pulse provides to minimizing the mean-squared error between the source signal and the quantized sequence of pulses.

Abstract: The apparatus of the invention eliminates objectionable popping and clicking when switching between conferees in a communications conferencing system employing signal compression techniques to reduce bandwidth requirements. The apparatus operates with the conventional audio decoder of such system, and with its conventional audio encoder of the type employing a look-ahead-buffer in converting a frame of sampled voice or data to a compressed bit stream. The apparatus of the invention employs its own audio decoder and audio encoder in broadcasting the digital signal samples to the various conferees on the system, but its audio encoder is devoid of any look-ahead-buffer, to achieve synchronization and time matching between the audio encoder delay and its relevant bit stream.

Abstract: An improved method of speech coding for use in conjunction with speech coding methods wherein speech is digitized into a plurality of temporally defined frames, each frame including a plurality of sub-frames, and the digitized speech is partitioned into periodic components and a residual signal. For each of a plurality of sub-frames of the residual signal, the improved method of speech coding selects and applies a time shift T to the sub-frame by applying a matching criterion to (a) the current sub-frame of the residual signal, and (b) a sample-to-sample (subframe-to-subframe) pitch delay determined by applying linear interpolation to known pitch delays occurring at or near frame-to-frame boundaries of previous frames.The matching criterion is applied by minimizing .epsilon.

Abstract: In a speech coding system which encodes speech parameters into a plurality of frames, each frame having a predetermined number of bits, a predefined number of bits per frame are employed to transmit a speech parameter delta. The speech parameter delta specifies the amount by which the value of a given parameter has changed from a previous frame to the present frame. According to a preferred embodiment disclosed herein, a speech parameter delta representing change in pitch delay from the present frame to the immediately preceding frame is transmitted in the present frame, and the predefined number of bits is in the approximate range of four to six. The speech parameter delta is used to update a memory table in the speech coding system when a frame erasure occurs.

Abstract: A compressed bit stream converter is utilized for converting the video signal parameter bit stream representations of a video signal that are encoded using the temporal and spatial resolution characteristics of a first video transmission standard to the video signal parameter bit stream representations of the video signal that are encoded using the temporal and spatial resolution characteristics of a second video transmission standard. The representation of the video signal remains in compressed form and is not converted back into video signal form using a video encoder during the conversion between the compressed bit streams encoded using the first and second video transmission standard.

Abstract: A conferencing bridging technique for compressed information signals which monitors the signal energy transmitted by each conferee. When there is only one conferee speaking, the summing circuit in the bridge unit is bypassed along with the speech decoding/coding apparatus and the talking conferee's signal is broadcast to all conferees. If there is more than one conferee speaking at any given time, then only the speaking conferee's signals are routed to an associated speech decoder wherein those conferees' compressed signals are decompressed. These decompressed signals are then combined by the summing circuit and the sum is recompressed and broadcast to all conferees. Advantageously, the above-described technique eliminates the signal degradation associated with tandeming of signal decoding and coding operations and permits the sharing of a smaller number of signal decoders among a larger number of conferees.